Merge branch 'for-next' into for-linus

262 files changed, 20305 insertions, 2841 deletions

diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 38590b322c54..fbd5dad0c484 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -15,6 +15,7 @@
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/dmaengine.h>
+#include <linux/dma/pxa-dma.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -43,7 +44,11 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
 	.reset	= pxa2xx_ac97_reset,
 };
 
-static unsigned long pxa2xx_ac97_pcm_out_req = 12;
+static struct pxad_param pxa2xx_ac97_pcm_out_req = {
+	.prio = PXAD_PRIO_LOWEST,
+	.drcmr = 12,
+};
+
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = {
 	.addr		= __PREG(PCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
@@ -51,7 +56,11 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = {
 	.filter_data	= &pxa2xx_ac97_pcm_out_req,
 };
 
-static unsigned long pxa2xx_ac97_pcm_in_req = 11;
+static struct pxad_param pxa2xx_ac97_pcm_in_req = {
+	.prio = PXAD_PRIO_LOWEST,
+	.drcmr = 11,
+};
+
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = {
 	.addr		= __PREG(PCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 01f8fdc42b1b..e9b98af6b52c 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -8,6 +8,7 @@
 #include <linux/module.h>
 #include <linux/dma-mapping.h>
 #include <linux/dmaengine.h>
+#include <linux/dma/pxa-dma.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -15,8 +16,6 @@
 #include <sound/pxa2xx-lib.h>
 #include <sound/dmaengine_pcm.h>
 
-#include <mach/dma.h>
-
 #include "pxa2xx-pcm.h"
 
 static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
@@ -31,7 +30,7 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 8192 - 32,
 	.periods_min		= 1,
-	.periods_max		= PAGE_SIZE/sizeof(pxa_dma_desc),
+	.periods_max		= 256,
 	.buffer_bytes_max	= 128 * 1024,
 	.fifo_size		= 32,
 };
@@ -39,65 +38,29 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
 int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct pxa2xx_runtime_data *rtd = runtime->private_data;
-	size_t totsize = params_buffer_bytes(params);
-	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;
+	struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_dmaengine_dai_dma_data *dma_params;
+	struct dma_slave_config config;
+	int ret;
 
-	/* 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;
-	}
+	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	if (!dma_params)
+		return 0;
 
-	switch (rtd->params->maxburst) {
-	case 8:
-		dcmd |= DCMD_BURST8;
-		break;
-	case 16:
-		dcmd |= DCMD_BURST16;
-		break;
-	case 32:
-		dcmd |= DCMD_BURST32;
-		break;
-	}
+	ret = snd_hwparams_to_dma_slave_config(substream, params, &config);
+	if (ret)
+		return ret;
 
-	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-	runtime->dma_bytes = totsize;
+	snd_dmaengine_pcm_set_config_from_dai_data(substream,
+			snd_soc_dai_get_dma_data(rtd->cpu_dai, substream),
+			&config);
 
-	dma_desc = rtd->dma_desc_array;
-	next_desc_phys = rtd->dma_desc_array_phys;
-	dma_buff_phys = runtime->dma_addr;
-	do {
-		next_desc_phys += sizeof(pxa_dma_desc);
-		dma_desc->ddadr = next_desc_phys;
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			dma_desc->dsadr = dma_buff_phys;
-			dma_desc->dtadr = rtd->params->addr;
-		} else {
-			dma_desc->dsadr = rtd->params->addr;
-			dma_desc->dtadr = dma_buff_phys;
-		}
-		if (period > totsize)
-			period = totsize;
-		dma_desc->dcmd = dcmd | period | DCMD_ENDIRQEN;
-		dma_desc++;
-		dma_buff_phys += period;
-	} while (totsize -= period);
-	dma_desc[-1].ddadr = rtd->dma_desc_array_phys;
+	ret = dmaengine_slave_config(chan, &config);
+	if (ret)
+		return ret;
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
 
 	return 0;
 }
@@ -105,13 +68,6 @@ EXPORT_SYMBOL(__pxa2xx_pcm_hw_params);
 
 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->filter_data) {
-		unsigned long req = *(unsigned long *) rtd->params->filter_data;
-		DRCMR(req) = 0;
-	}
-
 	snd_pcm_set_runtime_buffer(substream, NULL);
 	return 0;
 }
@@ -119,100 +75,36 @@ EXPORT_SYMBOL(__pxa2xx_pcm_hw_free);
 
 int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 {
-	struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
-	int ret = 0;
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-		DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys;
-		DCSR(prtd->dma_ch) = DCSR_RUN;
-		break;
-
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		DCSR(prtd->dma_ch) &= ~DCSR_RUN;
-		break;
-
-	case SNDRV_PCM_TRIGGER_RESUME:
-		DCSR(prtd->dma_ch) |= DCSR_RUN;
-		break;
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys;
-		DCSR(prtd->dma_ch) |= DCSR_RUN;
-		break;
-
-	default:
-		ret = -EINVAL;
-	}
-
-	return ret;
+	return snd_dmaengine_pcm_trigger(substream, cmd);
 }
 EXPORT_SYMBOL(pxa2xx_pcm_trigger);
 
 snd_pcm_uframes_t
 pxa2xx_pcm_pointer(struct snd_pcm_substream *substream)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct pxa2xx_runtime_data *prtd = runtime->private_data;
-
-	dma_addr_t ptr = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
-			 DSADR(prtd->dma_ch) : DTADR(prtd->dma_ch);
-	snd_pcm_uframes_t x = bytes_to_frames(runtime, ptr - runtime->dma_addr);
-
-	if (x == runtime->buffer_size)
-		x = 0;
-	return x;
+	return snd_dmaengine_pcm_pointer(substream);
 }
 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;
-
-	if (prtd->dma_ch == -1)
-		return -EINVAL;
-
-	DCSR(prtd->dma_ch) &= ~DCSR_RUN;
-	DCSR(prtd->dma_ch) = 0;
-	DCMD(prtd->dma_ch) = 0;
-	req = *(unsigned long *) prtd->params->filter_data;
-	DRCMR(req) = prtd->dma_ch | DRCMR_MAPVLD;
-
 	return 0;
 }
 EXPORT_SYMBOL(__pxa2xx_pcm_prepare);
 
-void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id)
-{
-	struct snd_pcm_substream *substream = dev_id;
-	int dcsr;
-
-	dcsr = DCSR(dma_ch);
-	DCSR(dma_ch) = dcsr & ~DCSR_STOPIRQEN;
-
-	if (dcsr & DCSR_ENDINTR) {
-		snd_pcm_period_elapsed(substream);
-	} else {
-		printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n",
-			dma_ch, dcsr);
-		snd_pcm_stop_xrun(substream);
-	}
-}
-EXPORT_SYMBOL(pxa2xx_pcm_dma_irq);
-
 int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
 {
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct pxa2xx_runtime_data *rtd;
+	struct snd_dmaengine_dai_dma_data *dma_params;
 	int ret;
 
 	runtime->hw = pxa2xx_pcm_hardware;
 
+	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	if (!dma_params)
+		return 0;
+
 	/*
 	 * For mysterious reasons (and despite what the manual says)
 	 * playback samples are lost if the DMA count is not a multiple
@@ -221,48 +113,27 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
 	ret = snd_pcm_hw_constraint_step(runtime, 0,
 		SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
 	if (ret)
-		goto out;
+		return ret;
 
 	ret = snd_pcm_hw_constraint_step(runtime, 0,
 		SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
 	if (ret)
-		goto out;
+		return ret;
 
 	ret = snd_pcm_hw_constraint_integer(runtime,
 					    SNDRV_PCM_HW_PARAM_PERIODS);
 	if (ret < 0)
-		goto out;
-
-	ret = -ENOMEM;
-	rtd = kzalloc(sizeof(*rtd), GFP_KERNEL);
-	if (!rtd)
-		goto out;
-	rtd->dma_desc_array =
-		dma_alloc_writecombine(substream->pcm->card->dev, PAGE_SIZE,
-				       &rtd->dma_desc_array_phys, GFP_KERNEL);
-	if (!rtd->dma_desc_array)
-		goto err1;
+		return ret;
 
-	rtd->dma_ch = -1;
-	runtime->private_data = rtd;
-	return 0;
-
- err1:
-	kfree(rtd);
- out:
-	return ret;
+	return snd_dmaengine_pcm_open_request_chan(substream,
+					pxad_filter_fn,
+					dma_params->filter_data);
 }
 EXPORT_SYMBOL(__pxa2xx_pcm_open);
 
 int __pxa2xx_pcm_close(struct snd_pcm_substream *substream)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct pxa2xx_runtime_data *rtd = runtime->private_data;
-
-	dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE,
-			      rtd->dma_desc_array, rtd->dma_desc_array_phys);
-	kfree(rtd);
-	return 0;
+	return snd_dmaengine_pcm_close_release_chan(substream);
 }
 EXPORT_SYMBOL(__pxa2xx_pcm_close);
 
diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c
index 83be8e3f095e..83fcfac97739 100644
--- a/sound/arm/pxa2xx-pcm.c
+++ b/sound/arm/pxa2xx-pcm.c
@@ -46,17 +46,13 @@ 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("dma", DMA_PRIO_LOW,
-			      pxa2xx_pcm_dma_irq, substream);
-	if (ret < 0)
-		goto err2;
-	rtd->dma_ch = ret;
 
 	ret = client->startup(substream);
 	if (!ret)
-		goto out;
+		goto err2;
+
+	return 0;
 
-	pxa_free_dma(rtd->dma_ch);
  err2:
 	__pxa2xx_pcm_close(substream);
  out:
@@ -66,9 +62,7 @@ static int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
 static int pxa2xx_pcm_close(struct snd_pcm_substream *substream)
 {
 	struct pxa2xx_pcm_client *client = substream->private_data;
-	struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
 
-	pxa_free_dma(rtd->dma_ch);
 	client->shutdown(substream);
 
 	return __pxa2xx_pcm_close(substream);
diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h
index 00330985beec..8fa2b7c9e6b8 100644
--- a/sound/arm/pxa2xx-pcm.h
+++ b/sound/arm/pxa2xx-pcm.h
@@ -13,8 +13,6 @@
 struct pxa2xx_runtime_data {
 	int dma_ch;
 	struct snd_dmaengine_dai_dma_data *params;
-	struct pxa_dma_desc *dma_desc_array;
-	dma_addr_t dma_desc_array_phys;
 };
 
 struct pxa2xx_pcm_client {
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index 6c96feeaf01e..e3e949126a56 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -4,7 +4,7 @@ config SND_TIMER
 
 config SND_PCM
 	tristate
-	select SND_TIMER
+	select SND_TIMER if SND_PCM_TIMER
 
 config SND_PCM_ELD
 	bool
@@ -93,6 +93,17 @@ config SND_PCM_OSS_PLUGINS
           support conversion of channels, formats and rates. It will
           behave like most of new OSS/Free drivers in 2.4/2.6 kernels.
 
+config SND_PCM_TIMER
+	bool "PCM timer interface" if EXPERT
+	default y
+	help
+	  If you disable this option, pcm timer will be inavailable, so
+	  those stubs used pcm timer (e.g. dmix, dsnoop & co) may work
+	  incorrectlly.
+
+	  For some embedded device, we may disable it to reduce memory
+	  footprint, about 20KB on x86_64 platform.
+
 config SND_SEQUENCER_OSS
 	bool "OSS Sequencer API"
 	depends on SND_SEQUENCER
diff --git a/sound/core/Makefile b/sound/core/Makefile
index 3354f91e003a..48ab4b8f8279 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -13,8 +13,9 @@ snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o
 snd-$(CONFIG_SND_VMASTER) += vmaster.o
 snd-$(CONFIG_SND_JACK)	  += ctljack.o jack.o
 
-snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \
+snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_misc.o \
 		pcm_memory.o memalloc.o
+snd-pcm-$(CONFIG_SND_PCM_TIMER) += pcm_timer.o
 snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o
 snd-pcm-$(CONFIG_SND_PCM_ELD) += pcm_drm_eld.o
 snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o
diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
index a99f7200ff3f..7a8c79dd9734 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -1177,7 +1177,8 @@ static void snd_mixer_oss_proc_write(struct snd_info_entry *entry,
 	struct snd_mixer_oss *mixer = entry->private_data;
 	char line[128], str[32], idxstr[16];
 	const char *cptr;
-	int ch, idx;
+	unsigned int idx;
+	int ch;
 	struct snd_mixer_oss_assign_table *tbl;
 	struct slot *slot;
 
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 02bd96954dc4..308c9ecf73db 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -1014,9 +1014,6 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream)
 	snd_free_pages((void*)runtime->control,
 		       PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control)));
 	kfree(runtime->hw_constraints.rules);
-#ifdef CONFIG_SND_PCM_XRUN_DEBUG
-	kfree(runtime->hwptr_log);
-#endif
 	kfree(runtime);
 	substream->runtime = NULL;
 	put_pid(substream->pid);
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 7d45645f10ba..6b5a811e01a5 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -801,7 +801,7 @@ void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k,
  * negative error code.
  */
 int snd_interval_ratnum(struct snd_interval *i,
-			unsigned int rats_count, struct snd_ratnum *rats,
+			unsigned int rats_count, const struct snd_ratnum *rats,
 			unsigned int *nump, unsigned int *denp)
 {
 	unsigned int best_num, best_den;
@@ -920,7 +920,8 @@ EXPORT_SYMBOL(snd_interval_ratnum);
  * negative error code.
  */
 static int snd_interval_ratden(struct snd_interval *i,
-			       unsigned int rats_count, struct snd_ratden *rats,
+			       unsigned int rats_count,
+			       const struct snd_ratden *rats,
 			       unsigned int *nump, unsigned int *denp)
 {
 	unsigned int best_num, best_diff, best_den;
@@ -1339,7 +1340,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_ranges);
 static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params,
 				   struct snd_pcm_hw_rule *rule)
 {
-	struct snd_pcm_hw_constraint_ratnums *r = rule->private;
+	const struct snd_pcm_hw_constraint_ratnums *r = rule->private;
 	unsigned int num = 0, den = 0;
 	int err;
 	err = snd_interval_ratnum(hw_param_interval(params, rule->var),
@@ -1363,10 +1364,10 @@ static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params,
 int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, 
 				  unsigned int cond,
 				  snd_pcm_hw_param_t var,
-				  struct snd_pcm_hw_constraint_ratnums *r)
+				  const struct snd_pcm_hw_constraint_ratnums *r)
 {
 	return snd_pcm_hw_rule_add(runtime, cond, var,
-				   snd_pcm_hw_rule_ratnums, r,
+				   snd_pcm_hw_rule_ratnums, (void *)r,
 				   var, -1);
 }
 
@@ -1375,7 +1376,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_ratnums);
 static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params,
 				   struct snd_pcm_hw_rule *rule)
 {
-	struct snd_pcm_hw_constraint_ratdens *r = rule->private;
+	const struct snd_pcm_hw_constraint_ratdens *r = rule->private;
 	unsigned int num = 0, den = 0;
 	int err = snd_interval_ratden(hw_param_interval(params, rule->var),
 				  r->nrats, r->rats, &num, &den);
@@ -1398,10 +1399,10 @@ static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params,
 int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, 
 				  unsigned int cond,
 				  snd_pcm_hw_param_t var,
-				  struct snd_pcm_hw_constraint_ratdens *r)
+				  const struct snd_pcm_hw_constraint_ratdens *r)
 {
 	return snd_pcm_hw_rule_add(runtime, cond, var,
-				   snd_pcm_hw_rule_ratdens, r,
+				   snd_pcm_hw_rule_ratdens, (void *)r,
 				   var, -1);
 }
 
@@ -1875,20 +1876,17 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream)
 		return;
 	runtime = substream->runtime;
 
-	if (runtime->transfer_ack_begin)
-		runtime->transfer_ack_begin(substream);
-
 	snd_pcm_stream_lock_irqsave(substream, flags);
 	if (!snd_pcm_running(substream) ||
 	    snd_pcm_update_hw_ptr0(substream, 1) < 0)
 		goto _end;
 
+#ifdef CONFIG_SND_PCM_TIMER
 	if (substream->timer_running)
 		snd_timer_interrupt(substream->timer, 1);
+#endif
  _end:
 	snd_pcm_stream_unlock_irqrestore(substream, flags);
-	if (runtime->transfer_ack_end)
-		runtime->transfer_ack_end(substream);
 	kill_fasync(&runtime->fasync, SIGIO, POLL_IN);
 }
 
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 75888dd38a7f..a8b27cdc2844 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -486,6 +486,16 @@ static void snd_pcm_set_state(struct snd_pcm_substream *substream, int state)
 	snd_pcm_stream_unlock_irq(substream);
 }
 
+static inline void snd_pcm_timer_notify(struct snd_pcm_substream *substream,
+					int event)
+{
+#ifdef CONFIG_SND_PCM_TIMER
+	if (substream->timer)
+		snd_timer_notify(substream->timer, event,
+					&substream->runtime->trigger_tstamp);
+#endif
+}
+
 static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params)
 {
@@ -650,7 +660,8 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream,
 	}
 	snd_pcm_stream_unlock_irq(substream);
 
-	if (params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST)
+	if (params->tstamp_mode < 0 ||
+	    params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST)
 		return -EINVAL;
 	if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12) &&
 	    params->tstamp_type > SNDRV_PCM_TSTAMP_TYPE_LAST)
@@ -1042,9 +1053,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
 	    runtime->silence_size > 0)
 		snd_pcm_playback_silence(substream, ULONG_MAX);
-	if (substream->timer)
-		snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTART,
-				 &runtime->trigger_tstamp);
+	snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART);
 }
 
 static struct action_ops snd_pcm_action_start = {
@@ -1092,9 +1101,7 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state)
 	if (runtime->status->state != state) {
 		snd_pcm_trigger_tstamp(substream);
 		runtime->status->state = state;
-		if (substream->timer)
-			snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTOP,
-					 &runtime->trigger_tstamp);
+		snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTOP);
 	}
 	wake_up(&runtime->sleep);
 	wake_up(&runtime->tsleep);
@@ -1208,18 +1215,12 @@ static void snd_pcm_post_pause(struct snd_pcm_substream *substream, int push)
 	snd_pcm_trigger_tstamp(substream);
 	if (push) {
 		runtime->status->state = SNDRV_PCM_STATE_PAUSED;
-		if (substream->timer)
-			snd_timer_notify(substream->timer,
-					 SNDRV_TIMER_EVENT_MPAUSE,
-					 &runtime->trigger_tstamp);
+		snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MPAUSE);
 		wake_up(&runtime->sleep);
 		wake_up(&runtime->tsleep);
 	} else {
 		runtime->status->state = SNDRV_PCM_STATE_RUNNING;
-		if (substream->timer)
-			snd_timer_notify(substream->timer,
-					 SNDRV_TIMER_EVENT_MCONTINUE,
-					 &runtime->trigger_tstamp);
+		snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MCONTINUE);
 	}
 }
 
@@ -1267,9 +1268,7 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state)
 	snd_pcm_trigger_tstamp(substream);
 	runtime->status->suspended_state = runtime->status->state;
 	runtime->status->state = SNDRV_PCM_STATE_SUSPENDED;
-	if (substream->timer)
-		snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSUSPEND,
-				 &runtime->trigger_tstamp);
+	snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSUSPEND);
 	wake_up(&runtime->sleep);
 	wake_up(&runtime->tsleep);
 }
@@ -1373,9 +1372,7 @@ static void snd_pcm_post_resume(struct snd_pcm_substream *substream, int state)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	snd_pcm_trigger_tstamp(substream);
 	runtime->status->state = runtime->status->suspended_state;
-	if (substream->timer)
-		snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MRESUME,
-				 &runtime->trigger_tstamp);
+	snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MRESUME);
 }
 
 static struct action_ops snd_pcm_action_resume = {
@@ -2226,7 +2223,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream)
 
 	snd_pcm_drop(substream);
 	if (substream->hw_opened) {
-		if (substream->ops->hw_free != NULL)
+		if (substream->ops->hw_free &&
+		    substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
 			substream->ops->hw_free(substream);
 		substream->ops->close(substream);
 		substream->hw_opened = 0;
diff --git a/sound/core/seq/oss/seq_oss_readq.c b/sound/core/seq/oss/seq_oss_readq.c
index ccd893566f1d..046cb586fb2f 100644
--- a/sound/core/seq/oss/seq_oss_readq.c
+++ b/sound/core/seq/oss/seq_oss_readq.c
@@ -91,8 +91,7 @@ snd_seq_oss_readq_clear(struct seq_oss_readq *q)
 		q->head = q->tail = 0;
 	}
 	/* if someone sleeping, wake'em up */
-	if (waitqueue_active(&q->midi_sleep))
-		wake_up(&q->midi_sleep);
+	wake_up(&q->midi_sleep);
 	q->input_time = (unsigned long)-1;
 }
 
@@ -138,8 +137,7 @@ snd_seq_oss_readq_put_event(struct seq_oss_readq *q, union evrec *ev)
 	q->qlen++;
 
 	/* wake up sleeper */
-	if (waitqueue_active(&q->midi_sleep))
-		wake_up(&q->midi_sleep);
+	wake_up(&q->midi_sleep);
 
 	spin_unlock_irqrestore(&q->lock, flags);
 
diff --git a/sound/core/seq/oss/seq_oss_writeq.c b/sound/core/seq/oss/seq_oss_writeq.c
index d50338bbc21f..1f6788a18444 100644
--- a/sound/core/seq/oss/seq_oss_writeq.c
+++ b/sound/core/seq/oss/seq_oss_writeq.c
@@ -138,9 +138,7 @@ snd_seq_oss_writeq_wakeup(struct seq_oss_writeq *q, abstime_t time)
 	spin_lock_irqsave(&q->sync_lock, flags);
 	q->sync_time = time;
 	q->sync_event_put = 0;
-	if (waitqueue_active(&q->sync_sleep)) {
-		wake_up(&q->sync_sleep);
-	}
+	wake_up(&q->sync_sleep);
 	spin_unlock_irqrestore(&q->sync_lock, flags);
 }
 
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index 8850b7de1d38..bee0e5f1a116 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -120,4 +120,31 @@ config SND_BEBOB
           To compile this driver as a module, choose M here: the module
           will be called snd-bebob.
 
+config SND_FIREWIRE_DIGI00X
+	tristate "Digidesign Digi 002/003 family support"
+	select SND_FIREWIRE_LIB
+	select SND_HWDEP
+	help
+	 Say Y here to include support for Digidesign Digi 002/003 family.
+	  * Digi 002 Console
+	  * Digi 002 Rack
+	  * Digi 003 Console
+	  * Digi 003 Rack
+	  * Digi 003 Rack+
+
+	 To compile this driver as a module, choose M here: the module
+	 will be called snd-firewire-digi00x.
+
+config SND_FIREWIRE_TASCAM
+	tristate "TASCAM FireWire series support"
+	select SND_FIREWIRE_LIB
+	select SND_HWDEP
+	help
+	 Say Y here to include support for TASCAM.
+	  * FW-1884
+	  * FW-1082
+
+	 To compile this driver as a module, choose M here: the module
+	 will be called snd-firewire-tascam.
+
 endif # SND_FIREWIRE
diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile
index 8b37f084b2ab..f5fb62551c60 100644
--- a/sound/firewire/Makefile
+++ b/sound/firewire/Makefile
@@ -1,6 +1,5 @@
 snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \
-			 fcp.o cmp.o amdtp.o
-snd-oxfw-objs := oxfw.o
+			 fcp.o cmp.o amdtp-stream.o amdtp-am824.o
 snd-isight-objs := isight.o
 snd-scs1x-objs := scs1x.o
 
@@ -11,3 +10,5 @@ obj-$(CONFIG_SND_ISIGHT) += snd-isight.o
 obj-$(CONFIG_SND_SCS1X) += snd-scs1x.o
 obj-$(CONFIG_SND_FIREWORKS) += fireworks/
 obj-$(CONFIG_SND_BEBOB) += bebob/
+obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += digi00x/
+obj-$(CONFIG_SND_FIREWIRE_TASCAM) += tascam/
diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
new file mode 100644
index 000000000000..bebddc60fde8
--- /dev/null
+++ b/sound/firewire/amdtp-am824.c
@@ -0,0 +1,465 @@
+/*
+ * AM824 format in Audio and Music Data Transmission Protocol (IEC 61883-6)
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ * Copyright (c) 2015 Takashi Sakamoto <o-takashi@sakamocchi.jp>
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <linux/slab.h>
+
+#include "amdtp-am824.h"
+
+#define CIP_FMT_AM		0x10
+
+/* "Clock-based rate control mode" is just supported. */
+#define AMDTP_FDF_AM824		0x00
+
+/*
+ * Nominally 3125 bytes/second, but the MIDI port's clock might be
+ * 1% too slow, and the bus clock 100 ppm too fast.
+ */
+#define MIDI_BYTES_PER_SECOND	3093
+
+/*
+ * Several devices look only at the first eight data blocks.
+ * In any case, this is more than enough for the MIDI data rate.
+ */
+#define MAX_MIDI_RX_BLOCKS	8
+
+struct amdtp_am824 {
+	struct snd_rawmidi_substream *midi[AM824_MAX_CHANNELS_FOR_MIDI * 8];
+	int midi_fifo_limit;
+	int midi_fifo_used[AM824_MAX_CHANNELS_FOR_MIDI * 8];
+	unsigned int pcm_channels;
+	unsigned int midi_ports;
+
+	u8 pcm_positions[AM824_MAX_CHANNELS_FOR_PCM];
+	u8 midi_position;
+
+	void (*transfer_samples)(struct amdtp_stream *s,
+				 struct snd_pcm_substream *pcm,
+				 __be32 *buffer, unsigned int frames);
+
+	unsigned int frame_multiplier;
+};
+
+/**
+ * amdtp_am824_set_parameters - set stream parameters
+ * @s: the AMDTP stream to configure
+ * @rate: the sample rate
+ * @pcm_channels: the number of PCM samples in each data block, to be encoded
+ *                as AM824 multi-bit linear audio
+ * @midi_ports: the number of MIDI ports (i.e., MPX-MIDI Data Channels)
+ * @double_pcm_frames: one data block transfers two PCM frames
+ *
+ * The parameters must be set before the stream is started, and must not be
+ * changed while the stream is running.
+ */
+int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate,
+			       unsigned int pcm_channels,
+			       unsigned int midi_ports,
+			       bool double_pcm_frames)
+{
+	struct amdtp_am824 *p = s->protocol;
+	unsigned int midi_channels;
+	unsigned int i;
+	int err;
+
+	if (amdtp_stream_running(s))
+		return -EINVAL;
+
+	if (pcm_channels > AM824_MAX_CHANNELS_FOR_PCM)
+		return -EINVAL;
+
+	midi_channels = DIV_ROUND_UP(midi_ports, 8);
+	if (midi_channels > AM824_MAX_CHANNELS_FOR_MIDI)
+		return -EINVAL;
+
+	if (WARN_ON(amdtp_stream_running(s)) ||
+	    WARN_ON(pcm_channels > AM824_MAX_CHANNELS_FOR_PCM) ||
+	    WARN_ON(midi_channels > AM824_MAX_CHANNELS_FOR_MIDI))
+		return -EINVAL;
+
+	err = amdtp_stream_set_parameters(s, rate,
+					  pcm_channels + midi_channels);
+	if (err < 0)
+		return err;
+
+	s->fdf = AMDTP_FDF_AM824 | s->sfc;
+
+	p->pcm_channels = pcm_channels;
+	p->midi_ports = midi_ports;
+
+	/*
+	 * In IEC 61883-6, one data block represents one event. In ALSA, one
+	 * event equals to one PCM frame. But Dice has a quirk at higher
+	 * sampling rate to transfer two PCM frames in one data block.
+	 */
+	if (double_pcm_frames)
+		p->frame_multiplier = 2;
+	else
+		p->frame_multiplier = 1;
+
+	/* init the position map for PCM and MIDI channels */
+	for (i = 0; i < pcm_channels; i++)
+		p->pcm_positions[i] = i;
+	p->midi_position = p->pcm_channels;
+
+	/*
+	 * We do not know the actual MIDI FIFO size of most devices.  Just
+	 * assume two bytes, i.e., one byte can be received over the bus while
+	 * the previous one is transmitted over MIDI.
+	 * (The value here is adjusted for midi_ratelimit_per_packet().)
+	 */
+	p->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_parameters);
+
+/**
+ * amdtp_am824_set_pcm_position - set an index of data channel for a channel
+ *				  of PCM frame
+ * @s: the AMDTP stream
+ * @index: the index of data channel in an data block
+ * @position: the channel of PCM frame
+ */
+void amdtp_am824_set_pcm_position(struct amdtp_stream *s, unsigned int index,
+				 unsigned int position)
+{
+	struct amdtp_am824 *p = s->protocol;
+
+	if (index < p->pcm_channels)
+		p->pcm_positions[index] = position;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_pcm_position);
+
+/**
+ * amdtp_am824_set_midi_position - set a index of data channel for MIDI
+ *				   conformant data channel
+ * @s: the AMDTP stream
+ * @position: the index of data channel in an data block
+ */
+void amdtp_am824_set_midi_position(struct amdtp_stream *s,
+				   unsigned int position)
+{
+	struct amdtp_am824 *p = s->protocol;
+
+	p->midi_position = position;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_midi_position);
+
+static void write_pcm_s32(struct amdtp_stream *s,
+			  struct snd_pcm_substream *pcm,
+			  __be32 *buffer, unsigned int frames)
+{
+	struct amdtp_am824 *p = s->protocol;
+	struct snd_pcm_runtime *runtime = pcm->runtime;
+	unsigned int channels, remaining_frames, i, c;
+	const u32 *src;
+
+	channels = p->pcm_channels;
+	src = (void *)runtime->dma_area +
+			frames_to_bytes(runtime, s->pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+	for (i = 0; i < frames; ++i) {
+		for (c = 0; c < channels; ++c) {
+			buffer[p->pcm_positions[c]] =
+					cpu_to_be32((*src >> 8) | 0x40000000);
+			src++;
+		}
+		buffer += s->data_block_quadlets;
+		if (--remaining_frames == 0)
+			src = (void *)runtime->dma_area;
+	}
+}
+
+static void write_pcm_s16(struct amdtp_stream *s,
+			  struct snd_pcm_substream *pcm,
+			  __be32 *buffer, unsigned int frames)
+{
+	struct amdtp_am824 *p = s->protocol;
+	struct snd_pcm_runtime *runtime = pcm->runtime;
+	unsigned int channels, remaining_frames, i, c;
+	const u16 *src;
+
+	channels = p->pcm_channels;
+	src = (void *)runtime->dma_area +
+			frames_to_bytes(runtime, s->pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+	for (i = 0; i < frames; ++i) {
+		for (c = 0; c < channels; ++c) {
+			buffer[p->pcm_positions[c]] =
+					cpu_to_be32((*src << 8) | 0x42000000);
+			src++;
+		}
+		buffer += s->data_block_quadlets;
+		if (--remaining_frames == 0)
+			src = (void *)runtime->dma_area;
+	}
+}
+
+static void read_pcm_s32(struct amdtp_stream *s,
+			 struct snd_pcm_substream *pcm,
+			 __be32 *buffer, unsigned int frames)
+{
+	struct amdtp_am824 *p = s->protocol;
+	struct snd_pcm_runtime *runtime = pcm->runtime;
+	unsigned int channels, remaining_frames, i, c;
+	u32 *dst;
+
+	channels = p->pcm_channels;
+	dst  = (void *)runtime->dma_area +
+			frames_to_bytes(runtime, s->pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+	for (i = 0; i < frames; ++i) {
+		for (c = 0; c < channels; ++c) {
+			*dst = be32_to_cpu(buffer[p->pcm_positions[c]]) << 8;
+			dst++;
+		}
+		buffer += s->data_block_quadlets;
+		if (--remaining_frames == 0)
+			dst = (void *)runtime->dma_area;
+	}
+}
+
+static void write_pcm_silence(struct amdtp_stream *s,
+			      __be32 *buffer, unsigned int frames)
+{
+	struct amdtp_am824 *p = s->protocol;
+	unsigned int i, c, channels = p->pcm_channels;
+
+	for (i = 0; i < frames; ++i) {
+		for (c = 0; c < channels; ++c)
+			buffer[p->pcm_positions[c]] = cpu_to_be32(0x40000000);
+		buffer += s->data_block_quadlets;
+	}
+}
+
+/**
+ * amdtp_am824_set_pcm_format - set the PCM format
+ * @s: the AMDTP stream to configure
+ * @format: the format of the ALSA PCM device
+ *
+ * The sample format must be set after the other parameters (rate/PCM channels/
+ * MIDI) and before the stream is started, and must not be changed while the
+ * stream is running.
+ */
+void amdtp_am824_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
+{
+	struct amdtp_am824 *p = s->protocol;
+
+	if (WARN_ON(amdtp_stream_pcm_running(s)))
+		return;
+
+	switch (format) {
+	default:
+		WARN_ON(1);
+		/* fall through */
+	case SNDRV_PCM_FORMAT_S16:
+		if (s->direction == AMDTP_OUT_STREAM) {
+			p->transfer_samples = write_pcm_s16;
+			break;
+		}
+		WARN_ON(1);
+		/* fall through */
+	case SNDRV_PCM_FORMAT_S32:
+		if (s->direction == AMDTP_OUT_STREAM)
+			p->transfer_samples = write_pcm_s32;
+		else
+			p->transfer_samples = read_pcm_s32;
+		break;
+	}
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_pcm_format);
+
+/**
+ * amdtp_am824_add_pcm_hw_constraints - add hw constraints for PCM substream
+ * @s:		the AMDTP stream for AM824 data block, must be initialized.
+ * @runtime:	the PCM substream runtime
+ *
+ */
+int amdtp_am824_add_pcm_hw_constraints(struct amdtp_stream *s,
+				       struct snd_pcm_runtime *runtime)
+{
+	int err;
+
+	err = amdtp_stream_add_pcm_hw_constraints(s, runtime);
+	if (err < 0)
+		return err;
+
+	/* AM824 in IEC 61883-6 can deliver 24bit data. */
+	return snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_add_pcm_hw_constraints);
+
+/**
+ * amdtp_am824_midi_trigger - start/stop playback/capture with a MIDI device
+ * @s: the AMDTP stream
+ * @port: index of MIDI port
+ * @midi: the MIDI device to be started, or %NULL to stop the current device
+ *
+ * Call this function on a running isochronous stream to enable the actual
+ * transmission of MIDI data.  This function should be called from the MIDI
+ * device's .trigger callback.
+ */
+void amdtp_am824_midi_trigger(struct amdtp_stream *s, unsigned int port,
+			      struct snd_rawmidi_substream *midi)
+{
+	struct amdtp_am824 *p = s->protocol;
+
+	if (port < p->midi_ports)
+		ACCESS_ONCE(p->midi[port]) = midi;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_midi_trigger);
+
+/*
+ * To avoid sending MIDI bytes at too high a rate, assume that the receiving
+ * device has a FIFO, and track how much it is filled.  This values increases
+ * by one whenever we send one byte in a packet, but the FIFO empties at
+ * a constant rate independent of our packet rate.  One packet has syt_interval
+ * samples, so the number of bytes that empty out of the FIFO, per packet(!),
+ * is MIDI_BYTES_PER_SECOND * syt_interval / sample_rate.  To avoid storing
+ * fractional values, the values in midi_fifo_used[] are measured in bytes
+ * multiplied by the sample rate.
+ */
+static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port)
+{
+	struct amdtp_am824 *p = s->protocol;
+	int used;
+
+	used = p->midi_fifo_used[port];
+	if (used == 0) /* common shortcut */
+		return true;
+
+	used -= MIDI_BYTES_PER_SECOND * s->syt_interval;
+	used = max(used, 0);
+	p->midi_fifo_used[port] = used;
+
+	return used < p->midi_fifo_limit;
+}
+
+static void midi_rate_use_one_byte(struct amdtp_stream *s, unsigned int port)
+{
+	struct amdtp_am824 *p = s->protocol;
+
+	p->midi_fifo_used[port] += amdtp_rate_table[s->sfc];
+}
+
+static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer,
+				unsigned int frames)
+{
+	struct amdtp_am824 *p = s->protocol;
+	unsigned int f, port;
+	u8 *b;
+
+	for (f = 0; f < frames; f++) {
+		b = (u8 *)&buffer[p->midi_position];
+
+		port = (s->data_block_counter + f) % 8;
+		if (f < MAX_MIDI_RX_BLOCKS &&
+		    midi_ratelimit_per_packet(s, port) &&
+		    p->midi[port] != NULL &&
+		    snd_rawmidi_transmit(p->midi[port], &b[1], 1) == 1) {
+			midi_rate_use_one_byte(s, port);
+			b[0] = 0x81;
+		} else {
+			b[0] = 0x80;
+			b[1] = 0;
+		}
+		b[2] = 0;
+		b[3] = 0;
+
+		buffer += s->data_block_quadlets;
+	}
+}
+
+static void read_midi_messages(struct amdtp_stream *s,
+			       __be32 *buffer, unsigned int frames)
+{
+	struct amdtp_am824 *p = s->protocol;
+	unsigned int f, port;
+	int len;
+	u8 *b;
+
+	for (f = 0; f < frames; f++) {
+		port = (s->data_block_counter + f) % 8;
+		b = (u8 *)&buffer[p->midi_position];
+
+		len = b[0] - 0x80;
+		if ((1 <= len) &&  (len <= 3) && (p->midi[port]))
+			snd_rawmidi_receive(p->midi[port], b + 1, len);
+
+		buffer += s->data_block_quadlets;
+	}
+}
+
+static unsigned int process_rx_data_blocks(struct amdtp_stream *s, __be32 *buffer,
+					   unsigned int data_blocks, unsigned int *syt)
+{
+	struct amdtp_am824 *p = s->protocol;
+	struct snd_pcm_substream *pcm = ACCESS_ONCE(s->pcm);
+	unsigned int pcm_frames;
+
+	if (pcm) {
+		p->transfer_samples(s, pcm, buffer, data_blocks);
+		pcm_frames = data_blocks * p->frame_multiplier;
+	} else {
+		write_pcm_silence(s, buffer, data_blocks);
+		pcm_frames = 0;
+	}
+
+	if (p->midi_ports)
+		write_midi_messages(s, buffer, data_blocks);
+
+	return pcm_frames;
+}
+
+static unsigned int process_tx_data_blocks(struct amdtp_stream *s, __be32 *buffer,
+					   unsigned int data_blocks, unsigned int *syt)
+{
+	struct amdtp_am824 *p = s->protocol;
+	struct snd_pcm_substream *pcm = ACCESS_ONCE(s->pcm);
+	unsigned int pcm_frames;
+
+	if (pcm) {
+		p->transfer_samples(s, pcm, buffer, data_blocks);
+		pcm_frames = data_blocks * p->frame_multiplier;
+	} else {
+		pcm_frames = 0;
+	}
+
+	if (p->midi_ports)
+		read_midi_messages(s, buffer, data_blocks);
+
+	return pcm_frames;
+}
+
+/**
+ * amdtp_am824_init - initialize an AMDTP stream structure to handle AM824
+ *		      data block
+ * @s: the AMDTP stream to initialize
+ * @unit: the target of the stream
+ * @dir: the direction of stream
+ * @flags: the packet transmission method to use
+ */
+int amdtp_am824_init(struct amdtp_stream *s, struct fw_unit *unit,
+		     enum amdtp_stream_direction dir, enum cip_flags flags)
+{
+	amdtp_stream_process_data_blocks_t process_data_blocks;
+
+	if (dir == AMDTP_IN_STREAM)
+		process_data_blocks = process_tx_data_blocks;
+	else
+		process_data_blocks = process_rx_data_blocks;
+
+	return amdtp_stream_init(s, unit, dir, flags, CIP_FMT_AM,
+				 process_data_blocks,
+				 sizeof(struct amdtp_am824));
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_init);
diff --git a/sound/firewire/amdtp-am824.h b/sound/firewire/amdtp-am824.h
new file mode 100644
index 000000000000..73b07b3109db
--- /dev/null
+++ b/sound/firewire/amdtp-am824.h
@@ -0,0 +1,52 @@
+#ifndef SOUND_FIREWIRE_AMDTP_AM824_H_INCLUDED
+#define SOUND_FIREWIRE_AMDTP_AM824_H_INCLUDED
+
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+
+#include "amdtp-stream.h"
+
+#define AM824_IN_PCM_FORMAT_BITS	SNDRV_PCM_FMTBIT_S32
+
+#define AM824_OUT_PCM_FORMAT_BITS	(SNDRV_PCM_FMTBIT_S16 | \
+					 SNDRV_PCM_FMTBIT_S32)
+
+/*
+ * This module supports maximum 64 PCM channels for one PCM stream
+ * This is for our convenience.
+ */
+#define AM824_MAX_CHANNELS_FOR_PCM	64
+
+/*
+ * AMDTP packet can include channels for MIDI conformant data.
+ * Each MIDI conformant data channel includes 8 MPX-MIDI data stream.
+ * Each MPX-MIDI data stream includes one data stream from/to MIDI ports.
+ *
+ * This module supports maximum 1 MIDI conformant data channels.
+ * Then this AMDTP packets can transfer maximum 8 MIDI data streams.
+ */
+#define AM824_MAX_CHANNELS_FOR_MIDI	1
+
+int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate,
+			       unsigned int pcm_channels,
+			       unsigned int midi_ports,
+			       bool double_pcm_frames);
+
+void amdtp_am824_set_pcm_position(struct amdtp_stream *s, unsigned int index,
+				 unsigned int position);
+
+void amdtp_am824_set_midi_position(struct amdtp_stream *s,
+				   unsigned int position);
+
+int amdtp_am824_add_pcm_hw_constraints(struct amdtp_stream *s,
+				       struct snd_pcm_runtime *runtime);
+
+void amdtp_am824_set_pcm_format(struct amdtp_stream *s,
+				snd_pcm_format_t format);
+
+void amdtp_am824_midi_trigger(struct amdtp_stream *s, unsigned int port,
+			      struct snd_rawmidi_substream *midi);
+
+int amdtp_am824_init(struct amdtp_stream *s, struct fw_unit *unit,
+		     enum amdtp_stream_direction dir, enum cip_flags flags);
+#endif
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp-stream.c
index 2a153d260836..ed2902609a4c 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp-stream.c
@@ -11,28 +11,14 @@
 #include <linux/firewire.h>
 #include <linux/module.h>
 #include <linux/slab.h>
-#include <linux/sched.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
-#include <sound/rawmidi.h>
-#include "amdtp.h"
+#include "amdtp-stream.h"
 
 #define TICKS_PER_CYCLE		3072
 #define CYCLES_PER_SECOND	8000
 #define TICKS_PER_SECOND	(TICKS_PER_CYCLE * CYCLES_PER_SECOND)
 
-/*
- * Nominally 3125 bytes/second, but the MIDI port's clock might be
- * 1% too slow, and the bus clock 100 ppm too fast.
- */
-#define MIDI_BYTES_PER_SECOND	3093
-
-/*
- * Several devices look only at the first eight data blocks.
- * In any case, this is more than enough for the MIDI data rate.
- */
-#define MAX_MIDI_RX_BLOCKS	8
-
 #define TRANSFER_DELAY_TICKS	0x2e00 /* 479.17 microseconds */
 
 /* isochronous header parameters */
@@ -55,12 +41,8 @@
 #define CIP_SYT_MASK		0x0000ffff
 #define CIP_SYT_NO_INFO		0xffff
 
-/*
- * Audio and Music transfer protocol specific parameters
- * only "Clock-based rate control mode" is supported
- */
-#define CIP_FMT_AM		(0x10 << CIP_FMT_SHIFT)
-#define AMDTP_FDF_AM824		(0 << (CIP_FDF_SHIFT + 3))
+/* Audio and Music transfer protocol specific parameters */
+#define CIP_FMT_AM		0x10
 #define AMDTP_FDF_NO_DATA	0xff
 
 /* TODO: make these configurable */
@@ -78,10 +60,23 @@ static void pcm_period_tasklet(unsigned long data);
  * @unit: the target of the stream
  * @dir: the direction of stream
  * @flags: the packet transmission method to use
+ * @fmt: the value of fmt field in CIP header
+ * @process_data_blocks: callback handler to process data blocks
+ * @protocol_size: the size to allocate newly for protocol
  */
 int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
-		      enum amdtp_stream_direction dir, enum cip_flags flags)
+		      enum amdtp_stream_direction dir, enum cip_flags flags,
+		      unsigned int fmt,
+		      amdtp_stream_process_data_blocks_t process_data_blocks,
+		      unsigned int protocol_size)
 {
+	if (process_data_blocks == NULL)
+		return -EINVAL;
+
+	s->protocol = kzalloc(protocol_size, GFP_KERNEL);
+	if (!s->protocol)
+		return -ENOMEM;
+
 	s->unit = unit;
 	s->direction = dir;
 	s->flags = flags;
@@ -94,6 +89,9 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
 	s->callbacked = false;
 	s->sync_slave = NULL;
 
+	s->fmt = fmt;
+	s->process_data_blocks = process_data_blocks;
+
 	return 0;
 }
 EXPORT_SYMBOL(amdtp_stream_init);
@@ -105,6 +103,7 @@ EXPORT_SYMBOL(amdtp_stream_init);
 void amdtp_stream_destroy(struct amdtp_stream *s)
 {
 	WARN_ON(amdtp_stream_running(s));
+	kfree(s->protocol);
 	mutex_destroy(&s->mutex);
 }
 EXPORT_SYMBOL(amdtp_stream_destroy);
@@ -141,11 +140,6 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
 {
 	int err;
 
-	/* AM824 in IEC 61883-6 can deliver 24bit data */
-	err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
-	if (err < 0)
-		goto end;
-
 	/*
 	 * Currently firewire-lib processes 16 packets in one software
 	 * interrupt callback. This equals to 2msec but actually the
@@ -190,39 +184,25 @@ EXPORT_SYMBOL(amdtp_stream_add_pcm_hw_constraints);
  * amdtp_stream_set_parameters - set stream parameters
  * @s: the AMDTP stream to configure
  * @rate: the sample rate
- * @pcm_channels: the number of PCM samples in each data block, to be encoded
- *                as AM824 multi-bit linear audio
- * @midi_ports: the number of MIDI ports (i.e., MPX-MIDI Data Channels)
+ * @data_block_quadlets: the size of a data block in quadlet unit
  *
  * The parameters must be set before the stream is started, and must not be
  * changed while the stream is running.
  */
-void amdtp_stream_set_parameters(struct amdtp_stream *s,
-				 unsigned int rate,
-				 unsigned int pcm_channels,
-				 unsigned int midi_ports)
+int amdtp_stream_set_parameters(struct amdtp_stream *s, unsigned int rate,
+				unsigned int data_block_quadlets)
 {
-	unsigned int i, sfc, midi_channels;
+	unsigned int sfc;
 
-	midi_channels = DIV_ROUND_UP(midi_ports, 8);
-
-	if (WARN_ON(amdtp_stream_running(s)) |
-	    WARN_ON(pcm_channels > AMDTP_MAX_CHANNELS_FOR_PCM) |
-	    WARN_ON(midi_channels > AMDTP_MAX_CHANNELS_FOR_MIDI))
-		return;
-
-	for (sfc = 0; sfc < ARRAY_SIZE(amdtp_rate_table); ++sfc)
+	for (sfc = 0; sfc < ARRAY_SIZE(amdtp_rate_table); ++sfc) {
 		if (amdtp_rate_table[sfc] == rate)
-			goto sfc_found;
-	WARN_ON(1);
-	return;
+			break;
+	}
+	if (sfc == ARRAY_SIZE(amdtp_rate_table))
+		return -EINVAL;
 
-sfc_found:
-	s->pcm_channels = pcm_channels;
 	s->sfc = sfc;
-	s->data_block_quadlets = s->pcm_channels + midi_channels;
-	s->midi_ports = midi_ports;
-
+	s->data_block_quadlets = data_block_quadlets;
 	s->syt_interval = amdtp_syt_intervals[sfc];
 
 	/* default buffering in the device */
@@ -231,18 +211,7 @@ sfc_found:
 		/* additional buffering needed to adjust for no-data packets */
 		s->transfer_delay += TICKS_PER_SECOND * s->syt_interval / rate;
 
-	/* init the position map for PCM and MIDI channels */
-	for (i = 0; i < pcm_channels; i++)
-		s->pcm_positions[i] = i;
-	s->midi_position = s->pcm_channels;
-
-	/*
-	 * We do not know the actual MIDI FIFO size of most devices.  Just
-	 * assume two bytes, i.e., one byte can be received over the bus while
-	 * the previous one is transmitted over MIDI.
-	 * (The value here is adjusted for midi_ratelimit_per_packet().)
-	 */
-	s->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1;
+	return 0;
 }
 EXPORT_SYMBOL(amdtp_stream_set_parameters);
 
@@ -264,52 +233,6 @@ unsigned int amdtp_stream_get_max_payload(struct amdtp_stream *s)
 }
 EXPORT_SYMBOL(amdtp_stream_get_max_payload);
 
-static void write_pcm_s16(struct amdtp_stream *s,
-			  struct snd_pcm_substream *pcm,
-			  __be32 *buffer, unsigned int frames);
-static void write_pcm_s32(struct amdtp_stream *s,
-			  struct snd_pcm_substream *pcm,
-			  __be32 *buffer, unsigned int frames);
-static void read_pcm_s32(struct amdtp_stream *s,
-			 struct snd_pcm_substream *pcm,
-			 __be32 *buffer, unsigned int frames);
-
-/**
- * amdtp_stream_set_pcm_format - set the PCM format
- * @s: the AMDTP stream to configure
- * @format: the format of the ALSA PCM device
- *
- * The sample format must be set after the other parameters (rate/PCM channels/
- * MIDI) and before the stream is started, and must not be changed while the
- * stream is running.
- */
-void amdtp_stream_set_pcm_format(struct amdtp_stream *s,
-				 snd_pcm_format_t format)
-{
-	if (WARN_ON(amdtp_stream_pcm_running(s)))
-		return;
-
-	switch (format) {
-	default:
-		WARN_ON(1);
-		/* fall through */
-	case SNDRV_PCM_FORMAT_S16:
-		if (s->direction == AMDTP_OUT_STREAM) {
-			s->transfer_samples = write_pcm_s16;
-			break;
-		}
-		WARN_ON(1);
-		/* fall through */
-	case SNDRV_PCM_FORMAT_S32:
-		if (s->direction == AMDTP_OUT_STREAM)
-			s->transfer_samples = write_pcm_s32;
-		else
-			s->transfer_samples = read_pcm_s32;
-		break;
-	}
-}
-EXPORT_SYMBOL(amdtp_stream_set_pcm_format);
-
 /**
  * amdtp_stream_pcm_prepare - prepare PCM device for running
  * @s: the AMDTP stream
@@ -412,182 +335,12 @@ static unsigned int calculate_syt(struct amdtp_stream *s,
 	}
 }
 
-static void write_pcm_s32(struct amdtp_stream *s,
-			  struct snd_pcm_substream *pcm,
-			  __be32 *buffer, unsigned int frames)
-{
-	struct snd_pcm_runtime *runtime = pcm->runtime;
-	unsigned int channels, remaining_frames, i, c;
-	const u32 *src;
-
-	channels = s->pcm_channels;
-	src = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
-
-	for (i = 0; i < frames; ++i) {
-		for (c = 0; c < channels; ++c) {
-			buffer[s->pcm_positions[c]] =
-					cpu_to_be32((*src >> 8) | 0x40000000);
-			src++;
-		}
-		buffer += s->data_block_quadlets;
-		if (--remaining_frames == 0)
-			src = (void *)runtime->dma_area;
-	}
-}
-
-static void write_pcm_s16(struct amdtp_stream *s,
-			  struct snd_pcm_substream *pcm,
-			  __be32 *buffer, unsigned int frames)
-{
-	struct snd_pcm_runtime *runtime = pcm->runtime;
-	unsigned int channels, remaining_frames, i, c;
-	const u16 *src;
-
-	channels = s->pcm_channels;
-	src = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
-
-	for (i = 0; i < frames; ++i) {
-		for (c = 0; c < channels; ++c) {
-			buffer[s->pcm_positions[c]] =
-					cpu_to_be32((*src << 8) | 0x42000000);
-			src++;
-		}
-		buffer += s->data_block_quadlets;
-		if (--remaining_frames == 0)
-			src = (void *)runtime->dma_area;
-	}
-}
-
-static void read_pcm_s32(struct amdtp_stream *s,
-			 struct snd_pcm_substream *pcm,
-			 __be32 *buffer, unsigned int frames)
-{
-	struct snd_pcm_runtime *runtime = pcm->runtime;
-	unsigned int channels, remaining_frames, i, c;
-	u32 *dst;
-
-	channels = s->pcm_channels;
-	dst  = (void *)runtime->dma_area +
-			frames_to_bytes(runtime, s->pcm_buffer_pointer);
-	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
-
-	for (i = 0; i < frames; ++i) {
-		for (c = 0; c < channels; ++c) {
-			*dst = be32_to_cpu(buffer[s->pcm_positions[c]]) << 8;
-			dst++;
-		}
-		buffer += s->data_block_quadlets;
-		if (--remaining_frames == 0)
-			dst = (void *)runtime->dma_area;
-	}
-}
-
-static void write_pcm_silence(struct amdtp_stream *s,
-			      __be32 *buffer, unsigned int frames)
-{
-	unsigned int i, c;
-
-	for (i = 0; i < frames; ++i) {
-		for (c = 0; c < s->pcm_channels; ++c)
-			buffer[s->pcm_positions[c]] = cpu_to_be32(0x40000000);
-		buffer += s->data_block_quadlets;
-	}
-}
-
-/*
- * To avoid sending MIDI bytes at too high a rate, assume that the receiving
- * device has a FIFO, and track how much it is filled.  This values increases
- * by one whenever we send one byte in a packet, but the FIFO empties at
- * a constant rate independent of our packet rate.  One packet has syt_interval
- * samples, so the number of bytes that empty out of the FIFO, per packet(!),
- * is MIDI_BYTES_PER_SECOND * syt_interval / sample_rate.  To avoid storing
- * fractional values, the values in midi_fifo_used[] are measured in bytes
- * multiplied by the sample rate.
- */
-static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port)
-{
-	int used;
-
-	used = s->midi_fifo_used[port];
-	if (used == 0) /* common shortcut */
-		return true;
-
-	used -= MIDI_BYTES_PER_SECOND * s->syt_interval;
-	used = max(used, 0);
-	s->midi_fifo_used[port] = used;
-
-	return used < s->midi_fifo_limit;
-}
-
-static void midi_rate_use_one_byte(struct amdtp_stream *s, unsigned int port)
-{
-	s->midi_fifo_used[port] += amdtp_rate_table[s->sfc];
-}
-
-static void write_midi_messages(struct amdtp_stream *s,
-				__be32 *buffer, unsigned int frames)
-{
-	unsigned int f, port;
-	u8 *b;
-
-	for (f = 0; f < frames; f++) {
-		b = (u8 *)&buffer[s->midi_position];
-
-		port = (s->data_block_counter + f) % 8;
-		if (f < MAX_MIDI_RX_BLOCKS &&
-		    midi_ratelimit_per_packet(s, port) &&
-		    s->midi[port] != NULL &&
-		    snd_rawmidi_transmit(s->midi[port], &b[1], 1) == 1) {
-			midi_rate_use_one_byte(s, port);
-			b[0] = 0x81;
-		} else {
-			b[0] = 0x80;
-			b[1] = 0;
-		}
-		b[2] = 0;
-		b[3] = 0;
-
-		buffer += s->data_block_quadlets;
-	}
-}
-
-static void read_midi_messages(struct amdtp_stream *s,
-			       __be32 *buffer, unsigned int frames)
-{
-	unsigned int f, port;
-	int len;
-	u8 *b;
-
-	for (f = 0; f < frames; f++) {
-		port = (s->data_block_counter + f) % 8;
-		b = (u8 *)&buffer[s->midi_position];
-
-		len = b[0] - 0x80;
-		if ((1 <= len) &&  (len <= 3) && (s->midi[port]))
-			snd_rawmidi_receive(s->midi[port], b + 1, len);
-
-		buffer += s->data_block_quadlets;
-	}
-}
-
 static void update_pcm_pointers(struct amdtp_stream *s,
 				struct snd_pcm_substream *pcm,
 				unsigned int frames)
 {
 	unsigned int ptr;
 
-	/*
-	 * In IEC 61883-6, one data block represents one event. In ALSA, one
-	 * event equals to one PCM frame. But Dice has a quirk to transfer
-	 * two PCM frames in one data block.
-	 */
-	if (s->double_pcm_frames)
-		frames *= 2;
-
 	ptr = s->pcm_buffer_pointer + frames;
 	if (ptr >= pcm->runtime->buffer_size)
 		ptr -= pcm->runtime->buffer_size;
@@ -656,23 +409,19 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks,
 {
 	__be32 *buffer;
 	unsigned int payload_length;
+	unsigned int pcm_frames;
 	struct snd_pcm_substream *pcm;
 
 	buffer = s->buffer.packets[s->packet_index].buffer;
+	pcm_frames = s->process_data_blocks(s, buffer + 2, data_blocks, &syt);
+
 	buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) |
 				(s->data_block_quadlets << CIP_DBS_SHIFT) |
 				s->data_block_counter);
-	buffer[1] = cpu_to_be32(CIP_EOH | CIP_FMT_AM | AMDTP_FDF_AM824 |
-				(s->sfc << CIP_FDF_SHIFT) | syt);
-	buffer += 2;
-
-	pcm = ACCESS_ONCE(s->pcm);
-	if (pcm)
-		s->transfer_samples(s, pcm, buffer, data_blocks);
-	else
-		write_pcm_silence(s, buffer, data_blocks);
-	if (s->midi_ports)
-		write_midi_messages(s, buffer, data_blocks);
+	buffer[1] = cpu_to_be32(CIP_EOH |
+				((s->fmt << CIP_FMT_SHIFT) & CIP_FMT_MASK) |
+				((s->fdf << CIP_FDF_SHIFT) & CIP_FDF_MASK) |
+				(syt & CIP_SYT_MASK));
 
 	s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff;
 
@@ -680,8 +429,9 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks,
 	if (queue_out_packet(s, payload_length, false) < 0)
 		return -EIO;
 
-	if (pcm)
-		update_pcm_pointers(s, pcm, data_blocks);
+	pcm = ACCESS_ONCE(s->pcm);
+	if (pcm && pcm_frames > 0)
+		update_pcm_pointers(s, pcm, pcm_frames);
 
 	/* No need to return the number of handled data blocks. */
 	return 0;
@@ -689,11 +439,13 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks,
 
 static int handle_in_packet(struct amdtp_stream *s,
 			    unsigned int payload_quadlets, __be32 *buffer,
-			    unsigned int *data_blocks)
+			    unsigned int *data_blocks, unsigned int syt)
 {
 	u32 cip_header[2];
+	unsigned int fmt, fdf;
 	unsigned int data_block_quadlets, data_block_counter, dbc_interval;
-	struct snd_pcm_substream *pcm = NULL;
+	struct snd_pcm_substream *pcm;
+	unsigned int pcm_frames;
 	bool lost;
 
 	cip_header[0] = be32_to_cpu(buffer[0]);
@@ -704,19 +456,30 @@ static int handle_in_packet(struct amdtp_stream *s,
 	 * For convenience, also check FMT field is AM824 or not.
 	 */
 	if (((cip_header[0] & CIP_EOH_MASK) == CIP_EOH) ||
-	    ((cip_header[1] & CIP_EOH_MASK) != CIP_EOH) ||
-	    ((cip_header[1] & CIP_FMT_MASK) != CIP_FMT_AM)) {
+	    ((cip_header[1] & CIP_EOH_MASK) != CIP_EOH)) {
 		dev_info_ratelimited(&s->unit->device,
 				"Invalid CIP header for AMDTP: %08X:%08X\n",
 				cip_header[0], cip_header[1]);
 		*data_blocks = 0;
+		pcm_frames = 0;
+		goto end;
+	}
+
+	/* Check valid protocol or not. */
+	fmt = (cip_header[1] & CIP_FMT_MASK) >> CIP_FMT_SHIFT;
+	if (fmt != s->fmt) {
+		dev_info_ratelimited(&s->unit->device,
+				     "Detect unexpected protocol: %08x %08x\n",
+				     cip_header[0], cip_header[1]);
+		*data_blocks = 0;
+		pcm_frames = 0;
 		goto end;
 	}
 
 	/* Calculate data blocks */
+	fdf = (cip_header[1] & CIP_FDF_MASK) >> CIP_FDF_SHIFT;
 	if (payload_quadlets < 3 ||
-	    ((cip_header[1] & CIP_FDF_MASK) ==
-				(AMDTP_FDF_NO_DATA << CIP_FDF_SHIFT))) {
+	    (fmt == CIP_FMT_AM && fdf == AMDTP_FDF_NO_DATA)) {
 		*data_blocks = 0;
 	} else {
 		data_block_quadlets =
@@ -763,16 +526,7 @@ static int handle_in_packet(struct amdtp_stream *s,
 		return -EIO;
 	}
 
-	if (*data_blocks > 0) {
-		buffer += 2;
-
-		pcm = ACCESS_ONCE(s->pcm);
-		if (pcm)
-			s->transfer_samples(s, pcm, buffer, *data_blocks);
-
-		if (s->midi_ports)
-			read_midi_messages(s, buffer, *data_blocks);
-	}
+	pcm_frames = s->process_data_blocks(s, buffer + 2, *data_blocks, &syt);
 
 	if (s->flags & CIP_DBC_IS_END_EVENT)
 		s->data_block_counter = data_block_counter;
@@ -783,8 +537,9 @@ end:
 	if (queue_in_packet(s) < 0)
 		return -EIO;
 
-	if (pcm)
-		update_pcm_pointers(s, pcm, *data_blocks);
+	pcm = ACCESS_ONCE(s->pcm);
+	if (pcm && pcm_frames > 0)
+		update_pcm_pointers(s, pcm, pcm_frames);
 
 	return 0;
 }
@@ -854,15 +609,15 @@ static void in_stream_callback(struct fw_iso_context *context, u32 cycle,
 			break;
 		}
 
+		syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK;
 		if (handle_in_packet(s, payload_quadlets, buffer,
-							&data_blocks) < 0) {
+						&data_blocks, syt) < 0) {
 			s->packet_index = -1;
 			break;
 		}
 
 		/* Process sync slave stream */
 		if (s->sync_slave && s->sync_slave->callbacked) {
-			syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK;
 			if (handle_out_packet(s->sync_slave,
 					      data_blocks, syt) < 0) {
 				s->packet_index = -1;
diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp-stream.h
index b2cf9e75693b..8775704a3665 100644
--- a/sound/firewire/amdtp.h
+++ b/sound/firewire/amdtp-stream.h
@@ -4,6 +4,7 @@
 #include <linux/err.h>
 #include <linux/interrupt.h>
 #include <linux/mutex.h>
+#include <linux/sched.h>
 #include <sound/asound.h>
 #include "packets-buffer.h"
 
@@ -80,100 +81,78 @@ enum cip_sfc {
 	CIP_SFC_COUNT
 };
 
-#define AMDTP_IN_PCM_FORMAT_BITS	SNDRV_PCM_FMTBIT_S32
-
-#define AMDTP_OUT_PCM_FORMAT_BITS	(SNDRV_PCM_FMTBIT_S16 | \
-					 SNDRV_PCM_FMTBIT_S32)
-
-
-/*
- * This module supports maximum 64 PCM channels for one PCM stream
- * This is for our convenience.
- */
-#define AMDTP_MAX_CHANNELS_FOR_PCM	64
-
-/*
- * AMDTP packet can include channels for MIDI conformant data.
- * Each MIDI conformant data channel includes 8 MPX-MIDI data stream.
- * Each MPX-MIDI data stream includes one data stream from/to MIDI ports.
- *
- * This module supports maximum 1 MIDI conformant data channels.
- * Then this AMDTP packets can transfer maximum 8 MIDI data streams.
- */
-#define AMDTP_MAX_CHANNELS_FOR_MIDI	1
-
 struct fw_unit;
 struct fw_iso_context;
 struct snd_pcm_substream;
 struct snd_pcm_runtime;
-struct snd_rawmidi_substream;
 
 enum amdtp_stream_direction {
 	AMDTP_OUT_STREAM = 0,
 	AMDTP_IN_STREAM
 };
 
+struct amdtp_stream;
+typedef unsigned int (*amdtp_stream_process_data_blocks_t)(
+						struct amdtp_stream *s,
+						__be32 *buffer,
+						unsigned int data_blocks,
+						unsigned int *syt);
 struct amdtp_stream {
 	struct fw_unit *unit;
 	enum cip_flags flags;
 	enum amdtp_stream_direction direction;
-	struct fw_iso_context *context;
 	struct mutex mutex;
 
-	enum cip_sfc sfc;
-	unsigned int data_block_quadlets;
-	unsigned int pcm_channels;
-	unsigned int midi_ports;
-	void (*transfer_samples)(struct amdtp_stream *s,
-				 struct snd_pcm_substream *pcm,
-				 __be32 *buffer, unsigned int frames);
-	u8 pcm_positions[AMDTP_MAX_CHANNELS_FOR_PCM];
-	u8 midi_position;
-
-	unsigned int syt_interval;
-	unsigned int transfer_delay;
-	unsigned int source_node_id_field;
+	/* For packet processing. */
+	struct fw_iso_context *context;
 	struct iso_packets_buffer buffer;
-
-	struct snd_pcm_substream *pcm;
-	struct tasklet_struct period_tasklet;
-
 	int packet_index;
+
+	/* For CIP headers. */
+	unsigned int source_node_id_field;
+	unsigned int data_block_quadlets;
 	unsigned int data_block_counter;
+	unsigned int fmt;
+	unsigned int fdf;
+	/* quirk: fixed interval of dbc between previos/current packets. */
+	unsigned int tx_dbc_interval;
+	/* quirk: indicate the value of dbc field in a first packet. */
+	unsigned int tx_first_dbc;
 
+	/* Internal flags. */
+	enum cip_sfc sfc;
+	unsigned int syt_interval;
+	unsigned int transfer_delay;
 	unsigned int data_block_state;
-
 	unsigned int last_syt_offset;
 	unsigned int syt_offset_state;
 
+	/* For a PCM substream processing. */
+	struct snd_pcm_substream *pcm;
+	struct tasklet_struct period_tasklet;
 	unsigned int pcm_buffer_pointer;
 	unsigned int pcm_period_pointer;
 	bool pointer_flush;
-	bool double_pcm_frames;
-
-	struct snd_rawmidi_substream *midi[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
-	int midi_fifo_limit;
-	int midi_fifo_used[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
-
-	/* quirk: fixed interval of dbc between previos/current packets. */
-	unsigned int tx_dbc_interval;
-	/* quirk: indicate the value of dbc field in a first packet. */
-	unsigned int tx_first_dbc;
 
+	/* To wait for first packet. */
 	bool callbacked;
 	wait_queue_head_t callback_wait;
 	struct amdtp_stream *sync_slave;
+
+	/* For backends to process data blocks. */
+	void *protocol;
+	amdtp_stream_process_data_blocks_t process_data_blocks;
 };
 
 int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
-		      enum amdtp_stream_direction dir,
-		      enum cip_flags flags);
+		      enum amdtp_stream_direction dir, enum cip_flags flags,
+		      unsigned int fmt,
+		      amdtp_stream_process_data_blocks_t process_data_blocks,
+		      unsigned int protocol_size);
 void amdtp_stream_destroy(struct amdtp_stream *s);
 
-void amdtp_stream_set_parameters(struct amdtp_stream *s,
-				 unsigned int rate,
-				 unsigned int pcm_channels,
-				 unsigned int midi_ports);
+int amdtp_stream_set_parameters(struct amdtp_stream *s, unsigned int rate,
+				unsigned int data_block_quadlets);
 unsigned int amdtp_stream_get_max_payload(struct amdtp_stream *s);
 
 int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed);
@@ -182,8 +161,7 @@ void amdtp_stream_stop(struct amdtp_stream *s);
 
 int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
 					struct snd_pcm_runtime *runtime);
-void amdtp_stream_set_pcm_format(struct amdtp_stream *s,
-				 snd_pcm_format_t format);
+
 void amdtp_stream_pcm_prepare(struct amdtp_stream *s);
 unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s);
 void amdtp_stream_pcm_abort(struct amdtp_stream *s);
@@ -240,24 +218,6 @@ static inline void amdtp_stream_pcm_trigger(struct amdtp_stream *s,
 	ACCESS_ONCE(s->pcm) = pcm;
 }
 
-/**
- * amdtp_stream_midi_trigger - start/stop playback/capture with a MIDI device
- * @s: the AMDTP stream
- * @port: index of MIDI port
- * @midi: the MIDI device to be started, or %NULL to stop the current device
- *
- * Call this function on a running isochronous stream to enable the actual
- * transmission of MIDI data.  This function should be called from the MIDI
- * device's .trigger callback.
- */
-static inline void amdtp_stream_midi_trigger(struct amdtp_stream *s,
-					     unsigned int port,
-					     struct snd_rawmidi_substream *midi)
-{
-	if (port < s->midi_ports)
-		ACCESS_ONCE(s->midi[port]) = midi;
-}
-
 static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc)
 {
 	return sfc & 1;
diff --git a/sound/firewire/bebob/Makefile b/sound/firewire/bebob/Makefile
index 6cf470c80d1f..af7ed6643266 100644
--- a/sound/firewire/bebob/Makefile
+++ b/sound/firewire/bebob/Makefile
@@ -1,4 +1,4 @@
 snd-bebob-objs := bebob_command.o bebob_stream.o bebob_proc.o bebob_midi.o \
 		  bebob_pcm.o bebob_hwdep.o bebob_terratec.o bebob_yamaha.o \
 		  bebob_focusrite.o bebob_maudio.o bebob.o
-obj-m += snd-bebob.o
+obj-$(CONFIG_SND_BEBOB) += snd-bebob.o
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 27a04ac8ffee..091290d1f3ea 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -41,7 +41,8 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS);
 #define VEN_EDIROL	0x000040ab
 #define VEN_PRESONUS	0x00000a92
 #define VEN_BRIDGECO	0x000007f5
-#define VEN_MACKIE	0x0000000f
+#define VEN_MACKIE1	0x0000000f
+#define VEN_MACKIE2	0x00000ff2
 #define VEN_STANTON	0x00001260
 #define VEN_TASCAM	0x0000022e
 #define VEN_BEHRINGER	0x00001564
@@ -334,7 +335,7 @@ static void bebob_remove(struct fw_unit *unit)
 	snd_card_free_when_closed(bebob->card);
 }
 
-static struct snd_bebob_rate_spec normal_rate_spec = {
+static const struct snd_bebob_rate_spec normal_rate_spec = {
 	.get	= &snd_bebob_stream_get_rate,
 	.set	= &snd_bebob_stream_set_rate
 };
@@ -360,9 +361,9 @@ static const struct ieee1394_device_id bebob_id_table[] = {
 	/* BridgeCo, Audio5 */
 	SND_BEBOB_DEV_ENTRY(VEN_BRIDGECO, 0x00010049, &spec_normal),
 	/* Mackie, Onyx 1220/1620/1640 (Firewire I/O Card) */
-	SND_BEBOB_DEV_ENTRY(VEN_MACKIE, 0x00010065, &spec_normal),
+	SND_BEBOB_DEV_ENTRY(VEN_MACKIE2, 0x00010065, &spec_normal),
 	/* Mackie, d.2 (Firewire Option) */
-	SND_BEBOB_DEV_ENTRY(VEN_MACKIE, 0x00010067, &spec_normal),
+	SND_BEBOB_DEV_ENTRY(VEN_MACKIE1, 0x00010067, &spec_normal),
 	/* Stanton, ScratchAmp */
 	SND_BEBOB_DEV_ENTRY(VEN_STANTON, 0x00000001, &spec_normal),
 	/* Tascam, IF-FW DM */
diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h
index d23caca7f369..4d8fcc78e747 100644
--- a/sound/firewire/bebob/bebob.h
+++ b/sound/firewire/bebob/bebob.h
@@ -31,7 +31,7 @@
 #include "../fcp.h"
 #include "../packets-buffer.h"
 #include "../iso-resources.h"
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
 #include "../cmp.h"
 
 /* basic register addresses on DM1000/DM1100/DM1500 */
@@ -70,9 +70,9 @@ struct snd_bebob_meter_spec {
 	int (*get)(struct snd_bebob *bebob, u32 *target, unsigned int size);
 };
 struct snd_bebob_spec {
-	struct snd_bebob_clock_spec *clock;
-	struct snd_bebob_rate_spec *rate;
-	struct snd_bebob_meter_spec *meter;
+	const struct snd_bebob_clock_spec *clock;
+	const struct snd_bebob_rate_spec *rate;
+	const struct snd_bebob_meter_spec *meter;
 };
 
 struct snd_bebob {
@@ -235,19 +235,19 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob);
 int snd_bebob_create_hwdep_device(struct snd_bebob *bebob);
 
 /* model specific operations */
-extern struct snd_bebob_spec phase88_rack_spec;
-extern struct snd_bebob_spec phase24_series_spec;
-extern struct snd_bebob_spec yamaha_go_spec;
-extern struct snd_bebob_spec saffirepro_26_spec;
-extern struct snd_bebob_spec saffirepro_10_spec;
-extern struct snd_bebob_spec saffire_le_spec;
-extern struct snd_bebob_spec saffire_spec;
-extern struct snd_bebob_spec maudio_fw410_spec;
-extern struct snd_bebob_spec maudio_audiophile_spec;
-extern struct snd_bebob_spec maudio_solo_spec;
-extern struct snd_bebob_spec maudio_ozonic_spec;
-extern struct snd_bebob_spec maudio_nrv10_spec;
-extern struct snd_bebob_spec maudio_special_spec;
+extern const struct snd_bebob_spec phase88_rack_spec;
+extern const struct snd_bebob_spec phase24_series_spec;
+extern const struct snd_bebob_spec yamaha_go_spec;
+extern const struct snd_bebob_spec saffirepro_26_spec;
+extern const struct snd_bebob_spec saffirepro_10_spec;
+extern const struct snd_bebob_spec saffire_le_spec;
+extern const struct snd_bebob_spec saffire_spec;
+extern const struct snd_bebob_spec maudio_fw410_spec;
+extern const struct snd_bebob_spec maudio_audiophile_spec;
+extern const struct snd_bebob_spec maudio_solo_spec;
+extern const struct snd_bebob_spec maudio_ozonic_spec;
+extern const struct snd_bebob_spec maudio_nrv10_spec;
+extern const struct snd_bebob_spec maudio_special_spec;
 int snd_bebob_maudio_special_discover(struct snd_bebob *bebob, bool is1814);
 int snd_bebob_maudio_load_firmware(struct fw_unit *unit);
 
diff --git a/sound/firewire/bebob/bebob_focusrite.c b/sound/firewire/bebob/bebob_focusrite.c
index a1a39494ea6c..f11090057949 100644
--- a/sound/firewire/bebob/bebob_focusrite.c
+++ b/sound/firewire/bebob/bebob_focusrite.c
@@ -200,7 +200,7 @@ end:
 	return err;
 }
 
-struct snd_bebob_spec saffire_le_spec;
+const struct snd_bebob_spec saffire_le_spec;
 static enum snd_bebob_clock_type saffire_both_clk_src_types[] = {
 	SND_BEBOB_CLOCK_TYPE_INTERNAL,
 	SND_BEBOB_CLOCK_TYPE_EXTERNAL,
@@ -229,7 +229,7 @@ static const char *const saffire_meter_labels[] = {
 static int
 saffire_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size)
 {
-	struct snd_bebob_meter_spec *spec = bebob->spec->meter;
+	const struct snd_bebob_meter_spec *spec = bebob->spec->meter;
 	unsigned int channels;
 	u64 offset;
 	int err;
@@ -260,60 +260,60 @@ saffire_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size)
 	return err;
 }
 
-static struct snd_bebob_rate_spec saffirepro_both_rate_spec = {
+static const struct snd_bebob_rate_spec saffirepro_both_rate_spec = {
 	.get	= &saffirepro_both_clk_freq_get,
 	.set	= &saffirepro_both_clk_freq_set,
 };
 /* Saffire Pro 26 I/O  */
-static struct snd_bebob_clock_spec saffirepro_26_clk_spec = {
+static const struct snd_bebob_clock_spec saffirepro_26_clk_spec = {
 	.num	= ARRAY_SIZE(saffirepro_26_clk_src_types),
 	.types	= saffirepro_26_clk_src_types,
 	.get	= &saffirepro_both_clk_src_get,
 };
-struct snd_bebob_spec saffirepro_26_spec = {
+const struct snd_bebob_spec saffirepro_26_spec = {
 	.clock	= &saffirepro_26_clk_spec,
 	.rate	= &saffirepro_both_rate_spec,
 	.meter	= NULL
 };
 /* Saffire Pro 10 I/O */
-static struct snd_bebob_clock_spec saffirepro_10_clk_spec = {
+static const struct snd_bebob_clock_spec saffirepro_10_clk_spec = {
 	.num	= ARRAY_SIZE(saffirepro_10_clk_src_types),
 	.types	= saffirepro_10_clk_src_types,
 	.get	= &saffirepro_both_clk_src_get,
 };
-struct snd_bebob_spec saffirepro_10_spec = {
+const struct snd_bebob_spec saffirepro_10_spec = {
 	.clock	= &saffirepro_10_clk_spec,
 	.rate	= &saffirepro_both_rate_spec,
 	.meter	= NULL
 };
 
-static struct snd_bebob_rate_spec saffire_both_rate_spec = {
+static const struct snd_bebob_rate_spec saffire_both_rate_spec = {
 	.get	= &snd_bebob_stream_get_rate,
 	.set	= &snd_bebob_stream_set_rate,
 };
-static struct snd_bebob_clock_spec saffire_both_clk_spec = {
+static const struct snd_bebob_clock_spec saffire_both_clk_spec = {
 	.num	= ARRAY_SIZE(saffire_both_clk_src_types),
 	.types	= saffire_both_clk_src_types,
 	.get	= &saffire_both_clk_src_get,
 };
 /* Saffire LE */
-static struct snd_bebob_meter_spec saffire_le_meter_spec = {
+static const struct snd_bebob_meter_spec saffire_le_meter_spec = {
 	.num	= ARRAY_SIZE(saffire_le_meter_labels),
 	.labels	= saffire_le_meter_labels,
 	.get	= &saffire_meter_get,
 };
-struct snd_bebob_spec saffire_le_spec = {
+const struct snd_bebob_spec saffire_le_spec = {
 	.clock	= &saffire_both_clk_spec,
 	.rate	= &saffire_both_rate_spec,
 	.meter	= &saffire_le_meter_spec
 };
 /* Saffire */
-static struct snd_bebob_meter_spec saffire_meter_spec = {
+static const struct snd_bebob_meter_spec saffire_meter_spec = {
 	.num	= ARRAY_SIZE(saffire_meter_labels),
 	.labels	= saffire_meter_labels,
 	.get	= &saffire_meter_get,
 };
-struct snd_bebob_spec saffire_spec = {
+const struct snd_bebob_spec saffire_spec = {
 	.clock	= &saffire_both_clk_spec,
 	.rate	= &saffire_both_rate_spec,
 	.meter	= &saffire_meter_spec
diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c
index 057495d54ab0..07e5abdbceb5 100644
--- a/sound/firewire/bebob/bebob_maudio.c
+++ b/sound/firewire/bebob/bebob_maudio.c
@@ -628,7 +628,7 @@ static const char *const special_meter_labels[] = {
 static int
 special_meter_get(struct snd_bebob *bebob, u32 *target, unsigned int size)
 {
-	u16 *buf;
+	__be16 *buf;
 	unsigned int i, c, channels;
 	int err;
 
@@ -687,7 +687,7 @@ static const char *const nrv10_meter_labels[] = {
 static int
 normal_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size)
 {
-	struct snd_bebob_meter_spec *spec = bebob->spec->meter;
+	const struct snd_bebob_meter_spec *spec = bebob->spec->meter;
 	unsigned int c, channels;
 	int err;
 
@@ -712,85 +712,85 @@ end:
 }
 
 /* for special customized devices */
-static struct snd_bebob_rate_spec special_rate_spec = {
+static const struct snd_bebob_rate_spec special_rate_spec = {
 	.get	= &special_get_rate,
 	.set	= &special_set_rate,
 };
-static struct snd_bebob_clock_spec special_clk_spec = {
+static const struct snd_bebob_clock_spec special_clk_spec = {
 	.num	= ARRAY_SIZE(special_clk_types),
 	.types	= special_clk_types,
 	.get	= &special_clk_get,
 };
-static struct snd_bebob_meter_spec special_meter_spec = {
+static const struct snd_bebob_meter_spec special_meter_spec = {
 	.num	= ARRAY_SIZE(special_meter_labels),
 	.labels	= special_meter_labels,
 	.get	= &special_meter_get
 };
-struct snd_bebob_spec maudio_special_spec = {
+const struct snd_bebob_spec maudio_special_spec = {
 	.clock	= &special_clk_spec,
 	.rate	= &special_rate_spec,
 	.meter	= &special_meter_spec
 };
 
 /* Firewire 410 specification */
-static struct snd_bebob_rate_spec usual_rate_spec = {
+static const struct snd_bebob_rate_spec usual_rate_spec = {
 	.get	= &snd_bebob_stream_get_rate,
 	.set	= &snd_bebob_stream_set_rate,
 };
-static struct snd_bebob_meter_spec fw410_meter_spec = {
+static const struct snd_bebob_meter_spec fw410_meter_spec = {
 	.num	= ARRAY_SIZE(fw410_meter_labels),
 	.labels	= fw410_meter_labels,
 	.get	= &normal_meter_get
 };
-struct snd_bebob_spec maudio_fw410_spec = {
+const struct snd_bebob_spec maudio_fw410_spec = {
 	.clock	= NULL,
 	.rate	= &usual_rate_spec,
 	.meter	= &fw410_meter_spec
 };
 
 /* Firewire Audiophile specification */
-static struct snd_bebob_meter_spec audiophile_meter_spec = {
+static const struct snd_bebob_meter_spec audiophile_meter_spec = {
 	.num	= ARRAY_SIZE(audiophile_meter_labels),
 	.labels	= audiophile_meter_labels,
 	.get	= &normal_meter_get
 };
-struct snd_bebob_spec maudio_audiophile_spec = {
+const struct snd_bebob_spec maudio_audiophile_spec = {
 	.clock	= NULL,
 	.rate	= &usual_rate_spec,
 	.meter	= &audiophile_meter_spec
 };
 
 /* Firewire Solo specification */
-static struct snd_bebob_meter_spec solo_meter_spec = {
+static const struct snd_bebob_meter_spec solo_meter_spec = {
 	.num	= ARRAY_SIZE(solo_meter_labels),
 	.labels	= solo_meter_labels,
 	.get	= &normal_meter_get
 };
-struct snd_bebob_spec maudio_solo_spec = {
+const struct snd_bebob_spec maudio_solo_spec = {
 	.clock	= NULL,
 	.rate	= &usual_rate_spec,
 	.meter	= &solo_meter_spec
 };
 
 /* Ozonic specification */
-static struct snd_bebob_meter_spec ozonic_meter_spec = {
+static const struct snd_bebob_meter_spec ozonic_meter_spec = {
 	.num	= ARRAY_SIZE(ozonic_meter_labels),
 	.labels	= ozonic_meter_labels,
 	.get	= &normal_meter_get
 };
-struct snd_bebob_spec maudio_ozonic_spec = {
+const struct snd_bebob_spec maudio_ozonic_spec = {
 	.clock	= NULL,
 	.rate	= &usual_rate_spec,
 	.meter	= &ozonic_meter_spec
 };
 
 /* NRV10 specification */
-static struct snd_bebob_meter_spec nrv10_meter_spec = {
+static const struct snd_bebob_meter_spec nrv10_meter_spec = {
 	.num	= ARRAY_SIZE(nrv10_meter_labels),
 	.labels	= nrv10_meter_labels,
 	.get	= &normal_meter_get
 };
-struct snd_bebob_spec maudio_nrv10_spec = {
+const struct snd_bebob_spec maudio_nrv10_spec = {
 	.clock	= NULL,
 	.rate	= &usual_rate_spec,
 	.meter	= &nrv10_meter_spec
diff --git a/sound/firewire/bebob/bebob_midi.c b/sound/firewire/bebob/bebob_midi.c
index 5681143925cd..90d95be499b0 100644
--- a/sound/firewire/bebob/bebob_midi.c
+++ b/sound/firewire/bebob/bebob_midi.c
@@ -72,11 +72,11 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
 	spin_lock_irqsave(&bebob->lock, flags);
 
 	if (up)
-		amdtp_stream_midi_trigger(&bebob->tx_stream,
-					  substrm->number, substrm);
+		amdtp_am824_midi_trigger(&bebob->tx_stream,
+					 substrm->number, substrm);
 	else
-		amdtp_stream_midi_trigger(&bebob->tx_stream,
-					  substrm->number, NULL);
+		amdtp_am824_midi_trigger(&bebob->tx_stream,
+					 substrm->number, NULL);
 
 	spin_unlock_irqrestore(&bebob->lock, flags);
 }
@@ -89,11 +89,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
 	spin_lock_irqsave(&bebob->lock, flags);
 
 	if (up)
-		amdtp_stream_midi_trigger(&bebob->rx_stream,
-					  substrm->number, substrm);
+		amdtp_am824_midi_trigger(&bebob->rx_stream,
+					 substrm->number, substrm);
 	else
-		amdtp_stream_midi_trigger(&bebob->rx_stream,
-					  substrm->number, NULL);
+		amdtp_am824_midi_trigger(&bebob->rx_stream,
+					 substrm->number, NULL);
 
 	spin_unlock_irqrestore(&bebob->lock, flags);
 }
diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c
index c0f018a61fdc..ef224d6f5c24 100644
--- a/sound/firewire/bebob/bebob_pcm.c
+++ b/sound/firewire/bebob/bebob_pcm.c
@@ -122,11 +122,11 @@ pcm_init_hw_params(struct snd_bebob *bebob,
 			   SNDRV_PCM_INFO_MMAP_VALID;
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
-		runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS;
+		runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS;
 		s = &bebob->tx_stream;
 		formations = bebob->tx_stream_formations;
 	} else {
-		runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS;
+		runtime->hw.formats = AM824_OUT_PCM_FORMAT_BITS;
 		s = &bebob->rx_stream;
 		formations = bebob->rx_stream_formations;
 	}
@@ -146,7 +146,7 @@ pcm_init_hw_params(struct snd_bebob *bebob,
 	if (err < 0)
 		goto end;
 
-	err = amdtp_stream_add_pcm_hw_constraints(s, runtime);
+	err = amdtp_am824_add_pcm_hw_constraints(s, runtime);
 end:
 	return err;
 }
@@ -155,7 +155,7 @@ static int
 pcm_open(struct snd_pcm_substream *substream)
 {
 	struct snd_bebob *bebob = substream->private_data;
-	struct snd_bebob_rate_spec *spec = bebob->spec->rate;
+	const struct snd_bebob_rate_spec *spec = bebob->spec->rate;
 	unsigned int sampling_rate;
 	enum snd_bebob_clock_type src;
 	int err;
@@ -220,8 +220,8 @@ pcm_capture_hw_params(struct snd_pcm_substream *substream,
 
 	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
 		atomic_inc(&bebob->substreams_counter);
-	amdtp_stream_set_pcm_format(&bebob->tx_stream,
-				    params_format(hw_params));
+
+	amdtp_am824_set_pcm_format(&bebob->tx_stream, params_format(hw_params));
 
 	return 0;
 }
@@ -239,8 +239,8 @@ pcm_playback_hw_params(struct snd_pcm_substream *substream,
 
 	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
 		atomic_inc(&bebob->substreams_counter);
-	amdtp_stream_set_pcm_format(&bebob->rx_stream,
-				    params_format(hw_params));
+
+	amdtp_am824_set_pcm_format(&bebob->rx_stream, params_format(hw_params));
 
 	return 0;
 }
diff --git a/sound/firewire/bebob/bebob_proc.c b/sound/firewire/bebob/bebob_proc.c
index 301cc6a93945..ec24f96794f5 100644
--- a/sound/firewire/bebob/bebob_proc.c
+++ b/sound/firewire/bebob/bebob_proc.c
@@ -73,7 +73,7 @@ proc_read_meters(struct snd_info_entry *entry,
 		 struct snd_info_buffer *buffer)
 {
 	struct snd_bebob *bebob = entry->private_data;
-	struct snd_bebob_meter_spec *spec = bebob->spec->meter;
+	const struct snd_bebob_meter_spec *spec = bebob->spec->meter;
 	u32 *buf;
 	unsigned int i, c, channels, size;
 
