Merge tag 'sound-3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai: "Changes are seen in a wide range of codes, mainly due to ASoC DAPM requirements; HD-audio shows a high peak in diffstat, it's just a removal of bunch of old static quirks. Some highlights: - HDPM: Updates for AIO/RayDAT support, TCO/sync support - RME96: Add PCM sync support - HD-audio: * A few HDMI/DP audio updates (CA assignment fix, stream switching fix, Intel DP device list support) * Device specific fixes (ASUS/CXT HP mic support, Thinkpad mic improvements, Chromebook fixes, STAC9228 Dell fixes) * Replace the all static quirks for AD codecs with the generic parser * WAKEEN support for handling irqs in the power saving mode - USB-audio: Clean up implicit fb handling and related codes - DAPM is now mandatory for ASoC CODEC drivers; all existing drivers have had some level of DAPM support added. In addition, a lot of cleanups and improvements in DAPM. - Support for ASoC cross-platform compile test - New drivers and support for Analog Devices ADAU1702 and ADAU1401(a), Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson Microelectronics WM8997 - DT bindings for kirkwood and i.MX S/PDIF - Clean up and bug fixes: ssm2602, rt5640 and sgtl5000. - Core helpers for bitbanged AC'97 reset" * tag 'sound-3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (375 commits) ALSA: hda - Re-setup HDMI pin and audio infoframe on stream switches ALSA: hda - hdmi: Fallback to ALSA allocation when selecting CA ASoC: mxs-sgtl5000: Configure the dai_links as unidirectional ASoC: soc-pcm: Allow to specify unidirectional dai_link ASoC: fsl_spdif: Staticse non-exported symbols ASoC: ssm2602: Fix cache sync ASoC: Remove unused sysfs_registered field from snd_soc_codec struct ASoC: Remove unused debugfs_dapm field from snd_soc_{platform,codec} struct ASoC: Remove unused control_type field from snd_soc_codec struct ASoC: fsl: Add one blank space after ':=' in Makefile ASoC: fsl: Add wrapping for dev_dbg() in fsl_spdif.c ASoC: rt5640: change widget sequence for depop ASoC: dapm: Fix auto-disable for inverted controls ASoC: fsl: Drop SND_SOC_FSL_UTILS from SND_SOC_IMX_SPDIF ASoC: Samsung: Do not queue cyclic buffers multiple times ASoC: ep93xx-i2s: Remove unnecessary dev_set_drvdata() ASoC: designware_i2s: Remove unnecessary dev_set_drvdata() ASoC: fsl_spdif: remove redundant dev_err call in fsl_spdif_probe() ASoC: fsl: Add S/PDIF machine driver ASoc: kirkwood: Use the Kirkwood audio driver in Dove boards ...
author: Linus Torvalds <torvalds@linux-foundation.org> 2013-09-04 16:26:56 -0700
committer: Linus Torvalds <torvalds@linux-foundation.org> 2013-09-04 16:26:56 -0700
commit: 977dbfcf8e9ff1783355b260d93101af315de18a (patch)
tree: b586ca678499d1ccc2d199a97d65996c630b25d8 /sound
parent: aa7054f5a5a9ff728ce291cb103afa19f4f849eb (diff)
parent: b054087dbacee30a9dddaef2c9a96312146be04e (diff)
download: linux-977dbfcf8e9ff1783355b260d93101af315de18a.tar.bz2
188 files changed, 11349 insertions, 7001 deletions
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index ce431e6e07cf..5066a3768b28 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -14,12 +14,14 @@
 #include <linux/io.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
+#include <linux/dmaengine.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/ac97_codec.h>
 #include <sound/initval.h>
 #include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
 
 #include <mach/regs-ac97.h>
 #include <mach/audio.h>
@@ -41,20 +43,20 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
 	.reset	= pxa2xx_ac97_reset,
 };
 
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_out = {
-	.name			= "AC97 PCM out",
-	.dev_addr		= __PREG(PCDR),
-	.drcmr			= &DRCMR(12),
-	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-				  DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_ac97_pcm_out_req = 12;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = {
+	.addr		= __PREG(PCDR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.maxburst	= 32,
+	.filter_data	= &pxa2xx_ac97_pcm_out_req,
 };
 
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_in = {
-	.name			= "AC97 PCM in",
-	.dev_addr		= __PREG(PCDR),
-	.drcmr			= &DRCMR(11),
-	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-				  DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_ac97_pcm_in_req = 11;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = {
+	.addr		= __PREG(PCDR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.maxburst	= 32,
+	.filter_data	= &pxa2xx_ac97_pcm_in_req,
 };
 
 static struct snd_pcm *pxa2xx_ac97_pcm;
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 823359ed95e1..a61d7a9a995e 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -7,11 +7,13 @@
 #include <linux/slab.h>
 #include <linux/module.h>
 #include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
 
 #include <mach/dma.h>
 
@@ -43,6 +45,35 @@ int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 	size_t period = params_period_bytes(params);
 	pxa_dma_desc *dma_desc;
 	dma_addr_t dma_buff_phys, next_desc_phys;
+	u32 dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG;
+
+	/* temporary transition hack */
+	switch (rtd->params->addr_width) {
+	case DMA_SLAVE_BUSWIDTH_1_BYTE:
+		dcmd |= DCMD_WIDTH1;
+		break;
+	case DMA_SLAVE_BUSWIDTH_2_BYTES:
+		dcmd |= DCMD_WIDTH2;
+		break;
+	case DMA_SLAVE_BUSWIDTH_4_BYTES:
+		dcmd |= DCMD_WIDTH4;
+		break;
+	default:
+		/* can't happen */
+		break;
+	}
+
+	switch (rtd->params->maxburst) {
+	case 8:
+		dcmd |= DCMD_BURST8;
+		break;
+	case 16:
+		dcmd |= DCMD_BURST16;
+		break;
+	case 32:
+		dcmd |= DCMD_BURST32;
+		break;
+	}
 
 	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
 	runtime->dma_bytes = totsize;
@@ -55,14 +86,14 @@ int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 		dma_desc->ddadr = next_desc_phys;
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 			dma_desc->dsadr = dma_buff_phys;
-			dma_desc->dtadr = rtd->params->dev_addr;
+			dma_desc->dtadr = rtd->params->addr;
 		} else {
-			dma_desc->dsadr = rtd->params->dev_addr;
+			dma_desc->dsadr = rtd->params->addr;
 			dma_desc->dtadr = dma_buff_phys;
 		}
 		if (period > totsize)
 			period = totsize;
-		dma_desc->dcmd = rtd->params->dcmd | period | DCMD_ENDIRQEN;
+		dma_desc->dcmd = dcmd | period | DCMD_ENDIRQEN;
 		dma_desc++;
 		dma_buff_phys += period;
 	} while (totsize -= period);
@@ -76,8 +107,10 @@ int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
 
-	if (rtd && rtd->params && rtd->params->drcmr)
-		*rtd->params->drcmr = 0;
+	if (rtd && rtd->params && rtd->params->filter_data) {
+		unsigned long req = *(unsigned long *) rtd->params->filter_data;
+		DRCMR(req) = 0;
+	}
 
 	snd_pcm_set_runtime_buffer(substream, NULL);
 	return 0;
@@ -136,6 +169,7 @@ EXPORT_SYMBOL(pxa2xx_pcm_pointer);
 int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
 {
 	struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+	unsigned long req;
 
 	if (!prtd || !prtd->params)
 		return 0;
@@ -146,7 +180,8 @@ int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
 	DCSR(prtd->dma_ch) &= ~DCSR_RUN;
 	DCSR(prtd->dma_ch) = 0;
 	DCMD(prtd->dma_ch) = 0;
-	*prtd->params->drcmr = prtd->dma_ch | DRCMR_MAPVLD;
+	req = *(unsigned long *) prtd->params->filter_data;
+	DRCMR(req) = prtd->dma_ch | DRCMR_MAPVLD;
 
 	return 0;
 }
@@ -155,7 +190,6 @@ EXPORT_SYMBOL(__pxa2xx_pcm_prepare);
 void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id)
 {
 	struct snd_pcm_substream *substream = dev_id;
-	struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
 	int dcsr;
 
 	dcsr = DCSR(dma_ch);
@@ -164,8 +198,8 @@ void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id)
 	if (dcsr & DCSR_ENDINTR) {
 		snd_pcm_period_elapsed(substream);
 	} else {
-		printk(KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n",
-			rtd->params->name, dma_ch, dcsr);
+		printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n",
+			dma_ch, dcsr);
 		snd_pcm_stream_lock(substream);
 		snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
 		snd_pcm_stream_unlock(substream);
diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c
index 26422a3584ea..69a2455b4472 100644
--- a/sound/arm/pxa2xx-pcm.c
+++ b/sound/arm/pxa2xx-pcm.c
@@ -11,8 +11,11 @@
  */
 
 #include <linux/module.h>
+#include <linux/dmaengine.h>
+
 #include <sound/core.h>
 #include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
 
 #include "pxa2xx-pcm.h"
 
@@ -40,7 +43,7 @@ static int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
 
 	rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
 		      client->playback_params : client->capture_params;
-	ret = pxa_request_dma(rtd->params->name, DMA_PRIO_LOW,
+	ret = pxa_request_dma("dma", DMA_PRIO_LOW,
 			      pxa2xx_pcm_dma_irq, substream);
 	if (ret < 0)
 		goto err2;
diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h
index 65f86b56ba42..2a8fc08d52a1 100644
--- a/sound/arm/pxa2xx-pcm.h
+++ b/sound/arm/pxa2xx-pcm.h
@@ -13,14 +13,14 @@
 
 struct pxa2xx_runtime_data {
 	int dma_ch;
-	struct pxa2xx_pcm_dma_params *params;
+	struct snd_dmaengine_dai_dma_data *params;
 	pxa_dma_desc *dma_desc_array;
 	dma_addr_t dma_desc_array_phys;
 };
 
 struct pxa2xx_pcm_client {
-	struct pxa2xx_pcm_dma_params *playback_params;
-	struct pxa2xx_pcm_dma_params *capture_params;
+	struct snd_dmaengine_dai_dma_data *playback_params;
+	struct snd_dmaengine_dai_dma_data *capture_params;
 	int (*startup)(struct snd_pcm_substream *);
 	void (*shutdown)(struct snd_pcm_substream *);
 	int (*prepare)(struct snd_pcm_substream *);
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index c0c2f57a0d6f..313f22e9d929 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -6,6 +6,9 @@ config SND_PCM
 	tristate
 	select SND_TIMER
 
+config SND_DMAENGINE_PCM
+	tristate
+
 config SND_HWDEP
 	tristate
 
diff --git a/sound/core/Makefile b/sound/core/Makefile
index 43d4117428ac..5e890cfed423 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -13,6 +13,8 @@ snd-$(CONFIG_SND_JACK)	  += jack.o
 snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \
 		pcm_memory.o
 
+snd-pcm-dmaengine-objs := pcm_dmaengine.o
+
 snd-page-alloc-y := memalloc.o
 snd-page-alloc-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o
 
@@ -30,6 +32,7 @@ obj-$(CONFIG_SND_TIMER)		+= snd-timer.o
 obj-$(CONFIG_SND_HRTIMER)	+= snd-hrtimer.o
 obj-$(CONFIG_SND_RTCTIMER)	+= snd-rtctimer.o
 obj-$(CONFIG_SND_PCM)		+= snd-pcm.o snd-page-alloc.o
+obj-$(CONFIG_SND_DMAENGINE_PCM)	+= snd-pcm-dmaengine.o
 obj-$(CONFIG_SND_RAWMIDI)	+= snd-rawmidi.o
 
 obj-$(CONFIG_SND_OSSEMUL)	+= oss/
diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/core/pcm_dmaengine.c
index aa924d9b7986..aa924d9b7986 100644
--- a/sound/soc/soc-dmaengine-pcm.c
+++ b/sound/core/pcm_dmaengine.c
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 82bb029d4414..6e03b465e44e 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -184,7 +184,7 @@ static void xrun(struct snd_pcm_substream *substream)
 	do {								\
 		if (xrun_debug(substream, XRUN_DEBUG_BASIC)) {		\
 			xrun_log_show(substream);			\
-			if (printk_ratelimit()) {			\
+			if (snd_printd_ratelimit()) {			\
 				snd_printd("PCM: " fmt, ##args);	\
 			}						\
 			dump_stack_on_xrun(substream);			\
@@ -342,7 +342,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 		return -EPIPE;
 	}
 	if (pos >= runtime->buffer_size) {
-		if (printk_ratelimit()) {
+		if (snd_printd_ratelimit()) {
 			char name[16];
 			snd_pcm_debug_name(substream, name, sizeof(name));
 			xrun_log_show(substream);
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 11048cc744d0..915b4d7fbb23 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -1022,7 +1022,7 @@ static void dummy_proc_write(struct snd_info_entry *entry,
 		if (i >= ARRAY_SIZE(fields))
 			continue;
 		snd_info_get_str(item, ptr, sizeof(item));
-		if (strict_strtoull(item, 0, &val))
+		if (kstrtoull(item, 0, &val))
 			continue;
 		if (fields[i].size == sizeof(int))
 			*get_dummy_int_ptr(dummy, fields[i].offset) = val;
diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c
index 2c6386503940..fe9e6e2f2c5b 100644
--- a/sound/firewire/speakers.c
+++ b/sound/firewire/speakers.c
@@ -49,7 +49,6 @@ struct fwspk {
 	struct snd_card *card;
 	struct fw_unit *unit;
 	const struct device_info *device_info;
-	struct snd_pcm_substream *pcm;
 	struct mutex mutex;
 	struct cmp_connection connection;
 	struct amdtp_out_stream stream;
@@ -363,8 +362,7 @@ static int fwspk_create_pcm(struct fwspk *fwspk)
 		return err;
 	pcm->private_data = fwspk;
 	strcpy(pcm->name, fwspk->device_info->short_name);
-	fwspk->pcm = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
-	fwspk->pcm->ops = &ops;
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops);
 	return 0;
 }
 
diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c
index 9942691cc0ca..afef0d738078 100644
--- a/sound/isa/gus/interwave.c
+++ b/sound/isa/gus/interwave.c
@@ -443,8 +443,7 @@ static void snd_interwave_detect_memory(struct snd_gus_card *gus)
 		for (i = 0; i < 8; ++i)
 			iwave[i] = snd_gf1_peek(gus, bank_pos + i);
 #ifdef CONFIG_SND_DEBUG_ROM
-		printk(KERN_DEBUG "ROM at 0x%06x = %*phC\n", bank_pos,
-				  8, iwave);
+		printk(KERN_DEBUG "ROM at 0x%06x = %8phC\n", bank_pos, iwave);
 #endif
 		if (strncmp(iwave, "INTRWAVE", 8))
 			continue;	/* first check */
diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c
index a59c88818f48..461d94cfecbe 100644
--- a/sound/oss/dmabuf.c
+++ b/sound/oss/dmabuf.c
@@ -557,7 +557,6 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock)
 	unsigned long flags;
 	int err = 0, n = 0;
 	struct dma_buffparms *dmap = adev->dmap_in;
-	int go;
 
 	if (!(adev->open_mode & OPEN_READ))
 		return -EIO;
@@ -584,7 +583,7 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock)
 			spin_unlock_irqrestore(&dmap->lock,flags);
 			return -EAGAIN;
 		}
-		if ((go = adev->go))
+		if (adev->go)
 			timeout = dmabuf_timeout(dmap);
 
 		spin_unlock_irqrestore(&dmap->lock,flags);
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 59c5e9c03d53..8de66ccd7279 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -152,14 +152,9 @@ config SND_HDA_CODEC_HDMI
 	  This module is automatically loaded at probing.
 
 config SND_HDA_I915
-	bool "Build Display HD-audio controller/codec power well support for i915 cards"
+	bool
+	default y
 	depends on DRM_I915
-	help
-	  Say Y here to include full HDMI and DisplayPort HD-audio controller/codec
-	  power-well support for Intel Haswell graphics cards based on the i915 driver.
-
-	  Note that this option must be enabled for Intel Haswell C+ stepping machines, otherwise
-	  the GPU audio controller/codecs will not be initialized or damaged when exit from S3 mode.
 
 config SND_HDA_CODEC_CIRRUS
 	bool "Build Cirrus Logic codec support"
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 8a005f0e5ca4..5b6c4e3c92ca 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -666,6 +666,64 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
 }
 EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
 
+
+/* return DEVLIST_LEN parameter of the given widget */
+static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid)
+{
+	unsigned int wcaps = get_wcaps(codec, nid);
+	unsigned int parm;
+
+	if (!codec->dp_mst || !(wcaps & AC_WCAP_DIGITAL) ||
+	    get_wcaps_type(wcaps) != AC_WID_PIN)
+		return 0;
+
+	parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN);
+	if (parm == -1 && codec->bus->rirb_error)
+		parm = 0;
+	return parm & AC_DEV_LIST_LEN_MASK;
+}
+
+/**
+ * snd_hda_get_devices - copy device list without cache
+ * @codec: the HDA codec
+ * @nid: NID of the pin to parse
+ * @dev_list: device list array
+ * @max_devices: max. number of devices to store
+ *
+ * Copy the device list. This info is dynamic and so not cached.
+ * Currently called only from hda_proc.c, so not exported.
+ */
+int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid,
+			u8 *dev_list, int max_devices)
+{
+	unsigned int parm;
+	int i, dev_len, devices;
+
+	parm = get_num_devices(codec, nid);
+	if (!parm)	/* not multi-stream capable */
+		return 0;
+
+	dev_len = parm + 1;
+	dev_len = dev_len < max_devices ? dev_len : max_devices;
+
+	devices = 0;
+	while (devices < dev_len) {
+		parm = snd_hda_codec_read(codec, nid, 0,
+					  AC_VERB_GET_DEVICE_LIST, devices);
+		if (parm == -1 && codec->bus->rirb_error)
+			break;
+
+		for (i = 0; i < 8; i++) {
+			dev_list[devices] = (u8)parm;
+			parm >>= 4;
+			devices++;
+			if (devices >= dev_len)
+				break;
+		}
+	}
+	return devices;
+}
+
 /**
  * snd_hda_queue_unsol_event - add an unsolicited event to queue
  * @bus: the BUS
@@ -1216,11 +1274,13 @@ static void hda_jackpoll_work(struct work_struct *work)
 {
 	struct hda_codec *codec =
 		container_of(work, struct hda_codec, jackpoll_work.work);
-	if (!codec->jackpoll_interval)
-		return;
 
 	snd_hda_jack_set_dirty_all(codec);
 	snd_hda_jack_poll_all(codec);
+
+	if (!codec->jackpoll_interval)
+		return;
+
 	queue_delayed_work(codec->bus->workq, &codec->jackpoll_work,
 			   codec->jackpoll_interval);
 }
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 701c2e069b10..7aa9870040c1 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -94,6 +94,8 @@ enum {
 #define AC_VERB_GET_HDMI_DIP_XMIT		0x0f32
 #define AC_VERB_GET_HDMI_CP_CTRL		0x0f33
 #define AC_VERB_GET_HDMI_CHAN_SLOT		0x0f34
+#define AC_VERB_GET_DEVICE_SEL			0xf35
+#define AC_VERB_GET_DEVICE_LIST			0xf36
 
 /*
  * SET verbs
@@ -133,6 +135,7 @@ enum {
 #define AC_VERB_SET_HDMI_DIP_XMIT		0x732
 #define AC_VERB_SET_HDMI_CP_CTRL		0x733
 #define AC_VERB_SET_HDMI_CHAN_SLOT		0x734
+#define AC_VERB_SET_DEVICE_SEL			0x735
 
 /*
  * Parameter IDs
@@ -154,6 +157,7 @@ enum {
 #define AC_PAR_GPIO_CAP			0x11
 #define AC_PAR_AMP_OUT_CAP		0x12
 #define AC_PAR_VOL_KNB_CAP		0x13
+#define AC_PAR_DEVLIST_LEN		0x15
 #define AC_PAR_HDMI_LPCM_CAP		0x20
 
 /*
@@ -251,6 +255,11 @@ enum {
 #define AC_UNSOL_RES_TAG_SHIFT		26
 #define AC_UNSOL_RES_SUBTAG		(0x1f<<21)
 #define AC_UNSOL_RES_SUBTAG_SHIFT	21
+#define AC_UNSOL_RES_DE			(0x3f<<15)  /* Device Entry
+						     * (for DP1.2 MST)
+						     */
+#define AC_UNSOL_RES_DE_SHIFT		15
+#define AC_UNSOL_RES_IA			(1<<2)	/* Inactive (for DP1.2 MST) */
 #define AC_UNSOL_RES_ELDV		(1<<1)	/* ELD Data valid (for HDMI) */
 #define AC_UNSOL_RES_PD			(1<<0)	/* pinsense detect */
 #define AC_UNSOL_RES_CP_STATE		(1<<1)	/* content protection */
@@ -352,6 +361,10 @@ enum {
 #define AC_LPCMCAP_44K			(1<<30)	/* 44.1kHz support */
 #define AC_LPCMCAP_44K_MS		(1<<31)	/* 44.1kHz-multiplies support */
 
+/* Display pin's device list length */
+#define AC_DEV_LIST_LEN_MASK		0x3f
+#define AC_MAX_DEV_LIST_LEN		64
+
 /*
  * Control Parameters
  */
@@ -460,6 +473,11 @@ enum {
 #define AC_DEFCFG_PORT_CONN		(0x3<<30)
 #define AC_DEFCFG_PORT_CONN_SHIFT	30
 
+/* Display pin's device list entry */
+#define AC_DE_PD			(1<<0)
+#define AC_DE_ELDV			(1<<1)
+#define AC_DE_IA			(1<<2)
+
 /* device device types (0x0-0xf) */
 enum {
 	AC_JACK_LINE_OUT,
@@ -885,6 +903,7 @@ struct hda_codec {
 	unsigned int pcm_format_first:1; /* PCM format must be set first */
 	unsigned int epss:1;		/* supporting EPSS? */
 	unsigned int cached_write:1;	/* write only to caches */
+	unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */
 #ifdef CONFIG_PM
 	unsigned int power_on :1;	/* current (global) power-state */
 	unsigned int d3_stop_clk:1;	/* support D3 operation without BCLK */
@@ -972,6 +991,8 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums,
 			  const hda_nid_t *list);
 int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
 			   hda_nid_t nid, int recursive);
+int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid,
+			u8 *dev_list, int max_devices);
 int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 				u32 *ratesp, u64 *formatsp, unsigned int *bpsp);
 
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index e3c7ba8d7582..ac41e9cdc976 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -142,6 +142,9 @@ static void parse_user_hints(struct hda_codec *codec)
 	val = snd_hda_get_bool_hint(codec, "primary_hp");
 	if (val >= 0)
 		spec->no_primary_hp = !val;
+	val = snd_hda_get_bool_hint(codec, "multi_io");
+	if (val >= 0)
+		spec->no_multi_io = !val;
 	val = snd_hda_get_bool_hint(codec, "multi_cap_vol");
 	if (val >= 0)
 		spec->multi_cap_vol = !!val;
@@ -813,6 +816,8 @@ static void resume_path_from_idx(struct hda_codec *codec, int path_idx)
 
 static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_value *ucontrol);
+static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol);
 
 enum {
 	HDA_CTL_WIDGET_VOL,
@@ -830,7 +835,13 @@ static const struct snd_kcontrol_new control_templates[] = {
 		.put = hda_gen_mixer_mute_put, /* replaced */
 		.private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0),
 	},
-	HDA_BIND_MUTE(NULL, 0, 0, 0),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.info = snd_hda_mixer_amp_switch_info,
+		.get = snd_hda_mixer_bind_switch_get,
+		.put = hda_gen_bind_mute_put, /* replaced */
+		.private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0),
+	},
 };
 
 /* add dynamic controls from template */
@@ -937,8 +948,8 @@ static int add_stereo_sw(struct hda_codec *codec, const char *pfx,
 }
 
 /* playback mute control with the software mute bit check */
-static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
-				  struct snd_ctl_elem_value *ucontrol)
+static void sync_auto_mute_bits(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct hda_gen_spec *spec = codec->spec;
@@ -949,10 +960,22 @@ static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
 		ucontrol->value.integer.value[0] &= enabled;
 		ucontrol->value.integer.value[1] &= enabled;
 	}
+}
 
+static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	sync_auto_mute_bits(kcontrol, ucontrol);
 	return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
 }
 
+static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	sync_auto_mute_bits(kcontrol, ucontrol);
+	return snd_hda_mixer_bind_switch_put(kcontrol, ucontrol);
+}
+
 /* any ctl assigned to the path with the given index? */
 static bool path_has_mixer(struct hda_codec *codec, int path_idx, int ctl_type)
 {
@@ -1541,7 +1564,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 					      cfg->speaker_pins,
 					      spec->multiout.extra_out_nid,
 					      spec->speaker_paths);
-			if (fill_mio_first && cfg->line_outs == 1 &&
+			if (!spec->no_multi_io &&
+			    fill_mio_first && cfg->line_outs == 1 &&
 			    cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
 				err = fill_multi_ios(codec, cfg->line_out_pins[0], true);
 				if (!err)
@@ -1554,7 +1578,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 				   spec->private_dac_nids, spec->out_paths,
 				   spec->main_out_badness);
 
-	if (fill_mio_first &&
+	if (!spec->no_multi_io && fill_mio_first &&
 	    cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
 		/* try to fill multi-io first */
 		err = fill_multi_ios(codec, cfg->line_out_pins[0], false);
@@ -1582,7 +1606,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 			return err;
 		badness += err;
 	}
-	if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
+	if (!spec->no_multi_io &&
+	    cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
 		err = fill_multi_ios(codec, cfg->line_out_pins[0], false);
 		if (err < 0)
 			return err;
@@ -1600,7 +1625,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 				check_aamix_out_path(codec, spec->speaker_paths[0]);
 	}
 
-	if (cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+	if (!spec->no_multi_io &&
+	    cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
 		if (count_multiio_pins(codec, cfg->hp_pins[0]) >= 2)
 			spec->multi_ios = 1; /* give badness */
 
@@ -3724,7 +3750,8 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx,
 /* check each pin in the given array; returns true if any of them is plugged */
 static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
 {
-	int i, present = 0;
+	int i;
+	bool present = false;
 
 	for (i = 0; i < num_pins; i++) {
 		hda_nid_t nid = pins[i];
@@ -3733,14 +3760,15 @@ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
 		/* don't detect pins retasked as inputs */
 		if (snd_hda_codec_get_pin_target(codec, nid) & AC_PINCTL_IN_EN)
 			continue;
-		present |= snd_hda_jack_detect(codec, nid);
+		if (snd_hda_jack_detect_state(codec, nid) == HDA_JACK_PRESENT)
+			present = true;
 	}
 	return present;
 }
 
 /* standard HP/line-out auto-mute helper */
 static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
-			bool mute)
+			int *paths, bool mute)
 {
 	struct hda_gen_spec *spec = codec->spec;
 	int i;
@@ -3752,10 +3780,19 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
 			break;
 
 		if (spec->auto_mute_via_amp) {
+			struct nid_path *path;
+			hda_nid_t mute_nid;
+
+			path = snd_hda_get_path_from_idx(codec, paths[i]);
+			if (!path)
+				continue;
+			mute_nid = get_amp_nid_(path->ctls[NID_PATH_MUTE_CTL]);
+			if (!mute_nid)
+				continue;
 			if (mute)
-				spec->mute_bits |= (1ULL << nid);
+				spec->mute_bits |= (1ULL << mute_nid);
 			else
-				spec->mute_bits &= ~(1ULL << nid);
+				spec->mute_bits &= ~(1ULL << mute_nid);
 			set_pin_eapd(codec, nid, !mute);
 			continue;
 		}
@@ -3786,14 +3823,19 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
 void snd_hda_gen_update_outputs(struct hda_codec *codec)
 {
 	struct hda_gen_spec *spec = codec->spec;
+	int *paths;
 	int on;
 
 	/* Control HP pins/amps depending on master_mute state;
 	 * in general, HP pins/amps control should be enabled in all cases,
 	 * but currently set only for master_mute, just to be safe
 	 */
+	if (spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)
+		paths = spec->out_paths;
+	else
+		paths = spec->hp_paths;
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
-		    spec->autocfg.hp_pins, spec->master_mute);
+		    spec->autocfg.hp_pins, paths, spec->master_mute);
 
 	if (!spec->automute_speaker)
 		on = 0;
@@ -3801,8 +3843,12 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec)
 		on = spec->hp_jack_present | spec->line_jack_present;
 	on |= spec->master_mute;
 	spec->speaker_muted = on;
+	if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
+		paths = spec->out_paths;
+	else
+		paths = spec->speaker_paths;
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins),
-		    spec->autocfg.speaker_pins, on);
+		    spec->autocfg.speaker_pins, paths, on);
 
 	/* toggle line-out mutes if needed, too */
 	/* if LO is a copy of either HP or Speaker, don't need to handle it */
@@ -3815,8 +3861,9 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec)
 		on = spec->hp_jack_present;
 	on |= spec->master_mute;
 	spec->line_out_muted = on;
+	paths = spec->out_paths;
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
-		    spec->autocfg.line_out_pins, on);
+		    spec->autocfg.line_out_pins, paths, on);
 }
 EXPORT_SYMBOL_HDA(snd_hda_gen_update_outputs);
 
@@ -3887,7 +3934,7 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja
 		/* don't detect pins retasked as outputs */
 		if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN)
 			continue;
-		if (snd_hda_jack_detect(codec, pin)) {
+		if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) {
 			mux_select(codec, 0, spec->am_entry[i].idx);
 			return;
 		}
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index e199a852388b..48d44026705b 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -220,6 +220,7 @@ struct hda_gen_spec {
 	unsigned int hp_mic:1; /* Allow HP as a mic-in */
 	unsigned int suppress_hp_mic_detect:1; /* Don't detect HP/mic */
 	unsigned int no_primary_hp:1; /* Don't prefer HP pins to speaker pins */
+	unsigned int no_multi_io:1; /* Don't try multi I/O config */
 	unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */
 	unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */
 	unsigned int own_eapd_ctl:1; /* set EAPD by own function */
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index ce67608734b5..fe0bda19de15 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -295,7 +295,7 @@ static ssize_t type##_store(struct device *dev,			\
 	struct snd_hwdep *hwdep = dev_get_drvdata(dev);		\
 	struct hda_codec *codec = hwdep->private_data;		\
 	unsigned long val;					\
-	int err = strict_strtoul(buf, 0, &val);			\
+	int err = kstrtoul(buf, 0, &val);			\
 	if (err < 0)						\
 		return err;					\
 	codec->type = val;					\
@@ -654,7 +654,7 @@ int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp)
 	p = snd_hda_get_hint(codec, key);
 	if (!p)
 		ret = -ENOENT;
-	else if (strict_strtoul(p, 0, &val))
+	else if (kstrtoul(p, 0, &val))
 		ret = -EINVAL;
 	else {
 		*valp = val;
@@ -751,7 +751,7 @@ static void parse_##name##_mode(char *buf, struct hda_bus *bus, \
 				 struct hda_codec **codecp) \
 { \
 	unsigned long val; \
-	if (!strict_strtoul(buf, 0, &val)) \
+	if (!kstrtoul(buf, 0, &val)) \
 		(*codecp)->name = val; \
 }
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 8860dd529520..c6c98298ac39 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1160,7 +1160,7 @@ static int azx_reset(struct azx *chip, int full_reset)
 		goto __skip;
 
 	/* clear STATESTS */
-	azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+	azx_writew(chip, STATESTS, STATESTS_INT_MASK);
 
 	/* reset controller */
 	azx_enter_link_reset(chip);
@@ -1242,7 +1242,7 @@ static void azx_int_clear(struct azx *chip)
 	}
 
 	/* clear STATESTS */
-	azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+	azx_writew(chip, STATESTS, STATESTS_INT_MASK);
 
 	/* clear rirb status */
 	azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
@@ -1451,8 +1451,8 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
 
 #if 0
 	/* clear state status int */
-	if (azx_readb(chip, STATESTS) & 0x04)
-		azx_writeb(chip, STATESTS, 0x04);
+	if (azx_readw(chip, STATESTS) & 0x04)
+		azx_writew(chip, STATESTS, 0x04);
 #endif
 	spin_unlock(&chip->reg_lock);
 	
@@ -2971,6 +2971,10 @@ static int azx_runtime_suspend(struct device *dev)
 	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;
 
+	/* enable controller wake up event */
+	azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
+		  STATESTS_INT_MASK);
+
 	azx_stop_chip(chip);
 	azx_enter_link_reset(chip);
 	azx_clear_irq_pending(chip);
@@ -2983,11 +2987,31 @@ static int azx_runtime_resume(struct device *dev)
 {
 	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;
+	struct hda_bus *bus;
+	struct hda_codec *codec;
+	int status;
 
 	if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
 		hda_display_power(true);
+
+	/* Read STATESTS before controller reset */
+	status = azx_readw(chip, STATESTS);
+
 	azx_init_pci(chip);
 	azx_init_chip(chip, 1);
+
+	bus = chip->bus;
+	if (status && bus) {
+		list_for_each_entry(codec, &bus->codec_list, list)
+			if (status & (1 << codec->addr))
+				queue_delayed_work(codec->bus->workq,
+						   &codec->jackpoll_work, codec->jackpoll_interval);
+	}
+
+	/* disable controller Wake Up event*/
+	azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
+			~STATESTS_INT_MASK);
+
 	return 0;
 }
 
@@ -3831,11 +3855,13 @@ static int azx_probe_continue(struct azx *chip)
 
 	/* Request power well for Haswell HDA controller and codec */
 	if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) {
+#ifdef CONFIG_SND_HDA_I915
 		err = hda_i915_init();
 		if (err < 0) {
 			snd_printk(KERN_ERR SFX "Error request power-well from i915\n");
 			goto out_free;
 		}
+#endif
 		hda_display_power(true);
 	}
 
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index 3fd2973183e2..05b3e3e9108f 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -194,18 +194,24 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid)
 EXPORT_SYMBOL_HDA(snd_hda_pin_sense);
 
 /**
- * snd_hda_jack_detect - query pin Presence Detect status
+ * snd_hda_jack_detect_state - query pin Presence Detect status
  * @codec: the CODEC to sense
  * @nid: the pin NID to sense
  *
- * Query and return the pin's Presence Detect status.
+ * Query and return the pin's Presence Detect status, as either
+ * HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT or HDA_JACK_PHANTOM.
  */
-int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
+int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid)
 {
-	u32 sense = snd_hda_pin_sense(codec, nid);
-	return get_jack_plug_state(sense);
+	struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid);
+	if (jack && jack->phantom_jack)
+		return HDA_JACK_PHANTOM;
+	else if (snd_hda_pin_sense(codec, nid) & AC_PINSENSE_PRESENCE)
+		return HDA_JACK_PRESENT;
+	else
+		return HDA_JACK_NOT_PRESENT;
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_detect);
+EXPORT_SYMBOL_HDA(snd_hda_jack_detect_state);
 
 /**
  * snd_hda_jack_detect_enable - enable the jack-detection
@@ -247,8 +253,8 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable);
 int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid,
 				 hda_nid_t gating_nid)
 {
-	struct hda_jack_tbl *gated = snd_hda_jack_tbl_get(codec, gated_nid);
-	struct hda_jack_tbl *gating = snd_hda_jack_tbl_get(codec, gating_nid);
+	struct hda_jack_tbl *gated = snd_hda_jack_tbl_new(codec, gated_nid);
+	struct hda_jack_tbl *gating = snd_hda_jack_tbl_new(codec, gating_nid);
 
 	if (!gated || !gating)
 		return -EINVAL;
diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index ec12abd45263..379420c44eef 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -75,7 +75,18 @@ int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid,
 				 hda_nid_t gating_nid);
 
 u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid);
-int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid);
+
+/* the jack state returned from snd_hda_jack_detect_state() */
+enum {
+	HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT, HDA_JACK_PHANTOM,
+};
+
+int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid);
+
+static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
+{
+	return snd_hda_jack_detect_state(codec, nid) != HDA_JACK_NOT_PRESENT;
+}
 
 bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid);
 
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 9760f001916d..a8cb22eec89e 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -582,6 +582,36 @@ static void print_gpio(struct snd_info_buffer *buffer,
 	print_nid_array(buffer, codec, nid, &codec->nids);
 }
 
+static void print_device_list(struct snd_info_buffer *buffer,
+			    struct hda_codec *codec, hda_nid_t nid)
+{
+	int i, curr = -1;
+	u8 dev_list[AC_MAX_DEV_LIST_LEN];
+	int devlist_len;
+
+	devlist_len = snd_hda_get_devices(codec, nid, dev_list,
+					AC_MAX_DEV_LIST_LEN);
+	snd_iprintf(buffer, "  Devices: %d\n", devlist_len);
+	if (devlist_len <= 0)
+		return;
+
+	curr = snd_hda_codec_read(codec, nid, 0,
+				AC_VERB_GET_DEVICE_SEL, 0);
+
+	for (i = 0; i < devlist_len; i++) {
+		if (i == curr)
+			snd_iprintf(buffer, "    *");
+		else
+			snd_iprintf(buffer, "     ");
+
+		snd_iprintf(buffer,
+			"Dev %02d: PD = %d, ELDV = %d, IA = %d\n", i,
+			!!(dev_list[i] & AC_DE_PD),
+			!!(dev_list[i] & AC_DE_ELDV),
+			!!(dev_list[i] & AC_DE_IA));
+	}
+}
+
 static void print_codec_info(struct snd_info_entry *entry,
 			     struct snd_info_buffer *buffer)
 {
@@ -751,6 +781,9 @@ static void print_codec_info(struct snd_info_entry *entry,
 				    (wid_caps & AC_WCAP_DELAY) >>
 				    AC_WCAP_DELAY_SHIFT);
 
+		if (wid_type == AC_WID_PIN && codec->dp_mst)
+			print_device_list(buffer, codec, nid);
+
 		if (wid_caps & AC_WCAP_CONN_LIST)
 			print_conn_list(buffer, codec, nid, wid_type,
 					conn, conn_len);
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index d97f0d61a15b..0cbdd87dde6d 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -32,7 +32,6 @@
 #include "hda_jack.h"
 #include "hda_generic.h"
 
-#define ENABLE_AD_STATIC_QUIRKS
 
 struct ad198x_spec {
 	struct hda_gen_spec gen;
@@ -43,114 +42,8 @@ struct ad198x_spec {
 	hda_nid_t eapd_nid;
 
 	unsigned int beep_amp;	/* beep amp value, set via set_beep_amp() */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-	const struct snd_kcontrol_new *mixers[6];
-	int num_mixers;
-	const struct hda_verb *init_verbs[6];	/* initialization verbs
-						 * don't forget NULL termination!
-						 */
-	unsigned int num_init_verbs;
-
-	/* playback */
-	struct hda_multi_out multiout;	/* playback set-up
-					 * max_channels, dacs must be set
-					 * dig_out_nid and hp_nid are optional
-					 */
-	unsigned int cur_eapd;
-	unsigned int need_dac_fix;
-
-	/* capture */
-	unsigned int num_adc_nids;
-	const hda_nid_t *adc_nids;
-	hda_nid_t dig_in_nid;		/* digital-in NID; optional */
-
-	/* capture source */
-	const struct hda_input_mux *input_mux;
-	const hda_nid_t *capsrc_nids;
-	unsigned int cur_mux[3];
-
-	/* channel model */
-	const struct hda_channel_mode *channel_mode;
-	int num_channel_mode;
-
-	/* PCM information */
-	struct hda_pcm pcm_rec[3];	/* used in alc_build_pcms() */
-
-	unsigned int spdif_route;
-
-	unsigned int jack_present: 1;
-	unsigned int inv_jack_detect: 1;/* inverted jack-detection */
-	unsigned int analog_beep: 1;	/* analog beep input present */
-	unsigned int avoid_init_slave_vol:1;
-
-#ifdef CONFIG_PM
-	struct hda_loopback_check loopback;
-#endif
-	/* for virtual master */
-	hda_nid_t vmaster_nid;
-	const char * const *slave_vols;
-	const char * const *slave_sws;
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-};
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-/*
- * input MUX handling (common part)
- */
-static int ad198x_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-
-	return snd_hda_input_mux_info(spec->input_mux, uinfo);
-}
-
-static int ad198x_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-
-	ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
-	return 0;
-}
-
-static int ad198x_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-
-	return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
-				     spec->capsrc_nids[adc_idx],
-				     &spec->cur_mux[adc_idx]);
-}
-
-/*
- * initialization (common callbacks)
- */
-static int ad198x_init(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	int i;
-
-	for (i = 0; i < spec->num_init_verbs; i++)
-		snd_hda_sequence_write(codec, spec->init_verbs[i]);
-	return 0;
-}
-
-static const char * const ad_slave_pfxs[] = {
-	"Front", "Surround", "Center", "LFE", "Side",
-	"Headphone", "Mono", "Speaker", "IEC958",
-	NULL
 };
 
-static const char * const ad1988_6stack_fp_slave_pfxs[] = {
-	"Front", "Surround", "Center", "LFE", "Side", "IEC958",
-	NULL
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
 
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
 /* additional beep mixers; the actual parameters are overwritten at build */
@@ -160,12 +53,6 @@ static const struct snd_kcontrol_new ad_beep_mixer[] = {
 	{ } /* end */
 };
 
-static const struct snd_kcontrol_new ad_beep2_mixer[] = {
-	HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_BEEP("Digital Beep Playback Switch", 0, 0, HDA_OUTPUT),
-	{ } /* end */
-};
-
 #define set_beep_amp(spec, nid, idx, dir) \
 	((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) /* mono */
 #else
@@ -181,8 +68,7 @@ static int create_beep_ctls(struct hda_codec *codec)
 	if (!spec->beep_amp)
 		return 0;
 
-	knew = spec->analog_beep ? ad_beep2_mixer : ad_beep_mixer;
-	for ( ; knew->name; knew++) {
+	for (knew = ad_beep_mixer ; knew->name; knew++) {
 		int err;
 		struct snd_kcontrol *kctl;
 		kctl = snd_ctl_new1(knew, codec);
@@ -199,268 +85,6 @@ static int create_beep_ctls(struct hda_codec *codec)
 #define create_beep_ctls(codec)		0
 #endif
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int ad198x_build_controls(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	struct snd_kcontrol *kctl;
-	unsigned int i;
-	int err;
-
-	for (i = 0; i < spec->num_mixers; i++) {
-		err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
-		if (err < 0)
-			return err;
-	}
-	if (spec->multiout.dig_out_nid) {
-		err = snd_hda_create_spdif_out_ctls(codec,
-						    spec->multiout.dig_out_nid,
-						    spec->multiout.dig_out_nid);
-		if (err < 0)
-			return err;
-		err = snd_hda_create_spdif_share_sw(codec,
-						    &spec->multiout);
-		if (err < 0)
-			return err;
-		spec->multiout.share_spdif = 1;
-	} 
-	if (spec->dig_in_nid) {
-		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
-		if (err < 0)
-			return err;
-	}
-
-	/* create beep controls if needed */
-	err = create_beep_ctls(codec);
-	if (err < 0)
-		return err;
-
-	/* if we have no master control, let's create it */
-	if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
-		unsigned int vmaster_tlv[4];
-		snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
-					HDA_OUTPUT, vmaster_tlv);
-		err = __snd_hda_add_vmaster(codec, "Master Playback Volume",
-					  vmaster_tlv,
-					  (spec->slave_vols ?
-					   spec->slave_vols : ad_slave_pfxs),
-					  "Playback Volume",
-					  !spec->avoid_init_slave_vol, NULL);
-		if (err < 0)
-			return err;
-	}
-	if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
-		err = snd_hda_add_vmaster(codec, "Master Playback Switch",
-					  NULL,
-					  (spec->slave_sws ?
-					   spec->slave_sws : ad_slave_pfxs),
-					  "Playback Switch");
-		if (err < 0)
-			return err;
-	}
-
-	/* assign Capture Source enums to NID */
-	kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
-	if (!kctl)
-		kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
-	for (i = 0; kctl && i < kctl->count; i++) {
-		err = snd_hda_add_nid(codec, kctl, i, spec->capsrc_nids[i]);
-		if (err < 0)
-			return err;
-	}
-
-	/* assign IEC958 enums to NID */
-	kctl = snd_hda_find_mixer_ctl(codec,
-			SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source");
-	if (kctl) {
-		err = snd_hda_add_nid(codec, kctl, 0,
-				      spec->multiout.dig_out_nid);
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
-#ifdef CONFIG_PM
-static int ad198x_check_power_status(struct hda_codec *codec, hda_nid_t nid)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
-}
-#endif
-
-/*
- * Analog playback callbacks
- */
-static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo,
-				    struct hda_codec *codec,
-				    struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
-					     hinfo);
-}
-
-static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
-				       struct hda_codec *codec,
-				       unsigned int stream_tag,
-				       unsigned int format,
-				       struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
-						format, substream);
-}
-
-static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
-				       struct hda_codec *codec,
-				       struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
-}
-
-/*
- * Digital out
- */
-static int ad198x_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
-					struct hda_codec *codec,
-					struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_dig_open(codec, &spec->multiout);
-}
-
-static int ad198x_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
-					 struct hda_codec *codec,
-					 struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
-}
-
-static int ad198x_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
-					   struct hda_codec *codec,
-					   unsigned int stream_tag,
-					   unsigned int format,
-					   struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
-					     format, substream);
-}
-
-static int ad198x_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
-					   struct hda_codec *codec,
-					   struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
-}
-
-/*
- * Analog capture
- */
-static int ad198x_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
-				      struct hda_codec *codec,
-				      unsigned int stream_tag,
-				      unsigned int format,
-				      struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
-				   stream_tag, 0, format);
-	return 0;
-}
-
-static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
-				      struct hda_codec *codec,
-				      struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
-	return 0;
-}
-
-/*
- */
-static const struct hda_pcm_stream ad198x_pcm_analog_playback = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 6, /* changed later */
-	.nid = 0, /* fill later */
-	.ops = {
-		.open = ad198x_playback_pcm_open,
-		.prepare = ad198x_playback_pcm_prepare,
-		.cleanup = ad198x_playback_pcm_cleanup,
-	},
-};
-
-static const struct hda_pcm_stream ad198x_pcm_analog_capture = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 2,
-	.nid = 0, /* fill later */
-	.ops = {
-		.prepare = ad198x_capture_pcm_prepare,
-		.cleanup = ad198x_capture_pcm_cleanup
-	},
-};
-
-static const struct hda_pcm_stream ad198x_pcm_digital_playback = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 2,
-	.nid = 0, /* fill later */
-	.ops = {
-		.open = ad198x_dig_playback_pcm_open,
-		.close = ad198x_dig_playback_pcm_close,
-		.prepare = ad198x_dig_playback_pcm_prepare,
-		.cleanup = ad198x_dig_playback_pcm_cleanup
-	},
-};
-
-static const struct hda_pcm_stream ad198x_pcm_digital_capture = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 2,
-	/* NID is set in alc_build_pcms */
-};
-
-static int ad198x_build_pcms(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	struct hda_pcm *info = spec->pcm_rec;
-
-	codec->num_pcms = 1;
-	codec->pcm_info = info;
-
-	info->name = "AD198x Analog";
-	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_analog_playback;
-	info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->multiout.max_channels;
-	info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
-	info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_analog_capture;
-	info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids;
-	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
-
-	if (spec->multiout.dig_out_nid) {
-		info++;
-		codec->num_pcms++;
-		codec->spdif_status_reset = 1;
-		info->name = "AD198x Digital";
-		info->pcm_type = HDA_PCM_TYPE_SPDIF;
-		info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback;
-		info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
-		if (spec->dig_in_nid) {
-			info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_digital_capture;
-			info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
-		}
-	}
-
-	return 0;
-}
-#endif /* ENABLE_AD_STATIC_QUIRKS */
 
 static void ad198x_power_eapd_write(struct hda_codec *codec, hda_nid_t front,
 				hda_nid_t hp)
@@ -507,18 +131,6 @@ static void ad198x_shutup(struct hda_codec *codec)
 	ad198x_power_eapd(codec);
 }
 
-static void ad198x_free(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-
-	if (!spec)
-		return;
-
-	snd_hda_gen_spec_free(&spec->gen);
-	kfree(spec);
-	snd_hda_detach_beep_device(codec);
-}
-
 #ifdef CONFIG_PM
 static int ad198x_suspend(struct hda_codec *codec)
 {
@@ -527,65 +139,6 @@ static int ad198x_suspend(struct hda_codec *codec)
 }
 #endif
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const struct hda_codec_ops ad198x_patch_ops = {
-	.build_controls = ad198x_build_controls,
-	.build_pcms = ad198x_build_pcms,
-	.init = ad198x_init,
-	.free = ad198x_free,
-#ifdef CONFIG_PM
-	.check_power_status = ad198x_check_power_status,
-	.suspend = ad198x_suspend,
-#endif
-	.reboot_notify = ad198x_shutup,
-};
-
-
-/*
- * EAPD control
- * the private value = nid
- */
-#define ad198x_eapd_info	snd_ctl_boolean_mono_info
-
-static int ad198x_eapd_get(struct snd_kcontrol *kcontrol,
-			   struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-	if (codec->inv_eapd)
-		ucontrol->value.integer.value[0] = ! spec->cur_eapd;
-	else
-		ucontrol->value.integer.value[0] = spec->cur_eapd;
-	return 0;
-}
-
-static int ad198x_eapd_put(struct snd_kcontrol *kcontrol,
-			   struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-	hda_nid_t nid = kcontrol->private_value & 0xff;
-	unsigned int eapd;
-	eapd = !!ucontrol->value.integer.value[0];
-	if (codec->inv_eapd)
-		eapd = !eapd;
-	if (eapd == spec->cur_eapd)
-		return 0;
-	spec->cur_eapd = eapd;
-	snd_hda_codec_write_cache(codec, nid,
-				  0, AC_VERB_SET_EAPD_BTLENABLE,
-				  eapd ? 0x02 : 0x00);
-	return 1;
-}
-
-static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol,
-			       struct snd_ctl_elem_info *uinfo);
-static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol,
-			      struct snd_ctl_elem_value *ucontrol);
-static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
-			      struct snd_ctl_elem_value *ucontrol);
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /*
  * Automatic parse of I/O pins from the BIOS configuration
@@ -646,537 +199,6 @@ static int ad198x_parse_auto_config(struct hda_codec *codec)
  * AD1986A specific
  */
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-#define AD1986A_SPDIF_OUT	0x02
-#define AD1986A_FRONT_DAC	0x03
-#define AD1986A_SURR_DAC	0x04
-#define AD1986A_CLFE_DAC	0x05
-#define AD1986A_ADC		0x06
-
-static const hda_nid_t ad1986a_dac_nids[3] = {
-	AD1986A_FRONT_DAC, AD1986A_SURR_DAC, AD1986A_CLFE_DAC
-};
-static const hda_nid_t ad1986a_adc_nids[1] = { AD1986A_ADC };
-static const hda_nid_t ad1986a_capsrc_nids[1] = { 0x12 };
-
-static const struct hda_input_mux ad1986a_capture_source = {
-	.num_items = 7,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "CD", 0x1 },
-		{ "Aux", 0x3 },
-		{ "Line", 0x4 },
-		{ "Mix", 0x5 },
-		{ "Mono", 0x6 },
-		{ "Phone", 0x7 },
-	},
-};
-
-
-static const struct hda_bind_ctls ad1986a_bind_pcm_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-static const struct hda_bind_ctls ad1986a_bind_pcm_sw = {
-	.ops = &snd_hda_bind_sw,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-/*
- * mixers
- */
-static const struct snd_kcontrol_new ad1986a_mixers[] = {
-	/*
-	 * bind volumes/mutes of 3 DACs as a single PCM control for simplicity
-	 */
-	HDA_BIND_VOL("PCM Playback Volume", &ad1986a_bind_pcm_vol),
-	HDA_BIND_SW("PCM Playback Switch", &ad1986a_bind_pcm_sw),
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Surround Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x1d, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x1d, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x1d, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x1d, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x1a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	HDA_CODEC_MUTE("Stereo Downmix Switch", 0x09, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-/* additional mixers for 3stack mode */
-static const struct snd_kcontrol_new ad1986a_3st_mixers[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Channel Mode",
-		.info = ad198x_ch_mode_info,
-		.get = ad198x_ch_mode_get,
-		.put = ad198x_ch_mode_put,
-	},
-	{ } /* end */
-};
-
-/* laptop model - 2ch only */
-static const hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC };
-
-/* master controls both pins 0x1a and 0x1b */
-static const struct hda_bind_ctls ad1986a_laptop_master_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
-		0,
-	},
-};
-
-static const struct hda_bind_ctls ad1986a_laptop_master_sw = {
-	.ops = &snd_hda_bind_sw,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
-		0,
-	},
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_mixers[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
-	HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
-	HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT),
-	/* 
-	   HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
-	   HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */
-	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-/* laptop-eapd model - 2ch only */
-
-static const struct hda_input_mux ad1986a_laptop_eapd_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Internal Mic", 0x4 },
-		{ "Mix", 0x5 },
-	},
-};
-
-static const struct hda_input_mux ad1986a_automic_capture_source = {
-	.num_items = 2,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Mix", 0x5 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_master_mixers[] = {
-	HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
-	HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "External Amplifier",
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x1b,
-		.info = ad198x_eapd_info,
-		.get = ad198x_eapd_get,
-		.put = ad198x_eapd_put,
-		.private_value = 0x1b, /* port-D */
-	},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = {
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-/* re-connect the mic boost input according to the jack sensing */
-static void ad1986a_automic(struct hda_codec *codec)
-{
-	unsigned int present;
-	present = snd_hda_jack_detect(codec, 0x1f);
-	/* 0 = 0x1f, 2 = 0x1d, 4 = mixed */
-	snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_CONNECT_SEL,
-			    present ? 0 : 2);
-}
-
-#define AD1986A_MIC_EVENT		0x36
-
-static void ad1986a_automic_unsol_event(struct hda_codec *codec,
-					    unsigned int res)
-{
-	if ((res >> 26) != AD1986A_MIC_EVENT)
-		return;
-	ad1986a_automic(codec);
-}
-
-static int ad1986a_automic_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1986a_automic(codec);
-	return 0;
-}
-
-/* laptop-automute - 2ch only */
-
-static void ad1986a_update_hp(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	unsigned int mute;
-
-	if (spec->jack_present)
-		mute = HDA_AMP_MUTE; /* mute internal speaker */
-	else
-		/* unmute internal speaker if necessary */
-		mute = snd_hda_codec_amp_read(codec, 0x1a, 0, HDA_OUTPUT, 0);
-	snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, mute);
-}
-
-static void ad1986a_hp_automute(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-
-	spec->jack_present = snd_hda_jack_detect(codec, 0x1a);
-	if (spec->inv_jack_detect)
-		spec->jack_present = !spec->jack_present;
-	ad1986a_update_hp(codec);
-}
-
-#define AD1986A_HP_EVENT		0x37
-
-static void ad1986a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	if ((res >> 26) != AD1986A_HP_EVENT)
-		return;
-	ad1986a_hp_automute(codec);
-}
-
-static int ad1986a_hp_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1986a_hp_automute(codec);
-	return 0;
-}
-
-/* bind hp and internal speaker mute (with plug check) */
-static int ad1986a_hp_master_sw_put(struct snd_kcontrol *kcontrol,
-				    struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
-	if (change)
-		ad1986a_update_hp(codec);
-	return change;
-}
-
-static const struct snd_kcontrol_new ad1986a_automute_master_mixers[] = {
-	HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = ad1986a_hp_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
-	},
-	{ } /* end */
-};
-
-
-/*
- * initialization verbs
- */
-static const struct hda_verb ad1986a_init_verbs[] = {
-	/* Front, Surround, CLFE DAC; mute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Downmix - off */
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* HP, Line-Out, Surround, CLFE selectors */
-	{0x0a, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Mono selector */
-	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Mic selector: Mic 1/2 pin */
-	{0x0f, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Line-in selector: Line-in */
-	{0x10, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Mic 1/2 swap */
-	{0x11, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Record selector: mic */
-	{0x12, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Mic, Phone, CD, Aux, Line-In amp; mute as default */
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* PC beep */
-	{0x18, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* HP, Line-Out, Surround, CLFE, Mono pins; mute as default */
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* HP Pin */
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
-	/* Front, Surround, CLFE Pins */
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* Mono Pin */
-	{0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* Mic Pin */
-	{0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* Line, Aux, CD, Beep-In Pin */
-	{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{ } /* end */
-};
-
-static const struct hda_verb ad1986a_ch2_init[] = {
-	/* Surround out -> Line In */
-	{ 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- 	/* Line-in selectors */
-	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x1 },
-	/* CLFE -> Mic in */
-	{ 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	/* Mic selector, mix C/LFE (backmic) and Mic (frontmic) */
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 },
-	{ } /* end */
-};
-
-static const struct hda_verb ad1986a_ch4_init[] = {
-	/* Surround out -> Surround */
-	{ 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
-	/* CLFE -> Mic in */
-	{ 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 },
-	{ } /* end */
-};
-
-static const struct hda_verb ad1986a_ch6_init[] = {
-	/* Surround out -> Surround out */
-	{ 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
-	/* CLFE -> CLFE */
-	{ 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x0 },
-	{ } /* end */
-};
-
-static const struct hda_channel_mode ad1986a_modes[3] = {
-	{ 2, ad1986a_ch2_init },
-	{ 4, ad1986a_ch4_init },
-	{ 6, ad1986a_ch6_init },
-};
-
-/* eapd initialization */
-static const struct hda_verb ad1986a_eapd_init_verbs[] = {
-	{0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 },
-	{}
-};
-
-static const struct hda_verb ad1986a_automic_verbs[] = {
-	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	/*{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},*/
-	{0x0f, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x1f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_MIC_EVENT},
-	{}
-};
-
-/* Ultra initialization */
-static const struct hda_verb ad1986a_ultra_init[] = {
-	/* eapd initialization */
-	{ 0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 },
-	/* CLFE -> Mic in */
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2 },
-	{ 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	{ 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
-	{ } /* end */
-};
-
-/* pin sensing on HP jack */
-static const struct hda_verb ad1986a_hp_init_verbs[] = {
-	{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_HP_EVENT},
-	{}
-};
-
-static void ad1986a_samsung_p50_unsol_event(struct hda_codec *codec,
-					    unsigned int res)
-{
-	switch (res >> 26) {
-	case AD1986A_HP_EVENT:
-		ad1986a_hp_automute(codec);
-		break;
-	case AD1986A_MIC_EVENT:
-		ad1986a_automic(codec);
-		break;
-	}
-}
-
-static int ad1986a_samsung_p50_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1986a_hp_automute(codec);
-	ad1986a_automic(codec);
-	return 0;
-}
-
-
-/* models */
-enum {
-	AD1986A_AUTO,
-	AD1986A_6STACK,
-	AD1986A_3STACK,
-	AD1986A_LAPTOP,
-	AD1986A_LAPTOP_EAPD,
-	AD1986A_LAPTOP_AUTOMUTE,
-	AD1986A_ULTRA,
-	AD1986A_SAMSUNG,
-	AD1986A_SAMSUNG_P50,
-	AD1986A_MODELS
-};
-
-static const char * const ad1986a_models[AD1986A_MODELS] = {
-	[AD1986A_AUTO]		= "auto",
-	[AD1986A_6STACK]	= "6stack",
-	[AD1986A_3STACK]	= "3stack",
-	[AD1986A_LAPTOP]	= "laptop",
-	[AD1986A_LAPTOP_EAPD]	= "laptop-eapd",
-	[AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute",
-	[AD1986A_ULTRA]		= "ultra",
-	[AD1986A_SAMSUNG]	= "samsung",
-	[AD1986A_SAMSUNG_P50]	= "samsung-p50",
-};
-
-static const struct snd_pci_quirk ad1986a_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x11f7, "ASUS U5A", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1213, "ASUS A6J", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1263, "ASUS U5F", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1297, "ASUS Z62F", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS V1j", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1302, "ASUS W3j", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1443, "ASUS VX1", AD1986A_LAPTOP),
-	SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8J", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x1043, 0x817f, "ASUS P5", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x1043, 0x818f, "ASUS P5", AD1986A_LAPTOP),
-	SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS P5", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS M2N", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x1043, 0x8234, "ASUS M2N", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba Satellite L40-10Q", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP),
-	SND_PCI_QUIRK(0x144d, 0xc024, "Samsung P50", AD1986A_SAMSUNG_P50),
-	SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA),
-	SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_SAMSUNG),
-	SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP),
-	SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE),
-	SND_PCI_QUIRK(0x17c0, 0x2017, "Samsung M50", AD1986A_LAPTOP),
-	{}
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1986a_loopbacks[] = {
-	{ 0x13, HDA_OUTPUT, 0 }, /* Mic */
-	{ 0x14, HDA_OUTPUT, 0 }, /* Phone */
-	{ 0x15, HDA_OUTPUT, 0 }, /* CD */
-	{ 0x16, HDA_OUTPUT, 0 }, /* Aux */
-	{ 0x17, HDA_OUTPUT, 0 }, /* Line */
-	{ } /* end */
-};
-#endif
-
-static int is_jack_available(struct hda_codec *codec, hda_nid_t nid)
-{
-	unsigned int conf = snd_hda_codec_get_pincfg(codec, nid);
-	return get_defcfg_connect(conf) != AC_JACK_PORT_NONE;
-}
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 static int alloc_ad_spec(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
@@ -1203,6 +225,11 @@ static void ad_fixup_inv_jack_detect(struct hda_codec *codec,
 
 enum {
 	AD1986A_FIXUP_INV_JACK_DETECT,
+	AD1986A_FIXUP_ULTRA,
+	AD1986A_FIXUP_SAMSUNG,
+	AD1986A_FIXUP_3STACK,
+	AD1986A_FIXUP_LAPTOP,
+	AD1986A_FIXUP_LAPTOP_IMIC,
 };
 
 static const struct hda_fixup ad1986a_fixups[] = {
@@ -1210,16 +237,86 @@ static const struct hda_fixup ad1986a_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = ad_fixup_inv_jack_detect,
 	},
+	[AD1986A_FIXUP_ULTRA] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1b, 0x90170110 }, /* speaker */
+			{ 0x1d, 0x90a7013e }, /* int mic */
+			{}
+		},
+	},
+	[AD1986A_FIXUP_SAMSUNG] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1b, 0x90170110 }, /* speaker */
+			{ 0x1d, 0x90a7013e }, /* int mic */
+			{ 0x20, 0x411111f0 }, /* N/A */
+			{ 0x24, 0x411111f0 }, /* N/A */
+			{}
+		},
+	},
+	[AD1986A_FIXUP_3STACK] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1a, 0x02214021 }, /* headphone */
+			{ 0x1b, 0x01014011 }, /* front */
+			{ 0x1c, 0x01013012 }, /* surround */
+			{ 0x1d, 0x01019015 }, /* clfe */
+			{ 0x1e, 0x411111f0 }, /* N/A */
+			{ 0x1f, 0x02a190f0 }, /* mic */
+			{ 0x20, 0x018130f0 }, /* line-in */
+			{}
+		},
+	},
+	[AD1986A_FIXUP_LAPTOP] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1a, 0x02214021 }, /* headphone */
+			{ 0x1b, 0x90170110 }, /* speaker */
+			{ 0x1c, 0x411111f0 }, /* N/A */
+			{ 0x1d, 0x411111f0 }, /* N/A */
+			{ 0x1e, 0x411111f0 }, /* N/A */
+			{ 0x1f, 0x02a191f0 }, /* mic */
+			{ 0x20, 0x411111f0 }, /* N/A */
+			{}
+		},
+	},
+	[AD1986A_FIXUP_LAPTOP_IMIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1d, 0x90a7013e }, /* int mic */
+			{}
+		},
+		.chained_before = 1,
+		.chain_id = AD1986A_FIXUP_LAPTOP,
+	},
 };
 
 static const struct snd_pci_quirk ad1986a_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC),
+	SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK),
+	SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK),
+	SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_FIXUP_3STACK),
+	SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_FIXUP_LAPTOP),
+	SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_FIXUP_SAMSUNG),
+	SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA),
 	SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_FIXUP_INV_JACK_DETECT),
+	SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_FIXUP_3STACK),
+	SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_FIXUP_3STACK),
+	{}
+};
+
+static const struct hda_model_fixup ad1986a_fixup_models[] = {
+	{ .id = AD1986A_FIXUP_3STACK, .name = "3stack" },
+	{ .id = AD1986A_FIXUP_LAPTOP, .name = "laptop" },
+	{ .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-imic" },
+	{ .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-eapd" }, /* alias */
 	{}
 };
 
 /*
  */
-static int ad1986a_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1986a(struct hda_codec *codec)
 {
 	int err;
 	struct ad198x_spec *spec;
@@ -1244,7 +341,8 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec)
 	 */
 	spec->gen.multiout.no_share_stream = 1;
 
-	snd_hda_pick_fixup(codec, NULL, ad1986a_fixup_tbl, ad1986a_fixups);
+	snd_hda_pick_fixup(codec, ad1986a_fixup_models, ad1986a_fixup_tbl,
+			   ad1986a_fixups);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
 	err = ad198x_parse_auto_config(codec);
@@ -1258,330 +356,11 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec)
 	return 0;
 }
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1986a(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err, board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1986A_MODELS,
-						  ad1986a_models,
-						  ad1986a_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1986A_AUTO;
-	}
-
-	if (board_config == AD1986A_AUTO)
-		return ad1986a_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x19);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x18, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 6;
-	spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids);
-	spec->multiout.dac_nids = ad1986a_dac_nids;
-	spec->multiout.dig_out_nid = AD1986A_SPDIF_OUT;
-	spec->num_adc_nids = 1;
-	spec->adc_nids = ad1986a_adc_nids;
-	spec->capsrc_nids = ad1986a_capsrc_nids;
-	spec->input_mux = &ad1986a_capture_source;
-	spec->num_mixers = 1;
-	spec->mixers[0] = ad1986a_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1986a_init_verbs;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1986a_loopbacks;
-#endif
-	spec->vmaster_nid = 0x1b;
-	codec->inv_eapd = 1; /* AD1986A has the inverted EAPD implementation */
-
-	codec->patch_ops = ad198x_patch_ops;
-
-	/* override some parameters */
-	switch (board_config) {
-	case AD1986A_3STACK:
-		spec->num_mixers = 2;
-		spec->mixers[1] = ad1986a_3st_mixers;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = ad1986a_ch2_init;
-		spec->channel_mode = ad1986a_modes;
-		spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes);
-		spec->need_dac_fix = 1;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		break;
-	case AD1986A_LAPTOP:
-		spec->mixers[0] = ad1986a_laptop_mixers;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		break;
-	case AD1986A_LAPTOP_EAPD:
-		spec->num_mixers = 3;
-		spec->mixers[0] = ad1986a_laptop_master_mixers;
-		spec->mixers[1] = ad1986a_laptop_eapd_mixers;
-		spec->mixers[2] = ad1986a_laptop_intmic_mixers;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = ad1986a_eapd_init_verbs;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		if (!is_jack_available(codec, 0x25))
-			spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1986a_laptop_eapd_capture_source;
-		break;
-	case AD1986A_SAMSUNG:
-		spec->num_mixers = 2;
-		spec->mixers[0] = ad1986a_laptop_master_mixers;
-		spec->mixers[1] = ad1986a_laptop_eapd_mixers;
-		spec->num_init_verbs = 3;
-		spec->init_verbs[1] = ad1986a_eapd_init_verbs;
-		spec->init_verbs[2] = ad1986a_automic_verbs;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		if (!is_jack_available(codec, 0x25))
-			spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1986a_automic_capture_source;
-		codec->patch_ops.unsol_event = ad1986a_automic_unsol_event;
-		codec->patch_ops.init = ad1986a_automic_init;
-		break;
-	case AD1986A_SAMSUNG_P50:
-		spec->num_mixers = 2;
-		spec->mixers[0] = ad1986a_automute_master_mixers;
-		spec->mixers[1] = ad1986a_laptop_eapd_mixers;
-		spec->num_init_verbs = 4;
-		spec->init_verbs[1] = ad1986a_eapd_init_verbs;
-		spec->init_verbs[2] = ad1986a_automic_verbs;
-		spec->init_verbs[3] = ad1986a_hp_init_verbs;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		if (!is_jack_available(codec, 0x25))
-			spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1986a_automic_capture_source;
-		codec->patch_ops.unsol_event = ad1986a_samsung_p50_unsol_event;
-		codec->patch_ops.init = ad1986a_samsung_p50_init;
-		break;
-	case AD1986A_LAPTOP_AUTOMUTE:
-		spec->num_mixers = 3;
-		spec->mixers[0] = ad1986a_automute_master_mixers;
-		spec->mixers[1] = ad1986a_laptop_eapd_mixers;
-		spec->mixers[2] = ad1986a_laptop_intmic_mixers;
-		spec->num_init_verbs = 3;
-		spec->init_verbs[1] = ad1986a_eapd_init_verbs;
-		spec->init_verbs[2] = ad1986a_hp_init_verbs;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		if (!is_jack_available(codec, 0x25))
-			spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1986a_laptop_eapd_capture_source;
-		codec->patch_ops.unsol_event = ad1986a_hp_unsol_event;
-		codec->patch_ops.init = ad1986a_hp_init;
-		/* Lenovo N100 seems to report the reversed bit
-		 * for HP jack-sensing
-		 */
-		spec->inv_jack_detect = 1;
-		break;
-	case AD1986A_ULTRA:
-		spec->mixers[0] = ad1986a_laptop_eapd_mixers;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = ad1986a_ultra_init;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		spec->multiout.dig_out_nid = 0;
-		break;
-	}
-
-	/* AD1986A has a hardware problem that it can't share a stream
-	 * with multiple output pins.  The copy of front to surrounds
-	 * causes noisy or silent outputs at a certain timing, e.g.
-	 * changing the volume.
-	 * So, let's disable the shared stream.
-	 */
-	spec->multiout.no_share_stream = 1;
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1986a	ad1986a_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
 
 /*
  * AD1983 specific
  */
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-#define AD1983_SPDIF_OUT	0x02
-#define AD1983_DAC		0x03
-#define AD1983_ADC		0x04
-
-static const hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC };
-static const hda_nid_t ad1983_adc_nids[1] = { AD1983_ADC };
-static const hda_nid_t ad1983_capsrc_nids[1] = { 0x15 };
-
-static const struct hda_input_mux ad1983_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Line", 0x1 },
-		{ "Mix", 0x2 },
-		{ "Mix Mono", 0x3 },
-	},
-};
-
-/*
- * SPDIF playback route
- */
-static int ad1983_spdif_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
-	static const char * const texts[] = { "PCM", "ADC" };
-
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item > 1)
-		uinfo->value.enumerated.item = 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
-}
-
-static int ad1983_spdif_route_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-
-	ucontrol->value.enumerated.item[0] = spec->spdif_route;
-	return 0;
-}
-
-static int ad1983_spdif_route_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-
-	if (ucontrol->value.enumerated.item[0] > 1)
-		return -EINVAL;
-	if (spec->spdif_route != ucontrol->value.enumerated.item[0]) {
-		spec->spdif_route = ucontrol->value.enumerated.item[0];
-		snd_hda_codec_write_cache(codec, spec->multiout.dig_out_nid, 0,
-					  AC_VERB_SET_CONNECT_SEL,
-					  spec->spdif_route);
-		return 1;
-	}
-	return 0;
-}
-
-static const struct snd_kcontrol_new ad1983_mixers[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-static const struct hda_verb ad1983_init_verbs[] = {
-	/* Front, HP, Mono; mute as default */
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Beep, PCM, Mic, Line-In: mute */
-	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Front, HP selectors; from Mix */
-	{0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x06, AC_VERB_SET_CONNECT_SEL, 0x01},
-	/* Mono selector; from Mix */
-	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x03},
-	/* Mic selector; Mic */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Line-in selector: Line-in */
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Mic boost: 0dB */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-	/* Record selector: mic */
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* SPDIF route: PCM */
-	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Front Pin */
-	{0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* HP Pin */
-	{0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
-	/* Mono Pin */
-	{0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* Mic Pin */
-	{0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* Line Pin */
-	{0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{ } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1983_loopbacks[] = {
-	{ 0x12, HDA_OUTPUT, 0 }, /* Mic */
-	{ 0x13, HDA_OUTPUT, 0 }, /* Line */
-	{ } /* end */
-};
-#endif
-
-/* models */
-enum {
-	AD1983_AUTO,
-	AD1983_BASIC,
-	AD1983_MODELS
-};
-
-static const char * const ad1983_models[AD1983_MODELS] = {
-	[AD1983_AUTO]		= "auto",
-	[AD1983_BASIC]		= "basic",
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
 /*
  * SPDIF mux control for AD1983 auto-parser
  */
@@ -1656,7 +435,7 @@ static int ad1983_add_spdif_mux_ctl(struct hda_codec *codec)
 	return 0;
 }
 
-static int ad1983_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1983(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
 	int err;
@@ -1681,432 +460,11 @@ static int ad1983_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1983(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int board_config;
-	int err;
-
-	board_config = snd_hda_check_board_config(codec, AD1983_MODELS,
-						  ad1983_models, NULL);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1983_AUTO;
-	}
-
-	if (board_config == AD1983_AUTO)
-		return ad1983_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 2;
-	spec->multiout.num_dacs = ARRAY_SIZE(ad1983_dac_nids);
-	spec->multiout.dac_nids = ad1983_dac_nids;
-	spec->multiout.dig_out_nid = AD1983_SPDIF_OUT;
-	spec->num_adc_nids = 1;
-	spec->adc_nids = ad1983_adc_nids;
-	spec->capsrc_nids = ad1983_capsrc_nids;
-	spec->input_mux = &ad1983_capture_source;
-	spec->num_mixers = 1;
-	spec->mixers[0] = ad1983_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1983_init_verbs;
-	spec->spdif_route = 0;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1983_loopbacks;
-#endif
-	spec->vmaster_nid = 0x05;
-
-	codec->patch_ops = ad198x_patch_ops;
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1983	ad1983_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /*
  * AD1981 HD specific
  */
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-#define AD1981_SPDIF_OUT	0x02
-#define AD1981_DAC		0x03
-#define AD1981_ADC		0x04
-
-static const hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC };
-static const hda_nid_t ad1981_adc_nids[1] = { AD1981_ADC };
-static const hda_nid_t ad1981_capsrc_nids[1] = { 0x15 };
-
-/* 0x0c, 0x09, 0x0e, 0x0f, 0x19, 0x05, 0x18, 0x17 */
-static const struct hda_input_mux ad1981_capture_source = {
-	.num_items = 7,
-	.items = {
-		{ "Front Mic", 0x0 },
-		{ "Line", 0x1 },
-		{ "Mix", 0x2 },
-		{ "Mix Mono", 0x3 },
-		{ "CD", 0x4 },
-		{ "Mic", 0x6 },
-		{ "Aux", 0x7 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1981_mixers[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Aux Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Aux Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* identical with AD1983 */
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-static const struct hda_verb ad1981_init_verbs[] = {
-	/* Front, HP, Mono; mute as default */
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Beep, PCM, Front Mic, Line, Rear Mic, Aux, CD-In: mute */
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Front, HP selectors; from Mix */
-	{0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x06, AC_VERB_SET_CONNECT_SEL, 0x01},
-	/* Mono selector; from Mix */
-	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x03},
-	/* Mic Mixer; select Front Mic */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-	{0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Mic boost: 0dB */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Record selector: Front mic */
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* SPDIF route: PCM */
-	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Front Pin */
-	{0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* HP Pin */
-	{0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
-	/* Mono Pin */
-	{0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* Front & Rear Mic Pins */
-	{0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* Line Pin */
-	{0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	/* Digital Beep */
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
-	/* Line-Out as Input: disabled */
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{ } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1981_loopbacks[] = {
-	{ 0x12, HDA_OUTPUT, 0 }, /* Front Mic */
-	{ 0x13, HDA_OUTPUT, 0 }, /* Line */
-	{ 0x1b, HDA_OUTPUT, 0 }, /* Aux */
-	{ 0x1c, HDA_OUTPUT, 0 }, /* Mic */
-	{ 0x1d, HDA_OUTPUT, 0 }, /* CD */
-	{ } /* end */
-};
-#endif
-
-/*
- * Patch for HP nx6320
- *
- * nx6320 uses EAPD in the reverse way - EAPD-on means the internal
- * speaker output enabled _and_ mute-LED off.
- */
-
-#define AD1981_HP_EVENT		0x37
-#define AD1981_MIC_EVENT	0x38
-
-static const struct hda_verb ad1981_hp_init_verbs[] = {
-	{0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, /* default off */
-	/* pin sensing on HP and Mic jacks */
-	{0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT},
-	{0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT},
-	{}
-};
-
-/* turn on/off EAPD (+ mute HP) as a master switch */
-static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol,
-				   struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-
-	if (! ad198x_eapd_put(kcontrol, ucontrol))
-		return 0;
-	/* change speaker pin appropriately */
-	snd_hda_set_pin_ctl(codec, 0x05, spec->cur_eapd ? PIN_OUT : 0);
-	/* toggle HP mute appropriately */
-	snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE,
-				 spec->cur_eapd ? 0 : HDA_AMP_MUTE);
-	return 1;
-}
-
-/* bind volumes of both NID 0x05 and 0x06 */
-static const struct hda_bind_ctls ad1981_hp_bind_master_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x06, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-/* mute internal speaker if HP is plugged */
-static void ad1981_hp_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x06);
-	snd_hda_codec_amp_stereo(codec, 0x05, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-/* toggle input of built-in and mic jack appropriately */
-static void ad1981_hp_automic(struct hda_codec *codec)
-{
-	static const struct hda_verb mic_jack_on[] = {
-		{0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-		{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-		{}
-	};
-	static const struct hda_verb mic_jack_off[] = {
-		{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-		{0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-		{}
-	};
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x08);
-	if (present)
-		snd_hda_sequence_write(codec, mic_jack_on);
-	else
-		snd_hda_sequence_write(codec, mic_jack_off);
-}
-
-/* unsolicited event for HP jack sensing */
-static void ad1981_hp_unsol_event(struct hda_codec *codec,
-				  unsigned int res)
-{
-	res >>= 26;
-	switch (res) {
-	case AD1981_HP_EVENT:
-		ad1981_hp_automute(codec);
-		break;
-	case AD1981_MIC_EVENT:
-		ad1981_hp_automic(codec);
-		break;
-	}
-}
-
-static const struct hda_input_mux ad1981_hp_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Dock Mic", 0x1 },
-		{ "Mix", 0x2 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1981_hp_mixers[] = {
-	HDA_BIND_VOL("Master Playback Volume", &ad1981_hp_bind_master_vol),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x05,
-		.name = "Master Playback Switch",
-		.info = ad198x_eapd_info,
-		.get = ad198x_eapd_get,
-		.put = ad1981_hp_master_sw_put,
-		.private_value = 0x05,
-	},
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-#if 0
-	/* FIXME: analog mic/line loopback doesn't work with my tests...
-	 *        (although recording is OK)
-	 */
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
-	/* FIXME: does this laptop have analog CD connection? */
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
-#endif
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x18, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-/* initialize jack-sensing, too */
-static int ad1981_hp_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1981_hp_automute(codec);
-	ad1981_hp_automic(codec);
-	return 0;
-}
-
-/* configuration for Toshiba Laptops */
-static const struct hda_verb ad1981_toshiba_init_verbs[] = {
-	{0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x01 }, /* default on */
-	/* pin sensing on HP and Mic jacks */
-	{0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT},
-	{0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT},
-	{}
-};
-
-static const struct snd_kcontrol_new ad1981_toshiba_mixers[] = {
-	HDA_CODEC_VOLUME("Amp Volume", 0x1a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Amp Switch", 0x1a, 0x0, HDA_OUTPUT),
-	{ }
-};
-
-/* configuration for Lenovo Thinkpad T60 */
-static const struct snd_kcontrol_new ad1981_thinkpad_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Master Playback Switch", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* identical with AD1983 */
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-static const struct hda_input_mux ad1981_thinkpad_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Mix", 0x2 },
-		{ "CD", 0x4 },
-	},
-};
-
-/* models */
-enum {
-	AD1981_AUTO,
-	AD1981_BASIC,
-	AD1981_HP,
-	AD1981_THINKPAD,
-	AD1981_TOSHIBA,
-	AD1981_MODELS
-};
-
-static const char * const ad1981_models[AD1981_MODELS] = {
-	[AD1981_AUTO]		= "auto",
-	[AD1981_HP]		= "hp",
-	[AD1981_THINKPAD]	= "thinkpad",
-	[AD1981_BASIC]		= "basic",
-	[AD1981_TOSHIBA]	= "toshiba"
-};
-
-static const struct snd_pci_quirk ad1981_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD),
-	SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD),
-	/* All HP models */
-	SND_PCI_QUIRK_VENDOR(0x103c, "HP nx", AD1981_HP),
-	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA),
-	/* Lenovo Thinkpad T60/X60/Z6xx */
-	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1981_THINKPAD),
-	/* HP nx6320 (reversed SSID, H/W bug) */
-	SND_PCI_QUIRK(0x30b0, 0x103c, "HP nx6320", AD1981_HP),
-	{}
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
 /* follow EAPD via vmaster hook */
 static void ad_vmaster_eapd_hook(void *private_data, int enabled)
 {
@@ -2172,7 +530,7 @@ static const struct snd_pci_quirk ad1981_fixup_tbl[] = {
 	{}
 };
 
-static int ad1981_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1981(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
 	int err;
@@ -2205,110 +563,6 @@ static int ad1981_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1981(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err, board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1981_MODELS,
-						  ad1981_models,
-						  ad1981_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1981_AUTO;
-	}
-
-	if (board_config == AD1981_AUTO)
-		return ad1981_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return -ENOMEM;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x0d, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 2;
-	spec->multiout.num_dacs = ARRAY_SIZE(ad1981_dac_nids);
-	spec->multiout.dac_nids = ad1981_dac_nids;
-	spec->multiout.dig_out_nid = AD1981_SPDIF_OUT;
-	spec->num_adc_nids = 1;
-	spec->adc_nids = ad1981_adc_nids;
-	spec->capsrc_nids = ad1981_capsrc_nids;
-	spec->input_mux = &ad1981_capture_source;
-	spec->num_mixers = 1;
-	spec->mixers[0] = ad1981_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1981_init_verbs;
-	spec->spdif_route = 0;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1981_loopbacks;
-#endif
-	spec->vmaster_nid = 0x05;
-
-	codec->patch_ops = ad198x_patch_ops;
-
-	/* override some parameters */
-	switch (board_config) {
-	case AD1981_HP:
-		spec->mixers[0] = ad1981_hp_mixers;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = ad1981_hp_init_verbs;
-		if (!is_jack_available(codec, 0x0a))
-			spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1981_hp_capture_source;
-
-		codec->patch_ops.init = ad1981_hp_init;
-		codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
-		/* set the upper-limit for mixer amp to 0dB for avoiding the
-		 * possible damage by overloading
-		 */
-		snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT,
-					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
-					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-					  (1 << AC_AMPCAP_MUTE_SHIFT));
-		break;
-	case AD1981_THINKPAD:
-		spec->mixers[0] = ad1981_thinkpad_mixers;
-		spec->input_mux = &ad1981_thinkpad_capture_source;
-		/* set the upper-limit for mixer amp to 0dB for avoiding the
-		 * possible damage by overloading
-		 */
-		snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT,
-					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
-					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-					  (1 << AC_AMPCAP_MUTE_SHIFT));
-		break;
-	case AD1981_TOSHIBA:
-		spec->mixers[0] = ad1981_hp_mixers;
-		spec->mixers[1] = ad1981_toshiba_mixers;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = ad1981_toshiba_init_verbs;
-		spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1981_hp_capture_source;
-		codec->patch_ops.init = ad1981_hp_init;
-		codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
-		break;
-	}
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1981	ad1981_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /*
  * AD1988
@@ -2395,90 +649,7 @@ static int patch_ad1981(struct hda_codec *codec)
  *      E/F quad mic array
  */
 
-
 #ifdef ENABLE_AD_STATIC_QUIRKS
-/* models */
-enum {
-	AD1988_AUTO,
-	AD1988_6STACK,
-	AD1988_6STACK_DIG,
-	AD1988_3STACK,
-	AD1988_3STACK_DIG,
-	AD1988_LAPTOP,
-	AD1988_LAPTOP_DIG,
-	AD1988_MODEL_LAST,
-};
-
-/* reivision id to check workarounds */
-#define AD1988A_REV2		0x100200
-
-#define is_rev2(codec) \
-	((codec)->vendor_id == 0x11d41988 && \
-	 (codec)->revision_id == AD1988A_REV2)
-
-/*
- * mixers
- */
-
-static const hda_nid_t ad1988_6stack_dac_nids[4] = {
-	0x04, 0x06, 0x05, 0x0a
-};
-
-static const hda_nid_t ad1988_3stack_dac_nids[3] = {
-	0x04, 0x05, 0x0a
-};
-
-/* for AD1988A revision-2, DAC2-4 are swapped */
-static const hda_nid_t ad1988_6stack_dac_nids_rev2[4] = {
-	0x04, 0x05, 0x0a, 0x06
-};
-
-static const hda_nid_t ad1988_alt_dac_nid[1] = {
-	0x03
-};
-
-static const hda_nid_t ad1988_3stack_dac_nids_rev2[3] = {
-	0x04, 0x0a, 0x06
-};
-
-static const hda_nid_t ad1988_adc_nids[3] = {
-	0x08, 0x09, 0x0f
-};
-
-static const hda_nid_t ad1988_capsrc_nids[3] = {
-	0x0c, 0x0d, 0x0e
-};
-
-#define AD1988_SPDIF_OUT		0x02
-#define AD1988_SPDIF_OUT_HDMI	0x0b
-#define AD1988_SPDIF_IN		0x07
-
-static const hda_nid_t ad1989b_slave_dig_outs[] = {
-	AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0
-};
-
-static const struct hda_input_mux ad1988_6stack_capture_source = {
-	.num_items = 5,
-	.items = {
-		{ "Front Mic", 0x1 },	/* port-B */
-		{ "Line", 0x2 },	/* port-C */
-		{ "Mic", 0x4 },		/* port-E */
-		{ "CD", 0x5 },
-		{ "Mix", 0x9 },
-	},
-};
-
-static const struct hda_input_mux ad1988_laptop_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic/Line", 0x1 },	/* port-B */
-		{ "CD", 0x5 },
-		{ "Mix", 0x9 },
-	},
-};
-
-/*
- */
 static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol,
 			       struct snd_ctl_elem_info *uinfo)
 {
@@ -2509,569 +680,6 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
 		spec->multiout.num_dacs = spec->multiout.max_channels / 2;
 	return err;
 }
-
-/* 6-stack mode */
-static const struct snd_kcontrol_new ad1988_6stack_mixers1[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Side Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0a, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Side Playback Volume", 0x06, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT),
-	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT),
-	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-/* 3-stack mode */
-static const struct snd_kcontrol_new ad1988_3stack_mixers1[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x06, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x06, 2, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_3stack_mixers2[] = {
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Surround Playback Switch", 0x2c, 2, HDA_INPUT),
-	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x26, 1, 2, HDA_INPUT),
-	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x26, 2, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Channel Mode",
-		.info = ad198x_ch_mode_info,
-		.get = ad198x_ch_mode_get,
-		.put = ad198x_ch_mode_put,
-	},
-
-	{ } /* end */
-};
-
-/* laptop mode */
-static const struct snd_kcontrol_new ad1988_laptop_mixers[] = {
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x29, 0x0, HDA_INPUT),
-	HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
-
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "External Amplifier",
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x12,
-		.info = ad198x_eapd_info,
-		.get = ad198x_eapd_get,
-		.put = ad198x_eapd_put,
-		.private_value = 0x12, /* port-D */
-	},
-
-	{ } /* end */
-};
-
-/* capture */
-static const struct snd_kcontrol_new ad1988_capture_mixers[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x0e, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x0e, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 3,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-static int ad1988_spdif_playback_source_info(struct snd_kcontrol *kcontrol,
-					     struct snd_ctl_elem_info *uinfo)
-{
-	static const char * const texts[] = {
-		"PCM", "ADC1", "ADC2", "ADC3"
-	};
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 4;
-	if (uinfo->value.enumerated.item >= 4)
-		uinfo->value.enumerated.item = 3;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
-}
-
-static int ad1988_spdif_playback_source_get(struct snd_kcontrol *kcontrol,
-					    struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	unsigned int sel;
-
-	sel = snd_hda_codec_read(codec, 0x1d, 0, AC_VERB_GET_AMP_GAIN_MUTE,
-				 AC_AMP_GET_INPUT);
-	if (!(sel & 0x80))
-		ucontrol->value.enumerated.item[0] = 0;
-	else {
-		sel = snd_hda_codec_read(codec, 0x0b, 0,
-					 AC_VERB_GET_CONNECT_SEL, 0);
-		if (sel < 3)
-			sel++;
-		else
-			sel = 0;
-		ucontrol->value.enumerated.item[0] = sel;
-	}
-	return 0;
-}
-
-static int ad1988_spdif_playback_source_put(struct snd_kcontrol *kcontrol,
-					    struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	unsigned int val, sel;
-	int change;
-
-	val = ucontrol->value.enumerated.item[0];
-	if (val > 3)
-		return -EINVAL;
-	if (!val) {
-		sel = snd_hda_codec_read(codec, 0x1d, 0,
-					 AC_VERB_GET_AMP_GAIN_MUTE,
-					 AC_AMP_GET_INPUT);
-		change = sel & 0x80;
-		if (change) {
-			snd_hda_codec_write_cache(codec, 0x1d, 0,
-						  AC_VERB_SET_AMP_GAIN_MUTE,
-						  AMP_IN_UNMUTE(0));
-			snd_hda_codec_write_cache(codec, 0x1d, 0,
-						  AC_VERB_SET_AMP_GAIN_MUTE,
-						  AMP_IN_MUTE(1));
-		}
-	} else {
-		sel = snd_hda_codec_read(codec, 0x1d, 0,
-					 AC_VERB_GET_AMP_GAIN_MUTE,
-					 AC_AMP_GET_INPUT | 0x01);
-		change = sel & 0x80;
-		if (change) {
-			snd_hda_codec_write_cache(codec, 0x1d, 0,
-						  AC_VERB_SET_AMP_GAIN_MUTE,
-						  AMP_IN_MUTE(0));
-			snd_hda_codec_write_cache(codec, 0x1d, 0,
-						  AC_VERB_SET_AMP_GAIN_MUTE,
-						  AMP_IN_UNMUTE(1));
-		}
-		sel = snd_hda_codec_read(codec, 0x0b, 0,
-					 AC_VERB_GET_CONNECT_SEL, 0) + 1;
-		change |= sel != val;
-		if (change)
-			snd_hda_codec_write_cache(codec, 0x0b, 0,
-						  AC_VERB_SET_CONNECT_SEL,
-						  val - 1);
-	}
-	return change;
-}
-
-static const struct snd_kcontrol_new ad1988_spdif_out_mixers[] = {
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "IEC958 Playback Source",
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x1b,
-		.info = ad1988_spdif_playback_source_info,
-		.get = ad1988_spdif_playback_source_get,
-		.put = ad1988_spdif_playback_source_put,
-	},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_spdif_in_mixers[] = {
-	HDA_CODEC_VOLUME("IEC958 Capture Volume", 0x1c, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1989_spdif_out_mixers[] = {
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("HDMI Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-/*
- * initialization verbs
- */
-
-/*
- * for 6-stack (+dig)
- */
-static const struct hda_verb ad1988_6stack_init_verbs[] = {
-	/* Front, Surround, CLFE, side DAC; unmute as default */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-A front headphon path */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* Port-D line-out path */
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Port-F surround path */
-	{0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Port-G CLFE path */
-	{0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Port-H side path */
-	{0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Mono out path */
-	{0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
-	/* Port-B front mic-in path */
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Port-C line-in path */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Port-E mic-in path */
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x34, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Analog CD Input */
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-
-	{ }
-};
-
-static const struct hda_verb ad1988_6stack_fp_init_verbs[] = {
-	/* Headphone; unmute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-A front headphon path */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-
-	{ }
-};
-
-static const struct hda_verb ad1988_capture_init_verbs[] = {
-	/* mute analog mix */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
-	/* select ADCs - front-mic */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-
-	{ }
-};
-
-static const struct hda_verb ad1988_spdif_init_verbs[] = {
-	/* SPDIF out sel */
-	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
-	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	/* SPDIF out pin */
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-
-	{ }
-};
-
-static const struct hda_verb ad1988_spdif_in_init_verbs[] = {
-	/* unmute SPDIF input pin */
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{ }
-};
-
-/* AD1989 has no ADC -> SPDIF route */
-static const struct hda_verb ad1989_spdif_init_verbs[] = {
-	/* SPDIF-1 out pin */
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	/* SPDIF-2/HDMI out pin */
-	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	{ }
-};
-
-/*
- * verbs for 3stack (+dig)
- */
-static const struct hda_verb ad1988_3stack_ch2_init[] = {
-	/* set port-C to line-in */
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
-	/* set port-E to mic-in */
-	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	{ } /* end */
-};
-
-static const struct hda_verb ad1988_3stack_ch6_init[] = {
-	/* set port-C to surround out */
-	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	/* set port-E to CLFE out */
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ } /* end */
-};
-
-static const struct hda_channel_mode ad1988_3stack_modes[2] = {
-	{ 2, ad1988_3stack_ch2_init },
-	{ 6, ad1988_3stack_ch6_init },
-};
-
-static const struct hda_verb ad1988_3stack_init_verbs[] = {
-	/* Front, Surround, CLFE, side DAC; unmute as default */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-A front headphon path */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* Port-D line-out path */
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Mono out path */
-	{0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
-	/* Port-B front mic-in path */
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Port-C line-in/surround path - 6ch mode as default */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x31, AC_VERB_SET_CONNECT_SEL, 0x0}, /* output sel: DAC 0x05 */
-	{0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Port-E mic-in/CLFE path - 6ch mode as default */
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x32, AC_VERB_SET_CONNECT_SEL, 0x1}, /* output sel: DAC 0x0a */
-	{0x34, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* mute analog mix */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
-	/* select ADCs - front-mic */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-	{ }
-};
-
-/*
- * verbs for laptop mode (+dig)
- */
-static const struct hda_verb ad1988_laptop_hp_on[] = {
-	/* unmute port-A and mute port-D */
-	{ 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ } /* end */
-};
-static const struct hda_verb ad1988_laptop_hp_off[] = {
-	/* mute port-A and unmute port-D */
-	{ 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ } /* end */
-};
-
-#define AD1988_HP_EVENT	0x01
-
-static const struct hda_verb ad1988_laptop_init_verbs[] = {
-	/* Front, Surround, CLFE, side DAC; unmute as default */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-A front headphon path */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* unsolicited event for pin-sense */
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1988_HP_EVENT },
-	/* Port-D line-out path + EAPD */
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x00}, /* EAPD-off */
-	/* Mono out path */
-	{0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
-	/* Port-B mic-in path */
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Port-C docking station - try to output */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* mute analog mix */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
-	/* select ADCs - mic */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-	{ }
-};
-
-static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	if ((res >> 26) != AD1988_HP_EVENT)
-		return;
-	if (snd_hda_jack_detect(codec, 0x11))
-		snd_hda_sequence_write(codec, ad1988_laptop_hp_on);
-	else
-		snd_hda_sequence_write(codec, ad1988_laptop_hp_off);
-} 
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1988_loopbacks[] = {
-	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
-	{ 0x20, HDA_INPUT, 1 }, /* Line */
-	{ 0x20, HDA_INPUT, 4 }, /* Mic */
-	{ 0x20, HDA_INPUT, 6 }, /* CD */
-	{ } /* end */
-};
-#endif
 #endif /* ENABLE_AD_STATIC_QUIRKS */
 
 static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol,
@@ -3220,7 +828,34 @@ static int ad1988_add_spdif_mux_ctl(struct hda_codec *codec)
 /*
  */
 
-static int ad1988_parse_auto_config(struct hda_codec *codec)
+enum {
+	AD1988_FIXUP_6STACK_DIG,
+};
+
+static const struct hda_fixup ad1988_fixups[] = {
+	[AD1988_FIXUP_6STACK_DIG] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x11, 0x02214130 }, /* front-hp */
+			{ 0x12, 0x01014010 }, /* line-out */
+			{ 0x14, 0x02a19122 }, /* front-mic */
+			{ 0x15, 0x01813021 }, /* line-in */
+			{ 0x16, 0x01011012 }, /* line-out */
+			{ 0x17, 0x01a19020 }, /* mic */
+			{ 0x1b, 0x0145f1f0 }, /* SPDIF */
+			{ 0x24, 0x01016011 }, /* line-out */
+			{ 0x25, 0x01012013 }, /* line-out */
+			{ }
+		}
+	},
+};
+
+static const struct hda_model_fixup ad1988_fixup_models[] = {
+	{ .id = AD1988_FIXUP_6STACK_DIG, .name = "6stack-dig" },
+	{}
+};
+
+static int patch_ad1988(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
 	int err;
@@ -3234,12 +869,19 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
 	spec->gen.mixer_merge_nid = 0x21;
 	spec->gen.beep_nid = 0x10;
 	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
+
+	snd_hda_pick_fixup(codec, ad1988_fixup_models, NULL, ad1988_fixups);
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
+
 	err = ad198x_parse_auto_config(codec);
 	if (err < 0)
 		goto error;
 	err = ad1988_add_spdif_mux_ctl(codec);
 	if (err < 0)
 		goto error;
+
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
+
 	return 0;
 
  error:
@@ -3247,169 +889,6 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-/*
- */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const char * const ad1988_models[AD1988_MODEL_LAST] = {
-	[AD1988_6STACK]		= "6stack",
-	[AD1988_6STACK_DIG]	= "6stack-dig",
-	[AD1988_3STACK]		= "3stack",
-	[AD1988_3STACK_DIG]	= "3stack-dig",
-	[AD1988_LAPTOP]		= "laptop",
-	[AD1988_LAPTOP_DIG]	= "laptop-dig",
-	[AD1988_AUTO]		= "auto",
-};
-
-static const struct snd_pci_quirk ad1988_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG),
-	SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG),
-	SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG),
-	SND_PCI_QUIRK(0x1043, 0x82c0, "Asus M3N-HT Deluxe", AD1988_6STACK_DIG),
-	SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG),
-	{}
-};
-
-static int patch_ad1988(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err, board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1988_MODEL_LAST,
-						  ad1988_models, ad1988_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1988_AUTO;
-	}
-
-	if (board_config == AD1988_AUTO)
-		return ad1988_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	if (is_rev2(codec))
-		snd_printk(KERN_INFO "patch_analog: AD1988A rev.2 is detected, enable workarounds\n");
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
-	if (!spec->multiout.hp_nid)
-		spec->multiout.hp_nid = ad1988_alt_dac_nid[0];
-	switch (board_config) {
-	case AD1988_6STACK:
-	case AD1988_6STACK_DIG:
-		spec->multiout.max_channels = 8;
-		spec->multiout.num_dacs = 4;
-		if (is_rev2(codec))
-			spec->multiout.dac_nids = ad1988_6stack_dac_nids_rev2;
-		else
-			spec->multiout.dac_nids = ad1988_6stack_dac_nids;
-		spec->input_mux = &ad1988_6stack_capture_source;
-		spec->num_mixers = 2;
-		if (is_rev2(codec))
-			spec->mixers[0] = ad1988_6stack_mixers1_rev2;
-		else
-			spec->mixers[0] = ad1988_6stack_mixers1;
-		spec->mixers[1] = ad1988_6stack_mixers2;
-		spec->num_init_verbs = 1;
-		spec->init_verbs[0] = ad1988_6stack_init_verbs;
-		if (board_config == AD1988_6STACK_DIG) {
-			spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
-			spec->dig_in_nid = AD1988_SPDIF_IN;
-		}
-		break;
-	case AD1988_3STACK:
-	case AD1988_3STACK_DIG:
-		spec->multiout.max_channels = 6;
-		spec->multiout.num_dacs = 3;
-		if (is_rev2(codec))
-			spec->multiout.dac_nids = ad1988_3stack_dac_nids_rev2;
-		else
-			spec->multiout.dac_nids = ad1988_3stack_dac_nids;
-		spec->input_mux = &ad1988_6stack_capture_source;
-		spec->channel_mode = ad1988_3stack_modes;
-		spec->num_channel_mode = ARRAY_SIZE(ad1988_3stack_modes);
-		spec->num_mixers = 2;
-		if (is_rev2(codec))
-			spec->mixers[0] = ad1988_3stack_mixers1_rev2;
-		else
-			spec->mixers[0] = ad1988_3stack_mixers1;
-		spec->mixers[1] = ad1988_3stack_mixers2;
-		spec->num_init_verbs = 1;
-		spec->init_verbs[0] = ad1988_3stack_init_verbs;
-		if (board_config == AD1988_3STACK_DIG)
-			spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
-		break;
-	case AD1988_LAPTOP:
-	case AD1988_LAPTOP_DIG:
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1988_3stack_dac_nids;
-		spec->input_mux = &ad1988_laptop_capture_source;
-		spec->num_mixers = 1;
-		spec->mixers[0] = ad1988_laptop_mixers;
-		codec->inv_eapd = 1; /* inverted EAPD */
-		spec->num_init_verbs = 1;
-		spec->init_verbs[0] = ad1988_laptop_init_verbs;
-		if (board_config == AD1988_LAPTOP_DIG)
-			spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
-		break;
-	}
-
-	spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids);
-	spec->adc_nids = ad1988_adc_nids;
-	spec->capsrc_nids = ad1988_capsrc_nids;
-	spec->mixers[spec->num_mixers++] = ad1988_capture_mixers;
-	spec->init_verbs[spec->num_init_verbs++] = ad1988_capture_init_verbs;
-	if (spec->multiout.dig_out_nid) {
-		if (codec->vendor_id >= 0x11d4989a) {
-			spec->mixers[spec->num_mixers++] =
-				ad1989_spdif_out_mixers;
-			spec->init_verbs[spec->num_init_verbs++] =
-				ad1989_spdif_init_verbs;
-			codec->slave_dig_outs = ad1989b_slave_dig_outs;
-		} else {
-			spec->mixers[spec->num_mixers++] =
-				ad1988_spdif_out_mixers;
-			spec->init_verbs[spec->num_init_verbs++] =
-				ad1988_spdif_init_verbs;
-		}
-	}
-	if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a) {
-		spec->mixers[spec->num_mixers++] = ad1988_spdif_in_mixers;
-		spec->init_verbs[spec->num_init_verbs++] =
-			ad1988_spdif_in_init_verbs;
-	}
-
-	codec->patch_ops = ad198x_patch_ops;
-	switch (board_config) {
-	case AD1988_LAPTOP:
-	case AD1988_LAPTOP_DIG:
-		codec->patch_ops.unsol_event = ad1988_laptop_unsol_event;
-		break;
-	}
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1988_loopbacks;
-#endif
-	spec->vmaster_nid = 0x04;
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1988	ad1988_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /*
  * AD1884 / AD1984
@@ -3423,167 +902,19 @@ static int patch_ad1988(struct hda_codec *codec)
  *
  * AD1984 = AD1884 + two digital mic-ins
  *
- * FIXME:
- * For simplicity, we share the single DAC for both HP and line-outs
- * right now.  The inidividual playbacks could be easily implemented,
- * but no build-up framework is given, so far.
- */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const hda_nid_t ad1884_dac_nids[1] = {
-	0x04,
-};
-
-static const hda_nid_t ad1884_adc_nids[2] = {
-	0x08, 0x09,
-};
-
-static const hda_nid_t ad1884_capsrc_nids[2] = {
-	0x0c, 0x0d,
-};
-
-#define AD1884_SPDIF_OUT	0x02
-
-static const struct hda_input_mux ad1884_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Front Mic", 0x0 },
-		{ "Mic", 0x1 },
-		{ "CD", 0x2 },
-		{ "Mix", 0x3 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1884_base_mixers[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	/* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* SPDIF controls */
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		/* identical with ad1983 */
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984_dmic_mixers[] = {
-	HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0,
-			     HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Digital Mic Capture Switch", 1, 0x06, 0x0,
-			   HDA_INPUT),
-	{ } /* end */
-};
-
-/*
- * initialization verbs
+ * AD1883 / AD1884A / AD1984A / AD1984B
+ *
+ * port-B (0x14) - front mic-in
+ * port-E (0x1c) - rear mic-in
+ * port-F (0x16) - CD / ext out
+ * port-C (0x15) - rear line-in
+ * port-D (0x12) - rear line-out
+ * port-A (0x11) - front hp-out
+ *
+ * AD1984A = AD1884A + digital-mic
+ * AD1883 = equivalent with AD1984A
+ * AD1984B = AD1984A + extra SPDIF-out
  */
-static const struct hda_verb ad1884_init_verbs[] = {
-	/* DACs; mute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Port-A (HP) mixer */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-A pin */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* HP selector - select DAC2 */
-	{0x22, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Port-D (Line-out) mixer */
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-D pin */
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Mono-out mixer */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Mono-out pin */
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Mono selector */
-	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Port-B (front mic) pin */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Port-C (rear mic) pin */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Analog mixer; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-	/* SPDIF output selector */
-	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	{ } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1884_loopbacks[] = {
-	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
-	{ 0x20, HDA_INPUT, 1 }, /* Mic */
-	{ 0x20, HDA_INPUT, 2 }, /* CD */
-	{ 0x20, HDA_INPUT, 4 }, /* Docking */
-	{ } /* end */
-};
-#endif
-
-static const char * const ad1884_slave_vols[] = {
-	"PCM", "Mic", "Mono", "Front Mic", "Mic", "CD",
-	"Internal Mic", "Dock Mic", /* "Beep", */ "IEC958",
-	NULL
-};
-
-enum {
-	AD1884_AUTO,
-	AD1884_BASIC,
-	AD1884_MODELS
-};
-
-static const char * const ad1884_models[AD1884_MODELS] = {
-	[AD1884_AUTO]		= "auto",
-	[AD1884_BASIC]		= "basic",
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /* set the upper-limit for mixer amp to 0dB for avoiding the possible
  * damage by overloading
@@ -3599,14 +930,34 @@ static void ad1884_fixup_amp_override(struct hda_codec *codec,
 					  (1 << AC_AMPCAP_MUTE_SHIFT));
 }
 
+/* toggle GPIO1 according to the mute state */
+static void ad1884_vmaster_hp_gpio_hook(void *private_data, int enabled)
+{
+	struct hda_codec *codec = private_data;
+	struct ad198x_spec *spec = codec->spec;
+
+	if (spec->eapd_nid)
+		ad_vmaster_eapd_hook(private_data, enabled);
+	snd_hda_codec_update_cache(codec, 0x01, 0,
+				   AC_VERB_SET_GPIO_DATA,
+				   enabled ? 0x00 : 0x02);
+}
+
 static void ad1884_fixup_hp_eapd(struct hda_codec *codec,
 				 const struct hda_fixup *fix, int action)
 {
 	struct ad198x_spec *spec = codec->spec;
+	static const struct hda_verb gpio_init_verbs[] = {
+		{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+		{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+		{0x01, AC_VERB_SET_GPIO_DATA, 0x02},
+		{},
+	};
 
 	switch (action) {
 	case HDA_FIXUP_ACT_PRE_PROBE:
-		spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook;
+		spec->gen.vmaster_mute.hook = ad1884_vmaster_hp_gpio_hook;
+		snd_hda_sequence_write_cache(codec, gpio_init_verbs);
 		break;
 	case HDA_FIXUP_ACT_PROBE:
 		if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
@@ -3617,9 +968,18 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec,
 	}
 }
 
+/* set magic COEFs for dmic */
+static const struct hda_verb ad1884_dmic_init_verbs[] = {
+	{0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
+	{0x01, AC_VERB_SET_PROC_COEF, 0x08},
+	{}
+};
+
 enum {
 	AD1884_FIXUP_AMP_OVERRIDE,
 	AD1884_FIXUP_HP_EAPD,
+	AD1884_FIXUP_DMIC_COEF,
+	AD1884_FIXUP_HP_TOUCHSMART,
 };
 
 static const struct hda_fixup ad1884_fixups[] = {
@@ -3633,15 +993,27 @@ static const struct hda_fixup ad1884_fixups[] = {
 		.chained = true,
 		.chain_id = AD1884_FIXUP_AMP_OVERRIDE,
 	},
+	[AD1884_FIXUP_DMIC_COEF] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = ad1884_dmic_init_verbs,
+	},
+	[AD1884_FIXUP_HP_TOUCHSMART] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = ad1884_dmic_init_verbs,
+		.chained = true,
+		.chain_id = AD1884_FIXUP_HP_EAPD,
+	},
 };
 
 static const struct snd_pci_quirk ad1884_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x103c, 0x2a82, "HP Touchsmart", AD1884_FIXUP_HP_TOUCHSMART),
 	SND_PCI_QUIRK_VENDOR(0x103c, "HP", AD1884_FIXUP_HP_EAPD),
+	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_DMIC_COEF),
 	{}
 };
 
 
-static int ad1884_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1884(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
 	int err;
@@ -3674,1170 +1046,6 @@ static int ad1884_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1884_basic(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err;
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 2;
-	spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids);
-	spec->multiout.dac_nids = ad1884_dac_nids;
-	spec->multiout.dig_out_nid = AD1884_SPDIF_OUT;
-	spec->num_adc_nids = ARRAY_SIZE(ad1884_adc_nids);
-	spec->adc_nids = ad1884_adc_nids;
-	spec->capsrc_nids = ad1884_capsrc_nids;
-	spec->input_mux = &ad1884_capture_source;
-	spec->num_mixers = 1;
-	spec->mixers[0] = ad1884_base_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1884_init_verbs;
-	spec->spdif_route = 0;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1884_loopbacks;
-#endif
-	spec->vmaster_nid = 0x04;
-	/* we need to cover all playback volumes */
-	spec->slave_vols = ad1884_slave_vols;
-	/* slaves may contain input volumes, so we can't raise to 0dB blindly */
-	spec->avoid_init_slave_vol = 1;
-
-	codec->patch_ops = ad198x_patch_ops;
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-
-static int patch_ad1884(struct hda_codec *codec)
-{
-	int board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1884_MODELS,
-						  ad1884_models, NULL);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1884_AUTO;
-	}
-
-	if (board_config == AD1884_AUTO)
-		return ad1884_parse_auto_config(codec);
-	else
-		return patch_ad1884_basic(codec);
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1884	ad1884_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-/*
- * Lenovo Thinkpad T61/X61
- */
-static const struct hda_input_mux ad1984_thinkpad_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Internal Mic", 0x1 },
-		{ "Mix", 0x3 },
-		{ "Dock Mic", 0x4 },
-	},
-};
-
-
-/*
- * Dell Precision T3400
- */
-static const struct hda_input_mux ad1984_dell_desktop_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Front Mic", 0x0 },
-		{ "Line-In", 0x1 },
-		{ "Mix", 0x3 },
-	},
-};
-
-
-static const struct snd_kcontrol_new ad1984_thinkpad_mixers[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	/* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* SPDIF controls */
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		/* identical with ad1983 */
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-/* additional verbs */
-static const struct hda_verb ad1984_thinkpad_init_verbs[] = {
-	/* Port-E (docking station mic) pin */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* docking mic boost */
-	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Analog PC Beeper - allow firmware/ACPI beeps */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3) | 0x1a},
-	/* Analog mixer - docking mic; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* enable EAPD bit */
-	{0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
-	{ } /* end */
-};
-
-/*
- * Dell Precision T3400
- */
-static const struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line-In Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Line-In Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line-In Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-/* Digial MIC ADC NID 0x05 + 0x06 */
-static int ad1984_pcm_dmic_prepare(struct hda_pcm_stream *hinfo,
-				   struct hda_codec *codec,
-				   unsigned int stream_tag,
-				   unsigned int format,
-				   struct snd_pcm_substream *substream)
-{
-	snd_hda_codec_setup_stream(codec, 0x05 + substream->number,
-				   stream_tag, 0, format);
-	return 0;
-}
-
-static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo,
-				   struct hda_codec *codec,
-				   struct snd_pcm_substream *substream)
-{
-	snd_hda_codec_cleanup_stream(codec, 0x05 + substream->number);
-	return 0;
-}
-
-static const struct hda_pcm_stream ad1984_pcm_dmic_capture = {
-	.substreams = 2,
-	.channels_min = 2,
-	.channels_max = 2,
-	.nid = 0x05,
-	.ops = {
-		.prepare = ad1984_pcm_dmic_prepare,
-		.cleanup = ad1984_pcm_dmic_cleanup
-	},
-};
-
-static int ad1984_build_pcms(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	struct hda_pcm *info;
-	int err;
-
-	err = ad198x_build_pcms(codec);
-	if (err < 0)
-		return err;
-
-	info = spec->pcm_rec + codec->num_pcms;
-	codec->num_pcms++;
-	info->name = "AD1984 Digital Mic";
-	info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad1984_pcm_dmic_capture;
-	return 0;
-}
-
-/* models */
-enum {
-	AD1984_AUTO,
-	AD1984_BASIC,
-	AD1984_THINKPAD,
-	AD1984_DELL_DESKTOP,
-	AD1984_MODELS
-};
-
-static const char * const ad1984_models[AD1984_MODELS] = {
-	[AD1984_AUTO]		= "auto",
-	[AD1984_BASIC]		= "basic",
-	[AD1984_THINKPAD]	= "thinkpad",
-	[AD1984_DELL_DESKTOP]	= "dell_desktop",
-};
-
-static const struct snd_pci_quirk ad1984_cfg_tbl[] = {
-	/* Lenovo Thinkpad T61/X61 */
-	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD),
-	SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP),
-	SND_PCI_QUIRK(0x1028, 0x0233, "Dell Latitude E6400", AD1984_DELL_DESKTOP),
-	{}
-};
-
-static int patch_ad1984(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int board_config, err;
-
-	board_config = snd_hda_check_board_config(codec, AD1984_MODELS,
-						  ad1984_models, ad1984_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1984_AUTO;
-	}
-
-	if (board_config == AD1984_AUTO)
-		return ad1884_parse_auto_config(codec);
-
-	err = patch_ad1884_basic(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	switch (board_config) {
-	case AD1984_BASIC:
-		/* additional digital mics */
-		spec->mixers[spec->num_mixers++] = ad1984_dmic_mixers;
-		codec->patch_ops.build_pcms = ad1984_build_pcms;
-		break;
-	case AD1984_THINKPAD:
-		if (codec->subsystem_id == 0x17aa20fb) {
-			/* Thinpad X300 does not have the ability to do SPDIF,
-			   or attach to docking station to use SPDIF */
-			spec->multiout.dig_out_nid = 0;
-		} else
-			spec->multiout.dig_out_nid = AD1884_SPDIF_OUT;
-		spec->input_mux = &ad1984_thinkpad_capture_source;
-		spec->mixers[0] = ad1984_thinkpad_mixers;
-		spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs;
-		spec->analog_beep = 1;
-		break;
-	case AD1984_DELL_DESKTOP:
-		spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1984_dell_desktop_capture_source;
-		spec->mixers[0] = ad1984_dell_desktop_mixers;
-		break;
-	}
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1984	ad1884_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
-/*
- * AD1883 / AD1884A / AD1984A / AD1984B
- *
- * port-B (0x14) - front mic-in
- * port-E (0x1c) - rear mic-in
- * port-F (0x16) - CD / ext out
- * port-C (0x15) - rear line-in
- * port-D (0x12) - rear line-out
- * port-A (0x11) - front hp-out
- *
- * AD1984A = AD1884A + digital-mic
- * AD1883 = equivalent with AD1984A
- * AD1984B = AD1984A + extra SPDIF-out
- *
- * FIXME:
- * We share the single DAC for both HP and line-outs (see AD1884/1984).
- */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const hda_nid_t ad1884a_dac_nids[1] = {
-	0x03,
-};
-
-#define ad1884a_adc_nids	ad1884_adc_nids
-#define ad1884a_capsrc_nids	ad1884_capsrc_nids
-
-#define AD1884A_SPDIF_OUT	0x02
-
-static const struct hda_input_mux ad1884a_capture_source = {
-	.num_items = 5,
-	.items = {
-		{ "Front Mic", 0x0 },
-		{ "Mic", 0x4 },
-		{ "Line", 0x1 },
-		{ "CD", 0x2 },
-		{ "Mix", 0x3 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1884a_base_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* SPDIF controls */
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		/* identical with ad1983 */
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-/*
- * initialization verbs
- */
-static const struct hda_verb ad1884a_init_verbs[] = {
-	/* DACs; unmute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	/* Port-A (HP) mixer - route only from analog mixer */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-A pin */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Port-D (Line-out) mixer - route only from analog mixer */
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-D pin */
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Mono-out mixer - route only from analog mixer */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Mono-out pin */
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Port-B (front mic) pin */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Port-C (rear line-in) pin */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Port-E (rear mic) pin */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* no boost */
-	/* Port-F (CD) pin */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Analog mixer; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, /* aux */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* capture sources */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* SPDIF output amp */
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	{ } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1884a_loopbacks[] = {
-	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
-	{ 0x20, HDA_INPUT, 1 }, /* Mic */
-	{ 0x20, HDA_INPUT, 2 }, /* CD */
-	{ 0x20, HDA_INPUT, 4 }, /* Docking */
-	{ } /* end */
-};
-#endif
-
-/*
- * Laptop model
- *
- * Port A: Headphone jack
- * Port B: MIC jack
- * Port C: Internal MIC
- * Port D: Dock Line Out (if enabled)
- * Port E: Dock Line In (if enabled)
- * Port F: Internal speakers
- */
-
-static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
-					struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
-	int mute = (!ucontrol->value.integer.value[0] &&
-		    !ucontrol->value.integer.value[1]);
-	/* toggle GPIO1 according to the mute state */
-	snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
-			    mute ? 0x02 : 0x0);
-	return ret;
-}
-
-static const struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = ad1884a_mobile_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
-	},
-	HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	/*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = ad1884a_mobile_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
-	},
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-/* mute internal speaker if HP is plugged */
-static void ad1884a_hp_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x11);
-	snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-	snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
-			    present ? 0x00 : 0x02);
-}
-
-/* switch to external mic if plugged */
-static void ad1884a_hp_automic(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x14);
-	snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL,
-			    present ? 0 : 1);
-}
-
-#define AD1884A_HP_EVENT		0x37
-#define AD1884A_MIC_EVENT		0x36
-
-/* unsolicited event for HP jack sensing */
-static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	switch (res >> 26) {
-	case AD1884A_HP_EVENT:
-		ad1884a_hp_automute(codec);
-		break;
-	case AD1884A_MIC_EVENT:
-		ad1884a_hp_automic(codec);
-		break;
-	}
-}
-
-/* initialize jack-sensing, too */
-static int ad1884a_hp_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1884a_hp_automute(codec);
-	ad1884a_hp_automic(codec);
-	return 0;
-}
-
-/* mute internal speaker if HP or docking HP is plugged */
-static void ad1884a_laptop_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x11);
-	if (!present)
-		present = snd_hda_jack_detect(codec, 0x12);
-	snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-	snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
-			    present ? 0x00 : 0x02);
-}
-
-/* switch to external mic if plugged */
-static void ad1884a_laptop_automic(struct hda_codec *codec)
-{
-	unsigned int idx;
-
-	if (snd_hda_jack_detect(codec, 0x14))
-		idx = 0;
-	else if (snd_hda_jack_detect(codec, 0x1c))
-		idx = 4;
-	else
-		idx = 1;
-	snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, idx);
-}
-
-/* unsolicited event for HP jack sensing */
-static void ad1884a_laptop_unsol_event(struct hda_codec *codec,
-				       unsigned int res)
-{
-	switch (res >> 26) {
-	case AD1884A_HP_EVENT:
-		ad1884a_laptop_automute(codec);
-		break;
-	case AD1884A_MIC_EVENT:
-		ad1884a_laptop_automic(codec);
-		break;
-	}
-}
-
-/* initialize jack-sensing, too */
-static int ad1884a_laptop_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1884a_laptop_automute(codec);
-	ad1884a_laptop_automic(codec);
-	return 0;
-}
-
-/* additional verbs for laptop model */
-static const struct hda_verb ad1884a_laptop_verbs[] = {
-	/* Port-A (HP) pin - always unmuted */
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-F (int speaker) mixer - route only from analog mixer */
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-F (int speaker) pin */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* required for compaq 6530s/6531s speaker output */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Port-C pin - internal mic-in */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
-	/* Port-D (docking line-out) pin - default unmuted */
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* analog mix */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* unsolicited event for pin-sense */
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	{0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
-	{0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
-	/* allow to touch GPIO1 (for mute control) */
-	{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
-	{ } /* end */
-};
-
-static const struct hda_verb ad1884a_mobile_verbs[] = {
-	/* DACs; unmute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	/* Port-A (HP) mixer - route only from analog mixer */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-A pin */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* Port-A (HP) pin - always unmuted */
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-B (mic jack) pin */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
-	/* Port-C (int mic) pin */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
-	/* Port-F (int speaker) mixer - route only from analog mixer */
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-F pin */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Analog mixer; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* capture sources */
-	/* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* unsolicited event for pin-sense */
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
-	/* allow to touch GPIO1 (for mute control) */
-	{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
-	{ } /* end */
-};
-
-/*
- * Thinkpad X300
- * 0x11 - HP
- * 0x12 - speaker
- * 0x14 - mic-in
- * 0x17 - built-in mic
- */
-
-static const struct hda_verb ad1984a_thinkpad_verbs[] = {
-	/* HP unmute */
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* analog mix */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* turn on EAPD */
-	{0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
-	/* unsolicited event for pin-sense */
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	/* internal mic - dmic */
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	/* set magic COEFs for dmic */
-	{0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
-	{0x01, AC_VERB_SET_PROC_COEF, 0x08},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-static const struct hda_input_mux ad1984a_thinkpad_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Internal Mic", 0x5 },
-		{ "Mix", 0x3 },
-	},
-};
-
-/* mute internal speaker if HP is plugged */
-static void ad1984a_thinkpad_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x11);
-	snd_hda_codec_amp_stereo(codec, 0x12, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-/* unsolicited event for HP jack sensing */
-static void ad1984a_thinkpad_unsol_event(struct hda_codec *codec,
-					 unsigned int res)
-{
-	if ((res >> 26) != AD1884A_HP_EVENT)
-		return;
-	ad1984a_thinkpad_automute(codec);
-}
-
-/* initialize jack-sensing, too */
-static int ad1984a_thinkpad_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1984a_thinkpad_automute(codec);
-	return 0;
-}
-
-/*
- * Precision R5500
- * 0x12 - HP/line-out
- * 0x13 - speaker (mono)
- * 0x15 - mic-in
- */
-
-static const struct hda_verb ad1984a_precision_verbs[] = {
-	/* Unmute main output path */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x1f}, /* 0dB */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) + 0x17}, /* 0dB */
-	/* Analog mixer; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* Select mic as input */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x27}, /* 0dB */
-	/* Configure as mic */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
-	/* HP unmute */
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* turn on EAPD */
-	{0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
-	/* unsolicited event for pin-sense */
-	{0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984a_precision_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-
-/* mute internal speaker if HP is plugged */
-static void ad1984a_precision_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x12);
-	snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-
-/* unsolicited event for HP jack sensing */
-static void ad1984a_precision_unsol_event(struct hda_codec *codec,
-					 unsigned int res)
-{
-	if ((res >> 26) != AD1884A_HP_EVENT)
-		return;
-	ad1984a_precision_automute(codec);
-}
-
-/* initialize jack-sensing, too */
-static int ad1984a_precision_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1984a_precision_automute(codec);
-	return 0;
-}
-
-
-/*
- * HP Touchsmart
- * port-A (0x11)      - front hp-out
- * port-B (0x14)      - unused
- * port-C (0x15)      - unused
- * port-D (0x12)      - rear line out
- * port-E (0x1c)      - front mic-in
- * port-F (0x16)      - Internal speakers
- * digital-mic (0x17) - Internal mic
- */
-
-static const struct hda_verb ad1984a_touchsmart_verbs[] = {
-	/* DACs; unmute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	/* Port-A (HP) mixer - route only from analog mixer */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-A pin */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* Port-A (HP) pin - always unmuted */
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-E (int speaker) mixer - route only from analog mixer */
-	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03},
-	/* Port-E pin */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	/* Port-F (int speaker) mixer - route only from analog mixer */
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-F pin */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Analog mixer; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* capture sources */
-	/* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* unsolicited event for pin-sense */
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	{0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
-	/* allow to touch GPIO1 (for mute control) */
-	{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
-	/* internal mic - dmic */
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	/* set magic COEFs for dmic */
-	{0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
-	{0x01, AC_VERB_SET_PROC_COEF, 0x08},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-/*	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.name = "Master Playback Switch",
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = ad1884a_mobile_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
-	},
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-/* switch to external mic if plugged */
-static void ad1984a_touchsmart_automic(struct hda_codec *codec)
-{
-	if (snd_hda_jack_detect(codec, 0x1c))
-		snd_hda_codec_write(codec, 0x0c, 0,
-				     AC_VERB_SET_CONNECT_SEL, 0x4);
-	else
-		snd_hda_codec_write(codec, 0x0c, 0,
-				     AC_VERB_SET_CONNECT_SEL, 0x5);
-}
-
-
-/* unsolicited event for HP jack sensing */
-static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec,
-	unsigned int res)
-{
-	switch (res >> 26) {
-	case AD1884A_HP_EVENT:
-		ad1884a_hp_automute(codec);
-		break;
-	case AD1884A_MIC_EVENT:
-		ad1984a_touchsmart_automic(codec);
-		break;
-	}
-}
-
-/* initialize jack-sensing, too */
-static int ad1984a_touchsmart_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1884a_hp_automute(codec);
-	ad1984a_touchsmart_automic(codec);
-	return 0;
-}
-
-
-/*
- */
-
-enum {
-	AD1884A_AUTO,
-	AD1884A_DESKTOP,
-	AD1884A_LAPTOP,
-	AD1884A_MOBILE,
-	AD1884A_THINKPAD,
-	AD1984A_TOUCHSMART,
-	AD1984A_PRECISION,
-	AD1884A_MODELS
-};
-
-static const char * const ad1884a_models[AD1884A_MODELS] = {
-	[AD1884A_AUTO]		= "auto",
-	[AD1884A_DESKTOP]	= "desktop",
-	[AD1884A_LAPTOP]	= "laptop",
-	[AD1884A_MOBILE]	= "mobile",
-	[AD1884A_THINKPAD]	= "thinkpad",
-	[AD1984A_TOUCHSMART]	= "touchsmart",
-	[AD1984A_PRECISION]	= "precision",
-};
-
-static const struct snd_pci_quirk ad1884a_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1028, 0x04ac, "Precision R5500", AD1984A_PRECISION),
-	SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
-	SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP),
-	SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
-	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE),
-	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP),
-	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP),
-	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
-	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE),
-	SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
-	SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART),
-	{}
-};
-
-static int patch_ad1884a(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err, board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1884A_MODELS,
-						  ad1884a_models,
-						  ad1884a_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1884A_AUTO;
-	}
-
-	if (board_config == AD1884A_AUTO)
-		return ad1884_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 2;
-	spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids);
-	spec->multiout.dac_nids = ad1884a_dac_nids;
-	spec->multiout.dig_out_nid = AD1884A_SPDIF_OUT;
-	spec->num_adc_nids = ARRAY_SIZE(ad1884a_adc_nids);
-	spec->adc_nids = ad1884a_adc_nids;
-	spec->capsrc_nids = ad1884a_capsrc_nids;
-	spec->input_mux = &ad1884a_capture_source;
-	spec->num_mixers = 1;
-	spec->mixers[0] = ad1884a_base_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1884a_init_verbs;
-	spec->spdif_route = 0;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1884a_loopbacks;
-#endif
-	codec->patch_ops = ad198x_patch_ops;
-
-	/* override some parameters */
-	switch (board_config) {
-	case AD1884A_LAPTOP:
-		spec->mixers[0] = ad1884a_laptop_mixers;
-		spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
-		spec->multiout.dig_out_nid = 0;
-		codec->patch_ops.unsol_event = ad1884a_laptop_unsol_event;
-		codec->patch_ops.init = ad1884a_laptop_init;
-		/* set the upper-limit for mixer amp to 0dB for avoiding the
-		 * possible damage by overloading
-		 */
-		snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
-					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
-					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-					  (1 << AC_AMPCAP_MUTE_SHIFT));
-		break;
-	case AD1884A_MOBILE:
-		spec->mixers[0] = ad1884a_mobile_mixers;
-		spec->init_verbs[0] = ad1884a_mobile_verbs;
-		spec->multiout.dig_out_nid = 0;
-		codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
-		codec->patch_ops.init = ad1884a_hp_init;
-		/* set the upper-limit for mixer amp to 0dB for avoiding the
-		 * possible damage by overloading
-		 */
-		snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
-					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
-					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-					  (1 << AC_AMPCAP_MUTE_SHIFT));
-		break;
-	case AD1884A_THINKPAD:
-		spec->mixers[0] = ad1984a_thinkpad_mixers;
-		spec->init_verbs[spec->num_init_verbs++] =
-			ad1984a_thinkpad_verbs;
-		spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1984a_thinkpad_capture_source;
-		codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event;
-		codec->patch_ops.init = ad1984a_thinkpad_init;
-		break;
-	case AD1984A_PRECISION:
-		spec->mixers[0] = ad1984a_precision_mixers;
-		spec->init_verbs[spec->num_init_verbs++] =
-			ad1984a_precision_verbs;
-		spec->multiout.dig_out_nid = 0;
-		codec->patch_ops.unsol_event = ad1984a_precision_unsol_event;
-		codec->patch_ops.init = ad1984a_precision_init;
-		break;
-	case AD1984A_TOUCHSMART:
-		spec->mixers[0] = ad1984a_touchsmart_mixers;
-		spec->init_verbs[0] = ad1984a_touchsmart_verbs;
-		spec->multiout.dig_out_nid = 0;
-		codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event;
-		codec->patch_ops.init = ad1984a_touchsmart_init;
-		/* set the upper-limit for mixer amp to 0dB for avoiding the
-		 * possible damage by overloading
-		 */
-		snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
-					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
-					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-					  (1 << AC_AMPCAP_MUTE_SHIFT));
-		break;
-	}
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1884a	ad1884_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
 /*
  * AD1882 / AD1882A
  *
@@ -4850,299 +1058,7 @@ static int patch_ad1884a(struct hda_codec *codec)
  * port-G - rear clfe-out (6stack)
  */
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const hda_nid_t ad1882_dac_nids[3] = {
-	0x04, 0x03, 0x05
-};
-
-static const hda_nid_t ad1882_adc_nids[2] = {
-	0x08, 0x09,
-};
-
-static const hda_nid_t ad1882_capsrc_nids[2] = {
-	0x0c, 0x0d,
-};
-
-#define AD1882_SPDIF_OUT	0x02
-
-/* list: 0x11, 0x39, 0x3a, 0x18, 0x3c, 0x3b, 0x12, 0x20 */
-static const struct hda_input_mux ad1882_capture_source = {
-	.num_items = 5,
-	.items = {
-		{ "Front Mic", 0x1 },
-		{ "Mic", 0x4 },
-		{ "Line", 0x2 },
-		{ "CD", 0x3 },
-		{ "Mix", 0x7 },
-	},
-};
-
-/* list: 0x11, 0x39, 0x3a, 0x3c, 0x18, 0x1f, 0x12, 0x20 */
-static const struct hda_input_mux ad1882a_capture_source = {
-	.num_items = 5,
-	.items = {
-		{ "Front Mic", 0x1 },
-		{ "Mic", 0x4},
-		{ "Line", 0x2 },
-		{ "Digital Mic", 0x06 },
-		{ "Mix", 0x7 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1882_base_mixers[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line-In Boost Volume", 0x3a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* SPDIF controls */
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		/* identical with ad1983 */
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1882_loopback_mixers[] = {
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1882a_loopback_mixers[] = {
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
-	HDA_CODEC_VOLUME("Digital Mic Boost Volume", 0x1f, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1882_3stack_mixers[] = {
-	HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x17, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x17, 2, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Channel Mode",
-		.info = ad198x_ch_mode_info,
-		.get = ad198x_ch_mode_get,
-		.put = ad198x_ch_mode_put,
-	},
-	{ } /* end */
-};
-
-/* simple auto-mute control for AD1882 3-stack board */
-#define AD1882_HP_EVENT	0x01
-
-static void ad1882_3stack_automute(struct hda_codec *codec)
-{
-	bool mute = snd_hda_jack_detect(codec, 0x11);
-	snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    mute ? 0 : PIN_OUT);
-}
-
-static int ad1882_3stack_automute_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1882_3stack_automute(codec);
-	return 0;
-}
-
-static void ad1882_3stack_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	switch (res >> 26) {
-	case AD1882_HP_EVENT:
-		ad1882_3stack_automute(codec);
-		break;
-	}
-}
-
-static const struct snd_kcontrol_new ad1882_6stack_mixers[] = {
-	HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x24, 2, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct hda_verb ad1882_ch2_init[] = {
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{ } /* end */
-};
-
-static const struct hda_verb ad1882_ch4_init[] = {
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{ } /* end */
-};
-
-static const struct hda_verb ad1882_ch6_init[] = {
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{ } /* end */
-};
-
-static const struct hda_channel_mode ad1882_modes[3] = {
-	{ 2, ad1882_ch2_init },
-	{ 4, ad1882_ch4_init },
-	{ 6, ad1882_ch6_init },
-};
-
-/*
- * initialization verbs
- */
-static const struct hda_verb ad1882_init_verbs[] = {
-	/* DACs; mute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Port-A (HP) mixer */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-A pin */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* HP selector - select DAC2 */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Port-D (Line-out) mixer */
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-D pin */
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Mono-out mixer */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Mono-out pin */
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Port-B (front mic) pin */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
-	/* Port-C (line-in) pin */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
-	/* Port-C mixer - mute as input */
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	/* Port-E (mic-in) pin */
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
-	/* Port-E mixer - mute as input */
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	/* Port-F (surround) */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Port-G (CLFE) */
-	{0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Analog mixer; mute as default */
-	/* list: 0x39, 0x3a, 0x11, 0x12, 0x3c, 0x3b, 0x18, 0x1a */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-	/* SPDIF output selector */
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	{ } /* end */
-};
-
-static const struct hda_verb ad1882_3stack_automute_verbs[] = {
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1882_HP_EVENT},
-	{ } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1882_loopbacks[] = {
-	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
-	{ 0x20, HDA_INPUT, 1 }, /* Mic */
-	{ 0x20, HDA_INPUT, 4 }, /* Line */
-	{ 0x20, HDA_INPUT, 6 }, /* CD */
-	{ } /* end */
-};
-#endif
-
-/* models */
-enum {
-	AD1882_AUTO,
-	AD1882_3STACK,
-	AD1882_6STACK,
-	AD1882_3STACK_AUTOMUTE,
-	AD1882_MODELS
-};
-
-static const char * const ad1882_models[AD1986A_MODELS] = {
-	[AD1882_AUTO]		= "auto",
-	[AD1882_3STACK]		= "3stack",
-	[AD1882_6STACK]		= "6stack",
-	[AD1882_3STACK_AUTOMUTE] = "3stack-automute",
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-static int ad1882_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1882(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
 	int err;
@@ -5169,110 +1085,20 @@ static int ad1882_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1882(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err, board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1882_MODELS,
-						  ad1882_models, NULL);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1882_AUTO;
-	}
-
-	if (board_config == AD1882_AUTO)
-		return ad1882_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 6;
-	spec->multiout.num_dacs = 3;
-	spec->multiout.dac_nids = ad1882_dac_nids;
-	spec->multiout.dig_out_nid = AD1882_SPDIF_OUT;
-	spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids);
-	spec->adc_nids = ad1882_adc_nids;
-	spec->capsrc_nids = ad1882_capsrc_nids;
-	if (codec->vendor_id == 0x11d41882)
-		spec->input_mux = &ad1882_capture_source;
-	else
-		spec->input_mux = &ad1882a_capture_source;
-	spec->num_mixers = 2;
-	spec->mixers[0] = ad1882_base_mixers;
-	if (codec->vendor_id == 0x11d41882)
-		spec->mixers[1] = ad1882_loopback_mixers;
-	else
-		spec->mixers[1] = ad1882a_loopback_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1882_init_verbs;
-	spec->spdif_route = 0;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1882_loopbacks;
-#endif
-	spec->vmaster_nid = 0x04;
-
-	codec->patch_ops = ad198x_patch_ops;
-
-	/* override some parameters */
-	switch (board_config) {
-	default:
-	case AD1882_3STACK:
-	case AD1882_3STACK_AUTOMUTE:
-		spec->num_mixers = 3;
-		spec->mixers[2] = ad1882_3stack_mixers;
-		spec->channel_mode = ad1882_modes;
-		spec->num_channel_mode = ARRAY_SIZE(ad1882_modes);
-		spec->need_dac_fix = 1;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		if (board_config != AD1882_3STACK) {
-			spec->init_verbs[spec->num_init_verbs++] =
-				ad1882_3stack_automute_verbs;
-			codec->patch_ops.unsol_event = ad1882_3stack_unsol_event;
-			codec->patch_ops.init = ad1882_3stack_automute_init;
-		}
-		break;
-	case AD1882_6STACK:
-		spec->num_mixers = 3;
-		spec->mixers[2] = ad1882_6stack_mixers;
-		break;
-	}
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1882	ad1882_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /*
  * patch entries
  */
 static const struct hda_codec_preset snd_hda_preset_analog[] = {
-	{ .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a },
+	{ .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884 },
 	{ .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 },
-	{ .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a },
+	{ .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884 },
 	{ .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 },
-	{ .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884a },
-	{ .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884a },
+	{ .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884 },
+	{ .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884 },
 	{ .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 },
 	{ .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 },
-	{ .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 },
+	{ .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1884 },
 	{ .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },
 	{ .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 },
 	{ .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 },
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index de00ce166470..4edd2d0f9a3c 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -66,6 +66,8 @@ struct conexant_spec {
 	hda_nid_t eapds[4];
 	bool dynamic_eapd;
 
+	unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */
+
 #ifdef ENABLE_CXT_STATIC_QUIRKS
 	const struct snd_kcontrol_new *mixers[5];
 	int num_mixers;
@@ -3200,6 +3202,9 @@ static int cx_auto_init(struct hda_codec *codec)
 	snd_hda_gen_init(codec);
 	if (!spec->dynamic_eapd)
 		cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
+
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT);
+
 	return 0;
 }
 
@@ -3224,6 +3229,8 @@ enum {
 	CXT_PINCFG_LEMOTE_A1205,
 	CXT_FIXUP_STEREO_DMIC,
 	CXT_FIXUP_INC_MIC_BOOST,
+	CXT_FIXUP_HEADPHONE_MIC_PIN,
+	CXT_FIXUP_HEADPHONE_MIC,
 };
 
 static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
@@ -3246,6 +3253,59 @@ static void cxt5066_increase_mic_boost(struct hda_codec *codec,
 				  (0 << AC_AMPCAP_MUTE_SHIFT));
 }
 
+static void cxt_update_headset_mode(struct hda_codec *codec)
+{
+	/* The verbs used in this function were tested on a Conexant CX20751/2 codec. */
+	int i;
+	bool mic_mode = false;
+	struct conexant_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+
+	hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]];
+
+	for (i = 0; i < cfg->num_inputs; i++)
+		if (cfg->inputs[i].pin == mux_pin) {
+			mic_mode = !!cfg->inputs[i].is_headphone_mic;
+			break;
+		}
+
+	if (mic_mode) {
+		snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x7c); /* enable merged mode for analog int-mic */
+		spec->gen.hp_jack_present = false;
+	} else {
+		snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x54); /* disable merged mode for analog int-mic */
+		spec->gen.hp_jack_present = snd_hda_jack_detect(codec, spec->gen.autocfg.hp_pins[0]);
+	}
+
+	snd_hda_gen_update_outputs(codec);
+}
+
+static void cxt_update_headset_mode_hook(struct hda_codec *codec,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	cxt_update_headset_mode(codec);
+}
+
+static void cxt_fixup_headphone_mic(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	struct conexant_spec *spec = codec->spec;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		spec->parse_flags |= HDA_PINCFG_HEADPHONE_MIC;
+		break;
+	case HDA_FIXUP_ACT_PROBE:
+		spec->gen.cap_sync_hook = cxt_update_headset_mode_hook;
+		spec->gen.automute_hook = cxt_update_headset_mode;
+		break;
+	case HDA_FIXUP_ACT_INIT:
+		cxt_update_headset_mode(codec);
+		break;
+	}
+}
+
+
 /* ThinkPad X200 & co with cxt5051 */
 static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = {
 	{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
@@ -3302,6 +3362,19 @@ static const struct hda_fixup cxt_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cxt5066_increase_mic_boost,
 	},
+	[CXT_FIXUP_HEADPHONE_MIC_PIN] = {
+		.type = HDA_FIXUP_PINS,
+		.chained = true,
+		.chain_id = CXT_FIXUP_HEADPHONE_MIC,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, 0x03a1913d }, /* use as headphone mic, without its own jack detect */
+			{ }
+		}
+	},
+	[CXT_FIXUP_HEADPHONE_MIC] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = cxt_fixup_headphone_mic,
+	},
 };
 
 static const struct snd_pci_quirk cxt5051_fixups[] = {
@@ -3311,6 +3384,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = {
 
 static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
@@ -3395,7 +3469,8 @@ static int patch_conexant_auto(struct hda_codec *codec)
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
-	err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0);
+	err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL,
+				       spec->parse_flags);
 	if (err < 0)
 		goto error;
 
@@ -3416,6 +3491,8 @@ static int patch_conexant_auto(struct hda_codec *codec)
 		codec->bus->allow_bus_reset = 1;
 	}
 
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
+
 	return 0;
 
  error:
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 9f3586276871..9a58893d52a7 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -67,6 +67,8 @@ struct hdmi_spec_per_pin {
 	struct delayed_work work;
 	struct snd_kcontrol *eld_ctl;
 	int repoll_count;
+	bool setup; /* the stream has been set up by prepare callback */
+	int channels; /* current number of channels */
 	bool non_pcm;
 	bool chmap_set;		/* channel-map override by ALSA API? */
 	unsigned char chmap[8]; /* ALSA API channel-map */
@@ -551,6 +553,17 @@ static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels)
 		}
 	}
 
+	if (!ca) {
+		/* if there was no match, select the regular ALSA channel
+		 * allocation with the matching number of channels */
+		for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) {
+			if (channels == channel_allocations[i].channels) {
+				ca = channel_allocations[i].ca_index;
+				break;
+			}
+		}
+	}
+
 	snd_print_channel_allocation(eld->info.spk_alloc, buf, sizeof(buf));
 	snd_printdd("HDMI: select CA 0x%x for %d-channel allocation: %s\n",
 		    ca, channels, buf);
@@ -868,18 +881,19 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid,
 	return true;
 }
 
-static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx,
-				       bool non_pcm,
-				       struct snd_pcm_substream *substream)
+static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
+				       struct hdmi_spec_per_pin *per_pin,
+				       bool non_pcm)
 {
-	struct hdmi_spec *spec = codec->spec;
-	struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
 	hda_nid_t pin_nid = per_pin->pin_nid;
-	int channels = substream->runtime->channels;
+	int channels = per_pin->channels;
 	struct hdmi_eld *eld;
 	int ca;
 	union audio_infoframe ai;
 
+	if (!channels)
+		return;
+
 	eld = &per_pin->sink_eld;
 	if (!eld->monitor_present)
 		return;
@@ -959,6 +973,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	int pin_nid;
 	int pin_idx;
 	struct hda_jack_tbl *jack;
+	int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT;
 
 	jack = snd_hda_jack_tbl_get_from_tag(codec, tag);
 	if (!jack)
@@ -967,8 +982,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	jack->jack_dirty = 1;
 
 	_snd_printd(SND_PR_VERBOSE,
-		"HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
-		codec->addr, pin_nid,
+		"HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n",
+		codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA),
 		!!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV));
 
 	pin_idx = pin_nid_to_pin_index(spec, pin_nid);
@@ -1329,6 +1344,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 		eld_changed = true;
 	}
 	if (update_eld) {
+		bool old_eld_valid = pin_eld->eld_valid;
 		pin_eld->eld_valid = eld->eld_valid;
 		eld_changed = pin_eld->eld_size != eld->eld_size ||
 			      memcmp(pin_eld->eld_buffer, eld->eld_buffer,
@@ -1338,6 +1354,18 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 			       eld->eld_size);
 		pin_eld->eld_size = eld->eld_size;
 		pin_eld->info = eld->info;
+
+		/* Haswell-specific workaround: re-setup when the transcoder is
+		 * changed during the stream playback
+		 */
+		if (codec->vendor_id == 0x80862807 &&
+		    eld->eld_valid && !old_eld_valid && per_pin->setup) {
+			snd_hda_codec_write(codec, pin_nid, 0,
+					    AC_VERB_SET_AMP_GAIN_MUTE,
+					    AMP_OUT_UNMUTE);
+			hdmi_setup_audio_infoframe(codec, per_pin,
+						   per_pin->non_pcm);
+		}
 	}
 	mutex_unlock(&pin_eld->lock);
 
@@ -1510,14 +1538,17 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 	hda_nid_t cvt_nid = hinfo->nid;
 	struct hdmi_spec *spec = codec->spec;
 	int pin_idx = hinfo_to_pin_index(spec, hinfo);
-	hda_nid_t pin_nid = get_pin(spec, pin_idx)->pin_nid;
+	struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
+	hda_nid_t pin_nid = per_pin->pin_nid;
 	bool non_pcm;
 
 	non_pcm = check_non_pcm_per_cvt(codec, cvt_nid);
+	per_pin->channels = substream->runtime->channels;
+	per_pin->setup = true;
 
 	hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels);
 
-	hdmi_setup_audio_infoframe(codec, pin_idx, non_pcm, substream);
+	hdmi_setup_audio_infoframe(codec, per_pin, non_pcm);
 
 	return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format);
 }
@@ -1557,6 +1588,9 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
 		snd_hda_spdif_ctls_unassign(codec, pin_idx);
 		per_pin->chmap_set = false;
 		memset(per_pin->chmap, 0, sizeof(per_pin->chmap));
+
+		per_pin->setup = false;
+		per_pin->channels = 0;
 	}
 
 	return 0;
@@ -1692,8 +1726,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol,
 	per_pin->chmap_set = true;
 	memcpy(per_pin->chmap, chmap, sizeof(chmap));
 	if (prepared)
-		hdmi_setup_audio_infoframe(codec, pin_idx, per_pin->non_pcm,
-					   substream);
+		hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm);
 
 	return 0;
 }
@@ -1992,8 +2025,10 @@ static int patch_generic_hdmi(struct hda_codec *codec)
 		return -EINVAL;
 	}
 	codec->patch_ops = generic_hdmi_patch_ops;
-	if (codec->vendor_id == 0x80862807)
+	if (codec->vendor_id == 0x80862807) {
 		codec->patch_ops.set_power_state = haswell_set_power_state;
+		codec->dp_mst = true;
+	}
 
 	generic_hdmi_init_per_pins(codec);
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 389db4c2801b..9e9378cde8fa 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -282,6 +282,7 @@ static void alc_eapd_shutup(struct hda_codec *codec)
 {
 	alc_auto_setup_eapd(codec, false);
 	msleep(200);
+	snd_hda_shutup_pins(codec);
 }
 
 /* generic EAPD initialization */
@@ -826,7 +827,8 @@ static inline void alc_shutup(struct hda_codec *codec)
 
 	if (spec && spec->shutup)
 		spec->shutup(codec);
-	snd_hda_shutup_pins(codec);
+	else
+		snd_hda_shutup_pins(codec);
 }
 
 #define alc_free	snd_hda_gen_free
@@ -1853,8 +1855,10 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec,
 				       const struct hda_fixup *fix, int action)
 {
 	struct alc_spec *spec = codec->spec;
-	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->gen.no_primary_hp = 1;
+		spec->gen.no_multi_io = 1;
+	}
 }
 
 static const struct hda_fixup alc882_fixups[] = {
@@ -2533,6 +2537,7 @@ enum {
 	ALC269_TYPE_ALC269VD,
 	ALC269_TYPE_ALC280,
 	ALC269_TYPE_ALC282,
+	ALC269_TYPE_ALC283,
 	ALC269_TYPE_ALC284,
 	ALC269_TYPE_ALC286,
 };
@@ -2558,6 +2563,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	case ALC269_TYPE_ALC269VB:
 	case ALC269_TYPE_ALC269VD:
 	case ALC269_TYPE_ALC282:
+	case ALC269_TYPE_ALC283:
 	case ALC269_TYPE_ALC286:
 		ssids = alc269_ssids;
 		break;
@@ -2583,15 +2589,81 @@ static void alc269_shutup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (spec->codec_variant != ALC269_TYPE_ALC269VB)
-		return;
-
 	if (spec->codec_variant == ALC269_TYPE_ALC269VB)
 		alc269vb_toggle_power_output(codec, 0);
 	if (spec->codec_variant == ALC269_TYPE_ALC269VB &&
 			(alc_get_coef0(codec) & 0x00ff) == 0x018) {
 		msleep(150);
 	}
+	snd_hda_shutup_pins(codec);
+}
+
+static void alc283_init(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0];
+	bool hp_pin_sense;
+	int val;
+
+	if (!hp_pin)
+		return;
+	hp_pin_sense = snd_hda_jack_detect(codec, hp_pin);
+
+	/* Index 0x43 Direct Drive HP AMP LPM Control 1 */
+	/* Headphone capless set to high power mode */
+	alc_write_coef_idx(codec, 0x43, 0x9004);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+	if (hp_pin_sense)
+		msleep(85);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+
+	if (hp_pin_sense)
+		msleep(85);
+	/* Index 0x46 Combo jack auto switch control 2 */
+	/* 3k pull low control for Headset jack. */
+	val = alc_read_coef_idx(codec, 0x46);
+	alc_write_coef_idx(codec, 0x46, val & ~(3 << 12));
+	/* Headphone capless set to normal mode */
+	alc_write_coef_idx(codec, 0x43, 0x9614);
+}
+
+static void alc283_shutup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0];
+	bool hp_pin_sense;
+	int val;
+
+	if (!hp_pin) {
+		alc269_shutup(codec);
+		return;
+	}
+
+	hp_pin_sense = snd_hda_jack_detect(codec, hp_pin);
+
+	alc_write_coef_idx(codec, 0x43, 0x9004);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+	if (hp_pin_sense)
+		msleep(85);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0);
+
+	val = alc_read_coef_idx(codec, 0x46);
+	alc_write_coef_idx(codec, 0x46, val | (3 << 12));
+
+	if (hp_pin_sense)
+		msleep(85);
+	snd_hda_shutup_pins(codec);
+	alc_write_coef_idx(codec, 0x43, 0x9614);
 }
 
 static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg,
@@ -2722,6 +2794,7 @@ static int alc269_resume(struct hda_codec *codec)
 	hda_call_check_power_status(codec, 0x01);
 	if (spec->has_alc5505_dsp)
 		alc5505_dsp_resume(codec);
+
 	return 0;
 }
 #endif /* CONFIG_PM */
@@ -3261,6 +3334,28 @@ static void alc_fixup_headset_mode_alc668(struct hda_codec *codec,
 	alc_fixup_headset_mode(codec, fix, action);
 }
 
+/* Returns the nid of the external mic input pin, or 0 if it cannot be found. */
+static int find_ext_mic_pin(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+	hda_nid_t nid;
+	unsigned int defcfg;
+	int i;
+
+	for (i = 0; i < cfg->num_inputs; i++) {
+		if (cfg->inputs[i].type != AUTO_PIN_MIC)
+			continue;
+		nid = cfg->inputs[i].pin;
+		defcfg = snd_hda_codec_get_pincfg(codec, nid);
+		if (snd_hda_get_input_pin_attr(defcfg) == INPUT_PIN_ATTR_INT)
+			continue;
+		return nid;
+	}
+
+	return 0;
+}
+
 static void alc271_hp_gate_mic_jack(struct hda_codec *codec,
 				    const struct hda_fixup *fix,
 				    int action)
@@ -3268,11 +3363,12 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec,
 	struct alc_spec *spec = codec->spec;
 
 	if (action == HDA_FIXUP_ACT_PROBE) {
-		if (snd_BUG_ON(!spec->gen.am_entry[1].pin ||
-			       !spec->gen.autocfg.hp_pins[0]))
+		int mic_pin = find_ext_mic_pin(codec);
+		int hp_pin = spec->gen.autocfg.hp_pins[0];
+
+		if (snd_BUG_ON(!mic_pin || !hp_pin))
 			return;
-		snd_hda_jack_set_gating_jack(codec, spec->gen.am_entry[1].pin,
-					     spec->gen.autocfg.hp_pins[0]);
+		snd_hda_jack_set_gating_jack(codec, mic_pin, hp_pin);
 	}
 }
 
@@ -3308,6 +3404,45 @@ static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec,
 	}
 }
 
+static void alc283_hp_automute_hook(struct hda_codec *codec,
+				    struct hda_jack_tbl *jack)
+{
+	struct alc_spec *spec = codec->spec;
+	int vref;
+
+	msleep(200);
+	snd_hda_gen_hp_automute(codec, jack);
+
+	vref = spec->gen.hp_jack_present ? PIN_VREF80 : 0;
+
+	msleep(600);
+	snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			    vref);
+}
+
+static void alc283_chromebook_caps(struct hda_codec *codec)
+{
+	snd_hda_override_wcaps(codec, 0x03, 0);
+}
+
+static void alc283_fixup_chromebook(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+	int val;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		alc283_chromebook_caps(codec);
+		spec->gen.hp_automute_hook = alc283_hp_automute_hook;
+		/* MIC2-VREF control */
+		/* Set to manual mode */
+		val = alc_read_coef_idx(codec, 0x06);
+		alc_write_coef_idx(codec, 0x06, val & ~0x000c);
+		break;
+	}
+}
+
 enum {
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -3344,6 +3479,7 @@ enum {
 	ALC269_FIXUP_ACER_AC700,
 	ALC269_FIXUP_LIMIT_INT_MIC_BOOST,
 	ALC269VB_FIXUP_ORDISSIMO_EVE2,
+	ALC283_FIXUP_CHROME_BOOK,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -3595,11 +3731,20 @@ static const struct hda_fixup alc269_fixups[] = {
 			{ }
 		},
 	},
+	[ALC283_FIXUP_CHROME_BOOK] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc283_fixup_chromebook,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC),
+	SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),
+	SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
+	SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
+	SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
+	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
 	SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05c4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
@@ -3637,6 +3782,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+	SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK),
 	SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -3655,11 +3801,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
-	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
-	SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),
-	SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
-	SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
-	SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
 	SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
@@ -3670,8 +3811,16 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
-	SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
 	SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
@@ -3840,11 +3989,15 @@ static int patch_alc269(struct hda_codec *codec)
 	case 0x10ec0290:
 		spec->codec_variant = ALC269_TYPE_ALC280;
 		break;
-	case 0x10ec0233:
 	case 0x10ec0282:
-	case 0x10ec0283:
 		spec->codec_variant = ALC269_TYPE_ALC282;
 		break;
+	case 0x10ec0233:
+	case 0x10ec0283:
+		spec->codec_variant = ALC269_TYPE_ALC283;
+		spec->shutup = alc283_shutup;
+		spec->init_hook = alc283_init;
+		break;
 	case 0x10ec0284:
 	case 0x10ec0292:
 		spec->codec_variant = ALC269_TYPE_ALC284;
@@ -3872,7 +4025,8 @@ static int patch_alc269(struct hda_codec *codec)
 	codec->patch_ops.suspend = alc269_suspend;
 	codec->patch_ops.resume = alc269_resume;
 #endif
-	spec->shutup = alc269_shutup;
+	if (!spec->shutup)
+		spec->shutup = alc269_shutup;
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6d1924c19abf..fba0cef1c47f 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -158,6 +158,7 @@ enum {
 	STAC_D965_VERBS,
 	STAC_DELL_3ST,
 	STAC_DELL_BIOS,
+	STAC_DELL_BIOS_AMIC,
 	STAC_DELL_BIOS_SPDIF,
 	STAC_927X_DELL_DMIC,
 	STAC_927X_VOLKNOB,
@@ -3231,8 +3232,6 @@ static const struct hda_fixup stac927x_fixups[] = {
 	[STAC_DELL_BIOS] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
-			/* configure the analog microphone on some laptops */
-			{ 0x0c, 0x90a79130 },
 			/* correct the front output jack as a hp out */
 			{ 0x0f, 0x0221101f },
 			/* correct the front input jack as a mic */
@@ -3242,6 +3241,16 @@ static const struct hda_fixup stac927x_fixups[] = {
 		.chained = true,
 		.chain_id = STAC_927X_DELL_DMIC,
 	},
+	[STAC_DELL_BIOS_AMIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			/* configure the analog microphone on some laptops */
+			{ 0x0c, 0x90a79130 },
+			{}
+		},
+		.chained = true,
+		.chain_id = STAC_DELL_BIOS,
+	},
 	[STAC_DELL_BIOS_SPDIF] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -3270,6 +3279,7 @@ static const struct hda_model_fixup stac927x_models[] = {
 	{ .id = STAC_D965_5ST_NO_FP, .name = "5stack-no-fp" },
 	{ .id = STAC_DELL_3ST, .name = "dell-3stack" },
 	{ .id = STAC_DELL_BIOS, .name = "dell-bios" },
+	{ .id = STAC_DELL_BIOS_AMIC, .name = "dell-bios-amic" },
 	{ .id = STAC_927X_VOLKNOB, .name = "volknob" },
 	{}
 };
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index e2481baddc70..0bc20ef5687a 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -207,9 +207,9 @@ static void vt1708_stop_hp_work(struct hda_codec *codec)
 		return;
 	if (spec->hp_work_active) {
 		snd_hda_codec_write(codec, 0x1, 0, 0xf81, 1);
+		codec->jackpoll_interval = 0;
 		cancel_delayed_work_sync(&codec->jackpoll_work);
 		spec->hp_work_active = false;
-		codec->jackpoll_interval = 0;
 	}
 }
 
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 2a8ad9d1a2ae..bb9ebc5543d7 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -28,6 +28,7 @@
 #include <linux/interrupt.h>
 #include <linux/pci.h>
 #include <linux/module.h>
+#include <linux/vmalloc.h>
 
 #include <sound/core.h>
 #include <sound/info.h>
@@ -198,6 +199,31 @@ MODULE_PARM_DESC(enable, "Enable RME Digi96 soundcard.");
 #define RME96_AD1852_VOL_BITS 14
 #define RME96_AD1855_VOL_BITS 10
 
+/* Defines for snd_rme96_trigger */
+#define RME96_TB_START_PLAYBACK 1
+#define RME96_TB_START_CAPTURE 2
+#define RME96_TB_STOP_PLAYBACK 4
+#define RME96_TB_STOP_CAPTURE 8
+#define RME96_TB_RESET_PLAYPOS 16
+#define RME96_TB_RESET_CAPTUREPOS 32
+#define RME96_TB_CLEAR_PLAYBACK_IRQ 64
+#define RME96_TB_CLEAR_CAPTURE_IRQ 128
+#define RME96_RESUME_PLAYBACK	(RME96_TB_START_PLAYBACK)
+#define RME96_RESUME_CAPTURE	(RME96_TB_START_CAPTURE)
+#define RME96_RESUME_BOTH	(RME96_RESUME_PLAYBACK \
+				| RME96_RESUME_CAPTURE)
+#define RME96_START_PLAYBACK	(RME96_TB_START_PLAYBACK \
+				| RME96_TB_RESET_PLAYPOS)
+#define RME96_START_CAPTURE	(RME96_TB_START_CAPTURE \
+				| RME96_TB_RESET_CAPTUREPOS)
+#define RME96_START_BOTH	(RME96_START_PLAYBACK \
+				| RME96_START_CAPTURE)
+#define RME96_STOP_PLAYBACK	(RME96_TB_STOP_PLAYBACK \
+				| RME96_TB_CLEAR_PLAYBACK_IRQ)
+#define RME96_STOP_CAPTURE	(RME96_TB_STOP_CAPTURE \
+				| RME96_TB_CLEAR_CAPTURE_IRQ)
+#define RME96_STOP_BOTH		(RME96_STOP_PLAYBACK \
+				| RME96_STOP_CAPTURE)
 
 struct rme96 {
 	spinlock_t    lock;
@@ -214,6 +240,13 @@ struct rme96 {
 
 	u8 rev; /* card revision number */
 
+#ifdef CONFIG_PM
+	u32 playback_pointer;
+	u32 capture_pointer;
+	void *playback_suspend_buffer;
+	void *capture_suspend_buffer;
+#endif
+
 	struct snd_pcm_substream *playback_substream;
 	struct snd_pcm_substream *capture_substream;
 
@@ -344,6 +377,8 @@ static struct snd_pcm_hardware snd_rme96_playback_spdif_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@@ -373,6 +408,8 @@ static struct snd_pcm_hardware snd_rme96_capture_spdif_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@@ -402,6 +439,8 @@ static struct snd_pcm_hardware snd_rme96_playback_adat_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@@ -427,6 +466,8 @@ static struct snd_pcm_hardware snd_rme96_capture_adat_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@@ -1045,54 +1086,35 @@ snd_rme96_capture_hw_params(struct snd_pcm_substream *substream,
 }
 
 static void
-snd_rme96_playback_start(struct rme96 *rme96,
-			 int from_pause)
+snd_rme96_trigger(struct rme96 *rme96,
+		  int op)
 {
-	if (!from_pause) {
+	if (op & RME96_TB_RESET_PLAYPOS)
 		writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS);
-	}
-
-	rme96->wcreg |= RME96_WCR_START;
-	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
-
-static void
-snd_rme96_capture_start(struct rme96 *rme96,
-			int from_pause)
-{
-	if (!from_pause) {
+	if (op & RME96_TB_RESET_CAPTUREPOS)
 		writel(0, rme96->iobase + RME96_IO_RESET_REC_POS);
-	}
-
-	rme96->wcreg |= RME96_WCR_START_2;
+	if (op & RME96_TB_CLEAR_PLAYBACK_IRQ) {
+		rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+		if (rme96->rcreg & RME96_RCR_IRQ)
+			writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ);
+	}
+	if (op & RME96_TB_CLEAR_CAPTURE_IRQ) {
+		rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+		if (rme96->rcreg & RME96_RCR_IRQ_2)
+			writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ);
+	}
+	if (op & RME96_TB_START_PLAYBACK)
+		rme96->wcreg |= RME96_WCR_START;
+	if (op & RME96_TB_STOP_PLAYBACK)
+		rme96->wcreg &= ~RME96_WCR_START;
+	if (op & RME96_TB_START_CAPTURE)
+		rme96->wcreg |= RME96_WCR_START_2;
+	if (op & RME96_TB_STOP_CAPTURE)
+		rme96->wcreg &= ~RME96_WCR_START_2;
 	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
 }
 
-static void
-snd_rme96_playback_stop(struct rme96 *rme96)
-{
-	/*
-	 * Check if there is an unconfirmed IRQ, if so confirm it, or else
-	 * the hardware will not stop generating interrupts
-	 */
-	rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
-	if (rme96->rcreg & RME96_RCR_IRQ) {
-		writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ);
-	}	
-	rme96->wcreg &= ~RME96_WCR_START;
-	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
 
-static void
-snd_rme96_capture_stop(struct rme96 *rme96)
-{
-	rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
-	if (rme96->rcreg & RME96_RCR_IRQ_2) {
-		writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ);
-	}	
-	rme96->wcreg &= ~RME96_WCR_START_2;
-	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
 
 static irqreturn_t
 snd_rme96_interrupt(int irq,
@@ -1155,6 +1177,7 @@ snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
+	snd_pcm_set_sync(substream);
 	spin_lock_irq(&rme96->lock);	
         if (rme96->playback_substream != NULL) {
 		spin_unlock_irq(&rme96->lock);
@@ -1191,6 +1214,7 @@ snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
+	snd_pcm_set_sync(substream);
 	runtime->hw = snd_rme96_capture_spdif_info;
         if (snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG &&
             (rate = snd_rme96_capture_getrate(rme96, &isadat)) > 0)
@@ -1222,6 +1246,7 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;        
 	
+	snd_pcm_set_sync(substream);
 	spin_lock_irq(&rme96->lock);	
         if (rme96->playback_substream != NULL) {
 		spin_unlock_irq(&rme96->lock);
@@ -1253,6 +1278,7 @@ snd_rme96_capture_adat_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
+	snd_pcm_set_sync(substream);
 	runtime->hw = snd_rme96_capture_adat_info;
         if (snd_rme96_getinputtype(rme96) == RME96_INPUT_ANALOG) {
                 /* makes no sense to use analog input. Note that analog
@@ -1288,7 +1314,7 @@ snd_rme96_playback_close(struct snd_pcm_substream *substream)
 
 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISPLAYING(rme96)) {
-		snd_rme96_playback_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK);
 	}
 	rme96->playback_substream = NULL;
 	rme96->playback_periodsize = 0;
@@ -1309,7 +1335,7 @@ snd_rme96_capture_close(struct snd_pcm_substream *substream)
 	
 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISRECORDING(rme96)) {
-		snd_rme96_capture_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_CAPTURE);
 	}
 	rme96->capture_substream = NULL;
 	rme96->capture_periodsize = 0;
@@ -1324,7 +1350,7 @@ snd_rme96_playback_prepare(struct snd_pcm_substream *substream)
 	
 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISPLAYING(rme96)) {
-		snd_rme96_playback_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK);
 	}
 	writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS);
 	spin_unlock_irq(&rme96->lock);
@@ -1338,7 +1364,7 @@ snd_rme96_capture_prepare(struct snd_pcm_substream *substream)
 	
 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISRECORDING(rme96)) {
-		snd_rme96_capture_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_CAPTURE);
 	}
 	writel(0, rme96->iobase + RME96_IO_RESET_REC_POS);
 	spin_unlock_irq(&rme96->lock);
@@ -1350,41 +1376,55 @@ snd_rme96_playback_trigger(struct snd_pcm_substream *substream,
 			   int cmd)
 {
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+	struct snd_pcm_substream *s;
+	bool sync;
+
+	snd_pcm_group_for_each_entry(s, substream) {
+		if (snd_pcm_substream_chip(s) == rme96)
+			snd_pcm_trigger_done(s, substream);
+	}
+
+	sync = (rme96->playback_substream && rme96->capture_substream) &&
+	       (rme96->playback_substream->group ==
+		rme96->capture_substream->group);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		if (!RME96_ISPLAYING(rme96)) {
-			if (substream != rme96->playback_substream) {
+			if (substream != rme96->playback_substream)
 				return -EBUSY;
-			}
-			snd_rme96_playback_start(rme96, 0);
+			snd_rme96_trigger(rme96, sync ? RME96_START_BOTH
+						 : RME96_START_PLAYBACK);
 		}
 		break;
 
+	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		if (RME96_ISPLAYING(rme96)) {
-			if (substream != rme96->playback_substream) {
+			if (substream != rme96->playback_substream)
 				return -EBUSY;
-			}
-			snd_rme96_playback_stop(rme96);
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 :  RME96_STOP_PLAYBACK);
 		}
 		break;
 
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		if (RME96_ISPLAYING(rme96)) {
-			snd_rme96_playback_stop(rme96);
-		}
+		if (RME96_ISPLAYING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 : RME96_STOP_PLAYBACK);
 		break;
 
+	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (!RME96_ISPLAYING(rme96)) {
-			snd_rme96_playback_start(rme96, 1);
-		}
+		if (!RME96_ISPLAYING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH
+						 : RME96_RESUME_PLAYBACK);
 		break;
-		
+
 	default:
 		return -EINVAL;
 	}
+
 	return 0;
 }
 
@@ -1393,38 +1433,51 @@ snd_rme96_capture_trigger(struct snd_pcm_substream *substream,
 			  int cmd)
 {
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+	struct snd_pcm_substream *s;
+	bool sync;
+
+	snd_pcm_group_for_each_entry(s, substream) {
+		if (snd_pcm_substream_chip(s) == rme96)
+			snd_pcm_trigger_done(s, substream);
+	}
+
+	sync = (rme96->playback_substream && rme96->capture_substream) &&
+	       (rme96->playback_substream->group ==
+		rme96->capture_substream->group);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		if (!RME96_ISRECORDING(rme96)) {
-			if (substream != rme96->capture_substream) {
+			if (substream != rme96->capture_substream)
 				return -EBUSY;
-			}
-			snd_rme96_capture_start(rme96, 0);
+			snd_rme96_trigger(rme96, sync ? RME96_START_BOTH
+						 : RME96_START_CAPTURE);
 		}
 		break;
 
+	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		if (RME96_ISRECORDING(rme96)) {
-			if (substream != rme96->capture_substream) {
+			if (substream != rme96->capture_substream)
 				return -EBUSY;
-			}
-			snd_rme96_capture_stop(rme96);
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 : RME96_STOP_CAPTURE);
 		}
 		break;
 
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		if (RME96_ISRECORDING(rme96)) {
-			snd_rme96_capture_stop(rme96);
-		}
+		if (RME96_ISRECORDING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 : RME96_STOP_CAPTURE);
 		break;
 
+	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (!RME96_ISRECORDING(rme96)) {
-			snd_rme96_capture_start(rme96, 1);
-		}
+		if (!RME96_ISRECORDING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH
+						 : RME96_RESUME_CAPTURE);
 		break;
-		
+
 	default:
 		return -EINVAL;
 	}
@@ -1505,8 +1558,7 @@ snd_rme96_free(void *private_data)
 	        return;
 	}
 	if (rme96->irq >= 0) {
-		snd_rme96_playback_stop(rme96);
-		snd_rme96_capture_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_BOTH);
 		rme96->areg &= ~RME96_AR_DAC_EN;
 		writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
 		free_irq(rme96->irq, (void *)rme96);
@@ -1520,6 +1572,10 @@ snd_rme96_free(void *private_data)
 		pci_release_regions(rme96->pci);
 		rme96->port = 0;
 	}
+#ifdef CONFIG_PM
+	vfree(rme96->playback_suspend_buffer);
+	vfree(rme96->capture_suspend_buffer);
+#endif
 	pci_disable_device(rme96->pci);
 }
 
@@ -1606,8 +1662,7 @@ snd_rme96_create(struct rme96 *rme96)
 	rme96->capture_periodsize = 0;
 	
 	/* make sure playback/capture is stopped, if by some reason active */
-	snd_rme96_playback_stop(rme96);
-	snd_rme96_capture_stop(rme96);
+	snd_rme96_trigger(rme96, RME96_STOP_BOTH);
 	
 	/* set default values in registers */
 	rme96->wcreg =
@@ -2319,6 +2374,87 @@ snd_rme96_create_switches(struct snd_card *card,
  * Card initialisation
  */
 
+#ifdef CONFIG_PM
+
+static int
+snd_rme96_suspend(struct pci_dev *pci,
+		  pm_message_t state)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct rme96 *rme96 = card->private_data;
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+	snd_pcm_suspend(rme96->playback_substream);
+	snd_pcm_suspend(rme96->capture_substream);
+
+	/* save capture & playback pointers */
+	rme96->playback_pointer = readl(rme96->iobase + RME96_IO_GET_PLAY_POS)
+				  & RME96_RCR_AUDIO_ADDR_MASK;
+	rme96->capture_pointer = readl(rme96->iobase + RME96_IO_GET_REC_POS)
+				 & RME96_RCR_AUDIO_ADDR_MASK;
+
+	/* save playback and capture buffers */
+	memcpy_fromio(rme96->playback_suspend_buffer,
+		      rme96->iobase + RME96_IO_PLAY_BUFFER, RME96_BUFFER_SIZE);
+	memcpy_fromio(rme96->capture_suspend_buffer,
+		      rme96->iobase + RME96_IO_REC_BUFFER, RME96_BUFFER_SIZE);
+
+	/* disable the DAC  */
+	rme96->areg &= ~RME96_AR_DAC_EN;
+	writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+
+	pci_disable_device(pci);
+	pci_save_state(pci);
+
+	return 0;
+}
+
+static int
+snd_rme96_resume(struct pci_dev *pci)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct rme96 *rme96 = card->private_data;
+
+	pci_restore_state(pci);
+	if (pci_enable_device(pci) < 0) {
+		printk(KERN_ERR "rme96: pci_enable_device failed, disabling device\n");
+		snd_card_disconnect(card);
+		return -EIO;
+	}
+
+	/* reset playback and record buffer pointers */
+	writel(0, rme96->iobase + RME96_IO_SET_PLAY_POS
+		  + rme96->playback_pointer);
+	writel(0, rme96->iobase + RME96_IO_SET_REC_POS
+		  + rme96->capture_pointer);
+
+	/* restore playback and capture buffers */
+	memcpy_toio(rme96->iobase + RME96_IO_PLAY_BUFFER,
+		    rme96->playback_suspend_buffer, RME96_BUFFER_SIZE);
+	memcpy_toio(rme96->iobase + RME96_IO_REC_BUFFER,
+		    rme96->capture_suspend_buffer, RME96_BUFFER_SIZE);
+
+	/* reset the ADC */
+	writel(rme96->areg | RME96_AR_PD2,
+	       rme96->iobase + RME96_IO_ADDITIONAL_REG);
+	writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+
+	/* reset and enable DAC, restore analog volume */
+	snd_rme96_reset_dac(rme96);
+	rme96->areg |= RME96_AR_DAC_EN;
+	writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+	if (RME96_HAS_ANALOG_OUT(rme96)) {
+		usleep_range(3000, 10000);
+		snd_rme96_apply_dac_volume(rme96);
+	}
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+
+	return 0;
+}
+
+#endif
+
 static void snd_rme96_card_free(struct snd_card *card)
 {
 	snd_rme96_free(card->private_data);
@@ -2355,6 +2491,23 @@ snd_rme96_probe(struct pci_dev *pci,
 		return err;
 	}
 	
+#ifdef CONFIG_PM
+	rme96->playback_suspend_buffer = vmalloc(RME96_BUFFER_SIZE);
+	if (!rme96->playback_suspend_buffer) {
+		snd_printk(KERN_ERR
+			   "Failed to allocate playback suspend buffer!\n");
+		snd_card_free(card);
+		return -ENOMEM;
+	}
+	rme96->capture_suspend_buffer = vmalloc(RME96_BUFFER_SIZE);
+	if (!rme96->capture_suspend_buffer) {
+		snd_printk(KERN_ERR
+			   "Failed to allocate capture suspend buffer!\n");
+		snd_card_free(card);
+		return -ENOMEM;
+	}
+#endif
+
 	strcpy(card->driver, "Digi96");
 	switch (rme96->pci->device) {
 	case PCI_DEVICE_ID_RME_DIGI96:
@@ -2397,6 +2550,10 @@ static struct pci_driver rme96_driver = {
 	.id_table = snd_rme96_ids,
 	.probe = snd_rme96_probe,
 	.remove = snd_rme96_remove,
+#ifdef CONFIG_PM
+	.suspend = snd_rme96_suspend,
+	.resume = snd_rme96_resume,
+#endif
 };
 
 module_pci_driver(rme96_driver);
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index bd501931ee23..3cde55b753e2 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -38,6 +38,97 @@
  *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
  *
  */
+
+/* *************    Register Documentation   *******************************************************
+ *
+ * Work in progress! Documentation is based on the code in this file.
+ *
+ * --------- HDSPM_controlRegister ---------
+ * :7654.3210:7654.3210:7654.3210:7654.3210: bit number per byte
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number
+ * :1098.7654:3210.9876:5432.1098:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit
+ * :    .    :    .    :    .    :  x .    :  HDSPM_AudioInterruptEnable \_ setting both bits
+ * :    .    :    .    :    .    :    .   x:  HDSPM_Start                /  enables audio IO
+ * :    .    :    .    :    .    :   x.    :  HDSPM_ClockModeMaster - 1: Master, 0: Slave
+ * :    .    :    .    :    .    :    .210 :  HDSPM_LatencyMask - 3 Bit value for latency
+ * :    .    :    .    :    .    :    .    :      0:64, 1:128, 2:256, 3:512,
+ * :    .    :    .    :    .    :    .    :      4:1024, 5:2048, 6:4096, 7:8192
+ * :x   .    :    .    :    .   x:xx  .    :  HDSPM_FrequencyMask
+ * :    .    :    .    :    .    :10  .    :  HDSPM_Frequency1|HDSPM_Frequency0: 1=32K,2=44.1K,3=48K,0=??
+ * :    .    :    .    :    .   x:    .    :  <MADI> HDSPM_DoubleSpeed
+ * :x   .    :    .    :    .    :    .    :  <MADI> HDSPM_QuadSpeed
+ * :    .  3 :    .  10:  2 .    :    .    :  HDSPM_SyncRefMask :
+ * :    .    :    .   x:    .    :    .    :  HDSPM_SyncRef0
+ * :    .    :    .  x :    .    :    .    :  HDSPM_SyncRef1
+ * :    .    :    .    :  x .    :    .    :  <AES32> HDSPM_SyncRef2
+ * :    .  x :    .    :    .    :    .    :  <AES32> HDSPM_SyncRef3
+ * :    .    :    .  10:    .    :    .    :  <MADI> sync ref: 0:WC, 1:Madi, 2:TCO, 3:SyncIn
+ * :    .  3 :    .  10:  2 .    :    .    :  <AES32>  0:WC, 1:AES1 ... 8:AES8, 9: TCO, 10:SyncIn?
+ * :    .  x :    .    :    .    :    .    :  <MADIe> HDSPe_FLOAT_FORMAT
+ * :    .    :    .    : x  .    :    .    :  <MADI> HDSPM_InputSelect0 : 0=optical,1=coax
+ * :    .    :    .    :x   .    :    .    :  <MADI> HDSPM_InputSelect1
+ * :    .    :    .x   :    .    :    .    :  <MADI> HDSPM_clr_tms
+ * :    .    :    .    :    . x  :    .    :  <MADI> HDSPM_TX_64ch
+ * :    .    :    .    :    . x  :    .    :  <AES32> HDSPM_Emphasis
+ * :    .    :    .    :    .x   :    .    :  <MADI> HDSPM_AutoInp
+ * :    .    :    . x  :    .    :    .    :  <MADI> HDSPM_SMUX
+ * :    .    :    .x   :    .    :    .    :  <MADI> HDSPM_clr_tms
+ * :    .    :   x.    :    .    :    .    :  <MADI> HDSPM_taxi_reset
+ * :    .   x:    .    :    .    :    .    :  <MADI> HDSPM_LineOut
+ * :    .   x:    .    :    .    :    .    :  <AES32> ??????????????????
+ * :    .    :   x.    :    .    :    .    :  <AES32> HDSPM_WCK48
+ * :    .    :    .    :    .x   :    .    :  <AES32> HDSPM_Dolby
+ * :    .    : x  .    :    .    :    .    :  HDSPM_Midi0InterruptEnable
+ * :    .    :x   .    :    .    :    .    :  HDSPM_Midi1InterruptEnable
+ * :    .    :  x .    :    .    :    .    :  HDSPM_Midi2InterruptEnable
+ * :    . x  :    .    :    .    :    .    :  <MADI> HDSPM_Midi3InterruptEnable
+ * :    . x  :    .    :    .    :    .    :  <AES32> HDSPM_DS_DoubleWire
+ * :    .x   :    .    :    .    :    .    :  <AES32> HDSPM_QS_DoubleWire
+ * :   x.    :    .    :    .    :    .    :  <AES32> HDSPM_QS_QuadWire
+ * :    .    :    .    :    .  x :    .    :  <AES32> HDSPM_Professional
+ * : x  .    :    .    :    .    :    .    :  HDSPM_wclk_sel
+ * :    .    :    .    :    .    :    .    :
+ * :7654.3210:7654.3210:7654.3210:7654.3210: bit number per byte
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number
+ * :1098.7654:3210.9876:5432.1098:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421:hex digit
+ *
+ *
+ *
+ * AIO / RayDAT only
+ *
+ * ------------ HDSPM_WR_SETTINGS ----------
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number per byte
+ * :1098.7654:3210.9876:5432.1098:7654.3210:
+ * :||||.||||:||||.||||:||||.||||:||||.||||: bit number
+ * :7654.3210:7654.3210:7654.3210:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit
+ * :    .    :    .    :    .    :    .   x: HDSPM_c0Master 1: Master, 0: Slave
+ * :    .    :    .    :    .    :    .  x : HDSPM_c0_SyncRef0
+ * :    .    :    .    :    .    :    . x  : HDSPM_c0_SyncRef1
+ * :    .    :    .    :    .    :    .x   : HDSPM_c0_SyncRef2
+ * :    .    :    .    :    .    :   x.    : HDSPM_c0_SyncRef3
+ * :    .    :    .    :    .    :   3.210 : HDSPM_c0_SyncRefMask:
+ * :    .    :    .    :    .    :    .    :  RayDat: 0:WC, 1:AES, 2:SPDIF, 3..6: ADAT1..4,
+ * :    .    :    .    :    .    :    .    :          9:TCO, 10:SyncIn
+ * :    .    :    .    :    .    :    .    :  AIO: 0:WC, 1:AES, 2: SPDIF, 3: ATAT,
+ * :    .    :    .    :    .    :    .    :          9:TCO, 10:SyncIn
+ * :    .    :    .    :    .    :    .    :
+ * :    .    :    .    :    .    :    .    :
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number per byte
+ * :1098.7654:3210.9876:5432.1098:7654.3210:
+ * :||||.||||:||||.||||:||||.||||:||||.||||: bit number
+ * :7654.3210:7654.3210:7654.3210:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit
+ *
+ */
 #include <linux/init.h>
 #include <linux/delay.h>
 #include <linux/interrupt.h>
@@ -95,7 +186,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 #define HDSPM_controlRegister	     64
 #define HDSPM_interruptConfirmation  96
 #define HDSPM_control2Reg	     256  /* not in specs ???????? */
-#define HDSPM_freqReg                256  /* for AES32 */
+#define HDSPM_freqReg                256  /* for setting arbitrary clock values (DDS feature) */
 #define HDSPM_midiDataOut0	     352  /* just believe in old code */
 #define HDSPM_midiDataOut1	     356
 #define HDSPM_eeprom_wr		     384  /* for AES32 */
@@ -258,6 +349,25 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 
 #define HDSPM_wclk_sel (1<<30)
 
+/* additional control register bits for AIO*/
+#define HDSPM_c0_Wck48				0x20 /* also RayDAT */
+#define HDSPM_c0_Input0				0x1000
+#define HDSPM_c0_Input1				0x2000
+#define HDSPM_c0_Spdif_Opt			0x4000
+#define HDSPM_c0_Pro				0x8000
+#define HDSPM_c0_clr_tms			0x10000
+#define HDSPM_c0_AEB1				0x20000
+#define HDSPM_c0_AEB2				0x40000
+#define HDSPM_c0_LineOut			0x80000
+#define HDSPM_c0_AD_GAIN0			0x100000
+#define HDSPM_c0_AD_GAIN1			0x200000
+#define HDSPM_c0_DA_GAIN0			0x400000
+#define HDSPM_c0_DA_GAIN1			0x800000
+#define HDSPM_c0_PH_GAIN0			0x1000000
+#define HDSPM_c0_PH_GAIN1			0x2000000
+#define HDSPM_c0_Sym6db				0x4000000
+
+
 /* --- bit helper defines */
 #define HDSPM_LatencyMask    (HDSPM_Latency0|HDSPM_Latency1|HDSPM_Latency2)
 #define HDSPM_FrequencyMask  (HDSPM_Frequency0|HDSPM_Frequency1|\
@@ -341,11 +451,11 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 #define HDSPM_madiLock           (1<<3)	/* MADI Locked =1, no=0 */
 #define HDSPM_madiSync          (1<<18) /* MADI is in sync */
 
-#define HDSPM_tcoLock    0x00000020 /* Optional TCO locked status FOR HDSPe MADI! */
-#define HDSPM_tcoSync    0x10000000 /* Optional TCO sync status */
+#define HDSPM_tcoLockMadi    0x00000020 /* Optional TCO locked status for HDSPe MADI*/
+#define HDSPM_tcoSync    0x10000000 /* Optional TCO sync status for HDSPe MADI and AES32!*/
 
-#define HDSPM_syncInLock 0x00010000 /* Sync In lock status FOR HDSPe MADI! */
-#define HDSPM_syncInSync 0x00020000 /* Sync In sync status FOR HDSPe MADI! */
+#define HDSPM_syncInLock 0x00010000 /* Sync In lock status for HDSPe MADI! */
+#define HDSPM_syncInSync 0x00020000 /* Sync In sync status for HDSPe MADI! */
 
 #define HDSPM_BufferPositionMask 0x000FFC0 /* Bit 6..15 : h/w buffer pointer */
 			/* since 64byte accurate, last 6 bits are not used */
@@ -363,7 +473,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 					 * Interrupt
 					 */
 #define HDSPM_tco_detect         0x08000000
-#define HDSPM_tco_lock	         0x20000000
+#define HDSPM_tcoLockAes         0x20000000 /* Optional TCO locked status for HDSPe AES */
 
 #define HDSPM_s2_tco_detect      0x00000040
 #define HDSPM_s2_AEBO_D          0x00000080
@@ -461,7 +571,9 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 #define HDSPM_AES32_AUTOSYNC_FROM_AES6 6
 #define HDSPM_AES32_AUTOSYNC_FROM_AES7 7
 #define HDSPM_AES32_AUTOSYNC_FROM_AES8 8
-#define HDSPM_AES32_AUTOSYNC_FROM_NONE 9
+#define HDSPM_AES32_AUTOSYNC_FROM_TCO 9
+#define HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN 10
+#define HDSPM_AES32_AUTOSYNC_FROM_NONE 11
 
 /*  status2 */
 /* HDSPM_LockAES_bit is given by HDSPM_LockAES >> (AES# - 1) */
@@ -537,36 +649,39 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 /* names for speed modes */
 static char *hdspm_speed_names[] = { "single", "double", "quad" };
 
-static char *texts_autosync_aes_tco[] = { "Word Clock",
+static const char *const texts_autosync_aes_tco[] = { "Word Clock",
 					  "AES1", "AES2", "AES3", "AES4",
 					  "AES5", "AES6", "AES7", "AES8",
-					  "TCO" };
-static char *texts_autosync_aes[] = { "Word Clock",
+					  "TCO", "Sync In"
+};
+static const char *const texts_autosync_aes[] = { "Word Clock",
 				      "AES1", "AES2", "AES3", "AES4",
-				      "AES5", "AES6", "AES7", "AES8" };
-static char *texts_autosync_madi_tco[] = { "Word Clock",
+				      "AES5", "AES6", "AES7", "AES8",
+				      "Sync In"
+};
+static const char *const texts_autosync_madi_tco[] = { "Word Clock",
 					   "MADI", "TCO", "Sync In" };
-static char *texts_autosync_madi[] = { "Word Clock",
+static const char *const texts_autosync_madi[] = { "Word Clock",
 				       "MADI", "Sync In" };
 
-static char *texts_autosync_raydat_tco[] = {
+static const char *const texts_autosync_raydat_tco[] = {
 	"Word Clock",
 	"ADAT 1", "ADAT 2", "ADAT 3", "ADAT 4",
 	"AES", "SPDIF", "TCO", "Sync In"
 };
-static char *texts_autosync_raydat[] = {
+static const char *const texts_autosync_raydat[] = {
 	"Word Clock",
 	"ADAT 1", "ADAT 2", "ADAT 3", "ADAT 4",
 	"AES", "SPDIF", "Sync In"
 };
-static char *texts_autosync_aio_tco[] = {
+static const char *const texts_autosync_aio_tco[] = {
 	"Word Clock",
 	"ADAT", "AES", "SPDIF", "TCO", "Sync In"
 };
-static char *texts_autosync_aio[] = { "Word Clock",
+static const char *const texts_autosync_aio[] = { "Word Clock",
 				      "ADAT", "AES", "SPDIF", "Sync In" };
 
-static char *texts_freq[] = {
+static const char *const texts_freq[] = {
 	"No Lock",
 	"32 kHz",
 	"44.1 kHz",
@@ -629,7 +744,8 @@ static char *texts_ports_aio_in_ss[] = {
 	"AES.L", "AES.R",
 	"SPDIF.L", "SPDIF.R",
 	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6",
-	"ADAT.7", "ADAT.8"
+	"ADAT.7", "ADAT.8",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aio_out_ss[] = {
@@ -638,14 +754,16 @@ static char *texts_ports_aio_out_ss[] = {
 	"SPDIF.L", "SPDIF.R",
 	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6",
 	"ADAT.7", "ADAT.8",
-	"Phone.L", "Phone.R"
+	"Phone.L", "Phone.R",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aio_in_ds[] = {
 	"Analogue.L", "Analogue.R",
 	"AES.L", "AES.R",
 	"SPDIF.L", "SPDIF.R",
-	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4"
+	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aio_out_ds[] = {
@@ -653,14 +771,16 @@ static char *texts_ports_aio_out_ds[] = {
 	"AES.L", "AES.R",
 	"SPDIF.L", "SPDIF.R",
 	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
-	"Phone.L", "Phone.R"
+	"Phone.L", "Phone.R",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aio_in_qs[] = {
 	"Analogue.L", "Analogue.R",
 	"AES.L", "AES.R",
 	"SPDIF.L", "SPDIF.R",
-	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4"
+	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aio_out_qs[] = {
@@ -668,7 +788,8 @@ static char *texts_ports_aio_out_qs[] = {
 	"AES.L", "AES.R",
 	"SPDIF.L", "SPDIF.R",
 	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
-	"Phone.L", "Phone.R"
+	"Phone.L", "Phone.R",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aes32[] = {
@@ -745,8 +866,8 @@ static char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = {
 	8, 9,			/* aes in, */
 	10, 11,			/* spdif in */
 	12, 13, 14, 15, 16, 17, 18, 19,	/* ADAT in */
-	-1, -1,
-	-1, -1, -1, -1, -1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
+	-1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -760,7 +881,8 @@ static char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = {
 	10, 11,			/* spdif out */
 	12, 13, 14, 15, 16, 17, 18, 19,	/* ADAT out */
 	6, 7,			/* phone out */
-	-1, -1, -1, -1, -1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
+	-1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -773,7 +895,8 @@ static char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = {
 	8, 9,			/* aes in */
 	10, 11,			/* spdif in */
 	12, 14, 16, 18,		/* adat in */
-	-1, -1, -1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
+	-1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -788,7 +911,7 @@ static char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = {
 	10, 11,			/* spdif out */
 	12, 14, 16, 18,		/* adat out */
 	6, 7,			/* phone out */
-	-1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -802,7 +925,8 @@ static char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = {
 	8, 9,			/* aes in */
 	10, 11,			/* spdif in */
 	12, 16,			/* adat in */
-	-1, -1, -1, -1, -1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
+	-1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -817,7 +941,8 @@ static char channel_map_aio_out_qs[HDSPM_MAX_CHANNELS] = {
 	10, 11,			/* spdif out */
 	12, 16,			/* adat out */
 	6, 7,			/* phone out */
-	-1, -1, -1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
+	-1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -856,11 +981,11 @@ struct hdspm_midi {
 };
 
 struct hdspm_tco {
-	int input;
-	int framerate;
-	int wordclock;
-	int samplerate;
-	int pull;
+	int input; /* 0: LTC, 1:Video, 2: WC*/
+	int framerate; /* 0=24, 1=25, 2=29.97, 3=29.97d, 4=30, 5=30d */
+	int wordclock; /* 0=1:1, 1=44.1->48, 2=48->44.1 */
+	int samplerate; /* 0=44.1, 1=48, 2= freq from app */
+	int pull; /*   0=0, 1=+0.1%, 2=-0.1%, 3=+4%, 4=-4%*/
 	int term; /* 0 = off, 1 = on */
 };
 
@@ -879,7 +1004,7 @@ struct hdspm {
 
 	u32 control_register;	/* cached value */
 	u32 control2_register;	/* cached value */
-	u32 settings_register;
+	u32 settings_register;  /* cached value for AIO / RayDat (sync reference, master/slave) */
 
 	struct hdspm_midi midi[4];
 	struct tasklet_struct midi_tasklet;
@@ -941,7 +1066,7 @@ struct hdspm {
 
 	struct hdspm_tco *tco;  /* NULL if no TCO detected */
 
-	char **texts_autosync;
+	const char *const *texts_autosync;
 	int texts_autosync_items;
 
 	cycles_t last_interrupt;
@@ -976,12 +1101,24 @@ static inline void snd_hdspm_initialize_midi_flush(struct hdspm *hdspm);
 static inline int hdspm_get_pll_freq(struct hdspm *hdspm);
 static int hdspm_update_simple_mixer_controls(struct hdspm *hdspm);
 static int hdspm_autosync_ref(struct hdspm *hdspm);
+static int hdspm_set_toggle_setting(struct hdspm *hdspm, u32 regmask, int out);
 static int snd_hdspm_set_defaults(struct hdspm *hdspm);
 static int hdspm_system_clock_mode(struct hdspm *hdspm);
 static void hdspm_set_sgbuf(struct hdspm *hdspm,
 			    struct snd_pcm_substream *substream,
 			     unsigned int reg, int channels);
 
+static int hdspm_aes_sync_check(struct hdspm *hdspm, int idx);
+static int hdspm_wc_sync_check(struct hdspm *hdspm);
+static int hdspm_tco_sync_check(struct hdspm *hdspm);
+static int hdspm_sync_in_sync_check(struct hdspm *hdspm);
+
+static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index);
+static int hdspm_get_tco_sample_rate(struct hdspm *hdspm);
+static int hdspm_get_wc_sample_rate(struct hdspm *hdspm);
+
+
+
 static inline int HDSPM_bit2freq(int n)
 {
 	static const int bit2freq_tab[] = {
@@ -992,6 +1129,12 @@ static inline int HDSPM_bit2freq(int n)
 	return bit2freq_tab[n];
 }
 
+static bool hdspm_is_raydat_or_aio(struct hdspm *hdspm)
+{
+	return ((AIO == hdspm->io_type) || (RayDAT == hdspm->io_type));
+}
+
+
 /* Write/read to/from HDSPM with Adresses in Bytes
    not words but only 32Bit writes are allowed */
 
@@ -1107,14 +1250,11 @@ static int hdspm_rate_multiplier(struct hdspm *hdspm, int rate)
 		else if (hdspm->control_register &
 				HDSPM_DoubleSpeed)
 			return rate * 2;
-	};
+	}
 	return rate;
 }
 
-static int hdspm_tco_sync_check(struct hdspm *hdspm);
-static int hdspm_sync_in_sync_check(struct hdspm *hdspm);
-
-/* check for external sample rate */
+/* check for external sample rate, returns the sample rate in Hz*/
 static int hdspm_external_sample_rate(struct hdspm *hdspm)
 {
 	unsigned int status, status2, timecode;
@@ -1127,17 +1267,36 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm)
 		timecode = hdspm_read(hdspm, HDSPM_timecodeRegister);
 
 		syncref = hdspm_autosync_ref(hdspm);
+		switch (syncref) {
+		case HDSPM_AES32_AUTOSYNC_FROM_WORD:
+		/* Check WC sync and get sample rate */
+			if (hdspm_wc_sync_check(hdspm))
+				return HDSPM_bit2freq(hdspm_get_wc_sample_rate(hdspm));
+			break;
 
-		if (syncref == HDSPM_AES32_AUTOSYNC_FROM_WORD &&
-				status & HDSPM_AES32_wcLock)
-			return HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF);
+		case HDSPM_AES32_AUTOSYNC_FROM_AES1:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES2:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES3:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES4:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES5:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES6:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES7:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES8:
+		/* Check AES sync and get sample rate */
+			if (hdspm_aes_sync_check(hdspm, syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1))
+				return HDSPM_bit2freq(hdspm_get_aes_sample_rate(hdspm,
+							syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1));
+			break;
 
-		if (syncref >= HDSPM_AES32_AUTOSYNC_FROM_AES1 &&
-				syncref <= HDSPM_AES32_AUTOSYNC_FROM_AES8 &&
-				status2 & (HDSPM_LockAES >>
-				(syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1)))
-			return HDSPM_bit2freq((timecode >> (4*(syncref-HDSPM_AES32_AUTOSYNC_FROM_AES1))) & 0xF);
-		return 0;
+
+		case HDSPM_AES32_AUTOSYNC_FROM_TCO:
+		/* Check TCO sync and get sample rate */
+			if (hdspm_tco_sync_check(hdspm))
+				return HDSPM_bit2freq(hdspm_get_tco_sample_rate(hdspm));
+			break;
+		default:
+			return 0;
+		} /* end switch(syncref) */
 		break;
 
 	case MADIface:
@@ -2129,6 +2288,9 @@ static int hdspm_get_wc_sample_rate(struct hdspm *hdspm)
 		status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);
 		return (status >> 16) & 0xF;
 		break;
+	case AES32:
+		status = hdspm_read(hdspm, HDSPM_statusRegister);
+		return (status >> HDSPM_AES32_wcFreq_bit) & 0xF;
 	default:
 		break;
 	}
@@ -2152,6 +2314,9 @@ static int hdspm_get_tco_sample_rate(struct hdspm *hdspm)
 			status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);
 			return (status >> 20) & 0xF;
 			break;
+		case AES32:
+			status = hdspm_read(hdspm, HDSPM_statusRegister);
+			return (status >> 1) & 0xF;
 		default:
 			break;
 		}
@@ -2183,6 +2348,23 @@ static int hdspm_get_sync_in_sample_rate(struct hdspm *hdspm)
 	return 0;
 }
 