@@ -138,8 +138,8 @@ proc_read_clock(struct snd_info_entry *entry,
 		"SYT-Match",
 	};
 	struct snd_bebob *bebob = entry->private_data;
-	struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
-	struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
+	const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
+	const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
 	enum snd_bebob_clock_type src;
 	unsigned int rate;
 
diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c
index 5be5242e1ed8..926e5dcbb66a 100644
--- a/sound/firewire/bebob/bebob_stream.c
+++ b/sound/firewire/bebob/bebob_stream.c
@@ -119,7 +119,7 @@ end:
 int snd_bebob_stream_get_clock_src(struct snd_bebob *bebob,
 				   enum snd_bebob_clock_type *src)
 {
-	struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
+	const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
 	u8 addr[AVC_BRIDGECO_ADDR_BYTES], input[7];
 	unsigned int id;
 	enum avc_bridgeco_plug_type type;
@@ -338,7 +338,7 @@ map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s)
 					err = -ENOSYS;
 					goto end;
 				}
-				s->midi_position = stm_pos;
+				amdtp_am824_set_midi_position(s, stm_pos);
 				midi = stm_pos;
 				break;
 			/* for PCM data channel */
@@ -354,11 +354,12 @@ map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s)
 			case 0x09:	/* Digital */
 			default:
 				location = pcm + sec_loc;
-				if (location >= AMDTP_MAX_CHANNELS_FOR_PCM) {
+				if (location >= AM824_MAX_CHANNELS_FOR_PCM) {
 					err = -ENOSYS;
 					goto end;
 				}
-				s->pcm_positions[location] = stm_pos;
+				amdtp_am824_set_pcm_position(s, location,
+							     stm_pos);
 				break;
 			}
 		}
@@ -427,12 +428,19 @@ make_both_connections(struct snd_bebob *bebob, unsigned int rate)
 	index = get_formation_index(rate);
 	pcm_channels = bebob->tx_stream_formations[index].pcm;
 	midi_channels = bebob->tx_stream_formations[index].midi;
-	amdtp_stream_set_parameters(&bebob->tx_stream,
-				    rate, pcm_channels, midi_channels * 8);
+	err = amdtp_am824_set_parameters(&bebob->tx_stream, rate,
+					 pcm_channels, midi_channels * 8,
+					 false);
+	if (err < 0)
+		goto end;
+
 	pcm_channels = bebob->rx_stream_formations[index].pcm;
 	midi_channels = bebob->rx_stream_formations[index].midi;
-	amdtp_stream_set_parameters(&bebob->rx_stream,
-				    rate, pcm_channels, midi_channels * 8);
+	err = amdtp_am824_set_parameters(&bebob->rx_stream, rate,
+					 pcm_channels, midi_channels * 8,
+					 false);
+	if (err < 0)
+		goto end;
 
 	/* establish connections for both streams */
 	err = cmp_connection_establish(&bebob->out_conn,
@@ -530,8 +538,8 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob)
 	if (err < 0)
 		goto end;
 
-	err = amdtp_stream_init(&bebob->tx_stream, bebob->unit,
-				AMDTP_IN_STREAM, CIP_BLOCKING);
+	err = amdtp_am824_init(&bebob->tx_stream, bebob->unit,
+			       AMDTP_IN_STREAM, CIP_BLOCKING);
 	if (err < 0) {
 		amdtp_stream_destroy(&bebob->tx_stream);
 		destroy_both_connections(bebob);
@@ -559,8 +567,8 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob)
 	if (bebob->maudio_special_quirk)
 		bebob->tx_stream.flags |= CIP_EMPTY_HAS_WRONG_DBC;
 
-	err = amdtp_stream_init(&bebob->rx_stream, bebob->unit,
-				AMDTP_OUT_STREAM, CIP_BLOCKING);
+	err = amdtp_am824_init(&bebob->rx_stream, bebob->unit,
+			       AMDTP_OUT_STREAM, CIP_BLOCKING);
 	if (err < 0) {
 		amdtp_stream_destroy(&bebob->tx_stream);
 		amdtp_stream_destroy(&bebob->rx_stream);
@@ -572,7 +580,7 @@ end:
 
 int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate)
 {
-	struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
+	const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
 	struct amdtp_stream *master, *slave;
 	enum cip_flags sync_mode;
 	unsigned int curr_rate;
@@ -864,8 +872,8 @@ parse_stream_formation(u8 *buf, unsigned int len,
 		}
 	}
 
-	if (formation[i].pcm  > AMDTP_MAX_CHANNELS_FOR_PCM ||
-	    formation[i].midi > AMDTP_MAX_CHANNELS_FOR_MIDI)
+	if (formation[i].pcm  > AM824_MAX_CHANNELS_FOR_PCM ||
+	    formation[i].midi > AM824_MAX_CHANNELS_FOR_MIDI)
 		return -ENOSYS;
 
 	return 0;
@@ -959,7 +967,7 @@ end:
 
 int snd_bebob_stream_discover(struct snd_bebob *bebob)
 {
-	struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
+	const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
 	u8 plugs[AVC_PLUG_INFO_BUF_BYTES], addr[AVC_BRIDGECO_ADDR_BYTES];
 	enum avc_bridgeco_plug_type type;
 	unsigned int i;
diff --git a/sound/firewire/bebob/bebob_terratec.c b/sound/firewire/bebob/bebob_terratec.c
index 9242e33d2cf1..c38358b82ada 100644
--- a/sound/firewire/bebob/bebob_terratec.c
+++ b/sound/firewire/bebob/bebob_terratec.c
@@ -55,30 +55,30 @@ phase24_series_clk_src_get(struct snd_bebob *bebob, unsigned int *id)
 	return 0;
 }
 
-static struct snd_bebob_rate_spec phase_series_rate_spec = {
+static const struct snd_bebob_rate_spec phase_series_rate_spec = {
 	.get	= &snd_bebob_stream_get_rate,
 	.set	= &snd_bebob_stream_set_rate,
 };
 
 /* PHASE 88 Rack FW */
-static struct snd_bebob_clock_spec phase88_rack_clk = {
+static const struct snd_bebob_clock_spec phase88_rack_clk = {
 	.num	= ARRAY_SIZE(phase88_rack_clk_src_types),
 	.types	= phase88_rack_clk_src_types,
 	.get	= &phase88_rack_clk_src_get,
 };
-struct snd_bebob_spec phase88_rack_spec = {
+const struct snd_bebob_spec phase88_rack_spec = {
 	.clock	= &phase88_rack_clk,
 	.rate	= &phase_series_rate_spec,
 	.meter	= NULL
 };
 
 /* 'PHASE 24 FW' and 'PHASE X24 FW' */
-static struct snd_bebob_clock_spec phase24_series_clk = {
+static const struct snd_bebob_clock_spec phase24_series_clk = {
 	.num	= ARRAY_SIZE(phase24_series_clk_src_types),
 	.types	= phase24_series_clk_src_types,
 	.get	= &phase24_series_clk_src_get,
 };
-struct snd_bebob_spec phase24_series_spec = {
+const struct snd_bebob_spec phase24_series_spec = {
 	.clock	= &phase24_series_clk,
 	.rate	= &phase_series_rate_spec,
 	.meter	= NULL
diff --git a/sound/firewire/bebob/bebob_yamaha.c b/sound/firewire/bebob/bebob_yamaha.c
index 58101702410b..90d4404f77ce 100644
--- a/sound/firewire/bebob/bebob_yamaha.c
+++ b/sound/firewire/bebob/bebob_yamaha.c
@@ -46,16 +46,16 @@ clk_src_get(struct snd_bebob *bebob, unsigned int *id)
 
 	return 0;
 }
-static struct snd_bebob_clock_spec clock_spec = {
+static const struct snd_bebob_clock_spec clock_spec = {
 	.num	= ARRAY_SIZE(clk_src_types),
 	.types	= clk_src_types,
 	.get	= &clk_src_get,
 };
-static struct snd_bebob_rate_spec rate_spec = {
+static const struct snd_bebob_rate_spec rate_spec = {
 	.get	= &snd_bebob_stream_get_rate,
 	.set	= &snd_bebob_stream_set_rate,
 };
-struct snd_bebob_spec yamaha_go_spec = {
+const struct snd_bebob_spec yamaha_go_spec = {
 	.clock	= &clock_spec,
 	.rate	= &rate_spec,
 	.meter	= NULL
diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile
index 9ef228ef7baf..55b4be9b0034 100644
--- a/sound/firewire/dice/Makefile
+++ b/sound/firewire/dice/Makefile
@@ -1,3 +1,3 @@
 snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \
 		 dice-pcm.o dice-hwdep.o dice.o
-obj-m += snd-dice.o
+obj-$(CONFIG_SND_DICE) += snd-dice.o
diff --git a/sound/firewire/dice/dice-midi.c b/sound/firewire/dice/dice-midi.c
index fe43ce791f84..151b09f240f2 100644
--- a/sound/firewire/dice/dice-midi.c
+++ b/sound/firewire/dice/dice-midi.c
@@ -52,10 +52,10 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
 	spin_lock_irqsave(&dice->lock, flags);
 
 	if (up)
-		amdtp_stream_midi_trigger(&dice->tx_stream,
+		amdtp_am824_midi_trigger(&dice->tx_stream,
 					  substrm->number, substrm);
 	else
-		amdtp_stream_midi_trigger(&dice->tx_stream,
+		amdtp_am824_midi_trigger(&dice->tx_stream,
 					  substrm->number, NULL);
 
 	spin_unlock_irqrestore(&dice->lock, flags);
@@ -69,11 +69,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
 	spin_lock_irqsave(&dice->lock, flags);
 
 	if (up)
-		amdtp_stream_midi_trigger(&dice->rx_stream,
-					  substrm->number, substrm);
+		amdtp_am824_midi_trigger(&dice->rx_stream,
+					 substrm->number, substrm);
 	else
-		amdtp_stream_midi_trigger(&dice->rx_stream,
-					  substrm->number, NULL);
+		amdtp_am824_midi_trigger(&dice->rx_stream,
+					 substrm->number, NULL);
 
 	spin_unlock_irqrestore(&dice->lock, flags);
 }
diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c
index 4e67b1da0fe6..9b3431999fc8 100644
--- a/sound/firewire/dice/dice-pcm.c
+++ b/sound/firewire/dice/dice-pcm.c
@@ -133,11 +133,11 @@ static int init_hw_info(struct snd_dice *dice,
 		   SNDRV_PCM_INFO_BLOCK_TRANSFER;
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
-		hw->formats = AMDTP_IN_PCM_FORMAT_BITS;
+		hw->formats = AM824_IN_PCM_FORMAT_BITS;
 		stream = &dice->tx_stream;
 		pcm_channels = dice->tx_channels;
 	} else {
-		hw->formats = AMDTP_OUT_PCM_FORMAT_BITS;
+		hw->formats = AM824_OUT_PCM_FORMAT_BITS;
 		stream = &dice->rx_stream;
 		pcm_channels = dice->rx_channels;
 	}
@@ -156,7 +156,7 @@ static int init_hw_info(struct snd_dice *dice,
 	if (err < 0)
 		goto end;
 
-	err = amdtp_stream_add_pcm_hw_constraints(stream, runtime);
+	err = amdtp_am824_add_pcm_hw_constraints(stream, runtime);
 end:
 	return err;
 }
@@ -243,8 +243,7 @@ static int capture_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&dice->mutex);
 	}
 
-	amdtp_stream_set_pcm_format(&dice->tx_stream,
-				    params_format(hw_params));
+	amdtp_am824_set_pcm_format(&dice->tx_stream, params_format(hw_params));
 
 	return 0;
 }
@@ -265,8 +264,7 @@ static int playback_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&dice->mutex);
 	}
 
-	amdtp_stream_set_pcm_format(&dice->rx_stream,
-				    params_format(hw_params));
+	amdtp_am824_set_pcm_format(&dice->rx_stream, params_format(hw_params));
 
 	return 0;
 }
diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index 07dbd01d7a6b..a6a39f7ef58d 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -44,16 +44,16 @@ int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate,
 static void release_resources(struct snd_dice *dice,
 			      struct fw_iso_resources *resources)
 {
-	unsigned int channel;
+	__be32 channel;
 
 	/* Reset channel number */
 	channel = cpu_to_be32((u32)-1);
 	if (resources == &dice->tx_resources)
 		snd_dice_transaction_write_tx(dice, TX_ISOCHRONOUS,
-					      &channel, 4);
+					      &channel, sizeof(channel));
 	else
 		snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS,
-					      &channel, 4);
+					      &channel, sizeof(channel));
 
 	fw_iso_resources_free(resources);
 }
@@ -62,7 +62,7 @@ static int keep_resources(struct snd_dice *dice,
 			  struct fw_iso_resources *resources,
 			  unsigned int max_payload_bytes)
 {
-	unsigned int channel;
+	__be32 channel;
 	int err;
 
 	err = fw_iso_resources_allocate(resources, max_payload_bytes,
@@ -74,10 +74,10 @@ static int keep_resources(struct snd_dice *dice,
 	channel = cpu_to_be32(resources->channel);
 	if (resources == &dice->tx_resources)
 		err = snd_dice_transaction_write_tx(dice, TX_ISOCHRONOUS,
-						    &channel, 4);
+						    &channel, sizeof(channel));
 	else
 		err = snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS,
-						    &channel, 4);
+						    &channel, sizeof(channel));
 	if (err < 0)
 		release_resources(dice, resources);
 end:
@@ -100,6 +100,7 @@ static int start_stream(struct snd_dice *dice, struct amdtp_stream *stream,
 {
 	struct fw_iso_resources *resources;
 	unsigned int i, mode, pcm_chs, midi_ports;
+	bool double_pcm_frames;
 	int err;
 
 	err = snd_dice_stream_get_rate_mode(dice, rate, &mode);
@@ -125,21 +126,24 @@ static int start_stream(struct snd_dice *dice, struct amdtp_stream *stream,
 	 * For this quirk, blocking mode is required and PCM buffer size should
 	 * be aligned to SYT_INTERVAL.
 	 */
-	if (mode > 1) {
+	double_pcm_frames = mode > 1;
+	if (double_pcm_frames) {
 		rate /= 2;
 		pcm_chs *= 2;
-		stream->double_pcm_frames = true;
-	} else {
-		stream->double_pcm_frames = false;
 	}
 
-	amdtp_stream_set_parameters(stream, rate, pcm_chs, midi_ports);
-	if (mode > 1) {
+	err = amdtp_am824_set_parameters(stream, rate, pcm_chs, midi_ports,
+					 double_pcm_frames);
+	if (err < 0)
+		goto end;
+
+	if (double_pcm_frames) {
 		pcm_chs /= 2;
 
 		for (i = 0; i < pcm_chs; i++) {
-			stream->pcm_positions[i] = i * 2;
-			stream->pcm_positions[i + pcm_chs] = i * 2 + 1;
+			amdtp_am824_set_pcm_position(stream, i, i * 2);
+			amdtp_am824_set_pcm_position(stream, i + pcm_chs,
+						     i * 2 + 1);
 		}
 	}
 
@@ -302,7 +306,7 @@ static int init_stream(struct snd_dice *dice, struct amdtp_stream *stream)
 		goto end;
 	resources->channels_mask = 0x00000000ffffffffuLL;
 
-	err = amdtp_stream_init(stream, dice->unit, dir, CIP_BLOCKING);
+	err = amdtp_am824_init(stream, dice->unit, dir, CIP_BLOCKING);
 	if (err < 0) {
 		amdtp_stream_destroy(stream);
 		fw_iso_resources_destroy(resources);
diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 70a111d7f428..5d99436dfcae 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -29,7 +29,8 @@ static int dice_interface_check(struct fw_unit *unit)
 	struct fw_csr_iterator it;
 	int key, val, vendor = -1, model = -1, err;
 	unsigned int category, i;
-	__be32 *pointers, value;
+	__be32 *pointers;
+	u32 value;
 	__be32 version;
 
 	pointers = kmalloc_array(ARRAY_SIZE(min_values), sizeof(__be32),
diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h
index ecf5dc862235..101550ac1a24 100644
--- a/sound/firewire/dice/dice.h
+++ b/sound/firewire/dice/dice.h
@@ -34,7 +34,7 @@
 #include <sound/pcm_params.h>
 #include <sound/rawmidi.h>
 
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
 #include "../iso-resources.h"
 #include "../lib.h"
 #include "dice-interface.h"
diff --git a/sound/firewire/digi00x/Makefile b/sound/firewire/digi00x/Makefile
new file mode 100644
index 000000000000..1123e68c8b28
--- /dev/null
+++ b/sound/firewire/digi00x/Makefile
@@ -0,0 +1,4 @@
+snd-firewire-digi00x-objs := amdtp-dot.o digi00x-stream.o digi00x-proc.o \
+			     digi00x-pcm.o digi00x-hwdep.o \
+			     digi00x-transaction.o digi00x-midi.o digi00x.o
+obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += snd-firewire-digi00x.o
diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c
new file mode 100644
index 000000000000..b02a5e8cad44
--- /dev/null
+++ b/sound/firewire/digi00x/amdtp-dot.c
@@ -0,0 +1,442 @@
+/*
+ * amdtp-dot.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ * Copyright (C) 2012 Robin Gareus <robin@gareus.org>
+ * Copyright (C) 2012 Damien Zammit <damien@zamaudio.com>
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <sound/pcm.h>
+#include "digi00x.h"
+
+#define CIP_FMT_AM		0x10
+
+/* 'Clock-based rate control mode' is just supported. */
+#define AMDTP_FDF_AM824		0x00
+
+/*
+ * Nominally 3125 bytes/second, but the MIDI port's clock might be
+ * 1% too slow, and the bus clock 100 ppm too fast.
+ */
+#define MIDI_BYTES_PER_SECOND	3093
+
+/*
+ * Several devices look only at the first eight data blocks.
+ * In any case, this is more than enough for the MIDI data rate.
+ */
+#define MAX_MIDI_RX_BLOCKS	8
+
+/*
+ * The double-oh-three algorithm was discovered by Robin Gareus and Damien
+ * Zammit in 2012, with reverse-engineering for Digi 003 Rack.
+ */
+struct dot_state {
+	u8 carry;
+	u8 idx;
+	unsigned int off;
+};
+
+struct amdtp_dot {
+	unsigned int pcm_channels;
+	struct dot_state state;
+
+	unsigned int midi_ports;
+	/* 2 = MAX(DOT_MIDI_IN_PORTS, DOT_MIDI_OUT_PORTS) */
+	struct snd_rawmidi_substream *midi[2];
+	int midi_fifo_used[2];
+	int midi_fifo_limit;
+
+	void (*transfer_samples)(struct amdtp_stream *s,
+				 struct snd_pcm_substream *pcm,
+				 __be32 *buffer, unsigned int frames);
+};
+
+/*
+ * double-oh-three look up table
+ *
+ * @param idx index byte (audio-sample data) 0x00..0xff
+ * @param off channel offset shift
+ * @return salt to XOR with given data
+ */
+#define BYTE_PER_SAMPLE (4)
+#define MAGIC_DOT_BYTE (2)
+#define MAGIC_BYTE_OFF(x) (((x) * BYTE_PER_SAMPLE) + MAGIC_DOT_BYTE)
+static const u8 dot_scrt(const u8 idx, const unsigned int off)
+{
+	/*
+	 * the length of the added pattern only depends on the lower nibble
+	 * of the last non-zero data
+	 */
+	static const u8 len[16] = {0, 1, 3, 5, 7, 9, 11, 13, 14,
+				   12, 10, 8, 6, 4, 2, 0};
+
+	/*
+	 * the lower nibble of the salt. Interleaved sequence.
+	 * this is walked backwards according to len[]
+	 */
+	static const u8 nib[15] = {0x8, 0x7, 0x9, 0x6, 0xa, 0x5, 0xb, 0x4,
+				   0xc, 0x3, 0xd, 0x2, 0xe, 0x1, 0xf};
+
+	/* circular list for the salt's hi nibble. */
+	static const u8 hir[15] = {0x0, 0x6, 0xf, 0x8, 0x7, 0x5, 0x3, 0x4,
+				   0xc, 0xd, 0xe, 0x1, 0x2, 0xb, 0xa};
+
+	/*
+	 * start offset for upper nibble mapping.
+	 * note: 9 is /special/. In the case where the high nibble == 0x9,
+	 * hir[] is not used and - coincidentally - the salt's hi nibble is
+	 * 0x09 regardless of the offset.
+	 */
+	static const u8 hio[16] = {0, 11, 12, 6, 7, 5, 1, 4,
+				   3, 0x00, 14, 13, 8, 9, 10, 2};
+
+	const u8 ln = idx & 0xf;
+	const u8 hn = (idx >> 4) & 0xf;
+	const u8 hr = (hn == 0x9) ? 0x9 : hir[(hio[hn] + off) % 15];
+
+	if (len[ln] < off)
+		return 0x00;
+
+	return ((nib[14 + off - len[ln]]) | (hr << 4));
+}
+
+static void dot_encode_step(struct dot_state *state, __be32 *const buffer)
+{
+	u8 * const data = (u8 *) buffer;
+
+	if (data[MAGIC_DOT_BYTE] != 0x00) {
+		state->off = 0;
+		state->idx = data[MAGIC_DOT_BYTE] ^ state->carry;
+	}
+	data[MAGIC_DOT_BYTE] ^= state->carry;
+	state->carry = dot_scrt(state->idx, ++(state->off));
+}
+
+int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate,
+			     unsigned int pcm_channels)
+{
+	struct amdtp_dot *p = s->protocol;
+	int err;
+
+	if (amdtp_stream_running(s))
+		return -EBUSY;
+
+	/*
+	 * A first data channel is for MIDI conformant data channel, the rest is
+	 * Multi Bit Linear Audio data channel.
+	 */
+	err = amdtp_stream_set_parameters(s, rate, pcm_channels + 1);
+	if (err < 0)
+		return err;
+
+	s->fdf = AMDTP_FDF_AM824 | s->sfc;
+
+	p->pcm_channels = pcm_channels;
+
+	if (s->direction == AMDTP_IN_STREAM)
+		p->midi_ports = DOT_MIDI_IN_PORTS;
+	else
+		p->midi_ports = DOT_MIDI_OUT_PORTS;
+
+	/*
+	 * We do not know the actual MIDI FIFO size of most devices.  Just
+	 * assume two bytes, i.e., one byte can be received over the bus while
+	 * the previous one is transmitted over MIDI.
+	 * (The value here is adjusted for midi_ratelimit_per_packet().)
+	 */
+	p->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1;
+
+	return 0;
+}
+
+static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+			  __be32 *buffer, unsigned int frames)
+{
+	struct amdtp_dot *p = s->protocol;
+	struct snd_pcm_runtime *runtime = pcm->runtime;
+	unsigned int channels, remaining_frames, i, c;
+	const u32 *src;
+
+	channels = p->pcm_channels;
+	src = (void *)runtime->dma_area +
+			frames_to_bytes(runtime, s->pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+	buffer++;
+	for (i = 0; i < frames; ++i) {
+		for (c = 0; c < channels; ++c) {
+			buffer[c] = cpu_to_be32((*src >> 8) | 0x40000000);
+			dot_encode_step(&p->state, &buffer[c]);
+			src++;
+		}
+		buffer += s->data_block_quadlets;
+		if (--remaining_frames == 0)
+			src = (void *)runtime->dma_area;
+	}
+}
+
+static void write_pcm_s16(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+			  __be32 *buffer, unsigned int frames)
+{
+	struct amdtp_dot *p = s->protocol;
+	struct snd_pcm_runtime *runtime = pcm->runtime;
+	unsigned int channels, remaining_frames, i, c;
+	const u16 *src;
+
+	channels = p->pcm_channels;
+	src = (void *)runtime->dma_area +
+			frames_to_bytes(runtime, s->pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+	buffer++;
+	for (i = 0; i < frames; ++i) {
+		for (c = 0; c < channels; ++c) {
+			buffer[c] = cpu_to_be32((*src << 8) | 0x40000000);
+			dot_encode_step(&p->state, &buffer[c]);
+			src++;
+		}
+		buffer += s->data_block_quadlets;
+		if (--remaining_frames == 0)
+			src = (void *)runtime->dma_area;
+	}
+}
+
+static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+			 __be32 *buffer, unsigned int frames)
+{
+	struct amdtp_dot *p = s->protocol;
+	struct snd_pcm_runtime *runtime = pcm->runtime;
+	unsigned int channels, remaining_frames, i, c;
+	u32 *dst;
+
+	channels = p->pcm_channels;
+	dst  = (void *)runtime->dma_area +
+			frames_to_bytes(runtime, s->pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+	buffer++;
+	for (i = 0; i < frames; ++i) {
+		for (c = 0; c < channels; ++c) {
+			*dst = be32_to_cpu(buffer[c]) << 8;
+			dst++;
+		}
+		buffer += s->data_block_quadlets;
+		if (--remaining_frames == 0)
+			dst = (void *)runtime->dma_area;
+	}
+}
+
+static void write_pcm_silence(struct amdtp_stream *s, __be32 *buffer,
+			      unsigned int data_blocks)
+{
+	struct amdtp_dot *p = s->protocol;
+	unsigned int channels, i, c;
+
+	channels = p->pcm_channels;
+
+	buffer++;
+	for (i = 0; i < data_blocks; ++i) {
+		for (c = 0; c < channels; ++c)
+			buffer[c] = cpu_to_be32(0x40000000);
+		buffer += s->data_block_quadlets;
+	}
+}
+
+static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port)
+{
+	struct amdtp_dot *p = s->protocol;
+	int used;
+
+	used = p->midi_fifo_used[port];
+	if (used == 0)
+		return true;
+
+	used -= MIDI_BYTES_PER_SECOND * s->syt_interval;
+	used = max(used, 0);
+	p->midi_fifo_used[port] = used;
+
+	return used < p->midi_fifo_limit;
+}
+
+static inline void midi_use_bytes(struct amdtp_stream *s,
+				  unsigned int port, unsigned int count)
+{
+	struct amdtp_dot *p = s->protocol;
+
+	p->midi_fifo_used[port] += amdtp_rate_table[s->sfc] * count;
+}
+
+static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer,
+				unsigned int data_blocks)
+{
+	struct amdtp_dot *p = s->protocol;
+	unsigned int f, port;
+	int len;
+	u8 *b;
+
+	for (f = 0; f < data_blocks; f++) {
+		port = (s->data_block_counter + f) % 8;
+		b = (u8 *)&buffer[0];
+
+		len = 0;
+		if (port < p->midi_ports &&
+		    midi_ratelimit_per_packet(s, port) &&
+		    p->midi[port] != NULL)
+			len = snd_rawmidi_transmit(p->midi[port], b + 1, 2);
+
+		if (len > 0) {
+			b[3] = (0x10 << port) | len;
+			midi_use_bytes(s, port, len);
+		} else {
+			b[1] = 0;
+			b[2] = 0;
+			b[3] = 0;
+		}
+		b[0] = 0x80;
+
+		buffer += s->data_block_quadlets;
+	}
+}
+
+static void read_midi_messages(struct amdtp_stream *s, __be32 *buffer,
+			       unsigned int data_blocks)
+{
+	struct amdtp_dot *p = s->protocol;
+	unsigned int f, port, len;
+	u8 *b;
+
+	for (f = 0; f < data_blocks; f++) {
+		b = (u8 *)&buffer[0];
+		port = b[3] >> 4;
+		len = b[3] & 0x0f;
+
+		if (port < p->midi_ports && p->midi[port] && len > 0)
+			snd_rawmidi_receive(p->midi[port], b + 1, len);
+
+		buffer += s->data_block_quadlets;
+	}
+}
+
+int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s,
+				     struct snd_pcm_runtime *runtime)
+{
+	int err;
+
+	/* This protocol delivers 24 bit data in 32bit data channel. */
+	err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+	if (err < 0)
+		return err;
+
+	return amdtp_stream_add_pcm_hw_constraints(s, runtime);
+}
+
+void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
+{
+	struct amdtp_dot *p = s->protocol;
+
+	if (WARN_ON(amdtp_stream_pcm_running(s)))
+		return;
+
+	switch (format) {
+	default:
+		WARN_ON(1);
+		/* fall through */
+	case SNDRV_PCM_FORMAT_S16:
+		if (s->direction == AMDTP_OUT_STREAM) {
+			p->transfer_samples = write_pcm_s16;
+			break;
+		}
+		WARN_ON(1);
+		/* fall through */
+	case SNDRV_PCM_FORMAT_S32:
+		if (s->direction == AMDTP_OUT_STREAM)
+			p->transfer_samples = write_pcm_s32;
+		else
+			p->transfer_samples = read_pcm_s32;
+		break;
+	}
+}
+
+void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port,
+			  struct snd_rawmidi_substream *midi)
+{
+	struct amdtp_dot *p = s->protocol;
+
+	if (port < p->midi_ports)
+		ACCESS_ONCE(p->midi[port]) = midi;
+}
+
+static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
+					   __be32 *buffer,
+					   unsigned int data_blocks,
+					   unsigned int *syt)
+{
+	struct amdtp_dot *p = (struct amdtp_dot *)s->protocol;
+	struct snd_pcm_substream *pcm;
+	unsigned int pcm_frames;
+
+	pcm = ACCESS_ONCE(s->pcm);
+	if (pcm) {
+		p->transfer_samples(s, pcm, buffer, data_blocks);
+		pcm_frames = data_blocks;
+	} else {
+		pcm_frames = 0;
+	}
+
+	read_midi_messages(s, buffer, data_blocks);
+
+	return pcm_frames;
+}
+
+static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
+					   __be32 *buffer,
+					   unsigned int data_blocks,
+					   unsigned int *syt)
+{
+	struct amdtp_dot *p = (struct amdtp_dot *)s->protocol;
+	struct snd_pcm_substream *pcm;
+	unsigned int pcm_frames;
+
+	pcm = ACCESS_ONCE(s->pcm);
+	if (pcm) {
+		p->transfer_samples(s, pcm, buffer, data_blocks);
+		pcm_frames = data_blocks;
+	} else {
+		write_pcm_silence(s, buffer, data_blocks);
+		pcm_frames = 0;
+	}
+
+	write_midi_messages(s, buffer, data_blocks);
+
+	return pcm_frames;
+}
+
+int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit,
+		 enum amdtp_stream_direction dir)
+{
+	amdtp_stream_process_data_blocks_t process_data_blocks;
+	enum cip_flags flags;
+
+	/* Use different mode between incoming/outgoing. */
+	if (dir == AMDTP_IN_STREAM) {
+		flags = CIP_NONBLOCKING | CIP_SKIP_INIT_DBC_CHECK;
+		process_data_blocks = process_tx_data_blocks;
+	} else {
+		flags = CIP_BLOCKING;
+		process_data_blocks = process_rx_data_blocks;
+	}
+
+	return amdtp_stream_init(s, unit, dir, flags, CIP_FMT_AM,
+				 process_data_blocks, sizeof(struct amdtp_dot));
+}
+
+void amdtp_dot_reset(struct amdtp_stream *s)
+{
+	struct amdtp_dot *p = s->protocol;
+
+	p->state.carry = 0x00;
+	p->state.idx = 0x00;
+	p->state.off = 0;
+}
diff --git a/sound/firewire/digi00x/digi00x-hwdep.c b/sound/firewire/digi00x/digi00x-hwdep.c
new file mode 100644
index 000000000000..f188e4758fd2
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-hwdep.c
@@ -0,0 +1,200 @@
+/*
+ * digi00x-hwdep.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+/*
+ * This codes give three functionality.
+ *
+ * 1.get firewire node information
+ * 2.get notification about starting/stopping stream
+ * 3.lock/unlock stream
+ * 4.get asynchronous messaging
+ */
+
+#include "digi00x.h"
+
+static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf,  long count,
+		       loff_t *offset)
+{
+	struct snd_dg00x *dg00x = hwdep->private_data;
+	DEFINE_WAIT(wait);
+	union snd_firewire_event event;
+
+	spin_lock_irq(&dg00x->lock);
+
+	while (!dg00x->dev_lock_changed && dg00x->msg == 0) {
+		prepare_to_wait(&dg00x->hwdep_wait, &wait, TASK_INTERRUPTIBLE);
+		spin_unlock_irq(&dg00x->lock);
+		schedule();
+		finish_wait(&dg00x->hwdep_wait, &wait);
+		if (signal_pending(current))
+			return -ERESTARTSYS;
+		spin_lock_irq(&dg00x->lock);
+	}
+
+	memset(&event, 0, sizeof(event));
+	if (dg00x->dev_lock_changed) {
+		event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS;
+		event.lock_status.status = (dg00x->dev_lock_count > 0);
+		dg00x->dev_lock_changed = false;
+
+		count = min_t(long, count, sizeof(event.lock_status));
+	} else {
+		event.digi00x_message.type =
+					SNDRV_FIREWIRE_EVENT_DIGI00X_MESSAGE;
+		event.digi00x_message.message = dg00x->msg;
+		dg00x->msg = 0;
+
+		count = min_t(long, count, sizeof(event.digi00x_message));
+	}
+
+	spin_unlock_irq(&dg00x->lock);
+
+	if (copy_to_user(buf, &event, count))
+		return -EFAULT;
+
+	return count;
+}
+
+static unsigned int hwdep_poll(struct snd_hwdep *hwdep, struct file *file,
+			       poll_table *wait)
+{
+	struct snd_dg00x *dg00x = hwdep->private_data;
+	unsigned int events;
+
+	poll_wait(file, &dg00x->hwdep_wait, wait);
+
+	spin_lock_irq(&dg00x->lock);
+	if (dg00x->dev_lock_changed || dg00x->msg)
+		events = POLLIN | POLLRDNORM;
+	else
+		events = 0;
+	spin_unlock_irq(&dg00x->lock);
+
+	return events;
+}
+
+static int hwdep_get_info(struct snd_dg00x *dg00x, void __user *arg)
+{
+	struct fw_device *dev = fw_parent_device(dg00x->unit);
+	struct snd_firewire_get_info info;
+
+	memset(&info, 0, sizeof(info));
+	info.type = SNDRV_FIREWIRE_TYPE_DIGI00X;
+	info.card = dev->card->index;
+	*(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]);
+	*(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]);
+	strlcpy(info.device_name, dev_name(&dev->device),
+		sizeof(info.device_name));
+
+	if (copy_to_user(arg, &info, sizeof(info)))
+		return -EFAULT;
+
+	return 0;
+}
+
+static int hwdep_lock(struct snd_dg00x *dg00x)
+{
+	int err;
+
+	spin_lock_irq(&dg00x->lock);
+
+	if (dg00x->dev_lock_count == 0) {
+		dg00x->dev_lock_count = -1;
+		err = 0;
+	} else {
+		err = -EBUSY;
+	}
+
+	spin_unlock_irq(&dg00x->lock);
+
+	return err;
+}
+
+static int hwdep_unlock(struct snd_dg00x *dg00x)
+{
+	int err;
+
+	spin_lock_irq(&dg00x->lock);
+
+	if (dg00x->dev_lock_count == -1) {
+		dg00x->dev_lock_count = 0;
+		err = 0;
+	} else {
+		err = -EBADFD;
+	}
+
+	spin_unlock_irq(&dg00x->lock);
+
+	return err;
+}
+
+static int hwdep_release(struct snd_hwdep *hwdep, struct file *file)
+{
+	struct snd_dg00x *dg00x = hwdep->private_data;
+
+	spin_lock_irq(&dg00x->lock);
+	if (dg00x->dev_lock_count == -1)
+		dg00x->dev_lock_count = 0;
+	spin_unlock_irq(&dg00x->lock);
+
+	return 0;
+}
+
+static int hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file,
+	    unsigned int cmd, unsigned long arg)
+{
+	struct snd_dg00x *dg00x = hwdep->private_data;
+
+	switch (cmd) {
+	case SNDRV_FIREWIRE_IOCTL_GET_INFO:
+		return hwdep_get_info(dg00x, (void __user *)arg);
+	case SNDRV_FIREWIRE_IOCTL_LOCK:
+		return hwdep_lock(dg00x);
+	case SNDRV_FIREWIRE_IOCTL_UNLOCK:
+		return hwdep_unlock(dg00x);
+	default:
+		return -ENOIOCTLCMD;
+	}
+}
+
+#ifdef CONFIG_COMPAT
+static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file,
+			      unsigned int cmd, unsigned long arg)
+{
+	return hwdep_ioctl(hwdep, file, cmd,
+			   (unsigned long)compat_ptr(arg));
+}
+#else
+#define hwdep_compat_ioctl NULL
+#endif
+
+static const struct snd_hwdep_ops hwdep_ops = {
+	.read		= hwdep_read,
+	.release	= hwdep_release,
+	.poll		= hwdep_poll,
+	.ioctl		= hwdep_ioctl,
+	.ioctl_compat	= hwdep_compat_ioctl,
+};
+
+int snd_dg00x_create_hwdep_device(struct snd_dg00x *dg00x)
+{
+	struct snd_hwdep *hwdep;
+	int err;
+
+	err = snd_hwdep_new(dg00x->card, "Digi00x", 0, &hwdep);
+	if (err < 0)
+		return err;
+
+	strcpy(hwdep->name, "Digi00x");
+	hwdep->iface = SNDRV_HWDEP_IFACE_FW_DIGI00X;
+	hwdep->ops = hwdep_ops;
+	hwdep->private_data = dg00x;
+	hwdep->exclusive = true;
+
+	return err;
+}
diff --git a/sound/firewire/digi00x/digi00x-midi.c b/sound/firewire/digi00x/digi00x-midi.c
new file mode 100644
index 000000000000..1a72a382b384
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-midi.c
@@ -0,0 +1,223 @@
+/*
+ * digi00x-midi.h - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+static int midi_phys_open(struct snd_rawmidi_substream *substream)
+{
+	struct snd_dg00x *dg00x = substream->rmidi->private_data;
+	int err;
+
+	err = snd_dg00x_stream_lock_try(dg00x);
+	if (err < 0)
+		return err;
+
+	mutex_lock(&dg00x->mutex);
+	dg00x->substreams_counter++;
+	err = snd_dg00x_stream_start_duplex(dg00x, 0);
+	mutex_unlock(&dg00x->mutex);
+	if (err < 0)
+		snd_dg00x_stream_lock_release(dg00x);
+
+	return err;
+}
+
+static int midi_phys_close(struct snd_rawmidi_substream *substream)
+{
+	struct snd_dg00x *dg00x = substream->rmidi->private_data;
+
+	mutex_lock(&dg00x->mutex);
+	dg00x->substreams_counter--;
+	snd_dg00x_stream_stop_duplex(dg00x);
+	mutex_unlock(&dg00x->mutex);
+
+	snd_dg00x_stream_lock_release(dg00x);
+	return 0;
+}
+
+static void midi_phys_capture_trigger(struct snd_rawmidi_substream *substream,
+				      int up)
+{
+	struct snd_dg00x *dg00x = substream->rmidi->private_data;
+	unsigned long flags;
+
+	spin_lock_irqsave(&dg00x->lock, flags);
+
+	if (up)
+		amdtp_dot_midi_trigger(&dg00x->tx_stream, substream->number,
+				       substream);
+	else
+		amdtp_dot_midi_trigger(&dg00x->tx_stream, substream->number,
+				       NULL);
+
+	spin_unlock_irqrestore(&dg00x->lock, flags);
+}
+
+static void midi_phys_playback_trigger(struct snd_rawmidi_substream *substream,
+				       int up)
+{
+	struct snd_dg00x *dg00x = substream->rmidi->private_data;
+	unsigned long flags;
+
+	spin_lock_irqsave(&dg00x->lock, flags);
+
+	if (up)
+		amdtp_dot_midi_trigger(&dg00x->rx_stream, substream->number,
+				       substream);
+	else
+		amdtp_dot_midi_trigger(&dg00x->rx_stream, substream->number,
+				       NULL);
+
+	spin_unlock_irqrestore(&dg00x->lock, flags);
+}
+
+static struct snd_rawmidi_ops midi_phys_capture_ops = {
+	.open		= midi_phys_open,
+	.close		= midi_phys_close,
+	.trigger	= midi_phys_capture_trigger,
+};
+
+static struct snd_rawmidi_ops midi_phys_playback_ops = {
+	.open		= midi_phys_open,
+	.close		= midi_phys_close,
+	.trigger	= midi_phys_playback_trigger,
+};
+
+static int midi_ctl_open(struct snd_rawmidi_substream *substream)
+{
+	/* Do nothing. */
+	return 0;
+}
+
+static int midi_ctl_capture_close(struct snd_rawmidi_substream *substream)
+{
+	/* Do nothing. */
+	return 0;
+}
+
+static int midi_ctl_playback_close(struct snd_rawmidi_substream *substream)
+{
+	struct snd_dg00x *dg00x = substream->rmidi->private_data;
+
+	snd_fw_async_midi_port_finish(&dg00x->out_control);
+
+	return 0;
+}
+
+static void midi_ctl_capture_trigger(struct snd_rawmidi_substream *substream,
+				     int up)
+{
+	struct snd_dg00x *dg00x = substream->rmidi->private_data;
+	unsigned long flags;
+
+	spin_lock_irqsave(&dg00x->lock, flags);
+
+	if (up)
+		dg00x->in_control = substream;
+	else
+		dg00x->in_control = NULL;
+
+	spin_unlock_irqrestore(&dg00x->lock, flags);
+}
+
+static void midi_ctl_playback_trigger(struct snd_rawmidi_substream *substream,
+				      int up)
+{
+	struct snd_dg00x *dg00x = substream->rmidi->private_data;
+	unsigned long flags;
+
+	spin_lock_irqsave(&dg00x->lock, flags);
+
+	if (up)
+		snd_fw_async_midi_port_run(&dg00x->out_control, substream);
+
+	spin_unlock_irqrestore(&dg00x->lock, flags);
+}
+
+static struct snd_rawmidi_ops midi_ctl_capture_ops = {
+	.open		= midi_ctl_open,
+	.close		= midi_ctl_capture_close,
+	.trigger	= midi_ctl_capture_trigger,
+};
+
+static struct snd_rawmidi_ops midi_ctl_playback_ops = {
+	.open		= midi_ctl_open,
+	.close		= midi_ctl_playback_close,
+	.trigger	= midi_ctl_playback_trigger,
+};
+
+static void set_midi_substream_names(struct snd_dg00x *dg00x,
+				     struct snd_rawmidi_str *str,
+				     bool is_ctl)
+{
+	struct snd_rawmidi_substream *subs;
+
+	list_for_each_entry(subs, &str->substreams, list) {
+		if (!is_ctl)
+			snprintf(subs->name, sizeof(subs->name),
+				 "%s MIDI %d",
+				 dg00x->card->shortname, subs->number + 1);
+		else
+			/* This port is for asynchronous transaction. */
+			snprintf(subs->name, sizeof(subs->name),
+				 "%s control",
+				 dg00x->card->shortname);
+	}
+}
+
+int snd_dg00x_create_midi_devices(struct snd_dg00x *dg00x)
+{
+	struct snd_rawmidi *rmidi[2];
+	struct snd_rawmidi_str *str;
+	unsigned int i;
+	int err;
+
+	/* Add physical midi ports. */
+	err = snd_rawmidi_new(dg00x->card, dg00x->card->driver, 0,
+			DOT_MIDI_OUT_PORTS, DOT_MIDI_IN_PORTS, &rmidi[0]);
+	if (err < 0)
+		return err;
+
+	snprintf(rmidi[0]->name, sizeof(rmidi[0]->name),
+		 "%s MIDI", dg00x->card->shortname);
+
+	snd_rawmidi_set_ops(rmidi[0], SNDRV_RAWMIDI_STREAM_INPUT,
+			    &midi_phys_capture_ops);
+	snd_rawmidi_set_ops(rmidi[0], SNDRV_RAWMIDI_STREAM_OUTPUT,
+			    &midi_phys_playback_ops);
+
+	/* Add a pair of control midi ports. */
+	err = snd_rawmidi_new(dg00x->card, dg00x->card->driver, 1,
+			      1, 1, &rmidi[1]);
+	if (err < 0)
+		return err;
+
+	snprintf(rmidi[1]->name, sizeof(rmidi[1]->name),
+		 "%s control", dg00x->card->shortname);
+
+	snd_rawmidi_set_ops(rmidi[1], SNDRV_RAWMIDI_STREAM_INPUT,
+			    &midi_ctl_capture_ops);
+	snd_rawmidi_set_ops(rmidi[1], SNDRV_RAWMIDI_STREAM_OUTPUT,
+			    &midi_ctl_playback_ops);
+
+	for (i = 0; i < ARRAY_SIZE(rmidi); i++) {
+		rmidi[i]->private_data = dg00x;
+
+		rmidi[i]->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
+		str = &rmidi[i]->streams[SNDRV_RAWMIDI_STREAM_INPUT];
+		set_midi_substream_names(dg00x, str, i);
+
+		rmidi[i]->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
+		str = &rmidi[i]->streams[SNDRV_RAWMIDI_STREAM_OUTPUT];
+		set_midi_substream_names(dg00x, str, i);
+
+		rmidi[i]->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX;
+	}
+
+	return 0;
+}
diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c
new file mode 100644
index 000000000000..cac28f70aef7
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-pcm.c
@@ -0,0 +1,373 @@
+/*
+ * digi00x-pcm.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+static int hw_rule_rate(struct snd_pcm_hw_params *params,
+			struct snd_pcm_hw_rule *rule)
+{
+	struct snd_interval *r =
+		hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	const struct snd_interval *c =
+		hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	struct snd_interval t = {
+		.min = UINT_MAX, .max = 0, .integer = 1,
+	};
+	unsigned int i;
+
+	for (i = 0; i < SND_DG00X_RATE_COUNT; i++) {
+		if (!snd_interval_test(c,
+				       snd_dg00x_stream_pcm_channels[i]))
+			continue;
+
+		t.min = min(t.min, snd_dg00x_stream_rates[i]);
+		t.max = max(t.max, snd_dg00x_stream_rates[i]);
+	}
+
+	return snd_interval_refine(r, &t);
+}
+
+static int hw_rule_channels(struct snd_pcm_hw_params *params,
+			    struct snd_pcm_hw_rule *rule)
+{
+	struct snd_interval *c =
+		hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	const struct snd_interval *r =
+		hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval t = {
+		.min = UINT_MAX, .max = 0, .integer = 1,
+	};
+	unsigned int i;
+
+	for (i = 0; i < SND_DG00X_RATE_COUNT; i++) {
+		if (!snd_interval_test(r, snd_dg00x_stream_rates[i]))
+			continue;
+
+		t.min = min(t.min, snd_dg00x_stream_pcm_channels[i]);
+		t.max = max(t.max, snd_dg00x_stream_pcm_channels[i]);
+	}
+
+	return snd_interval_refine(c, &t);
+}
+
+static int pcm_init_hw_params(struct snd_dg00x *dg00x,
+			      struct snd_pcm_substream *substream)
+{
+	static const struct snd_pcm_hardware hardware = {
+		.info = SNDRV_PCM_INFO_BATCH |
+			SNDRV_PCM_INFO_BLOCK_TRANSFER |
+			SNDRV_PCM_INFO_INTERLEAVED |
+			SNDRV_PCM_INFO_JOINT_DUPLEX |
+			SNDRV_PCM_INFO_MMAP |
+			SNDRV_PCM_INFO_MMAP_VALID,
+		.rates = SNDRV_PCM_RATE_44100 |
+			 SNDRV_PCM_RATE_48000 |
+			 SNDRV_PCM_RATE_88200 |
+			 SNDRV_PCM_RATE_96000,
+		.rate_min = 44100,
+		.rate_max = 96000,
+		.channels_min = 10,
+		.channels_max = 18,
+		.period_bytes_min = 4 * 18,
+		.period_bytes_max = 4 * 18 * 2048,
+		.buffer_bytes_max = 4 * 18 * 2048 * 2,
+		.periods_min = 2,
+		.periods_max = UINT_MAX,
+	};
+	struct amdtp_stream *s;
+	int err;
+
+	substream->runtime->hw = hardware;
+
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S32;
+		s = &dg00x->tx_stream;
+	} else {
+		substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16 |
+						 SNDRV_PCM_FMTBIT_S32;
+		s = &dg00x->rx_stream;
+	}
+
+	err = snd_pcm_hw_rule_add(substream->runtime, 0,
+				  SNDRV_PCM_HW_PARAM_CHANNELS,
+				  hw_rule_channels, NULL,
+				  SNDRV_PCM_HW_PARAM_RATE, -1);
+	if (err < 0)
+		return err;
+
+	err = snd_pcm_hw_rule_add(substream->runtime, 0,
+				  SNDRV_PCM_HW_PARAM_RATE,
+				  hw_rule_rate, NULL,
+				  SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+	if (err < 0)
+		return err;
+
+	return amdtp_dot_add_pcm_hw_constraints(s, substream->runtime);
+}
+
+static int pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_dg00x *dg00x = substream->private_data;
+	enum snd_dg00x_clock clock;
+	bool detect;
+	unsigned int rate;
+	int err;
+
+	err = snd_dg00x_stream_lock_try(dg00x);
+	if (err < 0)
+		goto end;
+
+	err = pcm_init_hw_params(dg00x, substream);
+	if (err < 0)
+		goto err_locked;
+
+	/* Check current clock source. */
+	err = snd_dg00x_stream_get_clock(dg00x, &clock);
+	if (err < 0)
+		goto err_locked;
+	if (clock != SND_DG00X_CLOCK_INTERNAL) {
+		err = snd_dg00x_stream_check_external_clock(dg00x, &detect);
+		if (err < 0)
+			goto err_locked;
+		if (!detect) {
+			err = -EBUSY;
+			goto err_locked;
+		}
+	}
+
+	if ((clock != SND_DG00X_CLOCK_INTERNAL) ||
+	    amdtp_stream_pcm_running(&dg00x->rx_stream) ||
+	    amdtp_stream_pcm_running(&dg00x->tx_stream)) {
+		err = snd_dg00x_stream_get_external_rate(dg00x, &rate);
+		if (err < 0)
+			goto err_locked;
+		substream->runtime->hw.rate_min = rate;
+		substream->runtime->hw.rate_max = rate;
+	}
+
+	snd_pcm_set_sync(substream);
+end:
+	return err;
+err_locked:
+	snd_dg00x_stream_lock_release(dg00x);
+	return err;
+}
+
+static int pcm_close(struct snd_pcm_substream *substream)
+{
+	struct snd_dg00x *dg00x = substream->private_data;
+
+	snd_dg00x_stream_lock_release(dg00x);
+
+	return 0;
+}
+
+static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *hw_params)
+{
+	struct snd_dg00x *dg00x = substream->private_data;
+	int err;
+
+	err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+					       params_buffer_bytes(hw_params));
+	if (err < 0)
+		return err;
+
+	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+		mutex_lock(&dg00x->mutex);
+		dg00x->substreams_counter++;
+		mutex_unlock(&dg00x->mutex);
+	}
+
+	amdtp_dot_set_pcm_format(&dg00x->tx_stream, params_format(hw_params));
+
+	return 0;
+}
+
+static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *hw_params)
+{
+	struct snd_dg00x *dg00x = substream->private_data;
+	int err;
+
+	err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+					       params_buffer_bytes(hw_params));
+	if (err < 0)
+		return err;
+
+	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+		mutex_lock(&dg00x->mutex);
+		dg00x->substreams_counter++;
+		mutex_unlock(&dg00x->mutex);
+	}
+
+	amdtp_dot_set_pcm_format(&dg00x->rx_stream, params_format(hw_params));
+
+	return 0;
+}
+
+static int pcm_capture_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_dg00x *dg00x = substream->private_data;
+
+	mutex_lock(&dg00x->mutex);
+
+	if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+		dg00x->substreams_counter--;
+
+	snd_dg00x_stream_stop_duplex(dg00x);
+
+	mutex_unlock(&dg00x->mutex);
+
+	return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_playback_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_dg00x *dg00x = substream->private_data;
+
+	mutex_lock(&dg00x->mutex);
+
+	if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+		dg00x->substreams_counter--;
+
+	snd_dg00x_stream_stop_duplex(dg00x);
+
+	mutex_unlock(&dg00x->mutex);
+
+	return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_dg00x *dg00x = substream->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	int err;
+
+	mutex_lock(&dg00x->mutex);
+
+	err = snd_dg00x_stream_start_duplex(dg00x, runtime->rate);
+	if (err >= 0)
+		amdtp_stream_pcm_prepare(&dg00x->tx_stream);
+
+	mutex_unlock(&dg00x->mutex);
+
+	return err;
+}
+
+static int pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_dg00x *dg00x = substream->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	int err;
+
+	mutex_lock(&dg00x->mutex);
+
+	err = snd_dg00x_stream_start_duplex(dg00x, runtime->rate);
+	if (err >= 0) {
+		amdtp_stream_pcm_prepare(&dg00x->rx_stream);
+		amdtp_dot_reset(&dg00x->rx_stream);
+	}
+
+	mutex_unlock(&dg00x->mutex);
+
+	return err;
+}
+
+static int pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_dg00x *dg00x = substream->private_data;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		amdtp_stream_pcm_trigger(&dg00x->tx_stream, substream);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		amdtp_stream_pcm_trigger(&dg00x->tx_stream, NULL);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_dg00x *dg00x = substream->private_data;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		amdtp_stream_pcm_trigger(&dg00x->rx_stream, substream);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		amdtp_stream_pcm_trigger(&dg00x->rx_stream, NULL);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstrm)
+{
+	struct snd_dg00x *dg00x = sbstrm->private_data;
+
+	return amdtp_stream_pcm_pointer(&dg00x->tx_stream);
+}
+
+static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
+{
+	struct snd_dg00x *dg00x = sbstrm->private_data;
+
+	return amdtp_stream_pcm_pointer(&dg00x->rx_stream);
+}
+
+static struct snd_pcm_ops pcm_capture_ops = {
+	.open		= pcm_open,
+	.close		= pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= pcm_capture_hw_params,
+	.hw_free	= pcm_capture_hw_free,
+	.prepare	= pcm_capture_prepare,
+	.trigger	= pcm_capture_trigger,
+	.pointer	= pcm_capture_pointer,
+	.page		= snd_pcm_lib_get_vmalloc_page,
+};
+
+static struct snd_pcm_ops pcm_playback_ops = {
+	.open		= pcm_open,
+	.close		= pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= pcm_playback_hw_params,
+	.hw_free	= pcm_playback_hw_free,
+	.prepare	= pcm_playback_prepare,
+	.trigger	= pcm_playback_trigger,
+	.pointer	= pcm_playback_pointer,
+	.page		= snd_pcm_lib_get_vmalloc_page,
+	.mmap		= snd_pcm_lib_mmap_vmalloc,
+};
+
+int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x)
+{
+	struct snd_pcm *pcm;
+	int err;
+
+	err = snd_pcm_new(dg00x->card, dg00x->card->driver, 0, 1, 1, &pcm);
+	if (err < 0)
+		return err;
+
+	pcm->private_data = dg00x;
+	snprintf(pcm->name, sizeof(pcm->name),
+		 "%s PCM", dg00x->card->shortname);
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops);
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops);
+
+	return 0;
+}
diff --git a/sound/firewire/digi00x/digi00x-proc.c b/sound/firewire/digi00x/digi00x-proc.c
new file mode 100644
index 000000000000..a1d601f31165
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-proc.c
@@ -0,0 +1,99 @@
+/*
+ * digi00x-proc.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+static int get_optical_iface_mode(struct snd_dg00x *dg00x,
+				  enum snd_dg00x_optical_mode *mode)
+{
+	__be32 data;
+	int err;
+
+	err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+				 DG00X_ADDR_BASE + DG00X_OFFSET_OPT_IFACE_MODE,
+				 &data, sizeof(data), 0);
+	if (err >= 0)
+		*mode = be32_to_cpu(data) & 0x01;
+
+	return err;
+}
+
+static void proc_read_clock(struct snd_info_entry *entry,
+			    struct snd_info_buffer *buf)
+{
+	static const char *const source_name[] = {
+		[SND_DG00X_CLOCK_INTERNAL] = "internal",
+		[SND_DG00X_CLOCK_SPDIF] = "s/pdif",
+		[SND_DG00X_CLOCK_ADAT] = "adat",
+		[SND_DG00X_CLOCK_WORD] = "word clock",
+	};
+	static const char *const optical_name[] = {
+		[SND_DG00X_OPT_IFACE_MODE_ADAT] = "adat",
+		[SND_DG00X_OPT_IFACE_MODE_SPDIF] = "s/pdif",
+	};
+	struct snd_dg00x *dg00x = entry->private_data;
+	enum snd_dg00x_optical_mode mode;
+	unsigned int rate;
+	enum snd_dg00x_clock clock;
+	bool detect;
+
+	if (get_optical_iface_mode(dg00x, &mode) < 0)
+		return;
+	if (snd_dg00x_stream_get_local_rate(dg00x, &rate) < 0)
+		return;
+	if (snd_dg00x_stream_get_clock(dg00x, &clock) < 0)
+		return;
+
+	snd_iprintf(buf, "Optical mode: %s\n", optical_name[mode]);
+	snd_iprintf(buf, "Sampling Rate: %d\n", rate);
+	snd_iprintf(buf, "Clock Source: %s\n", source_name[clock]);
+
+	if (clock == SND_DG00X_CLOCK_INTERNAL)
+		return;
+
+	if (snd_dg00x_stream_check_external_clock(dg00x, &detect) < 0)
+		return;
+	snd_iprintf(buf, "External source: %s\n", detect ? "detected" : "not");
+	if (!detect)
+		return;
+
+	if (snd_dg00x_stream_get_external_rate(dg00x, &rate) >= 0)
+		snd_iprintf(buf, "External sampling rate: %d\n", rate);
+}
+
+void snd_dg00x_proc_init(struct snd_dg00x *dg00x)
+{
+	struct snd_info_entry *root, *entry;
+
+	/*
+	 * All nodes are automatically removed at snd_card_disconnect(),
+	 * by following to link list.
+	 */
+	root = snd_info_create_card_entry(dg00x->card, "firewire",
+					  dg00x->card->proc_root);
+	if (root == NULL)
+		return;
+
+	root->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+	if (snd_info_register(root) < 0) {
+		snd_info_free_entry(root);
+		return;
+	}
+
+	entry = snd_info_create_card_entry(dg00x->card, "clock", root);
+	if (entry == NULL) {
+		snd_info_free_entry(root);
+		return;
+	}
+
+	snd_info_set_text_ops(entry, dg00x, proc_read_clock);
+	if (snd_info_register(entry) < 0) {
+		snd_info_free_entry(entry);
+		snd_info_free_entry(root);
+	}
+}
diff --git a/sound/firewire/digi00x/digi00x-stream.c b/sound/firewire/digi00x/digi00x-stream.c
new file mode 100644
index 000000000000..4d3b4ebbdd49
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-stream.c
@@ -0,0 +1,422 @@
+/*
+ * digi00x-stream.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+#define CALLBACK_TIMEOUT 500
+
+const unsigned int snd_dg00x_stream_rates[SND_DG00X_RATE_COUNT] = {
+	[SND_DG00X_RATE_44100] = 44100,
+	[SND_DG00X_RATE_48000] = 48000,
+	[SND_DG00X_RATE_88200] = 88200,
+	[SND_DG00X_RATE_96000] = 96000,
+};
+
+/* Multi Bit Linear Audio data channels for each sampling transfer frequency. */
+const unsigned int
+snd_dg00x_stream_pcm_channels[SND_DG00X_RATE_COUNT] = {
+	/* Analog/ADAT/SPDIF */
+	[SND_DG00X_RATE_44100] = (8 + 8 + 2),
+	[SND_DG00X_RATE_48000] = (8 + 8 + 2),
+	/* Analog/SPDIF */
+	[SND_DG00X_RATE_88200] = (8 + 2),
+	[SND_DG00X_RATE_96000] = (8 + 2),
+};
+
+int snd_dg00x_stream_get_local_rate(struct snd_dg00x *dg00x, unsigned int *rate)
+{
+	u32 data;
+	__be32 reg;
+	int err;
+
+	err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+				 DG00X_ADDR_BASE + DG00X_OFFSET_LOCAL_RATE,
+				 &reg, sizeof(reg), 0);
+	if (err < 0)
+		return err;
+
+	data = be32_to_cpu(reg) & 0x0f;
+	if (data < ARRAY_SIZE(snd_dg00x_stream_rates))
+		*rate = snd_dg00x_stream_rates[data];
+	else
+		err = -EIO;
+
+	return err;
+}
+
+int snd_dg00x_stream_set_local_rate(struct snd_dg00x *dg00x, unsigned int rate)
+{
+	__be32 reg;
+	unsigned int i;
+
+	for (i = 0; i < ARRAY_SIZE(snd_dg00x_stream_rates); i++) {
+		if (rate == snd_dg00x_stream_rates[i])
+			break;
+	}
+	if (i == ARRAY_SIZE(snd_dg00x_stream_rates))
+		return -EINVAL;
+
+	reg = cpu_to_be32(i);
+	return snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+				  DG00X_ADDR_BASE + DG00X_OFFSET_LOCAL_RATE,
+				  &reg, sizeof(reg), 0);
+}
+
+int snd_dg00x_stream_get_clock(struct snd_dg00x *dg00x,
+			       enum snd_dg00x_clock *clock)
+{
+	__be32 reg;
+	int err;
+
+	err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+				 DG00X_ADDR_BASE + DG00X_OFFSET_CLOCK_SOURCE,
+				 &reg, sizeof(reg), 0);
+	if (err < 0)
+		return err;
+
+	*clock = be32_to_cpu(reg) & 0x0f;
+	if (*clock >= SND_DG00X_CLOCK_COUNT)
+		err = -EIO;
+
+	return err;
+}
+
+int snd_dg00x_stream_check_external_clock(struct snd_dg00x *dg00x, bool *detect)
+{
+	__be32 reg;
+	int err;
+
+	err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+				 DG00X_ADDR_BASE + DG00X_OFFSET_DETECT_EXTERNAL,
+				 &reg, sizeof(reg), 0);
+	if (err >= 0)
+		*detect = be32_to_cpu(reg) > 0;
+
+	return err;
+}
+
+int snd_dg00x_stream_get_external_rate(struct snd_dg00x *dg00x,
+				       unsigned int *rate)
+{
+	u32 data;
+	__be32 reg;
+	int err;
+
+	err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+				 DG00X_ADDR_BASE + DG00X_OFFSET_EXTERNAL_RATE,
+				 &reg, sizeof(reg), 0);
+	if (err < 0)
+		return err;
+
+	data = be32_to_cpu(reg) & 0x0f;
+	if (data < ARRAY_SIZE(snd_dg00x_stream_rates))
+		*rate = snd_dg00x_stream_rates[data];
+	/* This means desync. */
+	else
+		err = -EBUSY;
+
+	return err;
+}
+
+static void finish_session(struct snd_dg00x *dg00x)
+{
+	__be32 data = cpu_to_be32(0x00000003);
+
+	snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+			   DG00X_ADDR_BASE + DG00X_OFFSET_STREAMING_SET,
+			   &data, sizeof(data), 0);
+}
+
+static int begin_session(struct snd_dg00x *dg00x)
+{
+	__be32 data;
+	u32 curr;
+	int err;
+
+	err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+				 DG00X_ADDR_BASE + DG00X_OFFSET_STREAMING_STATE,
+				 &data, sizeof(data), 0);
+	if (err < 0)
+		goto error;
+	curr = be32_to_cpu(data);
+
+	if (curr == 0)
+		curr = 2;
+
+	curr--;
+	while (curr > 0) {
+		data = cpu_to_be32(curr);
+		err = snd_fw_transaction(dg00x->unit,
+					 TCODE_WRITE_QUADLET_REQUEST,
+					 DG00X_ADDR_BASE +
+					 DG00X_OFFSET_STREAMING_SET,
+					 &data, sizeof(data), 0);
+		if (err < 0)
+			goto error;
+
+		msleep(20);
+		curr--;
+	}
+
+	return 0;
+error:
+	finish_session(dg00x);
+	return err;
+}
+
+static void release_resources(struct snd_dg00x *dg00x)
+{
+	__be32 data = 0;
+
+	/* Unregister isochronous channels for both direction. */
+	snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+			   DG00X_ADDR_BASE + DG00X_OFFSET_ISOC_CHANNELS,
+			   &data, sizeof(data), 0);
+
+	/* Release isochronous resources. */
+	fw_iso_resources_free(&dg00x->tx_resources);
+	fw_iso_resources_free(&dg00x->rx_resources);
+}
+
+static int keep_resources(struct snd_dg00x *dg00x, unsigned int rate)
+{
+	unsigned int i;
+	__be32 data;
+	int err;
+
+	/* Check sampling rate. */
+	for (i = 0; i < SND_DG00X_RATE_COUNT; i++) {
+		if (snd_dg00x_stream_rates[i] == rate)
+			break;
+	}
+	if (i == SND_DG00X_RATE_COUNT)
+		return -EINVAL;
+
+	/* Keep resources for out-stream. */
+	err = amdtp_dot_set_parameters(&dg00x->rx_stream, rate,
+				       snd_dg00x_stream_pcm_channels[i]);
+	if (err < 0)
+		return err;
+	err = fw_iso_resources_allocate(&dg00x->rx_resources,
+				amdtp_stream_get_max_payload(&dg00x->rx_stream),
+				fw_parent_device(dg00x->unit)->max_speed);
+	if (err < 0)
+		return err;
+
+	/* Keep resources for in-stream. */
+	err = amdtp_dot_set_parameters(&dg00x->tx_stream, rate,
+				       snd_dg00x_stream_pcm_channels[i]);
+	if (err < 0)
+		return err;
+	err = fw_iso_resources_allocate(&dg00x->tx_resources,
+				amdtp_stream_get_max_payload(&dg00x->tx_stream),
+				fw_parent_device(dg00x->unit)->max_speed);
+	if (err < 0)
+		goto error;
+
+	/* Register isochronous channels for both direction. */
+	data = cpu_to_be32((dg00x->tx_resources.channel << 16) |
+			   dg00x->rx_resources.channel);
+	err = snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+				 DG00X_ADDR_BASE + DG00X_OFFSET_ISOC_CHANNELS,
+				 &data, sizeof(data), 0);
+	if (err < 0)
+		goto error;
+
+	return 0;
+error:
+	release_resources(dg00x);
+	return err;
+}
+
+int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x)
+{
+	int err;
+
+	/* For out-stream. */
+	err = fw_iso_resources_init(&dg00x->rx_resources, dg00x->unit);
+	if (err < 0)
+		goto error;
+	err = amdtp_dot_init(&dg00x->rx_stream, dg00x->unit, AMDTP_OUT_STREAM);
+	if (err < 0)
+		goto error;
+
+	/* For in-stream. */
+	err = fw_iso_resources_init(&dg00x->tx_resources, dg00x->unit);
+	if (err < 0)
+		goto error;
+	err = amdtp_dot_init(&dg00x->tx_stream, dg00x->unit, AMDTP_IN_STREAM);
+	if (err < 0)
+		goto error;
+
+	return 0;
+error:
+	snd_dg00x_stream_destroy_duplex(dg00x);
+	return err;
+}
+
+/*
+ * This function should be called before starting streams or after stopping
+ * streams.
+ */
+void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x)
+{
+	amdtp_stream_destroy(&dg00x->rx_stream);
+	fw_iso_resources_destroy(&dg00x->rx_resources);
+
+	amdtp_stream_destroy(&dg00x->tx_stream);
+	fw_iso_resources_destroy(&dg00x->tx_resources);
+}
+
+int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x, unsigned int rate)
+{
+	unsigned int curr_rate;
+	int err = 0;
+
+	if (dg00x->substreams_counter == 0)
+		goto end;
+
+	/* Check current sampling rate. */
+	err = snd_dg00x_stream_get_local_rate(dg00x, &curr_rate);
+	if (err < 0)
+		goto error;
+	if (rate == 0)
+		rate = curr_rate;
+	if (curr_rate != rate ||
+	    amdtp_streaming_error(&dg00x->tx_stream) ||
+	    amdtp_streaming_error(&dg00x->rx_stream)) {
+		finish_session(dg00x);
+
+		amdtp_stream_stop(&dg00x->tx_stream);
+		amdtp_stream_stop(&dg00x->rx_stream);
+		release_resources(dg00x);
+	}
+
+	/*
+	 * No packets are transmitted without receiving packets, reagardless of
+	 * which source of clock is used.
+	 */
+	if (!amdtp_stream_running(&dg00x->rx_stream)) {
+		err = snd_dg00x_stream_set_local_rate(dg00x, rate);
+		if (err < 0)
+			goto error;
+
+		err = keep_resources(dg00x, rate);
+		if (err < 0)
+			goto error;
+
+		err = begin_session(dg00x);
+		if (err < 0)
+			goto error;
+
+		err = amdtp_stream_start(&dg00x->rx_stream,
+				dg00x->rx_resources.channel,
+				fw_parent_device(dg00x->unit)->max_speed);
+		if (err < 0)
+			goto error;
+
+		if (!amdtp_stream_wait_callback(&dg00x->rx_stream,
+					      CALLBACK_TIMEOUT)) {
+			err = -ETIMEDOUT;
+			goto error;
+		}
+	}
+
+	/*
+	 * The value of SYT field in transmitted packets is always 0x0000. Thus,
+	 * duplex streams with timestamp synchronization cannot be built.
+	 */
+	if (!amdtp_stream_running(&dg00x->tx_stream)) {
+		err = amdtp_stream_start(&dg00x->tx_stream,
+				dg00x->tx_resources.channel,
+				fw_parent_device(dg00x->unit)->max_speed);
+		if (err < 0)
+			goto error;
+
+		if (!amdtp_stream_wait_callback(&dg00x->tx_stream,
+					      CALLBACK_TIMEOUT)) {
+			err = -ETIMEDOUT;
+			goto error;
+		}
+	}
+end:
+	return err;
+error:
+	finish_session(dg00x);
+
+	amdtp_stream_stop(&dg00x->tx_stream);
+	amdtp_stream_stop(&dg00x->rx_stream);
+	release_resources(dg00x);
+
+	return err;
+}
+
+void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x)
+{
+	if (dg00x->substreams_counter > 0)
+		return;
+
+	amdtp_stream_stop(&dg00x->tx_stream);
+	amdtp_stream_stop(&dg00x->rx_stream);
+	finish_session(dg00x);
+	release_resources(dg00x);
+
+	/*
+	 * Just after finishing the session, the device may lost transmitting
+	 * functionality for a short time.
+	 */
+	msleep(50);
+}
+
+void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x)
+{
+	fw_iso_resources_update(&dg00x->tx_resources);
+	fw_iso_resources_update(&dg00x->rx_resources);
+
+	amdtp_stream_update(&dg00x->tx_stream);
+	amdtp_stream_update(&dg00x->rx_stream);
+}
+
+void snd_dg00x_stream_lock_changed(struct snd_dg00x *dg00x)
+{
+	dg00x->dev_lock_changed = true;
+	wake_up(&dg00x->hwdep_wait);
+}
+
+int snd_dg00x_stream_lock_try(struct snd_dg00x *dg00x)
+{
+	int err;
+
+	spin_lock_irq(&dg00x->lock);
+
+	/* user land lock this */
+	if (dg00x->dev_lock_count < 0) {
+		err = -EBUSY;
+		goto end;
+	}
+
+	/* this is the first time */
+	if (dg00x->dev_lock_count++ == 0)
+		snd_dg00x_stream_lock_changed(dg00x);
+	err = 0;
+end:
+	spin_unlock_irq(&dg00x->lock);
+	return err;
+}
+
+void snd_dg00x_stream_lock_release(struct snd_dg00x *dg00x)
+{
+	spin_lock_irq(&dg00x->lock);
+
+	if (WARN_ON(dg00x->dev_lock_count <= 0))
+		goto end;
+	if (--dg00x->dev_lock_count == 0)
+		snd_dg00x_stream_lock_changed(dg00x);
+end:
+	spin_unlock_irq(&dg00x->lock);
+}
diff --git a/sound/firewire/digi00x/digi00x-transaction.c b/sound/firewire/digi00x/digi00x-transaction.c
new file mode 100644
index 000000000000..554324d8c602
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-transaction.c
@@ -0,0 +1,137 @@
+/*
+ * digi00x-transaction.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <sound/asound.h>
+#include "digi00x.h"
+
+static int fill_midi_message(struct snd_rawmidi_substream *substream, u8 *buf)
+{
+	int bytes;
+
+	buf[0] = 0x80;
+	bytes = snd_rawmidi_transmit_peek(substream, buf + 1, 2);
+	if (bytes >= 0)
+		buf[3] = 0xc0 | bytes;
+
+	return bytes;
+}
+
+static void handle_midi_control(struct snd_dg00x *dg00x, __be32 *buf,
+				unsigned int length)
+{
+	struct snd_rawmidi_substream *substream;
+	unsigned int i;
+	unsigned int len;
+	u8 *b;
+
+	substream = ACCESS_ONCE(dg00x->in_control);
+	if (substream == NULL)
+		return;
+
+	length /= 4;
+
+	for (i = 0; i < length; i++) {
+		b = (u8 *)&buf[i];
+		len = b[3] & 0xf;
+		if (len > 0)
+			snd_rawmidi_receive(dg00x->in_control, b + 1, len);
+	}
+}
+
+static void handle_unknown_message(struct snd_dg00x *dg00x,
+				   unsigned long long offset, __be32 *buf)
+{
+	unsigned long flags;
+
+	spin_lock_irqsave(&dg00x->lock, flags);
+	dg00x->msg = be32_to_cpu(*buf);
+	spin_unlock_irqrestore(&dg00x->lock, flags);
+
+	wake_up(&dg00x->hwdep_wait);
+}
+
+static void handle_message(struct fw_card *card, struct fw_request *request,
+			   int tcode, int destination, int source,
+			   int generation, unsigned long long offset,
+			   void *data, size_t length, void *callback_data)
+{
+	struct snd_dg00x *dg00x = callback_data;
+	__be32 *buf = (__be32 *)data;
+
+	if (offset == dg00x->async_handler.offset)
+		handle_unknown_message(dg00x, offset, buf);
+	else if (offset == dg00x->async_handler.offset + 4)
+		handle_midi_control(dg00x, buf, length);
+
+	fw_send_response(card, request, RCODE_COMPLETE);
+}
+
+int snd_dg00x_transaction_reregister(struct snd_dg00x *dg00x)
+{
+	struct fw_device *device = fw_parent_device(dg00x->unit);
+	__be32 data[2];
+	int err;
+
+	/* Unknown. 4bytes. */
+	data[0] = cpu_to_be32((device->card->node_id << 16) |
+			      (dg00x->async_handler.offset >> 32));
+	data[1] = cpu_to_be32(dg00x->async_handler.offset);
+	err = snd_fw_transaction(dg00x->unit, TCODE_WRITE_BLOCK_REQUEST,
+				 DG00X_ADDR_BASE + DG00X_OFFSET_MESSAGE_ADDR,
+				 &data, sizeof(data), 0);
+	if (err < 0)
+		return err;
+
+	/* Asynchronous transactions for MIDI control message. */
+	data[0] = cpu_to_be32((device->card->node_id << 16) |
+			      (dg00x->async_handler.offset >> 32));
+	data[1] = cpu_to_be32(dg00x->async_handler.offset + 4);
+	return snd_fw_transaction(dg00x->unit, TCODE_WRITE_BLOCK_REQUEST,
+				  DG00X_ADDR_BASE + DG00X_OFFSET_MIDI_CTL_ADDR,
+				  &data, sizeof(data), 0);
+}
+
+int snd_dg00x_transaction_register(struct snd_dg00x *dg00x)
+{
+	static const struct fw_address_region resp_register_region = {
+		.start	= 0xffffe0000000ull,
+		.end	= 0xffffe000ffffull,
+	};
+	int err;
+
+	dg00x->async_handler.length = 12;
+	dg00x->async_handler.address_callback = handle_message;
+	dg00x->async_handler.callback_data = dg00x;
+
+	err = fw_core_add_address_handler(&dg00x->async_handler,
+					  &resp_register_region);
+	if (err < 0)
+		return err;
+
+	err = snd_dg00x_transaction_reregister(dg00x);
+	if (err < 0)
+		goto error;
+
+	err = snd_fw_async_midi_port_init(&dg00x->out_control, dg00x->unit,
+					  DG00X_ADDR_BASE + DG00X_OFFSET_MMC,
+					  4, fill_midi_message);
+	if (err < 0)
+		goto error;
+
+	return err;
+error:
+	fw_core_remove_address_handler(&dg00x->async_handler);
+	dg00x->async_handler.address_callback = NULL;
+	return err;
+}
+
+void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x)
+{
+	snd_fw_async_midi_port_destroy(&dg00x->out_control);
+	fw_core_remove_address_handler(&dg00x->async_handler);
+}
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
new file mode 100644
index 000000000000..1f33b7a1fca4
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x.c
@@ -0,0 +1,170 @@
+/*
+ * digi00x.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+MODULE_DESCRIPTION("Digidesign Digi 002/003 family Driver");
+MODULE_AUTHOR("Takashi Sakamoto <o-takashi@sakamocchi.jp>");
+MODULE_LICENSE("GPL v2");
+
+#define VENDOR_DIGIDESIGN	0x00a07e
+#define MODEL_DIGI00X		0x000002
+
+static int name_card(struct snd_dg00x *dg00x)
+{
+	struct fw_device *fw_dev = fw_parent_device(dg00x->unit);
+	char name[32] = {0};
+	char *model;
+	int err;
+
+	err = fw_csr_string(dg00x->unit->directory, CSR_MODEL, name,
+			    sizeof(name));
+	if (err < 0)
+		return err;
+
+	model = skip_spaces(name);
+
+	strcpy(dg00x->card->driver, "Digi00x");
+	strcpy(dg00x->card->shortname, model);
+	strcpy(dg00x->card->mixername, model);
+	snprintf(dg00x->card->longname, sizeof(dg00x->card->longname),
+		 "Digidesign %s, GUID %08x%08x at %s, S%d", model,
+		 fw_dev->config_rom[3], fw_dev->config_rom[4],
+		 dev_name(&dg00x->unit->device), 100 << fw_dev->max_speed);
+
+	return 0;
+}
+
+static void dg00x_card_free(struct snd_card *card)
+{
+	struct snd_dg00x *dg00x = card->private_data;
+
+	snd_dg00x_stream_destroy_duplex(dg00x);
+	snd_dg00x_transaction_unregister(dg00x);
+
+	fw_unit_put(dg00x->unit);
+
+	mutex_destroy(&dg00x->mutex);
+}
+
+static int snd_dg00x_probe(struct fw_unit *unit,
+			   const struct ieee1394_device_id *entry)
+{
+	struct snd_card *card;
+	struct snd_dg00x *dg00x;
+	int err;
+
+	/* create card */
+	err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE,
+			   sizeof(struct snd_dg00x), &card);
+	if (err < 0)
+		return err;
+	card->private_free = dg00x_card_free;
+
+	/* initialize myself */
+	dg00x = card->private_data;
+	dg00x->card = card;
+	dg00x->unit = fw_unit_get(unit);
+
+	mutex_init(&dg00x->mutex);
+	spin_lock_init(&dg00x->lock);
+	init_waitqueue_head(&dg00x->hwdep_wait);
+
+	err = name_card(dg00x);
+	if (err < 0)
+		goto error;
+
+	err = snd_dg00x_stream_init_duplex(dg00x);
+	if (err < 0)
+		goto error;
+
+	snd_dg00x_proc_init(dg00x);
+
+	err = snd_dg00x_create_pcm_devices(dg00x);
+	if (err < 0)
+		goto error;
+
+	err = snd_dg00x_create_midi_devices(dg00x);
+	if (err < 0)
+		goto error;
+
+	err = snd_dg00x_create_hwdep_device(dg00x);
+	if (err < 0)
+		goto error;
+
+	err = snd_dg00x_transaction_register(dg00x);
+	if (err < 0)
+		goto error;
+
+	err = snd_card_register(card);
+	if (err < 0)
+		goto error;
+
+	dev_set_drvdata(&unit->device, dg00x);
+
+	return err;
+error:
+	snd_card_free(card);
+	return err;
+}
+
+static void snd_dg00x_update(struct fw_unit *unit)
+{
+	struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device);
+
+	snd_dg00x_transaction_reregister(dg00x);
+
+	mutex_lock(&dg00x->mutex);
+	snd_dg00x_stream_update_duplex(dg00x);
+	mutex_unlock(&dg00x->mutex);
+}
+
+static void snd_dg00x_remove(struct fw_unit *unit)
+{
+	struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device);
+
+	/* No need to wait for releasing card object in this context. */
+	snd_card_free_when_closed(dg00x->card);
+}
+
+static const struct ieee1394_device_id snd_dg00x_id_table[] = {
+	/* Both of 002/003 use the same ID. */
+	{
+		.match_flags = IEEE1394_MATCH_VENDOR_ID |
+			       IEEE1394_MATCH_MODEL_ID,
+		.vendor_id = VENDOR_DIGIDESIGN,
+		.model_id = MODEL_DIGI00X,
+	},
+	{}
+};
+MODULE_DEVICE_TABLE(ieee1394, snd_dg00x_id_table);
+
+static struct fw_driver dg00x_driver = {
+	.driver = {
+		.owner = THIS_MODULE,
+		.name = "snd-firewire-digi00x",
+		.bus = &fw_bus_type,
+	},
+	.probe    = snd_dg00x_probe,
+	.update   = snd_dg00x_update,
+	.remove   = snd_dg00x_remove,
+	.id_table = snd_dg00x_id_table,
+};
+
+static int __init snd_dg00x_init(void)
+{
+	return driver_register(&dg00x_driver.driver);
+}
+
+static void __exit snd_dg00x_exit(void)
+{
+	driver_unregister(&dg00x_driver.driver);
+}
+
+module_init(snd_dg00x_init);
+module_exit(snd_dg00x_exit);
diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h
new file mode 100644
index 000000000000..907e73993677
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x.h
@@ -0,0 +1,157 @@
+/*
+ * digi00x.h - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#ifndef SOUND_DIGI00X_H_INCLUDED
+#define SOUND_DIGI00X_H_INCLUDED
+
+#include <linux/compat.h>
+#include <linux/device.h>
+#include <linux/firewire.h>
+#include <linux/module.h>
+#include <linux/mod_devicetable.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/firewire.h>
+#include <sound/hwdep.h>
+#include <sound/rawmidi.h>
+
+#include "../lib.h"
+#include "../iso-resources.h"
+#include "../amdtp-stream.h"
+
+struct snd_dg00x {
+	struct snd_card *card;
+	struct fw_unit *unit;
+
+	struct mutex mutex;
+	spinlock_t lock;
+
+	struct amdtp_stream tx_stream;
+	struct fw_iso_resources tx_resources;
+
+	struct amdtp_stream rx_stream;
+	struct fw_iso_resources rx_resources;
+
+	unsigned int substreams_counter;
+
+	/* for uapi */
+	int dev_lock_count;
+	bool dev_lock_changed;
+	wait_queue_head_t hwdep_wait;
+
+	/* For asynchronous messages. */
+	struct fw_address_handler async_handler;
+	u32 msg;
+
+	/* For asynchronous MIDI controls. */
+	struct snd_rawmidi_substream *in_control;
+	struct snd_fw_async_midi_port out_control;
+};
+
+#define DG00X_ADDR_BASE		0xffffe0000000ull
+
+#define DG00X_OFFSET_STREAMING_STATE	0x0000
+#define DG00X_OFFSET_STREAMING_SET	0x0004
+#define DG00X_OFFSET_MIDI_CTL_ADDR	0x0008
+/* For LSB of the address		0x000c */
+/* unknown				0x0010 */
+#define DG00X_OFFSET_MESSAGE_ADDR	0x0014
+/* For LSB of the address		0x0018 */
+/* unknown				0x001c */
+/* unknown				0x0020 */
+/* not used			0x0024--0x00ff */
+#define DG00X_OFFSET_ISOC_CHANNELS	0x0100
+/* unknown				0x0104 */
+/* unknown				0x0108 */
+/* unknown				0x010c */
+#define DG00X_OFFSET_LOCAL_RATE		0x0110
+#define DG00X_OFFSET_EXTERNAL_RATE	0x0114
+#define DG00X_OFFSET_CLOCK_SOURCE	0x0118
+#define DG00X_OFFSET_OPT_IFACE_MODE	0x011c
+/* unknown				0x0120 */
+/* Mixer control on/off			0x0124 */
+/* unknown				0x0128 */
+#define DG00X_OFFSET_DETECT_EXTERNAL	0x012c
+/* unknown				0x0138 */
+#define DG00X_OFFSET_MMC		0x0400
+
+enum snd_dg00x_rate {
+	SND_DG00X_RATE_44100 = 0,
+	SND_DG00X_RATE_48000,
+	SND_DG00X_RATE_88200,
+	SND_DG00X_RATE_96000,
+	SND_DG00X_RATE_COUNT,
+};
+
+enum snd_dg00x_clock {
+	SND_DG00X_CLOCK_INTERNAL = 0,
+	SND_DG00X_CLOCK_SPDIF,
+	SND_DG00X_CLOCK_ADAT,
+	SND_DG00X_CLOCK_WORD,
+	SND_DG00X_CLOCK_COUNT,
+};
+
+enum snd_dg00x_optical_mode {
+	SND_DG00X_OPT_IFACE_MODE_ADAT = 0,
+	SND_DG00X_OPT_IFACE_MODE_SPDIF,
+	SND_DG00X_OPT_IFACE_MODE_COUNT,
+};
+
+#define DOT_MIDI_IN_PORTS	1
+#define DOT_MIDI_OUT_PORTS	2
+
+int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit,
+		   enum amdtp_stream_direction dir);
+int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate,
+			     unsigned int pcm_channels);
+void amdtp_dot_reset(struct amdtp_stream *s);
+int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s,
+				     struct snd_pcm_runtime *runtime);
+void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format);
+void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port,
+			  struct snd_rawmidi_substream *midi);
+
+int snd_dg00x_transaction_register(struct snd_dg00x *dg00x);
+int snd_dg00x_transaction_reregister(struct snd_dg00x *dg00x);
+void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x);
+
+extern const unsigned int snd_dg00x_stream_rates[SND_DG00X_RATE_COUNT];
+extern const unsigned int snd_dg00x_stream_pcm_channels[SND_DG00X_RATE_COUNT];
+int snd_dg00x_stream_get_external_rate(struct snd_dg00x *dg00x,
+				       unsigned int *rate);
+int snd_dg00x_stream_get_local_rate(struct snd_dg00x *dg00x,
+				    unsigned int *rate);
+int snd_dg00x_stream_set_local_rate(struct snd_dg00x *dg00x, unsigned int rate);
+int snd_dg00x_stream_get_clock(struct snd_dg00x *dg00x,
+			       enum snd_dg00x_clock *clock);
+int snd_dg00x_stream_check_external_clock(struct snd_dg00x *dg00x,
+					  bool *detect);
+int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x);
+int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x, unsigned int rate);
+void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x);
+void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x);
+void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x);
+
+void snd_dg00x_stream_lock_changed(struct snd_dg00x *dg00x);
+int snd_dg00x_stream_lock_try(struct snd_dg00x *dg00x);
+void snd_dg00x_stream_lock_release(struct snd_dg00x *dg00x);
+
+void snd_dg00x_proc_init(struct snd_dg00x *dg00x);
+
+int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x);
+
+int snd_dg00x_create_midi_devices(struct snd_dg00x *dg00x);
+
+int snd_dg00x_create_hwdep_device(struct snd_dg00x *dg00x);
+#endif
diff --git a/sound/firewire/fcp.c b/sound/firewire/fcp.c
index 0619597e3a3f..cce19768f43d 100644
--- a/sound/firewire/fcp.c
+++ b/sound/firewire/fcp.c
@@ -17,7 +17,7 @@
 #include <linux/delay.h>
 #include "fcp.h"
 #include "lib.h"
-#include "amdtp.h"
+#include "amdtp-stream.h"
 
 #define CTS_AVC 0x00
 
diff --git a/sound/firewire/fireworks/Makefile b/sound/firewire/fireworks/Makefile
index 0c7440826db8..15ef7f75a8ef 100644
--- a/sound/firewire/fireworks/Makefile
+++ b/sound/firewire/fireworks/Makefile
@@ -1,4 +1,4 @@
 snd-fireworks-objs := fireworks_transaction.o fireworks_command.o \
 		      fireworks_stream.o fireworks_proc.o fireworks_midi.o \
 		      fireworks_pcm.o fireworks_hwdep.o fireworks.o
-obj-m += snd-fireworks.o
+obj-$(CONFIG_SND_FIREWORKS) += snd-fireworks.o
diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c
index c94a432f7cc6..d5b19bc11e59 100644
--- a/sound/firewire/fireworks/fireworks.c
+++ b/sound/firewire/fireworks/fireworks.c
@@ -138,12 +138,12 @@ get_hardware_info(struct snd_efw *efw)
 	efw->midi_out_ports = hwinfo->midi_out_ports;
 	efw->midi_in_ports = hwinfo->midi_in_ports;
 
-	if (hwinfo->amdtp_tx_pcm_channels    > AMDTP_MAX_CHANNELS_FOR_PCM ||
-	    hwinfo->amdtp_tx_pcm_channels_2x > AMDTP_MAX_CHANNELS_FOR_PCM ||
-	    hwinfo->amdtp_tx_pcm_channels_4x > AMDTP_MAX_CHANNELS_FOR_PCM ||
-	    hwinfo->amdtp_rx_pcm_channels    > AMDTP_MAX_CHANNELS_FOR_PCM ||
-	    hwinfo->amdtp_rx_pcm_channels_2x > AMDTP_MAX_CHANNELS_FOR_PCM ||
-	    hwinfo->amdtp_rx_pcm_channels_4x > AMDTP_MAX_CHANNELS_FOR_PCM) {
+	if (hwinfo->amdtp_tx_pcm_channels    > AM824_MAX_CHANNELS_FOR_PCM ||
+	    hwinfo->amdtp_tx_pcm_channels_2x > AM824_MAX_CHANNELS_FOR_PCM ||
+	    hwinfo->amdtp_tx_pcm_channels_4x > AM824_MAX_CHANNELS_FOR_PCM ||
+	    hwinfo->amdtp_rx_pcm_channels    > AM824_MAX_CHANNELS_FOR_PCM ||
+	    hwinfo->amdtp_rx_pcm_channels_2x > AM824_MAX_CHANNELS_FOR_PCM ||
+	    hwinfo->amdtp_rx_pcm_channels_4x > AM824_MAX_CHANNELS_FOR_PCM) {
 		err = -ENOSYS;
 		goto end;
 	}
diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h
index 084d414b228c..c7cb7deafe48 100644
--- a/sound/firewire/fireworks/fireworks.h
+++ b/sound/firewire/fireworks/fireworks.h
@@ -29,7 +29,7 @@
 
 #include "../packets-buffer.h"
 #include "../iso-resources.h"
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
 #include "../cmp.h"
 #include "../lib.h"
 
diff --git a/sound/firewire/fireworks/fireworks_command.c b/sound/firewire/fireworks/fireworks_command.c
index 166f80584c2a..94bab0476a65 100644
--- a/sound/firewire/fireworks/fireworks_command.c
+++ b/sound/firewire/fireworks/fireworks_command.c
@@ -257,7 +257,7 @@ int snd_efw_command_get_phys_meters(struct snd_efw *efw,
 				    struct snd_efw_phys_meters *meters,
 				    unsigned int len)
 {
-	__be32 *buf = (__be32 *)meters;
+	u32 *buf = (u32 *)meters;
 	unsigned int i;
 	int err;
 
diff --git a/sound/firewire/fireworks/fireworks_midi.c b/sound/firewire/fireworks/fireworks_midi.c
index cf9c65260439..fba01bbba456 100644
--- a/sound/firewire/fireworks/fireworks_midi.c
+++ b/sound/firewire/fireworks/fireworks_midi.c
@@ -73,10 +73,10 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
 	spin_lock_irqsave(&efw->lock, flags);
 
 	if (up)
-		amdtp_stream_midi_trigger(&efw->tx_stream,
+		amdtp_am824_midi_trigger(&efw->tx_stream,
 					  substrm->number, substrm);
 	else
-		amdtp_stream_midi_trigger(&efw->tx_stream,
+		amdtp_am824_midi_trigger(&efw->tx_stream,
 					  substrm->number, NULL);
 
 	spin_unlock_irqrestore(&efw->lock, flags);
@@ -90,11 +90,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
 	spin_lock_irqsave(&efw->lock, flags);
 
 	if (up)
-		amdtp_stream_midi_trigger(&efw->rx_stream,
-					  substrm->number, substrm);
+		amdtp_am824_midi_trigger(&efw->rx_stream,
+					 substrm->number, substrm);
 	else
-		amdtp_stream_midi_trigger(&efw->rx_stream,
-					  substrm->number, NULL);
+		amdtp_am824_midi_trigger(&efw->rx_stream,
+					 substrm->number, NULL);
 
 	spin_unlock_irqrestore(&efw->lock, flags);
 }
diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c
index c30b2ffa8dfb..d27135bac513 100644
--- a/sound/firewire/fireworks/fireworks_pcm.c
+++ b/sound/firewire/fireworks/fireworks_pcm.c
@@ -159,11 +159,11 @@ pcm_init_hw_params(struct snd_efw *efw,
 			   SNDRV_PCM_INFO_MMAP_VALID;
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
-		runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS;
+		runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS;
 		s = &efw->tx_stream;
 		pcm_channels = efw->pcm_capture_channels;
 	} else {
-		runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS;
+		runtime->hw.formats = AM824_OUT_PCM_FORMAT_BITS;
 		s = &efw->rx_stream;
 		pcm_channels = efw->pcm_playback_channels;
 	}
@@ -187,7 +187,7 @@ pcm_init_hw_params(struct snd_efw *efw,
 	if (err < 0)
 		goto end;
 
-	err = amdtp_stream_add_pcm_hw_constraints(s, runtime);
+	err = amdtp_am824_add_pcm_hw_constraints(s, runtime);
 end:
 	return err;
 }
@@ -253,7 +253,8 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
 
 	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
 		atomic_inc(&efw->capture_substreams);
-	amdtp_stream_set_pcm_format(&efw->tx_stream, params_format(hw_params));
+
+	amdtp_am824_set_pcm_format(&efw->tx_stream, params_format(hw_params));
 
 	return 0;
 }
@@ -270,7 +271,8 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
 
 	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
 		atomic_inc(&efw->playback_substreams);
-	amdtp_stream_set_pcm_format(&efw->rx_stream, params_format(hw_params));
+
+	amdtp_am824_set_pcm_format(&efw->rx_stream, params_format(hw_params));
 
 	return 0;
 }
diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c
index 7e353f1f7bff..759f6e3ed44a 100644
--- a/sound/firewire/fireworks/fireworks_stream.c
+++ b/sound/firewire/fireworks/fireworks_stream.c
@@ -31,7 +31,7 @@ init_stream(struct snd_efw *efw, struct amdtp_stream *stream)
 	if (err < 0)
 		goto end;
 
-	err = amdtp_stream_init(stream, efw->unit, s_dir, CIP_BLOCKING);
+	err = amdtp_am824_init(stream, efw->unit, s_dir, CIP_BLOCKING);
 	if (err < 0) {
 		amdtp_stream_destroy(stream);
 		cmp_connection_destroy(conn);
@@ -73,8 +73,10 @@ start_stream(struct snd_efw *efw, struct amdtp_stream *stream,
 		midi_ports = efw->midi_in_ports;
 	}
 
-	amdtp_stream_set_parameters(stream, sampling_rate,
-				    pcm_channels, midi_ports);
+	err = amdtp_am824_set_parameters(stream, sampling_rate,
+					 pcm_channels, midi_ports, false);
+	if (err < 0)
+		goto end;
 
 	/*  establish connection via CMP */
 	err = cmp_connection_establish(conn,
diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c
index 7409edba9f06..f80aafa44c89 100644
--- a/sound/firewire/lib.c
+++ b/sound/firewire/lib.c
@@ -9,6 +9,7 @@
 #include <linux/device.h>
 #include <linux/firewire.h>
 #include <linux/module.h>
+#include <linux/slab.h>
 #include "lib.h"
 
 #define ERROR_RETRY_DELAY_MS	20
@@ -66,6 +67,147 @@ int snd_fw_transaction(struct fw_unit *unit, int tcode,
 }
 EXPORT_SYMBOL(snd_fw_transaction);
 
+static void async_midi_port_callback(struct fw_card *card, int rcode,
+				     void *data, size_t length,
+				     void *callback_data)
+{
+	struct snd_fw_async_midi_port *port = callback_data;
+	struct snd_rawmidi_substream *substream = ACCESS_ONCE(port->substream);
+
+	/* This port is closed. */
+	if (substream == NULL)
+		return;
+
+	if (rcode == RCODE_COMPLETE)
+		snd_rawmidi_transmit_ack(substream, port->consume_bytes);
+	else if (!rcode_is_permanent_error(rcode))
+		/* To start next transaction immediately for recovery. */
+		port->next_ktime = ktime_set(0, 0);
+	else
+		/* Don't continue processing. */
+		port->error = true;
+
+	port->idling = true;
+
+	if (!snd_rawmidi_transmit_empty(substream))
+		schedule_work(&port->work);
+}
+
+static void midi_port_work(struct work_struct *work)
+{
+	struct snd_fw_async_midi_port *port =
+			container_of(work, struct snd_fw_async_midi_port, work);
+	struct snd_rawmidi_substream *substream = ACCESS_ONCE(port->substream);
+	int generation;
+	int type;
+
+	/* Under transacting or error state. */
+	if (!port->idling || port->error)
+		return;
+
+	/* Nothing to do. */
+	if (substream == NULL || snd_rawmidi_transmit_empty(substream))
+		return;
+
+	/* Do it in next chance. */
+	if (ktime_after(port->next_ktime, ktime_get())) {
+		schedule_work(&port->work);
+		return;
+	}
+
+	/*
+	 * Fill the buffer. The callee must use snd_rawmidi_transmit_peek().
+	 * Later, snd_rawmidi_transmit_ack() is called.
+	 */
+	memset(port->buf, 0, port->len);
+	port->consume_bytes = port->fill(substream, port->buf);
+	if (port->consume_bytes <= 0) {
+		/* Do it in next chance, immediately. */
+		if (port->consume_bytes == 0) {
+			port->next_ktime = ktime_set(0, 0);
+			schedule_work(&port->work);
+		} else {
+			/* Fatal error. */
+			port->error = true;
+		}
+		return;
+	}
+
+	/* Calculate type of transaction. */
+	if (port->len == 4)
+		type = TCODE_WRITE_QUADLET_REQUEST;
+	else
+		type = TCODE_WRITE_BLOCK_REQUEST;
+
+	/* Set interval to next transaction. */
+	port->next_ktime = ktime_add_ns(ktime_get(),
+				port->consume_bytes * 8 * NSEC_PER_SEC / 31250);
+
+	/* Start this transaction. */
+	port->idling = false;
+
+	/*
+	 * In Linux FireWire core, when generation is updated with memory
+	 * barrier, node id has already been updated. In this module, After
+	 * this smp_rmb(), load/store instructions to memory are completed.
+	 * Thus, both of generation and node id are available with recent
+	 * values. This is a light-serialization solution to handle bus reset
+	 * events on IEEE 1394 bus.
+	 */
+	generation = port->parent->generation;
+	smp_rmb();
+
+	fw_send_request(port->parent->card, &port->transaction, type,
+			port->parent->node_id, generation,
+			port->parent->max_speed, port->addr,
+			port->buf, port->len, async_midi_port_callback,
+			port);
+}
+
+/**
+ * snd_fw_async_midi_port_init - initialize asynchronous MIDI port structure
+ * @port: the asynchronous MIDI port to initialize
+ * @unit: the target of the asynchronous transaction
+ * @addr: the address to which transactions are transferred
+ * @len: the length of transaction
+ * @fill: the callback function to fill given buffer, and returns the
+ *	       number of consumed bytes for MIDI message.
+ *
+ */
+int snd_fw_async_midi_port_init(struct snd_fw_async_midi_port *port,
+		struct fw_unit *unit, u64 addr, unsigned int len,
+		snd_fw_async_midi_port_fill fill)
+{
+	port->len = DIV_ROUND_UP(len, 4) * 4;
+	port->buf = kzalloc(port->len, GFP_KERNEL);
+	if (port->buf == NULL)
+		return -ENOMEM;
+
+	port->parent = fw_parent_device(unit);
+	port->addr = addr;
+	port->fill = fill;
+	port->idling = true;
+	port->next_ktime = ktime_set(0, 0);
+	port->error = false;
+
+	INIT_WORK(&port->work, midi_port_work);
+
+	return 0;
+}
+EXPORT_SYMBOL(snd_fw_async_midi_port_init);
+
+/**
+ * snd_fw_async_midi_port_destroy - free asynchronous MIDI port structure
+ * @port: the asynchronous MIDI port structure
+ */
+void snd_fw_async_midi_port_destroy(struct snd_fw_async_midi_port *port)
+{
+	snd_fw_async_midi_port_finish(port);
+	cancel_work_sync(&port->work);
+	kfree(port->buf);
+}
+EXPORT_SYMBOL(snd_fw_async_midi_port_destroy);
+
 MODULE_DESCRIPTION("FireWire audio helper functions");
 MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_LICENSE("GPL v2");
diff --git a/sound/firewire/lib.h b/sound/firewire/lib.h
index 02cfabc9c3c4..f3f6f84c48d6 100644
--- a/sound/firewire/lib.h
+++ b/sound/firewire/lib.h
@@ -3,6 +3,8 @@
 
 #include <linux/firewire-constants.h>
 #include <linux/types.h>
+#include <linux/sched.h>
+#include <sound/rawmidi.h>
 
 struct fw_unit;
 
@@ -20,4 +22,58 @@ static inline bool rcode_is_permanent_error(int rcode)
 	return rcode == RCODE_TYPE_ERROR || rcode == RCODE_ADDRESS_ERROR;
 }
 
+struct snd_fw_async_midi_port;
+typedef int (*snd_fw_async_midi_port_fill)(
+				struct snd_rawmidi_substream *substream,
+				u8 *buf);
+
+struct snd_fw_async_midi_port {
+	struct fw_device *parent;
+	struct work_struct work;
+	bool idling;
+	ktime_t next_ktime;
+	bool error;
+
+	u64 addr;
+	struct fw_transaction transaction;
+
+	u8 *buf;
+	unsigned int len;
+
+	struct snd_rawmidi_substream *substream;
+	snd_fw_async_midi_port_fill fill;
+	unsigned int consume_bytes;
+};
+
+int snd_fw_async_midi_port_init(struct snd_fw_async_midi_port *port,
+		struct fw_unit *unit, u64 addr, unsigned int len,
+		snd_fw_async_midi_port_fill fill);
+void snd_fw_async_midi_port_destroy(struct snd_fw_async_midi_port *port);
+
+/**
+ * snd_fw_async_midi_port_run - run transactions for the async MIDI port
+ * @port: the asynchronous MIDI port
+ * @substream: the MIDI substream
+ */
+static inline void
+snd_fw_async_midi_port_run(struct snd_fw_async_midi_port *port,
+			   struct snd_rawmidi_substream *substream)
+{
+	if (!port->error) {
+		port->substream = substream;
+		schedule_work(&port->work);
+	}
+}
+
+/**
+ * snd_fw_async_midi_port_finish - finish the asynchronous MIDI port
+ * @port: the asynchronous MIDI port
+ */
+static inline void
+snd_fw_async_midi_port_finish(struct snd_fw_async_midi_port *port)
+{
+	port->substream = NULL;
+	port->error = false;
+}
+
 #endif
diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile
index a926850864f6..06ff50f4e6c0 100644
--- a/sound/firewire/oxfw/Makefile
+++ b/sound/firewire/oxfw/Makefile
@@ -1,3 +1,3 @@
 snd-oxfw-objs := oxfw-command.o oxfw-stream.o oxfw-control.o oxfw-pcm.o \
 		 oxfw-proc.o oxfw-midi.o oxfw-hwdep.o oxfw.o
-obj-m += snd-oxfw.o
+obj-$(CONFIG_SND_OXFW) += snd-oxfw.o
diff --git a/sound/firewire/oxfw/oxfw-midi.c b/sound/firewire/oxfw/oxfw-midi.c
index 540a30338516..8665e1043d41 100644
--- a/sound/firewire/oxfw/oxfw-midi.c
+++ b/sound/firewire/oxfw/oxfw-midi.c
@@ -90,11 +90,11 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
 	spin_lock_irqsave(&oxfw->lock, flags);
 
 	if (up)
-		amdtp_stream_midi_trigger(&oxfw->tx_stream,
-					  substrm->number, substrm);
+		amdtp_am824_midi_trigger(&oxfw->tx_stream,
+					 substrm->number, substrm);
 	else
-		amdtp_stream_midi_trigger(&oxfw->tx_stream,
-					  substrm->number, NULL);
+		amdtp_am824_midi_trigger(&oxfw->tx_stream,
+					 substrm->number, NULL);
 
 	spin_unlock_irqrestore(&oxfw->lock, flags);
 }
@@ -107,11 +107,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
 	spin_lock_irqsave(&oxfw->lock, flags);
 
 	if (up)
-		amdtp_stream_midi_trigger(&oxfw->rx_stream,
-					  substrm->number, substrm);
+		amdtp_am824_midi_trigger(&oxfw->rx_stream,
+					 substrm->number, substrm);
 	else
-		amdtp_stream_midi_trigger(&oxfw->rx_stream,
-					  substrm->number, NULL);
+		amdtp_am824_midi_trigger(&oxfw->rx_stream,
+					 substrm->number, NULL);
 
 	spin_unlock_irqrestore(&oxfw->lock, flags);
 }
@@ -142,29 +142,11 @@ static void set_midi_substream_names(struct snd_oxfw *oxfw,
 
 int snd_oxfw_create_midi(struct snd_oxfw *oxfw)
 {
-	struct snd_oxfw_stream_formation formation;
 	struct snd_rawmidi *rmidi;
 	struct snd_rawmidi_str *str;
-	u8 *format;
-	int i, err;
-
-	/* If its stream has MIDI conformant data channel, add one MIDI port */
-	for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
-		format = oxfw->tx_stream_formats[i];
-		if (format != NULL) {
-			err = snd_oxfw_stream_parse_format(format, &formation);
-			if (err >= 0 && formation.midi > 0)
-				oxfw->midi_input_ports = 1;
-		}
-
-		format = oxfw->rx_stream_formats[i];
-		if (format != NULL) {
-			err = snd_oxfw_stream_parse_format(format, &formation);
-			if (err >= 0 && formation.midi > 0)
-				oxfw->midi_output_ports = 1;
-		}
-	}
-	if ((oxfw->midi_input_ports == 0) && (oxfw->midi_output_ports == 0))
+	int err;
+
+	if (oxfw->midi_input_ports == 0 && oxfw->midi_output_ports == 0)
 		return 0;
 
 	/* create midi ports */
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c
index 9c73930d0278..8d233417695d 100644
--- a/sound/firewire/oxfw/oxfw-pcm.c
+++ b/sound/firewire/oxfw/oxfw-pcm.c
@@ -134,11 +134,11 @@ static int init_hw_params(struct snd_oxfw *oxfw,
 			   SNDRV_PCM_INFO_MMAP_VALID;
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
-		runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS;
+		runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS;
 		stream = &oxfw->tx_stream;
 		formats = oxfw->tx_stream_formats;
 	} else {
-		runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS;
+		runtime->hw.formats = AM824_OUT_PCM_FORMAT_BITS;
 		stream = &oxfw->rx_stream;
 		formats = oxfw->rx_stream_formats;
 	}
@@ -158,7 +158,7 @@ static int init_hw_params(struct snd_oxfw *oxfw,
 	if (err < 0)
 		goto end;
 
-	err = amdtp_stream_add_pcm_hw_constraints(stream, runtime);
+	err = amdtp_am824_add_pcm_hw_constraints(stream, runtime);
 end:
 	return err;
 }
@@ -244,7 +244,7 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&oxfw->mutex);
 	}
 
-	amdtp_stream_set_pcm_format(&oxfw->tx_stream, params_format(hw_params));
+	amdtp_am824_set_pcm_format(&oxfw->tx_stream, params_format(hw_params));
 
 	return 0;
 }
@@ -265,7 +265,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
 		mutex_unlock(&oxfw->mutex);
 	}
 
-	amdtp_stream_set_pcm_format(&oxfw->rx_stream, params_format(hw_params));
+	amdtp_am824_set_pcm_format(&oxfw->rx_stream, params_format(hw_params));
 
 	return 0;
 }
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index 77ad5b98e806..7cb5743c073b 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -148,14 +148,17 @@ static int start_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream,
 	}
 
 	pcm_channels = formation.pcm;
-	midi_ports = DIV_ROUND_UP(formation.midi, 8);
+	midi_ports = formation.midi * 8;
 
 	/* The stream should have one pcm channels at least */
 	if (pcm_channels == 0) {
 		err = -EINVAL;
 		goto end;
 	}
-	amdtp_stream_set_parameters(stream, rate, pcm_channels, midi_ports);
+	err = amdtp_am824_set_parameters(stream, rate, pcm_channels, midi_ports,
+					 false);
+	if (err < 0)
+		goto end;
 
 	err = cmp_connection_establish(conn,
 				       amdtp_stream_get_max_payload(stream));
@@ -225,7 +228,7 @@ int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw,
 	if (err < 0)
 		goto end;
 
-	err = amdtp_stream_init(stream, oxfw->unit, s_dir, CIP_NONBLOCKING);
+	err = amdtp_am824_init(stream, oxfw->unit, s_dir, CIP_NONBLOCKING);
 	if (err < 0) {
 		amdtp_stream_destroy(stream);
 		cmp_connection_destroy(conn);
@@ -238,9 +241,12 @@ int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw,
 	 * packets. As a result, next isochronous packet includes more data
 	 * blocks than IEC 61883-6 defines.
 	 */
-	if (stream == &oxfw->tx_stream)
+	if (stream == &oxfw->tx_stream) {
 		oxfw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK |
 					 CIP_JUMBO_PAYLOAD;
+		if (oxfw->wrong_dbs)
+			oxfw->tx_stream.flags |= CIP_WRONG_DBS;
+	}
 end:
 	return err;
 }
@@ -480,8 +486,8 @@ int snd_oxfw_stream_parse_format(u8 *format,
 		}
 	}
 
-	if (formation->pcm  > AMDTP_MAX_CHANNELS_FOR_PCM ||
-	    formation->midi > AMDTP_MAX_CHANNELS_FOR_MIDI)
+	if (formation->pcm  > AM824_MAX_CHANNELS_FOR_PCM ||
+	    formation->midi > AM824_MAX_CHANNELS_FOR_MIDI)
 		return -ENOSYS;
 
 	return 0;
@@ -623,6 +629,9 @@ end:
 int snd_oxfw_stream_discover(struct snd_oxfw *oxfw)
 {
 	u8 plugs[AVC_PLUG_INFO_BUF_BYTES];
+	struct snd_oxfw_stream_formation formation;
+	u8 *format;
+	unsigned int i;
 	int err;
 
 	/* the number of plugs for isoc in/out, ext in/out  */
@@ -642,12 +651,42 @@ int snd_oxfw_stream_discover(struct snd_oxfw *oxfw)
 		err = fill_stream_formats(oxfw, AVC_GENERAL_PLUG_DIR_OUT, 0);
 		if (err < 0)
 			goto end;
+
+		for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
+			format = oxfw->tx_stream_formats[i];
+			if (format == NULL)
+				continue;
+			err = snd_oxfw_stream_parse_format(format, &formation);
+			if (err < 0)
+				continue;
+
+			/* Add one MIDI port. */
+			if (formation.midi > 0)
+				oxfw->midi_input_ports = 1;
+		}
+
 		oxfw->has_output = true;
 	}
 
 	/* use iPCR[0] if exists */
-	if (plugs[0] > 0)
+	if (plugs[0] > 0) {
 		err = fill_stream_formats(oxfw, AVC_GENERAL_PLUG_DIR_IN, 0);
+		if (err < 0)
+			goto end;
+
+		for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
+			format = oxfw->rx_stream_formats[i];
+			if (format == NULL)
+				continue;
+			err = snd_oxfw_stream_parse_format(format, &formation);
+			if (err < 0)
+				continue;
+
+			/* Add one MIDI port. */
+			if (formation.midi > 0)
+				oxfw->midi_output_ports = 1;
+		}
+	}
 end:
 	return err;
 }
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 8c6ce019f437..588b93f20c2e 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -18,6 +18,9 @@
 #define VENDOR_GRIFFIN		0x001292
 #define VENDOR_BEHRINGER	0x001564
 #define VENDOR_LACIE		0x00d04b
+#define VENDOR_TASCAM		0x00022e
+
+#define MODEL_SATELLITE		0x00200f
 
 #define SPECIFIER_1394TA	0x00a02d
 #define VERSION_AVC		0x010001
@@ -129,6 +132,40 @@ static void oxfw_card_free(struct snd_card *card)
 	mutex_destroy(&oxfw->mutex);
 }
 
+static void detect_quirks(struct snd_oxfw *oxfw)
+{
+	struct fw_device *fw_dev = fw_parent_device(oxfw->unit);
+	struct fw_csr_iterator it;
+	int key, val;
+	int vendor, model;
+
+	/* Seek from Root Directory of Config ROM. */
+	vendor = model = 0;
+	fw_csr_iterator_init(&it, fw_dev->config_rom + 5);
+	while (fw_csr_iterator_next(&it, &key, &val)) {
+		if (key == CSR_VENDOR)
+			vendor = val;
+		else if (key == CSR_MODEL)
+			model = val;
+	}
+
+	/*
+	 * Mackie Onyx Satellite with base station has a quirk to report a wrong
+	 * value in 'dbs' field of CIP header against its format information.
+	 */
+	if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE)
+		oxfw->wrong_dbs = true;
+
+	/*
+	 * TASCAM FireOne has physical control and requires a pair of additional
+	 * MIDI ports.
+	 */
+	if (vendor == VENDOR_TASCAM) {
+		oxfw->midi_input_ports++;
+		oxfw->midi_output_ports++;
+	}
+}
+
 static int oxfw_probe(struct fw_unit *unit,
 		       const struct ieee1394_device_id *id)
 {
@@ -157,6 +194,8 @@ static int oxfw_probe(struct fw_unit *unit,
 	if (err < 0)
 		goto error;
 
+	detect_quirks(oxfw);
+
 	err = name_card(oxfw);
 	if (err < 0)
 		goto error;
@@ -294,6 +333,13 @@ static const struct ieee1394_device_id oxfw_id_table[] = {
 		.specifier_id	= SPECIFIER_1394TA,
 		.version	= VERSION_AVC,
 	},
+	/* TASCAM, FireOne */
+	{
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_MODEL_ID,
+		.vendor_id	= VENDOR_TASCAM,
+		.model_id	= 0x800007,
+	},
 	{ }
 };
 MODULE_DEVICE_TABLE(ieee1394, oxfw_id_table);
diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h
index cace5ad4fe76..8392c424ad1d 100644
--- a/sound/firewire/oxfw/oxfw.h
+++ b/sound/firewire/oxfw/oxfw.h
@@ -28,7 +28,7 @@
 #include "../fcp.h"
 #include "../packets-buffer.h"
 #include "../iso-resources.h"
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
 #include "../cmp.h"
 
 struct device_info {
@@ -49,6 +49,7 @@ struct snd_oxfw {
 	struct mutex mutex;
 	spinlock_t lock;
 
+	bool wrong_dbs;
 	bool has_output;
 	u8 *tx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES];
 	u8 *rx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES];
diff --git a/sound/firewire/tascam/Makefile b/sound/firewire/tascam/Makefile
new file mode 100644
index 000000000000..0fc955d5bd15
--- /dev/null
+++ b/sound/firewire/tascam/Makefile
@@ -0,0 +1,4 @@
+snd-firewire-tascam-objs := tascam-proc.o amdtp-tascam.o tascam-stream.o \
+			    tascam-pcm.o tascam-hwdep.o tascam-transaction.o \
+			    tascam-midi.o tascam.o
+obj-$(CONFIG_SND_FIREWIRE_TASCAM) += snd-firewire-tascam.o
diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c
new file mode 100644
index 000000000000..9dd0fccd5ccc
--- /dev/null
+++ b/sound/firewire/tascam/amdtp-tascam.c
@@ -0,0 +1,243 @@
+/*
+ * amdtp-tascam.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <sound/pcm.h>
+#include "tascam.h"
+
+#define AMDTP_FMT_TSCM_TX	0x1e
+#define AMDTP_FMT_TSCM_RX	0x3e
+
+struct amdtp_tscm {
+	unsigned int pcm_channels;
+
+	void (*transfer_samples)(struct amdtp_stream *s,
+				 struct snd_pcm_substream *pcm,
+				 __be32 *buffer, unsigned int frames);
+};
+
+int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate)
+{
+	struct amdtp_tscm *p = s->protocol;
+	unsigned int data_channels;
+
+	if (amdtp_stream_running(s))
+		return -EBUSY;
+
+	data_channels = p->pcm_channels;
+
+	/* Packets in in-stream have extra 2 data channels. */
+	if (s->direction == AMDTP_IN_STREAM)
+		data_channels += 2;
+
+	return amdtp_stream_set_parameters(s, rate, data_channels);
+}
+
+static void write_pcm_s32(struct amdtp_stream *s,
+			  struct snd_pcm_substream *pcm,
+			  __be32 *buffer, unsigned int frames)
+{
+	struct amdtp_tscm *p = s->protocol;
+	struct snd_pcm_runtime *runtime = pcm->runtime;
+	unsigned int channels, remaining_frames, i, c;
+	const u32 *src;
+
+	channels = p->pcm_channels;
+	src = (void *)runtime->dma_area +
+			frames_to_bytes(runtime, s->pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+	for (i = 0; i < frames; ++i) {
+		for (c = 0; c < channels; ++c) {
+			buffer[c] = cpu_to_be32(*src);
+			src++;
+		}
+		buffer += s->data_block_quadlets;
+		if (--remaining_frames == 0)
+			src = (void *)runtime->dma_area;
+	}
+}
+
+static void write_pcm_s16(struct amdtp_stream *s,
+			  struct snd_pcm_substream *pcm,
+			  __be32 *buffer, unsigned int frames)
+{
+	struct amdtp_tscm *p = s->protocol;
+	struct snd_pcm_runtime *runtime = pcm->runtime;
+	unsigned int channels, remaining_frames, i, c;
+	const u16 *src;
+
+	channels = p->pcm_channels;
+	src = (void *)runtime->dma_area +
+			frames_to_bytes(runtime, s->pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+	for (i = 0; i < frames; ++i) {
+		for (c = 0; c < channels; ++c) {
+			buffer[c] = cpu_to_be32(*src << 16);
+			src++;
+		}
+		buffer += s->data_block_quadlets;
+		if (--remaining_frames == 0)
+			src = (void *)runtime->dma_area;
+	}
+}
+
+static void read_pcm_s32(struct amdtp_stream *s,
+			 struct snd_pcm_substream *pcm,
+			 __be32 *buffer, unsigned int frames)
+{
+	struct amdtp_tscm *p = s->protocol;
+	struct snd_pcm_runtime *runtime = pcm->runtime;
+	unsigned int channels, remaining_frames, i, c;
+	u32 *dst;
+
+	channels = p->pcm_channels;
+	dst  = (void *)runtime->dma_area +
+			frames_to_bytes(runtime, s->pcm_buffer_pointer);
+	remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+	/* The first data channel is for event counter. */
+	buffer += 1;
+
+	for (i = 0; i < frames; ++i) {
+		for (c = 0; c < channels; ++c) {
+			*dst = be32_to_cpu(buffer[c]);
+			dst++;
+		}
+		buffer += s->data_block_quadlets;
+		if (--remaining_frames == 0)
+			dst = (void *)runtime->dma_area;
+	}
+}
+
+static void write_pcm_silence(struct amdtp_stream *s, __be32 *buffer,
+			      unsigned int data_blocks)
+{
+	struct amdtp_tscm *p = s->protocol;
+	unsigned int channels, i, c;
+
+	channels = p->pcm_channels;
+
+	for (i = 0; i < data_blocks; ++i) {
+		for (c = 0; c < channels; ++c)
+			buffer[c] = 0x00000000;
+		buffer += s->data_block_quadlets;
+	}
+}
+
+int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s,
+				      struct snd_pcm_runtime *runtime)
+{
+	int err;
+
+	/*
+	 * Our implementation allows this protocol to deliver 24 bit sample in
+	 * 32bit data channel.
+	 */
+	err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+	if (err < 0)
+		return err;
+
+	return amdtp_stream_add_pcm_hw_constraints(s, runtime);
+}
+
+void amdtp_tscm_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
+{
+	struct amdtp_tscm *p = s->protocol;
+
+	if (WARN_ON(amdtp_stream_pcm_running(s)))
+		return;
+
+	switch (format) {
+	default:
+		WARN_ON(1);
+		/* fall through */
+	case SNDRV_PCM_FORMAT_S16:
+		if (s->direction == AMDTP_OUT_STREAM) {
+			p->transfer_samples = write_pcm_s16;
+			break;
+		}
+		WARN_ON(1);
+		/* fall through */
+	case SNDRV_PCM_FORMAT_S32:
+		if (s->direction == AMDTP_OUT_STREAM)
+			p->transfer_samples = write_pcm_s32;
+		else
+			p->transfer_samples = read_pcm_s32;
+		break;
+	}
+}
+
+static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
+					   __be32 *buffer,
+					   unsigned int data_blocks,
+					   unsigned int *syt)
+{
+	struct amdtp_tscm *p = (struct amdtp_tscm *)s->protocol;
+	struct snd_pcm_substream *pcm;
+
+	pcm = ACCESS_ONCE(s->pcm);
+	if (data_blocks > 0 && pcm)
+		p->transfer_samples(s, pcm, buffer, data_blocks);
+
+	/* A place holder for control messages. */
+
+	return data_blocks;
+}
+
+static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
+					   __be32 *buffer,
+					   unsigned int data_blocks,
+					   unsigned int *syt)
+{
+	struct amdtp_tscm *p = (struct amdtp_tscm *)s->protocol;
+	struct snd_pcm_substream *pcm;
+
+	/* This field is not used. */
+	*syt = 0x0000;
+
+	pcm = ACCESS_ONCE(s->pcm);
+	if (pcm)
+		p->transfer_samples(s, pcm, buffer, data_blocks);
+	else
+		write_pcm_silence(s, buffer, data_blocks);
+
+	return data_blocks;
+}
+
+int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit,
+		    enum amdtp_stream_direction dir, unsigned int pcm_channels)
+{
+	amdtp_stream_process_data_blocks_t process_data_blocks;
+	struct amdtp_tscm *p;
+	unsigned int fmt;
+	int err;
+
+	if (dir == AMDTP_IN_STREAM) {
+		fmt = AMDTP_FMT_TSCM_TX;
+		process_data_blocks = process_tx_data_blocks;
+	} else {
+		fmt = AMDTP_FMT_TSCM_RX;
+		process_data_blocks = process_rx_data_blocks;
+	}
+
+	err = amdtp_stream_init(s, unit, dir,
+				CIP_NONBLOCKING | CIP_SKIP_DBC_ZERO_CHECK, fmt,
+				process_data_blocks, sizeof(struct amdtp_tscm));
+	if (err < 0)
+		return 0;
+
+	/* Use fixed value for FDF field. */
+	s->fdf = 0x00;
+
+	/* This protocol uses fixed number of data channels for PCM samples. */
+	p = s->protocol;
+	p->pcm_channels = pcm_channels;
+
+	return 0;
+}
diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c
new file mode 100644
index 000000000000..131267c3a042
--- /dev/null
+++ b/sound/firewire/tascam/tascam-hwdep.c
@@ -0,0 +1,201 @@
+/*
+ * tascam-hwdep.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+/*
+ * This codes give three functionality.
+ *
+ * 1.get firewire node information
+ * 2.get notification about starting/stopping stream
+ * 3.lock/unlock stream
+ */
+
+#include "tascam.h"
+
+static long hwdep_read_locked(struct snd_tscm *tscm, char __user *buf,
+			      long count)
+{
+	union snd_firewire_event event;
+
+	memset(&event, 0, sizeof(event));
+
+	event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS;
+	event.lock_status.status = (tscm->dev_lock_count > 0);
+	tscm->dev_lock_changed = false;
+
+	count = min_t(long, count, sizeof(event.lock_status));
+
+	if (copy_to_user(buf, &event, count))
+		return -EFAULT;
+
+	return count;
+}
+
+static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count,
+		       loff_t *offset)
+{
+	struct snd_tscm *tscm = hwdep->private_data;
+	DEFINE_WAIT(wait);
+	union snd_firewire_event event;
+
+	spin_lock_irq(&tscm->lock);
+
+	while (!tscm->dev_lock_changed) {
+		prepare_to_wait(&tscm->hwdep_wait, &wait, TASK_INTERRUPTIBLE);
+		spin_unlock_irq(&tscm->lock);
+		schedule();
+		finish_wait(&tscm->hwdep_wait, &wait);
+		if (signal_pending(current))
+			return -ERESTARTSYS;
+		spin_lock_irq(&tscm->lock);
+	}
+
+	memset(&event, 0, sizeof(event));
+	count = hwdep_read_locked(tscm, buf, count);
+	spin_unlock_irq(&tscm->lock);
+
+	return count;
+}
+
+static unsigned int hwdep_poll(struct snd_hwdep *hwdep, struct file *file,
+			       poll_table *wait)
+{
+	struct snd_tscm *tscm = hwdep->private_data;
+	unsigned int events;
+
+	poll_wait(file, &tscm->hwdep_wait, wait);
+
+	spin_lock_irq(&tscm->lock);
+	if (tscm->dev_lock_changed)
+		events = POLLIN | POLLRDNORM;
+	else
+		events = 0;
+	spin_unlock_irq(&tscm->lock);
+
+	return events;
+}
+
+static int hwdep_get_info(struct snd_tscm *tscm, void __user *arg)
+{
+	struct fw_device *dev = fw_parent_device(tscm->unit);
+	struct snd_firewire_get_info info;
+
+	memset(&info, 0, sizeof(info));
+	info.type = SNDRV_FIREWIRE_TYPE_TASCAM;
+	info.card = dev->card->index;
+	*(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]);
+	*(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]);
+	strlcpy(info.device_name, dev_name(&dev->device),
+		sizeof(info.device_name));
+
+	if (copy_to_user(arg, &info, sizeof(info)))
+		return -EFAULT;
+
+	return 0;
+}
+
+static int hwdep_lock(struct snd_tscm *tscm)
+{
+	int err;
+
+	spin_lock_irq(&tscm->lock);
+
+	if (tscm->dev_lock_count == 0) {
+		tscm->dev_lock_count = -1;
+		err = 0;
+	} else {
+		err = -EBUSY;
+	}
+
+	spin_unlock_irq(&tscm->lock);
+
+	return err;
+}
+
+static int hwdep_unlock(struct snd_tscm *tscm)
+{
+	int err;
+
+	spin_lock_irq(&tscm->lock);
+
+	if (tscm->dev_lock_count == -1) {
+		tscm->dev_lock_count = 0;
+		err = 0;
+	} else {
+		err = -EBADFD;
+	}
+
+	spin_unlock_irq(&tscm->lock);
+
+	return err;
+}
+
+static int hwdep_release(struct snd_hwdep *hwdep, struct file *file)
+{
+	struct snd_tscm *tscm = hwdep->private_data;
+
+	spin_lock_irq(&tscm->lock);
+	if (tscm->dev_lock_count == -1)
+		tscm->dev_lock_count = 0;
+	spin_unlock_irq(&tscm->lock);
+
+	return 0;
+}
+
+static int hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file,
+	    unsigned int cmd, unsigned long arg)
+{
+	struct snd_tscm *tscm = hwdep->private_data;
+
+	switch (cmd) {
+	case SNDRV_FIREWIRE_IOCTL_GET_INFO:
+		return hwdep_get_info(tscm, (void __user *)arg);
+	case SNDRV_FIREWIRE_IOCTL_LOCK:
+		return hwdep_lock(tscm);
+	case SNDRV_FIREWIRE_IOCTL_UNLOCK:
+		return hwdep_unlock(tscm);
+	default:
+		return -ENOIOCTLCMD;
+	}
+}
+
+#ifdef CONFIG_COMPAT
+static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file,
+			      unsigned int cmd, unsigned long arg)
+{
+	return hwdep_ioctl(hwdep, file, cmd,
+			   (unsigned long)compat_ptr(arg));
+}
+#else
+#define hwdep_compat_ioctl NULL
+#endif
+
+static const struct snd_hwdep_ops hwdep_ops = {
+	.read		= hwdep_read,
+	.release	= hwdep_release,
+	.poll		= hwdep_poll,
+	.ioctl		= hwdep_ioctl,
+	.ioctl_compat	= hwdep_compat_ioctl,
+};
+
+int snd_tscm_create_hwdep_device(struct snd_tscm *tscm)
+{
+	struct snd_hwdep *hwdep;
+	int err;
+
+	err = snd_hwdep_new(tscm->card, "Tascam", 0, &hwdep);
+	if (err < 0)
+		return err;
+
+	strcpy(hwdep->name, "Tascam");
+	hwdep->iface = SNDRV_HWDEP_IFACE_FW_TASCAM;
+	hwdep->ops = hwdep_ops;
+	hwdep->private_data = tscm;
+	hwdep->exclusive = true;
+
+	return err;
+}
diff --git a/sound/firewire/tascam/tascam-midi.c b/sound/firewire/tascam/tascam-midi.c
new file mode 100644
index 000000000000..41f842079d9d
--- /dev/null
+++ b/sound/firewire/tascam/tascam-midi.c
@@ -0,0 +1,135 @@
+/*
+ * tascam-midi.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+static int midi_capture_open(struct snd_rawmidi_substream *substream)
+{
+	/* Do nothing. */
+	return 0;
+}
+
+static int midi_playback_open(struct snd_rawmidi_substream *substream)
+{
+	struct snd_tscm *tscm = substream->rmidi->private_data;
+
+	/* Initialize internal status. */
+	tscm->running_status[substream->number] = 0;
+	tscm->on_sysex[substream->number] = 0;
+	return 0;
+}
+
+static int midi_capture_close(struct snd_rawmidi_substream *substream)
+{
+	/* Do nothing. */
+	return 0;
+}
+
+static int midi_playback_close(struct snd_rawmidi_substream *substream)
+{
+	struct snd_tscm *tscm = substream->rmidi->private_data;
+
+	snd_fw_async_midi_port_finish(&tscm->out_ports[substream->number]);
+
+	return 0;
+}
+
+static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
+{
+	struct snd_tscm *tscm = substrm->rmidi->private_data;
+	unsigned long flags;
+
+	spin_lock_irqsave(&tscm->lock, flags);
+
+	if (up)
+		tscm->tx_midi_substreams[substrm->number] = substrm;
+	else
+		tscm->tx_midi_substreams[substrm->number] = NULL;
+
+	spin_unlock_irqrestore(&tscm->lock, flags);
+}
+
+static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
+{
+	struct snd_tscm *tscm = substrm->rmidi->private_data;
+	unsigned long flags;
+
+	spin_lock_irqsave(&tscm->lock, flags);
+
+	if (up)
+		snd_fw_async_midi_port_run(&tscm->out_ports[substrm->number],
+					   substrm);
+
+	spin_unlock_irqrestore(&tscm->lock, flags);
+}
+
+static struct snd_rawmidi_ops midi_capture_ops = {
+	.open		= midi_capture_open,
+	.close		= midi_capture_close,
+	.trigger	= midi_capture_trigger,
+};
+
+static struct snd_rawmidi_ops midi_playback_ops = {
+	.open		= midi_playback_open,
+	.close		= midi_playback_close,
+	.trigger	= midi_playback_trigger,
+};
+
+int snd_tscm_create_midi_devices(struct snd_tscm *tscm)
+{
+	struct snd_rawmidi *rmidi;
+	struct snd_rawmidi_str *stream;
+	struct snd_rawmidi_substream *subs;
+	int err;
+
+	err = snd_rawmidi_new(tscm->card, tscm->card->driver, 0,
+			      tscm->spec->midi_playback_ports,
+			      tscm->spec->midi_capture_ports,
+			      &rmidi);
+	if (err < 0)
+		return err;
+
+	snprintf(rmidi->name, sizeof(rmidi->name),
+		 "%s MIDI", tscm->card->shortname);
+	rmidi->private_data = tscm;
+
+	rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
+	snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+			    &midi_capture_ops);
+	stream = &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT];
+
+	/* Set port names for MIDI input. */
+	list_for_each_entry(subs, &stream->substreams, list) {
+		/* TODO: support virtual MIDI ports. */
+		if (subs->number < tscm->spec->midi_capture_ports) {
+			/* Hardware MIDI ports. */
+			snprintf(subs->name, sizeof(subs->name),
+				 "%s MIDI %d",
+				 tscm->card->shortname, subs->number + 1);
+		}
+	}
+
+	rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
+	snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+			    &midi_playback_ops);
+	stream = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT];
+
+	/* Set port names for MIDI ourput. */
+	list_for_each_entry(subs, &stream->substreams, list) {
+		if (subs->number < tscm->spec->midi_playback_ports) {
+			/* Hardware MIDI ports only. */
+			snprintf(subs->name, sizeof(subs->name),
+				 "%s MIDI %d",
+				 tscm->card->shortname, subs->number + 1);
+		}
+	}
+
+	rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX;
+
+	return 0;
+}
diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c
new file mode 100644
index 000000000000..380d3db969a5
--- /dev/null
+++ b/sound/firewire/tascam/tascam-pcm.c
@@ -0,0 +1,312 @@
+/*
+ * tascam-pcm.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+static void set_buffer_params(struct snd_pcm_hardware *hw)
+{
+	hw->period_bytes_min = 4 * hw->channels_min;
+	hw->period_bytes_max = hw->period_bytes_min * 2048;
+	hw->buffer_bytes_max = hw->period_bytes_max * 2;
+
+	hw->periods_min = 2;
+	hw->periods_max = UINT_MAX;
+}
+
+static int pcm_init_hw_params(struct snd_tscm *tscm,
+			      struct snd_pcm_substream *substream)
+{
+	static const struct snd_pcm_hardware hardware = {
+		.info = SNDRV_PCM_INFO_BATCH |
+			SNDRV_PCM_INFO_BLOCK_TRANSFER |
+			SNDRV_PCM_INFO_INTERLEAVED |
+			SNDRV_PCM_INFO_JOINT_DUPLEX |
+			SNDRV_PCM_INFO_MMAP |
+			SNDRV_PCM_INFO_MMAP_VALID,
+		.rates = SNDRV_PCM_RATE_44100 |
+			 SNDRV_PCM_RATE_48000 |
+			 SNDRV_PCM_RATE_88200 |
+			 SNDRV_PCM_RATE_96000,
+		.rate_min = 44100,
+		.rate_max = 96000,
+		.channels_min = 10,
+		.channels_max = 18,
+	};
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct amdtp_stream *stream;
+	unsigned int pcm_channels;
+
+	runtime->hw = hardware;
+
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		runtime->hw.formats = SNDRV_PCM_FMTBIT_S32;
+		stream = &tscm->tx_stream;
+		pcm_channels = tscm->spec->pcm_capture_analog_channels;
+	} else {
+		runtime->hw.formats =
+				SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S32;
+		stream = &tscm->rx_stream;
+		pcm_channels = tscm->spec->pcm_playback_analog_channels;
+	}
+
+	if (tscm->spec->has_adat)
+		pcm_channels += 8;
+	if (tscm->spec->has_spdif)
+		pcm_channels += 2;
+	runtime->hw.channels_min = runtime->hw.channels_max = pcm_channels;
+
+	set_buffer_params(&runtime->hw);
+
+	return amdtp_tscm_add_pcm_hw_constraints(stream, runtime);
+}
+
+static int pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_tscm *tscm = substream->private_data;
+	enum snd_tscm_clock clock;
+	unsigned int rate;
+	int err;
+
+	err = snd_tscm_stream_lock_try(tscm);
+	if (err < 0)
+		goto end;
+
+	err = pcm_init_hw_params(tscm, substream);
+	if (err < 0)
+		goto err_locked;
+
+	err = snd_tscm_stream_get_clock(tscm, &clock);
+	if (clock != SND_TSCM_CLOCK_INTERNAL ||
+	    amdtp_stream_pcm_running(&tscm->rx_stream) ||
+	    amdtp_stream_pcm_running(&tscm->tx_stream)) {
+		err = snd_tscm_stream_get_rate(tscm, &rate);
+		if (err < 0)
+			goto err_locked;
+		substream->runtime->hw.rate_min = rate;
+		substream->runtime->hw.rate_max = rate;
+	}
+
+	snd_pcm_set_sync(substream);
+end:
+	return err;
+err_locked:
+	snd_tscm_stream_lock_release(tscm);
+	return err;
+}
+
+static int pcm_close(struct snd_pcm_substream *substream)
+{
+	struct snd_tscm *tscm = substream->private_data;
+
+	snd_tscm_stream_lock_release(tscm);
+
+	return 0;
+}
+
+static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *hw_params)
+{
+	struct snd_tscm *tscm = substream->private_data;
+	int err;
+
+	err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+					       params_buffer_bytes(hw_params));
+	if (err < 0)
+		return err;
+
+	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+		mutex_lock(&tscm->mutex);
+		tscm->substreams_counter++;
+		mutex_unlock(&tscm->mutex);
+	}
+
+	amdtp_tscm_set_pcm_format(&tscm->tx_stream, params_format(hw_params));
+
+	return 0;
+}
+
+static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *hw_params)
+{
+	struct snd_tscm *tscm = substream->private_data;
+	int err;
+
+	err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+					       params_buffer_bytes(hw_params));
+	if (err < 0)
+		return err;
+
+	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+		mutex_lock(&tscm->mutex);
+		tscm->substreams_counter++;
+		mutex_unlock(&tscm->mutex);
+	}
+
+	amdtp_tscm_set_pcm_format(&tscm->rx_stream, params_format(hw_params));
+
+	return 0;
+}
+
+static int pcm_capture_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_tscm *tscm = substream->private_data;
+
+	mutex_lock(&tscm->mutex);
+
+	if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+		tscm->substreams_counter--;
+
+	snd_tscm_stream_stop_duplex(tscm);
+
+	mutex_unlock(&tscm->mutex);
+
+	return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_playback_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_tscm *tscm = substream->private_data;
+
+	mutex_lock(&tscm->mutex);
+
+	if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+		tscm->substreams_counter--;
+
+	snd_tscm_stream_stop_duplex(tscm);
+
+	mutex_unlock(&tscm->mutex);
+
+	return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_tscm *tscm = substream->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	int err;
+
+	mutex_lock(&tscm->mutex);
+
+	err = snd_tscm_stream_start_duplex(tscm, runtime->rate);
+	if (err >= 0)
+		amdtp_stream_pcm_prepare(&tscm->tx_stream);
+
+	mutex_unlock(&tscm->mutex);
+
+	return err;
+}
+
+static int pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_tscm *tscm = substream->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	int err;
+
+	mutex_lock(&tscm->mutex);
+
+	err = snd_tscm_stream_start_duplex(tscm, runtime->rate);
+	if (err >= 0)
+		amdtp_stream_pcm_prepare(&tscm->rx_stream);
+
+	mutex_unlock(&tscm->mutex);
+
+	return err;
+}
+
+static int pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_tscm *tscm = substream->private_data;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		amdtp_stream_pcm_trigger(&tscm->tx_stream, substream);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		amdtp_stream_pcm_trigger(&tscm->tx_stream, NULL);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_tscm *tscm = substream->private_data;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		amdtp_stream_pcm_trigger(&tscm->rx_stream, substream);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		amdtp_stream_pcm_trigger(&tscm->rx_stream, NULL);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstrm)
+{
+	struct snd_tscm *tscm = sbstrm->private_data;
+
+	return amdtp_stream_pcm_pointer(&tscm->tx_stream);
+}
+
+static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
+{
+	struct snd_tscm *tscm = sbstrm->private_data;
+
+	return amdtp_stream_pcm_pointer(&tscm->rx_stream);
+}
+
+static struct snd_pcm_ops pcm_capture_ops = {
+	.open		= pcm_open,
+	.close		= pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= pcm_capture_hw_params,
+	.hw_free	= pcm_capture_hw_free,
+	.prepare	= pcm_capture_prepare,
+	.trigger	= pcm_capture_trigger,
+	.pointer	= pcm_capture_pointer,
+	.page		= snd_pcm_lib_get_vmalloc_page,
+};
+
+static struct snd_pcm_ops pcm_playback_ops = {
+	.open		= pcm_open,
+	.close		= pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= pcm_playback_hw_params,
+	.hw_free	= pcm_playback_hw_free,
+	.prepare	= pcm_playback_prepare,
+	.trigger	= pcm_playback_trigger,
+	.pointer	= pcm_playback_pointer,
+	.page		= snd_pcm_lib_get_vmalloc_page,
+	.mmap		= snd_pcm_lib_mmap_vmalloc,
+};
+
+int snd_tscm_create_pcm_devices(struct snd_tscm *tscm)
+{
+	struct snd_pcm *pcm;
+	int err;
+
+	err = snd_pcm_new(tscm->card, tscm->card->driver, 0, 1, 1, &pcm);
+	if (err < 0)
+		return err;
+
+	pcm->private_data = tscm;
+	snprintf(pcm->name, sizeof(pcm->name),
+		 "%s PCM", tscm->card->shortname);
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops);
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops);
+
+	return 0;
+}
diff --git a/sound/firewire/tascam/tascam-proc.c b/sound/firewire/tascam/tascam-proc.c
new file mode 100644
index 000000000000..bfd4a4c06914
--- /dev/null
+++ b/sound/firewire/tascam/tascam-proc.c
@@ -0,0 +1,88 @@
+/*
+ * tascam-proc.h - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "./tascam.h"
+
+static void proc_read_firmware(struct snd_info_entry *entry,
+			       struct snd_info_buffer *buffer)
+{
+	struct snd_tscm *tscm = entry->private_data;
+	__be32 data;
+	unsigned int reg, fpga, arm, hw;
+	int err;
+
+	err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+			TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_REGISTER,
+			&data, sizeof(data), 0);
+	if (err < 0)
+		return;
+	reg = be32_to_cpu(data);
+
+	err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+			TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_FPGA,
+			&data, sizeof(data), 0);
+	if (err < 0)
+		return;
+	fpga = be32_to_cpu(data);
+
+	err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+			TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_ARM,
+			&data, sizeof(data), 0);
+	if (err < 0)
+		return;
+	arm = be32_to_cpu(data);
+
+	err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+			TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_HW,
+			&data, sizeof(data), 0);
+	if (err < 0)
+		return;
+	hw = be32_to_cpu(data);
+
+	snd_iprintf(buffer, "Register: %d (0x%08x)\n", reg & 0xffff, reg);
+	snd_iprintf(buffer, "FPGA:     %d (0x%08x)\n", fpga & 0xffff, fpga);
+	snd_iprintf(buffer, "ARM:      %d (0x%08x)\n", arm & 0xffff, arm);
+	snd_iprintf(buffer, "Hardware: %d (0x%08x)\n", hw >> 16, hw);
+}
+
+static void add_node(struct snd_tscm *tscm, struct snd_info_entry *root,
+		     const char *name,
+		     void (*op)(struct snd_info_entry *e,
+				struct snd_info_buffer *b))
+{
+	struct snd_info_entry *entry;
+
+	entry = snd_info_create_card_entry(tscm->card, name, root);
+	if (entry == NULL)
+		return;
+
+	snd_info_set_text_ops(entry, tscm, op);
+	if (snd_info_register(entry) < 0)
+		snd_info_free_entry(entry);
+}
+
+void snd_tscm_proc_init(struct snd_tscm *tscm)
+{
+	struct snd_info_entry *root;
+
+	/*
+	 * All nodes are automatically removed at snd_card_disconnect(),
+	 * by following to link list.
+	 */
+	root = snd_info_create_card_entry(tscm->card, "firewire",
+					  tscm->card->proc_root);
+	if (root == NULL)
+		return;
+	root->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+	if (snd_info_register(root) < 0) {
+		snd_info_free_entry(root);
+		return;
+	}
+
+	add_node(tscm, root, "firmware", proc_read_firmware);
+}
diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c
new file mode 100644
index 000000000000..0e6dd5c61f53
--- /dev/null
+++ b/sound/firewire/tascam/tascam-stream.c
@@ -0,0 +1,496 @@
+/*
+ * tascam-stream.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <linux/delay.h>
+#include "tascam.h"
+
+#define CALLBACK_TIMEOUT 500
+
+static int get_clock(struct snd_tscm *tscm, u32 *data)
+{
+	__be32 reg;
+	int err;
+
+	err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+				 TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS,
+				 &reg, sizeof(reg), 0);
+	if (err >= 0)
+		*data = be32_to_cpu(reg);
+
+	return err;
+}
+
+static int set_clock(struct snd_tscm *tscm, unsigned int rate,
+		     enum snd_tscm_clock clock)
+{
+	u32 data;
+	__be32 reg;
+	int err;
+
+	err = get_clock(tscm, &data);
+	if (err < 0)
+		return err;
+	data &= 0x0000ffff;
+
+	if (rate > 0) {
+		data &= 0x000000ff;
+		/* Base rate. */
+		if ((rate % 44100) == 0) {
+			data |= 0x00000100;
+			/* Multiplier. */
+			if (rate / 44100 == 2)
+				data |= 0x00008000;
+		} else if ((rate % 48000) == 0) {
+			data |= 0x00000200;
+			/* Multiplier. */
+			if (rate / 48000 == 2)
+				data |= 0x00008000;
+		} else {
+			return -EAGAIN;
+		}
+	}
+
+	if (clock != INT_MAX) {
+		data &= 0x0000ff00;
+		data |= clock + 1;
+	}
+
+	reg = cpu_to_be32(data);
+
+	err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+				 TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS,
+				 &reg, sizeof(reg), 0);
+	if (err < 0)
+		return err;
+
+	if (data & 0x00008000)
+		reg = cpu_to_be32(0x0000001a);
+	else
+		reg = cpu_to_be32(0x0000000d);
+
+	return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+				  TSCM_ADDR_BASE + TSCM_OFFSET_MULTIPLEX_MODE,
+				  &reg, sizeof(reg), 0);
+}
+
+int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate)
+{
+	u32 data = 0x0;
+	unsigned int trials = 0;
+	int err;
+
+	while (data == 0x0 || trials++ < 5) {
+		err = get_clock(tscm, &data);
+		if (err < 0)
+			return err;
+
+		data = (data & 0xff000000) >> 24;
+	}
+
+	/* Check base rate. */
+	if ((data & 0x0f) == 0x01)
+		*rate = 44100;
+	else if ((data & 0x0f) == 0x02)
+		*rate = 48000;
+	else
+		return -EAGAIN;
+
+	/* Check multiplier. */
+	if ((data & 0xf0) == 0x80)
+		*rate *= 2;
+	else if ((data & 0xf0) != 0x00)
+		return -EAGAIN;
+
+	return err;
+}
+
+int snd_tscm_stream_get_clock(struct snd_tscm *tscm, enum snd_tscm_clock *clock)
+{
+	u32 data;
+	int err;
+
+	err = get_clock(tscm, &data);
+	if (err < 0)
+		return err;
+
+	*clock = ((data & 0x00ff0000) >> 16) - 1;
+	if (*clock < 0 || *clock > SND_TSCM_CLOCK_ADAT)
+		return -EIO;
+
+	return 0;
+}
+
+static int enable_data_channels(struct snd_tscm *tscm)
+{
+	__be32 reg;
+	u32 data;
+	unsigned int i;
+	int err;
+
+	data = 0;
+	for (i = 0; i < tscm->spec->pcm_capture_analog_channels; ++i)
+		data |= BIT(i);
+	if (tscm->spec->has_adat)
+		data |= 0x0000ff00;
+	if (tscm->spec->has_spdif)
+		data |= 0x00030000;
+
+	reg = cpu_to_be32(data);
+	err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+				 TSCM_ADDR_BASE + TSCM_OFFSET_TX_PCM_CHANNELS,
+				 &reg, sizeof(reg), 0);
+	if (err < 0)
+		return err;
+
+	data = 0;
+	for (i = 0; i < tscm->spec->pcm_playback_analog_channels; ++i)
+		data |= BIT(i);
+	if (tscm->spec->has_adat)
+		data |= 0x0000ff00;
+	if (tscm->spec->has_spdif)
+		data |= 0x00030000;
+
+	reg = cpu_to_be32(data);
+	return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+				  TSCM_ADDR_BASE + TSCM_OFFSET_RX_PCM_CHANNELS,
+				  &reg, sizeof(reg), 0);
+}
+
+static int set_stream_formats(struct snd_tscm *tscm, unsigned int rate)
+{
+	__be32 reg;
+	int err;
+
+	/* Set an option for unknown purpose. */
+	reg = cpu_to_be32(0x00200000);
+	err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+				 TSCM_ADDR_BASE + TSCM_OFFSET_SET_OPTION,
+				 &reg, sizeof(reg), 0);
+	if (err < 0)
+		return err;
+
+	err = enable_data_channels(tscm);
+	if (err < 0)
+		return err;
+
+	return set_clock(tscm, rate, INT_MAX);
+}
+
+static void finish_session(struct snd_tscm *tscm)
+{
+	__be32 reg;
+
+	reg = 0;
+	snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+			   TSCM_ADDR_BASE + TSCM_OFFSET_START_STREAMING,
+			   &reg, sizeof(reg), 0);
+
+	reg = 0;
+	snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+			   TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_ON,
+			   &reg, sizeof(reg), 0);
+
+}
+
+static int begin_session(struct snd_tscm *tscm)
+{
+	__be32 reg;
+	int err;
+
+	reg = cpu_to_be32(0x00000001);
+	err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+				 TSCM_ADDR_BASE + TSCM_OFFSET_START_STREAMING,
+				 &reg, sizeof(reg), 0);
+	if (err < 0)
+		return err;
+
+	reg = cpu_to_be32(0x00000001);
+	err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+				 TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_ON,
+				 &reg, sizeof(reg), 0);
+	if (err < 0)
+		return err;
+
+	/* Set an option for unknown purpose. */
+	reg = cpu_to_be32(0x00002000);
+	err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+				 TSCM_ADDR_BASE + TSCM_OFFSET_SET_OPTION,
+				 &reg, sizeof(reg), 0);
+	if (err < 0)
+		return err;
+
+	/* Start multiplexing PCM samples on packets. */
+	reg = cpu_to_be32(0x00000001);
+	return snd_fw_transaction(tscm->unit,
+				  TCODE_WRITE_QUADLET_REQUEST,
+				  TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_TX_ON,
+				  &reg, sizeof(reg), 0);
+}
+
+static void release_resources(struct snd_tscm *tscm)
+{
+	__be32 reg;
+
+	/* Unregister channels. */
+	reg = cpu_to_be32(0x00000000);
+	snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+			   TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_TX_CH,
+			   &reg, sizeof(reg), 0);
+	reg = cpu_to_be32(0x00000000);
+	snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+			   TSCM_ADDR_BASE + TSCM_OFFSET_UNKNOWN,
+			   &reg, sizeof(reg), 0);
+	reg = cpu_to_be32(0x00000000);
+	snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+			   TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_CH,
+			   &reg, sizeof(reg), 0);
+
+	/* Release isochronous resources. */
+	fw_iso_resources_free(&tscm->tx_resources);
+	fw_iso_resources_free(&tscm->rx_resources);
+}
+
+static int keep_resources(struct snd_tscm *tscm, unsigned int rate)
+{
+	__be32 reg;
+	int err;
+
+	/* Keep resources for in-stream. */
+	err = amdtp_tscm_set_parameters(&tscm->tx_stream, rate);
+	if (err < 0)
+		return err;
+	err = fw_iso_resources_allocate(&tscm->tx_resources,
+			amdtp_stream_get_max_payload(&tscm->tx_stream),
+			fw_parent_device(tscm->unit)->max_speed);
+	if (err < 0)
+		goto error;
+
+	/* Keep resources for out-stream. */
+	err = amdtp_tscm_set_parameters(&tscm->rx_stream, rate);
+	if (err < 0)
+		return err;
+	err = fw_iso_resources_allocate(&tscm->rx_resources,
+			amdtp_stream_get_max_payload(&tscm->rx_stream),
+			fw_parent_device(tscm->unit)->max_speed);
+	if (err < 0)
+		return err;
+
+	/* Register the isochronous channel for transmitting stream. */
+	reg = cpu_to_be32(tscm->tx_resources.channel);
+	err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+				 TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_TX_CH,
+				 &reg, sizeof(reg), 0);
+	if (err < 0)
+		goto error;
+
+	/* Unknown */
+	reg = cpu_to_be32(0x00000002);
+	err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+				 TSCM_ADDR_BASE + TSCM_OFFSET_UNKNOWN,
+				 &reg, sizeof(reg), 0);
+	if (err < 0)
+		goto error;
+
+	/* Register the isochronous channel for receiving stream. */
+	reg = cpu_to_be32(tscm->rx_resources.channel);
+	err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+				 TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_CH,
+				 &reg, sizeof(reg), 0);
+	if (err < 0)
+		goto error;
+
+	return 0;
+error:
+	release_resources(tscm);
+	return err;
+}
+
+int snd_tscm_stream_init_duplex(struct snd_tscm *tscm)
+{
+	unsigned int pcm_channels;
+	int err;
+
+	/* For out-stream. */
+	err = fw_iso_resources_init(&tscm->rx_resources, tscm->unit);
+	if (err < 0)
+		return err;
+	pcm_channels = tscm->spec->pcm_playback_analog_channels;
+	if (tscm->spec->has_adat)
+		pcm_channels += 8;
+	if (tscm->spec->has_spdif)
+		pcm_channels += 2;
+	err = amdtp_tscm_init(&tscm->rx_stream, tscm->unit, AMDTP_OUT_STREAM,
+			      pcm_channels);
+	if (err < 0)
+		return err;
+
+	/* For in-stream. */
+	err = fw_iso_resources_init(&tscm->tx_resources, tscm->unit);
+	if (err < 0)
+		return err;
+	pcm_channels = tscm->spec->pcm_capture_analog_channels;
+	if (tscm->spec->has_adat)
+		pcm_channels += 8;
+	if (tscm->spec->has_spdif)
+		pcm_channels += 2;
+	err = amdtp_tscm_init(&tscm->tx_stream, tscm->unit, AMDTP_IN_STREAM,
+			      pcm_channels);
+	if (err < 0)
+		amdtp_stream_destroy(&tscm->rx_stream);
+
+	return 0;
+}
+
+/* At bus reset, streaming is stopped and some registers are clear. */
+void snd_tscm_stream_update_duplex(struct snd_tscm *tscm)
+{
+	amdtp_stream_pcm_abort(&tscm->tx_stream);
+	amdtp_stream_stop(&tscm->tx_stream);
+
+	amdtp_stream_pcm_abort(&tscm->rx_stream);
+	amdtp_stream_stop(&tscm->rx_stream);
+}
+
+/*
+ * This function should be called before starting streams or after stopping
+ * streams.
+ */
+void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm)
+{
+	amdtp_stream_destroy(&tscm->rx_stream);
+	amdtp_stream_destroy(&tscm->tx_stream);
+
+	fw_iso_resources_destroy(&tscm->rx_resources);
+	fw_iso_resources_destroy(&tscm->tx_resources);
+}
+
+int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate)
+{
+	unsigned int curr_rate;
+	int err;
+
+	if (tscm->substreams_counter == 0)
+		return 0;
+
+	err = snd_tscm_stream_get_rate(tscm, &curr_rate);
+	if (err < 0)
+		return err;
+	if (curr_rate != rate ||
+	    amdtp_streaming_error(&tscm->tx_stream) ||
+	    amdtp_streaming_error(&tscm->rx_stream)) {
+		finish_session(tscm);
+
+		amdtp_stream_stop(&tscm->tx_stream);
+		amdtp_stream_stop(&tscm->rx_stream);
+
+		release_resources(tscm);
+	}
+
+	if (!amdtp_stream_running(&tscm->tx_stream)) {
+		amdtp_stream_set_sync(CIP_SYNC_TO_DEVICE,
+				      &tscm->tx_stream, &tscm->rx_stream);
+		err = keep_resources(tscm, rate);
+		if (err < 0)
+			goto error;
+
+		err = set_stream_formats(tscm, rate);
+		if (err < 0)
+			goto error;
+
+		err = begin_session(tscm);
+		if (err < 0)
+			goto error;
+
+		err = amdtp_stream_start(&tscm->tx_stream,
+				tscm->tx_resources.channel,
+				fw_parent_device(tscm->unit)->max_speed);
+		if (err < 0)
+			goto error;
+
+		if (!amdtp_stream_wait_callback(&tscm->tx_stream,
+						CALLBACK_TIMEOUT)) {
+			err = -ETIMEDOUT;
+			goto error;
+		}
+	}
+
+	if (!amdtp_stream_running(&tscm->rx_stream)) {
+		err = amdtp_stream_start(&tscm->rx_stream,
+				tscm->rx_resources.channel,
+				fw_parent_device(tscm->unit)->max_speed);
+		if (err < 0)
+			goto error;
+
+		if (!amdtp_stream_wait_callback(&tscm->rx_stream,
+						CALLBACK_TIMEOUT)) {
+			err = -ETIMEDOUT;
+			goto error;
+		}
+	}
+
+	return 0;
+error:
+	amdtp_stream_stop(&tscm->tx_stream);
+	amdtp_stream_stop(&tscm->rx_stream);
+
+	finish_session(tscm);
+	release_resources(tscm);
+
+	return err;
+}
+
+void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm)
+{
+	if (tscm->substreams_counter > 0)
+		return;
+
+	amdtp_stream_stop(&tscm->tx_stream);
+	amdtp_stream_stop(&tscm->rx_stream);
+
+	finish_session(tscm);
+	release_resources(tscm);
+}
+
+void snd_tscm_stream_lock_changed(struct snd_tscm *tscm)
+{
+	tscm->dev_lock_changed = true;
+	wake_up(&tscm->hwdep_wait);
+}
+
+int snd_tscm_stream_lock_try(struct snd_tscm *tscm)
+{
+	int err;
+
+	spin_lock_irq(&tscm->lock);
+
+	/* user land lock this */
+	if (tscm->dev_lock_count < 0) {
+		err = -EBUSY;
+		goto end;
+	}
+
+	/* this is the first time */
+	if (tscm->dev_lock_count++ == 0)
+		snd_tscm_stream_lock_changed(tscm);
+	err = 0;
+end:
+	spin_unlock_irq(&tscm->lock);
+	return err;
+}
+
+void snd_tscm_stream_lock_release(struct snd_tscm *tscm)
+{
+	spin_lock_irq(&tscm->lock);
+
+	if (WARN_ON(tscm->dev_lock_count <= 0))
+		goto end;
+	if (--tscm->dev_lock_count == 0)
+		snd_tscm_stream_lock_changed(tscm);
+end:
+	spin_unlock_irq(&tscm->lock);
+}
diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c
new file mode 100644
index 000000000000..904ce0329fa1
--- /dev/null
+++ b/sound/firewire/tascam/tascam-transaction.c
@@ -0,0 +1,302 @@
+/*
+ * tascam-transaction.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+/*
+ * When return minus value, given argument is not MIDI status.
+ * When return 0, given argument is a beginning of system exclusive.
+ * When return the others, given argument is MIDI data.
+ */
+static inline int calculate_message_bytes(u8 status)
+{
+	switch (status) {
+	case 0xf6:	/* Tune request. */
+	case 0xf8:	/* Timing clock. */
+	case 0xfa:	/* Start. */
+	case 0xfb:	/* Continue. */
+	case 0xfc:	/* Stop. */
+	case 0xfe:	/* Active sensing. */
+	case 0xff:	/* System reset. */
+		return 1;
+	case 0xf1:	/* MIDI time code quarter frame. */
+	case 0xf3:	/* Song select. */
+		return 2;
+	case 0xf2:	/* Song position pointer. */
+		return 3;
+	case 0xf0:	/* Exclusive. */
+		return 0;
+	case 0xf7:	/* End of exclusive. */
+		break;
+	case 0xf4:	/* Undefined. */
+	case 0xf5:	/* Undefined. */
+	case 0xf9:	/* Undefined. */
+	case 0xfd:	/* Undefined. */
+		break;
+	default:
+		switch (status & 0xf0) {
+		case 0x80:	/* Note on. */
+		case 0x90:	/* Note off. */
+		case 0xa0:	/* Polyphonic key pressure. */
+		case 0xb0:	/* Control change and Mode change. */
+		case 0xe0:	/* Pitch bend change. */
+			return 3;
+		case 0xc0:	/* Program change. */
+		case 0xd0:	/* Channel pressure. */
+			return 2;
+		default:
+		break;
+		}
+	break;
+	}
+
+	return -EINVAL;
+}
+
+static int fill_message(struct snd_rawmidi_substream *substream, u8 *buf)
+{
+	struct snd_tscm *tscm = substream->rmidi->private_data;
+	unsigned int port = substream->number;
+	int i, len, consume;
+	u8 *label, *msg;
+	u8 status;
+
+	/* The first byte is used for label, the rest for MIDI bytes. */
+	label = buf;
+	msg = buf + 1;
+
+	consume = snd_rawmidi_transmit_peek(substream, msg, 3);
+	if (consume == 0)
+		return 0;
+
+	/* On exclusive message. */
+	if (tscm->on_sysex[port]) {
+		/* Seek the end of exclusives. */
+		for (i = 0; i < consume; ++i) {
+			if (msg[i] == 0xf7) {
+				tscm->on_sysex[port] = false;
+				break;
+			}
+		}
+
+		/* At the end of exclusive message, use label 0x07. */
+		if (!tscm->on_sysex[port]) {
+			consume = i + 1;
+			*label = (port << 4) | 0x07;
+		/* During exclusive message, use label 0x04. */
+		} else if (consume == 3) {
+			*label = (port << 4) | 0x04;
+		/* We need to fill whole 3 bytes. Go to next change. */
+		} else {
+			return 0;
+		}
+
+		len = consume;
+	} else {
+		/* The beginning of exclusives. */
+		if (msg[0] == 0xf0) {
+			/* Transfer it in next chance in another condition. */
+			tscm->on_sysex[port] = true;
+			return 0;
+		} else {
+			/* On running-status. */
+			if ((msg[0] & 0x80) != 0x80)
+				status = tscm->running_status[port];
+			else
+				status = msg[0];
+
+			/* Calculate consume bytes. */
+			len = calculate_message_bytes(status);
+			if (len <= 0)
+				return 0;
+
+			/* On running-status. */
+			if ((msg[0] & 0x80) != 0x80) {
+				/* Enough MIDI bytes were not retrieved. */
+				if (consume < len - 1)
+					return 0;
+				consume = len - 1;
+
+				msg[2] = msg[1];
+				msg[1] = msg[0];
+				msg[0] = tscm->running_status[port];
+			} else {
+				/* Enough MIDI bytes were not retrieved. */
+				if (consume < len)
+					return 0;
+				consume = len;
+
+				tscm->running_status[port] = msg[0];
+			}
+		}
+
+		*label = (port << 4) | (msg[0] >> 4);
+	}
+
+	if (len > 0 && len < 3)
+		memset(msg + len, 0, 3 - len);
+
+	return consume;
+}
+
+static void handle_midi_tx(struct fw_card *card, struct fw_request *request,
+			   int tcode, int destination, int source,
+			   int generation, unsigned long long offset,
+			   void *data, size_t length, void *callback_data)
+{
+	struct snd_tscm *tscm = callback_data;
+	u32 *buf = (u32 *)data;
+	unsigned int messages;
+	unsigned int i;
+	unsigned int port;
+	struct snd_rawmidi_substream *substream;
+	u8 *b;
+	int bytes;
+
+	if (offset != tscm->async_handler.offset)
+		goto end;
+
+	messages = length / 8;
+	for (i = 0; i < messages; i++) {
+		b = (u8 *)(buf + i * 2);
+
+		port = b[0] >> 4;
+		/* TODO: support virtual MIDI ports. */
+		if (port >= tscm->spec->midi_capture_ports)
+			goto end;
+
+		/* Assume the message length. */
+		bytes = calculate_message_bytes(b[1]);
+		/* On MIDI data or exclusives. */
+		if (bytes <= 0) {
+			/* Seek the end of exclusives. */
+			for (bytes = 1; bytes < 4; bytes++) {
+				if (b[bytes] == 0xf7)
+					break;
+			}
+			if (bytes == 4)
+				bytes = 3;
+		}
+
+		substream = ACCESS_ONCE(tscm->tx_midi_substreams[port]);
+		if (substream != NULL)
+			snd_rawmidi_receive(substream, b + 1, bytes);
+	}
+end:
+	fw_send_response(card, request, RCODE_COMPLETE);
+}
+
+int snd_tscm_transaction_register(struct snd_tscm *tscm)
+{
+	static const struct fw_address_region resp_register_region = {
+		.start	= 0xffffe0000000ull,
+		.end	= 0xffffe000ffffull,
+	};
+	unsigned int i;
+	int err;
+
+	/*
+	 * Usually, two quadlets are transferred by one transaction. The first
+	 * quadlet has MIDI messages, the rest includes timestamp.
+	 * Sometimes, 8 set of the data is transferred by a block transaction.
+	 */
+	tscm->async_handler.length = 8 * 8;
+	tscm->async_handler.address_callback = handle_midi_tx;
+	tscm->async_handler.callback_data = tscm;
+
+	err = fw_core_add_address_handler(&tscm->async_handler,
+					  &resp_register_region);
+	if (err < 0)
+		return err;
+
+	err = snd_tscm_transaction_reregister(tscm);
+	if (err < 0)
+		goto error;
+
+	for (i = 0; i < TSCM_MIDI_OUT_PORT_MAX; i++) {
+		err = snd_fw_async_midi_port_init(
+				&tscm->out_ports[i], tscm->unit,
+				TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_RX_QUAD,
+				4, fill_message);
+		if (err < 0)
+			goto error;
+	}
+
+	return err;
+error:
+	fw_core_remove_address_handler(&tscm->async_handler);
+	return err;
+}
+
+/* At bus reset, these registers are cleared. */
+int snd_tscm_transaction_reregister(struct snd_tscm *tscm)
+{
+	struct fw_device *device = fw_parent_device(tscm->unit);
+	__be32 reg;
+	int err;
+
+	/* Register messaging address. Block transaction is not allowed. */
+	reg = cpu_to_be32((device->card->node_id << 16) |
+			  (tscm->async_handler.offset >> 32));
+	err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+				 TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_HI,
+				 &reg, sizeof(reg), 0);
+	if (err < 0)
+		return err;
+
+	reg = cpu_to_be32(tscm->async_handler.offset);
+	err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+				 TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_LO,
+				 &reg, sizeof(reg), 0);
+	if (err < 0)
+		return err;
+
+	/* Turn on messaging. */
+	reg = cpu_to_be32(0x00000001);
+	err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+				  TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ON,
+				  &reg, sizeof(reg), 0);
+	if (err < 0)
+		return err;
+
+	/* Turn on FireWire LED. */
+	reg = cpu_to_be32(0x0001008e);
+	return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+				  TSCM_ADDR_BASE + TSCM_OFFSET_LED_POWER,
+				  &reg, sizeof(reg), 0);
+}
+
+void snd_tscm_transaction_unregister(struct snd_tscm *tscm)
+{
+	__be32 reg;
+	unsigned int i;
+
+	/* Turn off FireWire LED. */
+	reg = cpu_to_be32(0x0000008e);
+	snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+			   TSCM_ADDR_BASE + TSCM_OFFSET_LED_POWER,
+			   &reg, sizeof(reg), 0);
+
+	/* Turn off messaging. */
+	reg = cpu_to_be32(0x00000000);
+	snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+			   TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ON,
+			   &reg, sizeof(reg), 0);
+
+	/* Unregister the address. */
+	snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+			   TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_HI,
+			   &reg, sizeof(reg), 0);
+	snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+			   TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_LO,
+			   &reg, sizeof(reg), 0);
+
+	fw_core_remove_address_handler(&tscm->async_handler);
+	for (i = 0; i < TSCM_MIDI_OUT_PORT_MAX; i++)
+		snd_fw_async_midi_port_destroy(&tscm->out_ports[i]);
+}
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
new file mode 100644
index 000000000000..ee0bc1839508
--- /dev/null
+++ b/sound/firewire/tascam/tascam.c
@@ -0,0 +1,209 @@
+/*
+ * tascam.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+MODULE_DESCRIPTION("TASCAM FireWire series Driver");
+MODULE_AUTHOR("Takashi Sakamoto <o-takashi@sakamocchi.jp>");
+MODULE_LICENSE("GPL v2");
+
+static struct snd_tscm_spec model_specs[] = {
+	{
+		.name = "FW-1884",
+		.has_adat = true,
+		.has_spdif = true,
+		.pcm_capture_analog_channels = 8,
+		.pcm_playback_analog_channels = 8,
+		.midi_capture_ports = 4,
+		.midi_playback_ports = 4,
+		.is_controller = true,
+	},
+	{
+		.name = "FW-1082",
+		.has_adat = false,
+		.has_spdif = true,
+		.pcm_capture_analog_channels = 8,
+		.pcm_playback_analog_channels = 2,
+		.midi_capture_ports = 2,
+		.midi_playback_ports = 2,
+		.is_controller = true,
+	},
+	/* FW-1804 may be supported. */
+};
+
+static int identify_model(struct snd_tscm *tscm)
+{
+	struct fw_device *fw_dev = fw_parent_device(tscm->unit);
+	const u32 *config_rom = fw_dev->config_rom;
+	char model[9];
+	unsigned int i;
+	u8 c;
+
+	if (fw_dev->config_rom_length < 30) {
+		dev_err(&tscm->unit->device,
+			"Configuration ROM is too short.\n");
+		return -ENODEV;
+	}
+
+	/* Pick up model name from certain addresses. */
+	for (i = 0; i < 8; i++) {
+		c = config_rom[28 + i / 4] >> (24 - 8 * (i % 4));
+		if (c == '\0')
+			break;
+		model[i] = c;
+	}
+	model[i] = '\0';
+
+	for (i = 0; i < ARRAY_SIZE(model_specs); i++) {
+		if (strcmp(model, model_specs[i].name) == 0) {
+			tscm->spec = &model_specs[i];
+			break;
+		}
+	}
+	if (tscm->spec == NULL)
+		return -ENODEV;
+
+	strcpy(tscm->card->driver, "FW-TASCAM");
+	strcpy(tscm->card->shortname, model);
+	strcpy(tscm->card->mixername, model);
+	snprintf(tscm->card->longname, sizeof(tscm->card->longname),
+		 "TASCAM %s, GUID %08x%08x at %s, S%d", model,
+		 fw_dev->config_rom[3], fw_dev->config_rom[4],
+		 dev_name(&tscm->unit->device), 100 << fw_dev->max_speed);
+
+	return 0;
+}
+
+static void tscm_card_free(struct snd_card *card)
+{
+	struct snd_tscm *tscm = card->private_data;
+
+	snd_tscm_transaction_unregister(tscm);
+	snd_tscm_stream_destroy_duplex(tscm);
+
+	fw_unit_put(tscm->unit);
+
+	mutex_destroy(&tscm->mutex);
+}
+
+static int snd_tscm_probe(struct fw_unit *unit,
+			   const struct ieee1394_device_id *entry)
+{
+	struct snd_card *card;
+	struct snd_tscm *tscm;
+	int err;
+
+	/* create card */
+	err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE,
+			   sizeof(struct snd_tscm), &card);
+	if (err < 0)
+		return err;
+	card->private_free = tscm_card_free;
+
+	/* initialize myself */
+	tscm = card->private_data;
+	tscm->card = card;
+	tscm->unit = fw_unit_get(unit);
+
+	mutex_init(&tscm->mutex);
+	spin_lock_init(&tscm->lock);
+	init_waitqueue_head(&tscm->hwdep_wait);
+
+	err = identify_model(tscm);
+	if (err < 0)
+		goto error;
+
+	snd_tscm_proc_init(tscm);
+
+	err = snd_tscm_stream_init_duplex(tscm);
+	if (err < 0)
+		goto error;
+
+	err = snd_tscm_create_pcm_devices(tscm);
+	if (err < 0)
+		goto error;
+
+	err = snd_tscm_transaction_register(tscm);
+	if (err < 0)
+		goto error;
+
+	err = snd_tscm_create_midi_devices(tscm);
+	if (err < 0)
+		goto error;
+
+	err = snd_tscm_create_hwdep_device(tscm);
+	if (err < 0)
+		goto error;
+
+	err = snd_card_register(card);
+	if (err < 0)
+		goto error;
+
+	dev_set_drvdata(&unit->device, tscm);
+
+	return err;
+error:
+	snd_card_free(card);
+	return err;
+}
+
+static void snd_tscm_update(struct fw_unit *unit)
+{
+	struct snd_tscm *tscm = dev_get_drvdata(&unit->device);
+
+	snd_tscm_transaction_reregister(tscm);
+
+	mutex_lock(&tscm->mutex);
+	snd_tscm_stream_update_duplex(tscm);
+	mutex_unlock(&tscm->mutex);
+}
+
+static void snd_tscm_remove(struct fw_unit *unit)
+{
+	struct snd_tscm *tscm = dev_get_drvdata(&unit->device);
+
+	/* No need to wait for releasing card object in this context. */
+	snd_card_free_when_closed(tscm->card);
+}
+
+static const struct ieee1394_device_id snd_tscm_id_table[] = {
+	{
+		.match_flags = IEEE1394_MATCH_VENDOR_ID |
+			       IEEE1394_MATCH_SPECIFIER_ID,
+		.vendor_id = 0x00022e,
+		.specifier_id = 0x00022e,
+	},
+	/* FE-08 requires reverse-engineering because it just has faders. */
+	{}
+};
+MODULE_DEVICE_TABLE(ieee1394, snd_tscm_id_table);
+
+static struct fw_driver tscm_driver = {
+	.driver = {
+		.owner = THIS_MODULE,
+		.name = "snd-firewire-tascam",
+		.bus = &fw_bus_type,
+	},
+	.probe    = snd_tscm_probe,
+	.update   = snd_tscm_update,
+	.remove   = snd_tscm_remove,
+	.id_table = snd_tscm_id_table,
+};
+
+static int __init snd_tscm_init(void)
+{
+	return driver_register(&tscm_driver.driver);
+}
+
+static void __exit snd_tscm_exit(void)
+{
+	driver_unregister(&tscm_driver.driver);
+}
+
+module_init(snd_tscm_init);
+module_exit(snd_tscm_exit);
diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h
new file mode 100644
index 000000000000..2d028d2bd3bd
--- /dev/null
+++ b/sound/firewire/tascam/tascam.h
@@ -0,0 +1,147 @@
+/*
+ * tascam.h - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#ifndef SOUND_TASCAM_H_INCLUDED
+#define SOUND_TASCAM_H_INCLUDED
+
+#include <linux/device.h>
+#include <linux/firewire.h>
+#include <linux/firewire-constants.h>
+#include <linux/module.h>
+#include <linux/mod_devicetable.h>
+#include <linux/mutex.h>
+#include <linux/slab.h>
+#include <linux/compat.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/firewire.h>
+#include <sound/hwdep.h>
+#include <sound/rawmidi.h>
+
+#include "../lib.h"
+#include "../amdtp-stream.h"
+#include "../iso-resources.h"
+
+struct snd_tscm_spec {
+	const char *const name;
+	bool has_adat;
+	bool has_spdif;
+	unsigned int pcm_capture_analog_channels;
+	unsigned int pcm_playback_analog_channels;
+	unsigned int midi_capture_ports;
+	unsigned int midi_playback_ports;
+	bool is_controller;
+};
+
+#define TSCM_MIDI_IN_PORT_MAX	4
+#define TSCM_MIDI_OUT_PORT_MAX	4
+
+struct snd_tscm {
+	struct snd_card *card;
+	struct fw_unit *unit;
+
+	struct mutex mutex;
+	spinlock_t lock;
+
+	const struct snd_tscm_spec *spec;
+
+	struct fw_iso_resources tx_resources;
+	struct fw_iso_resources rx_resources;
+	struct amdtp_stream tx_stream;
+	struct amdtp_stream rx_stream;
+	unsigned int substreams_counter;
+
+	int dev_lock_count;
+	bool dev_lock_changed;
+	wait_queue_head_t hwdep_wait;
+
+	/* For MIDI message incoming transactions. */
+	struct fw_address_handler async_handler;
+	struct snd_rawmidi_substream *tx_midi_substreams[TSCM_MIDI_IN_PORT_MAX];
+
+	/* For MIDI message outgoing transactions. */
+	struct snd_fw_async_midi_port out_ports[TSCM_MIDI_OUT_PORT_MAX];
+	u8 running_status[TSCM_MIDI_OUT_PORT_MAX];
+	bool on_sysex[TSCM_MIDI_OUT_PORT_MAX];
+
+	/* For control messages. */
+	struct snd_firewire_tascam_status *status;
+};
+
+#define TSCM_ADDR_BASE			0xffff00000000ull
+
+#define TSCM_OFFSET_FIRMWARE_REGISTER	0x0000
+#define TSCM_OFFSET_FIRMWARE_FPGA	0x0004
+#define TSCM_OFFSET_FIRMWARE_ARM	0x0008
+#define TSCM_OFFSET_FIRMWARE_HW		0x000c
+
+#define TSCM_OFFSET_ISOC_TX_CH		0x0200
+#define TSCM_OFFSET_UNKNOWN		0x0204
+#define TSCM_OFFSET_START_STREAMING	0x0208
+#define TSCM_OFFSET_ISOC_RX_CH		0x020c
+#define TSCM_OFFSET_ISOC_RX_ON		0x0210	/* Little conviction. */
+#define TSCM_OFFSET_TX_PCM_CHANNELS	0x0214
+#define TSCM_OFFSET_RX_PCM_CHANNELS	0x0218
+#define TSCM_OFFSET_MULTIPLEX_MODE	0x021c
+#define TSCM_OFFSET_ISOC_TX_ON		0x0220
+/* Unknown				0x0224 */
+#define TSCM_OFFSET_CLOCK_STATUS	0x0228
+#define TSCM_OFFSET_SET_OPTION		0x022c
+
+#define TSCM_OFFSET_MIDI_TX_ON		0x0300
+#define TSCM_OFFSET_MIDI_TX_ADDR_HI	0x0304
+#define TSCM_OFFSET_MIDI_TX_ADDR_LO	0x0308
+
+#define TSCM_OFFSET_LED_POWER		0x0404
+
+#define TSCM_OFFSET_MIDI_RX_QUAD	0x4000
+
+enum snd_tscm_clock {
+	SND_TSCM_CLOCK_INTERNAL = 0,
+	SND_TSCM_CLOCK_WORD	= 1,
+	SND_TSCM_CLOCK_SPDIF	= 2,
+	SND_TSCM_CLOCK_ADAT	= 3,
+};
+
+int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit,
+		  enum amdtp_stream_direction dir, unsigned int pcm_channels);
+int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate);
+int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s,
+				      struct snd_pcm_runtime *runtime);
+void amdtp_tscm_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format);
+
+int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate);
+int snd_tscm_stream_get_clock(struct snd_tscm *tscm,
+			      enum snd_tscm_clock *clock);
+int snd_tscm_stream_init_duplex(struct snd_tscm *tscm);
+void snd_tscm_stream_update_duplex(struct snd_tscm *tscm);
+void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm);
+int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate);
+void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm);
+
+void snd_tscm_stream_lock_changed(struct snd_tscm *tscm);
+int snd_tscm_stream_lock_try(struct snd_tscm *tscm);
+void snd_tscm_stream_lock_release(struct snd_tscm *tscm);
+
+int snd_tscm_transaction_register(struct snd_tscm *tscm);
+int snd_tscm_transaction_reregister(struct snd_tscm *tscm);
+void snd_tscm_transaction_unregister(struct snd_tscm *tscm);
+
+void snd_tscm_proc_init(struct snd_tscm *tscm);
+
+int snd_tscm_create_pcm_devices(struct snd_tscm *tscm);
+
+int snd_tscm_create_midi_devices(struct snd_tscm *tscm);
+
+int snd_tscm_create_hwdep_device(struct snd_tscm *tscm);
+
+#endif
diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c
index 33ba77dd32f2..cb89ec7c8147 100644
--- a/sound/hda/ext/hdac_ext_stream.c
+++ b/sound/hda/ext/hdac_ext_stream.c
@@ -227,7 +227,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_link_stream_setup);
 void snd_hdac_ext_link_set_stream_id(struct hdac_ext_link *link,
 				 int stream)
 {
-	snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, (1 << stream), 0);
+	snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, (1 << stream), 1 << stream);
 }
 EXPORT_SYMBOL_GPL(snd_hdac_ext_link_set_stream_id);
 