+/**
+ * Returns the AES sample rate class for the given card.
+ **/
+static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index)
+{
+	int timecode;
+
+	switch (hdspm->io_type) {
+	case AES32:
+		timecode = hdspm_read(hdspm, HDSPM_timecodeRegister);
+		return (timecode >> (4*index)) & 0xF;
+		break;
+	default:
+		break;
+	}
+	return 0;
+}
 
 /**
  * Returns the sample rate class for input source <idx> for
@@ -2196,15 +2378,23 @@ static int hdspm_get_s1_sample_rate(struct hdspm *hdspm, unsigned int idx)
 }
 
 #define ENUMERATED_CTL_INFO(info, texts) \
-{ \
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; \
-	uinfo->count = 1; \
-	uinfo->value.enumerated.items = ARRAY_SIZE(texts); \
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) \
-		uinfo->value.enumerated.item =	uinfo->value.enumerated.items - 1; \
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); \
-}
+	snd_ctl_enum_info(info, 1, ARRAY_SIZE(texts), texts)
+
 
+/* Helper function to query the external sample rate and return the
+ * corresponding enum to be returned to userspace.
+ */
+static int hdspm_external_rate_to_enum(struct hdspm *hdspm)
+{
+	int rate = hdspm_external_sample_rate(hdspm);
+	int i, selected_rate = 0;
+	for (i = 1; i < 10; i++)
+		if (HDSPM_bit2freq(i) == rate) {
+			selected_rate = i;
+			break;
+		}
+	return selected_rate;
+}
 
 
 #define HDSPM_AUTOSYNC_SAMPLE_RATE(xname, xindex) \
@@ -2270,7 +2460,7 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol,
 		default:
 			ucontrol->value.enumerated.item[0] =
 				hdspm_get_s1_sample_rate(hdspm,
-						ucontrol->id.index-1);
+						kcontrol->private_value-1);
 		}
 		break;
 
@@ -2289,28 +2479,24 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol,
 			ucontrol->value.enumerated.item[0] =
 				hdspm_get_sync_in_sample_rate(hdspm);
 			break;
+		case 11: /* External Rate */
+			ucontrol->value.enumerated.item[0] =
+				hdspm_external_rate_to_enum(hdspm);
+			break;
 		default: /* AES1 to AES8 */
 			ucontrol->value.enumerated.item[0] =
-				hdspm_get_s1_sample_rate(hdspm,
-						kcontrol->private_value-1);
+				hdspm_get_aes_sample_rate(hdspm,
+						kcontrol->private_value -
+						HDSPM_AES32_AUTOSYNC_FROM_AES1);
 			break;
 		}
 		break;
 
 	case MADI:
 	case MADIface:
-		{
-			int rate = hdspm_external_sample_rate(hdspm);
-			int i, selected_rate = 0;
-			for (i = 1; i < 10; i++)
-				if (HDSPM_bit2freq(i) == rate) {
-					selected_rate = i;
-					break;
-				}
-			ucontrol->value.enumerated.item[0] = selected_rate;
-		}
+		ucontrol->value.enumerated.item[0] =
+			hdspm_external_rate_to_enum(hdspm);
 		break;
-
 	default:
 		break;
 	}
@@ -2359,33 +2545,17 @@ static int hdspm_system_clock_mode(struct hdspm *hdspm)
  **/
 static void hdspm_set_system_clock_mode(struct hdspm *hdspm, int mode)
 {
-	switch (hdspm->io_type) {
-	case AIO:
-	case RayDAT:
-		if (0 == mode)
-			hdspm->settings_register |= HDSPM_c0Master;
-		else
-			hdspm->settings_register &= ~HDSPM_c0Master;
-
-		hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register);
-		break;
-
-	default:
-		if (0 == mode)
-			hdspm->control_register |= HDSPM_ClockModeMaster;
-		else
-			hdspm->control_register &= ~HDSPM_ClockModeMaster;
-
-		hdspm_write(hdspm, HDSPM_controlRegister,
-				hdspm->control_register);
-	}
+	hdspm_set_toggle_setting(hdspm,
+			(hdspm_is_raydat_or_aio(hdspm)) ?
+			HDSPM_c0Master : HDSPM_ClockModeMaster,
+			(0 == mode));
 }
 
 
 static int snd_hdspm_info_system_clock_mode(struct snd_kcontrol *kcontrol,
 					    struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "Master", "AutoSync" };
+	static const char *const texts[] = { "Master", "AutoSync" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -2809,16 +2979,7 @@ static int snd_hdspm_info_pref_sync_ref(struct snd_kcontrol *kcontrol,
 {
 	struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = hdspm->texts_autosync_items;
-
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item =
-			uinfo->value.enumerated.items - 1;
-
-	strcpy(uinfo->value.enumerated.name,
-			hdspm->texts_autosync[uinfo->value.enumerated.item]);
+	snd_ctl_enum_info(uinfo, 1, hdspm->texts_autosync_items, hdspm->texts_autosync);
 
 	return 0;
 }
@@ -2873,19 +3034,20 @@ static int snd_hdspm_put_pref_sync_ref(struct snd_kcontrol *kcontrol,
 
 static int hdspm_autosync_ref(struct hdspm *hdspm)
 {
+	/* This looks at the autosync selected sync reference */
 	if (AES32 == hdspm->io_type) {
+
 		unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister);
-		unsigned int syncref =
-			(status >> HDSPM_AES32_syncref_bit) & 0xF;
-		if (syncref == 0)
-			return HDSPM_AES32_AUTOSYNC_FROM_WORD;
-		if (syncref <= 8)
+		unsigned int syncref = (status >> HDSPM_AES32_syncref_bit) & 0xF;
+		if ((syncref >= HDSPM_AES32_AUTOSYNC_FROM_WORD) &&
+				(syncref <= HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN)) {
 			return syncref;
+		}
 		return HDSPM_AES32_AUTOSYNC_FROM_NONE;
+
 	} else if (MADI == hdspm->io_type) {
-		/* This looks at the autosync selected sync reference */
-		unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
 
+		unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
 		switch (status2 & HDSPM_SelSyncRefMask) {
 		case HDSPM_SelSyncRef_WORD:
 			return HDSPM_AUTOSYNC_FROM_WORD;
@@ -2898,7 +3060,7 @@ static int hdspm_autosync_ref(struct hdspm *hdspm)
 		case HDSPM_SelSyncRef_NVALID:
 			return HDSPM_AUTOSYNC_FROM_NONE;
 		default:
-			return 0;
+			return HDSPM_AUTOSYNC_FROM_NONE;
 		}
 
 	}
@@ -2912,31 +3074,15 @@ static int snd_hdspm_info_autosync_ref(struct snd_kcontrol *kcontrol,
 	struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
 
 	if (AES32 == hdspm->io_type) {
-		static char *texts[] = { "WordClock", "AES1", "AES2", "AES3",
-			"AES4",	"AES5", "AES6", "AES7", "AES8", "None"};
-
-		uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-		uinfo->count = 1;
-		uinfo->value.enumerated.items = 10;
-		if (uinfo->value.enumerated.item >=
-		    uinfo->value.enumerated.items)
-			uinfo->value.enumerated.item =
-				uinfo->value.enumerated.items - 1;
-		strcpy(uinfo->value.enumerated.name,
-				texts[uinfo->value.enumerated.item]);
+		static const char *const texts[] = { "WordClock", "AES1", "AES2", "AES3",
+			"AES4",	"AES5", "AES6", "AES7", "AES8", "TCO", "Sync In", "None"};
+
+		ENUMERATED_CTL_INFO(uinfo, texts);
 	} else if (MADI == hdspm->io_type) {
-		static char *texts[] = {"Word Clock", "MADI", "TCO",
+		static const char *const texts[] = {"Word Clock", "MADI", "TCO",
 			"Sync In", "None" };
 
-		uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-		uinfo->count = 1;
-		uinfo->value.enumerated.items = 5;
-		if (uinfo->value.enumerated.item >=
-				uinfo->value.enumerated.items)
-			uinfo->value.enumerated.item =
-				uinfo->value.enumerated.items - 1;
-		strcpy(uinfo->value.enumerated.name,
-				texts[uinfo->value.enumerated.item]);
+		ENUMERATED_CTL_INFO(uinfo, texts);
 	}
 	return 0;
 }
@@ -2964,7 +3110,7 @@ static int snd_hdspm_get_autosync_ref(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_video_input_format(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"No video", "NTSC", "PAL"};
+	static const char *const texts[] = {"No video", "NTSC", "PAL"};
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3010,7 +3156,7 @@ static int snd_hdspm_get_tco_video_input_format(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_ltc_frames(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps",
+	static const char *const texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps",
 				"30 fps"};
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
@@ -3027,19 +3173,19 @@ static int hdspm_tco_ltc_frames(struct hdspm *hdspm)
 					HDSPM_TCO1_LTC_Format_MSB)) {
 		case 0:
 			/* 24 fps */
-			ret = 1;
+			ret = fps_24;
 			break;
 		case HDSPM_TCO1_LTC_Format_LSB:
 			/* 25 fps */
-			ret = 2;
+			ret = fps_25;
 			break;
 		case HDSPM_TCO1_LTC_Format_MSB:
-			/* 25 fps */
-			ret = 3;
+			/* 29.97 fps */
+			ret = fps_2997;
 			break;
 		default:
 			/* 30 fps */
-			ret = 4;
+			ret = fps_30;
 			break;
 		}
 	}
@@ -3067,16 +3213,35 @@ static int snd_hdspm_get_tco_ltc_frames(struct snd_kcontrol *kcontrol,
 
 static int hdspm_toggle_setting(struct hdspm *hdspm, u32 regmask)
 {
-	return (hdspm->control_register & regmask) ? 1 : 0;
+	u32 reg;
+
+	if (hdspm_is_raydat_or_aio(hdspm))
+		reg = hdspm->settings_register;
+	else
+		reg = hdspm->control_register;
+
+	return (reg & regmask) ? 1 : 0;
 }
 
 static int hdspm_set_toggle_setting(struct hdspm *hdspm, u32 regmask, int out)
 {
+	u32 *reg;
+	u32 target_reg;
+
+	if (hdspm_is_raydat_or_aio(hdspm)) {
+		reg = &(hdspm->settings_register);
+		target_reg = HDSPM_WR_SETTINGS;
+	} else {
+		reg = &(hdspm->control_register);
+		target_reg = HDSPM_controlRegister;
+	}
+
 	if (out)
-		hdspm->control_register |= regmask;
+		*reg |= regmask;
 	else
-		hdspm->control_register &= ~regmask;
-	hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+		*reg &= ~regmask;
+
+	hdspm_write(hdspm, target_reg, *reg);
 
 	return 0;
 }
@@ -3141,7 +3306,7 @@ static int hdspm_set_input_select(struct hdspm * hdspm, int out)
 static int snd_hdspm_info_input_select(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "optical", "coaxial" };
+	static const char *const texts[] = { "optical", "coaxial" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3203,7 +3368,7 @@ static int hdspm_set_ds_wire(struct hdspm * hdspm, int ds)
 static int snd_hdspm_info_ds_wire(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "Single", "Double" };
+	static const char *const texts[] = { "Single", "Double" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3276,7 +3441,7 @@ static int hdspm_set_qs_wire(struct hdspm * hdspm, int mode)
 static int snd_hdspm_info_qs_wire(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "Single", "Double", "Quad" };
+	static const char *const texts[] = { "Single", "Double", "Quad" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3313,6 +3478,84 @@ static int snd_hdspm_put_qs_wire(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
+#define HDSPM_CONTROL_TRISTATE(xname, xindex) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+	.name = xname, \
+	.private_value = xindex, \
+	.info = snd_hdspm_info_tristate, \
+	.get = snd_hdspm_get_tristate, \
+	.put = snd_hdspm_put_tristate \
+}
+
+static int hdspm_tristate(struct hdspm *hdspm, u32 regmask)
+{
+	u32 reg = hdspm->settings_register & (regmask * 3);
+	return reg / regmask;
+}
+
+static int hdspm_set_tristate(struct hdspm *hdspm, int mode, u32 regmask)
+{
+	hdspm->settings_register &= ~(regmask * 3);
+	hdspm->settings_register |= (regmask * mode);
+	hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register);
+
+	return 0;
+}
+
+static int snd_hdspm_info_tristate(struct snd_kcontrol *kcontrol,
+				       struct snd_ctl_elem_info *uinfo)
+{
+	u32 regmask = kcontrol->private_value;
+
+	static const char *const texts_spdif[] = { "Optical", "Coaxial", "Internal" };
+	static const char *const texts_levels[] = { "Hi Gain", "+4 dBu", "-10 dBV" };
+
+	switch (regmask) {
+	case HDSPM_c0_Input0:
+		ENUMERATED_CTL_INFO(uinfo, texts_spdif);
+		break;
+	default:
+		ENUMERATED_CTL_INFO(uinfo, texts_levels);
+		break;
+	}
+	return 0;
+}
+
+static int snd_hdspm_get_tristate(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+	u32 regmask = kcontrol->private_value;
+
+	spin_lock_irq(&hdspm->lock);
+	ucontrol->value.enumerated.item[0] = hdspm_tristate(hdspm, regmask);
+	spin_unlock_irq(&hdspm->lock);
+	return 0;
+}
+
+static int snd_hdspm_put_tristate(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+	u32 regmask = kcontrol->private_value;
+	int change;
+	int val;
+
+	if (!snd_hdspm_use_is_exclusive(hdspm))
+		return -EBUSY;
+	val = ucontrol->value.integer.value[0];
+	if (val < 0)
+		val = 0;
+	if (val > 2)
+		val = 2;
+
+	spin_lock_irq(&hdspm->lock);
+	change = val != hdspm_tristate(hdspm, regmask);
+	hdspm_set_tristate(hdspm, val, regmask);
+	spin_unlock_irq(&hdspm->lock);
+	return change;
+}
+
 #define HDSPM_MADI_SPEEDMODE(xname, xindex) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
 	.name = xname, \
@@ -3352,7 +3595,7 @@ static int hdspm_set_madi_speedmode(struct hdspm *hdspm, int mode)
 static int snd_hdspm_info_madi_speedmode(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "Single", "Double", "Quad" };
+	static const char *const texts[] = { "Single", "Double", "Quad" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3587,7 +3830,7 @@ static int snd_hdspm_put_playback_mixer(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol,
 				     struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "No Lock", "Lock", "Sync", "N/A" };
+	static const char *const texts[] = { "No Lock", "Lock", "Sync", "N/A" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3595,7 +3838,7 @@ static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_tco_info_lock_check(struct snd_kcontrol *kcontrol,
 				     struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "No Lock", "Lock" };
+	static const char *const texts[] = { "No Lock", "Lock" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3745,9 +3988,18 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm)
 	if (hdspm->tco) {
 		switch (hdspm->io_type) {
 		case MADI:
+			status = hdspm_read(hdspm, HDSPM_statusRegister);
+			if (status & HDSPM_tcoLockMadi) {
+				if (status & HDSPM_tcoSync)
+					return 2;
+				else
+					return 1;
+			}
+			return 0;
+			break;
 		case AES32:
 			status = hdspm_read(hdspm, HDSPM_statusRegister);
-			if (status & HDSPM_tcoLock) {
+			if (status & HDSPM_tcoLockAes) {
 				if (status & HDSPM_tcoSync)
 					return 2;
 				else
@@ -3807,7 +4059,8 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol,
 		case 5: /* SYNC IN */
 			val = hdspm_sync_in_sync_check(hdspm); break;
 		default:
-			val = hdspm_s1_sync_check(hdspm, ucontrol->id.index-1);
+			val = hdspm_s1_sync_check(hdspm,
+					kcontrol->private_value-1);
 		}
 		break;
 
@@ -3975,7 +4228,8 @@ static void hdspm_tco_write(struct hdspm *hdspm)
 static int snd_hdspm_info_tco_sample_rate(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "44.1 kHz", "48 kHz" };
+	/* TODO freq from app could be supported here, see tco->samplerate */
+	static const char *const texts[] = { "44.1 kHz", "48 kHz" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -4021,7 +4275,8 @@ static int snd_hdspm_put_tco_sample_rate(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_pull(struct snd_kcontrol *kcontrol,
 				   struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "0", "+ 0.1 %", "- 0.1 %", "+ 4 %", "- 4 %" };
+	static const char *const texts[] = { "0", "+ 0.1 %", "- 0.1 %",
+		"+ 4 %", "- 4 %" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -4066,7 +4321,7 @@ static int snd_hdspm_put_tco_pull(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_wck_conversion(struct snd_kcontrol *kcontrol,
 					     struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" };
+	static const char *const texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -4112,7 +4367,7 @@ static int snd_hdspm_put_tco_wck_conversion(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_frame_rate(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "24 fps", "25 fps", "29.97fps",
+	static const char *const texts[] = { "24 fps", "25 fps", "29.97fps",
 		"29.97 dfps", "30 fps", "30 dfps" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
@@ -4159,7 +4414,7 @@ static int snd_hdspm_put_tco_frame_rate(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_sync_source(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "LTC", "Video", "WCK" };
+	static const char *const texts[] = { "LTC", "Video", "WCK" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -4284,7 +4539,6 @@ static struct snd_kcontrol_new snd_hdspm_controls_aio[] = {
 	HDSPM_INTERNAL_CLOCK("Internal Clock", 0),
 	HDSPM_SYSTEM_CLOCK_MODE("System Clock Mode", 0),
 	HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0),
-	HDSPM_AUTOSYNC_REF("AutoSync Reference", 0),
 	HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0),
 	HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0),
 	HDSPM_SYNC_CHECK("WC SyncCheck", 0),
@@ -4298,7 +4552,16 @@ static struct snd_kcontrol_new snd_hdspm_controls_aio[] = {
 	HDSPM_AUTOSYNC_SAMPLE_RATE("SPDIF Frequency", 2),
 	HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT Frequency", 3),
 	HDSPM_AUTOSYNC_SAMPLE_RATE("TCO Frequency", 4),
-	HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 5)
+	HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 5),
+	HDSPM_CONTROL_TRISTATE("S/PDIF Input", HDSPM_c0_Input0),
+	HDSPM_TOGGLE_SETTING("S/PDIF Out Optical", HDSPM_c0_Spdif_Opt),
+	HDSPM_TOGGLE_SETTING("S/PDIF Out Professional", HDSPM_c0_Pro),
+	HDSPM_TOGGLE_SETTING("ADAT internal (AEB/TEB)", HDSPM_c0_AEB1),
+	HDSPM_TOGGLE_SETTING("XLR Breakout Cable", HDSPM_c0_Sym6db),
+	HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48),
+	HDSPM_CONTROL_TRISTATE("Input Level", HDSPM_c0_AD_GAIN0),
+	HDSPM_CONTROL_TRISTATE("Output Level", HDSPM_c0_DA_GAIN0),
+	HDSPM_CONTROL_TRISTATE("Phones Level", HDSPM_c0_PH_GAIN0)
 
 		/*
 		   HDSPM_INPUT_SELECT("Input Select", 0),
@@ -4335,7 +4598,9 @@ static struct snd_kcontrol_new snd_hdspm_controls_raydat[] = {
 	HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT3 Frequency", 5),
 	HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT4 Frequency", 6),
 	HDSPM_AUTOSYNC_SAMPLE_RATE("TCO Frequency", 7),
-	HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 8)
+	HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 8),
+	HDSPM_TOGGLE_SETTING("S/PDIF Out Professional", HDSPM_c0_Pro),
+	HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48)
 };
 
 static struct snd_kcontrol_new snd_hdspm_controls_aes32[] = {
@@ -4345,7 +4610,7 @@ static struct snd_kcontrol_new snd_hdspm_controls_aes32[] = {
 	HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0),
 	HDSPM_AUTOSYNC_REF("AutoSync Reference", 0),
 	HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0),
-	HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0),
+	HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 11),
 	HDSPM_SYNC_CHECK("WC Sync Check", 0),
 	HDSPM_SYNC_CHECK("AES1 Sync Check", 1),
 	HDSPM_SYNC_CHECK("AES2 Sync Check", 2),
@@ -4501,77 +4766,22 @@ static int snd_hdspm_create_controls(struct snd_card *card,
  ------------------------------------------------------------*/
 
 static void
-snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
-			 struct snd_info_buffer *buffer)
+snd_hdspm_proc_read_tco(struct snd_info_entry *entry,
+					struct snd_info_buffer *buffer)
 {
 	struct hdspm *hdspm = entry->private_data;
-	unsigned int status, status2, control, freq;
-
-	char *pref_sync_ref;
-	char *autosync_ref;
-	char *system_clock_mode;
-	char *insel;
-	int x, x2;
-
-	/* TCO stuff */
+	unsigned int status, control;
 	int a, ltc, frames, seconds, minutes, hours;
 	unsigned int period;
 	u64 freq_const = 0;
 	u32 rate;
 
+	snd_iprintf(buffer, "--- TCO ---\n");
+
 	status = hdspm_read(hdspm, HDSPM_statusRegister);
-	status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
 	control = hdspm->control_register;
-	freq = hdspm_read(hdspm, HDSPM_timecodeRegister);
 
-	snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n",
-			hdspm->card_name, hdspm->card->number + 1,
-			hdspm->firmware_rev,
-			(status2 & HDSPM_version0) |
-			(status2 & HDSPM_version1) | (status2 &
-				HDSPM_version2));
 
-	snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n",
-			(hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF,
-			hdspm->serial);
-
-	snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
-			hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase);
-
-	snd_iprintf(buffer, "--- System ---\n");
-
-	snd_iprintf(buffer,
-		"IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n",
-		status & HDSPM_audioIRQPending,
-		(status & HDSPM_midi0IRQPending) ? 1 : 0,
-		(status & HDSPM_midi1IRQPending) ? 1 : 0,
-		hdspm->irq_count);
-	snd_iprintf(buffer,
-		"HW pointer: id = %d, rawptr = %d (%d->%d) "
-		"estimated= %ld (bytes)\n",
-		((status & HDSPM_BufferID) ? 1 : 0),
-		(status & HDSPM_BufferPositionMask),
-		(status & HDSPM_BufferPositionMask) %
-		(2 * (int)hdspm->period_bytes),
-		((status & HDSPM_BufferPositionMask) - 64) %
-		(2 * (int)hdspm->period_bytes),
-		(long) hdspm_hw_pointer(hdspm) * 4);
-
-	snd_iprintf(buffer,
-		"MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n",
-		hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF,
-		hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF,
-		hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF,
-		hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF);
-	snd_iprintf(buffer,
-		"MIDIoverMADI FIFO: In=0x%x, Out=0x%x \n",
-		hdspm_read(hdspm, HDSPM_midiStatusIn2) & 0xFF,
-		hdspm_read(hdspm, HDSPM_midiStatusOut2) & 0xFF);
-	snd_iprintf(buffer,
-		"Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, "
-		"status2=0x%x\n",
-		hdspm->control_register, hdspm->control2_register,
-		status, status2);
 	if (status & HDSPM_tco_detect) {
 		snd_iprintf(buffer, "TCO module detected.\n");
 		a = hdspm_read(hdspm, HDSPM_RD_TCO+4);
@@ -4665,6 +4875,75 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
 	} else {
 		snd_iprintf(buffer, "No TCO module detected.\n");
 	}
+}
+
+static void
+snd_hdspm_proc_read_madi(struct snd_info_entry *entry,
+			 struct snd_info_buffer *buffer)
+{
+	struct hdspm *hdspm = entry->private_data;
+	unsigned int status, status2, control, freq;
+
+	char *pref_sync_ref;
+	char *autosync_ref;
+	char *system_clock_mode;
+	char *insel;
+	int x, x2;
+
+	status = hdspm_read(hdspm, HDSPM_statusRegister);
+	status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+	control = hdspm->control_register;
+	freq = hdspm_read(hdspm, HDSPM_timecodeRegister);
+
+	snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n",
+			hdspm->card_name, hdspm->card->number + 1,
+			hdspm->firmware_rev,
+			(status2 & HDSPM_version0) |
+			(status2 & HDSPM_version1) | (status2 &
+				HDSPM_version2));
+
+	snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n",
+			(hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF,
+			hdspm->serial);
+
+	snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
+			hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase);
+
+	snd_iprintf(buffer, "--- System ---\n");
+
+	snd_iprintf(buffer,
+		"IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n",
+		status & HDSPM_audioIRQPending,
+		(status & HDSPM_midi0IRQPending) ? 1 : 0,
+		(status & HDSPM_midi1IRQPending) ? 1 : 0,
+		hdspm->irq_count);
+	snd_iprintf(buffer,
+		"HW pointer: id = %d, rawptr = %d (%d->%d) "
+		"estimated= %ld (bytes)\n",
+		((status & HDSPM_BufferID) ? 1 : 0),
+		(status & HDSPM_BufferPositionMask),
+		(status & HDSPM_BufferPositionMask) %
+		(2 * (int)hdspm->period_bytes),
+		((status & HDSPM_BufferPositionMask) - 64) %
+		(2 * (int)hdspm->period_bytes),
+		(long) hdspm_hw_pointer(hdspm) * 4);
+
+	snd_iprintf(buffer,
+		"MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n",
+		hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF,
+		hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF,
+		hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF,
+		hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF);
+	snd_iprintf(buffer,
+		"MIDIoverMADI FIFO: In=0x%x, Out=0x%x \n",
+		hdspm_read(hdspm, HDSPM_midiStatusIn2) & 0xFF,
+		hdspm_read(hdspm, HDSPM_midiStatusOut2) & 0xFF);
+	snd_iprintf(buffer,
+		"Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, "
+		"status2=0x%x\n",
+		hdspm->control_register, hdspm->control2_register,
+		status, status2);
+
 
 	snd_iprintf(buffer, "--- Settings ---\n");
 
@@ -4768,6 +5047,9 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
 		(status & HDSPM_RX_64ch) ? "64 channels" :
 		"56 channels");
 
+	/* call readout function for TCO specific status */
+	snd_hdspm_proc_read_tco(entry, buffer);
+
 	snd_iprintf(buffer, "\n");
 }
 
@@ -4909,11 +5191,18 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
 		autosync_ref = "AES7"; break;
 	case HDSPM_AES32_AUTOSYNC_FROM_AES8:
 		autosync_ref = "AES8"; break;
+	case HDSPM_AES32_AUTOSYNC_FROM_TCO:
+		autosync_ref = "TCO"; break;
+	case HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN:
+		autosync_ref = "Sync In"; break;
 	default:
 		autosync_ref = "---"; break;
 	}
 	snd_iprintf(buffer, "AutoSync ref = %s\n", autosync_ref);
 
+	/* call readout function for TCO specific status */
+	snd_hdspm_proc_read_tco(entry, buffer);
+
 	snd_iprintf(buffer, "\n");
 }
 
@@ -5097,7 +5386,7 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm)
 
 	case AES32:
 		hdspm->control_register =
-			HDSPM_ClockModeMaster |	/* Master Cloack Mode on */
+			HDSPM_ClockModeMaster |	/* Master Clock Mode on */
 			hdspm_encode_latency(7) | /* latency max=8192samples */
 			HDSPM_SyncRef0 |	/* AES1 is syncclock */
 			HDSPM_LineOut |	/* Analog output in */
@@ -5123,9 +5412,8 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm)
 
 	all_in_all_mixer(hdspm, 0 * UNITY_GAIN);
 
-	if (hdspm->io_type == AIO || hdspm->io_type == RayDAT) {
+	if (hdspm_is_raydat_or_aio(hdspm))
 		hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register);
-	}
 
 	/* set a default rate so that the channel map is set up. */
 	hdspm_set_rate(hdspm, 48000, 1);
@@ -5371,6 +5659,16 @@ static int snd_hdspm_hw_params(struct snd_pcm_substream *substream,
 	   */
 
 
+	/*  For AES cards, the float format bit is the same as the
+	 *  preferred sync reference. Since we don't want to break
+	 *  sync settings, we have to skip the remaining part of this
+	 *  function.
+	 */
+	if (hdspm->io_type == AES32) {
+		return 0;
+	}
+
+
 	/* Switch to native float format if requested */
 	if (SNDRV_PCM_FORMAT_FLOAT_LE == params_format(params)) {
 		if (!(hdspm->control_register & HDSPe_FLOAT_FORMAT))
@@ -6013,7 +6311,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
 				ltc.format = fps_2997;
 				break;
 			default:
-				ltc.format = 30;
+				ltc.format = fps_30;
 				break;
 			}
 			if (i & HDSPM_TCO1_set_drop_frame_flag) {
@@ -6479,10 +6777,6 @@ static int snd_hdspm_create(struct snd_card *card,
 		break;
 
 	case AIO:
-		if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) {
-			snd_printk(KERN_INFO "HDSPM: AEB input board found, but not supported\n");
-		}
-
 		hdspm->ss_in_channels = AIO_IN_SS_CHANNELS;
 		hdspm->ds_in_channels = AIO_IN_DS_CHANNELS;
 		hdspm->qs_in_channels = AIO_IN_QS_CHANNELS;
@@ -6490,6 +6784,20 @@ static int snd_hdspm_create(struct snd_card *card,
 		hdspm->ds_out_channels = AIO_OUT_DS_CHANNELS;
 		hdspm->qs_out_channels = AIO_OUT_QS_CHANNELS;
 
+		if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) {
+			snd_printk(KERN_INFO "HDSPM: AEB input board found\n");
+			hdspm->ss_in_channels += 4;
+			hdspm->ds_in_channels += 4;
+			hdspm->qs_in_channels += 4;
+		}
+
+		if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBO_D)) {
+			snd_printk(KERN_INFO "HDSPM: AEB output board found\n");
+			hdspm->ss_out_channels += 4;
+			hdspm->ds_out_channels += 4;
+			hdspm->qs_out_channels += 4;
+		}
+
 		hdspm->channel_map_out_ss = channel_map_aio_out_ss;
 		hdspm->channel_map_out_ds = channel_map_aio_out_ds;
 		hdspm->channel_map_out_qs = channel_map_aio_out_qs;
@@ -6558,6 +6866,7 @@ static int snd_hdspm_create(struct snd_card *card,
 		break;
 
 	case MADI:
+	case AES32:
 		if (hdspm_read(hdspm, HDSPM_statusRegister) & HDSPM_tco_detect) {
 			hdspm->midiPorts++;
 			hdspm->tco = kzalloc(sizeof(struct hdspm_tco),
@@ -6565,7 +6874,7 @@ static int snd_hdspm_create(struct snd_card *card,
 			if (NULL != hdspm->tco) {
 				hdspm_tco_write(hdspm);
 			}
-			snd_printk(KERN_INFO "HDSPM: MADI TCO module found\n");
+			snd_printk(KERN_INFO "HDSPM: MADI/AES TCO module found\n");
 		} else {
 			hdspm->tco = NULL;
 		}
@@ -6580,10 +6889,12 @@ static int snd_hdspm_create(struct snd_card *card,
 	case AES32:
 		if (hdspm->tco) {
 			hdspm->texts_autosync = texts_autosync_aes_tco;
-			hdspm->texts_autosync_items = 10;
+			hdspm->texts_autosync_items =
+				ARRAY_SIZE(texts_autosync_aes_tco);
 		} else {
 			hdspm->texts_autosync = texts_autosync_aes;
-			hdspm->texts_autosync_items = 9;
+			hdspm->texts_autosync_items =
+				ARRAY_SIZE(texts_autosync_aes);
 		}
 		break;
 
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 45eeaa9f7fec..5138b8493051 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -26,12 +26,9 @@ if SND_SOC
 config SND_SOC_AC97_BUS
 	bool
 
-config SND_SOC_DMAENGINE_PCM
-	bool
-
 config SND_SOC_GENERIC_DMAENGINE_PCM
 	bool
-	select SND_SOC_DMAENGINE_PCM
+	select SND_DMAENGINE_PCM
 
 # All the supported SoCs
 source "sound/soc/atmel/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index bc0261476d7a..61a64d281905 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,10 +1,6 @@
 snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
 snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o
 
-ifneq ($(CONFIG_SND_SOC_DMAENGINE_PCM),)
-snd-soc-core-objs += soc-dmaengine-pcm.o
-endif
-
 ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),)
 snd-soc-core-objs += soc-generic-dmaengine-pcm.o
 endif
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index 3fdd87fa18a9..e48d38a1b95c 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -13,6 +13,7 @@ config SND_ATMEL_SOC_PDC
 config SND_ATMEL_SOC_DMA
 	tristate
 	depends on SND_ATMEL_SOC
+	select SND_SOC_GENERIC_DMAENGINE_PCM
 
 config SND_ATMEL_SOC_SSC
 	tristate
@@ -32,6 +33,26 @@ config SND_AT91_SOC_SAM9G20_WM8731
 	  Say Y if you want to add support for SoC audio on WM8731-based
 	  AT91sam9g20 evaluation board.
 
+config SND_ATMEL_SOC_WM8904
+	tristate "Atmel ASoC driver for boards using WM8904 codec"
+	depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC
+	select SND_ATMEL_SOC_SSC
+	select SND_ATMEL_SOC_DMA
+	select SND_SOC_WM8904
+	help
+	  Say Y if you want to add support for Atmel ASoC driver for boards using
+	  WM8904 codec.
+
+config SND_AT91_SOC_SAM9X5_WM8731
+	tristate "SoC Audio support for WM8731-based at91sam9x5 board"
+	depends on ATMEL_SSC && SND_ATMEL_SOC && SOC_AT91SAM9X5
+	select SND_ATMEL_SOC_SSC
+	select SND_ATMEL_SOC_DMA
+	select SND_SOC_WM8731
+	help
+	  Say Y if you want to add support for audio SoC on an
+	  at91sam9x5 based board that is using WM8731 codec.
+
 config SND_AT91_SOC_AFEB9260
 	tristate "SoC Audio support for AFEB9260 board"
 	depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index 41967ccb6f41..5baabc8bde3a 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -11,6 +11,10 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
 
 # AT91 Machine Support
 snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
+snd-atmel-soc-wm8904-objs := atmel_wm8904.o
+snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
 
 obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
+obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o
+obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o
 obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c
index d12826526798..06082e5e5dcb 100644
--- a/sound/soc/atmel/atmel-pcm-dma.c
+++ b/sound/soc/atmel/atmel-pcm-dma.c
@@ -91,138 +91,52 @@ static void atmel_pcm_dma_irq(u32 ssc_sr,
 	}
 }
 
-/*--------------------------------------------------------------------------*\
- * DMAENGINE operations
-\*--------------------------------------------------------------------------*/
-static bool filter(struct dma_chan *chan, void *slave)
-{
-	struct at_dma_slave *sl = slave;
-
-	if (sl->dma_dev == chan->device->dev) {
-		chan->private = sl;
-		return true;
-	} else {
-		return false;
-	}
-}
-
 static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params, struct atmel_pcm_dma_params *prtd)
+	struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config)
 {
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct atmel_pcm_dma_params *prtd;
 	struct ssc_device *ssc;
-	struct dma_chan *dma_chan;
-	struct dma_slave_config slave_config;
 	int ret;
 
+	prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 	ssc = prtd->ssc;
 
-	ret = snd_hwparams_to_dma_slave_config(substream, params,
-			&slave_config);
+	ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config);
 	if (ret) {
 		pr_err("atmel-pcm: hwparams to dma slave configure failed\n");
 		return ret;
 	}
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		slave_config.dst_addr = (dma_addr_t)ssc->phybase + SSC_THR;
-		slave_config.dst_maxburst = 1;
+		slave_config->dst_addr = ssc->phybase + SSC_THR;
+		slave_config->dst_maxburst = 1;
 	} else {
-		slave_config.src_addr = (dma_addr_t)ssc->phybase + SSC_RHR;
-		slave_config.src_maxburst = 1;
-	}
-
-	dma_chan = snd_dmaengine_pcm_get_chan(substream);
-	if (dmaengine_slave_config(dma_chan, &slave_config)) {
-		pr_err("atmel-pcm: failed to configure dma channel\n");
-		ret = -EBUSY;
-		return ret;
-	}
-
-	return 0;
-}
-
-static int atmel_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct atmel_pcm_dma_params *prtd;
-	struct ssc_device *ssc;
-	struct at_dma_slave *sdata = NULL;
-	int ret;
-
-	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-
-	prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-	ssc = prtd->ssc;
-	if (ssc->pdev)
-		sdata = ssc->pdev->dev.platform_data;
-
-	ret = snd_dmaengine_pcm_open_request_chan(substream, filter, sdata);
-	if (ret) {
-		pr_err("atmel-pcm: dmaengine pcm open failed\n");
-		return -EINVAL;
-	}
-
-	ret = atmel_pcm_configure_dma(substream, params, prtd);
-	if (ret) {
-		pr_err("atmel-pcm: failed to configure dmai\n");
-		goto err;
+		slave_config->src_addr = ssc->phybase + SSC_RHR;
+		slave_config->src_maxburst = 1;
 	}
 
 	prtd->dma_intr_handler = atmel_pcm_dma_irq;
 
 	return 0;
-err:
-	snd_dmaengine_pcm_close_release_chan(substream);
-	return ret;
 }
 
-static int atmel_pcm_dma_prepare(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct atmel_pcm_dma_params *prtd;
-
-	prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
-	ssc_writex(prtd->ssc->regs, SSC_IER, prtd->mask->ssc_error);
-	ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_enable);
-
-	return 0;
-}
-
-static int atmel_pcm_open(struct snd_pcm_substream *substream)
-{
-	snd_soc_set_runtime_hwparams(substream, &atmel_pcm_dma_hardware);
-
-	return 0;
-}
-
-static struct snd_pcm_ops atmel_pcm_ops = {
-	.open		= atmel_pcm_open,
-	.close		= snd_dmaengine_pcm_close_release_chan,
-	.ioctl		= snd_pcm_lib_ioctl,
-	.hw_params	= atmel_pcm_hw_params,
-	.prepare	= atmel_pcm_dma_prepare,
-	.trigger	= snd_dmaengine_pcm_trigger,
-	.pointer	= snd_dmaengine_pcm_pointer_no_residue,
-	.mmap		= atmel_pcm_mmap,
-};
-
-static struct snd_soc_platform_driver atmel_soc_platform = {
-	.ops		= &atmel_pcm_ops,
-	.pcm_new	= atmel_pcm_new,
-	.pcm_free	= atmel_pcm_free,
+static const struct snd_dmaengine_pcm_config atmel_dmaengine_pcm_config = {
+	.prepare_slave_config = atmel_pcm_configure_dma,
+	.pcm_hardware = &atmel_pcm_dma_hardware,
+	.prealloc_buffer_size = ATMEL_SSC_DMABUF_SIZE,
 };
 
 int atmel_pcm_dma_platform_register(struct device *dev)
 {
-	return snd_soc_register_platform(dev, &atmel_soc_platform);
+	return snd_dmaengine_pcm_register(dev, &atmel_dmaengine_pcm_config,
+			SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
 }
 EXPORT_SYMBOL(atmel_pcm_dma_platform_register);
 
 void atmel_pcm_dma_platform_unregister(struct device *dev)
 {
-	snd_soc_unregister_platform(dev);
+	snd_dmaengine_pcm_unregister(dev);
 }
 EXPORT_SYMBOL(atmel_pcm_dma_platform_unregister);
 
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index f3fdfa07fcb9..0ecf356027f6 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -73,6 +73,7 @@ static struct atmel_ssc_mask ssc_tx_mask = {
 	.ssc_disable	= SSC_BIT(CR_TXDIS),
 	.ssc_endx	= SSC_BIT(SR_ENDTX),
 	.ssc_endbuf	= SSC_BIT(SR_TXBUFE),
+	.ssc_error	= SSC_BIT(SR_OVRUN),
 	.pdc_enable	= ATMEL_PDC_TXTEN,
 	.pdc_disable	= ATMEL_PDC_TXTDIS,
 };
@@ -82,6 +83,7 @@ static struct atmel_ssc_mask ssc_rx_mask = {
 	.ssc_disable	= SSC_BIT(CR_RXDIS),
 	.ssc_endx	= SSC_BIT(SR_ENDRX),
 	.ssc_endbuf	= SSC_BIT(SR_RXBUFF),
+	.ssc_error	= SSC_BIT(SR_OVRUN),
 	.pdc_enable	= ATMEL_PDC_RXTEN,
 	.pdc_disable	= ATMEL_PDC_RXTDIS,
 };
@@ -196,15 +198,27 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *dai)
 {
 	struct atmel_ssc_info *ssc_p = &ssc_info[dai->id];
-	int dir_mask;
+	struct atmel_pcm_dma_params *dma_params;
+	int dir, dir_mask;
 
 	pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n",
 		ssc_readl(ssc_p->ssc->regs, SR));
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		dir = 0;
 		dir_mask = SSC_DIR_MASK_PLAYBACK;
-	else
+	} else {
+		dir = 1;
 		dir_mask = SSC_DIR_MASK_CAPTURE;
+	}
+
+	dma_params = &ssc_dma_params[dai->id][dir];
+	dma_params->ssc = ssc_p->ssc;
+	dma_params->substream = substream;
+
+	ssc_p->dma_params[dir] = dma_params;
+
+	snd_soc_dai_set_dma_data(dai, substream, dma_params);
 
 	spin_lock_irq(&ssc_p->lock);
 	if (ssc_p->dir_mask & dir_mask) {
@@ -325,7 +339,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params,
 	struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
 	int id = dai->id;
 	struct atmel_ssc_info *ssc_p = &ssc_info[id];
 	struct atmel_pcm_dma_params *dma_params;
@@ -344,19 +357,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 	else
 		dir = 1;
 
-	dma_params = &ssc_dma_params[id][dir];
-	dma_params->ssc = ssc_p->ssc;
-	dma_params->substream = substream;
-
-	ssc_p->dma_params[dir] = dma_params;
-
-	/*
-	 * The snd_soc_pcm_stream->dma_data field is only used to communicate
-	 * the appropriate DMA parameters to the pcm driver hw_params()
-	 * function.  It should not be used for other purposes
-	 * as it is common to all substreams.
-	 */
-	snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_params);
+	dma_params = ssc_p->dma_params[dir];
 
 	channels = params_channels(params);
 
@@ -648,6 +649,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
 	dma_params = ssc_p->dma_params[dir];
 
 	ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+	ssc_writel(ssc_p->ssc->regs, IER, dma_params->mask->ssc_error);
 
 	pr_debug("%s enabled SSC_SR=0x%08x\n",
 			dir ? "receive" : "transmit",
diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c
new file mode 100644
index 000000000000..7222380131ea
--- /dev/null
+++ b/sound/soc/atmel/atmel_wm8904.c
@@ -0,0 +1,254 @@
+/*
+ * atmel_wm8904 - Atmel ASoC driver for boards with WM8904 codec.
+ *
+ * Copyright (C) 2012 Atmel
+ *
+ * Author: Bo Shen <voice.shen@atmel.com>
+ *
+ * GPLv2 or later
+ */
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <linux/pinctrl/consumer.h>
+
+#include <sound/soc.h>
+
+#include "../codecs/wm8904.h"
+#include "atmel_ssc_dai.h"
+
+#define MCLK_RATE 32768
+
+static struct clk *mclk;
+
+static const struct snd_soc_dapm_widget atmel_asoc_wm8904_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Mic", NULL),
+	SND_SOC_DAPM_LINE("Line In Jack", NULL),
+};
+
+static int atmel_asoc_wm8904_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;
+	int ret;
+
+	ret = snd_soc_dai_set_pll(codec_dai, WM8904_FLL_MCLK, WM8904_FLL_MCLK,
+		32768, params_rate(params) * 256);
+	if (ret < 0) {
+		pr_err("%s - failed to set wm8904 codec PLL.", __func__);
+		return ret;
+	}
+
+	/*
+	 * As here wm8904 use FLL output as its system clock
+	 * so calling set_sysclk won't care freq parameter
+	 * then we pass 0
+	 */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8904_CLK_FLL,
+			0, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		pr_err("%s -failed to set wm8904 SYSCLK\n", __func__);
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops atmel_asoc_wm8904_ops = {
+	.hw_params = atmel_asoc_wm8904_hw_params,
+};
+
+static int atmel_set_bias_level(struct snd_soc_card *card,
+		struct snd_soc_dapm_context *dapm,
+		enum snd_soc_bias_level level)
+{
+	if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
+		switch (level) {
+		case SND_SOC_BIAS_PREPARE:
+			clk_prepare_enable(mclk);
+			break;
+		case SND_SOC_BIAS_OFF:
+			clk_disable_unprepare(mclk);
+			break;
+		default:
+			break;
+		}
+	}
+
+	return 0;
+};
+
+static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = {
+	.name = "WM8904",
+	.stream_name = "WM8904 PCM",
+	.codec_dai_name = "wm8904-hifi",
+	.dai_fmt = SND_SOC_DAIFMT_I2S
+		| SND_SOC_DAIFMT_NB_NF
+		| SND_SOC_DAIFMT_CBM_CFM,
+	.ops = &atmel_asoc_wm8904_ops,
+};
+
+static struct snd_soc_card atmel_asoc_wm8904_card = {
+	.name = "atmel_asoc_wm8904",
+	.owner = THIS_MODULE,
+	.set_bias_level = atmel_set_bias_level,
+	.dai_link = &atmel_asoc_wm8904_dailink,
+	.num_links = 1,
+	.dapm_widgets = atmel_asoc_wm8904_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(atmel_asoc_wm8904_dapm_widgets),
+	.fully_routed = true,
+};
+
+static int atmel_asoc_wm8904_dt_init(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *codec_np, *cpu_np;
+	struct snd_soc_card *card = &atmel_asoc_wm8904_card;
+	struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink;
+	int ret;
+
+	if (!np) {
+		dev_err(&pdev->dev, "only device tree supported\n");
+		return -EINVAL;
+	}
+
+	ret = snd_soc_of_parse_card_name(card, "atmel,model");
+	if (ret) {
+		dev_err(&pdev->dev, "failed to parse card name\n");
+		return ret;
+	}
+
+	ret = snd_soc_of_parse_audio_routing(card, "atmel,audio-routing");
+	if (ret) {
+		dev_err(&pdev->dev, "failed to parse audio routing\n");
+		return ret;
+	}
+
+	cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
+	if (!cpu_np) {
+		dev_err(&pdev->dev, "failed to get dai and pcm info\n");
+		ret = -EINVAL;
+		return ret;
+	}
+	dailink->cpu_of_node = cpu_np;
+	dailink->platform_of_node = cpu_np;
+	of_node_put(cpu_np);
+
+	codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
+	if (!codec_np) {
+		dev_err(&pdev->dev, "failed to get codec info\n");
+		ret = -EINVAL;
+		return ret;
+	}
+	dailink->codec_of_node = codec_np;
+	of_node_put(codec_np);
+
+	return 0;
+}
+
+static int atmel_asoc_wm8904_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &atmel_asoc_wm8904_card;
+	struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink;
+	struct clk *clk_src;
+	struct pinctrl *pinctrl;
+	int id, ret;
+
+	pinctrl = devm_pinctrl_get_select_default(&pdev->dev);
+	if (IS_ERR(pinctrl)) {
+		dev_err(&pdev->dev, "failed to request pinctrl\n");
+		return PTR_ERR(pinctrl);
+	}
+
+	card->dev = &pdev->dev;
+	ret = atmel_asoc_wm8904_dt_init(pdev);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to init dt info\n");
+		return ret;
+	}
+
+	id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc");
+	ret = atmel_ssc_set_audio(id);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "failed to set SSC %d for audio\n", id);
+		return ret;
+	}
+
+	mclk = clk_get(NULL, "pck0");
+	if (IS_ERR(mclk)) {
+		dev_err(&pdev->dev, "failed to get pck0\n");
+		ret = PTR_ERR(mclk);
+		goto err_set_audio;
+	}
+
+	clk_src = clk_get(NULL, "clk32k");
+	if (IS_ERR(clk_src)) {
+		dev_err(&pdev->dev, "failed to get clk32k\n");
+		ret = PTR_ERR(clk_src);
+		goto err_set_audio;
+	}
+
+	ret = clk_set_parent(mclk, clk_src);
+	clk_put(clk_src);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "failed to set MCLK parent\n");
+		goto err_set_audio;
+	}
+
+	dev_info(&pdev->dev, "setting pck0 to %dHz\n", MCLK_RATE);
+	clk_set_rate(mclk, MCLK_RATE);
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed\n");
+		goto err_set_audio;
+	}
+
+	return 0;
+
+err_set_audio:
+	atmel_ssc_put_audio(id);
+	return ret;
+}
+
+static int atmel_asoc_wm8904_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink;
+	int id;
+
+	id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc");
+
+	snd_soc_unregister_card(card);
+	atmel_ssc_put_audio(id);
+
+	return 0;
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = {
+	{ .compatible = "atmel,asoc-wm8904", },
+	{ }
+};
+#endif
+
+static struct platform_driver atmel_asoc_wm8904_driver = {
+	.driver = {
+		.name = "atmel-wm8904-audio",
+		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(atmel_asoc_wm8904_dt_ids),
+	},
+	.probe = atmel_asoc_wm8904_probe,
+	.remove = atmel_asoc_wm8904_remove,
+};
+
+module_platform_driver(atmel_asoc_wm8904_driver);
+
+/* Module information */
+MODULE_AUTHOR("Bo Shen <voice.shen@atmel.com>");
+MODULE_DESCRIPTION("ALSA SoC machine driver for Atmel EK with WM8904 codec");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c
new file mode 100644
index 000000000000..992ae38d5a15
--- /dev/null
+++ b/sound/soc/atmel/sam9x5_wm8731.c
@@ -0,0 +1,208 @@
+/*
+ * sam9x5_wm8731   --	SoC audio for AT91SAM9X5-based boards
+ *			that are using WM8731 as codec.
+ *
+ *  Copyright (C) 2011 Atmel,
+ *		  Nicolas Ferre <nicolas.ferre@atmel.com>
+ *
+ *  Copyright (C) 2013 Paratronic,
+ *		  Richard Genoud <richard.genoud@gmail.com>
+ *
+ * Based on sam9g20_wm8731.c by:
+ * Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+#include <linux/of.h>
+#include <linux/export.h>
+#include <linux/module.h>
+#include <linux/mod_devicetable.h>
+#include <linux/platform_device.h>
+#include <linux/device.h>
+
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
+
+#include "../codecs/wm8731.h"
+#include "atmel_ssc_dai.h"
+
+
+#define MCLK_RATE 12288000
+
+#define DRV_NAME "sam9x5-snd-wm8731"
+
+struct sam9x5_drvdata {
+	int ssc_id;
+};
+
+/*
+ * Logic for a wm8731 as connected on a at91sam9x5ek based board.
+ */
+static int sam9x5_wm8731_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct device *dev = rtd->dev;
+	int ret;
+
+	dev_dbg(dev, "ASoC: %s called\n", __func__);
+
+	/* set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
+				     MCLK_RATE, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(dev, "ASoC: Failed to set WM8731 SYSCLK: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+/*
+ * Audio paths on at91sam9x5ek board:
+ *
+ *  |A| ------------> |      | ---R----> Headphone Jack
+ *  |T| <----\        |  WM  | ---L--/
+ *  |9| ---> CLK <--> | 8731 | <--R----- Line In Jack
+ *  |1| <------------ |      | <--L--/
+ */
+static const struct snd_soc_dapm_widget sam9x5_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In Jack", NULL),
+};
+
+static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *codec_np, *cpu_np;
+	struct snd_soc_card *card;
+	struct snd_soc_dai_link *dai;
+	struct sam9x5_drvdata *priv;
+	int ret;
+
+	if (!np) {
+		dev_err(&pdev->dev, "No device node supplied\n");
+		return -EINVAL;
+	}
+
+	card = devm_kzalloc(&pdev->dev, sizeof(*card), GFP_KERNEL);
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	dai = devm_kzalloc(&pdev->dev, sizeof(*dai), GFP_KERNEL);
+	if (!dai || !card || !priv) {
+		ret = -ENOMEM;
+		goto out;
+	}
+
+	card->dev = &pdev->dev;
+	card->owner = THIS_MODULE;
+	card->dai_link = dai;
+	card->num_links = 1;
+	card->dapm_widgets = sam9x5_dapm_widgets;
+	card->num_dapm_widgets = ARRAY_SIZE(sam9x5_dapm_widgets);
+	dai->name = "WM8731";
+	dai->stream_name = "WM8731 PCM";
+	dai->codec_dai_name = "wm8731-hifi";
+	dai->init = sam9x5_wm8731_init;
+	dai->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+		| SND_SOC_DAIFMT_CBM_CFM;
+
+	ret = snd_soc_of_parse_card_name(card, "atmel,model");
+	if (ret) {
+		dev_err(&pdev->dev, "atmel,model node missing\n");
+		goto out;
+	}
+
+	ret = snd_soc_of_parse_audio_routing(card, "atmel,audio-routing");
+	if (ret) {
+		dev_err(&pdev->dev, "atmel,audio-routing node missing\n");
+		goto out;
+	}
+
+	codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
+	if (!codec_np) {
+		dev_err(&pdev->dev, "atmel,audio-codec node missing\n");
+		ret = -EINVAL;
+		goto out;
+	}
+
+	dai->codec_of_node = codec_np;
+
+	cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
+	if (!cpu_np) {
+		dev_err(&pdev->dev, "atmel,ssc-controller node missing\n");
+		ret = -EINVAL;
+		goto out;
+	}
+	dai->cpu_of_node = cpu_np;
+	dai->platform_of_node = cpu_np;
+
+	priv->ssc_id = of_alias_get_id(cpu_np, "ssc");
+
+	ret = atmel_ssc_set_audio(priv->ssc_id);
+	if (ret != 0) {
+		dev_err(&pdev->dev,
+			"ASoC: Failed to set SSC %d for audio: %d\n",
+			ret, priv->ssc_id);
+		goto out;
+	}
+
+	of_node_put(codec_np);
+	of_node_put(cpu_np);
+
+	platform_set_drvdata(pdev, card);
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev,
+			"ASoC: Platform device allocation failed\n");
+		goto out_put_audio;
+	}
+
+	dev_dbg(&pdev->dev, "ASoC: %s ok\n", __func__);
+
+	return ret;
+
+out_put_audio:
+	atmel_ssc_put_audio(priv->ssc_id);
+out:
+	return ret;
+}
+
+static int sam9x5_wm8731_driver_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct sam9x5_drvdata *priv = card->drvdata;
+
+	snd_soc_unregister_card(card);
+	atmel_ssc_put_audio(priv->ssc_id);
+
+	return 0;
+}
+
+static const struct of_device_id sam9x5_wm8731_of_match[] = {
+	{ .compatible = "atmel,sam9x5-wm8731-audio", },
+	{},
+};
+MODULE_DEVICE_TABLE(of, sam9x5_wm8731_of_match);
+
+static struct platform_driver sam9x5_wm8731_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(sam9x5_wm8731_of_match),
+	},
+	.probe = sam9x5_wm8731_driver_probe,
+	.remove = sam9x5_wm8731_driver_remove,
+};
+module_platform_driver(sam9x5_wm8731_driver);
+
+/* Module information */
+MODULE_AUTHOR("Nicolas Ferre <nicolas.ferre@atmel.com>");
+MODULE_AUTHOR("Richard Genoud <richard.genoud@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC machine driver for AT91SAM9x5 - WM8731");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index a497a0cfeba1..decba87a074c 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -73,12 +73,14 @@ static struct snd_soc_dai_link db1300_ac97_dai = {
 
 static struct snd_soc_card db1300_ac97_machine = {
 	.name		= "DB1300_AC97",
+	.owner		= THIS_MODULE,
 	.dai_link	= &db1300_ac97_dai,
 	.num_links	= 1,
 };
 
 static struct snd_soc_card db1550_ac97_machine = {
 	.name		= "DB1550_AC97",
+	.owner		= THIS_MODULE,
 	.dai_link	= &db1200_ac97_dai,
 	.num_links	= 1,
 };
@@ -145,6 +147,7 @@ static struct snd_soc_dai_link db1300_i2s_dai = {
 
 static struct snd_soc_card db1300_i2s_machine = {
 	.name		= "DB1300_I2S",
+	.owner		= THIS_MODULE,
 	.dai_link	= &db1300_i2s_dai,
 	.num_links	= 1,
 };
@@ -161,6 +164,7 @@ static struct snd_soc_dai_link db1550_i2s_dai = {
 
 static struct snd_soc_card db1550_i2s_machine = {
 	.name		= "DB1550_I2S",
+	.owner		= THIS_MODULE,
 	.dai_link	= &db1550_i2s_dai,
 	.num_links	= 1,
 };
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index a822ab822bb7..986dcec79fa0 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -379,9 +379,6 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev)
 	mutex_init(&wd->lock);
 
 	iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!iores)
-		return -ENODEV;
-
 	wd->mmio = devm_ioremap_resource(&pdev->dev, iores);
 	if (IS_ERR(wd->mmio))
 		return PTR_ERR(wd->mmio);
diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h
index 0c3e22d90a8d..a680fdc9bb42 100644
--- a/sound/soc/blackfin/bf5xx-ac97.h
+++ b/sound/soc/blackfin/bf5xx-ac97.h
@@ -9,7 +9,6 @@
 #ifndef _BF5XX_AC97_H
 #define _BF5XX_AC97_H
 
-extern struct snd_ac97 *ac97;
 /* Frame format in memory, only support stereo currently */
 struct ac97_frame {
 	u16 ac97_tag;		/* slot 0 */
diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c
index 04491f0e8d1b..efa75b5086a4 100644
--- a/sound/soc/cirrus/ep93xx-ac97.c
+++ b/sound/soc/cirrus/ep93xx-ac97.c
@@ -363,9 +363,6 @@ static int ep93xx_ac97_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!res)
-		return -ENODEV;
-
 	info->regs = devm_ioremap_resource(&pdev->dev, res);
 	if (IS_ERR(info->regs))
 		return PTR_ERR(info->regs);
diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c
index 17ad70bca9fe..a57643d6402f 100644
--- a/sound/soc/cirrus/ep93xx-i2s.c
+++ b/sound/soc/cirrus/ep93xx-i2s.c
@@ -376,9 +376,6 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!res)
-		return -ENODEV;
-
 	info->regs = devm_ioremap_resource(&pdev->dev, res);
 	if (IS_ERR(info->regs))
 		return PTR_ERR(info->regs);
@@ -411,7 +408,6 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
 	return 0;
 
 fail_put_lrclk:
-	dev_set_drvdata(&pdev->dev, NULL);
 	clk_put(info->lrclk);
 fail_put_sclk:
 	clk_put(info->sclk);
@@ -426,7 +422,6 @@ static int ep93xx_i2s_remove(struct platform_device *pdev)
 	struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev);
 
 	snd_soc_unregister_component(&pdev->dev);
-	dev_set_drvdata(&pdev->dev, NULL);
 	clk_put(info->lrclk);
 	clk_put(info->sclk);
 	clk_put(info->mclk);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index badb6fbacaa6..15106c045478 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI
 
 config SND_SOC_ALL_CODECS
 	tristate "Build all ASoC CODEC drivers"
+	depends on COMPILE_TEST
 	select SND_SOC_88PM860X if MFD_88PM860X
 	select SND_SOC_L3
 	select SND_SOC_AB8500_CODEC if ABX500_CORE
@@ -20,6 +21,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AD73311
 	select SND_SOC_ADAU1373 if I2C
 	select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI
+	select SND_SOC_ADAU1701 if I2C
 	select SND_SOC_ADS117X
 	select SND_SOC_AK4104 if SPI_MASTER
 	select SND_SOC_AK4535 if I2C
@@ -54,6 +56,8 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MC13783 if MFD_MC13XXX
 	select SND_SOC_ML26124 if I2C
 	select SND_SOC_HDMI_CODEC
+	select SND_SOC_PCM1681 if I2C
+	select SND_SOC_PCM1792A if SPI_MASTER
 	select SND_SOC_PCM3008
 	select SND_SOC_RT5631 if I2C
 	select SND_SOC_RT5640 if I2C
@@ -122,6 +126,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8994 if MFD_WM8994
 	select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8996 if I2C
+	select SND_SOC_WM8997 if MFD_WM8997
 	select SND_SOC_WM9081 if I2C
 	select SND_SOC_WM9090 if I2C
 	select SND_SOC_WM9705 if SND_SOC_AC97_BUS
@@ -145,8 +150,10 @@ config SND_SOC_ARIZONA
 	tristate
 	default y if SND_SOC_WM5102=y
 	default y if SND_SOC_WM5110=y
+	default y if SND_SOC_WM8997=y
 	default m if SND_SOC_WM5102=m
 	default m if SND_SOC_WM5110=m
+	default m if SND_SOC_WM8997=m
 
 config SND_SOC_WM_HUBS
 	tristate
@@ -198,6 +205,9 @@ config SND_SOC_AK4104
 config SND_SOC_AK4535
 	tristate
 
+config SND_SOC_AK4554
+	tristate
+
 config SND_SOC_AK4641
 	tristate
 
@@ -292,6 +302,12 @@ config SND_SOC_MAX9850
 config SND_SOC_HDMI_CODEC
        tristate
 
+config SND_SOC_PCM1681
+       tristate
+
+config SND_SOC_PCM1792A
+       tristate
+
 config SND_SOC_PCM3008
        tristate
 
@@ -500,6 +516,9 @@ config SND_SOC_WM8995
 config SND_SOC_WM8996
 	tristate
 
+config SND_SOC_WM8997
+	tristate
+
 config SND_SOC_WM9081
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 70fd8066f546..bc126764a44d 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -11,6 +11,7 @@ snd-soc-adav80x-objs := adav80x.o
 snd-soc-ads117x-objs := ads117x.o
 snd-soc-ak4104-objs := ak4104.o
 snd-soc-ak4535-objs := ak4535.o
+snd-soc-ak4554-objs := ak4554.o
 snd-soc-ak4641-objs := ak4641.o
 snd-soc-ak4642-objs := ak4642.o
 snd-soc-ak4671-objs := ak4671.o
@@ -42,6 +43,8 @@ snd-soc-max9850-objs := max9850.o
 snd-soc-mc13783-objs := mc13783.o
 snd-soc-ml26124-objs := ml26124.o
 snd-soc-hdmi-codec-objs := hdmi.o
+snd-soc-pcm1681-objs := pcm1681.o
+snd-soc-pcm1792a-codec-objs := pcm1792a.o
 snd-soc-pcm3008-objs := pcm3008.o
 snd-soc-rt5631-objs := rt5631.o
 snd-soc-rt5640-objs := rt5640.o
@@ -114,6 +117,7 @@ snd-soc-wm8991-objs := wm8991.o
 snd-soc-wm8993-objs := wm8993.o
 snd-soc-wm8994-objs := wm8994.o wm8958-dsp2.o
 snd-soc-wm8995-objs := wm8995.o
+snd-soc-wm8997-objs := wm8997.o
 snd-soc-wm9081-objs := wm9081.o
 snd-soc-wm9090-objs := wm9090.o
 snd-soc-wm9705-objs := wm9705.o
@@ -138,6 +142,7 @@ obj-$(CONFIG_SND_SOC_ADAV80X)  += snd-soc-adav80x.o
 obj-$(CONFIG_SND_SOC_ADS117X)	+= snd-soc-ads117x.o
 obj-$(CONFIG_SND_SOC_AK4104)	+= snd-soc-ak4104.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
+obj-$(CONFIG_SND_SOC_AK4554)	+= snd-soc-ak4554.o
 obj-$(CONFIG_SND_SOC_AK4641)	+= snd-soc-ak4641.o
 obj-$(CONFIG_SND_SOC_AK4642)	+= snd-soc-ak4642.o
 obj-$(CONFIG_SND_SOC_AK4671)	+= snd-soc-ak4671.o
@@ -171,6 +176,8 @@ obj-$(CONFIG_SND_SOC_MAX9850)	+= snd-soc-max9850.o
 obj-$(CONFIG_SND_SOC_MC13783)	+= snd-soc-mc13783.o
 obj-$(CONFIG_SND_SOC_ML26124)	+= snd-soc-ml26124.o
 obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o
+obj-$(CONFIG_SND_SOC_PCM1681)	+= snd-soc-pcm1681.o
+obj-$(CONFIG_SND_SOC_PCM1792A)	+= snd-soc-pcm1792a-codec.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
 obj-$(CONFIG_SND_SOC_RT5631)	+= snd-soc-rt5631.o
 obj-$(CONFIG_SND_SOC_RT5640)	+= snd-soc-rt5640.o
@@ -239,6 +246,7 @@ obj-$(CONFIG_SND_SOC_WM8991)	+= snd-soc-wm8991.o
 obj-$(CONFIG_SND_SOC_WM8993)	+= snd-soc-wm8993.o
 obj-$(CONFIG_SND_SOC_WM8994)	+= snd-soc-wm8994.o
 obj-$(CONFIG_SND_SOC_WM8995)	+= snd-soc-wm8995.o
+obj-$(CONFIG_SND_SOC_WM8997)	+= snd-soc-wm8997.o
 obj-$(CONFIG_SND_SOC_WM9081)	+= snd-soc-wm9081.o
 obj-$(CONFIG_SND_SOC_WM9090)	+= snd-soc-wm9090.o
 obj-$(CONFIG_SND_SOC_WM9705)	+= snd-soc-wm9705.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index ec7351803c24..8d9ba4ba4bfe 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -23,6 +23,16 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
+static const struct snd_soc_dapm_widget ac97_widgets[] = {
+	SND_SOC_DAPM_INPUT("RX"),
+	SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route ac97_routes[] = {
+	{ "AC97 Capture", NULL, "RX" },
+	{ "TX", NULL, "AC97 Playback" },
+};
+
 static int ac97_prepare(struct snd_pcm_substream *substream,
 			struct snd_soc_dai *dai)
 {
@@ -117,6 +127,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = {
 	.probe = 	ac97_soc_probe,
 	.suspend =	ac97_soc_suspend,
 	.resume =	ac97_soc_resume,
+
+	.dapm_widgets = ac97_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ac97_widgets),
+	.dapm_routes = ac97_routes,
+	.num_dapm_routes = ARRAY_SIZE(ac97_routes),
 };
 
 static int ac97_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 89fcf7d6e7b8..7257a8885f42 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -96,6 +96,44 @@ SOC_ENUM("Capture Source", ad1980_cap_src),
 SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
 };
 
+static const struct snd_soc_dapm_widget ad1980_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("MIC1"),
+SND_SOC_DAPM_INPUT("MIC2"),
+SND_SOC_DAPM_INPUT("CD_L"),
+SND_SOC_DAPM_INPUT("CD_R"),
+SND_SOC_DAPM_INPUT("AUX_L"),
+SND_SOC_DAPM_INPUT("AUX_R"),
+SND_SOC_DAPM_INPUT("LINE_IN_L"),
+SND_SOC_DAPM_INPUT("LINE_IN_R"),
+
+SND_SOC_DAPM_OUTPUT("LFE_OUT"),
+SND_SOC_DAPM_OUTPUT("CENTER_OUT"),
+SND_SOC_DAPM_OUTPUT("LINE_OUT_L"),
+SND_SOC_DAPM_OUTPUT("LINE_OUT_R"),
+SND_SOC_DAPM_OUTPUT("MONO_OUT"),
+SND_SOC_DAPM_OUTPUT("HP_OUT_L"),
+SND_SOC_DAPM_OUTPUT("HP_OUT_R"),
+};
+
+static const struct snd_soc_dapm_route ad1980_dapm_routes[] = {
+	{ "Capture", NULL, "MIC1" },
+	{ "Capture", NULL, "MIC2" },
+	{ "Capture", NULL, "CD_L" },
+	{ "Capture", NULL, "CD_R" },
+	{ "Capture", NULL, "AUX_L" },
+	{ "Capture", NULL, "AUX_R" },
+	{ "Capture", NULL, "LINE_IN_L" },
+	{ "Capture", NULL, "LINE_IN_R" },
+
+	{ "LFE_OUT", NULL, "Playback" },
+	{ "CENTER_OUT", NULL, "Playback" },
+	{ "LINE_OUT_L", NULL, "Playback" },
+	{ "LINE_OUT_R", NULL, "Playback" },
+	{ "MONO_OUT", NULL, "Playback" },
+	{ "HP_OUT_L", NULL, "Playback" },
+	{ "HP_OUT_R", NULL, "Playback" },
+};
+
 static unsigned int ac97_read(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
@@ -253,6 +291,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ad1980 = {
 	.reg_cache_step = 2,
 	.write = ac97_write,
 	.read = ac97_read,
+
+	.dapm_widgets = ad1980_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ad1980_dapm_widgets),
+	.dapm_routes = ad1980_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(ad1980_dapm_routes),
 };
 
 static int ad1980_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
index b1f2baf42b48..5fac8adbc136 100644
--- a/sound/soc/codecs/ad73311.c
+++ b/sound/soc/codecs/ad73311.c
@@ -23,6 +23,21 @@
 
 #include "ad73311.h"
 
+static const struct snd_soc_dapm_widget ad73311_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("VINP"),
+SND_SOC_DAPM_INPUT("VINN"),
+SND_SOC_DAPM_OUTPUT("VOUTN"),
+SND_SOC_DAPM_OUTPUT("VOUTP"),
+};
+
+static const struct snd_soc_dapm_route ad73311_dapm_routes[] = {
+	{ "Capture", NULL, "VINP" },
+	{ "Capture", NULL, "VINN" },
+
+	{ "VOUTN", NULL, "Playback" },
+	{ "VOUTP", NULL, "Playback" },
+};
+
 static struct snd_soc_dai_driver ad73311_dai = {
 	.name = "ad73311-hifi",
 	.playback = {
@@ -39,7 +54,12 @@ static struct snd_soc_dai_driver ad73311_dai = {
 		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_ad73311;
+static struct snd_soc_codec_driver soc_codec_dev_ad73311 = {
+	.dapm_widgets = ad73311_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ad73311_dapm_widgets),
+	.dapm_routes = ad73311_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(ad73311_dapm_routes),
+};
 
 static int ad73311_probe(struct platform_device *pdev)
 {
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index d1124a5b3471..ebff1128be59 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -91,7 +91,7 @@
 #define ADAU1701_OSCIPOW_OPD		0x04
 #define ADAU1701_DACSET_DACINIT		1
 
-#define ADAU1707_CLKDIV_UNSET		(-1UL)
+#define ADAU1707_CLKDIV_UNSET		(-1U)
 
 #define ADAU1701_FIRMWARE "adau1701.bin"
 
@@ -247,21 +247,21 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv)
 	    gpio_is_valid(adau1701->gpio_pll_mode[1])) {
 		switch (clkdiv) {
 		case 64:
-			gpio_set_value(adau1701->gpio_pll_mode[0], 0);
-			gpio_set_value(adau1701->gpio_pll_mode[1], 0);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0);
 			break;
 		case 256:
-			gpio_set_value(adau1701->gpio_pll_mode[0], 0);
-			gpio_set_value(adau1701->gpio_pll_mode[1], 1);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1);
 			break;
 		case 384:
-			gpio_set_value(adau1701->gpio_pll_mode[0], 1);
-			gpio_set_value(adau1701->gpio_pll_mode[1], 0);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0);
 			break;
 		case 0:	/* fallback */
 		case 512:
-			gpio_set_value(adau1701->gpio_pll_mode[0], 1);
-			gpio_set_value(adau1701->gpio_pll_mode[1], 1);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1);
+			gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1);
 			break;
 		}
 	}
@@ -269,10 +269,10 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv)
 	adau1701->pll_clkdiv = clkdiv;
 
 	if (gpio_is_valid(adau1701->gpio_nreset)) {
-		gpio_set_value(adau1701->gpio_nreset, 0);
+		gpio_set_value_cansleep(adau1701->gpio_nreset, 0);
 		/* minimum reset time is 20ns */
 		udelay(1);
-		gpio_set_value(adau1701->gpio_nreset, 1);
+		gpio_set_value_cansleep(adau1701->gpio_nreset, 1);
 		/* power-up time may be as long as 85ms */
 		mdelay(85);
 	}
@@ -734,7 +734,10 @@ static int adau1701_i2c_remove(struct i2c_client *client)
 }
 
 static const struct i2c_device_id adau1701_i2c_id[] = {
+	{ "adau1401", 0 },
+	{ "adau1401a", 0 },
 	{ "adau1701", 0 },
+	{ "adau1702", 0 },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id);
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index 3c839cc4e00e..15b012d0f226 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -868,6 +868,12 @@ static int adav80x_bus_remove(struct device *dev)
 }
 
 #if defined(CONFIG_SPI_MASTER)
+static const struct spi_device_id adav80x_spi_id[] = {
+	{ "adav801", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(spi, adav80x_spi_id);
+
 static int adav80x_spi_probe(struct spi_device *spi)
 {
 	return adav80x_bus_probe(&spi->dev, SND_SOC_SPI);
@@ -885,15 +891,16 @@ static struct spi_driver adav80x_spi_driver = {
 	},
 	.probe		= adav80x_spi_probe,
 	.remove		= adav80x_spi_remove,
+	.id_table	= adav80x_spi_id,
 };
 #endif
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-static const struct i2c_device_id adav80x_id[] = {
+static const struct i2c_device_id adav80x_i2c_id[] = {
 	{ "adav803", 0 },
 	{ }
 };
-MODULE_DEVICE_TABLE(i2c, adav80x_id);
+MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id);
 
 static int adav80x_i2c_probe(struct i2c_client *client,
 			     const struct i2c_device_id *id)
@@ -913,7 +920,7 @@ static struct i2c_driver adav80x_i2c_driver = {
 	},
 	.probe = adav80x_i2c_probe,
 	.remove = adav80x_i2c_remove,
-	.id_table = adav80x_id,
+	.id_table = adav80x_i2c_id,
 };
 #endif
 
diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c
index 506d474c4d22..8f388edff586 100644
--- a/sound/soc/codecs/ads117x.c
+++ b/sound/soc/codecs/ads117x.c
@@ -23,6 +23,28 @@
 #define ADS117X_RATES (SNDRV_PCM_RATE_8000_48000)
 #define ADS117X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
 
+static const struct snd_soc_dapm_widget ads117x_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("Input1"),
+SND_SOC_DAPM_INPUT("Input2"),
+SND_SOC_DAPM_INPUT("Input3"),
+SND_SOC_DAPM_INPUT("Input4"),
+SND_SOC_DAPM_INPUT("Input5"),
+SND_SOC_DAPM_INPUT("Input6"),
+SND_SOC_DAPM_INPUT("Input7"),
+SND_SOC_DAPM_INPUT("Input8"),
+};
+
+static const struct snd_soc_dapm_route ads117x_dapm_routes[] = {
+	{ "Capture", NULL, "Input1" },
+	{ "Capture", NULL, "Input2" },
+	{ "Capture", NULL, "Input3" },
+	{ "Capture", NULL, "Input4" },
+	{ "Capture", NULL, "Input5" },
+	{ "Capture", NULL, "Input6" },
+	{ "Capture", NULL, "Input7" },
+	{ "Capture", NULL, "Input8" },
+};
+
 static struct snd_soc_dai_driver ads117x_dai = {
 /* ADC */
 	.name = "ads117x-hifi",
@@ -34,7 +56,12 @@ static struct snd_soc_dai_driver ads117x_dai = {
 		.formats = ADS117X_FORMATS,},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_ads117x;
+static struct snd_soc_codec_driver soc_codec_dev_ads117x = {
+	.dapm_widgets = ads117x_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ads117x_dapm_widgets),
+	.dapm_routes = ads117x_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(ads117x_dapm_routes),
+};
 
 static int ads117x_probe(struct platform_device *pdev)
 {
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index c7cfdf957e4d..71059c07ae7b 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -51,6 +51,17 @@ struct ak4104_private {
 	struct regmap *regmap;
 };
 
+static const struct snd_soc_dapm_widget ak4104_dapm_widgets[] = {
+SND_SOC_DAPM_PGA("TXE", AK4104_REG_TX, AK4104_TX_TXE, 0, NULL, 0),
+
+SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route ak4104_dapm_routes[] = {
+	{ "TXE", NULL, "Playback" },
+	{ "TX", NULL, "TXE" },
+};
+
 static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			      unsigned int format)
 {
@@ -138,29 +149,11 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
-	/* enable transmitter */
-	ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX,
-				 AK4104_TX_TXE, AK4104_TX_TXE);
-	if (ret < 0)
-		return ret;
-
 	return 0;
 }
 
-static int ak4104_hw_free(struct snd_pcm_substream *substream,
-			  struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-	struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
-
-	/* disable transmitter */
-	return regmap_update_bits(ak4104->regmap, AK4104_REG_TX,
-				  AK4104_TX_TXE, 0);
-}
-
 static const struct snd_soc_dai_ops ak4101_dai_ops = {
 	.hw_params = ak4104_hw_params,
-	.hw_free = ak4104_hw_free,
 	.set_fmt = ak4104_set_dai_fmt,
 };
 
@@ -214,6 +207,11 @@ static int ak4104_remove(struct snd_soc_codec *codec)
 static struct snd_soc_codec_driver soc_codec_device_ak4104 = {
 	.probe =	ak4104_probe,
 	.remove =	ak4104_remove,
+
+	.dapm_widgets = ak4104_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ak4104_dapm_widgets),
+	.dapm_routes = ak4104_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(ak4104_dapm_routes),
 };
 
 static const struct regmap_config ak4104_regmap = {
diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c
new file mode 100644
index 000000000000..79e9555766c0
--- /dev/null
+++ b/sound/soc/codecs/ak4554.c
@@ -0,0 +1,106 @@
+/*
+ * ak4554.c
+ *
+ * Copyright (C) 2013 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <sound/soc.h>
+
+/*
+ * ak4554 is very simple DA/AD converter which has no setting register.
+ *
+ * CAUTION
+ *
+ * ak4554 playback format is SND_SOC_DAIFMT_RIGHT_J,
+ * and,   capture  format is SND_SOC_DAIFMT_LEFT_J
+ * on same bit clock, LR clock.
+ * But, this driver doesn't have snd_soc_dai_ops :: set_fmt
+ *
+ * CPU/Codec DAI image
+ *
+ * CPU-DAI1 (plaback only fmt = RIGHT_J) --+-- ak4554
+ *					   |
+ * CPU-DAI2 (capture only fmt = LEFT_J) ---+
+ */
+
+static const struct snd_soc_dapm_widget ak4554_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("AINL"),
+SND_SOC_DAPM_INPUT("AINR"),
+
+SND_SOC_DAPM_OUTPUT("AOUTL"),
+SND_SOC_DAPM_OUTPUT("AOUTR"),
+};
+
+static const struct snd_soc_dapm_route ak4554_dapm_routes[] = {
+	{ "Capture", NULL, "AINL" },
+	{ "Capture", NULL, "AINR" },
+
+	{ "AOUTL", NULL, "Playback" },
+	{ "AOUTR", NULL, "Playback" },
+};
+
+static struct snd_soc_dai_driver ak4554_dai = {
+	.name = "ak4554-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.symmetric_rates = 1,
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_ak4554 = {
+	.dapm_widgets = ak4554_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ak4554_dapm_widgets),
+	.dapm_routes = ak4554_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(ak4554_dapm_routes),
+};
+
+static int ak4554_soc_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_codec(&pdev->dev,
+				      &soc_codec_dev_ak4554,
+				      &ak4554_dai, 1);
+}
+
+static int ak4554_soc_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+static struct of_device_id ak4554_of_match[] = {
+	{ .compatible = "asahi-kasei,ak4554" },
+	{},
+};
+MODULE_DEVICE_TABLE(of, ak4554_of_match);
+
+static struct platform_driver ak4554_driver = {
+	.driver = {
+		.name = "ak4554-adc-dac",
+		.owner = THIS_MODULE,
+		.of_match_table = ak4554_of_match,
+	},
+	.probe	= ak4554_soc_probe,
+	.remove	= ak4554_soc_remove,
+};
+module_platform_driver(ak4554_driver);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("SoC AK4554 driver");
+MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c
index 1f303983ae02..72e953b2cb41 100644
--- a/sound/soc/codecs/ak5386.c
+++ b/sound/soc/codecs/ak5386.c
@@ -22,7 +22,22 @@ struct ak5386_priv {
 	int reset_gpio;
 };
 
-static struct snd_soc_codec_driver soc_codec_ak5386;
+static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("AINL"),
+SND_SOC_DAPM_INPUT("AINR"),
+};
+
+static const struct snd_soc_dapm_route ak5386_dapm_routes[] = {
+	{ "Capture", NULL, "AINL" },
+	{ "Capture", NULL, "AINR" },
+};
+
+static struct snd_soc_codec_driver soc_codec_ak5386 = {
+	.dapm_widgets = ak5386_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ak5386_dapm_widgets),
+	.dapm_routes = ak5386_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(ak5386_dapm_routes),
+};
 
 static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			      unsigned int format)
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index de625813c0e6..657808ba1418 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -19,6 +19,7 @@
 #include <sound/tlv.h>
 
 #include <linux/mfd/arizona/core.h>
+#include <linux/mfd/arizona/gpio.h>
 #include <linux/mfd/arizona/registers.h>
 
 #include "arizona.h"
@@ -199,9 +200,16 @@ int arizona_init_spk(struct snd_soc_codec *codec)
 	if (ret != 0)
 		return ret;
 
-	ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkr, 1);
-	if (ret != 0)
-		return ret;
+	switch (arizona->type) {
+	case WM8997:
+		break;
+	default:
+		ret = snd_soc_dapm_new_controls(&codec->dapm,
+						&arizona_spkr, 1);
+		if (ret != 0)
+			return ret;
+		break;
+	}
 
 	ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN_WARN,
 				  "Thermal warning", arizona_thermal_warn,
@@ -223,6 +231,41 @@ int arizona_init_spk(struct snd_soc_codec *codec)
 }
 EXPORT_SYMBOL_GPL(arizona_init_spk);
 
+int arizona_init_gpio(struct snd_soc_codec *codec)
+{
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->arizona;
+	int i;
+
+	switch (arizona->type) {
+	case WM5110:
+		snd_soc_dapm_disable_pin(&codec->dapm, "DRC2 Signal Activity");
+		break;
+	default:
+		break;
+	}
+
+	snd_soc_dapm_disable_pin(&codec->dapm, "DRC1 Signal Activity");
+
+	for (i = 0; i < ARRAY_SIZE(arizona->pdata.gpio_defaults); i++) {
+		switch (arizona->pdata.gpio_defaults[i] & ARIZONA_GPN_FN_MASK) {
+		case ARIZONA_GP_FN_DRC1_SIGNAL_DETECT:
+			snd_soc_dapm_enable_pin(&codec->dapm,
+						"DRC1 Signal Activity");
+			break;
+		case ARIZONA_GP_FN_DRC2_SIGNAL_DETECT:
+			snd_soc_dapm_enable_pin(&codec->dapm,
+						"DRC2 Signal Activity");
+			break;
+		default:
+			break;
+		}
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_gpio);
+
 const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
 	"None",
 	"Tone Generator 1",
@@ -517,6 +560,26 @@ const struct soc_enum arizona_ng_hold =
 			4, arizona_ng_hold_text);
 EXPORT_SYMBOL_GPL(arizona_ng_hold);
 
+static const char * const arizona_in_dmic_osr_text[] = {
+	"1.536MHz", "3.072MHz", "6.144MHz",
+};
+
+const struct soc_enum arizona_in_dmic_osr[] = {
+	SOC_ENUM_SINGLE(ARIZONA_IN1L_CONTROL, ARIZONA_IN1_OSR_SHIFT,
+			ARRAY_SIZE(arizona_in_dmic_osr_text),
+			arizona_in_dmic_osr_text),
+	SOC_ENUM_SINGLE(ARIZONA_IN2L_CONTROL, ARIZONA_IN2_OSR_SHIFT,
+			ARRAY_SIZE(arizona_in_dmic_osr_text),
+			arizona_in_dmic_osr_text),
+	SOC_ENUM_SINGLE(ARIZONA_IN3L_CONTROL, ARIZONA_IN3_OSR_SHIFT,
+			ARRAY_SIZE(arizona_in_dmic_osr_text),
+			arizona_in_dmic_osr_text),
+	SOC_ENUM_SINGLE(ARIZONA_IN4L_CONTROL, ARIZONA_IN4_OSR_SHIFT,
+			ARRAY_SIZE(arizona_in_dmic_osr_text),
+			arizona_in_dmic_osr_text),
+};
+EXPORT_SYMBOL_GPL(arizona_in_dmic_osr);
+
 static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena)
 {
 	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index b60b08ccc1d0..9e81b6392692 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -150,7 +150,8 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
 	ARIZONA_MUX(name_str " Aux 5", &name##_aux5_mux), \
 	ARIZONA_MUX(name_str " Aux 6", &name##_aux6_mux)
 
-#define ARIZONA_MUX_ROUTES(name) \
+#define ARIZONA_MUX_ROUTES(widget, name) \
+	{ widget, NULL, name " Input" }, \
 	ARIZONA_MIXER_INPUT_ROUTES(name " Input")
 
 #define ARIZONA_MIXER_ROUTES(widget, name) \
@@ -198,6 +199,7 @@ extern const struct soc_enum arizona_lhpf3_mode;
 extern const struct soc_enum arizona_lhpf4_mode;
 
 extern const struct soc_enum arizona_ng_hold;
+extern const struct soc_enum arizona_in_dmic_osr[];
 
 extern int arizona_in_ev(struct snd_soc_dapm_widget *w,
 			 struct snd_kcontrol *kcontrol,
@@ -242,6 +244,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source,
 			   unsigned int Fref, unsigned int Fout);
 
 extern int arizona_init_spk(struct snd_soc_codec *codec);
+extern int arizona_init_gpio(struct snd_soc_codec *codec);
 
 extern int arizona_init_dai(struct arizona_priv *priv, int dai);
 
diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c
index a081d9fcb166..c4cf0699e77f 100644
--- a/sound/soc/codecs/bt-sco.c
+++ b/sound/soc/codecs/bt-sco.c
@@ -15,15 +15,27 @@
 
 #include <sound/soc.h>
 
+static const struct snd_soc_dapm_widget bt_sco_widgets[] = {
+	SND_SOC_DAPM_INPUT("RX"),
+	SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route bt_sco_routes[] = {
+	{ "Capture", NULL, "RX" },
+	{ "TX", NULL, "Playback" },
+};
+
 static struct snd_soc_dai_driver bt_sco_dai = {
 	.name = "bt-sco-pcm",
 	.playback = {
+		.stream_name = "Playback",
 		.channels_min = 1,
 		.channels_max = 1,
 		.rates = SNDRV_PCM_RATE_8000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.capture = {
+		 .stream_name = "Capture",
 		.channels_min = 1,
 		.channels_max = 1,
 		.rates = SNDRV_PCM_RATE_8000,
@@ -31,7 +43,12 @@ static struct snd_soc_dai_driver bt_sco_dai = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_bt_sco;
+static struct snd_soc_codec_driver soc_codec_dev_bt_sco = {
+	.dapm_widgets = bt_sco_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(bt_sco_widgets),
+	.dapm_routes = bt_sco_routes,
+	.num_dapm_routes = ARRAY_SIZE(bt_sco_routes),
+};
 
 static int bt_sco_probe(struct platform_device *pdev)
 {
@@ -50,6 +67,9 @@ static struct platform_device_id bt_sco_driver_ids[] = {
 	{
 		.name		= "dfbmcs320",
 	},
+	{
+		.name		= "bt-sco",
+	},
 	{},
 };
 MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids);
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 8e4779812b96..83c835d9fd88 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -139,6 +139,22 @@ struct cs4270_private {
 	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
 };
 
+static const struct snd_soc_dapm_widget cs4270_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("AINL"),
+SND_SOC_DAPM_INPUT("AINR"),
+
+SND_SOC_DAPM_OUTPUT("AOUTL"),
+SND_SOC_DAPM_OUTPUT("AOUTR"),
+};
+
+static const struct snd_soc_dapm_route cs4270_dapm_routes[] = {
+	{ "Capture", NULL, "AINA" },
+	{ "Capture", NULL, "AINB" },
+
+	{ "AOUTA", NULL, "Playback" },
+	{ "AOUTB", NULL, "Playback" },
+};
+
 /**
  * struct cs4270_mode_ratios - clock ratio tables
  * @ratio: the ratio of MCLK to the sample rate
@@ -612,6 +628,10 @@ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = {
 
 	.controls =		cs4270_snd_controls,
 	.num_controls =		ARRAY_SIZE(cs4270_snd_controls),
+	.dapm_widgets =		cs4270_dapm_widgets,
+	.num_dapm_widgets =	ARRAY_SIZE(cs4270_dapm_widgets),
+	.dapm_routes =		cs4270_dapm_routes,
+	.num_dapm_routes =	ARRAY_SIZE(cs4270_dapm_routes),
 };
 
 /*
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 03036b326732..a20f1bb8f071 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -173,6 +173,26 @@ struct cs4271_private {
 	bool				enable_soft_reset;
 };
 
+static const struct snd_soc_dapm_widget cs4271_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("AINA"),
+SND_SOC_DAPM_INPUT("AINB"),
+
+SND_SOC_DAPM_OUTPUT("AOUTA+"),
+SND_SOC_DAPM_OUTPUT("AOUTA-"),
+SND_SOC_DAPM_OUTPUT("AOUTB+"),
+SND_SOC_DAPM_OUTPUT("AOUTB-"),
+};
+
+static const struct snd_soc_dapm_route cs4271_dapm_routes[] = {
+	{ "Capture", NULL, "AINA" },
+	{ "Capture", NULL, "AINB" },
+
+	{ "AOUTA+", NULL, "Playback" },
+	{ "AOUTA-", NULL, "Playback" },
+	{ "AOUTB+", NULL, "Playback" },
+	{ "AOUTB-", NULL, "Playback" },
+};
+
 /*
  * @freq is the desired MCLK rate
  * MCLK rate should (c) be the sample rate, multiplied by one of the
@@ -576,8 +596,7 @@ static int cs4271_probe(struct snd_soc_codec *codec)
 				   CS4271_MODE2_MUTECAEQUB,
 				   CS4271_MODE2_MUTECAEQUB);
 
-	return snd_soc_add_codec_controls(codec, cs4271_snd_controls,
-		ARRAY_SIZE(cs4271_snd_controls));
+	return 0;
 }
 
 static int cs4271_remove(struct snd_soc_codec *codec)
@@ -596,6 +615,13 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = {
 	.remove			= cs4271_remove,
 	.suspend		= cs4271_soc_suspend,
 	.resume			= cs4271_soc_resume,
+
+	.controls		= cs4271_snd_controls,
+	.num_controls		= ARRAY_SIZE(cs4271_snd_controls),
+	.dapm_widgets		= cs4271_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(cs4271_dapm_widgets),
+	.dapm_routes		= cs4271_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(cs4271_dapm_routes),
 };
 
 #if defined(CONFIG_SPI_MASTER)
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
index 66967ba6f757..b2090b2a5e2d 100644
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@@ -50,20 +50,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"DMIC AIF", NULL, "DMic"},
 };
 
-static int dmic_probe(struct snd_soc_codec *codec)
-{
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
-				  ARRAY_SIZE(dmic_dapm_widgets));
-        snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-	snd_soc_dapm_new_widgets(dapm);
-
-	return 0;
-}
-
 static struct snd_soc_codec_driver soc_dmic = {
-	.probe	= dmic_probe,
+	.dapm_widgets = dmic_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets),
+	.dapm_routes = intercon,
+	.num_dapm_routes = ARRAY_SIZE(intercon),
 };
 
 static int dmic_dev_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c
index 2bcae2b40c92..68342b121c96 100644
--- a/sound/soc/codecs/hdmi.c
+++ b/sound/soc/codecs/hdmi.c
@@ -23,11 +23,20 @@
 
 #define DRV_NAME "hdmi-audio-codec"
 
-static struct snd_soc_codec_driver hdmi_codec;
+static const struct snd_soc_dapm_widget hdmi_widgets[] = {
+	SND_SOC_DAPM_INPUT("RX"),
+	SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route hdmi_routes[] = {
+	{ "Capture", NULL, "RX" },
+	{ "TX", NULL, "Playback" },
+};
 
 static struct snd_soc_dai_driver hdmi_codec_dai = {
 	.name = "hdmi-hifi",
 	.playback = {
+		.stream_name = "Playback",
 		.channels_min = 2,
 		.channels_max = 8,
 		.rates = SNDRV_PCM_RATE_32000 |
@@ -37,6 +46,25 @@ static struct snd_soc_dai_driver hdmi_codec_dai = {
 		.formats = SNDRV_PCM_FMTBIT_S16_LE |
 			SNDRV_PCM_FMTBIT_S24_LE,
 	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_32000 |
+			SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+			SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+			SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE |
+			SNDRV_PCM_FMTBIT_S24_LE,
+	},
+
+};
+
+static struct snd_soc_codec_driver hdmi_codec = {
+	.dapm_widgets = hdmi_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(hdmi_widgets),
+	.dapm_routes = hdmi_routes,
+	.num_dapm_routes = ARRAY_SIZE(hdmi_routes),
 };
 
 static int hdmi_codec_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
index 9f9f59573f72..0e5743ea79df 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/sound/soc/codecs/lm4857.c
@@ -16,6 +16,7 @@
 #include <linux/init.h>
 #include <linux/module.h>
 #include <linux/i2c.h>
+#include <linux/regmap.h>
 #include <linux/slab.h>
 
 #include <sound/core.h>
@@ -23,12 +24,15 @@
 #include <sound/tlv.h>
 
 struct lm4857 {
-	struct i2c_client *i2c;
+	struct regmap *regmap;
 	uint8_t mode;
 };
 
-static const uint8_t lm4857_default_regs[] = {
-	0x00, 0x00, 0x00, 0x00,
+static const struct reg_default lm4857_default_regs[] = {
+	{ 0x0, 0x00 },
+	{ 0x1, 0x00 },
+	{ 0x2, 0x00 },
+	{ 0x3, 0x00 },
 };
 
 /* The register offsets in the cache array */
@@ -42,39 +46,6 @@ static const uint8_t lm4857_default_regs[] = {
 #define LM4857_WAKEUP 5
 #define LM4857_EPGAIN 4
 
-static int lm4857_write(struct snd_soc_codec *codec, unsigned int reg,
-		unsigned int value)
-{
-	uint8_t data;
-	int ret;
-
-	ret = snd_soc_cache_write(codec, reg, value);
-	if (ret < 0)
-		return ret;
-
-	data = (reg << 6) | value;
-	ret = i2c_master_send(codec->control_data, &data, 1);
-	if (ret != 1) {
-		dev_err(codec->dev, "Failed to write register: %d\n", ret);
-		return ret;
-	}
-
-	return 0;
-}
-
-static unsigned int lm4857_read(struct snd_soc_codec *codec,
-		unsigned int reg)
-{
-	unsigned int val;
-	int ret;
-
-	ret = snd_soc_cache_read(codec, reg, &val);
-	if (ret)
-		return -1;
-
-	return val;
-}
-
 static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
@@ -96,7 +67,7 @@ static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
 	lm4857->mode = value;
 
 	if (codec->dapm.bias_level == SND_SOC_BIAS_ON)
-		snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, value + 6);
+		regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, value + 6);
 
 	return 1;
 }
@@ -108,10 +79,11 @@ static int lm4857_set_bias_level(struct snd_soc_codec *codec,
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, lm4857->mode + 6);
+		regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F,
+			lm4857->mode + 6);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, 0);
+		regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, 0);
 		break;
 	default:
 		break;
@@ -171,49 +143,32 @@ static const struct snd_soc_dapm_route lm4857_routes[] = {
 	{"EP", NULL, "IN"},
 };
 
-static int lm4857_probe(struct snd_soc_codec *codec)
-{
-	struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int ret;
-
-	codec->control_data = lm4857->i2c;
-
-	ret = snd_soc_add_codec_controls(codec, lm4857_controls,
-			ARRAY_SIZE(lm4857_controls));
-	if (ret)
-		return ret;
-
-	ret = snd_soc_dapm_new_controls(dapm, lm4857_dapm_widgets,
-			ARRAY_SIZE(lm4857_dapm_widgets));
-	if (ret)
-		return ret;
+static struct snd_soc_codec_driver soc_codec_dev_lm4857 = {
+	.set_bias_level = lm4857_set_bias_level,
 
-	ret = snd_soc_dapm_add_routes(dapm, lm4857_routes,
-			ARRAY_SIZE(lm4857_routes));
-	if (ret)
-		return ret;
+	.controls = lm4857_controls,
+	.num_controls = ARRAY_SIZE(lm4857_controls),
+	.dapm_widgets = lm4857_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(lm4857_dapm_widgets),
+	.dapm_routes = lm4857_routes,
+	.num_dapm_routes = ARRAY_SIZE(lm4857_routes),
+};
 
-	snd_soc_dapm_new_widgets(dapm);
+static const struct regmap_config lm4857_regmap_config = {
+	.val_bits = 6,
+	.reg_bits = 2,
 
-	return 0;
-}
+	.max_register = LM4857_CTRL,
 
-static struct snd_soc_codec_driver soc_codec_dev_lm4857 = {
-	.write = lm4857_write,
-	.read = lm4857_read,
-	.probe = lm4857_probe,
-	.reg_cache_size = ARRAY_SIZE(lm4857_default_regs),
-	.reg_word_size = sizeof(uint8_t),
-	.reg_cache_default = lm4857_default_regs,
-	.set_bias_level = lm4857_set_bias_level,
+	.cache_type = REGCACHE_FLAT,
+	.reg_defaults = lm4857_default_regs,
+	.num_reg_defaults = ARRAY_SIZE(lm4857_default_regs),
 };
 
 static int lm4857_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct lm4857 *lm4857;
-	int ret;
 
 	lm4857 = devm_kzalloc(&i2c->dev, sizeof(*lm4857), GFP_KERNEL);
 	if (!lm4857)
@@ -221,11 +176,11 @@ static int lm4857_i2c_probe(struct i2c_client *i2c,
 
 	i2c_set_clientdata(i2c, lm4857);
 
-	lm4857->i2c = i2c;
-
-	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0);
+	lm4857->regmap = devm_regmap_init_i2c(i2c, &lm4857_regmap_config);
+	if (IS_ERR(lm4857->regmap))
+		return PTR_ERR(lm4857->regmap);
 
-	return ret;
+	return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0);
 }
 
 static int lm4857_i2c_remove(struct i2c_client *i2c)
diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c
index a6ac2313047d..31f91560e9f6 100644
--- a/sound/soc/codecs/max9768.c
+++ b/sound/soc/codecs/max9768.c
@@ -118,6 +118,18 @@ static const struct snd_kcontrol_new max9768_mute[] = {
 	SOC_SINGLE_BOOL_EXT("Playback Switch", 0, max9768_get_gpio, max9768_set_gpio),
 };
 