@@ -385,14 +385,13 @@ void snd_hdac_ext_stream_release(struct hdac_ext_stream *stream, int type)
 		break;
 
 	case HDAC_EXT_STREAM_TYPE_HOST:
-		if (stream->decoupled) {
+		if (stream->decoupled && !stream->link_locked)
 			snd_hdac_ext_stream_decouple(ebus, stream, false);
-			snd_hdac_stream_release(&stream->hstream);
-		}
+		snd_hdac_stream_release(&stream->hstream);
 		break;
 
 	case HDAC_EXT_STREAM_TYPE_LINK:
-		if (stream->decoupled)
+		if (stream->decoupled && !stream->hstream.opened)
 			snd_hdac_ext_stream_decouple(ebus, stream, false);
 		spin_lock_irq(&bus->reg_lock);
 		stream->link_locked = 0;
diff --git a/sound/hda/hda_bus_type.c b/sound/hda/hda_bus_type.c
index 89c2711baaaf..3060e2aee36f 100644
--- a/sound/hda/hda_bus_type.c
+++ b/sound/hda/hda_bus_type.c
@@ -4,6 +4,7 @@
 #include <linux/init.h>
 #include <linux/device.h>
 #include <linux/module.h>
+#include <linux/mod_devicetable.h>
 #include <linux/export.h>
 #include <sound/hdaudio.h>
 
@@ -63,9 +64,21 @@ static int hda_bus_match(struct device *dev, struct device_driver *drv)
 	return 1;
 }
 
+static int hda_uevent(struct device *dev, struct kobj_uevent_env *env)
+{
+	char modalias[32];
+
+	snd_hdac_codec_modalias(dev_to_hdac_dev(dev), modalias,
+				sizeof(modalias));
+	if (add_uevent_var(env, "MODALIAS=%s", modalias))
+		return -ENOMEM;
+	return 0;
+}
+
 struct bus_type snd_hda_bus_type = {
 	.name = "hdaudio",
 	.match = hda_bus_match,
+	.uevent = hda_uevent,
 };
 EXPORT_SYMBOL_GPL(snd_hda_bus_type);
 
diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c
index 27c447e4fe5c..0e81ea89a596 100644
--- a/sound/hda/hdac_bus.c
+++ b/sound/hda/hdac_bus.c
@@ -172,6 +172,15 @@ static void process_unsol_events(struct work_struct *work)
 	}
 }
 
+/**
+ * snd_hdac_bus_add_device - Add a codec to bus
+ * @bus: HDA core bus
+ * @codec: HDA core device to add
+ *
+ * Adds the given codec to the list in the bus.  The caddr_tbl array
+ * and codec_powered bits are updated, as well.
+ * Returns zero if success, or a negative error code.
+ */
 int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec)
 {
 	if (bus->caddr_tbl[codec->addr]) {
@@ -188,6 +197,11 @@ int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec)
 }
 EXPORT_SYMBOL_GPL(snd_hdac_bus_add_device);
 
+/**
+ * snd_hdac_bus_remove_device - Remove a codec from bus
+ * @bus: HDA core bus
+ * @codec: HDA core device to remove
+ */
 void snd_hdac_bus_remove_device(struct hdac_bus *bus,
 				struct hdac_device *codec)
 {
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index db96042a497f..e361024eabb6 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -164,6 +164,43 @@ void snd_hdac_device_unregister(struct hdac_device *codec)
 EXPORT_SYMBOL_GPL(snd_hdac_device_unregister);
 
 /**
+ * snd_hdac_device_set_chip_name - set/update the codec name
+ * @codec: the HDAC device
+ * @name: name string to set
+ *
+ * Returns 0 if the name is set or updated, or a negative error code.
+ */
+int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name)
+{
+	char *newname;
+
+	if (!name)
+		return 0;
+	newname = kstrdup(name, GFP_KERNEL);
+	if (!newname)
+		return -ENOMEM;
+	kfree(codec->chip_name);
+	codec->chip_name = newname;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_device_set_chip_name);
+
+/**
+ * snd_hdac_codec_modalias - give the module alias name
+ * @codec: HDAC device
+ * @buf: string buffer to store
+ * @size: string buffer size
+ *
+ * Returns the size of string, like snprintf(), or a negative error code.
+ */
+int snd_hdac_codec_modalias(struct hdac_device *codec, char *buf, size_t size)
+{
+	return snprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n",
+			codec->vendor_id, codec->revision_id, codec->type);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_codec_modalias);
+
+/**
  * snd_hdac_make_cmd - compose a 32bit command word to be sent to the
  *	HD-audio controller
  * @codec: the codec object
@@ -592,8 +629,10 @@ int snd_hdac_power_down_pm(struct hdac_device *codec)
 EXPORT_SYMBOL_GPL(snd_hdac_power_down_pm);
 #endif
 
-/*
- * Enable/disable the link power for a codec.
+/**
+ * snd_hdac_link_power - Enable/disable the link power for a codec
+ * @codec: the codec object
+ * @bool: enable or disable the link power
  */
 int snd_hdac_link_power(struct hdac_device *codec, bool enable)
 {
@@ -952,3 +991,84 @@ bool snd_hdac_is_supported_format(struct hdac_device *codec, hda_nid_t nid,
 	return true;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_is_supported_format);
+
+static unsigned int codec_read(struct hdac_device *hdac, hda_nid_t nid,
+			int flags, unsigned int verb, unsigned int parm)
+{
+	unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm);
+	unsigned int res;
+
+	if (snd_hdac_exec_verb(hdac, cmd, flags, &res))
+		return -1;
+
+	return res;
+}
+
+static int codec_write(struct hdac_device *hdac, hda_nid_t nid,
+			int flags, unsigned int verb, unsigned int parm)
+{
+	unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm);
+
+	return snd_hdac_exec_verb(hdac, cmd, flags, NULL);
+}
+
+/**
+ * snd_hdac_codec_read - send a command and get the response
+ * @hdac: the HDAC device
+ * @nid: NID to send the command
+ * @flags: optional bit flags
+ * @verb: the verb to send
+ * @parm: the parameter for the verb
+ *
+ * Send a single command and read the corresponding response.
+ *
+ * Returns the obtained response value, or -1 for an error.
+ */
+int snd_hdac_codec_read(struct hdac_device *hdac, hda_nid_t nid,
+			int flags, unsigned int verb, unsigned int parm)
+{
+	return codec_read(hdac, nid, flags, verb, parm);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_codec_read);
+
+/**
+ * snd_hdac_codec_write - send a single command without waiting for response
+ * @hdac: the HDAC device
+ * @nid: NID to send the command
+ * @flags: optional bit flags
+ * @verb: the verb to send
+ * @parm: the parameter for the verb
+ *
+ * Send a single command without waiting for response.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid,
+			int flags, unsigned int verb, unsigned int parm)
+{
+	return codec_write(hdac, nid, flags, verb, parm);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_codec_write);
+
+/**
+ * snd_hdac_check_power_state - check whether the actual power state matches
+ * with the target state
+ *
+ * @hdac: the HDAC device
+ * @nid: NID to send the command
+ * @target_state: target state to check for
+ *
+ * Return true if state matches, false if not
+ */
+bool snd_hdac_check_power_state(struct hdac_device *hdac,
+		hda_nid_t nid, unsigned int target_state)
+{
+	unsigned int state = codec_read(hdac, nid, 0,
+				AC_VERB_GET_POWER_STATE, 0);
+
+	if (state & AC_PWRST_ERROR)
+		return true;
+	state = (state >> 4) & 0x0f;
+	return (state == target_state);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_check_power_state);
diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index 55c3df4458f7..8fef1b8d1fd8 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -23,6 +23,19 @@
 
 static struct i915_audio_component *hdac_acomp;
 
+/**
+ * snd_hdac_set_codec_wakeup - Enable / disable HDMI/DP codec wakeup
+ * @bus: HDA core bus
+ * @enable: enable or disable the wakeup
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function should be called during the chip reset, also called at
+ * resume for updating STATESTS register read.
+ *
+ * Returns zero for success or a negative error code.
+ */
 int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable)
 {
 	struct i915_audio_component *acomp = bus->audio_component;
@@ -45,6 +58,19 @@ int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable)
 }
 EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup);
 
+/**
+ * snd_hdac_display_power - Power up / down the power refcount
+ * @bus: HDA core bus
+ * @enable: power up or down
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function manages a refcount and calls the i915 get_power() and
+ * put_power() ops accordingly, toggling the codec wakeup, too.
+ *
+ * Returns zero for success or a negative error code.
+ */
 int snd_hdac_display_power(struct hdac_bus *bus, bool enable)
 {
 	struct i915_audio_component *acomp = bus->audio_component;
@@ -71,6 +97,16 @@ int snd_hdac_display_power(struct hdac_bus *bus, bool enable)
 }
 EXPORT_SYMBOL_GPL(snd_hdac_display_power);
 
+/**
+ * snd_hdac_get_display_clk - Get CDCLK in kHz
+ * @bus: HDA core bus
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function queries CDCLK value in kHz from the graphics driver and
+ * returns the value.  A negative code is returned in error.
+ */
 int snd_hdac_get_display_clk(struct hdac_bus *bus)
 {
 	struct i915_audio_component *acomp = bus->audio_component;
@@ -134,6 +170,17 @@ static int hdac_component_master_match(struct device *dev, void *data)
 	return !strcmp(dev->driver->name, "i915");
 }
 
+/**
+ * snd_hdac_i915_register_notifier - Register i915 audio component ops
+ * @aops: i915 audio component ops
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function sets the given ops to be called by the i915 graphics driver.
+ *
+ * Returns zero for success or a negative error code.
+ */
 int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *aops)
 {
 	if (WARN_ON(!hdac_acomp))
@@ -144,6 +191,18 @@ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops
 }
 EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier);
 
+/**
+ * snd_hdac_i915_init - Initialize i915 audio component
+ * @bus: HDA core bus
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function initializes and sets up the audio component to communicate
+ * with i915 graphics driver.
+ *
+ * Returns zero for success or a negative error code.
+ */
 int snd_hdac_i915_init(struct hdac_bus *bus)
 {
 	struct component_match *match = NULL;
@@ -187,6 +246,17 @@ out_err:
 }
 EXPORT_SYMBOL_GPL(snd_hdac_i915_init);
 
+/**
+ * snd_hdac_i915_exit - Finalize i915 audio component
+ * @bus: HDA core bus
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function releases the i915 audio component that has been used.
+ *
+ * Returns zero for success or a negative error code.
+ */
 int snd_hdac_i915_exit(struct hdac_bus *bus)
 {
 	struct device *dev = bus->dev;
diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c
index b0ed870ffb88..eb8f7c30cb09 100644
--- a/sound/hda/hdac_regmap.c
+++ b/sound/hda/hdac_regmap.c
@@ -339,6 +339,12 @@ static const struct regmap_config hda_regmap_cfg = {
 	.use_single_rw = true,
 };
 
+/**
+ * snd_hdac_regmap_init - Initialize regmap for HDA register accesses
+ * @codec: the codec object
+ *
+ * Returns zero for success or a negative error code.
+ */
 int snd_hdac_regmap_init(struct hdac_device *codec)
 {
 	struct regmap *regmap;
@@ -352,6 +358,10 @@ int snd_hdac_regmap_init(struct hdac_device *codec)
 }
 EXPORT_SYMBOL_GPL(snd_hdac_regmap_init);
 
+/**
+ * snd_hdac_regmap_init - Release the regmap from HDA codec
+ * @codec: the codec object
+ */
 void snd_hdac_regmap_exit(struct hdac_device *codec)
 {
 	if (codec->regmap) {
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index 8981159813ef..38990a77d7b7 100644
--- a/sound/hda/hdac_stream.c
+++ b/sound/hda/hdac_stream.c
@@ -426,7 +426,8 @@ int snd_hdac_stream_setup_periods(struct hdac_stream *azx_dev)
 }
 EXPORT_SYMBOL_GPL(snd_hdac_stream_setup_periods);
 
-/* snd_hdac_stream_set_params - set stream parameters
+/**
+ * snd_hdac_stream_set_params - set stream parameters
  * @azx_dev: HD-audio core stream for which parameters are to be set
  * @format_val: format value parameter
  *
diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c
index c71142dea98a..42d61bf41969 100644
--- a/sound/hda/hdac_sysfs.c
+++ b/sound/hda/hdac_sysfs.c
@@ -45,6 +45,13 @@ CODEC_ATTR(mfg);
 CODEC_ATTR_STR(vendor_name);
 CODEC_ATTR_STR(chip_name);
 
+static ssize_t modalias_show(struct device *dev, struct device_attribute *attr,
+			     char *buf)
+{
+	return snd_hdac_codec_modalias(dev_to_hdac_dev(dev), buf, 256);
+}
+static DEVICE_ATTR_RO(modalias);
+
 static struct attribute *hdac_dev_attrs[] = {
 	&dev_attr_type.attr,
 	&dev_attr_vendor_id.attr,
@@ -54,6 +61,7 @@ static struct attribute *hdac_dev_attrs[] = {
 	&dev_attr_mfg.attr,
 	&dev_attr_vendor_name.attr,
 	&dev_attr_chip_name.attr,
+	&dev_attr_modalias.attr,
 	NULL
 };
 
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 2a9f4a345171..2706f271a83b 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -1864,7 +1864,7 @@ int snd_cs46xx_pcm_iec958(struct snd_cs46xx *chip, int device)
 	/* global setup */
 	pcm->info_flags = 0;
 	strcpy(pcm->name, "CS46xx - IEC958");
-	chip->pcm_rear = pcm;
+	chip->pcm_iec958 = pcm;
 
 	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
 					      snd_dma_pci_data(chip->pci), 64*1024, 256*1024);
@@ -2528,7 +2528,7 @@ int snd_cs46xx_mixer(struct snd_cs46xx *chip, int spdif_device)
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
 	if (chip->nr_ac97_codecs == 1) {
 		unsigned int id2 = chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]->id & 0xffff;
-		if (id2 == 0x592b || id2 == 0x592d) {
+		if ((id2 & 0xfff0) == 0x5920) {	/* CS4294 and CS4298 */
 			err = snd_ctl_add(card, snd_ctl_new1(&snd_cs46xx_front_dup_ctl, chip));
 			if (err < 0)
 				return err;
@@ -3780,6 +3780,11 @@ static int snd_cs46xx_suspend(struct device *dev)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
 	chip->in_suspend = 1;
 	snd_pcm_suspend_all(chip->pcm);
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+	snd_pcm_suspend_all(chip->pcm_rear);
+	snd_pcm_suspend_all(chip->pcm_center_lfe);
+	snd_pcm_suspend_all(chip->pcm_iec958);
+#endif
 	// chip->ac97_powerdown = snd_cs46xx_codec_read(chip, AC97_POWER_CONTROL);
 	// chip->ac97_general_purpose = snd_cs46xx_codec_read(chip, BA0_AC97_GENERAL_PURPOSE);
 
diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index d5ac25cc7fee..70671ad65d24 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -15,21 +15,22 @@
 #include "hda_local.h"
 
 /*
- * find a matching codec preset
+ * find a matching codec id
  */
 static int hda_codec_match(struct hdac_device *dev, struct hdac_driver *drv)
 {
 	struct hda_codec *codec = container_of(dev, struct hda_codec, core);
 	struct hda_codec_driver *driver =
 		container_of(drv, struct hda_codec_driver, core);
-	const struct hda_codec_preset *preset;
+	const struct hda_device_id *list;
 	/* check probe_id instead of vendor_id if set */
 	u32 id = codec->probe_id ? codec->probe_id : codec->core.vendor_id;
+	u32 rev_id = codec->core.revision_id;
 
-	for (preset = driver->preset; preset->id; preset++) {
-		if (preset->id == id &&
-		    (!preset->rev || preset->rev == codec->core.revision_id)) {
-			codec->preset = preset;
+	for (list = driver->id; list->vendor_id; list++) {
+		if (list->vendor_id == id &&
+		    (!list->rev_id || list->rev_id == rev_id)) {
+			codec->preset = list;
 			return 1;
 		}
 	}
@@ -45,26 +46,45 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev)
 		codec->patch_ops.unsol_event(codec, ev);
 }
 
-/* reset the codec name from the preset */
-static int codec_refresh_name(struct hda_codec *codec, const char *name)
+/**
+ * snd_hda_codec_set_name - set the codec name
+ * @codec: the HDA codec
+ * @name: name string to set
+ */
+int snd_hda_codec_set_name(struct hda_codec *codec, const char *name)
 {
-	if (name) {
-		kfree(codec->core.chip_name);
-		codec->core.chip_name = kstrdup(name, GFP_KERNEL);
+	int err;
+
+	if (!name)
+		return 0;
+	err = snd_hdac_device_set_chip_name(&codec->core, name);
+	if (err < 0)
+		return err;
+
+	/* update the mixer name */
+	if (!*codec->card->mixername ||
+	    codec->bus->mixer_assigned >= codec->core.addr) {
+		snprintf(codec->card->mixername,
+			 sizeof(codec->card->mixername), "%s %s",
+			 codec->core.vendor_name, codec->core.chip_name);
+		codec->bus->mixer_assigned = codec->core.addr;
 	}
-	return codec->core.chip_name ? 0 : -ENOMEM;
+
+	return 0;
 }
+EXPORT_SYMBOL_GPL(snd_hda_codec_set_name);
 
 static int hda_codec_driver_probe(struct device *dev)
 {
 	struct hda_codec *codec = dev_to_hda_codec(dev);
 	struct module *owner = dev->driver->owner;
+	hda_codec_patch_t patch;
 	int err;
 
 	if (WARN_ON(!codec->preset))
 		return -EINVAL;
 
-	err = codec_refresh_name(codec, codec->preset->name);
+	err = snd_hda_codec_set_name(codec, codec->preset->name);
 	if (err < 0)
 		goto error;
 	err = snd_hdac_regmap_init(&codec->core);
@@ -76,9 +96,12 @@ static int hda_codec_driver_probe(struct device *dev)
 		goto error;
 	}
 
-	err = codec->preset->patch(codec);
-	if (err < 0)
-		goto error_module;
+	patch = (hda_codec_patch_t)codec->preset->driver_data;
+	if (patch) {
+		err = patch(codec);
+		if (err < 0)
+			goto error_module;
+	}
 
 	err = snd_hda_codec_build_pcms(codec);
 	if (err < 0)
@@ -155,11 +178,10 @@ static inline bool codec_probed(struct hda_codec *codec)
 static void codec_bind_module(struct hda_codec *codec)
 {
 #ifdef MODULE
-	request_module("snd-hda-codec-id:%08x", codec->core.vendor_id);
-	if (codec_probed(codec))
-		return;
-	request_module("snd-hda-codec-id:%04x*",
-		       (codec->core.vendor_id >> 16) & 0xffff);
+	char modalias[32];
+
+	snd_hdac_codec_modalias(&codec->core, modalias, sizeof(modalias));
+	request_module(modalias);
 	if (codec_probed(codec))
 		return;
 #endif
@@ -251,11 +273,6 @@ int snd_hda_codec_configure(struct hda_codec *codec)
 		}
 	}
 
-	/* audio codec should override the mixer name */
-	if (codec->core.afg || !*codec->card->mixername)
-		snprintf(codec->card->mixername,
-			 sizeof(codec->card->mixername), "%s %s",
-			 codec->core.vendor_name, codec->core.chip_name);
 	return 0;
 
  error:
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a249d5486889..83741887faa1 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -91,50 +91,6 @@ static int codec_exec_verb(struct hdac_device *dev, unsigned int cmd,
 }
 
 /**
- * snd_hda_codec_read - send a command and get the response
- * @codec: the HDA codec
- * @nid: NID to send the command
- * @flags: optional bit flags
- * @verb: the verb to send
- * @parm: the parameter for the verb
- *
- * Send a single command and read the corresponding response.
- *
- * Returns the obtained response value, or -1 for an error.
- */
-unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
-				int flags,
-				unsigned int verb, unsigned int parm)
-{
-	unsigned int cmd = snd_hdac_make_cmd(&codec->core, nid, verb, parm);
-	unsigned int res;
-	if (snd_hdac_exec_verb(&codec->core, cmd, flags, &res))
-		return -1;
-	return res;
-}
-EXPORT_SYMBOL_GPL(snd_hda_codec_read);
-
-/**
- * snd_hda_codec_write - send a single command without waiting for response
- * @codec: the HDA codec
- * @nid: NID to send the command
- * @flags: optional bit flags
- * @verb: the verb to send
- * @parm: the parameter for the verb
- *
- * Send a single command without waiting for response.
- *
- * Returns 0 if successful, or a negative error code.
- */
-int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags,
-			unsigned int verb, unsigned int parm)
-{
-	unsigned int cmd = snd_hdac_make_cmd(&codec->core, nid, verb, parm);
-	return snd_hdac_exec_verb(&codec->core, cmd, flags, NULL);
-}
-EXPORT_SYMBOL_GPL(snd_hda_codec_write);
-
-/**
  * snd_hda_sequence_write - sequence writes
  * @codec: the HDA codec
  * @seq: VERB array to send
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 2970413f18a0..373fcad840ea 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -22,6 +22,7 @@
 #define __SOUND_HDA_CODEC_H
 
 #include <linux/kref.h>
+#include <linux/mod_devicetable.h>
 #include <sound/info.h>
 #include <sound/control.h>
 #include <sound/pcm.h>
@@ -69,6 +70,7 @@ struct hda_bus {
 	unsigned int no_response_fallback:1; /* don't fallback at RIRB error */
 
 	int primary_dig_out_type;	/* primary digital out PCM type */
+	unsigned int mixer_assigned;	/* codec addr for mixer name */
 };
 
 /* from hdac_bus to hda_bus */
@@ -80,19 +82,21 @@ struct hda_bus {
  * Known codecs have the patch to build and set up the controls/PCMs
  * better than the generic parser.
  */
-struct hda_codec_preset {
-	unsigned int id;
-	unsigned int rev;
-	const char *name;
-	int (*patch)(struct hda_codec *codec);
-};
+typedef int (*hda_codec_patch_t)(struct hda_codec *);
 	
 #define HDA_CODEC_ID_GENERIC_HDMI	0x00000101
 #define HDA_CODEC_ID_GENERIC		0x00000201
 
+#define HDA_CODEC_REV_ENTRY(_vid, _rev, _name, _patch) \
+	{ .vendor_id = (_vid), .rev_id = (_rev), .name = (_name), \
+	  .api_version = HDA_DEV_LEGACY, \
+	  .driver_data = (unsigned long)(_patch) }
+#define HDA_CODEC_ENTRY(_vid, _name, _patch) \
+	HDA_CODEC_REV_ENTRY(_vid, 0, _name, _patch)
+
 struct hda_codec_driver {
 	struct hdac_driver core;
-	const struct hda_codec_preset *preset;
+	const struct hda_device_id *id;
 };
 
 int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name,
@@ -183,7 +187,7 @@ struct hda_codec {
 	u32 probe_id; /* overridden id for probing */
 
 	/* detected preset */
-	const struct hda_codec_preset *preset;
+	const struct hda_device_id *preset;
 	const char *modelname;	/* model name for preset */
 
 	/* set by patch */
@@ -297,10 +301,6 @@ struct hda_codec {
 /*
  * constructors
  */
-int snd_hda_bus_new(struct snd_card *card,
-		    const struct hdac_bus_ops *ops,
-		    const struct hdac_io_ops *io_ops,
-		    struct hda_bus **busp);
 int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card,
 		      unsigned int codec_addr, struct hda_codec **codecp);
 int snd_hda_codec_configure(struct hda_codec *codec);
@@ -309,11 +309,21 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec);
 /*
  * low level functions
  */
-unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
+static inline unsigned int
+snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
 				int flags,
-				unsigned int verb, unsigned int parm);
-int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags,
-			unsigned int verb, unsigned int parm);
+				unsigned int verb, unsigned int parm)
+{
+	return snd_hdac_codec_read(&codec->core, nid, flags, verb, parm);
+}
+
+static inline int
+snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags,
+			unsigned int verb, unsigned int parm)
+{
+	return snd_hdac_codec_write(&codec->core, nid, flags, verb, parm);
+}
+
 #define snd_hda_param_read(codec, nid, param) \
 	snd_hdac_read_parm(&(codec)->core, nid, param)
 #define snd_hda_get_sub_nodes(codec, nid, start_nid) \
@@ -453,6 +463,8 @@ void snd_hda_unlock_devices(struct hda_bus *bus);
 void snd_hda_bus_reset(struct hda_bus *bus);
 void snd_hda_bus_reset_codecs(struct hda_bus *bus);
 
+int snd_hda_codec_set_name(struct hda_codec *codec, const char *name);
+
 /*
  * power management
  */
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 944455997fdc..d6b93a20361b 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -1045,6 +1045,7 @@ int azx_bus_init(struct azx *chip, const char *model,
 	mutex_init(&bus->prepare_mutex);
 	bus->pci = chip->pci;
 	bus->modelname = model;
+	bus->mixer_assigned = -1;
 	bus->core.snoop = azx_snoop(chip);
 	if (chip->get_position[0] != azx_get_pos_lpib ||
 	    chip->get_position[1] != azx_get_pos_lpib)
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 24f91114a32c..c6e8a651cea1 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -5877,13 +5877,14 @@ error:
 	return err;
 }
 
-static const struct hda_codec_preset snd_hda_preset_generic[] = {
-	{ .id = HDA_CODEC_ID_GENERIC, .patch = snd_hda_parse_generic_codec },
+static const struct hda_device_id snd_hda_id_generic[] = {
+	HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC, "Generic", snd_hda_parse_generic_codec),
 	{} /* terminator */
 };
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_generic);
 
 static struct hda_codec_driver generic_driver = {
-	.preset = snd_hda_preset_generic,
+	.id = snd_hda_id_generic,
 };
 
 module_hda_codec_driver(generic_driver);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 4a21c2199e02..d0e066e4c985 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -681,12 +681,7 @@ static inline bool
 snd_hda_check_power_state(struct hda_codec *codec, hda_nid_t nid,
 			  unsigned int target_state)
 {
-	unsigned int state = snd_hda_codec_read(codec, nid, 0,
-						AC_VERB_GET_POWER_STATE, 0);
-	if (state & AC_PWRST_ERROR)
-		return true;
-	state = (state >> 4) & 0x0f;
-	return (state == target_state);
+	return snd_hdac_check_power_state(&codec->core, nid, target_state);
 }
 
 unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec,
diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c
index a6e3d9b511ab..64e0d1d81ca5 100644
--- a/sound/pci/hda/hda_sysfs.c
+++ b/sound/pci/hda/hda_sysfs.c
@@ -595,8 +595,7 @@ static void parse_model_mode(char *buf, struct hda_bus *bus,
 static void parse_chip_name_mode(char *buf, struct hda_bus *bus,
 				 struct hda_codec **codecp)
 {
-	kfree((*codecp)->core.chip_name);
-	(*codecp)->core.chip_name = kstrdup(buf, GFP_KERNEL);
+	snd_hda_codec_set_name(*codecp, buf);
 }
 
 #define DEFINE_PARSE_ID_MODE(name) \
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index c033a4ee6547..e0fb8c6d1bc2 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1165,32 +1165,31 @@ static int patch_ad1882(struct hda_codec *codec)
 /*
  * patch entries
  */
-static const struct hda_codec_preset snd_hda_preset_analog[] = {
-	{ .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884 },
-	{ .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 },
-	{ .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884 },
-	{ .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 },
-	{ .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_ad1884 },
-	{ .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },
-	{ .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 },
-	{ .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 },
-	{ .id = 0x11d4882a, .name = "AD1882A", .patch = patch_ad1882 },
-	{ .id = 0x11d4989a, .name = "AD1989A", .patch = patch_ad1988 },
-	{ .id = 0x11d4989b, .name = "AD1989B", .patch = patch_ad1988 },
+static const struct hda_device_id snd_hda_id_analog[] = {
+	HDA_CODEC_ENTRY(0x11d4184a, "AD1884A", patch_ad1884),
+	HDA_CODEC_ENTRY(0x11d41882, "AD1882", patch_ad1882),
+	HDA_CODEC_ENTRY(0x11d41883, "AD1883", patch_ad1884),
+	HDA_CODEC_ENTRY(0x11d41884, "AD1884", patch_ad1884),
+	HDA_CODEC_ENTRY(0x11d4194a, "AD1984A", patch_ad1884),
+	HDA_CODEC_ENTRY(0x11d4194b, "AD1984B", patch_ad1884),
+	HDA_CODEC_ENTRY(0x11d41981, "AD1981", patch_ad1981),
+	HDA_CODEC_ENTRY(0x11d41983, "AD1983", patch_ad1983),
+	HDA_CODEC_ENTRY(0x11d41984, "AD1984", patch_ad1884),
+	HDA_CODEC_ENTRY(0x11d41986, "AD1986A", patch_ad1986a),
+	HDA_CODEC_ENTRY(0x11d41988, "AD1988", patch_ad1988),
+	HDA_CODEC_ENTRY(0x11d4198b, "AD1988B", patch_ad1988),
+	HDA_CODEC_ENTRY(0x11d4882a, "AD1882A", patch_ad1882),
+	HDA_CODEC_ENTRY(0x11d4989a, "AD1989A", patch_ad1988),
+	HDA_CODEC_ENTRY(0x11d4989b, "AD1989B", patch_ad1988),
 	{} /* terminator */
 };
-
-MODULE_ALIAS("snd-hda-codec-id:11d4*");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_analog);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Analog Devices HD-audio codec");
 
 static struct hda_codec_driver analog_driver = {
-	.preset = snd_hda_preset_analog,
+	.id = snd_hda_id_analog,
 };
 
 module_hda_codec_driver(analog_driver);
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index 484bbf4134cd..c2d9ee9cfdc0 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -83,22 +83,19 @@ static int patch_ca0110(struct hda_codec *codec)
 /*
  * patch entries
  */
-static const struct hda_codec_preset snd_hda_preset_ca0110[] = {
-	{ .id = 0x1102000a, .name = "CA0110-IBG", .patch = patch_ca0110 },
-	{ .id = 0x1102000b, .name = "CA0110-IBG", .patch = patch_ca0110 },
-	{ .id = 0x1102000d, .name = "SB0880 X-Fi", .patch = patch_ca0110 },
+static const struct hda_device_id snd_hda_id_ca0110[] = {
+	HDA_CODEC_ENTRY(0x1102000a, "CA0110-IBG", patch_ca0110),
+	HDA_CODEC_ENTRY(0x1102000b, "CA0110-IBG", patch_ca0110),
+	HDA_CODEC_ENTRY(0x1102000d, "SB0880 X-Fi", patch_ca0110),
 	{} /* terminator */
 };
-
-MODULE_ALIAS("snd-hda-codec-id:1102000a");
-MODULE_ALIAS("snd-hda-codec-id:1102000b");
-MODULE_ALIAS("snd-hda-codec-id:1102000d");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_ca0110);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Creative CA0110-IBG HD-audio codec");
 
 static struct hda_codec_driver ca0110_driver = {
-	.preset = snd_hda_preset_ca0110,
+	.id = snd_hda_id_ca0110,
 };
 
 module_hda_codec_driver(ca0110_driver);
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 186792fe226e..f8a12ca477f1 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -2673,13 +2673,13 @@ static bool dspload_wait_loaded(struct hda_codec *codec)
 
 	do {
 		if (dspload_is_loaded(codec)) {
-			pr_info("ca0132 DOWNLOAD OK :-) DSP IS RUNNING.\n");
+			codec_info(codec, "ca0132 DSP downloaded and running\n");
 			return true;
 		}
 		msleep(20);
 	} while (time_before(jiffies, timeout));
 
-	pr_err("ca0132 DOWNLOAD FAILED!!! DSP IS NOT RUNNING.\n");
+	codec_err(codec, "ca0132 failed to download DSP\n");
 	return false;
 }
 
@@ -4375,7 +4375,7 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec)
 
 	dsp_os_image = (struct dsp_image_seg *)(fw_entry->data);
 	if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) {
-		pr_err("ca0132 dspload_image failed.\n");
+		codec_err(codec, "ca0132 DSP load image failed\n");
 		goto exit_download;
 	}
 
@@ -4778,18 +4778,17 @@ static int patch_ca0132(struct hda_codec *codec)
 /*
  * patch entries
  */
-static struct hda_codec_preset snd_hda_preset_ca0132[] = {
-	{ .id = 0x11020011, .name = "CA0132",     .patch = patch_ca0132 },
+static struct hda_device_id snd_hda_id_ca0132[] = {
+	HDA_CODEC_ENTRY(0x11020011, "CA0132", patch_ca0132),
 	{} /* terminator */
 };
-
-MODULE_ALIAS("snd-hda-codec-id:11020011");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_ca0132);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Creative Sound Core3D codec");
 
 static struct hda_codec_driver ca0132_driver = {
-	.preset = snd_hda_preset_ca0132,
+	.id = snd_hda_id_ca0132,
 };
 
 module_hda_codec_driver(ca0132_driver);
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 85813de26da8..a12ae8ac0914 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -570,6 +570,7 @@ static struct cs_spec *cs_alloc_spec(struct hda_codec *codec, int vendor_nid)
 		return NULL;
 	codec->spec = spec;
 	spec->vendor_nid = vendor_nid;
+	codec->power_save_node = 1;
 	snd_hda_gen_spec_init(&spec->gen);
 
 	return spec;
@@ -1200,26 +1201,21 @@ static int patch_cs4213(struct hda_codec *codec)
 /*
  * patch entries
  */
-static const struct hda_codec_preset snd_hda_preset_cirrus[] = {
-	{ .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x },
-	{ .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x },
-	{ .id = 0x10134208, .name = "CS4208", .patch = patch_cs4208 },
-	{ .id = 0x10134210, .name = "CS4210", .patch = patch_cs4210 },
-	{ .id = 0x10134213, .name = "CS4213", .patch = patch_cs4213 },
+static const struct hda_device_id snd_hda_id_cirrus[] = {
+	HDA_CODEC_ENTRY(0x10134206, "CS4206", patch_cs420x),
+	HDA_CODEC_ENTRY(0x10134207, "CS4207", patch_cs420x),
+	HDA_CODEC_ENTRY(0x10134208, "CS4208", patch_cs4208),
+	HDA_CODEC_ENTRY(0x10134210, "CS4210", patch_cs4210),
+	HDA_CODEC_ENTRY(0x10134213, "CS4213", patch_cs4213),
 	{} /* terminator */
 };
-
-MODULE_ALIAS("snd-hda-codec-id:10134206");
-MODULE_ALIAS("snd-hda-codec-id:10134207");
-MODULE_ALIAS("snd-hda-codec-id:10134208");
-MODULE_ALIAS("snd-hda-codec-id:10134210");
-MODULE_ALIAS("snd-hda-codec-id:10134213");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_cirrus);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Cirrus Logic HD-audio codec");
 
 static struct hda_codec_driver cirrus_driver = {
-	.preset = snd_hda_preset_cirrus,
+	.id = snd_hda_id_cirrus,
 };
 
 module_hda_codec_driver(cirrus_driver);
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index f5ed078710f8..1b2195dd2b26 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -123,22 +123,19 @@ static int patch_cmi8888(struct hda_codec *codec)
 /*
  * patch entries
  */
-static const struct hda_codec_preset snd_hda_preset_cmedia[] = {
-	{ .id = 0x13f68888, .name = "CMI8888", .patch = patch_cmi8888 },
-	{ .id = 0x13f69880, .name = "CMI9880", .patch = patch_cmi9880 },
- 	{ .id = 0x434d4980, .name = "CMI9880", .patch = patch_cmi9880 },
+static const struct hda_device_id snd_hda_id_cmedia[] = {
+	HDA_CODEC_ENTRY(0x13f68888, "CMI8888", patch_cmi8888),
+	HDA_CODEC_ENTRY(0x13f69880, "CMI9880", patch_cmi9880),
+	HDA_CODEC_ENTRY(0x434d4980, "CMI9880", patch_cmi9880),
 	{} /* terminator */
 };
-
-MODULE_ALIAS("snd-hda-codec-id:13f68888");
-MODULE_ALIAS("snd-hda-codec-id:13f69880");
-MODULE_ALIAS("snd-hda-codec-id:434d4980");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_cmedia);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("C-Media HD-audio codec");
 
 static struct hda_codec_driver cmedia_driver = {
-	.preset = snd_hda_preset_cmedia,
+	.id = snd_hda_id_cmedia,
 };
 
 module_hda_codec_driver(cmedia_driver);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 2f0ec7c45fc7..c8b8ef5246a6 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -954,100 +954,44 @@ static int patch_conexant_auto(struct hda_codec *codec)
 /*
  */
 
-static const struct hda_codec_preset snd_hda_preset_conexant[] = {
-	{ .id = 0x14f15045, .name = "CX20549 (Venice)",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f15047, .name = "CX20551 (Waikiki)",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f15051, .name = "CX20561 (Hermosa)",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f15066, .name = "CX20582 (Pebble)",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f15067, .name = "CX20583 (Pebble HSF)",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f15068, .name = "CX20584",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f15069, .name = "CX20585",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f1506c, .name = "CX20588",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f1506e, .name = "CX20590",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f15097, .name = "CX20631",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f15098, .name = "CX20632",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f150a1, .name = "CX20641",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f150a2, .name = "CX20642",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f150ab, .name = "CX20651",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f150ac, .name = "CX20652",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f150b8, .name = "CX20664",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f150b9, .name = "CX20665",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f150f1, .name = "CX20721",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f150f2, .name = "CX20722",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f150f3, .name = "CX20723",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f150f4, .name = "CX20724",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f1510f, .name = "CX20751/2",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f15110, .name = "CX20751/2",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f15111, .name = "CX20753/4",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f15113, .name = "CX20755",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f15114, .name = "CX20756",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f15115, .name = "CX20757",
-	  .patch = patch_conexant_auto },
-	{ .id = 0x14f151d7, .name = "CX20952",
-	  .patch = patch_conexant_auto },
+static const struct hda_device_id snd_hda_id_conexant[] = {
+	HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f15066, "CX20582 (Pebble)", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f15067, "CX20583 (Pebble HSF)", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f15068, "CX20584", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f15069, "CX20585", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f1506c, "CX20588", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f1506e, "CX20590", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f15097, "CX20631", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f15098, "CX20632", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f150a1, "CX20641", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f150a2, "CX20642", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f150ab, "CX20651", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f150ac, "CX20652", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f150b8, "CX20664", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f150b9, "CX20665", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f150f1, "CX20721", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f150f2, "CX20722", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f150f3, "CX20723", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f150f4, "CX20724", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f1510f, "CX20751/2", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f15110, "CX20751/2", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f15111, "CX20753/4", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f15113, "CX20755", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f15114, "CX20756", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f15115, "CX20757", patch_conexant_auto),
+	HDA_CODEC_ENTRY(0x14f151d7, "CX20952", patch_conexant_auto),
 	{} /* terminator */
 };
-
-MODULE_ALIAS("snd-hda-codec-id:14f15045");
-MODULE_ALIAS("snd-hda-codec-id:14f15047");
-MODULE_ALIAS("snd-hda-codec-id:14f15051");
-MODULE_ALIAS("snd-hda-codec-id:14f15066");
-MODULE_ALIAS("snd-hda-codec-id:14f15067");
-MODULE_ALIAS("snd-hda-codec-id:14f15068");
-MODULE_ALIAS("snd-hda-codec-id:14f15069");
-MODULE_ALIAS("snd-hda-codec-id:14f1506c");
-MODULE_ALIAS("snd-hda-codec-id:14f1506e");
-MODULE_ALIAS("snd-hda-codec-id:14f15097");
-MODULE_ALIAS("snd-hda-codec-id:14f15098");
-MODULE_ALIAS("snd-hda-codec-id:14f150a1");
-MODULE_ALIAS("snd-hda-codec-id:14f150a2");
-MODULE_ALIAS("snd-hda-codec-id:14f150ab");
-MODULE_ALIAS("snd-hda-codec-id:14f150ac");
-MODULE_ALIAS("snd-hda-codec-id:14f150b8");
-MODULE_ALIAS("snd-hda-codec-id:14f150b9");
-MODULE_ALIAS("snd-hda-codec-id:14f150f1");
-MODULE_ALIAS("snd-hda-codec-id:14f150f2");
-MODULE_ALIAS("snd-hda-codec-id:14f150f3");
-MODULE_ALIAS("snd-hda-codec-id:14f150f4");
-MODULE_ALIAS("snd-hda-codec-id:14f1510f");
-MODULE_ALIAS("snd-hda-codec-id:14f15110");
-MODULE_ALIAS("snd-hda-codec-id:14f15111");
-MODULE_ALIAS("snd-hda-codec-id:14f15113");
-MODULE_ALIAS("snd-hda-codec-id:14f15114");
-MODULE_ALIAS("snd-hda-codec-id:14f15115");
-MODULE_ALIAS("snd-hda-codec-id:14f151d7");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_conexant);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Conexant HD-audio codec");
 
 static struct hda_codec_driver conexant_driver = {
-	.preset = snd_hda_preset_conexant,
+	.id = snd_hda_id_conexant,
 };
 
 module_hda_codec_driver(conexant_driver);
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index acbfbe087ee8..f503a883bef3 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1775,6 +1775,16 @@ static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
 	return non_pcm;
 }
 
+/* There is a fixed mapping between audio pin node and display port
+ * on current Intel platforms:
+ * Pin Widget 5 - PORT B (port = 1 in i915 driver)
+ * Pin Widget 6 - PORT C (port = 2 in i915 driver)
+ * Pin Widget 7 - PORT D (port = 3 in i915 driver)
+ */
+static int intel_pin2port(hda_nid_t pin_nid)
+{
+	return pin_nid - 4;
+}
 
 /*
  * HDMI callbacks
@@ -1791,6 +1801,8 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 	int pin_idx = hinfo_to_pin_index(codec, hinfo);
 	struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
 	hda_nid_t pin_nid = per_pin->pin_nid;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct i915_audio_component *acomp = codec->bus->core.audio_component;
 	bool non_pcm;
 	int pinctl;
 
@@ -1807,6 +1819,13 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 		intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx);
 	}
 
+	/* Call sync_audio_rate to set the N/CTS/M manually if necessary */
+	/* Todo: add DP1.2 MST audio support later */
+	if (acomp && acomp->ops && acomp->ops->sync_audio_rate)
+		acomp->ops->sync_audio_rate(acomp->dev,
+				intel_pin2port(pin_nid),
+				runtime->rate);
+
 	non_pcm = check_non_pcm_per_cvt(codec, cvt_nid);
 	mutex_lock(&per_pin->lock);
 	per_pin->channels = substream->runtime->channels;
@@ -2561,7 +2580,7 @@ static int simple_playback_pcm_open(struct hda_pcm_stream *hinfo,
 	struct hdmi_spec *spec = codec->spec;
 	struct snd_pcm_hw_constraint_list *hw_constraints_channels = NULL;
 
-	switch (codec->preset->id) {
+	switch (codec->preset->vendor_id) {
 	case 0x10de0002:
 	case 0x10de0003:
 	case 0x10de0005:
@@ -2879,7 +2898,7 @@ static int nvhdmi_7x_8ch_build_controls(struct hda_codec *codec)
 				     snd_pcm_alt_chmaps, 8, 0, &chmap);
 	if (err < 0)
 		return err;
-	switch (codec->preset->id) {
+	switch (codec->preset->vendor_id) {
 	case 0x10de0002:
 	case 0x10de0003:
 	case 0x10de0005:
@@ -3487,138 +3506,77 @@ static int patch_via_hdmi(struct hda_codec *codec)
 /*
  * patch entries
  */
-static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
-{ .id = 0x1002793c, .name = "RS600 HDMI",	.patch = patch_atihdmi },
-{ .id = 0x10027919, .name = "RS600 HDMI",	.patch = patch_atihdmi },
-{ .id = 0x1002791a, .name = "RS690/780 HDMI",	.patch = patch_atihdmi },
-{ .id = 0x1002aa01, .name = "R6xx HDMI",	.patch = patch_atihdmi },
-{ .id = 0x10951390, .name = "SiI1390 HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x10951392, .name = "SiI1392 HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x17e80047, .name = "Chrontel HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x10de0002, .name = "MCP77/78 HDMI",	.patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de0003, .name = "MCP77/78 HDMI",	.patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de0005, .name = "MCP77/78 HDMI",	.patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de0006, .name = "MCP77/78 HDMI",	.patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de0007, .name = "MCP79/7A HDMI",	.patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de000a, .name = "GPU 0a HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de000b, .name = "GPU 0b HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de000c, .name = "MCP89 HDMI",	.patch = patch_nvhdmi },
-{ .id = 0x10de000d, .name = "GPU 0d HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0010, .name = "GPU 10 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0011, .name = "GPU 11 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0012, .name = "GPU 12 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0013, .name = "GPU 13 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0014, .name = "GPU 14 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0015, .name = "GPU 15 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP",	.patch = patch_nvhdmi },
+static const struct hda_device_id snd_hda_id_hdmi[] = {
+HDA_CODEC_ENTRY(0x1002793c, "RS600 HDMI",	patch_atihdmi),
+HDA_CODEC_ENTRY(0x10027919, "RS600 HDMI",	patch_atihdmi),
+HDA_CODEC_ENTRY(0x1002791a, "RS690/780 HDMI",	patch_atihdmi),
+HDA_CODEC_ENTRY(0x1002aa01, "R6xx HDMI",	patch_atihdmi),
+HDA_CODEC_ENTRY(0x10951390, "SiI1390 HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x10951392, "SiI1392 HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x17e80047, "Chrontel HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x10de0002, "MCP77/78 HDMI",	patch_nvhdmi_8ch_7x),
+HDA_CODEC_ENTRY(0x10de0003, "MCP77/78 HDMI",	patch_nvhdmi_8ch_7x),
+HDA_CODEC_ENTRY(0x10de0005, "MCP77/78 HDMI",	patch_nvhdmi_8ch_7x),
+HDA_CODEC_ENTRY(0x10de0006, "MCP77/78 HDMI",	patch_nvhdmi_8ch_7x),
+HDA_CODEC_ENTRY(0x10de0007, "MCP79/7A HDMI",	patch_nvhdmi_8ch_7x),
+HDA_CODEC_ENTRY(0x10de000a, "GPU 0a HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de000b, "GPU 0b HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de000c, "MCP89 HDMI",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de000d, "GPU 0d HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0010, "GPU 10 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0011, "GPU 11 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0012, "GPU 12 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0013, "GPU 13 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0014, "GPU 14 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0015, "GPU 15 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0016, "GPU 16 HDMI/DP",	patch_nvhdmi),
 /* 17 is known to be absent */
-{ .id = 0x10de0018, .name = "GPU 18 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0019, .name = "GPU 19 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de001a, .name = "GPU 1a HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de001b, .name = "GPU 1b HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de001c, .name = "GPU 1c HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0020, .name = "Tegra30 HDMI",	.patch = patch_tegra_hdmi },
-{ .id = 0x10de0022, .name = "Tegra114 HDMI",	.patch = patch_tegra_hdmi },
-{ .id = 0x10de0028, .name = "Tegra124 HDMI",	.patch = patch_tegra_hdmi },
-{ .id = 0x10de0029, .name = "Tegra210 HDMI/DP",	.patch = patch_tegra_hdmi },
-{ .id = 0x10de0040, .name = "GPU 40 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0041, .name = "GPU 41 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0042, .name = "GPU 42 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0043, .name = "GPU 43 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0044, .name = "GPU 44 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0067, .name = "MCP67 HDMI",	.patch = patch_nvhdmi_2ch },
-{ .id = 0x10de0070, .name = "GPU 70 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0071, .name = "GPU 71 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de0072, .name = "GPU 72 HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de007d, .name = "GPU 7d HDMI/DP",	.patch = patch_nvhdmi },
-{ .id = 0x10de8001, .name = "MCP73 HDMI",	.patch = patch_nvhdmi_2ch },
-{ .id = 0x11069f80, .name = "VX900 HDMI/DP",	.patch = patch_via_hdmi },
-{ .id = 0x11069f81, .name = "VX900 HDMI/DP",	.patch = patch_via_hdmi },
-{ .id = 0x11069f84, .name = "VX11 HDMI/DP",	.patch = patch_generic_hdmi },
-{ .id = 0x11069f85, .name = "VX11 HDMI/DP",	.patch = patch_generic_hdmi },
-{ .id = 0x80860054, .name = "IbexPeak HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x80862801, .name = "Bearlake HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x80862802, .name = "Cantiga HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x80862803, .name = "Eaglelake HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x80862804, .name = "IbexPeak HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x80862805, .name = "CougarPoint HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862807, .name = "Haswell HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x80862808, .name = "Broadwell HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x80862809, .name = "Skylake HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x8086280a, .name = "Broxton HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x80862880, .name = "CedarTrail HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x80862882, .name = "Valleyview2 HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x80862883, .name = "Braswell HDMI",	.patch = patch_generic_hdmi },
-{ .id = 0x808629fb, .name = "Crestline HDMI",	.patch = patch_generic_hdmi },
+HDA_CODEC_ENTRY(0x10de0018, "GPU 18 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0019, "GPU 19 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de001a, "GPU 1a HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de001b, "GPU 1b HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de001c, "GPU 1c HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0020, "Tegra30 HDMI",	patch_tegra_hdmi),
+HDA_CODEC_ENTRY(0x10de0022, "Tegra114 HDMI",	patch_tegra_hdmi),
+HDA_CODEC_ENTRY(0x10de0028, "Tegra124 HDMI",	patch_tegra_hdmi),
+HDA_CODEC_ENTRY(0x10de0029, "Tegra210 HDMI/DP",	patch_tegra_hdmi),
+HDA_CODEC_ENTRY(0x10de0040, "GPU 40 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0041, "GPU 41 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0042, "GPU 42 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0043, "GPU 43 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0044, "GPU 44 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0051, "GPU 51 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0060, "GPU 60 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0067, "MCP67 HDMI",	patch_nvhdmi_2ch),
+HDA_CODEC_ENTRY(0x10de0070, "GPU 70 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0071, "GPU 71 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0072, "GPU 72 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de007d, "GPU 7d HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI",	patch_nvhdmi_2ch),
+HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP",	patch_via_hdmi),
+HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP",	patch_via_hdmi),
+HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x11069f85, "VX11 HDMI/DP",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862807, "Haswell HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x808629fb, "Crestline HDMI",	patch_generic_hdmi),
 /* special ID for generic HDMI */
-{ .id = HDA_CODEC_ID_GENERIC_HDMI, .patch = patch_generic_hdmi },
+HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC_HDMI, "Generic HDMI", patch_generic_hdmi),
 {} /* terminator */
 };
-
-MODULE_ALIAS("snd-hda-codec-id:1002793c");
-MODULE_ALIAS("snd-hda-codec-id:10027919");
-MODULE_ALIAS("snd-hda-codec-id:1002791a");
-MODULE_ALIAS("snd-hda-codec-id:1002aa01");
-MODULE_ALIAS("snd-hda-codec-id:10951390");
-MODULE_ALIAS("snd-hda-codec-id:10951392");
-MODULE_ALIAS("snd-hda-codec-id:10de0002");
-MODULE_ALIAS("snd-hda-codec-id:10de0003");
-MODULE_ALIAS("snd-hda-codec-id:10de0005");
-MODULE_ALIAS("snd-hda-codec-id:10de0006");
-MODULE_ALIAS("snd-hda-codec-id:10de0007");
-MODULE_ALIAS("snd-hda-codec-id:10de000a");
-MODULE_ALIAS("snd-hda-codec-id:10de000b");
-MODULE_ALIAS("snd-hda-codec-id:10de000c");
-MODULE_ALIAS("snd-hda-codec-id:10de000d");
-MODULE_ALIAS("snd-hda-codec-id:10de0010");
-MODULE_ALIAS("snd-hda-codec-id:10de0011");
-MODULE_ALIAS("snd-hda-codec-id:10de0012");
-MODULE_ALIAS("snd-hda-codec-id:10de0013");
-MODULE_ALIAS("snd-hda-codec-id:10de0014");
-MODULE_ALIAS("snd-hda-codec-id:10de0015");
-MODULE_ALIAS("snd-hda-codec-id:10de0016");
-MODULE_ALIAS("snd-hda-codec-id:10de0018");
-MODULE_ALIAS("snd-hda-codec-id:10de0019");
-MODULE_ALIAS("snd-hda-codec-id:10de001a");
-MODULE_ALIAS("snd-hda-codec-id:10de001b");
-MODULE_ALIAS("snd-hda-codec-id:10de001c");
-MODULE_ALIAS("snd-hda-codec-id:10de0028");
-MODULE_ALIAS("snd-hda-codec-id:10de0040");
-MODULE_ALIAS("snd-hda-codec-id:10de0041");
-MODULE_ALIAS("snd-hda-codec-id:10de0042");
-MODULE_ALIAS("snd-hda-codec-id:10de0043");
-MODULE_ALIAS("snd-hda-codec-id:10de0044");
-MODULE_ALIAS("snd-hda-codec-id:10de0051");
-MODULE_ALIAS("snd-hda-codec-id:10de0060");
-MODULE_ALIAS("snd-hda-codec-id:10de0067");
-MODULE_ALIAS("snd-hda-codec-id:10de0070");
-MODULE_ALIAS("snd-hda-codec-id:10de0071");
-MODULE_ALIAS("snd-hda-codec-id:10de0072");
-MODULE_ALIAS("snd-hda-codec-id:10de007d");
-MODULE_ALIAS("snd-hda-codec-id:10de8001");
-MODULE_ALIAS("snd-hda-codec-id:11069f80");
-MODULE_ALIAS("snd-hda-codec-id:11069f81");
-MODULE_ALIAS("snd-hda-codec-id:11069f84");
-MODULE_ALIAS("snd-hda-codec-id:11069f85");
-MODULE_ALIAS("snd-hda-codec-id:17e80047");
-MODULE_ALIAS("snd-hda-codec-id:80860054");
-MODULE_ALIAS("snd-hda-codec-id:80862801");
-MODULE_ALIAS("snd-hda-codec-id:80862802");
-MODULE_ALIAS("snd-hda-codec-id:80862803");
-MODULE_ALIAS("snd-hda-codec-id:80862804");
-MODULE_ALIAS("snd-hda-codec-id:80862805");
-MODULE_ALIAS("snd-hda-codec-id:80862806");
-MODULE_ALIAS("snd-hda-codec-id:80862807");
-MODULE_ALIAS("snd-hda-codec-id:80862808");
-MODULE_ALIAS("snd-hda-codec-id:80862809");
-MODULE_ALIAS("snd-hda-codec-id:8086280a");
-MODULE_ALIAS("snd-hda-codec-id:80862880");
-MODULE_ALIAS("snd-hda-codec-id:80862882");
-MODULE_ALIAS("snd-hda-codec-id:80862883");
-MODULE_ALIAS("snd-hda-codec-id:808629fb");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_hdmi);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("HDMI HD-audio codec");
@@ -3627,7 +3585,7 @@ MODULE_ALIAS("snd-hda-codec-nvhdmi");
 MODULE_ALIAS("snd-hda-codec-atihdmi");
 
 static struct hda_codec_driver hdmi_driver = {
-	.preset = snd_hda_preset_hdmi,
+	.id = snd_hda_id_hdmi,
 };
 
 module_hda_codec_driver(hdmi_driver);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 720a9fb32e20..2f7b065f9ac4 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -822,17 +822,7 @@ static const struct hda_codec_ops alc_patch_ops = {
 };
 
 
-/* replace the codec chip_name with the given string */
-static int alc_codec_rename(struct hda_codec *codec, const char *name)
-{
-	kfree(codec->core.chip_name);
-	codec->core.chip_name = kstrdup(name, GFP_KERNEL);
-	if (!codec->core.chip_name) {
-		alc_free(codec);
-		return -ENOMEM;
-	}
-	return 0;
-}
+#define alc_codec_rename(codec, name) snd_hda_codec_set_name(codec, name)
 
 /*
  * Rename codecs appropriately from COEF value or subvendor id
@@ -6640,78 +6630,70 @@ static int patch_alc680(struct hda_codec *codec)
 /*
  * patch entries
  */
-static const struct hda_codec_preset snd_hda_preset_realtek[] = {
-	{ .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 },
-	{ .id = 0x10ec0231, .name = "ALC231", .patch = patch_alc269 },
-	{ .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 },
-	{ .id = 0x10ec0235, .name = "ALC233", .patch = patch_alc269 },
-	{ .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 },
-	{ .id = 0x10ec0256, .name = "ALC256", .patch = patch_alc269 },
-	{ .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
-	{ .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
-	{ .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 },
-	{ .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 },
-	{ .id = 0x10ec0269, .name = "ALC269", .patch = patch_alc269 },
-	{ .id = 0x10ec0270, .name = "ALC270", .patch = patch_alc269 },
-	{ .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 },
-	{ .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 },
-	{ .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 },
-	{ .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 },
-	{ .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 },
-	{ .id = 0x10ec0283, .name = "ALC283", .patch = patch_alc269 },
-	{ .id = 0x10ec0284, .name = "ALC284", .patch = patch_alc269 },
-	{ .id = 0x10ec0285, .name = "ALC285", .patch = patch_alc269 },
-	{ .id = 0x10ec0286, .name = "ALC286", .patch = patch_alc269 },
-	{ .id = 0x10ec0288, .name = "ALC288", .patch = patch_alc269 },
-	{ .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 },
-	{ .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 },
-	{ .id = 0x10ec0293, .name = "ALC293", .patch = patch_alc269 },
-	{ .id = 0x10ec0298, .name = "ALC298", .patch = patch_alc269 },
-	{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
-	  .patch = patch_alc861 },
-	{ .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
-	{ .id = 0x10ec0861, .name = "ALC861", .patch = patch_alc861 },
-	{ .id = 0x10ec0862, .name = "ALC861-VD", .patch = patch_alc861vd },
-	{ .id = 0x10ec0662, .rev = 0x100002, .name = "ALC662 rev2",
-	  .patch = patch_alc882 },
-	{ .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1",
-	  .patch = patch_alc662 },
-	{ .id = 0x10ec0662, .rev = 0x100300, .name = "ALC662 rev3",
-	  .patch = patch_alc662 },
-	{ .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 },
-	{ .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 },
-	{ .id = 0x10ec0667, .name = "ALC667", .patch = patch_alc662 },
-	{ .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 },
-	{ .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 },
-	{ .id = 0x10ec0671, .name = "ALC671", .patch = patch_alc662 },
-	{ .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 },
-	{ .id = 0x10ec0867, .name = "ALC891", .patch = patch_alc882 },
-	{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
-	{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
-	{ .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 },
-	{ .id = 0x10ec0885, .rev = 0x100101, .name = "ALC889A",
-	  .patch = patch_alc882 },
-	{ .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
-	  .patch = patch_alc882 },
-	{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
-	{ .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 },
-	{ .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
-	  .patch = patch_alc882 },
-	{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc882 },
-	{ .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 },
-	{ .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 },
-	{ .id = 0x10ec0899, .name = "ALC898", .patch = patch_alc882 },
-	{ .id = 0x10ec0900, .name = "ALC1150", .patch = patch_alc882 },
+static const struct hda_device_id snd_hda_id_realtek[] = {
+	HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0235, "ALC233", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0260, "ALC260", patch_alc260),
+	HDA_CODEC_ENTRY(0x10ec0262, "ALC262", patch_alc262),
+	HDA_CODEC_ENTRY(0x10ec0267, "ALC267", patch_alc268),
+	HDA_CODEC_ENTRY(0x10ec0268, "ALC268", patch_alc268),
+	HDA_CODEC_ENTRY(0x10ec0269, "ALC269", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0270, "ALC270", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0272, "ALC272", patch_alc662),
+	HDA_CODEC_ENTRY(0x10ec0275, "ALC275", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0276, "ALC276", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0280, "ALC280", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0282, "ALC282", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0283, "ALC283", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0284, "ALC284", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0285, "ALC285", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0286, "ALC286", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0288, "ALC288", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0290, "ALC290", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0292, "ALC292", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0293, "ALC293", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269),
+	HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861),
+	HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd),
+	HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861),
+	HDA_CODEC_ENTRY(0x10ec0862, "ALC861-VD", patch_alc861vd),
+	HDA_CODEC_REV_ENTRY(0x10ec0662, 0x100002, "ALC662 rev2", patch_alc882),
+	HDA_CODEC_REV_ENTRY(0x10ec0662, 0x100101, "ALC662 rev1", patch_alc662),
+	HDA_CODEC_REV_ENTRY(0x10ec0662, 0x100300, "ALC662 rev3", patch_alc662),
+	HDA_CODEC_ENTRY(0x10ec0663, "ALC663", patch_alc662),
+	HDA_CODEC_ENTRY(0x10ec0665, "ALC665", patch_alc662),
+	HDA_CODEC_ENTRY(0x10ec0667, "ALC667", patch_alc662),
+	HDA_CODEC_ENTRY(0x10ec0668, "ALC668", patch_alc662),
+	HDA_CODEC_ENTRY(0x10ec0670, "ALC670", patch_alc662),
+	HDA_CODEC_ENTRY(0x10ec0671, "ALC671", patch_alc662),
+	HDA_CODEC_ENTRY(0x10ec0680, "ALC680", patch_alc680),
+	HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc882),
+	HDA_CODEC_ENTRY(0x10ec0880, "ALC880", patch_alc880),
+	HDA_CODEC_ENTRY(0x10ec0882, "ALC882", patch_alc882),
+	HDA_CODEC_ENTRY(0x10ec0883, "ALC883", patch_alc882),
+	HDA_CODEC_REV_ENTRY(0x10ec0885, 0x100101, "ALC889A", patch_alc882),
+	HDA_CODEC_REV_ENTRY(0x10ec0885, 0x100103, "ALC889A", patch_alc882),
+	HDA_CODEC_ENTRY(0x10ec0885, "ALC885", patch_alc882),
+	HDA_CODEC_ENTRY(0x10ec0887, "ALC887", patch_alc882),
+	HDA_CODEC_REV_ENTRY(0x10ec0888, 0x100101, "ALC1200", patch_alc882),
+	HDA_CODEC_ENTRY(0x10ec0888, "ALC888", patch_alc882),
+	HDA_CODEC_ENTRY(0x10ec0889, "ALC889", patch_alc882),
+	HDA_CODEC_ENTRY(0x10ec0892, "ALC892", patch_alc662),
+	HDA_CODEC_ENTRY(0x10ec0899, "ALC898", patch_alc882),
+	HDA_CODEC_ENTRY(0x10ec0900, "ALC1150", patch_alc882),
 	{} /* terminator */
 };
-
-MODULE_ALIAS("snd-hda-codec-id:10ec*");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Realtek HD-audio codec");
 
 static struct hda_codec_driver realtek_driver = {
-	.preset = snd_hda_preset_realtek,
+	.id = snd_hda_id_realtek,
 };
 
 module_hda_codec_driver(realtek_driver);
diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c
index 5104bebb2286..ffda38c45509 100644
--- a/sound/pci/hda/patch_si3054.c
+++ b/sound/pci/hda/patch_si3054.c
@@ -289,41 +289,30 @@ static int patch_si3054(struct hda_codec *codec)
 /*
  * patch entries
  */
-static const struct hda_codec_preset snd_hda_preset_si3054[] = {
- 	{ .id = 0x163c3055, .name = "Si3054", .patch = patch_si3054 },
- 	{ .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 },
- 	{ .id = 0x11c13026, .name = "Si3054", .patch = patch_si3054 },
- 	{ .id = 0x11c13055, .name = "Si3054", .patch = patch_si3054 },
- 	{ .id = 0x11c13155, .name = "Si3054", .patch = patch_si3054 },
- 	{ .id = 0x10573055, .name = "Si3054", .patch = patch_si3054 },
- 	{ .id = 0x10573057, .name = "Si3054", .patch = patch_si3054 },
- 	{ .id = 0x10573155, .name = "Si3054", .patch = patch_si3054 },
+static const struct hda_device_id snd_hda_id_si3054[] = {
+	HDA_CODEC_ENTRY(0x163c3055, "Si3054", patch_si3054),
+	HDA_CODEC_ENTRY(0x163c3155, "Si3054", patch_si3054),
+	HDA_CODEC_ENTRY(0x11c13026, "Si3054", patch_si3054),
+	HDA_CODEC_ENTRY(0x11c13055, "Si3054", patch_si3054),
+	HDA_CODEC_ENTRY(0x11c13155, "Si3054", patch_si3054),
+	HDA_CODEC_ENTRY(0x10573055, "Si3054", patch_si3054),
+	HDA_CODEC_ENTRY(0x10573057, "Si3054", patch_si3054),
+	HDA_CODEC_ENTRY(0x10573155, "Si3054", patch_si3054),
 	/* VIA HDA on Clevo m540 */
-	{ .id = 0x11063288, .name = "Si3054", .patch = patch_si3054 },
+	HDA_CODEC_ENTRY(0x11063288, "Si3054", patch_si3054),
 	/* Asus A8J Modem (SM56) */
-	{ .id = 0x15433155, .name = "Si3054", .patch = patch_si3054 },
+	HDA_CODEC_ENTRY(0x15433155, "Si3054", patch_si3054),
 	/* LG LW20 modem */
-	{ .id = 0x18540018, .name = "Si3054", .patch = patch_si3054 },
+	HDA_CODEC_ENTRY(0x18540018, "Si3054", patch_si3054),
 	{}
 };
-
-MODULE_ALIAS("snd-hda-codec-id:163c3055");
-MODULE_ALIAS("snd-hda-codec-id:163c3155");
-MODULE_ALIAS("snd-hda-codec-id:11c13026");
-MODULE_ALIAS("snd-hda-codec-id:11c13055");
-MODULE_ALIAS("snd-hda-codec-id:11c13155");
-MODULE_ALIAS("snd-hda-codec-id:10573055");
-MODULE_ALIAS("snd-hda-codec-id:10573057");
-MODULE_ALIAS("snd-hda-codec-id:10573155");
-MODULE_ALIAS("snd-hda-codec-id:11063288");
-MODULE_ALIAS("snd-hda-codec-id:15433155");
-MODULE_ALIAS("snd-hda-codec-id:18540018");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_si3054);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Si3054 HD-audio modem codec");
 
 static struct hda_codec_driver si3054_driver = {
-	.preset = snd_hda_preset_si3054,
+	.id = snd_hda_id_si3054,
 };
 
 module_hda_codec_driver(si3054_driver);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index def5cc8dff02..08a0f6a35cca 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5012,121 +5012,119 @@ static int patch_stac9872(struct hda_codec *codec)
 /*
  * patch entries
  */
-static const struct hda_codec_preset snd_hda_preset_sigmatel[] = {
- 	{ .id = 0x83847690, .name = "STAC9200", .patch = patch_stac9200 },
- 	{ .id = 0x83847882, .name = "STAC9220 A1", .patch = patch_stac922x },
- 	{ .id = 0x83847680, .name = "STAC9221 A1", .patch = patch_stac922x },
- 	{ .id = 0x83847880, .name = "STAC9220 A2", .patch = patch_stac922x },
- 	{ .id = 0x83847681, .name = "STAC9220D/9223D A2", .patch = patch_stac922x },
- 	{ .id = 0x83847682, .name = "STAC9221 A2", .patch = patch_stac922x },
- 	{ .id = 0x83847683, .name = "STAC9221D A2", .patch = patch_stac922x },
- 	{ .id = 0x83847618, .name = "STAC9227", .patch = patch_stac927x },
- 	{ .id = 0x83847619, .name = "STAC9227", .patch = patch_stac927x },
- 	{ .id = 0x83847616, .name = "STAC9228", .patch = patch_stac927x },
- 	{ .id = 0x83847617, .name = "STAC9228", .patch = patch_stac927x },
- 	{ .id = 0x83847614, .name = "STAC9229", .patch = patch_stac927x },
- 	{ .id = 0x83847615, .name = "STAC9229", .patch = patch_stac927x },
- 	{ .id = 0x83847620, .name = "STAC9274", .patch = patch_stac927x },
- 	{ .id = 0x83847621, .name = "STAC9274D", .patch = patch_stac927x },
- 	{ .id = 0x83847622, .name = "STAC9273X", .patch = patch_stac927x },
- 	{ .id = 0x83847623, .name = "STAC9273D", .patch = patch_stac927x },
- 	{ .id = 0x83847624, .name = "STAC9272X", .patch = patch_stac927x },
- 	{ .id = 0x83847625, .name = "STAC9272D", .patch = patch_stac927x },
- 	{ .id = 0x83847626, .name = "STAC9271X", .patch = patch_stac927x },
- 	{ .id = 0x83847627, .name = "STAC9271D", .patch = patch_stac927x },
- 	{ .id = 0x83847628, .name = "STAC9274X5NH", .patch = patch_stac927x },
- 	{ .id = 0x83847629, .name = "STAC9274D5NH", .patch = patch_stac927x },
-	{ .id = 0x83847632, .name = "STAC9202",  .patch = patch_stac925x },
-	{ .id = 0x83847633, .name = "STAC9202D", .patch = patch_stac925x },
-	{ .id = 0x83847634, .name = "STAC9250", .patch = patch_stac925x },
-	{ .id = 0x83847635, .name = "STAC9250D", .patch = patch_stac925x },
-	{ .id = 0x83847636, .name = "STAC9251", .patch = patch_stac925x },
-	{ .id = 0x83847637, .name = "STAC9250D", .patch = patch_stac925x },
-	{ .id = 0x83847645, .name = "92HD206X", .patch = patch_stac927x },
-	{ .id = 0x83847646, .name = "92HD206D", .patch = patch_stac927x },
- 	/* The following does not take into account .id=0x83847661 when subsys =
- 	 * 104D0C00 which is STAC9225s. Because of this, some SZ Notebooks are
- 	 * currently not fully supported.
- 	 */
- 	{ .id = 0x83847661, .name = "CXD9872RD/K", .patch = patch_stac9872 },
- 	{ .id = 0x83847662, .name = "STAC9872AK", .patch = patch_stac9872 },
- 	{ .id = 0x83847664, .name = "CXD9872AKD", .patch = patch_stac9872 },
-	{ .id = 0x83847698, .name = "STAC9205", .patch = patch_stac9205 },
- 	{ .id = 0x838476a0, .name = "STAC9205", .patch = patch_stac9205 },
- 	{ .id = 0x838476a1, .name = "STAC9205D", .patch = patch_stac9205 },
- 	{ .id = 0x838476a2, .name = "STAC9204", .patch = patch_stac9205 },
- 	{ .id = 0x838476a3, .name = "STAC9204D", .patch = patch_stac9205 },
- 	{ .id = 0x838476a4, .name = "STAC9255", .patch = patch_stac9205 },
- 	{ .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 },
- 	{ .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 },
- 	{ .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 },
-	{ .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx},
-	{ .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76d4, .name = "92HD83C1C5", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76d1, .name = "92HD87B1/3", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76d9, .name = "92HD87B2/4", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d7666, .name = "92HD88B3", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d7667, .name = "92HD88B1", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d7668, .name = "92HD88B2", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d7669, .name = "92HD88B4", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx},
-	{ .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d7695, .name = "92HD95", .patch = patch_stac92hd95 },
-	{ .id = 0x111d76b0, .name = "92HD71B8X", .patch = patch_stac92hd71bxx },
-	{ .id = 0x111d76b1, .name = "92HD71B8X", .patch = patch_stac92hd71bxx },
-	{ .id = 0x111d76b2, .name = "92HD71B7X", .patch = patch_stac92hd71bxx },
-	{ .id = 0x111d76b3, .name = "92HD71B7X", .patch = patch_stac92hd71bxx },
-	{ .id = 0x111d76b4, .name = "92HD71B6X", .patch = patch_stac92hd71bxx },
-	{ .id = 0x111d76b5, .name = "92HD71B6X", .patch = patch_stac92hd71bxx },
-	{ .id = 0x111d76b6, .name = "92HD71B5X", .patch = patch_stac92hd71bxx },
-	{ .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx },
-	{ .id = 0x111d76c0, .name = "92HD89C3", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76c1, .name = "92HD89C2", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76c2, .name = "92HD89C1", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76c3, .name = "92HD89B3", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76c4, .name = "92HD89B2", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76c5, .name = "92HD89B1", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76c6, .name = "92HD89E3", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76c7, .name = "92HD89E2", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76c8, .name = "92HD89E1", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76c9, .name = "92HD89D3", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76ca, .name = "92HD89D2", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76cb, .name = "92HD89D1", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76e7, .name = "92HD90BXX", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76e8, .name = "92HD66B1X5", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76e9, .name = "92HD66B2X5", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76ea, .name = "92HD66B3X5", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76eb, .name = "92HD66C1X5", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76ec, .name = "92HD66C2X5", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76ed, .name = "92HD66C3X5", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76ee, .name = "92HD66B1X3", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76ef, .name = "92HD66B2X3", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76f0, .name = "92HD66B3X3", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76f1, .name = "92HD66C1X3", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76f2, .name = "92HD66C2X3", .patch = patch_stac92hd83xxx},
-	{ .id = 0x111d76f3, .name = "92HD66C3/65", .patch = patch_stac92hd83xxx},
+static const struct hda_device_id snd_hda_id_sigmatel[] = {
+	HDA_CODEC_ENTRY(0x83847690, "STAC9200", patch_stac9200),
+	HDA_CODEC_ENTRY(0x83847882, "STAC9220 A1", patch_stac922x),
+	HDA_CODEC_ENTRY(0x83847680, "STAC9221 A1", patch_stac922x),
+	HDA_CODEC_ENTRY(0x83847880, "STAC9220 A2", patch_stac922x),
+	HDA_CODEC_ENTRY(0x83847681, "STAC9220D/9223D A2", patch_stac922x),
+	HDA_CODEC_ENTRY(0x83847682, "STAC9221 A2", patch_stac922x),
+	HDA_CODEC_ENTRY(0x83847683, "STAC9221D A2", patch_stac922x),
+	HDA_CODEC_ENTRY(0x83847618, "STAC9227", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847619, "STAC9227", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847616, "STAC9228", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847617, "STAC9228", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847614, "STAC9229", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847615, "STAC9229", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847620, "STAC9274", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847621, "STAC9274D", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847622, "STAC9273X", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847623, "STAC9273D", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847624, "STAC9272X", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847625, "STAC9272D", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847626, "STAC9271X", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847627, "STAC9271D", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847628, "STAC9274X5NH", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847629, "STAC9274D5NH", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847632, "STAC9202",  patch_stac925x),
+	HDA_CODEC_ENTRY(0x83847633, "STAC9202D", patch_stac925x),
+	HDA_CODEC_ENTRY(0x83847634, "STAC9250", patch_stac925x),
+	HDA_CODEC_ENTRY(0x83847635, "STAC9250D", patch_stac925x),
+	HDA_CODEC_ENTRY(0x83847636, "STAC9251", patch_stac925x),
+	HDA_CODEC_ENTRY(0x83847637, "STAC9250D", patch_stac925x),
+	HDA_CODEC_ENTRY(0x83847645, "92HD206X", patch_stac927x),
+	HDA_CODEC_ENTRY(0x83847646, "92HD206D", patch_stac927x),
+	/* The following does not take into account .id=0x83847661 when subsys =
+	 * 104D0C00 which is STAC9225s. Because of this, some SZ Notebooks are
+	 * currently not fully supported.
+	 */
+	HDA_CODEC_ENTRY(0x83847661, "CXD9872RD/K", patch_stac9872),
+	HDA_CODEC_ENTRY(0x83847662, "STAC9872AK", patch_stac9872),
+	HDA_CODEC_ENTRY(0x83847664, "CXD9872AKD", patch_stac9872),
+	HDA_CODEC_ENTRY(0x83847698, "STAC9205", patch_stac9205),
+	HDA_CODEC_ENTRY(0x838476a0, "STAC9205", patch_stac9205),
+	HDA_CODEC_ENTRY(0x838476a1, "STAC9205D", patch_stac9205),
+	HDA_CODEC_ENTRY(0x838476a2, "STAC9204", patch_stac9205),
+	HDA_CODEC_ENTRY(0x838476a3, "STAC9204D", patch_stac9205),
+	HDA_CODEC_ENTRY(0x838476a4, "STAC9255", patch_stac9205),
+	HDA_CODEC_ENTRY(0x838476a5, "STAC9255D", patch_stac9205),
+	HDA_CODEC_ENTRY(0x838476a6, "STAC9254", patch_stac9205),
+	HDA_CODEC_ENTRY(0x838476a7, "STAC9254D", patch_stac9205),
+	HDA_CODEC_ENTRY(0x111d7603, "92HD75B3X5", patch_stac92hd71bxx),
+	HDA_CODEC_ENTRY(0x111d7604, "92HD83C1X5", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76d4, "92HD83C1C5", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d7605, "92HD81B1X5", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76d5, "92HD81B1C5", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76d1, "92HD87B1/3", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76d9, "92HD87B2/4", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d7666, "92HD88B3", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d7667, "92HD88B1", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d7668, "92HD88B2", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d7669, "92HD88B4", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d7608, "92HD75B2X5", patch_stac92hd71bxx),
+	HDA_CODEC_ENTRY(0x111d7674, "92HD73D1X5", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d7675, "92HD73C1X5", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d7676, "92HD73E1X5", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d7695, "92HD95", patch_stac92hd95),
+	HDA_CODEC_ENTRY(0x111d76b0, "92HD71B8X", patch_stac92hd71bxx),
+	HDA_CODEC_ENTRY(0x111d76b1, "92HD71B8X", patch_stac92hd71bxx),
+	HDA_CODEC_ENTRY(0x111d76b2, "92HD71B7X", patch_stac92hd71bxx),
+	HDA_CODEC_ENTRY(0x111d76b3, "92HD71B7X", patch_stac92hd71bxx),
+	HDA_CODEC_ENTRY(0x111d76b4, "92HD71B6X", patch_stac92hd71bxx),
+	HDA_CODEC_ENTRY(0x111d76b5, "92HD71B6X", patch_stac92hd71bxx),
+	HDA_CODEC_ENTRY(0x111d76b6, "92HD71B5X", patch_stac92hd71bxx),
+	HDA_CODEC_ENTRY(0x111d76b7, "92HD71B5X", patch_stac92hd71bxx),
+	HDA_CODEC_ENTRY(0x111d76c0, "92HD89C3", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d76c1, "92HD89C2", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d76c2, "92HD89C1", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d76c3, "92HD89B3", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d76c4, "92HD89B2", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d76c5, "92HD89B1", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d76c6, "92HD89E3", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d76c7, "92HD89E2", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d76c8, "92HD89E1", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d76c9, "92HD89D3", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d76ca, "92HD89D2", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d76cb, "92HD89D1", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d76cc, "92HD89F3", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d76cd, "92HD89F2", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d76ce, "92HD89F1", patch_stac92hd73xx),
+	HDA_CODEC_ENTRY(0x111d76df, "92HD93BXX", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76e0, "92HD91BXX", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76e3, "92HD98BXX", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76e5, "92HD99BXX", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76e7, "92HD90BXX", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76e8, "92HD66B1X5", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76e9, "92HD66B2X5", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76ea, "92HD66B3X5", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76eb, "92HD66C1X5", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76ec, "92HD66C2X5", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76ed, "92HD66C3X5", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76ee, "92HD66B1X3", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76ef, "92HD66B2X3", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76f0, "92HD66B3X3", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76f1, "92HD66C1X3", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76f2, "92HD66C2X3", patch_stac92hd83xxx),
+	HDA_CODEC_ENTRY(0x111d76f3, "92HD66C3/65", patch_stac92hd83xxx),
 	{} /* terminator */
 };
-
-MODULE_ALIAS("snd-hda-codec-id:8384*");
-MODULE_ALIAS("snd-hda-codec-id:111d*");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_sigmatel);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("IDT/Sigmatel HD-audio codec");
 
 static struct hda_codec_driver sigmatel_driver = {
-	.preset = snd_hda_preset_sigmatel,
+	.id = snd_hda_id_sigmatel,
 };
 
 module_hda_codec_driver(sigmatel_driver);
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index da5366405eda..fc30d1e8aa76 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -785,21 +785,11 @@ static int patch_vt1708S(struct hda_codec *codec)
 	override_mic_boost(codec, 0x1e, 0, 3, 40);
 
 	/* correct names for VT1708BCE */
-	if (get_codec_type(codec) == VT1708BCE)	{
-		kfree(codec->core.chip_name);
-		codec->core.chip_name = kstrdup("VT1708BCE", GFP_KERNEL);
-		snprintf(codec->card->mixername,
-			 sizeof(codec->card->mixername),
-			 "%s %s", codec->core.vendor_name, codec->core.chip_name);
-	}
+	if (get_codec_type(codec) == VT1708BCE)
+		snd_hda_codec_set_name(codec, "VT1708BCE");
 	/* correct names for VT1705 */
-	if (codec->core.vendor_id == 0x11064397) {
-		kfree(codec->core.chip_name);
-		codec->core.chip_name = kstrdup("VT1705", GFP_KERNEL);
-		snprintf(codec->card->mixername,
-			 sizeof(codec->card->mixername),
-			 "%s %s", codec->core.vendor_name, codec->core.chip_name);
-	}
+	if (codec->core.vendor_id == 0x11064397)
+		snd_hda_codec_set_name(codec, "VT1705");
 
 	/* automatic parse from the BIOS config */
 	err = via_parse_auto_config(codec);
@@ -1210,109 +1200,64 @@ static int patch_vt3476(struct hda_codec *codec)
 /*
  * patch entries
  */
-static const struct hda_codec_preset snd_hda_preset_via[] = {
-	{ .id = 0x11061708, .name = "VT1708", .patch = patch_vt1708},
-	{ .id = 0x11061709, .name = "VT1708", .patch = patch_vt1708},
-	{ .id = 0x1106170a, .name = "VT1708", .patch = patch_vt1708},
-	{ .id = 0x1106170b, .name = "VT1708", .patch = patch_vt1708},
-	{ .id = 0x1106e710, .name = "VT1709 10-Ch",
-	  .patch = patch_vt1709},
-	{ .id = 0x1106e711, .name = "VT1709 10-Ch",
-	  .patch = patch_vt1709},
-	{ .id = 0x1106e712, .name = "VT1709 10-Ch",
-	  .patch = patch_vt1709},
-	{ .id = 0x1106e713, .name = "VT1709 10-Ch",
-	  .patch = patch_vt1709},
-	{ .id = 0x1106e714, .name = "VT1709 6-Ch",
-	  .patch = patch_vt1709},
-	{ .id = 0x1106e715, .name = "VT1709 6-Ch",
-	  .patch = patch_vt1709},
-	{ .id = 0x1106e716, .name = "VT1709 6-Ch",
-	  .patch = patch_vt1709},
-	{ .id = 0x1106e717, .name = "VT1709 6-Ch",
-	  .patch = patch_vt1709},
-	{ .id = 0x1106e720, .name = "VT1708B 8-Ch",
-	  .patch = patch_vt1708B},
-	{ .id = 0x1106e721, .name = "VT1708B 8-Ch",
-	  .patch = patch_vt1708B},
-	{ .id = 0x1106e722, .name = "VT1708B 8-Ch",
-	  .patch = patch_vt1708B},
-	{ .id = 0x1106e723, .name = "VT1708B 8-Ch",
-	  .patch = patch_vt1708B},
-	{ .id = 0x1106e724, .name = "VT1708B 4-Ch",
-	  .patch = patch_vt1708B},
-	{ .id = 0x1106e725, .name = "VT1708B 4-Ch",
-	  .patch = patch_vt1708B},
-	{ .id = 0x1106e726, .name = "VT1708B 4-Ch",
-	  .patch = patch_vt1708B},
-	{ .id = 0x1106e727, .name = "VT1708B 4-Ch",
-	  .patch = patch_vt1708B},
-	{ .id = 0x11060397, .name = "VT1708S",
-	  .patch = patch_vt1708S},
-	{ .id = 0x11061397, .name = "VT1708S",
-	  .patch = patch_vt1708S},
-	{ .id = 0x11062397, .name = "VT1708S",
-	  .patch = patch_vt1708S},
-	{ .id = 0x11063397, .name = "VT1708S",
-	  .patch = patch_vt1708S},
-	{ .id = 0x11064397, .name = "VT1705",
-	  .patch = patch_vt1708S},
-	{ .id = 0x11065397, .name = "VT1708S",
-	  .patch = patch_vt1708S},
-	{ .id = 0x11066397, .name = "VT1708S",
-	  .patch = patch_vt1708S},
-	{ .id = 0x11067397, .name = "VT1708S",
-	  .patch = patch_vt1708S},
-	{ .id = 0x11060398, .name = "VT1702",
-	  .patch = patch_vt1702},
-	{ .id = 0x11061398, .name = "VT1702",
-	  .patch = patch_vt1702},
-	{ .id = 0x11062398, .name = "VT1702",
-	  .patch = patch_vt1702},
-	{ .id = 0x11063398, .name = "VT1702",
-	  .patch = patch_vt1702},
-	{ .id = 0x11064398, .name = "VT1702",
-	  .patch = patch_vt1702},
-	{ .id = 0x11065398, .name = "VT1702",
-	  .patch = patch_vt1702},
-	{ .id = 0x11066398, .name = "VT1702",
-	  .patch = patch_vt1702},
-	{ .id = 0x11067398, .name = "VT1702",
-	  .patch = patch_vt1702},
-	{ .id = 0x11060428, .name = "VT1718S",
-	  .patch = patch_vt1718S},
-	{ .id = 0x11064428, .name = "VT1718S",
-	  .patch = patch_vt1718S},
-	{ .id = 0x11060441, .name = "VT2020",
-	  .patch = patch_vt1718S},
-	{ .id = 0x11064441, .name = "VT1828S",
-	  .patch = patch_vt1718S},
-	{ .id = 0x11060433, .name = "VT1716S",
-	  .patch = patch_vt1716S},
-	{ .id = 0x1106a721, .name = "VT1716S",
-	  .patch = patch_vt1716S},
-	{ .id = 0x11060438, .name = "VT2002P", .patch = patch_vt2002P},
-	{ .id = 0x11064438, .name = "VT2002P", .patch = patch_vt2002P},
-	{ .id = 0x11060448, .name = "VT1812", .patch = patch_vt1812},
-	{ .id = 0x11060440, .name = "VT1818S",
-	  .patch = patch_vt1708S},
-	{ .id = 0x11060446, .name = "VT1802",
-		.patch = patch_vt2002P},
-	{ .id = 0x11068446, .name = "VT1802",
-		.patch = patch_vt2002P},
-	{ .id = 0x11064760, .name = "VT1705CF",
-		.patch = patch_vt3476},
-	{ .id = 0x11064761, .name = "VT1708SCE",
-		.patch = patch_vt3476},
-	{ .id = 0x11064762, .name = "VT1808",
-		.patch = patch_vt3476},
+static const struct hda_device_id snd_hda_id_via[] = {
+	HDA_CODEC_ENTRY(0x11061708, "VT1708", patch_vt1708),
+	HDA_CODEC_ENTRY(0x11061709, "VT1708", patch_vt1708),
+	HDA_CODEC_ENTRY(0x1106170a, "VT1708", patch_vt1708),
+	HDA_CODEC_ENTRY(0x1106170b, "VT1708", patch_vt1708),
+	HDA_CODEC_ENTRY(0x1106e710, "VT1709 10-Ch", patch_vt1709),
+	HDA_CODEC_ENTRY(0x1106e711, "VT1709 10-Ch", patch_vt1709),
+	HDA_CODEC_ENTRY(0x1106e712, "VT1709 10-Ch", patch_vt1709),
+	HDA_CODEC_ENTRY(0x1106e713, "VT1709 10-Ch", patch_vt1709),
+	HDA_CODEC_ENTRY(0x1106e714, "VT1709 6-Ch", patch_vt1709),
+	HDA_CODEC_ENTRY(0x1106e715, "VT1709 6-Ch", patch_vt1709),
+	HDA_CODEC_ENTRY(0x1106e716, "VT1709 6-Ch", patch_vt1709),
+	HDA_CODEC_ENTRY(0x1106e717, "VT1709 6-Ch", patch_vt1709),
+	HDA_CODEC_ENTRY(0x1106e720, "VT1708B 8-Ch", patch_vt1708B),
+	HDA_CODEC_ENTRY(0x1106e721, "VT1708B 8-Ch", patch_vt1708B),
+	HDA_CODEC_ENTRY(0x1106e722, "VT1708B 8-Ch", patch_vt1708B),
+	HDA_CODEC_ENTRY(0x1106e723, "VT1708B 8-Ch", patch_vt1708B),
+	HDA_CODEC_ENTRY(0x1106e724, "VT1708B 4-Ch", patch_vt1708B),
+	HDA_CODEC_ENTRY(0x1106e725, "VT1708B 4-Ch", patch_vt1708B),
+	HDA_CODEC_ENTRY(0x1106e726, "VT1708B 4-Ch", patch_vt1708B),
+	HDA_CODEC_ENTRY(0x1106e727, "VT1708B 4-Ch", patch_vt1708B),
+	HDA_CODEC_ENTRY(0x11060397, "VT1708S", patch_vt1708S),
+	HDA_CODEC_ENTRY(0x11061397, "VT1708S", patch_vt1708S),
+	HDA_CODEC_ENTRY(0x11062397, "VT1708S", patch_vt1708S),
+	HDA_CODEC_ENTRY(0x11063397, "VT1708S", patch_vt1708S),
+	HDA_CODEC_ENTRY(0x11064397, "VT1705", patch_vt1708S),
+	HDA_CODEC_ENTRY(0x11065397, "VT1708S", patch_vt1708S),
+	HDA_CODEC_ENTRY(0x11066397, "VT1708S", patch_vt1708S),
+	HDA_CODEC_ENTRY(0x11067397, "VT1708S", patch_vt1708S),
+	HDA_CODEC_ENTRY(0x11060398, "VT1702", patch_vt1702),
+	HDA_CODEC_ENTRY(0x11061398, "VT1702", patch_vt1702),
+	HDA_CODEC_ENTRY(0x11062398, "VT1702", patch_vt1702),
+	HDA_CODEC_ENTRY(0x11063398, "VT1702", patch_vt1702),
+	HDA_CODEC_ENTRY(0x11064398, "VT1702", patch_vt1702),
+	HDA_CODEC_ENTRY(0x11065398, "VT1702", patch_vt1702),
+	HDA_CODEC_ENTRY(0x11066398, "VT1702", patch_vt1702),
+	HDA_CODEC_ENTRY(0x11067398, "VT1702", patch_vt1702),
+	HDA_CODEC_ENTRY(0x11060428, "VT1718S", patch_vt1718S),
+	HDA_CODEC_ENTRY(0x11064428, "VT1718S", patch_vt1718S),
+	HDA_CODEC_ENTRY(0x11060441, "VT2020", patch_vt1718S),
+	HDA_CODEC_ENTRY(0x11064441, "VT1828S", patch_vt1718S),
+	HDA_CODEC_ENTRY(0x11060433, "VT1716S", patch_vt1716S),
+	HDA_CODEC_ENTRY(0x1106a721, "VT1716S", patch_vt1716S),
+	HDA_CODEC_ENTRY(0x11060438, "VT2002P", patch_vt2002P),
+	HDA_CODEC_ENTRY(0x11064438, "VT2002P", patch_vt2002P),
+	HDA_CODEC_ENTRY(0x11060448, "VT1812", patch_vt1812),
+	HDA_CODEC_ENTRY(0x11060440, "VT1818S", patch_vt1708S),
+	HDA_CODEC_ENTRY(0x11060446, "VT1802", patch_vt2002P),
+	HDA_CODEC_ENTRY(0x11068446, "VT1802", patch_vt2002P),
+	HDA_CODEC_ENTRY(0x11064760, "VT1705CF", patch_vt3476),
+	HDA_CODEC_ENTRY(0x11064761, "VT1708SCE", patch_vt3476),
+	HDA_CODEC_ENTRY(0x11064762, "VT1808", patch_vt3476),
 	{} /* terminator */
 };
-
-MODULE_ALIAS("snd-hda-codec-id:1106*");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_via);
 
 static struct hda_codec_driver via_driver = {
-	.preset = snd_hda_preset_via,
+	.id = snd_hda_id_via,
 };
 
 MODULE_LICENSE("GPL");
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 7acbc21d642a..9e1ad119a3ce 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -1394,7 +1394,9 @@ static int snd_korg1212_playback_open(struct snd_pcm_substream *substream)
 
         spin_unlock_irqrestore(&korg1212->lock, flags);
 
-        snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, kPlayBufferFrames, kPlayBufferFrames);
+	snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+				     kPlayBufferFrames);
+
         return 0;
 }
 
@@ -1422,8 +1424,8 @@ static int snd_korg1212_capture_open(struct snd_pcm_substream *substream)
 
         spin_unlock_irqrestore(&korg1212->lock, flags);
 
-        snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
-				     kPlayBufferFrames, kPlayBufferFrames);
+	snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+				     kPlayBufferFrames);
         return 0;
 }
 
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index cba89beb2b38..8b8e2e54fba3 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -234,8 +234,8 @@ static int lx_pcm_open(struct snd_pcm_substream *substream)
 
 	/* the clock rate cannot be changed */
 	board_rate = chip->board_sample_rate;
-	err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE,
-					   board_rate, board_rate);
+	err = snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_RATE,
+					   board_rate);
 
 	if (err < 0) {
 		dev_warn(chip->card->dev, "could not constrain periods\n");
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 72e89cedc52d..17ae92613de4 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -1929,15 +1929,32 @@ snd_m3_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val)
 		return;
 	snd_m3_outw(chip, val, CODEC_DATA);
 	snd_m3_outb(chip, reg & 0x7f, CODEC_COMMAND);
+	/*
+	 * Workaround for buggy ES1988 integrated AC'97 codec. It remains silent
+	 * until the MASTER volume or mute is touched (alsactl restore does not
+	 * work).
+	 */
+	if (ac97->id == 0x45838308 && reg == AC97_MASTER) {
+		snd_m3_ac97_wait(chip);
+		snd_m3_outw(chip, val, CODEC_DATA);
+		snd_m3_outb(chip, reg & 0x7f, CODEC_COMMAND);
+	}
 }
 
 
-static void snd_m3_remote_codec_config(int io, int isremote)
+static void snd_m3_remote_codec_config(struct snd_m3 *chip, int isremote)
 {
+	int io = chip->iobase;
+	u16 tmp;
+
 	isremote = isremote ? 1 : 0;
 
-	outw((inw(io + RING_BUS_CTRL_B) & ~SECOND_CODEC_ID_MASK) | isremote,
-	     io + RING_BUS_CTRL_B);
+	tmp = inw(io + RING_BUS_CTRL_B) & ~SECOND_CODEC_ID_MASK;
+	/* enable dock on Dell Latitude C810 */
+	if (chip->pci->subsystem_vendor == 0x1028 &&
+	    chip->pci->subsystem_device == 0x00e5)
+		tmp |= M3I_DOCK_ENABLE;
+	outw(tmp | isremote, io + RING_BUS_CTRL_B);
 	outw((inw(io + SDO_OUT_DEST_CTRL) & ~COMMAND_ADDR_OUT) | isremote,
 	     io + SDO_OUT_DEST_CTRL);
 	outw((inw(io + SDO_IN_DEST_CTRL) & ~STATUS_ADDR_IN) | isremote,
@@ -1989,7 +2006,7 @@ static void snd_m3_ac97_reset(struct snd_m3 *chip)
 		if (!chip->irda_workaround)
 			dir |= 0x10; /* assuming pci bus master? */
 
-		snd_m3_remote_codec_config(io, 0);
+		snd_m3_remote_codec_config(chip, 0);
 
 		outw(IO_SRAM_ENABLE, io + RING_BUS_CTRL_A);
 		udelay(20);
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index 23d7f5d30c41..cd94ac548ba3 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -831,9 +831,9 @@ static struct snd_pcm_hw_constraint_list hw_constraints_period_bytes = {
 static void snd_rme32_set_buffer_constraint(struct rme32 *rme32, struct snd_pcm_runtime *runtime)
 {
 	if (! rme32->fullduplex_mode) {
-		snd_pcm_hw_constraint_minmax(runtime,
+		snd_pcm_hw_constraint_single(runtime,
 					     SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
-					     RME32_BUFFER_SIZE, RME32_BUFFER_SIZE);
+					     RME32_BUFFER_SIZE);
 		snd_pcm_hw_constraint_list(runtime, 0,
 					   SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
 					   &hw_constraints_period_bytes);
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 2306ccf7281e..714df906249e 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -1152,13 +1152,13 @@ rme96_set_buffer_size_constraint(struct rme96 *rme96,
 {
 	unsigned int size;
 
-	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
-				     RME96_BUFFER_SIZE, RME96_BUFFER_SIZE);
+	snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+				     RME96_BUFFER_SIZE);
 	if ((size = rme96->playback_periodsize) != 0 ||
 	    (size = rme96->capture_periodsize) != 0)
-		snd_pcm_hw_constraint_minmax(runtime,
+		snd_pcm_hw_constraint_single(runtime,
 					     SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
-					     size, size);
+					     size);
 	else
 		snd_pcm_hw_constraint_list(runtime, 0,
 					   SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 9bba275b4c9b..2875b4f6d8c9 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -5112,6 +5112,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp)
 		dev_err(hdsp->card->dev,
 			"too short firmware size %d (expected %d)\n",
 			   (int)fw->size, HDSP_FIRMWARE_SIZE);
+		release_firmware(fw);
 		return -EINVAL;
 	}
 
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index cb666c73712d..8bc8016c173d 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6080,18 +6080,17 @@ static int snd_hdspm_open(struct snd_pcm_substream *substream)
 					     SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
 					     32, 4096);
 		/* RayDAT & AIO have a fixed buffer of 16384 samples per channel */
-		snd_pcm_hw_constraint_minmax(runtime,
+		snd_pcm_hw_constraint_single(runtime,
 					     SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
-					     16384, 16384);
+					     16384);
 		break;
 
 	default:
 		snd_pcm_hw_constraint_minmax(runtime,
 					     SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
 					     64, 8192);
-		snd_pcm_hw_constraint_minmax(runtime,
-					     SNDRV_PCM_HW_PARAM_PERIODS,
-					     2, 2);
+		snd_pcm_hw_constraint_single(runtime,
+					     SNDRV_PCM_HW_PARAM_PERIODS, 2);
 		break;
 	}
 
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 225bfda414e9..7ff7d88e46dd 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -9,7 +9,6 @@ menuconfig SND_SOC
 	select SND_JACK if INPUT=y || INPUT=SND
 	select REGMAP_I2C if I2C
 	select REGMAP_SPI if SPI_MASTER
-	select SND_COMPRESS_OFFLOAD
 	---help---
 
 	  If you want ASoC support, you should say Y here and also to the
@@ -30,6 +29,10 @@ config SND_SOC_GENERIC_DMAENGINE_PCM
 	bool
 	select SND_DMAENGINE_PCM
 
+config SND_SOC_COMPRESS
+	bool
+	select SND_COMPRESS_OFFLOAD
+
 config SND_SOC_TOPOLOGY
 	bool
 
@@ -58,6 +61,7 @@ source "sound/soc/sh/Kconfig"
 source "sound/soc/sirf/Kconfig"
 source "sound/soc/spear/Kconfig"
 source "sound/soc/sti/Kconfig"
+source "sound/soc/sunxi/Kconfig"
 source "sound/soc/tegra/Kconfig"
 source "sound/soc/txx9/Kconfig"
 source "sound/soc/ux500/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 134aca150a50..8eb06db32fa0 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,5 +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 soc-devres.o soc-ops.o
+snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o
+snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o
 
 ifneq ($(CONFIG_SND_SOC_TOPOLOGY),)
 snd-soc-core-objs += soc-topology.o
@@ -40,6 +41,7 @@ obj-$(CONFIG_SND_SOC)	+= sh/
 obj-$(CONFIG_SND_SOC)	+= sirf/
 obj-$(CONFIG_SND_SOC)	+= spear/
 obj-$(CONFIG_SND_SOC)	+= sti/
+obj-$(CONFIG_SND_SOC)	+= sunxi/
 obj-$(CONFIG_SND_SOC)	+= tegra/
 obj-$(CONFIG_SND_SOC)	+= txx9/
 obj-$(CONFIG_SND_SOC)	+= ux500/
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index 1489cd461aec..2d30464b81ce 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -59,4 +59,13 @@ config SND_AT91_SOC_SAM9X5_WM8731
 	help
 	  Say Y if you want to add support for audio SoC on an
 	  at91sam9x5 based board that is using WM8731 codec.
+
+config SND_ATMEL_SOC_CLASSD
+	tristate "Atmel ASoC driver for boards using CLASSD"
+	depends on ARCH_AT91 || COMPILE_TEST
+	select SND_ATMEL_SOC_DMA
+	select REGMAP_MMIO
+	help
+	  Say Y if you want to add support for Atmel ASoC driver for boards using
+	  CLASSD.
 endif
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index b327e5cc8de3..f6f7db428216 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -11,7 +11,9 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
 snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
 snd-atmel-soc-wm8904-objs := atmel_wm8904.o
 snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
+snd-atmel-soc-classd-objs := atmel-classd.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_ATMEL_SOC_CLASSD) += snd-atmel-soc-classd.o
diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c
new file mode 100644
index 000000000000..8276675730ef
--- /dev/null
+++ b/sound/soc/atmel/atmel-classd.c
@@ -0,0 +1,679 @@
+/* Atmel ALSA SoC Audio Class D Amplifier (CLASSD) driver
+ *
+ * Copyright (C) 2015 Atmel
+ *
+ * Author: Songjun Wu <songjun.wu@atmel.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 or later
+ * as published by the Free Software Foundation.
+ */
+
+#include <linux/of.h>
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include "atmel-classd.h"
+
+struct atmel_classd_pdata {
+	bool non_overlap_enable;
+	int non_overlap_time;
+	int pwm_type;
+	const char *card_name;
+};
+
+struct atmel_classd {
+	dma_addr_t phy_base;
+	struct regmap *regmap;
+	struct clk *pclk;
+	struct clk *gclk;
+	struct clk *aclk;
+	int irq;
+	const struct atmel_classd_pdata *pdata;
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id atmel_classd_of_match[] = {
+	{
+		.compatible = "atmel,sama5d2-classd",
+	}, {
+		/* sentinel */
+	}
+};
+MODULE_DEVICE_TABLE(of, atmel_classd_of_match);
+
+static struct atmel_classd_pdata *atmel_classd_dt_init(struct device *dev)
+{
+	struct device_node *np = dev->of_node;
+	struct atmel_classd_pdata *pdata;
+	const char *pwm_type;
+	int ret;
+
+	if (!np) {
+		dev_err(dev, "device node not found\n");
+		return ERR_PTR(-EINVAL);
+	}
+
+	pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL);
+	if (!pdata)
+		return ERR_PTR(-ENOMEM);
+
+	ret = of_property_read_string(np, "atmel,pwm-type", &pwm_type);
+	if ((ret == 0) && (strcmp(pwm_type, "diff") == 0))
+		pdata->pwm_type = CLASSD_MR_PWMTYP_DIFF;
+	else
+		pdata->pwm_type = CLASSD_MR_PWMTYP_SINGLE;
+
+	ret = of_property_read_u32(np,
+			"atmel,non-overlap-time", &pdata->non_overlap_time);
+	if (ret)
+		pdata->non_overlap_enable = false;
+	else
+		pdata->non_overlap_enable = true;
+
+	ret = of_property_read_string(np, "atmel,model", &pdata->card_name);
+	if (ret)
+		pdata->card_name = "CLASSD";
+
+	return pdata;
+}
+#else
+static inline struct atmel_classd_pdata *
+atmel_classd_dt_init(struct device *dev)
+{
+	return ERR_PTR(-EINVAL);
+}
+#endif
+
+#define ATMEL_CLASSD_RATES (SNDRV_PCM_RATE_8000 \
+			| SNDRV_PCM_RATE_16000	| SNDRV_PCM_RATE_22050 \
+			| SNDRV_PCM_RATE_32000	| SNDRV_PCM_RATE_44100 \
+			| SNDRV_PCM_RATE_48000	| SNDRV_PCM_RATE_88200 \
+			| SNDRV_PCM_RATE_96000)
+
+static const struct snd_pcm_hardware atmel_classd_hw = {
+	.info			= SNDRV_PCM_INFO_MMAP
+				| SNDRV_PCM_INFO_MMAP_VALID
+				| SNDRV_PCM_INFO_INTERLEAVED
+				| SNDRV_PCM_INFO_RESUME
+				| SNDRV_PCM_INFO_PAUSE,
+	.formats		= (SNDRV_PCM_FMTBIT_S16_LE),
+	.rates			= ATMEL_CLASSD_RATES,
+	.rate_min		= 8000,
+	.rate_max		= 96000,
+	.channels_min		= 2,
+	.channels_max		= 2,
+	.buffer_bytes_max	= 64 * 1024,
+	.period_bytes_min	= 256,
+	.period_bytes_max	= 32 * 1024,
+	.periods_min		= 2,
+	.periods_max		= 256,
+};
+
+#define ATMEL_CLASSD_PREALLOC_BUF_SIZE  (64 * 1024)
+
+/* cpu dai component */
+static int atmel_classd_cpu_dai_startup(struct snd_pcm_substream *substream,
+					struct snd_soc_dai *cpu_dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+
+	regmap_write(dd->regmap, CLASSD_THR, 0x0);
+
+	return clk_prepare_enable(dd->pclk);
+}
+
+static void atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream,
+					struct snd_soc_dai *cpu_dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+
+	clk_disable_unprepare(dd->pclk);
+}
+
+static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = {
+	.startup	= atmel_classd_cpu_dai_startup,
+	.shutdown	= atmel_classd_cpu_dai_shutdown,
+};
+
+static struct snd_soc_dai_driver atmel_classd_cpu_dai = {
+	.playback = {
+		.channels_min	= 2,
+		.channels_max	= 2,
+		.rates		= ATMEL_CLASSD_RATES,
+		.formats	= SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = &atmel_classd_cpu_dai_ops,
+};
+
+static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = {
+	.name = "atmel-classd",
+};
+
+/* platform */
+static int
+atmel_classd_platform_configure_dma(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params,
+	struct dma_slave_config *slave_config)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+
+	if (params_physical_width(params) != 16) {
+		dev_err(rtd->platform->dev,
+			"only supports 16-bit audio data\n");
+		return -EINVAL;
+	}
+
+	slave_config->direction		= DMA_MEM_TO_DEV;
+	slave_config->dst_addr		= dd->phy_base + CLASSD_THR;
+	slave_config->dst_addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES;
+	slave_config->dst_maxburst	= 1;
+	slave_config->src_maxburst	= 1;
+	slave_config->device_fc		= false;
+
+	return 0;
+}
+
+static const struct snd_dmaengine_pcm_config
+atmel_classd_dmaengine_pcm_config = {
+	.prepare_slave_config	= atmel_classd_platform_configure_dma,
+	.pcm_hardware		= &atmel_classd_hw,
+	.prealloc_buffer_size	= ATMEL_CLASSD_PREALLOC_BUF_SIZE,
+};
+
+/* codec */
+static const char * const mono_mode_text[] = {
+	"mix", "sat", "left", "right"
+};
+
+static SOC_ENUM_SINGLE_DECL(classd_mono_mode_enum,
+			CLASSD_INTPMR, CLASSD_INTPMR_MONO_MODE_SHIFT,
+			mono_mode_text);
+
+static const char * const eqcfg_text[] = {
+	"Treble-12dB", "Treble-6dB",
+	"Medium-8dB", "Medium-3dB",
+	"Bass-12dB", "Bass-6dB",
+	"0 dB",
+	"Bass+6dB", "Bass+12dB",
+	"Medium+3dB", "Medium+8dB",
+	"Treble+6dB", "Treble+12dB",
+};
+
+static const unsigned int eqcfg_value[] = {
+	CLASSD_INTPMR_EQCFG_T_CUT_12, CLASSD_INTPMR_EQCFG_T_CUT_6,
+	CLASSD_INTPMR_EQCFG_M_CUT_8, CLASSD_INTPMR_EQCFG_M_CUT_3,
+	CLASSD_INTPMR_EQCFG_B_CUT_12, CLASSD_INTPMR_EQCFG_B_CUT_6,
+	CLASSD_INTPMR_EQCFG_FLAT,
+	CLASSD_INTPMR_EQCFG_B_BOOST_6, CLASSD_INTPMR_EQCFG_B_BOOST_12,
+	CLASSD_INTPMR_EQCFG_M_BOOST_3, CLASSD_INTPMR_EQCFG_M_BOOST_8,
+	CLASSD_INTPMR_EQCFG_T_BOOST_6, CLASSD_INTPMR_EQCFG_T_BOOST_12,
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(classd_eqcfg_enum,
+		CLASSD_INTPMR, CLASSD_INTPMR_EQCFG_SHIFT, 0xf,
+		eqcfg_text, eqcfg_value);
+
+static const DECLARE_TLV_DB_SCALE(classd_digital_tlv, -7800, 100, 1);
+
+static const struct snd_kcontrol_new atmel_classd_snd_controls[] = {
+SOC_DOUBLE_TLV("Playback Volume", CLASSD_INTPMR,
+		CLASSD_INTPMR_ATTL_SHIFT, CLASSD_INTPMR_ATTR_SHIFT,
+		78, 1, classd_digital_tlv),
+
+SOC_SINGLE("Deemphasis Switch", CLASSD_INTPMR,
+		CLASSD_INTPMR_DEEMP_SHIFT, 1, 0),
+
+SOC_SINGLE("Mono Switch", CLASSD_INTPMR, CLASSD_INTPMR_MONO_SHIFT, 1, 0),
+
+SOC_SINGLE("Swap Switch", CLASSD_INTPMR, CLASSD_INTPMR_SWAP_SHIFT, 1, 0),
+
+SOC_ENUM("Mono Mode", classd_mono_mode_enum),
+
+SOC_ENUM("EQ", classd_eqcfg_enum),
+};
+
+static const char * const pwm_type[] = {
+	"Single ended", "Differential"
+};
+
+static int atmel_classd_codec_probe(struct snd_soc_codec *codec)
+{
+	struct snd_soc_card *card = snd_soc_codec_get_drvdata(codec);
+	struct atmel_classd *dd = snd_soc_card_get_drvdata(card);
+	const struct atmel_classd_pdata *pdata = dd->pdata;
+	u32 mask, val;
+
+	mask = CLASSD_MR_PWMTYP_MASK;
+	val = pdata->pwm_type << CLASSD_MR_PWMTYP_SHIFT;
+
+	mask |= CLASSD_MR_NON_OVERLAP_MASK;
+	if (pdata->non_overlap_enable) {
+		val |= (CLASSD_MR_NON_OVERLAP_EN
+			<< CLASSD_MR_NON_OVERLAP_SHIFT);
+
+		mask |= CLASSD_MR_NOVR_VAL_MASK;
+		switch (pdata->non_overlap_time) {
+		case 5:
+			val |= (CLASSD_MR_NOVR_VAL_5NS
+				<< CLASSD_MR_NOVR_VAL_SHIFT);
+			break;
+		case 10:
+			val |= (CLASSD_MR_NOVR_VAL_10NS
+				<< CLASSD_MR_NOVR_VAL_SHIFT);
+			break;
+		case 15:
+			val |= (CLASSD_MR_NOVR_VAL_15NS
+				<< CLASSD_MR_NOVR_VAL_SHIFT);
+			break;
+		case 20:
+			val |= (CLASSD_MR_NOVR_VAL_20NS
+				<< CLASSD_MR_NOVR_VAL_SHIFT);
+			break;
+		default:
+			val |= (CLASSD_MR_NOVR_VAL_10NS
+				<< CLASSD_MR_NOVR_VAL_SHIFT);
+			dev_warn(codec->dev,
+				"non-overlapping value %d is invalid, the default value 10 is specified\n",
+				pdata->non_overlap_time);
+			break;
+		}
+	}
+
+	snd_soc_update_bits(codec, CLASSD_MR, mask, val);
+
+	dev_info(codec->dev,
+		"PWM modulation type is %s, non-overlapping is %s\n",
+		pwm_type[pdata->pwm_type],
+		pdata->non_overlap_enable?"enabled":"disabled");
+
+	return 0;
+}
+
+static struct regmap *atmel_classd_codec_get_remap(struct device *dev)
+{
+	return dev_get_regmap(dev, NULL);
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_classd = {
+	.probe		= atmel_classd_codec_probe,
+	.controls	= atmel_classd_snd_controls,
+	.num_controls	= ARRAY_SIZE(atmel_classd_snd_controls),
+	.get_regmap	= atmel_classd_codec_get_remap,
+};
+
+/* codec dai component */
+static int atmel_classd_codec_dai_startup(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *codec_dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+	int ret;
+
+	ret = clk_prepare_enable(dd->aclk);
+	if (ret)
+		return ret;
+
+	return clk_prepare_enable(dd->gclk);
+}
+
+static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai,
+	int mute)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u32 mask, val;
+
+	mask = CLASSD_MR_LMUTE_MASK | CLASSD_MR_RMUTE_MASK;
+
+	if (mute)
+		val = mask;
+	else
+		val = 0;
+
+	snd_soc_update_bits(codec, CLASSD_MR, mask, val);
+
+	return 0;
+}
+
+#define CLASSD_ACLK_RATE_11M2896_MPY_8 (112896 * 100 * 8)
+#define CLASSD_ACLK_RATE_12M288_MPY_8  (12228 * 1000 * 8)
+
+static struct {
+	int rate;
+	int sample_rate;
+	int dsp_clk;
+	unsigned long aclk_rate;
+} const sample_rates[] = {
+	{ 8000,  CLASSD_INTPMR_FRAME_8K,
+	CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 },
+	{ 16000, CLASSD_INTPMR_FRAME_16K,
+	CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 },
+	{ 32000, CLASSD_INTPMR_FRAME_32K,
+	CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 },
+	{ 48000, CLASSD_INTPMR_FRAME_48K,
+	CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 },
+	{ 96000, CLASSD_INTPMR_FRAME_96K,
+	CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 },
+	{ 22050, CLASSD_INTPMR_FRAME_22K,
+	CLASSD_INTPMR_DSP_CLK_FREQ_11M2896, CLASSD_ACLK_RATE_11M2896_MPY_8 },
+	{ 44100, CLASSD_INTPMR_FRAME_44K,
+	CLASSD_INTPMR_DSP_CLK_FREQ_11M2896, CLASSD_ACLK_RATE_11M2896_MPY_8 },
+	{ 88200, CLASSD_INTPMR_FRAME_88K,
+	CLASSD_INTPMR_DSP_CLK_FREQ_11M2896, CLASSD_ACLK_RATE_11M2896_MPY_8 },
+};
+
+static int
+atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *codec_dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int fs;
+	int i, best, best_val, cur_val, ret;
+	u32 mask, val;
+
+	fs = params_rate(params);
+
+	best = 0;
+	best_val = abs(fs - sample_rates[0].rate);
+	for (i = 1; i < ARRAY_SIZE(sample_rates); i++) {
+		/* Closest match */
+		cur_val = abs(fs - sample_rates[i].rate);
+		if (cur_val < best_val) {
+			best = i;
+			best_val = cur_val;
+		}
+	}
+
+	dev_dbg(codec->dev,
+		"Selected SAMPLE_RATE of %dHz, ACLK_RATE of %ldHz\n",
+		sample_rates[best].rate, sample_rates[best].aclk_rate);
+
+	clk_disable_unprepare(dd->gclk);
+	clk_disable_unprepare(dd->aclk);
+
+	ret = clk_set_rate(dd->aclk, sample_rates[best].aclk_rate);
+	if (ret)
+		return ret;
+
+	mask = CLASSD_INTPMR_DSP_CLK_FREQ_MASK | CLASSD_INTPMR_FRAME_MASK;
+	val = (sample_rates[best].dsp_clk << CLASSD_INTPMR_DSP_CLK_FREQ_SHIFT)
+	| (sample_rates[best].sample_rate << CLASSD_INTPMR_FRAME_SHIFT);
+
+	snd_soc_update_bits(codec, CLASSD_INTPMR, mask, val);
+
+	ret = clk_prepare_enable(dd->aclk);
+	if (ret)
+		return ret;
+
+	return clk_prepare_enable(dd->gclk);
+}
+
+static void
+atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *codec_dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+
+	clk_disable_unprepare(dd->gclk);
+	clk_disable_unprepare(dd->aclk);
+}
+
+static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream,
+					struct snd_soc_dai *codec_dai)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+
+	snd_soc_update_bits(codec, CLASSD_MR,
+				CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK,
+				(CLASSD_MR_LEN_DIS << CLASSD_MR_LEN_SHIFT)
+				|(CLASSD_MR_REN_DIS << CLASSD_MR_REN_SHIFT));
+
+	return 0;
+}
+
+static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream,
+					int cmd, struct snd_soc_dai *codec_dai)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u32 mask, val;
+
+	mask = CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		val = mask;
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		val = (CLASSD_MR_LEN_DIS << CLASSD_MR_LEN_SHIFT)
+			| (CLASSD_MR_REN_DIS << CLASSD_MR_REN_SHIFT);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, CLASSD_MR, mask, val);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops atmel_classd_codec_dai_ops = {
+	.digital_mute	= atmel_classd_codec_dai_digital_mute,
+	.startup	= atmel_classd_codec_dai_startup,
+	.shutdown	= atmel_classd_codec_dai_shutdown,
+	.hw_params	= atmel_classd_codec_dai_hw_params,
+	.prepare	= atmel_classd_codec_dai_prepare,
+	.trigger	= atmel_classd_codec_dai_trigger,
+};
+
+#define ATMEL_CLASSD_CODEC_DAI_NAME  "atmel-classd-hifi"
+
+static struct snd_soc_dai_driver atmel_classd_codec_dai = {
+	.name = ATMEL_CLASSD_CODEC_DAI_NAME,
+	.playback = {
+		.stream_name	= "Playback",
+		.channels_min	= 2,
+		.channels_max	= 2,
+		.rates		= ATMEL_CLASSD_RATES,
+		.formats	= SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = &atmel_classd_codec_dai_ops,
+};
+
+/* ASoC sound card */
+static int atmel_classd_asoc_card_init(struct device *dev,
+					struct snd_soc_card *card)
+{
+	struct snd_soc_dai_link *dai_link;
+	struct atmel_classd *dd = snd_soc_card_get_drvdata(card);
+
+	dai_link = devm_kzalloc(dev, sizeof(*dai_link), GFP_KERNEL);
+	if (!dai_link)
+		return -ENOMEM;
+
+	dai_link->name			= "CLASSD";
+	dai_link->stream_name		= "CLASSD PCM";
+	dai_link->codec_dai_name	= ATMEL_CLASSD_CODEC_DAI_NAME;
+	dai_link->cpu_dai_name		= dev_name(dev);
+	dai_link->codec_name		= dev_name(dev);
+	dai_link->platform_name		= dev_name(dev);
+
+	card->dai_link	= dai_link;
+	card->num_links	= 1;
+	card->name	= dd->pdata->card_name;
+	card->dev	= dev;
+
+	return 0;
+};
+
+/* regmap configuration */
+static const struct reg_default atmel_classd_reg_defaults[] = {
+	{ CLASSD_INTPMR,   0x00301212 },
+};
+
+#define ATMEL_CLASSD_REG_MAX    0xE4
+static const struct regmap_config atmel_classd_regmap_config = {
+	.reg_bits	= 32,
+	.reg_stride	= 4,
+	.val_bits	= 32,
+	.max_register	= ATMEL_CLASSD_REG_MAX,
+
+	.cache_type		= REGCACHE_FLAT,
+	.reg_defaults		= atmel_classd_reg_defaults,
+	.num_reg_defaults	= ARRAY_SIZE(atmel_classd_reg_defaults),
+};
+
+static int atmel_classd_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct atmel_classd *dd;
+	struct resource *res;
+	void __iomem *io_base;
+	const struct atmel_classd_pdata *pdata;
+	struct snd_soc_card *card;
+	int ret;
+
+	pdata = dev_get_platdata(dev);
+	if (!pdata) {
+		pdata = atmel_classd_dt_init(dev);
+		if (IS_ERR(pdata))
+			return PTR_ERR(pdata);
+	}
+
+	dd = devm_kzalloc(dev, sizeof(*dd), GFP_KERNEL);
+	if (!dd)
+		return -ENOMEM;
+
+	dd->pdata = pdata;
+
+	dd->irq = platform_get_irq(pdev, 0);
+	if (dd->irq < 0) {
+		ret = dd->irq;
+		dev_err(dev, "failed to could not get irq: %d\n", ret);
+		return ret;
+	}
+
+	dd->pclk = devm_clk_get(dev, "pclk");
+	if (IS_ERR(dd->pclk)) {
+		ret = PTR_ERR(dd->pclk);
+		dev_err(dev, "failed to get peripheral clock: %d\n", ret);
+		return ret;
+	}
+
+	dd->gclk = devm_clk_get(dev, "gclk");
+	if (IS_ERR(dd->gclk)) {
+		ret = PTR_ERR(dd->gclk);
+		dev_err(dev, "failed to get GCK clock: %d\n", ret);
+		return ret;
+	}
+
+	dd->aclk = devm_clk_get(dev, "aclk");
+	if (IS_ERR(dd->aclk)) {
+		ret = PTR_ERR(dd->aclk);
+		dev_err(dev, "failed to get audio clock: %d\n", ret);
+		return ret;
+	}
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!res) {
+		dev_err(dev, "no memory resource\n");
+		return -ENXIO;
+	}
+
+	io_base = devm_ioremap_resource(dev, res);
+	if (IS_ERR(io_base)) {
+		ret =  PTR_ERR(io_base);
+		dev_err(dev, "failed to remap register memory: %d\n", ret);
+		return ret;
+	}
+
+	dd->phy_base = res->start;
+
+	dd->regmap = devm_regmap_init_mmio(dev, io_base,
+					&atmel_classd_regmap_config);
+	if (IS_ERR(dd->regmap)) {
+		ret = PTR_ERR(dd->regmap);
+		dev_err(dev, "failed to init register map: %d\n", ret);
+		return ret;
+	}
+
+	ret = devm_snd_soc_register_component(dev,
+					&atmel_classd_cpu_dai_component,
+					&atmel_classd_cpu_dai, 1);
+	if (ret) {
+		dev_err(dev, "could not register CPU DAI: %d\n", ret);
+		return ret;
+	}
+
+	ret = devm_snd_dmaengine_pcm_register(dev,
+					&atmel_classd_dmaengine_pcm_config,
+					0);
+	if (ret) {
+		dev_err(dev, "could not register platform: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_register_codec(dev, &soc_codec_dev_classd,
+					&atmel_classd_codec_dai, 1);
+	if (ret) {
+		dev_err(dev, "could not register codec: %d\n", ret);
+		return ret;
+	}
+
+	/* register sound card */
+	card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
+	if (!card)
+		return -ENOMEM;
+
+	snd_soc_card_set_drvdata(card, dd);
+	platform_set_drvdata(pdev, card);
+
+	ret = atmel_classd_asoc_card_init(dev, card);
+	if (ret) {
+		dev_err(dev, "failed to init sound card\n");
+		return ret;
+	}
+
+	ret = devm_snd_soc_register_card(dev, card);
+	if (ret) {
+		dev_err(dev, "failed to register sound card: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int atmel_classd_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver atmel_classd_driver = {
+	.driver	= {
+		.name		= "atmel-classd",
+		.of_match_table	= of_match_ptr(atmel_classd_of_match),
+		.pm		= &snd_soc_pm_ops,
+	},
+	.probe	= atmel_classd_probe,
+	.remove	= atmel_classd_remove,
+};
+module_platform_driver(atmel_classd_driver);
+
+MODULE_DESCRIPTION("Atmel ClassD driver under ALSA SoC architecture");
+MODULE_AUTHOR("Songjun Wu <songjun.wu@atmel.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/atmel/atmel-classd.h b/sound/soc/atmel/atmel-classd.h
new file mode 100644
index 000000000000..73f8fdd1ca83
--- /dev/null
+++ b/sound/soc/atmel/atmel-classd.h
@@ -0,0 +1,120 @@
+#ifndef __ATMEL_CLASSD_H_
+#define __ATMEL_CLASSD_H_
+
+#define CLASSD_CR		0x00000000
+#define CLASSD_CR_RESET		0x1
+
+#define CLASSD_MR			0x00000004
+
+#define CLASSD_MR_LEN_DIS		0x0
+#define CLASSD_MR_LEN_EN		0x1
+#define CLASSD_MR_LEN_MASK		(0x1 << 0)
+#define CLASSD_MR_LEN_SHIFT		(0)
+
+#define CLASSD_MR_LMUTE_DIS		0x0
+#define CLASSD_MR_LMUTE_EN		0x1
+#define CLASSD_MR_LMUTE_SHIFT		(0x1)
+#define CLASSD_MR_LMUTE_MASK		(0x1 << 1)
+
+#define CLASSD_MR_REN_DIS		0x0
+#define CLASSD_MR_REN_EN		0x1
+#define CLASSD_MR_REN_MASK		(0x1 << 4)
+#define CLASSD_MR_REN_SHIFT		(4)
+
+#define CLASSD_MR_RMUTE_DIS		0x0
+#define CLASSD_MR_RMUTE_EN		0x1
+#define CLASSD_MR_RMUTE_SHIFT		(0x5)
+#define CLASSD_MR_RMUTE_MASK		(0x1 << 5)
+
+#define CLASSD_MR_PWMTYP_SINGLE		0x0
+#define CLASSD_MR_PWMTYP_DIFF		0x1
+#define CLASSD_MR_PWMTYP_MASK		(0x1 << 8)
+#define CLASSD_MR_PWMTYP_SHIFT		(8)
+
+#define CLASSD_MR_NON_OVERLAP_DIS	0x0
+#define CLASSD_MR_NON_OVERLAP_EN	0x1
+#define CLASSD_MR_NON_OVERLAP_MASK	(0x1 << 16)
+#define CLASSD_MR_NON_OVERLAP_SHIFT	(16)
+
+#define CLASSD_MR_NOVR_VAL_5NS		0x0
+#define CLASSD_MR_NOVR_VAL_10NS		0x1
+#define CLASSD_MR_NOVR_VAL_15NS		0x2
+#define CLASSD_MR_NOVR_VAL_20NS		0x3
+#define CLASSD_MR_NOVR_VAL_MASK		(0x3 << 20)
+#define CLASSD_MR_NOVR_VAL_SHIFT	(20)
+
+#define CLASSD_INTPMR				0x00000008
+
+#define CLASSD_INTPMR_ATTL_MASK			(0x3f << 0)
+#define CLASSD_INTPMR_ATTL_SHIFT		(0)
+#define CLASSD_INTPMR_ATTR_MASK			(0x3f << 8)
+#define CLASSD_INTPMR_ATTR_SHIFT		(8)
+
+#define CLASSD_INTPMR_DSP_CLK_FREQ_12M288	0x0
+#define CLASSD_INTPMR_DSP_CLK_FREQ_11M2896	0x1
+#define CLASSD_INTPMR_DSP_CLK_FREQ_MASK		(0x1 << 16)
+#define CLASSD_INTPMR_DSP_CLK_FREQ_SHIFT	(16)
+
+#define CLASSD_INTPMR_DEEMP_DIS			0x0
+#define CLASSD_INTPMR_DEEMP_EN			0x1
+#define CLASSD_INTPMR_DEEMP_MASK		(0x1 << 18)
+#define CLASSD_INTPMR_DEEMP_SHIFT		(18)
+
+#define CLASSD_INTPMR_SWAP_LEFT_ON_LSB		0x0
+#define CLASSD_INTPMR_SWAP_RIGHT_ON_LSB		0x1
+#define CLASSD_INTPMR_SWAP_MASK			(0x1 << 19)
+#define CLASSD_INTPMR_SWAP_SHIFT		(19)
+
+#define CLASSD_INTPMR_FRAME_8K			0x0
+#define CLASSD_INTPMR_FRAME_16K			0x1
+#define CLASSD_INTPMR_FRAME_32K			0x2
+#define CLASSD_INTPMR_FRAME_48K			0x3
+#define CLASSD_INTPMR_FRAME_96K			0x4
+#define CLASSD_INTPMR_FRAME_22K			0x5
+#define CLASSD_INTPMR_FRAME_44K			0x6
+#define CLASSD_INTPMR_FRAME_88K			0x7
+#define CLASSD_INTPMR_FRAME_MASK		(0x7 << 20)
+#define CLASSD_INTPMR_FRAME_SHIFT		(20)
+
+#define CLASSD_INTPMR_EQCFG_FLAT		0x0
+#define CLASSD_INTPMR_EQCFG_B_BOOST_12		0x1
+#define CLASSD_INTPMR_EQCFG_B_BOOST_6		0x2
+#define CLASSD_INTPMR_EQCFG_B_CUT_12		0x3
+#define CLASSD_INTPMR_EQCFG_B_CUT_6		0x4
+#define CLASSD_INTPMR_EQCFG_M_BOOST_3		0x5
+#define CLASSD_INTPMR_EQCFG_M_BOOST_8		0x6
+#define CLASSD_INTPMR_EQCFG_M_CUT_3		0x7
+#define CLASSD_INTPMR_EQCFG_M_CUT_8		0x8
+#define CLASSD_INTPMR_EQCFG_T_BOOST_12		0x9
+#define CLASSD_INTPMR_EQCFG_T_BOOST_6		0xa
+#define CLASSD_INTPMR_EQCFG_T_CUT_12		0xb
+#define CLASSD_INTPMR_EQCFG_T_CUT_6		0xc
+#define CLASSD_INTPMR_EQCFG_SHIFT		(24)
+
+#define CLASSD_INTPMR_MONO_DIS			0x0
+#define CLASSD_INTPMR_MONO_EN			0x1
+#define CLASSD_INTPMR_MONO_MASK			(0x1 << 28)
+#define CLASSD_INTPMR_MONO_SHIFT		(28)
+
+#define CLASSD_INTPMR_MONO_MODE_MIX		0x0
+#define CLASSD_INTPMR_MONO_MODE_SAT		0x1
+#define CLASSD_INTPMR_MONO_MODE_LEFT		0x2
+#define CLASSD_INTPMR_MONO_MODE_RIGHT		0x3
+#define CLASSD_INTPMR_MONO_MODE_MASK		(0x3 << 29)
+#define CLASSD_INTPMR_MONO_MODE_SHIFT		(29)
+
+#define CLASSD_INTSR	0x0000000c
+
+#define CLASSD_THR	0x00000010
+
+#define CLASSD_IER	0x00000014
+
+#define CLASSD_IDR	0x00000018
+
+#define CLASSD_IMR	0x0000001c
+
+#define CLASSD_ISR	0x00000020
+
+#define CLASSD_WPMR	0x000000e4
+
+#endif
diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c
index aa354e1c6ff7..1933bcd46cca 100644
--- a/sound/soc/atmel/atmel_wm8904.c
+++ b/sound/soc/atmel/atmel_wm8904.c
@@ -176,6 +176,7 @@ static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = {
 	{ .compatible = "atmel,asoc-wm8904", },
 	{ }
 };
+MODULE_DEVICE_TABLE(of, atmel_asoc_wm8904_dt_ids);
 #endif
 
 static struct platform_driver atmel_asoc_wm8904_driver = {
diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c
index 452f404abfd2..e97c32798e98 100644
--- a/sound/soc/au1x/db1000.c
+++ b/sound/soc/au1x/db1000.c
@@ -38,14 +38,7 @@ static int db1000_audio_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &db1000_ac97;
 	card->dev = &pdev->dev;
-	return snd_soc_register_card(card);
-}
-
-static int db1000_audio_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-	snd_soc_unregister_card(card);
-	return 0;
+	return devm_snd_soc_register_card(&pdev->dev, card);
 }
 
 static struct platform_driver db1000_audio_driver = {
@@ -54,7 +47,6 @@ static struct platform_driver db1000_audio_driver = {
 		.pm	= &snd_soc_pm_ops,
 	},
 	.probe		= db1000_audio_probe,
-	.remove		= db1000_audio_remove,
 };
 
 module_platform_driver(db1000_audio_driver);
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 8c907ebea189..5c73061d912a 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -178,14 +178,7 @@ static int db1200_audio_probe(struct platform_device *pdev)
 
 	card = db1200_cards[pid->driver_data];
 	card->dev = &pdev->dev;
-	return snd_soc_register_card(card);
-}
-
-static int db1200_audio_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-	snd_soc_unregister_card(card);
-	return 0;
+	return devm_snd_soc_register_card(&pdev->dev, card);
 }
 
 static struct platform_driver db1200_audio_driver = {
@@ -195,7 +188,6 @@ static struct platform_driver db1200_audio_driver = {
 	},
 	.id_table	= db1200_pids,
 	.probe		= db1200_audio_probe,
-	.remove		= db1200_audio_remove,
 };
 
 module_platform_driver(db1200_audio_driver);
diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
index 5bf1501e5e3c..864df2616e10 100644
--- a/sound/soc/blackfin/bf5xx-ad1836.c
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -87,27 +87,18 @@ static int bf5xx_ad1836_driver_probe(struct platform_device *pdev)
 	card->dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
 
-	ret = snd_soc_register_card(card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret)
 		dev_err(&pdev->dev, "Failed to register card\n");
 	return ret;
 }
 
-static int bf5xx_ad1836_driver_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-	return 0;
-}
-
 static struct platform_driver bf5xx_ad1836_driver = {
 	.driver = {
 		.name = "bfin-snd-ad1836",
 		.pm = &snd_soc_pm_ops,
 	},
 	.probe = bf5xx_ad1836_driver_probe,
-	.remove = bf5xx_ad1836_driver_remove,
 };
 module_platform_driver(bf5xx_ad1836_driver);
 
diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c
index 523baf5820d7..72ac78988426 100644
--- a/sound/soc/blackfin/bfin-eval-adau1373.c
+++ b/sound/soc/blackfin/bfin-eval-adau1373.c
@@ -154,16 +154,7 @@ static int bfin_eval_adau1373_probe(struct platform_device *pdev)
 
 	card->dev = &pdev->dev;
 
-	return snd_soc_register_card(&bfin_eval_adau1373);
-}
-
-static int bfin_eval_adau1373_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-
-	return 0;
+	return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1373);
 }
 
 static struct platform_driver bfin_eval_adau1373_driver = {
@@ -172,7 +163,6 @@ static struct platform_driver bfin_eval_adau1373_driver = {
 		.pm = &snd_soc_pm_ops,
 	},
 	.probe = bfin_eval_adau1373_probe,
-	.remove = bfin_eval_adau1373_remove,
 };
 
 module_platform_driver(bfin_eval_adau1373_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c
index f9e926dfd4ef..5c67f72cf9a9 100644
--- a/sound/soc/blackfin/bfin-eval-adau1701.c
+++ b/sound/soc/blackfin/bfin-eval-adau1701.c
@@ -94,16 +94,7 @@ static int bfin_eval_adau1701_probe(struct platform_device *pdev)
 
 	card->dev = &pdev->dev;
 
-	return snd_soc_register_card(&bfin_eval_adau1701);
-}
-
-static int bfin_eval_adau1701_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-
-	return 0;
+	return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1701);
 }
 
 static struct platform_driver bfin_eval_adau1701_driver = {
@@ -112,7 +103,6 @@ static struct platform_driver bfin_eval_adau1701_driver = {
 		.pm = &snd_soc_pm_ops,
 	},
 	.probe = bfin_eval_adau1701_probe,
-	.remove = bfin_eval_adau1701_remove,
 };
 
 module_platform_driver(bfin_eval_adau1701_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c
index 27eee66afdb2..1037477d10b2 100644
--- a/sound/soc/blackfin/bfin-eval-adav80x.c
+++ b/sound/soc/blackfin/bfin-eval-adav80x.c
@@ -119,16 +119,7 @@ static int bfin_eval_adav80x_probe(struct platform_device *pdev)
 
 	card->dev = &pdev->dev;
 
-	return snd_soc_register_card(&bfin_eval_adav80x);
-}
-
-static int bfin_eval_adav80x_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-
-	return 0;
+	return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adav80x);
 }
 
 static const struct platform_device_id bfin_eval_adav80x_ids[] = {
@@ -144,7 +135,6 @@ static struct platform_driver bfin_eval_adav80x_driver = {
 		.pm = &snd_soc_pm_ops,
 	},
 	.probe = bfin_eval_adav80x_probe,
-	.remove = bfin_eval_adav80x_remove,
 	.id_table = bfin_eval_adav80x_ids,
 };
 
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 0c9733ecd17f..cfdafc4c11ea 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AK4104 if SPI_MASTER
 	select SND_SOC_AK4535 if I2C
 	select SND_SOC_AK4554
+	select SND_SOC_AK4613 if I2C
 	select SND_SOC_AK4641 if I2C
 	select SND_SOC_AK4642 if I2C
 	select SND_SOC_AK4671 if I2C
@@ -57,6 +58,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CX20442 if TTY
 	select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_DA7213 if I2C
+	select SND_SOC_DA7219 if I2C
 	select SND_SOC_DA732X if I2C
 	select SND_SOC_DA9055 if I2C
 	select SND_SOC_DMIC
@@ -79,7 +81,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MAX9877 if I2C
 	select SND_SOC_MC13783 if MFD_MC13XXX
 	select SND_SOC_ML26124 if I2C
-	select SND_SOC_HDMI_CODEC
+	select SND_SOC_NAU8825 if I2C
 	select SND_SOC_PCM1681 if I2C
 	select SND_SOC_PCM1792A if SPI_MASTER
 	select SND_SOC_PCM3008
@@ -171,6 +173,7 @@ config SND_SOC_ALL_CODECS
 	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_WM8998 if MFD_WM8998
 	select SND_SOC_WM9081 if I2C
 	select SND_SOC_WM9090 if I2C
 	select SND_SOC_WM9705 if SND_SOC_AC97_BUS
@@ -195,9 +198,11 @@ config SND_SOC_ARIZONA
 	default y if SND_SOC_WM5102=y
 	default y if SND_SOC_WM5110=y
 	default y if SND_SOC_WM8997=y
+	default y if SND_SOC_WM8998=y
 	default m if SND_SOC_WM5102=m
 	default m if SND_SOC_WM5110=m
 	default m if SND_SOC_WM8997=m
+	default m if SND_SOC_WM8998=m
 
 config SND_SOC_WM_HUBS
 	tristate
@@ -319,6 +324,10 @@ config SND_SOC_AK4535
 config SND_SOC_AK4554
 	tristate "AKM AK4554 CODEC"
 
+config SND_SOC_AK4613
+	tristate "AKM AK4613 CODEC"
+	depends on I2C
+
 config SND_SOC_AK4641
 	tristate
 
@@ -430,6 +439,9 @@ config SND_SOC_DA7210
 config SND_SOC_DA7213
         tristate
 
+config SND_SOC_DA7219
+        tristate
+
 config SND_SOC_DA732X
         tristate
 
@@ -442,9 +454,6 @@ config SND_SOC_BT_SCO
 config SND_SOC_DMIC
 	tristate
 
-config SND_SOC_HDMI_CODEC
-       tristate "HDMI stub CODEC"
-
 config SND_SOC_ES8328
 	tristate "Everest Semi ES8328 CODEC"
 
@@ -865,6 +874,9 @@ config SND_SOC_WM8996
 config SND_SOC_WM8997
 	tristate
 
+config SND_SOC_WM8998
+	tristate
+
 config SND_SOC_WM9081
 	tristate
 
@@ -896,6 +908,9 @@ config SND_SOC_MC13783
 config SND_SOC_ML26124
 	tristate
 
+config SND_SOC_NAU8825
+	tristate
+
 config SND_SOC_TPA6130A2
 	tristate "Texas Instruments TPA6130A2 headphone amplifier"
 	depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 4a32077954ae..f632fc42f59f 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -26,6 +26,7 @@ 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-ak4613-objs := ak4613.o
 snd-soc-ak4641-objs := ak4641.o
 snd-soc-ak4642-objs := ak4642.o
 snd-soc-ak4671-objs := ak4671.o
@@ -49,6 +50,7 @@ snd-soc-cs4349-objs := cs4349.o
 snd-soc-cx20442-objs := cx20442.o
 snd-soc-da7210-objs := da7210.o
 snd-soc-da7213-objs := da7213.o
+snd-soc-da7219-objs := da7219.o da7219-aad.o
 snd-soc-da732x-objs := da732x.o
 snd-soc-da9055-objs := da9055.o
 snd-soc-bt-sco-objs := bt-sco.o
@@ -72,7 +74,7 @@ snd-soc-max98925-objs := max98925.o
 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-nau8825-objs := nau8825.o
 snd-soc-pcm1681-objs := pcm1681.o
 snd-soc-pcm1792a-codec-objs := pcm1792a.o
 snd-soc-pcm3008-objs := pcm3008.o
@@ -176,6 +178,7 @@ 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-wm8998-objs := wm8998.o
 snd-soc-wm9081-objs := wm9081.o
 snd-soc-wm9090-objs := wm9090.o
 snd-soc-wm9705-objs := wm9705.o
@@ -216,6 +219,7 @@ 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-$(CONFIG_SND_SOC_AK4613)	+= snd-soc-ak4613.o
 obj-$(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
@@ -241,6 +245,7 @@ obj-$(CONFIG_SND_SOC_CS4349)	+= snd-soc-cs4349.o
 obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
 obj-$(CONFIG_SND_SOC_DA7210)	+= snd-soc-da7210.o
 obj-$(CONFIG_SND_SOC_DA7213)	+= snd-soc-da7213.o
+obj-$(CONFIG_SND_SOC_DA7219)	+= snd-soc-da7219.o
 obj-$(CONFIG_SND_SOC_DA732X)	+= snd-soc-da732x.o
 obj-$(CONFIG_SND_SOC_DA9055)	+= snd-soc-da9055.o
 obj-$(CONFIG_SND_SOC_BT_SCO)	+= snd-soc-bt-sco.o
@@ -264,7 +269,7 @@ obj-$(CONFIG_SND_SOC_MAX98925)	+= snd-soc-max98925.o
 obj-$(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_NAU8825)   += snd-soc-nau8825.o
 obj-$(CONFIG_SND_SOC_PCM1681)	+= snd-soc-pcm1681.o
 obj-$(CONFIG_SND_SOC_PCM1792A)	+= snd-soc-pcm1792a-codec.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
@@ -364,6 +369,7 @@ obj-$(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_WM8998)	+= snd-soc-wm8998.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/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c
index df3a1a415825..171313664bc8 100644
--- a/sound/soc/codecs/ad193x-i2c.c
+++ b/sound/soc/codecs/ad193x-i2c.c
@@ -15,8 +15,8 @@
 #include "ad193x.h"
 
 static const struct i2c_device_id ad193x_id[] = {
-	{ "ad1936", 0 },
-	{ "ad1937", 0 },
+	{ "ad1936", AD193X },
+	{ "ad1937", AD193X },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, ad193x_id);
@@ -30,7 +30,9 @@ static int ad193x_i2c_probe(struct i2c_client *client,
 	config.val_bits = 8;
 	config.reg_bits = 8;
 
-	return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config));
+	return ad193x_probe(&client->dev,
+			    devm_regmap_init_i2c(client, &config),
+			    (enum ad193x_type)id->driver_data);
 }
 
 static int ad193x_i2c_remove(struct i2c_client *client)
diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c
index 390cef9b9dc2..431f95da1de1 100644
--- a/sound/soc/codecs/ad193x-spi.c
+++ b/sound/soc/codecs/ad193x-spi.c
@@ -16,6 +16,7 @@
 
 static int ad193x_spi_probe(struct spi_device *spi)
 {
+	const struct spi_device_id *id = spi_get_device_id(spi);
 	struct regmap_config config;
 
 	config = ad193x_regmap_config;
@@ -24,7 +25,8 @@ static int ad193x_spi_probe(struct spi_device *spi)
 	config.read_flag_mask = 0x09;
 	config.write_flag_mask = 0x08;
 
-	return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config));
+	return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config),
+			    (enum ad193x_type)id->driver_data);
 }
 
 static int ad193x_spi_remove(struct spi_device *spi)
@@ -33,6 +35,17 @@ static int ad193x_spi_remove(struct spi_device *spi)
 	return 0;
 }
 
+static const struct spi_device_id ad193x_spi_id[] = {
+	{ "ad193x", AD193X },
+	{ "ad1933", AD1933 },
+	{ "ad1934", AD1934 },
+	{ "ad1938", AD193X },
+	{ "ad1939", AD193X },
+	{ "adau1328", AD193X },
+	{ }
+};
+MODULE_DEVICE_TABLE(spi, ad193x_spi_id);
+
 static struct spi_driver ad193x_spi_driver = {
 	.driver = {
 		.name	= "ad193x",
@@ -40,6 +53,7 @@ static struct spi_driver ad193x_spi_driver = {
 	},
 	.probe		= ad193x_spi_probe,
 	.remove		= ad193x_spi_remove,
+	.id_table	= ad193x_spi_id,
 };
 module_spi_driver(ad193x_spi_driver);
 
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 17c953595660..3a3f3f2343d7 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -23,6 +23,7 @@
 /* codec private data */
 struct ad193x_priv {
 	struct regmap *regmap;
+	enum ad193x_type type;
 	int sysclk;
 };
 
@@ -47,12 +48,6 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = {
 	SOC_DOUBLE_R_TLV("DAC4 Volume", AD193X_DAC_L4_VOL,
 			AD193X_DAC_R4_VOL, 0, 0xFF, 1, adau193x_tlv),
 
-	/* ADC switch control */
-	SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE,
-		AD193X_ADCR1_MUTE, 1, 1),
-	SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE,
-		AD193X_ADCR2_MUTE, 1, 1),
-
 	/* DAC switch control */
 	SOC_DOUBLE("DAC1 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL1_MUTE,
 		AD193X_DACR1_MUTE, 1, 1),
@@ -63,26 +58,37 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = {
 	SOC_DOUBLE("DAC4 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL4_MUTE,
 		AD193X_DACR4_MUTE, 1, 1),
 
+	/* DAC de-emphasis */
+	SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum),
+};
+
+static const struct snd_kcontrol_new ad193x_adc_snd_controls[] = {
+	/* ADC switch control */
+	SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE,
+		AD193X_ADCR1_MUTE, 1, 1),
+	SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE,
+		AD193X_ADCR2_MUTE, 1, 1),
+
 	/* ADC high-pass filter */
 	SOC_SINGLE("ADC High Pass Filter Switch", AD193X_ADC_CTRL0,
 			AD193X_ADC_HIGHPASS_FILTER, 1, 0),
-
-	/* DAC de-emphasis */
-	SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum),
 };
 
 static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = {
 	SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0),
-	SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0),
-	SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0),
 	SND_SOC_DAPM_VMID("VMID"),
 	SND_SOC_DAPM_OUTPUT("DAC1OUT"),
 	SND_SOC_DAPM_OUTPUT("DAC2OUT"),
 	SND_SOC_DAPM_OUTPUT("DAC3OUT"),
 	SND_SOC_DAPM_OUTPUT("DAC4OUT"),
+};
+
+static const struct snd_soc_dapm_widget ad193x_adc_widgets[] = {
+	SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0),
 	SND_SOC_DAPM_INPUT("ADC1IN"),
 	SND_SOC_DAPM_INPUT("ADC2IN"),
 };
@@ -91,18 +97,33 @@ static const struct snd_soc_dapm_route audio_paths[] = {
 	{ "DAC", NULL, "SYSCLK" },
 	{ "DAC Output", NULL, "DAC" },
 	{ "DAC Output", NULL, "VMID" },
-	{ "ADC", NULL, "SYSCLK" },
-	{ "DAC", NULL, "ADC_PWR" },
-	{ "ADC", NULL, "ADC_PWR" },
 	{ "DAC1OUT", NULL, "DAC Output" },
 	{ "DAC2OUT", NULL, "DAC Output" },
 	{ "DAC3OUT", NULL, "DAC Output" },
 	{ "DAC4OUT", NULL, "DAC Output" },
+	{ "SYSCLK", NULL, "PLL_PWR" },
+};
+
+static const struct snd_soc_dapm_route ad193x_adc_audio_paths[] = {
+	{ "ADC", NULL, "SYSCLK" },
+	{ "ADC", NULL, "ADC_PWR" },
 	{ "ADC", NULL, "ADC1IN" },
 	{ "ADC", NULL, "ADC2IN" },
-	{ "SYSCLK", NULL, "PLL_PWR" },
 };
 
+static inline bool ad193x_has_adc(const struct ad193x_priv *ad193x)
+{
+	switch (ad193x->type) {
+	case AD1933:
+	case AD1934:
+		return false;
+	default:
+		break;
+	}
+
+	return true;
+}
+
 /*
  * DAI ops entries
  */
@@ -147,8 +168,10 @@ static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 
 	regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1,
 		AD193X_DAC_CHAN_MASK, channels << AD193X_DAC_CHAN_SHFT);
-	regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2,
-		AD193X_ADC_CHAN_MASK, channels << AD193X_ADC_CHAN_SHFT);
+	if (ad193x_has_adc(ad193x))
+		regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2,
+				   AD193X_ADC_CHAN_MASK,
+				   channels << AD193X_ADC_CHAN_SHFT);
 
 	return 0;
 }
@@ -172,7 +195,9 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		adc_serfmt |= AD193X_ADC_SERFMT_AUX;
 		break;
 	default:
-		return -EINVAL;
+		if (ad193x_has_adc(ad193x))
+			return -EINVAL;
+		break;
 	}
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
@@ -217,10 +242,12 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		return -EINVAL;
 	}
 
-	regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1,
-		AD193X_ADC_SERFMT_MASK, adc_serfmt);
-	regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2,
-		AD193X_ADC_FMT_MASK, adc_fmt);
+	if (ad193x_has_adc(ad193x)) {
+		regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1,
+				   AD193X_ADC_SERFMT_MASK, adc_serfmt);
+		regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2,
+				   AD193X_ADC_FMT_MASK, adc_fmt);
+	}
 	regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1,
 		AD193X_DAC_FMT_MASK, dac_fmt);
 
@@ -287,8 +314,9 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
 			    AD193X_DAC_WORD_LEN_MASK,
 			    word_len << AD193X_DAC_WORD_LEN_SHFT);
 
-	regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1,
-			    AD193X_ADC_WORD_LEN_MASK, word_len);
+	if (ad193x_has_adc(ad193x))
+		regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1,
+				   AD193X_ADC_WORD_LEN_MASK, word_len);
 
 	return 0;
 }
@@ -326,6 +354,8 @@ static struct snd_soc_dai_driver ad193x_dai = {
 static int ad193x_codec_probe(struct snd_soc_codec *codec)
 {
 	struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+	int num, ret;
 
 	/* default setting for ad193x */
 
@@ -335,14 +365,46 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec)
 	regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A);
 	/* dac in tdm mode */
 	regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40);
-	/* high-pass filter enable */
-	regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3);
-	/* sata delay=1, adc aux mode */
-	regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43);
+
+	/* adc only */
+	if (ad193x_has_adc(ad193x)) {
+		/* high-pass filter enable */
+		regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3);
+		/* sata delay=1, adc aux mode */
+		regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43);
+	}
+
 	/* pll input: mclki/xi */
 	regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */
 	regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04);
 
+	/* adc only */
+	if (ad193x_has_adc(ad193x)) {
+		/* add adc controls */
+		num = ARRAY_SIZE(ad193x_adc_snd_controls);
+		ret = snd_soc_add_codec_controls(codec,
+						 ad193x_adc_snd_controls,
+						 num);
+		if (ret)
+			return ret;
+
+		/* add adc widgets */
+		num = ARRAY_SIZE(ad193x_adc_widgets);
+		ret = snd_soc_dapm_new_controls(dapm,
+						ad193x_adc_widgets,
+						num);
+		if (ret)
+			return ret;
+
+		/* add adc routes */
+		num = ARRAY_SIZE(ad193x_adc_audio_paths);
+		ret = snd_soc_dapm_add_routes(dapm,
+					      ad193x_adc_audio_paths,
+					      num);
+		if (ret)
+			return ret;
+	}
+
 	return 0;
 }
 
@@ -356,18 +418,13 @@ static struct snd_soc_codec_driver soc_codec_dev_ad193x = {
 	.num_dapm_routes = ARRAY_SIZE(audio_paths),
 };
 
-static bool adau193x_reg_volatile(struct device *dev, unsigned int reg)
-{
-	return false;
-}
-
 const struct regmap_config ad193x_regmap_config = {
 	.max_register = AD193X_NUM_REGS - 1,
-	.volatile_reg = adau193x_reg_volatile,
 };
 EXPORT_SYMBOL_GPL(ad193x_regmap_config);
 
-int ad193x_probe(struct device *dev, struct regmap *regmap)
+int ad193x_probe(struct device *dev, struct regmap *regmap,
+		 enum ad193x_type type)
 {
 	struct ad193x_priv *ad193x;
 
@@ -379,6 +436,7 @@ int ad193x_probe(struct device *dev, struct regmap *regmap)
 		return -ENOMEM;
 
 	ad193x->regmap = regmap;
+	ad193x->type = type;
 
 	dev_set_drvdata(dev, ad193x);
 
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index ab9a998f15be..8b1e65f928d2 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -13,8 +13,15 @@
 
 struct device;
 
+enum ad193x_type {
+	AD193X,
+	AD1933,
+	AD1934,
+};
+
 extern const struct regmap_config ad193x_regmap_config;
-int ad193x_probe(struct device *dev, struct regmap *regmap);
+int ad193x_probe(struct device *dev, struct regmap *regmap,
+		 enum ad193x_type type);
 
 #define AD193X_PLL_CLK_CTRL0    0x00
 #define AD193X_PLL_POWERDOWN           0x01
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index 198c924551b7..acff8d62059c 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -728,8 +728,8 @@ static int adav80x_dai_startup(struct snd_pcm_substream *substream,
 	if (!snd_soc_codec_is_active(codec) || !adav80x->rate)
 		return 0;
 
-	return snd_pcm_hw_constraint_minmax(substream->runtime,
-			SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate);
+	return snd_pcm_hw_constraint_single(substream->runtime,
+			SNDRV_PCM_HW_PARAM_RATE, adav80x->rate);
 }
 
 static void adav80x_dai_shutdown(struct snd_pcm_substream *substream,
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c
new file mode 100644
index 000000000000..07a266460ec3
--- /dev/null
+++ b/sound/soc/codecs/ak4613.c
@@ -0,0 +1,497 @@
+/*
+ * ak4613.c  --  Asahi Kasei ALSA Soc Audio driver
+ *
+ * Copyright (C) 2015 Renesas Electronics Corporation
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * Based on ak4642.c by Kuninori Morimoto
+ * Based on wm8731.c by Richard Purdie
+ * Based on ak4535.c by Richard Purdie
+ * Based on wm8753.c by Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/of_device.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+
+#define PW_MGMT1	0x00 /* Power Management 1 */
+#define PW_MGMT2	0x01 /* Power Management 2 */
+#define PW_MGMT3	0x02 /* Power Management 3 */
+#define CTRL1		0x03 /* Control 1 */
+#define CTRL2		0x04 /* Control 2 */
+#define DEMP1		0x05 /* De-emphasis1 */
+#define DEMP2		0x06 /* De-emphasis2 */
+#define OFD		0x07 /* Overflow Detect */
+#define ZRD		0x08 /* Zero Detect */
+#define ICTRL		0x09 /* Input Control */
+#define OCTRL		0x0a /* Output Control */
+#define LOUT1		0x0b /* LOUT1 Volume Control */
+#define ROUT1		0x0c /* ROUT1 Volume Control */
+#define LOUT2		0x0d /* LOUT2 Volume Control */
+#define ROUT2		0x0e /* ROUT2 Volume Control */
+#define LOUT3		0x0f /* LOUT3 Volume Control */
+#define ROUT3		0x10 /* ROUT3 Volume Control */
+#define LOUT4		0x11 /* LOUT4 Volume Control */
+#define ROUT4		0x12 /* ROUT4 Volume Control */
+#define LOUT5		0x13 /* LOUT5 Volume Control */
+#define ROUT5		0x14 /* ROUT5 Volume Control */
+#define LOUT6		0x15 /* LOUT6 Volume Control */
+#define ROUT6		0x16 /* ROUT6 Volume Control */
+
+/* PW_MGMT1 */
+#define RSTN		BIT(0)
+#define PMDAC		BIT(1)
+#define PMADC		BIT(2)
+#define PMVR		BIT(3)
+
+/* PW_MGMT2 */
+#define PMAD_ALL	0x7
+
+/* PW_MGMT3 */
+#define PMDA_ALL	0x3f
+
+/* CTRL1 */
+#define DIF0		BIT(3)
+#define DIF1		BIT(4)
+#define DIF2		BIT(5)
+#define TDM0		BIT(6)
+#define TDM1		BIT(7)
+#define NO_FMT		(0xff)
+#define FMT_MASK	(0xf8)
+
+/* CTRL2 */
+#define DFS_NORMAL_SPEED	(0 << 2)
+#define DFS_DOUBLE_SPEED	(1 << 2)
+#define DFS_QUAD_SPEED		(2 << 2)
+
+struct ak4613_priv {
+	struct mutex lock;
+
+	unsigned int fmt;
+	u8 fmt_ctrl;
+	int cnt;
+};
+
+struct ak4613_formats {
+	unsigned int width;
+	unsigned int fmt;
+};
+
+struct ak4613_interface {
+	struct ak4613_formats capture;
+	struct ak4613_formats playback;
+};
+
+/*
+ * Playback Volume
+ *
+ * max : 0x00 : 0 dB
+ *       ( 0.5 dB step )
+ * min : 0xFE : -127.0 dB
+ * mute: 0xFF
+ */
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12750, 50, 1);
+
+static const struct snd_kcontrol_new ak4613_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("Digital Playback Volume1", LOUT1, ROUT1,
+			 0, 0xFF, 1, out_tlv),
+	SOC_DOUBLE_R_TLV("Digital Playback Volume2", LOUT2, ROUT2,
+			 0, 0xFF, 1, out_tlv),
+	SOC_DOUBLE_R_TLV("Digital Playback Volume3", LOUT3, ROUT3,
+			 0, 0xFF, 1, out_tlv),
+	SOC_DOUBLE_R_TLV("Digital Playback Volume4", LOUT4, ROUT4,
+			 0, 0xFF, 1, out_tlv),
+	SOC_DOUBLE_R_TLV("Digital Playback Volume5", LOUT5, ROUT5,
+			 0, 0xFF, 1, out_tlv),
+	SOC_DOUBLE_R_TLV("Digital Playback Volume6", LOUT6, ROUT6,
+			 0, 0xFF, 1, out_tlv),
+};
+
+static const struct reg_default ak4613_reg[] = {
+	{ 0x0,  0x0f }, { 0x1,  0x07 }, { 0x2,  0x3f }, { 0x3,  0x20 },
+	{ 0x4,  0x20 }, { 0x5,  0x55 }, { 0x6,  0x05 }, { 0x7,  0x07 },
+	{ 0x8,  0x0f }, { 0x9,  0x07 }, { 0xa,  0x3f }, { 0xb,  0x00 },
+	{ 0xc,  0x00 }, { 0xd,  0x00 }, { 0xe,  0x00 }, { 0xf,  0x00 },
+	{ 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, { 0x13, 0x00 },
+	{ 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 },
+};
+
+#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3)
+#define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt }
+static const struct ak4613_interface ak4613_iface[] = {
+	/* capture */				/* playback */
+	[0] = {	AUDIO_IFACE(24, LEFT_J),	AUDIO_IFACE(16, RIGHT_J) },
+	[1] = {	AUDIO_IFACE(24, LEFT_J),	AUDIO_IFACE(20, RIGHT_J) },
+	[2] = {	AUDIO_IFACE(24, LEFT_J),	AUDIO_IFACE(24, RIGHT_J) },
+	[3] = {	AUDIO_IFACE(24, LEFT_J),	AUDIO_IFACE(24, LEFT_J) },
+	[4] = {	AUDIO_IFACE(24, I2S),		AUDIO_IFACE(24, I2S) },
+};
+
+static const struct regmap_config ak4613_regmap_cfg = {
+	.reg_bits		= 8,
+	.val_bits		= 8,
+	.max_register		= 0x16,
+	.reg_defaults		= ak4613_reg,
+	.num_reg_defaults	= ARRAY_SIZE(ak4613_reg),
+};
+
+static const struct of_device_id ak4613_of_match[] = {
+	{ .compatible = "asahi-kasei,ak4613",	.data = &ak4613_regmap_cfg },
+	{},
+};
+MODULE_DEVICE_TABLE(of, ak4613_of_match);
+
+static const struct i2c_device_id ak4613_i2c_id[] = {
+	{ "ak4613", (kernel_ulong_t)&ak4613_regmap_cfg },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, ak4613_i2c_id);
+
+static const struct snd_soc_dapm_widget ak4613_dapm_widgets[] = {
+
+	/* Outputs */
+	SND_SOC_DAPM_OUTPUT("LOUT1"),
+	SND_SOC_DAPM_OUTPUT("LOUT2"),
+	SND_SOC_DAPM_OUTPUT("LOUT3"),
+	SND_SOC_DAPM_OUTPUT("LOUT4"),
+	SND_SOC_DAPM_OUTPUT("LOUT5"),
+	SND_SOC_DAPM_OUTPUT("LOUT6"),
+
+	SND_SOC_DAPM_OUTPUT("ROUT1"),
+	SND_SOC_DAPM_OUTPUT("ROUT2"),
+	SND_SOC_DAPM_OUTPUT("ROUT3"),
+	SND_SOC_DAPM_OUTPUT("ROUT4"),
+	SND_SOC_DAPM_OUTPUT("ROUT5"),
+	SND_SOC_DAPM_OUTPUT("ROUT6"),
+
+	/* Inputs */
+	SND_SOC_DAPM_INPUT("LIN1"),
+	SND_SOC_DAPM_INPUT("LIN2"),
+
+	SND_SOC_DAPM_INPUT("RIN1"),
+	SND_SOC_DAPM_INPUT("RIN2"),
+
+	/* DAC */
+	SND_SOC_DAPM_DAC("DAC1", NULL, PW_MGMT3, 0, 0),
+	SND_SOC_DAPM_DAC("DAC2", NULL, PW_MGMT3, 1, 0),
+	SND_SOC_DAPM_DAC("DAC3", NULL, PW_MGMT3, 2, 0),
+	SND_SOC_DAPM_DAC("DAC4", NULL, PW_MGMT3, 3, 0),
+	SND_SOC_DAPM_DAC("DAC5", NULL, PW_MGMT3, 4, 0),
+	SND_SOC_DAPM_DAC("DAC6", NULL, PW_MGMT3, 5, 0),
+
+	/* ADC */
+	SND_SOC_DAPM_ADC("ADC1", NULL, PW_MGMT2, 0, 0),
+	SND_SOC_DAPM_ADC("ADC2", NULL, PW_MGMT2, 1, 0),
+};
+
+static const struct snd_soc_dapm_route ak4613_intercon[] = {
+	{"LOUT1", NULL, "DAC1"},
+	{"LOUT2", NULL, "DAC2"},
+	{"LOUT3", NULL, "DAC3"},
+	{"LOUT4", NULL, "DAC4"},
+	{"LOUT5", NULL, "DAC5"},
+	{"LOUT6", NULL, "DAC6"},
+
+	{"ROUT1", NULL, "DAC1"},
+	{"ROUT2", NULL, "DAC2"},
+	{"ROUT3", NULL, "DAC3"},
+	{"ROUT4", NULL, "DAC4"},
+	{"ROUT5", NULL, "DAC5"},
+	{"ROUT6", NULL, "DAC6"},
+
+	{"DAC1", NULL, "Playback"},
+	{"DAC2", NULL, "Playback"},
+	{"DAC3", NULL, "Playback"},
+	{"DAC4", NULL, "Playback"},
+	{"DAC5", NULL, "Playback"},
+	{"DAC6", NULL, "Playback"},
+
+	{"Capture", NULL, "ADC1"},
+	{"Capture", NULL, "ADC2"},
+
+	{"ADC1", NULL, "LIN1"},
+	{"ADC2", NULL, "LIN2"},
+
+	{"ADC1", NULL, "RIN1"},
+	{"ADC2", NULL, "RIN2"},
+};
+
+static void ak4613_dai_shutdown(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct device *dev = codec->dev;
+
+	mutex_lock(&priv->lock);
+	priv->cnt--;
+	if (priv->cnt < 0) {
+		dev_err(dev, "unexpected counter error\n");
+		priv->cnt = 0;
+	}
+	if (!priv->cnt)
+		priv->fmt_ctrl = NO_FMT;
+	mutex_unlock(&priv->lock);
+}
+
+static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	fmt &= SND_SOC_DAIFMT_FORMAT_MASK;
+
+	switch (fmt) {
+	case SND_SOC_DAIFMT_RIGHT_J:
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_I2S:
+		priv->fmt = fmt;
+
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int ak4613_dai_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 ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+	const struct ak4613_formats *fmts;
+	struct device *dev = codec->dev;
+	unsigned int width = params_width(params);
+	unsigned int fmt = priv->fmt;
+	unsigned int rate;
+	int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	int i, ret;
+	u8 fmt_ctrl, ctrl2;
+
+	rate = params_rate(params);
+	switch (rate) {
+	case 32000:
+	case 44100:
+	case 48000:
+		ctrl2 = DFS_NORMAL_SPEED;
+		break;
+	case 88200:
+	case 96000:
+		ctrl2 = DFS_DOUBLE_SPEED;
+		break;
+	case 176400:
+	case 192000:
+		ctrl2 = DFS_QUAD_SPEED;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/*
+	 * FIXME
+	 *
+	 * It doesn't support TDM at this point
+	 */
+	fmt_ctrl = NO_FMT;
+	for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) {
+		fmts = (is_play) ?	&ak4613_iface[i].playback :
+					&ak4613_iface[i].capture;
+
+		if (fmts->fmt != fmt)
+			continue;
+
+		if (fmt == SND_SOC_DAIFMT_RIGHT_J) {
+			if (fmts->width != width)
+				continue;
+		} else {
+			if (fmts->width < width)
+				continue;
+		}
+
+		fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i);
+		break;
+	}
+
+	ret = -EINVAL;
+	if (fmt_ctrl == NO_FMT)
+		goto hw_params_end;
+
+	mutex_lock(&priv->lock);
+	if ((priv->fmt_ctrl == NO_FMT) ||
+	    (priv->fmt_ctrl == fmt_ctrl)) {
+		priv->fmt_ctrl = fmt_ctrl;
+		priv->cnt++;
+		ret = 0;
+	}
+	mutex_unlock(&priv->lock);
+
+	if (ret < 0)
+		goto hw_params_end;
+
+	snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl);
+	snd_soc_write(codec, CTRL2, ctrl2);
+
+hw_params_end:
+	if (ret < 0)
+		dev_warn(dev, "unsupported data width/format combination\n");
+
+	return ret;
+}
+
+static int ak4613_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	u8 mgmt1 = 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		mgmt1 |= RSTN;
+		/* fall through */
+	case SND_SOC_BIAS_PREPARE:
+		mgmt1 |= PMADC | PMDAC;
+		/* fall through */
+	case SND_SOC_BIAS_STANDBY:
+		mgmt1 |= PMVR;
+		/* fall through */
+	case SND_SOC_BIAS_OFF:
+	default:
+		break;
+	}
+
+	snd_soc_write(codec, PW_MGMT1, mgmt1);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops ak4613_dai_ops = {
+	.shutdown	= ak4613_dai_shutdown,
+	.set_fmt	= ak4613_dai_set_fmt,
+	.hw_params	= ak4613_dai_hw_params,
+};
+
+#define AK4613_PCM_RATE		(SNDRV_PCM_RATE_32000  |\
+				 SNDRV_PCM_RATE_44100  |\
+				 SNDRV_PCM_RATE_48000  |\
+				 SNDRV_PCM_RATE_64000  |\
+				 SNDRV_PCM_RATE_88200  |\
+				 SNDRV_PCM_RATE_96000  |\
+				 SNDRV_PCM_RATE_176400 |\
+				 SNDRV_PCM_RATE_192000)
+#define AK4613_PCM_FMTBIT	(SNDRV_PCM_FMTBIT_S16_LE |\
+				 SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_driver ak4613_dai = {
+	.name = "ak4613-hifi",
+	.playback = {
+		.stream_name	= "Playback",
+		.channels_min	= 2,
+		.channels_max	= 2,
+		.rates		= AK4613_PCM_RATE,
+		.formats	= AK4613_PCM_FMTBIT,
+	},
+	.capture = {
+		.stream_name	= "Capture",
+		.channels_min	= 2,
+		.channels_max	= 2,
+		.rates		= AK4613_PCM_RATE,
+		.formats	= AK4613_PCM_FMTBIT,
+	},
+	.ops = &ak4613_dai_ops,
+	.symmetric_rates = 1,
+};
+
+static int ak4613_resume(struct snd_soc_codec *codec)
+{
+	struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+
+	regcache_mark_dirty(regmap);
+	return regcache_sync(regmap);
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_ak4613 = {
+	.resume			= ak4613_resume,
+	.set_bias_level		= ak4613_set_bias_level,
+	.controls		= ak4613_snd_controls,
+	.num_controls		= ARRAY_SIZE(ak4613_snd_controls),
+	.dapm_widgets		= ak4613_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(ak4613_dapm_widgets),
+	.dapm_routes		= ak4613_intercon,
+	.num_dapm_routes	= ARRAY_SIZE(ak4613_intercon),
+};
+
+static int ak4613_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	struct device *dev = &i2c->dev;
+	struct device_node *np = dev->of_node;
+	const struct regmap_config *regmap_cfg;
+	struct regmap *regmap;
+	struct ak4613_priv *priv;
+
+	regmap_cfg = NULL;
+	if (np) {
+		const struct of_device_id *of_id;
+
+		of_id = of_match_device(ak4613_of_match, dev);
+		if (of_id)
+			regmap_cfg = of_id->data;
+	} else {
+		regmap_cfg = (const struct regmap_config *)id->driver_data;
+	}
+
+	if (!regmap_cfg)
+		return -EINVAL;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->fmt_ctrl		= NO_FMT;
+	priv->cnt		= 0;
+
+	mutex_init(&priv->lock);
+
+	i2c_set_clientdata(i2c, priv);
+
+	regmap = devm_regmap_init_i2c(i2c, regmap_cfg);
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
+
+	return snd_soc_register_codec(dev, &soc_codec_dev_ak4613,
+				      &ak4613_dai, 1);
+}
+
+static int ak4613_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	return 0;
+}
+
+static struct i2c_driver ak4613_i2c_driver = {
+	.driver = {
+		.name = "ak4613-codec",
+		.owner = THIS_MODULE,
+		.of_match_table = ak4613_of_match,
+	},
+	.probe		= ak4613_i2c_probe,
+	.remove		= ak4613_i2c_remove,
+	.id_table	= ak4613_i2c_id,
+};
+
+module_i2c_driver(ak4613_i2c_driver);
+
+MODULE_DESCRIPTION("Soc AK4613 driver");
+MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4a90143d0e90..cda27c22812a 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -23,6 +23,8 @@
  * AK4648 is tested.
  */
 
+#include <linux/clk.h>
+#include <linux/clk-provider.h>
 #include <linux/delay.h>
 #include <linux/i2c.h>
 #include <linux/slab.h>
@@ -128,11 +130,8 @@
 #define I2S		(3 << 0)
 
 /* MD_CTL2 */
-#define FS0		(1 << 0)
-#define FS1		(1 << 1)
-#define FS2		(1 << 2)
-#define FS3		(1 << 5)
-#define FS_MASK		(FS0 | FS1 | FS2 | FS3)
+#define FSs(val)	(((val & 0x7) << 0) | ((val & 0x8) << 2))
+#define PSs(val)	((val & 0x3) << 6)
 
 /* MD_CTL3 */
 #define BST1		(1 << 3)
@@ -147,6 +146,7 @@ struct ak4642_drvdata {
 
 struct ak4642_priv {
 	const struct ak4642_drvdata *drvdata;
+	struct clk *mcko;
 };
 
 /*
@@ -430,56 +430,56 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
+static int ak4642_set_mcko(struct snd_soc_codec *codec,
+			   u32 frequency)
+{
+	u32 fs_list[] = {
+		[0] = 8000,
+		[1] = 12000,
+		[2] = 16000,
+		[3] = 24000,
+		[4] = 7350,
+		[5] = 11025,
+		[6] = 14700,
+		[7] = 22050,
+		[10] = 32000,
+		[11] = 48000,
+		[14] = 29400,
+		[15] = 44100,
+	};
+	u32 ps_list[] = {
+		[0] = 256,
+		[1] = 128,
+		[2] = 64,
+		[3] = 32
+	};
+	int ps, fs;
+
+	for (ps = 0; ps < ARRAY_SIZE(ps_list); ps++) {
+		for (fs = 0; fs < ARRAY_SIZE(fs_list); fs++) {
+			if (frequency == ps_list[ps] * fs_list[fs]) {
+				snd_soc_write(codec, MD_CTL2,
+					      PSs(ps) | FSs(fs));
+				return 0;
+			}
+		}
+	}
+
+	return 0;
+}
+
 static int ak4642_dai_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	u8 rate;
+	struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
+	u32 rate = clk_get_rate(priv->mcko);
 