+static const struct snd_soc_dapm_widget max9768_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("IN"),
+
+SND_SOC_DAPM_OUTPUT("OUT+"),
+SND_SOC_DAPM_OUTPUT("OUT-"),
+};
+
+static const struct snd_soc_dapm_route max9768_dapm_routes[] = {
+	{ "OUT+", NULL, "IN" },
+	{ "OUT-", NULL, "IN" },
+};
+
 static int max9768_probe(struct snd_soc_codec *codec)
 {
 	struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec);
@@ -148,6 +160,10 @@ static struct snd_soc_codec_driver max9768_codec_driver = {
 	.probe = max9768_probe,
 	.controls = max9768_volume,
 	.num_controls = ARRAY_SIZE(max9768_volume),
+	.dapm_widgets = max9768_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(max9768_dapm_widgets),
+	.dapm_routes = max9768_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(max9768_dapm_routes),
 };
 
 static const struct regmap_config max9768_i2c_regmap_config = {
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index ad5313f98f28..0569a4c3ae00 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2084,8 +2084,9 @@ static irqreturn_t max98090_interrupt(int irq, void *data)
 
 		pm_wakeup_event(codec->dev, 100);
 
-		schedule_delayed_work(&max98090->jack_work,
-			msecs_to_jiffies(100));
+		queue_delayed_work(system_power_efficient_wq,
+				   &max98090->jack_work,
+				   msecs_to_jiffies(100));
 	}
 
 	if (active & M98090_DRCACT_MASK)
@@ -2132,8 +2133,9 @@ int max98090_mic_detect(struct snd_soc_codec *codec,
 	snd_soc_jack_report(max98090->jack, 0,
 			    SND_JACK_HEADSET | SND_JACK_BTN_0);
 
-	schedule_delayed_work(&max98090->jack_work,
-		msecs_to_jiffies(100));
+	queue_delayed_work(system_power_efficient_wq,
+			   &max98090->jack_work,
+			   msecs_to_jiffies(100));
 
 	return 0;
 }
diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c
index 6b6c74cd83e2..29549cdbf4c1 100644
--- a/sound/soc/codecs/max9877.c
+++ b/sound/soc/codecs/max9877.c
@@ -14,170 +14,21 @@
 #include <linux/module.h>
 #include <linux/init.h>
 #include <linux/i2c.h>
+#include <linux/regmap.h>
 #include <sound/soc.h>
 #include <sound/tlv.h>
 
 #include "max9877.h"
 
-static struct i2c_client *i2c;
+static struct regmap *regmap;
 
-static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 };
-
-static void max9877_write_regs(void)
-{
-	unsigned int i;
-	u8 data[6];
-
-	data[0] = MAX9877_INPUT_MODE;
-	for (i = 0; i < ARRAY_SIZE(max9877_regs); i++)
-		data[i + 1] = max9877_regs[i];
-
-	if (i2c_master_send(i2c, data, 6) != 6)
-		dev_err(&i2c->dev, "i2c write failed\n");
-}
-
-static int max9877_get_reg(struct snd_kcontrol *kcontrol,
-		struct snd_ctl_elem_value *ucontrol)
-{
-	struct soc_mixer_control *mc =
-		(struct soc_mixer_control *)kcontrol->private_value;
-	unsigned int reg = mc->reg;
-	unsigned int shift = mc->shift;
-	unsigned int mask = mc->max;
-	unsigned int invert = mc->invert;
-
-	ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask;
-
-	if (invert)
-		ucontrol->value.integer.value[0] =
-			mask - ucontrol->value.integer.value[0];
-
-	return 0;
-}
-
-static int max9877_set_reg(struct snd_kcontrol *kcontrol,
-		struct snd_ctl_elem_value *ucontrol)
-{
-	struct soc_mixer_control *mc =
-		(struct soc_mixer_control *)kcontrol->private_value;
-	unsigned int reg = mc->reg;
-	unsigned int shift = mc->shift;
-	unsigned int mask = mc->max;
-	unsigned int invert = mc->invert;
-	unsigned int val = (ucontrol->value.integer.value[0] & mask);
-
-	if (invert)
-		val = mask - val;
-
-	if (((max9877_regs[reg] >> shift) & mask) == val)
-		return 0;
-
-	max9877_regs[reg] &= ~(mask << shift);
-	max9877_regs[reg] |= val << shift;
-	max9877_write_regs();
-
-	return 1;
-}
-
-static int max9877_get_2reg(struct snd_kcontrol *kcontrol,
-		struct snd_ctl_elem_value *ucontrol)
-{
-	struct soc_mixer_control *mc =
-		(struct soc_mixer_control *)kcontrol->private_value;
-	unsigned int reg = mc->reg;
-	unsigned int reg2 = mc->rreg;
-	unsigned int shift = mc->shift;
-	unsigned int mask = mc->max;
-
-	ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask;
-	ucontrol->value.integer.value[1] = (max9877_regs[reg2] >> shift) & mask;
-
-	return 0;
-}
-
-static int max9877_set_2reg(struct snd_kcontrol *kcontrol,
-		struct snd_ctl_elem_value *ucontrol)
-{
-	struct soc_mixer_control *mc =
-		(struct soc_mixer_control *)kcontrol->private_value;
-	unsigned int reg = mc->reg;
-	unsigned int reg2 = mc->rreg;
-	unsigned int shift = mc->shift;
-	unsigned int mask = mc->max;
-	unsigned int val = (ucontrol->value.integer.value[0] & mask);
-	unsigned int val2 = (ucontrol->value.integer.value[1] & mask);
-	unsigned int change = 0;
-
-	if (((max9877_regs[reg] >> shift) & mask) != val)
-		change = 1;
-
-	if (((max9877_regs[reg2] >> shift) & mask) != val2)
-		change = 1;
-
-	if (change) {
-		max9877_regs[reg] &= ~(mask << shift);
-		max9877_regs[reg] |= val << shift;
-		max9877_regs[reg2] &= ~(mask << shift);
-		max9877_regs[reg2] |= val2 << shift;
-		max9877_write_regs();
-	}
-
-	return change;
-}
-
-static int max9877_get_out_mode(struct snd_kcontrol *kcontrol,
-		struct snd_ctl_elem_value *ucontrol)
-{
-	u8 value = max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK;
-
-	if (value)
-		value -= 1;
-
-	ucontrol->value.integer.value[0] = value;
-	return 0;
-}
-
-static int max9877_set_out_mode(struct snd_kcontrol *kcontrol,
-		struct snd_ctl_elem_value *ucontrol)
-{
-	u8 value = ucontrol->value.integer.value[0];
-
-	value += 1;
-
-	if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value)
-		return 0;
-
-	max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OUTMODE_MASK;
-	max9877_regs[MAX9877_OUTPUT_MODE] |= value;
-	max9877_write_regs();
-	return 1;
-}
-
-static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol,
-		struct snd_ctl_elem_value *ucontrol)
-{
-	u8 value = (max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK);
-
-	value = value >> MAX9877_OSC_OFFSET;
-
-	ucontrol->value.integer.value[0] = value;
-	return 0;
-}
-
-static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol,
-		struct snd_ctl_elem_value *ucontrol)
-{
-	u8 value = ucontrol->value.integer.value[0];
-
-	value = value << MAX9877_OSC_OFFSET;
-	if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value)
-		return 0;
-
-	max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OSC_MASK;
-	max9877_regs[MAX9877_OUTPUT_MODE] |= value;
-	max9877_write_regs();
-	return 1;
-}
+static struct reg_default max9877_regs[] = {
+	{ 0, 0x40 },
+	{ 1, 0x00 },
+	{ 2, 0x00 },
+	{ 3, 0x00 },
+	{ 4, 0x49 },
+};
 
 static const unsigned int max9877_pgain_tlv[] = {
 	TLV_DB_RANGE_HEAD(2),
@@ -212,65 +63,104 @@ static const char *max9877_osc_mode[] = {
 };
 
 static const struct soc_enum max9877_enum[] = {
-	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_out_mode), max9877_out_mode),
-	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode),
+	SOC_ENUM_SINGLE(MAX9877_OUTPUT_MODE, 0, ARRAY_SIZE(max9877_out_mode),
+			max9877_out_mode),
+	SOC_ENUM_SINGLE(MAX9877_OUTPUT_MODE, MAX9877_OSC_OFFSET,
+			ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode),
 };
 
 static const struct snd_kcontrol_new max9877_controls[] = {
-	SOC_SINGLE_EXT_TLV("MAX9877 PGAINA Playback Volume",
-			MAX9877_INPUT_MODE, 0, 2, 0,
-			max9877_get_reg, max9877_set_reg, max9877_pgain_tlv),
-	SOC_SINGLE_EXT_TLV("MAX9877 PGAINB Playback Volume",
-			MAX9877_INPUT_MODE, 2, 2, 0,
-			max9877_get_reg, max9877_set_reg, max9877_pgain_tlv),
-	SOC_SINGLE_EXT_TLV("MAX9877 Amp Speaker Playback Volume",
-			MAX9877_SPK_VOLUME, 0, 31, 0,
-			max9877_get_reg, max9877_set_reg, max9877_output_tlv),
-	SOC_DOUBLE_R_EXT_TLV("MAX9877 Amp HP Playback Volume",
-			MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0,
-			max9877_get_2reg, max9877_set_2reg, max9877_output_tlv),
-	SOC_SINGLE_EXT("MAX9877 INB Stereo Switch",
-			MAX9877_INPUT_MODE, 4, 1, 1,
-			max9877_get_reg, max9877_set_reg),
-	SOC_SINGLE_EXT("MAX9877 INA Stereo Switch",
-			MAX9877_INPUT_MODE, 5, 1, 1,
-			max9877_get_reg, max9877_set_reg),
-	SOC_SINGLE_EXT("MAX9877 Zero-crossing detection Switch",
-			MAX9877_INPUT_MODE, 6, 1, 0,
-			max9877_get_reg, max9877_set_reg),
-	SOC_SINGLE_EXT("MAX9877 Bypass Mode Switch",
-			MAX9877_OUTPUT_MODE, 6, 1, 0,
-			max9877_get_reg, max9877_set_reg),
-	SOC_SINGLE_EXT("MAX9877 Shutdown Mode Switch",
-			MAX9877_OUTPUT_MODE, 7, 1, 1,
-			max9877_get_reg, max9877_set_reg),
-	SOC_ENUM_EXT("MAX9877 Output Mode", max9877_enum[0],
-			max9877_get_out_mode, max9877_set_out_mode),
-	SOC_ENUM_EXT("MAX9877 Oscillator Mode", max9877_enum[1],
-			max9877_get_osc_mode, max9877_set_osc_mode),
+	SOC_SINGLE_TLV("MAX9877 PGAINA Playback Volume",
+		       MAX9877_INPUT_MODE, 0, 2, 0, max9877_pgain_tlv),
+	SOC_SINGLE_TLV("MAX9877 PGAINB Playback Volume",
+		       MAX9877_INPUT_MODE, 2, 2, 0, max9877_pgain_tlv),
+	SOC_SINGLE_TLV("MAX9877 Amp Speaker Playback Volume",
+		       MAX9877_SPK_VOLUME, 0, 31, 0, max9877_output_tlv),
+	SOC_DOUBLE_R_TLV("MAX9877 Amp HP Playback Volume",
+			 MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0,
+			 max9877_output_tlv),
+	SOC_SINGLE("MAX9877 INB Stereo Switch",
+		   MAX9877_INPUT_MODE, 4, 1, 1),
+	SOC_SINGLE("MAX9877 INA Stereo Switch",
+		   MAX9877_INPUT_MODE, 5, 1, 1),
+	SOC_SINGLE("MAX9877 Zero-crossing detection Switch",
+		   MAX9877_INPUT_MODE, 6, 1, 0),
+	SOC_SINGLE("MAX9877 Bypass Mode Switch",
+		   MAX9877_OUTPUT_MODE, 6, 1, 0),
+	SOC_ENUM("MAX9877 Output Mode", max9877_enum[0]),
+	SOC_ENUM("MAX9877 Oscillator Mode", max9877_enum[1]),
 };
 
-/* This function is called from ASoC machine driver */
-int max9877_add_controls(struct snd_soc_codec *codec)
-{
-	return snd_soc_add_codec_controls(codec, max9877_controls,
-			ARRAY_SIZE(max9877_controls));
-}
-EXPORT_SYMBOL_GPL(max9877_add_controls);
+static const struct snd_soc_dapm_widget max9877_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("INA1"),
+SND_SOC_DAPM_INPUT("INA2"),
+SND_SOC_DAPM_INPUT("INB1"),
+SND_SOC_DAPM_INPUT("INB2"),
+SND_SOC_DAPM_INPUT("RXIN+"),
+SND_SOC_DAPM_INPUT("RXIN-"),
+
+SND_SOC_DAPM_PGA("SHDN", MAX9877_OUTPUT_MODE, 7, 1, NULL, 0),
+
+SND_SOC_DAPM_OUTPUT("OUT+"),
+SND_SOC_DAPM_OUTPUT("OUT-"),
+SND_SOC_DAPM_OUTPUT("HPL"),
+SND_SOC_DAPM_OUTPUT("HPR"),
+};
+
+static const struct snd_soc_dapm_route max9877_dapm_routes[] = {
+	{ "SHDN", NULL, "INA1" },
+	{ "SHDN", NULL, "INA2" },
+	{ "SHDN", NULL, "INB1" },
+	{ "SHDN", NULL, "INB2" },
+
+	{ "OUT+", NULL, "RXIN+" },
+	{ "OUT+", NULL, "SHDN" },
+
+	{ "OUT-", NULL, "SHDN" },
+	{ "OUT-", NULL, "RXIN-" },
+
+	{ "HPL", NULL, "SHDN" },
+	{ "HPR", NULL, "SHDN" },
+};
+
+static const struct snd_soc_codec_driver max9877_codec = {
+	.controls = max9877_controls,
+	.num_controls = ARRAY_SIZE(max9877_controls),
+
+	.dapm_widgets = max9877_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(max9877_dapm_widgets),
+	.dapm_routes = max9877_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(max9877_dapm_routes),
+};
+
+static const struct regmap_config max9877_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.reg_defaults = max9877_regs,
+	.num_reg_defaults = ARRAY_SIZE(max9877_regs),
+	.cache_type = REGCACHE_RBTREE,
+};
 
 static int max9877_i2c_probe(struct i2c_client *client,
 			     const struct i2c_device_id *id)
 {
-	i2c = client;
+	int i;
 
-	max9877_write_regs();
+	regmap = devm_regmap_init_i2c(client, &max9877_regmap);
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
 
-	return 0;
+	/* Ensure the device is in reset state */
+	for (i = 0; i < ARRAY_SIZE(max9877_regs); i++)
+		regmap_write(regmap, max9877_regs[i].reg, max9877_regs[i].def);
+
+	return snd_soc_register_codec(&client->dev, &max9877_codec, NULL, 0);
 }
 
 static int max9877_i2c_remove(struct i2c_client *client)
 {
-	i2c = NULL;
+	snd_soc_unregister_codec(&client->dev);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index 5402dfbbb716..4d3c8fd8c5db 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -94,7 +94,6 @@
 #define AUDIO_DAC_CFS_DLY_B		(1 << 10)
 
 struct mc13783_priv {
-	struct snd_soc_codec codec;
 	struct mc13xxx *mc13xxx;
 
 	enum mc13783_ssi_port adc_ssi_port;
diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
new file mode 100644
index 000000000000..651ce0923675
--- /dev/null
+++ b/sound/soc/codecs/pcm1681.c
@@ -0,0 +1,339 @@
+/*
+ * PCM1681 ASoC codec driver
+ *
+ * Copyright (c) StreamUnlimited GmbH 2013
+ *	Marek Belisko <marek.belisko@streamunlimited.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/of_device.h>
+#include <linux/of_gpio.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#define PCM1681_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE  |		\
+			     SNDRV_PCM_FMTBIT_S24_LE)
+
+#define PCM1681_PCM_RATES   (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \
+			     SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100  | \
+			     SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200  | \
+			     SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000)
+
+#define PCM1681_SOFT_MUTE_ALL		0xff
+#define PCM1681_DEEMPH_RATE_MASK	0x18
+#define PCM1681_DEEMPH_MASK		0x01
+
+#define PCM1681_ATT_CONTROL(X)	(X <= 6 ? X : X + 9) /* Attenuation level */
+#define PCM1681_SOFT_MUTE	0x07	/* Soft mute control register */
+#define PCM1681_DAC_CONTROL	0x08	/* DAC operation control */
+#define PCM1681_FMT_CONTROL	0x09	/* Audio interface data format */
+#define PCM1681_DEEMPH_CONTROL	0x0a	/* De-emphasis control */
+#define PCM1681_ZERO_DETECT_STATUS	0x0e	/* Zero detect status reg */
+
+static const struct reg_default pcm1681_reg_defaults[] = {
+	{ 0x01,	0xff },
+	{ 0x02,	0xff },
+	{ 0x03,	0xff },
+	{ 0x04,	0xff },
+	{ 0x05,	0xff },
+	{ 0x06,	0xff },
+	{ 0x07,	0x00 },
+	{ 0x08,	0x00 },
+	{ 0x09,	0x06 },
+	{ 0x0A,	0x00 },
+	{ 0x0B,	0xff },
+	{ 0x0C,	0x0f },
+	{ 0x0D,	0x00 },
+	{ 0x10,	0xff },
+	{ 0x11,	0xff },
+	{ 0x12,	0x00 },
+	{ 0x13,	0x00 },
+};
+
+static bool pcm1681_accessible_reg(struct device *dev, unsigned int reg)
+{
+	return !((reg == 0x00) || (reg == 0x0f));
+}
+
+static bool pcm1681_writeable_reg(struct device *dev, unsigned register reg)
+{
+	return pcm1681_accessible_reg(dev, reg) &&
+		(reg != PCM1681_ZERO_DETECT_STATUS);
+}
+
+struct pcm1681_private {
+	struct regmap *regmap;
+	unsigned int format;
+	/* Current deemphasis status */
+	unsigned int deemph;
+	/* Current rate for deemphasis control */
+	unsigned int rate;
+};
+
+static const int pcm1681_deemph[] = { 44100, 48000, 32000 };
+
+static int pcm1681_set_deemph(struct snd_soc_codec *codec)
+{
+	struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
+	int i = 0, val = -1, enable = 0;
+
+	if (priv->deemph)
+		for (i = 0; i < ARRAY_SIZE(pcm1681_deemph); i++)
+			if (pcm1681_deemph[i] == priv->rate)
+				val = i;
+
+	if (val != -1) {
+		regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL,
+					PCM1681_DEEMPH_RATE_MASK, val);
+		enable = 1;
+	} else
+		enable = 0;
+
+	/* enable/disable deemphasis functionality */
+	return regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL,
+					PCM1681_DEEMPH_MASK, enable);
+}
+
+static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.enumerated.item[0] = priv->deemph;
+
+	return 0;
+}
+
+static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
+
+	priv->deemph = ucontrol->value.enumerated.item[0];
+
+	return pcm1681_set_deemph(codec);
+}
+
+static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			      unsigned int format)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
+
+	/* The PCM1681 can only be slave to all clocks */
+	if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
+		dev_err(codec->dev, "Invalid clocking mode\n");
+		return -EINVAL;
+	}
+
+	priv->format = format;
+
+	return 0;
+}
+
+static int pcm1681_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
+	int val;
+
+	if (mute)
+		val = PCM1681_SOFT_MUTE_ALL;
+	else
+		val = 0;
+
+	return regmap_write(priv->regmap, PCM1681_SOFT_MUTE, val);
+}
+
+static int pcm1681_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 pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
+	int val = 0, ret;
+	int pcm_format = params_format(params);
+
+	priv->rate = params_rate(params);
+
+	switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_RIGHT_J:
+		if (pcm_format == SNDRV_PCM_FORMAT_S24_LE)
+			val = 0x00;
+		else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE)
+			val = 0x03;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		val = 0x04;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		val = 0x05;
+		break;
+	default:
+		dev_err(codec->dev, "Invalid DAI format\n");
+		return -EINVAL;
+	}
+
+	ret = regmap_update_bits(priv->regmap, PCM1681_FMT_CONTROL, 0x0f, val);
+	if (ret < 0)
+		return ret;
+
+	return pcm1681_set_deemph(codec);
+}
+
+static const struct snd_soc_dai_ops pcm1681_dai_ops = {
+	.set_fmt	= pcm1681_set_dai_fmt,
+	.hw_params	= pcm1681_hw_params,
+	.digital_mute	= pcm1681_digital_mute,
+};
+
+static const struct snd_soc_dapm_widget pcm1681_dapm_widgets[] = {
+SND_SOC_DAPM_OUTPUT("VOUT1"),
+SND_SOC_DAPM_OUTPUT("VOUT2"),
+SND_SOC_DAPM_OUTPUT("VOUT3"),
+SND_SOC_DAPM_OUTPUT("VOUT4"),
+SND_SOC_DAPM_OUTPUT("VOUT5"),
+SND_SOC_DAPM_OUTPUT("VOUT6"),
+SND_SOC_DAPM_OUTPUT("VOUT7"),
+SND_SOC_DAPM_OUTPUT("VOUT8"),
+};
+
+static const struct snd_soc_dapm_route pcm1681_dapm_routes[] = {
+	{ "VOUT1", NULL, "Playback" },
+	{ "VOUT2", NULL, "Playback" },
+	{ "VOUT3", NULL, "Playback" },
+	{ "VOUT4", NULL, "Playback" },
+	{ "VOUT5", NULL, "Playback" },
+	{ "VOUT6", NULL, "Playback" },
+	{ "VOUT7", NULL, "Playback" },
+	{ "VOUT8", NULL, "Playback" },
+};
+
+static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1);
+
+static const struct snd_kcontrol_new pcm1681_controls[] = {
+	SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume",
+			PCM1681_ATT_CONTROL(1), PCM1681_ATT_CONTROL(2), 0,
+			0x7f, 0, pcm1681_dac_tlv),
+	SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume",
+			PCM1681_ATT_CONTROL(3), PCM1681_ATT_CONTROL(4), 0,
+			0x7f, 0, pcm1681_dac_tlv),
+	SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume",
+			PCM1681_ATT_CONTROL(5), PCM1681_ATT_CONTROL(6), 0,
+			0x7f, 0, pcm1681_dac_tlv),
+	SOC_DOUBLE_R_TLV("Channel 7/8 Playback Volume",
+			PCM1681_ATT_CONTROL(7), PCM1681_ATT_CONTROL(8), 0,
+			0x7f, 0, pcm1681_dac_tlv),
+	SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0,
+			    pcm1681_get_deemph, pcm1681_put_deemph),
+};
+
+static struct snd_soc_dai_driver pcm1681_dai = {
+	.name = "pcm1681-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 8,
+		.rates = PCM1681_PCM_RATES,
+		.formats = PCM1681_PCM_FORMATS,
+	},
+	.ops = &pcm1681_dai_ops,
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id pcm1681_dt_ids[] = {
+	{ .compatible = "ti,pcm1681", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, pcm1681_dt_ids);
+#endif
+
+static const struct regmap_config pcm1681_regmap = {
+	.reg_bits		= 8,
+	.val_bits		= 8,
+	.max_register		= ARRAY_SIZE(pcm1681_reg_defaults) + 1,
+	.reg_defaults		= pcm1681_reg_defaults,
+	.num_reg_defaults	= ARRAY_SIZE(pcm1681_reg_defaults),
+	.writeable_reg		= pcm1681_writeable_reg,
+	.readable_reg		= pcm1681_accessible_reg,
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = {
+	.controls		= pcm1681_controls,
+	.num_controls		= ARRAY_SIZE(pcm1681_controls),
+	.dapm_widgets		= pcm1681_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(pcm1681_dapm_widgets),
+	.dapm_routes		= pcm1681_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(pcm1681_dapm_routes),
+};
+
+static const struct i2c_device_id pcm1681_i2c_id[] = {
+	{"pcm1681", 0},
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id);
+
+static int pcm1681_i2c_probe(struct i2c_client *client,
+			      const struct i2c_device_id *id)
+{
+	int ret;
+	struct pcm1681_private *priv;
+
+	priv = devm_kzalloc(&client->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->regmap = devm_regmap_init_i2c(client, &pcm1681_regmap);
+	if (IS_ERR(priv->regmap)) {
+		ret = PTR_ERR(priv->regmap);
+		dev_err(&client->dev, "Failed to create regmap: %d\n", ret);
+		return ret;
+	}
+
+	i2c_set_clientdata(client, priv);
+
+	return snd_soc_register_codec(&client->dev, &soc_codec_dev_pcm1681,
+		&pcm1681_dai, 1);
+}
+
+static int pcm1681_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	return 0;
+}
+
+static struct i2c_driver pcm1681_i2c_driver = {
+	.driver = {
+		.name	= "pcm1681",
+		.owner	= THIS_MODULE,
+		.of_match_table = of_match_ptr(pcm1681_dt_ids),
+	},
+	.id_table	= pcm1681_i2c_id,
+	.probe		= pcm1681_i2c_probe,
+	.remove		= pcm1681_i2c_remove,
+};
+
+module_i2c_driver(pcm1681_i2c_driver);
+
+MODULE_DESCRIPTION("Texas Instruments PCM1681 ALSA SoC Codec Driver");
+MODULE_AUTHOR("Marek Belisko <marek.belisko@streamunlimited.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c
new file mode 100644
index 000000000000..2a8eccf64c76
--- /dev/null
+++ b/sound/soc/codecs/pcm1792a.c
@@ -0,0 +1,257 @@
+/*
+ * PCM1792A ASoC codec driver
+ *
+ * Copyright (c) Amarula Solutions B.V. 2013
+ *
+ *     Michael Trimarchi <michael@amarulasolutions.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <linux/spi/spi.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <linux/of_device.h>
+
+#include "pcm1792a.h"
+
+#define PCM1792A_DAC_VOL_LEFT	0x10
+#define PCM1792A_DAC_VOL_RIGHT	0x11
+#define PCM1792A_FMT_CONTROL	0x12
+#define PCM1792A_SOFT_MUTE	PCM1792A_FMT_CONTROL
+
+#define PCM1792A_FMT_MASK	0x70
+#define PCM1792A_FMT_SHIFT	4
+#define PCM1792A_MUTE_MASK	0x01
+#define PCM1792A_MUTE_SHIFT	0
+#define PCM1792A_ATLD_ENABLE	(1 << 7)
+
+static const struct reg_default pcm1792a_reg_defaults[] = {
+	{ 0x10, 0xff },
+	{ 0x11, 0xff },
+	{ 0x12, 0x50 },
+	{ 0x13, 0x00 },
+	{ 0x14, 0x00 },
+	{ 0x15, 0x01 },
+	{ 0x16, 0x00 },
+	{ 0x17, 0x00 },
+};
+
+static bool pcm1792a_accessible_reg(struct device *dev, unsigned int reg)
+{
+	return reg >= 0x10 && reg <= 0x17;
+}
+
+static bool pcm1792a_writeable_reg(struct device *dev, unsigned register reg)
+{
+	bool accessible;
+
+	accessible = pcm1792a_accessible_reg(dev, reg);
+
+	return accessible && reg != 0x16 && reg != 0x17;
+}
+
+struct pcm1792a_private {
+	struct regmap *regmap;
+	unsigned int format;
+	unsigned int rate;
+};
+
+static int pcm1792a_set_dai_fmt(struct snd_soc_dai *codec_dai,
+                             unsigned int format)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec);
+
+	priv->format = format;
+
+	return 0;
+}
+
+static int pcm1792a_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	ret = regmap_update_bits(priv->regmap, PCM1792A_SOFT_MUTE,
+				 PCM1792A_MUTE_MASK, !!mute);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static int pcm1792a_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 pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec);
+	int val = 0, ret;
+	int pcm_format = params_format(params);
+
+	priv->rate = params_rate(params);
+
+	switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_RIGHT_J:
+		if (pcm_format == SNDRV_PCM_FORMAT_S24_LE ||
+		    pcm_format == SNDRV_PCM_FORMAT_S32_LE)
+			val = 0x02;
+		else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE)
+			val = 0x00;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		if (pcm_format == SNDRV_PCM_FORMAT_S24_LE ||
+		    pcm_format == SNDRV_PCM_FORMAT_S32_LE)
+			val = 0x05;
+		else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE)
+			val = 0x04;
+		break;
+	default:
+		dev_err(codec->dev, "Invalid DAI format\n");
+		return -EINVAL;
+	}
+
+	val = val << PCM1792A_FMT_SHIFT | PCM1792A_ATLD_ENABLE;
+
+	ret = regmap_update_bits(priv->regmap, PCM1792A_FMT_CONTROL,
+				 PCM1792A_FMT_MASK | PCM1792A_ATLD_ENABLE, val);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops pcm1792a_dai_ops = {
+	.set_fmt	= pcm1792a_set_dai_fmt,
+	.hw_params	= pcm1792a_hw_params,
+	.digital_mute	= pcm1792a_digital_mute,
+};
+
+static const DECLARE_TLV_DB_SCALE(pcm1792a_dac_tlv, -12000, 50, 1);
+
+static const struct snd_kcontrol_new pcm1792a_controls[] = {
+	SOC_DOUBLE_R_RANGE_TLV("DAC Playback Volume", PCM1792A_DAC_VOL_LEFT,
+			 PCM1792A_DAC_VOL_RIGHT, 0, 0xf, 0xff, 0,
+			 pcm1792a_dac_tlv),
+};
+
+static const struct snd_soc_dapm_widget pcm1792a_dapm_widgets[] = {
+SND_SOC_DAPM_OUTPUT("IOUTL+"),
+SND_SOC_DAPM_OUTPUT("IOUTL-"),
+SND_SOC_DAPM_OUTPUT("IOUTR+"),
+SND_SOC_DAPM_OUTPUT("IOUTR-"),
+};
+
+static const struct snd_soc_dapm_route pcm1792a_dapm_routes[] = {
+	{ "IOUTL+", NULL, "Playback" },
+	{ "IOUTL-", NULL, "Playback" },
+	{ "IOUTR+", NULL, "Playback" },
+	{ "IOUTR-", NULL, "Playback" },
+};
+
+static struct snd_soc_dai_driver pcm1792a_dai = {
+	.name = "pcm1792a-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = PCM1792A_RATES,
+		.formats = PCM1792A_FORMATS, },
+	.ops = &pcm1792a_dai_ops,
+};
+
+static const struct of_device_id pcm1792a_of_match[] = {
+	{ .compatible = "ti,pcm1792a", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, pcm1792a_of_match);
+
+static const struct regmap_config pcm1792a_regmap = {
+	.reg_bits		= 8,
+	.val_bits		= 8,
+	.max_register		= 24,
+	.reg_defaults		= pcm1792a_reg_defaults,
+	.num_reg_defaults	= ARRAY_SIZE(pcm1792a_reg_defaults),
+	.writeable_reg		= pcm1792a_writeable_reg,
+	.readable_reg		= pcm1792a_accessible_reg,
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_pcm1792a = {
+	.controls		= pcm1792a_controls,
+	.num_controls		= ARRAY_SIZE(pcm1792a_controls),
+	.dapm_widgets		= pcm1792a_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(pcm1792a_dapm_widgets),
+	.dapm_routes		= pcm1792a_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(pcm1792a_dapm_routes),
+};
+
+static int pcm1792a_spi_probe(struct spi_device *spi)
+{
+	struct pcm1792a_private *pcm1792a;
+	int ret;
+
+	pcm1792a = devm_kzalloc(&spi->dev, sizeof(struct pcm1792a_private),
+				GFP_KERNEL);
+	if (!pcm1792a)
+		return -ENOMEM;
+
+	spi_set_drvdata(spi, pcm1792a);
+
+	pcm1792a->regmap = devm_regmap_init_spi(spi, &pcm1792a_regmap);
+	if (IS_ERR(pcm1792a->regmap)) {
+		ret = PTR_ERR(pcm1792a->regmap);
+		dev_err(&spi->dev, "Failed to register regmap: %d\n", ret);
+		return ret;
+	}
+
+	return snd_soc_register_codec(&spi->dev,
+			&soc_codec_dev_pcm1792a, &pcm1792a_dai, 1);
+}
+
+static int pcm1792a_spi_remove(struct spi_device *spi)
+{
+	snd_soc_unregister_codec(&spi->dev);
+	return 0;
+}
+
+static const struct spi_device_id pcm1792a_spi_ids[] = {
+	{ "pcm1792a", 0 },
+	{ },
+};
+MODULE_DEVICE_TABLE(spi, pcm1792a_spi_ids);
+
+static struct spi_driver pcm1792a_codec_driver = {
+	.driver = {
+		.name = "pcm1792a",
+		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(pcm1792a_of_match),
+	},
+	.id_table = pcm1792a_spi_ids,
+	.probe = pcm1792a_spi_probe,
+	.remove = pcm1792a_spi_remove,
+};
+
+module_spi_driver(pcm1792a_codec_driver);
+
+MODULE_DESCRIPTION("ASoC PCM1792A driver");
+MODULE_AUTHOR("Michael Trimarchi <michael@amarulasolutions.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/pcm1792a.h b/sound/soc/codecs/pcm1792a.h
new file mode 100644
index 000000000000..7a83d1fc102a
--- /dev/null
+++ b/sound/soc/codecs/pcm1792a.h
@@ -0,0 +1,26 @@
+/*
+ * definitions for PCM1792A
+ *
+ * Copyright 2013 Amarula Solutions
+ *
+  * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ */
+
+#ifndef __PCM1792A_H__
+#define __PCM1792A_H__
+
+#define PCM1792A_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_8000_48000 | \
+			SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000)
+
+#define PCM1792A_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \
+			  SNDRV_PCM_FMTBIT_S16_LE)
+
+#endif
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
index f2a6282b41f4..b6618c4a7597 100644
--- a/sound/soc/codecs/pcm3008.c
+++ b/sound/soc/codecs/pcm3008.c
@@ -28,7 +28,54 @@
 
 #include "pcm3008.h"
 
-#define PCM3008_VERSION "0.2"
+static int pcm3008_dac_ev(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *kcontrol,
+			  int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct pcm3008_setup_data *setup = codec->dev->platform_data;
+
+	gpio_set_value_cansleep(setup->pdda_pin,
+				SND_SOC_DAPM_EVENT_ON(event));
+
+	return 0;
+}
+
+static int pcm3008_adc_ev(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *kcontrol,
+			  int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct pcm3008_setup_data *setup = codec->dev->platform_data;
+
+	gpio_set_value_cansleep(setup->pdad_pin,
+				SND_SOC_DAPM_EVENT_ON(event));
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget pcm3008_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("VINL"),
+SND_SOC_DAPM_INPUT("VINR"),
+
+SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, pcm3008_dac_ev,
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_ADC_E("ADC", NULL, SND_SOC_NOPM, 0, 0, pcm3008_adc_ev,
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+SND_SOC_DAPM_OUTPUT("VOUTL"),
+SND_SOC_DAPM_OUTPUT("VOUTR"),
+};
+
+static const struct snd_soc_dapm_route pcm3008_dapm_routes[] = {
+	{ "PCM3008 Capture", NULL, "ADC" },
+	{ "ADC", NULL, "VINL" },
+	{ "ADC", NULL, "VINR" },
+
+	{ "DAC", NULL, "PCM3008 Playback" },
+	{ "VOUTL", NULL, "DAC" },
+	{ "VOUTR", NULL, "DAC" },
+};
 
 #define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |	\
 		       SNDRV_PCM_RATE_48000)
@@ -51,20 +98,20 @@ static struct snd_soc_dai_driver pcm3008_dai = {
 	},
 };
 
-static void pcm3008_gpio_free(struct pcm3008_setup_data *setup)
-{
-	gpio_free(setup->dem0_pin);
-	gpio_free(setup->dem1_pin);
-	gpio_free(setup->pdad_pin);
-	gpio_free(setup->pdda_pin);
-}
+static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = {
+	.dapm_widgets = pcm3008_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(pcm3008_dapm_widgets),
+	.dapm_routes = pcm3008_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(pcm3008_dapm_routes),
+};
 
-static int pcm3008_soc_probe(struct snd_soc_codec *codec)
+static int pcm3008_codec_probe(struct platform_device *pdev)
 {
-	struct pcm3008_setup_data *setup = codec->dev->platform_data;
-	int ret = 0;
+	struct pcm3008_setup_data *setup = pdev->dev.platform_data;
+	int ret;
 
-	printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION);
+	if (!setup)
+		return -EINVAL;
 
 	/* DEM1  DEM0  DE-EMPHASIS_MODE
 	 * Low   Low   De-emphasis 44.1 kHz ON
@@ -74,83 +121,29 @@ static int pcm3008_soc_probe(struct snd_soc_codec *codec)
 	 */
 
 	/* Configure DEM0 GPIO (turning OFF DAC De-emphasis). */
-	ret = gpio_request(setup->dem0_pin, "codec_dem0");
-	if (ret == 0)
-		ret = gpio_direction_output(setup->dem0_pin, 1);
+	ret = devm_gpio_request_one(&pdev->dev, setup->dem0_pin,
+				    GPIOF_OUT_INIT_HIGH, "codec_dem0");
 	if (ret != 0)
-		goto gpio_err;
+		return ret;
 
 	/* Configure DEM1 GPIO (turning OFF DAC De-emphasis). */
-	ret = gpio_request(setup->dem1_pin, "codec_dem1");
-	if (ret == 0)
-		ret = gpio_direction_output(setup->dem1_pin, 0);
+	ret = devm_gpio_request_one(&pdev->dev, setup->dem1_pin,
+				    GPIOF_OUT_INIT_LOW, "codec_dem1");
 	if (ret != 0)
-		goto gpio_err;
+		return ret;
 
 	/* Configure PDAD GPIO. */
-	ret = gpio_request(setup->pdad_pin, "codec_pdad");
-	if (ret == 0)
-		ret = gpio_direction_output(setup->pdad_pin, 1);
+	ret = devm_gpio_request_one(&pdev->dev, setup->pdad_pin,
+				    GPIOF_OUT_INIT_LOW, "codec_pdad");
 	if (ret != 0)
-		goto gpio_err;
+		return ret;
 
 	/* Configure PDDA GPIO. */
-	ret = gpio_request(setup->pdda_pin, "codec_pdda");
-	if (ret == 0)
-		ret = gpio_direction_output(setup->pdda_pin, 1);
+	ret = devm_gpio_request_one(&pdev->dev, setup->pdda_pin,
+				    GPIOF_OUT_INIT_LOW, "codec_pdda");
 	if (ret != 0)
-		goto gpio_err;
-
-	return ret;
-
-gpio_err:
-	pcm3008_gpio_free(setup);
+		return ret;
 
-	return ret;
-}
-
-static int pcm3008_soc_remove(struct snd_soc_codec *codec)
-{
-	struct pcm3008_setup_data *setup = codec->dev->platform_data;
-
-	pcm3008_gpio_free(setup);
-	return 0;
-}
-
-#ifdef CONFIG_PM
-static int pcm3008_soc_suspend(struct snd_soc_codec *codec)
-{
-	struct pcm3008_setup_data *setup = codec->dev->platform_data;
-
-	gpio_set_value(setup->pdad_pin, 0);
-	gpio_set_value(setup->pdda_pin, 0);
-
-	return 0;
-}
-
-static int pcm3008_soc_resume(struct snd_soc_codec *codec)
-{
-	struct pcm3008_setup_data *setup = codec->dev->platform_data;
-
-	gpio_set_value(setup->pdad_pin, 1);
-	gpio_set_value(setup->pdda_pin, 1);
-
-	return 0;
-}
-#else
-#define pcm3008_soc_suspend NULL
-#define pcm3008_soc_resume NULL
-#endif
-
-static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = {
-	.probe = 	pcm3008_soc_probe,
-	.remove = 	pcm3008_soc_remove,
-	.suspend =	pcm3008_soc_suspend,
-	.resume =	pcm3008_soc_resume,
-};
-
-static int pcm3008_codec_probe(struct platform_device *pdev)
-{
 	return snd_soc_register_codec(&pdev->dev,
 			&soc_codec_dev_pcm3008, &pcm3008_dai, 1);
 }
@@ -158,6 +151,7 @@ static int pcm3008_codec_probe(struct platform_device *pdev)
 static int pcm3008_codec_remove(struct platform_device *pdev)
 {
 	snd_soc_unregister_codec(&pdev->dev);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index ce585e37e38a..c26a8f814b18 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -50,8 +50,6 @@ static const struct regmap_range_cfg rt5640_ranges[] = {
 
 static struct reg_default init_list[] = {
 	{RT5640_PR_BASE + 0x3d,	0x3600},
-	{RT5640_PR_BASE + 0x1c,	0x0D21},
-	{RT5640_PR_BASE + 0x1b,	0x0000},
 	{RT5640_PR_BASE + 0x12,	0x0aa8},
 	{RT5640_PR_BASE + 0x14,	0x0aaa},
 	{RT5640_PR_BASE + 0x20,	0x6110},
@@ -384,15 +382,11 @@ static const SOC_ENUM_SINGLE_DECL(
 
 static const struct snd_kcontrol_new rt5640_snd_controls[] = {
 	/* Speaker Output Volume */
-	SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL,
-		RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1),
 	SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL,
 		RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1),
 	SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL,
 		RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv),
 	/* Headphone Output Volume */
-	SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL,
-		RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1),
 	SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL,
 		RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1),
 	SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL,
@@ -737,28 +731,21 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = {
 			RT5640_M_BST1_MM_SFT, 1, 1),
 };
 
-/* INL/R source */
-static const char * const rt5640_inl_src[] = {
-	"IN2P", "MONOP"
-};
+static const struct snd_kcontrol_new spk_l_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL,
+		RT5640_L_MUTE_SFT, 1, 1);
 
-static const SOC_ENUM_SINGLE_DECL(
-	rt5640_inl_enum, RT5640_INL_INR_VOL,
-	RT5640_INL_SEL_SFT, rt5640_inl_src);
+static const struct snd_kcontrol_new spk_r_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL,
+		RT5640_R_MUTE_SFT, 1, 1);
 
-static const struct snd_kcontrol_new rt5640_inl_mux =
-	SOC_DAPM_ENUM("INL source", rt5640_inl_enum);
+static const struct snd_kcontrol_new hp_l_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL,
+		RT5640_L_MUTE_SFT, 1, 1);
 
-static const char * const rt5640_inr_src[] = {
-	"IN2N", "MONON"
-};
-
-static const SOC_ENUM_SINGLE_DECL(
-	rt5640_inr_enum, RT5640_INL_INR_VOL,
-	RT5640_INR_SEL_SFT, rt5640_inr_src);
-
-static const struct snd_kcontrol_new rt5640_inr_mux =
-	SOC_DAPM_ENUM("INR source", rt5640_inr_enum);
+static const struct snd_kcontrol_new hp_r_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL,
+		RT5640_R_MUTE_SFT, 1, 1);
 
 /* Stereo ADC source */
 static const char * const rt5640_stereo_adc1_src[] = {
@@ -891,33 +878,6 @@ static const SOC_ENUM_SINGLE_DECL(
 static const struct snd_kcontrol_new rt5640_sdi_mux =
 	SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum);
 
-static int spk_event(struct snd_soc_dapm_widget *w,
-	struct snd_kcontrol *kcontrol, int event)
-{
-	struct snd_soc_codec *codec = w->codec;
-	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
-
-	switch (event) {
-	case SND_SOC_DAPM_POST_PMU:
-		regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1,
-					0x0001, 0x0001);
-		regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c,
-					0xf000, 0xf000);
-		break;
-
-	case SND_SOC_DAPM_PRE_PMD:
-		regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c,
-					0xf000, 0x0000);
-		regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1,
-					0x0001, 0x0000);
-		break;
-
-	default:
-		return 0;
-	}
-	return 0;
-}
-
 static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -966,6 +926,117 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+void hp_amp_power_on(struct snd_soc_codec *codec)
+{
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	/* depop parameters */
+	regmap_update_bits(rt5640->regmap, RT5640_PR_BASE +
+		RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200);
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2,
+		RT5640_DEPOP_MASK, RT5640_DEPOP_MAN);
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1,
+		RT5640_HP_CP_MASK | RT5640_HP_SG_MASK | RT5640_HP_CB_MASK,
+		RT5640_HP_CP_PU | RT5640_HP_SG_DIS | RT5640_HP_CB_PU);
+	regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1,
+			   0x9f00);
+	/* headphone amp power on */
+	regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1,
+		RT5640_PWR_FV1 | RT5640_PWR_FV2, 0);
+	regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1,
+		RT5640_PWR_HA,
+		RT5640_PWR_HA);
+	usleep_range(10000, 15000);
+	regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1,
+		RT5640_PWR_FV1 | RT5640_PWR_FV2 ,
+		RT5640_PWR_FV1 | RT5640_PWR_FV2);
+}
+
+static void rt5640_pmu_depop(struct snd_soc_codec *codec)
+{
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2,
+		RT5640_DEPOP_MASK | RT5640_DIG_DP_MASK,
+		RT5640_DEPOP_AUTO | RT5640_DIG_DP_EN);
+	regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP,
+		RT5640_PM_HP_MASK, RT5640_PM_HP_HV);
+
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3,
+		RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | RT5640_CP_FQ3_MASK,
+		(RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) |
+		(RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) |
+		(RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT));
+
+	regmap_write(rt5640->regmap, RT5640_PR_BASE +
+		RT5640_MAMP_INT_REG2, 0x1c00);
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1,
+		RT5640_HP_CP_MASK | RT5640_HP_SG_MASK,
+		RT5640_HP_CP_PD | RT5640_HP_SG_EN);
+	regmap_update_bits(rt5640->regmap, RT5640_PR_BASE +
+		RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400);
+}
+
+static int rt5640_hp_event(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		rt5640_pmu_depop(codec);
+		rt5640->hp_mute = 0;
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		rt5640->hp_mute = 1;
+		usleep_range(70000, 75000);
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
+static int rt5640_hp_power_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_POST_PMU:
+		hp_amp_power_on(codec);
+		break;
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
+static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		if (!rt5640->hp_mute)
+			usleep_range(80000, 85000);
+
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2,
 			RT5640_PWR_PLL_BIT, 0, NULL, 0),
@@ -1005,9 +1076,6 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
 		RT5640_PWR_IN_L_BIT, 0, NULL, 0),
 	SND_SOC_DAPM_PGA("INR VOL", RT5640_PWR_VOL,
 		RT5640_PWR_IN_R_BIT, 0, NULL, 0),
-	/* IN Mux */
-	SND_SOC_DAPM_MUX("INL Mux", SND_SOC_NOPM, 0, 0, &rt5640_inl_mux),
-	SND_SOC_DAPM_MUX("INR Mux", SND_SOC_NOPM, 0, 0, &rt5640_inr_mux),
 	/* REC Mixer */
 	SND_SOC_DAPM_MIXER("RECMIXL", RT5640_PWR_MIXER, RT5640_PWR_RM_L_BIT, 0,
 			rt5640_rec_l_mix, ARRAY_SIZE(rt5640_rec_l_mix)),
@@ -1158,15 +1226,28 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
 		rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)),
 	SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1,
 		RT5640_PWR_MA_BIT, 0, NULL, 0),
-	SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1,
-		SND_SOC_NOPM, 0, NULL, 0),
-	SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1,
+	SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM,
+		0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0,
+		rt5640_hp_event,
+		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1,
 		RT5640_PWR_HP_L_BIT, 0, NULL, 0),
-	SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1,
+	SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1,
 		RT5640_PWR_HP_R_BIT, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1,
-		SND_SOC_NOPM, 0, spk_event,
-		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		RT5640_PWR_CLS_D_BIT, 0, NULL, 0),
+
+	/* Output Switch */
+	SND_SOC_DAPM_SWITCH("Speaker L Playback", SND_SOC_NOPM, 0, 0,
+			&spk_l_enable_control),
+	SND_SOC_DAPM_SWITCH("Speaker R Playback", SND_SOC_NOPM, 0, 0,
+			&spk_r_enable_control),
+	SND_SOC_DAPM_SWITCH("HP L Playback", SND_SOC_NOPM, 0, 0,
+			&hp_l_enable_control),
+	SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0,
+			&hp_r_enable_control),
+	SND_SOC_DAPM_POST("HP Post", rt5640_hp_post_event),
 	/* Output Lines */
 	SND_SOC_DAPM_OUTPUT("SPOLP"),
 	SND_SOC_DAPM_OUTPUT("SPOLN"),
@@ -1407,9 +1488,11 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
 	{"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"},
 	{"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"},
 	{"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"},
+	{"HPO MIX L", NULL, "HP L Amp"},
 	{"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"},
 	{"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"},
 	{"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"},
+	{"HPO MIX R", NULL, "HP R Amp"},
 
 	{"LOUT MIX", "DAC L1 Switch", "DAC L1"},
 	{"LOUT MIX", "DAC R1 Switch", "DAC R1"},
@@ -1422,13 +1505,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
 	{"Mono MIX", "OUTVOL L Switch", "OUTVOL L"},
 	{"Mono MIX", "BST1 Switch", "BST1"},
 
-	{"HP L Amp", NULL, "HPO MIX L"},
-	{"HP R Amp", NULL, "HPO MIX R"},
+	{"HP Amp", NULL, "HPO MIX L"},
+	{"HP Amp", NULL, "HPO MIX R"},
 
-	{"SPOLP", NULL, "SPOL MIX"},
-	{"SPOLN", NULL, "SPOL MIX"},
-	{"SPORP", NULL, "SPOR MIX"},
-	{"SPORN", NULL, "SPOR MIX"},
+	{"Speaker L Playback", "Switch", "SPOL MIX"},
+	{"Speaker R Playback", "Switch", "SPOR MIX"},
+	{"SPOLP", NULL, "Speaker L Playback"},
+	{"SPOLN", NULL, "Speaker L Playback"},
+	{"SPORP", NULL, "Speaker R Playback"},
+	{"SPORN", NULL, "Speaker R Playback"},
 
 	{"SPOLP", NULL, "Improve SPK Amp Drv"},
 	{"SPOLN", NULL, "Improve SPK Amp Drv"},
@@ -1438,8 +1523,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
 	{"HPOL", NULL, "Improve HP Amp Drv"},
 	{"HPOR", NULL, "Improve HP Amp Drv"},
 
-	{"HPOL", NULL, "HP L Amp"},
-	{"HPOR", NULL, "HP R Amp"},
+	{"HP L Playback", "Switch", "HP Amp"},
+	{"HP R Playback", "Switch", "HP Amp"},
+	{"HPOL", NULL, "HP L Playback"},
+	{"HPOR", NULL, "HP R Playback"},
 	{"LOUTL", NULL, "LOUT MIX"},
 	{"LOUTR", NULL, "LOUT MIX"},
 	{"MONOP", NULL, "Mono MIX"},
@@ -1818,17 +1905,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec,
 				RT5640_PWR_BG | RT5640_PWR_VREF2,
 				RT5640_PWR_VREF1 | RT5640_PWR_MB |
 				RT5640_PWR_BG | RT5640_PWR_VREF2);
-			mdelay(10);
+			usleep_range(10000, 15000);
 			snd_soc_update_bits(codec, RT5640_PWR_ANLG1,
 				RT5640_PWR_FV1 | RT5640_PWR_FV2,
 				RT5640_PWR_FV1 | RT5640_PWR_FV2);
 			regcache_sync(rt5640->regmap);
 			snd_soc_update_bits(codec, RT5640_DUMMY1,
 						0x0301, 0x0301);
-			snd_soc_update_bits(codec, RT5640_DEPOP_M1,
-						0x001d, 0x0019);
-			snd_soc_update_bits(codec, RT5640_DEPOP_M2,
-						0x2000, 0x2000);
 			snd_soc_update_bits(codec, RT5640_MICBIAS,
 						0x0030, 0x0030);
 		}
@@ -1872,8 +1955,6 @@ static int rt5640_probe(struct snd_soc_codec *codec)
 	rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301);
-	snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019);
-	snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000);
 	snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030);
 	snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00);
 
@@ -2095,6 +2176,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c,
 		regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4,
 					RT5640_IN_DF2, RT5640_IN_DF2);
 
+	rt5640->hp_mute = 1;
+
 	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640,
 			rt5640_dai, ARRAY_SIZE(rt5640_dai));
 	if (ret < 0)
diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h
index c48286d7118f..5e8df25a13f3 100644
--- a/sound/soc/codecs/rt5640.h
+++ b/sound/soc/codecs/rt5640.h
@@ -145,6 +145,8 @@
 
 
 /* Index of Codec Private Register definition */
+#define RT5640_CHPUMP_INT_REG1			0x24
+#define RT5640_MAMP_INT_REG2			0x37
 #define RT5640_3D_SPK				0x63
 #define RT5640_WND_1				0x6c
 #define RT5640_WND_2				0x6d
@@ -153,6 +155,7 @@
 #define RT5640_WND_5				0x70
 #define RT5640_WND_8				0x73
 #define RT5640_DIP_SPK_INF			0x75
+#define RT5640_HP_DCC_INT1			0x77
 #define RT5640_EQ_BW_LOP			0xa0
 #define RT5640_EQ_GN_LOP			0xa1
 #define RT5640_EQ_FC_BP1			0xa2
@@ -1201,6 +1204,14 @@
 #define RT5640_CP_FQ2_SFT			4
 #define RT5640_CP_FQ3_MASK			(0x7)
 #define RT5640_CP_FQ3_SFT			0
+#define RT5640_CP_FQ_1_5_KHZ			0
+#define RT5640_CP_FQ_3_KHZ			1
+#define RT5640_CP_FQ_6_KHZ			2
+#define RT5640_CP_FQ_12_KHZ			3
+#define RT5640_CP_FQ_24_KHZ			4
+#define RT5640_CP_FQ_48_KHZ			5
+#define RT5640_CP_FQ_96_KHZ			6
+#define RT5640_CP_FQ_192_KHZ			7
 
 /* HPOUT charge pump (0x91) */
 #define RT5640_OSW_L_MASK			(0x1 << 11)
@@ -2087,6 +2098,7 @@ struct rt5640_priv {
 	int pll_out;
 
 	int dmic_en;
+	bool hp_mute;
 };
 
 #endif
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 760e8bfeacaa..1f4093f3f3a1 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -654,16 +654,19 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate)
 		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
 			SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP,
 			SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP);
+
+		/* if using pll, clk_ctrl must be set after pll power up */
+		snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl);
 	} else {
+		/* otherwise, clk_ctrl must be set before pll power down */
+		snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl);
+
 		/* power down pll */
 		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
 			SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP,
 			0);
 	}
 
-	/* if using pll, clk_ctrl must be set after pll power up */
-	snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl);
-
 	return 0;
 }
 
@@ -1480,6 +1483,7 @@ static struct snd_soc_codec_driver sgtl5000_driver = {
 static const struct regmap_config sgtl5000_regmap = {
 	.reg_bits = 16,
 	.val_bits = 16,
+	.reg_stride = 2,
 
 	.max_register = SGTL5000_MAX_REG_OFFSET,
 	.volatile_reg = sgtl5000_volatile,
diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c
index 73e205c892a0..38f3b105c17d 100644
--- a/sound/soc/codecs/si476x.c
+++ b/sound/soc/codecs/si476x.c
@@ -102,6 +102,16 @@ static int si476x_codec_write(struct snd_soc_codec *codec,
 	return err;
 }
 
+static const struct snd_soc_dapm_widget si476x_dapm_widgets[] = {
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+};
+
+static const struct snd_soc_dapm_route si476x_dapm_routes[] = {
+	{ "Capture", NULL, "LOUT" },
+	{ "Capture", NULL, "ROUT" },
+};
+
 static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai,
 				    unsigned int fmt)
 {
@@ -260,6 +270,10 @@ static struct snd_soc_codec_driver soc_codec_dev_si476x = {
 	.probe  = si476x_codec_probe,
 	.read   = si476x_codec_read,
 	.write  = si476x_codec_write,
+	.dapm_widgets = si476x_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(si476x_dapm_widgets),
+	.dapm_routes = si476x_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(si476x_dapm_routes),
 };
 
 static int si476x_platform_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c
index e9d7881ed2c8..e3501f40c7b3 100644
--- a/sound/soc/codecs/spdif_receiver.c
+++ b/sound/soc/codecs/spdif_receiver.c
@@ -23,11 +23,26 @@
 #include <sound/initval.h>
 #include <linux/of.h>
 
+static const struct snd_soc_dapm_widget dir_widgets[] = {
+	SND_SOC_DAPM_INPUT("spdif-in"),
+};
+
+static const struct snd_soc_dapm_route dir_routes[] = {
+	{ "Capture", NULL, "spdif-in" },
+};
+
 #define STUB_RATES	SNDRV_PCM_RATE_8000_192000
 #define STUB_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | \
+			SNDRV_PCM_FMTBIT_S20_3LE | \
+			SNDRV_PCM_FMTBIT_S24_LE | \
 			SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
 
-static struct snd_soc_codec_driver soc_codec_spdif_dir;
+static struct snd_soc_codec_driver soc_codec_spdif_dir = {
+	.dapm_widgets = dir_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(dir_widgets),
+	.dapm_routes = dir_routes,
+	.num_dapm_routes = ARRAY_SIZE(dir_routes),
+};
 
 static struct snd_soc_dai_driver dir_stub_dai = {
 	.name		= "dir-hifi",
diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c
index 18280499fd55..a078aa31052a 100644
--- a/sound/soc/codecs/spdif_transmitter.c
+++ b/sound/soc/codecs/spdif_transmitter.c
@@ -25,10 +25,24 @@
 #define DRV_NAME "spdif-dit"
 
 #define STUB_RATES	SNDRV_PCM_RATE_8000_96000
-#define STUB_FORMATS	SNDRV_PCM_FMTBIT_S16_LE
+#define STUB_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | \
+			SNDRV_PCM_FMTBIT_S20_3LE | \
+			SNDRV_PCM_FMTBIT_S24_LE)
 
+static const struct snd_soc_dapm_widget dit_widgets[] = {
+	SND_SOC_DAPM_OUTPUT("spdif-out"),
+};
+
+static const struct snd_soc_dapm_route dit_routes[] = {
+	{ "spdif-out", NULL, "Playback" },
+};
 
-static struct snd_soc_codec_driver soc_codec_spdif_dit;
+static struct snd_soc_codec_driver soc_codec_spdif_dit = {
+	.dapm_widgets = dit_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(dit_widgets),
+	.dapm_routes = dit_routes,
+	.num_dapm_routes = ARRAY_SIZE(dit_routes),
+};
 
 static struct snd_soc_dai_driver dit_stub_dai = {
 	.name		= "dit-hifi",
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index f8d30e5f6371..492644e67ace 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -561,8 +561,9 @@ static int ssm2602_suspend(struct snd_soc_codec *codec)
 
 static int ssm2602_resume(struct snd_soc_codec *codec)
 {
-	snd_soc_cache_sync(codec);
+	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
 
+	regcache_sync(ssm2602->regmap);
 	ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index cfb55fe35e98..06edb396e733 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -363,16 +363,18 @@ static void sta32x_watchdog(struct work_struct *work)
 	}
 
 	if (!sta32x->shutdown)
-		schedule_delayed_work(&sta32x->watchdog_work,
-				      round_jiffies_relative(HZ));
+		queue_delayed_work(system_power_efficient_wq,
+				   &sta32x->watchdog_work,
+				   round_jiffies_relative(HZ));
 }
 
 static void sta32x_watchdog_start(struct sta32x_priv *sta32x)
 {
 	if (sta32x->pdata->needs_esd_watchdog) {
 		sta32x->shutdown = 0;
-		schedule_delayed_work(&sta32x->watchdog_work,
-				      round_jiffies_relative(HZ));
+		queue_delayed_work(system_power_efficient_wq,
+				   &sta32x->watchdog_work,
+				   round_jiffies_relative(HZ));
 	}
 }
 
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index b1f6982c7c9c..7b8f3d965f43 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -29,7 +29,7 @@ MODULE_LICENSE("GPL");
 /* AIC26 driver private data */
 struct aic26 {
 	struct spi_device *spi;
-	struct snd_soc_codec codec;
+	struct snd_soc_codec *codec;
 	int master;
 	int datfm;
 	int mclk;
@@ -119,6 +119,22 @@ static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg,
 	return 0;
 }
 
+static const struct snd_soc_dapm_widget tlv320aic26_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("MICIN"),
+SND_SOC_DAPM_INPUT("AUX"),
+
+SND_SOC_DAPM_OUTPUT("HPL"),
+SND_SOC_DAPM_OUTPUT("HPR"),
+};
+
+static const struct snd_soc_dapm_route tlv320aic26_dapm_routes[] = {
+	{ "Capture", NULL, "MICIN" },
+	{ "Capture", NULL, "AUX" },
+
+	{ "HPL", NULL, "Playback" },
+	{ "HPR", NULL, "Playback" },
+};
+
 /* ---------------------------------------------------------------------
  * Digital Audio Interface Operations
  */
@@ -174,9 +190,9 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
 	dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval);
 	qval = 0;
 	reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
-	aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
+	snd_soc_write(codec, AIC26_REG_PLL_PROG1, reg);
 	reg = dval << 2;
-	aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg);
+	snd_soc_write(codec, AIC26_REG_PLL_PROG2, reg);
 
 	/* Audio Control 3 (master mode, fsref rate) */
 	reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3);
@@ -185,13 +201,13 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
 		reg |= 0x0800;
 	if (fsref == 48000)
 		reg |= 0x2000;
-	aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
+	snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
 
 	/* Audio Control 1 (FSref divisor) */
 	reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1);
 	reg &= ~0x0fff;
 	reg |= wlen | aic26->datfm | (divisor << 3) | divisor;
-	aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg);
+	snd_soc_write(codec, AIC26_REG_AUDIO_CTRL1, reg);
 
 	return 0;
 }
@@ -212,7 +228,7 @@ static int aic26_mute(struct snd_soc_dai *dai, int mute)
 		reg |= 0x8080;
 	else
 		reg &= ~0x8080;
-	aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg);
+	snd_soc_write(codec, AIC26_REG_DAC_GAIN, reg);
 
 	return 0;
 }
@@ -330,7 +346,7 @@ static ssize_t aic26_keyclick_show(struct device *dev,
 	struct aic26 *aic26 = dev_get_drvdata(dev);
 	int val, amp, freq, len;
 
-	val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
+	val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2);
 	amp = (val >> 12) & 0x7;
 	freq = (125 << ((val >> 8) & 0x7)) >> 1;
 	len = 2 * (1 + ((val >> 4) & 0xf));
@@ -346,9 +362,9 @@ static ssize_t aic26_keyclick_set(struct device *dev,
 	struct aic26 *aic26 = dev_get_drvdata(dev);
 	int val;
 
-	val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
+	val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2);
 	val |= 0x8000;
-	aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val);
+	snd_soc_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val);
 
 	return count;
 }
@@ -360,25 +376,26 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set);
  */
 static int aic26_probe(struct snd_soc_codec *codec)
 {
+	struct aic26 *aic26 = dev_get_drvdata(codec->dev);
 	int ret, err, i, reg;
 
-	dev_info(codec->dev, "Probing AIC26 SoC CODEC driver\n");
+	aic26->codec = codec;
 
 	/* Reset the codec to power on defaults */
-	aic26_reg_write(codec, AIC26_REG_RESET, 0xBB00);
+	snd_soc_write(codec, AIC26_REG_RESET, 0xBB00);
 
 	/* Power up CODEC */
-	aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0);
+	snd_soc_write(codec, AIC26_REG_POWER_CTRL, 0);
 
 	/* Audio Control 3 (master mode, fsref rate) */
-	reg = aic26_reg_read(codec, AIC26_REG_AUDIO_CTRL3);
+	reg = snd_soc_read(codec, AIC26_REG_AUDIO_CTRL3);
 	reg &= ~0xf800;
 	reg |= 0x0800; /* set master mode */
-	aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
+	snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
 
 	/* Fill register cache */
 	for (i = 0; i < codec->driver->reg_cache_size; i++)
-		aic26_reg_read(codec, i);
+		snd_soc_read(codec, i);
 
 	/* Register the sysfs files for debugging */
 	/* Create SysFS files */
@@ -401,6 +418,10 @@ static struct snd_soc_codec_driver aic26_soc_codec_dev = {
 	.write = aic26_reg_write,
 	.reg_cache_size = AIC26_NUM_REGS,
 	.reg_word_size = sizeof(u16),
+	.dapm_widgets = tlv320aic26_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(tlv320aic26_dapm_widgets),
+	.dapm_routes = tlv320aic26_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(tlv320aic26_dapm_routes),
 };
 
 /* ---------------------------------------------------------------------
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 17df4e32feac..2ed57d4aa445 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -338,18 +338,6 @@ static inline int aic32x4_get_divs(int mclk, int rate)
 	return -EINVAL;
 }
 
-static int aic32x4_add_widgets(struct snd_soc_codec *codec)
-{
-	snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets,
-				  ARRAY_SIZE(aic32x4_dapm_widgets));
-
-	snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes,
-				ARRAY_SIZE(aic32x4_dapm_routes));
-
-	snd_soc_dapm_new_widgets(&codec->dapm);
-	return 0;
-}
-
 static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				  int clk_id, unsigned int freq, int dir)
 {
@@ -683,9 +671,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
 	}
 
 	aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	snd_soc_add_codec_controls(codec, aic32x4_snd_controls,
-			     ARRAY_SIZE(aic32x4_snd_controls));
-	aic32x4_add_widgets(codec);
 
 	/*
 	 * Workaround: for an unknown reason, the ADC needs to be powered up
@@ -714,6 +699,13 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
 	.suspend = aic32x4_suspend,
 	.resume = aic32x4_resume,
 	.set_bias_level = aic32x4_set_bias_level,
+
+	.controls = aic32x4_snd_controls,
+	.num_controls = ARRAY_SIZE(aic32x4_snd_controls),
+	.dapm_widgets = aic32x4_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets),
+	.dapm_routes = aic32x4_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes),
 };
 
 static int aic32x4_i2c_probe(struct i2c_client *i2c,
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index e5b926883131..6e3f269243e0 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -138,8 +138,7 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = {
 static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
 					struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
 	unsigned int reg = mc->reg;
@@ -147,10 +146,9 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
 	int max = mc->max;
 	unsigned int mask = (1 << fls(max)) - 1;
 	unsigned int invert = mc->invert;
-	unsigned short val, val_mask;
-	int ret;
-	struct snd_soc_dapm_path *path;
-	int found = 0;
+	unsigned short val;
+	struct snd_soc_dapm_update update;
+	int connect, change;
 
 	val = (ucontrol->value.integer.value[0] & mask);
 
@@ -158,42 +156,26 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
 	if (val)
 		val = mask;
 
+	connect = !!val;
+
 	if (invert)
 		val = mask - val;
-	val_mask = mask << shift;
-	val = val << shift;
-
-	mutex_lock(&widget->codec->mutex);
 
-	if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) {
-		/* find dapm widget path assoc with kcontrol */
-		list_for_each_entry(path, &widget->dapm->card->paths, list) {
-			if (path->kcontrol != kcontrol)
-				continue;
+	mask <<= shift;
+	val <<= shift;
 
-			/* found, now check type */
-			found = 1;
-			if (val)
-				/* new connection */
-				path->connect = invert ? 0 : 1;
-			else
-				/* old connection must be powered down */
-				path->connect = invert ? 1 : 0;
+	change = snd_soc_test_bits(codec, val, mask, reg);
+	if (change) {
+		update.kcontrol = kcontrol;
+		update.reg = reg;
+		update.mask = mask;
+		update.val = val;
 
-			dapm_mark_dirty(path->source, "tlv320aic3x source");
-			dapm_mark_dirty(path->sink, "tlv320aic3x sink");
-
-			break;
-		}
+		snd_soc_dapm_mixer_update_power(&codec->dapm, kcontrol, connect,
+			&update);
 	}
 
-	mutex_unlock(&widget->codec->mutex);
-
-	if (found)
-		snd_soc_dapm_sync(widget->dapm);
-
-	ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val);
-	return ret;
+	return change;
 }
 
 /*
@@ -1492,6 +1474,7 @@ static const struct i2c_device_id aic3x_i2c_id[] = {
 	{ "tlv320aic3x", AIC3X_MODEL_3X },
 	{ "tlv320aic33", AIC3X_MODEL_33 },
 	{ "tlv320aic3007", AIC3X_MODEL_3007 },
+	{ "tlv320aic3106", AIC3X_MODEL_3X },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id);
@@ -1582,6 +1565,9 @@ static int aic3x_i2c_remove(struct i2c_client *client)
 #if defined(CONFIG_OF)
 static const struct of_device_id tlv320aic3x_of_match[] = {
 	{ .compatible = "ti,tlv320aic3x", },
+	{ .compatible = "ti,tlv320aic33" },
+	{ .compatible = "ti,tlv320aic3007" },
+	{ .compatible = "ti,tlv320aic3106" },
 	{},
 };
 MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match);
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 8e6e5b016021..1e3884d6b3fb 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -137,8 +137,6 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
 
 /* codec private data */
 struct twl4030_priv {
-	struct snd_soc_codec codec;
-
 	unsigned int codec_powered;
 
 	/* reference counts of AIF/APLL users */
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 44621ddc332d..3c79dbb6c323 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -429,7 +429,8 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data)
 	struct snd_soc_codec *codec = data;
 	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 
-	schedule_delayed_work(&priv->hs_jack.work, msecs_to_jiffies(200));
+	queue_delayed_work(system_power_efficient_wq,
+			   &priv->hs_jack.work, msecs_to_jiffies(200));
 
 	return IRQ_HANDLED;
 }
@@ -437,9 +438,7 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data)
 static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
-	struct snd_soc_codec *codec = widget->codec;
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int val;
 
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 6d0aa44c3757..c94d4c1e3dac 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -325,7 +325,6 @@ static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 static int uda134x_set_bias_level(struct snd_soc_codec *codec,
 				  enum snd_soc_bias_level level)
 {
-	u8 reg;
 	struct uda134x_platform_data *pd = codec->control_data;
 	int i;
 	u8 *cache = codec->reg_cache;
@@ -334,23 +333,6 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec,
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		/* ADC, DAC on */
-		switch (pd->model) {
-		case UDA134X_UDA1340:
-		case UDA134X_UDA1344:
-		case UDA134X_UDA1345:
-			reg = uda134x_read_reg_cache(codec, UDA134X_DATA011);
-			uda134x_write(codec, UDA134X_DATA011, reg | 0x03);
-			break;
-		case UDA134X_UDA1341:
-			reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
-			uda134x_write(codec, UDA134X_STATUS1, reg | 0x03);
-			break;
-		default:
-			printk(KERN_ERR "UDA134X SoC codec: "
-			       "unsupported model %d\n", pd->model);
-			return -EINVAL;
-		}
 		break;
 	case SND_SOC_BIAS_PREPARE:
 		/* power on */
@@ -362,23 +344,6 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec,
 		}
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		/* ADC, DAC power off */
-		switch (pd->model) {
-		case UDA134X_UDA1340:
-		case UDA134X_UDA1344:
-		case UDA134X_UDA1345:
-			reg = uda134x_read_reg_cache(codec, UDA134X_DATA011);
-			uda134x_write(codec, UDA134X_DATA011, reg & ~(0x03));
-			break;
-		case UDA134X_UDA1341:
-			reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
-			uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03));
-			break;
-		default:
-			printk(KERN_ERR "UDA134X SoC codec: "
-			       "unsupported model %d\n", pd->model);
-			return -EINVAL;
-		}
 		break;
 	case SND_SOC_BIAS_OFF:
 		/* power off */
@@ -450,6 +415,37 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
 SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
 };
 
+/* UDA1341 has the DAC/ADC power down in STATUS1 */
+static const struct snd_soc_dapm_widget uda1341_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_STATUS1, 0, 0),
+	SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_STATUS1, 1, 0),
+};
+
+/* UDA1340/4/5 has the DAC/ADC pwoer down in DATA0 11 */
+static const struct snd_soc_dapm_widget uda1340_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_DATA011, 0, 0),
+	SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_DATA011, 1, 0),
+};
+
+/* Common DAPM widgets */
+static const struct snd_soc_dapm_widget uda134x_dapm_widgets[] = {
+	SND_SOC_DAPM_INPUT("VINL1"),
+	SND_SOC_DAPM_INPUT("VINR1"),
+	SND_SOC_DAPM_INPUT("VINL2"),
+	SND_SOC_DAPM_INPUT("VINR2"),
+	SND_SOC_DAPM_OUTPUT("VOUTL"),
+	SND_SOC_DAPM_OUTPUT("VOUTR"),
+};
+
+static const struct snd_soc_dapm_route uda134x_dapm_routes[] = {
+	{ "ADC", NULL, "VINL1" },
+	{ "ADC", NULL, "VINR1" },
+	{ "ADC", NULL, "VINL2" },
+	{ "ADC", NULL, "VINR2" },
+	{ "VOUTL", NULL, "DAC" },
+	{ "VOUTR", NULL, "DAC" },
+};
+
 static const struct snd_soc_dai_ops uda134x_dai_ops = {
 	.startup	= uda134x_startup,
 	.shutdown	= uda134x_shutdown,
@@ -485,6 +481,8 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec)
 {
 	struct uda134x_priv *uda134x;
 	struct uda134x_platform_data *pd = codec->card->dev->platform_data;
+	const struct snd_soc_dapm_widget *widgets;
+	unsigned num_widgets;
 
 	int ret;
 
@@ -526,6 +524,22 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec)
 	else
 		uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
+	if (pd->model == UDA134X_UDA1341) {
+		widgets = uda1341_dapm_widgets;
+		num_widgets = ARRAY_SIZE(uda1341_dapm_widgets);
+	} else {
+		widgets = uda1340_dapm_widgets;
+		num_widgets = ARRAY_SIZE(uda1340_dapm_widgets);
+	}
+
+	ret = snd_soc_dapm_new_controls(&codec->dapm, widgets, num_widgets);
+	if (ret) {
+		printk(KERN_ERR "%s failed to register dapm controls: %d",
+			__func__, ret);
+		kfree(uda134x);
+		return ret;
+	}
+
 	switch (pd->model) {
 	case UDA134X_UDA1340:
 	case UDA134X_UDA1344:
@@ -599,6 +613,10 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = {
 	.read = uda134x_read_reg_cache,
 	.write = uda134x_write,
 	.set_bias_level = uda134x_set_bias_level,
+	.dapm_widgets = uda134x_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(uda134x_dapm_widgets),
+	.dapm_routes = uda134x_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(uda134x_dapm_routes),
 };
 
 static int uda134x_codec_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index 54cd3da09abd..b7ab2ef567c8 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -290,6 +290,18 @@ static const struct snd_kcontrol_new wl1273_controls[] = {
 		       snd_wl1273_fm_volume_get, snd_wl1273_fm_volume_put),
 };
 
+static const struct snd_soc_dapm_widget wl1273_dapm_widgets[] = {
+	SND_SOC_DAPM_INPUT("RX"),
+
+	SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route wl1273_dapm_routes[] = {
+	{ "Capture", NULL, "RX" },
+
+	{ "TX", NULL, "Playback" },
+};
+
 static int wl1273_startup(struct snd_pcm_substream *substream,
 			  struct snd_soc_dai *dai)
 {
@@ -483,6 +495,11 @@ static int wl1273_remove(struct snd_soc_codec *codec)
 static struct snd_soc_codec_driver soc_codec_dev_wl1273 = {
 	.probe = wl1273_probe,
 	.remove = wl1273_remove,
+
+	.dapm_widgets = wl1273_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wl1273_dapm_widgets),
+	.dapm_routes = wl1273_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(wl1273_dapm_routes),
 };
 
 static int wl1273_platform_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index 10adc4145d46..d5ebcb00019b 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -420,7 +420,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec)
 		xfer->codec = codec;
 		list_add_tail(&xfer->list, &xfer_list);
 
-		out = kzalloc(len, GFP_KERNEL);
+		out = kzalloc(len, GFP_KERNEL | GFP_DMA);
 		if (!out) {
 			dev_err(codec->dev,
 				"Failed to allocate RX buffer\n");
@@ -429,7 +429,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec)
 		}
 		xfer->t.rx_buf = out;
 
-		img = kzalloc(len, GFP_KERNEL);
+		img = kzalloc(len, GFP_KERNEL | GFP_DMA);
 		if (!img) {
 			dev_err(codec->dev,
 				"Failed to allocate image buffer\n");
@@ -523,14 +523,14 @@ static int wm0010_stage2_load(struct snd_soc_codec *codec)
 	dev_dbg(codec->dev, "Downloading %zu byte stage 2 loader\n", fw->size);
 
 	/* Copy to local buffer first as vmalloc causes problems for dma */
-	img = kzalloc(fw->size, GFP_KERNEL);
+	img = kzalloc(fw->size, GFP_KERNEL | GFP_DMA);
 	if (!img) {
 		dev_err(codec->dev, "Failed to allocate image buffer\n");
 		ret = -ENOMEM;
 		goto abort2;
 	}
 
-	out = kzalloc(fw->size, GFP_KERNEL);
+	out = kzalloc(fw->size, GFP_KERNEL | GFP_DMA);
 	if (!out) {
 		dev_err(codec->dev, "Failed to allocate output buffer\n");
 		ret = -ENOMEM;
@@ -670,14 +670,14 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 
 		ret = -ENOMEM;
 		len = pll_rec.length + 8;
-		out = kzalloc(len, GFP_KERNEL);
+		out = kzalloc(len, GFP_KERNEL | GFP_DMA);
 		if (!out) {
 			dev_err(codec->dev,
 				"Failed to allocate RX buffer\n");
 			goto abort;
 		}
 
-		img_swap = kzalloc(len, GFP_KERNEL);
+		img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA);
 		if (!img_swap) {
 			dev_err(codec->dev,
 				"Failed to allocate image buffer\n");
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 282fd232cdf7..8bbddc151aa8 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -998,6 +998,8 @@ SND_SOC_DAPM_INPUT("IN2R"),
 SND_SOC_DAPM_INPUT("IN3L"),
 SND_SOC_DAPM_INPUT("IN3R"),
 
+SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"),
+
 SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
@@ -1421,9 +1423,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
 	{ "Tone Generator 1", NULL, "TONE" },
 	{ "Tone Generator 2", NULL, "TONE" },
 
-	{ "Mic Mute Mixer", NULL, "Noise Mixer" },
-	{ "Mic Mute Mixer", NULL, "Mic Mixer" },
-
 	{ "AIF1 Capture", NULL, "AIF1TX1" },
 	{ "AIF1 Capture", NULL, "AIF1TX2" },
 	{ "AIF1 Capture", NULL, "AIF1TX3" },
@@ -1499,23 +1498,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
 	{ "IN3L PGA", NULL, "IN3L" },
 	{ "IN3R PGA", NULL, "IN3R" },
 
-	{ "ASRC1L", NULL, "ASRC1L Input" },
-	{ "ASRC1R", NULL, "ASRC1R Input" },
-	{ "ASRC2L", NULL, "ASRC2L Input" },
-	{ "ASRC2R", NULL, "ASRC2R Input" },
-
-	{ "ISRC1DEC1", NULL, "ISRC1DEC1 Input" },
-	{ "ISRC1DEC2", NULL, "ISRC1DEC2 Input" },
-
-	{ "ISRC1INT1", NULL, "ISRC1INT1 Input" },
-	{ "ISRC1INT2", NULL, "ISRC1INT2 Input" },
-
-	{ "ISRC2DEC1", NULL, "ISRC2DEC1 Input" },
-	{ "ISRC2DEC2", NULL, "ISRC2DEC2 Input" },
-
-	{ "ISRC2INT1", NULL, "ISRC2INT1 Input" },
-	{ "ISRC2INT2", NULL, "ISRC2INT2 Input" },
-
 	ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
 	ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
 	ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
@@ -1567,22 +1549,25 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
 	ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"),
 	ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"),
 
-	ARIZONA_MUX_ROUTES("ASRC1L"),
-	ARIZONA_MUX_ROUTES("ASRC1R"),
-	ARIZONA_MUX_ROUTES("ASRC2L"),
-	ARIZONA_MUX_ROUTES("ASRC2R"),
+	ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"),
+	ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"),
 
-	ARIZONA_MUX_ROUTES("ISRC1INT1"),
-	ARIZONA_MUX_ROUTES("ISRC1INT2"),
+	ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"),
+	ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"),
+	ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"),
+	ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"),
 
-	ARIZONA_MUX_ROUTES("ISRC1DEC1"),
-	ARIZONA_MUX_ROUTES("ISRC1DEC2"),
+	ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"),
+	ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"),
 
-	ARIZONA_MUX_ROUTES("ISRC2INT1"),
-	ARIZONA_MUX_ROUTES("ISRC2INT2"),
+	ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"),
+	ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"),
 
-	ARIZONA_MUX_ROUTES("ISRC2DEC1"),
-	ARIZONA_MUX_ROUTES("ISRC2DEC2"),
+	ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"),
+	ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"),
+
+	ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"),
+	ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"),
 
 	ARIZONA_DSP_ROUTES("DSP1"),
 
@@ -1614,6 +1599,9 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
 	{ "SPKDAT1R", NULL, "OUT5R" },
 
 	{ "MICSUPP", NULL, "SYSCLK" },
+
+	{ "DRC1 Signal Activity", NULL, "DRC1L" },
+	{ "DRC1 Signal Activity", NULL, "DRC1R" },
 };
 
 static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
@@ -1781,6 +1769,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec)
 		return ret;
 
 	arizona_init_spk(codec);
+	arizona_init_gpio(codec);
 
 	snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
 
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 2e7cb4ba161a..bbd64384ca1c 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -58,14 +58,10 @@ static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0);
 	SOC_SINGLE(name " NG SPKDAT2R Switch", base, 11, 1, 0)
 
 static const struct snd_kcontrol_new wm5110_snd_controls[] = {
-SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL,
-	   ARIZONA_IN1_OSR_SHIFT, 1, 0),
-SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL,
-	   ARIZONA_IN2_OSR_SHIFT, 1, 0),
-SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL,
-	   ARIZONA_IN3_OSR_SHIFT, 1, 0),
-SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL,
-	   ARIZONA_IN4_OSR_SHIFT, 1, 0),
+SOC_ENUM("IN1 OSR", arizona_in_dmic_osr[0]),
+SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]),
+SOC_ENUM("IN3 OSR", arizona_in_dmic_osr[2]),
+SOC_ENUM("IN4 OSR", arizona_in_dmic_osr[3]),
 
 SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL,
 		     ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
@@ -432,6 +428,9 @@ SND_SOC_DAPM_INPUT("IN3R"),
 SND_SOC_DAPM_INPUT("IN4L"),
 SND_SOC_DAPM_INPUT("IN4R"),
 
+SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"),
+SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"),
+
 SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
@@ -842,9 +841,6 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
 	{ "Tone Generator 1", NULL, "TONE" },
 	{ "Tone Generator 2", NULL, "TONE" },
 
-	{ "Mic Mute Mixer", NULL, "Noise Mixer" },
-	{ "Mic Mute Mixer", NULL, "Mic Mixer" },
-
 	{ "AIF1 Capture", NULL, "AIF1TX1" },
 	{ "AIF1 Capture", NULL, "AIF1TX2" },
 	{ "AIF1 Capture", NULL, "AIF1TX3" },
@@ -979,10 +975,13 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
 	ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"),
 	ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"),
 
-	ARIZONA_MUX_ROUTES("ASRC1L"),
-	ARIZONA_MUX_ROUTES("ASRC1R"),
-	ARIZONA_MUX_ROUTES("ASRC2L"),
-	ARIZONA_MUX_ROUTES("ASRC2R"),
+	ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"),
+	ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"),
+
+	ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"),
+	ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"),
+	ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"),
+	ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"),
 
 	{ "HPOUT1L", NULL, "OUT1L" },
 	{ "HPOUT1R", NULL, "OUT1R" },
@@ -1006,6 +1005,11 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
 	{ "SPKDAT2R", NULL, "OUT6R" },
 
 	{ "MICSUPP", NULL, "SYSCLK" },
+
+	{ "DRC1 Signal Activity", NULL, "DRC1L" },
+	{ "DRC1 Signal Activity", NULL, "DRC1R" },
+	{ "DRC2 Signal Activity", NULL, "DRC2L" },
+	{ "DRC2 Signal Activity", NULL, "DRC2R" },
 };
 
 static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
@@ -1170,6 +1174,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
 		return ret;
 
 	arizona_init_spk(codec);
+	arizona_init_gpio(codec);
 
 	snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
 
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 0e8b3aaf6c8d..af1318ddb062 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1301,7 +1301,8 @@ static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data)
 	if (device_may_wakeup(wm8350->dev))
 		pm_wakeup_event(wm8350->dev, 250);
 
-	schedule_delayed_work(&priv->hpl.work, msecs_to_jiffies(200));
+	queue_delayed_work(system_power_efficient_wq,
+			   &priv->hpl.work, msecs_to_jiffies(200));
 
 	return IRQ_HANDLED;
 }
@@ -1318,7 +1319,8 @@ static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data)
 	if (device_may_wakeup(wm8350->dev))
 		pm_wakeup_event(wm8350->dev, 250);
 
-	schedule_delayed_work(&priv->hpr.work, msecs_to_jiffies(200));
+	queue_delayed_work(system_power_efficient_wq,
+			   &priv->hpr.work, msecs_to_jiffies(200));
 
 	return IRQ_HANDLED;
 }
diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c
index 462f5e4d5c05..7b1a6d5c11c6 100644
--- a/sound/soc/codecs/wm8727.c
+++ b/sound/soc/codecs/wm8727.c
@@ -23,6 +23,16 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
+static const struct snd_soc_dapm_widget wm8727_dapm_widgets[] = {
+SND_SOC_DAPM_OUTPUT("VOUTL"),
+SND_SOC_DAPM_OUTPUT("VOUTR"),
+};
+
+static const struct snd_soc_dapm_route wm8727_dapm_routes[] = {
+	{ "VOUTL", NULL, "Playback" },
+	{ "VOUTR", NULL, "Playback" },
+};
+
 /*
  * Note this is a simple chip with no configuration interface, sample rate is
  * determined automatically by examining the Master clock and Bit clock ratios
@@ -43,7 +53,12 @@ static struct snd_soc_dai_driver wm8727_dai = {
 		},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_wm8727;
+static struct snd_soc_codec_driver soc_codec_dev_wm8727 = {
+	.dapm_widgets = wm8727_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm8727_dapm_widgets),
+	.dapm_routes = wm8727_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(wm8727_dapm_routes),
+};
 
 static int wm8727_probe(struct platform_device *pdev)
 {
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 5276062d6c79..456bb8c6d759 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -45,6 +45,7 @@ static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = {
 struct wm8731_priv {
 	struct regmap *regmap;
 	struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES];
+	const struct snd_pcm_hw_constraint_list *constraints;
 	unsigned int sysclk;
 	int sysclk_type;
 	int playback_fs;
@@ -290,6 +291,36 @@ static const struct _coeff_div coeff_div[] = {
 	{12000000, 88200, 136, 0xf, 0x1, 0x1},
 };
 
+/* rates constraints */
+static const unsigned int wm8731_rates_12000000[] = {
+	8000, 32000, 44100, 48000, 96000, 88200,
+};
+
+static const unsigned int wm8731_rates_12288000_18432000[] = {
+	8000, 32000, 48000, 96000,
+};
+
+static const unsigned int wm8731_rates_11289600_16934400[] = {
+	8000, 44100, 88200,
+};
+
+static const struct snd_pcm_hw_constraint_list wm8731_constraints_12000000 = {
+	.list = wm8731_rates_12000000,
+	.count = ARRAY_SIZE(wm8731_rates_12000000),
+};
+
+static const
+struct snd_pcm_hw_constraint_list wm8731_constraints_12288000_18432000 = {
+	.list = wm8731_rates_12288000_18432000,
+	.count = ARRAY_SIZE(wm8731_rates_12288000_18432000),
+};
+
+static const
+struct snd_pcm_hw_constraint_list wm8731_constraints_11289600_16934400 = {
+	.list = wm8731_rates_11289600_16934400,
+	.count = ARRAY_SIZE(wm8731_rates_11289600_16934400),
+};
+
 static inline int get_coeff(int mclk, int rate)
 {
 	int i;
@@ -362,17 +393,26 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	}
 
 	switch (freq) {
-	case 11289600:
+	case 0:
+		wm8731->constraints = NULL;
+		break;
 	case 12000000:
+		wm8731->constraints = &wm8731_constraints_12000000;
+		break;
 	case 12288000:
-	case 16934400:
 	case 18432000:
-		wm8731->sysclk = freq;
+		wm8731->constraints = &wm8731_constraints_12288000_18432000;
+		break;
+	case 16934400:
+	case 11289600:
+		wm8731->constraints = &wm8731_constraints_11289600_16934400;
 		break;
 	default:
 		return -EINVAL;
 	}
 
+	wm8731->sysclk = freq;
+
 	snd_soc_dapm_sync(&codec->dapm);
 
 	return 0;
@@ -475,12 +515,26 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
+static int wm8731_startup(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(dai->codec);
+
+	if (wm8731->constraints)
+		snd_pcm_hw_constraint_list(substream->runtime, 0,
+					   SNDRV_PCM_HW_PARAM_RATE,
+					   wm8731->constraints);
+
+	return 0;
+}
+
 #define WM8731_RATES SNDRV_PCM_RATE_8000_96000
 
 #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
 static const struct snd_soc_dai_ops wm8731_dai_ops = {
+	.startup	= wm8731_startup,
 	.hw_params	= wm8731_hw_params,
 	.digital_mute	= wm8731_mute,
 	.set_sysclk	= wm8731_set_dai_sysclk,
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 0a4ab4c423d1..d96ebf52d953 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1456,8 +1456,9 @@ static int wm8753_resume(struct snd_soc_codec *codec)
 	if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
 		wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
 		codec->dapm.bias_level = SND_SOC_BIAS_ON;
-		schedule_delayed_work(&codec->dapm.delayed_work,
-			msecs_to_jiffies(caps_charge));
+		queue_delayed_work(system_power_efficient_wq,
+				   &codec->dapm.delayed_work,
+				   msecs_to_jiffies(caps_charge));
 	}
 
 	return 0;
diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c
index f1fdbf63abb4..8092495605ce 100644
--- a/sound/soc/codecs/wm8782.c
+++ b/sound/soc/codecs/wm8782.c
@@ -26,6 +26,16 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
+static const struct snd_soc_dapm_widget wm8782_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("AINL"),
+SND_SOC_DAPM_INPUT("AINR"),
+};
+
+static const struct snd_soc_dapm_route wm8782_dapm_routes[] = {
+	{ "Capture", NULL, "AINL" },
+	{ "Capture", NULL, "AINR" },
+};
+
 static struct snd_soc_dai_driver wm8782_dai = {
 	.name = "wm8782",
 	.capture = {
@@ -40,7 +50,12 @@ static struct snd_soc_dai_driver wm8782_dai = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_wm8782;
+static struct snd_soc_codec_driver soc_codec_dev_wm8782 = {
+	.dapm_widgets = wm8782_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm8782_dapm_widgets),
+	.dapm_routes = wm8782_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(wm8782_dapm_routes),
+};
 
 static int wm8782_probe(struct platform_device *pdev)
 {
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index fa24cedee687..eebcb1da3b7b 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -364,9 +364,7 @@ static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm,
 static int wm8903_class_w_put(struct snd_kcontrol *kcontrol,
 			      struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
-	struct snd_soc_codec *codec = widget->codec;
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
 	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 	u16 reg;
 	int ret;
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 4c9fb142cb2d..4dfa8dceeabf 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1012,7 +1012,7 @@ static const struct soc_enum liner_enum =
 	SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 0, 2, out_mux_text);
 
 static const struct snd_kcontrol_new liner_mux =
-	SOC_DAPM_ENUM("LINEL Mux", liner_enum);
+	SOC_DAPM_ENUM("LINER Mux", liner_enum);
 
 static const char *sidetone_text[] = {
 	"None", "Left", "Right"
@@ -1202,7 +1202,6 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec)
 		break;
 	}
 
-	snd_soc_dapm_new_widgets(dapm);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 0a4ffdd1d2a7..f156010e52bc 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -263,8 +263,8 @@ SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0),
 SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0),
 SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0),
 
-SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH,
-	0, 127, 0),
+SOC_DOUBLE_R_TLV("ADC PCM Capture Volume", WM8960_LADC, WM8960_RADC,
+	0, 255, 0, adc_tlv),
 
 SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume",
 	       WM8960_BYPASS1, 4, 7, 1, bypass_tlv),
@@ -857,9 +857,9 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 	if (pll_div.k) {
 		reg |= 0x20;
 
-		snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f);
-		snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff);
-		snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0x1ff);
+		snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 16) & 0xff);
+		snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 8) & 0xff);
+		snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0xff);
 	}
 	snd_soc_write(codec, WM8960_PLL1, reg);
 
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index e2de9ecfd641..11d80f3b6137 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2621,8 +2621,6 @@ static int wm8962_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
 
 	wm8962->sysclk_rate = freq;
 
-	wm8962_configure_bclk(codec);
-
 	return 0;
 }
 
@@ -3046,8 +3044,9 @@ static irqreturn_t wm8962_irq(int irq, void *data)
 
 		pm_wakeup_event(dev, 300);
 
-		schedule_delayed_work(&wm8962->mic_work,
-				      msecs_to_jiffies(250));
+		queue_delayed_work(system_power_efficient_wq,
+				   &wm8962->mic_work,
+				   msecs_to_jiffies(250));
 	}
 
 	return IRQ_HANDLED;
@@ -3175,7 +3174,7 @@ static ssize_t wm8962_beep_set(struct device *dev,
 	long int time;
 	int ret;
 
-	ret = strict_strtol(buf, 10, &time);
+	ret = kstrtol(buf, 10, &time);
 	if (ret != 0)
 		return ret;
 
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index ba832b77c543..86426a117b07 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -819,8 +819,9 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w,
 		 * don't want false reports.
 		 */
 		if (wm8994->jackdet && !wm8994->clk_has_run) {
-			schedule_delayed_work(&wm8994->jackdet_bootstrap,
-					      msecs_to_jiffies(1000));
+			queue_delayed_work(system_power_efficient_wq,
+					   &wm8994->jackdet_bootstrap,
+					   msecs_to_jiffies(1000));
 			wm8994->clk_has_run = true;
 		}
 		break;
@@ -1432,14 +1433,12 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING,
 
 #define WM8994_CLASS_W_SWITCH(xname, reg, shift, max, invert) \
 	SOC_SINGLE_EXT(xname, reg, shift, max, invert, \
-		snd_soc_get_volsw, wm8994_put_class_w)
+		snd_soc_dapm_get_volsw, wm8994_put_class_w)
 
 static int wm8994_put_class_w(struct snd_kcontrol *kcontrol,
 			      struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *w = wlist->widgets[0];
-	struct snd_soc_codec *codec = w->codec;
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
 	int ret;
 
 	ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol);
@@ -3487,7 +3486,8 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data)
 
 	pm_wakeup_event(codec->dev, 300);
 
-	schedule_delayed_work(&priv->mic_work, msecs_to_jiffies(250));
+	queue_delayed_work(system_power_efficient_wq,
+			   &priv->mic_work, msecs_to_jiffies(250));
 
 	return IRQ_HANDLED;
 }
@@ -3575,8 +3575,9 @@ static void wm8958_mic_id(void *data, u16 status)
 		/* If nothing present then clear our statuses */
 		dev_dbg(codec->dev, "Detected open circuit\n");
 
-		schedule_delayed_work(&wm8994->open_circuit_work,
-				      msecs_to_jiffies(2500));
+		queue_delayed_work(system_power_efficient_wq,
+				   &wm8994->open_circuit_work,
+				   msecs_to_jiffies(2500));
 		return;
 	}
 
@@ -3690,8 +3691,9 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data)
 				    WM1811_JACKDET_DB, 0);
 
 		delay = control->pdata.micdet_delay;
-		schedule_delayed_work(&wm8994->mic_work,
-				      msecs_to_jiffies(delay));
+		queue_delayed_work(system_power_efficient_wq,
+				   &wm8994->mic_work,
+				   msecs_to_jiffies(delay));
 	} else {
 		dev_dbg(codec->dev, "Jack not detected\n");
 
@@ -3936,8 +3938,9 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
 	id_delay = wm8994->wm8994->pdata.mic_id_delay;
 
 	if (wm8994->mic_detecting)
-		schedule_delayed_work(&wm8994->mic_complete_work,
-				      msecs_to_jiffies(id_delay));
+		queue_delayed_work(system_power_efficient_wq,
+				   &wm8994->mic_complete_work,
+				   msecs_to_jiffies(id_delay));
 	else
 		wm8958_button_det(codec, reg);
 
@@ -4010,9 +4013,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 
 	wm8994->micdet_irq = control->pdata.micdet_irq;
 
-	pm_runtime_enable(codec->dev);
-	pm_runtime_idle(codec->dev);
-
 	/* By default use idle_bias_off, will override for WM8994 */
 	codec->dapm.idle_bias_off = 1;
 
@@ -4385,8 +4385,6 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec)
 
 	wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
-	pm_runtime_disable(codec->dev);
-
 	for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++)
 		wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FLL1_LOCK + i,
 				&wm8994->fll_locked[i]);
@@ -4445,6 +4443,9 @@ static int wm8994_probe(struct platform_device *pdev)
 
 	wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent);
 
+	pm_runtime_enable(&pdev->dev);
+	pm_runtime_idle(&pdev->dev);
+
 	return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8994,
 			wm8994_dai, ARRAY_SIZE(wm8994_dai));
 }
@@ -4452,6 +4453,8 @@ static int wm8994_probe(struct platform_device *pdev)
 static int wm8994_remove(struct platform_device *pdev)
 {
 	snd_soc_unregister_codec(&pdev->dev);
+	pm_runtime_disable(&pdev->dev);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 90a65c427541..da2899e6c401 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -549,12 +549,9 @@ static int check_clk_sys(struct snd_soc_dapm_widget *source,
 static int wm8995_put_class_w(struct snd_kcontrol *kcontrol,
 			      struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *w = wlist->widgets[0];
-	struct snd_soc_codec *codec;
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
 	int ret;
 