-	switch (params_rate(params)) {
-	case 7350:
-		rate = FS2;
-		break;
-	case 8000:
-		rate = 0;
-		break;
-	case 11025:
-		rate = FS2 | FS0;
-		break;
-	case 12000:
-		rate = FS0;
-		break;
-	case 14700:
-		rate = FS2 | FS1;
-		break;
-	case 16000:
-		rate = FS1;
-		break;
-	case 22050:
-		rate = FS2 | FS1 | FS0;
-		break;
-	case 24000:
-		rate = FS1 | FS0;
-		break;
-	case 29400:
-		rate = FS3 | FS2 | FS1;
-		break;
-	case 32000:
-		rate = FS3 | FS1;
-		break;
-	case 44100:
-		rate = FS3 | FS2 | FS1 | FS0;
-		break;
-	case 48000:
-		rate = FS3 | FS1 | FS0;
-		break;
-	default:
-		return -EINVAL;
-	}
-	snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
+	if (!rate)
+		rate = params_rate(params) * 256;
 
-	return 0;
+	return ak4642_set_mcko(codec, rate);
 }
 
 static int ak4642_set_bias_level(struct snd_soc_codec *codec,
@@ -532,7 +532,18 @@ static int ak4642_resume(struct snd_soc_codec *codec)
 	return 0;
 }
 
+static int ak4642_probe(struct snd_soc_codec *codec)
+{
+	struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	if (priv->mcko)
+		ak4642_set_mcko(codec, clk_get_rate(priv->mcko));
+
+	return 0;
+}
+
 static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
+	.probe			= ak4642_probe,
 	.resume			= ak4642_resume,
 	.set_bias_level		= ak4642_set_bias_level,
 	.controls		= ak4642_snd_controls,
@@ -580,19 +591,54 @@ static const struct ak4642_drvdata ak4648_drvdata = {
 	.extended_frequencies = 1,
 };
 
+#ifdef CONFIG_COMMON_CLK
+static struct clk *ak4642_of_parse_mcko(struct device *dev)
+{
+	struct device_node *np = dev->of_node;
+	struct clk *clk;
+	const char *clk_name = np->name;
+	const char *parent_clk_name = NULL;
+	u32 rate;
+
+	if (of_property_read_u32(np, "clock-frequency", &rate))
+		return NULL;
+
+	if (of_property_read_bool(np, "clocks"))
+		parent_clk_name = of_clk_get_parent_name(np, 0);
+
+	of_property_read_string(np, "clock-output-names", &clk_name);
+
+	clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name,
+				      (parent_clk_name) ? 0 : CLK_IS_ROOT,
+				      rate);
+	if (!IS_ERR(clk))
+		of_clk_add_provider(np, of_clk_src_simple_get, clk);
+
+	return clk;
+}
+#else
+#define ak4642_of_parse_mcko(d) 0
+#endif
+
 static const struct of_device_id ak4642_of_match[];
 static int ak4642_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-	struct device_node *np = i2c->dev.of_node;
+	struct device *dev = &i2c->dev;
+	struct device_node *np = dev->of_node;
 	const struct ak4642_drvdata *drvdata = NULL;
 	struct regmap *regmap;
 	struct ak4642_priv *priv;
+	struct clk *mcko = NULL;
 
 	if (np) {
 		const struct of_device_id *of_id;
 
-		of_id = of_match_device(ak4642_of_match, &i2c->dev);
+		mcko = ak4642_of_parse_mcko(dev);
+		if (IS_ERR(mcko))
+			mcko = NULL;
+
+		of_id = of_match_device(ak4642_of_match, dev);
 		if (of_id)
 			drvdata = of_id->data;
 	} else {
@@ -600,15 +646,16 @@ static int ak4642_i2c_probe(struct i2c_client *i2c,
 	}
 
 	if (!drvdata) {
-		dev_err(&i2c->dev, "Unknown device type\n");
+		dev_err(dev, "Unknown device type\n");
 		return -EINVAL;
 	}
 
-	priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
 	if (!priv)
 		return -ENOMEM;
 
 	priv->drvdata = drvdata;
+	priv->mcko = mcko;
 
 	i2c_set_clientdata(i2c, priv);
 
@@ -616,7 +663,7 @@ static int ak4642_i2c_probe(struct i2c_client *i2c,
 	if (IS_ERR(regmap))
 		return PTR_ERR(regmap);
 
-	return snd_soc_register_codec(&i2c->dev,
+	return snd_soc_register_codec(dev,
 				      &soc_codec_dev_ak4642, &ak4642_dai, 1);
 }
 
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 8a2221ab3d10..9929efc6b9aa 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -147,6 +147,8 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w,
 						   0x4f5, 0x0da);
 		}
 		break;
+	default:
+		break;
 	}
 
 	return 0;
@@ -314,6 +316,7 @@ const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
 	"Tone Generator 2",
 	"Haptics",
 	"AEC",
+	"AEC2",
 	"Mic Mute Mixer",
 	"Noise Generator",
 	"IN1L",
@@ -421,6 +424,7 @@ int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = {
 	0x05,
 	0x06,  /* Haptics */
 	0x08,  /* AEC */
+	0x09,  /* AEC2 */
 	0x0c,  /* Noise mixer */
 	0x0d,  /* Comfort noise */
 	0x10,  /* IN1L */
@@ -525,6 +529,32 @@ EXPORT_SYMBOL_GPL(arizona_mixer_values);
 const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0);
 EXPORT_SYMBOL_GPL(arizona_mixer_tlv);
 
+const char * const arizona_sample_rate_text[ARIZONA_SAMPLE_RATE_ENUM_SIZE] = {
+	"12kHz", "24kHz", "48kHz", "96kHz", "192kHz",
+	"11.025kHz", "22.05kHz", "44.1kHz", "88.2kHz", "176.4kHz",
+	"4kHz", "8kHz", "16kHz", "32kHz",
+};
+EXPORT_SYMBOL_GPL(arizona_sample_rate_text);
+
+const unsigned int arizona_sample_rate_val[ARIZONA_SAMPLE_RATE_ENUM_SIZE] = {
+	0x01, 0x02, 0x03, 0x04, 0x05, 0x09, 0x0A, 0x0B, 0x0C, 0x0D,
+	0x10, 0x11, 0x12, 0x13,
+};
+EXPORT_SYMBOL_GPL(arizona_sample_rate_val);
+
+const char *arizona_sample_rate_val_to_name(unsigned int rate_val)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(arizona_sample_rate_val); ++i) {
+		if (arizona_sample_rate_val[i] == rate_val)
+			return arizona_sample_rate_text[i];
+	}
+
+	return "Illegal";
+}
+EXPORT_SYMBOL_GPL(arizona_sample_rate_val_to_name);
+
 const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE] = {
 	"SYNCCLK rate", "8kHz", "16kHz", "ASYNCCLK rate",
 };
@@ -689,6 +719,15 @@ static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena)
 				    ARIZONA_IN_VU, val);
 }
 
+bool arizona_input_analog(struct snd_soc_codec *codec, int shift)
+{
+	unsigned int reg = ARIZONA_IN1L_CONTROL + ((shift / 2) * 8);
+	unsigned int val = snd_soc_read(codec, reg);
+
+	return !(val & ARIZONA_IN1_MODE_MASK);
+}
+EXPORT_SYMBOL_GPL(arizona_input_analog);
+
 int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
 		  int event)
 {
@@ -725,6 +764,9 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
 		reg = snd_soc_read(codec, ARIZONA_INPUT_ENABLES);
 		if (reg == 0)
 			arizona_in_set_vu(codec, 0);
+		break;
+	default:
+		break;
 	}
 
 	return 0;
@@ -806,6 +848,8 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w,
 			break;
 		}
 		break;
+	default:
+		break;
 	}
 
 	return 0;
@@ -1868,6 +1912,11 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
 		if (fll->arizona->rev < 3 || sync)
 			return init_ratio;
 		break;
+	case WM8998:
+	case WM1814:
+		if (sync)
+			return init_ratio;
+		break;
 	default:
 		return init_ratio;
 	}
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index ada0a418ff4b..fea8b8ae8e1a 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -93,12 +93,17 @@ struct arizona_priv {
 	bool dvfs_cached;
 };
 
-#define ARIZONA_NUM_MIXER_INPUTS 103
+#define ARIZONA_NUM_MIXER_INPUTS 104
 
 extern const unsigned int arizona_mixer_tlv[];
 extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS];
 extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
 
+#define ARIZONA_GAINMUX_CONTROLS(name, base) \
+	SOC_SINGLE_RANGE_TLV(name " Input Volume", base + 1,		\
+			     ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0,	\
+			     arizona_mixer_tlv)
+
 #define ARIZONA_MIXER_CONTROLS(name, base) \
 	SOC_SINGLE_RANGE_TLV(name " Input 1 Volume", base + 1,		\
 			     ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0,	\
@@ -209,8 +214,12 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
 	 .num_regs = 1 }) }
 
 #define ARIZONA_RATE_ENUM_SIZE 4
+#define ARIZONA_SAMPLE_RATE_ENUM_SIZE 14
+
 extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE];
 extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE];
+extern const char * const arizona_sample_rate_text[ARIZONA_SAMPLE_RATE_ENUM_SIZE];
+extern const unsigned int arizona_sample_rate_val[ARIZONA_SAMPLE_RATE_ENUM_SIZE];
 
 extern const struct soc_enum arizona_isrc_fsl[];
 extern const struct soc_enum arizona_isrc_fsh[];
@@ -294,4 +303,7 @@ extern int arizona_init_dai(struct arizona_priv *priv, int dai);
 int arizona_set_output_mode(struct snd_soc_codec *codec, int output,
 			    bool diff);
 
+extern bool arizona_input_analog(struct snd_soc_codec *codec, int shift);
+
+extern const char *arizona_sample_rate_val_to_name(unsigned int rate_val);
 #endif
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index a9c86efb3187..7278f93460c1 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -12,6 +12,7 @@
  * option) any later version.
  */
 
+#include <linux/clk.h>
 #include <linux/delay.h>
 #include <linux/i2c.h>
 #include <linux/regmap.h>
@@ -1222,23 +1223,44 @@ static int da7213_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	if ((da7213->clk_src == clk_id) && (da7213->mclk_rate == freq))
+		return 0;
+
+	if (((freq < 5000000) && (freq != 32768)) || (freq > 54000000)) {
+		dev_err(codec_dai->dev, "Unsupported MCLK value %d\n",
+			freq);
+		return -EINVAL;
+	}
 
 	switch (clk_id) {
 	case DA7213_CLKSRC_MCLK:
-		if ((freq == 32768) ||
-		    ((freq >= 5000000) && (freq <= 54000000))) {
-			da7213->mclk_rate = freq;
-			return 0;
-		} else {
-			dev_err(codec_dai->dev, "Unsupported MCLK value %d\n",
-				freq);
-			return -EINVAL;
-		}
+		da7213->mclk_squarer_en = false;
+		break;
+	case DA7213_CLKSRC_MCLK_SQR:
+		da7213->mclk_squarer_en = true;
 		break;
 	default:
 		dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id);
 		return -EINVAL;
 	}
+
+	da7213->clk_src = clk_id;
+
+	if (da7213->mclk) {
+		freq = clk_round_rate(da7213->mclk, freq);
+		ret = clk_set_rate(da7213->mclk, freq);
+		if (ret) {
+			dev_err(codec_dai->dev, "Failed to set clock rate %d\n",
+				freq);
+			return ret;
+		}
+	}
+
+	da7213->mclk_rate = freq;
+
+	return 0;
 }
 
 /* Supported PLL input frequencies are 5MHz - 54MHz. */
@@ -1366,12 +1388,25 @@ static struct snd_soc_dai_driver da7213_dai = {
 static int da7213_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
+	struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
+			/* MCLK */
+			if (da7213->mclk) {
+				ret = clk_prepare_enable(da7213->mclk);
+				if (ret) {
+					dev_err(codec->dev,
+						"Failed to enable mclk\n");
+					return ret;
+				}
+			}
+
 			/* Enable VMID reference & master bias */
 			snd_soc_update_bits(codec, DA7213_REFERENCES,
 					    DA7213_VMID_EN | DA7213_BIAS_EN,
@@ -1382,15 +1417,127 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec,
 		/* Disable VMID reference & master bias */
 		snd_soc_update_bits(codec, DA7213_REFERENCES,
 				    DA7213_VMID_EN | DA7213_BIAS_EN, 0);
+
+		/* MCLK */
+		if (da7213->mclk)
+			clk_disable_unprepare(da7213->mclk);
 		break;
 	}
 	return 0;
 }
 
+/* DT */
+static const struct of_device_id da7213_of_match[] = {
+	{ .compatible = "dlg,da7213", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, da7213_of_match);
+
+static enum da7213_micbias_voltage
+	da7213_of_micbias_lvl(struct snd_soc_codec *codec, u32 val)
+{
+	switch (val) {
+	case 1600:
+		return DA7213_MICBIAS_1_6V;
+	case 2200:
+		return DA7213_MICBIAS_2_2V;
+	case 2500:
+		return DA7213_MICBIAS_2_5V;
+	case 3000:
+		return DA7213_MICBIAS_3_0V;
+	default:
+		dev_warn(codec->dev, "Invalid micbias level\n");
+		return DA7213_MICBIAS_2_2V;
+	}
+}
+
+static enum da7213_dmic_data_sel
+	da7213_of_dmic_data_sel(struct snd_soc_codec *codec, const char *str)
+{
+	if (!strcmp(str, "lrise_rfall")) {
+		return DA7213_DMIC_DATA_LRISE_RFALL;
+	} else if (!strcmp(str, "lfall_rrise")) {
+		return DA7213_DMIC_DATA_LFALL_RRISE;
+	} else {
+		dev_warn(codec->dev, "Invalid DMIC data select type\n");
+		return DA7213_DMIC_DATA_LRISE_RFALL;
+	}
+}
+
+static enum da7213_dmic_samplephase
+	da7213_of_dmic_samplephase(struct snd_soc_codec *codec, const char *str)
+{
+	if (!strcmp(str, "on_clkedge")) {
+		return DA7213_DMIC_SAMPLE_ON_CLKEDGE;
+	} else if (!strcmp(str, "between_clkedge")) {
+		return DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE;
+	} else {
+		dev_warn(codec->dev, "Invalid DMIC sample phase\n");
+		return DA7213_DMIC_SAMPLE_ON_CLKEDGE;
+	}
+}
+
+static enum da7213_dmic_clk_rate
+	da7213_of_dmic_clkrate(struct snd_soc_codec *codec, u32 val)
+{
+	switch (val) {
+	case 1500000:
+		return DA7213_DMIC_CLK_1_5MHZ;
+	case 3000000:
+		return DA7213_DMIC_CLK_3_0MHZ;
+	default:
+		dev_warn(codec->dev, "Invalid DMIC clock rate\n");
+		return DA7213_DMIC_CLK_1_5MHZ;
+	}
+}
+
+static struct da7213_platform_data
+	*da7213_of_to_pdata(struct snd_soc_codec *codec)
+{
+	struct device_node *np = codec->dev->of_node;
+	struct da7213_platform_data *pdata;
+	const char *of_str;
+	u32 of_val32;
+
+	pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL);
+	if (!pdata) {
+		dev_warn(codec->dev, "Failed to allocate memory for pdata\n");
+		return NULL;
+	}
+
+	if (of_property_read_u32(np, "dlg,micbias1-lvl", &of_val32) >= 0)
+		pdata->micbias1_lvl = da7213_of_micbias_lvl(codec, of_val32);
+	else
+		pdata->micbias1_lvl = DA7213_MICBIAS_2_2V;
+
+	if (of_property_read_u32(np, "dlg,micbias2-lvl", &of_val32) >= 0)
+		pdata->micbias2_lvl = da7213_of_micbias_lvl(codec, of_val32);
+	else
+		pdata->micbias2_lvl = DA7213_MICBIAS_2_2V;
+
+	if (!of_property_read_string(np, "dlg,dmic-data-sel", &of_str))
+		pdata->dmic_data_sel = da7213_of_dmic_data_sel(codec, of_str);
+	else
+		pdata->dmic_data_sel = DA7213_DMIC_DATA_LRISE_RFALL;
+
+	if (!of_property_read_string(np, "dlg,dmic-samplephase", &of_str))
+		pdata->dmic_samplephase =
+			da7213_of_dmic_samplephase(codec, of_str);
+	else
+		pdata->dmic_samplephase = DA7213_DMIC_SAMPLE_ON_CLKEDGE;
+
+	if (of_property_read_u32(np, "dlg,dmic-clkrate", &of_val32) >= 0)
+		pdata->dmic_clk_rate = da7213_of_dmic_clkrate(codec, of_val32);
+	else
+		pdata->dmic_clk_rate = DA7213_DMIC_CLK_3_0MHZ;
+
+	return pdata;
+}
+
+
 static int da7213_probe(struct snd_soc_codec *codec)
 {
 	struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec);
-	struct da7213_platform_data *pdata = da7213->pdata;
 
 	/* Default to using ALC auto offset calibration mode. */
 	snd_soc_update_bits(codec, DA7213_ALC_CTRL1,
@@ -1450,8 +1597,15 @@ static int da7213_probe(struct snd_soc_codec *codec)
 	snd_soc_update_bits(codec, DA7213_LINE_CTRL,
 			    DA7213_LINE_AMP_OE, DA7213_LINE_AMP_OE);
 
+	/* Handle DT/Platform data */
+	if (codec->dev->of_node)
+		da7213->pdata = da7213_of_to_pdata(codec);
+	else
+		da7213->pdata = dev_get_platdata(codec->dev);
+
 	/* Set platform data values */
 	if (da7213->pdata) {
+		struct da7213_platform_data *pdata = da7213->pdata;
 		u8 micbias_lvl = 0, dmic_cfg = 0;
 
 		/* Set Mic Bias voltages */
@@ -1503,10 +1657,17 @@ static int da7213_probe(struct snd_soc_codec *codec)
 				    DA7213_DMIC_DATA_SEL_MASK |
 				    DA7213_DMIC_SAMPLEPHASE_MASK |
 				    DA7213_DMIC_CLK_RATE_MASK, dmic_cfg);
+	}
 
-		/* Set MCLK squaring */
-		da7213->mclk_squarer_en = pdata->mclk_squaring;
+	/* Check if MCLK provided */
+	da7213->mclk = devm_clk_get(codec->dev, "mclk");
+	if (IS_ERR(da7213->mclk)) {
+		if (PTR_ERR(da7213->mclk) != -ENOENT)
+			return PTR_ERR(da7213->mclk);
+		else
+			da7213->mclk = NULL;
 	}
+
 	return 0;
 }
 
@@ -1537,7 +1698,6 @@ static int da7213_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct da7213_priv *da7213;
-	struct da7213_platform_data *pdata = dev_get_platdata(&i2c->dev);
 	int ret;
 
 	da7213 = devm_kzalloc(&i2c->dev, sizeof(struct da7213_priv),
@@ -1545,9 +1705,6 @@ static int da7213_i2c_probe(struct i2c_client *i2c,
 	if (!da7213)
 		return -ENOMEM;
 
-	if (pdata)
-		da7213->pdata = pdata;
-
 	i2c_set_clientdata(i2c, da7213);
 
 	da7213->regmap = devm_regmap_init_i2c(i2c, &da7213_regmap_config);
@@ -1582,6 +1739,7 @@ MODULE_DEVICE_TABLE(i2c, da7213_i2c_id);
 static struct i2c_driver da7213_i2c_driver = {
 	.driver = {
 		.name = "da7213",
+		.of_match_table = of_match_ptr(da7213_of_match),
 	},
 	.probe		= da7213_i2c_probe,
 	.remove		= da7213_remove,
diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h
index 9cb9ddd01282..030fd691b076 100644
--- a/sound/soc/codecs/da7213.h
+++ b/sound/soc/codecs/da7213.h
@@ -13,6 +13,7 @@
 #ifndef _DA7213_H
 #define _DA7213_H
 
+#include <linux/clk.h>
 #include <linux/regmap.h>
 #include <sound/da7213.h>
 
@@ -504,14 +505,17 @@
 #define DA7213_PLL_INDIV_20_40_MHZ_VAL	8
 #define DA7213_PLL_INDIV_40_54_MHZ_VAL	16
 
-enum clk_src {
-	DA7213_CLKSRC_MCLK
+enum da7213_clk_src {
+	DA7213_CLKSRC_MCLK = 0,
+	DA7213_CLKSRC_MCLK_SQR,
 };
 