-	codec = w->codec;
 	ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol);
 	wm8995_update_class_w(codec);
 	return ret;
diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c
new file mode 100644
index 000000000000..6ec3de3efa4f
--- /dev/null
+++ b/sound/soc/codecs/wm8997.c
@@ -0,0 +1,1175 @@
+/*
+ * wm8997.c  --  WM8997 ALSA SoC Audio driver
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include <linux/mfd/arizona/core.h>
+#include <linux/mfd/arizona/registers.h>
+
+#include "arizona.h"
+#include "wm8997.h"
+
+struct wm8997_priv {
+	struct arizona_priv core;
+	struct arizona_fll fll[2];
+};
+
+static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
+static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
+static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0);
+static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0);
+
+static const struct reg_default wm8997_sysclk_reva_patch[] = {
+	{ 0x301D, 0x7B15 },
+	{ 0x301B, 0x0050 },
+	{ 0x305D, 0x7B17 },
+	{ 0x305B, 0x0050 },
+	{ 0x3001, 0x08FE },
+	{ 0x3003, 0x00F4 },
+	{ 0x3041, 0x08FF },
+	{ 0x3043, 0x0005 },
+	{ 0x3020, 0x0225 },
+	{ 0x3021, 0x0A00 },
+	{ 0x3022, 0xE24D },
+	{ 0x3023, 0x0800 },
+	{ 0x3024, 0xE24D },
+	{ 0x3025, 0xF000 },
+	{ 0x3060, 0x0226 },
+	{ 0x3061, 0x0A00 },
+	{ 0x3062, 0xE252 },
+	{ 0x3063, 0x0800 },
+	{ 0x3064, 0xE252 },
+	{ 0x3065, 0xF000 },
+	{ 0x3116, 0x022B },
+	{ 0x3117, 0xFA00 },
+	{ 0x3110, 0x246C },
+	{ 0x3111, 0x0A03 },
+	{ 0x3112, 0x246E },
+	{ 0x3113, 0x0A03 },
+	{ 0x3114, 0x2470 },
+	{ 0x3115, 0x0A03 },
+	{ 0x3126, 0x246C },
+	{ 0x3127, 0x0A02 },
+	{ 0x3128, 0x246E },
+	{ 0x3129, 0x0A02 },
+	{ 0x312A, 0x2470 },
+	{ 0x312B, 0xFA02 },
+	{ 0x3125, 0x0800 },
+};
+
+static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+	struct regmap *regmap = codec->control_data;
+	const struct reg_default *patch = NULL;
+	int i, patch_size;
+
+	switch (arizona->rev) {
+	case 0:
+		patch = wm8997_sysclk_reva_patch;
+		patch_size = ARRAY_SIZE(wm8997_sysclk_reva_patch);
+		break;
+	default:
+		break;
+	}
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		if (patch)
+			for (i = 0; i < patch_size; i++)
+				regmap_write(regmap, patch[i].reg,
+					     patch[i].def);
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static const char *wm8997_osr_text[] = {
+	"Low power", "Normal", "High performance",
+};
+
+static const unsigned int wm8997_osr_val[] = {
+	0x0, 0x3, 0x5,
+};
+
+static const struct soc_enum wm8997_hpout_osr[] = {
+	SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L,
+			      ARIZONA_OUT1_OSR_SHIFT, 0x7, 3,
+			      wm8997_osr_text, wm8997_osr_val),
+	SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L,
+			      ARIZONA_OUT3_OSR_SHIFT, 0x7, 3,
+			      wm8997_osr_text, wm8997_osr_val),
+};
+
+#define WM8997_NG_SRC(name, base) \
+	SOC_SINGLE(name " NG HPOUT1L Switch",  base, 0, 1, 0), \
+	SOC_SINGLE(name " NG HPOUT1R Switch",  base, 1, 1, 0), \
+	SOC_SINGLE(name " NG EPOUT Switch",    base, 4, 1, 0), \
+	SOC_SINGLE(name " NG SPKOUT Switch",   base, 6, 1, 0), \
+	SOC_SINGLE(name " NG SPKDAT1L Switch", base, 8, 1, 0), \
+	SOC_SINGLE(name " NG SPKDAT1R Switch", base, 9, 1, 0)
+
+static const struct snd_kcontrol_new wm8997_snd_controls[] = {
+SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL,
+	   ARIZONA_IN1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL,
+	   ARIZONA_IN2_OSR_SHIFT, 1, 0),
+
+SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL,
+		     ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL,
+		     ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL,
+		     ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL,
+		     ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+
+SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L,
+	       ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R,
+	       ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+	       ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R,
+	       ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv),
+
+SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp),
+SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp),
+
+ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES_MASK("EQ1 Coefficeints", ARIZONA_EQ1_1, 21,
+		   ARIZONA_EQ1_ENA_MASK),
+SND_SOC_BYTES_MASK("EQ2 Coefficeints", ARIZONA_EQ2_1, 21,
+		   ARIZONA_EQ2_ENA_MASK),
+SND_SOC_BYTES_MASK("EQ3 Coefficeints", ARIZONA_EQ3_1, 21,
+		   ARIZONA_EQ3_ENA_MASK),
+SND_SOC_BYTES_MASK("EQ4 Coefficeints", ARIZONA_EQ4_1, 21,
+		   ARIZONA_EQ4_ENA_MASK),
+
+SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5,
+		   ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA),
+
+ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE),
+
+SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode),
+SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode),
+SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode),
+SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode),
+
+SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1),
+SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1),
+SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1),
+SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1),
+
+SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]),
+SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]),
+
+ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR,
+	       ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv),
+
+ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUT", ARIZONA_OUT4LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
+	   ARIZONA_OUT4_OSR_SHIFT, 1, 0),
+SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
+	   ARIZONA_OUT5_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
+SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	   ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
+SOC_SINGLE("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+	   ARIZONA_OUT4L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	       ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+	       ARIZONA_OUT4L_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+
+SOC_VALUE_ENUM("HPOUT1 OSR", wm8997_hpout_osr[0]),
+SOC_VALUE_ENUM("EPOUT OSR", wm8997_hpout_osr[1]),
+
+SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp),
+SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp),
+
+SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
+	   ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
+
+SOC_SINGLE("Noise Gate Switch", ARIZONA_NOISE_GATE_CONTROL,
+	   ARIZONA_NGATE_ENA_SHIFT, 1, 0),
+SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL,
+	       ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv),
+SOC_ENUM("Noise Gate Hold", arizona_ng_hold),
+
+WM8997_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L),
+WM8997_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R),
+WM8997_NG_SRC("EPOUT", ARIZONA_NOISE_GATE_SELECT_3L),
+WM8997_NG_SRC("SPKOUT", ARIZONA_NOISE_GATE_SELECT_4L),
+WM8997_NG_SRC("SPKDAT1L", ARIZONA_NOISE_GATE_SELECT_5L),
+WM8997_NG_SRC("SPKDAT1R", ARIZONA_NOISE_GATE_SELECT_5R),
+
+ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("SLIMTX1", ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SLIMTX2", ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SLIMTX3", ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SLIMTX4", ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE),
+};
+
+ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUT, ARIZONA_OUT4LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(SLIMTX1, ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SLIMTX2, ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SLIMTX3, ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SLIMTX4, ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SLIMTX5, ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SLIMTX6, ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SLIMTX7, ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SLIMTX8, ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC2INT1, ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC2INT2, ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC2DEC1, ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC2DEC2, ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE);
+
+static const char *wm8997_aec_loopback_texts[] = {
+	"HPOUT1L", "HPOUT1R", "EPOUT", "SPKOUT", "SPKDAT1L", "SPKDAT1R",
+};
+
+static const unsigned int wm8997_aec_loopback_values[] = {
+	0, 1, 4, 6, 8, 9,
+};
+
+static const struct soc_enum wm8997_aec_loopback =
+	SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1,
+			      ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf,
+			      ARRAY_SIZE(wm8997_aec_loopback_texts),
+			      wm8997_aec_loopback_texts,
+			      wm8997_aec_loopback_values);
+
+static const struct snd_kcontrol_new wm8997_aec_loopback_mux =
+	SOC_DAPM_VALUE_ENUM("AEC Loopback", wm8997_aec_loopback);
+
+static const struct snd_soc_dapm_widget wm8997_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT,
+		    0, wm8997_sysclk_ev, SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1,
+		    ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK,
+		    ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK,
+		    ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDD", 0, 0),
+
+SND_SOC_DAPM_SIGGEN("TONE"),
+SND_SOC_DAPM_SIGGEN("NOISE"),
+SND_SOC_DAPM_SIGGEN("HAPTICS"),
+
+SND_SOC_DAPM_INPUT("IN1L"),
+SND_SOC_DAPM_INPUT("IN1R"),
+SND_SOC_DAPM_INPUT("IN2L"),
+SND_SOC_DAPM_INPUT("IN2R"),
+
+SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+
+SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1,
+		    ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2,
+		    ARIZONA_MICB2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3,
+		    ARIZONA_MICB3_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR,
+		 ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1,
+		 ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1,
+		 ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1,
+		 ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT,
+		 0, NULL, 0),
+SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT,
+		 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1INT2", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC1DEC1", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_DEC0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1DEC2", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC2INT1", ARIZONA_ISRC_2_CTRL_3,
+		 ARIZONA_ISRC2_INT0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2INT2", ARIZONA_ISRC_2_CTRL_3,
+		 ARIZONA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC2DEC1", ARIZONA_ISRC_2_CTRL_3,
+		 ARIZONA_ISRC2_DEC0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3,
+		 ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("SLIMTX1", NULL, 0,
+		     ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     ARIZONA_SLIMTX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX2", NULL, 0,
+		     ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     ARIZONA_SLIMTX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX3", NULL, 0,
+		     ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     ARIZONA_SLIMTX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX4", NULL, 0,
+		     ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     ARIZONA_SLIMTX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX5", NULL, 0,
+		     ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     ARIZONA_SLIMTX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX6", NULL, 0,
+		     ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     ARIZONA_SLIMTX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX7", NULL, 0,
+		     ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     ARIZONA_SLIMTX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("SLIMTX8", NULL, 0,
+		     ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE,
+		     ARIZONA_SLIMTX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0,
+		    ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    ARIZONA_SLIMRX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX2", NULL, 0,
+		    ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    ARIZONA_SLIMRX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX3", NULL, 0,
+		    ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    ARIZONA_SLIMRX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX4", NULL, 0,
+		    ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    ARIZONA_SLIMRX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX5", NULL, 0,
+		    ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    ARIZONA_SLIMRX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX6", NULL, 0,
+		    ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    ARIZONA_SLIMRX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX7", NULL, 0,
+		    ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    ARIZONA_SLIMRX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0,
+		    ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
+		    ARIZONA_SLIMRX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
+		       ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+		       &wm8997_aec_loopback_mux),
+
+SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM,
+		   ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM,
+		   ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"),
+ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"),
+ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"),
+ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"),
+
+ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"),
+ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"),
+
+ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"),
+ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"),
+ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"),
+ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"),
+
+ARIZONA_MIXER_WIDGETS(Mic, "Mic"),
+ARIZONA_MIXER_WIDGETS(Noise, "Noise"),
+
+ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"),
+ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"),
+
+ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"),
+ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"),
+ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"),
+ARIZONA_MIXER_WIDGETS(SPKOUT, "SPKOUT"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"),
+
+ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"),
+ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"),
+ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"),
+ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"),
+ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"),
+ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"),
+ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"),
+ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"),
+
+ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"),
+ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"),
+
+ARIZONA_MIXER_WIDGETS(SLIMTX1, "SLIMTX1"),
+ARIZONA_MIXER_WIDGETS(SLIMTX2, "SLIMTX2"),
+ARIZONA_MIXER_WIDGETS(SLIMTX3, "SLIMTX3"),
+ARIZONA_MIXER_WIDGETS(SLIMTX4, "SLIMTX4"),
+ARIZONA_MIXER_WIDGETS(SLIMTX5, "SLIMTX5"),
+ARIZONA_MIXER_WIDGETS(SLIMTX6, "SLIMTX6"),
+ARIZONA_MIXER_WIDGETS(SLIMTX7, "SLIMTX7"),
+ARIZONA_MIXER_WIDGETS(SLIMTX8, "SLIMTX8"),
+
+ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"),
+ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"),
+
+ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"),
+ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"),
+
+ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"),
+ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"),
+
+ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"),
+ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"),
+
+SND_SOC_DAPM_OUTPUT("HPOUT1L"),
+SND_SOC_DAPM_OUTPUT("HPOUT1R"),
+SND_SOC_DAPM_OUTPUT("EPOUTN"),
+SND_SOC_DAPM_OUTPUT("EPOUTP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTP"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1L"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1R"),
+
+SND_SOC_DAPM_OUTPUT("MICSUPP"),
+};
+
+#define ARIZONA_MIXER_INPUT_ROUTES(name)	\
+	{ name, "Noise Generator", "Noise Generator" }, \
+	{ name, "Tone Generator 1", "Tone Generator 1" }, \
+	{ name, "Tone Generator 2", "Tone Generator 2" }, \
+	{ name, "Haptics", "HAPTICS" }, \
+	{ name, "AEC", "AEC Loopback" }, \
+	{ name, "IN1L", "IN1L PGA" }, \
+	{ name, "IN1R", "IN1R PGA" }, \
+	{ name, "IN2L", "IN2L PGA" }, \
+	{ name, "IN2R", "IN2R PGA" }, \
+	{ name, "Mic Mute Mixer", "Mic Mute Mixer" }, \
+	{ name, "AIF1RX1", "AIF1RX1" }, \
+	{ name, "AIF1RX2", "AIF1RX2" }, \
+	{ name, "AIF1RX3", "AIF1RX3" }, \
+	{ name, "AIF1RX4", "AIF1RX4" }, \
+	{ name, "AIF1RX5", "AIF1RX5" }, \
+	{ name, "AIF1RX6", "AIF1RX6" }, \
+	{ name, "AIF1RX7", "AIF1RX7" }, \
+	{ name, "AIF1RX8", "AIF1RX8" }, \
+	{ name, "AIF2RX1", "AIF2RX1" }, \
+	{ name, "AIF2RX2", "AIF2RX2" }, \
+	{ name, "SLIMRX1", "SLIMRX1" }, \
+	{ name, "SLIMRX2", "SLIMRX2" }, \
+	{ name, "SLIMRX3", "SLIMRX3" }, \
+	{ name, "SLIMRX4", "SLIMRX4" }, \
+	{ name, "SLIMRX5", "SLIMRX5" }, \
+	{ name, "SLIMRX6", "SLIMRX6" }, \
+	{ name, "SLIMRX7", "SLIMRX7" }, \
+	{ name, "SLIMRX8", "SLIMRX8" }, \
+	{ name, "EQ1", "EQ1" }, \
+	{ name, "EQ2", "EQ2" }, \
+	{ name, "EQ3", "EQ3" }, \
+	{ name, "EQ4", "EQ4" }, \
+	{ name, "DRC1L", "DRC1L" }, \
+	{ name, "DRC1R", "DRC1R" }, \
+	{ name, "LHPF1", "LHPF1" }, \
+	{ name, "LHPF2", "LHPF2" }, \
+	{ name, "LHPF3", "LHPF3" }, \
+	{ name, "LHPF4", "LHPF4" }, \
+	{ name, "ISRC1DEC1", "ISRC1DEC1" }, \
+	{ name, "ISRC1DEC2", "ISRC1DEC2" }, \
+	{ name, "ISRC1INT1", "ISRC1INT1" }, \
+	{ name, "ISRC1INT2", "ISRC1INT2" }, \
+	{ name, "ISRC2DEC1", "ISRC2DEC1" }, \
+	{ name, "ISRC2DEC2", "ISRC2DEC2" }, \
+	{ name, "ISRC2INT1", "ISRC2INT1" }, \
+	{ name, "ISRC2INT2", "ISRC2INT2" }
+
+static const struct snd_soc_dapm_route wm8997_dapm_routes[] = {
+	{ "AIF2 Capture", NULL, "DBVDD2" },
+	{ "AIF2 Playback", NULL, "DBVDD2" },
+
+	{ "OUT1L", NULL, "CPVDD" },
+	{ "OUT1R", NULL, "CPVDD" },
+	{ "OUT3L", NULL, "CPVDD" },
+
+	{ "OUT4L", NULL, "SPKVDD" },
+
+	{ "OUT1L", NULL, "SYSCLK" },
+	{ "OUT1R", NULL, "SYSCLK" },
+	{ "OUT3L", NULL, "SYSCLK" },
+	{ "OUT4L", NULL, "SYSCLK" },
+
+	{ "IN1L", NULL, "SYSCLK" },
+	{ "IN1R", NULL, "SYSCLK" },
+	{ "IN2L", NULL, "SYSCLK" },
+	{ "IN2R", NULL, "SYSCLK" },
+
+	{ "MICBIAS1", NULL, "MICVDD" },
+	{ "MICBIAS2", NULL, "MICVDD" },
+	{ "MICBIAS3", NULL, "MICVDD" },
+
+	{ "Noise Generator", NULL, "SYSCLK" },
+	{ "Tone Generator 1", NULL, "SYSCLK" },
+	{ "Tone Generator 2", NULL, "SYSCLK" },
+
+	{ "Noise Generator", NULL, "NOISE" },
+	{ "Tone Generator 1", NULL, "TONE" },
+	{ "Tone Generator 2", NULL, "TONE" },
+
+	{ "AIF1 Capture", NULL, "AIF1TX1" },
+	{ "AIF1 Capture", NULL, "AIF1TX2" },
+	{ "AIF1 Capture", NULL, "AIF1TX3" },
+	{ "AIF1 Capture", NULL, "AIF1TX4" },
+	{ "AIF1 Capture", NULL, "AIF1TX5" },
+	{ "AIF1 Capture", NULL, "AIF1TX6" },
+	{ "AIF1 Capture", NULL, "AIF1TX7" },
+	{ "AIF1 Capture", NULL, "AIF1TX8" },
+
+	{ "AIF1RX1", NULL, "AIF1 Playback" },
+	{ "AIF1RX2", NULL, "AIF1 Playback" },
+	{ "AIF1RX3", NULL, "AIF1 Playback" },
+	{ "AIF1RX4", NULL, "AIF1 Playback" },
+	{ "AIF1RX5", NULL, "AIF1 Playback" },
+	{ "AIF1RX6", NULL, "AIF1 Playback" },
+	{ "AIF1RX7", NULL, "AIF1 Playback" },
+	{ "AIF1RX8", NULL, "AIF1 Playback" },
+
+	{ "AIF2 Capture", NULL, "AIF2TX1" },
+	{ "AIF2 Capture", NULL, "AIF2TX2" },
+
+	{ "AIF2RX1", NULL, "AIF2 Playback" },
+	{ "AIF2RX2", NULL, "AIF2 Playback" },
+
+	{ "Slim1 Capture", NULL, "SLIMTX1" },
+	{ "Slim1 Capture", NULL, "SLIMTX2" },
+	{ "Slim1 Capture", NULL, "SLIMTX3" },
+	{ "Slim1 Capture", NULL, "SLIMTX4" },
+
+	{ "SLIMRX1", NULL, "Slim1 Playback" },
+	{ "SLIMRX2", NULL, "Slim1 Playback" },
+	{ "SLIMRX3", NULL, "Slim1 Playback" },
+	{ "SLIMRX4", NULL, "Slim1 Playback" },
+
+	{ "Slim2 Capture", NULL, "SLIMTX5" },
+	{ "Slim2 Capture", NULL, "SLIMTX6" },
+
+	{ "SLIMRX5", NULL, "Slim2 Playback" },
+	{ "SLIMRX6", NULL, "Slim2 Playback" },
+
+	{ "Slim3 Capture", NULL, "SLIMTX7" },
+	{ "Slim3 Capture", NULL, "SLIMTX8" },
+
+	{ "SLIMRX7", NULL, "Slim3 Playback" },
+	{ "SLIMRX8", NULL, "Slim3 Playback" },
+
+	{ "AIF1 Playback", NULL, "SYSCLK" },
+	{ "AIF2 Playback", NULL, "SYSCLK" },
+	{ "Slim1 Playback", NULL, "SYSCLK" },
+	{ "Slim2 Playback", NULL, "SYSCLK" },
+	{ "Slim3 Playback", NULL, "SYSCLK" },
+
+	{ "AIF1 Capture", NULL, "SYSCLK" },
+	{ "AIF2 Capture", NULL, "SYSCLK" },
+	{ "Slim1 Capture", NULL, "SYSCLK" },
+	{ "Slim2 Capture", NULL, "SYSCLK" },
+	{ "Slim3 Capture", NULL, "SYSCLK" },
+
+	{ "IN1L PGA", NULL, "IN1L" },
+	{ "IN1R PGA", NULL, "IN1R" },
+
+	{ "IN2L PGA", NULL, "IN2L" },
+	{ "IN2R PGA", NULL, "IN2R" },
+
+	ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
+	ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
+	ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"),
+
+	ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUT"),
+	ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"),
+	ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"),
+
+	ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"),
+	ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"),
+
+	ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"),
+	ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"),
+	ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"),
+	ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"),
+	ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"),
+	ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"),
+	ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"),
+	ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"),
+
+	ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"),
+	ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"),
+
+	ARIZONA_MIXER_ROUTES("SLIMTX1", "SLIMTX1"),
+	ARIZONA_MIXER_ROUTES("SLIMTX2", "SLIMTX2"),
+	ARIZONA_MIXER_ROUTES("SLIMTX3", "SLIMTX3"),
+	ARIZONA_MIXER_ROUTES("SLIMTX4", "SLIMTX4"),
+	ARIZONA_MIXER_ROUTES("SLIMTX5", "SLIMTX5"),
+	ARIZONA_MIXER_ROUTES("SLIMTX6", "SLIMTX6"),
+	ARIZONA_MIXER_ROUTES("SLIMTX7", "SLIMTX7"),
+	ARIZONA_MIXER_ROUTES("SLIMTX8", "SLIMTX8"),
+
+	ARIZONA_MIXER_ROUTES("EQ1", "EQ1"),
+	ARIZONA_MIXER_ROUTES("EQ2", "EQ2"),
+	ARIZONA_MIXER_ROUTES("EQ3", "EQ3"),
+	ARIZONA_MIXER_ROUTES("EQ4", "EQ4"),
+
+	ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"),
+	ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"),
+
+	ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"),
+	ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"),
+	ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"),
+	ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"),
+
+	ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"),
+	ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"),
+
+	ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"),
+	ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC2INT2"),
+
+	ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"),
+	ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"),
+
+	ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"),
+	ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"),
+
+	ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"),
+	ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"),
+
+	{ "AEC Loopback", "HPOUT1L", "OUT1L" },
+	{ "AEC Loopback", "HPOUT1R", "OUT1R" },
+	{ "HPOUT1L", NULL, "OUT1L" },
+	{ "HPOUT1R", NULL, "OUT1R" },
+
+	{ "AEC Loopback", "EPOUT", "OUT3L" },
+	{ "EPOUTN", NULL, "OUT3L" },
+	{ "EPOUTP", NULL, "OUT3L" },
+
+	{ "AEC Loopback", "SPKOUT", "OUT4L" },
+	{ "SPKOUTN", NULL, "OUT4L" },
+	{ "SPKOUTP", NULL, "OUT4L" },
+
+	{ "AEC Loopback", "SPKDAT1L", "OUT5L" },
+	{ "AEC Loopback", "SPKDAT1R", "OUT5R" },
+	{ "SPKDAT1L", NULL, "OUT5L" },
+	{ "SPKDAT1R", NULL, "OUT5R" },
+
+	{ "MICSUPP", NULL, "SYSCLK" },
+};
+
+static int wm8997_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
+			  unsigned int Fref, unsigned int Fout)
+{
+	struct wm8997_priv *wm8997 = snd_soc_codec_get_drvdata(codec);
+
+	switch (fll_id) {
+	case WM8997_FLL1:
+		return arizona_set_fll(&wm8997->fll[0], source, Fref, Fout);
+	case WM8997_FLL2:
+		return arizona_set_fll(&wm8997->fll[1], source, Fref, Fout);
+	case WM8997_FLL1_REFCLK:
+		return arizona_set_fll_refclk(&wm8997->fll[0], source, Fref,
+					      Fout);
+	case WM8997_FLL2_REFCLK:
+		return arizona_set_fll_refclk(&wm8997->fll[1], source, Fref,
+					      Fout);
+	default:
+		return -EINVAL;
+	}
+}
+
+#define WM8997_RATES SNDRV_PCM_RATE_8000_192000
+
+#define WM8997_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 wm8997_dai[] = {
+	{
+		.name = "wm8997-aif1",
+		.id = 1,
+		.base = ARIZONA_AIF1_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF1 Playback",
+			.channels_min = 1,
+			.channels_max = 8,
+			.rates = WM8997_RATES,
+			.formats = WM8997_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF1 Capture",
+			 .channels_min = 1,
+			 .channels_max = 8,
+			 .rates = WM8997_RATES,
+			 .formats = WM8997_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = "wm8997-aif2",
+		.id = 2,
+		.base = ARIZONA_AIF2_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF2 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM8997_RATES,
+			.formats = WM8997_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF2 Capture",
+			 .channels_min = 1,
+			 .channels_max = 2,
+			 .rates = WM8997_RATES,
+			 .formats = WM8997_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = "wm8997-slim1",
+		.id = 3,
+		.playback = {
+			.stream_name = "Slim1 Playback",
+			.channels_min = 1,
+			.channels_max = 4,
+			.rates = WM8997_RATES,
+			.formats = WM8997_FORMATS,
+		},
+		.capture = {
+			.stream_name = "Slim1 Capture",
+			.channels_min = 1,
+			.channels_max = 4,
+			.rates = WM8997_RATES,
+			.formats = WM8997_FORMATS,
+		},
+		.ops = &arizona_simple_dai_ops,
+	},
+	{
+		.name = "wm8997-slim2",
+		.id = 4,
+		.playback = {
+			.stream_name = "Slim2 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM8997_RATES,
+			.formats = WM8997_FORMATS,
+		},
+		.capture = {
+			.stream_name = "Slim2 Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM8997_RATES,
+			.formats = WM8997_FORMATS,
+		},
+		.ops = &arizona_simple_dai_ops,
+	},
+	{
+		.name = "wm8997-slim3",
+		.id = 5,
+		.playback = {
+			.stream_name = "Slim3 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM8997_RATES,
+			.formats = WM8997_FORMATS,
+		},
+		.capture = {
+			.stream_name = "Slim3 Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM8997_RATES,
+			.formats = WM8997_FORMATS,
+		},
+		.ops = &arizona_simple_dai_ops,
+	},
+};
+
+static int wm8997_codec_probe(struct snd_soc_codec *codec)
+{
+	struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	codec->control_data = priv->core.arizona->regmap;
+
+	ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP);
+	if (ret != 0)
+		return ret;
+
+	arizona_init_spk(codec);
+
+	snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
+
+	priv->core.arizona->dapm = &codec->dapm;
+
+	return 0;
+}
+
+static int wm8997_codec_remove(struct snd_soc_codec *codec)
+{
+	struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	priv->core.arizona->dapm = NULL;
+
+	return 0;
+}
+
+#define WM8997_DIG_VU 0x0200
+
+static unsigned int wm8997_digital_vu[] = {
+	ARIZONA_DAC_DIGITAL_VOLUME_1L,
+	ARIZONA_DAC_DIGITAL_VOLUME_1R,
+	ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	ARIZONA_DAC_DIGITAL_VOLUME_4L,
+	ARIZONA_DAC_DIGITAL_VOLUME_5L,
+	ARIZONA_DAC_DIGITAL_VOLUME_5R,
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_wm8997 = {
+	.probe = wm8997_codec_probe,
+	.remove = wm8997_codec_remove,
+
+	.idle_bias_off = true,
+
+	.set_sysclk = arizona_set_sysclk,
+	.set_pll = wm8997_set_fll,
+
+	.controls = wm8997_snd_controls,
+	.num_controls = ARRAY_SIZE(wm8997_snd_controls),
+	.dapm_widgets = wm8997_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm8997_dapm_widgets),
+	.dapm_routes = wm8997_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(wm8997_dapm_routes),
+};
+
+static int wm8997_probe(struct platform_device *pdev)
+{
+	struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
+	struct wm8997_priv *wm8997;
+	int i;
+
+	wm8997 = devm_kzalloc(&pdev->dev, sizeof(struct wm8997_priv),
+			      GFP_KERNEL);
+	if (wm8997 == NULL)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, wm8997);
+
+	wm8997->core.arizona = arizona;
+	wm8997->core.num_inputs = 4;
+
+	for (i = 0; i < ARRAY_SIZE(wm8997->fll); i++)
+		wm8997->fll[i].vco_mult = 1;
+
+	arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1,
+			 ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK,
+			 &wm8997->fll[0]);
+	arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1,
+			 ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK,
+			 &wm8997->fll[1]);
+
+	/* SR2 fixed at 8kHz, SR3 fixed at 16kHz */
+	regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_2,
+			   ARIZONA_SAMPLE_RATE_2_MASK, 0x11);
+	regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_3,
+			   ARIZONA_SAMPLE_RATE_3_MASK, 0x12);
+
+	for (i = 0; i < ARRAY_SIZE(wm8997_dai); i++)
+		arizona_init_dai(&wm8997->core, i);
+
+	/* Latch volume update bits */
+	for (i = 0; i < ARRAY_SIZE(wm8997_digital_vu); i++)
+		regmap_update_bits(arizona->regmap, wm8997_digital_vu[i],
+				   WM8997_DIG_VU, WM8997_DIG_VU);
+
+	pm_runtime_enable(&pdev->dev);
+	pm_runtime_idle(&pdev->dev);
+
+	return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8997,
+				      wm8997_dai, ARRAY_SIZE(wm8997_dai));
+}
+
+static int wm8997_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	pm_runtime_disable(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver wm8997_codec_driver = {
+	.driver = {
+		.name = "wm8997-codec",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm8997_probe,
+	.remove = wm8997_remove,
+};
+
+module_platform_driver(wm8997_codec_driver);
+
+MODULE_DESCRIPTION("ASoC WM8997 driver");
+MODULE_AUTHOR("Charles Keepax <ckeepax@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm8997-codec");
diff --git a/sound/soc/codecs/wm8997.h b/sound/soc/codecs/wm8997.h
new file mode 100644
index 000000000000..5e91c6a7d567
--- /dev/null
+++ b/sound/soc/codecs/wm8997.h
@@ -0,0 +1,23 @@
+/*
+ * wm8997.h  --  WM8997 ALSA SoC Audio driver
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8997_H
+#define _WM8997_H
+
+#include "arizona.h"
+
+#define WM8997_FLL1        1
+#define WM8997_FLL2        2
+#define WM8997_FLL1_REFCLK 3
+#define WM8997_FLL2_REFCLK 4
+
+#endif
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 05252ac936a3..b38f3506418f 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -225,15 +225,8 @@ struct wm_coeff_ctl_ops {
 		     struct snd_ctl_elem_info *uinfo);
 };
 
-struct wm_coeff {
-	struct device *dev;
-	struct list_head ctl_list;
-	struct regmap *regmap;
-};
-
 struct wm_coeff_ctl {
 	const char *name;
-	struct snd_card *card;
 	struct wm_adsp_alg_region region;
 	struct wm_coeff_ctl_ops ops;
 	struct wm_adsp *adsp;
@@ -378,7 +371,6 @@ static int wm_coeff_info(struct snd_kcontrol *kcontrol,
 static int wm_coeff_write_control(struct snd_kcontrol *kcontrol,
 				  const void *buf, size_t len)
 {
-	struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol);
 	struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value;
 	struct wm_adsp_alg_region *region = &ctl->region;
 	const struct wm_adsp_region *mem;
@@ -401,7 +393,7 @@ static int wm_coeff_write_control(struct snd_kcontrol *kcontrol,
 	if (!scratch)
 		return -ENOMEM;
 
-	ret = regmap_raw_write(wm_coeff->regmap, reg, scratch,
+	ret = regmap_raw_write(adsp->regmap, reg, scratch,
 			       ctl->len);
 	if (ret) {
 		adsp_err(adsp, "Failed to write %zu bytes to %x\n",
@@ -434,7 +426,6 @@ static int wm_coeff_put(struct snd_kcontrol *kcontrol,
 static int wm_coeff_read_control(struct snd_kcontrol *kcontrol,
 				 void *buf, size_t len)
 {
-	struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol);
 	struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value;
 	struct wm_adsp_alg_region *region = &ctl->region;
 	const struct wm_adsp_region *mem;
@@ -457,7 +448,7 @@ static int wm_coeff_read_control(struct snd_kcontrol *kcontrol,
 	if (!scratch)
 		return -ENOMEM;
 
-	ret = regmap_raw_read(wm_coeff->regmap, reg, scratch, ctl->len);
+	ret = regmap_raw_read(adsp->regmap, reg, scratch, ctl->len);
 	if (ret) {
 		adsp_err(adsp, "Failed to read %zu bytes from %x\n",
 			 ctl->len, reg);
@@ -481,37 +472,18 @@ static int wm_coeff_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static int wm_coeff_add_kcontrol(struct wm_coeff *wm_coeff,
-				 struct wm_coeff_ctl *ctl,
-				 const struct snd_kcontrol_new *kctl)
-{
-	int ret;
-	struct snd_kcontrol *kcontrol;
-
-	kcontrol = snd_ctl_new1(kctl, wm_coeff);
-	ret = snd_ctl_add(ctl->card, kcontrol);
-	if (ret < 0) {
-		dev_err(wm_coeff->dev, "Failed to add %s: %d\n",
-			kctl->name, ret);
-		return ret;
-	}
-	ctl->kcontrol = kcontrol;
-	return 0;
-}
-
 struct wmfw_ctl_work {
-	struct wm_coeff *wm_coeff;
+	struct wm_adsp *adsp;
 	struct wm_coeff_ctl *ctl;
 	struct work_struct work;
 };
 
-static int wmfw_add_ctl(struct wm_coeff *wm_coeff,
-			struct wm_coeff_ctl *ctl)
+static int wmfw_add_ctl(struct wm_adsp *adsp, struct wm_coeff_ctl *ctl)
 {
 	struct snd_kcontrol_new *kcontrol;
 	int ret;
 
-	if (!wm_coeff || !ctl || !ctl->name || !ctl->card)
+	if (!ctl || !ctl->name)
 		return -EINVAL;
 
 	kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL);
@@ -525,14 +497,17 @@ static int wmfw_add_ctl(struct wm_coeff *wm_coeff,
 	kcontrol->put = wm_coeff_put;
 	kcontrol->private_value = (unsigned long)ctl;
 
-	ret = wm_coeff_add_kcontrol(wm_coeff,
-				    ctl, kcontrol);
+	ret = snd_soc_add_card_controls(adsp->card,
+					kcontrol, 1);
 	if (ret < 0)
 		goto err_kcontrol;
 
 	kfree(kcontrol);
 
-	list_add(&ctl->list, &wm_coeff->ctl_list);
+	ctl->kcontrol = snd_soc_card_get_kcontrol(adsp->card,
+						  ctl->name);
+
+	list_add(&ctl->list, &adsp->ctl_list);
 	return 0;
 
 err_kcontrol:
@@ -753,13 +728,12 @@ out:
 	return ret;
 }
 
-static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff)
+static int wm_coeff_init_control_caches(struct wm_adsp *adsp)
 {
 	struct wm_coeff_ctl *ctl;
 	int ret;
 
-	list_for_each_entry(ctl, &wm_coeff->ctl_list,
-			    list) {
+	list_for_each_entry(ctl, &adsp->ctl_list, list) {
 		if (!ctl->enabled || ctl->set)
 			continue;
 		ret = wm_coeff_read_control(ctl->kcontrol,
@@ -772,13 +746,12 @@ static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff)
 	return 0;
 }
 
-static int wm_coeff_sync_controls(struct wm_coeff *wm_coeff)
+static int wm_coeff_sync_controls(struct wm_adsp *adsp)
 {
 	struct wm_coeff_ctl *ctl;
 	int ret;
 
-	list_for_each_entry(ctl, &wm_coeff->ctl_list,
-			    list) {
+	list_for_each_entry(ctl, &adsp->ctl_list, list) {
 		if (!ctl->enabled)
 			continue;
 		if (ctl->set) {
@@ -799,15 +772,14 @@ static void wm_adsp_ctl_work(struct work_struct *work)
 						      struct wmfw_ctl_work,
 						      work);
 
-	wmfw_add_ctl(ctl_work->wm_coeff, ctl_work->ctl);
+	wmfw_add_ctl(ctl_work->adsp, ctl_work->ctl);
 	kfree(ctl_work);
 }
 
-static int wm_adsp_create_control(struct snd_soc_codec *codec,
+static int wm_adsp_create_control(struct wm_adsp *dsp,
 				  const struct wm_adsp_alg_region *region)
 
 {
-	struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec);
 	struct wm_coeff_ctl *ctl;
 	struct wmfw_ctl_work *ctl_work;
 	char *name;
@@ -842,7 +814,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec,
 	snprintf(name, PAGE_SIZE, "DSP%d %s %x",
 		 dsp->num, region_name, region->alg);
 
-	list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list,
+	list_for_each_entry(ctl, &dsp->ctl_list,
 			    list) {
 		if (!strcmp(ctl->name, name)) {
 			if (!ctl->enabled)
@@ -866,7 +838,6 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec,
 	ctl->set = 0;
 	ctl->ops.xget = wm_coeff_get;
 	ctl->ops.xput = wm_coeff_put;
-	ctl->card = codec->card->snd_card;
 	ctl->adsp = dsp;
 
 	ctl->len = region->len;
@@ -882,7 +853,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec,
 		goto err_ctl_cache;
 	}
 
-	ctl_work->wm_coeff = dsp->wm_coeff;
+	ctl_work->adsp = dsp;
 	ctl_work->ctl = ctl;
 	INIT_WORK(&ctl_work->work, wm_adsp_ctl_work);
 	schedule_work(&ctl_work->work);
@@ -903,7 +874,7 @@ err_name:
 	return ret;
 }
 
-static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec)
+static int wm_adsp_setup_algs(struct wm_adsp *dsp)
 {
 	struct regmap *regmap = dsp->regmap;
 	struct wmfw_adsp1_id_hdr adsp1_id;
@@ -1091,7 +1062,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec)
 			if (i + 1 < algs) {
 				region->len = be32_to_cpu(adsp1_alg[i + 1].dm);
 				region->len -= be32_to_cpu(adsp1_alg[i].dm);
-				wm_adsp_create_control(codec, region);
+				wm_adsp_create_control(dsp, region);
 			} else {
 				adsp_warn(dsp, "Missing length info for region DM with ID %x\n",
 					  be32_to_cpu(adsp1_alg[i].alg.id));
@@ -1108,7 +1079,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec)
 			if (i + 1 < algs) {
 				region->len = be32_to_cpu(adsp1_alg[i + 1].zm);
 				region->len -= be32_to_cpu(adsp1_alg[i].zm);
-				wm_adsp_create_control(codec, region);
+				wm_adsp_create_control(dsp, region);
 			} else {
 				adsp_warn(dsp, "Missing length info for region ZM with ID %x\n",
 					  be32_to_cpu(adsp1_alg[i].alg.id));
@@ -1137,7 +1108,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec)
 			if (i + 1 < algs) {
 				region->len = be32_to_cpu(adsp2_alg[i + 1].xm);
 				region->len -= be32_to_cpu(adsp2_alg[i].xm);
-				wm_adsp_create_control(codec, region);
+				wm_adsp_create_control(dsp, region);
 			} else {
 				adsp_warn(dsp, "Missing length info for region XM with ID %x\n",
 					  be32_to_cpu(adsp2_alg[i].alg.id));
@@ -1154,7 +1125,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec)
 			if (i + 1 < algs) {
 				region->len = be32_to_cpu(adsp2_alg[i + 1].ym);
 				region->len -= be32_to_cpu(adsp2_alg[i].ym);
-				wm_adsp_create_control(codec, region);
+				wm_adsp_create_control(dsp, region);
 			} else {
 				adsp_warn(dsp, "Missing length info for region YM with ID %x\n",
 					  be32_to_cpu(adsp2_alg[i].alg.id));
@@ -1171,7 +1142,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec)
 			if (i + 1 < algs) {
 				region->len = be32_to_cpu(adsp2_alg[i + 1].zm);
 				region->len -= be32_to_cpu(adsp2_alg[i].zm);
-				wm_adsp_create_control(codec, region);
+				wm_adsp_create_control(dsp, region);
 			} else {
 				adsp_warn(dsp, "Missing length info for region ZM with ID %x\n",
 					  be32_to_cpu(adsp2_alg[i].alg.id));
@@ -1391,6 +1362,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
 	int ret;
 	int val;
 
+	dsp->card = codec->card;
+
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
 		regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30,
@@ -1425,7 +1398,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
 		if (ret != 0)
 			goto err;
 
-		ret = wm_adsp_setup_algs(dsp, codec);
+		ret = wm_adsp_setup_algs(dsp);
 		if (ret != 0)
 			goto err;
 
@@ -1434,12 +1407,12 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
 			goto err;
 
 		/* Initialize caches for enabled and unset controls */
-		ret = wm_coeff_init_control_caches(dsp->wm_coeff);
+		ret = wm_coeff_init_control_caches(dsp);
 		if (ret != 0)
 			goto err;
 
 		/* Sync set controls */
-		ret = wm_coeff_sync_controls(dsp->wm_coeff);
+		ret = wm_coeff_sync_controls(dsp);
 		if (ret != 0)
 			goto err;
 
@@ -1460,10 +1433,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
 		regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30,
 				   ADSP1_SYS_ENA, 0);
 
-		list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list,
-				    list) {
+		list_for_each_entry(ctl, &dsp->ctl_list, list)
 			ctl->enabled = 0;
-		}
 		break;
 
 	default:
@@ -1520,6 +1491,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
 	unsigned int val;
 	int ret;
 
+	dsp->card = codec->card;
+
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
 		/*
@@ -1582,7 +1555,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
 		if (ret != 0)
 			goto err;
 
-		ret = wm_adsp_setup_algs(dsp, codec);
+		ret = wm_adsp_setup_algs(dsp);
 		if (ret != 0)
 			goto err;
 
@@ -1591,12 +1564,12 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
 			goto err;
 
 		/* Initialize caches for enabled and unset controls */
-		ret = wm_coeff_init_control_caches(dsp->wm_coeff);
+		ret = wm_coeff_init_control_caches(dsp);
 		if (ret != 0)
 			goto err;
 
 		/* Sync set controls */
-		ret = wm_coeff_sync_controls(dsp->wm_coeff);
+		ret = wm_coeff_sync_controls(dsp);
 		if (ret != 0)
 			goto err;
 
@@ -1637,10 +1610,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
 					ret);
 		}
 
-		list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list,
-				    list) {
+		list_for_each_entry(ctl, &dsp->ctl_list, list)
 			ctl->enabled = 0;
-		}
 
 		while (!list_empty(&dsp->alg_regions)) {
 			alg_region = list_first_entry(&dsp->alg_regions,
@@ -1679,49 +1650,38 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs)
 	}
 
 	INIT_LIST_HEAD(&adsp->alg_regions);
-
-	adsp->wm_coeff = kzalloc(sizeof(*adsp->wm_coeff),
-				 GFP_KERNEL);
-	if (!adsp->wm_coeff)
-		return -ENOMEM;
-	adsp->wm_coeff->regmap = adsp->regmap;
-	adsp->wm_coeff->dev = adsp->dev;
-	INIT_LIST_HEAD(&adsp->wm_coeff->ctl_list);
+	INIT_LIST_HEAD(&adsp->ctl_list);
 
 	if (dvfs) {
 		adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD");
 		if (IS_ERR(adsp->dvfs)) {
 			ret = PTR_ERR(adsp->dvfs);
 			dev_err(adsp->dev, "Failed to get DCVDD: %d\n", ret);
-			goto out_coeff;
+			return ret;
 		}
 
 		ret = regulator_enable(adsp->dvfs);
 		if (ret != 0) {
 			dev_err(adsp->dev, "Failed to enable DCVDD: %d\n",
 				ret);
-			goto out_coeff;
+			return ret;
 		}
 
 		ret = regulator_set_voltage(adsp->dvfs, 1200000, 1800000);
 		if (ret != 0) {
 			dev_err(adsp->dev, "Failed to initialise DVFS: %d\n",
 				ret);
-			goto out_coeff;
+			return ret;
 		}
 
 		ret = regulator_disable(adsp->dvfs);
 		if (ret != 0) {
 			dev_err(adsp->dev, "Failed to disable DCVDD: %d\n",
 				ret);
-			goto out_coeff;
+			return ret;
 		}
 	}
 
 	return 0;
-
-out_coeff:
-	kfree(adsp->wm_coeff);
-	return ret;
 }
 EXPORT_SYMBOL_GPL(wm_adsp2_init);
diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h
index 9f922c82536c..d018dea6254d 100644
--- a/sound/soc/codecs/wm_adsp.h
+++ b/sound/soc/codecs/wm_adsp.h
@@ -39,6 +39,7 @@ struct wm_adsp {
 	int type;
 	struct device *dev;
 	struct regmap *regmap;
+	struct snd_soc_card *card;
 
 	int base;
 	int sysclk_reg;
@@ -57,7 +58,7 @@ struct wm_adsp {
 
 	struct regulator *dvfs;
 
-	struct wm_coeff *wm_coeff;
+	struct list_head ctl_list;
 };
 
 #define WM_ADSP1(wname, num) \
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 2d9e099415a5..8b50e5958de5 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -699,9 +699,7 @@ EXPORT_SYMBOL_GPL(wm_hubs_update_class_w);
 static int class_w_put_volsw(struct snd_kcontrol *kcontrol,
 			      struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
-	struct snd_soc_codec *codec = widget->codec;
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
 	int ret;
 
 	ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol);
@@ -721,9 +719,7 @@ static int class_w_put_volsw(struct snd_kcontrol *kcontrol,
 static int class_w_put_double(struct snd_kcontrol *kcontrol,
 			      struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
-	struct snd_soc_codec *codec = widget->codec;
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
 	int ret;
 
 	ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol);
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index 70eb37a5dd16..25c31f1655f6 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -421,13 +421,11 @@ static int dw_i2s_probe(struct platform_device *pdev)
 					 dw_i2s_dai, 1);
 	if (ret != 0) {
 		dev_err(&pdev->dev, "not able to register dai\n");
-		goto err_set_drvdata;
+		goto err_clk_disable;
 	}
 
 	return 0;
 
-err_set_drvdata:
-	dev_set_drvdata(&pdev->dev, NULL);
 err_clk_disable:
 	clk_disable(dev->clk);
 err_clk_put:
@@ -440,7 +438,6 @@ static int dw_i2s_remove(struct platform_device *pdev)
 	struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
 
 	snd_soc_unregister_component(&pdev->dev);
-	dev_set_drvdata(&pdev->dev, NULL);
 
 	clk_put(dev->clk);
 
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index aa438546c912..704e246f5b1e 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,6 +1,9 @@
 config SND_SOC_FSL_SSI
 	tristate
 
+config SND_SOC_FSL_SPDIF
+	tristate
+
 config SND_SOC_FSL_UTILS
 	tristate
 
@@ -98,7 +101,7 @@ endif # SND_POWERPC_SOC
 
 menuconfig SND_IMX_SOC
 	tristate "SoC Audio for Freescale i.MX CPUs"
-	depends on ARCH_MXC
+	depends on ARCH_MXC || COMPILE_TEST
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the i.MX CPUs.
@@ -109,11 +112,11 @@ config SND_SOC_IMX_SSI
 	tristate
 
 config SND_SOC_IMX_PCM_FIQ
-	bool
+	tristate
 	select FIQ
 
 config SND_SOC_IMX_PCM_DMA
-	bool
+	tristate
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 
 config SND_SOC_IMX_AUDMUX
@@ -175,7 +178,6 @@ config SND_SOC_IMX_WM8962
 	select SND_SOC_IMX_PCM_DMA
 	select SND_SOC_IMX_AUDMUX
 	select SND_SOC_FSL_SSI
-	select SND_SOC_FSL_UTILS
 	help
 	  Say Y if you want to add support for SoC audio on an i.MX board with
 	  a wm8962 codec.
@@ -187,14 +189,23 @@ config SND_SOC_IMX_SGTL5000
 	select SND_SOC_IMX_PCM_DMA
 	select SND_SOC_IMX_AUDMUX
 	select SND_SOC_FSL_SSI
-	select SND_SOC_FSL_UTILS
 	help
 	  Say Y if you want to add support for SoC audio on an i.MX board with
 	  a sgtl5000 codec.
 
+config SND_SOC_IMX_SPDIF
+	tristate "SoC Audio support for i.MX boards with S/PDIF"
+	select SND_SOC_IMX_PCM_DMA
+	select SND_SOC_FSL_SPDIF
+	select SND_SOC_SPDIF
+	help
+	  SoC Audio support for i.MX boards with S/PDIF
+	  Say Y if you want to add support for SoC audio on an i.MX board with
+	  a S/DPDIF.
+
 config SND_SOC_IMX_MC13783
 	tristate "SoC Audio support for I.MX boards with mc13783"
-	depends on MFD_MC13783
+	depends on MFD_MC13783 && ARM
 	select SND_SOC_IMX_SSI
 	select SND_SOC_IMX_AUDMUX
 	select SND_SOC_MC13783
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index d4b4aa8b5649..8db705b0fdf9 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -12,9 +12,11 @@ obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o
 
 # Freescale PowerPC SSI/DMA Platform Support
 snd-soc-fsl-ssi-objs := fsl_ssi.o
+snd-soc-fsl-spdif-objs := fsl_spdif.o
 snd-soc-fsl-utils-objs := fsl_utils.o
 snd-soc-fsl-dma-objs := fsl_dma.o
 obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
+obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o
 obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o
 obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o
 
@@ -43,6 +45,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
 snd-soc-wm1133-ev1-objs := wm1133-ev1.o
 snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
 snd-soc-imx-wm8962-objs := imx-wm8962.o
+snd-soc-imx-spdif-objs := imx-spdif.o
 snd-soc-imx-mc13783-objs := imx-mc13783.o
 
 obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
@@ -51,4 +54,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
 obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
 obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
 obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
+obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
 obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
new file mode 100644
index 000000000000..3920c3e849ce
--- /dev/null
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -0,0 +1,1225 @@
+/*
+ * Freescale S/PDIF ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Copyright (C) 2013 Freescale Semiconductor, Inc.
+ *
+ * Based on stmp3xxx_spdif_dai.c
+ * Vladimir Barinov <vbarinov@embeddedalley.com>
+ * Copyright 2008 SigmaTel, Inc
+ * Copyright 2008 Embedded Alley Solutions, Inc
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2.  This program  is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/module.h>
+#include <linux/clk.h>
+#include <linux/clk-private.h>
+#include <linux/bitrev.h>
+#include <linux/regmap.h>
+#include <linux/of_address.h>
+#include <linux/of_device.h>
+#include <linux/of_irq.h>
+
+#include <sound/asoundef.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "fsl_spdif.h"
+#include "imx-pcm.h"
+
+#define FSL_SPDIF_TXFIFO_WML	0x8
+#define FSL_SPDIF_RXFIFO_WML	0x8
+
+#define INTR_FOR_PLAYBACK (INT_TXFIFO_RESYNC)
+#define INTR_FOR_CAPTURE (INT_SYM_ERR | INT_BIT_ERR | INT_URX_FUL | INT_URX_OV|\
+		INT_QRX_FUL | INT_QRX_OV | INT_UQ_SYNC | INT_UQ_ERR |\
+		INT_RXFIFO_RESYNC | INT_LOSS_LOCK | INT_DPLL_LOCKED)
+
+/* Index list for the values that has if (DPLL Locked) condition */
+static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb };
+#define SRPC_NODPLL_START1	0x5
+#define SRPC_NODPLL_START2	0xc
+
+#define DEFAULT_RXCLK_SRC	1
+
+/*
+ * SPDIF control structure
+ * Defines channel status, subcode and Q sub
+ */
+struct spdif_mixer_control {
+	/* spinlock to access control data */
+	spinlock_t ctl_lock;
+
+	/* IEC958 channel tx status bit */
+	unsigned char ch_status[4];
+
+	/* User bits */
+	unsigned char subcode[2 * SPDIF_UBITS_SIZE];
+
+	/* Q subcode part of user bits */
+	unsigned char qsub[2 * SPDIF_QSUB_SIZE];
+
+	/* Buffer offset for U/Q */
+	u32 upos;
+	u32 qpos;
+
+	/* Ready buffer index of the two buffers */
+	u32 ready_buf;
+};
+
+struct fsl_spdif_priv {
+	struct spdif_mixer_control fsl_spdif_control;
+	struct snd_soc_dai_driver cpu_dai_drv;
+	struct platform_device *pdev;
+	struct regmap *regmap;
+	bool dpll_locked;
+	u8 txclk_div[SPDIF_TXRATE_MAX];
+	u8 txclk_src[SPDIF_TXRATE_MAX];
+	u8 rxclk_src;
+	struct clk *txclk[SPDIF_TXRATE_MAX];
+	struct clk *rxclk;
+	struct snd_dmaengine_dai_dma_data dma_params_tx;
+	struct snd_dmaengine_dai_dma_data dma_params_rx;
+
+	/* The name space will be allocated dynamically */
+	char name[0];
+};
+
+
+/* DPLL locked and lock loss interrupt handler */
+static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv)
+{
+	struct regmap *regmap = spdif_priv->regmap;
+	struct platform_device *pdev = spdif_priv->pdev;
+	u32 locked;
+
+	regmap_read(regmap, REG_SPDIF_SRPC, &locked);
+	locked &= SRPC_DPLL_LOCKED;
+
+	dev_dbg(&pdev->dev, "isr: Rx dpll %s \n",
+			locked ? "locked" : "loss lock");
+
+	spdif_priv->dpll_locked = locked ? true : false;
+}
+
+/* Receiver found illegal symbol interrupt handler */
+static void spdif_irq_sym_error(struct fsl_spdif_priv *spdif_priv)
+{
+	struct regmap *regmap = spdif_priv->regmap;
+	struct platform_device *pdev = spdif_priv->pdev;
+
+	dev_dbg(&pdev->dev, "isr: receiver found illegal symbol\n");
+
+	if (!spdif_priv->dpll_locked) {
+		/* DPLL unlocked seems no audio stream */
+		regmap_update_bits(regmap, REG_SPDIF_SIE, INT_SYM_ERR, 0);
+	}
+}
+
+/* U/Q Channel receive register full */
+static void spdif_irq_uqrx_full(struct fsl_spdif_priv *spdif_priv, char name)
+{
+	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+	struct regmap *regmap = spdif_priv->regmap;
+	struct platform_device *pdev = spdif_priv->pdev;
+	u32 *pos, size, val, reg;
+
+	switch (name) {
+	case 'U':
+		pos = &ctrl->upos;
+		size = SPDIF_UBITS_SIZE;
+		reg = REG_SPDIF_SRU;
+		break;
+	case 'Q':
+		pos = &ctrl->qpos;
+		size = SPDIF_QSUB_SIZE;
+		reg = REG_SPDIF_SRQ;
+		break;
+	default:
+		dev_err(&pdev->dev, "unsupported channel name\n");
+		return;
+	}
+
+	dev_dbg(&pdev->dev, "isr: %c Channel receive register full\n", name);
+
+	if (*pos >= size * 2) {
+		*pos = 0;
+	} else if (unlikely((*pos % size) + 3 > size)) {
+		dev_err(&pdev->dev, "User bit receivce buffer overflow\n");
+		return;
+	}
+
+	regmap_read(regmap, reg, &val);
+	ctrl->subcode[*pos++] = val >> 16;
+	ctrl->subcode[*pos++] = val >> 8;
+	ctrl->subcode[*pos++] = val;
+}
+
+/* U/Q Channel sync found */
+static void spdif_irq_uq_sync(struct fsl_spdif_priv *spdif_priv)
+{
+	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+	struct platform_device *pdev = spdif_priv->pdev;
+
+	dev_dbg(&pdev->dev, "isr: U/Q Channel sync found\n");
+
+	/* U/Q buffer reset */
+	if (ctrl->qpos == 0)
+		return;
+
+	/* Set ready to this buffer */
+	ctrl->ready_buf = (ctrl->qpos - 1) / SPDIF_QSUB_SIZE + 1;
+}
+
+/* U/Q Channel framing error */
+static void spdif_irq_uq_err(struct fsl_spdif_priv *spdif_priv)
+{
+	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+	struct regmap *regmap = spdif_priv->regmap;
+	struct platform_device *pdev = spdif_priv->pdev;
+	u32 val;
+
+	dev_dbg(&pdev->dev, "isr: U/Q Channel framing error\n");
+
+	/* Read U/Q data to clear the irq and do buffer reset */
+	regmap_read(regmap, REG_SPDIF_SRU, &val);
+	regmap_read(regmap, REG_SPDIF_SRQ, &val);
+
+	/* Drop this U/Q buffer */
+	ctrl->ready_buf = 0;
+	ctrl->upos = 0;
+	ctrl->qpos = 0;
+}
+
+/* Get spdif interrupt status and clear the interrupt */
+static u32 spdif_intr_status_clear(struct fsl_spdif_priv *spdif_priv)
+{
+	struct regmap *regmap = spdif_priv->regmap;
+	u32 val, val2;
+
+	regmap_read(regmap, REG_SPDIF_SIS, &val);
+	regmap_read(regmap, REG_SPDIF_SIE, &val2);
+
+	regmap_write(regmap, REG_SPDIF_SIC, val & val2);
+
+	return val;
+}
+
+static irqreturn_t spdif_isr(int irq, void *devid)
+{
+	struct fsl_spdif_priv *spdif_priv = (struct fsl_spdif_priv *)devid;
+	struct platform_device *pdev = spdif_priv->pdev;
+	u32 sis;
+
+	sis = spdif_intr_status_clear(spdif_priv);
+
+	if (sis & INT_DPLL_LOCKED)
+		spdif_irq_dpll_lock(spdif_priv);
+
+	if (sis & INT_TXFIFO_UNOV)
+		dev_dbg(&pdev->dev, "isr: Tx FIFO under/overrun\n");
+
+	if (sis & INT_TXFIFO_RESYNC)
+		dev_dbg(&pdev->dev, "isr: Tx FIFO resync\n");
+
+	if (sis & INT_CNEW)
+		dev_dbg(&pdev->dev, "isr: cstatus new\n");
+
+	if (sis & INT_VAL_NOGOOD)
+		dev_dbg(&pdev->dev, "isr: validity flag no good\n");
+
+	if (sis & INT_SYM_ERR)
+		spdif_irq_sym_error(spdif_priv);
+
+	if (sis & INT_BIT_ERR)
+		dev_dbg(&pdev->dev, "isr: receiver found parity bit error\n");
+
+	if (sis & INT_URX_FUL)
+		spdif_irq_uqrx_full(spdif_priv, 'U');
+
+	if (sis & INT_URX_OV)
+		dev_dbg(&pdev->dev, "isr: U Channel receive register overrun\n");
+
+	if (sis & INT_QRX_FUL)
+		spdif_irq_uqrx_full(spdif_priv, 'Q');
+
+	if (sis & INT_QRX_OV)
+		dev_dbg(&pdev->dev, "isr: Q Channel receive register overrun\n");
+
+	if (sis & INT_UQ_SYNC)
+		spdif_irq_uq_sync(spdif_priv);
+
+	if (sis & INT_UQ_ERR)
+		spdif_irq_uq_err(spdif_priv);
+
+	if (sis & INT_RXFIFO_UNOV)
+		dev_dbg(&pdev->dev, "isr: Rx FIFO under/overrun\n");
+
+	if (sis & INT_RXFIFO_RESYNC)
+		dev_dbg(&pdev->dev, "isr: Rx FIFO resync\n");
+
+	if (sis & INT_LOSS_LOCK)
+		spdif_irq_dpll_lock(spdif_priv);
+
+	/* FIXME: Write Tx FIFO to clear TxEm */
+	if (sis & INT_TX_EM)
+		dev_dbg(&pdev->dev, "isr: Tx FIFO empty\n");
+
+	/* FIXME: Read Rx FIFO to clear RxFIFOFul */
+	if (sis & INT_RXFIFO_FUL)
+		dev_dbg(&pdev->dev, "isr: Rx FIFO full\n");
+
+	return IRQ_HANDLED;
+}
+
+static int spdif_softreset(struct fsl_spdif_priv *spdif_priv)
+{
+	struct regmap *regmap = spdif_priv->regmap;
+	u32 val, cycle = 1000;
+
+	regmap_write(regmap, REG_SPDIF_SCR, SCR_SOFT_RESET);
+
+	/*
+	 * RESET bit would be cleared after finishing its reset procedure,
+	 * which typically lasts 8 cycles. 1000 cycles will keep it safe.
+	 */
+	do {
+		regmap_read(regmap, REG_SPDIF_SCR, &val);
+	} while ((val & SCR_SOFT_RESET) && cycle--);
+
+	if (cycle)
+		return 0;
+	else
+		return -EBUSY;
+}
+
+static void spdif_set_cstatus(struct spdif_mixer_control *ctrl,
+				u8 mask, u8 cstatus)
+{
+	ctrl->ch_status[3] &= ~mask;
+	ctrl->ch_status[3] |= cstatus & mask;
+}
+
+static void spdif_write_channel_status(struct fsl_spdif_priv *spdif_priv)
+{
+	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+	struct regmap *regmap = spdif_priv->regmap;
+	struct platform_device *pdev = spdif_priv->pdev;
+	u32 ch_status;
+
+	ch_status = (bitrev8(ctrl->ch_status[0]) << 16) |
+		(bitrev8(ctrl->ch_status[1]) << 8) |
+		bitrev8(ctrl->ch_status[2]);
+	regmap_write(regmap, REG_SPDIF_STCSCH, ch_status);
+
+	dev_dbg(&pdev->dev, "STCSCH: 0x%06x\n", ch_status);
+
+	ch_status = bitrev8(ctrl->ch_status[3]) << 16;
+	regmap_write(regmap, REG_SPDIF_STCSCL, ch_status);
+
+	dev_dbg(&pdev->dev, "STCSCL: 0x%06x\n", ch_status);
+}
+
+/* Set SPDIF PhaseConfig register for rx clock */
+static int spdif_set_rx_clksrc(struct fsl_spdif_priv *spdif_priv,
+				enum spdif_gainsel gainsel, int dpll_locked)
+{
+	struct regmap *regmap = spdif_priv->regmap;
+	u8 clksrc = spdif_priv->rxclk_src;
+
+	if (clksrc >= SRPC_CLKSRC_MAX || gainsel >= GAINSEL_MULTI_MAX)
+		return -EINVAL;
+
+	regmap_update_bits(regmap, REG_SPDIF_SRPC,
+			SRPC_CLKSRC_SEL_MASK | SRPC_GAINSEL_MASK,
+			SRPC_CLKSRC_SEL_SET(clksrc) | SRPC_GAINSEL_SET(gainsel));
+
+	return 0;
+}
+
+static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
+				int sample_rate)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+	struct regmap *regmap = spdif_priv->regmap;
+	struct platform_device *pdev = spdif_priv->pdev;
+	unsigned long csfs = 0;
+	u32 stc, mask, rate;
+	u8 clk, div;
+	int ret;
+
+	switch (sample_rate) {
+	case 32000:
+		rate = SPDIF_TXRATE_32000;
+		csfs = IEC958_AES3_CON_FS_32000;
+		break;
+	case 44100:
+		rate = SPDIF_TXRATE_44100;
+		csfs = IEC958_AES3_CON_FS_44100;
+		break;
+	case 48000:
+		rate = SPDIF_TXRATE_48000;
+		csfs = IEC958_AES3_CON_FS_48000;
+		break;
+	default:
+		dev_err(&pdev->dev, "unsupported sample rate %d\n", sample_rate);
+		return -EINVAL;
+	}
+
+	clk = spdif_priv->txclk_src[rate];
+	if (clk >= STC_TXCLK_SRC_MAX) {
+		dev_err(&pdev->dev, "tx clock source is out of range\n");
+		return -EINVAL;
+	}
+
+	div = spdif_priv->txclk_div[rate];
+	if (div == 0) {
+		dev_err(&pdev->dev, "the divisor can't be zero\n");
+		return -EINVAL;
+	}
+
+	/*
+	 * The S/PDIF block needs a clock of 64 * fs * div.  The S/PDIF block
+	 * will divide by (div).  So request 64 * fs * (div+1) which will
+	 * get rounded.
+	 */
+	ret = clk_set_rate(spdif_priv->txclk[rate], 64 * sample_rate * (div + 1));
+	if (ret) {
+		dev_err(&pdev->dev, "failed to set tx clock rate\n");
+		return ret;
+	}
+
+	dev_dbg(&pdev->dev, "expected clock rate = %d\n",
+			(64 * sample_rate * div));
+	dev_dbg(&pdev->dev, "actual clock rate = %ld\n",
+			clk_get_rate(spdif_priv->txclk[rate]));
+
+	/* set fs field in consumer channel status */
+	spdif_set_cstatus(ctrl, IEC958_AES3_CON_FS, csfs);
+
+	/* select clock source and divisor */
+	stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) | STC_TXCLK_DIV(div);
+	mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK | STC_TXCLK_DIV_MASK;
+	regmap_update_bits(regmap, REG_SPDIF_STC, mask, stc);
+
+	dev_dbg(&pdev->dev, "set sample rate to %d\n", sample_rate);
+
+	return 0;
+}
+
+static int fsl_spdif_startup(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *cpu_dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct platform_device *pdev = spdif_priv->pdev;
+	struct regmap *regmap = spdif_priv->regmap;
+	u32 scr, mask, i;
+	int ret;
+
+	/* Reset module and interrupts only for first initialization */
+	if (!cpu_dai->active) {
+		ret = spdif_softreset(spdif_priv);
+		if (ret) {
+			dev_err(&pdev->dev, "failed to soft reset\n");
+			return ret;
+		}
+
+		/* Disable all the interrupts */
+		regmap_update_bits(regmap, REG_SPDIF_SIE, 0xffffff, 0);
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		scr = SCR_TXFIFO_AUTOSYNC | SCR_TXFIFO_CTRL_NORMAL |
+			SCR_TXSEL_NORMAL | SCR_USRC_SEL_CHIP |
+			SCR_TXFIFO_FSEL_IF8;
+		mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK |
+			SCR_TXSEL_MASK | SCR_USRC_SEL_MASK |
+			SCR_TXFIFO_FSEL_MASK;
+		for (i = 0; i < SPDIF_TXRATE_MAX; i++)
+			clk_prepare_enable(spdif_priv->txclk[i]);
+	} else {
+		scr = SCR_RXFIFO_FSEL_IF8 | SCR_RXFIFO_AUTOSYNC;
+		mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK|
+			SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK;
+		clk_prepare_enable(spdif_priv->rxclk);
+	}
+	regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr);
+
+	/* Power up SPDIF module */
+	regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_LOW_POWER, 0);
+
+	return 0;
+}
+
+static void fsl_spdif_shutdown(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *cpu_dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct regmap *regmap = spdif_priv->regmap;
+	u32 scr, mask, i;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		scr = 0;
+		mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK |
+			SCR_TXSEL_MASK | SCR_USRC_SEL_MASK |
+			SCR_TXFIFO_FSEL_MASK;
+		for (i = 0; i < SPDIF_TXRATE_MAX; i++)
+			clk_disable_unprepare(spdif_priv->txclk[i]);
+	} else {
+		scr = SCR_RXFIFO_OFF | SCR_RXFIFO_CTL_ZERO;
+		mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK|
+			SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK;
+		clk_disable_unprepare(spdif_priv->rxclk);
+	}
+	regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr);
+
+	/* Power down SPDIF module only if tx&rx are both inactive */
+	if (!cpu_dai->active) {
+		spdif_intr_status_clear(spdif_priv);
+		regmap_update_bits(regmap, REG_SPDIF_SCR,
+				SCR_LOW_POWER, SCR_LOW_POWER);
+	}
+}
+
+static int fsl_spdif_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+	struct platform_device *pdev = spdif_priv->pdev;
+	u32 sample_rate = params_rate(params);
+	int ret = 0;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		ret  = spdif_set_sample_rate(substream, sample_rate);
+		if (ret) {
+			dev_err(&pdev->dev, "%s: set sample rate failed: %d\n",
+					__func__, sample_rate);
+			return ret;
+		}
+		spdif_set_cstatus(ctrl, IEC958_AES3_CON_CLOCK,
+				IEC958_AES3_CON_CLOCK_1000PPM);
+		spdif_write_channel_status(spdif_priv);
+	} else {
+		/* Setup rx clock source */
+		ret = spdif_set_rx_clksrc(spdif_priv, SPDIF_DEFAULT_GAINSEL, 1);
+	}
+
+	return ret;
+}
+
+static int fsl_spdif_trigger(struct snd_pcm_substream *substream,
+				int cmd, struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct regmap *regmap = spdif_priv->regmap;
+	int is_playack = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	u32 intr = is_playack ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE;
+	u32 dmaen = is_playack ? SCR_DMA_TX_EN : SCR_DMA_RX_EN;;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		regmap_update_bits(regmap, REG_SPDIF_SIE, intr, intr);
+		regmap_update_bits(regmap, REG_SPDIF_SCR, dmaen, dmaen);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		regmap_update_bits(regmap, REG_SPDIF_SCR, dmaen, 0);
+		regmap_update_bits(regmap, REG_SPDIF_SIE, intr, 0);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_dai_ops fsl_spdif_dai_ops = {
+	.startup = fsl_spdif_startup,
+	.hw_params = fsl_spdif_hw_params,
+	.trigger = fsl_spdif_trigger,
+	.shutdown = fsl_spdif_shutdown,
+};
+
+
+/*
+ * FSL SPDIF IEC958 controller(mixer) functions
+ *
+ *	Channel status get/put control
+ *	User bit value get/put control
+ *	Valid bit value get control
+ *	DPLL lock status get control
+ *	User bit sync mode selection control
+ */
+
+static int fsl_spdif_info(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+	uinfo->count = 1;
+
+	return 0;
+}
+
+static int fsl_spdif_pb_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *uvalue)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+
+	uvalue->value.iec958.status[0] = ctrl->ch_status[0];
+	uvalue->value.iec958.status[1] = ctrl->ch_status[1];
+	uvalue->value.iec958.status[2] = ctrl->ch_status[2];
+	uvalue->value.iec958.status[3] = ctrl->ch_status[3];
+
+	return 0;
+}
+
+static int fsl_spdif_pb_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *uvalue)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+
+	ctrl->ch_status[0] = uvalue->value.iec958.status[0];
+	ctrl->ch_status[1] = uvalue->value.iec958.status[1];
+	ctrl->ch_status[2] = uvalue->value.iec958.status[2];
+	ctrl->ch_status[3] = uvalue->value.iec958.status[3];
+
+	spdif_write_channel_status(spdif_priv);
+
+	return 0;
+}
+
+/* Get channel status from SPDIF_RX_CCHAN register */
+static int fsl_spdif_capture_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct regmap *regmap = spdif_priv->regmap;
+	u32 cstatus, val;
+
+	regmap_read(regmap, REG_SPDIF_SIS, &val);
+	if (!(val & INT_CNEW)) {
+		return -EAGAIN;
+	}
+
+	regmap_read(regmap, REG_SPDIF_SRCSH, &cstatus);
+	ucontrol->value.iec958.status[0] = (cstatus >> 16) & 0xFF;
+	ucontrol->value.iec958.status[1] = (cstatus >> 8) & 0xFF;
+	ucontrol->value.iec958.status[2] = cstatus & 0xFF;
+
+	regmap_read(regmap, REG_SPDIF_SRCSL, &cstatus);
+	ucontrol->value.iec958.status[3] = (cstatus >> 16) & 0xFF;
+	ucontrol->value.iec958.status[4] = (cstatus >> 8) & 0xFF;
+	ucontrol->value.iec958.status[5] = cstatus & 0xFF;
+
+	/* Clear intr */
+	regmap_write(regmap, REG_SPDIF_SIC, INT_CNEW);
+
+	return 0;
+}
+
+/*
+ * Get User bits (subcode) from chip value which readed out
+ * in UChannel register.
+ */
+static int fsl_spdif_subcode_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+	unsigned long flags;
+	int ret = 0;
+
+	spin_lock_irqsave(&ctrl->ctl_lock, flags);
+	if (ctrl->ready_buf) {
+		int idx = (ctrl->ready_buf - 1) * SPDIF_UBITS_SIZE;
+		memcpy(&ucontrol->value.iec958.subcode[0],
+				&ctrl->subcode[idx], SPDIF_UBITS_SIZE);
+	} else {
+		ret = -EAGAIN;
+	}
+	spin_unlock_irqrestore(&ctrl->ctl_lock, flags);
+
+	return ret;
+}
+
+/* Q-subcode infomation. The byte size is SPDIF_UBITS_SIZE/8 */
+static int fsl_spdif_qinfo(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+	uinfo->count = SPDIF_QSUB_SIZE;
+
+	return 0;
+}
+
+/* Get Q subcode from chip value which readed out in QChannel register */
+static int fsl_spdif_qget(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+	unsigned long flags;
+	int ret = 0;
+
+	spin_lock_irqsave(&ctrl->ctl_lock, flags);
+	if (ctrl->ready_buf) {
+		int idx = (ctrl->ready_buf - 1) * SPDIF_QSUB_SIZE;
+		memcpy(&ucontrol->value.bytes.data[0],
+				&ctrl->qsub[idx], SPDIF_QSUB_SIZE);
+	} else {
+		ret = -EAGAIN;
+	}
+	spin_unlock_irqrestore(&ctrl->ctl_lock, flags);
+
+	return ret;
+}
+
+/* Valid bit infomation */
+static int fsl_spdif_vbit_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;
+}
+
+/* Get valid good bit from interrupt status register */
+static int fsl_spdif_vbit_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct regmap *regmap = spdif_priv->regmap;
+	u32 val;
+
+	val = regmap_read(regmap, REG_SPDIF_SIS, &val);
+	ucontrol->value.integer.value[0] = (val & INT_VAL_NOGOOD) != 0;
+	regmap_write(regmap, REG_SPDIF_SIC, INT_VAL_NOGOOD);
+
+	return 0;
+}
+
+/* DPLL lock infomation */
+static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 16000;
+	uinfo->value.integer.max = 96000;
+
+	return 0;
+}
+
+static u32 gainsel_multi[GAINSEL_MULTI_MAX] = {
+	24, 16, 12, 8, 6, 4, 3,
+};
+
+/* Get RX data clock rate given the SPDIF bus_clk */
+static int spdif_get_rxclk_rate(struct fsl_spdif_priv *spdif_priv,
+				enum spdif_gainsel gainsel)
+{
+	struct regmap *regmap = spdif_priv->regmap;
+	struct platform_device *pdev = spdif_priv->pdev;
+	u64 tmpval64, busclk_freq = 0;
+	u32 freqmeas, phaseconf;
+	u8 clksrc;
+
+	regmap_read(regmap, REG_SPDIF_SRFM, &freqmeas);
+	regmap_read(regmap, REG_SPDIF_SRPC, &phaseconf);
+
+	clksrc = (phaseconf >> SRPC_CLKSRC_SEL_OFFSET) & 0xf;
+	if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED)) {
+		/* Get bus clock from system */
+		busclk_freq = clk_get_rate(spdif_priv->rxclk);
+	}
+
+	/* FreqMeas_CLK = (BUS_CLK * FreqMeas) / 2 ^ 10 / GAINSEL / 128 */
+	tmpval64 = (u64) busclk_freq * freqmeas;
+	do_div(tmpval64, gainsel_multi[gainsel] * 1024);
+	do_div(tmpval64, 128 * 1024);
+
+	dev_dbg(&pdev->dev, "FreqMeas: %d\n", freqmeas);
+	dev_dbg(&pdev->dev, "BusclkFreq: %lld\n", busclk_freq);
+	dev_dbg(&pdev->dev, "RxRate: %lld\n", tmpval64);
+
+	return (int)tmpval64;
+}
+
+/*
+ * Get DPLL lock or not info from stable interrupt status register.
+ * User application must use this control to get locked,
+ * then can do next PCM operation
+ */
+static int fsl_spdif_rxrate_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+	int rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL);
+
+	if (spdif_priv->dpll_locked)
+		ucontrol->value.integer.value[0] = rate;
+	else
+		ucontrol->value.integer.value[0] = 0;
+
+	return 0;
+}
+
+/* User bit sync mode info */
+static int fsl_spdif_usync_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;
+}
+
+/*
+ * User bit sync mode:
+ * 1 CD User channel subcode
+ * 0 Non-CD data
+ */
+static int fsl_spdif_usync_get(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct regmap *regmap = spdif_priv->regmap;
+	u32 val;
+
+	regmap_read(regmap, REG_SPDIF_SRCD, &val);
+	ucontrol->value.integer.value[0] = (val & SRCD_CD_USER) != 0;
+
+	return 0;
+}
+
+/*
+ * User bit sync mode:
+ * 1 CD User channel subcode
+ * 0 Non-CD data
+ */
+static int fsl_spdif_usync_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct regmap *regmap = spdif_priv->regmap;
+	u32 val = ucontrol->value.integer.value[0] << SRCD_CD_USER_OFFSET;
+
+	regmap_update_bits(regmap, REG_SPDIF_SRCD, SRCD_CD_USER, val);
+
+	return 0;
+}
+
+/* FSL SPDIF IEC958 controller defines */
+static struct snd_kcontrol_new fsl_spdif_ctrls[] = {
+	/* Status cchanel controller */
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
+		.access = SNDRV_CTL_ELEM_ACCESS_READ |
+			SNDRV_CTL_ELEM_ACCESS_WRITE |
+			SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = fsl_spdif_info,
+		.get = fsl_spdif_pb_get,
+		.put = fsl_spdif_pb_put,
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+		.name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT),
+		.access = SNDRV_CTL_ELEM_ACCESS_READ |
+			SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = fsl_spdif_info,
+		.get = fsl_spdif_capture_get,
+	},
+	/* User bits controller */
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+		.name = "IEC958 Subcode Capture Default",
+		.access = SNDRV_CTL_ELEM_ACCESS_READ |
+			SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = fsl_spdif_info,
+		.get = fsl_spdif_subcode_get,
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+		.name = "IEC958 Q-subcode Capture Default",
+		.access = SNDRV_CTL_ELEM_ACCESS_READ |
+			SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = fsl_spdif_qinfo,
+		.get = fsl_spdif_qget,
+	},
+	/* Valid bit error controller */
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+		.name = "IEC958 V-Bit Errors",
+		.access = SNDRV_CTL_ELEM_ACCESS_READ |
+			SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = fsl_spdif_vbit_info,
+		.get = fsl_spdif_vbit_get,
+	},
+	/* DPLL lock info get controller */
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+		.name = "RX Sample Rate",
+		.access = SNDRV_CTL_ELEM_ACCESS_READ |
+			SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = fsl_spdif_rxrate_info,
+		.get = fsl_spdif_rxrate_get,
+	},
+	/* User bit sync mode set/get controller */
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+		.name = "IEC958 USyncMode CDText",
+		.access = SNDRV_CTL_ELEM_ACCESS_READ |
+			SNDRV_CTL_ELEM_ACCESS_WRITE |
+			SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = fsl_spdif_usync_info,
+		.get = fsl_spdif_usync_get,
+		.put = fsl_spdif_usync_put,
+	},
+};
+
+static int fsl_spdif_dai_probe(struct snd_soc_dai *dai)
+{
+	struct fsl_spdif_priv *spdif_private = snd_soc_dai_get_drvdata(dai);
+
+	dai->playback_dma_data = &spdif_private->dma_params_tx;
+	dai->capture_dma_data = &spdif_private->dma_params_rx;
+
+	snd_soc_add_dai_controls(dai, fsl_spdif_ctrls, ARRAY_SIZE(fsl_spdif_ctrls));
+
+	return 0;
+}
+
+static struct snd_soc_dai_driver fsl_spdif_dai = {
+	.probe = &fsl_spdif_dai_probe,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = FSL_SPDIF_RATES_PLAYBACK,
+		.formats = FSL_SPDIF_FORMATS_PLAYBACK,
+	},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = FSL_SPDIF_RATES_CAPTURE,
+		.formats = FSL_SPDIF_FORMATS_CAPTURE,
+	},
+	.ops = &fsl_spdif_dai_ops,
+};
+
+static const struct snd_soc_component_driver fsl_spdif_component = {
+	.name		= "fsl-spdif",
+};
+
+/* FSL SPDIF REGMAP */
+
+static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case REG_SPDIF_SCR:
+	case REG_SPDIF_SRCD:
+	case REG_SPDIF_SRPC:
+	case REG_SPDIF_SIE:
+	case REG_SPDIF_SIS:
+	case REG_SPDIF_SRL:
+	case REG_SPDIF_SRR:
+	case REG_SPDIF_SRCSH:
+	case REG_SPDIF_SRCSL:
+	case REG_SPDIF_SRU:
+	case REG_SPDIF_SRQ:
+	case REG_SPDIF_STCSCH:
+	case REG_SPDIF_STCSCL:
+	case REG_SPDIF_SRFM:
+	case REG_SPDIF_STC:
+		return true;
+	default:
+		return false;
+	};
+}
+
+static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case REG_SPDIF_SCR:
+	case REG_SPDIF_SRCD:
+	case REG_SPDIF_SRPC:
+	case REG_SPDIF_SIE:
+	case REG_SPDIF_SIC:
+	case REG_SPDIF_STL:
+	case REG_SPDIF_STR:
+	case REG_SPDIF_STCSCH:
+	case REG_SPDIF_STCSCL:
+	case REG_SPDIF_STC:
+		return true;
+	default:
+		return false;
+	};
+}
+
+static const struct regmap_config fsl_spdif_regmap_config = {
+	.reg_bits = 32,
+	.reg_stride = 4,
+	.val_bits = 32,
+
+	.max_register = REG_SPDIF_STC,
+	.readable_reg = fsl_spdif_readable_reg,
+	.writeable_reg = fsl_spdif_writeable_reg,
+};
+
+static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
+				struct clk *clk, u64 savesub,
+				enum spdif_txrate index)
+{
+	const u32 rate[] = { 32000, 44100, 48000 };
+	u64 rate_ideal, rate_actual, sub;
+	u32 div, arate;
+
+	for (div = 1; div <= 128; div++) {
+		rate_ideal = rate[index] * (div + 1) * 64;
+		rate_actual = clk_round_rate(clk, rate_ideal);
+
+		arate = rate_actual / 64;
+		arate /= div;
+
+		if (arate == rate[index]) {
+			/* We are lucky */
+			savesub = 0;
+			spdif_priv->txclk_div[index] = div;
+			break;
+		} else if (arate / rate[index] == 1) {
+			/* A little bigger than expect */
+			sub = (arate - rate[index]) * 100000;
+			do_div(sub, rate[index]);
+			if (sub < savesub) {
+				savesub = sub;
+				spdif_priv->txclk_div[index] = div;
+			}
+		} else if (rate[index] / arate == 1) {
+			/* A little smaller than expect */
+			sub = (rate[index] - arate) * 100000;
+			do_div(sub, rate[index]);
+			if (sub < savesub) {
+				savesub = sub;
+				spdif_priv->txclk_div[index] = div;
+			}
+		}
+	}
+
+	return savesub;
+}
+
+static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
+				enum spdif_txrate index)
+{
+	const u32 rate[] = { 32000, 44100, 48000 };
+	struct platform_device *pdev = spdif_priv->pdev;
+	struct device *dev = &pdev->dev;
+	u64 savesub = 100000, ret;
+	struct clk *clk;
+	char tmp[16];
+	int i;
+
+	for (i = 0; i < STC_TXCLK_SRC_MAX; i++) {
+		sprintf(tmp, "rxtx%d", i);
+		clk = devm_clk_get(&pdev->dev, tmp);
+		if (IS_ERR(clk)) {
+			dev_err(dev, "no rxtx%d clock in devicetree\n", i);
+			return PTR_ERR(clk);
+		}
+		if (!clk_get_rate(clk))
+			continue;
+
+		ret = fsl_spdif_txclk_caldiv(spdif_priv, clk, savesub, index);
+		if (savesub == ret)
+			continue;
+
+		savesub = ret;
+		spdif_priv->txclk[index] = clk;
+		spdif_priv->txclk_src[index] = i;
+
+		/* To quick catch a divisor, we allow a 0.1% deviation */
+		if (savesub < 100)
+			break;
+	}
+
+	dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate\n",
+			spdif_priv->txclk_src[index], rate[index]);
+	dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate\n",
+			spdif_priv->txclk_div[index], rate[index]);
+
+	return 0;
+}
+
+static int fsl_spdif_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct fsl_spdif_priv *spdif_priv;
+	struct spdif_mixer_control *ctrl;
+	struct resource *res;
+	void __iomem *regs;
+	int irq, ret, i;
+
+	if (!np)
+		return -ENODEV;
+
+	spdif_priv = devm_kzalloc(&pdev->dev,
+			sizeof(struct fsl_spdif_priv) + strlen(np->name) + 1,
+			GFP_KERNEL);
+	if (!spdif_priv)
+		return -ENOMEM;
+
+	strcpy(spdif_priv->name, np->name);
+
+	spdif_priv->pdev = pdev;
+
+	/* Initialize this copy of the CPU DAI driver structure */
+	memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai));
+	spdif_priv->cpu_dai_drv.name = spdif_priv->name;
+
+	/* Get the addresses and IRQ */
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (IS_ERR(res)) {
+		dev_err(&pdev->dev, "could not determine device resources\n");
+		return PTR_ERR(res);
+	}
+
+	regs = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	spdif_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev,
+			"core", regs, &fsl_spdif_regmap_config);
+	if (IS_ERR(spdif_priv->regmap)) {
+		dev_err(&pdev->dev, "regmap init failed\n");
+		return PTR_ERR(spdif_priv->regmap);
+	}
+
+	irq = platform_get_irq(pdev, 0);
+	if (irq < 0) {
+		dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+		return irq;
+	}
+
+	ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0,
+			spdif_priv->name, spdif_priv);
+	if (ret) {
+		dev_err(&pdev->dev, "could not claim irq %u\n", irq);
+		return ret;
+	}
+
+	/* Select clock source for rx/tx clock */
+	spdif_priv->rxclk = devm_clk_get(&pdev->dev, "rxtx1");
+	if (IS_ERR(spdif_priv->rxclk)) {
+		dev_err(&pdev->dev, "no rxtx1 clock in devicetree\n");
+		return PTR_ERR(spdif_priv->rxclk);
+	}
+	spdif_priv->rxclk_src = DEFAULT_RXCLK_SRC;
+
+	for (i = 0; i < SPDIF_TXRATE_MAX; i++) {
+		ret = fsl_spdif_probe_txclk(spdif_priv, i);
+		if (ret)
+			return ret;
+	}
+
+	/* Initial spinlock for control data */
+	ctrl = &spdif_priv->fsl_spdif_control;
+	spin_lock_init(&ctrl->ctl_lock);
+
+	/* Init tx channel status default value */
+	ctrl->ch_status[0] =
+		IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_5015;
+	ctrl->ch_status[1] = IEC958_AES1_CON_DIGDIGCONV_ID;
+	ctrl->ch_status[2] = 0x00;
+	ctrl->ch_status[3] =
+		IEC958_AES3_CON_FS_44100 | IEC958_AES3_CON_CLOCK_1000PPM;
+
+	spdif_priv->dpll_locked = false;
+
+	spdif_priv->dma_params_tx.maxburst = FSL_SPDIF_TXFIFO_WML;
+	spdif_priv->dma_params_rx.maxburst = FSL_SPDIF_RXFIFO_WML;
+	spdif_priv->dma_params_tx.addr = res->start + REG_SPDIF_STL;
+	spdif_priv->dma_params_rx.addr = res->start + REG_SPDIF_SRL;
+
+	/* Register with ASoC */
+	dev_set_drvdata(&pdev->dev, spdif_priv);
+
+	ret = snd_soc_register_component(&pdev->dev, &fsl_spdif_component,
+					 &spdif_priv->cpu_dai_drv, 1);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
+		return ret;
+	}
+
+	ret = imx_pcm_dma_init(pdev);
+	if (ret) {
+		dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret);
+		goto error_component;
+	}
+
+	return ret;
+
+error_component:
+	snd_soc_unregister_component(&pdev->dev);
+
+	return ret;
+}
+
+static int fsl_spdif_remove(struct platform_device *pdev)
+{
+	imx_pcm_dma_exit(pdev);
+	snd_soc_unregister_component(&pdev->dev);
+
+	return 0;
+}
+
+static const struct of_device_id fsl_spdif_dt_ids[] = {
+	{ .compatible = "fsl,imx35-spdif", },
+	{}
+};
+MODULE_DEVICE_TABLE(of, fsl_spdif_dt_ids);
+
+static struct platform_driver fsl_spdif_driver = {
+	.driver = {
+		.name = "fsl-spdif-dai",
+		.owner = THIS_MODULE,
+		.of_match_table = fsl_spdif_dt_ids,
+	},
+	.probe = fsl_spdif_probe,
+	.remove = fsl_spdif_remove,
+};
+
+module_platform_driver(fsl_spdif_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale S/PDIF CPU DAI Driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:fsl-spdif-dai");
diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h
new file mode 100644
index 000000000000..b1266790d117
--- /dev/null
+++ b/sound/soc/fsl/fsl_spdif.h
@@ -0,0 +1,191 @@
+/*
+ * fsl_spdif.h - ALSA S/PDIF interface for the Freescale i.MX SoC
+ *
+ * Copyright (C) 2013 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <b42378@freescale.com>
+ *
+ * Based on fsl_ssi.h
+ * Author: Timur Tabi <timur@freescale.com>
+ * Copyright 2007-2008 Freescale Semiconductor, Inc.
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2.  This program  is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#ifndef _FSL_SPDIF_DAI_H
+#define _FSL_SPDIF_DAI_H
+
+/* S/PDIF Register Map */
+#define REG_SPDIF_SCR 			0x0	/* SPDIF Configuration Register */
+#define REG_SPDIF_SRCD		 	0x4	/* CDText Control Register */
+#define REG_SPDIF_SRPC			0x8	/* PhaseConfig Register */
+#define REG_SPDIF_SIE			0xc	/* InterruptEn Register */
+#define REG_SPDIF_SIS			0x10	/* InterruptStat Register */
+#define REG_SPDIF_SIC			0x10	/* InterruptClear Register */
+#define REG_SPDIF_SRL			0x14	/* SPDIFRxLeft Register */
+#define REG_SPDIF_SRR			0x18	/* SPDIFRxRight Register */
+#define REG_SPDIF_SRCSH			0x1c	/* SPDIFRxCChannel_h Register */
+#define REG_SPDIF_SRCSL			0x20	/* SPDIFRxCChannel_l Register */
+#define REG_SPDIF_SRU			0x24	/* UchannelRx Register */
+#define REG_SPDIF_SRQ			0x28	/* QchannelRx Register */
+#define REG_SPDIF_STL			0x2C	/* SPDIFTxLeft Register */
+#define REG_SPDIF_STR			0x30	/* SPDIFTxRight Register */
+#define REG_SPDIF_STCSCH		0x34	/* SPDIFTxCChannelCons_h Register */
+#define REG_SPDIF_STCSCL		0x38	/* SPDIFTxCChannelCons_l Register */
+#define REG_SPDIF_SRFM			0x44	/* FreqMeas Register */
+#define REG_SPDIF_STC			0x50	/* SPDIFTxClk Register */
+
+
+/* SPDIF Configuration register */
+#define SCR_RXFIFO_CTL_OFFSET		23
+#define SCR_RXFIFO_CTL_MASK		(1 << SCR_RXFIFO_CTL_OFFSET)
+#define SCR_RXFIFO_CTL_ZERO		(1 << SCR_RXFIFO_CTL_OFFSET)
+#define SCR_RXFIFO_OFF_OFFSET		22
+#define SCR_RXFIFO_OFF_MASK		(1 << SCR_RXFIFO_OFF_OFFSET)
+#define SCR_RXFIFO_OFF			(1 << SCR_RXFIFO_OFF_OFFSET)
+#define SCR_RXFIFO_RST_OFFSET		21
+#define SCR_RXFIFO_RST_MASK		(1 << SCR_RXFIFO_RST_OFFSET)
+#define SCR_RXFIFO_RST			(1 << SCR_RXFIFO_RST_OFFSET)
+#define SCR_RXFIFO_FSEL_OFFSET		19
+#define SCR_RXFIFO_FSEL_MASK		(0x3 << SCR_RXFIFO_FSEL_OFFSET)
+#define SCR_RXFIFO_FSEL_IF0		(0x0 << SCR_RXFIFO_FSEL_OFFSET)
+#define SCR_RXFIFO_FSEL_IF4		(0x1 << SCR_RXFIFO_FSEL_OFFSET)
+#define SCR_RXFIFO_FSEL_IF8		(0x2 << SCR_RXFIFO_FSEL_OFFSET)
+#define SCR_RXFIFO_FSEL_IF12		(0x3 << SCR_RXFIFO_FSEL_OFFSET)
+#define SCR_RXFIFO_AUTOSYNC_OFFSET	18
+#define SCR_RXFIFO_AUTOSYNC_MASK	(1 << SCR_RXFIFO_AUTOSYNC_OFFSET)
+#define SCR_RXFIFO_AUTOSYNC		(1 << SCR_RXFIFO_AUTOSYNC_OFFSET)
+#define SCR_TXFIFO_AUTOSYNC_OFFSET	17
+#define SCR_TXFIFO_AUTOSYNC_MASK	(1 << SCR_TXFIFO_AUTOSYNC_OFFSET)
+#define SCR_TXFIFO_AUTOSYNC		(1 << SCR_TXFIFO_AUTOSYNC_OFFSET)
+#define SCR_TXFIFO_FSEL_OFFSET		15
+#define SCR_TXFIFO_FSEL_MASK		(0x3 << SCR_TXFIFO_FSEL_OFFSET)
+#define SCR_TXFIFO_FSEL_IF0		(0x0 << SCR_TXFIFO_FSEL_OFFSET)
+#define SCR_TXFIFO_FSEL_IF4		(0x1 << SCR_TXFIFO_FSEL_OFFSET)
+#define SCR_TXFIFO_FSEL_IF8		(0x2 << SCR_TXFIFO_FSEL_OFFSET)
+#define SCR_TXFIFO_FSEL_IF12		(0x3 << SCR_TXFIFO_FSEL_OFFSET)
+#define SCR_LOW_POWER			(1 << 13)
+#define SCR_SOFT_RESET			(1 << 12)
+#define SCR_TXFIFO_CTRL_OFFSET		10
+#define SCR_TXFIFO_CTRL_MASK		(0x3 << SCR_TXFIFO_CTRL_OFFSET)
+#define SCR_TXFIFO_CTRL_ZERO		(0x0 << SCR_TXFIFO_CTRL_OFFSET)
+#define SCR_TXFIFO_CTRL_NORMAL		(0x1 << SCR_TXFIFO_CTRL_OFFSET)
+#define SCR_TXFIFO_CTRL_ONESAMPLE	(0x2 << SCR_TXFIFO_CTRL_OFFSET)
+#define SCR_DMA_RX_EN_OFFSET		9
+#define SCR_DMA_RX_EN_MASK		(1 << SCR_DMA_RX_EN_OFFSET)
+#define SCR_DMA_RX_EN			(1 << SCR_DMA_RX_EN_OFFSET)
+#define SCR_DMA_TX_EN_OFFSET		8
+#define SCR_DMA_TX_EN_MASK		(1 << SCR_DMA_TX_EN_OFFSET)
+#define SCR_DMA_TX_EN			(1 << SCR_DMA_TX_EN_OFFSET)
+#define SCR_VAL_OFFSET			5
+#define SCR_VAL_MASK			(1 << SCR_VAL_OFFSET)
+#define SCR_VAL_CLEAR			(1 << SCR_VAL_OFFSET)
+#define SCR_TXSEL_OFFSET		2
+#define SCR_TXSEL_MASK			(0x7 << SCR_TXSEL_OFFSET)
+#define SCR_TXSEL_OFF			(0 << SCR_TXSEL_OFFSET)
+#define SCR_TXSEL_RX			(1 << SCR_TXSEL_OFFSET)
+#define SCR_TXSEL_NORMAL		(0x5 << SCR_TXSEL_OFFSET)
+#define SCR_USRC_SEL_OFFSET		0x0
+#define SCR_USRC_SEL_MASK		(0x3 << SCR_USRC_SEL_OFFSET)
+#define SCR_USRC_SEL_NONE		(0x0 << SCR_USRC_SEL_OFFSET)
+#define SCR_USRC_SEL_RECV		(0x1 << SCR_USRC_SEL_OFFSET)
+#define SCR_USRC_SEL_CHIP		(0x3 << SCR_USRC_SEL_OFFSET)
+
+/* SPDIF CDText control */
+#define SRCD_CD_USER_OFFSET		1
+#define SRCD_CD_USER			(1 << SRCD_CD_USER_OFFSET)
+
+/* SPDIF Phase Configuration register */
+#define SRPC_DPLL_LOCKED		(1 << 6)
+#define SRPC_CLKSRC_SEL_OFFSET		7
+#define SRPC_CLKSRC_SEL_MASK		(0xf << SRPC_CLKSRC_SEL_OFFSET)
+#define SRPC_CLKSRC_SEL_SET(x)		((x << SRPC_CLKSRC_SEL_OFFSET) & SRPC_CLKSRC_SEL_MASK)
+#define SRPC_CLKSRC_SEL_LOCKED_OFFSET1	5
+#define SRPC_CLKSRC_SEL_LOCKED_OFFSET2	2
+#define SRPC_GAINSEL_OFFSET		3
+#define SRPC_GAINSEL_MASK		(0x7 << SRPC_GAINSEL_OFFSET)
+#define SRPC_GAINSEL_SET(x)		((x << SRPC_GAINSEL_OFFSET) & SRPC_GAINSEL_MASK)
+
+#define SRPC_CLKSRC_MAX			16
+
+enum spdif_gainsel {
+	GAINSEL_MULTI_24 = 0,
+	GAINSEL_MULTI_16,
+	GAINSEL_MULTI_12,
+	GAINSEL_MULTI_8,
+	GAINSEL_MULTI_6,
+	GAINSEL_MULTI_4,
+	GAINSEL_MULTI_3,
+};
+#define GAINSEL_MULTI_MAX		(GAINSEL_MULTI_3 + 1)
+#define SPDIF_DEFAULT_GAINSEL		GAINSEL_MULTI_8
+
+/* SPDIF interrupt mask define */
+#define INT_DPLL_LOCKED			(1 << 20)
+#define INT_TXFIFO_UNOV			(1 << 19)
+#define INT_TXFIFO_RESYNC		(1 << 18)
+#define INT_CNEW			(1 << 17)
+#define INT_VAL_NOGOOD			(1 << 16)
+#define INT_SYM_ERR			(1 << 15)
+#define INT_BIT_ERR			(1 << 14)
+#define INT_URX_FUL			(1 << 10)
+#define INT_URX_OV			(1 << 9)
+#define INT_QRX_FUL			(1 << 8)
+#define INT_QRX_OV			(1 << 7)
+#define INT_UQ_SYNC			(1 << 6)
+#define INT_UQ_ERR			(1 << 5)
+#define INT_RXFIFO_UNOV			(1 << 4)
+#define INT_RXFIFO_RESYNC		(1 << 3)
+#define INT_LOSS_LOCK			(1 << 2)
+#define INT_TX_EM			(1 << 1)
+#define INT_RXFIFO_FUL			(1 << 0)
+
+/* SPDIF Clock register */
+#define STC_SYSCLK_DIV_OFFSET		11
+#define STC_SYSCLK_DIV_MASK		(0x1ff << STC_TXCLK_SRC_OFFSET)
+#define STC_SYSCLK_DIV(x)		((((x) - 1) << STC_TXCLK_DIV_OFFSET) & STC_SYSCLK_DIV_MASK)
+#define STC_TXCLK_SRC_OFFSET		8
+#define STC_TXCLK_SRC_MASK		(0x7 << STC_TXCLK_SRC_OFFSET)
+#define STC_TXCLK_SRC_SET(x)		((x << STC_TXCLK_SRC_OFFSET) & STC_TXCLK_SRC_MASK)
+#define STC_TXCLK_ALL_EN_OFFSET		7
+#define STC_TXCLK_ALL_EN_MASK		(1 << STC_TXCLK_ALL_EN_OFFSET)
+#define STC_TXCLK_ALL_EN		(1 << STC_TXCLK_ALL_EN_OFFSET)
+#define STC_TXCLK_DIV_OFFSET		0
+#define STC_TXCLK_DIV_MASK		(0x7ff << STC_TXCLK_DIV_OFFSET)
+#define STC_TXCLK_DIV(x)		((((x) - 1) << STC_TXCLK_DIV_OFFSET) & STC_TXCLK_DIV_MASK)
+#define STC_TXCLK_SRC_MAX		8
+
+/* SPDIF tx rate */
+enum spdif_txrate {
+	SPDIF_TXRATE_32000 = 0,
+	SPDIF_TXRATE_44100,
+	SPDIF_TXRATE_48000,
+};
+#define SPDIF_TXRATE_MAX		(SPDIF_TXRATE_48000 + 1)
+
+
+#define SPDIF_CSTATUS_BYTE		6
+#define SPDIF_UBITS_SIZE		96
+#define SPDIF_QSUB_SIZE			(SPDIF_UBITS_SIZE / 8)
+
+
+#define FSL_SPDIF_RATES_PLAYBACK	(SNDRV_PCM_RATE_32000 |	\
+					 SNDRV_PCM_RATE_44100 |	\
+					 SNDRV_PCM_RATE_48000)
+
+#define FSL_SPDIF_RATES_CAPTURE		(SNDRV_PCM_RATE_16000 | \
+					 SNDRV_PCM_RATE_32000 |	\
+					 SNDRV_PCM_RATE_44100 | \
+					 SNDRV_PCM_RATE_48000 |	\
+					 SNDRV_PCM_RATE_64000 | \
+					 SNDRV_PCM_RATE_96000)
+
+#define FSL_SPDIF_FORMATS_PLAYBACK	(SNDRV_PCM_FMTBIT_S16_LE | \
+					 SNDRV_PCM_FMTBIT_S20_3LE | \
+					 SNDRV_PCM_FMTBIT_S24_LE)
+
+#define FSL_SPDIF_FORMATS_CAPTURE	(SNDRV_PCM_FMTBIT_S24_LE)
+
+#endif /* _FSL_SPDIF_DAI_H */
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 2f2d837df07f..c6b743978d5e 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -8,6 +8,26 @@
  * This file is licensed under the terms of the GNU General Public License
  * version 2.  This program is licensed "as is" without any warranty of any
  * kind, whether express or implied.
+ *
+ *
+ * Some notes why imx-pcm-fiq is used instead of DMA on some boards:
+ *
+ * The i.MX SSI core has some nasty limitations in AC97 mode. While most
+ * sane processor vendors have a FIFO per AC97 slot, the i.MX has only
+ * one FIFO which combines all valid receive slots. We cannot even select
+ * which slots we want to receive. The WM9712 with which this driver
+ * was developed with always sends GPIO status data in slot 12 which
+ * we receive in our (PCM-) data stream. The only chance we have is to
+ * manually skip this data in the FIQ handler. With sampling rates different
+ * from 48000Hz not every frame has valid receive data, so the ratio
+ * between pcm data and GPIO status data changes. Our FIQ handler is not
+ * able to handle this, hence this driver only works with 48000Hz sampling
+ * rate.
+ * Reading and writing AC97 registers is another challenge. The core
+ * provides us status bits when the read register is updated with *another*
+ * value. When we read the same register two times (and the register still
+ * contains the same value) these status bits are not set. We work
+ * around this by not polling these bits but only wait a fixed delay.
  */
 
 #include <linux/init.h>
@@ -36,7 +56,7 @@
 #define read_ssi(addr)			 in_be32(addr)
 #define write_ssi(val, addr)		 out_be32(addr, val)
 #define write_ssi_mask(addr, clear, set) clrsetbits_be32(addr, clear, set)
-#elif defined ARM
+#else
 #define read_ssi(addr)			 readl(addr)
 #define write_ssi(val, addr)		 writel(val, addr)
 /*
@@ -121,11 +141,14 @@ struct fsl_ssi_private {
 
 	bool new_binding;
 	bool ssi_on_imx;
+	bool imx_ac97;
+	bool use_dma;
 	struct clk *clk;
 	struct snd_dmaengine_dai_dma_data dma_params_tx;
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
 	struct imx_dma_data filter_data_tx;
 	struct imx_dma_data filter_data_rx;
+	struct imx_pcm_fiq_params fiq_params;
 
 	struct {
 		unsigned int rfrc;
@@ -298,6 +321,102 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
 	return ret;
 }
 
+static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private)
+{
+	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+	u8 i2s_mode;
+	u8 wm;
+	int synchronous = ssi_private->cpu_dai_drv.symmetric_rates;
+
+	if (ssi_private->imx_ac97)
+		i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET;
+	else
+		i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE;
+
+	/*
+	 * Section 16.5 of the MPC8610 reference manual says that the SSI needs
+	 * to be disabled before updating the registers we set here.
+	 */
+	write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0);
+
+	/*
+	 * Program the SSI into I2S Slave Non-Network Synchronous mode. Also
+	 * enable the transmit and receive FIFO.
+	 *
+	 * FIXME: Little-endian samples require a different shift dir
+	 */
+	write_ssi_mask(&ssi->scr,
+		CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN,
+		CCSR_SSI_SCR_TFR_CLK_DIS |
+		i2s_mode |
+		(synchronous ? CCSR_SSI_SCR_SYN : 0));
+
+	write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
+		 CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS |
+		 CCSR_SSI_STCR_TSCKP, &ssi->stcr);
+
+	write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 |
+		 CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS |
+		 CCSR_SSI_SRCR_RSCKP, &ssi->srcr);
+	/*
+	 * The DC and PM bits are only used if the SSI is the clock master.
+	 */
+
+	/*
+	 * Set the watermark for transmit FIFI 0 and receive FIFO 0. We don't
+	 * use FIFO 1. We program the transmit water to signal a DMA transfer
+	 * if there are only two (or fewer) elements left in the FIFO. Two
+	 * elements equals one frame (left channel, right channel). This value,
+	 * however, depends on the depth of the transmit buffer.
+	 *
+	 * We set the watermark on the same level as the DMA burstsize.  For
+	 * fiq it is probably better to use the biggest possible watermark
+	 * size.
+	 */
+	if (ssi_private->use_dma)
+		wm = ssi_private->fifo_depth - 2;
+	else
+		wm = ssi_private->fifo_depth;
+
+	write_ssi(CCSR_SSI_SFCSR_TFWM0(wm) | CCSR_SSI_SFCSR_RFWM0(wm) |
+		CCSR_SSI_SFCSR_TFWM1(wm) | CCSR_SSI_SFCSR_RFWM1(wm),
+		&ssi->sfcsr);
+
+	/*
+	 * For ac97 interrupts are enabled with the startup of the substream
+	 * because it is also running without an active substream. Normally SSI
+	 * is only enabled when there is a substream.
+	 */
+	if (ssi_private->imx_ac97) {
+		/*
+		 * Setup the clock control register
+		 */
+		write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13),
+				&ssi->stccr);
+		write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13),
+				&ssi->srccr);
+
+		/*
+		 * Enable AC97 mode and startup the SSI
+		 */
+		write_ssi(CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV,
+				&ssi->sacnt);
+		write_ssi(0xff, &ssi->saccdis);
+		write_ssi(0x300, &ssi->saccen);
+
+		/*
+		 * Enable SSI, Transmit and Receive
+		 */
+		write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN |
+				CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE);
+
+		write_ssi(CCSR_SSI_SOR_WAIT(3), &ssi->sor);
+	}
+
+	return 0;
+}
+
+
 /**
  * fsl_ssi_startup: create a new substream
  *
@@ -319,70 +438,14 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 	 * and initialize the SSI registers.
 	 */
 	if (!ssi_private->first_stream) {
-		struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
-
 		ssi_private->first_stream = substream;
 
 		/*
-		 * Section 16.5 of the MPC8610 reference manual says that the
-		 * SSI needs to be disabled before updating the registers we set
-		 * here.
-		 */
-		write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0);
-
-		/*
-		 * Program the SSI into I2S Slave Non-Network Synchronous mode.
-		 * Also enable the transmit and receive FIFO.
-		 *
-		 * FIXME: Little-endian samples require a different shift dir
-		 */
-		write_ssi_mask(&ssi->scr,
-			CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN,
-			CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE
-			| (synchronous ? CCSR_SSI_SCR_SYN : 0));
-
-		write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
-			 CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS |
-			 CCSR_SSI_STCR_TSCKP, &ssi->stcr);
-
-		write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 |
-			 CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS |
-			 CCSR_SSI_SRCR_RSCKP, &ssi->srcr);
-
-		/*
-		 * The DC and PM bits are only used if the SSI is the clock
-		 * master.
-		 */
-
-		/* Enable the interrupts and DMA requests */
-		write_ssi(SIER_FLAGS, &ssi->sier);
-
-		/*
-		 * Set the watermark for transmit FIFI 0 and receive FIFO 0. We
-		 * don't use FIFO 1.  We program the transmit water to signal a
-		 * DMA transfer if there are only two (or fewer) elements left
-		 * in the FIFO.  Two elements equals one frame (left channel,
-		 * right channel).  This value, however, depends on the depth of
-		 * the transmit buffer.
-		 *
-		 * We program the receive FIFO to notify us if at least two
-		 * elements (one frame) have been written to the FIFO.  We could
-		 * make this value larger (and maybe we should), but this way
-		 * data will be written to memory as soon as it's available.
-		 */
-		write_ssi(CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) |
-			CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2),
-			&ssi->sfcsr);
-
-		/*
-		 * We keep the SSI disabled because if we enable it, then the
-		 * DMA controller will start.  It's not supposed to start until
-		 * the SCR.TE (or SCR.RE) bit is set, but it does anyway.  The
-		 * DMA controller will transfer one "BWC" of data (i.e. the
-		 * amount of data that the MR.BWC bits are set to).  The reason
-		 * this is bad is because at this point, the PCM driver has not
-		 * finished initializing the DMA controller.
+		 * fsl_ssi_setup was already called by ac97_init earlier if
+		 * the driver is in ac97 mode.
 		 */
+		if (!ssi_private->imx_ac97)
+			fsl_ssi_setup(ssi_private);
 	} else {
 		if (synchronous) {
 			struct snd_pcm_runtime *first_runtime =
@@ -492,6 +555,27 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai);
 	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+	unsigned int sier_bits;
+
+	/*
+	 *  Enable only the interrupts and DMA requests
+	 *  that are needed for the channel. As the fiq
+	 *  is polling for this bits, we have to ensure
+	 *  that this are aligned with the preallocated
+	 *  buffers
+	 */
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		if (ssi_private->use_dma)
+			sier_bits = SIER_FLAGS;
+		else
+			sier_bits = CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TFE0_EN;
+	} else {
+		if (ssi_private->use_dma)
+			sier_bits = SIER_FLAGS;
+		else
+			sier_bits = CCSR_SSI_SIER_RIE | CCSR_SSI_SIER_RFF0_EN;
+	}
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -510,12 +594,18 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 			write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0);
 		else
 			write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0);
+
+		if (!ssi_private->imx_ac97 && (read_ssi(&ssi->scr) &
+					(CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0)
+			write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0);
 		break;
 
 	default:
 		return -EINVAL;
 	}
 
+	write_ssi(sier_bits, &ssi->sier);
+
 	return 0;
 }
 
@@ -534,22 +624,13 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
 		ssi_private->first_stream = ssi_private->second_stream;
 
 	ssi_private->second_stream = NULL;
-
-	/*
-	 * If this is the last active substream, disable the SSI.
-	 */
-	if (!ssi_private->first_stream) {
-		struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
-
-		write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0);
-	}
 }
 
 static int fsl_ssi_dai_probe(struct snd_soc_dai *dai)
 {
 	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(dai);
 
-	if (ssi_private->ssi_on_imx) {
+	if (ssi_private->ssi_on_imx && ssi_private->use_dma) {
 		dai->playback_dma_data = &ssi_private->dma_params_tx;
 		dai->capture_dma_data = &ssi_private->dma_params_rx;
 	}
@@ -587,6 +668,133 @@ static const struct snd_soc_component_driver fsl_ssi_component = {
 	.name		= "fsl-ssi",
 };
 
+/**
+ * fsl_ssi_ac97_trigger: start and stop the AC97 receive/transmit.
+ *
+ * This function is called by ALSA to start, stop, pause, and resume the
+ * transfer of data.
+ */
+static int fsl_ssi_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
+			   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 ccsr_ssi __iomem *ssi = ssi_private->ssi;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_TIE |
+					CCSR_SSI_SIER_TFE0_EN);
+		else
+			write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_RIE |
+					CCSR_SSI_SIER_RFF0_EN);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_TIE |
+					CCSR_SSI_SIER_TFE0_EN, 0);
+		else
+			write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_RIE |
+					CCSR_SSI_SIER_RFF0_EN, 0);
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		write_ssi(CCSR_SSI_SOR_TX_CLR, &ssi->sor);
+	else
+		write_ssi(CCSR_SSI_SOR_RX_CLR, &ssi->sor);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops fsl_ssi_ac97_dai_ops = {
+	.startup	= fsl_ssi_startup,
+	.shutdown	= fsl_ssi_shutdown,
+	.trigger	= fsl_ssi_ac97_trigger,
+};
+
+static struct snd_soc_dai_driver fsl_ssi_ac97_dai = {
+	.ac97_control = 1,
+	.playback = {
+		.stream_name = "AC97 Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "AC97 Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = &fsl_ssi_ac97_dai_ops,
+};
+
+
+static struct fsl_ssi_private *fsl_ac97_data;
+
+static void fsl_ssi_ac97_init(void)
+{
+	fsl_ssi_setup(fsl_ac97_data);
+}
+
+void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+		unsigned short val)
+{
+	struct ccsr_ssi *ssi = fsl_ac97_data->ssi;
+	unsigned int lreg;
+	unsigned int lval;
+
+	if (reg > 0x7f)
+		return;
+
+
+	lreg = reg <<  12;
+	write_ssi(lreg, &ssi->sacadd);
+
+	lval = val << 4;
+	write_ssi(lval , &ssi->sacdat);
+
+	write_ssi_mask(&ssi->sacnt, CCSR_SSI_SACNT_RDWR_MASK,
+			CCSR_SSI_SACNT_WR);
+	udelay(100);
+}
+
+unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97,
+		unsigned short reg)
+{
+	struct ccsr_ssi *ssi = fsl_ac97_data->ssi;
+
+	unsigned short val = -1;
+	unsigned int lreg;
+
+	lreg = (reg & 0x7f) <<  12;
+	write_ssi(lreg, &ssi->sacadd);
+	write_ssi_mask(&ssi->sacnt, CCSR_SSI_SACNT_RDWR_MASK,
+			CCSR_SSI_SACNT_RD);
+
+	udelay(100);
+
+	val = (read_ssi(&ssi->sacdat) >> 4) & 0xffff;
+
+	return val;
+}
+
+static struct snd_ac97_bus_ops fsl_ssi_ac97_ops = {
+	.read		= fsl_ssi_ac97_read,
+	.write		= fsl_ssi_ac97_write,
+};
+
 /* Show the statistics of a flag only if its interrupt is enabled.  The
  * compiler will optimze this code to a no-op if the interrupt is not
  * enabled.
@@ -663,6 +871,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	struct resource res;
 	char name[64];
 	bool shared;
+	bool ac97 = false;
 
 	/* SSIs that are not connected on the board should have a
 	 *      status = "disabled"
@@ -673,14 +882,20 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 
 	/* We only support the SSI in "I2S Slave" mode */
 	sprop = of_get_property(np, "fsl,mode", NULL);
-	if (!sprop || strcmp(sprop, "i2s-slave")) {
+	if (!sprop) {
+		dev_err(&pdev->dev, "fsl,mode property is necessary\n");
+		return -EINVAL;
+	}
+	if (!strcmp(sprop, "ac97-slave")) {
+		ac97 = true;
+	} else if (strcmp(sprop, "i2s-slave")) {
 		dev_notice(&pdev->dev, "mode %s is unsupported\n", sprop);
 		return -ENODEV;
 	}
 
 	/* The DAI name is the last part of the full name of the node. */
 	p = strrchr(np->full_name, '/') + 1;
-	ssi_private = kzalloc(sizeof(struct fsl_ssi_private) + strlen(p),
+	ssi_private = devm_kzalloc(&pdev->dev, sizeof(*ssi_private) + strlen(p),
 			      GFP_KERNEL);
 	if (!ssi_private) {
 		dev_err(&pdev->dev, "could not allocate DAI object\n");
@@ -689,38 +904,41 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 
 	strcpy(ssi_private->name, p);
 
-	/* Initialize this copy of the CPU DAI driver structure */
-	memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template,
-	       sizeof(fsl_ssi_dai_template));
+	ssi_private->use_dma = !of_property_read_bool(np,
+			"fsl,fiq-stream-filter");
+
+	if (ac97) {
+		memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_ac97_dai,
+				sizeof(fsl_ssi_ac97_dai));
+
+		fsl_ac97_data = ssi_private;
+		ssi_private->imx_ac97 = true;
+
+		snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev);
+	} else {
+		/* Initialize this copy of the CPU DAI driver structure */
+		memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template,
+		       sizeof(fsl_ssi_dai_template));
+	}
 	ssi_private->cpu_dai_drv.name = ssi_private->name;
 
 	/* Get the addresses and IRQ */
 	ret = of_address_to_resource(np, 0, &res);
 	if (ret) {
 		dev_err(&pdev->dev, "could not determine device resources\n");
-		goto error_kmalloc;
+		return ret;
 	}
 	ssi_private->ssi = of_iomap(np, 0);
 	if (!ssi_private->ssi) {
 		dev_err(&pdev->dev, "could not map device resources\n");
-		ret = -ENOMEM;
-		goto error_kmalloc;
+		return -ENOMEM;
 	}
 	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;
+		return -ENXIO;
 	}
 
 	/* Are the RX and the TX clocks locked? */
@@ -739,13 +957,18 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 		u32 dma_events[2];
 		ssi_private->ssi_on_imx = true;
 
-		ssi_private->clk = clk_get(&pdev->dev, NULL);
+		ssi_private->clk = devm_clk_get(&pdev->dev, NULL);
 		if (IS_ERR(ssi_private->clk)) {
 			ret = PTR_ERR(ssi_private->clk);
 			dev_err(&pdev->dev, "could not get clock: %d\n", ret);
-			goto error_irq;
+			goto error_irqmap;
+		}
+		ret = clk_prepare_enable(ssi_private->clk);
+		if (ret) {
+			dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n",
+				ret);
+			goto error_irqmap;
 		}
-		clk_prepare_enable(ssi_private->clk);
 
 		/*
 		 * We have burstsize be "fifo_depth - 2" to match the SSI
@@ -763,24 +986,38 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 			&ssi_private->filter_data_tx;
 		ssi_private->dma_params_rx.filter_data =
 			&ssi_private->filter_data_rx;
-		/*
-		 * TODO: This is a temporary solution and should be changed
-		 * to use generic DMA binding later when the helplers get in.
-		 */
-		ret = of_property_read_u32_array(pdev->dev.of_node,
+		if (!of_property_read_bool(pdev->dev.of_node, "dmas") &&
+				ssi_private->use_dma) {
+			/*
+			 * FIXME: This is a temporary solution until all
+			 * necessary dma drivers support the generic dma
+			 * bindings.
+			 */
+			ret = of_property_read_u32_array(pdev->dev.of_node,
 					"fsl,ssi-dma-events", dma_events, 2);
-		if (ret) {
-			dev_err(&pdev->dev, "could not get dma events\n");
-			goto error_clk;
+			if (ret && ssi_private->use_dma) {
+				dev_err(&pdev->dev, "could not get dma events but fsl-ssi is configured to use DMA\n");
+				goto error_clk;
+			}
 		}
 
 		shared = of_device_is_compatible(of_get_parent(np),
 			    "fsl,spba-bus");
 
 		imx_pcm_dma_params_init_data(&ssi_private->filter_data_tx,
-			dma_events[0], shared);
+			dma_events[0], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI);
 		imx_pcm_dma_params_init_data(&ssi_private->filter_data_rx,
-			dma_events[1], shared);
+			dma_events[1], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI);
+	} else if (ssi_private->use_dma) {
+		/* The 'name' should not have any slashes in it. */
+		ret = devm_request_irq(&pdev->dev, 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;
+		}
 	}
 
 	/* Initialize the the device_attribute structure */
@@ -794,7 +1031,7 @@ static int 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_irq;
+		goto error_clk;
 	}
 
 	/* Register with ASoC */
@@ -808,9 +1045,30 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	}
 
 	if (ssi_private->ssi_on_imx) {
-		ret = imx_pcm_dma_init(pdev);
-		if (ret)
-			goto error_dev;
+		if (!ssi_private->use_dma) {
+
+			/*
+			 * Some boards use an incompatible codec. To get it
+			 * working, we are using imx-fiq-pcm-audio, that
+			 * can handle those codecs. DMA is not possible in this
+			 * situation.
+			 */
+
+			ssi_private->fiq_params.irq = ssi_private->irq;
+			ssi_private->fiq_params.base = ssi_private->ssi;
+			ssi_private->fiq_params.dma_params_rx =
+				&ssi_private->dma_params_rx;
+			ssi_private->fiq_params.dma_params_tx =
+				&ssi_private->dma_params_tx;
+
+			ret = imx_pcm_fiq_init(pdev, &ssi_private->fiq_params);
+			if (ret)
+				goto error_dev;
+		} else {
+			ret = imx_pcm_dma_init(pdev);
+			if (ret)
+				goto error_dev;
+		}
 	}
 
 	/*
@@ -845,6 +1103,9 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	}
 
 done:
+	if (ssi_private->imx_ac97)
+		fsl_ssi_ac97_init();
+
 	return 0;
 
 error_dai:
@@ -853,27 +1114,15 @@ error_dai:
 	snd_soc_unregister_component(&pdev->dev);
 
 error_dev:
-	dev_set_drvdata(&pdev->dev, NULL);
 	device_remove_file(&pdev->dev, dev_attr);
 
 error_clk:
-	if (ssi_private->ssi_on_imx) {
+	if (ssi_private->ssi_on_imx)
 		clk_disable_unprepare(ssi_private->clk);
-		clk_put(ssi_private->clk);
-	}
-
-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;
 }
 
@@ -883,20 +1132,15 @@ static int fsl_ssi_remove(struct platform_device *pdev)
 
 	if (!ssi_private->new_binding)
 		platform_device_unregister(ssi_private->pdev);
-	if (ssi_private->ssi_on_imx) {
+	if (ssi_private->ssi_on_imx)
 		imx_pcm_dma_exit(pdev);
-		clk_disable_unprepare(ssi_private->clk);
-		clk_put(ssi_private->clk);
-	}
 	snd_soc_unregister_component(&pdev->dev);
+	dev_set_drvdata(&pdev->dev, NULL);
 	device_remove_file(&pdev->dev, &ssi_private->dev_attr);
-
-	free_irq(ssi_private->irq, ssi_private);
+	if (ssi_private->ssi_on_imx)
+		clk_disable_unprepare(ssi_private->clk);
 	irq_dispose_mapping(ssi_private->irq);
 
-	kfree(ssi_private);
-	dev_set_drvdata(&pdev->dev, NULL);
-
 	return 0;
 }
 
@@ -919,6 +1163,7 @@ static struct platform_driver fsl_ssi_driver = {
 
 module_platform_driver(fsl_ssi_driver);
 
+MODULE_ALIAS("platform:fsl-ssi-dai");
 MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
 MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver");
 MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index e260f1f899db..ab17381cc981 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -73,8 +73,11 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf,
 	if (!buf)
 		return -ENOMEM;
 
-	if (audmux_clk)
-		clk_prepare_enable(audmux_clk);
+	if (audmux_clk) {
+		ret = clk_prepare_enable(audmux_clk);
+		if (ret)
+			return ret;
+	}
 
 	ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port));
 	pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port));
@@ -224,14 +227,19 @@ EXPORT_SYMBOL_GPL(imx_audmux_v1_configure_port);
 int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr,
 		unsigned int pdcr)
 {
+	int ret;
+
 	if (audmux_type != IMX31_AUDMUX)
 		return -EINVAL;
 
 	if (!audmux_base)
 		return -ENOSYS;
 
-	if (audmux_clk)
-		clk_prepare_enable(audmux_clk);
+	if (audmux_clk) {
+		ret = clk_prepare_enable(audmux_clk);
+		if (ret)
+			return ret;
+	}
 
 	writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port));
 	writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port));
@@ -243,6 +251,66 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr,
 }
 EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port);
 
+static int imx_audmux_parse_dt_defaults(struct platform_device *pdev,
+		struct device_node *of_node)
+{
+	struct device_node *child;
+
+	for_each_available_child_of_node(of_node, child) {
+		unsigned int port;
+		unsigned int ptcr = 0;
+		unsigned int pdcr = 0;
+		unsigned int pcr = 0;
+		unsigned int val;
+		int ret;
+		int i = 0;
+
+		ret = of_property_read_u32(child, "fsl,audmux-port", &port);
+		if (ret) {
+			dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%s\"\n",
+					child->full_name);
+			continue;
+		}
+		if (!of_property_read_bool(child, "fsl,port-config")) {
+			dev_warn(&pdev->dev, "child node \"%s\" does not have property fsl,port-config\n",
+					child->full_name);
+			continue;
+		}
+
+		for (i = 0; (ret = of_property_read_u32_index(child,
+					"fsl,port-config", i, &val)) == 0;
+				++i) {
+			if (audmux_type == IMX31_AUDMUX) {
+				if (i % 2)
+					pdcr |= val;
+				else
+					ptcr |= val;
+			} else {
+				pcr |= val;
+			}
+		}
+
+		if (ret != -EOVERFLOW) {
+			dev_err(&pdev->dev, "Failed to read u32 at index %d of child %s\n",
+					i, child->full_name);
+			continue;
+		}
+
+		if (audmux_type == IMX31_AUDMUX) {
+			if (i % 2) {
+				dev_err(&pdev->dev, "One pdcr value is missing in child node %s\n",
+						child->full_name);
+				continue;
+			}
+			imx_audmux_v2_configure_port(port, ptcr, pdcr);
+		} else {
+			imx_audmux_v1_configure_port(port, pcr);
+		}
+	}
+
+	return 0;
+}
+
 static int imx_audmux_probe(struct platform_device *pdev)
 {
 	struct resource *res;
@@ -267,6 +335,8 @@ static int imx_audmux_probe(struct platform_device *pdev)
 	if (audmux_type == IMX31_AUDMUX)
 		audmux_debugfs_init();
 
+	imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node);
+
 	return 0;
 }
 
diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h
index b8ff44b9dafa..38a4209af7c6 100644
--- a/sound/soc/fsl/imx-audmux.h
+++ b/sound/soc/fsl/imx-audmux.h
@@ -1,57 +1,7 @@
 #ifndef __IMX_AUDMUX_H
 #define __IMX_AUDMUX_H
 
-#define MX27_AUDMUX_HPCR1_SSI0		0
-#define MX27_AUDMUX_HPCR2_SSI1		1
-#define MX27_AUDMUX_HPCR3_SSI_PINS_4	2
-#define MX27_AUDMUX_PPCR1_SSI_PINS_1	3
-#define MX27_AUDMUX_PPCR2_SSI_PINS_2	4
-#define MX27_AUDMUX_PPCR3_SSI_PINS_3	5
-
-#define MX31_AUDMUX_PORT1_SSI0		0
-#define MX31_AUDMUX_PORT2_SSI1		1
-#define MX31_AUDMUX_PORT3_SSI_PINS_3	2
-#define MX31_AUDMUX_PORT4_SSI_PINS_4	3
-#define MX31_AUDMUX_PORT5_SSI_PINS_5	4
-#define MX31_AUDMUX_PORT6_SSI_PINS_6	5
-#define MX31_AUDMUX_PORT7_SSI_PINS_7	6
-
-#define MX51_AUDMUX_PORT1_SSI0		0
-#define MX51_AUDMUX_PORT2_SSI1		1
-#define MX51_AUDMUX_PORT3		2
-#define MX51_AUDMUX_PORT4		3
-#define MX51_AUDMUX_PORT5		4
-#define MX51_AUDMUX_PORT6		5
-#define MX51_AUDMUX_PORT7		6
-
-/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */
-#define IMX_AUDMUX_V1_PCR_INMMASK(x)	((x) & 0xff)
-#define IMX_AUDMUX_V1_PCR_INMEN		(1 << 8)
-#define IMX_AUDMUX_V1_PCR_TXRXEN	(1 << 10)
-#define IMX_AUDMUX_V1_PCR_SYN		(1 << 12)
-#define IMX_AUDMUX_V1_PCR_RXDSEL(x)	(((x) & 0x7) << 13)
-#define IMX_AUDMUX_V1_PCR_RFCSEL(x)	(((x) & 0xf) << 20)
-#define IMX_AUDMUX_V1_PCR_RCLKDIR	(1 << 24)
-#define IMX_AUDMUX_V1_PCR_RFSDIR	(1 << 25)
-#define IMX_AUDMUX_V1_PCR_TFCSEL(x)	(((x) & 0xf) << 26)
-#define IMX_AUDMUX_V1_PCR_TCLKDIR	(1 << 30)
-#define IMX_AUDMUX_V1_PCR_TFSDIR	(1 << 31)
-
-/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */
-#define IMX_AUDMUX_V2_PTCR_TFSDIR	(1 << 31)
-#define IMX_AUDMUX_V2_PTCR_TFSEL(x)	(((x) & 0xf) << 27)
-#define IMX_AUDMUX_V2_PTCR_TCLKDIR	(1 << 26)
-#define IMX_AUDMUX_V2_PTCR_TCSEL(x)	(((x) & 0xf) << 22)
-#define IMX_AUDMUX_V2_PTCR_RFSDIR	(1 << 21)
-#define IMX_AUDMUX_V2_PTCR_RFSEL(x)	(((x) & 0xf) << 17)
-#define IMX_AUDMUX_V2_PTCR_RCLKDIR	(1 << 16)
-#define IMX_AUDMUX_V2_PTCR_RCSEL(x)	(((x) & 0xf) << 12)
-#define IMX_AUDMUX_V2_PTCR_SYN		(1 << 11)
-
-#define IMX_AUDMUX_V2_PDCR_RXDSEL(x)	(((x) & 0x7) << 13)
-#define IMX_AUDMUX_V2_PDCR_TXRXEN	(1 << 12)
-#define IMX_AUDMUX_V2_PDCR_MODE(x)	(((x) & 0x3) << 8)
-#define IMX_AUDMUX_V2_PDCR_INMMASK(x)	((x) & 0xff)
+#include <dt-bindings/sound/fsl-imx-audmux.h>
 
 int imx_audmux_v1_configure_port(unsigned int port, unsigned int pcr);
 
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index 9df173c091a6..a3d60d4bea4c 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -90,6 +90,7 @@ static const struct snd_soc_dapm_route imx_mc13783_routes[] = {
 
 static struct snd_soc_card imx_mc13783 = {
 	.name		= "imx_mc13783",
+	.owner		= THIS_MODULE,
 	.dai_link	= imx_mc13783_dai_mc13783,
 	.num_links	= ARRAY_SIZE(imx_mc13783_dai_mc13783),
 	.dapm_widgets	= imx_mc13783_widget,
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
index fde4d2ea68c8..4dc1296688e9 100644
--- a/sound/soc/fsl/imx-pcm-dma.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -14,6 +14,7 @@
 #include <linux/platform_device.h>
 #include <linux/dmaengine.h>
 #include <linux/types.h>
+#include <linux/module.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -64,7 +65,6 @@ int imx_pcm_dma_init(struct platform_device *pdev)
 {
 	return snd_dmaengine_pcm_register(&pdev->dev, &imx_dmaengine_pcm_config,
 		SND_DMAENGINE_PCM_FLAG_NO_RESIDUE |
-		SND_DMAENGINE_PCM_FLAG_NO_DT |
 		SND_DMAENGINE_PCM_FLAG_COMPAT);
 }
 EXPORT_SYMBOL_GPL(imx_pcm_dma_init);
@@ -74,3 +74,5 @@ void imx_pcm_dma_exit(struct platform_device *pdev)
 	snd_dmaengine_pcm_unregister(&pdev->dev);
 }
 EXPORT_SYMBOL_GPL(imx_pcm_dma_exit);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index 310d90290320..34043c55f2a6 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -22,6 +22,7 @@
 #include <linux/slab.h>
 
 #include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
 #include <sound/initval.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -32,6 +33,7 @@
 #include <linux/platform_data/asoc-imx-ssi.h>
 
 #include "imx-ssi.h"
+#include "imx-pcm.h"
 
 struct imx_pcm_runtime_data {
 	unsigned int period;
@@ -366,9 +368,9 @@ static struct snd_soc_platform_driver imx_soc_platform_fiq = {
 	.pcm_free	= imx_pcm_fiq_free,
 };
 
-int imx_pcm_fiq_init(struct platform_device *pdev)
+int imx_pcm_fiq_init(struct platform_device *pdev,
+		struct imx_pcm_fiq_params *params)
 {
-	struct imx_ssi *ssi = platform_get_drvdata(pdev);
 	int ret;
 
 	ret = claim_fiq(&fh);
@@ -377,15 +379,15 @@ int imx_pcm_fiq_init(struct platform_device *pdev)
 		return ret;
 	}
 
-	mxc_set_irq_fiq(ssi->irq, 1);
-	ssi_irq = ssi->irq;
+	mxc_set_irq_fiq(params->irq, 1);
+	ssi_irq = params->irq;
 
-	imx_pcm_fiq = ssi->irq;
+	imx_pcm_fiq = params->irq;
 
-	imx_ssi_fiq_base = (unsigned long)ssi->base;
+	imx_ssi_fiq_base = (unsigned long)params->base;
 
-	ssi->dma_params_tx.maxburst = 4;
-	ssi->dma_params_rx.maxburst = 6;
+	params->dma_params_tx->maxburst = 4;
+	params->dma_params_rx->maxburst = 6;
 
 	ret = snd_soc_register_platform(&pdev->dev, &imx_soc_platform_fiq);
 	if (ret)
@@ -406,3 +408,5 @@ void imx_pcm_fiq_exit(struct platform_device *pdev)
 	snd_soc_unregister_platform(&pdev->dev);
 }
 EXPORT_SYMBOL_GPL(imx_pcm_fiq_exit);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h
index 67f656c7c320..5d5b73303e11 100644
--- a/sound/soc/fsl/imx-pcm.h
+++ b/sound/soc/fsl/imx-pcm.h
@@ -22,17 +22,23 @@
 
 static inline void
 imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data,
-	int dma, bool shared)
+	int dma, enum sdma_peripheral_type peripheral_type)
 {
 	dma_data->dma_request = dma;
 	dma_data->priority = DMA_PRIO_HIGH;
-	if (shared)
-		dma_data->peripheral_type = IMX_DMATYPE_SSI_SP;
-	else
-		dma_data->peripheral_type = IMX_DMATYPE_SSI;
+	dma_data->peripheral_type = peripheral_type;
 }
 
-#ifdef CONFIG_SND_SOC_IMX_PCM_DMA
+struct imx_pcm_fiq_params {
+	int irq;
+	void __iomem *base;
+
+	/* Pointer to original ssi driver to setup tx rx sizes */
+	struct snd_dmaengine_dai_dma_data *dma_params_rx;
+	struct snd_dmaengine_dai_dma_data *dma_params_tx;
+};
+
+#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA)
 int imx_pcm_dma_init(struct platform_device *pdev);
 void imx_pcm_dma_exit(struct platform_device *pdev);
 #else
@@ -46,11 +52,13 @@ static inline void imx_pcm_dma_exit(struct platform_device *pdev)
 }
 #endif
 
-#ifdef CONFIG_SND_SOC_IMX_PCM_FIQ
-int imx_pcm_fiq_init(struct platform_device *pdev);
+#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_FIQ)
+int imx_pcm_fiq_init(struct platform_device *pdev,
+		struct imx_pcm_fiq_params *params);
 void imx_pcm_fiq_exit(struct platform_device *pdev);
 #else
-static inline int imx_pcm_fiq_init(struct platform_device *pdev)
+static inline int imx_pcm_fiq_init(struct platform_device *pdev,
+		struct imx_pcm_fiq_params *params)
 {
 	return -ENODEV;
 }
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index 3f726e4f88db..389cbfa6dca7 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -129,8 +129,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
 	}
 
 	data->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
-	if (IS_ERR(data->codec_clk))
+	if (IS_ERR(data->codec_clk)) {
+		ret = PTR_ERR(data->codec_clk);
 		goto fail;
+	}
 
 	data->clk_frequency = clk_get_rate(data->codec_clk);
 
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
new file mode 100644
index 000000000000..816013b0ebba
--- /dev/null
+++ b/sound/soc/fsl/imx-spdif.c
@@ -0,0 +1,148 @@
+/*
+ * Copyright (C) 2013 Freescale Semiconductor, Inc.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+
+struct imx_spdif_data {
+	struct snd_soc_dai_link dai[2];
+	struct snd_soc_card card;
+	struct platform_device *txdev;
+	struct platform_device *rxdev;
+};
+
+static int imx_spdif_audio_probe(struct platform_device *pdev)
+{
+	struct device_node *spdif_np, *np = pdev->dev.of_node;
+	struct imx_spdif_data *data;
+	int ret = 0, num_links = 0;
+
+	spdif_np = of_parse_phandle(np, "spdif-controller", 0);
+	if (!spdif_np) {
+		dev_err(&pdev->dev, "failed to find spdif-controller\n");
+		ret = -EINVAL;
+		goto end;
+	}
+
+	data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+	if (!data) {
+		dev_err(&pdev->dev, "failed to allocate memory\n");
+		ret = -ENOMEM;
+		goto end;
+	}
+
+	if (of_property_read_bool(np, "spdif-out")) {
+		data->dai[num_links].name = "S/PDIF TX";
+		data->dai[num_links].stream_name = "S/PDIF PCM Playback";
+		data->dai[num_links].codec_dai_name = "dit-hifi";
+		data->dai[num_links].codec_name = "spdif-dit";
+		data->dai[num_links].cpu_of_node = spdif_np;
+		data->dai[num_links].platform_of_node = spdif_np;
+		num_links++;
+
+		data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0);
+		if (IS_ERR(data->txdev)) {
+			ret = PTR_ERR(data->txdev);
+			dev_err(&pdev->dev, "register dit failed: %d\n", ret);
+			goto end;
+		}
+	}
+
+	if (of_property_read_bool(np, "spdif-in")) {
+		data->dai[num_links].name = "S/PDIF RX";
+		data->dai[num_links].stream_name = "S/PDIF PCM Capture";
+		data->dai[num_links].codec_dai_name = "dir-hifi";
+		data->dai[num_links].codec_name = "spdif-dir";
+		data->dai[num_links].cpu_of_node = spdif_np;
+		data->dai[num_links].platform_of_node = spdif_np;
+		num_links++;
+
+		data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0);
+		if (IS_ERR(data->rxdev)) {
+			ret = PTR_ERR(data->rxdev);
+			dev_err(&pdev->dev, "register dir failed: %d\n", ret);
+			goto error_dit;
+		}
+	}
+
+	if (!num_links) {
+		dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n");
+		goto error_dir;
+	}
+
+	data->card.dev = &pdev->dev;
+	data->card.num_links = num_links;
+	data->card.dai_link = data->dai;
+
+	ret = snd_soc_of_parse_card_name(&data->card, "model");
+	if (ret)
+		goto error_dir;
+
+	ret = snd_soc_register_card(&data->card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret);
+		goto error_dir;
+	}
+
+	platform_set_drvdata(pdev, data);
+
+	goto end;
+
+error_dir:
+	if (data->rxdev)
+		platform_device_unregister(data->rxdev);
+error_dit:
+	if (data->txdev)
+		platform_device_unregister(data->txdev);
+end:
+	if (spdif_np)
+		of_node_put(spdif_np);
+
+	return ret;
+}
+
+static int imx_spdif_audio_remove(struct platform_device *pdev)
+{
+	struct imx_spdif_data *data = platform_get_drvdata(pdev);
+
+	if (data->rxdev)
+		platform_device_unregister(data->rxdev);
+	if (data->txdev)
+		platform_device_unregister(data->txdev);
+
+	snd_soc_unregister_card(&data->card);
+
+	return 0;
+}
+
+static const struct of_device_id imx_spdif_dt_ids[] = {
+	{ .compatible = "fsl,imx-audio-spdif", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids);
+
+static struct platform_driver imx_spdif_driver = {
+	.driver = {
+		.name = "imx-spdif",
+		.owner = THIS_MODULE,
+		.of_match_table = imx_spdif_dt_ids,
+	},
+	.probe = imx_spdif_audio_probe,
+	.remove = imx_spdif_audio_remove,
+};
+
+module_platform_driver(imx_spdif_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-spdif");
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 51be3772cba9..f58bcd85c07f 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -571,13 +571,13 @@ static int imx_ssi_probe(struct platform_device *pdev)
 	res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0");
 	if (res) {
 		imx_pcm_dma_params_init_data(&ssi->filter_data_tx, res->start,
-			false);
+			IMX_DMATYPE_SSI);
 	}
 
 	res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx0");
 	if (res) {
 		imx_pcm_dma_params_init_data(&ssi->filter_data_rx, res->start,
-			false);
+			IMX_DMATYPE_SSI);
 	}
 
 	platform_set_drvdata(pdev, ssi);
@@ -595,7 +595,12 @@ static int imx_ssi_probe(struct platform_device *pdev)
 		goto failed_register;
 	}
 
-	ret = imx_pcm_fiq_init(pdev);
+	ssi->fiq_params.irq = ssi->irq;
+	ssi->fiq_params.base = ssi->base;
+	ssi->fiq_params.dma_params_rx = &ssi->dma_params_rx;
+	ssi->fiq_params.dma_params_tx = &ssi->dma_params_tx;
+
+	ret = imx_pcm_fiq_init(pdev, &ssi->fiq_params);
 	if (ret)
 		goto failed_pcm_fiq;
 
diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h
index d5003cefca8d..fb1616ba8c59 100644
--- a/sound/soc/fsl/imx-ssi.h
+++ b/sound/soc/fsl/imx-ssi.h
@@ -209,6 +209,7 @@ struct imx_ssi {
 	struct snd_dmaengine_dai_dma_data dma_params_tx;
 	struct imx_dma_data filter_data_tx;
 	struct imx_dma_data filter_data_rx;
+	struct imx_pcm_fiq_params fiq_params;
 
 	int enabled;
 };
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 52a36a90f4f4..1d70e278e915 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -217,7 +217,8 @@ static int imx_wm8962_probe(struct platform_device *pdev)
 	codec_dev = of_find_i2c_device_by_node(codec_np);
 	if (!codec_dev || !codec_dev->driver) {
 		dev_err(&pdev->dev, "failed to find codec platform device\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto fail;
 	}
 
 	data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 6cf8355a8542..8c49147db84c 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -105,6 +105,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev)
 static struct platform_driver asoc_simple_card = {
 	.driver = {
 		.name	= "asoc-simple-card",
+		.owner = THIS_MODULE,
 	},
 	.probe		= asoc_simple_card_probe,
 	.remove		= asoc_simple_card_remove,
@@ -112,6 +113,7 @@ static struct platform_driver asoc_simple_card = {
 
 module_platform_driver(asoc_simple_card);
 
+MODULE_ALIAS("platform:asoc-simple-card");
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("ASoC Simple Sound Card");
 MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index c62d715235e2..78ed4a42ad21 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -1,19 +1,15 @@
 config SND_KIRKWOOD_SOC
-	tristate "SoC Audio for the Marvell Kirkwood chip"
-	depends on ARCH_KIRKWOOD
+	tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
+	depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Kirkwood I2S interface. You will also need to select the
 	  audio interfaces to support below.
 
-config SND_KIRKWOOD_SOC_I2S
-	tristate
-
 config SND_KIRKWOOD_SOC_OPENRD
 	tristate "SoC Audio support for Kirkwood Openrd Client"
-	depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE)
+	depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST)
 	depends on I2C
-	select SND_KIRKWOOD_SOC_I2S
 	select SND_SOC_CS42L51
 	help
 	  Say Y if you want to add support for SoC audio on
@@ -21,8 +17,7 @@ config SND_KIRKWOOD_SOC_OPENRD
 
 config SND_KIRKWOOD_SOC_T5325
 	tristate "SoC Audio support for HP t5325"
-	depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C
-	select SND_KIRKWOOD_SOC_I2S
+	depends on SND_KIRKWOOD_SOC && (MACH_T5325 || COMPILE_TEST) && I2C
 	select SND_SOC_ALC5623
 	help
 	  Say Y if you want to add support for SoC audio on
diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile
index 3e62ae9e7bbe..9e781385cb88 100644
--- a/sound/soc/kirkwood/Makefile
+++ b/sound/soc/kirkwood/Makefile
@@ -1,8 +1,6 @@
-snd-soc-kirkwood-objs := kirkwood-dma.o
-snd-soc-kirkwood-i2s-objs := kirkwood-i2s.o
+snd-soc-kirkwood-objs := kirkwood-dma.o kirkwood-i2s.o
 
 obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o
-obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o
 
 snd-soc-openrd-objs := kirkwood-openrd.o
 snd-soc-t5325-objs := kirkwood-t5325.o
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index a9f14530c3db..b238434f92b0 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -33,11 +33,11 @@
 	 SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE | \
 	 SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE)
 
-struct kirkwood_dma_priv {
-	struct snd_pcm_substream *play_stream;
-	struct snd_pcm_substream *rec_stream;
-	struct kirkwood_dma_data *data;
-};
+static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs)
+{
+	struct snd_soc_pcm_runtime *soc_runtime = subs->private_data;
+	return snd_soc_dai_get_drvdata(soc_runtime->cpu_dai);
+}
 
 static struct snd_pcm_hardware kirkwood_dma_snd_hw = {
 	.info = (SNDRV_PCM_INFO_INTERLEAVED |
@@ -51,7 +51,7 @@ static struct snd_pcm_hardware kirkwood_dma_snd_hw = {
 	.rate_max		= 384000,
 	.channels_min		= 1,
 	.channels_max		= 8,
-	.buffer_bytes_max	= KIRKWOOD_SND_MAX_PERIOD_BYTES * KIRKWOOD_SND_MAX_PERIODS,
+	.buffer_bytes_max	= KIRKWOOD_SND_MAX_BUFFER_BYTES,
 	.period_bytes_min	= KIRKWOOD_SND_MIN_PERIOD_BYTES,
 	.period_bytes_max	= KIRKWOOD_SND_MAX_PERIOD_BYTES,
 	.periods_min		= KIRKWOOD_SND_MIN_PERIODS,
@@ -63,8 +63,7 @@ static u64 kirkwood_dma_dmamask = DMA_BIT_MASK(32);
 
 static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id)
 {
-	struct kirkwood_dma_priv *prdata = dev_id;
-	struct kirkwood_dma_data *priv = prdata->data;
+	struct kirkwood_dma_data *priv = dev_id;
 	unsigned long mask, status, cause;
 
 	mask = readl(priv->io + KIRKWOOD_INT_MASK);
@@ -89,10 +88,10 @@ static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id)
 	writel(status, priv->io + KIRKWOOD_INT_CAUSE);
 
 	if (status & KIRKWOOD_INT_CAUSE_PLAY_BYTES)
-		snd_pcm_period_elapsed(prdata->play_stream);
+		snd_pcm_period_elapsed(priv->substream_play);
 
 	if (status & KIRKWOOD_INT_CAUSE_REC_BYTES)
-		snd_pcm_period_elapsed(prdata->rec_stream);
+		snd_pcm_period_elapsed(priv->substream_rec);
 
 	return IRQ_HANDLED;
 }
@@ -126,15 +125,10 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream)
 {
 	int err;
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct snd_soc_platform *platform = soc_runtime->platform;
-	struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai;
-	struct kirkwood_dma_data *priv;
-	struct kirkwood_dma_priv *prdata = snd_soc_platform_get_drvdata(platform);
+	struct kirkwood_dma_data *priv = kirkwood_priv(substream);
 	const struct mbus_dram_target_info *dram;
 	unsigned long addr;
 
-	priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
 	snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw);
 
 	/* Ensure that all constraints linked to dma burst are fulfilled */
@@ -157,21 +151,11 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream)
 	if (err < 0)
 		return err;
 
-	if (prdata == NULL) {
-		prdata = kzalloc(sizeof(struct kirkwood_dma_priv), GFP_KERNEL);
-		if (prdata == NULL)
-			return -ENOMEM;
-
-		prdata->data = priv;
-
+	if (!priv->substream_play && !priv->substream_rec) {
 		err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED,
-				  "kirkwood-i2s", prdata);
-		if (err) {
-			kfree(prdata);
+				  "kirkwood-i2s", priv);
+		if (err)
 			return -EBUSY;
-		}
-
-		snd_soc_platform_set_drvdata(platform, prdata);
 
 		/*
 		 * Enable Error interrupts. We're only ack'ing them but
@@ -183,11 +167,11 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream)
 	dram = mv_mbus_dram_info();
 	addr = substream->dma_buffer.addr;
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		prdata->play_stream = substream;
+		priv->substream_play = substream;
 		kirkwood_dma_conf_mbus_windows(priv->io,
 			KIRKWOOD_PLAYBACK_WIN, addr, dram);
 	} else {
-		prdata->rec_stream = substream;
+		priv->substream_rec = substream;
 		kirkwood_dma_conf_mbus_windows(priv->io,
 			KIRKWOOD_RECORD_WIN, addr, dram);
 	}
@@ -197,27 +181,19 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream)
 
 static int kirkwood_dma_close(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai;
-	struct snd_soc_platform *platform = soc_runtime->platform;
-	struct kirkwood_dma_priv *prdata = snd_soc_platform_get_drvdata(platform);
-	struct kirkwood_dma_data *priv;
-
-	priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
+	struct kirkwood_dma_data *priv = kirkwood_priv(substream);
 
-	if (!prdata || !priv)
+	if (!priv)
 		return 0;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		prdata->play_stream = NULL;
+		priv->substream_play = NULL;
 	else
-		prdata->rec_stream = NULL;
+		priv->substream_rec = NULL;
 
-	if (!prdata->play_stream && !prdata->rec_stream) {
+	if (!priv->substream_play && !priv->substream_rec) {
 		writel(0, priv->io + KIRKWOOD_ERR_MASK);
-		free_irq(priv->irq, prdata);
-		kfree(prdata);
-		snd_soc_platform_set_drvdata(platform, NULL);
+		free_irq(priv->irq, priv);
 	}
 
 	return 0;
@@ -243,13 +219,9 @@ static int kirkwood_dma_hw_free(struct snd_pcm_substream *substream)
 static int kirkwood_dma_prepare(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai;
-	struct kirkwood_dma_data *priv;
+	struct kirkwood_dma_data *priv = kirkwood_priv(substream);
 	unsigned long size, count;
 
-	priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
-
 	/* compute buffer size in term of "words" as requested in specs */
 	size = frames_to_bytes(runtime, runtime->buffer_size);
 	size = (size>>2)-1;
@@ -272,13 +244,9 @@ static int kirkwood_dma_prepare(struct snd_pcm_substream *substream)
 static snd_pcm_uframes_t kirkwood_dma_pointer(struct snd_pcm_substream
 						*substream)
 {
-	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai;
-	struct kirkwood_dma_data *priv;
+	struct kirkwood_dma_data *priv = kirkwood_priv(substream);
 	snd_pcm_uframes_t count;
 
-	priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
-
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		count = bytes_to_frames(substream->runtime,
 			readl(priv->io + KIRKWOOD_PLAY_BYTE_COUNT));
@@ -366,36 +334,8 @@ static void kirkwood_dma_free_dma_buffers(struct snd_pcm *pcm)
 	}
 }
 
-static struct snd_soc_platform_driver kirkwood_soc_platform = {
+struct snd_soc_platform_driver kirkwood_soc_platform = {
 	.ops		= &kirkwood_dma_ops,
 	.pcm_new	= kirkwood_dma_new,
 	.pcm_free	= kirkwood_dma_free_dma_buffers,
 };
-
-static int kirkwood_soc_platform_probe(struct platform_device *pdev)
-{
-	return snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform);
-}
-
-static int kirkwood_soc_platform_remove(struct platform_device *pdev)
-{
-	snd_soc_unregister_platform(&pdev->dev);
-	return 0;
-}
-
-static struct platform_driver kirkwood_pcm_driver = {
-	.driver = {
-			.name = "kirkwood-pcm-audio",
-			.owner = THIS_MODULE,
-	},
-
-	.probe = kirkwood_soc_platform_probe,
-	.remove = kirkwood_soc_platform_remove,
-};
-
-module_platform_driver(kirkwood_pcm_driver);
-
-MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
-MODULE_DESCRIPTION("Marvell Kirkwood Audio DMA module");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:kirkwood-pcm-audio");
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 4c9dad3263c5..7fce340ab3ef 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -22,13 +22,12 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <linux/platform_data/asoc-kirkwood.h>
+#include <linux/of.h>
+
 #include "kirkwood.h"
 
-#define DRV_NAME	"kirkwood-i2s"
+#define DRV_NAME	"mvebu-audio"
 
-#define KIRKWOOD_I2S_RATES \
-	(SNDRV_PCM_RATE_44100 | \
-	 SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
 #define KIRKWOOD_I2S_FORMATS \
 	(SNDRV_PCM_FMTBIT_S16_LE | \
 	 SNDRV_PCM_FMTBIT_S24_LE | \
@@ -105,14 +104,16 @@ static void kirkwood_set_rate(struct snd_soc_dai *dai,
 	uint32_t clks_ctrl;
 
 	if (rate == 44100 || rate == 48000 || rate == 96000) {
-		/* use internal dco for supported rates */
+		/* use internal dco for the supported rates
+		 * defined in kirkwood_i2s_dai */
 		dev_dbg(dai->dev, "%s: dco set rate = %lu\n",
 			__func__, rate);
 		kirkwood_set_dco(priv->io, rate);
 
 		clks_ctrl = KIRKWOOD_MCLK_SOURCE_DCO;
-	} else if (!IS_ERR(priv->extclk)) {
-		/* use optional external clk for other rates */
+	} else {
+		/* use the external clock for the other rates
+		 * defined in kirkwood_i2s_dai_extclk */
 		dev_dbg(dai->dev, "%s: extclk set rate = %lu -> %lu\n",
 			__func__, rate, 256 * rate);
 		clk_set_rate(priv->extclk, 256 * rate);
@@ -199,8 +200,7 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream,
 			ctl_play |= KIRKWOOD_PLAYCTL_MONO_OFF;
 
 		priv->ctl_play &= ~(KIRKWOOD_PLAYCTL_MONO_MASK |
-				    KIRKWOOD_PLAYCTL_I2S_EN |
-				    KIRKWOOD_PLAYCTL_SPDIF_EN |
+				    KIRKWOOD_PLAYCTL_ENABLE_MASK |
 				    KIRKWOOD_PLAYCTL_SIZE_MASK);
 		priv->ctl_play |= ctl_play;
 	} else {
@@ -244,8 +244,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_START:
 		/* configure */
 		ctl = priv->ctl_play;
-		value = ctl & ~(KIRKWOOD_PLAYCTL_I2S_EN |
-				KIRKWOOD_PLAYCTL_SPDIF_EN);
+		value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK;
 		writel(value, priv->io + KIRKWOOD_PLAYCTL);
 
 		/* enable interrupts */
@@ -267,7 +266,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
 		writel(value, priv->io + KIRKWOOD_INT_MASK);
 
 		/* disable all playbacks */
-		ctl &= ~(KIRKWOOD_PLAYCTL_I2S_EN | KIRKWOOD_PLAYCTL_SPDIF_EN);
+		ctl &= ~KIRKWOOD_PLAYCTL_ENABLE_MASK;
 		writel(ctl, priv->io + KIRKWOOD_PLAYCTL);
 		break;
 
@@ -387,7 +386,7 @@ static int kirkwood_i2s_probe(struct snd_soc_dai *dai)
 
 	/* disable playback/record */
 	value = readl(priv->io + KIRKWOOD_PLAYCTL);
-	value &= ~(KIRKWOOD_PLAYCTL_I2S_EN|KIRKWOOD_PLAYCTL_SPDIF_EN);
+	value &= ~KIRKWOOD_PLAYCTL_ENABLE_MASK;
 	writel(value, priv->io + KIRKWOOD_PLAYCTL);
 
 	value = readl(priv->io + KIRKWOOD_RECCTL);
@@ -398,11 +397,6 @@ static int kirkwood_i2s_probe(struct snd_soc_dai *dai)
 
 }
 
-static int kirkwood_i2s_remove(struct snd_soc_dai *dai)
-{
-	return 0;
-}
-
 static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = {
 	.startup	= kirkwood_i2s_startup,
 	.trigger	= kirkwood_i2s_trigger,
@@ -413,17 +407,18 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = {
 
 static struct snd_soc_dai_driver kirkwood_i2s_dai = {
 	.probe = kirkwood_i2s_probe,
-	.remove = kirkwood_i2s_remove,
 	.playback = {
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = KIRKWOOD_I2S_RATES,
+		.rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+				SNDRV_PCM_RATE_96000,
 		.formats = KIRKWOOD_I2S_FORMATS,
 	},
 	.capture = {
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = KIRKWOOD_I2S_RATES,
+		.rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+				SNDRV_PCM_RATE_96000,
 		.formats = KIRKWOOD_I2S_FORMATS,
 	},
 	.ops = &kirkwood_i2s_dai_ops,
@@ -431,7 +426,6 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai = {
 
 static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = {
 	.probe = kirkwood_i2s_probe,
-	.remove = kirkwood_i2s_remove,
 	.playback = {
 		.channels_min = 1,
 		.channels_max = 2,
@@ -461,6 +455,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 	struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai;
 	struct kirkwood_dma_data *priv;
 	struct resource *mem;
+	struct device_node *np = pdev->dev.of_node;
 	int err;
 
 	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
@@ -481,14 +476,16 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 		return -ENXIO;
 	}
 
-	if (!data) {
-		dev_err(&pdev->dev, "no platform data ?!\n");
+	if (np) {
+		priv->burst = 128;		/* might be 32 or 128 */
+	} else if (data) {
+		priv->burst = data->burst;
+	} else {
+		dev_err(&pdev->dev, "no DT nor platform data ?!\n");
 		return -EINVAL;
 	}
 
-	priv->burst = data->burst;
-
-	priv->clk = devm_clk_get(&pdev->dev, NULL);
+	priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL);
 	if (IS_ERR(priv->clk)) {
 		dev_err(&pdev->dev, "no clock\n");
 		return PTR_ERR(priv->clk);
@@ -498,10 +495,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 	if (err < 0)
 		return err;
 
-	priv->extclk = clk_get(&pdev->dev, "extclk");
+	priv->extclk = devm_clk_get(&pdev->dev, "extclk");
 	if (!IS_ERR(priv->extclk)) {
 		if (priv->extclk == priv->clk) {
-			clk_put(priv->extclk);
+			devm_clk_put(&pdev->dev, priv->extclk);
 			priv->extclk = ERR_PTR(-EINVAL);
 		} else {
 			dev_info(&pdev->dev, "found external clock\n");
@@ -515,7 +512,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 	priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24;
 
 	/* Select the burst size */
-	if (data->burst == 32) {
+	if (priv->burst == 32) {
 		priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32;
 		priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32;
 	} else {
@@ -525,14 +522,22 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 
 	err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component,
 					 soc_dai, 1);
-	if (!err)
-		return 0;
-	dev_err(&pdev->dev, "snd_soc_register_component failed\n");
+	if (err) {
+		dev_err(&pdev->dev, "snd_soc_register_component failed\n");
+		goto err_component;
+	}
 
-	if (!IS_ERR(priv->extclk)) {
-		clk_disable_unprepare(priv->extclk);
-		clk_put(priv->extclk);
+	err = snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform);
+	if (err) {
+		dev_err(&pdev->dev, "snd_soc_register_platform failed\n");
+		goto err_platform;
 	}
+	return 0;
+ err_platform:
+	snd_soc_unregister_component(&pdev->dev);
+ err_component:
+	if (!IS_ERR(priv->extclk))
+		clk_disable_unprepare(priv->extclk);
 	clk_disable_unprepare(priv->clk);
 
 	return err;
@@ -542,23 +547,31 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev)
 {
 	struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev);
 
+	snd_soc_unregister_platform(&pdev->dev);
 	snd_soc_unregister_component(&pdev->dev);
 
-	if (!IS_ERR(priv->extclk)) {
+	if (!IS_ERR(priv->extclk))
 		clk_disable_unprepare(priv->extclk);
-		clk_put(priv->extclk);
-	}
 	clk_disable_unprepare(priv->clk);
 
 	return 0;
 }
 
+#ifdef CONFIG_OF
+static struct of_device_id mvebu_audio_of_match[] = {
+	{ .compatible = "marvell,mvebu-audio" },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, mvebu_audio_of_match);
+#endif
+
 static struct platform_driver kirkwood_i2s_driver = {
 	.probe  = kirkwood_i2s_dev_probe,
 	.remove = kirkwood_i2s_dev_remove,
 	.driver = {
 		.name = DRV_NAME,
 		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(mvebu_audio_of_match),
 	},
 };
 
@@ -568,4 +581,4 @@ module_platform_driver(kirkwood_i2s_driver);
 MODULE_AUTHOR("Arnaud Patard, <arnaud.patard@rtp-net.org>");
 MODULE_DESCRIPTION("Kirkwood I2S SoC Interface");
 MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:kirkwood-i2s");
+MODULE_ALIAS("platform:mvebu-audio");
diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c
index b979c7154715..025be0e97164 100644
--- a/sound/soc/kirkwood/kirkwood-openrd.c
+++ b/sound/soc/kirkwood/kirkwood-openrd.c
@@ -16,9 +16,7 @@
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <sound/soc.h>
-#include <mach/kirkwood.h>
 #include <linux/platform_data/asoc-kirkwood.h>
-#include <asm/mach-types.h>
 #include "../codecs/cs42l51.h"
 
 static int openrd_client_hw_params(struct snd_pcm_substream *substream,
@@ -54,8 +52,8 @@ static struct snd_soc_dai_link openrd_client_dai[] = {
 {
 	.name = "CS42L51",
 	.stream_name = "CS42L51 HiFi",
-	.cpu_dai_name = "kirkwood-i2s",
-	.platform_name = "kirkwood-pcm-audio",
+	.cpu_dai_name = "mvebu-audio",
+	.platform_name = "mvebu-audio",
 	.codec_dai_name = "cs42l51-hifi",
 	.codec_name = "cs42l51-codec.0-004a",
 	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c
index 1d0ed6f8add7..27545b0c4856 100644
--- a/sound/soc/kirkwood/kirkwood-t5325.c
+++ b/sound/soc/kirkwood/kirkwood-t5325.c
@@ -15,9 +15,7 @@
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <sound/soc.h>
-#include <mach/kirkwood.h>
 #include <linux/platform_data/asoc-kirkwood.h>
-#include <asm/mach-types.h>
 #include "../codecs/alc5623.h"
 
 static int t5325_hw_params(struct snd_pcm_substream *substream,
@@ -70,8 +68,8 @@ static struct snd_soc_dai_link t5325_dai[] = {
 {
 	.name = "ALC5621",
 	.stream_name = "ALC5621 HiFi",
-	.cpu_dai_name = "kirkwood-i2s",
-	.platform_name = "kirkwood-pcm-audio",
+	.cpu_dai_name = "mvebu-audio",
+	.platform_name = "mvebu-audio",
 	.codec_dai_name = "alc5621-hifi",
 	.codec_name = "alc562x-codec.0-001a",
 	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h
index 4d92637ddb3f..f8e1ccc1c58c 100644
--- a/sound/soc/kirkwood/kirkwood.h
+++ b/sound/soc/kirkwood/kirkwood.h
@@ -54,7 +54,7 @@
 #define KIRKWOOD_PLAYCTL_MONO_OFF		(0<<5)
 #define KIRKWOOD_PLAYCTL_I2S_MUTE		(1<<7)
 #define KIRKWOOD_PLAYCTL_SPDIF_EN		(1<<4)
-#define KIRKWOOD_PLAYCTL_I2S_EN		(1<<3)
+#define KIRKWOOD_PLAYCTL_I2S_EN			(1<<3)
 #define KIRKWOOD_PLAYCTL_SIZE_MASK		(7<<0)
 #define KIRKWOOD_PLAYCTL_SIZE_16		(7<<0)
 #define KIRKWOOD_PLAYCTL_SIZE_16_C		(3<<0)
@@ -62,6 +62,9 @@
 #define KIRKWOOD_PLAYCTL_SIZE_24		(1<<0)
 #define KIRKWOOD_PLAYCTL_SIZE_32		(0<<0)
 
+#define KIRKWOOD_PLAYCTL_ENABLE_MASK		(KIRKWOOD_PLAYCTL_SPDIF_EN | \
+						 KIRKWOOD_PLAYCTL_I2S_EN)
+
 #define KIRKWOOD_PLAY_BUF_ADDR			0x1104
 #define KIRKWOOD_PLAY_BUF_SIZE			0x1108
 #define KIRKWOOD_PLAY_BYTE_COUNT		0x110C
@@ -122,6 +125,8 @@
 #define KIRKWOOD_SND_MAX_PERIODS		16
 #define KIRKWOOD_SND_MIN_PERIOD_BYTES		0x4000
 #define KIRKWOOD_SND_MAX_PERIOD_BYTES		0x4000
+#define KIRKWOOD_SND_MAX_BUFFER_BYTES		(KIRKWOOD_SND_MAX_PERIOD_BYTES \
+						 * KIRKWOOD_SND_MAX_PERIODS)
 
 struct kirkwood_dma_data {
 	void __iomem *io;
@@ -129,8 +134,12 @@ struct kirkwood_dma_data {
 	struct clk *extclk;
 	uint32_t ctl_play;
 	uint32_t ctl_rec;
+	struct snd_pcm_substream *substream_play;
+	struct snd_pcm_substream *substream_rec;
 	int irq;
 	int burst;
 };
 
+extern struct snd_soc_platform_driver kirkwood_soc_platform;
+
 #endif
diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig
index 78d321cbe8b4..219235c02212 100644
--- a/sound/soc/mxs/Kconfig
+++ b/sound/soc/mxs/Kconfig
@@ -1,6 +1,7 @@
 menuconfig SND_MXS_SOC
 	tristate "SoC Audio for Freescale MXS CPUs"
-	depends on ARCH_MXS
+	depends on ARCH_MXS || COMPILE_TEST
+	depends on COMMON_CLK
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 	help
 	  Say Y or M if you want to add support for codecs attached to
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index 54511c5e6a7c..b56b8a0e8deb 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -31,7 +31,6 @@
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
-#include <asm/mach-types.h>
 
 #include "mxs-saif.h"
 
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 1b134d72f120..4bb273786ff3 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -25,7 +25,6 @@
 #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"
@@ -51,18 +50,27 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	/* Sgtl5000 sysclk should be >= 8MHz and <= 27M */
-	if (mclk < 8000000 || mclk > 27000000)
+	if (mclk < 8000000 || mclk > 27000000) {
+		dev_err(codec_dai->dev, "Invalid mclk frequency: %u.%03uMHz\n",
+			mclk / 1000000, mclk / 1000 % 1000);
 		return -EINVAL;
+	}
 
 	/* Set SGTL5000's SYSCLK (provided by SAIF MCLK) */
 	ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, mclk, 0);
-	if (ret)
+	if (ret) {
+		dev_err(codec_dai->dev, "Failed to set sysclk to %u.%03uMHz\n",
+			mclk / 1000000, mclk / 1000 % 1000);
 		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)
+	if (ret) {
+		dev_err(cpu_dai->dev, "Failed to set sysclk to %u.%03uMHz\n",
+			mclk / 1000000, mclk / 1000 % 1000);
 		return ret;
+	}
 
 	/* set codec to slave mode */
 	dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
@@ -70,13 +78,19 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream,
 
 	/* set codec DAI configuration */
 	ret = snd_soc_dai_set_fmt(codec_dai, dai_format);
-	if (ret)
+	if (ret) {
+		dev_err(codec_dai->dev, "Failed to set dai format to %08x\n",
+			dai_format);
 		return ret;
+	}
 
 	/* set cpu DAI configuration */
 	ret = snd_soc_dai_set_fmt(cpu_dai, dai_format);
-	if (ret)
+	if (ret) {
+		dev_err(cpu_dai->dev, "Failed to set dai format to %08x\n",
+			dai_format);
 		return ret;
+	}
 
 	return 0;
 }
@@ -91,11 +105,13 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
 		.stream_name	= "HiFi Playback",
 		.codec_dai_name	= "sgtl5000",
 		.ops		= &mxs_sgtl5000_hifi_ops,
+		.playback_only	= true,
 	}, {
 		.name		= "HiFi Rx",
 		.stream_name	= "HiFi Capture",
 		.codec_dai_name	= "sgtl5000",
 		.ops		= &mxs_sgtl5000_hifi_ops,
+		.capture_only	= true,
 	},
 };
 
@@ -154,8 +170,10 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev)
 	 * should be >= 8MHz and <= 27M.
 	 */
 	ret = mxs_saif_get_mclk(0, 44100 * 256, 44100);
-	if (ret)
+	if (ret) {
+		dev_err(&pdev->dev, "failed to get mclk\n");
 		return ret;
+	}
 
 	card->dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index f4c2417a8730..8987bf987e58 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -333,9 +333,6 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev)
 	spin_lock_init(&nuc900_audio->lock);
 
 	nuc900_audio->res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!nuc900_audio->res)
-		return ret;
-
 	nuc900_audio->mmio = devm_ioremap_resource(&pdev->dev,
 						   nuc900_audio->res);
 	if (IS_ERR(nuc900_audio->mmio))
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 9f5d55e6b17a..daa78a0095fa 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -1,7 +1,7 @@
 config SND_OMAP_SOC
 	tristate "SoC Audio for the Texas Instruments OMAP chips"
-	depends on ARCH_OMAP && DMA_OMAP
-	select SND_SOC_DMAENGINE_PCM
+	depends on (ARCH_OMAP && DMA_OMAP) || (ARCH_ARM && COMPILE_TEST)
+	select SND_DMAENGINE_PCM
 
 config SND_OMAP_SOC_DMIC
 	tristate
@@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810
 
 config SND_OMAP_SOC_RX51
 	tristate "SoC Audio support for Nokia RX-51"
-	depends on SND_OMAP_SOC && MACH_NOKIA_RX51
+	depends on SND_OMAP_SOC && ARCH_ARM && (MACH_NOKIA_RX51 || COMPILE_TEST)
 	select SND_OMAP_SOC_MCBSP
 	select SND_SOC_TLV320AIC3X
 	select SND_SOC_TPA6130A2
@@ -87,7 +87,7 @@ config SND_OMAP_SOC_OMAP_TWL4030
 
 config SND_OMAP_SOC_OMAP_ABE_TWL6040
 	tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec"
-	depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4
+	depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST)
 	select SND_OMAP_SOC_DMIC
 	select SND_OMAP_SOC_MCPDM
 	select SND_SOC_TWL6040
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index 361e4c03646e..83433fdea32a 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -781,7 +781,7 @@ static ssize_t prop##_store(struct device *dev,				\
 	unsigned long val;						\
 	int status;							\
 									\
-	status = strict_strtoul(buf, 0, &val);				\
+	status = kstrtoul(buf, 0, &val);				\
 	if (status)							\
 		return status;						\
 									\
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 70cd5c7b2e14..ebb13906b3a0 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -23,7 +23,6 @@
 #include <linux/clk.h>
 #include <linux/platform_device.h>
 #include <linux/mfd/twl6040.h>
-#include <linux/platform_data/omap-abe-twl6040.h>
 #include <linux/module.h>
 #include <linux/of.h>
 
@@ -166,19 +165,10 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"AFMR", NULL, "Line In"},
 };
 
-static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm,
-					  int connected, char *pin)
-{
-	if (!connected)
-		snd_soc_dapm_disable_pin(dapm, pin);
-}
-
 static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_card *card = codec->card;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
 	struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
 	int hs_trim;
 	int ret = 0;
@@ -203,24 +193,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
 		twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
 	}
 
-	/*
-	 * NULL pdata means we booted with DT. In this case the routing is
-	 * provided and the card is fully routed, no need to mark pins.
-	 */
-	if (!pdata)
-		return ret;
-
-	/* Disable not connected paths if not used */
-	twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone");
-	twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
-	twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
-	twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
-	twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator");
-	twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
-	twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
-	twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
-	twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In");
-
 	return ret;
 }
 
@@ -274,13 +246,18 @@ static struct snd_soc_card omap_abe_card = {
 
 static int omap_abe_probe(struct platform_device *pdev)
 {
-	struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev);
 	struct device_node *node = pdev->dev.of_node;
 	struct snd_soc_card *card = &omap_abe_card;
+	struct device_node *dai_node;
 	struct abe_twl6040 *priv;
 	int num_links = 0;
 	int ret = 0;
 
+	if (!node) {
+		dev_err(&pdev->dev, "of node is missing.\n");
+		return -ENODEV;
+	}
+
 	card->dev = &pdev->dev;
 
 	priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
@@ -289,78 +266,50 @@ static int omap_abe_probe(struct platform_device *pdev)
 
 	priv->dmic_codec_dev = ERR_PTR(-EINVAL);
 
-	if (node) {
-		struct device_node *dai_node;
-
-		if (snd_soc_of_parse_card_name(card, "ti,model")) {
-			dev_err(&pdev->dev, "Card name is not provided\n");
-			return -ENODEV;
-		}
+	if (snd_soc_of_parse_card_name(card, "ti,model")) {
+		dev_err(&pdev->dev, "Card name is not provided\n");
+		return -ENODEV;
+	}
 
-		ret = snd_soc_of_parse_audio_routing(card,
-						"ti,audio-routing");
-		if (ret) {
-			dev_err(&pdev->dev,
-				"Error while parsing DAPM routing\n");
-			return ret;
-		}
+	ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing");
+	if (ret) {
+		dev_err(&pdev->dev, "Error while parsing DAPM routing\n");
+		return ret;
+	}
 
-		dai_node = of_parse_phandle(node, "ti,mcpdm", 0);
-		if (!dai_node) {
-			dev_err(&pdev->dev, "McPDM node is not provided\n");
-			return -EINVAL;
-		}
-		abe_twl6040_dai_links[0].cpu_dai_name  = NULL;
-		abe_twl6040_dai_links[0].cpu_of_node = dai_node;
+	dai_node = of_parse_phandle(node, "ti,mcpdm", 0);
+	if (!dai_node) {
+		dev_err(&pdev->dev, "McPDM node is not provided\n");
+		return -EINVAL;
+	}
+	abe_twl6040_dai_links[0].cpu_dai_name  = NULL;
+	abe_twl6040_dai_links[0].cpu_of_node = dai_node;
 
-		dai_node = of_parse_phandle(node, "ti,dmic", 0);
-		if (dai_node) {
-			num_links = 2;
-			abe_twl6040_dai_links[1].cpu_dai_name  = NULL;
-			abe_twl6040_dai_links[1].cpu_of_node = dai_node;
+	dai_node = of_parse_phandle(node, "ti,dmic", 0);
+	if (dai_node) {
+		num_links = 2;
+		abe_twl6040_dai_links[1].cpu_dai_name  = NULL;
+		abe_twl6040_dai_links[1].cpu_of_node = dai_node;
 
-			priv->dmic_codec_dev = platform_device_register_simple(
+		priv->dmic_codec_dev = platform_device_register_simple(
 						"dmic-codec", -1, NULL, 0);
-			if (IS_ERR(priv->dmic_codec_dev)) {
-				dev_err(&pdev->dev,
-					"Can't instantiate dmic-codec\n");
-				return PTR_ERR(priv->dmic_codec_dev);
-			}
-		} else {
-			num_links = 1;
-		}
-
-		priv->jack_detection = of_property_read_bool(node,
-							   "ti,jack-detection");
-		of_property_read_u32(node, "ti,mclk-freq",
-				     &priv->mclk_freq);
-		if (!priv->mclk_freq) {
-			dev_err(&pdev->dev, "MCLK frequency not provided\n");
-			ret = -EINVAL;
-			goto err_unregister;
+		if (IS_ERR(priv->dmic_codec_dev)) {
+			dev_err(&pdev->dev, "Can't instantiate dmic-codec\n");
+			return PTR_ERR(priv->dmic_codec_dev);
 		}
-
-		omap_abe_card.fully_routed = 1;
-	} else if (pdata) {
-		if (pdata->card_name) {
-			card->name = pdata->card_name;
-		} else {
-			dev_err(&pdev->dev, "Card name is not provided\n");
-			return -ENODEV;
-		}
-
-		if (pdata->has_dmic)
-			num_links = 2;
-		else
-			num_links = 1;
-
-		priv->jack_detection = pdata->jack_detection;
-		priv->mclk_freq = pdata->mclk_freq;
 	} else {
-		dev_err(&pdev->dev, "Missing pdata\n");
-		return -ENODEV;
+		num_links = 1;
+	}
+
+	priv->jack_detection = of_property_read_bool(node, "ti,jack-detection");
+	of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq);
+	if (!priv->mclk_freq) {
+		dev_err(&pdev->dev, "MCLK frequency not provided\n");
+		ret = -EINVAL;
+		goto err_unregister;
 	}
 
+	card->fully_routed = 1;
 
 	if (!priv->mclk_freq) {
 		dev_err(&pdev->dev, "MCLK frequency missing\n");
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 4db1f8e6e172..12e566be3793 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -480,15 +480,12 @@ static int asoc_dmic_probe(struct platform_device *pdev)
 	dmic->dma_data.filter_data = "up_link";
 
 	res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
-	if (!res) {
-		dev_err(dmic->dev, "invalid memory resource\n");
-		ret = -ENODEV;
+	dmic->io_base = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(dmic->io_base)) {
+		ret = PTR_ERR(dmic->io_base);
 		goto err_put_clk;
 	}
 
-	dmic->io_base = devm_ioremap_resource(&pdev->dev, res);
-	if (IS_ERR(dmic->io_base))
-		return PTR_ERR(dmic->io_base);
 
 	ret = snd_soc_register_component(&pdev->dev, &omap_dmic_component,
 					 &omap_dmic_dai, 1);
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 7483efb6dc67..6c19bba23570 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -433,6 +433,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		/* Sample rate generator drives the FS */
 		regs->srgr2	|= FSGM;
 		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		/* McBSP slave. FS clock as output */
+		regs->srgr2	|= FSGM;
+		regs->pcr0	|= FSXM;
+		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		/* McBSP slave */
 		break;
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index a49dc52f8abc..90d2a7cd2563 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -480,9 +480,6 @@ static int asoc_mcpdm_probe(struct platform_device *pdev)
 	mcpdm->dma_data[1].filter_data = "up_link";
 
 	res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
-	if (res == NULL)
-		return -ENOMEM;
-
 	mcpdm->io_base = devm_ioremap_resource(&pdev->dev, res);
 	if (IS_ERR(mcpdm->io_base))
 		return PTR_ERR(mcpdm->io_base);
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index b35809467547..4db74a083db1 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -11,7 +11,7 @@ config SND_PXA2XX_SOC
 config SND_MMP_SOC
 	bool "Soc Audio for Marvell MMP chips"
 	depends on ARCH_MMP
-	select SND_SOC_DMAENGINE_PCM
+	select SND_DMAENGINE_PCM
 	select SND_ARM
 	help
 	  Say Y if you want to add support for codecs attached to
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
index 4ad76099dd43..5b7d969f89a9 100644
--- a/sound/soc/pxa/brownstone.c
+++ b/sound/soc/pxa/brownstone.c
@@ -129,6 +129,7 @@ static struct snd_soc_dai_link brownstone_wm8994_dai[] = {
 /* audio machine driver */
 static struct snd_soc_card brownstone = {
 	.name         = "brownstone",
+	.owner        = THIS_MODULE,
 	.dai_link     = brownstone_wm8994_dai,
 	.num_links    = ARRAY_SIZE(brownstone_wm8994_dai),
 
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 97b711e12821..bbea7780eac6 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -56,8 +56,6 @@
 #include "pxa2xx-ac97.h"
 #include "../codecs/wm9713.h"
 
-#define ARRAY_AND_SIZE(x)	(x), ARRAY_SIZE(x)
-
 #define AC97_GPIO_PULL		0x58
 
 /* Use GPIO8 for rear speaker amplifier */
@@ -133,10 +131,11 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd)
 	unsigned short reg;
 
 	/* Add mioa701 specific widgets */
-	snd_soc_dapm_new_controls(dapm, ARRAY_AND_SIZE(mioa701_dapm_widgets));
+	snd_soc_dapm_new_controls(dapm, mioa701_dapm_widgets,
+				  ARRAY_SIZE(mioa701_dapm_widgets));
 
 	/* Set up mioa701 specific audio path audio_mapnects */
-	snd_soc_dapm_add_routes(dapm, ARRAY_AND_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
 	/* Prepare GPIO8 for rear speaker amplifier */
 	reg = codec->driver->read(codec, AC97_GPIO_CFG);
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
index 5d57e071cdf5..8235e231d89c 100644
--- a/sound/soc/pxa/mmp-pcm.c
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -17,6 +17,7 @@
 #include <linux/dmaengine.h>
 #include <linux/platform_data/dma-mmp_tdma.h>
 #include <linux/platform_data/mmp_audio.h>
+
 #include <sound/pxa2xx-lib.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -67,7 +68,7 @@ static int mmp_pcm_hw_params(struct snd_pcm_substream *substream,
 {
 	struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct pxa2xx_pcm_dma_params *dma_params;
+	struct snd_dmaengine_dai_dma_data *dma_params;
 	struct dma_slave_config slave_config;
 	int ret;
 
@@ -80,10 +81,10 @@ static int mmp_pcm_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		slave_config.dst_addr     = dma_params->dev_addr;
+		slave_config.dst_addr     = dma_params->addr;
 		slave_config.dst_maxburst = 4;
 	} else {
-		slave_config.src_addr	  = dma_params->dev_addr;
+		slave_config.src_addr	  = dma_params->addr;
 		slave_config.src_maxburst = 4;
 	}
 
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
index 62142ce367c7..41752a5fe3b0 100644
--- a/sound/soc/pxa/mmp-sspa.c
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -27,12 +27,15 @@
 #include <linux/slab.h>
 #include <linux/pxa2xx_ssp.h>
 #include <linux/io.h>
+#include <linux/dmaengine.h>
+
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/initval.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
 #include "mmp-sspa.h"
 
 /*
@@ -40,7 +43,7 @@
  */
 struct sspa_priv {
 	struct ssp_device *sspa;
-	struct pxa2xx_pcm_dma_params *dma_params;
+	struct snd_dmaengine_dai_dma_data *dma_params;
 	struct clk *audio_clk;
 	struct clk *sysclk;
 	int dai_fmt;
@@ -266,7 +269,7 @@ static int mmp_sspa_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
 	struct ssp_device *sspa = sspa_priv->sspa;
-	struct pxa2xx_pcm_dma_params *dma_params;
+	struct snd_dmaengine_dai_dma_data *dma_params;
 	u32 sspa_ctrl;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -309,7 +312,7 @@ static int mmp_sspa_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	dma_params = &sspa_priv->dma_params[substream->stream];
-	dma_params->dev_addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+	dma_params->addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
 				(sspa->phys_base + SSPA_TXD) :
 				(sspa->phys_base + SSPA_RXD);
 	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_params);
@@ -425,14 +428,12 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	priv->dma_params = devm_kzalloc(&pdev->dev,
-			2 * sizeof(struct pxa2xx_pcm_dma_params), GFP_KERNEL);
+			2 * sizeof(struct snd_dmaengine_dai_dma_data),
+			GFP_KERNEL);
 	if (priv->dma_params == NULL)
 		return -ENOMEM;
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (res == NULL)
-		return -ENOMEM;
-
 	priv->sspa->mmio_base = devm_ioremap_resource(&pdev->dev, res);
 	if (IS_ERR(priv->sspa->mmio_base))
 		return PTR_ERR(priv->sspa->mmio_base);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 6f4dd7543e82..a3119a00d8fa 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -21,6 +21,8 @@
 #include <linux/clk.h>
 #include <linux/io.h>
 #include <linux/pxa2xx_ssp.h>
+#include <linux/of.h>
+#include <linux/dmaengine.h>
 
 #include <asm/irq.h>
 
@@ -30,9 +32,9 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
 
 #include <mach/hardware.h>
-#include <mach/dma.h>
 
 #include "../../arm/pxa2xx-pcm.h"
 #include "pxa-ssp.h"
@@ -79,27 +81,13 @@ static void pxa_ssp_disable(struct ssp_device *ssp)
 	__raw_writel(sscr0, ssp->mmio_base + SSCR0);
 }
 
-struct pxa2xx_pcm_dma_data {
-	struct pxa2xx_pcm_dma_params params;
-	char name[20];
-};
-
 static void pxa_ssp_set_dma_params(struct ssp_device *ssp, int width4,
-			int out, struct pxa2xx_pcm_dma_params *dma_data)
+			int out, struct snd_dmaengine_dai_dma_data *dma)
 {
-	struct pxa2xx_pcm_dma_data *dma;
-
-	dma = container_of(dma_data, struct pxa2xx_pcm_dma_data, params);
-
-	snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id,
-			width4 ? "32-bit" : "16-bit", out ? "out" : "in");
-
-	dma->params.name = dma->name;
-	dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx);
-	dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) :
-				  (DCMD_INCTRGADDR | DCMD_FLOWSRC)) |
-			(width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16;
-	dma->params.dev_addr = ssp->phys_base + SSDR;
+	dma->addr_width = width4 ? DMA_SLAVE_BUSWIDTH_4_BYTES :
+				   DMA_SLAVE_BUSWIDTH_2_BYTES;
+	dma->maxburst = 16;
+	dma->addr = ssp->phys_base + SSDR;
 }
 
 static int pxa_ssp_startup(struct snd_pcm_substream *substream,
@@ -107,7 +95,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 {
 	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
-	struct pxa2xx_pcm_dma_data *dma;
+	struct snd_dmaengine_dai_dma_data *dma;
 	int ret = 0;
 
 	if (!cpu_dai->active) {
@@ -115,10 +103,14 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 		pxa_ssp_disable(ssp);
 	}
 
-	dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL);
+	dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL);
 	if (!dma)
 		return -ENOMEM;
-	snd_soc_dai_set_dma_data(cpu_dai, substream, &dma->params);
+
+	dma->filter_data = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+				&ssp->drcmr_tx : &ssp->drcmr_rx;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma);
 
 	return ret;
 }
@@ -559,7 +551,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	u32 sspsp;
 	int width = snd_pcm_format_physical_width(params_format(params));
 	int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf;
-	struct pxa2xx_pcm_dma_params *dma_data;
+	struct snd_dmaengine_dai_dma_data *dma_data;
 
 	dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream);
 
@@ -719,6 +711,7 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
 
 static int pxa_ssp_probe(struct snd_soc_dai *dai)
 {
+	struct device *dev = dai->dev;
 	struct ssp_priv *priv;
 	int ret;
 
@@ -726,10 +719,26 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai)
 	if (!priv)
 		return -ENOMEM;
 
-	priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio");
-	if (priv->ssp == NULL) {
-		ret = -ENODEV;
-		goto err_priv;
+	if (dev->of_node) {
+		struct device_node *ssp_handle;
+
+		ssp_handle = of_parse_phandle(dev->of_node, "port", 0);
+		if (!ssp_handle) {
+			dev_err(dev, "unable to get 'port' phandle\n");
+			return -ENODEV;
+		}
+
+		priv->ssp = pxa_ssp_request_of(ssp_handle, "SoC audio");
+		if (priv->ssp == NULL) {
+			ret = -ENODEV;
+			goto err_priv;
+		}
+	} else {
+		priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio");
+		if (priv->ssp == NULL) {
+			ret = -ENODEV;
+			goto err_priv;
+		}
 	}
 
 	priv->dai_fmt = (unsigned int) -1;
@@ -798,6 +807,12 @@ static const struct snd_soc_component_driver pxa_ssp_component = {
 	.name		= "pxa-ssp",
 };
 
+#ifdef CONFIG_OF
+static const struct of_device_id pxa_ssp_of_ids[] = {
+	{ .compatible = "mrvl,pxa-ssp-dai" },
+};
+#endif
+
 static int asoc_ssp_probe(struct platform_device *pdev)
 {
 	return snd_soc_register_component(&pdev->dev, &pxa_ssp_component,
@@ -812,8 +827,9 @@ static int asoc_ssp_remove(struct platform_device *pdev)
 
 static struct platform_driver asoc_ssp_driver = {
 	.driver = {
-			.name = "pxa-ssp-dai",
-			.owner = THIS_MODULE,
+		.name = "pxa-ssp-dai",
+		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(pxa_ssp_of_ids),
 	},
 
 	.probe = asoc_ssp_probe,
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 1475515712e6..f1059d999de6 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -14,15 +14,16 @@
 #include <linux/io.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
+#include <linux/dmaengine.h>
 
 #include <sound/core.h>
 #include <sound/ac97_codec.h>
 #include <sound/soc.h>
 #include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
 
 #include <mach/hardware.h>
 #include <mach/regs-ac97.h>
-#include <mach/dma.h>
 #include <mach/audio.h>
 
 #include "pxa2xx-ac97.h"
@@ -48,44 +49,44 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
 	.reset	= pxa2xx_ac97_cold_reset,
 };
 
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = {
-	.name			= "AC97 PCM Stereo out",
-	.dev_addr		= __PREG(PCDR),
-	.drcmr			= &DRCMR(12),
-	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-				  DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
+	.addr		= __PREG(PCDR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.maxburst	= 32,
+	.filter_data	= &pxa2xx_ac97_pcm_stereo_in_req,
 };
 
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = {
-	.name			= "AC97 PCM Stereo in",
-	.dev_addr		= __PREG(PCDR),
-	.drcmr			= &DRCMR(11),
-	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-				  DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
+	.addr		= __PREG(PCDR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.maxburst	= 32,
+	.filter_data	= &pxa2xx_ac97_pcm_stereo_out_req,
 };
 
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = {
-	.name			= "AC97 Aux PCM (Slot 5) Mono out",
-	.dev_addr		= __PREG(MODR),
-	.drcmr			= &DRCMR(10),
-	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-				  DCMD_BURST16 | DCMD_WIDTH2,
+static unsigned long pxa2xx_ac97_pcm_aux_mono_out_req = 10;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = {
+	.addr		= __PREG(MODR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_2_BYTES,
+	.maxburst	= 16,
+	.filter_data	= &pxa2xx_ac97_pcm_aux_mono_out_req,
 };
 
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = {
-	.name			= "AC97 Aux PCM (Slot 5) Mono in",
-	.dev_addr		= __PREG(MODR),
-	.drcmr			= &DRCMR(9),
-	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-				  DCMD_BURST16 | DCMD_WIDTH2,
+static unsigned long pxa2xx_ac97_pcm_aux_mono_in_req = 9;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = {
+	.addr		= __PREG(MODR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_2_BYTES,
+	.maxburst	= 16,
+	.filter_data	= &pxa2xx_ac97_pcm_aux_mono_in_req,
 };
 
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
-	.name			= "AC97 Mic PCM (Slot 6) Mono in",
-	.dev_addr		= __PREG(MCDR),
-	.drcmr			= &DRCMR(8),
-	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-				  DCMD_BURST16 | DCMD_WIDTH2,
+static unsigned long pxa2xx_ac97_pcm_aux_mic_mono_req = 8;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = {
+	.addr		= __PREG(MCDR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_2_BYTES,
+	.maxburst	= 16,
+	.filter_data	= &pxa2xx_ac97_pcm_aux_mic_mono_req,
 };
 
 #ifdef CONFIG_PM
@@ -119,7 +120,7 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *params,
 				 struct snd_soc_dai *cpu_dai)
 {
-	struct pxa2xx_pcm_dma_params *dma_data;
+	struct snd_dmaengine_dai_dma_data *dma_data;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		dma_data = &pxa2xx_ac97_pcm_stereo_out;
@@ -135,7 +136,7 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
 				     struct snd_pcm_hw_params *params,
 				     struct snd_soc_dai *cpu_dai)
 {
-	struct pxa2xx_pcm_dma_params *dma_data;
+	struct snd_dmaengine_dai_dma_data *dma_data;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index f7ca71664112..d5340a088858 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -23,9 +23,9 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 #include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
 
 #include <mach/hardware.h>
-#include <mach/dma.h>
 #include <mach/audio.h>
 
 #include "pxa2xx-i2s.h"
@@ -82,20 +82,20 @@ static struct pxa_i2s_port pxa_i2s;
 static struct clk *clk_i2s;
 static int clk_ena = 0;
 
-static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = {
-	.name			= "I2S PCM Stereo out",
-	.dev_addr		= __PREG(SADR),
-	.drcmr			= &DRCMR(3),
-	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-				  DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3;
+static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = {
+	.addr		= __PREG(SADR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.maxburst	= 32,
+	.filter_data	= &pxa2xx_i2s_pcm_stereo_out_req,
 };
 
-static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = {
-	.name			= "I2S PCM Stereo in",
-	.dev_addr		= __PREG(SADR),
-	.drcmr			= &DRCMR(2),
-	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-				  DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2;
+static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = {
+	.addr		= __PREG(SADR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.maxburst	= 32,
+	.filter_data	= &pxa2xx_i2s_pcm_stereo_in_req,
 };
 
 static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
@@ -163,7 +163,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
 {
-	struct pxa2xx_pcm_dma_params *dma_data;
+	struct snd_dmaengine_dai_dma_data *dma_data;
 
 	BUG_ON(IS_ERR(clk_i2s));
 	clk_prepare_enable(clk_i2s);
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index ecff116cb7b0..806da27b8b67 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -12,10 +12,13 @@
 
 #include <linux/dma-mapping.h>
 #include <linux/module.h>
+#include <linux/dmaengine.h>
+#include <linux/of.h>
 
 #include <sound/core.h>
 #include <sound/soc.h>
 #include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
 
 #include "../../arm/pxa2xx-pcm.h"
 
@@ -25,7 +28,7 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct pxa2xx_runtime_data *prtd = runtime->private_data;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct pxa2xx_pcm_dma_params *dma;
+	struct snd_dmaengine_dai_dma_data *dma;
 	int ret;
 
 	dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
@@ -39,7 +42,7 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 	 * with different params */
 	if (prtd->params == NULL) {
 		prtd->params = dma;
-		ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW,
+		ret = pxa_request_dma("name", DMA_PRIO_LOW,
 			      pxa2xx_pcm_dma_irq, substream);
 		if (ret < 0)
 			return ret;
@@ -47,7 +50,7 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 	} else if (prtd->params != dma) {
 		pxa_free_dma(prtd->dma_ch);
 		prtd->params = dma;
-		ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW,
+		ret = pxa_request_dma("name", DMA_PRIO_LOW,
 			      pxa2xx_pcm_dma_irq, substream);
 		if (ret < 0)
 			return ret;
@@ -131,10 +134,18 @@ static int pxa2xx_soc_platform_remove(struct platform_device *pdev)
 	return 0;
 }
 
+#ifdef CONFIG_OF
+static const struct of_device_id snd_soc_pxa_audio_match[] = {
+	{ .compatible   = "mrvl,pxa-pcm-audio" },
+	{ }
+};
+#endif
+
 static struct platform_driver pxa_pcm_driver = {
 	.driver = {
-			.name = "pxa-pcm-audio",
-			.owner = THIS_MODULE,
+		.name = "pxa-pcm-audio",
+		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(snd_soc_pxa_audio_match),
 	},
 
 	.probe = pxa2xx_soc_platform_probe,
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
index f4ea4f6663a2..13c9ee0cb83b 100644
--- a/sound/soc/pxa/ttc-dkb.c
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -122,6 +122,7 @@ static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = {
 /* ttc/td audio machine driver */
 static struct snd_soc_card ttc_dkb_card = {
 	.name = "ttc-dkb-hifi",
+	.owner = THIS_MODULE,
 	.dai_link = ttc_pm860x_hifi_dai,
 	.num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai),
 
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
index 58cfb1eb7dd3..945e8abdc10f 100644
--- a/sound/soc/s6000/s6105-ipcam.c
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -192,7 +192,7 @@ static struct snd_soc_card snd_soc_card_s6105 = {
 	.num_links = 1,
 };
 
-static struct s6000_snd_platform_data __initdata s6105_snd_data = {
+static struct s6000_snd_platform_data s6105_snd_data __initdata = {
 	.wide		= 0,
 	.channel_in	= 0,
 	.channel_out	= 1,
diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c
index 2dd623fa3882..2acf987844e8 100644
--- a/sound/soc/samsung/ac97.c
+++ b/sound/soc/samsung/ac97.c
@@ -404,18 +404,13 @@ static int s3c_ac97_probe(struct platform_device *pdev)
 		return -ENXIO;
 	}
 
-	mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!mem_res) {
-		dev_err(&pdev->dev, "Unable to get register resource\n");
-		return -ENXIO;
-	}
-
 	irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0);
 	if (!irq_res) {
 		dev_err(&pdev->dev, "AC97 IRQ not provided!\n");
 		return -ENXIO;
 	}
 
+	mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	s3c_ac97.regs = devm_ioremap_resource(&pdev->dev, mem_res);
 	if (IS_ERR(s3c_ac97.regs))
 		return PTR_ERR(s3c_ac97.regs);
@@ -462,7 +457,7 @@ static int s3c_ac97_probe(struct platform_device *pdev)
 	if (ret)
 		goto err5;
 
-	ret = asoc_dma_platform_register(&pdev->dev);
+	ret = samsung_asoc_dma_platform_register(&pdev->dev);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to get register DMA: %d\n", ret);
 		goto err6;
@@ -485,7 +480,7 @@ static int s3c_ac97_remove(struct platform_device *pdev)
 {
 	struct resource *irq_res;
 
-	asoc_dma_platform_unregister(&pdev->dev);
+	samsung_asoc_dma_platform_unregister(&pdev->dev);
 	snd_soc_unregister_component(&pdev->dev);
 
 	irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0);
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index 21b79262010e..9338d11e9216 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -90,6 +90,13 @@ static void dma_enqueue(struct snd_pcm_substream *substream)
 	dma_info.period = prtd->dma_period;
 	dma_info.len = prtd->dma_period*limit;
 