 /* Codec private data */
 struct da7213_priv {
 	struct regmap *regmap;
+	struct clk *mclk;
 	unsigned int mclk_rate;
+	int clk_src;
 	bool master;
 	bool mclk_squarer_en;
 	bool srm_en;
diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c
new file mode 100644
index 000000000000..9459593eef13
--- /dev/null
+++ b/sound/soc/codecs/da7219-aad.c
@@ -0,0 +1,823 @@
+/*
+ * da7219-aad.c - Dialog DA7219 ALSA SoC AAD Driver
+ *
+ * Copyright (c) 2015 Dialog Semiconductor Ltd.
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/of_device.h>
+#include <linux/of_irq.h>
+#include <linux/pm_wakeirq.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/workqueue.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/da7219.h>
+
+#include "da7219.h"
+#include "da7219-aad.h"
+
+
+/*
+ * Detection control
+ */
+
+void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
+{
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+	da7219->aad->jack = jack;
+	da7219->aad->jack_inserted = false;
+
+	/* Send an initial empty report */
+	snd_soc_jack_report(jack, 0, DA7219_AAD_REPORT_ALL_MASK);
+
+	/* Enable/Disable jack detection */
+	snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1,
+			    DA7219_ACCDET_EN_MASK,
+			    (jack ? DA7219_ACCDET_EN_MASK : 0));
+}
+EXPORT_SYMBOL_GPL(da7219_aad_jack_det);
+
+/*
+ * Button/HPTest work
+ */
+
+static void da7219_aad_btn_det_work(struct work_struct *work)
+{
+	struct da7219_aad_priv *da7219_aad =
+		container_of(work, struct da7219_aad_priv, btn_det_work);
+	struct snd_soc_codec *codec = da7219_aad->codec;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+	u8 statusa, micbias_ctrl;
+	bool micbias_up = false;
+	int retries = 0;
+
+	/* Drive headphones/lineout */
+	snd_soc_update_bits(codec, DA7219_HP_L_CTRL,
+			    DA7219_HP_L_AMP_OE_MASK,
+			    DA7219_HP_L_AMP_OE_MASK);
+	snd_soc_update_bits(codec, DA7219_HP_R_CTRL,
+			    DA7219_HP_R_AMP_OE_MASK,
+			    DA7219_HP_R_AMP_OE_MASK);
+
+	/* Make sure mic bias is up */
+	snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
+	snd_soc_dapm_sync(dapm);
+
+	do {
+		statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A);
+		if (statusa & DA7219_MICBIAS_UP_STS_MASK)
+			micbias_up = true;
+		else if (retries++ < DA7219_AAD_MICBIAS_CHK_RETRIES)
+			msleep(DA7219_AAD_MICBIAS_CHK_DELAY);
+	} while ((!micbias_up) && (retries < DA7219_AAD_MICBIAS_CHK_RETRIES));
+
+	if (retries >= DA7219_AAD_MICBIAS_CHK_RETRIES)
+		dev_warn(codec->dev, "Mic bias status check timed out");
+
+	/*
+	 * Mic bias pulse required to enable mic, must be done before enabling
+	 * button detection to prevent erroneous button readings.
+	 */
+	if (da7219_aad->micbias_pulse_lvl && da7219_aad->micbias_pulse_time) {
+		/* Pulse higher level voltage */
+		micbias_ctrl = snd_soc_read(codec, DA7219_MICBIAS_CTRL);
+		snd_soc_update_bits(codec, DA7219_MICBIAS_CTRL,
+				    DA7219_MICBIAS1_LEVEL_MASK,
+				    da7219_aad->micbias_pulse_lvl);
+		msleep(da7219_aad->micbias_pulse_time);
+		snd_soc_write(codec, DA7219_MICBIAS_CTRL, micbias_ctrl);
+
+	}
+
+	snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1,
+			    DA7219_BUTTON_CONFIG_MASK,
+			    da7219_aad->btn_cfg);
+}
+
+static void da7219_aad_hptest_work(struct work_struct *work)
+{
+	struct da7219_aad_priv *da7219_aad =
+		container_of(work, struct da7219_aad_priv, hptest_work);
+	struct snd_soc_codec *codec = da7219_aad->codec;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+	u16 tonegen_freq_hptest;
+	u8 accdet_cfg8;
+	int report = 0;
+
+	/* Lock DAPM and any Kcontrols that are affected by this test */
+	snd_soc_dapm_mutex_lock(dapm);
+	mutex_lock(&da7219->lock);
+
+	/* Bypass cache so it saves current settings */
+	regcache_cache_bypass(da7219->regmap, true);
+
+	/* Make sure Tone Generator is disabled */
+	snd_soc_write(codec, DA7219_TONE_GEN_CFG1, 0);
+
+	/* Enable HPTest block, 1KOhms check */
+	snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_8,
+			    DA7219_HPTEST_EN_MASK | DA7219_HPTEST_RES_SEL_MASK,
+			    DA7219_HPTEST_EN_MASK |
+			    DA7219_HPTEST_RES_SEL_1KOHMS);
+
+	/* Set gains to 0db */
+	snd_soc_write(codec, DA7219_DAC_L_GAIN, DA7219_DAC_DIGITAL_GAIN_0DB);
+	snd_soc_write(codec, DA7219_DAC_R_GAIN, DA7219_DAC_DIGITAL_GAIN_0DB);
+	snd_soc_write(codec, DA7219_HP_L_GAIN, DA7219_HP_AMP_GAIN_0DB);
+	snd_soc_write(codec, DA7219_HP_R_GAIN, DA7219_HP_AMP_GAIN_0DB);
+
+	/* Disable DAC filters, EQs and soft mute */
+	snd_soc_update_bits(codec, DA7219_DAC_FILTERS1, DA7219_HPF_MODE_MASK,
+			    0);
+	snd_soc_update_bits(codec, DA7219_DAC_FILTERS4, DA7219_DAC_EQ_EN_MASK,
+			    0);
+	snd_soc_update_bits(codec, DA7219_DAC_FILTERS5,
+			    DA7219_DAC_SOFTMUTE_EN_MASK, 0);
+
+	/* Enable HP left & right paths */
+	snd_soc_update_bits(codec, DA7219_CP_CTRL, DA7219_CP_EN_MASK,
+			    DA7219_CP_EN_MASK);
+	snd_soc_update_bits(codec, DA7219_DIG_ROUTING_DAC,
+			    DA7219_DAC_L_SRC_MASK | DA7219_DAC_R_SRC_MASK,
+			    DA7219_DAC_L_SRC_TONEGEN |
+			    DA7219_DAC_R_SRC_TONEGEN);
+	snd_soc_update_bits(codec, DA7219_DAC_L_CTRL,
+			    DA7219_DAC_L_EN_MASK | DA7219_DAC_L_MUTE_EN_MASK,
+			    DA7219_DAC_L_EN_MASK);
+	snd_soc_update_bits(codec, DA7219_DAC_R_CTRL,
+			    DA7219_DAC_R_EN_MASK | DA7219_DAC_R_MUTE_EN_MASK,
+			    DA7219_DAC_R_EN_MASK);
+	snd_soc_update_bits(codec, DA7219_MIXOUT_L_SELECT,
+			    DA7219_MIXOUT_L_MIX_SELECT_MASK,
+			    DA7219_MIXOUT_L_MIX_SELECT_MASK);
+	snd_soc_update_bits(codec, DA7219_MIXOUT_R_SELECT,
+			    DA7219_MIXOUT_R_MIX_SELECT_MASK,
+			    DA7219_MIXOUT_R_MIX_SELECT_MASK);
+	snd_soc_update_bits(codec, DA7219_DROUTING_ST_OUTFILT_1L,
+			    DA7219_OUTFILT_ST_1L_SRC_MASK,
+			    DA7219_DMIX_ST_SRC_OUTFILT1L);
+	snd_soc_update_bits(codec, DA7219_DROUTING_ST_OUTFILT_1R,
+			    DA7219_OUTFILT_ST_1R_SRC_MASK,
+			    DA7219_DMIX_ST_SRC_OUTFILT1R);
+	snd_soc_update_bits(codec, DA7219_MIXOUT_L_CTRL,
+			    DA7219_MIXOUT_L_AMP_EN_MASK,
+			    DA7219_MIXOUT_L_AMP_EN_MASK);
+	snd_soc_update_bits(codec, DA7219_MIXOUT_R_CTRL,
+			    DA7219_MIXOUT_R_AMP_EN_MASK,
+			    DA7219_MIXOUT_R_AMP_EN_MASK);
+	snd_soc_write(codec, DA7219_HP_L_CTRL,
+		      DA7219_HP_L_AMP_OE_MASK | DA7219_HP_L_AMP_EN_MASK);
+	snd_soc_write(codec, DA7219_HP_R_CTRL,
+		      DA7219_HP_R_AMP_OE_MASK | DA7219_HP_R_AMP_EN_MASK);
+
+	/* Configure & start Tone Generator */
+	snd_soc_write(codec, DA7219_TONE_GEN_ON_PER, DA7219_BEEP_ON_PER_MASK);
+	tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ);
+	regmap_raw_write(da7219->regmap, DA7219_TONE_GEN_FREQ1_L,
+			 &tonegen_freq_hptest, sizeof(tonegen_freq_hptest));
+	snd_soc_update_bits(codec, DA7219_TONE_GEN_CFG2,
+			    DA7219_SWG_SEL_MASK | DA7219_TONE_GEN_GAIN_MASK,
+			    DA7219_SWG_SEL_SRAMP |
+			    DA7219_TONE_GEN_GAIN_MINUS_15DB);
+	snd_soc_write(codec, DA7219_TONE_GEN_CFG1, DA7219_START_STOPN_MASK);
+
+	msleep(DA7219_AAD_HPTEST_PERIOD);
+
+	/* Grab comparator reading */
+	accdet_cfg8 = snd_soc_read(codec, DA7219_ACCDET_CONFIG_8);
+	if (accdet_cfg8 & DA7219_HPTEST_COMP_MASK)
+		report |= SND_JACK_HEADPHONE;
+	else
+		report |= SND_JACK_LINEOUT;
+
+	/* Stop tone generator */
+	snd_soc_write(codec, DA7219_TONE_GEN_CFG1, 0);
+
+	msleep(DA7219_AAD_HPTEST_PERIOD);
+
+	/* Restore original settings from cache */
+	regcache_mark_dirty(da7219->regmap);
+	regcache_sync_region(da7219->regmap, DA7219_HP_L_CTRL,
+			     DA7219_HP_R_CTRL);
+	regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_CTRL,
+			     DA7219_MIXOUT_R_CTRL);
+	regcache_sync_region(da7219->regmap, DA7219_DROUTING_ST_OUTFILT_1L,
+			     DA7219_DROUTING_ST_OUTFILT_1R);
+	regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_SELECT,
+			     DA7219_MIXOUT_R_SELECT);
+	regcache_sync_region(da7219->regmap, DA7219_DAC_L_CTRL,
+			     DA7219_DAC_R_CTRL);
+	regcache_sync_region(da7219->regmap, DA7219_DIG_ROUTING_DAC,
+			     DA7219_DIG_ROUTING_DAC);
+	regcache_sync_region(da7219->regmap, DA7219_CP_CTRL, DA7219_CP_CTRL);
+	regcache_sync_region(da7219->regmap, DA7219_DAC_FILTERS5,
+			     DA7219_DAC_FILTERS5);
+	regcache_sync_region(da7219->regmap, DA7219_DAC_FILTERS4,
+			     DA7219_DAC_FILTERS1);
+	regcache_sync_region(da7219->regmap, DA7219_HP_L_GAIN,
+			     DA7219_HP_R_GAIN);
+	regcache_sync_region(da7219->regmap, DA7219_DAC_L_GAIN,
+			     DA7219_DAC_R_GAIN);
+	regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_ON_PER,
+			     DA7219_TONE_GEN_ON_PER);
+	regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_FREQ1_L,
+			     DA7219_TONE_GEN_FREQ1_U);
+	regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_CFG1,
+			     DA7219_TONE_GEN_CFG2);
+
+	regcache_cache_bypass(da7219->regmap, false);
+
+	/* Disable HPTest block */
+	snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_8,
+			    DA7219_HPTEST_EN_MASK, 0);
+
+	/* Drive Headphones/lineout */
+	snd_soc_update_bits(codec, DA7219_HP_L_CTRL, DA7219_HP_L_AMP_OE_MASK,
+			    DA7219_HP_L_AMP_OE_MASK);
+	snd_soc_update_bits(codec, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_OE_MASK,
+			    DA7219_HP_R_AMP_OE_MASK);
+
+	mutex_unlock(&da7219->lock);
+	snd_soc_dapm_mutex_unlock(dapm);
+
+	/*
+	 * Only send report if jack hasn't been removed during process,
+	 * otherwise it's invalid and we drop it.
+	 */
+	if (da7219_aad->jack_inserted)
+		snd_soc_jack_report(da7219_aad->jack, report,
+				    SND_JACK_HEADSET | SND_JACK_LINEOUT);
+}
+
+
+/*
+ * IRQ
+ */
+
+static irqreturn_t da7219_aad_irq_thread(int irq, void *data)
+{
+	struct da7219_aad_priv *da7219_aad = data;
+	struct snd_soc_codec *codec = da7219_aad->codec;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	u8 events[DA7219_AAD_IRQ_REG_MAX];
+	u8 statusa;
+	int i, report = 0, mask = 0;
+
+	/* Read current IRQ events */
+	regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A,
+			 events, DA7219_AAD_IRQ_REG_MAX);
+
+	if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B])
+		return IRQ_NONE;
+
+	/* Read status register for jack insertion & type status */
+	statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A);
+
+	/* Clear events */
+	regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A,
+			  events, DA7219_AAD_IRQ_REG_MAX);
+
+	dev_dbg(codec->dev, "IRQ events = 0x%x|0x%x, status = 0x%x\n",
+		events[DA7219_AAD_IRQ_REG_A], events[DA7219_AAD_IRQ_REG_B],
+		statusa);
+
+	if (statusa & DA7219_JACK_INSERTION_STS_MASK) {
+		/* Jack Insertion */
+		if (events[DA7219_AAD_IRQ_REG_A] &
+		    DA7219_E_JACK_INSERTED_MASK) {
+			report |= SND_JACK_MECHANICAL;
+			mask |= SND_JACK_MECHANICAL;
+			da7219_aad->jack_inserted = true;
+		}
+
+		/* Jack type detection */
+		if (events[DA7219_AAD_IRQ_REG_A] &
+		    DA7219_E_JACK_DETECT_COMPLETE_MASK) {
+			/*
+			 * If 4-pole, then enable button detection, else perform
+			 * HP impedance test to determine output type to report.
+			 *
+			 * We schedule work here as the tasks themselves can
+			 * take time to complete, and in particular for hptest
+			 * we want to be able to check if the jack was removed
+			 * during the procedure as this will invalidate the
+			 * result. By doing this as work, the IRQ thread can
+			 * handle a removal, and we can check at the end of
+			 * hptest if we have a valid result or not.
+			 */
+			if (statusa & DA7219_JACK_TYPE_STS_MASK) {
+				report |= SND_JACK_HEADSET;
+				mask |=	SND_JACK_HEADSET | SND_JACK_LINEOUT;
+				schedule_work(&da7219_aad->btn_det_work);
+			} else {
+				schedule_work(&da7219_aad->hptest_work);
+			}
+		}
+
+		/* Button support for 4-pole jack */
+		if (statusa & DA7219_JACK_TYPE_STS_MASK) {
+			for (i = 0; i < DA7219_AAD_MAX_BUTTONS; ++i) {
+				/* Button Press */
+				if (events[DA7219_AAD_IRQ_REG_B] &
+				    (DA7219_E_BUTTON_A_PRESSED_MASK << i)) {
+					report |= SND_JACK_BTN_0 >> i;
+					mask |= SND_JACK_BTN_0 >> i;
+				}
+			}
+			snd_soc_jack_report(da7219_aad->jack, report, mask);
+
+			for (i = 0; i < DA7219_AAD_MAX_BUTTONS; ++i) {
+				/* Button Release */
+				if (events[DA7219_AAD_IRQ_REG_B] &
+				    (DA7219_E_BUTTON_A_RELEASED_MASK >> i)) {
+					report &= ~(SND_JACK_BTN_0 >> i);
+					mask |= SND_JACK_BTN_0 >> i;
+				}
+			}
+		}
+	} else {
+		/* Jack removal */
+		if (events[DA7219_AAD_IRQ_REG_A] & DA7219_E_JACK_REMOVED_MASK) {
+			report = 0;
+			mask |= DA7219_AAD_REPORT_ALL_MASK;
+			da7219_aad->jack_inserted = false;
+
+			/* Un-drive headphones/lineout */
+			snd_soc_update_bits(codec, DA7219_HP_R_CTRL,
+					    DA7219_HP_R_AMP_OE_MASK, 0);
+			snd_soc_update_bits(codec, DA7219_HP_L_CTRL,
+					    DA7219_HP_L_AMP_OE_MASK, 0);
+
+			/* Ensure button detection disabled */
+			snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1,
+					    DA7219_BUTTON_CONFIG_MASK, 0);
+
+			/* Disable mic bias */
+			snd_soc_dapm_disable_pin(dapm, "Mic Bias");
+			snd_soc_dapm_sync(dapm);
+
+			/* Cancel any pending work */
+			cancel_work_sync(&da7219_aad->btn_det_work);
+			cancel_work_sync(&da7219_aad->hptest_work);
+		}
+	}
+
+	snd_soc_jack_report(da7219_aad->jack, report, mask);
+
+	return IRQ_HANDLED;
+}
+
+/*
+ * DT to pdata conversion
+ */
+
+static enum da7219_aad_micbias_pulse_lvl
+	da7219_aad_of_micbias_pulse_lvl(struct snd_soc_codec *codec, u32 val)
+{
+	switch (val) {
+	case 2800:
+		return DA7219_AAD_MICBIAS_PULSE_LVL_2_8V;
+	case 2900:
+		return DA7219_AAD_MICBIAS_PULSE_LVL_2_9V;
+	default:
+		dev_warn(codec->dev, "Invalid micbias pulse level");
+		return DA7219_AAD_MICBIAS_PULSE_LVL_OFF;
+	}
+}
+
+static enum da7219_aad_btn_cfg
+	da7219_aad_of_btn_cfg(struct snd_soc_codec *codec, u32 val)
+{
+	switch (val) {
+	case 2:
+		return DA7219_AAD_BTN_CFG_2MS;
+	case 5:
+		return DA7219_AAD_BTN_CFG_5MS;
+	case 10:
+		return DA7219_AAD_BTN_CFG_10MS;
+	case 50:
+		return DA7219_AAD_BTN_CFG_50MS;
+	case 100:
+		return DA7219_AAD_BTN_CFG_100MS;
+	case 200:
+		return DA7219_AAD_BTN_CFG_200MS;
+	case 500:
+		return DA7219_AAD_BTN_CFG_500MS;
+	default:
+		dev_warn(codec->dev, "Invalid button config");
+		return DA7219_AAD_BTN_CFG_10MS;
+	}
+}
+
+static enum da7219_aad_mic_det_thr
+	da7219_aad_of_mic_det_thr(struct snd_soc_codec *codec, u32 val)
+{
+	switch (val) {
+	case 200:
+		return DA7219_AAD_MIC_DET_THR_200_OHMS;
+	case 500:
+		return DA7219_AAD_MIC_DET_THR_500_OHMS;
+	case 750:
+		return DA7219_AAD_MIC_DET_THR_750_OHMS;
+	case 1000:
+		return DA7219_AAD_MIC_DET_THR_1000_OHMS;
+	default:
+		dev_warn(codec->dev, "Invalid mic detect threshold");
+		return DA7219_AAD_MIC_DET_THR_500_OHMS;
+	}
+}
+
+static enum da7219_aad_jack_ins_deb
+	da7219_aad_of_jack_ins_deb(struct snd_soc_codec *codec, u32 val)
+{
+	switch (val) {
+	case 5:
+		return DA7219_AAD_JACK_INS_DEB_5MS;
+	case 10:
+		return DA7219_AAD_JACK_INS_DEB_10MS;
+	case 20:
+		return DA7219_AAD_JACK_INS_DEB_20MS;
+	case 50:
+		return DA7219_AAD_JACK_INS_DEB_50MS;
+	case 100:
+		return DA7219_AAD_JACK_INS_DEB_100MS;
+	case 200:
+		return DA7219_AAD_JACK_INS_DEB_200MS;
+	case 500:
+		return DA7219_AAD_JACK_INS_DEB_500MS;
+	case 1000:
+		return DA7219_AAD_JACK_INS_DEB_1S;
+	default:
+		dev_warn(codec->dev, "Invalid jack insert debounce");
+		return DA7219_AAD_JACK_INS_DEB_20MS;
+	}
+}
+
+static enum da7219_aad_jack_det_rate
+	da7219_aad_of_jack_det_rate(struct snd_soc_codec *codec, const char *str)
+{
+	if (!strcmp(str, "32ms_64ms")) {
+		return DA7219_AAD_JACK_DET_RATE_32_64MS;
+	} else if (!strcmp(str, "64ms_128ms")) {
+		return DA7219_AAD_JACK_DET_RATE_64_128MS;
+	} else if (!strcmp(str, "128ms_256ms")) {
+		return DA7219_AAD_JACK_DET_RATE_128_256MS;
+	} else if (!strcmp(str, "256ms_512ms")) {
+		return DA7219_AAD_JACK_DET_RATE_256_512MS;
+	} else {
+		dev_warn(codec->dev, "Invalid jack detect rate");
+		return DA7219_AAD_JACK_DET_RATE_256_512MS;
+	}
+}
+
+static enum da7219_aad_jack_rem_deb
+	da7219_aad_of_jack_rem_deb(struct snd_soc_codec *codec, u32 val)
+{
+	switch (val) {
+	case 1:
+		return DA7219_AAD_JACK_REM_DEB_1MS;
+	case 5:
+		return DA7219_AAD_JACK_REM_DEB_5MS;
+	case 10:
+		return DA7219_AAD_JACK_REM_DEB_10MS;
+	case 20:
+		return DA7219_AAD_JACK_REM_DEB_20MS;
+	default:
+		dev_warn(codec->dev, "Invalid jack removal debounce");
+		return DA7219_AAD_JACK_REM_DEB_1MS;
+	}
+}
+
+static enum da7219_aad_btn_avg
+	da7219_aad_of_btn_avg(struct snd_soc_codec *codec, u32 val)
+{
+	switch (val) {
+	case 1:
+		return DA7219_AAD_BTN_AVG_1;
+	case 2:
+		return DA7219_AAD_BTN_AVG_2;
+	case 4:
+		return DA7219_AAD_BTN_AVG_4;
+	case 8:
+		return DA7219_AAD_BTN_AVG_8;
+	default:
+		dev_warn(codec->dev, "Invalid button average value");
+		return DA7219_AAD_BTN_AVG_2;
+	}
+}
+
+static enum da7219_aad_adc_1bit_rpt
+	da7219_aad_of_adc_1bit_rpt(struct snd_soc_codec *codec, u32 val)
+{
+	switch (val) {
+	case 1:
+		return DA7219_AAD_ADC_1BIT_RPT_1;
+	case 2:
+		return DA7219_AAD_ADC_1BIT_RPT_2;
+	case 4:
+		return DA7219_AAD_ADC_1BIT_RPT_4;
+	case 8:
+		return DA7219_AAD_ADC_1BIT_RPT_8;
+	default:
+		dev_warn(codec->dev, "Invalid ADC 1-bit repeat value");
+		return DA7219_AAD_ADC_1BIT_RPT_1;
+	}
+}
+
+static struct da7219_aad_pdata *da7219_aad_of_to_pdata(struct snd_soc_codec *codec)
+{
+	struct device_node *np = codec->dev->of_node;
+	struct device_node *aad_np = of_find_node_by_name(np, "da7219_aad");
+	struct da7219_aad_pdata *aad_pdata;
+	const char *of_str;
+	u32 of_val32;
+
+	if (!aad_np)
+		return NULL;
+
+	aad_pdata = devm_kzalloc(codec->dev, sizeof(*aad_pdata), GFP_KERNEL);
+	if (!aad_pdata)
+		goto out;
+
+	aad_pdata->irq = irq_of_parse_and_map(np, 0);
+
+	if (of_property_read_u32(aad_np, "dlg,micbias-pulse-lvl",
+				 &of_val32) >= 0)
+		aad_pdata->micbias_pulse_lvl =
+			da7219_aad_of_micbias_pulse_lvl(codec, of_val32);
+	else
+		aad_pdata->micbias_pulse_lvl = DA7219_AAD_MICBIAS_PULSE_LVL_OFF;
+
+	if (of_property_read_u32(aad_np, "dlg,micbias-pulse-time",
+				 &of_val32) >= 0)
+		aad_pdata->micbias_pulse_time = of_val32;
+
+	if (of_property_read_u32(aad_np, "dlg,btn-cfg", &of_val32) >= 0)
+		aad_pdata->btn_cfg = da7219_aad_of_btn_cfg(codec, of_val32);
+	else
+		aad_pdata->btn_cfg = DA7219_AAD_BTN_CFG_10MS;
+
+	if (of_property_read_u32(aad_np, "dlg,mic-det-thr", &of_val32) >= 0)
+		aad_pdata->mic_det_thr =
+			da7219_aad_of_mic_det_thr(codec, of_val32);
+	else
+		aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_500_OHMS;
+
+	if (of_property_read_u32(aad_np, "dlg,jack-ins-deb", &of_val32) >= 0)
+		aad_pdata->jack_ins_deb =
+			da7219_aad_of_jack_ins_deb(codec, of_val32);
+	else
+		aad_pdata->jack_ins_deb = DA7219_AAD_JACK_INS_DEB_20MS;
+
+	if (!of_property_read_string(aad_np, "dlg,jack-det-rate", &of_str))
+		aad_pdata->jack_det_rate =
+			da7219_aad_of_jack_det_rate(codec, of_str);
+	else
+		aad_pdata->jack_det_rate = DA7219_AAD_JACK_DET_RATE_256_512MS;
+
+	if (of_property_read_u32(aad_np, "dlg,jack-rem-deb", &of_val32) >= 0)
+		aad_pdata->jack_rem_deb =
+			da7219_aad_of_jack_rem_deb(codec, of_val32);
+	else
+		aad_pdata->jack_rem_deb = DA7219_AAD_JACK_REM_DEB_1MS;
+
+	if (of_property_read_u32(aad_np, "dlg,a-d-btn-thr", &of_val32) >= 0)
+		aad_pdata->a_d_btn_thr = (u8) of_val32;
+	else
+		aad_pdata->a_d_btn_thr = 0xA;
+
+	if (of_property_read_u32(aad_np, "dlg,d-b-btn-thr", &of_val32) >= 0)
+		aad_pdata->d_b_btn_thr = (u8) of_val32;
+	else
+		aad_pdata->d_b_btn_thr = 0x16;
+
+	if (of_property_read_u32(aad_np, "dlg,b-c-btn-thr", &of_val32) >= 0)
+		aad_pdata->b_c_btn_thr = (u8) of_val32;
+	else
+		aad_pdata->b_c_btn_thr = 0x21;
+
+	if (of_property_read_u32(aad_np, "dlg,c-mic-btn-thr", &of_val32) >= 0)
+		aad_pdata->c_mic_btn_thr = (u8) of_val32;
+	else
+		aad_pdata->c_mic_btn_thr = 0x3E;
+
+	if (of_property_read_u32(aad_np, "dlg,btn-avg", &of_val32) >= 0)
+		aad_pdata->btn_avg = da7219_aad_of_btn_avg(codec, of_val32);
+	else
+		aad_pdata->btn_avg = DA7219_AAD_BTN_AVG_2;
+
+	if (of_property_read_u32(aad_np, "dlg,adc-1bit-rpt", &of_val32) >= 0)
+		aad_pdata->adc_1bit_rpt =
+			da7219_aad_of_adc_1bit_rpt(codec, of_val32);
+	else
+		aad_pdata->adc_1bit_rpt = DA7219_AAD_ADC_1BIT_RPT_1;
+
+out:
+	of_node_put(aad_np);
+
+	return aad_pdata;
+}
+
+static void da7219_aad_handle_pdata(struct snd_soc_codec *codec)
+{
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	struct da7219_aad_priv *da7219_aad = da7219->aad;
+	struct da7219_pdata *pdata = da7219->pdata;
+
+	if ((pdata) && (pdata->aad_pdata)) {
+		struct da7219_aad_pdata *aad_pdata = pdata->aad_pdata;
+		u8 cfg, mask;
+
+		da7219_aad->irq = aad_pdata->irq;
+
+		switch (aad_pdata->micbias_pulse_lvl) {
+		case DA7219_AAD_MICBIAS_PULSE_LVL_2_8V:
+		case DA7219_AAD_MICBIAS_PULSE_LVL_2_9V:
+			da7219_aad->micbias_pulse_lvl =
+				(aad_pdata->micbias_pulse_lvl <<
+				 DA7219_MICBIAS1_LEVEL_SHIFT);
+			break;
+		default:
+			break;
+		}
+
+		da7219_aad->micbias_pulse_time = aad_pdata->micbias_pulse_time;
+
+		switch (aad_pdata->btn_cfg) {
+		case DA7219_AAD_BTN_CFG_2MS:
+		case DA7219_AAD_BTN_CFG_5MS:
+		case DA7219_AAD_BTN_CFG_10MS:
+		case DA7219_AAD_BTN_CFG_50MS:
+		case DA7219_AAD_BTN_CFG_100MS:
+		case DA7219_AAD_BTN_CFG_200MS:
+		case DA7219_AAD_BTN_CFG_500MS:
+			da7219_aad->btn_cfg  = (aad_pdata->btn_cfg <<
+						DA7219_BUTTON_CONFIG_SHIFT);
+		}
+
+		cfg = 0;
+		mask = 0;
+		switch (aad_pdata->mic_det_thr) {
+		case DA7219_AAD_MIC_DET_THR_200_OHMS:
+		case DA7219_AAD_MIC_DET_THR_500_OHMS:
+		case DA7219_AAD_MIC_DET_THR_750_OHMS:
+		case DA7219_AAD_MIC_DET_THR_1000_OHMS:
+			cfg |= (aad_pdata->mic_det_thr <<
+				DA7219_MIC_DET_THRESH_SHIFT);
+			mask |= DA7219_MIC_DET_THRESH_MASK;
+		}
+		snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, mask, cfg);
+
+		cfg = 0;
+		mask = 0;
+		switch (aad_pdata->jack_ins_deb) {
+		case DA7219_AAD_JACK_INS_DEB_5MS:
+		case DA7219_AAD_JACK_INS_DEB_10MS:
+		case DA7219_AAD_JACK_INS_DEB_20MS:
+		case DA7219_AAD_JACK_INS_DEB_50MS:
+		case DA7219_AAD_JACK_INS_DEB_100MS:
+		case DA7219_AAD_JACK_INS_DEB_200MS:
+		case DA7219_AAD_JACK_INS_DEB_500MS:
+		case DA7219_AAD_JACK_INS_DEB_1S:
+			cfg |= (aad_pdata->jack_ins_deb <<
+				DA7219_JACKDET_DEBOUNCE_SHIFT);
+			mask |= DA7219_JACKDET_DEBOUNCE_MASK;
+		}
+		switch (aad_pdata->jack_det_rate) {
+		case DA7219_AAD_JACK_DET_RATE_32_64MS:
+		case DA7219_AAD_JACK_DET_RATE_64_128MS:
+		case DA7219_AAD_JACK_DET_RATE_128_256MS:
+		case DA7219_AAD_JACK_DET_RATE_256_512MS:
+			cfg |= (aad_pdata->jack_det_rate <<
+				DA7219_JACK_DETECT_RATE_SHIFT);
+			mask |= DA7219_JACK_DETECT_RATE_MASK;
+		}
+		switch (aad_pdata->jack_rem_deb) {
+		case DA7219_AAD_JACK_REM_DEB_1MS:
+		case DA7219_AAD_JACK_REM_DEB_5MS:
+		case DA7219_AAD_JACK_REM_DEB_10MS:
+		case DA7219_AAD_JACK_REM_DEB_20MS:
+			cfg |= (aad_pdata->jack_rem_deb <<
+				DA7219_JACKDET_REM_DEB_SHIFT);
+			mask |= DA7219_JACKDET_REM_DEB_MASK;
+		}
+		snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_2, mask, cfg);
+
+		snd_soc_write(codec, DA7219_ACCDET_CONFIG_3,
+			      aad_pdata->a_d_btn_thr);
+		snd_soc_write(codec, DA7219_ACCDET_CONFIG_4,
+			      aad_pdata->d_b_btn_thr);
+		snd_soc_write(codec, DA7219_ACCDET_CONFIG_5,
+			      aad_pdata->b_c_btn_thr);
+		snd_soc_write(codec, DA7219_ACCDET_CONFIG_6,
+			      aad_pdata->c_mic_btn_thr);
+
+		cfg = 0;
+		mask = 0;
+		switch (aad_pdata->btn_avg) {
+		case DA7219_AAD_BTN_AVG_1:
+		case DA7219_AAD_BTN_AVG_2:
+		case DA7219_AAD_BTN_AVG_4:
+		case DA7219_AAD_BTN_AVG_8:
+			cfg |= (aad_pdata->btn_avg <<
+				DA7219_BUTTON_AVERAGE_SHIFT);
+			mask |= DA7219_BUTTON_AVERAGE_MASK;
+		}
+		switch (aad_pdata->adc_1bit_rpt) {
+		case DA7219_AAD_ADC_1BIT_RPT_1:
+		case DA7219_AAD_ADC_1BIT_RPT_2:
+		case DA7219_AAD_ADC_1BIT_RPT_4:
+		case DA7219_AAD_ADC_1BIT_RPT_8:
+			cfg |= (aad_pdata->adc_1bit_rpt <<
+			       DA7219_ADC_1_BIT_REPEAT_SHIFT);
+			mask |= DA7219_ADC_1_BIT_REPEAT_MASK;
+		}
+		snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_7, mask, cfg);
+	}
+}
+
+
+/*
+ * Init/Exit
+ */
+
+int da7219_aad_init(struct snd_soc_codec *codec)
+{
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	struct da7219_aad_priv *da7219_aad;
+	u8 mask[DA7219_AAD_IRQ_REG_MAX];
+	int ret;
+
+	da7219_aad = devm_kzalloc(codec->dev, sizeof(*da7219_aad), GFP_KERNEL);
+	if (!da7219_aad)
+		return -ENOMEM;
+
+	da7219->aad = da7219_aad;
+	da7219_aad->codec = codec;
+
+	/* Handle any DT/platform data */
+	if ((codec->dev->of_node) && (da7219->pdata))
+		da7219->pdata->aad_pdata = da7219_aad_of_to_pdata(codec);
+
+	da7219_aad_handle_pdata(codec);
+
+	/* Disable button detection */
+	snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1,
+			    DA7219_BUTTON_CONFIG_MASK, 0);
+
+	INIT_WORK(&da7219_aad->btn_det_work, da7219_aad_btn_det_work);
+	INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work);
+
+	ret = request_threaded_irq(da7219_aad->irq, NULL,
+				   da7219_aad_irq_thread,
+				   IRQF_TRIGGER_LOW | IRQF_ONESHOT,
+				   "da7219-aad", da7219_aad);
+	if (ret) {
+		dev_err(codec->dev, "Failed to request IRQ: %d\n", ret);
+		return ret;
+	}
+
+	/* Unmask AAD IRQs */
+	memset(mask, 0, DA7219_AAD_IRQ_REG_MAX);
+	regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A,
+			  &mask, DA7219_AAD_IRQ_REG_MAX);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(da7219_aad_init);
+
+void da7219_aad_exit(struct snd_soc_codec *codec)
+{
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	struct da7219_aad_priv *da7219_aad = da7219->aad;
+	u8 mask[DA7219_AAD_IRQ_REG_MAX];
+
+	/* Mask off AAD IRQs */
+	memset(mask, DA7219_BYTE_MASK, DA7219_AAD_IRQ_REG_MAX);
+	regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A,
+			  mask, DA7219_AAD_IRQ_REG_MAX);
+
+	free_irq(da7219_aad->irq, da7219_aad);
+
+	cancel_work_sync(&da7219_aad->btn_det_work);
+	cancel_work_sync(&da7219_aad->hptest_work);
+}
+EXPORT_SYMBOL_GPL(da7219_aad_exit);
+
+MODULE_DESCRIPTION("ASoC DA7219 AAD Driver");
+MODULE_AUTHOR("Adam Thomson <Adam.Thomson.Opensource@diasemi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/da7219-aad.h b/sound/soc/codecs/da7219-aad.h
new file mode 100644
index 000000000000..4fccf677cd06
--- /dev/null
+++ b/sound/soc/codecs/da7219-aad.h
@@ -0,0 +1,212 @@
+/*
+ * da7219-aad.h - DA7322 ASoC AAD Driver
+ *
+ * Copyright (c) 2015 Dialog Semiconductor Ltd.
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+
+#ifndef __DA7219_AAD_H
+#define __DA7219_AAD_H
+
+#include <linux/timer.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/da7219-aad.h>
+
+/*
+ * Registers
+ */
+
+#define DA7219_ACCDET_STATUS_A		0xC0
+#define DA7219_ACCDET_STATUS_B		0xC1
+#define DA7219_ACCDET_IRQ_EVENT_A	0xC2
+#define DA7219_ACCDET_IRQ_EVENT_B	0xC3
+#define DA7219_ACCDET_IRQ_MASK_A	0xC4
+#define DA7219_ACCDET_IRQ_MASK_B	0xC5
+#define DA7219_ACCDET_CONFIG_1		0xC6
+#define DA7219_ACCDET_CONFIG_2		0xC7
+#define DA7219_ACCDET_CONFIG_3		0xC8
+#define DA7219_ACCDET_CONFIG_4		0xC9
+#define DA7219_ACCDET_CONFIG_5		0xCA
+#define DA7219_ACCDET_CONFIG_6		0xCB
+#define DA7219_ACCDET_CONFIG_7		0xCC
+#define DA7219_ACCDET_CONFIG_8		0xCD
+
+
+/*
+ * Bit Fields
+ */
+
+/* DA7219_ACCDET_STATUS_A = 0xC0 */
+#define DA7219_JACK_INSERTION_STS_SHIFT	0
+#define DA7219_JACK_INSERTION_STS_MASK	(0x1 << 0)
+#define DA7219_JACK_TYPE_STS_SHIFT	1
+#define DA7219_JACK_TYPE_STS_MASK	(0x1 << 1)
+#define DA7219_JACK_PIN_ORDER_STS_SHIFT	2
+#define DA7219_JACK_PIN_ORDER_STS_MASK	(0x1 << 2)
+#define DA7219_MICBIAS_UP_STS_SHIFT	3
+#define DA7219_MICBIAS_UP_STS_MASK	(0x1 << 3)
+
+/* DA7219_ACCDET_STATUS_B = 0xC1 */
+#define DA7219_BUTTON_TYPE_STS_SHIFT	0
+#define DA7219_BUTTON_TYPE_STS_MASK	(0xFF << 0)
+
+/* DA7219_ACCDET_IRQ_EVENT_A = 0xC2 */
+#define DA7219_E_JACK_INSERTED_SHIFT		0
+#define DA7219_E_JACK_INSERTED_MASK		(0x1 << 0)
+#define DA7219_E_JACK_REMOVED_SHIFT		1
+#define DA7219_E_JACK_REMOVED_MASK		(0x1 << 1)
+#define DA7219_E_JACK_DETECT_COMPLETE_SHIFT	2
+#define DA7219_E_JACK_DETECT_COMPLETE_MASK	(0x1 << 2)
+
+/* DA7219_ACCDET_IRQ_EVENT_B = 0xC3 */
+#define DA7219_E_BUTTON_A_PRESSED_SHIFT		0
+#define DA7219_E_BUTTON_A_PRESSED_MASK		(0x1 << 0)
+#define DA7219_E_BUTTON_B_PRESSED_SHIFT		1
+#define DA7219_E_BUTTON_B_PRESSED_MASK		(0x1 << 1)
+#define DA7219_E_BUTTON_C_PRESSED_SHIFT		2
+#define DA7219_E_BUTTON_C_PRESSED_MASK		(0x1 << 2)
+#define DA7219_E_BUTTON_D_PRESSED_SHIFT		3
+#define DA7219_E_BUTTON_D_PRESSED_MASK		(0x1 << 3)
+#define DA7219_E_BUTTON_D_RELEASED_SHIFT	4
+#define DA7219_E_BUTTON_D_RELEASED_MASK		(0x1 << 4)
+#define DA7219_E_BUTTON_C_RELEASED_SHIFT	5
+#define DA7219_E_BUTTON_C_RELEASED_MASK		(0x1 << 5)
+#define DA7219_E_BUTTON_B_RELEASED_SHIFT	6
+#define DA7219_E_BUTTON_B_RELEASED_MASK		(0x1 << 6)
+#define DA7219_E_BUTTON_A_RELEASED_SHIFT	7
+#define DA7219_E_BUTTON_A_RELEASED_MASK		(0x1 << 7)
+
+/* DA7219_ACCDET_IRQ_MASK_A = 0xC4 */
+#define DA7219_M_JACK_INSERTED_SHIFT		0
+#define DA7219_M_JACK_INSERTED_MASK		(0x1 << 0)
+#define DA7219_M_JACK_REMOVED_SHIFT		1
+#define DA7219_M_JACK_REMOVED_MASK		(0x1 << 1)
+#define DA7219_M_JACK_DETECT_COMPLETE_SHIFT	2
+#define DA7219_M_JACK_DETECT_COMPLETE_MASK	(0x1 << 2)
+
+/* DA7219_ACCDET_IRQ_MASK_B = 0xC5 */
+#define DA7219_M_BUTTON_A_PRESSED_SHIFT		0
+#define DA7219_M_BUTTON_A_PRESSED_MASK		(0x1 << 0)
+#define DA7219_M_BUTTON_B_PRESSED_SHIFT		1
+#define DA7219_M_BUTTON_B_PRESSED_MASK		(0x1 << 1)
+#define DA7219_M_BUTTON_C_PRESSED_SHIFT		2
+#define DA7219_M_BUTTON_C_PRESSED_MASK		(0x1 << 2)
+#define DA7219_M_BUTTON_D_PRESSED_SHIFT		3
+#define DA7219_M_BUTTON_D_PRESSED_MASK		(0x1 << 3)
+#define DA7219_M_BUTTON_D_RELEASED_SHIFT	4
+#define DA7219_M_BUTTON_D_RELEASED_MASK		(0x1 << 4)
+#define DA7219_M_BUTTON_C_RELEASED_SHIFT	5
+#define DA7219_M_BUTTON_C_RELEASED_MASK		(0x1 << 5)
+#define DA7219_M_BUTTON_B_RELEASED_SHIFT	6
+#define DA7219_M_BUTTON_B_RELEASED_MASK		(0x1 << 6)
+#define DA7219_M_BUTTON_A_RELEASED_SHIFT	7
+#define DA7219_M_BUTTON_A_RELEASED_MASK		(0x1 << 7)
+
+/* DA7219_ACCDET_CONFIG_1 = 0xC6 */
+#define DA7219_ACCDET_EN_SHIFT		0
+#define DA7219_ACCDET_EN_MASK		(0x1 << 0)
+#define DA7219_BUTTON_CONFIG_SHIFT	1
+#define DA7219_BUTTON_CONFIG_MASK	(0x7 << 1)
+#define DA7219_MIC_DET_THRESH_SHIFT	4
+#define DA7219_MIC_DET_THRESH_MASK	(0x3 << 4)
+#define DA7219_JACK_TYPE_DET_EN_SHIFT	6
+#define DA7219_JACK_TYPE_DET_EN_MASK	(0x1 << 6)
+#define DA7219_PIN_ORDER_DET_EN_SHIFT	7
+#define DA7219_PIN_ORDER_DET_EN_MASK	(0x1 << 7)
+
+/* DA7219_ACCDET_CONFIG_2 = 0xC7 */
+#define DA7219_ACCDET_PAUSE_SHIFT	0
+#define DA7219_ACCDET_PAUSE_MASK	(0x1 << 0)
+#define DA7219_JACKDET_DEBOUNCE_SHIFT	1
+#define DA7219_JACKDET_DEBOUNCE_MASK	(0x7 << 1)
+#define DA7219_JACK_DETECT_RATE_SHIFT	4
+#define DA7219_JACK_DETECT_RATE_MASK	(0x3 << 4)
+#define DA7219_JACKDET_REM_DEB_SHIFT	6
+#define DA7219_JACKDET_REM_DEB_MASK	(0x3 << 6)
+
+/* DA7219_ACCDET_CONFIG_3 = 0xC8 */
+#define DA7219_A_D_BUTTON_THRESH_SHIFT	0
+#define DA7219_A_D_BUTTON_THRESH_MASK	(0xFF << 0)
+
+/* DA7219_ACCDET_CONFIG_4 = 0xC9 */
+#define DA7219_D_B_BUTTON_THRESH_SHIFT	0
+#define DA7219_D_B_BUTTON_THRESH_MASK	(0xFF << 0)
+
+/* DA7219_ACCDET_CONFIG_5 = 0xCA */
+#define DA7219_B_C_BUTTON_THRESH_SHIFT	0
+#define DA7219_B_C_BUTTON_THRESH_MASK	(0xFF << 0)
+
+/* DA7219_ACCDET_CONFIG_6 = 0xCB */
+#define DA7219_C_MIC_BUTTON_THRESH_SHIFT	0
+#define DA7219_C_MIC_BUTTON_THRESH_MASK		(0xFF << 0)
+
+/* DA7219_ACCDET_CONFIG_7 = 0xCC */
+#define DA7219_BUTTON_AVERAGE_SHIFT	0
+#define DA7219_BUTTON_AVERAGE_MASK	(0x3 << 0)
+#define DA7219_ADC_1_BIT_REPEAT_SHIFT	2
+#define DA7219_ADC_1_BIT_REPEAT_MASK	(0x3 << 2)
+#define DA7219_PIN_ORDER_FORCE_SHIFT	4
+#define DA7219_PIN_ORDER_FORCE_MASK	(0x1 << 4)
+#define DA7219_JACK_TYPE_FORCE_SHIFT	5
+#define DA7219_JACK_TYPE_FORCE_MASK	(0x1 << 5)
+
+/* DA7219_ACCDET_CONFIG_8 = 0xCD */
+#define DA7219_HPTEST_EN_SHIFT		0
+#define DA7219_HPTEST_EN_MASK		(0x1 << 0)
+#define DA7219_HPTEST_RES_SEL_SHIFT	1
+#define DA7219_HPTEST_RES_SEL_MASK	(0x3 << 1)
+#define DA7219_HPTEST_RES_SEL_1KOHMS	(0x0 << 1)
+#define DA7219_HPTEST_COMP_SHIFT	4
+#define DA7219_HPTEST_COMP_MASK		(0x1 << 4)
+
+
+#define DA7219_AAD_MAX_BUTTONS		4
+#define DA7219_AAD_REPORT_ALL_MASK	(SND_JACK_MECHANICAL |			\
+					 SND_JACK_HEADSET | SND_JACK_LINEOUT |	\
+					 SND_JACK_BTN_0 | SND_JACK_BTN_1 |	\
+					 SND_JACK_BTN_2 | SND_JACK_BTN_3)
+
+#define DA7219_AAD_MICBIAS_CHK_DELAY	10
+#define DA7219_AAD_MICBIAS_CHK_RETRIES	5
+
+#define DA7219_AAD_HPTEST_RAMP_FREQ	0x28
+#define DA7219_AAD_HPTEST_PERIOD	65
+
+enum da7219_aad_event_regs {
+	DA7219_AAD_IRQ_REG_A = 0,
+	DA7219_AAD_IRQ_REG_B,
+	DA7219_AAD_IRQ_REG_MAX,
+};
+
+/* Private data */
+struct da7219_aad_priv {
+	struct snd_soc_codec *codec;
+	int irq;
+
+	u8 micbias_pulse_lvl;
+	u32 micbias_pulse_time;
+
+	u8 btn_cfg;
+
+	struct work_struct btn_det_work;
+	struct work_struct hptest_work;
+
+	struct snd_soc_jack *jack;
+	bool jack_inserted;
+};
+
+/* AAD control */
+void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack);
+
+/* Init/Exit */
+int da7219_aad_init(struct snd_soc_codec *codec);
+void da7219_aad_exit(struct snd_soc_codec *codec);
+
+#endif /* __DA7219_AAD_H */
diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c
new file mode 100644
index 000000000000..f238c1e8a69c
--- /dev/null
+++ b/sound/soc/codecs/da7219.c
@@ -0,0 +1,1955 @@
+/*
+ * da7219.c - DA7219 ALSA SoC Codec Driver
+ *
+ * Copyright (c) 2015 Dialog Semiconductor
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.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/clk.h>
+#include <linux/i2c.h>
+#include <linux/of_device.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/pm.h>
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/regulator/consumer.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <asm/div64.h>
+
+#include <sound/da7219.h>
+#include "da7219.h"
+#include "da7219-aad.h"
+
+
+/*
+ * TLVs and Enums
+ */
+
+/* Input TLVs */
+static const DECLARE_TLV_DB_SCALE(da7219_mic_gain_tlv, -600, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_mixin_gain_tlv, -450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_adc_dig_gain_tlv, -8325, 75, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_alc_threshold_tlv, -9450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_alc_gain_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_alc_ana_gain_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_sidetone_gain_tlv, -4200, 300, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_tonegen_gain_tlv, -4500, 300, 0);
+
+/* Output TLVs */
+static const DECLARE_TLV_DB_SCALE(da7219_dac_eq_band_tlv, -1050, 150, 0);
+
+static const DECLARE_TLV_DB_RANGE(da7219_dac_dig_gain_tlv,
+	0x0, 0x07, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1),
+	/* -77.25dB to 12dB */
+	0x08, 0x7f, TLV_DB_SCALE_ITEM(-7725, 75, 0)
+);
+
+static const DECLARE_TLV_DB_SCALE(da7219_dac_ng_threshold_tlv, -10200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_hp_gain_tlv, -5700, 100, 0);
+
+/* Input Enums */
+static const char * const da7219_alc_attack_rate_txt[] = {
+	"7.33/fs", "14.66/fs", "29.32/fs", "58.64/fs", "117.3/fs", "234.6/fs",
+	"469.1/fs", "938.2/fs", "1876/fs", "3753/fs", "7506/fs", "15012/fs",
+	"30024/fs"
+};
+
+static const struct soc_enum da7219_alc_attack_rate =
+	SOC_ENUM_SINGLE(DA7219_ALC_CTRL2, DA7219_ALC_ATTACK_SHIFT,
+			DA7219_ALC_ATTACK_MAX, da7219_alc_attack_rate_txt);
+
+static const char * const da7219_alc_release_rate_txt[] = {
+	"28.66/fs", "57.33/fs", "114.6/fs", "229.3/fs", "458.6/fs", "917.1/fs",
+	"1834/fs", "3668/fs", "7337/fs", "14674/fs", "29348/fs"
+};
+
+static const struct soc_enum da7219_alc_release_rate =
+	SOC_ENUM_SINGLE(DA7219_ALC_CTRL2, DA7219_ALC_RELEASE_SHIFT,
+			DA7219_ALC_RELEASE_MAX, da7219_alc_release_rate_txt);
+
+static const char * const da7219_alc_hold_time_txt[] = {
+	"62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs",
+	"7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs",
+	"253952/fs", "507904/fs", "1015808/fs", "2031616/fs"
+};
+
+static const struct soc_enum da7219_alc_hold_time =
+	SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_HOLD_SHIFT,
+			DA7219_ALC_HOLD_MAX, da7219_alc_hold_time_txt);
+
+static const char * const da7219_alc_env_rate_txt[] = {
+	"1/4", "1/16", "1/256", "1/65536"
+};
+
+static const struct soc_enum da7219_alc_env_attack_rate =
+	SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_INTEG_ATTACK_SHIFT,
+			DA7219_ALC_INTEG_MAX, da7219_alc_env_rate_txt);
+
+static const struct soc_enum da7219_alc_env_release_rate =
+	SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_INTEG_RELEASE_SHIFT,
+			DA7219_ALC_INTEG_MAX, da7219_alc_env_rate_txt);
+
+static const char * const da7219_alc_anticlip_step_txt[] = {
+	"0.034dB/fs", "0.068dB/fs", "0.136dB/fs", "0.272dB/fs"
+};
+
+static const struct soc_enum da7219_alc_anticlip_step =
+	SOC_ENUM_SINGLE(DA7219_ALC_ANTICLIP_CTRL,
+			DA7219_ALC_ANTICLIP_STEP_SHIFT,
+			DA7219_ALC_ANTICLIP_STEP_MAX,
+			da7219_alc_anticlip_step_txt);
+
+/* Input/Output Enums */
+static const char * const da7219_gain_ramp_rate_txt[] = {
+	"Nominal Rate * 8", "Nominal Rate", "Nominal Rate / 8",
+	"Nominal Rate / 16"
+};
+
+static const struct soc_enum da7219_gain_ramp_rate =
+	SOC_ENUM_SINGLE(DA7219_GAIN_RAMP_CTRL, DA7219_GAIN_RAMP_RATE_SHIFT,
+			DA7219_GAIN_RAMP_RATE_MAX, da7219_gain_ramp_rate_txt);
+
+static const char * const da7219_hpf_mode_txt[] = {
+	"Disabled", "Audio", "Voice"
+};
+
+static const unsigned int da7219_hpf_mode_val[] = {
+	DA7219_HPF_DISABLED, DA7219_HPF_AUDIO_EN, DA7219_HPF_VOICE_EN,
+};
+
+static const struct soc_enum da7219_adc_hpf_mode =
+	SOC_VALUE_ENUM_SINGLE(DA7219_ADC_FILTERS1, DA7219_HPF_MODE_SHIFT,
+			      DA7219_HPF_MODE_MASK, DA7219_HPF_MODE_MAX,
+			      da7219_hpf_mode_txt, da7219_hpf_mode_val);
+
+static const struct soc_enum da7219_dac_hpf_mode =
+	SOC_VALUE_ENUM_SINGLE(DA7219_DAC_FILTERS1, DA7219_HPF_MODE_SHIFT,
+			      DA7219_HPF_MODE_MASK, DA7219_HPF_MODE_MAX,
+			      da7219_hpf_mode_txt, da7219_hpf_mode_val);
+
+static const char * const da7219_audio_hpf_corner_txt[] = {
+	"2Hz", "4Hz", "8Hz", "16Hz"
+};
+
+static const struct soc_enum da7219_adc_audio_hpf_corner =
+	SOC_ENUM_SINGLE(DA7219_ADC_FILTERS1,
+			DA7219_ADC_AUDIO_HPF_CORNER_SHIFT,
+			DA7219_AUDIO_HPF_CORNER_MAX,
+			da7219_audio_hpf_corner_txt);
+
+static const struct soc_enum da7219_dac_audio_hpf_corner =
+	SOC_ENUM_SINGLE(DA7219_DAC_FILTERS1,
+			DA7219_DAC_AUDIO_HPF_CORNER_SHIFT,
+			DA7219_AUDIO_HPF_CORNER_MAX,
+			da7219_audio_hpf_corner_txt);
+
+static const char * const da7219_voice_hpf_corner_txt[] = {
+	"2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz"
+};
+
+static const struct soc_enum da7219_adc_voice_hpf_corner =
+	SOC_ENUM_SINGLE(DA7219_ADC_FILTERS1,
+			DA7219_ADC_VOICE_HPF_CORNER_SHIFT,
+			DA7219_VOICE_HPF_CORNER_MAX,
+			da7219_voice_hpf_corner_txt);
+
+static const struct soc_enum da7219_dac_voice_hpf_corner =
+	SOC_ENUM_SINGLE(DA7219_DAC_FILTERS1,
+			DA7219_DAC_VOICE_HPF_CORNER_SHIFT,
+			DA7219_VOICE_HPF_CORNER_MAX,
+			da7219_voice_hpf_corner_txt);
+
+static const char * const da7219_tonegen_dtmf_key_txt[] = {
+	"0", "1", "2", "3", "4", "5", "6", "7", "8", "9", "A", "B", "C", "D",
+	"*", "#"
+};
+
+static const struct soc_enum da7219_tonegen_dtmf_key =
+	SOC_ENUM_SINGLE(DA7219_TONE_GEN_CFG1, DA7219_DTMF_REG_SHIFT,
+			DA7219_DTMF_REG_MAX, da7219_tonegen_dtmf_key_txt);
+
+static const char * const da7219_tonegen_swg_sel_txt[] = {
+	"Sum", "SWG1", "SWG2", "SWG1_1-Cos"
+};
+
+static const struct soc_enum da7219_tonegen_swg_sel =
+	SOC_ENUM_SINGLE(DA7219_TONE_GEN_CFG2, DA7219_SWG_SEL_SHIFT,
+			DA7219_SWG_SEL_MAX, da7219_tonegen_swg_sel_txt);
+
+/* Output Enums */
+static const char * const da7219_dac_softmute_rate_txt[] = {
+	"1 Sample", "2 Samples", "4 Samples", "8 Samples", "16 Samples",
+	"32 Samples", "64 Samples"
+};
+
+static const struct soc_enum da7219_dac_softmute_rate =
+	SOC_ENUM_SINGLE(DA7219_DAC_FILTERS5, DA7219_DAC_SOFTMUTE_RATE_SHIFT,
+			DA7219_DAC_SOFTMUTE_RATE_MAX,
+			da7219_dac_softmute_rate_txt);
+
+static const char * const da7219_dac_ng_setup_time_txt[] = {
+	"256 Samples", "512 Samples", "1024 Samples", "2048 Samples"
+};
+
+static const struct soc_enum da7219_dac_ng_setup_time =
+	SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME,
+			DA7219_DAC_NG_SETUP_TIME_SHIFT,
+			DA7219_DAC_NG_SETUP_TIME_MAX,
+			da7219_dac_ng_setup_time_txt);
+
+static const char * const da7219_dac_ng_rampup_txt[] = {
+	"0.22ms/dB", "0.0138ms/dB"
+};
+
+static const struct soc_enum da7219_dac_ng_rampup_rate =
+	SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME,
+			DA7219_DAC_NG_RAMPUP_RATE_SHIFT,
+			DA7219_DAC_NG_RAMP_RATE_MAX,
+			da7219_dac_ng_rampup_txt);
+
+static const char * const da7219_dac_ng_rampdown_txt[] = {
+	"0.88ms/dB", "14.08ms/dB"
+};
+
+static const struct soc_enum da7219_dac_ng_rampdown_rate =
+	SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME,
+			DA7219_DAC_NG_RAMPDN_RATE_SHIFT,
+			DA7219_DAC_NG_RAMP_RATE_MAX,
+			da7219_dac_ng_rampdown_txt);
+
+
+static const char * const da7219_cp_track_mode_txt[] = {
+	"Largest Volume", "DAC Volume", "Signal Magnitude"
+};
+
+static const unsigned int da7219_cp_track_mode_val[] = {
+	DA7219_CP_MCHANGE_LARGEST_VOL, DA7219_CP_MCHANGE_DAC_VOL,
+	DA7219_CP_MCHANGE_SIG_MAG
+};
+
+static const struct soc_enum da7219_cp_track_mode =
+	SOC_VALUE_ENUM_SINGLE(DA7219_CP_CTRL, DA7219_CP_MCHANGE_SHIFT,
+			      DA7219_CP_MCHANGE_REL_MASK, DA7219_CP_MCHANGE_MAX,
+			      da7219_cp_track_mode_txt,
+			      da7219_cp_track_mode_val);
+
+
+/*
+ * Control Functions
+ */
+
+/* Locked Kcontrol calls */
+static int da7219_volsw_locked_get(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	mutex_lock(&da7219->lock);
+	ret = snd_soc_get_volsw(kcontrol, ucontrol);
+	mutex_unlock(&da7219->lock);
+
+	return ret;
+}
+
+static int da7219_volsw_locked_put(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	mutex_lock(&da7219->lock);
+	ret = snd_soc_put_volsw(kcontrol, ucontrol);
+	mutex_unlock(&da7219->lock);
+
+	return ret;
+}
+
+static int da7219_enum_locked_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	mutex_lock(&da7219->lock);
+	ret = snd_soc_get_enum_double(kcontrol, ucontrol);
+	mutex_unlock(&da7219->lock);
+
+	return ret;
+}
+
+static int da7219_enum_locked_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	mutex_lock(&da7219->lock);
+	ret = snd_soc_put_enum_double(kcontrol, ucontrol);
+	mutex_unlock(&da7219->lock);
+
+	return ret;
+}
+
+/* ALC */
+static void da7219_alc_calib(struct snd_soc_codec *codec)
+{
+	u8 mic_ctrl, mixin_ctrl, adc_ctrl, calib_ctrl;
+
+	/* Save current state of mic control register */
+	mic_ctrl = snd_soc_read(codec, DA7219_MIC_1_CTRL);
+
+	/* Save current state of input mixer control register */
+	mixin_ctrl = snd_soc_read(codec, DA7219_MIXIN_L_CTRL);
+
+	/* Save current state of input ADC control register */
+	adc_ctrl = snd_soc_read(codec, DA7219_ADC_L_CTRL);
+
+	/* Enable then Mute MIC PGAs */
+	snd_soc_update_bits(codec, DA7219_MIC_1_CTRL, DA7219_MIC_1_AMP_EN_MASK,
+			    DA7219_MIC_1_AMP_EN_MASK);
+	snd_soc_update_bits(codec, DA7219_MIC_1_CTRL,
+			    DA7219_MIC_1_AMP_MUTE_EN_MASK,
+			    DA7219_MIC_1_AMP_MUTE_EN_MASK);
+
+	/* Enable input mixers unmuted */
+	snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL,
+			    DA7219_MIXIN_L_AMP_EN_MASK |
+			    DA7219_MIXIN_L_AMP_MUTE_EN_MASK,
+			    DA7219_MIXIN_L_AMP_EN_MASK);
+
+	/* Enable input filters unmuted */
+	snd_soc_update_bits(codec, DA7219_ADC_L_CTRL,
+			    DA7219_ADC_L_MUTE_EN_MASK | DA7219_ADC_L_EN_MASK,
+			    DA7219_ADC_L_EN_MASK);
+
+	/* Perform auto calibration */
+	snd_soc_update_bits(codec, DA7219_ALC_CTRL1,
+			    DA7219_ALC_AUTO_CALIB_EN_MASK,
+			    DA7219_ALC_AUTO_CALIB_EN_MASK);
+	do {
+		calib_ctrl = snd_soc_read(codec, DA7219_ALC_CTRL1);
+	} while (calib_ctrl & DA7219_ALC_AUTO_CALIB_EN_MASK);
+
+	/* If auto calibration fails, disable DC offset, hybrid ALC */
+	if (calib_ctrl & DA7219_ALC_CALIB_OVERFLOW_MASK) {
+		dev_warn(codec->dev,
+			 "ALC auto calibration failed with overflow\n");
+		snd_soc_update_bits(codec, DA7219_ALC_CTRL1,
+				    DA7219_ALC_OFFSET_EN_MASK |
+				    DA7219_ALC_SYNC_MODE_MASK, 0);
+	} else {
+		/* Enable DC offset cancellation, hybrid mode */
+		snd_soc_update_bits(codec, DA7219_ALC_CTRL1,
+				    DA7219_ALC_OFFSET_EN_MASK |
+				    DA7219_ALC_SYNC_MODE_MASK,
+				    DA7219_ALC_OFFSET_EN_MASK |
+				    DA7219_ALC_SYNC_MODE_MASK);
+	}
+
+	/* Restore input filter control register to original state */
+	snd_soc_write(codec, DA7219_ADC_L_CTRL, adc_ctrl);
+
+	/* Restore input mixer control registers to original state */
+	snd_soc_write(codec, DA7219_MIXIN_L_CTRL, mixin_ctrl);
+
+	/* Restore MIC control registers to original states */
+	snd_soc_write(codec, DA7219_MIC_1_CTRL, mic_ctrl);
+}
+
+static int da7219_mixin_gain_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	ret = snd_soc_put_volsw(kcontrol, ucontrol);
+
+	/*
+	 * If ALC in operation and value of control has been updated,
+	 * make sure calibrated offsets are updated.
+	 */
+	if ((ret == 1) && (da7219->alc_en))
+		da7219_alc_calib(codec);
+
+	return ret;
+}
+
+static int da7219_alc_sw_put(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+
+	/* Force ALC offset calibration if enabling ALC */
+	if ((ucontrol->value.integer.value[0]) && (!da7219->alc_en)) {
+		da7219_alc_calib(codec);
+		da7219->alc_en = true;
+	} else {
+		da7219->alc_en = false;
+	}
+
+	return snd_soc_put_volsw(kcontrol, ucontrol);
+}
+
+/* ToneGen */
+static int da7219_tonegen_freq_get(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	struct soc_mixer_control *mixer_ctrl =
+		(struct soc_mixer_control *) kcontrol->private_value;
+	unsigned int reg = mixer_ctrl->reg;
+	u16 val;
+	int ret;
+
+	mutex_lock(&da7219->lock);
+	ret = regmap_raw_read(da7219->regmap, reg, &val, sizeof(val));
+	mutex_unlock(&da7219->lock);
+
+	if (ret)
+		return ret;
+
+	/*
+	 * Frequency value spans two 8-bit registers, lower then upper byte.
+	 * Therefore we need to convert to host endianness here.
+	 */
+	ucontrol->value.integer.value[0] = le16_to_cpu(val);
+
+	return 0;
+}
+
+static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	struct soc_mixer_control *mixer_ctrl =
+		(struct soc_mixer_control *) kcontrol->private_value;
+	unsigned int reg = mixer_ctrl->reg;
+	u16 val;
+	int ret;
+
+	/*
+	 * Frequency value spans two 8-bit registers, lower then upper byte.
+	 * Therefore we need to convert to little endian here to align with
+	 * HW registers.
+	 */
+	val = cpu_to_le16(ucontrol->value.integer.value[0]);
+
+	mutex_lock(&da7219->lock);
+	ret = regmap_raw_write(da7219->regmap, reg, &val, sizeof(val));
+	mutex_unlock(&da7219->lock);
+
+	return ret;
+}
+
+
+/*
+ * KControls
+ */
+
+static const struct snd_kcontrol_new da7219_snd_controls[] = {
+	/* Mics */
+	SOC_SINGLE_TLV("Mic Volume", DA7219_MIC_1_GAIN,
+		       DA7219_MIC_1_AMP_GAIN_SHIFT, DA7219_MIC_1_AMP_GAIN_MAX,
+		       DA7219_NO_INVERT, da7219_mic_gain_tlv),
+	SOC_SINGLE("Mic Switch", DA7219_MIC_1_CTRL,
+		   DA7219_MIC_1_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+		   DA7219_INVERT),
+
+	/* Mixer Input */
+	SOC_SINGLE_EXT_TLV("Mixin Volume", DA7219_MIXIN_L_GAIN,
+			   DA7219_MIXIN_L_AMP_GAIN_SHIFT,
+			   DA7219_MIXIN_L_AMP_GAIN_MAX, DA7219_NO_INVERT,
+			   snd_soc_get_volsw, da7219_mixin_gain_put,
+			   da7219_mixin_gain_tlv),
+	SOC_SINGLE("Mixin Switch", DA7219_MIXIN_L_CTRL,
+		   DA7219_MIXIN_L_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+		   DA7219_INVERT),
+	SOC_SINGLE("Mixin Gain Ramp Switch", DA7219_MIXIN_L_CTRL,
+		   DA7219_MIXIN_L_AMP_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+		   DA7219_NO_INVERT),
+	SOC_SINGLE("Mixin ZC Gain Switch", DA7219_MIXIN_L_CTRL,
+		   DA7219_MIXIN_L_AMP_ZC_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+		   DA7219_NO_INVERT),
+
+	/* ADC */
+	SOC_SINGLE_TLV("Capture Digital Volume", DA7219_ADC_L_GAIN,
+		       DA7219_ADC_L_DIGITAL_GAIN_SHIFT,
+		       DA7219_ADC_L_DIGITAL_GAIN_MAX, DA7219_NO_INVERT,
+		       da7219_adc_dig_gain_tlv),
+	SOC_SINGLE("Capture Digital Switch", DA7219_ADC_L_CTRL,
+		   DA7219_ADC_L_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+		   DA7219_INVERT),
+	SOC_SINGLE("Capture Digital Gain Ramp Switch", DA7219_ADC_L_CTRL,
+		   DA7219_ADC_L_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+		   DA7219_NO_INVERT),
+
+	/* ALC */
+	SOC_ENUM("ALC Attack Rate", da7219_alc_attack_rate),
+	SOC_ENUM("ALC Release Rate", da7219_alc_release_rate),
+	SOC_ENUM("ALC Hold Time", da7219_alc_hold_time),
+	SOC_ENUM("ALC Envelope Attack Rate", da7219_alc_env_attack_rate),
+	SOC_ENUM("ALC Envelope Release Rate", da7219_alc_env_release_rate),
+	SOC_SINGLE_TLV("ALC Noise Threshold", DA7219_ALC_NOISE,
+		       DA7219_ALC_NOISE_SHIFT, DA7219_ALC_THRESHOLD_MAX,
+		       DA7219_INVERT, da7219_alc_threshold_tlv),
+	SOC_SINGLE_TLV("ALC Min Threshold", DA7219_ALC_TARGET_MIN,
+		       DA7219_ALC_THRESHOLD_MIN_SHIFT, DA7219_ALC_THRESHOLD_MAX,
+		       DA7219_INVERT, da7219_alc_threshold_tlv),
+	SOC_SINGLE_TLV("ALC Max Threshold", DA7219_ALC_TARGET_MAX,
+		       DA7219_ALC_THRESHOLD_MAX_SHIFT, DA7219_ALC_THRESHOLD_MAX,
+		       DA7219_INVERT, da7219_alc_threshold_tlv),
+	SOC_SINGLE_TLV("ALC Max Attenuation", DA7219_ALC_GAIN_LIMITS,
+		       DA7219_ALC_ATTEN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX,
+		       DA7219_NO_INVERT, da7219_alc_gain_tlv),
+	SOC_SINGLE_TLV("ALC Max Volume", DA7219_ALC_GAIN_LIMITS,
+		       DA7219_ALC_GAIN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX,
+		       DA7219_NO_INVERT, da7219_alc_gain_tlv),
+	SOC_SINGLE_RANGE_TLV("ALC Min Analog Volume", DA7219_ALC_ANA_GAIN_LIMITS,
+			     DA7219_ALC_ANA_GAIN_MIN_SHIFT,
+			     DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX,
+			     DA7219_NO_INVERT, da7219_alc_ana_gain_tlv),
+	SOC_SINGLE_RANGE_TLV("ALC Max Analog Volume", DA7219_ALC_ANA_GAIN_LIMITS,
+			     DA7219_ALC_ANA_GAIN_MAX_SHIFT,
+			     DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX,
+			     DA7219_NO_INVERT, da7219_alc_ana_gain_tlv),
+	SOC_ENUM("ALC Anticlip Step", da7219_alc_anticlip_step),
+	SOC_SINGLE("ALC Anticlip Switch", DA7219_ALC_ANTICLIP_CTRL,
+		   DA7219_ALC_ANTIPCLIP_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+		   DA7219_NO_INVERT),
+	SOC_SINGLE_EXT("ALC Switch", DA7219_ALC_CTRL1, DA7219_ALC_EN_SHIFT,
+		       DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT,
+		       snd_soc_get_volsw, da7219_alc_sw_put),
+
+	/* Input High-Pass Filters */
+	SOC_ENUM("ADC HPF Mode", da7219_adc_hpf_mode),
+	SOC_ENUM("ADC HPF Corner Audio", da7219_adc_audio_hpf_corner),
+	SOC_ENUM("ADC HPF Corner Voice", da7219_adc_voice_hpf_corner),
+
+	/* Sidetone Filter */
+	SOC_SINGLE_TLV("Sidetone Volume", DA7219_SIDETONE_GAIN,
+		       DA7219_SIDETONE_GAIN_SHIFT, DA7219_SIDETONE_GAIN_MAX,
+		       DA7219_NO_INVERT, da7219_sidetone_gain_tlv),
+	SOC_SINGLE("Sidetone Switch", DA7219_SIDETONE_CTRL,
+		   DA7219_SIDETONE_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+		   DA7219_INVERT),
+
+	/* Tone Generator */
+	SOC_SINGLE_EXT_TLV("ToneGen Volume", DA7219_TONE_GEN_CFG2,
+			   DA7219_TONE_GEN_GAIN_SHIFT, DA7219_TONE_GEN_GAIN_MAX,
+			   DA7219_NO_INVERT, da7219_volsw_locked_get,
+			   da7219_volsw_locked_put, da7219_tonegen_gain_tlv),
+	SOC_ENUM_EXT("ToneGen DTMF Key", da7219_tonegen_dtmf_key,
+		     da7219_enum_locked_get, da7219_enum_locked_put),
+	SOC_SINGLE_EXT("ToneGen DTMF Switch", DA7219_TONE_GEN_CFG1,
+		       DA7219_DTMF_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+		       DA7219_NO_INVERT, da7219_volsw_locked_get,
+		       da7219_volsw_locked_put),
+	SOC_ENUM_EXT("ToneGen Sinewave Gen Type", da7219_tonegen_swg_sel,
+		     da7219_enum_locked_get, da7219_enum_locked_put),
+	SOC_SINGLE_EXT("ToneGen Sinewave1 Freq", DA7219_TONE_GEN_FREQ1_L,
+		       DA7219_FREQ1_L_SHIFT, DA7219_FREQ_MAX, DA7219_NO_INVERT,
+		       da7219_tonegen_freq_get, da7219_tonegen_freq_put),
+	SOC_SINGLE_EXT("ToneGen Sinewave2 Freq", DA7219_TONE_GEN_FREQ2_L,
+		       DA7219_FREQ2_L_SHIFT, DA7219_FREQ_MAX, DA7219_NO_INVERT,
+		       da7219_tonegen_freq_get, da7219_tonegen_freq_put),
+	SOC_SINGLE_EXT("ToneGen On Time", DA7219_TONE_GEN_ON_PER,
+		       DA7219_BEEP_ON_PER_SHIFT, DA7219_BEEP_ON_OFF_MAX,
+		       DA7219_NO_INVERT, da7219_volsw_locked_get,
+		       da7219_volsw_locked_put),
+	SOC_SINGLE("ToneGen Off Time", DA7219_TONE_GEN_OFF_PER,
+		   DA7219_BEEP_OFF_PER_SHIFT, DA7219_BEEP_ON_OFF_MAX,
+		   DA7219_NO_INVERT),
+
+	/* Gain ramping */
+	SOC_ENUM("Gain Ramp Rate", da7219_gain_ramp_rate),
+
+	/* DAC High-Pass Filter */
+	SOC_ENUM_EXT("DAC HPF Mode", da7219_dac_hpf_mode,
+		     da7219_enum_locked_get, da7219_enum_locked_put),
+	SOC_ENUM("DAC HPF Corner Audio", da7219_dac_audio_hpf_corner),
+	SOC_ENUM("DAC HPF Corner Voice", da7219_dac_voice_hpf_corner),
+
+	/* DAC 5-Band Equaliser */
+	SOC_SINGLE_TLV("DAC EQ Band1 Volume", DA7219_DAC_FILTERS2,
+		       DA7219_DAC_EQ_BAND1_SHIFT, DA7219_DAC_EQ_BAND_MAX,
+		       DA7219_NO_INVERT, da7219_dac_eq_band_tlv),
+	SOC_SINGLE_TLV("DAC EQ Band2 Volume", DA7219_DAC_FILTERS2,
+		       DA7219_DAC_EQ_BAND2_SHIFT, DA7219_DAC_EQ_BAND_MAX,
+		       DA7219_NO_INVERT, da7219_dac_eq_band_tlv),
+	SOC_SINGLE_TLV("DAC EQ Band3 Volume", DA7219_DAC_FILTERS3,
+		       DA7219_DAC_EQ_BAND3_SHIFT, DA7219_DAC_EQ_BAND_MAX,
+		       DA7219_NO_INVERT, da7219_dac_eq_band_tlv),
+	SOC_SINGLE_TLV("DAC EQ Band4 Volume", DA7219_DAC_FILTERS3,
+		       DA7219_DAC_EQ_BAND4_SHIFT, DA7219_DAC_EQ_BAND_MAX,
+		       DA7219_NO_INVERT, da7219_dac_eq_band_tlv),
+	SOC_SINGLE_TLV("DAC EQ Band5 Volume", DA7219_DAC_FILTERS4,
+		       DA7219_DAC_EQ_BAND5_SHIFT, DA7219_DAC_EQ_BAND_MAX,
+		       DA7219_NO_INVERT, da7219_dac_eq_band_tlv),
+	SOC_SINGLE_EXT("DAC EQ Switch", DA7219_DAC_FILTERS4,
+		       DA7219_DAC_EQ_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+		       DA7219_NO_INVERT, da7219_volsw_locked_get,
+		       da7219_volsw_locked_put),
+
+	/* DAC Softmute */
+	SOC_ENUM("DAC Soft Mute Rate", da7219_dac_softmute_rate),
+	SOC_SINGLE_EXT("DAC Soft Mute Switch", DA7219_DAC_FILTERS5,
+		       DA7219_DAC_SOFTMUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+		       DA7219_NO_INVERT, da7219_volsw_locked_get,
+		       da7219_volsw_locked_put),
+
+	/* DAC Noise Gate */
+	SOC_ENUM("DAC NG Setup Time", da7219_dac_ng_setup_time),
+	SOC_ENUM("DAC NG Rampup Rate", da7219_dac_ng_rampup_rate),
+	SOC_ENUM("DAC NG Rampdown Rate", da7219_dac_ng_rampdown_rate),
+	SOC_SINGLE_TLV("DAC NG Off Threshold", DA7219_DAC_NG_OFF_THRESH,
+		       DA7219_DAC_NG_OFF_THRESHOLD_SHIFT,
+		       DA7219_DAC_NG_THRESHOLD_MAX, DA7219_NO_INVERT,
+		       da7219_dac_ng_threshold_tlv),
+	SOC_SINGLE_TLV("DAC NG On Threshold", DA7219_DAC_NG_ON_THRESH,
+		       DA7219_DAC_NG_ON_THRESHOLD_SHIFT,
+		       DA7219_DAC_NG_THRESHOLD_MAX, DA7219_NO_INVERT,
+		       da7219_dac_ng_threshold_tlv),
+	SOC_SINGLE("DAC NG Switch", DA7219_DAC_NG_CTRL, DA7219_DAC_NG_EN_SHIFT,
+		   DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+
+	/* DACs */
+	SOC_DOUBLE_R_EXT_TLV("Playback Digital Volume", DA7219_DAC_L_GAIN,
+			     DA7219_DAC_R_GAIN, DA7219_DAC_L_DIGITAL_GAIN_SHIFT,
+			     DA7219_DAC_DIGITAL_GAIN_MAX, DA7219_NO_INVERT,
+			     da7219_volsw_locked_get, da7219_volsw_locked_put,
+			     da7219_dac_dig_gain_tlv),
+	SOC_DOUBLE_R_EXT("Playback Digital Switch", DA7219_DAC_L_CTRL,
+			 DA7219_DAC_R_CTRL, DA7219_DAC_L_MUTE_EN_SHIFT,
+			 DA7219_SWITCH_EN_MAX, DA7219_INVERT,
+			 da7219_volsw_locked_get, da7219_volsw_locked_put),
+	SOC_DOUBLE_R("Playback Digital Gain Ramp Switch", DA7219_DAC_L_CTRL,
+		     DA7219_DAC_R_CTRL, DA7219_DAC_L_RAMP_EN_SHIFT,
+		     DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+
+	/* CP */
+	SOC_ENUM("Charge Pump Track Mode", da7219_cp_track_mode),
+	SOC_SINGLE("Charge Pump Threshold", DA7219_CP_VOL_THRESHOLD1,
+		   DA7219_CP_THRESH_VDD2_SHIFT, DA7219_CP_THRESH_VDD2_MAX,
+		   DA7219_NO_INVERT),
+
+	/* Headphones */
+	SOC_DOUBLE_R_EXT_TLV("Headphone Volume", DA7219_HP_L_GAIN,
+			     DA7219_HP_R_GAIN, DA7219_HP_L_AMP_GAIN_SHIFT,
+			     DA7219_HP_AMP_GAIN_MAX, DA7219_NO_INVERT,
+			     da7219_volsw_locked_get, da7219_volsw_locked_put,
+			     da7219_hp_gain_tlv),
+	SOC_DOUBLE_R_EXT("Headphone Switch", DA7219_HP_L_CTRL, DA7219_HP_R_CTRL,
+			 DA7219_HP_L_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+			 DA7219_INVERT, da7219_volsw_locked_get,
+			 da7219_volsw_locked_put),
+	SOC_DOUBLE_R("Headphone Gain Ramp Switch", DA7219_HP_L_CTRL,
+		     DA7219_HP_R_CTRL, DA7219_HP_L_AMP_RAMP_EN_SHIFT,
+		     DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+	SOC_DOUBLE_R("Headphone ZC Gain Switch", DA7219_HP_L_CTRL,
+		     DA7219_HP_R_CTRL, DA7219_HP_L_AMP_ZC_EN_SHIFT,
+		     DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+};
+
+
+/*
+ * DAPM Mux Controls
+ */
+
+static const char * const da7219_out_sel_txt[] = {
+	"ADC", "Tone Generator", "DAIL", "DAIR"
+};
+
+static const struct soc_enum da7219_out_dail_sel =
+	SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAI,
+			DA7219_DAI_L_SRC_SHIFT,
+			DA7219_OUT_SRC_MAX,
+			da7219_out_sel_txt);
+
+static const struct snd_kcontrol_new da7219_out_dail_sel_mux =
+	SOC_DAPM_ENUM("Out DAIL Mux", da7219_out_dail_sel);
+
+static const struct soc_enum da7219_out_dair_sel =
+	SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAI,
+			DA7219_DAI_R_SRC_SHIFT,
+			DA7219_OUT_SRC_MAX,
+			da7219_out_sel_txt);
+
+static const struct snd_kcontrol_new da7219_out_dair_sel_mux =
+	SOC_DAPM_ENUM("Out DAIR Mux", da7219_out_dair_sel);
+
+static const struct soc_enum da7219_out_dacl_sel =
+	SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAC,
+			DA7219_DAC_L_SRC_SHIFT,
+			DA7219_OUT_SRC_MAX,
+			da7219_out_sel_txt);
+
+static const struct snd_kcontrol_new da7219_out_dacl_sel_mux =
+	SOC_DAPM_ENUM("Out DACL Mux", da7219_out_dacl_sel);
+
+static const struct soc_enum da7219_out_dacr_sel =
+	SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAC,
+			DA7219_DAC_R_SRC_SHIFT,
+			DA7219_OUT_SRC_MAX,
+			da7219_out_sel_txt);
+
+static const struct snd_kcontrol_new da7219_out_dacr_sel_mux =
+	SOC_DAPM_ENUM("Out DACR Mux", da7219_out_dacr_sel);
+
+
+/*
+ * DAPM Mixer Controls
+ */
+
+static const struct snd_kcontrol_new da7219_mixin_controls[] = {
+	SOC_DAPM_SINGLE("Mic Switch", DA7219_MIXIN_L_SELECT,
+			DA7219_MIXIN_L_MIX_SELECT_SHIFT,
+			DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+};
+
+static const struct snd_kcontrol_new da7219_mixout_l_controls[] = {
+	SOC_DAPM_SINGLE("DACL Switch", DA7219_MIXOUT_L_SELECT,
+			DA7219_MIXOUT_L_MIX_SELECT_SHIFT,
+			DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+};
+
+static const struct snd_kcontrol_new da7219_mixout_r_controls[] = {
+	SOC_DAPM_SINGLE("DACR Switch", DA7219_MIXOUT_R_SELECT,
+			DA7219_MIXOUT_R_MIX_SELECT_SHIFT,
+			DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+};
+
+#define DA7219_DMIX_ST_CTRLS(reg)					\
+	SOC_DAPM_SINGLE("Out FilterL Switch", reg,			\
+			DA7219_DMIX_ST_SRC_OUTFILT1L_SHIFT,		\
+			DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),	\
+	SOC_DAPM_SINGLE("Out FilterR Switch", reg,			\
+			DA7219_DMIX_ST_SRC_OUTFILT1R_SHIFT,		\
+			DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),	\
+	SOC_DAPM_SINGLE("Sidetone Switch", reg,				\
+			DA7219_DMIX_ST_SRC_SIDETONE_SHIFT,		\
+			DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT)		\
+
+static const struct snd_kcontrol_new da7219_st_out_filtl_mix_controls[] = {
+	DA7219_DMIX_ST_CTRLS(DA7219_DROUTING_ST_OUTFILT_1L),
+};
+
+static const struct snd_kcontrol_new da7219_st_out_filtr_mix_controls[] = {
+	DA7219_DMIX_ST_CTRLS(DA7219_DROUTING_ST_OUTFILT_1R),
+};
+
+
+/*
+ * DAPM Events
+ */
+
+static int da7219_dai_event(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	u8 pll_ctrl, pll_status;
+	int i = 0;
+	bool srm_lock = false;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		if (da7219->master)
+			/* Enable DAI clks for master mode */
+			snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE,
+					    DA7219_DAI_CLK_EN_MASK,
+					    DA7219_DAI_CLK_EN_MASK);
+
+		/* PC synchronised to DAI */
+		snd_soc_update_bits(codec, DA7219_PC_COUNT,
+				    DA7219_PC_FREERUN_MASK, 0);
+
+		/* Slave mode, if SRM not enabled no need for status checks */
+		pll_ctrl = snd_soc_read(codec, DA7219_PLL_CTRL);
+		if ((pll_ctrl & DA7219_PLL_MODE_MASK) != DA7219_PLL_MODE_SRM)
+			return 0;
+
+		/* Check SRM has locked */
+		do {
+			pll_status = snd_soc_read(codec, DA7219_PLL_SRM_STS);
+			if (pll_status & DA7219_PLL_SRM_STS_SRM_LOCK) {
+				srm_lock = true;
+			} else {
+				++i;
+				msleep(50);
+			}
+		} while ((i < DA7219_SRM_CHECK_RETRIES) & (!srm_lock));
+
+		if (!srm_lock)
+			dev_warn(codec->dev, "SRM failed to lock\n");
+
+		return 0;
+	case SND_SOC_DAPM_POST_PMD:
+		/* PC free-running */
+		snd_soc_update_bits(codec, DA7219_PC_COUNT,
+				    DA7219_PC_FREERUN_MASK,
+				    DA7219_PC_FREERUN_MASK);
+
+		/* Disable DAI clks if in master mode */
+		if (da7219->master)
+			snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE,
+					    DA7219_DAI_CLK_EN_MASK, 0);
+		return 0;
+	default:
+		return -EINVAL;
+	}
+}
+
+
+/*
+ * DAPM Widgets
+ */
+
+static const struct snd_soc_dapm_widget da7219_dapm_widgets[] = {
+	/* Input Supplies */
+	SND_SOC_DAPM_SUPPLY("Mic Bias", DA7219_MICBIAS_CTRL,
+			    DA7219_MICBIAS1_EN_SHIFT, DA7219_NO_INVERT,
+			    NULL, 0),
+
+	/* Inputs */
+	SND_SOC_DAPM_INPUT("MIC"),
+
+	/* Input PGAs */
+	SND_SOC_DAPM_PGA("Mic PGA", DA7219_MIC_1_CTRL,
+			 DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT,
+			 NULL, 0),
+	SND_SOC_DAPM_PGA("Mixin PGA", DA7219_MIXIN_L_CTRL,
+			 DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT,
+			 NULL, 0),
+
+	/* Input Filters */
+	SND_SOC_DAPM_ADC("ADC", NULL, DA7219_ADC_L_CTRL, DA7219_ADC_L_EN_SHIFT,
+			 DA7219_NO_INVERT),
+
+	/* Tone Generator */
+	SND_SOC_DAPM_SIGGEN("TONE"),
+	SND_SOC_DAPM_PGA("Tone Generator", DA7219_TONE_GEN_CFG1,
+			 DA7219_START_STOPN_SHIFT, DA7219_NO_INVERT, NULL, 0),
+
+	/* Sidetone Input */
+	SND_SOC_DAPM_ADC("Sidetone Filter", NULL, DA7219_SIDETONE_CTRL,
+			 DA7219_SIDETONE_EN_SHIFT, DA7219_NO_INVERT),
+
+	/* Input Mixer Supply */
+	SND_SOC_DAPM_SUPPLY("Mixer In Supply", DA7219_MIXIN_L_CTRL,
+			    DA7219_MIXIN_L_MIX_EN_SHIFT, DA7219_NO_INVERT,
+			    NULL, 0),
+
+	/* Input Mixer */
+	SND_SOC_DAPM_MIXER("Mixer In", SND_SOC_NOPM, 0, 0,
+			   da7219_mixin_controls,
+			   ARRAY_SIZE(da7219_mixin_controls)),
+
+	/* Input Muxes */
+	SND_SOC_DAPM_MUX("Out DAIL Mux", SND_SOC_NOPM, 0, 0,
+			 &da7219_out_dail_sel_mux),
+	SND_SOC_DAPM_MUX("Out DAIR Mux", SND_SOC_NOPM, 0, 0,
+			 &da7219_out_dair_sel_mux),
+
+	/* DAI Supply */
+	SND_SOC_DAPM_SUPPLY("DAI", DA7219_DAI_CTRL, DA7219_DAI_EN_SHIFT,
+			    DA7219_NO_INVERT, da7219_dai_event,
+			    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* DAI */
+	SND_SOC_DAPM_AIF_OUT("DAIOUT", "Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("DAIIN", "Playback", 0, SND_SOC_NOPM, 0, 0),
+
+	/* Output Muxes */
+	SND_SOC_DAPM_MUX("Out DACL Mux", SND_SOC_NOPM, 0, 0,
+			 &da7219_out_dacl_sel_mux),
+	SND_SOC_DAPM_MUX("Out DACR Mux", SND_SOC_NOPM, 0, 0,
+			 &da7219_out_dacr_sel_mux),
+
+	/* Output Mixers */
+	SND_SOC_DAPM_MIXER("Mixer Out FilterL", SND_SOC_NOPM, 0, 0,
+			   da7219_mixout_l_controls,
+			   ARRAY_SIZE(da7219_mixout_l_controls)),
+	SND_SOC_DAPM_MIXER("Mixer Out FilterR", SND_SOC_NOPM, 0, 0,
+			   da7219_mixout_r_controls,
+			   ARRAY_SIZE(da7219_mixout_r_controls)),
+
+	/* Sidetone Mixers */
+	SND_SOC_DAPM_MIXER("ST Mixer Out FilterL", SND_SOC_NOPM, 0, 0,
+			   da7219_st_out_filtl_mix_controls,
+			   ARRAY_SIZE(da7219_st_out_filtl_mix_controls)),
+	SND_SOC_DAPM_MIXER("ST Mixer Out FilterR", SND_SOC_NOPM, 0,
+			   0, da7219_st_out_filtr_mix_controls,
+			   ARRAY_SIZE(da7219_st_out_filtr_mix_controls)),
+
+	/* DACs */
+	SND_SOC_DAPM_DAC("DACL", NULL, DA7219_DAC_L_CTRL, DA7219_DAC_L_EN_SHIFT,
+			 DA7219_NO_INVERT),
+	SND_SOC_DAPM_DAC("DACR", NULL, DA7219_DAC_R_CTRL, DA7219_DAC_R_EN_SHIFT,
+			 DA7219_NO_INVERT),
+
+	/* Output PGAs */
+	SND_SOC_DAPM_PGA("Mixout Left PGA", DA7219_MIXOUT_L_CTRL,
+			 DA7219_MIXOUT_L_AMP_EN_SHIFT, DA7219_NO_INVERT,
+			 NULL, 0),
+	SND_SOC_DAPM_PGA("Mixout Right PGA", DA7219_MIXOUT_R_CTRL,
+			 DA7219_MIXOUT_R_AMP_EN_SHIFT, DA7219_NO_INVERT,
+			 NULL, 0),
+	SND_SOC_DAPM_PGA("Headphone Left PGA", DA7219_HP_L_CTRL,
+			 DA7219_HP_L_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0),
+	SND_SOC_DAPM_PGA("Headphone Right PGA", DA7219_HP_R_CTRL,
+			 DA7219_HP_R_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0),
+
+	/* Output Supplies */
+	SND_SOC_DAPM_SUPPLY("Charge Pump", DA7219_CP_CTRL, DA7219_CP_EN_SHIFT,
+			    DA7219_NO_INVERT, NULL, 0),
+
+	/* Outputs */
+	SND_SOC_DAPM_OUTPUT("HPL"),
+	SND_SOC_DAPM_OUTPUT("HPR"),
+};
+
+
+/*
+ * DAPM Mux Routes
+ */
+
+#define DA7219_OUT_DAI_MUX_ROUTES(name)			\
+	{name, "ADC", "Mixer In"},			\
+	{name, "Tone Generator", "Tone Generator"},	\
+	{name, "DAIL", "DAIOUT"},			\
+	{name, "DAIR", "DAIOUT"}
+
+#define DA7219_OUT_DAC_MUX_ROUTES(name)			\
+	{name, "ADC", "Mixer In"},			\
+	{name, "Tone Generator", "Tone Generator"},		\
+	{name, "DAIL", "DAIIN"},			\
+	{name, "DAIR", "DAIIN"}
+
+/*
+ * DAPM Mixer Routes
+ */
+
+#define DA7219_DMIX_ST_ROUTES(name)				\
+	{name, "Out FilterL Switch", "Mixer Out FilterL"},	\
+	{name, "Out FilterR Switch", "Mixer Out FilterR"},	\
+	{name, "Sidetone Switch", "Sidetone Filter"}
+
+
+/*
+ * DAPM audio route definition
+ */
+
+static const struct snd_soc_dapm_route da7219_audio_map[] = {
+	/* Input paths */
+	{"MIC", NULL, "Mic Bias"},
+	{"Mic PGA", NULL, "MIC"},
+	{"Mixin PGA", NULL, "Mic PGA"},
+	{"ADC", NULL, "Mixin PGA"},
+
+	{"Sidetone Filter", NULL, "ADC"},
+	{"Mixer In", NULL, "Mixer In Supply"},
+	{"Mixer In", "Mic Switch", "ADC"},
+
+	{"Tone Generator", NULL, "TONE"},
+
+	DA7219_OUT_DAI_MUX_ROUTES("Out DAIL Mux"),
+	DA7219_OUT_DAI_MUX_ROUTES("Out DAIR Mux"),
+
+	{"DAIOUT", NULL, "Out DAIL Mux"},
+	{"DAIOUT", NULL, "Out DAIR Mux"},
+	{"DAIOUT", NULL, "DAI"},
+
+	/* Output paths */
+	{"DAIIN", NULL, "DAI"},
+
+	DA7219_OUT_DAC_MUX_ROUTES("Out DACL Mux"),
+	DA7219_OUT_DAC_MUX_ROUTES("Out DACR Mux"),
+
+	{"Mixer Out FilterL", "DACL Switch", "Out DACL Mux"},
+	{"Mixer Out FilterR", "DACR Switch", "Out DACR Mux"},
+
+	DA7219_DMIX_ST_ROUTES("ST Mixer Out FilterL"),
+	DA7219_DMIX_ST_ROUTES("ST Mixer Out FilterR"),
+
+	{"DACL", NULL, "ST Mixer Out FilterL"},
+	{"DACR", NULL, "ST Mixer Out FilterR"},
+
+	{"Mixout Left PGA", NULL, "DACL"},
+	{"Mixout Right PGA", NULL, "DACR"},
+
+	{"Headphone Left PGA", NULL, "Mixout Left PGA"},
+	{"Headphone Right PGA", NULL, "Mixout Right PGA"},
+
+	{"HPL", NULL, "Headphone Left PGA"},
+	{"HPR", NULL, "Headphone Right PGA"},
+
+	{"HPL", NULL, "Charge Pump"},
+	{"HPR", NULL, "Charge Pump"},
+};
+
+
+/*
+ * DAI operations
+ */
+
+static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				 int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq))
+		return 0;
+
+	if (((freq < 2000000) && (freq != 32768)) || (freq > 54000000)) {
+		dev_err(codec_dai->dev, "Unsupported MCLK value %d\n",
+			freq);
+		return -EINVAL;
+	}
+
+	switch (clk_id) {
+	case DA7219_CLKSRC_MCLK_SQR:
+		snd_soc_update_bits(codec, DA7219_PLL_CTRL,
+				    DA7219_PLL_MCLK_SQR_EN_MASK,
+				    DA7219_PLL_MCLK_SQR_EN_MASK);
+		break;
+	case DA7219_CLKSRC_MCLK:
+		snd_soc_update_bits(codec, DA7219_PLL_CTRL,
+				    DA7219_PLL_MCLK_SQR_EN_MASK, 0);
+		break;
+	default:
+		dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id);
+		return -EINVAL;
+	}
+
+	da7219->clk_src = clk_id;
+
+	if (da7219->mclk) {
+		freq = clk_round_rate(da7219->mclk, freq);
+		ret = clk_set_rate(da7219->mclk, freq);
+		if (ret) {
+			dev_err(codec_dai->dev, "Failed to set clock rate %d\n",
+				freq);
+			return ret;
+		}
+	}
+
+	da7219->mclk_rate = freq;
+
+	return 0;
+}
+
+static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+			      int source, unsigned int fref, unsigned int fout)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+	u8 pll_ctrl, indiv_bits, indiv;
+	u8 pll_frac_top, pll_frac_bot, pll_integer;
+	u32 freq_ref;
+	u64 frac_div;
+
+	/* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */
+	if (da7219->mclk_rate == 32768) {
+		indiv_bits = DA7219_PLL_INDIV_2_5_MHZ;
+		indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL;
+	} else if (da7219->mclk_rate < 2000000) {
+		dev_err(codec->dev, "PLL input clock %d below valid range\n",
+			da7219->mclk_rate);
+		return -EINVAL;
+	} else if (da7219->mclk_rate <= 5000000) {
+		indiv_bits = DA7219_PLL_INDIV_2_5_MHZ;
+		indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL;
+	} else if (da7219->mclk_rate <= 10000000) {
+		indiv_bits = DA7219_PLL_INDIV_5_10_MHZ;
+		indiv = DA7219_PLL_INDIV_5_10_MHZ_VAL;
+	} else if (da7219->mclk_rate <= 20000000) {
+		indiv_bits = DA7219_PLL_INDIV_10_20_MHZ;
+		indiv = DA7219_PLL_INDIV_10_20_MHZ_VAL;
+	} else if (da7219->mclk_rate <= 40000000) {
+		indiv_bits = DA7219_PLL_INDIV_20_40_MHZ;
+		indiv = DA7219_PLL_INDIV_20_40_MHZ_VAL;
+	} else if (da7219->mclk_rate <= 54000000) {
+		indiv_bits = DA7219_PLL_INDIV_40_54_MHZ;
+		indiv = DA7219_PLL_INDIV_40_54_MHZ_VAL;
+	} else {
+		dev_err(codec->dev, "PLL input clock %d above valid range\n",
+			da7219->mclk_rate);
+		return -EINVAL;
+	}
+	freq_ref = (da7219->mclk_rate / indiv);
+	pll_ctrl = indiv_bits;
+
+	/* Configure PLL */
+	switch (source) {
+	case DA7219_SYSCLK_MCLK:
+		pll_ctrl |= DA7219_PLL_MODE_BYPASS;
+		snd_soc_update_bits(codec, DA7219_PLL_CTRL,
+				    DA7219_PLL_INDIV_MASK |
+				    DA7219_PLL_MODE_MASK, pll_ctrl);
+		return 0;
+	case DA7219_SYSCLK_PLL:
+		pll_ctrl |= DA7219_PLL_MODE_NORMAL;
+		break;
+	case DA7219_SYSCLK_PLL_SRM:
+		pll_ctrl |= DA7219_PLL_MODE_SRM;
+		break;
+	case DA7219_SYSCLK_PLL_32KHZ:
+		pll_ctrl |= DA7219_PLL_MODE_32KHZ;
+		break;
+	default:
+		dev_err(codec->dev, "Invalid PLL config\n");
+		return -EINVAL;
+	}
+
+	/* Calculate dividers for PLL */
+	pll_integer = fout / freq_ref;
+	frac_div = (u64)(fout % freq_ref) * 8192ULL;
+	do_div(frac_div, freq_ref);
+	pll_frac_top = (frac_div >> DA7219_BYTE_SHIFT) & DA7219_BYTE_MASK;
+	pll_frac_bot = (frac_div) & DA7219_BYTE_MASK;
+
+	/* Write PLL config & dividers */
+	snd_soc_write(codec, DA7219_PLL_FRAC_TOP, pll_frac_top);
+	snd_soc_write(codec, DA7219_PLL_FRAC_BOT, pll_frac_bot);
+	snd_soc_write(codec, DA7219_PLL_INTEGER, pll_integer);
+	snd_soc_update_bits(codec, DA7219_PLL_CTRL,
+			    DA7219_PLL_INDIV_MASK | DA7219_PLL_MODE_MASK,
+			    pll_ctrl);
+
+	return 0;
+}
+
+static int da7219_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	u8 dai_clk_mode = 0, dai_ctrl = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		da7219->master = true;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		da7219->master = false;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		dai_clk_mode |= DA7219_DAI_WCLK_POL_INV;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		dai_clk_mode |= DA7219_DAI_CLK_POL_INV;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		dai_clk_mode |= DA7219_DAI_WCLK_POL_INV |
+				DA7219_DAI_CLK_POL_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		dai_ctrl |= DA7219_DAI_FORMAT_I2S;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		dai_ctrl |= DA7219_DAI_FORMAT_LEFT_J;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		dai_ctrl |= DA7219_DAI_FORMAT_RIGHT_J;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		dai_ctrl |= DA7219_DAI_FORMAT_DSP;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* By default 64 BCLKs per WCLK is supported */
+	dai_clk_mode |= DA7219_DAI_BCLKS_PER_WCLK_64;
+
+	snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE,
+			    DA7219_DAI_BCLKS_PER_WCLK_MASK |
+			    DA7219_DAI_CLK_POL_MASK | DA7219_DAI_WCLK_POL_MASK,
+			    dai_clk_mode);
+	snd_soc_update_bits(codec, DA7219_DAI_CTRL, DA7219_DAI_FORMAT_MASK,
+			    dai_ctrl);
+
+	return 0;
+}
+
+static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai,
+				   unsigned int tx_mask, unsigned int rx_mask,
+				   int slots, int slot_width)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	u8 dai_bclks_per_wclk;
+	u16 offset;
+	u32 frame_size;
+
+	/* No channels enabled so disable TDM, revert to 64-bit frames */
+	if (!tx_mask) {
+		snd_soc_update_bits(codec, DA7219_DAI_TDM_CTRL,
+				    DA7219_DAI_TDM_CH_EN_MASK |
+				    DA7219_DAI_TDM_MODE_EN_MASK, 0);
+		snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE,
+				    DA7219_DAI_BCLKS_PER_WCLK_MASK,
+				    DA7219_DAI_BCLKS_PER_WCLK_64);
+		return 0;
+	}
+
+	/* Check we have valid slots */
+	if (fls(tx_mask) > DA7219_DAI_TDM_MAX_SLOTS) {
+		dev_err(codec->dev, "Invalid number of slots, max = %d\n",
+			DA7219_DAI_TDM_MAX_SLOTS);
+		return -EINVAL;
+	}
+
+	/* Check we have a valid offset given */
+	if (rx_mask > DA7219_DAI_OFFSET_MAX) {
+		dev_err(codec->dev, "Invalid slot offset, max = %d\n",
+			DA7219_DAI_OFFSET_MAX);
+		return -EINVAL;
+	}
+
+	/* Calculate & validate frame size based on slot info provided. */
+	frame_size = slots * slot_width;
+	switch (frame_size) {
+	case 32:
+		dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_32;
+		break;
+	case 64:
+		dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_64;
+		break;
+	case 128:
+		dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_128;
+		break;
+	case 256:
+		dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_256;
+		break;
+	default:
+		dev_err(codec->dev, "Invalid frame size %d\n", frame_size);
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE,
+			    DA7219_DAI_BCLKS_PER_WCLK_MASK,
+			    dai_bclks_per_wclk);
+
+	offset = cpu_to_le16(rx_mask);
+	regmap_bulk_write(da7219->regmap, DA7219_DAI_OFFSET_LOWER,
+			  &offset, sizeof(offset));
+
+	snd_soc_update_bits(codec, DA7219_DAI_TDM_CTRL,
+			    DA7219_DAI_TDM_CH_EN_MASK |
+			    DA7219_DAI_TDM_MODE_EN_MASK,
+			    (tx_mask << DA7219_DAI_TDM_CH_EN_SHIFT) |
+			    DA7219_DAI_TDM_MODE_EN_MASK);
+
+	return 0;
+}
+
+static int da7219_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u8 dai_ctrl = 0, fs;
+	unsigned int channels;
+
+	switch (params_width(params)) {
+	case 16:
+		dai_ctrl |= DA7219_DAI_WORD_LENGTH_S16_LE;
+		break;
+	case 20:
+		dai_ctrl |= DA7219_DAI_WORD_LENGTH_S20_LE;
+		break;
+	case 24:
+		dai_ctrl |= DA7219_DAI_WORD_LENGTH_S24_LE;
+		break;
+	case 32:
+		dai_ctrl |= DA7219_DAI_WORD_LENGTH_S32_LE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	channels = params_channels(params);
+	if ((channels < 1) | (channels > DA7219_DAI_CH_NUM_MAX)) {
+		dev_err(codec->dev,
+			"Invalid number of channels, only 1 to %d supported\n",
+			DA7219_DAI_CH_NUM_MAX);
+		return -EINVAL;
+	}
+	dai_ctrl |= channels << DA7219_DAI_CH_NUM_SHIFT;
+
+	switch (params_rate(params)) {
+	case 8000:
+		fs = DA7219_SR_8000;
+		break;
+	case 11025:
+		fs = DA7219_SR_11025;
+		break;
+	case 12000:
+		fs = DA7219_SR_12000;
+		break;
+	case 16000:
+		fs = DA7219_SR_16000;
+		break;
+	case 22050:
+		fs = DA7219_SR_22050;
+		break;
+	case 24000:
+		fs = DA7219_SR_24000;
+		break;
+	case 32000:
+		fs = DA7219_SR_32000;
+		break;
+	case 44100:
+		fs = DA7219_SR_44100;
+		break;
+	case 48000:
+		fs = DA7219_SR_48000;
+		break;
+	case 88200:
+		fs = DA7219_SR_88200;
+		break;
+	case 96000:
+		fs = DA7219_SR_96000;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, DA7219_DAI_CTRL,
+			    DA7219_DAI_WORD_LENGTH_MASK |
+			    DA7219_DAI_CH_NUM_MASK,
+			    dai_ctrl);
+	snd_soc_write(codec, DA7219_SR, fs);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops da7219_dai_ops = {
+	.hw_params	= da7219_hw_params,
+	.set_sysclk	= da7219_set_dai_sysclk,
+	.set_pll	= da7219_set_dai_pll,
+	.set_fmt	= da7219_set_dai_fmt,
+	.set_tdm_slot	= da7219_set_dai_tdm_slot,
+};
+
+#define DA7219_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 da7219_dai = {
+	.name = "da7219-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = DA7219_DAI_CH_NUM_MAX,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = DA7219_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = DA7219_DAI_CH_NUM_MAX,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = DA7219_FORMATS,
+	},
+	.ops = &da7219_dai_ops,
+	.symmetric_rates = 1,
+	.symmetric_channels = 1,
+	.symmetric_samplebits = 1,
+};
+
+
+/*
+ * DT
+ */
+
+static const struct of_device_id da7219_of_match[] = {
+	{ .compatible = "dlg,da7219", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, da7219_of_match);
+
+static enum da7219_ldo_lvl_sel da7219_of_ldo_lvl(struct snd_soc_codec *codec,
+						 u32 val)
+{
+	switch (val) {
+	case 1050:
+		return DA7219_LDO_LVL_SEL_1_05V;
+	case 1100:
+		return DA7219_LDO_LVL_SEL_1_10V;
+	case 1200:
+		return DA7219_LDO_LVL_SEL_1_20V;
+	case 1400:
+		return DA7219_LDO_LVL_SEL_1_40V;
+	default:
+		dev_warn(codec->dev, "Invalid LDO level");
+		return DA7219_LDO_LVL_SEL_1_05V;
+	}
+}
+
+static enum da7219_micbias_voltage
+	da7219_of_micbias_lvl(struct snd_soc_codec *codec, u32 val)
+{
+	switch (val) {
+	case 1800:
+		return DA7219_MICBIAS_1_8V;
+	case 2000:
+		return DA7219_MICBIAS_2_0V;
+	case 2200:
+		return DA7219_MICBIAS_2_2V;
+	case 2400:
+		return DA7219_MICBIAS_2_4V;
+	case 2600:
+		return DA7219_MICBIAS_2_6V;
+	default:
+		dev_warn(codec->dev, "Invalid micbias level");
+		return DA7219_MICBIAS_2_2V;
+	}
+}
+
+static enum da7219_mic_amp_in_sel
+	da7219_of_mic_amp_in_sel(struct snd_soc_codec *codec, const char *str)
+{
+	if (!strcmp(str, "diff")) {
+		return DA7219_MIC_AMP_IN_SEL_DIFF;
+	} else if (!strcmp(str, "se_p")) {
+		return DA7219_MIC_AMP_IN_SEL_SE_P;
+	} else if (!strcmp(str, "se_n")) {
+		return DA7219_MIC_AMP_IN_SEL_SE_N;
+	} else {
+		dev_warn(codec->dev, "Invalid mic input type selection");
+		return DA7219_MIC_AMP_IN_SEL_DIFF;
+	}
+}
+
+static struct da7219_pdata *da7219_of_to_pdata(struct snd_soc_codec *codec)
+{
+	struct device_node *np = codec->dev->of_node;
+	struct da7219_pdata *pdata;
+	const char *of_str;
+	u32 of_val32;
+
+	pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL);
+	if (!pdata)
+		return NULL;
+
+	if (of_property_read_u32(np, "dlg,ldo-lvl", &of_val32) >= 0)
+		pdata->ldo_lvl_sel = da7219_of_ldo_lvl(codec, of_val32);
+
+	if (of_property_read_u32(np, "dlg,micbias-lvl", &of_val32) >= 0)
+		pdata->micbias_lvl = da7219_of_micbias_lvl(codec, of_val32);
+	else
+		pdata->micbias_lvl = DA7219_MICBIAS_2_2V;
+
+	if (!of_property_read_string(np, "dlg,mic-amp-in-sel", &of_str))
+		pdata->mic_amp_in_sel = da7219_of_mic_amp_in_sel(codec, of_str);
+	else
+		pdata->mic_amp_in_sel = DA7219_MIC_AMP_IN_SEL_DIFF;
+
+	return pdata;
+}
+
+
+/*
+ * Codec driver functions
+ */
+
+static int da7219_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
+			/* MCLK */
+			if (da7219->mclk) {
+				ret = clk_prepare_enable(da7219->mclk);
+				if (ret) {
+					dev_err(codec->dev,
+						"Failed to enable mclk\n");
+					return ret;
+				}
+			}
+
+			/* Master bias */
+			snd_soc_update_bits(codec, DA7219_REFERENCES,
+					    DA7219_BIAS_EN_MASK,
+					    DA7219_BIAS_EN_MASK);
+
+			/* Enable Internal Digital LDO */
+			snd_soc_update_bits(codec, DA7219_LDO_CTRL,
+					    DA7219_LDO_EN_MASK,
+					    DA7219_LDO_EN_MASK);
+		}
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* Only disable if jack detection not active */
+		if (!da7219->aad->jack) {
+			/* Bypass Internal Digital LDO */
+			snd_soc_update_bits(codec, DA7219_LDO_CTRL,
+					    DA7219_LDO_EN_MASK, 0);
+
+			/* Master bias */
+			snd_soc_update_bits(codec, DA7219_REFERENCES,
+					    DA7219_BIAS_EN_MASK, 0);
+		}
+
+		/* MCLK */
+		if (da7219->mclk)
+			clk_disable_unprepare(da7219->mclk);
+		break;
+	}
+
+	return 0;
+}
+
+static const char *da7219_supply_names[DA7219_NUM_SUPPLIES] = {
+	[DA7219_SUPPLY_VDD] = "VDD",
+	[DA7219_SUPPLY_VDDMIC] = "VDDMIC",
+	[DA7219_SUPPLY_VDDIO] = "VDDIO",
+};
+
+static int da7219_handle_supplies(struct snd_soc_codec *codec)
+{
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	struct regulator *vddio;
+	u8 io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V;
+	int i, ret;
+
+	/* Get required supplies */
+	for (i = 0; i < DA7219_NUM_SUPPLIES; ++i)
+		da7219->supplies[i].supply = da7219_supply_names[i];
+
+	ret = devm_regulator_bulk_get(codec->dev, DA7219_NUM_SUPPLIES,
+				      da7219->supplies);
+	if (ret) {
+		dev_err(codec->dev, "Failed to get supplies");
+		return ret;
+	}
+
+	/* Determine VDDIO voltage provided */
+	vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer;
+	ret = regulator_get_voltage(vddio);
+	if (ret < 1200000)
+		dev_warn(codec->dev, "Invalid VDDIO voltage\n");
+	else if (ret < 2800000)
+		io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V;
+
+	/* Enable main supplies */
+	ret = regulator_bulk_enable(DA7219_NUM_SUPPLIES, da7219->supplies);
+	if (ret) {
+		dev_err(codec->dev, "Failed to enable supplies");
+		return ret;
+	}
+
+	/* Ensure device in active mode */
+	snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, DA7219_SYSTEM_ACTIVE_MASK);
+
+	/* Update IO voltage level range */
+	snd_soc_write(codec, DA7219_IO_CTRL, io_voltage_lvl);
+
+	return 0;
+}
+
+static void da7219_handle_pdata(struct snd_soc_codec *codec)
+{
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	struct da7219_pdata *pdata = da7219->pdata;
+
+	if (pdata) {
+		u8 micbias_lvl = 0;
+
+		/* Internal LDO */
+		switch (pdata->ldo_lvl_sel) {
+		case DA7219_LDO_LVL_SEL_1_05V:
+		case DA7219_LDO_LVL_SEL_1_10V:
+		case DA7219_LDO_LVL_SEL_1_20V:
+		case DA7219_LDO_LVL_SEL_1_40V:
+			snd_soc_update_bits(codec, DA7219_LDO_CTRL,
+					    DA7219_LDO_LEVEL_SELECT_MASK,
+					    (pdata->ldo_lvl_sel <<
+					     DA7219_LDO_LEVEL_SELECT_SHIFT));
+			break;
+		}
+
+		/* Mic Bias voltages */
+		switch (pdata->micbias_lvl) {
+		case DA7219_MICBIAS_1_8V:
+		case DA7219_MICBIAS_2_0V:
+		case DA7219_MICBIAS_2_2V:
+		case DA7219_MICBIAS_2_4V:
+		case DA7219_MICBIAS_2_6V:
+			micbias_lvl |= (pdata->micbias_lvl <<
+					DA7219_MICBIAS1_LEVEL_SHIFT);
+			break;
+		}
+
+		snd_soc_write(codec, DA7219_MICBIAS_CTRL, micbias_lvl);
+
+		/* Mic */
+		switch (pdata->mic_amp_in_sel) {
+		case DA7219_MIC_AMP_IN_SEL_DIFF:
+		case DA7219_MIC_AMP_IN_SEL_SE_P:
+		case DA7219_MIC_AMP_IN_SEL_SE_N:
+			snd_soc_write(codec, DA7219_MIC_1_SELECT,
+				      pdata->mic_amp_in_sel);
+			break;
+		}
+	}
+}
+
+static int da7219_probe(struct snd_soc_codec *codec)
+{
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	mutex_init(&da7219->lock);
+
+	/* Regulator configuration */
+	ret = da7219_handle_supplies(codec);
+	if (ret)
+		return ret;
+
+	/* Handle DT/Platform data */
+	if (codec->dev->of_node)
+		da7219->pdata = da7219_of_to_pdata(codec);
+	else
+		da7219->pdata = dev_get_platdata(codec->dev);
+
+	da7219_handle_pdata(codec);
+
+	/* Check if MCLK provided */
+	da7219->mclk = devm_clk_get(codec->dev, "mclk");
+	if (IS_ERR(da7219->mclk)) {
+		if (PTR_ERR(da7219->mclk) != -ENOENT)
+			return PTR_ERR(da7219->mclk);
+		else
+			da7219->mclk = NULL;
+	}
+
+	/* Default PC counter to free-running */
+	snd_soc_update_bits(codec, DA7219_PC_COUNT, DA7219_PC_FREERUN_MASK,
+			    DA7219_PC_FREERUN_MASK);
+
+	/* Default gain ramping */
+	snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL,
+			    DA7219_MIXIN_L_AMP_RAMP_EN_MASK,
+			    DA7219_MIXIN_L_AMP_RAMP_EN_MASK);
+	snd_soc_update_bits(codec, DA7219_ADC_L_CTRL, DA7219_ADC_L_RAMP_EN_MASK,
+			    DA7219_ADC_L_RAMP_EN_MASK);
+	snd_soc_update_bits(codec, DA7219_DAC_L_CTRL, DA7219_DAC_L_RAMP_EN_MASK,
+			    DA7219_DAC_L_RAMP_EN_MASK);
+	snd_soc_update_bits(codec, DA7219_DAC_R_CTRL, DA7219_DAC_R_RAMP_EN_MASK,
+			    DA7219_DAC_R_RAMP_EN_MASK);
+	snd_soc_update_bits(codec, DA7219_HP_L_CTRL,
+			    DA7219_HP_L_AMP_RAMP_EN_MASK,
+			    DA7219_HP_L_AMP_RAMP_EN_MASK);
+	snd_soc_update_bits(codec, DA7219_HP_R_CTRL,
+			    DA7219_HP_R_AMP_RAMP_EN_MASK,
+			    DA7219_HP_R_AMP_RAMP_EN_MASK);
+
+	/* Default infinite tone gen, start/stop by Kcontrol */
+	snd_soc_write(codec, DA7219_TONE_GEN_CYCLES, DA7219_BEEP_CYCLES_MASK);
+
+	/* Initialise AAD block */
+	return da7219_aad_init(codec);
+}
+
+static int da7219_remove(struct snd_soc_codec *codec)
+{
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+	da7219_aad_exit(codec);
+
+	/* Supplies */
+	return regulator_bulk_disable(DA7219_NUM_SUPPLIES, da7219->supplies);
+}
+
+#ifdef CONFIG_PM
+static int da7219_suspend(struct snd_soc_codec *codec)
+{
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	/* Put device into standby mode if jack detection disabled */
+	if (!da7219->aad->jack)
+		snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, 0);
+
+	return 0;
+}
+
+static int da7219_resume(struct snd_soc_codec *codec)
+{
+	struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+	/* Put device into active mode if previously pushed to standby */
+	if (!da7219->aad->jack)
+		snd_soc_write(codec, DA7219_SYSTEM_ACTIVE,
+			      DA7219_SYSTEM_ACTIVE_MASK);
+
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+#else
+#define da7219_suspend NULL
+#define da7219_resume NULL
+#endif
+
+static struct snd_soc_codec_driver soc_codec_dev_da7219 = {
+	.probe			= da7219_probe,
+	.remove			= da7219_remove,
+	.suspend		= da7219_suspend,
+	.resume			= da7219_resume,
+	.set_bias_level		= da7219_set_bias_level,
+
+	.controls		= da7219_snd_controls,
+	.num_controls		= ARRAY_SIZE(da7219_snd_controls),
+
+	.dapm_widgets		= da7219_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(da7219_dapm_widgets),
+	.dapm_routes		= da7219_audio_map,
+	.num_dapm_routes	= ARRAY_SIZE(da7219_audio_map),
+};
+
+
+/*
+ * Regmap configs
+ */
+
+static struct reg_default da7219_reg_defaults[] = {
+	{ DA7219_MIC_1_SELECT, 0x00 },
+	{ DA7219_CIF_TIMEOUT_CTRL, 0x01 },
+	{ DA7219_SR_24_48, 0x00 },
+	{ DA7219_SR, 0x0A },
+	{ DA7219_CIF_I2C_ADDR_CFG, 0x02 },
+	{ DA7219_PLL_CTRL, 0x10 },
+	{ DA7219_PLL_FRAC_TOP, 0x00 },
+	{ DA7219_PLL_FRAC_BOT, 0x00 },
+	{ DA7219_PLL_INTEGER, 0x20 },
+	{ DA7219_DIG_ROUTING_DAI, 0x10 },
+	{ DA7219_DAI_CLK_MODE, 0x01 },
+	{ DA7219_DAI_CTRL, 0x28 },
+	{ DA7219_DAI_TDM_CTRL, 0x40 },
+	{ DA7219_DIG_ROUTING_DAC, 0x32 },
+	{ DA7219_DAI_OFFSET_LOWER, 0x00 },
+	{ DA7219_DAI_OFFSET_UPPER, 0x00 },
+	{ DA7219_REFERENCES, 0x00 },
+	{ DA7219_MIXIN_L_SELECT, 0x00 },
+	{ DA7219_MIXIN_L_GAIN, 0x03 },
+	{ DA7219_ADC_L_GAIN, 0x6F },
+	{ DA7219_ADC_FILTERS1, 0x80 },
+	{ DA7219_MIC_1_GAIN, 0x01 },
+	{ DA7219_SIDETONE_CTRL, 0x40 },
+	{ DA7219_SIDETONE_GAIN, 0x0E },
+	{ DA7219_DROUTING_ST_OUTFILT_1L, 0x01 },
+	{ DA7219_DROUTING_ST_OUTFILT_1R, 0x02 },
+	{ DA7219_DAC_FILTERS5, 0x00 },
+	{ DA7219_DAC_FILTERS2, 0x88 },
+	{ DA7219_DAC_FILTERS3, 0x88 },
+	{ DA7219_DAC_FILTERS4, 0x08 },
+	{ DA7219_DAC_FILTERS1, 0x80 },
+	{ DA7219_DAC_L_GAIN, 0x6F },
+	{ DA7219_DAC_R_GAIN, 0x6F },
+	{ DA7219_CP_CTRL, 0x20 },
+	{ DA7219_HP_L_GAIN, 0x39 },
+	{ DA7219_HP_R_GAIN, 0x39 },
+	{ DA7219_MIXOUT_L_SELECT, 0x00 },
+	{ DA7219_MIXOUT_R_SELECT, 0x00 },
+	{ DA7219_MICBIAS_CTRL, 0x03 },
+	{ DA7219_MIC_1_CTRL, 0x40 },
+	{ DA7219_MIXIN_L_CTRL, 0x40 },
+	{ DA7219_ADC_L_CTRL, 0x40 },
+	{ DA7219_DAC_L_CTRL, 0x40 },
+	{ DA7219_DAC_R_CTRL, 0x40 },
+	{ DA7219_HP_L_CTRL, 0x40 },
+	{ DA7219_HP_R_CTRL, 0x40 },
+	{ DA7219_MIXOUT_L_CTRL, 0x10 },
+	{ DA7219_MIXOUT_R_CTRL, 0x10 },
+	{ DA7219_CHIP_ID1, 0x23 },
+	{ DA7219_CHIP_ID2, 0x93 },
+	{ DA7219_CHIP_REVISION, 0x00 },
+	{ DA7219_LDO_CTRL, 0x00 },
+	{ DA7219_IO_CTRL, 0x00 },
+	{ DA7219_GAIN_RAMP_CTRL, 0x00 },
+	{ DA7219_PC_COUNT, 0x02 },
+	{ DA7219_CP_VOL_THRESHOLD1, 0x0E },
+	{ DA7219_DIG_CTRL, 0x00 },
+	{ DA7219_ALC_CTRL2, 0x00 },
+	{ DA7219_ALC_CTRL3, 0x00 },
+	{ DA7219_ALC_NOISE, 0x3F },
+	{ DA7219_ALC_TARGET_MIN, 0x3F },
+	{ DA7219_ALC_TARGET_MAX, 0x00 },
+	{ DA7219_ALC_GAIN_LIMITS, 0xFF },
+	{ DA7219_ALC_ANA_GAIN_LIMITS, 0x71 },
+	{ DA7219_ALC_ANTICLIP_CTRL, 0x00 },
+	{ DA7219_ALC_ANTICLIP_LEVEL, 0x00 },
+	{ DA7219_DAC_NG_SETUP_TIME, 0x00 },
+	{ DA7219_DAC_NG_OFF_THRESH, 0x00 },
+	{ DA7219_DAC_NG_ON_THRESH, 0x00 },
+	{ DA7219_DAC_NG_CTRL, 0x00 },
+	{ DA7219_TONE_GEN_CFG1, 0x00 },
+	{ DA7219_TONE_GEN_CFG2, 0x00 },
+	{ DA7219_TONE_GEN_CYCLES, 0x00 },
+	{ DA7219_TONE_GEN_FREQ1_L, 0x55 },
+	{ DA7219_TONE_GEN_FREQ1_U, 0x15 },
+	{ DA7219_TONE_GEN_FREQ2_L, 0x00 },
+	{ DA7219_TONE_GEN_FREQ2_U, 0x40 },
+	{ DA7219_TONE_GEN_ON_PER, 0x02 },
+	{ DA7219_TONE_GEN_OFF_PER, 0x01 },
+	{ DA7219_ACCDET_IRQ_MASK_A, 0x00 },
+	{ DA7219_ACCDET_IRQ_MASK_B, 0x00 },
+	{ DA7219_ACCDET_CONFIG_1, 0xD6 },
+	{ DA7219_ACCDET_CONFIG_2, 0x34 },
+	{ DA7219_ACCDET_CONFIG_3, 0x0A },
+	{ DA7219_ACCDET_CONFIG_4, 0x16 },
+	{ DA7219_ACCDET_CONFIG_5, 0x21 },
+	{ DA7219_ACCDET_CONFIG_6, 0x3E },
+	{ DA7219_ACCDET_CONFIG_7, 0x01 },
+	{ DA7219_SYSTEM_ACTIVE, 0x00 },
+};
+
+static bool da7219_volatile_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case DA7219_MIC_1_GAIN_STATUS:
+	case DA7219_MIXIN_L_GAIN_STATUS:
+	case DA7219_ADC_L_GAIN_STATUS:
+	case DA7219_DAC_L_GAIN_STATUS:
+	case DA7219_DAC_R_GAIN_STATUS:
+	case DA7219_HP_L_GAIN_STATUS:
+	case DA7219_HP_R_GAIN_STATUS:
+	case DA7219_CIF_CTRL:
+	case DA7219_PLL_SRM_STS:
+	case DA7219_ALC_CTRL1:
+	case DA7219_SYSTEM_MODES_INPUT:
+	case DA7219_SYSTEM_MODES_OUTPUT:
+	case DA7219_ALC_OFFSET_AUTO_M_L:
+	case DA7219_ALC_OFFSET_AUTO_U_L:
+	case DA7219_TONE_GEN_CFG1:
+	case DA7219_ACCDET_STATUS_A:
+	case DA7219_ACCDET_STATUS_B:
+	case DA7219_ACCDET_IRQ_EVENT_A:
+	case DA7219_ACCDET_IRQ_EVENT_B:
+	case DA7219_ACCDET_CONFIG_8:
+	case DA7219_SYSTEM_STATUS:
+		return 1;
+	default:
+		return 0;
+	}
+}
+
+static const struct regmap_config da7219_regmap_config = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.max_register = DA7219_SYSTEM_ACTIVE,
+	.reg_defaults = da7219_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(da7219_reg_defaults),
+	.volatile_reg = da7219_volatile_register,
+	.cache_type = REGCACHE_RBTREE,
+};
+
+
+/*
+ * I2C layer
+ */
+
+static int da7219_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	struct da7219_priv *da7219;
+	int ret;
+
+	da7219 = devm_kzalloc(&i2c->dev, sizeof(struct da7219_priv),
+			      GFP_KERNEL);
+	if (!da7219)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, da7219);
+
+	da7219->regmap = devm_regmap_init_i2c(i2c, &da7219_regmap_config);
+	if (IS_ERR(da7219->regmap)) {
+		ret = PTR_ERR(da7219->regmap);
+		dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7219,
+				     &da7219_dai, 1);
+	if (ret < 0) {
+		dev_err(&i2c->dev, "Failed to register da7219 codec: %d\n",
+			ret);
+	}
+	return ret;
+}
+
+static int da7219_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	return 0;
+}
+
+static const struct i2c_device_id da7219_i2c_id[] = {
+	{ "da7219", },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, da7219_i2c_id);
+
+static struct i2c_driver da7219_i2c_driver = {
+	.driver = {
+		.name = "da7219",
+		.of_match_table = of_match_ptr(da7219_of_match),
+	},
+	.probe		= da7219_i2c_probe,
+	.remove		= da7219_i2c_remove,
+	.id_table	= da7219_i2c_id,
+};
+
+module_i2c_driver(da7219_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC DA7219 Codec Driver");
+MODULE_AUTHOR("Adam Thomson <Adam.Thomson.Opensource@diasemi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h
new file mode 100644
index 000000000000..b514268c6c56
--- /dev/null
+++ b/sound/soc/codecs/da7219.h
@@ -0,0 +1,820 @@
+/*
+ * da7219.h - DA7219 ALSA SoC Codec Driver
+ *
+ * Copyright (c) 2015 Dialog Semiconductor
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+
+#ifndef __DA7219_H
+#define __DA7219_H
+
+#include <linux/regmap.h>
+#include <linux/regulator/consumer.h>
+#include <sound/da7219.h>
+
+/*
+ * Registers
+ */
+
+#define DA7219_MIC_1_GAIN_STATUS	0x6
+#define DA7219_MIXIN_L_GAIN_STATUS	0x8
+#define DA7219_ADC_L_GAIN_STATUS	0xA
+#define DA7219_DAC_L_GAIN_STATUS	0xC
+#define DA7219_DAC_R_GAIN_STATUS	0xD
+#define DA7219_HP_L_GAIN_STATUS		0xE
+#define DA7219_HP_R_GAIN_STATUS		0xF
+#define DA7219_MIC_1_SELECT		0x10
+#define DA7219_CIF_TIMEOUT_CTRL		0x12
+#define DA7219_CIF_CTRL			0x13
+#define DA7219_SR_24_48			0x16
+#define DA7219_SR			0x17
+#define DA7219_CIF_I2C_ADDR_CFG		0x1B
+#define DA7219_PLL_CTRL			0x20
+#define DA7219_PLL_FRAC_TOP		0x22
+#define DA7219_PLL_FRAC_BOT		0x23
+#define DA7219_PLL_INTEGER		0x24
+#define DA7219_PLL_SRM_STS		0x25
+#define DA7219_DIG_ROUTING_DAI		0x2A
+#define DA7219_DAI_CLK_MODE		0x2B
+#define DA7219_DAI_CTRL			0x2C
+#define DA7219_DAI_TDM_CTRL		0x2D
+#define DA7219_DIG_ROUTING_DAC		0x2E
+#define DA7219_ALC_CTRL1		0x2F
+#define DA7219_DAI_OFFSET_LOWER		0x30
+#define DA7219_DAI_OFFSET_UPPER		0x31
+#define DA7219_REFERENCES		0x32
+#define DA7219_MIXIN_L_SELECT		0x33
+#define DA7219_MIXIN_L_GAIN		0x34
+#define DA7219_ADC_L_GAIN		0x36
+#define DA7219_ADC_FILTERS1		0x38
+#define DA7219_MIC_1_GAIN		0x39
+#define DA7219_SIDETONE_CTRL		0x3A
+#define DA7219_SIDETONE_GAIN		0x3B
+#define DA7219_DROUTING_ST_OUTFILT_1L	0x3C
+#define DA7219_DROUTING_ST_OUTFILT_1R	0x3D
+#define DA7219_DAC_FILTERS5		0x40
+#define DA7219_DAC_FILTERS2		0x41
+#define DA7219_DAC_FILTERS3		0x42
+#define DA7219_DAC_FILTERS4		0x43
+#define DA7219_DAC_FILTERS1		0x44
+#define DA7219_DAC_L_GAIN		0x45
+#define DA7219_DAC_R_GAIN		0x46
+#define DA7219_CP_CTRL			0x47
+#define DA7219_HP_L_GAIN		0x48
+#define DA7219_HP_R_GAIN		0x49
+#define DA7219_MIXOUT_L_SELECT		0x4B
+#define DA7219_MIXOUT_R_SELECT		0x4C
+#define DA7219_SYSTEM_MODES_INPUT	0x50
+#define DA7219_SYSTEM_MODES_OUTPUT	0x51
+#define DA7219_MICBIAS_CTRL		0x62
+#define DA7219_MIC_1_CTRL		0x63
+#define DA7219_MIXIN_L_CTRL		0x65
+#define DA7219_ADC_L_CTRL		0x67
+#define DA7219_DAC_L_CTRL		0x69
+#define DA7219_DAC_R_CTRL		0x6A
+#define DA7219_HP_L_CTRL		0x6B
+#define DA7219_HP_R_CTRL		0x6C
+#define DA7219_MIXOUT_L_CTRL		0x6E
+#define DA7219_MIXOUT_R_CTRL		0x6F
+#define DA7219_CHIP_ID1			0x81
+#define DA7219_CHIP_ID2			0x82
+#define DA7219_CHIP_REVISION		0x83
+#define DA7219_LDO_CTRL			0x90
+#define DA7219_IO_CTRL			0x91
+#define DA7219_GAIN_RAMP_CTRL		0x92
+#define DA7219_PC_COUNT			0x94
+#define DA7219_CP_VOL_THRESHOLD1	0x95
+#define DA7219_CP_DELAY			0x96
+#define DA7219_DIG_CTRL			0x99
+#define DA7219_ALC_CTRL2		0x9A
+#define DA7219_ALC_CTRL3		0x9B
+#define DA7219_ALC_NOISE		0x9C
+#define DA7219_ALC_TARGET_MIN		0x9D
+#define DA7219_ALC_TARGET_MAX		0x9E
+#define DA7219_ALC_GAIN_LIMITS		0x9F
+#define DA7219_ALC_ANA_GAIN_LIMITS	0xA0
+#define DA7219_ALC_ANTICLIP_CTRL	0xA1
+#define DA7219_ALC_ANTICLIP_LEVEL	0xA2
+#define DA7219_ALC_OFFSET_AUTO_M_L	0xA3
+#define DA7219_ALC_OFFSET_AUTO_U_L	0xA4
+#define DA7219_DAC_NG_SETUP_TIME	0xAF
+#define DA7219_DAC_NG_OFF_THRESH	0xB0
+#define DA7219_DAC_NG_ON_THRESH		0xB1
+#define DA7219_DAC_NG_CTRL		0xB2
+#define DA7219_TONE_GEN_CFG1		0xB4
+#define DA7219_TONE_GEN_CFG2		0xB5
+#define DA7219_TONE_GEN_CYCLES		0xB6
+#define DA7219_TONE_GEN_FREQ1_L		0xB7
+#define DA7219_TONE_GEN_FREQ1_U		0xB8
+#define DA7219_TONE_GEN_FREQ2_L		0xB9
+#define DA7219_TONE_GEN_FREQ2_U		0xBA
+#define DA7219_TONE_GEN_ON_PER		0xBB
+#define DA7219_TONE_GEN_OFF_PER		0xBC
+#define DA7219_SYSTEM_STATUS		0xE0
+#define DA7219_SYSTEM_ACTIVE		0xFD
+
+
+/*
+ * Bit Fields
+ */
+
+#define DA7219_SWITCH_EN_MAX		0x1
+
+/* DA7219_MIC_1_GAIN_STATUS = 0x6 */
+#define DA7219_MIC_1_AMP_GAIN_STATUS_SHIFT	0
+#define DA7219_MIC_1_AMP_GAIN_STATUS_MASK	(0x7 << 0)
+#define DA7219_MIC_1_AMP_GAIN_MAX		0x7
+
+/* DA7219_MIXIN_L_GAIN_STATUS = 0x8 */
+#define DA7219_MIXIN_L_AMP_GAIN_STATUS_SHIFT	0
+#define DA7219_MIXIN_L_AMP_GAIN_STATUS_MASK	(0xF << 0)
+
+/* DA7219_ADC_L_GAIN_STATUS = 0xA */
+#define DA7219_ADC_L_DIGITAL_GAIN_STATUS_SHIFT	0
+#define DA7219_ADC_L_DIGITAL_GAIN_STATUS_MASK	(0x7F << 0)
+
+/* DA7219_DAC_L_GAIN_STATUS = 0xC */
+#define DA7219_DAC_L_DIGITAL_GAIN_STATUS_SHIFT	0
+#define DA7219_DAC_L_DIGITAL_GAIN_STATUS_MASK	(0x7F << 0)
+
+/* DA7219_DAC_R_GAIN_STATUS = 0xD */
+#define DA7219_DAC_R_DIGITAL_GAIN_STATUS_SHIFT	0
+#define DA7219_DAC_R_DIGITAL_GAIN_STATUS_MASK	(0x7F << 0)
+
+/* DA7219_HP_L_GAIN_STATUS = 0xE */
+#define DA7219_HP_L_AMP_GAIN_STATUS_SHIFT	0
+#define DA7219_HP_L_AMP_GAIN_STATUS_MASK	(0x3F << 0)
+
+/* DA7219_HP_R_GAIN_STATUS = 0xF */
+#define DA7219_HP_R_AMP_GAIN_STATUS_SHIFT	0
+#define DA7219_HP_R_AMP_GAIN_STATUS_MASK	(0x3F << 0)
+
+/* DA7219_MIC_1_SELECT = 0x10 */
+#define DA7219_MIC_1_AMP_IN_SEL_SHIFT	0
+#define DA7219_MIC_1_AMP_IN_SEL_MASK	(0x3 << 0)
+
+/* DA7219_CIF_TIMEOUT_CTRL = 0x12 */
+#define DA7219_I2C_TIMEOUT_EN_SHIFT	0
+#define DA7219_I2C_TIMEOUT_EN_MASK	(0x1 << 0)
+
+/* DA7219_CIF_CTRL = 0x13 */
+#define DA7219_CIF_I2C_WRITE_MODE_SHIFT		0
+#define DA7219_CIF_I2C_WRITE_MODE_MASK		(0x1 << 0)
+#define DA7219_CIF_REG_SOFT_RESET_SHIFT		7
+#define DA7219_CIF_REG_SOFT_RESET_MASK		(0x1 << 7)
+
+/* DA7219_SR_24_48 = 0x16 */
+#define DA7219_SR_24_48_SHIFT	0
+#define DA7219_SR_24_48_MASK	(0x1 << 0)
+
+/* DA7219_SR = 0x17 */
+#define DA7219_SR_SHIFT		0
+#define DA7219_SR_MASK		(0xF << 0)
+#define DA7219_SR_8000		(0x01 << 0)
+#define DA7219_SR_11025		(0x02 << 0)
+#define DA7219_SR_12000		(0x03 << 0)
+#define DA7219_SR_16000		(0x05 << 0)
+#define DA7219_SR_22050		(0x06 << 0)
+#define DA7219_SR_24000		(0x07 << 0)
+#define DA7219_SR_32000		(0x09 << 0)
+#define DA7219_SR_44100		(0x0A << 0)
+#define DA7219_SR_48000		(0x0B << 0)
+#define DA7219_SR_88200		(0x0E << 0)
+#define DA7219_SR_96000		(0x0F << 0)
+
+/* DA7219_CIF_I2C_ADDR_CFG = 0x1B */
+#define DA7219_CIF_I2C_ADDR_CFG_SHIFT	0
+#define DA7219_CIF_I2C_ADDR_CFG_MASK	(0x3 << 0)
+
+/* DA7219_PLL_CTRL = 0x20 */
+#define DA7219_PLL_INDIV_SHIFT		2
+#define DA7219_PLL_INDIV_MASK		(0x7 << 2)
+#define DA7219_PLL_INDIV_2_5_MHZ	(0x0 << 2)
+#define DA7219_PLL_INDIV_5_10_MHZ	(0x1 << 2)
+#define DA7219_PLL_INDIV_10_20_MHZ	(0x2 << 2)
+#define DA7219_PLL_INDIV_20_40_MHZ	(0x3 << 2)
+#define DA7219_PLL_INDIV_40_54_MHZ	(0x4 << 2)
+#define DA7219_PLL_MCLK_SQR_EN_SHIFT	5
+#define DA7219_PLL_MCLK_SQR_EN_MASK	(0x1 << 5)
+#define DA7219_PLL_MODE_SHIFT		6
+#define DA7219_PLL_MODE_MASK		(0x3 << 6)
+#define DA7219_PLL_MODE_BYPASS		(0x0 << 6)
+#define DA7219_PLL_MODE_NORMAL		(0x1 << 6)
+#define DA7219_PLL_MODE_SRM		(0x2 << 6)
+#define DA7219_PLL_MODE_32KHZ		(0x3 << 6)
+
+/* DA7219_PLL_FRAC_TOP = 0x22 */
+#define DA7219_PLL_FBDIV_FRAC_TOP_SHIFT	0
+#define DA7219_PLL_FBDIV_FRAC_TOP_MASK	(0x1F << 0)
+
+/* DA7219_PLL_FRAC_BOT = 0x23 */
+#define DA7219_PLL_FBDIV_FRAC_BOT_SHIFT	0
+#define DA7219_PLL_FBDIV_FRAC_BOT_MASK	(0xFF << 0)
+
+/* DA7219_PLL_INTEGER = 0x24 */
+#define DA7219_PLL_FBDIV_INTEGER_SHIFT	0
+#define DA7219_PLL_FBDIV_INTEGER_MASK	(0x7F << 0)
+
+/* DA7219_PLL_SRM_STS = 0x25 */
+#define DA7219_PLL_SRM_STATE_SHIFT	0
+#define DA7219_PLL_SRM_STATE_MASK	(0xF << 0)
+#define DA7219_PLL_SRM_STATUS_SHIFT	4
+#define DA7219_PLL_SRM_STATUS_MASK	(0xF << 4)
+#define DA7219_PLL_SRM_STS_SRM_LOCK	(0x1 << 7)
+
+/* DA7219_DIG_ROUTING_DAI = 0x2A */
+#define DA7219_DAI_L_SRC_SHIFT	0
+#define DA7219_DAI_L_SRC_MASK	(0x3 << 0)
+#define DA7219_DAI_R_SRC_SHIFT	4
+#define DA7219_DAI_R_SRC_MASK	(0x3 << 4)
+#define DA7219_OUT_SRC_MAX	4
+
+/* DA7219_DAI_CLK_MODE = 0x2B */
+#define DA7219_DAI_BCLKS_PER_WCLK_SHIFT	0
+#define DA7219_DAI_BCLKS_PER_WCLK_MASK	(0x3 << 0)
+#define DA7219_DAI_BCLKS_PER_WCLK_32	(0x0 << 0)
+#define DA7219_DAI_BCLKS_PER_WCLK_64	(0x1 << 0)
+#define DA7219_DAI_BCLKS_PER_WCLK_128	(0x2 << 0)
+#define DA7219_DAI_BCLKS_PER_WCLK_256	(0x3 << 0)
+#define DA7219_DAI_CLK_POL_SHIFT	2
+#define DA7219_DAI_CLK_POL_MASK		(0x1 << 2)
+#define DA7219_DAI_CLK_POL_INV		(0x1 << 2)
+#define DA7219_DAI_WCLK_POL_SHIFT	3
+#define DA7219_DAI_WCLK_POL_MASK	(0x1 << 3)
+#define DA7219_DAI_WCLK_POL_INV		(0x1 << 3)
+#define DA7219_DAI_WCLK_TRI_STATE_SHIFT	4
+#define DA7219_DAI_WCLK_TRI_STATE_MASK	(0x1 << 4)
+#define DA7219_DAI_CLK_EN_SHIFT		7
+#define DA7219_DAI_CLK_EN_MASK		(0x1 << 7)
+
+/* DA7219_DAI_CTRL = 0x2C */
+#define DA7219_DAI_FORMAT_SHIFT		0
+#define DA7219_DAI_FORMAT_MASK		(0x3 << 0)
+#define DA7219_DAI_FORMAT_I2S		(0x0 << 0)
+#define DA7219_DAI_FORMAT_LEFT_J	(0x1 << 0)
+#define DA7219_DAI_FORMAT_RIGHT_J	(0x2 << 0)
+#define DA7219_DAI_FORMAT_DSP		(0x3 << 0)
+#define DA7219_DAI_WORD_LENGTH_SHIFT	2
+#define DA7219_DAI_WORD_LENGTH_MASK	(0x3 << 2)
+#define DA7219_DAI_WORD_LENGTH_S16_LE	(0x0 << 2)
+#define DA7219_DAI_WORD_LENGTH_S20_LE	(0x1 << 2)
+#define DA7219_DAI_WORD_LENGTH_S24_LE	(0x2 << 2)
+#define DA7219_DAI_WORD_LENGTH_S32_LE	(0x3 << 2)
+#define DA7219_DAI_CH_NUM_SHIFT		4
+#define DA7219_DAI_CH_NUM_MASK		(0x3 << 4)
+#define DA7219_DAI_CH_NUM_MAX		2
+#define DA7219_DAI_EN_SHIFT		7
+#define DA7219_DAI_EN_MASK		(0x1 << 7)
+
+/* DA7219_DAI_TDM_CTRL = 0x2D */
+#define DA7219_DAI_TDM_CH_EN_SHIFT	0
+#define DA7219_DAI_TDM_CH_EN_MASK	(0x3 << 0)
+#define DA7219_DAI_OE_SHIFT		6
+#define DA7219_DAI_OE_MASK		(0x1 << 6)
+#define DA7219_DAI_TDM_MODE_EN_SHIFT	7
+#define DA7219_DAI_TDM_MODE_EN_MASK	(0x1 << 7)
+#define DA7219_DAI_TDM_MAX_SLOTS	2
+
+/* DA7219_DIG_ROUTING_DAC = 0x2E */
+#define DA7219_DAC_L_SRC_SHIFT		0
+#define DA7219_DAC_L_SRC_MASK		(0x3 << 0)
+#define DA7219_DAC_L_SRC_TONEGEN	(0x1 << 0)
+#define DA7219_DAC_L_MONO_SHIFT		3
+#define DA7219_DAC_L_MONO_MASK		(0x1 << 3)
+#define DA7219_DAC_R_SRC_SHIFT		4
+#define DA7219_DAC_R_SRC_MASK		(0x3 << 4)
+#define DA7219_DAC_R_SRC_TONEGEN	(0x1 << 4)
+#define DA7219_DAC_R_MONO_SHIFT		7
+#define DA7219_DAC_R_MONO_MASK		(0x1 << 7)
+
+/* DA7219_ALC_CTRL1 = 0x2F */
+#define DA7219_ALC_OFFSET_EN_SHIFT	0
+#define DA7219_ALC_OFFSET_EN_MASK	(0x1 << 0)
+#define DA7219_ALC_SYNC_MODE_SHIFT	1
+#define DA7219_ALC_SYNC_MODE_MASK	(0x1 << 1)
+#define DA7219_ALC_EN_SHIFT		3
+#define DA7219_ALC_EN_MASK		(0x1 << 3)
+#define DA7219_ALC_AUTO_CALIB_EN_SHIFT	4
+#define DA7219_ALC_AUTO_CALIB_EN_MASK	(0x1 << 4)
+#define DA7219_ALC_CALIB_OVERFLOW_SHIFT	5
+#define DA7219_ALC_CALIB_OVERFLOW_MASK	(0x1 << 5)
+
+/* DA7219_DAI_OFFSET_LOWER = 0x30 */
+#define DA7219_DAI_OFFSET_LOWER_SHIFT	0
+#define DA7219_DAI_OFFSET_LOWER_MASK	(0xFF << 0)
+
+/* DA7219_DAI_OFFSET_UPPER = 0x31 */
+#define DA7219_DAI_OFFSET_UPPER_SHIFT	0
+#define DA7219_DAI_OFFSET_UPPER_MASK	(0x7 << 0)
+#define DA7219_DAI_OFFSET_MAX		0x2FF
+
+/* DA7219_REFERENCES = 0x32 */
+#define DA7219_BIAS_EN_SHIFT		3
+#define DA7219_BIAS_EN_MASK		(0x1 << 3)
+#define DA7219_VMID_FAST_CHARGE_SHIFT	4
+#define DA7219_VMID_FAST_CHARGE_MASK	(0x1 << 4)
+
+/* DA7219_MIXIN_L_SELECT = 0x33 */
+#define DA7219_MIXIN_L_MIX_SELECT_SHIFT	0
+#define DA7219_MIXIN_L_MIX_SELECT_MASK	(0x1 << 0)
+
+/* DA7219_MIXIN_L_GAIN = 0x34 */
+#define DA7219_MIXIN_L_AMP_GAIN_SHIFT	0
+#define DA7219_MIXIN_L_AMP_GAIN_MASK	(0xF << 0)
+#define DA7219_MIXIN_L_AMP_GAIN_MAX	0xF
+
+/* DA7219_ADC_L_GAIN = 0x36 */
+#define DA7219_ADC_L_DIGITAL_GAIN_SHIFT	0
+#define DA7219_ADC_L_DIGITAL_GAIN_MASK	(0x7F << 0)
+#define DA7219_ADC_L_DIGITAL_GAIN_MAX	0x7F
+
+/* DA7219_ADC_FILTERS1 = 0x38 */
+#define DA7219_ADC_VOICE_HPF_CORNER_SHIFT	0
+#define DA7219_ADC_VOICE_HPF_CORNER_MASK	(0x7 << 0)
+#define DA7219_VOICE_HPF_CORNER_MAX		8
+#define DA7219_ADC_VOICE_EN_SHIFT		3
+#define DA7219_ADC_VOICE_EN_MASK		(0x1 << 3)
+#define DA7219_ADC_AUDIO_HPF_CORNER_SHIFT	4
+#define DA7219_ADC_AUDIO_HPF_CORNER_MASK	(0x3 << 4)
+#define DA7219_AUDIO_HPF_CORNER_MAX		4
+#define DA7219_ADC_HPF_EN_SHIFT			7
+#define DA7219_ADC_HPF_EN_MASK			(0x1 << 7)
+#define DA7219_HPF_MODE_SHIFT			0
+#define DA7219_HPF_DISABLED			((0x0 << 3) | (0x0 << 7))
+#define DA7219_HPF_AUDIO_EN			((0x0 << 3) | (0x1 << 7))
+#define DA7219_HPF_VOICE_EN			((0x1 << 3) | (0x1 << 7))
+#define DA7219_HPF_MODE_MASK			((0x1 << 3) | (0x1 << 7))
+#define DA7219_HPF_MODE_MAX			3
+
+/* DA7219_MIC_1_GAIN = 0x39 */
+#define DA7219_MIC_1_AMP_GAIN_SHIFT	0
+#define DA7219_MIC_1_AMP_GAIN_MASK	(0x7 << 0)
+
+/* DA7219_SIDETONE_CTRL = 0x3A */
+#define DA7219_SIDETONE_MUTE_EN_SHIFT	6
+#define DA7219_SIDETONE_MUTE_EN_MASK	(0x1 << 6)
+#define DA7219_SIDETONE_EN_SHIFT	7
+#define DA7219_SIDETONE_EN_MASK		(0x1 << 7)
+
+/* DA7219_SIDETONE_GAIN = 0x3B */
+#define DA7219_SIDETONE_GAIN_SHIFT	0
+#define DA7219_SIDETONE_GAIN_MASK	(0xF << 0)
+#define DA7219_SIDETONE_GAIN_MAX	0xE
+
+/* DA7219_DROUTING_ST_OUTFILT_1L = 0x3C */
+#define DA7219_OUTFILT_ST_1L_SRC_SHIFT		0
+#define DA7219_OUTFILT_ST_1L_SRC_MASK		(0x7 << 0)
+#define DA7219_DMIX_ST_SRC_OUTFILT1L_SHIFT	0
+#define DA7219_DMIX_ST_SRC_OUTFILT1R_SHIFT	1
+#define DA7219_DMIX_ST_SRC_SIDETONE_SHIFT	2
+#define DA7219_DMIX_ST_SRC_OUTFILT1L		(0x1 << 0)
+#define DA7219_DMIX_ST_SRC_OUTFILT1R		(0x1 << 1)
+
+/* DA7219_DROUTING_ST_OUTFILT_1R = 0x3D */
+#define DA7219_OUTFILT_ST_1R_SRC_SHIFT	0
+#define DA7219_OUTFILT_ST_1R_SRC_MASK	(0x7 << 0)
+
+/* DA7219_DAC_FILTERS5 = 0x40 */
+#define DA7219_DAC_SOFTMUTE_RATE_SHIFT	4
+#define DA7219_DAC_SOFTMUTE_RATE_MASK	(0x7 << 4)
+#define DA7219_DAC_SOFTMUTE_RATE_MAX	7
+#define DA7219_DAC_SOFTMUTE_EN_SHIFT	7
+#define DA7219_DAC_SOFTMUTE_EN_MASK	(0x1 << 7)
+
+/* DA7219_DAC_FILTERS2 = 0x41 */
+#define DA7219_DAC_EQ_BAND1_SHIFT	0
+#define DA7219_DAC_EQ_BAND1_MASK	(0xF << 0)
+#define DA7219_DAC_EQ_BAND2_SHIFT	4
+#define DA7219_DAC_EQ_BAND2_MASK	(0xF << 4)
+#define DA7219_DAC_EQ_BAND_MAX		0xF
+
+/* DA7219_DAC_FILTERS3 = 0x42 */
+#define DA7219_DAC_EQ_BAND3_SHIFT	0
+#define DA7219_DAC_EQ_BAND3_MASK	(0xF << 0)
+#define DA7219_DAC_EQ_BAND4_SHIFT	4
+#define DA7219_DAC_EQ_BAND4_MASK	(0xF << 4)
+
+/* DA7219_DAC_FILTERS4 = 0x43 */
+#define DA7219_DAC_EQ_BAND5_SHIFT	0
+#define DA7219_DAC_EQ_BAND5_MASK	(0xF << 0)
+#define DA7219_DAC_EQ_EN_SHIFT		7
+#define DA7219_DAC_EQ_EN_MASK		(0x1 << 7)
+
+/* DA7219_DAC_FILTERS1 = 0x44 */
+#define DA7219_DAC_VOICE_HPF_CORNER_SHIFT	0
+#define DA7219_DAC_VOICE_HPF_CORNER_MASK	(0x7 << 0)
+#define DA7219_DAC_VOICE_EN_SHIFT		3
+#define DA7219_DAC_VOICE_EN_MASK		(0x1 << 3)
+#define DA7219_DAC_AUDIO_HPF_CORNER_SHIFT	4
+#define DA7219_DAC_AUDIO_HPF_CORNER_MASK	(0x3 << 4)
+#define DA7219_DAC_HPF_EN_SHIFT			7
+#define DA7219_DAC_HPF_EN_MASK			(0x1 << 7)
+
+/* DA7219_DAC_L_GAIN = 0x45 */
+#define DA7219_DAC_L_DIGITAL_GAIN_SHIFT	0
+#define DA7219_DAC_L_DIGITAL_GAIN_MASK	(0x7F << 0)
+#define DA7219_DAC_DIGITAL_GAIN_MAX	0x7F
+#define DA7219_DAC_DIGITAL_GAIN_0DB	(0x6F << 0)
+
+/* DA7219_DAC_R_GAIN = 0x46 */
+#define DA7219_DAC_R_DIGITAL_GAIN_SHIFT	0
+#define DA7219_DAC_R_DIGITAL_GAIN_MASK	(0x7F << 0)
+
+/* DA7219_CP_CTRL = 0x47 */
+#define DA7219_CP_MCHANGE_SHIFT		4
+#define DA7219_CP_MCHANGE_MASK		(0x3 << 4)
+#define DA7219_CP_MCHANGE_REL_MASK	0x3
+#define DA7219_CP_MCHANGE_MAX		3
+#define DA7219_CP_MCHANGE_LARGEST_VOL	0x1
+#define DA7219_CP_MCHANGE_DAC_VOL	0x2
+#define DA7219_CP_MCHANGE_SIG_MAG	0x3
+#define DA7219_CP_EN_SHIFT		7
+#define DA7219_CP_EN_MASK		(0x1 << 7)
+
+/* DA7219_HP_L_GAIN = 0x48 */
+#define DA7219_HP_L_AMP_GAIN_SHIFT	0
+#define DA7219_HP_L_AMP_GAIN_MASK	(0x3F << 0)
+#define DA7219_HP_AMP_GAIN_MAX		0x3F
+#define DA7219_HP_AMP_GAIN_0DB		(0x39 << 0)
+
+/* DA7219_HP_R_GAIN = 0x49 */
+#define DA7219_HP_R_AMP_GAIN_SHIFT	0
+#define DA7219_HP_R_AMP_GAIN_MASK	(0x3F << 0)
+
+/* DA7219_MIXOUT_L_SELECT = 0x4B */
+#define DA7219_MIXOUT_L_MIX_SELECT_SHIFT	0
+#define DA7219_MIXOUT_L_MIX_SELECT_MASK		(0x1 << 0)
+
+/* DA7219_MIXOUT_R_SELECT = 0x4C */
+#define DA7219_MIXOUT_R_MIX_SELECT_SHIFT	0
+#define DA7219_MIXOUT_R_MIX_SELECT_MASK		(0x1 << 0)
+
+/* DA7219_SYSTEM_MODES_INPUT = 0x50 */
+#define DA7219_MODE_SUBMIT_SHIFT	0
+#define DA7219_MODE_SUBMIT_MASK		(0x1 << 0)
+#define DA7219_ADC_MODE_SHIFT		1
+#define DA7219_ADC_MODE_MASK		(0x7F << 1)
+
+/* DA7219_SYSTEM_MODES_OUTPUT = 0x51 */
+#define DA7219_MODE_SUBMIT_SHIFT	0
+#define DA7219_MODE_SUBMIT_MASK		(0x1 << 0)
+#define DA7219_DAC_MODE_SHIFT		1
+#define DA7219_DAC_MODE_MASK		(0x7F << 1)
+
+/* DA7219_MICBIAS_CTRL = 0x62 */
+#define DA7219_MICBIAS1_LEVEL_SHIFT	0
+#define DA7219_MICBIAS1_LEVEL_MASK	(0x7 << 0)
+#define DA7219_MICBIAS1_EN_SHIFT	3
+#define DA7219_MICBIAS1_EN_MASK		(0x1 << 3)
+
+/* DA7219_MIC_1_CTRL = 0x63 */
+#define DA7219_MIC_1_AMP_RAMP_EN_SHIFT	5
+#define DA7219_MIC_1_AMP_RAMP_EN_MASK	(0x1 << 5)
+#define DA7219_MIC_1_AMP_MUTE_EN_SHIFT	6
+#define DA7219_MIC_1_AMP_MUTE_EN_MASK	(0x1 << 6)
+#define DA7219_MIC_1_AMP_EN_SHIFT	7
+#define DA7219_MIC_1_AMP_EN_MASK	(0x1 << 7)
+
+/* DA7219_MIXIN_L_CTRL = 0x65 */
+#define DA7219_MIXIN_L_MIX_EN_SHIFT		3
+#define DA7219_MIXIN_L_MIX_EN_MASK		(0x1 << 3)
+#define DA7219_MIXIN_L_AMP_ZC_EN_SHIFT		4
+#define DA7219_MIXIN_L_AMP_ZC_EN_MASK		(0x1 << 4)
+#define DA7219_MIXIN_L_AMP_RAMP_EN_SHIFT	5
+#define DA7219_MIXIN_L_AMP_RAMP_EN_MASK		(0x1 << 5)
+#define DA7219_MIXIN_L_AMP_MUTE_EN_SHIFT	6
+#define DA7219_MIXIN_L_AMP_MUTE_EN_MASK		(0x1 << 6)
+#define DA7219_MIXIN_L_AMP_EN_SHIFT		7
+#define DA7219_MIXIN_L_AMP_EN_MASK		(0x1 << 7)
+
+/* DA7219_ADC_L_CTRL = 0x67 */
+#define DA7219_ADC_L_BIAS_SHIFT		0
+#define DA7219_ADC_L_BIAS_MASK		(0x3 << 0)
+#define DA7219_ADC_L_RAMP_EN_SHIFT	5
+#define DA7219_ADC_L_RAMP_EN_MASK	(0x1 << 5)
+#define DA7219_ADC_L_MUTE_EN_SHIFT	6
+#define DA7219_ADC_L_MUTE_EN_MASK	(0x1 << 6)
+#define DA7219_ADC_L_EN_SHIFT		7
+#define DA7219_ADC_L_EN_MASK		(0x1 << 7)
+
+/* DA7219_DAC_L_CTRL = 0x69 */
+#define DA7219_DAC_L_RAMP_EN_SHIFT	5
+#define DA7219_DAC_L_RAMP_EN_MASK	(0x1 << 5)
+#define DA7219_DAC_L_MUTE_EN_SHIFT	6
+#define DA7219_DAC_L_MUTE_EN_MASK	(0x1 << 6)
+#define DA7219_DAC_L_EN_SHIFT		7
+#define DA7219_DAC_L_EN_MASK		(0x1 << 7)
+
+/* DA7219_DAC_R_CTRL = 0x6A */
+#define DA7219_DAC_R_RAMP_EN_SHIFT	5
+#define DA7219_DAC_R_RAMP_EN_MASK	(0x1 << 5)
+#define DA7219_DAC_R_MUTE_EN_SHIFT	6
+#define DA7219_DAC_R_MUTE_EN_MASK	(0x1 << 6)
+#define DA7219_DAC_R_EN_SHIFT		7
+#define DA7219_DAC_R_EN_MASK		(0x1 << 7)
+
+/* DA7219_HP_L_CTRL = 0x6B */
+#define DA7219_HP_L_AMP_MIN_GAIN_EN_SHIFT	2
+#define DA7219_HP_L_AMP_MIN_GAIN_EN_MASK	(0x1 << 2)
+#define DA7219_HP_L_AMP_OE_SHIFT		3
+#define DA7219_HP_L_AMP_OE_MASK			(0x1 << 3)
+#define DA7219_HP_L_AMP_ZC_EN_SHIFT		4
+#define DA7219_HP_L_AMP_ZC_EN_MASK		(0x1 << 4)
+#define DA7219_HP_L_AMP_RAMP_EN_SHIFT		5
+#define DA7219_HP_L_AMP_RAMP_EN_MASK		(0x1 << 5)
+#define DA7219_HP_L_AMP_MUTE_EN_SHIFT		6
+#define DA7219_HP_L_AMP_MUTE_EN_MASK		(0x1 << 6)
+#define DA7219_HP_L_AMP_EN_SHIFT		7
+#define DA7219_HP_L_AMP_EN_MASK			(0x1 << 7)
+
+/* DA7219_HP_R_CTRL = 0x6C */
+#define DA7219_HP_R_AMP_MIN_GAIN_EN_SHIFT	2
+#define DA7219_HP_R_AMP_MIN_GAIN_EN_MASK	(0x1 << 2)
+#define DA7219_HP_R_AMP_OE_SHIFT		3
+#define DA7219_HP_R_AMP_OE_MASK			(0x1 << 3)
+#define DA7219_HP_R_AMP_ZC_EN_SHIFT		4
+#define DA7219_HP_R_AMP_ZC_EN_MASK		(0x1 << 4)
+#define DA7219_HP_R_AMP_RAMP_EN_SHIFT		5
+#define DA7219_HP_R_AMP_RAMP_EN_MASK		(0x1 << 5)
+#define DA7219_HP_R_AMP_MUTE_EN_SHIFT		6
+#define DA7219_HP_R_AMP_MUTE_EN_MASK		(0x1 << 6)
+#define DA7219_HP_R_AMP_EN_SHIFT		7
+#define DA7219_HP_R_AMP_EN_MASK			(0x1 << 7)
+
+/* DA7219_MIXOUT_L_CTRL = 0x6E */
+#define DA7219_MIXOUT_L_AMP_EN_SHIFT	7
+#define DA7219_MIXOUT_L_AMP_EN_MASK	(0x1 << 7)
+
+/* DA7219_MIXOUT_R_CTRL = 0x6F */
+#define DA7219_MIXOUT_R_AMP_EN_SHIFT	7
+#define DA7219_MIXOUT_R_AMP_EN_MASK	(0x1 << 7)
+
+/* DA7219_CHIP_ID1 = 0x81 */
+#define DA7219_CHIP_ID1_SHIFT	0
+#define DA7219_CHIP_ID1_MASK	(0xFF << 0)
+
+/* DA7219_CHIP_ID2 = 0x82 */
+#define DA7219_CHIP_ID2_SHIFT	0
+#define DA7219_CHIP_ID2_MASK	(0xFF << 0)
+
+/* DA7219_CHIP_REVISION = 0x83 */
+#define DA7219_CHIP_MINOR_SHIFT	0
+#define DA7219_CHIP_MINOR_MASK	(0xF << 0)
+#define DA7219_CHIP_MAJOR_SHIFT	4
+#define DA7219_CHIP_MAJOR_MASK	(0xF << 4)
+
+/* DA7219_LDO_CTRL = 0x90 */
+#define DA7219_LDO_LEVEL_SELECT_SHIFT	4
+#define DA7219_LDO_LEVEL_SELECT_MASK	(0x3 << 4)
+#define DA7219_LDO_EN_SHIFT		7
+#define DA7219_LDO_EN_MASK		(0x1 << 7)
+
+/* DA7219_IO_CTRL = 0x91 */
+#define DA7219_IO_VOLTAGE_LEVEL_SHIFT		0
+#define DA7219_IO_VOLTAGE_LEVEL_MASK		(0x1 << 0)
+#define DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V	0
+#define DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V	1
+
+/* DA7219_GAIN_RAMP_CTRL = 0x92 */
+#define DA7219_GAIN_RAMP_RATE_SHIFT	0
+#define DA7219_GAIN_RAMP_RATE_MASK	(0x3 << 0)
+#define DA7219_GAIN_RAMP_RATE_MAX	4
+
+/* DA7219_PC_COUNT = 0x94 */
+#define DA7219_PC_FREERUN_SHIFT		0
+#define DA7219_PC_FREERUN_MASK		(0x1 << 0)
+#define DA7219_PC_RESYNC_AUTO_SHIFT	1
+#define DA7219_PC_RESYNC_AUTO_MASK	(0x1 << 1)
+
+/* DA7219_CP_VOL_THRESHOLD1 = 0x95 */
+#define DA7219_CP_THRESH_VDD2_SHIFT	0
+#define DA7219_CP_THRESH_VDD2_MASK	(0x3F << 0)
+#define DA7219_CP_THRESH_VDD2_MAX	0x3F
+
+/* DA7219_DIG_CTRL = 0x99 */
+#define DA7219_DAC_L_INV_SHIFT	3
+#define DA7219_DAC_L_INV_MASK	(0x1 << 3)
+#define DA7219_DAC_R_INV_SHIFT	7
+#define DA7219_DAC_R_INV_MASK	(0x1 << 7)
+
+/* DA7219_ALC_CTRL2 = 0x9A */
+#define DA7219_ALC_ATTACK_SHIFT		0
+#define DA7219_ALC_ATTACK_MASK		(0xF << 0)
+#define DA7219_ALC_ATTACK_MAX		13
+#define DA7219_ALC_RELEASE_SHIFT	4
+#define DA7219_ALC_RELEASE_MASK		(0xF << 4)
+#define DA7219_ALC_RELEASE_MAX		11
+
+/* DA7219_ALC_CTRL3 = 0x9B */
+#define DA7219_ALC_HOLD_SHIFT		0
+#define DA7219_ALC_HOLD_MASK		(0xF << 0)
+#define DA7219_ALC_HOLD_MAX		16
+#define DA7219_ALC_INTEG_ATTACK_SHIFT	4
+#define DA7219_ALC_INTEG_ATTACK_MASK	(0x3 << 4)
+#define DA7219_ALC_INTEG_RELEASE_SHIFT	6
+#define DA7219_ALC_INTEG_RELEASE_MASK	(0x3 << 6)
+#define DA7219_ALC_INTEG_MAX		4
+
+/* DA7219_ALC_NOISE = 0x9C */
+#define DA7219_ALC_NOISE_SHIFT		0
+#define DA7219_ALC_NOISE_MASK		(0x3F << 0)
+#define DA7219_ALC_THRESHOLD_MAX	0x3F
+
+/* DA7219_ALC_TARGET_MIN = 0x9D */
+#define DA7219_ALC_THRESHOLD_MIN_SHIFT	0
+#define DA7219_ALC_THRESHOLD_MIN_MASK	(0x3F << 0)
+
+/* DA7219_ALC_TARGET_MAX = 0x9E */
+#define DA7219_ALC_THRESHOLD_MAX_SHIFT	0
+#define DA7219_ALC_THRESHOLD_MAX_MASK	(0x3F << 0)
+
+/* DA7219_ALC_GAIN_LIMITS = 0x9F */
+#define DA7219_ALC_ATTEN_MAX_SHIFT	0
+#define DA7219_ALC_ATTEN_MAX_MASK	(0xF << 0)
+#define DA7219_ALC_GAIN_MAX_SHIFT	4
+#define DA7219_ALC_GAIN_MAX_MASK	(0xF << 4)
+#define DA7219_ALC_ATTEN_GAIN_MAX	0xF
+
+/* DA7219_ALC_ANA_GAIN_LIMITS = 0xA0 */
+#define DA7219_ALC_ANA_GAIN_MIN_SHIFT	0
+#define DA7219_ALC_ANA_GAIN_MIN_MASK	(0x7 << 0)
+#define DA7219_ALC_ANA_GAIN_MIN		0x1
+#define DA7219_ALC_ANA_GAIN_MAX_SHIFT	4
+#define DA7219_ALC_ANA_GAIN_MAX_MASK	(0x7 << 4)
+#define DA7219_ALC_ANA_GAIN_MAX		0x7
+
+/* DA7219_ALC_ANTICLIP_CTRL = 0xA1 */
+#define DA7219_ALC_ANTICLIP_STEP_SHIFT	0
+#define DA7219_ALC_ANTICLIP_STEP_MASK	(0x3 << 0)
+#define DA7219_ALC_ANTICLIP_STEP_MAX	4
+#define DA7219_ALC_ANTIPCLIP_EN_SHIFT	7
+#define DA7219_ALC_ANTIPCLIP_EN_MASK	(0x1 << 7)
+
+/* DA7219_ALC_ANTICLIP_LEVEL = 0xA2 */
+#define DA7219_ALC_ANTICLIP_LEVEL_SHIFT	0
+#define DA7219_ALC_ANTICLIP_LEVEL_MASK	(0x7F << 0)
+
+/* DA7219_ALC_OFFSET_AUTO_M_L = 0xA3 */
+#define DA7219_ALC_OFFSET_AUTO_M_L_SHIFT	0
+#define DA7219_ALC_OFFSET_AUTO_M_L_MASK		(0xFF << 0)
+
+/* DA7219_ALC_OFFSET_AUTO_U_L = 0xA4 */
+#define DA7219_ALC_OFFSET_AUTO_U_L_SHIFT	0
+#define DA7219_ALC_OFFSET_AUTO_U_L_MASK		(0xF << 0)
+
+/* DA7219_DAC_NG_SETUP_TIME = 0xAF */
+#define DA7219_DAC_NG_SETUP_TIME_SHIFT	0
+#define DA7219_DAC_NG_SETUP_TIME_MASK	(0x3 << 0)
+#define DA7219_DAC_NG_SETUP_TIME_MAX	4
+#define DA7219_DAC_NG_RAMPUP_RATE_SHIFT	2
+#define DA7219_DAC_NG_RAMPUP_RATE_MASK	(0x1 << 2)
+#define DA7219_DAC_NG_RAMPDN_RATE_SHIFT	3
+#define DA7219_DAC_NG_RAMPDN_RATE_MASK	(0x1 << 3)
+#define DA7219_DAC_NG_RAMP_RATE_MAX	2
+
+/* DA7219_DAC_NG_OFF_THRESH = 0xB0 */
+#define DA7219_DAC_NG_OFF_THRESHOLD_SHIFT	0
+#define DA7219_DAC_NG_OFF_THRESHOLD_MASK	(0x7 << 0)
+#define DA7219_DAC_NG_THRESHOLD_MAX		0x7
+
+/* DA7219_DAC_NG_ON_THRESH = 0xB1 */
+#define DA7219_DAC_NG_ON_THRESHOLD_SHIFT	0
+#define DA7219_DAC_NG_ON_THRESHOLD_MASK		(0x7 << 0)
+
+/* DA7219_DAC_NG_CTRL = 0xB2 */
+#define DA7219_DAC_NG_EN_SHIFT	7
+#define DA7219_DAC_NG_EN_MASK	(0x1 << 7)
+
+/* DA7219_TONE_GEN_CFG1 = 0xB4 */
+#define DA7219_DTMF_REG_SHIFT		0
+#define DA7219_DTMF_REG_MASK		(0xF << 0)
+#define DA7219_DTMF_REG_MAX		16
+#define DA7219_DTMF_EN_SHIFT		4
+#define DA7219_DTMF_EN_MASK		(0x1 << 4)
+#define DA7219_START_STOPN_SHIFT	7
+#define DA7219_START_STOPN_MASK		(0x1 << 7)
+
+/* DA7219_TONE_GEN_CFG2 = 0xB5 */
+#define DA7219_SWG_SEL_SHIFT		0
+#define DA7219_SWG_SEL_MASK		(0x3 << 0)
+#define DA7219_SWG_SEL_MAX		4
+#define DA7219_SWG_SEL_SRAMP		(0x3 << 0)
+#define DA7219_TONE_GEN_GAIN_SHIFT	4
+#define DA7219_TONE_GEN_GAIN_MASK	(0xF << 4)
+#define DA7219_TONE_GEN_GAIN_MAX	0xF
+#define DA7219_TONE_GEN_GAIN_MINUS_9DB	(0x3 << 4)
+#define DA7219_TONE_GEN_GAIN_MINUS_15DB	(0x5 << 4)
+
+/* DA7219_TONE_GEN_CYCLES = 0xB6 */
+#define DA7219_BEEP_CYCLES_SHIFT	0
+#define DA7219_BEEP_CYCLES_MASK		(0x7 << 0)
+
+/* DA7219_TONE_GEN_FREQ1_L = 0xB7 */
+#define DA7219_FREQ1_L_SHIFT	0
+#define DA7219_FREQ1_L_MASK	(0xFF << 0)
+#define DA7219_FREQ_MAX		0xFFFF
+
+/* DA7219_TONE_GEN_FREQ1_U = 0xB8 */
+#define DA7219_FREQ1_U_SHIFT	0
+#define DA7219_FREQ1_U_MASK	(0xFF << 0)
+
+/* DA7219_TONE_GEN_FREQ2_L = 0xB9 */
+#define DA7219_FREQ2_L_SHIFT	0
+#define DA7219_FREQ2_L_MASK	(0xFF << 0)
+
+/* DA7219_TONE_GEN_FREQ2_U = 0xBA */
+#define DA7219_FREQ2_U_SHIFT	0
+#define DA7219_FREQ2_U_MASK	(0xFF << 0)
+
+/* DA7219_TONE_GEN_ON_PER = 0xBB */
+#define DA7219_BEEP_ON_PER_SHIFT	0
+#define DA7219_BEEP_ON_PER_MASK		(0x3F << 0)
+#define DA7219_BEEP_ON_OFF_MAX		0x3F
+
+/* DA7219_TONE_GEN_OFF_PER = 0xBC */
+#define DA7219_BEEP_OFF_PER_SHIFT	0
+#define DA7219_BEEP_OFF_PER_MASK	(0x3F << 0)
+
+/* DA7219_SYSTEM_STATUS = 0xE0 */
+#define DA7219_SC1_BUSY_SHIFT	0
+#define DA7219_SC1_BUSY_MASK	(0x1 << 0)
+#define DA7219_SC2_BUSY_SHIFT	1
+#define DA7219_SC2_BUSY_MASK	(0x1 << 1)
+
+/* DA7219_SYSTEM_ACTIVE = 0xFD */
+#define DA7219_SYSTEM_ACTIVE_SHIFT	0
+#define DA7219_SYSTEM_ACTIVE_MASK	(0x1 << 0)
+
+
+/*
+ * General defines & data
+ */
+
+/* Register inversion */
+#define DA7219_NO_INVERT	0
+#define DA7219_INVERT		1
+
+/* Byte related defines */
+#define DA7219_BYTE_SHIFT	8
+#define DA7219_BYTE_MASK	0xFF
+
+/* PLL Output Frequencies */
+#define DA7219_PLL_FREQ_OUT_90316	90316800
+#define DA7219_PLL_FREQ_OUT_98304	98304000
+
+/* PLL Frequency Dividers */
+#define DA7219_PLL_INDIV_2_5_MHZ_VAL	1
+#define DA7219_PLL_INDIV_5_10_MHZ_VAL	2
+#define DA7219_PLL_INDIV_10_20_MHZ_VAL	4
+#define DA7219_PLL_INDIV_20_40_MHZ_VAL	8
+#define DA7219_PLL_INDIV_40_54_MHZ_VAL	16
+
+/* SRM */
+#define DA7219_SRM_CHECK_RETRIES	8
+
+enum da7219_clk_src {
+	DA7219_CLKSRC_MCLK = 0,
+	DA7219_CLKSRC_MCLK_SQR,
+};
+
+enum da7219_sys_clk {
+	DA7219_SYSCLK_MCLK = 0,
+	DA7219_SYSCLK_PLL,
+	DA7219_SYSCLK_PLL_SRM,
+	DA7219_SYSCLK_PLL_32KHZ
+};
+
+/* Regulators */
+enum da7219_supplies {
+	DA7219_SUPPLY_VDD = 0,
+	DA7219_SUPPLY_VDDMIC,
+	DA7219_SUPPLY_VDDIO,
+	DA7219_NUM_SUPPLIES,
+};
+
+struct da7219_aad_priv;
+
+/* Private data */
+struct da7219_priv {
+	struct da7219_aad_priv *aad;
+	struct da7219_pdata *pdata;
+
+	struct regulator_bulk_data supplies[DA7219_NUM_SUPPLIES];
+	struct regmap *regmap;
+	struct mutex lock;
+
+	struct clk *mclk;
+	unsigned int mclk_rate;
+	int clk_src;
+
+	bool master;
+	bool alc_en;
+};
+
+#endif /* __DA7219_H */
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 6a091016e0fc..969e337dc17c 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -129,7 +129,7 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
-	int deemph = ucontrol->value.integer.value[0];
+	unsigned int deemph = ucontrol->value.integer.value[0];
 	int ret;
 