+	if (dma_info.cap == DMA_CYCLIC) {
+		dma_info.buf = pos;
+		prtd->params->ops->prepare(prtd->params->ch, &dma_info);
+		prtd->dma_loaded += limit;
+		return;
+	}
+
 	while (prtd->dma_loaded < limit) {
 		pr_debug("dma_loaded: %d\n", prtd->dma_loaded);
 
@@ -176,6 +183,10 @@ static int dma_hw_params(struct snd_pcm_substream *substream,
 		prtd->params->ch = prtd->params->ops->request(
 				prtd->params->channel, &req, rtd->cpu_dai->dev,
 				prtd->params->ch_name);
+		if (!prtd->params->ch) {
+			pr_err("Failed to allocate DMA channel\n");
+			return -ENXIO;
+		}
 		prtd->params->ops->config(prtd->params->ch, &config);
 	}
 
@@ -433,17 +444,17 @@ static struct snd_soc_platform_driver samsung_asoc_platform = {
 	.pcm_free	= dma_free_dma_buffers,
 };
 
-int asoc_dma_platform_register(struct device *dev)
+int samsung_asoc_dma_platform_register(struct device *dev)
 {
 	return snd_soc_register_platform(dev, &samsung_asoc_platform);
 }
-EXPORT_SYMBOL_GPL(asoc_dma_platform_register);
+EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register);
 
-void asoc_dma_platform_unregister(struct device *dev)
+void samsung_asoc_dma_platform_unregister(struct device *dev)
 {
 	snd_soc_unregister_platform(dev);
 }
-EXPORT_SYMBOL_GPL(asoc_dma_platform_unregister);
+EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_unregister);
 
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("Samsung ASoC DMA Driver");
diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h
index 189a7a6d5020..0e86315a3eaf 100644
--- a/sound/soc/samsung/dma.h
+++ b/sound/soc/samsung/dma.h
@@ -22,7 +22,7 @@ struct s3c_dma_params {
 	char *ch_name;
 };
 
-int asoc_dma_platform_register(struct device *dev);
-void asoc_dma_platform_unregister(struct device *dev);
+int samsung_asoc_dma_platform_register(struct device *dev);
+void samsung_asoc_dma_platform_unregister(struct device *dev);
 
 #endif
diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h
index c0e6d9a19efc..821a50231002 100644
--- a/sound/soc/samsung/i2s-regs.h
+++ b/sound/soc/samsung/i2s-regs.h
@@ -31,6 +31,10 @@
 #define I2SLVL1ADDR	0x34
 #define I2SLVL2ADDR	0x38
 #define I2SLVL3ADDR	0x3c
+#define I2SSTR1		0x40
+#define I2SVER		0x44
+#define I2SFIC2		0x48
+#define I2STDM		0x4c
 
 #define CON_RSTCLR		(1 << 31)
 #define CON_FRXOFSTATUS		(1 << 26)
@@ -95,24 +99,39 @@
 #define MOD_RXONLY		(1 << 8)
 #define MOD_TXRX		(2 << 8)
 #define MOD_MASK		(3 << 8)
-#define MOD_LR_LLOW		(0 << 7)
-#define MOD_LR_RLOW		(1 << 7)
-#define MOD_SDF_IIS		(0 << 5)
-#define MOD_SDF_MSB		(1 << 5)
-#define MOD_SDF_LSB		(2 << 5)
-#define MOD_SDF_MASK		(3 << 5)
-#define MOD_RCLK_256FS		(0 << 3)
-#define MOD_RCLK_512FS		(1 << 3)
-#define MOD_RCLK_384FS		(2 << 3)
-#define MOD_RCLK_768FS		(3 << 3)
-#define MOD_RCLK_MASK		(3 << 3)
-#define MOD_BCLK_32FS		(0 << 1)
-#define MOD_BCLK_48FS		(1 << 1)
-#define MOD_BCLK_16FS		(2 << 1)
-#define MOD_BCLK_24FS		(3 << 1)
-#define MOD_BCLK_MASK		(3 << 1)
+#define MOD_LRP_SHIFT		7
+#define MOD_LR_LLOW		0
+#define MOD_LR_RLOW		1
+#define MOD_SDF_SHIFT		5
+#define MOD_SDF_IIS		0
+#define MOD_SDF_MSB		1
+#define MOD_SDF_LSB		2
+#define MOD_SDF_MASK		3
+#define MOD_RCLK_SHIFT		3
+#define MOD_RCLK_256FS		0
+#define MOD_RCLK_512FS		1
+#define MOD_RCLK_384FS		2
+#define MOD_RCLK_768FS		3
+#define MOD_RCLK_MASK		3
+#define MOD_BCLK_SHIFT		1
+#define MOD_BCLK_32FS		0
+#define MOD_BCLK_48FS		1
+#define MOD_BCLK_16FS		2
+#define MOD_BCLK_24FS		3
+#define MOD_BCLK_MASK		3
 #define MOD_8BIT		(1 << 0)
 
+#define EXYNOS5420_MOD_LRP_SHIFT	15
+#define EXYNOS5420_MOD_SDF_SHIFT	6
+#define EXYNOS5420_MOD_RCLK_SHIFT	4
+#define EXYNOS5420_MOD_BCLK_SHIFT	0
+#define EXYNOS5420_MOD_BCLK_64FS	4
+#define EXYNOS5420_MOD_BCLK_96FS	5
+#define EXYNOS5420_MOD_BCLK_128FS	6
+#define EXYNOS5420_MOD_BCLK_192FS	7
+#define EXYNOS5420_MOD_BCLK_256FS	8
+#define EXYNOS5420_MOD_BCLK_MASK	0xf
+
 #define MOD_CDCLKCON		(1 << 12)
 
 #define PSR_PSREN		(1 << 15)
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 959c702235c8..b302f3b7a587 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -40,6 +40,7 @@ enum samsung_dai_type {
 
 struct samsung_i2s_dai_data {
 	int dai_type;
+	u32 quirks;
 };
 
 struct i2s_dai {
@@ -198,7 +199,13 @@ static inline bool is_manager(struct i2s_dai *i2s)
 /* Read RCLK of I2S (in multiples of LRCLK) */
 static inline unsigned get_rfs(struct i2s_dai *i2s)
 {
-	u32 rfs = (readl(i2s->addr + I2SMOD) >> 3) & 0x3;
+	u32 rfs;
+
+	if (i2s->quirks & QUIRK_SUPPORTS_TDM)
+		rfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_RCLK_SHIFT;
+	else
+		rfs = (readl(i2s->addr + I2SMOD) >> MOD_RCLK_SHIFT);
+	rfs &= MOD_RCLK_MASK;
 
 	switch (rfs) {
 	case 3:	return 768;
@@ -212,21 +219,26 @@ static inline unsigned get_rfs(struct i2s_dai *i2s)
 static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs)
 {
 	u32 mod = readl(i2s->addr + I2SMOD);
+	int rfs_shift;
 
-	mod &= ~MOD_RCLK_MASK;
+	if (i2s->quirks & QUIRK_SUPPORTS_TDM)
+		rfs_shift = EXYNOS5420_MOD_RCLK_SHIFT;
+	else
+		rfs_shift = MOD_RCLK_SHIFT;
+	mod &= ~(MOD_RCLK_MASK << rfs_shift);
 
 	switch (rfs) {
 	case 768:
-		mod |= MOD_RCLK_768FS;
+		mod |= (MOD_RCLK_768FS << rfs_shift);
 		break;
 	case 512:
-		mod |= MOD_RCLK_512FS;
+		mod |= (MOD_RCLK_512FS << rfs_shift);
 		break;
 	case 384:
-		mod |= MOD_RCLK_384FS;
+		mod |= (MOD_RCLK_384FS << rfs_shift);
 		break;
 	default:
-		mod |= MOD_RCLK_256FS;
+		mod |= (MOD_RCLK_256FS << rfs_shift);
 		break;
 	}
 
@@ -236,9 +248,22 @@ static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs)
 /* Read Bit-Clock of I2S (in multiples of LRCLK) */
 static inline unsigned get_bfs(struct i2s_dai *i2s)
 {
-	u32 bfs = (readl(i2s->addr + I2SMOD) >> 1) & 0x3;
+	u32 bfs;
+
+	if (i2s->quirks & QUIRK_SUPPORTS_TDM) {
+		bfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_BCLK_SHIFT;
+		bfs &= EXYNOS5420_MOD_BCLK_MASK;
+	} else {
+		bfs =  readl(i2s->addr + I2SMOD) >> MOD_BCLK_SHIFT;
+		bfs &= MOD_BCLK_MASK;
+	}
 
 	switch (bfs) {
+	case 8: return 256;
+	case 7: return 192;
+	case 6: return 128;
+	case 5: return 96;
+	case 4: return 64;
 	case 3: return 24;
 	case 2: return 16;
 	case 1:	return 48;
@@ -250,21 +275,50 @@ static inline unsigned get_bfs(struct i2s_dai *i2s)
 static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs)
 {
 	u32 mod = readl(i2s->addr + I2SMOD);
+	int bfs_shift;
+	int tdm = i2s->quirks & QUIRK_SUPPORTS_TDM;
 
-	mod &= ~MOD_BCLK_MASK;
+	if (i2s->quirks & QUIRK_SUPPORTS_TDM) {
+		bfs_shift = EXYNOS5420_MOD_BCLK_SHIFT;
+		mod &= ~(EXYNOS5420_MOD_BCLK_MASK << bfs_shift);
+	} else {
+		bfs_shift = MOD_BCLK_SHIFT;
+		mod &= ~(MOD_BCLK_MASK << bfs_shift);
+	}
+
+	/* Non-TDM I2S controllers do not support BCLK > 48 * FS */
+	if (!tdm && bfs > 48) {
+		dev_err(&i2s->pdev->dev, "Unsupported BCLK divider\n");
+		return;
+	}
 
 	switch (bfs) {
 	case 48:
-		mod |= MOD_BCLK_48FS;
+		mod |= (MOD_BCLK_48FS << bfs_shift);
 		break;
 	case 32:
-		mod |= MOD_BCLK_32FS;
+		mod |= (MOD_BCLK_32FS << bfs_shift);
 		break;
 	case 24:
-		mod |= MOD_BCLK_24FS;
+		mod |= (MOD_BCLK_24FS << bfs_shift);
 		break;
 	case 16:
-		mod |= MOD_BCLK_16FS;
+		mod |= (MOD_BCLK_16FS << bfs_shift);
+		break;
+	case 64:
+		mod |= (EXYNOS5420_MOD_BCLK_64FS << bfs_shift);
+		break;
+	case 96:
+		mod |= (EXYNOS5420_MOD_BCLK_96FS << bfs_shift);
+		break;
+	case 128:
+		mod |= (EXYNOS5420_MOD_BCLK_128FS << bfs_shift);
+		break;
+	case 192:
+		mod |= (EXYNOS5420_MOD_BCLK_192FS << bfs_shift);
+		break;
+	case 256:
+		mod |= (EXYNOS5420_MOD_BCLK_256FS << bfs_shift);
 		break;
 	default:
 		dev_err(&i2s->pdev->dev, "Wrong BCLK Divider!\n");
@@ -491,20 +545,32 @@ static int i2s_set_fmt(struct snd_soc_dai *dai,
 {
 	struct i2s_dai *i2s = to_info(dai);
 	u32 mod = readl(i2s->addr + I2SMOD);
+	int lrp_shift, sdf_shift, sdf_mask, lrp_rlow;
 	u32 tmp = 0;
 
+	if (i2s->quirks & QUIRK_SUPPORTS_TDM) {
+		lrp_shift = EXYNOS5420_MOD_LRP_SHIFT;
+		sdf_shift = EXYNOS5420_MOD_SDF_SHIFT;
+	} else {
+		lrp_shift = MOD_LRP_SHIFT;
+		sdf_shift = MOD_SDF_SHIFT;
+	}
+
+	sdf_mask = MOD_SDF_MASK << sdf_shift;
+	lrp_rlow = MOD_LR_RLOW << lrp_shift;
+
 	/* Format is priority */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_RIGHT_J:
-		tmp |= MOD_LR_RLOW;
-		tmp |= MOD_SDF_MSB;
+		tmp |= lrp_rlow;
+		tmp |= (MOD_SDF_MSB << sdf_shift);
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
-		tmp |= MOD_LR_RLOW;
-		tmp |= MOD_SDF_LSB;
+		tmp |= lrp_rlow;
+		tmp |= (MOD_SDF_LSB << sdf_shift);
 		break;
 	case SND_SOC_DAIFMT_I2S:
-		tmp |= MOD_SDF_IIS;
+		tmp |= (MOD_SDF_IIS << sdf_shift);
 		break;
 	default:
 		dev_err(&i2s->pdev->dev, "Format not supported\n");
@@ -519,10 +585,10 @@ static int i2s_set_fmt(struct snd_soc_dai *dai,
 	case SND_SOC_DAIFMT_NB_NF:
 		break;
 	case SND_SOC_DAIFMT_NB_IF:
-		if (tmp & MOD_LR_RLOW)
-			tmp &= ~MOD_LR_RLOW;
+		if (tmp & lrp_rlow)
+			tmp &= ~lrp_rlow;
 		else
-			tmp |= MOD_LR_RLOW;
+			tmp |= lrp_rlow;
 		break;
 	default:
 		dev_err(&i2s->pdev->dev, "Polarity not supported\n");
@@ -544,15 +610,18 @@ static int i2s_set_fmt(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
+	/*
+	 * Don't change the I2S mode if any controller is active on this
+	 * channel.
+	 */
 	if (any_active(i2s) &&
-			((mod & (MOD_SDF_MASK | MOD_LR_RLOW
-				| MOD_SLAVE)) != tmp)) {
+		((mod & (sdf_mask | lrp_rlow | MOD_SLAVE)) != tmp)) {
 		dev_err(&i2s->pdev->dev,
 				"%s:%d Other DAI busy\n", __func__, __LINE__);
 		return -EAGAIN;
 	}
 
-	mod &= ~(MOD_SDF_MASK | MOD_LR_RLOW | MOD_SLAVE);
+	mod &= ~(sdf_mask | lrp_rlow | MOD_SLAVE);
 	mod |= tmp;
 	writel(mod, i2s->addr + I2SMOD);
 
@@ -1007,6 +1076,8 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec)
 		if (IS_ERR(i2s->pdev))
 			return NULL;
 
+		i2s->pdev->dev.parent = &pdev->dev;
+
 		platform_set_drvdata(i2s->pdev, i2s);
 		ret = platform_device_add(i2s->pdev);
 		if (ret < 0)
@@ -1018,18 +1089,18 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec)
 
 static const struct of_device_id exynos_i2s_match[];
 
-static inline int samsung_i2s_get_driver_data(struct platform_device *pdev)
+static inline const struct samsung_i2s_dai_data *samsung_i2s_get_driver_data(
+						struct platform_device *pdev)
 {
 #ifdef CONFIG_OF
-	struct samsung_i2s_dai_data *data;
 	if (pdev->dev.of_node) {
 		const struct of_device_id *match;
 		match = of_match_node(exynos_i2s_match, pdev->dev.of_node);
-		data = (struct samsung_i2s_dai_data *) match->data;
-		return data->dai_type;
+		return match->data;
 	} else
 #endif
-		return platform_get_device_id(pdev)->driver_data;
+		return (struct samsung_i2s_dai_data *)
+				platform_get_device_id(pdev)->driver_data;
 }
 
 #ifdef CONFIG_PM_RUNTIME
@@ -1060,13 +1131,13 @@ static int samsung_i2s_probe(struct platform_device *pdev)
 	struct resource *res;
 	u32 regs_base, quirks = 0, idma_addr = 0;
 	struct device_node *np = pdev->dev.of_node;
-	enum samsung_dai_type samsung_dai_type;
+	const struct samsung_i2s_dai_data *i2s_dai_data;
 	int ret = 0;
 
 	/* Call during Seconday interface registration */
-	samsung_dai_type = samsung_i2s_get_driver_data(pdev);
+	i2s_dai_data = samsung_i2s_get_driver_data(pdev);
 
-	if (samsung_dai_type == TYPE_SEC) {
+	if (i2s_dai_data->dai_type == TYPE_SEC) {
 		sec_dai = dev_get_drvdata(&pdev->dev);
 		if (!sec_dai) {
 			dev_err(&pdev->dev, "Unable to get drvdata\n");
@@ -1075,7 +1146,7 @@ static int samsung_i2s_probe(struct platform_device *pdev)
 		snd_soc_register_component(&sec_dai->pdev->dev,
 					   &samsung_i2s_component,
 					   &sec_dai->i2s_dai_drv, 1);
-		asoc_dma_platform_register(&pdev->dev);
+		samsung_asoc_dma_platform_register(&pdev->dev);
 		return 0;
 	}
 
@@ -1115,15 +1186,7 @@ static int samsung_i2s_probe(struct platform_device *pdev)
 			idma_addr = i2s_cfg->idma_addr;
 		}
 	} else {
-		if (of_find_property(np, "samsung,supports-6ch", NULL))
-			quirks |= QUIRK_PRI_6CHAN;
-
-		if (of_find_property(np, "samsung,supports-secdai", NULL))
-			quirks |= QUIRK_SEC_DAI;
-
-		if (of_find_property(np, "samsung,supports-rstclr", NULL))
-			quirks |= QUIRK_NEED_RSTCLR;
-
+		quirks = i2s_dai_data->quirks;
 		if (of_property_read_u32(np, "samsung,idma-addr",
 					 &idma_addr)) {
 			if (quirks & QUIRK_SEC_DAI) {
@@ -1200,7 +1263,7 @@ static int samsung_i2s_probe(struct platform_device *pdev)
 
 	pm_runtime_enable(&pdev->dev);
 
-	asoc_dma_platform_register(&pdev->dev);
+	samsung_asoc_dma_platform_register(&pdev->dev);
 
 	return 0;
 err:
@@ -1230,33 +1293,59 @@ static int samsung_i2s_remove(struct platform_device *pdev)
 	i2s->pri_dai = NULL;
 	i2s->sec_dai = NULL;
 
-	asoc_dma_platform_unregister(&pdev->dev);
+	samsung_asoc_dma_platform_unregister(&pdev->dev);
 	snd_soc_unregister_component(&pdev->dev);
 
 	return 0;
 }
 
+static const struct samsung_i2s_dai_data i2sv3_dai_type = {
+	.dai_type = TYPE_PRI,
+	.quirks = QUIRK_NO_MUXPSR,
+};
+
+static const struct samsung_i2s_dai_data i2sv5_dai_type = {
+	.dai_type = TYPE_PRI,
+	.quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR,
+};
+
+static const struct samsung_i2s_dai_data i2sv6_dai_type = {
+	.dai_type = TYPE_PRI,
+	.quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR |
+			QUIRK_SUPPORTS_TDM,
+};
+
+static const struct samsung_i2s_dai_data samsung_dai_type_pri = {
+	.dai_type = TYPE_PRI,
+};
+
+static const struct samsung_i2s_dai_data samsung_dai_type_sec = {
+	.dai_type = TYPE_SEC,
+};
+
 static struct platform_device_id samsung_i2s_driver_ids[] = {
 	{
 		.name           = "samsung-i2s",
-		.driver_data	= TYPE_PRI,
+		.driver_data    = (kernel_ulong_t)&samsung_dai_type_pri,
 	}, {
 		.name           = "samsung-i2s-sec",
-		.driver_data	= TYPE_SEC,
+		.driver_data    = (kernel_ulong_t)&samsung_dai_type_sec,
 	},
 	{},
 };
 MODULE_DEVICE_TABLE(platform, samsung_i2s_driver_ids);
 
 #ifdef CONFIG_OF
-static struct samsung_i2s_dai_data samsung_i2s_dai_data_array[] = {
-	[TYPE_PRI] = { TYPE_PRI },
-	[TYPE_SEC] = { TYPE_SEC },
-};
-
 static const struct of_device_id exynos_i2s_match[] = {
-	{ .compatible = "samsung,i2s-v5",
-	  .data = &samsung_i2s_dai_data_array[TYPE_PRI],
+	{
+		.compatible = "samsung,s3c6410-i2s",
+		.data = &i2sv3_dai_type,
+	}, {
+		.compatible = "samsung,s5pv210-i2s",
+		.data = &i2sv5_dai_type,
+	}, {
+		.compatible = "samsung,exynos5420-i2s",
+		.data = &i2sv6_dai_type,
 	},
 	{},
 };
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index 1566afe9ef52..e54256fc4b2c 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -594,7 +594,7 @@ static int s3c_pcm_dev_probe(struct platform_device *pdev)
 		goto err5;
 	}
 
-	ret = asoc_dma_platform_register(&pdev->dev);
+	ret = samsung_asoc_dma_platform_register(&pdev->dev);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to get register DMA: %d\n", ret);
 		goto err6;
@@ -623,7 +623,7 @@ static int s3c_pcm_dev_remove(struct platform_device *pdev)
 	struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id];
 	struct resource *mem_res;
 
-	asoc_dma_platform_unregister(&pdev->dev);
+	samsung_asoc_dma_platform_unregister(&pdev->dev);
 	snd_soc_unregister_component(&pdev->dev);
 
 	pm_runtime_disable(&pdev->dev);
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 47e23864ea72..ea885cb9f76c 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -176,7 +176,7 @@ static int s3c2412_iis_dev_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	ret = asoc_dma_platform_register(&pdev->dev);
+	ret = samsung_asoc_dma_platform_register(&pdev->dev);
 	if (ret) {
 		pr_err("failed to register the DMA: %d\n", ret);
 		goto err;
@@ -190,7 +190,7 @@ err:
 
 static int s3c2412_iis_dev_remove(struct platform_device *pdev)
 {
-	asoc_dma_platform_unregister(&pdev->dev);
+	samsung_asoc_dma_platform_unregister(&pdev->dev);
 	snd_soc_unregister_component(&pdev->dev);
 	return 0;
 }
diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c
index 8b3414551a62..9c8ebd872fac 100644
--- a/sound/soc/samsung/s3c24xx-i2s.c
+++ b/sound/soc/samsung/s3c24xx-i2s.c
@@ -480,7 +480,7 @@ static int s3c24xx_iis_dev_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	ret = asoc_dma_platform_register(&pdev->dev);
+	ret = samsung_asoc_dma_platform_register(&pdev->dev);
 	if (ret) {
 		pr_err("failed to register the dma: %d\n", ret);
 		goto err;
@@ -494,7 +494,7 @@ err:
 
 static int s3c24xx_iis_dev_remove(struct platform_device *pdev)
 {
-	asoc_dma_platform_unregister(&pdev->dev);
+	samsung_asoc_dma_platform_unregister(&pdev->dev);
 	snd_soc_unregister_component(&pdev->dev);
 	return 0;
 }
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index 581ea4a06fc6..5fd7a05a9b9e 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -11,6 +11,7 @@
 #include <sound/pcm_params.h>
 #include <linux/module.h>
 #include <linux/of.h>
+#include <linux/of_device.h>
 
  /*
   * Default CFG switch settings to use this driver:
@@ -37,11 +38,19 @@
 /* SMDK has a 16.934MHZ crystal attached to WM8994 */
 #define SMDK_WM8994_FREQ 16934000
 
+struct smdk_wm8994_data {
+	int mclk1_rate;
+};
+
+/* Default SMDKs */
+static struct smdk_wm8994_data smdk_board_data = {
+	.mclk1_rate = SMDK_WM8994_FREQ,
+};
+
 static int smdk_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;
 	unsigned int pll_out;
 	int ret;
@@ -54,18 +63,6 @@ static int smdk_hw_params(struct snd_pcm_substream *substream,
 	else
 		pll_out = params_rate(params) * 256;
 
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
-					 | SND_SOC_DAIFMT_NB_NF
-					 | SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S
-					 | SND_SOC_DAIFMT_NB_NF
-					 | SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
 	ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1,
 					SMDK_WM8994_FREQ, pll_out);
 	if (ret < 0)
@@ -131,6 +128,8 @@ static struct snd_soc_dai_link smdk_dai[] = {
 		.platform_name = "samsung-i2s.0",
 		.codec_name = "wm8994-codec",
 		.init = smdk_wm8994_init_paiftx,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBM_CFM,
 		.ops = &smdk_ops,
 	}, { /* Sec_Fifo Playback i/f */
 		.name = "Sec_FIFO TX",
@@ -139,6 +138,8 @@ static struct snd_soc_dai_link smdk_dai[] = {
 		.codec_dai_name = "wm8994-aif1",
 		.platform_name = "samsung-i2s-sec",
 		.codec_name = "wm8994-codec",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBM_CFM,
 		.ops = &smdk_ops,
 	},
 };
@@ -150,15 +151,28 @@ static struct snd_soc_card smdk = {
 	.num_links = ARRAY_SIZE(smdk_dai),
 };
 
+#ifdef CONFIG_OF
+static const struct of_device_id samsung_wm8994_of_match[] = {
+	{ .compatible = "samsung,smdk-wm8994", .data = &smdk_board_data },
+	{},
+};
+MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match);
+#endif /* CONFIG_OF */
 
 static int smdk_audio_probe(struct platform_device *pdev)
 {
 	int ret;
 	struct device_node *np = pdev->dev.of_node;
 	struct snd_soc_card *card = &smdk;
+	struct smdk_wm8994_data *board;
+	const struct of_device_id *id;
 
 	card->dev = &pdev->dev;
 
+	board = devm_kzalloc(&pdev->dev, sizeof(*board), GFP_KERNEL);
+	if (!board)
+		return -ENOMEM;
+
 	if (np) {
 		smdk_dai[0].cpu_dai_name = NULL;
 		smdk_dai[0].cpu_of_node = of_parse_phandle(np,
@@ -173,6 +187,12 @@ static int smdk_audio_probe(struct platform_device *pdev)
 		smdk_dai[0].platform_of_node = smdk_dai[0].cpu_of_node;
 	}
 
+	id = of_match_device(samsung_wm8994_of_match, &pdev->dev);
+	if (id)
+		*board = *((struct smdk_wm8994_data *)id->data);
+
+	platform_set_drvdata(pdev, board);
+
 	ret = snd_soc_register_card(card);
 
 	if (ret)
@@ -190,17 +210,9 @@ static int smdk_audio_remove(struct platform_device *pdev)
 	return 0;
 }
 
-#ifdef CONFIG_OF
-static const struct of_device_id samsung_wm8994_of_match[] = {
-	{ .compatible = "samsung,smdk-wm8994", },
-	{},
-};
-MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match);
-#endif /* CONFIG_OF */
-
 static struct platform_driver smdk_audio_driver = {
 	.driver		= {
-		.name	= "smdk-audio",
+		.name	= "smdk-audio-wm8894",
 		.owner	= THIS_MODULE,
 		.of_match_table = of_match_ptr(samsung_wm8994_of_match),
 	},
@@ -212,4 +224,4 @@ module_platform_driver(smdk_audio_driver);
 
 MODULE_DESCRIPTION("ALSA SoC SMDK WM8994");
 MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:smdk-audio");
+MODULE_ALIAS("platform:smdk-audio-wm8994");
diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c
index 2e5ebb2f1982..28487dcc4538 100644
--- a/sound/soc/samsung/spdif.c
+++ b/sound/soc/samsung/spdif.c
@@ -395,7 +395,7 @@ static int spdif_probe(struct platform_device *pdev)
 
 	spin_lock_init(&spdif->lock);
 
-	spdif->pclk = clk_get(&pdev->dev, "spdif");
+	spdif->pclk = devm_clk_get(&pdev->dev, "spdif");
 	if (IS_ERR(spdif->pclk)) {
 		dev_err(&pdev->dev, "failed to get peri-clock\n");
 		ret = -ENOENT;
@@ -403,7 +403,7 @@ static int spdif_probe(struct platform_device *pdev)
 	}
 	clk_prepare_enable(spdif->pclk);
 
-	spdif->sclk = clk_get(&pdev->dev, "sclk_spdif");
+	spdif->sclk = devm_clk_get(&pdev->dev, "sclk_spdif");
 	if (IS_ERR(spdif->sclk)) {
 		dev_err(&pdev->dev, "failed to get internal source clock\n");
 		ret = -ENOENT;
@@ -442,7 +442,7 @@ static int spdif_probe(struct platform_device *pdev)
 
 	spdif->dma_playback = &spdif_stereo_out;
 
-	ret = asoc_dma_platform_register(&pdev->dev);
+	ret = samsung_asoc_dma_platform_register(&pdev->dev);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to register DMA: %d\n", ret);
 		goto err5;
@@ -457,10 +457,8 @@ err3:
 	release_mem_region(mem_res->start, resource_size(mem_res));
 err2:
 	clk_disable_unprepare(spdif->sclk);
-	clk_put(spdif->sclk);
 err1:
 	clk_disable_unprepare(spdif->pclk);
-	clk_put(spdif->pclk);
 err0:
 	return ret;
 }
@@ -470,7 +468,7 @@ static int spdif_remove(struct platform_device *pdev)
 	struct samsung_spdif_info *spdif = &spdif_info;
 	struct resource *mem_res;
 
-	asoc_dma_platform_unregister(&pdev->dev);
+	samsung_asoc_dma_platform_unregister(&pdev->dev);
 	snd_soc_unregister_component(&pdev->dev);
 
 	iounmap(spdif->regs);
@@ -480,9 +478,7 @@ static int spdif_remove(struct platform_device *pdev)
 		release_mem_region(mem_res->start, resource_size(mem_res));
 
 	clk_disable_unprepare(spdif->sclk);
-	clk_put(spdif->sclk);
 	clk_disable_unprepare(spdif->pclk);
-	clk_put(spdif->pclk);
 
 	return 0;
 }
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 6bcb1164d599..56d8ff6a402d 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -34,6 +34,13 @@ config SND_SOC_SH4_SIU
 	select SH_DMAE
 	select FW_LOADER
 
+config SND_SOC_RCAR
+	tristate "R-Car series SRU/SCU/SSIU/SSI support"
+	select SND_SIMPLE_CARD
+	select RCAR_CLK_ADG
+	help
+	  This option enables R-Car SUR/SCU/SSIU/SSI sound support
+
 ##
 ## Boards
 ##
diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile
index 849b387d17d9..aaf3dcd1ee2a 100644
--- a/sound/soc/sh/Makefile
+++ b/sound/soc/sh/Makefile
@@ -12,6 +12,9 @@ obj-$(CONFIG_SND_SOC_SH4_SSI)	+= snd-soc-ssi.o
 obj-$(CONFIG_SND_SOC_SH4_FSI)	+= snd-soc-fsi.o
 obj-$(CONFIG_SND_SOC_SH4_SIU)	+= snd-soc-siu.o
 
+## audio units for R-Car
+obj-$(CONFIG_SND_SOC_RCAR)	+= rcar/
+
 ## boards
 snd-soc-sh7760-ac97-objs	:= sh7760-ac97.o
 snd-soc-migor-objs		:= migor.o
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 30390260bb67..b33ca7cd085b 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -235,6 +235,8 @@ struct fsi_stream {
 	struct sh_dmae_slave	slave; /* see fsi_handler_init() */
 	struct work_struct	work;
 	dma_addr_t		dma;
+	int			loop_cnt;
+	int			additional_pos;
 };
 
 struct fsi_clk {
@@ -1289,6 +1291,8 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io)
 	io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
 			 BUSOP_SET(16, PACKAGE_16BITBUS_STREAM);
 
+	io->loop_cnt = 2; /* push 1st, 2nd period first, then 3rd, 4th... */
+	io->additional_pos = 0;
 	io->dma = dma_map_single(dai->dev, runtime->dma_area,
 				 snd_pcm_lib_buffer_bytes(io->substream), dir);
 	return 0;
@@ -1305,11 +1309,15 @@ static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io)
 	return 0;
 }
 
-static dma_addr_t fsi_dma_get_area(struct fsi_stream *io)
+static dma_addr_t fsi_dma_get_area(struct fsi_stream *io, int additional)
 {
 	struct snd_pcm_runtime *runtime = io->substream->runtime;
+	int period = io->period_pos + additional;
 
-	return io->dma + samples_to_bytes(runtime, io->buff_sample_pos);
+	if (period >= runtime->periods)
+		period = 0;
+
+	return io->dma + samples_to_bytes(runtime, period * io->period_samples);
 }
 
 static void fsi_dma_complete(void *data)
@@ -1321,7 +1329,7 @@ static void fsi_dma_complete(void *data)
 	enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ?
 		DMA_TO_DEVICE : DMA_FROM_DEVICE;
 
-	dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io),
+	dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io, 0),
 			samples_to_bytes(runtime, io->period_samples), dir);
 
 	io->buff_sample_pos += io->period_samples;
@@ -1347,7 +1355,7 @@ static void fsi_dma_do_work(struct work_struct *work)
 	struct snd_pcm_runtime *runtime;
 	enum dma_data_direction dir;
 	int is_play = fsi_stream_is_play(fsi, io);
-	int len;
+	int len, i;
 	dma_addr_t buf;
 
 	if (!fsi_stream_is_working(fsi, io))
@@ -1357,26 +1365,33 @@ static void fsi_dma_do_work(struct work_struct *work)
 	runtime	= io->substream->runtime;
 	dir	= is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
 	len	= samples_to_bytes(runtime, io->period_samples);
-	buf	= fsi_dma_get_area(io);
 
-	dma_sync_single_for_device(dai->dev, buf, len, dir);
+	for (i = 0; i < io->loop_cnt; i++) {
+		buf	= fsi_dma_get_area(io, io->additional_pos);
 
-	desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
-					   DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
-	if (!desc) {
-		dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
-		return;
-	}
+		dma_sync_single_for_device(dai->dev, buf, len, dir);
 
-	desc->callback		= fsi_dma_complete;
-	desc->callback_param	= io;
+		desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
+					DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+		if (!desc) {
+			dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
+			return;
+		}
 
-	if (dmaengine_submit(desc) < 0) {
-		dev_err(dai->dev, "tx_submit() fail\n");
-		return;
+		desc->callback		= fsi_dma_complete;
+		desc->callback_param	= io;
+
+		if (dmaengine_submit(desc) < 0) {
+			dev_err(dai->dev, "tx_submit() fail\n");
+			return;
+		}
+
+		dma_async_issue_pending(io->chan);
+
+		io->additional_pos = 1;
 	}
 
-	dma_async_issue_pending(io->chan);
+	io->loop_cnt = 1;
 
 	/*
 	 * FIXME
diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile
new file mode 100644
index 000000000000..0ff492df7929
--- /dev/null
+++ b/sound/soc/sh/rcar/Makefile
@@ -0,0 +1,2 @@
+snd-soc-rcar-objs	:= core.o gen.o scu.o adg.o ssi.o
+obj-$(CONFIG_SND_SOC_RCAR)	+= snd-soc-rcar.o
+\ No newline at end of file
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
new file mode 100644
index 000000000000..d80deb7ccf13
--- /dev/null
+++ b/sound/soc/sh/rcar/adg.c
@@ -0,0 +1,234 @@
+/*
+ * Helper routines for R-Car sound ADG.
+ *
+ *  Copyright (C) 2013  Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * This file is subject to the terms and conditions of the GNU General Public
+ * License.  See the file "COPYING" in the main directory of this archive
+ * for more details.
+ */
+#include <linux/sh_clk.h>
+#include <mach/clock.h>
+#include "rsnd.h"
+
+#define CLKA	0
+#define CLKB	1
+#define CLKC	2
+#define CLKI	3
+#define CLKMAX	4
+
+struct rsnd_adg {
+	struct clk *clk[CLKMAX];
+
+	int rate_of_441khz_div_6;
+	int rate_of_48khz_div_6;
+};
+
+#define for_each_rsnd_clk(pos, adg, i)		\
+	for (i = 0, (pos) = adg->clk[i];	\
+	     i < CLKMAX;			\
+	     i++, (pos) = adg->clk[i])
+#define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg)
+
+static enum rsnd_reg rsnd_adg_ssi_reg_get(int id)
+{
+	enum rsnd_reg reg;
+
+	/*
+	 * SSI 8 is not connected to ADG.
+	 * it works with SSI 7
+	 */
+	if (id == 8)
+		return RSND_REG_MAX;
+
+	if (0 <= id && id <= 3)
+		reg = RSND_REG_AUDIO_CLK_SEL0;
+	else if (4 <= id && id <= 7)
+		reg = RSND_REG_AUDIO_CLK_SEL1;
+	else
+		reg = RSND_REG_AUDIO_CLK_SEL2;
+
+	return reg;
+}
+
+int rsnd_adg_ssi_clk_stop(struct rsnd_mod *mod)
+{
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	enum rsnd_reg reg;
+	int id;
+
+	/*
+	 * "mod" = "ssi" here.
+	 * we can get "ssi id" from mod
+	 */
+	id  = rsnd_mod_id(mod);
+	reg = rsnd_adg_ssi_reg_get(id);
+
+	rsnd_write(priv, mod, reg, 0);
+
+	return 0;
+}
+
+int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate)
+{
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+	struct device *dev = rsnd_priv_to_dev(priv);
+	struct clk *clk;
+	enum rsnd_reg reg;
+	int id, shift, i;
+	u32 data;
+	int sel_table[] = {
+		[CLKA] = 0x1,
+		[CLKB] = 0x2,
+		[CLKC] = 0x3,
+		[CLKI] = 0x0,
+	};
+
+	dev_dbg(dev, "request clock = %d\n", rate);
+
+	/*
+	 * find suitable clock from
+	 * AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI.
+	 */
+	data = 0;
+	for_each_rsnd_clk(clk, adg, i) {
+		if (rate == clk_get_rate(clk)) {
+			data = sel_table[i];
+			goto found_clock;
+		}
+	}
+
+	/*
+	 * find 1/6 clock from BRGA/BRGB
+	 */
+	if (rate == adg->rate_of_441khz_div_6) {
+		data = 0x10;
+		goto found_clock;
+	}
+
+	if (rate == adg->rate_of_48khz_div_6) {
+		data = 0x20;
+		goto found_clock;
+	}
+
+	return -EIO;
+
+found_clock:
+
+	/*
+	 * This "mod" = "ssi" here.
+	 * we can get "ssi id" from mod
+	 */
+	id  = rsnd_mod_id(mod);
+	reg = rsnd_adg_ssi_reg_get(id);
+
+	dev_dbg(dev, "ADG: ssi%d selects clk%d = %d", id, i, rate);
+
+	/*
+	 * Enable SSIx clock
+	 */
+	shift = (id % 4) * 8;
+
+	rsnd_bset(priv, mod, reg,
+		   0xFF << shift,
+		   data << shift);
+
+	return 0;
+}
+
+static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
+{
+	struct clk *clk;
+	unsigned long rate;
+	u32 ckr;
+	int i;
+	int brg_table[] = {
+		[CLKA] = 0x0,
+		[CLKB] = 0x1,
+		[CLKC] = 0x4,
+		[CLKI] = 0x2,
+	};
+
+	/*
+	 * This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC
+	 * have 44.1kHz or 48kHz base clocks for now.
+	 *
+	 * SSI itself can divide parent clock by 1/1 - 1/16
+	 * So,  BRGA outputs 44.1kHz base parent clock 1/32,
+	 * and, BRGB outputs 48.0kHz base parent clock 1/32 here.
+	 * see
+	 *	rsnd_adg_ssi_clk_try_start()
+	 */
+	ckr = 0;
+	adg->rate_of_441khz_div_6 = 0;
+	adg->rate_of_48khz_div_6  = 0;
+	for_each_rsnd_clk(clk, adg, i) {
+		rate = clk_get_rate(clk);
+
+		if (0 == rate) /* not used */
+			continue;
+
+		/* RBGA */
+		if (!adg->rate_of_441khz_div_6 && (0 == rate % 44100)) {
+			adg->rate_of_441khz_div_6 = rate / 6;
+			ckr |= brg_table[i] << 20;
+		}
+
+		/* RBGB */
+		if (!adg->rate_of_48khz_div_6 && (0 == rate % 48000)) {
+			adg->rate_of_48khz_div_6 = rate / 6;
+			ckr |= brg_table[i] << 16;
+		}
+	}
+
+	rsnd_priv_bset(priv, SSICKR, 0x00FF0000, ckr);
+	rsnd_priv_write(priv, BRRA,  0x00000002); /* 1/6 */
+	rsnd_priv_write(priv, BRRB,  0x00000002); /* 1/6 */
+}
+
+int rsnd_adg_probe(struct platform_device *pdev,
+		   struct rcar_snd_info *info,
+		   struct rsnd_priv *priv)
+{
+	struct rsnd_adg *adg;
+	struct device *dev = rsnd_priv_to_dev(priv);
+	struct clk *clk;
+	int i;
+
+	adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL);
+	if (!adg) {
+		dev_err(dev, "ADG allocate failed\n");
+		return -ENOMEM;
+	}
+
+	adg->clk[CLKA] = clk_get(NULL, "audio_clk_a");
+	adg->clk[CLKB] = clk_get(NULL, "audio_clk_b");
+	adg->clk[CLKC] = clk_get(NULL, "audio_clk_c");
+	adg->clk[CLKI] = clk_get(NULL, "audio_clk_internal");
+	for_each_rsnd_clk(clk, adg, i) {
+		if (IS_ERR(clk)) {
+			dev_err(dev, "Audio clock failed\n");
+			return -EIO;
+		}
+	}
+
+	rsnd_adg_ssi_clk_init(priv, adg);
+
+	priv->adg = adg;
+
+	dev_dbg(dev, "adg probed\n");
+
+	return 0;
+}
+
+void rsnd_adg_remove(struct platform_device *pdev,
+		     struct rsnd_priv *priv)
+{
+	struct rsnd_adg *adg = priv->adg;
+	struct clk *clk;
+	int i;
+
+	for_each_rsnd_clk(clk, adg, i)
+		clk_put(clk);
+}
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
new file mode 100644
index 000000000000..a35706028514
--- /dev/null
+++ b/sound/soc/sh/rcar/core.c
@@ -0,0 +1,861 @@
+/*
+ * Renesas R-Car SRU/SCU/SSIU/SSI support
+ *
+ * Copyright (C) 2013 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * Based on fsi.c
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+/*
+ * Renesas R-Car sound device structure
+ *
+ * Gen1
+ *
+ * SRU		: Sound Routing Unit
+ *  - SRC	: Sampling Rate Converter
+ *  - CMD
+ *    - CTU	: Channel Count Conversion Unit
+ *    - MIX	: Mixer
+ *    - DVC	: Digital Volume and Mute Function
+ *  - SSI	: Serial Sound Interface
+ *
+ * Gen2
+ *
+ * SCU		: Sampling Rate Converter Unit
+ *  - SRC	: Sampling Rate Converter
+ *  - CMD
+ *   - CTU	: Channel Count Conversion Unit
+ *   - MIX	: Mixer
+ *   - DVC	: Digital Volume and Mute Function
+ * SSIU		: Serial Sound Interface Unit
+ *  - SSI	: Serial Sound Interface
+ */
+
+/*
+ *	driver data Image
+ *
+ * rsnd_priv
+ *   |
+ *   | ** this depends on Gen1/Gen2
+ *   |
+ *   +- gen
+ *   |
+ *   | ** these depend on data path
+ *   | ** gen and platform data control it
+ *   |
+ *   +- rdai[0]
+ *   |   |		 sru     ssiu      ssi
+ *   |   +- playback -> [mod] -> [mod] -> [mod] -> ...
+ *   |   |
+ *   |   |		 sru     ssiu      ssi
+ *   |   +- capture  -> [mod] -> [mod] -> [mod] -> ...
+ *   |
+ *   +- rdai[1]
+ *   |   |		 sru     ssiu      ssi
+ *   |   +- playback -> [mod] -> [mod] -> [mod] -> ...
+ *   |   |
+ *   |   |		 sru     ssiu      ssi
+ *   |   +- capture  -> [mod] -> [mod] -> [mod] -> ...
+ *   ...
+ *   |
+ *   | ** these control ssi
+ *   |
+ *   +- ssi
+ *   |  |
+ *   |  +- ssi[0]
+ *   |  +- ssi[1]
+ *   |  +- ssi[2]
+ *   |  ...
+ *   |
+ *   | ** these control scu
+ *   |
+ *   +- scu
+ *      |
+ *      +- scu[0]
+ *      +- scu[1]
+ *      +- scu[2]
+ *      ...
+ *
+ *
+ * for_each_rsnd_dai(xx, priv, xx)
+ *  rdai[0] => rdai[1] => rdai[2] => ...
+ *
+ * for_each_rsnd_mod(xx, rdai, xx)
+ *  [mod] => [mod] => [mod] => ...
+ *
+ * rsnd_dai_call(xxx, fn )
+ *  [mod]->fn() -> [mod]->fn() -> [mod]->fn()...
+ *
+ */
+#include <linux/pm_runtime.h>
+#include "rsnd.h"
+
+#define RSND_RATES SNDRV_PCM_RATE_8000_96000
+#define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
+
+/*
+ *	rsnd_platform functions
+ */
+#define rsnd_platform_call(priv, dai, func, param...)	\
+	(!(priv->info->func) ? -ENODEV :		\
+	 priv->info->func(param))
+
+
+/*
+ *	basic function
+ */
+u32 rsnd_read(struct rsnd_priv *priv,
+	      struct rsnd_mod *mod, enum rsnd_reg reg)
+{
+	void __iomem *base = rsnd_gen_reg_get(priv, mod, reg);
+
+	BUG_ON(!base);
+
+	return ioread32(base);
+}
+
+void rsnd_write(struct rsnd_priv *priv,
+		struct rsnd_mod *mod,
+		enum rsnd_reg reg, u32 data)
+{
+	void __iomem *base = rsnd_gen_reg_get(priv, mod, reg);
+	struct device *dev = rsnd_priv_to_dev(priv);
+
+	BUG_ON(!base);
+
+	dev_dbg(dev, "w %p : %08x\n", base, data);
+
+	iowrite32(data, base);
+}
+
+void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod,
+	       enum rsnd_reg reg, u32 mask, u32 data)
+{
+	void __iomem *base = rsnd_gen_reg_get(priv, mod, reg);
+	struct device *dev = rsnd_priv_to_dev(priv);
+	u32 val;
+
+	BUG_ON(!base);
+
+	val = ioread32(base);
+	val &= ~mask;
+	val |= data & mask;
+	iowrite32(val, base);
+
+	dev_dbg(dev, "s %p : %08x\n", base, val);
+}
+
+/*
+ *	rsnd_mod functions
+ */
+char *rsnd_mod_name(struct rsnd_mod *mod)
+{
+	if (!mod || !mod->ops)
+		return "unknown";
+
+	return mod->ops->name;
+}
+
+void rsnd_mod_init(struct rsnd_priv *priv,
+		   struct rsnd_mod *mod,
+		   struct rsnd_mod_ops *ops,
+		   int id)
+{
+	mod->priv	= priv;
+	mod->id		= id;
+	mod->ops	= ops;
+	INIT_LIST_HEAD(&mod->list);
+}
+
+/*
+ *	rsnd_dma functions
+ */
+static void rsnd_dma_continue(struct rsnd_dma *dma)
+{
+	/* push next A or B plane */
+	dma->submit_loop = 1;
+	schedule_work(&dma->work);
+}
+
+void rsnd_dma_start(struct rsnd_dma *dma)
+{
+	/* push both A and B plane*/
+	dma->submit_loop = 2;
+	schedule_work(&dma->work);
+}
+
+void rsnd_dma_stop(struct rsnd_dma *dma)
+{
+	dma->submit_loop = 0;
+	cancel_work_sync(&dma->work);
+	dmaengine_terminate_all(dma->chan);
+}
+
+static void rsnd_dma_complete(void *data)
+{
+	struct rsnd_dma *dma = (struct rsnd_dma *)data;
+	struct rsnd_priv *priv = dma->priv;
+	unsigned long flags;
+
+	rsnd_lock(priv, flags);
+
+	dma->complete(dma);
+
+	if (dma->submit_loop)
+		rsnd_dma_continue(dma);
+
+	rsnd_unlock(priv, flags);
+}
+
+static void rsnd_dma_do_work(struct work_struct *work)
+{
+	struct rsnd_dma *dma = container_of(work, struct rsnd_dma, work);
+	struct rsnd_priv *priv = dma->priv;
+	struct device *dev = rsnd_priv_to_dev(priv);
+	struct dma_async_tx_descriptor *desc;
+	dma_addr_t buf;
+	size_t len;
+	int i;
+
+	for (i = 0; i < dma->submit_loop; i++) {
+
+		if (dma->inquiry(dma, &buf, &len) < 0)
+			return;
+
+		desc = dmaengine_prep_slave_single(
+			dma->chan, buf, len, dma->dir,
+			DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+		if (!desc) {
+			dev_err(dev, "dmaengine_prep_slave_sg() fail\n");
+			return;
+		}
+
+		desc->callback		= rsnd_dma_complete;
+		desc->callback_param	= dma;
+
+		if (dmaengine_submit(desc) < 0) {
+			dev_err(dev, "dmaengine_submit() fail\n");
+			return;
+		}
+
+	}
+
+	dma_async_issue_pending(dma->chan);
+}
+
+int rsnd_dma_available(struct rsnd_dma *dma)
+{
+	return !!dma->chan;
+}
+
+static bool rsnd_dma_filter(struct dma_chan *chan, void *param)
+{
+	chan->private = param;
+
+	return true;
+}
+
+int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
+		  int is_play, int id,
+		  int (*inquiry)(struct rsnd_dma *dma,
+				  dma_addr_t *buf, int *len),
+		  int (*complete)(struct rsnd_dma *dma))
+{
+	struct device *dev = rsnd_priv_to_dev(priv);
+	dma_cap_mask_t mask;
+
+	if (dma->chan) {
+		dev_err(dev, "it already has dma channel\n");
+		return -EIO;
+	}
+
+	dma_cap_zero(mask);
+	dma_cap_set(DMA_SLAVE, mask);
+
+	dma->slave.shdma_slave.slave_id = id;
+
+	dma->chan = dma_request_channel(mask, rsnd_dma_filter,
+					&dma->slave.shdma_slave);
+	if (!dma->chan) {
+		dev_err(dev, "can't get dma channel\n");
+		return -EIO;
+	}
+
+	dma->dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
+	dma->priv = priv;
+	dma->inquiry = inquiry;
+	dma->complete = complete;
+	INIT_WORK(&dma->work, rsnd_dma_do_work);
+
+	return 0;
+}
+
+void  rsnd_dma_quit(struct rsnd_priv *priv,
+		    struct rsnd_dma *dma)
+{
+	if (dma->chan)
+		dma_release_channel(dma->chan);
+
+	dma->chan = NULL;
+}
+
+/*
+ *	rsnd_dai functions
+ */
+#define rsnd_dai_call(rdai, io, fn)			\
+({							\
+	struct rsnd_mod *mod, *n;			\
+	int ret = 0;					\
+	for_each_rsnd_mod(mod, n, io) {			\
+		ret = rsnd_mod_call(mod, fn, rdai, io);	\
+		if (ret < 0)				\
+			break;				\
+	}						\
+	ret;						\
+})
+
+int rsnd_dai_connect(struct rsnd_dai *rdai,
+		     struct rsnd_mod *mod,
+		     struct rsnd_dai_stream *io)
+{
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct device *dev = rsnd_priv_to_dev(priv);
+
+	if (!mod) {
+		dev_err(dev, "NULL mod\n");
+		return -EIO;
+	}
+
+	if (!list_empty(&mod->list)) {
+		dev_err(dev, "%s%d is not empty\n",
+			rsnd_mod_name(mod),
+			rsnd_mod_id(mod));
+		return -EIO;
+	}
+
+	list_add_tail(&mod->list, &io->head);
+
+	return 0;
+}
+
+int rsnd_dai_disconnect(struct rsnd_mod *mod)
+{
+	list_del_init(&mod->list);
+
+	return 0;
+}
+
+int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai)
+{
+	int id = rdai - priv->rdai;
+
+	if ((id < 0) || (id >= rsnd_dai_nr(priv)))
+		return -EINVAL;
+
+	return id;
+}
+
+struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id)
+{
+	return priv->rdai + id;
+}
+
+static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai)
+{
+	struct rsnd_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+	return rsnd_dai_get(priv, dai->id);
+}
+
+int rsnd_dai_is_play(struct rsnd_dai *rdai, struct rsnd_dai_stream *io)
+{
+	return &rdai->playback == io;
+}
+
+/*
+ *	rsnd_soc_dai functions
+ */
+int rsnd_dai_pointer_offset(struct rsnd_dai_stream *io, int additional)
+{
+	struct snd_pcm_substream *substream = io->substream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	int pos = io->byte_pos + additional;
+
+	pos %= (runtime->periods * io->byte_per_period);
+
+	return pos;
+}
+
+void rsnd_dai_pointer_update(struct rsnd_dai_stream *io, int byte)
+{
+	io->byte_pos += byte;
+
+	if (io->byte_pos >= io->next_period_byte) {
+		struct snd_pcm_substream *substream = io->substream;
+		struct snd_pcm_runtime *runtime = substream->runtime;
+
+		io->period_pos++;
+		io->next_period_byte += io->byte_per_period;
+
+		if (io->period_pos >= runtime->periods) {
+			io->byte_pos = 0;
+			io->period_pos = 0;
+			io->next_period_byte = io->byte_per_period;
+		}
+
+		snd_pcm_period_elapsed(substream);
+	}
+}
+
+static int rsnd_dai_stream_init(struct rsnd_dai_stream *io,
+				struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	if (!list_empty(&io->head))
+		return -EIO;
+
+	INIT_LIST_HEAD(&io->head);
+	io->substream		= substream;
+	io->byte_pos		= 0;
+	io->period_pos		= 0;
+	io->byte_per_period	= runtime->period_size *
+				  runtime->channels *
+				  samples_to_bytes(runtime, 1);
+	io->next_period_byte	= io->byte_per_period;
+
+	return 0;
+}
+
+static
+struct snd_soc_dai *rsnd_substream_to_dai(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	return  rtd->cpu_dai;
+}
+
+static
+struct rsnd_dai_stream *rsnd_rdai_to_io(struct rsnd_dai *rdai,
+					struct snd_pcm_substream *substream)
+{
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		return &rdai->playback;
+	else
+		return &rdai->capture;
+}
+
+static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+			    struct snd_soc_dai *dai)
+{
+	struct rsnd_priv *priv = snd_soc_dai_get_drvdata(dai);
+	struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
+	struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream);
+	struct rsnd_mod *mod = rsnd_ssi_mod_get_frm_dai(priv,
+						rsnd_dai_id(priv, rdai),
+						rsnd_dai_is_play(rdai, io));
+	int ssi_id = rsnd_mod_id(mod);
+	int ret;
+	unsigned long flags;
+
+	rsnd_lock(priv, flags);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		ret = rsnd_dai_stream_init(io, substream);
+		if (ret < 0)
+			goto dai_trigger_end;
+
+		ret = rsnd_platform_call(priv, dai, start, ssi_id);
+		if (ret < 0)
+			goto dai_trigger_end;
+
+		ret = rsnd_gen_path_init(priv, rdai, io);
+		if (ret < 0)
+			goto dai_trigger_end;
+
+		ret = rsnd_dai_call(rdai, io, init);
+		if (ret < 0)
+			goto dai_trigger_end;
+
+		ret = rsnd_dai_call(rdai, io, start);
+		if (ret < 0)
+			goto dai_trigger_end;
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		ret = rsnd_dai_call(rdai, io, stop);
+		if (ret < 0)
+			goto dai_trigger_end;
+
+		ret = rsnd_dai_call(rdai, io, quit);
+		if (ret < 0)
+			goto dai_trigger_end;
+
+		ret = rsnd_gen_path_exit(priv, rdai, io);
+		if (ret < 0)
+			goto dai_trigger_end;
+
+		ret = rsnd_platform_call(priv, dai, stop, ssi_id);
+		if (ret < 0)
+			goto dai_trigger_end;
+		break;
+	default:
+		ret = -EINVAL;
+	}
+
+dai_trigger_end:
+	rsnd_unlock(priv, flags);
+
+	return ret;
+}
+
+static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		rdai->clk_master = 1;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		rdai->clk_master = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* set clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_IF:
+		rdai->bit_clk_inv = 0;
+		rdai->frm_clk_inv = 1;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		rdai->bit_clk_inv = 1;
+		rdai->frm_clk_inv = 0;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		rdai->bit_clk_inv = 1;
+		rdai->frm_clk_inv = 1;
+		break;
+	case SND_SOC_DAIFMT_NB_NF:
+	default:
+		rdai->bit_clk_inv = 0;
+		rdai->frm_clk_inv = 0;
+		break;
+	}
+
+	/* set format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		rdai->sys_delay = 0;
+		rdai->data_alignment = 0;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		rdai->sys_delay = 1;
+		rdai->data_alignment = 0;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		rdai->sys_delay = 1;
+		rdai->data_alignment = 1;
+		break;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops rsnd_soc_dai_ops = {
+	.trigger	= rsnd_soc_dai_trigger,
+	.set_fmt	= rsnd_soc_dai_set_fmt,
+};
+
+static int rsnd_dai_probe(struct platform_device *pdev,
+			  struct rcar_snd_info *info,
+			  struct rsnd_priv *priv)
+{
+	struct snd_soc_dai_driver *drv;
+	struct rsnd_dai *rdai;
+	struct rsnd_mod *pmod, *cmod;
+	struct device *dev = rsnd_priv_to_dev(priv);
+	int dai_nr;
+	int i;
+
+	/* get max dai nr */
+	for (dai_nr = 0; dai_nr < 32; dai_nr++) {
+		pmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 1);
+		cmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 0);
+
+		if (!pmod && !cmod)
+			break;
+	}
+
+	if (!dai_nr) {
+		dev_err(dev, "no dai\n");
+		return -EIO;
+	}
+
+	drv  = devm_kzalloc(dev, sizeof(*drv)  * dai_nr, GFP_KERNEL);
+	rdai = devm_kzalloc(dev, sizeof(*rdai) * dai_nr, GFP_KERNEL);
+	if (!drv || !rdai) {
+		dev_err(dev, "dai allocate failed\n");
+		return -ENOMEM;
+	}
+
+	for (i = 0; i < dai_nr; i++) {
+
+		pmod = rsnd_ssi_mod_get_frm_dai(priv, i, 1);
+		cmod = rsnd_ssi_mod_get_frm_dai(priv, i, 0);
+
+		/*
+		 *	init rsnd_dai
+		 */
+		INIT_LIST_HEAD(&rdai[i].playback.head);
+		INIT_LIST_HEAD(&rdai[i].capture.head);
+
+		snprintf(rdai[i].name, RSND_DAI_NAME_SIZE, "rsnd-dai.%d", i);
+
+		/*
+		 *	init snd_soc_dai_driver
+		 */
+		drv[i].name	= rdai[i].name;
+		drv[i].ops	= &rsnd_soc_dai_ops;
+		if (pmod) {
+			drv[i].playback.rates		= RSND_RATES;
+			drv[i].playback.formats		= RSND_FMTS;
+			drv[i].playback.channels_min	= 2;
+			drv[i].playback.channels_max	= 2;
+		}
+		if (cmod) {
+			drv[i].capture.rates		= RSND_RATES;
+			drv[i].capture.formats		= RSND_FMTS;
+			drv[i].capture.channels_min	= 2;
+			drv[i].capture.channels_max	= 2;
+		}
+
+		dev_dbg(dev, "%s (%s/%s)\n", rdai[i].name,
+			pmod ? "play"    : " -- ",
+			cmod ? "capture" : "  --   ");
+	}
+
+	priv->dai_nr	= dai_nr;
+	priv->daidrv	= drv;
+	priv->rdai	= rdai;
+
+	return 0;
+}
+
+static void rsnd_dai_remove(struct platform_device *pdev,
+			  struct rsnd_priv *priv)
+{
+}
+
+/*
+ *		pcm ops
+ */
+static struct snd_pcm_hardware rsnd_pcm_hardware = {
+	.info =		SNDRV_PCM_INFO_INTERLEAVED	|
+			SNDRV_PCM_INFO_MMAP		|
+			SNDRV_PCM_INFO_MMAP_VALID	|
+			SNDRV_PCM_INFO_PAUSE,
+	.formats		= RSND_FMTS,
+	.rates			= RSND_RATES,
+	.rate_min		= 8000,
+	.rate_max		= 192000,
+	.channels_min		= 2,
+	.channels_max		= 2,
+	.buffer_bytes_max	= 64 * 1024,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= 8192,
+	.periods_min		= 1,
+	.periods_max		= 32,
+	.fifo_size		= 256,
+};
+
+static int rsnd_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	int ret = 0;
+
+	snd_soc_set_runtime_hwparams(substream, &rsnd_pcm_hardware);
+
+	ret = snd_pcm_hw_constraint_integer(runtime,
+					    SNDRV_PCM_HW_PARAM_PERIODS);
+
+	return ret;
+}
+
+static int rsnd_hw_params(struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *hw_params)
+{
+	return snd_pcm_lib_malloc_pages(substream,
+					params_buffer_bytes(hw_params));
+}
+
+static snd_pcm_uframes_t rsnd_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_dai *dai = rsnd_substream_to_dai(substream);
+	struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
+	struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream);
+
+	return bytes_to_frames(runtime, io->byte_pos);
+}
+
+static struct snd_pcm_ops rsnd_pcm_ops = {
+	.open		= rsnd_pcm_open,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= rsnd_hw_params,
+	.hw_free	= snd_pcm_lib_free_pages,
+	.pointer	= rsnd_pointer,
+};
+
+/*
+ *		snd_soc_platform
+ */
+
+#define PREALLOC_BUFFER		(32 * 1024)
+#define PREALLOC_BUFFER_MAX	(32 * 1024)
+
+static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	return snd_pcm_lib_preallocate_pages_for_all(
+		rtd->pcm,
+		SNDRV_DMA_TYPE_DEV,
+		rtd->card->snd_card->dev,
+		PREALLOC_BUFFER, PREALLOC_BUFFER_MAX);
+}
+
+static void rsnd_pcm_free(struct snd_pcm *pcm)
+{
+	snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static struct snd_soc_platform_driver rsnd_soc_platform = {
+	.ops		= &rsnd_pcm_ops,
+	.pcm_new	= rsnd_pcm_new,
+	.pcm_free	= rsnd_pcm_free,
+};
+
+static const struct snd_soc_component_driver rsnd_soc_component = {
+	.name		= "rsnd",
+};
+
+/*
+ *	rsnd probe
+ */
+static int rsnd_probe(struct platform_device *pdev)
+{
+	struct rcar_snd_info *info;
+	struct rsnd_priv *priv;
+	struct device *dev = &pdev->dev;
+	int ret;
+
+	info = pdev->dev.platform_data;
+	if (!info) {
+		dev_err(dev, "driver needs R-Car sound information\n");
+		return -ENODEV;
+	}
+
+	/*
+	 *	init priv data
+	 */
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv) {
+		dev_err(dev, "priv allocate failed\n");
+		return -ENODEV;
+	}
+
+	priv->dev	= dev;
+	priv->info	= info;
+	spin_lock_init(&priv->lock);
+
+	/*
+	 *	init each module
+	 */
+	ret = rsnd_gen_probe(pdev, info, priv);
+	if (ret < 0)
+		return ret;
+
+	ret = rsnd_scu_probe(pdev, info, priv);
+	if (ret < 0)
+		return ret;
+
+	ret = rsnd_adg_probe(pdev, info, priv);
+	if (ret < 0)
+		return ret;
+
+	ret = rsnd_ssi_probe(pdev, info, priv);
+	if (ret < 0)
+		return ret;
+
+	ret = rsnd_dai_probe(pdev, info, priv);
+	if (ret < 0)
+		return ret;
+
+	/*
+	 *	asoc register
+	 */
+	ret = snd_soc_register_platform(dev, &rsnd_soc_platform);
+	if (ret < 0) {
+		dev_err(dev, "cannot snd soc register\n");
+		return ret;
+	}
+
+	ret = snd_soc_register_component(dev, &rsnd_soc_component,
+					 priv->daidrv, rsnd_dai_nr(priv));
+	if (ret < 0) {
+		dev_err(dev, "cannot snd dai register\n");
+		goto exit_snd_soc;
+	}
+
+	dev_set_drvdata(dev, priv);
+
+	pm_runtime_enable(dev);
+
+	dev_info(dev, "probed\n");
+	return ret;
+
+exit_snd_soc:
+	snd_soc_unregister_platform(dev);
+
+	return ret;
+}
+
+static int rsnd_remove(struct platform_device *pdev)
+{
+	struct rsnd_priv *priv = dev_get_drvdata(&pdev->dev);
+
+	pm_runtime_disable(&pdev->dev);
+
+	/*
+	 *	remove each module
+	 */
+	rsnd_ssi_remove(pdev, priv);
+	rsnd_adg_remove(pdev, priv);
+	rsnd_scu_remove(pdev, priv);
+	rsnd_dai_remove(pdev, priv);
+	rsnd_gen_remove(pdev, priv);
+
+	return 0;
+}
+
+static struct platform_driver rsnd_driver = {
+	.driver	= {
+		.name	= "rcar_sound",
+	},
+	.probe		= rsnd_probe,
+	.remove		= rsnd_remove,
+};
+module_platform_driver(rsnd_driver);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Renesas R-Car audio driver");
+MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
+MODULE_ALIAS("platform:rcar-pcm-audio");
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
new file mode 100644
index 000000000000..babb203b43b7
--- /dev/null
+++ b/sound/soc/sh/rcar/gen.c
@@ -0,0 +1,280 @@
+/*
+ * Renesas R-Car Gen1 SRU/SSI support
+ *
+ * Copyright (C) 2013 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#include "rsnd.h"
+
+struct rsnd_gen_ops {
+	int (*path_init)(struct rsnd_priv *priv,
+			 struct rsnd_dai *rdai,
+			 struct rsnd_dai_stream *io);
+	int (*path_exit)(struct rsnd_priv *priv,
+			 struct rsnd_dai *rdai,
+			 struct rsnd_dai_stream *io);
+};
+
+struct rsnd_gen_reg_map {
+	int index;	/* -1 : not supported */
+	u32 offset_id;	/* offset of ssi0, ssi1, ssi2... */
+	u32 offset_adr;	/* offset of SSICR, SSISR, ... */
+};
+
+struct rsnd_gen {
+	void __iomem *base[RSND_BASE_MAX];
+
+	struct rsnd_gen_reg_map reg_map[RSND_REG_MAX];
+	struct rsnd_gen_ops *ops;
+};
+
+#define rsnd_priv_to_gen(p)	((struct rsnd_gen *)(p)->gen)
+
+/*
+ *		Gen2
+ *		will be filled in the future
+ */
+
+/*
+ *		Gen1
+ */
+static int rsnd_gen1_path_init(struct rsnd_priv *priv,
+			       struct rsnd_dai *rdai,
+			       struct rsnd_dai_stream *io)
+{
+	struct rsnd_mod *mod;
+	int ret;
+	int id;
+
+	/*
+	 * Gen1 is created by SRU/SSI, and this SRU is base module of
+	 * Gen2's SCU/SSIU/SSI. (Gen2 SCU/SSIU came from SRU)
+	 *
+	 * Easy image is..
+	 *	Gen1 SRU = Gen2 SCU + SSIU + etc
+	 *
+	 * Gen2 SCU path is very flexible, but, Gen1 SRU (SCU parts) is
+	 * using fixed path.
+	 *
+	 * Then, SSI id = SCU id here
+	 */
+
+	/* get SSI's ID */
+	mod = rsnd_ssi_mod_get_frm_dai(priv,
+				       rsnd_dai_id(priv, rdai),
+				       rsnd_dai_is_play(rdai, io));
+	id = rsnd_mod_id(mod);
+
+	/* SSI */
+	mod = rsnd_ssi_mod_get(priv, id);
+	ret = rsnd_dai_connect(rdai, mod, io);
+	if (ret < 0)
+		return ret;
+
+	/* SCU */
+	mod = rsnd_scu_mod_get(priv, id);
+	ret = rsnd_dai_connect(rdai, mod, io);
+
+	return ret;
+}
+
+static int rsnd_gen1_path_exit(struct rsnd_priv *priv,
+			       struct rsnd_dai *rdai,
+			       struct rsnd_dai_stream *io)
+{
+	struct rsnd_mod *mod, *n;
+	int ret = 0;
+
+	/*
+	 * remove all mod from rdai
+	 */
+	for_each_rsnd_mod(mod, n, io)
+		ret |= rsnd_dai_disconnect(mod);
+
+	return ret;
+}
+
+static struct rsnd_gen_ops rsnd_gen1_ops = {
+	.path_init	= rsnd_gen1_path_init,
+	.path_exit	= rsnd_gen1_path_exit,
+};
+
+#define RSND_GEN1_REG_MAP(g, s, i, oi, oa)				\
+	do {								\
+		(g)->reg_map[RSND_REG_##i].index  = RSND_GEN1_##s;	\
+		(g)->reg_map[RSND_REG_##i].offset_id = oi;		\
+		(g)->reg_map[RSND_REG_##i].offset_adr = oa;		\
+	} while (0)
+
+static void rsnd_gen1_reg_map_init(struct rsnd_gen *gen)
+{
+	RSND_GEN1_REG_MAP(gen, SRU,	SRC_ROUTE_SEL,	0x0,	0x00);
+	RSND_GEN1_REG_MAP(gen, SRU,	SRC_TMG_SEL0,	0x0,	0x08);
+	RSND_GEN1_REG_MAP(gen, SRU,	SRC_TMG_SEL1,	0x0,	0x0c);
+	RSND_GEN1_REG_MAP(gen, SRU,	SRC_TMG_SEL2,	0x0,	0x10);
+	RSND_GEN1_REG_MAP(gen, SRU,	SRC_CTRL,	0x0,	0xc0);
+	RSND_GEN1_REG_MAP(gen, SRU,	SSI_MODE0,	0x0,	0xD0);
+	RSND_GEN1_REG_MAP(gen, SRU,	SSI_MODE1,	0x0,	0xD4);
+	RSND_GEN1_REG_MAP(gen, SRU,	BUSIF_MODE,	0x4,	0x20);
+	RSND_GEN1_REG_MAP(gen, SRU,	BUSIF_ADINR,	0x40,	0x214);
+
+	RSND_GEN1_REG_MAP(gen, ADG,	BRRA,		0x0,	0x00);
+	RSND_GEN1_REG_MAP(gen, ADG,	BRRB,		0x0,	0x04);
+	RSND_GEN1_REG_MAP(gen, ADG,	SSICKR,		0x0,	0x08);
+	RSND_GEN1_REG_MAP(gen, ADG,	AUDIO_CLK_SEL0,	0x0,	0x0c);
+	RSND_GEN1_REG_MAP(gen, ADG,	AUDIO_CLK_SEL1,	0x0,	0x10);
+	RSND_GEN1_REG_MAP(gen, ADG,	AUDIO_CLK_SEL3,	0x0,	0x18);
+	RSND_GEN1_REG_MAP(gen, ADG,	AUDIO_CLK_SEL4,	0x0,	0x1c);
+	RSND_GEN1_REG_MAP(gen, ADG,	AUDIO_CLK_SEL5,	0x0,	0x20);
+
+	RSND_GEN1_REG_MAP(gen, SSI,	SSICR,		0x40,	0x00);
+	RSND_GEN1_REG_MAP(gen, SSI,	SSISR,		0x40,	0x04);
+	RSND_GEN1_REG_MAP(gen, SSI,	SSITDR,		0x40,	0x08);
+	RSND_GEN1_REG_MAP(gen, SSI,	SSIRDR,		0x40,	0x0c);
+	RSND_GEN1_REG_MAP(gen, SSI,	SSIWSR,		0x40,	0x20);
+}
+
+static int rsnd_gen1_probe(struct platform_device *pdev,
+			   struct rcar_snd_info *info,
+			   struct rsnd_priv *priv)
+{
+	struct device *dev = rsnd_priv_to_dev(priv);
+	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
+	struct resource *sru_res;
+	struct resource *adg_res;
+	struct resource *ssi_res;
+
+	/*
+	 * map address
+	 */
+	sru_res	= platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SRU);
+	adg_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_ADG);
+	ssi_res	= platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SSI);
+
+	gen->base[RSND_GEN1_SRU] = devm_ioremap_resource(dev, sru_res);
+	gen->base[RSND_GEN1_ADG] = devm_ioremap_resource(dev, adg_res);
+	gen->base[RSND_GEN1_SSI] = devm_ioremap_resource(dev, ssi_res);
+	if (IS_ERR(gen->base[RSND_GEN1_SRU]) ||
+	    IS_ERR(gen->base[RSND_GEN1_ADG]) ||
+	    IS_ERR(gen->base[RSND_GEN1_SSI]))
+		return -ENODEV;
+
+	gen->ops = &rsnd_gen1_ops;
+	rsnd_gen1_reg_map_init(gen);
+
+	dev_dbg(dev, "Gen1 device probed\n");
+	dev_dbg(dev, "SRU : %08x => %p\n",	sru_res->start,
+						gen->base[RSND_GEN1_SRU]);
+	dev_dbg(dev, "ADG : %08x => %p\n",	adg_res->start,
+						gen->base[RSND_GEN1_ADG]);
+	dev_dbg(dev, "SSI : %08x => %p\n",	ssi_res->start,
+						gen->base[RSND_GEN1_SSI]);
+
+	return 0;
+
+}
+
+static void rsnd_gen1_remove(struct platform_device *pdev,
+			     struct rsnd_priv *priv)
+{
+}
+
+/*
+ *		Gen
+ */
+int rsnd_gen_path_init(struct rsnd_priv *priv,
+		       struct rsnd_dai *rdai,
+		       struct rsnd_dai_stream *io)
+{
+	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
+
+	return gen->ops->path_init(priv, rdai, io);
+}
+
+int rsnd_gen_path_exit(struct rsnd_priv *priv,
+		       struct rsnd_dai *rdai,
+		       struct rsnd_dai_stream *io)
+{
+	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
+
+	return gen->ops->path_exit(priv, rdai, io);
+}
+
+void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv,
+			       struct rsnd_mod *mod,
+			       enum rsnd_reg reg)
+{
+	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
+	struct device *dev = rsnd_priv_to_dev(priv);
+	int index;
+	u32 offset_id, offset_adr;
+
+	if (reg >= RSND_REG_MAX) {
+		dev_err(dev, "rsnd_reg reg error\n");
+		return NULL;
+	}
+
+	index		= gen->reg_map[reg].index;
+	offset_id	= gen->reg_map[reg].offset_id;
+	offset_adr	= gen->reg_map[reg].offset_adr;
+
+	if (index < 0) {
+		dev_err(dev, "unsupported reg access %d\n", reg);
+		return NULL;
+	}
+
+	if (offset_id && mod)
+		offset_id *= rsnd_mod_id(mod);
+
+	/*
+	 * index/offset were set on gen1/gen2
+	 */
+
+	return gen->base[index] + offset_id + offset_adr;
+}
+
+int rsnd_gen_probe(struct platform_device *pdev,
+		   struct rcar_snd_info *info,
+		   struct rsnd_priv *priv)
+{
+	struct device *dev = rsnd_priv_to_dev(priv);
+	struct rsnd_gen *gen;
+	int i;
+
+	gen = devm_kzalloc(dev, sizeof(*gen), GFP_KERNEL);
+	if (!gen) {
+		dev_err(dev, "GEN allocate failed\n");
+		return -ENOMEM;
+	}
+
+	priv->gen = gen;
+
+	/*
+	 * see
+	 *	rsnd_reg_get()
+	 *	rsnd_gen_probe()
+	 */
+	for (i = 0; i < RSND_REG_MAX; i++)
+		gen->reg_map[i].index = -1;
+
+	/*
+	 *	init each module
+	 */
+	if (rsnd_is_gen1(priv))
+		return rsnd_gen1_probe(pdev, info, priv);
+
+	dev_err(dev, "unknown generation R-Car sound device\n");
+
+	return -ENODEV;
+}
+
+void rsnd_gen_remove(struct platform_device *pdev,
+		     struct rsnd_priv *priv)
+{
+	if (rsnd_is_gen1(priv))
+		rsnd_gen1_remove(pdev, priv);
+}
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
new file mode 100644
index 000000000000..9cc6986a8cfb
--- /dev/null
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -0,0 +1,302 @@
+/*
+ * Renesas R-Car
+ *
+ * Copyright (C) 2013 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#ifndef RSND_H
+#define RSND_H
+
+#include <linux/clk.h>
+#include <linux/device.h>
+#include <linux/dma-mapping.h>
+#include <linux/io.h>
+#include <linux/list.h>
+#include <linux/module.h>
+#include <linux/sh_dma.h>
+#include <linux/workqueue.h>
+#include <sound/rcar_snd.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+/*
+ *	pseudo register
+ *
+ * The register address offsets SRU/SCU/SSIU on Gen1/Gen2 are very different.
+ * This driver uses pseudo register in order to hide it.
+ * see gen1/gen2 for detail
+ */
+enum rsnd_reg {
+	/* SRU/SCU */
+	RSND_REG_SRC_ROUTE_SEL,
+	RSND_REG_SRC_TMG_SEL0,
+	RSND_REG_SRC_TMG_SEL1,
+	RSND_REG_SRC_TMG_SEL2,
+	RSND_REG_SRC_CTRL,
+	RSND_REG_SSI_MODE0,
+	RSND_REG_SSI_MODE1,
+	RSND_REG_BUSIF_MODE,
+	RSND_REG_BUSIF_ADINR,
+
+	/* ADG */
+	RSND_REG_BRRA,
+	RSND_REG_BRRB,
+	RSND_REG_SSICKR,
+	RSND_REG_AUDIO_CLK_SEL0,
+	RSND_REG_AUDIO_CLK_SEL1,
+	RSND_REG_AUDIO_CLK_SEL2,
+	RSND_REG_AUDIO_CLK_SEL3,
+	RSND_REG_AUDIO_CLK_SEL4,
+	RSND_REG_AUDIO_CLK_SEL5,
+
+	/* SSI */
+	RSND_REG_SSICR,
+	RSND_REG_SSISR,
+	RSND_REG_SSITDR,
+	RSND_REG_SSIRDR,
+	RSND_REG_SSIWSR,
+
+	RSND_REG_MAX,
+};
+
+struct rsnd_priv;
+struct rsnd_mod;
+struct rsnd_dai;
+struct rsnd_dai_stream;
+
+/*
+ *	R-Car basic functions
+ */
+#define rsnd_mod_read(m, r) \
+	rsnd_read(rsnd_mod_to_priv(m), m, RSND_REG_##r)
+#define rsnd_mod_write(m, r, d) \
+	rsnd_write(rsnd_mod_to_priv(m), m, RSND_REG_##r, d)
+#define rsnd_mod_bset(m, r, s, d) \
+	rsnd_bset(rsnd_mod_to_priv(m), m, RSND_REG_##r, s, d)
+
+#define rsnd_priv_read(p, r)		rsnd_read(p, NULL, RSND_REG_##r)
+#define rsnd_priv_write(p, r, d)	rsnd_write(p, NULL, RSND_REG_##r, d)
+#define rsnd_priv_bset(p, r, s, d)	rsnd_bset(p, NULL, RSND_REG_##r, s, d)
+
+u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg);
+void rsnd_write(struct rsnd_priv *priv, struct rsnd_mod *mod,
+		enum rsnd_reg reg, u32 data);
+void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg,
+		    u32 mask, u32 data);
+
+/*
+ *	R-Car DMA
+ */
+struct rsnd_dma {
+	struct rsnd_priv	*priv;
+	struct sh_dmae_slave	slave;
+	struct work_struct	work;
+	struct dma_chan		*chan;
+	enum dma_data_direction dir;
+	int (*inquiry)(struct rsnd_dma *dma, dma_addr_t *buf, int *len);
+	int (*complete)(struct rsnd_dma *dma);
+
+	int submit_loop;
+};
+
+void rsnd_dma_start(struct rsnd_dma *dma);
+void rsnd_dma_stop(struct rsnd_dma *dma);
+int rsnd_dma_available(struct rsnd_dma *dma);
+int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
+	int is_play, int id,
+	int (*inquiry)(struct rsnd_dma *dma, dma_addr_t *buf, int *len),
+	int (*complete)(struct rsnd_dma *dma));
+void  rsnd_dma_quit(struct rsnd_priv *priv,
+		    struct rsnd_dma *dma);
+
+
+/*
+ *	R-Car sound mod
+ */
+
+struct rsnd_mod_ops {
+	char *name;
+	int (*init)(struct rsnd_mod *mod,
+		    struct rsnd_dai *rdai,
+		    struct rsnd_dai_stream *io);
+	int (*quit)(struct rsnd_mod *mod,
+		    struct rsnd_dai *rdai,
+		    struct rsnd_dai_stream *io);
+	int (*start)(struct rsnd_mod *mod,
+		     struct rsnd_dai *rdai,
+		     struct rsnd_dai_stream *io);
+	int (*stop)(struct rsnd_mod *mod,
+		    struct rsnd_dai *rdai,
+		    struct rsnd_dai_stream *io);
+};
+
+struct rsnd_mod {
+	int id;
+	struct rsnd_priv *priv;
+	struct rsnd_mod_ops *ops;
+	struct list_head list; /* connect to rsnd_dai playback/capture */
+	struct rsnd_dma dma;
+};
+
+#define rsnd_mod_to_priv(mod) ((mod)->priv)
+#define rsnd_mod_to_dma(mod) (&(mod)->dma)
+#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma)
+#define rsnd_mod_id(mod) ((mod)->id)
+#define for_each_rsnd_mod(pos, n, io)	\
+	list_for_each_entry_safe(pos, n, &(io)->head, list)
+#define rsnd_mod_call(mod, func, rdai, io)	\
+	(!(mod) ? -ENODEV :			\
+	 !((mod)->ops->func) ? 0 :		\
+	 (mod)->ops->func(mod, rdai, io))
+
+void rsnd_mod_init(struct rsnd_priv *priv,
+		   struct rsnd_mod *mod,
+		   struct rsnd_mod_ops *ops,
+		   int id);
+char *rsnd_mod_name(struct rsnd_mod *mod);
+
+/*
+ *	R-Car sound DAI
+ */
+#define RSND_DAI_NAME_SIZE	16
+struct rsnd_dai_stream {
+	struct list_head head; /* head of rsnd_mod list */
+	struct snd_pcm_substream *substream;
+	int byte_pos;
+	int period_pos;
+	int byte_per_period;
+	int next_period_byte;
+};
+
+struct rsnd_dai {
+	char name[RSND_DAI_NAME_SIZE];
+	struct rsnd_dai_platform_info *info; /* rcar_snd.h */
+	struct rsnd_dai_stream playback;
+	struct rsnd_dai_stream capture;
+
+	int clk_master:1;
+	int bit_clk_inv:1;
+	int frm_clk_inv:1;
+	int sys_delay:1;
+	int data_alignment:1;
+};
+
+#define rsnd_dai_nr(priv) ((priv)->dai_nr)
+#define for_each_rsnd_dai(rdai, priv, i)		\
+	for (i = 0, (rdai) = rsnd_dai_get(priv, i);	\
+	     i < rsnd_dai_nr(priv);			\
+	     i++, (rdai) = rsnd_dai_get(priv, i))
+
+struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id);
+int rsnd_dai_disconnect(struct rsnd_mod *mod);
+int rsnd_dai_connect(struct rsnd_dai *rdai, struct rsnd_mod *mod,
+		     struct rsnd_dai_stream *io);
+int rsnd_dai_is_play(struct rsnd_dai *rdai, struct rsnd_dai_stream *io);
+int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai);
+#define rsnd_dai_get_platform_info(rdai) ((rdai)->info)
+#define rsnd_io_to_runtime(io) ((io)->substream->runtime)
+
+void rsnd_dai_pointer_update(struct rsnd_dai_stream *io, int cnt);
+int rsnd_dai_pointer_offset(struct rsnd_dai_stream *io, int additional);
+
+/*
+ *	R-Car Gen1/Gen2
+ */
+int rsnd_gen_probe(struct platform_device *pdev,
+		   struct rcar_snd_info *info,
+		   struct rsnd_priv *priv);
+void rsnd_gen_remove(struct platform_device *pdev,
+		     struct rsnd_priv *priv);
+int rsnd_gen_path_init(struct rsnd_priv *priv,
+		       struct rsnd_dai *rdai,
+		       struct rsnd_dai_stream *io);
+int rsnd_gen_path_exit(struct rsnd_priv *priv,
+		       struct rsnd_dai *rdai,
+		       struct rsnd_dai_stream *io);
+void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv,
+			       struct rsnd_mod *mod,
+			       enum rsnd_reg reg);
+#define rsnd_is_gen1(s)		((s)->info->flags & RSND_GEN1)
+#define rsnd_is_gen2(s)		((s)->info->flags & RSND_GEN2)
+
+/*
+ *	R-Car ADG
+ */
+int rsnd_adg_ssi_clk_stop(struct rsnd_mod *mod);
+int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate);
+int rsnd_adg_probe(struct platform_device *pdev,
+		   struct rcar_snd_info *info,
+		   struct rsnd_priv *priv);
+void rsnd_adg_remove(struct platform_device *pdev,
+		   struct rsnd_priv *priv);
+
+/*
+ *	R-Car sound priv
+ */
+struct rsnd_priv {
+
+	struct device *dev;
+	struct rcar_snd_info *info;
+	spinlock_t lock;
+
+	/*
+	 * below value will be filled on rsnd_gen_probe()
+	 */
+	void *gen;
+
+	/*
+	 * below value will be filled on rsnd_scu_probe()
+	 */
+	void *scu;
+	int scu_nr;
+
+	/*
+	 * below value will be filled on rsnd_adg_probe()
+	 */
+	void *adg;
+
+	/*
+	 * below value will be filled on rsnd_ssi_probe()
+	 */
+	void *ssiu;
+
+	/*
+	 * below value will be filled on rsnd_dai_probe()
+	 */
+	struct snd_soc_dai_driver *daidrv;
+	struct rsnd_dai *rdai;
+	int dai_nr;
+};
+
+#define rsnd_priv_to_dev(priv)	((priv)->dev)
+#define rsnd_lock(priv, flags) spin_lock_irqsave(&priv->lock, flags)
+#define rsnd_unlock(priv, flags) spin_unlock_irqrestore(&priv->lock, flags)
+
+/*
+ *	R-Car SCU
+ */
+int rsnd_scu_probe(struct platform_device *pdev,
+		   struct rcar_snd_info *info,
+		   struct rsnd_priv *priv);
+void rsnd_scu_remove(struct platform_device *pdev,
+		     struct rsnd_priv *priv);
+struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id);
+#define rsnd_scu_nr(priv) ((priv)->scu_nr)
+
+/*
+ *	R-Car SSI
+ */
+int rsnd_ssi_probe(struct platform_device *pdev,
+		   struct rcar_snd_info *info,
+		   struct rsnd_priv *priv);
+void rsnd_ssi_remove(struct platform_device *pdev,
+		   struct rsnd_priv *priv);
+struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id);
+struct rsnd_mod *rsnd_ssi_mod_get_frm_dai(struct rsnd_priv *priv,
+					  int dai_id, int is_play);
+
+#endif
diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c
new file mode 100644
index 000000000000..184d9008cecd
--- /dev/null
+++ b/sound/soc/sh/rcar/scu.c
@@ -0,0 +1,236 @@
+/*
+ * Renesas R-Car SCU support
+ *
+ * Copyright (C) 2013 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#include "rsnd.h"
+
+struct rsnd_scu {
+	struct rsnd_scu_platform_info *info; /* rcar_snd.h */
+	struct rsnd_mod mod;
+};
+
+#define rsnd_scu_mode_flags(p) ((p)->info->flags)
+
+/*
+ * ADINR
+ */
+#define OTBL_24		(0 << 16)
+#define OTBL_22		(2 << 16)
+#define OTBL_20		(4 << 16)
+#define OTBL_18		(6 << 16)
+#define OTBL_16		(8 << 16)
+
+
+#define rsnd_mod_to_scu(_mod)	\
+	container_of((_mod), struct rsnd_scu, mod)
+
+#define for_each_rsnd_scu(pos, priv, i)					\
+	for ((i) = 0;							\
+	     ((i) < rsnd_scu_nr(priv)) &&				\
+		     ((pos) = (struct rsnd_scu *)(priv)->scu + i);	\
+	     i++)
+
+static int rsnd_scu_set_route(struct rsnd_priv *priv,
+			      struct rsnd_mod *mod,
+			      struct rsnd_dai *rdai,
+			      struct rsnd_dai_stream *io)
+{
+	struct scu_route_config {
+		u32 mask;
+		int shift;
+	} routes[] = {
+		{ 0xF,  0, }, /* 0 */
+		{ 0xF,  4, }, /* 1 */
+		{ 0xF,  8, }, /* 2 */
+		{ 0x7, 12, }, /* 3 */
+		{ 0x7, 16, }, /* 4 */
+		{ 0x7, 20, }, /* 5 */
+		{ 0x7, 24, }, /* 6 */
+		{ 0x3, 28, }, /* 7 */
+		{ 0x3, 30, }, /* 8 */
+	};
+
+	u32 mask;
+	u32 val;
+	int shift;
+	int id;
+
+	/*
+	 * Gen1 only
+	 */
+	if (!rsnd_is_gen1(priv))
+		return 0;
+
+	id = rsnd_mod_id(mod);
+	if (id < 0 || id > ARRAY_SIZE(routes))
+		return -EIO;
+
+	/*
+	 * SRC_ROUTE_SELECT
+	 */
+	val = rsnd_dai_is_play(rdai, io) ? 0x1 : 0x2;
+	val = val		<< routes[id].shift;
+	mask = routes[id].mask	<< routes[id].shift;
+
+	rsnd_mod_bset(mod, SRC_ROUTE_SEL, mask, val);
+
+	/*
+	 * SRC_TIMING_SELECT
+	 */
+	shift	= (id % 4) * 8;
+	mask	= 0x1F << shift;
+	if (8 == id) /* SRU8 is very special */
+		val = id << shift;
+	else
+		val = (id + 1) << shift;
+
+	switch (id / 4) {
+	case 0:
+		rsnd_mod_bset(mod, SRC_TMG_SEL0, mask, val);
+		break;
+	case 1:
+		rsnd_mod_bset(mod, SRC_TMG_SEL1, mask, val);
+		break;
+	case 2:
+		rsnd_mod_bset(mod, SRC_TMG_SEL2, mask, val);
+		break;
+	}
+
+	return 0;
+}
+
+static int rsnd_scu_set_mode(struct rsnd_priv *priv,
+			     struct rsnd_mod *mod,
+			     struct rsnd_dai *rdai,
+			     struct rsnd_dai_stream *io)
+{
+	int id = rsnd_mod_id(mod);
+	u32 val;
+
+	if (rsnd_is_gen1(priv)) {
+		val = (1 << id);
+		rsnd_mod_bset(mod, SRC_CTRL, val, val);
+	}
+
+	return 0;
+}
+
+static int rsnd_scu_set_hpbif(struct rsnd_priv *priv,
+			      struct rsnd_mod *mod,
+			      struct rsnd_dai *rdai,
+			      struct rsnd_dai_stream *io)
+{
+	struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+	u32 adinr = runtime->channels;
+
+	switch (runtime->sample_bits) {
+	case 16:
+		adinr |= OTBL_16;
+		break;
+	case 32:
+		adinr |= OTBL_24;
+		break;
+	default:
+		return -EIO;
+	}
+
+	rsnd_mod_write(mod, BUSIF_MODE, 1);
+	rsnd_mod_write(mod, BUSIF_ADINR, adinr);
+
+	return 0;
+}
+
+static int rsnd_scu_start(struct rsnd_mod *mod,
+			  struct rsnd_dai *rdai,
+			  struct rsnd_dai_stream *io)
+{
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
+	struct device *dev = rsnd_priv_to_dev(priv);
+	u32 flags = rsnd_scu_mode_flags(scu);
+	int ret;
+
+	/*
+	 * SCU will be used if it has RSND_SCU_USB_HPBIF flags
+	 */
+	if (!(flags & RSND_SCU_USB_HPBIF)) {
+		/* it use PIO transter */
+		dev_dbg(dev, "%s%d is not used\n",
+			rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+		return 0;
+	}
+
+	/* it use DMA transter */
+	ret = rsnd_scu_set_route(priv, mod, rdai, io);
+	if (ret < 0)
+		return ret;
+
+	ret = rsnd_scu_set_mode(priv, mod, rdai, io);
+	if (ret < 0)
+		return ret;
+
+	ret = rsnd_scu_set_hpbif(priv, mod, rdai, io);
+	if (ret < 0)
+		return ret;
+
+	dev_dbg(dev, "%s%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+	return 0;
+}
+
+static struct rsnd_mod_ops rsnd_scu_ops = {
+	.name	= "scu",
+	.start	= rsnd_scu_start,
+};
+
+struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id)
+{
+	BUG_ON(id < 0 || id >= rsnd_scu_nr(priv));
+
+	return &((struct rsnd_scu *)(priv->scu) + id)->mod;
+}
+
+int rsnd_scu_probe(struct platform_device *pdev,
+		   struct rcar_snd_info *info,
+		   struct rsnd_priv *priv)
+{
+	struct device *dev = rsnd_priv_to_dev(priv);
+	struct rsnd_scu *scu;
+	int i, nr;
+
+	/*
+	 * init SCU
+	 */
+	nr	= info->scu_info_nr;
+	scu	= devm_kzalloc(dev, sizeof(*scu) * nr, GFP_KERNEL);
+	if (!scu) {
+		dev_err(dev, "SCU allocate failed\n");
+		return -ENOMEM;
+	}
+
+	priv->scu_nr	= nr;
+	priv->scu	= scu;
+
+	for_each_rsnd_scu(scu, priv, i) {
+		rsnd_mod_init(priv, &scu->mod,
+			      &rsnd_scu_ops, i);
+		scu->info = &info->scu_info[i];
+
+		dev_dbg(dev, "SCU%d probed\n", i);
+	}
+	dev_dbg(dev, "scu probed\n");
+
+	return 0;
+}
+
+void rsnd_scu_remove(struct platform_device *pdev,
+		     struct rsnd_priv *priv)
+{
+}
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
new file mode 100644
index 000000000000..fae26d3f79d2
--- /dev/null
+++ b/sound/soc/sh/rcar/ssi.c
@@ -0,0 +1,728 @@
+/*
+ * Renesas R-Car SSIU/SSI support
+ *
+ * Copyright (C) 2013 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * Based on fsi.c
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#include <linux/delay.h>
+#include "rsnd.h"
+#define RSND_SSI_NAME_SIZE 16
+
+/*
+ * SSICR
+ */
+#define	FORCE		(1 << 31)	/* Fixed */
+#define	DMEN		(1 << 28)	/* DMA Enable */
+#define	UIEN		(1 << 27)	/* Underflow Interrupt Enable */
+#define	OIEN		(1 << 26)	/* Overflow Interrupt Enable */
+#define	IIEN		(1 << 25)	/* Idle Mode Interrupt Enable */
+#define	DIEN		(1 << 24)	/* Data Interrupt Enable */
+
+#define	DWL_8		(0 << 19)	/* Data Word Length */
+#define	DWL_16		(1 << 19)	/* Data Word Length */
+#define	DWL_18		(2 << 19)	/* Data Word Length */
+#define	DWL_20		(3 << 19)	/* Data Word Length */
+#define	DWL_22		(4 << 19)	/* Data Word Length */
+#define	DWL_24		(5 << 19)	/* Data Word Length */
+#define	DWL_32		(6 << 19)	/* Data Word Length */
+
+#define	SWL_32		(3 << 16)	/* R/W System Word Length */
+#define	SCKD		(1 << 15)	/* Serial Bit Clock Direction */
+#define	SWSD		(1 << 14)	/* Serial WS Direction */
+#define	SCKP		(1 << 13)	/* Serial Bit Clock Polarity */
+#define	SWSP		(1 << 12)	/* Serial WS Polarity */
+#define	SDTA		(1 << 10)	/* Serial Data Alignment */
+#define	DEL		(1 <<  8)	/* Serial Data Delay */
+#define	CKDV(v)		(v <<  4)	/* Serial Clock Division Ratio */
+#define	TRMD		(1 <<  1)	/* Transmit/Receive Mode Select */
+#define	EN		(1 <<  0)	/* SSI Module Enable */
+
+/*
+ * SSISR
+ */
+#define	UIRQ		(1 << 27)	/* Underflow Error Interrupt Status */
+#define	OIRQ		(1 << 26)	/* Overflow Error Interrupt Status */
+#define	IIRQ		(1 << 25)	/* Idle Mode Interrupt Status */
+#define	DIRQ		(1 << 24)	/* Data Interrupt Status Flag */
+
+/*
+ * SSIWSR
+ */
+#define CONT		(1 << 8)	/* WS Continue Function */
+
+struct rsnd_ssi {
+	struct clk *clk;
+	struct rsnd_ssi_platform_info *info; /* rcar_snd.h */
+	struct rsnd_ssi *parent;
+	struct rsnd_mod mod;
+
+	struct rsnd_dai *rdai;
+	struct rsnd_dai_stream *io;
+	u32 cr_own;
+	u32 cr_clk;
+	u32 cr_etc;
+	int err;
+	int dma_offset;
+	unsigned int usrcnt;
+	unsigned int rate;
+};
+
+struct rsnd_ssiu {
+	u32 ssi_mode0;
+	u32 ssi_mode1;
+
+	int ssi_nr;
+	struct rsnd_ssi *ssi;
+};
+
+#define for_each_rsnd_ssi(pos, priv, i)					\
+	for (i = 0;							\
+	     (i < rsnd_ssi_nr(priv)) &&					\
+		((pos) = ((struct rsnd_ssiu *)((priv)->ssiu))->ssi + i); \
+	     i++)
+
+#define rsnd_ssi_nr(priv) (((struct rsnd_ssiu *)((priv)->ssiu))->ssi_nr)
+#define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod)
+#define rsnd_dma_to_ssi(dma)  rsnd_mod_to_ssi(rsnd_dma_to_mod(dma))
+#define rsnd_ssi_pio_available(ssi) ((ssi)->info->pio_irq > 0)
+#define rsnd_ssi_dma_available(ssi) \
+	rsnd_dma_available(rsnd_mod_to_dma(&(ssi)->mod))
+#define rsnd_ssi_clk_from_parent(ssi) ((ssi)->parent)
+#define rsnd_rdai_is_clk_master(rdai) ((rdai)->clk_master)
+#define rsnd_ssi_mode_flags(p) ((p)->info->flags)
+#define rsnd_ssi_dai_id(ssi) ((ssi)->info->dai_id)
+#define rsnd_ssi_to_ssiu(ssi)\
+	(((struct rsnd_ssiu *)((ssi) - rsnd_mod_id(&(ssi)->mod))) - 1)
+
+static void rsnd_ssi_mode_init(struct rsnd_priv *priv,
+			       struct rsnd_ssiu *ssiu)
+{
+	struct device *dev = rsnd_priv_to_dev(priv);
+	struct rsnd_ssi *ssi;
+	u32 flags;
+	u32 val;
+	int i;
+
+	/*
+	 * SSI_MODE0
+	 */
+	ssiu->ssi_mode0 = 0;
+	for_each_rsnd_ssi(ssi, priv, i) {
+		flags = rsnd_ssi_mode_flags(ssi);
+
+		/* see also BUSIF_MODE */
+		if (!(flags & RSND_SSI_DEPENDENT)) {
+			ssiu->ssi_mode0 |= (1 << i);
+			dev_dbg(dev, "SSI%d uses INDEPENDENT mode\n", i);
+		} else {
+			dev_dbg(dev, "SSI%d uses DEPENDENT mode\n", i);
+		}
+	}
+
+	/*
+	 * SSI_MODE1
+	 */
+#define ssi_parent_set(p, sync, adg, ext)		\
+	do {						\
+		ssi->parent = ssiu->ssi + p;		\
+		if (flags & RSND_SSI_CLK_FROM_ADG)	\
+			val = adg;			\
+		else					\
+			val = ext;			\
+		if (flags & RSND_SSI_SYNC)		\
+			val |= sync;			\
+	} while (0)
+
+	ssiu->ssi_mode1 = 0;
+	for_each_rsnd_ssi(ssi, priv, i) {
+		flags = rsnd_ssi_mode_flags(ssi);
+
+		if (!(flags & RSND_SSI_CLK_PIN_SHARE))
+			continue;
+
+		val = 0;
+		switch (i) {
+		case 1:
+			ssi_parent_set(0, (1 << 4), (0x2 << 0), (0x1 << 0));
+			break;
+		case 2:
+			ssi_parent_set(0, (1 << 4), (0x2 << 2), (0x1 << 2));
+			break;
+		case 4:
+			ssi_parent_set(3, (1 << 20), (0x2 << 16), (0x1 << 16));
+			break;
+		case 8:
+			ssi_parent_set(7, 0, 0, 0);
+			break;
+		}
+
+		ssiu->ssi_mode1 |= val;
+	}
+}
+
+static void rsnd_ssi_mode_set(struct rsnd_ssi *ssi)
+{
+	struct rsnd_ssiu *ssiu = rsnd_ssi_to_ssiu(ssi);
+
+	rsnd_mod_write(&ssi->mod, SSI_MODE0, ssiu->ssi_mode0);
+	rsnd_mod_write(&ssi->mod, SSI_MODE1, ssiu->ssi_mode1);
+}
+
+static void rsnd_ssi_status_check(struct rsnd_mod *mod,
+				  u32 bit)
+{
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct device *dev = rsnd_priv_to_dev(priv);
+	u32 status;
+	int i;
+
+	for (i = 0; i < 1024; i++) {
+		status = rsnd_mod_read(mod, SSISR);
+		if (status & bit)
+			return;
+
+		udelay(50);
+	}
+
+	dev_warn(dev, "status check failed\n");
+}
+
+static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
+				     unsigned int rate)
+{
+	struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod);
+	struct device *dev = rsnd_priv_to_dev(priv);
+	int i, j, ret;
+	int adg_clk_div_table[] = {
+		1, 6, /* see adg.c */
+	};
+	int ssi_clk_mul_table[] = {
+		1, 2, 4, 8, 16, 6, 12,
+	};
+	unsigned int main_rate;
+
+	/*
+	 * Find best clock, and try to start ADG
+	 */
+	for (i = 0; i < ARRAY_SIZE(adg_clk_div_table); i++) {
+		for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) {
+
+			/*
+			 * this driver is assuming that
+			 * system word is 64fs (= 2 x 32bit)
+			 * see rsnd_ssi_start()
+			 */
+			main_rate = rate / adg_clk_div_table[i]
+				* 32 * 2 * ssi_clk_mul_table[j];
+
+			ret = rsnd_adg_ssi_clk_try_start(&ssi->mod, main_rate);
+			if (0 == ret) {
+				ssi->rate	= rate;
+				ssi->cr_clk	= FORCE | SWL_32 |
+						  SCKD | SWSD | CKDV(j);
+
+				dev_dbg(dev, "ssi%d outputs %u Hz\n",
+					rsnd_mod_id(&ssi->mod), rate);
+
+				return 0;
+			}
+		}
+	}
+
+	dev_err(dev, "unsupported clock rate\n");
+	return -EIO;
+}
+
+static void rsnd_ssi_master_clk_stop(struct rsnd_ssi *ssi)
+{
+	ssi->rate = 0;
+	ssi->cr_clk = 0;
+	rsnd_adg_ssi_clk_stop(&ssi->mod);
+}
+
+static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
+			      struct rsnd_dai *rdai,
+			      struct rsnd_dai_stream *io)
+{
+	struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod);
+	struct device *dev = rsnd_priv_to_dev(priv);
+	u32 cr;
+
+	if (0 == ssi->usrcnt) {
+		clk_enable(ssi->clk);
+
+		if (rsnd_rdai_is_clk_master(rdai)) {
+			struct snd_pcm_runtime *runtime;
+
+			runtime = rsnd_io_to_runtime(io);
+
+			if (rsnd_ssi_clk_from_parent(ssi))
+				rsnd_ssi_hw_start(ssi->parent, rdai, io);
+			else
+				rsnd_ssi_master_clk_start(ssi, runtime->rate);
+		}
+	}
+
+	cr  =	ssi->cr_own	|
+		ssi->cr_clk	|
+		ssi->cr_etc	|
+		EN;
+
+	rsnd_mod_write(&ssi->mod, SSICR, cr);
+
+	ssi->usrcnt++;
+
+	dev_dbg(dev, "ssi%d hw started\n", rsnd_mod_id(&ssi->mod));
+}
+
+static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi,
+			     struct rsnd_dai *rdai)
+{
+	struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod);
+	struct device *dev = rsnd_priv_to_dev(priv);
+	u32 cr;
+
+	if (0 == ssi->usrcnt) /* stop might be called without start */
+		return;
+
+	ssi->usrcnt--;
+
+	if (0 == ssi->usrcnt) {
+		/*
+		 * disable all IRQ,
+		 * and, wait all data was sent
+		 */
+		cr  =	ssi->cr_own	|
+			ssi->cr_clk;
+
+		rsnd_mod_write(&ssi->mod, SSICR, cr | EN);
+		rsnd_ssi_status_check(&ssi->mod, DIRQ);
+
+		/*
+		 * disable SSI,
+		 * and, wait idle state
+		 */
+		rsnd_mod_write(&ssi->mod, SSICR, cr);	/* disabled all */
+		rsnd_ssi_status_check(&ssi->mod, IIRQ);
+
+		if (rsnd_rdai_is_clk_master(rdai)) {
+			if (rsnd_ssi_clk_from_parent(ssi))
+				rsnd_ssi_hw_stop(ssi->parent, rdai);
+			else
+				rsnd_ssi_master_clk_stop(ssi);
+		}
+
+		clk_disable(ssi->clk);
+	}
+
+	dev_dbg(dev, "ssi%d hw stopped\n", rsnd_mod_id(&ssi->mod));
+}
+
+/*
+ *	SSI mod common functions
+ */
+static int rsnd_ssi_init(struct rsnd_mod *mod,
+			 struct rsnd_dai *rdai,
+			 struct rsnd_dai_stream *io)
+{
+	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct device *dev = rsnd_priv_to_dev(priv);
+	struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+	u32 cr;
+
+	cr = FORCE;
+
+	/*
+	 * always use 32bit system word for easy clock calculation.
+	 * see also rsnd_ssi_master_clk_enable()
+	 */
+	cr |= SWL_32;
+
+	/*
+	 * init clock settings for SSICR
+	 */
+	switch (runtime->sample_bits) {
+	case 16:
+		cr |= DWL_16;
+		break;
+	case 32:
+		cr |= DWL_24;
+		break;
+	default:
+		return -EIO;
+	}
+
+	if (rdai->bit_clk_inv)
+		cr |= SCKP;
+	if (rdai->frm_clk_inv)
+		cr |= SWSP;
+	if (rdai->data_alignment)
+		cr |= SDTA;
+	if (rdai->sys_delay)
+		cr |= DEL;
+	if (rsnd_dai_is_play(rdai, io))
+		cr |= TRMD;
+
+	/*
+	 * set ssi parameter
+	 */
+	ssi->rdai	= rdai;
+	ssi->io		= io;
+	ssi->cr_own	= cr;
+	ssi->err	= -1; /* ignore 1st error */
+
+	rsnd_ssi_mode_set(ssi);
+
+	dev_dbg(dev, "%s.%d init\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+	return 0;
+}
+
+static int rsnd_ssi_quit(struct rsnd_mod *mod,
+			 struct rsnd_dai *rdai,
+			 struct rsnd_dai_stream *io)
+{
+	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct device *dev = rsnd_priv_to_dev(priv);
+
+	dev_dbg(dev, "%s.%d quit\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+	if (ssi->err > 0)
+		dev_warn(dev, "ssi under/over flow err = %d\n", ssi->err);
+
+	ssi->rdai	= NULL;
+	ssi->io		= NULL;
+	ssi->cr_own	= 0;
+	ssi->err	= 0;
+
+	return 0;
+}
+
+static void rsnd_ssi_record_error(struct rsnd_ssi *ssi, u32 status)
+{
+	/* under/over flow error */
+	if (status & (UIRQ | OIRQ)) {
+		ssi->err++;
+
+		/* clear error status */
+		rsnd_mod_write(&ssi->mod, SSISR, 0);
+	}
+}
+
+/*
+ *		SSI PIO
+ */
+static irqreturn_t rsnd_ssi_pio_interrupt(int irq, void *data)
+{
+	struct rsnd_ssi *ssi = data;
+	struct rsnd_dai_stream *io = ssi->io;
+	u32 status = rsnd_mod_read(&ssi->mod, SSISR);
+	irqreturn_t ret = IRQ_NONE;
+
+	if (io && (status & DIRQ)) {
+		struct rsnd_dai *rdai = ssi->rdai;
+		struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+		u32 *buf = (u32 *)(runtime->dma_area +
+				   rsnd_dai_pointer_offset(io, 0));
+
+		rsnd_ssi_record_error(ssi, status);
+
+		/*
+		 * 8/16/32 data can be assesse to TDR/RDR register
+		 * directly as 32bit data
+		 * see rsnd_ssi_init()
+		 */
+		if (rsnd_dai_is_play(rdai, io))
+			rsnd_mod_write(&ssi->mod, SSITDR, *buf);
+		else
+			*buf = rsnd_mod_read(&ssi->mod, SSIRDR);
+
+		rsnd_dai_pointer_update(io, sizeof(*buf));
+
+		ret = IRQ_HANDLED;
+	}
+
+	return ret;
+}
+
+static int rsnd_ssi_pio_start(struct rsnd_mod *mod,
+			      struct rsnd_dai *rdai,
+			      struct rsnd_dai_stream *io)
+{
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+	struct device *dev = rsnd_priv_to_dev(priv);
+
+	/* enable PIO IRQ */
+	ssi->cr_etc = UIEN | OIEN | DIEN;
+
+	rsnd_ssi_hw_start(ssi, rdai, io);
+
+	dev_dbg(dev, "%s.%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+	return 0;
+}
+
+static int rsnd_ssi_pio_stop(struct rsnd_mod *mod,
+			     struct rsnd_dai *rdai,
+			     struct rsnd_dai_stream *io)
+{
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct device *dev = rsnd_priv_to_dev(priv);
+	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+
+	dev_dbg(dev, "%s.%d stop\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+	ssi->cr_etc = 0;
+
+	rsnd_ssi_hw_stop(ssi, rdai);
+
+	return 0;
+}
+
+static struct rsnd_mod_ops rsnd_ssi_pio_ops = {
+	.name	= "ssi (pio)",
+	.init	= rsnd_ssi_init,
+	.quit	= rsnd_ssi_quit,
+	.start	= rsnd_ssi_pio_start,
+	.stop	= rsnd_ssi_pio_stop,
+};
+
+static int rsnd_ssi_dma_inquiry(struct rsnd_dma *dma, dma_addr_t *buf, int *len)
+{
+	struct rsnd_ssi *ssi = rsnd_dma_to_ssi(dma);
+	struct rsnd_dai_stream *io = ssi->io;
+	struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+
+	*len = io->byte_per_period;
+	*buf = runtime->dma_addr +
+		rsnd_dai_pointer_offset(io, ssi->dma_offset + *len);
+	ssi->dma_offset = *len; /* it cares A/B plane */
+
+	return 0;
+}
+
+static int rsnd_ssi_dma_complete(struct rsnd_dma *dma)
+{
+	struct rsnd_ssi *ssi = rsnd_dma_to_ssi(dma);
+	struct rsnd_dai_stream *io = ssi->io;
+	u32 status = rsnd_mod_read(&ssi->mod, SSISR);
+
+	rsnd_ssi_record_error(ssi, status);
+
+	rsnd_dai_pointer_update(ssi->io, io->byte_per_period);
+
+	return 0;
+}
+
+static int rsnd_ssi_dma_start(struct rsnd_mod *mod,
+			      struct rsnd_dai *rdai,
+			      struct rsnd_dai_stream *io)
+{
+	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+	struct rsnd_dma *dma = rsnd_mod_to_dma(&ssi->mod);
+
+	/* enable DMA transfer */
+	ssi->cr_etc = DMEN;
+	ssi->dma_offset = 0;
+
+	rsnd_dma_start(dma);
+
+	rsnd_ssi_hw_start(ssi, ssi->rdai, io);
+
+	/* enable WS continue */
+	if (rsnd_rdai_is_clk_master(rdai))
+		rsnd_mod_write(&ssi->mod, SSIWSR, CONT);
+
+	return 0;
+}
+
+static int rsnd_ssi_dma_stop(struct rsnd_mod *mod,
+			     struct rsnd_dai *rdai,
+			     struct rsnd_dai_stream *io)
+{
+	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+	struct rsnd_dma *dma = rsnd_mod_to_dma(&ssi->mod);
+
+	ssi->cr_etc = 0;
+
+	rsnd_ssi_hw_stop(ssi, rdai);
+
+	rsnd_dma_stop(dma);
+
+	return 0;
+}
+
+static struct rsnd_mod_ops rsnd_ssi_dma_ops = {
+	.name	= "ssi (dma)",
+	.init	= rsnd_ssi_init,
+	.quit	= rsnd_ssi_quit,
+	.start	= rsnd_ssi_dma_start,
+	.stop	= rsnd_ssi_dma_stop,
+};
+
+/*
+ *		Non SSI
+ */
+static int rsnd_ssi_non(struct rsnd_mod *mod,
+			struct rsnd_dai *rdai,
+			struct rsnd_dai_stream *io)
+{
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct device *dev = rsnd_priv_to_dev(priv);
+
+	dev_dbg(dev, "%s\n", __func__);
+
+	return 0;
+}
+
+static struct rsnd_mod_ops rsnd_ssi_non_ops = {
+	.name	= "ssi (non)",
+	.init	= rsnd_ssi_non,
+	.quit	= rsnd_ssi_non,
+	.start	= rsnd_ssi_non,
+	.stop	= rsnd_ssi_non,
+};
+
+/*
+ *		ssi mod function
+ */
+struct rsnd_mod *rsnd_ssi_mod_get_frm_dai(struct rsnd_priv *priv,
+					  int dai_id, int is_play)
+{
+	struct rsnd_ssi *ssi;
+	int i, has_play;
+
+	is_play = !!is_play;
+
+	for_each_rsnd_ssi(ssi, priv, i) {
+		if (rsnd_ssi_dai_id(ssi) != dai_id)
+			continue;
+
+		has_play = !!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_PLAY);
+
+		if (is_play == has_play)
+			return &ssi->mod;
+	}
+
+	return NULL;
+}
+
+struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id)
+{
+	BUG_ON(id < 0 || id >= rsnd_ssi_nr(priv));
+
+	return &(((struct rsnd_ssiu *)(priv->ssiu))->ssi + id)->mod;
+}
+
+int rsnd_ssi_probe(struct platform_device *pdev,
+		   struct rcar_snd_info *info,
+		   struct rsnd_priv *priv)
+{
+	struct rsnd_ssi_platform_info *pinfo;
+	struct device *dev = rsnd_priv_to_dev(priv);
+	struct rsnd_mod_ops *ops;
+	struct clk *clk;
+	struct rsnd_ssiu *ssiu;
+	struct rsnd_ssi *ssi;
+	char name[RSND_SSI_NAME_SIZE];
+	int i, nr, ret;
+
+	/*
+	 *	init SSI
+	 */
+	nr	= info->ssi_info_nr;
+	ssiu	= devm_kzalloc(dev, sizeof(*ssiu) + (sizeof(*ssi) * nr),
+			       GFP_KERNEL);
+	if (!ssiu) {
+		dev_err(dev, "SSI allocate failed\n");
+		return -ENOMEM;
+	}
+
+	priv->ssiu	= ssiu;
+	ssiu->ssi	= (struct rsnd_ssi *)(ssiu + 1);
+	ssiu->ssi_nr	= nr;
+
+	for_each_rsnd_ssi(ssi, priv, i) {
+		pinfo = &info->ssi_info[i];
+
+		snprintf(name, RSND_SSI_NAME_SIZE, "ssi.%d", i);
+
+		clk = clk_get(dev, name);
+		if (IS_ERR(clk))
+			return PTR_ERR(clk);
+
+		ssi->info	= pinfo;
+		ssi->clk	= clk;
+
+		ops = &rsnd_ssi_non_ops;
+
+		/*
+		 * SSI DMA case
+		 */
+		if (pinfo->dma_id > 0) {
+			ret = rsnd_dma_init(
+				priv, rsnd_mod_to_dma(&ssi->mod),
+				(rsnd_ssi_mode_flags(ssi) & RSND_SSI_PLAY),
+				pinfo->dma_id,
+				rsnd_ssi_dma_inquiry,
+				rsnd_ssi_dma_complete);
+			if (ret < 0)
+				dev_info(dev, "SSI DMA failed. try PIO transter\n");
+			else
+				ops	= &rsnd_ssi_dma_ops;
+
+			dev_dbg(dev, "SSI%d use DMA transfer\n", i);
+		}
+
+		/*
+		 * SSI PIO case
+		 */
+		if (!rsnd_ssi_dma_available(ssi) &&
+		     rsnd_ssi_pio_available(ssi)) {
+			ret = devm_request_irq(dev, pinfo->pio_irq,
+					       &rsnd_ssi_pio_interrupt,
+					       IRQF_SHARED,
+					       dev_name(dev), ssi);
+			if (ret) {
+				dev_err(dev, "SSI request interrupt failed\n");
+				return ret;
+			}
+
+			ops	= &rsnd_ssi_pio_ops;
+
+			dev_dbg(dev, "SSI%d use PIO transfer\n", i);
+		}
+
+		rsnd_mod_init(priv, &ssi->mod, ops, i);
+	}
+
+	rsnd_ssi_mode_init(priv, ssiu);
+
+	dev_dbg(dev, "ssi probed\n");
+
+	return 0;
+}
+
+void rsnd_ssi_remove(struct platform_device *pdev,
+		   struct rsnd_priv *priv)
+{
+	struct rsnd_ssi *ssi;
+	int i;
+
+	for_each_rsnd_ssi(ssi, priv, i) {
+		clk_put(ssi->clk);
+		if (rsnd_ssi_dma_available(ssi))
+			rsnd_dma_quit(priv, rsnd_mod_to_dma(&ssi->mod));
+	}
+
+}
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 06a8000aa07b..53c9ecdd119f 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -149,8 +149,9 @@ static int soc_compr_free(struct snd_compr_stream *cstream)
 					SND_SOC_DAPM_STREAM_STOP);
 		} else {
 			rtd->pop_wait = 1;
-			schedule_delayed_work(&rtd->delayed_work,
-				msecs_to_jiffies(rtd->pmdown_time));
+			queue_delayed_work(system_power_efficient_wq,
+					   &rtd->delayed_work,
+					   msecs_to_jiffies(rtd->pmdown_time));
 		}
 	} else {
 		/* capture streams can be powered down now */
@@ -334,7 +335,7 @@ static int soc_compr_copy(struct snd_compr_stream *cstream,
 	return ret;
 }
 
-static int sst_compr_set_metadata(struct snd_compr_stream *cstream,
+static int soc_compr_set_metadata(struct snd_compr_stream *cstream,
 				struct snd_compr_metadata *metadata)
 {
 	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
@@ -347,7 +348,7 @@ static int sst_compr_set_metadata(struct snd_compr_stream *cstream,
 	return ret;
 }
 
-static int sst_compr_get_metadata(struct snd_compr_stream *cstream,
+static int soc_compr_get_metadata(struct snd_compr_stream *cstream,
 				struct snd_compr_metadata *metadata)
 {
 	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
@@ -364,8 +365,8 @@ static struct snd_compr_ops soc_compr_ops = {
 	.open		= soc_compr_open,
 	.free		= soc_compr_free,
 	.set_params	= soc_compr_set_params,
-	.set_metadata   = sst_compr_set_metadata,
-	.get_metadata	= sst_compr_get_metadata,
+	.set_metadata   = soc_compr_set_metadata,
+	.get_metadata	= soc_compr_get_metadata,
 	.get_params	= soc_compr_get_params,
 	.trigger	= soc_compr_trigger,
 	.pointer	= soc_compr_pointer,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d82ee386eab5..4d0561312f3b 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -30,9 +30,12 @@
 #include <linux/bitops.h>
 #include <linux/debugfs.h>
 #include <linux/platform_device.h>
+#include <linux/pinctrl/consumer.h>
 #include <linux/ctype.h>
 #include <linux/slab.h>
 #include <linux/of.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
 #include <sound/ac97_codec.h>
 #include <sound/core.h>
 #include <sound/jack.h>
@@ -47,8 +50,6 @@
 
 #define NAME_SIZE	32
 
-static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
-
 #ifdef CONFIG_DEBUG_FS
 struct dentry *snd_soc_debugfs_root;
 EXPORT_SYMBOL_GPL(snd_soc_debugfs_root);
@@ -69,6 +70,16 @@ static int pmdown_time = 5000;
 module_param(pmdown_time, int, 0);
 MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
 
+struct snd_ac97_reset_cfg {
+	struct pinctrl *pctl;
+	struct pinctrl_state *pstate_reset;
+	struct pinctrl_state *pstate_warm_reset;
+	struct pinctrl_state *pstate_run;
+	int gpio_sdata;
+	int gpio_sync;
+	int gpio_reset;
+};
+
 /* returns the minimum number of bytes needed to represent
  * a particular given value */
 static int min_bytes_needed(unsigned long val)
@@ -192,7 +203,7 @@ static ssize_t pmdown_time_set(struct device *dev,
 	struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
 	int ret;
 
-	ret = strict_strtol(buf, 10, &rtd->pmdown_time);
+	ret = kstrtol(buf, 10, &rtd->pmdown_time);
 	if (ret)
 		return ret;
 
@@ -237,6 +248,7 @@ static ssize_t codec_reg_write_file(struct file *file,
 	char *start = buf;
 	unsigned long reg, value;
 	struct snd_soc_codec *codec = file->private_data;
+	int ret;
 
 	buf_size = min(count, (sizeof(buf)-1));
 	if (copy_from_user(buf, user_buf, buf_size))
@@ -248,8 +260,9 @@ static ssize_t codec_reg_write_file(struct file *file,
 	reg = simple_strtoul(start, &start, 16);
 	while (*start == ' ')
 		start++;
-	if (strict_strtoul(start, 16, &value))
-		return -EINVAL;
+	ret = kstrtoul(start, 16, &value);
+	if (ret)
+		return ret;
 
 	/* Userspace has been fiddling around behind the kernel's back */
 	add_taint(TAINT_USER, LOCKDEP_NOW_UNRELIABLE);
@@ -530,6 +543,15 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
 }
 #endif
 
+static void codec2codec_close_delayed_work(struct work_struct *work)
+{
+	/* Currently nothing to do for c2c links
+	 * Since c2c links are internal nodes in the DAPM graph and
+	 * don't interface with the outside world or application layer
+	 * we don't have to do any special handling on close.
+	 */
+}
+
 #ifdef CONFIG_PM_SLEEP
 /* powers down audio subsystem for suspend */
 int snd_soc_suspend(struct device *dev)
@@ -1223,9 +1245,6 @@ static int soc_post_component_init(struct snd_soc_card *card,
 	}
 	rtd->card = card;
 
-	/* Make sure all DAPM widgets are instantiated */
-	snd_soc_dapm_new_widgets(&codec->dapm);
-
 	/* machine controls, routes and widgets are not prefixed */
 	temp = codec->name_prefix;
 	codec->name_prefix = NULL;
@@ -1428,6 +1447,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order)
 				return ret;
 			}
 		} else {
+			INIT_DELAYED_WORK(&rtd->delayed_work,
+						codec2codec_close_delayed_work);
+
 			/* link the DAI widgets */
 			play_w = codec_dai->playback_widget;
 			capture_w = cpu_dai->capture_widget;
@@ -1718,8 +1740,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 		snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
 					card->num_dapm_routes);
 
-	snd_soc_dapm_new_widgets(&card->dapm);
-
 	for (i = 0; i < card->num_links; i++) {
 		dai_link = &card->dai_link[i];
 		dai_fmt = dai_link->dai_fmt;
@@ -1798,12 +1818,12 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 		}
 	}
 
-	snd_soc_dapm_new_widgets(&card->dapm);
-
 	if (card->fully_routed)
 		list_for_each_entry(codec, &card->codec_dev_list, card_list)
 			snd_soc_dapm_auto_nc_codec_pins(codec);
 
+	snd_soc_dapm_new_widgets(card);
+
 	ret = snd_card_register(card->snd_card);
 	if (ret < 0) {
 		dev_err(card->dev, "ASoC: failed to register soundcard %d\n",
@@ -2080,6 +2100,117 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec);
 
+static struct snd_ac97_reset_cfg snd_ac97_rst_cfg;
+
+static void snd_soc_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+	struct pinctrl *pctl = snd_ac97_rst_cfg.pctl;
+
+	pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_warm_reset);
+
+	gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 1);
+
+	udelay(10);
+
+	gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 0);
+
+	pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_run);
+	msleep(2);
+}
+
+static void snd_soc_ac97_reset(struct snd_ac97 *ac97)
+{
+	struct pinctrl *pctl = snd_ac97_rst_cfg.pctl;
+
+	pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_reset);
+
+	gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 0);
+	gpio_direction_output(snd_ac97_rst_cfg.gpio_sdata, 0);
+	gpio_direction_output(snd_ac97_rst_cfg.gpio_reset, 0);
+
+	udelay(10);
+
+	gpio_direction_output(snd_ac97_rst_cfg.gpio_reset, 1);
+
+	pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_run);
+	msleep(2);
+}
+
+static int snd_soc_ac97_parse_pinctl(struct device *dev,
+		struct snd_ac97_reset_cfg *cfg)
+{
+	struct pinctrl *p;
+	struct pinctrl_state *state;
+	int gpio;
+	int ret;
+
+	p = devm_pinctrl_get(dev);
+	if (IS_ERR(p)) {
+		dev_err(dev, "Failed to get pinctrl\n");
+		return PTR_RET(p);
+	}
+	cfg->pctl = p;
+
+	state = pinctrl_lookup_state(p, "ac97-reset");
+	if (IS_ERR(state)) {
+		dev_err(dev, "Can't find pinctrl state ac97-reset\n");
+		return PTR_RET(state);
+	}
+	cfg->pstate_reset = state;
+
+	state = pinctrl_lookup_state(p, "ac97-warm-reset");
+	if (IS_ERR(state)) {
+		dev_err(dev, "Can't find pinctrl state ac97-warm-reset\n");
+		return PTR_RET(state);
+	}
+	cfg->pstate_warm_reset = state;
+
+	state = pinctrl_lookup_state(p, "ac97-running");
+	if (IS_ERR(state)) {
+		dev_err(dev, "Can't find pinctrl state ac97-running\n");
+		return PTR_RET(state);
+	}
+	cfg->pstate_run = state;
+
+	gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 0);
+	if (gpio < 0) {
+		dev_err(dev, "Can't find ac97-sync gpio\n");
+		return gpio;
+	}
+	ret = devm_gpio_request(dev, gpio, "AC97 link sync");
+	if (ret) {
+		dev_err(dev, "Failed requesting ac97-sync gpio\n");
+		return ret;
+	}
+	cfg->gpio_sync = gpio;
+
+	gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 1);
+	if (gpio < 0) {
+		dev_err(dev, "Can't find ac97-sdata gpio %d\n", gpio);
+		return gpio;
+	}
+	ret = devm_gpio_request(dev, gpio, "AC97 link sdata");
+	if (ret) {
+		dev_err(dev, "Failed requesting ac97-sdata gpio\n");
+		return ret;
+	}
+	cfg->gpio_sdata = gpio;
+
+	gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 2);
+	if (gpio < 0) {
+		dev_err(dev, "Can't find ac97-reset gpio\n");
+		return gpio;
+	}
+	ret = devm_gpio_request(dev, gpio, "AC97 link reset");
+	if (ret) {
+		dev_err(dev, "Failed requesting ac97-reset gpio\n");
+		return ret;
+	}
+	cfg->gpio_reset = gpio;
+
+	return 0;
+}
+
 struct snd_ac97_bus_ops *soc_ac97_ops;
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
@@ -2098,6 +2229,35 @@ int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops)
 EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops);
 
 /**
+ * snd_soc_set_ac97_ops_of_reset - Set ac97 ops with generic ac97 reset functions
+ *
+ * This function sets the reset and warm_reset properties of ops and parses
+ * the device node of pdev to get pinctrl states and gpio numbers to use.
+ */
+int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops,
+		struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct snd_ac97_reset_cfg cfg;
+	int ret;
+
+	ret = snd_soc_ac97_parse_pinctl(dev, &cfg);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_set_ac97_ops(ops);
+	if (ret)
+		return ret;
+
+	ops->warm_reset = snd_soc_ac97_warm_reset;
+	ops->reset = snd_soc_ac97_reset;
+
+	snd_ac97_rst_cfg = cfg;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops_of_reset);
+
+/**
  * snd_soc_free_ac97_codec - free AC97 codec device
  * @codec: audio codec
  *
@@ -2299,6 +2459,22 @@ static int snd_soc_add_controls(struct snd_card *card, struct device *dev,
 	return 0;
 }
 
+struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card,
+					       const char *name)
+{
+	struct snd_card *card = soc_card->snd_card;
+	struct snd_kcontrol *kctl;
+
+	if (unlikely(!name))
+		return NULL;
+
+	list_for_each_entry(kctl, &card->controls, list)
+		if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name)))
+			return kctl;
+	return NULL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol);
+
 /**
  * snd_soc_add_codec_controls - add an array of controls to a codec.
  * Convenience function to add a list of controls. Many codecs were
@@ -2541,59 +2717,6 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol,
 EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_double);
 
 /**
- * snd_soc_info_enum_ext - external enumerated single mixer info callback
- * @kcontrol: mixer control
- * @uinfo: control element information
- *
- * Callback to provide information about an external enumerated
- * single mixer.
- *
- * Returns 0 for success.
- */
-int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_info *uinfo)
-{
-	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
-
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = e->max;
-
-	if (uinfo->value.enumerated.item > e->max - 1)
-		uinfo->value.enumerated.item = e->max - 1;
-	strcpy(uinfo->value.enumerated.name,
-		e->texts[uinfo->value.enumerated.item]);
-	return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext);
-
-/**
- * snd_soc_info_volsw_ext - external single mixer info callback
- * @kcontrol: mixer control
- * @uinfo: control element information
- *
- * Callback to provide information about a single external mixer control.
- *
- * Returns 0 for success.
- */
-int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_info *uinfo)
-{
-	int max = kcontrol->private_value;
-
-	if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
-		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
-	else
-		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
-
-	uinfo->count = 1;
-	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = max;
-	return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext);
-
-/**
  * snd_soc_info_volsw - single mixer info callback
  * @kcontrol: mixer control
  * @uinfo: control element information
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 4375c9f2b791..c17c14c394df 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -47,6 +47,15 @@
 
 #define DAPM_UPDATE_STAT(widget, val) widget->dapm->card->dapm_stats.val++;
 
+static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
+	struct snd_soc_dapm_widget *wsource, struct snd_soc_dapm_widget *wsink,
+	const char *control,
+	int (*connected)(struct snd_soc_dapm_widget *source,
+			 struct snd_soc_dapm_widget *sink));
+static struct snd_soc_dapm_widget *
+snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
+			 const struct snd_soc_dapm_widget *widget);
+
 /* dapm power sequences - make this per codec in the future */
 static int dapm_up_seq[] = {
 	[snd_soc_dapm_pre] = 0,
@@ -73,16 +82,18 @@ static int dapm_up_seq[] = {
 	[snd_soc_dapm_hp] = 10,
 	[snd_soc_dapm_spk] = 10,
 	[snd_soc_dapm_line] = 10,
-	[snd_soc_dapm_post] = 11,
+	[snd_soc_dapm_kcontrol] = 11,
+	[snd_soc_dapm_post] = 12,
 };
 
 static int dapm_down_seq[] = {
 	[snd_soc_dapm_pre] = 0,
-	[snd_soc_dapm_adc] = 1,
-	[snd_soc_dapm_hp] = 2,
-	[snd_soc_dapm_spk] = 2,
-	[snd_soc_dapm_line] = 2,
-	[snd_soc_dapm_out_drv] = 2,
+	[snd_soc_dapm_kcontrol] = 1,
+	[snd_soc_dapm_adc] = 2,
+	[snd_soc_dapm_hp] = 3,
+	[snd_soc_dapm_spk] = 3,
+	[snd_soc_dapm_line] = 3,
+	[snd_soc_dapm_out_drv] = 3,
 	[snd_soc_dapm_pga] = 4,
 	[snd_soc_dapm_switch] = 5,
 	[snd_soc_dapm_mixer_named_ctl] = 5,
@@ -174,35 +185,177 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget(
 	return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL);
 }
 
-/* get snd_card from DAPM context */
-static inline struct snd_card *dapm_get_snd_card(
-	struct snd_soc_dapm_context *dapm)
+struct dapm_kcontrol_data {
+	unsigned int value;
+	struct snd_soc_dapm_widget *widget;
+	struct list_head paths;
+	struct snd_soc_dapm_widget_list *wlist;
+};
+
+static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
+	struct snd_kcontrol *kcontrol)
 {
-	if (dapm->codec)
-		return dapm->codec->card->snd_card;
-	else if (dapm->platform)
-		return dapm->platform->card->snd_card;
-	else
-		BUG();
+	struct dapm_kcontrol_data *data;
+	struct soc_mixer_control *mc;
 
-	/* unreachable */
-	return NULL;
+	data = kzalloc(sizeof(*data), GFP_KERNEL);
+	if (!data) {
+		dev_err(widget->dapm->dev,
+				"ASoC: can't allocate kcontrol data for %s\n",
+				widget->name);
+		return -ENOMEM;
+	}
+
+	INIT_LIST_HEAD(&data->paths);
+
+	switch (widget->id) {
+	case snd_soc_dapm_switch:
+	case snd_soc_dapm_mixer:
+	case snd_soc_dapm_mixer_named_ctl:
+		mc = (struct soc_mixer_control *)kcontrol->private_value;
+
+		if (mc->autodisable) {
+			struct snd_soc_dapm_widget template;
+
+			memset(&template, 0, sizeof(template));
+			template.reg = mc->reg;
+			template.mask = (1 << fls(mc->max)) - 1;
+			template.shift = mc->shift;
+			if (mc->invert)
+				template.off_val = mc->max;
+			else
+				template.off_val = 0;
+			template.on_val = template.off_val;
+			template.id = snd_soc_dapm_kcontrol;
+			template.name = kcontrol->id.name;
+
+			data->value = template.on_val;
+
+			data->widget = snd_soc_dapm_new_control(widget->dapm,
+				&template);
+			if (!data->widget) {
+				kfree(data);
+				return -ENOMEM;
+			}
+		}
+		break;
+	default:
+		break;
+	}
+
+	kcontrol->private_data = data;
+
+	return 0;
 }
 
-/* get soc_card from DAPM context */
-static inline struct snd_soc_card *dapm_get_soc_card(
-		struct snd_soc_dapm_context *dapm)
+static void dapm_kcontrol_free(struct snd_kcontrol *kctl)
 {
-	if (dapm->codec)
-		return dapm->codec->card;
-	else if (dapm->platform)
-		return dapm->platform->card;
+	struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl);
+	kfree(data->widget);
+	kfree(data->wlist);
+	kfree(data);
+}
+
+static struct snd_soc_dapm_widget_list *dapm_kcontrol_get_wlist(
+	const struct snd_kcontrol *kcontrol)
+{
+	struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol);
+
+	return data->wlist;
+}
+
+static int dapm_kcontrol_add_widget(struct snd_kcontrol *kcontrol,
+	struct snd_soc_dapm_widget *widget)
+{
+	struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget_list *new_wlist;
+	unsigned int n;
+
+	if (data->wlist)
+		n = data->wlist->num_widgets + 1;
 	else
-		BUG();
+		n = 1;
 
-	/* unreachable */
-	return NULL;
+	new_wlist = krealloc(data->wlist,
+			sizeof(*new_wlist) + sizeof(widget) * n, GFP_KERNEL);
+	if (!new_wlist)
+		return -ENOMEM;
+
+	new_wlist->widgets[n - 1] = widget;
+	new_wlist->num_widgets = n;
+
+	data->wlist = new_wlist;
+
+	return 0;
+}
+
+static void dapm_kcontrol_add_path(const struct snd_kcontrol *kcontrol,
+	struct snd_soc_dapm_path *path)
+{
+	struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol);
+
+	list_add_tail(&path->list_kcontrol, &data->paths);
+
+	if (data->widget) {
+		snd_soc_dapm_add_path(data->widget->dapm, data->widget,
+		    path->source, NULL, NULL);
+	}
+}
+
+static bool dapm_kcontrol_is_powered(const struct snd_kcontrol *kcontrol)
+{
+	struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol);
+
+	if (!data->widget)
+		return true;
+
+	return data->widget->power;
+}
+
+static struct list_head *dapm_kcontrol_get_path_list(
+	const struct snd_kcontrol *kcontrol)
+{
+	struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol);
+
+	return &data->paths;
+}
+
+#define dapm_kcontrol_for_each_path(path, kcontrol) \
+	list_for_each_entry(path, dapm_kcontrol_get_path_list(kcontrol), \
+		list_kcontrol)
+
+static unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol)
+{
+	struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol);
+
+	return data->value;
+}
+
+static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol,
+	unsigned int value)
+{
+	struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol);
+
+	if (data->value == value)
+		return false;
+
+	if (data->widget)
+		data->widget->on_val = value;
+
+	data->value = value;
+
+	return true;
+}
+
+/**
+ * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol
+ * @kcontrol: The kcontrol
+ */
+struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol)
+{
+	return dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->codec;
 }
+EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_codec);
 
 static void dapm_reset(struct snd_soc_card *card)
 {
@@ -211,6 +364,7 @@ static void dapm_reset(struct snd_soc_card *card)
 	memset(&card->dapm_stats, 0, sizeof(card->dapm_stats));
 
 	list_for_each_entry(w, &card->widgets, list) {
+		w->new_power = w->power;
 		w->power_checked = false;
 		w->inputs = -1;
 		w->outputs = -1;
@@ -428,6 +582,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 	case snd_soc_dapm_spk:
 	case snd_soc_dapm_line:
 	case snd_soc_dapm_dai_link:
+	case snd_soc_dapm_kcontrol:
 		p->connect = 1;
 	break;
 	/* does affect routing - dynamically connected */
@@ -507,17 +662,12 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm,
 	return 0;
 }
 
-static void dapm_kcontrol_free(struct snd_kcontrol *kctl)
-{
-	kfree(kctl->private_data);
-}
-
 /*
  * Determine if a kcontrol is shared. If it is, look it up. If it isn't,
  * create it. Either way, add the widget into the control's widget list
  */
 static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w,
-	int kci, struct snd_soc_dapm_path *path)
+	int kci)
 {
 	struct snd_soc_dapm_context *dapm = w->dapm;
 	struct snd_card *card = dapm->card->snd_card;
@@ -525,9 +675,6 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w,
 	size_t prefix_len;
 	int shared;
 	struct snd_kcontrol *kcontrol;
-	struct snd_soc_dapm_widget_list *wlist;
-	int wlistentries;
-	size_t wlistsize;
 	bool wname_in_long_name, kcname_in_long_name;
 	char *long_name;
 	const char *name;
@@ -546,25 +693,6 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w,
 	shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[kci],
 					 &kcontrol);
 
-	if (kcontrol) {
-		wlist = kcontrol->private_data;
-		wlistentries = wlist->num_widgets + 1;
-	} else {
-		wlist = NULL;
-		wlistentries = 1;
-	}
-
-	wlistsize = sizeof(struct snd_soc_dapm_widget_list) +
-			wlistentries * sizeof(struct snd_soc_dapm_widget *);
-	wlist = krealloc(wlist, wlistsize, GFP_KERNEL);
-	if (wlist == NULL) {
-		dev_err(dapm->dev, "ASoC: can't allocate widget list for %s\n",
-			w->name);
-		return -ENOMEM;
-	}
-	wlist->num_widgets = wlistentries;
-	wlist->widgets[wlistentries - 1] = w;
-
 	if (!kcontrol) {
 		if (shared) {
 			wname_in_long_name = false;
@@ -587,7 +715,6 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w,
 				kcname_in_long_name = false;
 				break;
 			default:
-				kfree(wlist);
 				return -EINVAL;
 			}
 		}
@@ -602,10 +729,8 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w,
 			long_name = kasprintf(GFP_KERNEL, "%s %s",
 				 w->name + prefix_len,
 				 w->kcontrol_news[kci].name);
-			if (long_name == NULL) {
-				kfree(wlist);
+			if (long_name == NULL)
 				return -ENOMEM;
-			}
 
 			name = long_name;
 		} else if (wname_in_long_name) {
@@ -616,23 +741,33 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w,
 			name = w->kcontrol_news[kci].name;
 		}
 
-		kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], wlist, name,
+		kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], NULL, name,
 					prefix);
-		kcontrol->private_free = dapm_kcontrol_free;
 		kfree(long_name);
+		if (!kcontrol)
+			return -ENOMEM;
+		kcontrol->private_free = dapm_kcontrol_free;
+
+		ret = dapm_kcontrol_data_alloc(w, kcontrol);
+		if (ret) {
+			snd_ctl_free_one(kcontrol);
+			return ret;
+		}
+
 		ret = snd_ctl_add(card, kcontrol);
 		if (ret < 0) {
 			dev_err(dapm->dev,
 				"ASoC: failed to add widget %s dapm kcontrol %s: %d\n",
 				w->name, name, ret);
-			kfree(wlist);
 			return ret;
 		}
 	}
 
-	kcontrol->private_data = wlist;
+	ret = dapm_kcontrol_add_widget(kcontrol, w);
+	if (ret)
+		return ret;
+
 	w->kcontrols[kci] = kcontrol;
-	path->kcontrol = kcontrol;
 
 	return 0;
 }
@@ -652,13 +787,15 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w)
 				continue;
 
 			if (w->kcontrols[i]) {
-				path->kcontrol = w->kcontrols[i];
+				dapm_kcontrol_add_path(w->kcontrols[i], path);
 				continue;
 			}
 
-			ret = dapm_create_or_share_mixmux_kcontrol(w, i, path);
+			ret = dapm_create_or_share_mixmux_kcontrol(w, i);
 			if (ret < 0)
 				return ret;
+
+			dapm_kcontrol_add_path(w->kcontrols[i], path);
 		}
 	}
 
@@ -684,15 +821,12 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w)
 		return -EINVAL;
 	}
 
-	path = list_first_entry(&w->sources, struct snd_soc_dapm_path,
-				list_sink);
-
-	ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path);
+	ret = dapm_create_or_share_mixmux_kcontrol(w, 0);
 	if (ret < 0)
 		return ret;
 
 	list_for_each_entry(path, &w->sources, list_sink)
-		path->kcontrol = w->kcontrols[0];
+		dapm_kcontrol_add_path(w->kcontrols[0], path);
 
 	return 0;
 }
@@ -813,6 +947,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
 	case snd_soc_dapm_clock_supply:
+	case snd_soc_dapm_kcontrol:
 		return 0;
 	default:
 		break;
@@ -908,6 +1043,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
 	case snd_soc_dapm_clock_supply:
+	case snd_soc_dapm_kcontrol:
 		return 0;
 	default:
 		break;
@@ -1062,7 +1198,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
 	int ret;
 
 	if (SND_SOC_DAPM_EVENT_ON(event)) {
-		if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) {
+		if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) {
 			ret = regulator_allow_bypass(w->regulator, false);
 			if (ret != 0)
 				dev_warn(w->dapm->dev,
@@ -1072,7 +1208,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
 
 		return regulator_enable(w->regulator);
 	} else {
-		if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) {
+		if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) {
 			ret = regulator_allow_bypass(w->regulator, true);
 			if (ret != 0)
 				dev_warn(w->dapm->dev,
@@ -1244,10 +1380,9 @@ static void dapm_seq_insert(struct snd_soc_dapm_widget *new_widget,
 	list_add_tail(&new_widget->power_list, list);
 }
 
-static void dapm_seq_check_event(struct snd_soc_dapm_context *dapm,
+static void dapm_seq_check_event(struct snd_soc_card *card,
 				 struct snd_soc_dapm_widget *w, int event)
 {
-	struct snd_soc_card *card = dapm->card;
 	const char *ev_name;
 	int power, ret;
 
@@ -1281,55 +1416,50 @@ static void dapm_seq_check_event(struct snd_soc_dapm_context *dapm,
 		return;
 	}
 
-	if (w->power != power)
+	if (w->new_power != power)
 		return;
 
 	if (w->event && (w->event_flags & event)) {
-		pop_dbg(dapm->dev, card->pop_time, "pop test : %s %s\n",
+		pop_dbg(w->dapm->dev, card->pop_time, "pop test : %s %s\n",
 			w->name, ev_name);
 		trace_snd_soc_dapm_widget_event_start(w, event);
 		ret = w->event(w, NULL, event);
 		trace_snd_soc_dapm_widget_event_done(w, event);
 		if (ret < 0)
-			dev_err(dapm->dev, "ASoC: %s: %s event failed: %d\n",
+			dev_err(w->dapm->dev, "ASoC: %s: %s event failed: %d\n",
 			       ev_name, w->name, ret);
 	}
 }
 
 /* Apply the coalesced changes from a DAPM sequence */
-static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm,
+static void dapm_seq_run_coalesced(struct snd_soc_card *card,
 				   struct list_head *pending)
 {
-	struct snd_soc_card *card = dapm->card;
 	struct snd_soc_dapm_widget *w;
-	int reg, power;
+	int reg;
 	unsigned int value = 0;
 	unsigned int mask = 0;
-	unsigned int cur_mask;
 
 	reg = list_first_entry(pending, struct snd_soc_dapm_widget,
 			       power_list)->reg;
 
 	list_for_each_entry(w, pending, power_list) {
-		cur_mask = 1 << w->shift;
 		BUG_ON(reg != w->reg);
+		w->power = w->new_power;
 
-		if (w->invert)
-			power = !w->power;
+		mask |= w->mask << w->shift;
+		if (w->power)
+			value |= w->on_val << w->shift;
 		else
-			power = w->power;
+			value |= w->off_val << w->shift;
 
-		mask |= cur_mask;
-		if (power)
-			value |= cur_mask;
-
-		pop_dbg(dapm->dev, card->pop_time,
+		pop_dbg(w->dapm->dev, card->pop_time,
 			"pop test : Queue %s: reg=0x%x, 0x%x/0x%x\n",
 			w->name, reg, value, mask);
 
 		/* Check for events */
-		dapm_seq_check_event(dapm, w, SND_SOC_DAPM_PRE_PMU);
-		dapm_seq_check_event(dapm, w, SND_SOC_DAPM_PRE_PMD);
+		dapm_seq_check_event(card, w, SND_SOC_DAPM_PRE_PMU);
+		dapm_seq_check_event(card, w, SND_SOC_DAPM_PRE_PMD);
 	}
 
 	if (reg >= 0) {
@@ -1339,7 +1469,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm,
 		w = list_first_entry(pending, struct snd_soc_dapm_widget,
 				     power_list);
 
-		pop_dbg(dapm->dev, card->pop_time,
+		pop_dbg(w->dapm->dev, card->pop_time,
 			"pop test : Applying 0x%x/0x%x to %x in %dms\n",
 			value, mask, reg, card->pop_time);
 		pop_wait(card->pop_time);
@@ -1347,8 +1477,8 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm,
 	}
 
 	list_for_each_entry(w, pending, power_list) {
-		dapm_seq_check_event(dapm, w, SND_SOC_DAPM_POST_PMU);
-		dapm_seq_check_event(dapm, w, SND_SOC_DAPM_POST_PMD);
+		dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMU);
+		dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMD);
 	}
 }
 
@@ -1360,8 +1490,8 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm,
  * Currently anything that requires more than a single write is not
  * handled.
  */
-static void dapm_seq_run(struct snd_soc_dapm_context *dapm,
-			 struct list_head *list, int event, bool power_up)
+static void dapm_seq_run(struct snd_soc_card *card,
+	struct list_head *list, int event, bool power_up)
 {
 	struct snd_soc_dapm_widget *w, *n;
 	LIST_HEAD(pending);
@@ -1384,7 +1514,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm,
 		if (sort[w->id] != cur_sort || w->reg != cur_reg ||
 		    w->dapm != cur_dapm || w->subseq != cur_subseq) {
 			if (!list_empty(&pending))
-				dapm_seq_run_coalesced(cur_dapm, &pending);
+				dapm_seq_run_coalesced(card, &pending);
 
 			if (cur_dapm && cur_dapm->seq_notifier) {
 				for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++)
@@ -1444,7 +1574,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm,
 	}
 
 	if (!list_empty(&pending))
-		dapm_seq_run_coalesced(cur_dapm, &pending);
+		dapm_seq_run_coalesced(card, &pending);
 
 	if (cur_dapm && cur_dapm->seq_notifier) {
 		for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++)
@@ -1454,37 +1584,48 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm,
 	}
 }
 
-static void dapm_widget_update(struct snd_soc_dapm_context *dapm)
+static void dapm_widget_update(struct snd_soc_card *card)
 {
-	struct snd_soc_dapm_update *update = dapm->update;
-	struct snd_soc_dapm_widget *w;
+	struct snd_soc_dapm_update *update = card->update;
+	struct snd_soc_dapm_widget_list *wlist;
+	struct snd_soc_dapm_widget *w = NULL;
+	unsigned int wi;
 	int ret;
 
-	if (!update)
+	if (!update || !dapm_kcontrol_is_powered(update->kcontrol))
 		return;
 
-	w = update->widget;
+	wlist = dapm_kcontrol_get_wlist(update->kcontrol);
 
-	if (w->event &&
-	    (w->event_flags & SND_SOC_DAPM_PRE_REG)) {
-		ret = w->event(w, update->kcontrol, SND_SOC_DAPM_PRE_REG);
-		if (ret != 0)
-			dev_err(dapm->dev, "ASoC: %s DAPM pre-event failed: %d\n",
-			       w->name, ret);
+	for (wi = 0; wi < wlist->num_widgets; wi++) {
+		w = wlist->widgets[wi];
+
+		if (w->event && (w->event_flags & SND_SOC_DAPM_PRE_REG)) {
+			ret = w->event(w, update->kcontrol, SND_SOC_DAPM_PRE_REG);
+			if (ret != 0)
+				dev_err(w->dapm->dev, "ASoC: %s DAPM pre-event failed: %d\n",
+					   w->name, ret);
+		}
 	}
 
+	if (!w)
+		return;
+
 	ret = soc_widget_update_bits_locked(w, update->reg, update->mask,
 				  update->val);
 	if (ret < 0)
-		dev_err(dapm->dev, "ASoC: %s DAPM update failed: %d\n",
+		dev_err(w->dapm->dev, "ASoC: %s DAPM update failed: %d\n",
 			w->name, ret);
 
-	if (w->event &&
-	    (w->event_flags & SND_SOC_DAPM_POST_REG)) {
-		ret = w->event(w, update->kcontrol, SND_SOC_DAPM_POST_REG);
-		if (ret != 0)
-			dev_err(dapm->dev, "ASoC: %s DAPM post-event failed: %d\n",
-			       w->name, ret);
+	for (wi = 0; wi < wlist->num_widgets; wi++) {
+		w = wlist->widgets[wi];
+
+		if (w->event && (w->event_flags & SND_SOC_DAPM_POST_REG)) {
+			ret = w->event(w, update->kcontrol, SND_SOC_DAPM_POST_REG);
+			if (ret != 0)
+				dev_err(w->dapm->dev, "ASoC: %s DAPM post-event failed: %d\n",
+					   w->name, ret);
+		}
 	}
 }
 
@@ -1596,6 +1737,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power,
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
 	case snd_soc_dapm_clock_supply:
+	case snd_soc_dapm_kcontrol:
 		/* Supplies can't affect their outputs, only their inputs */
 		break;
 	default:
@@ -1612,8 +1754,6 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power,
 		dapm_seq_insert(w, up_list, true);
 	else
 		dapm_seq_insert(w, down_list, false);
-
-	w->power = power;
 }
 
 static void dapm_power_one_widget(struct snd_soc_dapm_widget *w,
@@ -1647,9 +1787,8 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w,
  *  o Input pin to Output pin (bypass, sidetone)
  *  o DAC to ADC (loopback).
  */
-static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
+static int dapm_power_widgets(struct snd_soc_card *card, int event)
 {
-	struct snd_soc_card *card = dapm->card;
 	struct snd_soc_dapm_widget *w;
 	struct snd_soc_dapm_context *d;
 	LIST_HEAD(up_list);
@@ -1689,7 +1828,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 			break;
 		}
 
-		if (w->power) {
+		if (w->new_power) {
 			d = w->dapm;
 
 			/* Supplies and micbiases only bring the
@@ -1731,29 +1870,29 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 	trace_snd_soc_dapm_walk_done(card);
 
 	/* Run all the bias changes in parallel */
-	list_for_each_entry(d, &dapm->card->dapm_list, list)
+	list_for_each_entry(d, &card->dapm_list, list)
 		async_schedule_domain(dapm_pre_sequence_async, d,
 					&async_domain);
 	async_synchronize_full_domain(&async_domain);
 
 	list_for_each_entry(w, &down_list, power_list) {
-		dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMD);
+		dapm_seq_check_event(card, w, SND_SOC_DAPM_WILL_PMD);
 	}
 
 	list_for_each_entry(w, &up_list, power_list) {
-		dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMU);
+		dapm_seq_check_event(card, w, SND_SOC_DAPM_WILL_PMU);
 	}
 
 	/* Power down widgets first; try to avoid amplifying pops. */
-	dapm_seq_run(dapm, &down_list, event, false);
+	dapm_seq_run(card, &down_list, event, false);
 
-	dapm_widget_update(dapm);
+	dapm_widget_update(card);
 
 	/* Now power up. */
-	dapm_seq_run(dapm, &up_list, event, true);
+	dapm_seq_run(card, &up_list, event, true);
 
 	/* Run all the bias changes in parallel */
-	list_for_each_entry(d, &dapm->card->dapm_list, list)
+	list_for_each_entry(d, &card->dapm_list, list)
 		async_schedule_domain(dapm_post_sequence_async, d,
 					&async_domain);
 	async_synchronize_full_domain(&async_domain);
@@ -1764,7 +1903,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 			d->stream_event(d, event);
 	}
 
-	pop_dbg(dapm->dev, card->pop_time,
+	pop_dbg(card->dev, card->pop_time,
 		"DAPM sequencing finished, waiting %dms\n", card->pop_time);
 	pop_wait(card->pop_time);
 
@@ -1799,8 +1938,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 
 	if (w->reg >= 0)
 		ret += snprintf(buf + ret, PAGE_SIZE - ret,
-				" - R%d(0x%x) bit %d",
-				w->reg, w->reg, w->shift);
+				" - R%d(0x%x) mask 0x%x",
+				w->reg, w->reg, w->mask << w->shift);
 
 	ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n");
 
@@ -1937,22 +2076,14 @@ static inline void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm)
 #endif
 
 /* test and update the power status of a mux widget */
-static int soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
+static int soc_dapm_mux_update_power(struct snd_soc_card *card,
 				 struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e)
 {
 	struct snd_soc_dapm_path *path;
 	int found = 0;
 
-	if (widget->id != snd_soc_dapm_mux &&
-	    widget->id != snd_soc_dapm_virt_mux &&
-	    widget->id != snd_soc_dapm_value_mux)
-		return -ENODEV;
-
 	/* find dapm widget path assoc with kcontrol */
-	list_for_each_entry(path, &widget->dapm->card->paths, list) {
-		if (path->kcontrol != kcontrol)
-			continue;
-
+	dapm_kcontrol_for_each_path(path, kcontrol) {
 		if (!path->name || !e->texts[mux])
 			continue;
 
@@ -1967,73 +2098,68 @@ static int soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
 						"mux disconnection");
 			path->connect = 0; /* old connection must be powered down */
 		}
+		dapm_mark_dirty(path->sink, "mux change");
 	}
 
-	if (found) {
-		dapm_mark_dirty(widget, "mux change");
-		dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP);
-	}
+	if (found)
+		dapm_power_widgets(card, SND_SOC_DAPM_STREAM_NOP);
 
 	return found;
 }
 
-int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
-		struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e)
+int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm,
+	struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e,
+	struct snd_soc_dapm_update *update)
 {
-	struct snd_soc_card *card = widget->dapm->card;
+	struct snd_soc_card *card = dapm->card;
 	int ret;
 
 	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
-	ret = soc_dapm_mux_update_power(widget, kcontrol, mux, e);
+	card->update = update;
+	ret = soc_dapm_mux_update_power(card, kcontrol, mux, e);
+	card->update = NULL;
 	mutex_unlock(&card->dapm_mutex);
 	if (ret > 0)
-		soc_dpcm_runtime_update(widget);
+		soc_dpcm_runtime_update(card);
 	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power);
 
 /* test and update the power status of a mixer or switch widget */
-static int soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
+static int soc_dapm_mixer_update_power(struct snd_soc_card *card,
 				   struct snd_kcontrol *kcontrol, int connect)
 {
 	struct snd_soc_dapm_path *path;
 	int found = 0;
 
-	if (widget->id != snd_soc_dapm_mixer &&
-	    widget->id != snd_soc_dapm_mixer_named_ctl &&
-	    widget->id != snd_soc_dapm_switch)
-		return -ENODEV;
-
 	/* find dapm widget path assoc with kcontrol */
-	list_for_each_entry(path, &widget->dapm->card->paths, list) {
-		if (path->kcontrol != kcontrol)
-			continue;
-
-		/* found, now check type */
+	dapm_kcontrol_for_each_path(path, kcontrol) {
 		found = 1;
 		path->connect = connect;
 		dapm_mark_dirty(path->source, "mixer connection");
+		dapm_mark_dirty(path->sink, "mixer update");
 	}
 
-	if (found) {
-		dapm_mark_dirty(widget, "mixer update");
-		dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP);
-	}
+	if (found)
+		dapm_power_widgets(card, SND_SOC_DAPM_STREAM_NOP);
 
 	return found;
 }
 
-int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
-				struct snd_kcontrol *kcontrol, int connect)
+int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm,
+	struct snd_kcontrol *kcontrol, int connect,
+	struct snd_soc_dapm_update *update)
 {
-	struct snd_soc_card *card = widget->dapm->card;
+	struct snd_soc_card *card = dapm->card;
 	int ret;
 
 	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
-	ret = soc_dapm_mixer_update_power(widget, kcontrol, connect);
+	card->update = update;
+	ret = soc_dapm_mixer_update_power(card, kcontrol, connect);
+	card->update = NULL;
 	mutex_unlock(&card->dapm_mutex);
 	if (ret > 0)
-		soc_dpcm_runtime_update(widget);
+		soc_dpcm_runtime_update(card);
 	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power);
@@ -2112,6 +2238,7 @@ static void dapm_free_path(struct snd_soc_dapm_path *path)
 {
 	list_del(&path->list_sink);
 	list_del(&path->list_source);
+	list_del(&path->list_kcontrol);
 	list_del(&path->list);
 	kfree(path);
 }
@@ -2206,70 +2333,20 @@ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm)
 		return 0;
 
 	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
-	ret = dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP);
+	ret = dapm_power_widgets(dapm->card, SND_SOC_DAPM_STREAM_NOP);
 	mutex_unlock(&dapm->card->dapm_mutex);
 	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_sync);
 
-static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
-				  const struct snd_soc_dapm_route *route)
+static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
+	struct snd_soc_dapm_widget *wsource, struct snd_soc_dapm_widget *wsink,
+	const char *control,
+	int (*connected)(struct snd_soc_dapm_widget *source,
+			 struct snd_soc_dapm_widget *sink))
 {
 	struct snd_soc_dapm_path *path;
-	struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w;
-	struct snd_soc_dapm_widget *wtsource = NULL, *wtsink = NULL;
-	const char *sink;
-	const char *control = route->control;
-	const char *source;
-	char prefixed_sink[80];
-	char prefixed_source[80];
-	int ret = 0;
-
-	if (dapm->codec && dapm->codec->name_prefix) {
-		snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s",
-			 dapm->codec->name_prefix, route->sink);
-		sink = prefixed_sink;
-		snprintf(prefixed_source, sizeof(prefixed_source), "%s %s",
-			 dapm->codec->name_prefix, route->source);
-		source = prefixed_source;
-	} else {
-		sink = route->sink;
-		source = route->source;
-	}
-
-	/*
-	 * find src and dest widgets over all widgets but favor a widget from
-	 * current DAPM context
-	 */
-	list_for_each_entry(w, &dapm->card->widgets, list) {
-		if (!wsink && !(strcmp(w->name, sink))) {
-			wtsink = w;
-			if (w->dapm == dapm)
-				wsink = w;
-			continue;
-		}
-		if (!wsource && !(strcmp(w->name, source))) {
-			wtsource = w;
-			if (w->dapm == dapm)
-				wsource = w;
-		}
-	}
-	/* use widget from another DAPM context if not found from this */
-	if (!wsink)
-		wsink = wtsink;
-	if (!wsource)
-		wsource = wtsource;
-
-	if (wsource == NULL) {
-		dev_err(dapm->dev, "ASoC: no source widget found for %s\n",
-			route->source);
-		return -ENODEV;
-	}
-	if (wsink == NULL) {
-		dev_err(dapm->dev, "ASoC: no sink widget found for %s\n",
-			route->sink);
-		return -ENODEV;
-	}
+	int ret;
 
 	path = kzalloc(sizeof(struct snd_soc_dapm_path), GFP_KERNEL);
 	if (!path)
@@ -2277,8 +2354,9 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 
 	path->source = wsource;
 	path->sink = wsink;
-	path->connected = route->connected;
+	path->connected = connected;
 	INIT_LIST_HEAD(&path->list);
+	INIT_LIST_HEAD(&path->list_kcontrol);
 	INIT_LIST_HEAD(&path->list_source);
 	INIT_LIST_HEAD(&path->list_sink);
 
@@ -2298,6 +2376,9 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 			wsource->ext = 1;
 	}
 
+	dapm_mark_dirty(wsource, "Route added");
+	dapm_mark_dirty(wsink, "Route added");
+
 	/* connect static paths */
 	if (control == NULL) {
 		list_add(&path->list, &dapm->card->paths);
@@ -2328,6 +2409,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 	case snd_soc_dapm_dai_in:
 	case snd_soc_dapm_dai_out:
 	case snd_soc_dapm_dai_link:
+	case snd_soc_dapm_kcontrol:
 		list_add(&path->list, &dapm->card->paths);
 		list_add(&path->list_sink, &wsink->sources);
 		list_add(&path->list_source, &wsource->sinks);
@@ -2359,15 +2441,78 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 		return 0;
 	}
 
-	dapm_mark_dirty(wsource, "Route added");
-	dapm_mark_dirty(wsink, "Route added");
-
 	return 0;
+err:
+	kfree(path);
+	return ret;
+}
+
+static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
+				  const struct snd_soc_dapm_route *route)
+{
+	struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w;
+	struct snd_soc_dapm_widget *wtsource = NULL, *wtsink = NULL;
+	const char *sink;
+	const char *source;
+	char prefixed_sink[80];
+	char prefixed_source[80];
+	int ret;
+
+	if (dapm->codec && dapm->codec->name_prefix) {
+		snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s",
+			 dapm->codec->name_prefix, route->sink);
+		sink = prefixed_sink;
+		snprintf(prefixed_source, sizeof(prefixed_source), "%s %s",
+			 dapm->codec->name_prefix, route->source);
+		source = prefixed_source;
+	} else {
+		sink = route->sink;
+		source = route->source;
+	}
+
+	/*
+	 * find src and dest widgets over all widgets but favor a widget from
+	 * current DAPM context
+	 */
+	list_for_each_entry(w, &dapm->card->widgets, list) {
+		if (!wsink && !(strcmp(w->name, sink))) {
+			wtsink = w;
+			if (w->dapm == dapm)
+				wsink = w;
+			continue;
+		}
+		if (!wsource && !(strcmp(w->name, source))) {
+			wtsource = w;
+			if (w->dapm == dapm)
+				wsource = w;
+		}
+	}
+	/* use widget from another DAPM context if not found from this */
+	if (!wsink)
+		wsink = wtsink;
+	if (!wsource)
+		wsource = wtsource;
 
+	if (wsource == NULL) {
+		dev_err(dapm->dev, "ASoC: no source widget found for %s\n",
+			route->source);
+		return -ENODEV;
+	}
+	if (wsink == NULL) {
+		dev_err(dapm->dev, "ASoC: no sink widget found for %s\n",
+			route->sink);
+		return -ENODEV;
+	}
+
+	ret = snd_soc_dapm_add_path(dapm, wsource, wsink, route->control,
+		route->connected);
+	if (ret)
+		goto err;
+
+	return 0;
 err:
 	dev_warn(dapm->dev, "ASoC: no dapm match for %s --> %s --> %s\n",
-		 source, control, sink);
-	kfree(path);
+		 source, route->control, sink);
 	return ret;
 }
 
@@ -2569,14 +2714,14 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes);
  *
  * Returns 0 for success.
  */
-int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
+int snd_soc_dapm_new_widgets(struct snd_soc_card *card)
 {
 	struct snd_soc_dapm_widget *w;
 	unsigned int val;
 
-	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
 
-	list_for_each_entry(w, &dapm->card->widgets, list)
+	list_for_each_entry(w, &card->widgets, list)
 	{
 		if (w->new)
 			continue;
@@ -2586,7 +2731,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
 						sizeof(struct snd_kcontrol *),
 						GFP_KERNEL);
 			if (!w->kcontrols) {
-				mutex_unlock(&dapm->card->dapm_mutex);
+				mutex_unlock(&card->dapm_mutex);
 				return -ENOMEM;
 			}
 		}
@@ -2612,12 +2757,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
 
 		/* Read the initial power state from the device */
 		if (w->reg >= 0) {
-			val = soc_widget_read(w, w->reg);
-			val &= 1 << w->shift;
-			if (w->invert)
-				val = !val;
-
-			if (val)
+			val = soc_widget_read(w, w->reg) >> w->shift;
+			val &= w->mask;
+			if (val == w->on_val)
 				w->power = 1;
 		}
 
@@ -2627,8 +2769,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
 		dapm_debugfs_add_widget(w);
 	}
 
-	dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP);
-	mutex_unlock(&dapm->card->dapm_mutex);
+	dapm_power_widgets(card, SND_SOC_DAPM_STREAM_NOP);
+	mutex_unlock(&card->dapm_mutex);
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets);
@@ -2645,8 +2787,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets);
 int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+	struct snd_soc_card *card = codec->card;
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
 	unsigned int reg = mc->reg;
@@ -2654,17 +2796,24 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
 	int max = mc->max;
 	unsigned int mask = (1 << fls(max)) - 1;
 	unsigned int invert = mc->invert;
+	unsigned int val;
 
 	if (snd_soc_volsw_is_stereo(mc))
-		dev_warn(widget->dapm->dev,
+		dev_warn(codec->dapm.dev,
 			 "ASoC: Control '%s' is stereo, which is not supported\n",
 			 kcontrol->id.name);
 
-	ucontrol->value.integer.value[0] =
-		(snd_soc_read(widget->codec, reg) >> shift) & mask;
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+	if (dapm_kcontrol_is_powered(kcontrol))
+		val = (snd_soc_read(codec, reg) >> shift) & mask;
+	else
+		val = dapm_kcontrol_get_value(kcontrol);
+	mutex_unlock(&card->dapm_mutex);
+
 	if (invert)
-		ucontrol->value.integer.value[0] =
-			max - ucontrol->value.integer.value[0];
+		ucontrol->value.integer.value[0] = max - val;
+	else
+		ucontrol->value.integer.value[0] = val;
 
 	return 0;
 }
@@ -2682,9 +2831,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw);
 int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
-	struct snd_soc_codec *codec = widget->codec;
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
 	struct snd_soc_card *card = codec->card;
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
@@ -2696,10 +2843,9 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 	unsigned int val;
 	int connect, change;
 	struct snd_soc_dapm_update update;
-	int wi;
 
 	if (snd_soc_volsw_is_stereo(mc))
-		dev_warn(widget->dapm->dev,
+		dev_warn(codec->dapm.dev,
 			 "ASoC: Control '%s' is stereo, which is not supported\n",
 			 kcontrol->id.name);
 
@@ -2708,29 +2854,26 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 
 	if (invert)
 		val = max - val;
-	mask = mask << shift;
-	val = val << shift;
 
 	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 
-	change = snd_soc_test_bits(widget->codec, reg, mask, val);
-	if (change) {
-		for (wi = 0; wi < wlist->num_widgets; wi++) {
-			widget = wlist->widgets[wi];
+	dapm_kcontrol_set_value(kcontrol, val);
 
-			widget->value = val;
+	mask = mask << shift;
+	val = val << shift;
 
-			update.kcontrol = kcontrol;
-			update.widget = widget;
-			update.reg = reg;
-			update.mask = mask;
-			update.val = val;
-			widget->dapm->update = &update;
+	change = snd_soc_test_bits(codec, reg, mask, val);
+	if (change) {
+		update.kcontrol = kcontrol;
+		update.reg = reg;
+		update.mask = mask;
+		update.val = val;
 
-			soc_dapm_mixer_update_power(widget, kcontrol, connect);
+		card->update = &update;
 
-			widget->dapm->update = NULL;
-		}
+		soc_dapm_mixer_update_power(card, kcontrol, connect);
+
+		card->update = NULL;
 	}
 
 	mutex_unlock(&card->dapm_mutex);
@@ -2750,12 +2893,11 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw);
 int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int val;
 
-	val = snd_soc_read(widget->codec, e->reg);
+	val = snd_soc_read(codec, e->reg);
 	ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & e->mask;
 	if (e->shift_l != e->shift_r)
 		ucontrol->value.enumerated.item[1] =
@@ -2777,15 +2919,12 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);
 int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
-	struct snd_soc_codec *codec = widget->codec;
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
 	struct snd_soc_card *card = codec->card;
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int val, mux, change;
 	unsigned int mask;
 	struct snd_soc_dapm_update update;
-	int wi;
 
 	if (ucontrol->value.enumerated.item[0] > e->max - 1)
 		return -EINVAL;
@@ -2801,24 +2940,17 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
 
 	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 
-	change = snd_soc_test_bits(widget->codec, e->reg, mask, val);
+	change = snd_soc_test_bits(codec, e->reg, mask, val);
 	if (change) {
-		for (wi = 0; wi < wlist->num_widgets; wi++) {
-			widget = wlist->widgets[wi];
-
-			widget->value = val;
+		update.kcontrol = kcontrol;
+		update.reg = e->reg;
+		update.mask = mask;
+		update.val = val;
+		card->update = &update;
 
-			update.kcontrol = kcontrol;
-			update.widget = widget;
-			update.reg = e->reg;
-			update.mask = mask;
-			update.val = val;
-			widget->dapm->update = &update;
+		soc_dapm_mux_update_power(card, kcontrol, mux, e);
 
-			soc_dapm_mux_update_power(widget, kcontrol, mux, e);
-
-			widget->dapm->update = NULL;
-		}
+		card->update = NULL;
 	}
 
 	mutex_unlock(&card->dapm_mutex);
@@ -2836,11 +2968,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double);
 int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol,
 			       struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
-
-	ucontrol->value.enumerated.item[0] = widget->value;
-
+	ucontrol->value.enumerated.item[0] = dapm_kcontrol_get_value(kcontrol);
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt);
@@ -2855,30 +2983,22 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt);
 int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol,
 			       struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
-	struct snd_soc_codec *codec = widget->codec;
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
 	struct snd_soc_card *card = codec->card;
+	unsigned int value;
 	struct soc_enum *e =
 		(struct soc_enum *)kcontrol->private_value;
 	int change;
-	int wi;
 
 	if (ucontrol->value.enumerated.item[0] >= e->max)
 		return -EINVAL;
 
 	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 
-	change = widget->value != ucontrol->value.enumerated.item[0];
-	if (change) {
-		for (wi = 0; wi < wlist->num_widgets; wi++) {
-			widget = wlist->widgets[wi];
-
-			widget->value = ucontrol->value.enumerated.item[0];
-
-			soc_dapm_mux_update_power(widget, kcontrol, widget->value, e);
-		}
-	}
+	value = ucontrol->value.enumerated.item[0];
+	change = dapm_kcontrol_set_value(kcontrol, value);
+	if (change)
+		soc_dapm_mux_update_power(card, kcontrol, value, e);
 
 	mutex_unlock(&card->dapm_mutex);
 	return change;
@@ -2901,12 +3021,11 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt);
 int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int reg_val, val, mux;
 
-	reg_val = snd_soc_read(widget->codec, e->reg);
+	reg_val = snd_soc_read(codec, e->reg);
 	val = (reg_val >> e->shift_l) & e->mask;
 	for (mux = 0; mux < e->max; mux++) {
 		if (val == e->values[mux])
@@ -2942,15 +3061,12 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_value_enum_double);
 int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
-	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
-	struct snd_soc_codec *codec = widget->codec;
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
 	struct snd_soc_card *card = codec->card;
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int val, mux, change;
 	unsigned int mask;
 	struct snd_soc_dapm_update update;
-	int wi;
 
 	if (ucontrol->value.enumerated.item[0] > e->max - 1)
 		return -EINVAL;
@@ -2966,24 +3082,17 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
 
 	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 
-	change = snd_soc_test_bits(widget->codec, e->reg, mask, val);
+	change = snd_soc_test_bits(codec, e->reg, mask, val);
 	if (change) {
-		for (wi = 0; wi < wlist->num_widgets; wi++) {
-			widget = wlist->widgets[wi];
-
-			widget->value = val;
+		update.kcontrol = kcontrol;
+		update.reg = e->reg;
+		update.mask = mask;
+		update.val = val;
+		card->update = &update;
 
-			update.kcontrol = kcontrol;
-			update.widget = widget;
-			update.reg = e->reg;
-			update.mask = mask;
-			update.val = val;
-			widget->dapm->update = &update;
+		soc_dapm_mux_update_power(card, kcontrol, mux, e);
 
-			soc_dapm_mux_update_power(widget, kcontrol, mux, e);
-
-			widget->dapm->update = NULL;
-		}
+		card->update = NULL;
 	}
 
 	mutex_unlock(&card->dapm_mutex);
@@ -3080,7 +3189,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 			return NULL;
 		}
 
-		if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) {
+		if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) {
 			ret = regulator_allow_bypass(w->regulator, true);
 			if (ret != 0)
 				dev_warn(w->dapm->dev,
@@ -3127,16 +3236,16 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 	case snd_soc_dapm_value_mux:
 		w->power_check = dapm_generic_check_power;
 		break;
-	case snd_soc_dapm_adc:
-	case snd_soc_dapm_aif_out:
 	case snd_soc_dapm_dai_out:
 		w->power_check = dapm_adc_check_power;
 		break;
-	case snd_soc_dapm_dac:
-	case snd_soc_dapm_aif_in:
 	case snd_soc_dapm_dai_in:
 		w->power_check = dapm_dac_check_power;
 		break;
+	case snd_soc_dapm_adc:
+	case snd_soc_dapm_aif_out:
+	case snd_soc_dapm_dac:
+	case snd_soc_dapm_aif_in:
 	case snd_soc_dapm_pga:
 	case snd_soc_dapm_out_drv:
 	case snd_soc_dapm_input:
@@ -3152,6 +3261,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
 	case snd_soc_dapm_clock_supply:
+	case snd_soc_dapm_kcontrol:
 		w->power_check = dapm_supply_check_power;
 		break;
 	default:
@@ -3416,9 +3526,6 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
 {
 	struct snd_soc_dapm_widget *dai_w, *w;
 	struct snd_soc_dai *dai;
-	struct snd_soc_dapm_route r;
-
-	memset(&r, 0, sizeof(r));
 
 	/* For each DAI widget... */
 	list_for_each_entry(dai_w, &card->widgets, list) {
@@ -3445,29 +3552,27 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
 				break;
 			}
 
-			if (!w->sname)
+			if (!w->sname || !strstr(w->sname, dai_w->name))
 				continue;
 
 			if (dai->driver->playback.stream_name &&
 			    strstr(w->sname,
 				   dai->driver->playback.stream_name)) {
-				r.source = dai->playback_widget->name;
-				r.sink = w->name;
 				dev_dbg(dai->dev, "%s -> %s\n",
-					 r.source, r.sink);
+					 dai->playback_widget->name, w->name);
 
-				snd_soc_dapm_add_route(w->dapm, &r);
+				snd_soc_dapm_add_path(w->dapm,
+					dai->playback_widget, w, NULL, NULL);
 			}
 
 			if (dai->driver->capture.stream_name &&
 			    strstr(w->sname,
 				   dai->driver->capture.stream_name)) {
-				r.source = w->name;
-				r.sink = dai->capture_widget->name;
 				dev_dbg(dai->dev, "%s -> %s\n",
-					r.source, r.sink);
+					w->name, dai->capture_widget->name);
 
-				snd_soc_dapm_add_route(w->dapm, &r);
+				snd_soc_dapm_add_path(w->dapm, w,
+					dai->capture_widget, NULL, NULL);
 			}
 		}
 	}
@@ -3529,7 +3634,7 @@ static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
 		}
 	}
 
-	dapm_power_widgets(&rtd->card->dapm, event);
+	dapm_power_widgets(rtd->card, event);
 }
 
 /**
@@ -3798,7 +3903,7 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
 		if (dapm->bias_level == SND_SOC_BIAS_ON)
 			snd_soc_dapm_set_bias_level(dapm,
 						    SND_SOC_BIAS_PREPARE);
-		dapm_seq_run(dapm, &down_list, 0, false);
+		dapm_seq_run(card, &down_list, 0, false);
 		if (dapm->bias_level == SND_SOC_BIAS_PREPARE)
 			snd_soc_dapm_set_bias_level(dapm,
 						    SND_SOC_BIAS_STANDBY);
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 0bb5cccd7766..71358e3b54d9 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -183,8 +183,6 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
 		list_add(&(pins[i].list), &jack->pins);
 	}
 
-	snd_soc_dapm_new_widgets(&jack->codec->card->dapm);
-
 	/* Update to reflect the last reported status; canned jack
 	 * implementations are likely to set their state before the
 	 * card has an opportunity to associate pins.
@@ -263,7 +261,7 @@ static irqreturn_t gpio_handler(int irq, void *data)
 	if (device_may_wakeup(dev))
 		pm_wakeup_event(dev, gpio->debounce_time + 50);
 
-	schedule_delayed_work(&gpio->work,
+	queue_delayed_work(system_power_efficient_wq, &gpio->work,
 			      msecs_to_jiffies(gpio->debounce_time));
 
 	return IRQ_HANDLED;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index b6c640332a17..330c9a6b5cb5 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -411,8 +411,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 		} else {
 			/* start delayed pop wq here for playback streams */
 			rtd->pop_wait = 1;
-			schedule_delayed_work(&rtd->delayed_work,
-				msecs_to_jiffies(rtd->pmdown_time));
+			queue_delayed_work(system_power_efficient_wq,
+					   &rtd->delayed_work,
+					   msecs_to_jiffies(rtd->pmdown_time));
 		}
 	} else {
 		/* capture streams can be powered down now */
@@ -1832,18 +1833,10 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream)
 /* Called by DAPM mixer/mux changes to update audio routing between PCMs and
  * any DAI links.
  */
-int soc_dpcm_runtime_update(struct snd_soc_dapm_widget *widget)
+int soc_dpcm_runtime_update(struct snd_soc_card *card)
 {
-	struct snd_soc_card *card;
 	int i, old, new, paths;
 
-	if (widget->codec)
-		card = widget->codec->card;
-	else if (widget->platform)
-		card = widget->platform->card;
-	else
-		return -EINVAL;
-
 	mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
 	for (i = 0; i < card->num_rtd; i++) {
 		struct snd_soc_dapm_widget_list *list;
@@ -2027,6 +2020,16 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 			capture = 1;
 	}
 
+	if (rtd->dai_link->playback_only) {
+		playback = 1;
+		capture = 0;
+	}
+
+	if (rtd->dai_link->capture_only) {
+		playback = 0;
+		capture = 1;
+	}
+
 	/* create the PCM */
 	if (rtd->dai_link->no_pcm) {
 		snprintf(new_name, sizeof(new_name), "(%s)",
diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig
index 3567d73b218e..0a53053495f3 100644
--- a/sound/soc/spear/Kconfig
+++ b/sound/soc/spear/Kconfig
@@ -1,6 +1,6 @@
 config SND_SPEAR_SOC
 	tristate
-	select SND_SOC_DMAENGINE_PCM
+	select SND_DMAENGINE_PCM
 
 config SND_SPEAR_SPDIF_OUT
 	tristate
diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index 995b120c2cd0..8fc653ca3ab4 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -1,8 +1,8 @@
 config SND_SOC_TEGRA
 	tristate "SoC Audio for the Tegra System-on-Chip"
-	depends on ARCH_TEGRA && TEGRA20_APB_DMA
+	depends on (ARCH_TEGRA && TEGRA20_APB_DMA) || COMPILE_TEST
 	select REGMAP_MMIO
-	select SND_SOC_GENERIC_DMAENGINE_PCM if TEGRA20_APB_DMA
+	select SND_SOC_GENERIC_DMAENGINE_PCM
 	help
 	  Say Y or M here if you want support for SoC audio on Tegra.
 
@@ -61,7 +61,7 @@ config SND_SOC_TEGRA30_I2S
 
 config SND_SOC_TEGRA_RT5640
 	tristate "SoC Audio support for Tegra boards using an RT5640 codec"
-	depends on SND_SOC_TEGRA && I2C
+	depends on SND_SOC_TEGRA && I2C && GPIOLIB
 	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
 	select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
 	select SND_SOC_RT5640
@@ -71,7 +71,7 @@ config SND_SOC_TEGRA_RT5640
 
 config SND_SOC_TEGRA_WM8753
 	tristate "SoC Audio support for Tegra boards using a WM8753 codec"
-	depends on SND_SOC_TEGRA && I2C
+	depends on SND_SOC_TEGRA && I2C && GPIOLIB
 	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
 	select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
 	select SND_SOC_WM8753
@@ -81,7 +81,7 @@ config SND_SOC_TEGRA_WM8753
 
 config SND_SOC_TEGRA_WM8903
 	tristate "SoC Audio support for Tegra boards using a WM8903 codec"
-	depends on SND_SOC_TEGRA && I2C
+	depends on SND_SOC_TEGRA && I2C && GPIOLIB
 	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
 	select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
 	select SND_SOC_WM8903
@@ -92,7 +92,7 @@ config SND_SOC_TEGRA_WM8903
 
 config SND_SOC_TEGRA_WM9712
 	tristate "SoC Audio support for Tegra boards using a WM9712 codec"
-	depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC
+	depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC && GPIOLIB
 	select SND_SOC_TEGRA20_AC97
 	select SND_SOC_WM9712
 	help
@@ -110,7 +110,7 @@ config SND_SOC_TEGRA_TRIMSLICE
 
 config SND_SOC_TEGRA_ALC5632
 	tristate "SoC Audio support for Tegra boards using an ALC5632 codec"
-	depends on SND_SOC_TEGRA && I2C
+	depends on SND_SOC_TEGRA && I2C && GPIOLIB
 	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
 	select SND_SOC_ALC5632
 	help
diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c
index 6c486625321b..ae27bcd586d2 100644
--- a/sound/soc/tegra/tegra20_ac97.c
+++ b/sound/soc/tegra/tegra20_ac97.c
@@ -334,12 +334,6 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev)
 	}
 
 	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!mem) {
-		dev_err(&pdev->dev, "No memory resource\n");
-		ret = -ENODEV;
-		goto err_clk_put;
-	}
-
 	regs = devm_ioremap_resource(&pdev->dev, mem);
 	if (IS_ERR(regs)) {
 		ret = PTR_ERR(regs);
@@ -432,8 +426,6 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev)
 
 	return 0;
 
-err_unregister_pcm:
-	tegra_pcm_platform_unregister(&pdev->dev);
 err_unregister_component:
 	snd_soc_unregister_component(&pdev->dev);
 err_asoc_utils_fini:
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 48d05d9e1002..c61ea3a1030f 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -13,8 +13,6 @@
  * published by the Free Software Foundation.
  */
 
-#include <asm/mach-types.h>
-
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c
index 08794f915a94..4511c5a875ec 100644
--- a/sound/soc/tegra/tegra_rt5640.c
+++ b/sound/soc/tegra/tegra_rt5640.c
@@ -99,6 +99,7 @@ static struct snd_soc_jack_gpio tegra_rt5640_hp_jack_gpio = {
 static const struct snd_soc_dapm_widget tegra_rt5640_dapm_widgets[] = {
 	SND_SOC_DAPM_HP("Headphones", NULL),
 	SND_SOC_DAPM_SPK("Speakers", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
 };
 
 static const struct snd_kcontrol_new tegra_rt5640_controls[] = {
diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c
index f87fc53e9b8c..8e774d1a243c 100644
--- a/sound/soc/tegra/tegra_wm8753.c
+++ b/sound/soc/tegra/tegra_wm8753.c
@@ -28,8 +28,6 @@
  *
  */
 
-#include <asm/mach-types.h>
-
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index 05c68aab5cf0..734bfcd21148 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -24,8 +24,6 @@
  *
  */
 
-#include <asm/mach-types.h>
-
 #include <linux/module.h>
 #include <linux/of.h>
 #include <linux/platform_device.h>
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
index 4bcce8a3cded..e0305a148568 100644
--- a/sound/soc/txx9/txx9aclc-ac97.c
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -184,9 +184,6 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev)
 	if (irq < 0)
 		return irq;
 	r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!r)
-		return -EBUSY;
-
 	drvdata->base = devm_ioremap_resource(&pdev->dev, r);
 	if (IS_ERR(drvdata->base))
 		return PTR_ERR(drvdata->base);
diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c
index 8f5cd00a6e46..178d1bad6259 100644
--- a/sound/soc/ux500/mop500.c
+++ b/sound/soc/ux500/mop500.c
@@ -52,6 +52,7 @@ static struct snd_soc_dai_link mop500_dai_links[] = {
 
 static struct snd_soc_card mop500_card = {
 	.name = "MOP500-card",
+	.owner = THIS_MODULE,
 	.probe = NULL,
 	.dai_link = mop500_dai_links,
 	.num_links = ARRAY_SIZE(mop500_dai_links),
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index b9defcdeb7ef..780bf3f62d28 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -346,10 +346,10 @@ static int usb6fire_fw_check(u8 *version)
 		if (!memcmp(version, known_fw_versions + i, 2))
 			return 0;
 
-	snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. "
+	snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %4ph. "
 			"please reconnect to power. if this failure "
 			"still happens, check your firmware installation.",
-			4, version);
+			version);
 	return -EINVAL;
 }
 
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 659950e5b94f..93e970f2b3c0 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -418,6 +418,9 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
 	struct snd_usb_endpoint *ep;
 	int is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK;
 
+	if (WARN_ON(!alts))
+		return NULL;
+
 	mutex_lock(&chip->mutex);
 
 	list_for_each_entry(ep, &chip->ep_list, list) {
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 15b151ed4899..b375d58871e7 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -327,6 +327,137 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum,
 	return 0;
 }
 
+static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
+					 struct usb_device *dev,
+					 struct usb_interface_descriptor *altsd,
+					 unsigned int attr)
+{
+	struct usb_host_interface *alts;
+	struct usb_interface *iface;
+	unsigned int ep;
+
+	/* Implicit feedback sync EPs consumers are always playback EPs */
+	if (subs->direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
+	switch (subs->stream->chip->usb_id) {
+	case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
+	case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
+		ep = 0x81;
+		iface = usb_ifnum_to_if(dev, 3);
+
+		if (!iface || iface->num_altsetting == 0)
+			return -EINVAL;
+
+		alts = &iface->altsetting[1];
+		goto add_sync_ep;
+		break;
+	case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
+	case USB_ID(0x0763, 0x2081):
+		ep = 0x81;
+		iface = usb_ifnum_to_if(dev, 2);
+
+		if (!iface || iface->num_altsetting == 0)
+			return -EINVAL;
+
+		alts = &iface->altsetting[1];
+		goto add_sync_ep;
+	}
+	if (attr == USB_ENDPOINT_SYNC_ASYNC &&
+	    altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
+	    altsd->bInterfaceProtocol == 2 &&
+	    altsd->bNumEndpoints == 1 &&
+	    USB_ID_VENDOR(subs->stream->chip->usb_id) == 0x0582 /* Roland */ &&
+	    search_roland_implicit_fb(dev, altsd->bInterfaceNumber + 1,
+				      altsd->bAlternateSetting,
+				      &alts, &ep) >= 0) {
+		goto add_sync_ep;
+	}
+
+	/* No quirk */
+	return 0;
+
+add_sync_ep:
+	subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
+						   alts, ep, !subs->direction,
+						   SND_USB_ENDPOINT_TYPE_DATA);
+	if (!subs->sync_endpoint)
+		return -EINVAL;
+
+	subs->data_endpoint->sync_master = subs->sync_endpoint;
+
+	return 0;
+}
+
+static int set_sync_endpoint(struct snd_usb_substream *subs,
+			     struct audioformat *fmt,
+			     struct usb_device *dev,
+			     struct usb_host_interface *alts,
+			     struct usb_interface_descriptor *altsd)
+{
+	int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
+	unsigned int ep, attr;
+	bool implicit_fb;
+	int err;
+
+	/* we need a sync pipe in async OUT or adaptive IN mode */
+	/* check the number of EP, since some devices have broken
+	 * descriptors which fool us.  if it has only one EP,
+	 * assume it as adaptive-out or sync-in.
+	 */
+	attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE;
+
+	err = set_sync_ep_implicit_fb_quirk(subs, dev, altsd, attr);
+	if (err < 0)
+		return err;
+
+	if (altsd->bNumEndpoints < 2)
+		return 0;
+
+	if ((is_playback && attr != USB_ENDPOINT_SYNC_ASYNC) ||
+	    (!is_playback && attr != USB_ENDPOINT_SYNC_ADAPTIVE))
+		return 0;
+
+	/* check sync-pipe endpoint */
+	/* ... and check descriptor size before accessing bSynchAddress
+	   because there is a version of the SB Audigy 2 NX firmware lacking
+	   the audio fields in the endpoint descriptors */
+	if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC ||
+	    (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+	     get_endpoint(alts, 1)->bSynchAddress != 0)) {
+		snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n",
+			   dev->devnum, fmt->iface, fmt->altsetting,
+			   get_endpoint(alts, 1)->bmAttributes,
+			   get_endpoint(alts, 1)->bLength,
+			   get_endpoint(alts, 1)->bSynchAddress);
+		return -EINVAL;
+	}
+	ep = get_endpoint(alts, 1)->bEndpointAddress;
+	if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+	    ((is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
+	     (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
+		snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n",
+			   dev->devnum, fmt->iface, fmt->altsetting,
+			   is_playback, ep, get_endpoint(alts, 0)->bSynchAddress);
+		return -EINVAL;
+	}
+
+	implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK)
+			== USB_ENDPOINT_USAGE_IMPLICIT_FB;
+
+	subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
+						   alts, ep, !subs->direction,
+						   implicit_fb ?
+							SND_USB_ENDPOINT_TYPE_DATA :
+							SND_USB_ENDPOINT_TYPE_SYNC);
+	if (!subs->sync_endpoint)
+		return -EINVAL;
+
+	subs->data_endpoint->sync_master = subs->sync_endpoint;
+
+	return 0;
+}
+
 /*
  * find a matching format and set up the interface
  */
@@ -336,9 +467,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 	struct usb_host_interface *alts;
 	struct usb_interface_descriptor *altsd;
 	struct usb_interface *iface;
-	unsigned int ep, attr;
-	int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
-	int err, implicit_fb = 0;
+	int err;
 
 	iface = usb_ifnum_to_if(dev, fmt->iface);
 	if (WARN_ON(!iface))
@@ -383,118 +512,22 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 	subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip,
 						   alts, fmt->endpoint, subs->direction,
 						   SND_USB_ENDPOINT_TYPE_DATA);
+
 	if (!subs->data_endpoint)
 		return -EINVAL;
 
-	/* we need a sync pipe in async OUT or adaptive IN mode */
-	/* check the number of EP, since some devices have broken
-	 * descriptors which fool us.  if it has only one EP,
-	 * assume it as adaptive-out or sync-in.
-	 */
-	attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE;
-
-	switch (subs->stream->chip->usb_id) {
-	case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
-	case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
-		if (is_playback) {
-			implicit_fb = 1;
-			ep = 0x81;
-			iface = usb_ifnum_to_if(dev, 3);
-
-			if (!iface || iface->num_altsetting == 0)
-				return -EINVAL;
-
-			alts = &iface->altsetting[1];
-			goto add_sync_ep;
-		}
-		break;
-	case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
-	case USB_ID(0x0763, 0x2081):
-		if (is_playback) {
-			implicit_fb = 1;
-			ep = 0x81;
-			iface = usb_ifnum_to_if(dev, 2);
-
-			if (!iface || iface->num_altsetting == 0)
-				return -EINVAL;
-
-			alts = &iface->altsetting[1];
-			goto add_sync_ep;
-		}
-	}
-	if (is_playback &&
-	    attr == USB_ENDPOINT_SYNC_ASYNC &&
-	    altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
-	    altsd->bInterfaceProtocol == 2 &&
-	    altsd->bNumEndpoints == 1 &&
-	    USB_ID_VENDOR(subs->stream->chip->usb_id) == 0x0582 /* Roland */ &&
-	    search_roland_implicit_fb(dev, altsd->bInterfaceNumber + 1,
-				      altsd->bAlternateSetting,
-				      &alts, &ep) >= 0) {
-		implicit_fb = 1;
-		goto add_sync_ep;
-	}
-
-	if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) ||
-	     (!is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) &&
-	    altsd->bNumEndpoints >= 2) {
-		/* check sync-pipe endpoint */
-		/* ... and check descriptor size before accessing bSynchAddress
-		   because there is a version of the SB Audigy 2 NX firmware lacking
-		   the audio fields in the endpoint descriptors */
-		if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC ||
-		    (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
-		     get_endpoint(alts, 1)->bSynchAddress != 0 &&
-		     !implicit_fb)) {
-			snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n",
-				   dev->devnum, fmt->iface, fmt->altsetting,
-				   get_endpoint(alts, 1)->bmAttributes,
-				   get_endpoint(alts, 1)->bLength,
-				   get_endpoint(alts, 1)->bSynchAddress);
-			return -EINVAL;
-		}
-		ep = get_endpoint(alts, 1)->bEndpointAddress;
-		if (!implicit_fb &&
-		    get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
-		    (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
-		     (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
-			snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n",
-				   dev->devnum, fmt->iface, fmt->altsetting,
-				   is_playback, ep, get_endpoint(alts, 0)->bSynchAddress);
-			return -EINVAL;
-		}
-
-		implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK)
-				== USB_ENDPOINT_USAGE_IMPLICIT_FB;
-
-add_sync_ep:
-		subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
-							   alts, ep, !subs->direction,
-							   implicit_fb ?
-								SND_USB_ENDPOINT_TYPE_DATA :
-								SND_USB_ENDPOINT_TYPE_SYNC);
-		if (!subs->sync_endpoint)
-			return -EINVAL;
-
-		subs->data_endpoint->sync_master = subs->sync_endpoint;
-	}
+	err = set_sync_endpoint(subs, fmt, dev, alts, altsd);
+	if (err < 0)
+		return err;
 
-	if ((err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt)) < 0)
+	err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt);
+	if (err < 0)
 		return err;
 
 	subs->cur_audiofmt = fmt;
 
 	snd_usb_set_format_quirk(subs, fmt);
 
-#if 0
-	printk(KERN_DEBUG
-	       "setting done: format = %d, rate = %d..%d, channels = %d\n",
-	       fmt->format, fmt->rate_min, fmt->rate_max, fmt->channels);
-	printk(KERN_DEBUG
-	       "  datapipe = 0x%0x, syncpipe = 0x%0x\n",
-	       subs->datapipe, subs->syncpipe);
-#endif
-
 	return 0;
 }
 
diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c
index 1f9bbd55553f..5a51b18c50fe 100644
--- a/sound/usb/usx2y/usbusx2y.c
+++ b/sound/usb/usx2y/usbusx2y.c
@@ -305,11 +305,9 @@ static void usX2Y_unlinkSeq(struct snd_usX2Y_AsyncSeq *S)
 {
 	int	i;
 	for (i = 0; i < URBS_AsyncSeq; ++i) {
-		if (S[i].urb) {
-			usb_kill_urb(S->urb[i]);
-			usb_free_urb(S->urb[i]);
-			S->urb[i] = NULL;
-		}
+		usb_kill_urb(S->urb[i]);
+		usb_free_urb(S->urb[i]);
+		S->urb[i] = NULL;
 	}
 	kfree(S->buffer);
 }
author	Linus Torvalds <torvalds@linux-foundation.org>	2013-09-04 16:26:56 -0700
committer	Linus Torvalds <torvalds@linux-foundation.org>	2013-09-04 16:26:56 -0700
commit	977dbfcf8e9ff1783355b260d93101af315de18a (patch)
tree	b586ca678499d1ccc2d199a97d65996c630b25d8 /sound
parent	aa7054f5a5a9ff728ce291cb103afa19f4f849eb (diff)
parent	b054087dbacee30a9dddaef2c9a96312146be04e (diff)
download	linux-977dbfcf8e9ff1783355b260d93101af315de18a.tar.bz2