 	if (deemph > 1)
diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c
deleted file mode 100644
index bd42ad34e004..000000000000
--- a/sound/soc/codecs/hdmi.c
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- * ALSA SoC codec driver for HDMI audio codecs.
- * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/
- * Author: Ricardo Neri <ricardo.neri@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-#include <linux/module.h>
-#include <sound/soc.h>
-#include <linux/of.h>
-#include <linux/of_device.h>
-
-#define DRV_NAME "hdmi-audio-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 |
-			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 | SNDRV_PCM_FMTBIT_S32_LE,
-		.sig_bits = 24,
-	},
-	.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,
-	},
-
-};
-
-#ifdef CONFIG_OF
-static const struct of_device_id hdmi_audio_codec_ids[] = {
-	{ .compatible = "linux,hdmi-audio", },
-	{ }
-};
-MODULE_DEVICE_TABLE(of, hdmi_audio_codec_ids);
-#endif
-
-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),
-	.ignore_pmdown_time = true,
-};
-
-static int hdmi_codec_probe(struct platform_device *pdev)
-{
-	return snd_soc_register_codec(&pdev->dev, &hdmi_codec,
-			&hdmi_codec_dai, 1);
-}
-
-static int hdmi_codec_remove(struct platform_device *pdev)
-{
-	snd_soc_unregister_codec(&pdev->dev);
-	return 0;
-}
-
-static struct platform_driver hdmi_codec_driver = {
-	.driver		= {
-		.name	= DRV_NAME,
-		.of_match_table = of_match_ptr(hdmi_audio_codec_ids),
-	},
-
-	.probe		= hdmi_codec_probe,
-	.remove		= hdmi_codec_remove,
-};
-
-module_platform_driver(hdmi_codec_driver);
-
-MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
-MODULE_DESCRIPTION("ASoC generic HDMI codec driver");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
new file mode 100644
index 000000000000..7fc7b4e3f444
--- /dev/null
+++ b/sound/soc/codecs/nau8825.c
@@ -0,0 +1,1309 @@
+/*
+ * Nuvoton NAU8825 audio codec driver
+ *
+ * Copyright 2015 Google Chromium project.
+ *  Author: Anatol Pomozov <anatol@chromium.org>
+ * Copyright 2015 Nuvoton Technology Corp.
+ *  Co-author: Meng-Huang Kuo <mhkuo@nuvoton.com>
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/clk.h>
+#include <linux/acpi.h>
+#include <linux/math64.h>
+
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+
+#include "nau8825.h"
+
+#define NAU_FREF_MAX 13500000
+#define NAU_FVCO_MAX 100000000
+#define NAU_FVCO_MIN 90000000
+
+struct nau8825_fll {
+	int mclk_src;
+	int ratio;
+	int fll_frac;
+	int fll_int;
+	int clk_ref_div;
+};
+
+struct nau8825_fll_attr {
+	unsigned int param;
+	unsigned int val;
+};
+
+/* scaling for mclk from sysclk_src output */
+static const struct nau8825_fll_attr mclk_src_scaling[] = {
+	{ 1, 0x0 },
+	{ 2, 0x2 },
+	{ 4, 0x3 },
+	{ 8, 0x4 },
+	{ 16, 0x5 },
+	{ 32, 0x6 },
+	{ 3, 0x7 },
+	{ 6, 0xa },
+	{ 12, 0xb },
+	{ 24, 0xc },
+	{ 48, 0xd },
+	{ 96, 0xe },
+	{ 5, 0xf },
+};
+
+/* ratio for input clk freq */
+static const struct nau8825_fll_attr fll_ratio[] = {
+	{ 512000, 0x01 },
+	{ 256000, 0x02 },
+	{ 128000, 0x04 },
+	{ 64000, 0x08 },
+	{ 32000, 0x10 },
+	{ 8000, 0x20 },
+	{ 4000, 0x40 },
+};
+
+static const struct nau8825_fll_attr fll_pre_scalar[] = {
+	{ 1, 0x0 },
+	{ 2, 0x1 },
+	{ 4, 0x2 },
+	{ 8, 0x3 },
+};
+
+static const struct reg_default nau8825_reg_defaults[] = {
+	{ NAU8825_REG_ENA_CTRL, 0x00ff },
+	{ NAU8825_REG_CLK_DIVIDER, 0x0050 },
+	{ NAU8825_REG_FLL1, 0x0 },
+	{ NAU8825_REG_FLL2, 0x3126 },
+	{ NAU8825_REG_FLL3, 0x0008 },
+	{ NAU8825_REG_FLL4, 0x0010 },
+	{ NAU8825_REG_FLL5, 0x0 },
+	{ NAU8825_REG_FLL6, 0x6000 },
+	{ NAU8825_REG_FLL_VCO_RSV, 0xf13c },
+	{ NAU8825_REG_HSD_CTRL, 0x000c },
+	{ NAU8825_REG_JACK_DET_CTRL, 0x0 },
+	{ NAU8825_REG_INTERRUPT_MASK, 0x0 },
+	{ NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff },
+	{ NAU8825_REG_SAR_CTRL, 0x0015 },
+	{ NAU8825_REG_KEYDET_CTRL, 0x0110 },
+	{ NAU8825_REG_VDET_THRESHOLD_1, 0x0 },
+	{ NAU8825_REG_VDET_THRESHOLD_2, 0x0 },
+	{ NAU8825_REG_VDET_THRESHOLD_3, 0x0 },
+	{ NAU8825_REG_VDET_THRESHOLD_4, 0x0 },
+	{ NAU8825_REG_GPIO34_CTRL, 0x0 },
+	{ NAU8825_REG_GPIO12_CTRL, 0x0 },
+	{ NAU8825_REG_TDM_CTRL, 0x0 },
+	{ NAU8825_REG_I2S_PCM_CTRL1, 0x000b },
+	{ NAU8825_REG_I2S_PCM_CTRL2, 0x8010 },
+	{ NAU8825_REG_LEFT_TIME_SLOT, 0x0 },
+	{ NAU8825_REG_RIGHT_TIME_SLOT, 0x0 },
+	{ NAU8825_REG_BIQ_CTRL, 0x0 },
+	{ NAU8825_REG_BIQ_COF1, 0x0 },
+	{ NAU8825_REG_BIQ_COF2, 0x0 },
+	{ NAU8825_REG_BIQ_COF3, 0x0 },
+	{ NAU8825_REG_BIQ_COF4, 0x0 },
+	{ NAU8825_REG_BIQ_COF5, 0x0 },
+	{ NAU8825_REG_BIQ_COF6, 0x0 },
+	{ NAU8825_REG_BIQ_COF7, 0x0 },
+	{ NAU8825_REG_BIQ_COF8, 0x0 },
+	{ NAU8825_REG_BIQ_COF9, 0x0 },
+	{ NAU8825_REG_BIQ_COF10, 0x0 },
+	{ NAU8825_REG_ADC_RATE, 0x0010 },
+	{ NAU8825_REG_DAC_CTRL1, 0x0001 },
+	{ NAU8825_REG_DAC_CTRL2, 0x0 },
+	{ NAU8825_REG_DAC_DGAIN_CTRL, 0x0 },
+	{ NAU8825_REG_ADC_DGAIN_CTRL, 0x00cf },
+	{ NAU8825_REG_MUTE_CTRL, 0x0 },
+	{ NAU8825_REG_HSVOL_CTRL, 0x0 },
+	{ NAU8825_REG_DACL_CTRL, 0x02cf },
+	{ NAU8825_REG_DACR_CTRL, 0x00cf },
+	{ NAU8825_REG_ADC_DRC_KNEE_IP12, 0x1486 },
+	{ NAU8825_REG_ADC_DRC_KNEE_IP34, 0x0f12 },
+	{ NAU8825_REG_ADC_DRC_SLOPES, 0x25ff },
+	{ NAU8825_REG_ADC_DRC_ATKDCY, 0x3457 },
+	{ NAU8825_REG_DAC_DRC_KNEE_IP12, 0x1486 },
+	{ NAU8825_REG_DAC_DRC_KNEE_IP34, 0x0f12 },
+	{ NAU8825_REG_DAC_DRC_SLOPES, 0x25f9 },
+	{ NAU8825_REG_DAC_DRC_ATKDCY, 0x3457 },
+	{ NAU8825_REG_IMM_MODE_CTRL, 0x0 },
+	{ NAU8825_REG_CLASSG_CTRL, 0x0 },
+	{ NAU8825_REG_OPT_EFUSE_CTRL, 0x0 },
+	{ NAU8825_REG_MISC_CTRL, 0x0 },
+	{ NAU8825_REG_BIAS_ADJ, 0x0 },
+	{ NAU8825_REG_TRIM_SETTINGS, 0x0 },
+	{ NAU8825_REG_ANALOG_CONTROL_1, 0x0 },
+	{ NAU8825_REG_ANALOG_CONTROL_2, 0x0 },
+	{ NAU8825_REG_ANALOG_ADC_1, 0x0011 },
+	{ NAU8825_REG_ANALOG_ADC_2, 0x0020 },
+	{ NAU8825_REG_RDAC, 0x0008 },
+	{ NAU8825_REG_MIC_BIAS, 0x0006 },
+	{ NAU8825_REG_BOOST, 0x0 },
+	{ NAU8825_REG_FEPGA, 0x0 },
+	{ NAU8825_REG_POWER_UP_CONTROL, 0x0 },
+	{ NAU8825_REG_CHARGE_PUMP, 0x0 },
+};
+
+static bool nau8825_readable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case NAU8825_REG_ENA_CTRL:
+	case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV:
+	case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL:
+	case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL:
+	case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL:
+	case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY:
+	case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY:
+	case NAU8825_REG_IMM_MODE_CTRL ... NAU8825_REG_IMM_RMS_R:
+	case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL:
+	case NAU8825_REG_MISC_CTRL:
+	case NAU8825_REG_I2C_DEVICE_ID ... NAU8825_REG_SARDOUT_RAM_STATUS:
+	case NAU8825_REG_BIAS_ADJ:
+	case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2:
+	case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS:
+	case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA:
+	case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_GENERAL_STATUS:
+		return true;
+	default:
+		return false;
+	}
+
+}
+
+static bool nau8825_writeable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL:
+	case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV:
+	case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL:
+	case NAU8825_REG_INTERRUPT_MASK:
+	case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL:
+	case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL:
+	case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY:
+	case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY:
+	case NAU8825_REG_IMM_MODE_CTRL:
+	case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL:
+	case NAU8825_REG_MISC_CTRL:
+	case NAU8825_REG_BIAS_ADJ:
+	case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2:
+	case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS:
+	case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA:
+	case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_CHARGE_PUMP:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool nau8825_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case NAU8825_REG_RESET:
+	case NAU8825_REG_IRQ_STATUS:
+	case NAU8825_REG_INT_CLR_KEY_STATUS:
+	case NAU8825_REG_IMM_RMS_L:
+	case NAU8825_REG_IMM_RMS_R:
+	case NAU8825_REG_I2C_DEVICE_ID:
+	case NAU8825_REG_SARDOUT_RAM_STATUS:
+	case NAU8825_REG_CHARGE_PUMP_INPUT_READ:
+	case NAU8825_REG_GENERAL_STATUS:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static int nau8825_pump_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		/* Prevent startup click by letting charge pump to ramp up */
+		msleep(10);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static const char * const nau8825_adc_decimation[] = {
+	"32", "64", "128", "256"
+};
+
+static const struct soc_enum nau8825_adc_decimation_enum =
+	SOC_ENUM_SINGLE(NAU8825_REG_ADC_RATE, NAU8825_ADC_SYNC_DOWN_SFT,
+		ARRAY_SIZE(nau8825_adc_decimation), nau8825_adc_decimation);
+
+static const char * const nau8825_dac_oversampl[] = {
+	"64", "256", "128", "", "32"
+};
+
+static const struct soc_enum nau8825_dac_oversampl_enum =
+	SOC_ENUM_SINGLE(NAU8825_REG_DAC_CTRL1, NAU8825_DAC_OVERSAMPLE_SFT,
+		ARRAY_SIZE(nau8825_dac_oversampl), nau8825_dac_oversampl);
+
+static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -10300, 2400);
+static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0);
+static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -5400, 0);
+static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600);
+static const DECLARE_TLV_DB_MINMAX_MUTE(crosstalk_vol_tlv, -9600, 2400);
+
+static const struct snd_kcontrol_new nau8825_controls[] = {
+	SOC_SINGLE_TLV("Mic Volume", NAU8825_REG_ADC_DGAIN_CTRL,
+		0, 0xff, 0, adc_vol_tlv),
+	SOC_DOUBLE_TLV("Headphone Bypass Volume", NAU8825_REG_ADC_DGAIN_CTRL,
+		12, 8, 0x0f, 0, sidetone_vol_tlv),
+	SOC_DOUBLE_TLV("Headphone Volume", NAU8825_REG_HSVOL_CTRL,
+		6, 0, 0x3f, 1, dac_vol_tlv),
+	SOC_SINGLE_TLV("Frontend PGA Volume", NAU8825_REG_POWER_UP_CONTROL,
+		8, 37, 0, fepga_gain_tlv),
+	SOC_DOUBLE_TLV("Headphone Crosstalk Volume", NAU8825_REG_DAC_DGAIN_CTRL,
+		0, 8, 0xff, 0, crosstalk_vol_tlv),
+
+	SOC_ENUM("ADC Decimation Rate", nau8825_adc_decimation_enum),
+	SOC_ENUM("DAC Oversampling Rate", nau8825_dac_oversampl_enum),
+};
+
+/* DAC Mux 0x33[9] and 0x34[9] */
+static const char * const nau8825_dac_src[] = {
+	"DACL", "DACR",
+};
+
+static SOC_ENUM_SINGLE_DECL(
+	nau8825_dacl_enum, NAU8825_REG_DACL_CTRL,
+	NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src);
+
+static SOC_ENUM_SINGLE_DECL(
+	nau8825_dacr_enum, NAU8825_REG_DACR_CTRL,
+	NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src);
+
+static const struct snd_kcontrol_new nau8825_dacl_mux =
+	SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum);
+
+static const struct snd_kcontrol_new nau8825_dacr_mux =
+	SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum);
+
+
+static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8825_REG_I2S_PCM_CTRL2,
+		15, 1),
+
+	SND_SOC_DAPM_INPUT("MIC"),
+	SND_SOC_DAPM_MICBIAS("MICBIAS", NAU8825_REG_MIC_BIAS, 8, 0),
+
+	SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0,
+		NULL, 0),
+
+	SND_SOC_DAPM_ADC("ADC", NULL, NAU8825_REG_ENA_CTRL, 8, 0),
+	SND_SOC_DAPM_SUPPLY("ADC Clock", NAU8825_REG_ENA_CTRL, 7, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL,
+		0),
+
+	/* ADC for button press detection */
+	SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL,
+		NAU8825_SAR_ADC_EN_SFT, 0),
+
+	SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0),
+	SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0),
+	SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0),
+
+	SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL,
+		NAU8825_ENABLE_DACR_SFT, 0),
+	SND_SOC_DAPM_DAC("DDACL", NULL, NAU8825_REG_ENA_CTRL,
+		NAU8825_ENABLE_DACL_SFT, 0),
+	SND_SOC_DAPM_SUPPLY("DDAC Clock", NAU8825_REG_ENA_CTRL, 6, 0, NULL, 0),
+
+	SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacl_mux),
+	SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacr_mux),
+
+	SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL,
+		0),
+
+	SND_SOC_DAPM_SUPPLY("Charge Pump", NAU8825_REG_CHARGE_PUMP, 5, 0,
+		nau8825_pump_event, SND_SOC_DAPM_POST_PMU),
+
+	SND_SOC_DAPM_PGA("Output Driver R Stage 1",
+		NAU8825_REG_POWER_UP_CONTROL, 5, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Output Driver L Stage 1",
+		NAU8825_REG_POWER_UP_CONTROL, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Output Driver R Stage 2",
+		NAU8825_REG_POWER_UP_CONTROL, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Output Driver L Stage 2",
+		NAU8825_REG_POWER_UP_CONTROL, 2, 0, NULL, 0),
+	SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 1,
+		NAU8825_REG_POWER_UP_CONTROL, 1, 0, NULL, 0),
+	SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 1,
+		NAU8825_REG_POWER_UP_CONTROL, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA_S("Output DACL", 2, NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, 0),
+	SND_SOC_DAPM_PGA_S("Output DACR", 2, NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("HPOL"),
+	SND_SOC_DAPM_OUTPUT("HPOR"),
+};
+
+static const struct snd_soc_dapm_route nau8825_dapm_routes[] = {
+	{"Frontend PGA", NULL, "MIC"},
+	{"ADC", NULL, "Frontend PGA"},
+	{"ADC", NULL, "ADC Clock"},
+	{"ADC", NULL, "ADC Power"},
+	{"AIFTX", NULL, "ADC"},
+
+	{"DDACL", NULL, "Playback"},
+	{"DDACR", NULL, "Playback"},
+	{"DDACL", NULL, "DDAC Clock"},
+	{"DDACR", NULL, "DDAC Clock"},
+	{"DACL Mux", "DACL", "DDACL"},
+	{"DACL Mux", "DACR", "DDACR"},
+	{"DACR Mux", "DACL", "DDACL"},
+	{"DACR Mux", "DACR", "DDACR"},
+	{"HP amp L", NULL, "DACL Mux"},
+	{"HP amp R", NULL, "DACR Mux"},
+	{"HP amp L", NULL, "HP amp power"},
+	{"HP amp R", NULL, "HP amp power"},
+	{"ADACL", NULL, "HP amp L"},
+	{"ADACR", NULL, "HP amp R"},
+	{"ADACL", NULL, "ADACL Clock"},
+	{"ADACR", NULL, "ADACR Clock"},
+	{"Output Driver L Stage 1", NULL, "ADACL"},
+	{"Output Driver R Stage 1", NULL, "ADACR"},
+	{"Output Driver L Stage 2", NULL, "Output Driver L Stage 1"},
+	{"Output Driver R Stage 2", NULL, "Output Driver R Stage 1"},
+	{"Output Driver L Stage 3", NULL, "Output Driver L Stage 2"},
+	{"Output Driver R Stage 3", NULL, "Output Driver R Stage 2"},
+	{"Output DACL", NULL, "Output Driver L Stage 3"},
+	{"Output DACR", NULL, "Output Driver R Stage 3"},
+	{"HPOL", NULL, "Output DACL"},
+	{"HPOR", NULL, "Output DACR"},
+	{"HPOL", NULL, "Charge Pump"},
+	{"HPOR", NULL, "Charge Pump"},
+};
+
+static int nau8825_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 nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+	unsigned int val_len = 0;
+
+	switch (params_width(params)) {
+	case 16:
+		val_len |= NAU8825_I2S_DL_16;
+		break;
+	case 20:
+		val_len |= NAU8825_I2S_DL_20;
+		break;
+	case 24:
+		val_len |= NAU8825_I2S_DL_24;
+		break;
+	case 32:
+		val_len |= NAU8825_I2S_DL_32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1,
+		NAU8825_I2S_DL_MASK, val_len);
+
+	return 0;
+}
+
+static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+	unsigned int ctrl1_val = 0, ctrl2_val = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		ctrl2_val |= NAU8825_I2S_MS_MASTER;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		ctrl1_val |= NAU8825_I2S_BP_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		ctrl1_val |= NAU8825_I2S_DF_I2S;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		ctrl1_val |= NAU8825_I2S_DF_LEFT;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		ctrl1_val |= NAU8825_I2S_DF_RIGTH;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		ctrl1_val |= NAU8825_I2S_DF_PCM_AB;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		ctrl1_val |= NAU8825_I2S_DF_PCM_AB;
+		ctrl1_val |= NAU8825_I2S_PCMB_EN;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1,
+		NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK |
+		NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK,
+		ctrl1_val);
+	regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2,
+		NAU8825_I2S_MS_MASK, ctrl2_val);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops nau8825_dai_ops = {
+	.hw_params	= nau8825_hw_params,
+	.set_fmt	= nau8825_set_dai_fmt,
+};
+
+#define NAU8825_RATES	SNDRV_PCM_RATE_8000_192000
+#define NAU8825_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
+			 | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver nau8825_dai = {
+	.name = "nau8825-hifi",
+	.playback = {
+		.stream_name	 = "Playback",
+		.channels_min	 = 1,
+		.channels_max	 = 2,
+		.rates		 = NAU8825_RATES,
+		.formats	 = NAU8825_FORMATS,
+	},
+	.capture = {
+		.stream_name	 = "Capture",
+		.channels_min	 = 1,
+		.channels_max	 = 1,
+		.rates		 = NAU8825_RATES,
+		.formats	 = NAU8825_FORMATS,
+	},
+	.ops = &nau8825_dai_ops,
+};
+
+/**
+ * nau8825_enable_jack_detect - Specify a jack for event reporting
+ *
+ * @component:  component to register the jack with
+ * @jack: jack to use to report headset and button events on
+ *
+ * After this function has been called the headset insert/remove and button
+ * events will be routed to the given jack.  Jack can be null to stop
+ * reporting.
+ */
+int nau8825_enable_jack_detect(struct snd_soc_codec *codec,
+				struct snd_soc_jack *jack)
+{
+	struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+	struct regmap *regmap = nau8825->regmap;
+
+	nau8825->jack = jack;
+
+	/* Ground HP Outputs[1:0], needed for headset auto detection
+	 * Enable Automatic Mic/Gnd switching reading on insert interrupt[6]
+	 */
+	regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL,
+		NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L,
+		NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L);
+
+	regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK,
+		NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect);
+
+
+static bool nau8825_is_jack_inserted(struct regmap *regmap)
+{
+	int status;
+
+	regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status);
+	return !(status & NAU8825_GPIO2JD1);
+}
+
+static void nau8825_restart_jack_detection(struct regmap *regmap)
+{
+	/* this will restart the entire jack detection process including MIC/GND
+	 * switching and create interrupts. We have to go from 0 to 1 and back
+	 * to 0 to restart.
+	 */
+	regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL,
+		NAU8825_JACK_DET_RESTART, NAU8825_JACK_DET_RESTART);
+	regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL,
+		NAU8825_JACK_DET_RESTART, 0);
+}
+
+static void nau8825_eject_jack(struct nau8825 *nau8825)
+{
+	struct snd_soc_dapm_context *dapm = nau8825->dapm;
+	struct regmap *regmap = nau8825->regmap;
+
+	snd_soc_dapm_disable_pin(dapm, "SAR");
+	snd_soc_dapm_disable_pin(dapm, "MICBIAS");
+	/* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */
+	regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS,
+		NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0);
+	/* ground HPL/HPR, MICGRND1/2 */
+	regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf);
+
+	snd_soc_dapm_sync(dapm);
+}
+
+static int nau8825_button_decode(int value)
+{
+	int buttons = 0;
+
+	/* The chip supports up to 8 buttons, but ALSA defines only 6 buttons */
+	if (value & BIT(0))
+		buttons |= SND_JACK_BTN_0;
+	if (value & BIT(1))
+		buttons |= SND_JACK_BTN_1;
+	if (value & BIT(2))
+		buttons |= SND_JACK_BTN_2;
+	if (value & BIT(3))
+		buttons |= SND_JACK_BTN_3;
+	if (value & BIT(4))
+		buttons |= SND_JACK_BTN_4;
+	if (value & BIT(5))
+		buttons |= SND_JACK_BTN_5;
+
+	return buttons;
+}
+
+static int nau8825_jack_insert(struct nau8825 *nau8825)
+{
+	struct regmap *regmap = nau8825->regmap;
+	struct snd_soc_dapm_context *dapm = nau8825->dapm;
+	int jack_status_reg, mic_detected;
+	int type = 0;
+
+	regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg);
+	mic_detected = (jack_status_reg >> 10) & 3;
+
+	switch (mic_detected) {
+	case 0:
+		/* no mic */
+		type = SND_JACK_HEADPHONE;
+		break;
+	case 1:
+		dev_dbg(nau8825->dev, "OMTP (micgnd1) mic connected\n");
+		type = SND_JACK_HEADSET;
+
+		/* Unground MICGND1 */
+		regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2,
+			1 << 2);
+		/* Attach 2kOhm Resistor from MICBIAS to MICGND1 */
+		regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS,
+			NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2,
+			NAU8825_MICBIAS_JKR2);
+		/* Attach SARADC to MICGND1 */
+		regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL,
+			NAU8825_SAR_INPUT_MASK,
+			NAU8825_SAR_INPUT_JKR2);
+
+		snd_soc_dapm_force_enable_pin(dapm, "MICBIAS");
+		snd_soc_dapm_force_enable_pin(dapm, "SAR");
+		snd_soc_dapm_sync(dapm);
+		break;
+	case 2:
+	case 3:
+		dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n");
+		type = SND_JACK_HEADSET;
+
+		/* Unground MICGND2 */
+		regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2,
+			2 << 2);
+		/* Attach 2kOhm Resistor from MICBIAS to MICGND2 */
+		regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS,
+			NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2,
+			NAU8825_MICBIAS_JKSLV);
+		/* Attach SARADC to MICGND2 */
+		regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL,
+			NAU8825_SAR_INPUT_MASK,
+			NAU8825_SAR_INPUT_JKSLV);
+
+		snd_soc_dapm_force_enable_pin(dapm, "MICBIAS");
+		snd_soc_dapm_force_enable_pin(dapm, "SAR");
+		snd_soc_dapm_sync(dapm);
+		break;
+	}
+
+	if (type & SND_JACK_HEADPHONE) {
+		/* Unground HPL/R */
+		regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0);
+	}
+
+	return type;
+}
+
+#define NAU8825_BUTTONS (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \
+		SND_JACK_BTN_2 | SND_JACK_BTN_3)
+
+static irqreturn_t nau8825_interrupt(int irq, void *data)
+{
+	struct nau8825 *nau8825 = (struct nau8825 *)data;
+	struct regmap *regmap = nau8825->regmap;
+	int active_irq, clear_irq = 0, event = 0, event_mask = 0;
+
+	regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq);
+
+	if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) ==
+		NAU8825_JACK_EJECTION_DETECTED) {
+
+		nau8825_eject_jack(nau8825);
+		event_mask |= SND_JACK_HEADSET;
+		clear_irq = NAU8825_JACK_EJECTION_IRQ_MASK;
+	} else if (active_irq & NAU8825_KEY_SHORT_PRESS_IRQ) {
+		int key_status;
+
+		regmap_read(regmap, NAU8825_REG_INT_CLR_KEY_STATUS,
+			&key_status);
+
+		/* upper 8 bits of the register are for short pressed keys,
+		 * lower 8 bits - for long pressed buttons
+		 */
+		nau8825->button_pressed = nau8825_button_decode(
+			key_status >> 8);
+
+		event |= nau8825->button_pressed;
+		event_mask |= NAU8825_BUTTONS;
+		clear_irq = NAU8825_KEY_SHORT_PRESS_IRQ;
+	} else if (active_irq & NAU8825_KEY_RELEASE_IRQ) {
+		event_mask = NAU8825_BUTTONS;
+		clear_irq = NAU8825_KEY_RELEASE_IRQ;
+	} else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) {
+		if (nau8825_is_jack_inserted(regmap)) {
+			event |= nau8825_jack_insert(nau8825);
+		} else {
+			dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n");
+			nau8825_eject_jack(nau8825);
+		}
+
+		event_mask |= SND_JACK_HEADSET;
+		clear_irq = NAU8825_HEADSET_COMPLETION_IRQ;
+	}
+
+	if (!clear_irq)
+		clear_irq = active_irq;
+	/* clears the rightmost interruption */
+	regmap_write(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq);
+
+	if (event_mask)
+		snd_soc_jack_report(nau8825->jack, event, event_mask);
+
+	return IRQ_HANDLED;
+}
+
+static void nau8825_setup_buttons(struct nau8825 *nau8825)
+{
+	struct regmap *regmap = nau8825->regmap;
+
+	regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL,
+		NAU8825_SAR_TRACKING_GAIN_MASK,
+		nau8825->sar_voltage << NAU8825_SAR_TRACKING_GAIN_SFT);
+	regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL,
+		NAU8825_SAR_COMPARE_TIME_MASK,
+		nau8825->sar_compare_time << NAU8825_SAR_COMPARE_TIME_SFT);
+	regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL,
+		NAU8825_SAR_SAMPLING_TIME_MASK,
+		nau8825->sar_sampling_time << NAU8825_SAR_SAMPLING_TIME_SFT);
+
+	regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL,
+		NAU8825_KEYDET_LEVELS_NR_MASK,
+		(nau8825->sar_threshold_num - 1) << NAU8825_KEYDET_LEVELS_NR_SFT);
+	regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL,
+		NAU8825_KEYDET_HYSTERESIS_MASK,
+		nau8825->sar_hysteresis << NAU8825_KEYDET_HYSTERESIS_SFT);
+	regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL,
+		NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK,
+		nau8825->key_debounce << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT);
+
+	regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_1,
+		(nau8825->sar_threshold[0] << 8) | nau8825->sar_threshold[1]);
+	regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_2,
+		(nau8825->sar_threshold[2] << 8) | nau8825->sar_threshold[3]);
+	regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_3,
+		(nau8825->sar_threshold[4] << 8) | nau8825->sar_threshold[5]);
+	regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_4,
+		(nau8825->sar_threshold[6] << 8) | nau8825->sar_threshold[7]);
+
+	/* Enable short press and release interruptions */
+	regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK,
+		NAU8825_IRQ_KEY_SHORT_PRESS_EN | NAU8825_IRQ_KEY_RELEASE_EN,
+		0);
+}
+
+static void nau8825_init_regs(struct nau8825 *nau8825)
+{
+	struct regmap *regmap = nau8825->regmap;
+
+	/* Enable Bias/Vmid */
+	regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ,
+		NAU8825_BIAS_VMID, NAU8825_BIAS_VMID);
+	regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST,
+		NAU8825_GLOBAL_BIAS_EN, NAU8825_GLOBAL_BIAS_EN);
+
+	/* VMID Tieoff */
+	regmap_update_bits(regmap, NAU8825_REG_BIAS_ADJ,
+		NAU8825_BIAS_VMID_SEL_MASK,
+		nau8825->vref_impedance << NAU8825_BIAS_VMID_SEL_SFT);
+	/* Disable Boost Driver, Automatic Short circuit protection enable */
+	regmap_update_bits(regmap, NAU8825_REG_BOOST,
+		NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS |
+		NAU8825_SHORT_SHUTDOWN_EN,
+		NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS |
+		NAU8825_SHORT_SHUTDOWN_EN);
+
+	regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL,
+		NAU8825_JKDET_OUTPUT_EN,
+		nau8825->jkdet_enable ? 0 : NAU8825_JKDET_OUTPUT_EN);
+	regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL,
+		NAU8825_JKDET_PULL_EN,
+		nau8825->jkdet_pull_enable ? 0 : NAU8825_JKDET_PULL_EN);
+	regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL,
+		NAU8825_JKDET_PULL_UP,
+		nau8825->jkdet_pull_up ? NAU8825_JKDET_PULL_UP : 0);
+	regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL,
+		NAU8825_JACK_POLARITY,
+		/* jkdet_polarity - 1  is for active-low */
+		nau8825->jkdet_polarity ? 0 : NAU8825_JACK_POLARITY);
+
+	regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL,
+		NAU8825_JACK_INSERT_DEBOUNCE_MASK,
+		nau8825->jack_insert_debounce << NAU8825_JACK_INSERT_DEBOUNCE_SFT);
+	regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL,
+		NAU8825_JACK_EJECT_DEBOUNCE_MASK,
+		nau8825->jack_eject_debounce << NAU8825_JACK_EJECT_DEBOUNCE_SFT);
+
+	/* Mask unneeded IRQs: 1 - disable, 0 - enable */
+	regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, 0x7ff, 0x7ff);
+
+	regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS,
+		NAU8825_MICBIAS_VOLTAGE_MASK, nau8825->micbias_voltage);
+
+	if (nau8825->sar_threshold_num)
+		nau8825_setup_buttons(nau8825);
+
+	/* Default oversampling/decimations settings are unusable
+	 * (audible hiss). Set it to something better.
+	 */
+	regmap_update_bits(regmap, NAU8825_REG_ADC_RATE,
+		NAU8825_ADC_SYNC_DOWN_MASK, NAU8825_ADC_SYNC_DOWN_128);
+	regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1,
+		NAU8825_DAC_OVERSAMPLE_MASK, NAU8825_DAC_OVERSAMPLE_128);
+}
+
+static const struct regmap_config nau8825_regmap_config = {
+	.val_bits = 16,
+	.reg_bits = 16,
+
+	.max_register = NAU8825_REG_MAX,
+	.readable_reg = nau8825_readable_reg,
+	.writeable_reg = nau8825_writeable_reg,
+	.volatile_reg = nau8825_volatile_reg,
+
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = nau8825_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(nau8825_reg_defaults),
+};
+
+static int nau8825_codec_probe(struct snd_soc_codec *codec)
+{
+	struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+
+	nau8825->dapm = dapm;
+
+	/* The interrupt clock is gated by x1[10:8],
+	 * one of them needs to be enabled all the time for
+	 * interrupts to happen.
+	 */
+	snd_soc_dapm_force_enable_pin(dapm, "DDACR");
+	snd_soc_dapm_sync(dapm);
+
+	/* Unmask interruptions. Handler uses dapm object so we can enable
+	 * interruptions only after dapm is fully initialized.
+	 */
+	regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0);
+	nau8825_restart_jack_detection(nau8825->regmap);
+
+	return 0;
+}
+
+/**
+ * nau8825_calc_fll_param - Calculate FLL parameters.
+ * @fll_in: external clock provided to codec.
+ * @fs: sampling rate.
+ * @fll_param: Pointer to structure of FLL parameters.
+ *
+ * Calculate FLL parameters to configure codec.
+ *
+ * Returns 0 for success or negative error code.
+ */
+static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs,
+		struct nau8825_fll *fll_param)
+{
+	u64 fvco;
+	unsigned int fref, i;
+
+	/* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing
+	 * freq_in by 1, 2, 4, or 8 using FLL pre-scalar.
+	 * FREF = freq_in / NAU8825_FLL_REF_DIV_MASK
+	 */
+	for (i = 0; i < ARRAY_SIZE(fll_pre_scalar); i++) {
+		fref = fll_in / fll_pre_scalar[i].param;
+		if (fref <= NAU_FREF_MAX)
+			break;
+	}
+	if (i == ARRAY_SIZE(fll_pre_scalar))
+		return -EINVAL;
+	fll_param->clk_ref_div = fll_pre_scalar[i].val;
+
+	/* Choose the FLL ratio based on FREF */
+	for (i = 0; i < ARRAY_SIZE(fll_ratio); i++) {
+		if (fref >= fll_ratio[i].param)
+			break;
+	}
+	if (i == ARRAY_SIZE(fll_ratio))
+		return -EINVAL;
+	fll_param->ratio = fll_ratio[i].val;
+
+	/* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs.
+	 * FDCO must be within the 90MHz - 100MHz or the FFL cannot be
+	 * guaranteed across the full range of operation.
+	 * FDCO = freq_out * 2 * mclk_src_scaling
+	 */
+	for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) {
+		fvco = 256 * fs * 2 * mclk_src_scaling[i].param;
+		if (NAU_FVCO_MIN < fvco && fvco < NAU_FVCO_MAX)
+			break;
+	}
+	if (i == ARRAY_SIZE(mclk_src_scaling))
+		return -EINVAL;
+	fll_param->mclk_src = mclk_src_scaling[i].val;
+
+	/* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional
+	 * input based on FDCO, FREF and FLL ratio.
+	 */
+	fvco = div_u64(fvco << 16, fref * fll_param->ratio);
+	fll_param->fll_int = (fvco >> 16) & 0x3FF;
+	fll_param->fll_frac = fvco & 0xFFFF;
+	return 0;
+}
+
+static void nau8825_fll_apply(struct nau8825 *nau8825,
+		struct nau8825_fll *fll_param)
+{
+	regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER,
+		NAU8825_CLK_MCLK_SRC_MASK, fll_param->mclk_src);
+	regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL1,
+			NAU8825_FLL_RATIO_MASK, fll_param->ratio);
+	/* FLL 16-bit fractional input */
+	regmap_write(nau8825->regmap, NAU8825_REG_FLL2, fll_param->fll_frac);
+	/* FLL 10-bit integer input */
+	regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL3,
+			NAU8825_FLL_INTEGER_MASK, fll_param->fll_int);
+	/* FLL pre-scaler */
+	regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL4,
+			NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div);
+	/* select divided VCO input */
+	regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5,
+			NAU8825_FLL_FILTER_SW_MASK, 0x0000);
+	/* FLL sigma delta modulator enable */
+	regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6,
+			NAU8825_SDM_EN_MASK, NAU8825_SDM_EN);
+}
+
+/* freq_out must be 256*Fs in order to achieve the best performance */
+static int nau8825_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
+		unsigned int freq_in, unsigned int freq_out)
+{
+	struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+	struct nau8825_fll fll_param;
+	int ret, fs;
+
+	fs = freq_out / 256;
+	ret = nau8825_calc_fll_param(freq_in, fs, &fll_param);
+	if (ret < 0) {
+		dev_err(codec->dev, "Unsupported input clock %d\n", freq_in);
+		return ret;
+	}
+	dev_dbg(codec->dev, "mclk_src=%x ratio=%x fll_frac=%x fll_int=%x clk_ref_div=%x\n",
+		fll_param.mclk_src, fll_param.ratio, fll_param.fll_frac,
+		fll_param.fll_int, fll_param.clk_ref_div);
+
+	nau8825_fll_apply(nau8825, &fll_param);
+	mdelay(2);
+	regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER,
+			NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO);
+	return 0;
+}
+
+static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id,
+	unsigned int freq)
+{
+	struct regmap *regmap = nau8825->regmap;
+	int ret;
+
+	switch (clk_id) {
+	case NAU8825_CLK_MCLK:
+		regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER,
+			NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK);
+		regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0);
+
+		/* We selected MCLK source but the clock itself managed externally */
+		if (!nau8825->mclk)
+			break;
+
+		if (!nau8825->mclk_freq) {
+			ret = clk_prepare_enable(nau8825->mclk);
+			if (ret) {
+				dev_err(nau8825->dev, "Unable to prepare codec mclk\n");
+				return ret;
+			}
+		}
+
+		if (nau8825->mclk_freq != freq) {
+			nau8825->mclk_freq = freq;
+
+			freq = clk_round_rate(nau8825->mclk, freq);
+			ret = clk_set_rate(nau8825->mclk, freq);
+			if (ret) {
+				dev_err(nau8825->dev, "Unable to set mclk rate\n");
+				return ret;
+			}
+		}
+
+		break;
+	case NAU8825_CLK_INTERNAL:
+		regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN,
+			NAU8825_DCO_EN);
+		regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER,
+			NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO);
+
+		if (nau8825->mclk_freq) {
+			clk_disable_unprepare(nau8825->mclk);
+			nau8825->mclk_freq = 0;
+		}
+
+		break;
+	default:
+		dev_err(nau8825->dev, "Invalid clock id (%d)\n", clk_id);
+		return -EINVAL;
+	}
+
+	dev_dbg(nau8825->dev, "Sysclk is %dHz and clock id is %d\n", freq,
+		clk_id);
+	return 0;
+}
+
+static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+	int source, unsigned int freq, int dir)
+{
+	struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+
+	return nau8825_configure_sysclk(nau8825, clk_id, freq);
+}
+
+static int nau8825_set_bias_level(struct snd_soc_codec *codec,
+				   enum snd_soc_bias_level level)
+{
+	struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
+			if (nau8825->mclk_freq) {
+				ret = clk_prepare_enable(nau8825->mclk);
+				if (ret) {
+					dev_err(nau8825->dev, "Unable to prepare codec mclk\n");
+					return ret;
+				}
+			}
+
+			ret = regcache_sync(nau8825->regmap);
+			if (ret) {
+				dev_err(codec->dev,
+					"Failed to sync cache: %d\n", ret);
+				return ret;
+			}
+		}
+
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		if (nau8825->mclk_freq)
+			clk_disable_unprepare(nau8825->mclk);
+
+		regcache_mark_dirty(nau8825->regmap);
+		break;
+	}
+	return 0;
+}
+
+static struct snd_soc_codec_driver nau8825_codec_driver = {
+	.probe = nau8825_codec_probe,
+	.set_sysclk = nau8825_set_sysclk,
+	.set_pll = nau8825_set_pll,
+	.set_bias_level = nau8825_set_bias_level,
+	.suspend_bias_off = true,
+
+	.controls = nau8825_controls,
+	.num_controls = ARRAY_SIZE(nau8825_controls),
+	.dapm_widgets = nau8825_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets),
+	.dapm_routes = nau8825_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes),
+};
+
+static void nau8825_reset_chip(struct regmap *regmap)
+{
+	regmap_write(regmap, NAU8825_REG_RESET, 0x00);
+	regmap_write(regmap, NAU8825_REG_RESET, 0x00);
+}
+
+static void nau8825_print_device_properties(struct nau8825 *nau8825)
+{
+	int i;
+	struct device *dev = nau8825->dev;
+
+	dev_dbg(dev, "jkdet-enable:         %d\n", nau8825->jkdet_enable);
+	dev_dbg(dev, "jkdet-pull-enable:    %d\n", nau8825->jkdet_pull_enable);
+	dev_dbg(dev, "jkdet-pull-up:        %d\n", nau8825->jkdet_pull_up);
+	dev_dbg(dev, "jkdet-polarity:       %d\n", nau8825->jkdet_polarity);
+	dev_dbg(dev, "micbias-voltage:      %d\n", nau8825->micbias_voltage);
+	dev_dbg(dev, "vref-impedance:       %d\n", nau8825->vref_impedance);
+
+	dev_dbg(dev, "sar-threshold-num:    %d\n", nau8825->sar_threshold_num);
+	for (i = 0; i < nau8825->sar_threshold_num; i++)
+		dev_dbg(dev, "sar-threshold[%d]=%d\n", i,
+				nau8825->sar_threshold[i]);
+
+	dev_dbg(dev, "sar-hysteresis:       %d\n", nau8825->sar_hysteresis);
+	dev_dbg(dev, "sar-voltage:          %d\n", nau8825->sar_voltage);
+	dev_dbg(dev, "sar-compare-time:     %d\n", nau8825->sar_compare_time);
+	dev_dbg(dev, "sar-sampling-time:    %d\n", nau8825->sar_sampling_time);
+	dev_dbg(dev, "short-key-debounce:   %d\n", nau8825->key_debounce);
+	dev_dbg(dev, "jack-insert-debounce: %d\n",
+			nau8825->jack_insert_debounce);
+	dev_dbg(dev, "jack-eject-debounce:  %d\n",
+			nau8825->jack_eject_debounce);
+}
+
+static int nau8825_read_device_properties(struct device *dev,
+	struct nau8825 *nau8825) {
+
+	nau8825->jkdet_enable = device_property_read_bool(dev,
+		"nuvoton,jkdet-enable");
+	nau8825->jkdet_pull_enable = device_property_read_bool(dev,
+		"nuvoton,jkdet-pull-enable");
+	nau8825->jkdet_pull_up = device_property_read_bool(dev,
+		"nuvoton,jkdet-pull-up");
+	device_property_read_u32(dev, "nuvoton,jkdet-polarity",
+		&nau8825->jkdet_polarity);
+	device_property_read_u32(dev, "nuvoton,micbias-voltage",
+		&nau8825->micbias_voltage);
+	device_property_read_u32(dev, "nuvoton,vref-impedance",
+		&nau8825->vref_impedance);
+	device_property_read_u32(dev, "nuvoton,sar-threshold-num",
+		&nau8825->sar_threshold_num);
+	device_property_read_u32_array(dev, "nuvoton,sar-threshold",
+		nau8825->sar_threshold, nau8825->sar_threshold_num);
+	device_property_read_u32(dev, "nuvoton,sar-hysteresis",
+		&nau8825->sar_hysteresis);
+	device_property_read_u32(dev, "nuvoton,sar-voltage",
+		&nau8825->sar_voltage);
+	device_property_read_u32(dev, "nuvoton,sar-compare-time",
+		&nau8825->sar_compare_time);
+	device_property_read_u32(dev, "nuvoton,sar-sampling-time",
+		&nau8825->sar_sampling_time);
+	device_property_read_u32(dev, "nuvoton,short-key-debounce",
+		&nau8825->key_debounce);
+	device_property_read_u32(dev, "nuvoton,jack-insert-debounce",
+		&nau8825->jack_insert_debounce);
+	device_property_read_u32(dev, "nuvoton,jack-eject-debounce",
+		&nau8825->jack_eject_debounce);
+
+	nau8825->mclk = devm_clk_get(dev, "mclk");
+	if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) {
+		return -EPROBE_DEFER;
+	} else if (PTR_ERR(nau8825->mclk) == -ENOENT) {
+		/* The MCLK is managed externally or not used at all */
+		nau8825->mclk = NULL;
+		dev_info(dev, "No 'mclk' clock found, assume MCLK is managed externally");
+	} else if (IS_ERR(nau8825->mclk)) {
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int nau8825_setup_irq(struct nau8825 *nau8825)
+{
+	struct regmap *regmap = nau8825->regmap;
+	int ret;
+
+	/* IRQ Output Enable */
+	regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK,
+		NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN);
+
+	/* Enable internal VCO needed for interruptions */
+	nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0);
+
+	/* Enable DDACR needed for interrupts
+	 * It is the same as force_enable_pin("DDACR") we do later
+	 */
+	regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL,
+		NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR);
+
+	/* Chip needs one FSCLK cycle in order to generate interrupts,
+	 * as we cannot guarantee one will be provided by the system. Turning
+	 * master mode on then off enables us to generate that FSCLK cycle
+	 * with a minimum of contention on the clock bus.
+	 */
+	regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2,
+		NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER);
+	regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2,
+		NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE);
+
+	ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL,
+		nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT,
+		"nau8825", nau8825);
+
+	if (ret) {
+		dev_err(nau8825->dev, "Cannot request irq %d (%d)\n",
+			nau8825->irq, ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int nau8825_i2c_probe(struct i2c_client *i2c,
+	const struct i2c_device_id *id)
+{
+	struct device *dev = &i2c->dev;
+	struct nau8825 *nau8825 = dev_get_platdata(&i2c->dev);
+	int ret, value;
+
+	if (!nau8825) {
+		nau8825 = devm_kzalloc(dev, sizeof(*nau8825), GFP_KERNEL);
+		if (!nau8825)
+			return -ENOMEM;
+		ret = nau8825_read_device_properties(dev, nau8825);
+		if (ret)
+			return ret;
+	}
+
+	i2c_set_clientdata(i2c, nau8825);
+
+	nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap_config);
+	if (IS_ERR(nau8825->regmap))
+		return PTR_ERR(nau8825->regmap);
+	nau8825->dev = dev;
+	nau8825->irq = i2c->irq;
+
+	nau8825_print_device_properties(nau8825);
+
+	nau8825_reset_chip(nau8825->regmap);
+	ret = regmap_read(nau8825->regmap, NAU8825_REG_I2C_DEVICE_ID, &value);
+	if (ret < 0) {
+		dev_err(dev, "Failed to read device id from the NAU8825: %d\n",
+			ret);
+		return ret;
+	}
+	if ((value & NAU8825_SOFTWARE_ID_MASK) !=
+			NAU8825_SOFTWARE_ID_NAU8825) {
+		dev_err(dev, "Not a NAU8825 chip\n");
+		return -ENODEV;
+	}
+
+	nau8825_init_regs(nau8825);
+
+	if (i2c->irq)
+		nau8825_setup_irq(nau8825);
+
+	return snd_soc_register_codec(&i2c->dev, &nau8825_codec_driver,
+		&nau8825_dai, 1);
+}
+
+static int nau8825_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	return 0;
+}
+
+static const struct i2c_device_id nau8825_i2c_ids[] = {
+	{ "nau8825", 0 },
+	{ }
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id nau8825_of_ids[] = {
+	{ .compatible = "nuvoton,nau8825", },
+	{}
+};
+MODULE_DEVICE_TABLE(of, nau8825_of_ids);
+#endif
+
+#ifdef CONFIG_ACPI
+static const struct acpi_device_id nau8825_acpi_match[] = {
+	{ "10508825", 0 },
+	{},
+};
+MODULE_DEVICE_TABLE(acpi, nau8825_acpi_match);
+#endif
+
+static struct i2c_driver nau8825_driver = {
+	.driver = {
+		.name = "nau8825",
+		.of_match_table = of_match_ptr(nau8825_of_ids),
+		.acpi_match_table = ACPI_PTR(nau8825_acpi_match),
+	},
+	.probe = nau8825_i2c_probe,
+	.remove = nau8825_i2c_remove,
+	.id_table = nau8825_i2c_ids,
+};
+module_i2c_driver(nau8825_driver);
+
+MODULE_DESCRIPTION("ASoC nau8825 driver");
+MODULE_AUTHOR("Anatol Pomozov <anatol@chromium.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h
new file mode 100644
index 000000000000..dff8edb83bfd
--- /dev/null
+++ b/sound/soc/codecs/nau8825.h
@@ -0,0 +1,341 @@
+/*
+ * NAU8825 ALSA SoC audio driver
+ *
+ * Copyright 2015 Google Inc.
+ * Author: Anatol Pomozov <anatol.pomozov@chrominium.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __NAU8825_H__
+#define __NAU8825_H__
+
+#define NAU8825_REG_RESET		0x00
+#define NAU8825_REG_ENA_CTRL		0x01
+#define NAU8825_REG_CLK_DIVIDER		0x03
+#define NAU8825_REG_FLL1		0x04
+#define NAU8825_REG_FLL2		0x05
+#define NAU8825_REG_FLL3		0x06
+#define NAU8825_REG_FLL4		0x07
+#define NAU8825_REG_FLL5		0x08
+#define NAU8825_REG_FLL6		0x09
+#define NAU8825_REG_FLL_VCO_RSV		0x0a
+#define NAU8825_REG_HSD_CTRL		0x0c
+#define NAU8825_REG_JACK_DET_CTRL		0x0d
+#define NAU8825_REG_INTERRUPT_MASK		0x0f
+#define NAU8825_REG_IRQ_STATUS		0x10
+#define NAU8825_REG_INT_CLR_KEY_STATUS		0x11
+#define NAU8825_REG_INTERRUPT_DIS_CTRL		0x12
+#define NAU8825_REG_SAR_CTRL		0x13
+#define NAU8825_REG_KEYDET_CTRL		0x14
+#define NAU8825_REG_VDET_THRESHOLD_1		0x15
+#define NAU8825_REG_VDET_THRESHOLD_2		0x16
+#define NAU8825_REG_VDET_THRESHOLD_3		0x17
+#define NAU8825_REG_VDET_THRESHOLD_4		0x18
+#define NAU8825_REG_GPIO34_CTRL		0x19
+#define NAU8825_REG_GPIO12_CTRL		0x1a
+#define NAU8825_REG_TDM_CTRL		0x1b
+#define NAU8825_REG_I2S_PCM_CTRL1		0x1c
+#define NAU8825_REG_I2S_PCM_CTRL2		0x1d
+#define NAU8825_REG_LEFT_TIME_SLOT		0x1e
+#define NAU8825_REG_RIGHT_TIME_SLOT		0x1f
+#define NAU8825_REG_BIQ_CTRL		0x20
+#define NAU8825_REG_BIQ_COF1		0x21
+#define NAU8825_REG_BIQ_COF2		0x22
+#define NAU8825_REG_BIQ_COF3		0x23
+#define NAU8825_REG_BIQ_COF4		0x24
+#define NAU8825_REG_BIQ_COF5		0x25
+#define NAU8825_REG_BIQ_COF6		0x26
+#define NAU8825_REG_BIQ_COF7		0x27
+#define NAU8825_REG_BIQ_COF8		0x28
+#define NAU8825_REG_BIQ_COF9		0x29
+#define NAU8825_REG_BIQ_COF10		0x2a
+#define NAU8825_REG_ADC_RATE		0x2b
+#define NAU8825_REG_DAC_CTRL1		0x2c
+#define NAU8825_REG_DAC_CTRL2		0x2d
+#define NAU8825_REG_DAC_DGAIN_CTRL		0x2f
+#define NAU8825_REG_ADC_DGAIN_CTRL		0x30
+#define NAU8825_REG_MUTE_CTRL		0x31
+#define NAU8825_REG_HSVOL_CTRL		0x32
+#define NAU8825_REG_DACL_CTRL		0x33
+#define NAU8825_REG_DACR_CTRL		0x34
+#define NAU8825_REG_ADC_DRC_KNEE_IP12		0x38
+#define NAU8825_REG_ADC_DRC_KNEE_IP34		0x39
+#define NAU8825_REG_ADC_DRC_SLOPES		0x3a
+#define NAU8825_REG_ADC_DRC_ATKDCY		0x3b
+#define NAU8825_REG_DAC_DRC_KNEE_IP12		0x45
+#define NAU8825_REG_DAC_DRC_KNEE_IP34		0x46
+#define NAU8825_REG_DAC_DRC_SLOPES		0x47
+#define NAU8825_REG_DAC_DRC_ATKDCY		0x48
+#define NAU8825_REG_IMM_MODE_CTRL		0x4c
+#define NAU8825_REG_IMM_RMS_L		0x4d
+#define NAU8825_REG_IMM_RMS_R		0x4e
+#define NAU8825_REG_CLASSG_CTRL		0x50
+#define NAU8825_REG_OPT_EFUSE_CTRL		0x51
+#define NAU8825_REG_MISC_CTRL		0x55
+#define NAU8825_REG_I2C_DEVICE_ID		0x58
+#define NAU8825_REG_SARDOUT_RAM_STATUS		0x59
+#define NAU8825_REG_BIAS_ADJ		0x66
+#define NAU8825_REG_TRIM_SETTINGS		0x68
+#define NAU8825_REG_ANALOG_CONTROL_1		0x69
+#define NAU8825_REG_ANALOG_CONTROL_2		0x6a
+#define NAU8825_REG_ANALOG_ADC_1		0x71
+#define NAU8825_REG_ANALOG_ADC_2		0x72
+#define NAU8825_REG_RDAC		0x73
+#define NAU8825_REG_MIC_BIAS		0x74
+#define NAU8825_REG_BOOST		0x76
+#define NAU8825_REG_FEPGA		0x77
+#define NAU8825_REG_POWER_UP_CONTROL		0x7f
+#define NAU8825_REG_CHARGE_PUMP		0x80
+#define NAU8825_REG_CHARGE_PUMP_INPUT_READ		0x81
+#define NAU8825_REG_GENERAL_STATUS		0x82
+#define NAU8825_REG_MAX		NAU8825_REG_GENERAL_STATUS
+
+/* ENA_CTRL (0x1) */
+#define NAU8825_ENABLE_DACR_SFT	10
+#define NAU8825_ENABLE_DACR	(1 << NAU8825_ENABLE_DACR_SFT)
+#define NAU8825_ENABLE_DACL_SFT	9
+#define NAU8825_ENABLE_ADC_SFT	8
+#define NAU8825_ENABLE_SAR_SFT	1
+
+/* CLK_DIVIDER (0x3) */
+#define NAU8825_CLK_SRC_SFT			15
+#define NAU8825_CLK_SRC_MASK			(1 << NAU8825_CLK_SRC_SFT)
+#define NAU8825_CLK_SRC_VCO			(1 << NAU8825_CLK_SRC_SFT)
+#define NAU8825_CLK_SRC_MCLK			(0 << NAU8825_CLK_SRC_SFT)
+#define NAU8825_CLK_MCLK_SRC_MASK		(0xf << 0)
+
+/* FLL1 (0x04) */
+#define NAU8825_FLL_RATIO_MASK			(0x7f << 0)
+
+/* FLL3 (0x06) */
+#define NAU8825_FLL_INTEGER_MASK		(0x3ff << 0)
+
+/* FLL4 (0x07) */
+#define NAU8825_FLL_REF_DIV_MASK		(0x3 << 10)
+
+/* FLL5 (0x08) */
+#define NAU8825_FLL_FILTER_SW_MASK		(0x1 << 14)
+
+/* FLL6 (0x9) */
+#define NAU8825_DCO_EN_MASK			(0x1 << 15)
+#define NAU8825_DCO_EN				(0x1 << 15)
+#define NAU8825_DCO_DIS				(0x0 << 15)
+#define NAU8825_SDM_EN_MASK			(0x1 << 14)
+#define NAU8825_SDM_EN				(0x1 << 14)
+#define NAU8825_SDM_DIS				(0x0 << 14)
+
+/* HSD_CTRL (0xc) */
+#define NAU8825_HSD_AUTO_MODE	(1 << 6)
+/* 0 - short to GND, 1 - open */
+#define NAU8825_SPKR_DWN1R	(1 << 1)
+#define NAU8825_SPKR_DWN1L	(1 << 0)
+
+/* JACK_DET_CTRL (0xd) */
+#define NAU8825_JACK_DET_RESTART	(1 << 9)
+#define NAU8825_JACK_INSERT_DEBOUNCE_SFT	5
+#define NAU8825_JACK_INSERT_DEBOUNCE_MASK	(0x7 << NAU8825_JACK_INSERT_DEBOUNCE_SFT)
+#define NAU8825_JACK_EJECT_DEBOUNCE_SFT		2
+#define NAU8825_JACK_EJECT_DEBOUNCE_MASK	(0x7 << NAU8825_JACK_EJECT_DEBOUNCE_SFT)
+#define NAU8825_JACK_POLARITY	(1 << 1) /* 0 - active low, 1 - active high */
+
+/* INTERRUPT_MASK (0xf) */
+#define NAU8825_IRQ_OUTPUT_EN (1 << 11)
+#define NAU8825_IRQ_HEADSET_COMPLETE_EN (1 << 10)
+#define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7)
+#define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5)
+#define NAU8825_IRQ_EJECT_EN (1 << 2)
+
+/* IRQ_STATUS (0x10) */
+#define NAU8825_HEADSET_COMPLETION_IRQ	(1 << 10)
+#define NAU8825_SHORT_CIRCUIT_IRQ	(1 << 9)
+#define NAU8825_IMPEDANCE_MEAS_IRQ	(1 << 8)
+#define NAU8825_KEY_IRQ_MASK	(0x7 << 5)
+#define NAU8825_KEY_RELEASE_IRQ	(1 << 7)
+#define NAU8825_KEY_LONG_PRESS_IRQ	(1 << 6)
+#define NAU8825_KEY_SHORT_PRESS_IRQ	(1 << 5)
+#define NAU8825_MIC_DETECTION_IRQ	(1 << 4)
+#define NAU8825_JACK_EJECTION_IRQ_MASK	(3 << 2)
+#define NAU8825_JACK_EJECTION_DETECTED	(1 << 2)
+#define NAU8825_JACK_INSERTION_IRQ_MASK	(3 << 0)
+#define NAU8825_JACK_INSERTION_DETECTED	(1 << 0)
+
+/* INTERRUPT_DIS_CTRL (0x12) */
+#define NAU8825_IRQ_HEADSET_COMPLETE_DIS (1 << 10)
+#define NAU8825_IRQ_KEY_RELEASE_DIS (1 << 7)
+#define NAU8825_IRQ_KEY_SHORT_PRESS_DIS (1 << 5)
+#define NAU8825_IRQ_EJECT_DIS (1 << 2)
+
+/* SAR_CTRL (0x13) */
+#define NAU8825_SAR_ADC_EN_SFT	12
+#define NAU8825_SAR_ADC_EN	(1 << NAU8825_SAR_ADC_EN_SFT)
+#define NAU8825_SAR_INPUT_MASK	(1 << 11)
+#define NAU8825_SAR_INPUT_JKSLV	(1 << 11)
+#define NAU8825_SAR_INPUT_JKR2	(0 << 11)
+#define NAU8825_SAR_TRACKING_GAIN_SFT	8
+#define NAU8825_SAR_TRACKING_GAIN_MASK	(0x7 << NAU8825_SAR_TRACKING_GAIN_SFT)
+#define NAU8825_SAR_COMPARE_TIME_SFT	2
+#define NAU8825_SAR_COMPARE_TIME_MASK	(3 << 2)
+#define NAU8825_SAR_SAMPLING_TIME_SFT	0
+#define NAU8825_SAR_SAMPLING_TIME_MASK	(3 << 0)
+
+/* KEYDET_CTRL (0x14) */
+#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT	12
+#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK	(0x3 << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT)
+#define NAU8825_KEYDET_LEVELS_NR_SFT	8
+#define NAU8825_KEYDET_LEVELS_NR_MASK	(0x7 << 8)
+#define NAU8825_KEYDET_HYSTERESIS_SFT	0
+#define NAU8825_KEYDET_HYSTERESIS_MASK	0xf
+
+/* GPIO12_CTRL (0x1a) */
+#define NAU8825_JKDET_PULL_UP	(1 << 11) /* 0 - pull down, 1 - pull up */
+#define NAU8825_JKDET_PULL_EN	(1 << 9) /* 0 - enable pull, 1 - disable */
+#define NAU8825_JKDET_OUTPUT_EN	(1 << 8) /* 0 - enable input, 1 - enable output */
+
+/* I2S_PCM_CTRL1 (0x1c) */
+#define NAU8825_I2S_BP_SFT	7
+#define NAU8825_I2S_BP_MASK	(1 << NAU8825_I2S_BP_SFT)
+#define NAU8825_I2S_BP_INV	(1 << NAU8825_I2S_BP_SFT)
+#define NAU8825_I2S_PCMB_SFT	6
+#define NAU8825_I2S_PCMB_MASK	(1 << NAU8825_I2S_PCMB_SFT)
+#define NAU8825_I2S_PCMB_EN	(1 << NAU8825_I2S_PCMB_SFT)
+#define NAU8825_I2S_DL_SFT	2
+#define NAU8825_I2S_DL_MASK	(0x3 << NAU8825_I2S_DL_SFT)
+#define NAU8825_I2S_DL_16	(0 << NAU8825_I2S_DL_SFT)
+#define NAU8825_I2S_DL_20	(1 << NAU8825_I2S_DL_SFT)
+#define NAU8825_I2S_DL_24	(2 << NAU8825_I2S_DL_SFT)
+#define NAU8825_I2S_DL_32	(3 << NAU8825_I2S_DL_SFT)
+#define NAU8825_I2S_DF_SFT	0
+#define NAU8825_I2S_DF_MASK	(0x3 << NAU8825_I2S_DF_SFT)
+#define NAU8825_I2S_DF_RIGTH	(0 << NAU8825_I2S_DF_SFT)
+#define NAU8825_I2S_DF_LEFT	(1 << NAU8825_I2S_DF_SFT)
+#define NAU8825_I2S_DF_I2S	(2 << NAU8825_I2S_DF_SFT)
+#define NAU8825_I2S_DF_PCM_AB	(3 << NAU8825_I2S_DF_SFT)
+
+/* I2S_PCM_CTRL2 (0x1d) */
+#define NAU8825_I2S_TRISTATE	(1 << 15) /* 0 - normal mode, 1 - Hi-Z output */
+#define NAU8825_I2S_MS_SFT	3
+#define NAU8825_I2S_MS_MASK	(1 << NAU8825_I2S_MS_SFT)
+#define NAU8825_I2S_MS_MASTER	(1 << NAU8825_I2S_MS_SFT)
+#define NAU8825_I2S_MS_SLAVE	(0 << NAU8825_I2S_MS_SFT)
+
+/* ADC_RATE (0x2b) */
+#define NAU8825_ADC_SYNC_DOWN_SFT	0
+#define NAU8825_ADC_SYNC_DOWN_MASK	0x3
+#define NAU8825_ADC_SYNC_DOWN_32	0
+#define NAU8825_ADC_SYNC_DOWN_64	1
+#define NAU8825_ADC_SYNC_DOWN_128	2
+#define NAU8825_ADC_SYNC_DOWN_256	3
+
+/* DAC_CTRL1 (0x2c) */
+#define NAU8825_DAC_CLIP_OFF	(1 << 7)
+#define NAU8825_DAC_OVERSAMPLE_SFT	0
+#define NAU8825_DAC_OVERSAMPLE_MASK	0x7
+#define NAU8825_DAC_OVERSAMPLE_64	0
+#define NAU8825_DAC_OVERSAMPLE_256	1
+#define NAU8825_DAC_OVERSAMPLE_128	2
+#define NAU8825_DAC_OVERSAMPLE_32	4
+
+/* MUTE_CTRL (0x31) */
+#define NAU8825_DAC_ZERO_CROSSING_EN	(1 << 9)
+#define NAU8825_DAC_SOFT_MUTE	(1 << 9)
+
+/* HSVOL_CTRL (0x32) */
+#define NAU8825_HP_MUTE	(1 << 15)
+
+/* DACL_CTRL (0x33) */
+#define NAU8825_DACL_CH_SEL_SFT	9
+
+/* DACR_CTRL (0x34) */
+#define NAU8825_DACR_CH_SEL_SFT	9
+
+/* I2C_DEVICE_ID (0x58) */
+#define NAU8825_GPIO2JD1	(1 << 7)
+#define NAU8825_SOFTWARE_ID_MASK	0x3
+#define NAU8825_SOFTWARE_ID_NAU8825	0x0
+
+/* BIAS_ADJ (0x66) */
+#define NAU8825_BIAS_VMID	(1 << 6)
+#define NAU8825_BIAS_VMID_SEL_SFT	4
+#define NAU8825_BIAS_VMID_SEL_MASK	(3 << NAU8825_BIAS_VMID_SEL_SFT)
+
+/* ANALOG_CONTROL_2 (0x6a) */
+#define NAU8825_HP_NON_CLASSG_CURRENT_2xADJ (1 << 12)
+#define NAU8825_DAC_CAPACITOR_MSB (1 << 1)
+#define NAU8825_DAC_CAPACITOR_LSB (1 << 0)
+
+/* ANALOG_ADC_2 (0x72) */
+#define NAU8825_ADC_VREFSEL_MASK	(0x3 << 8)
+#define NAU8825_ADC_VREFSEL_ANALOG	(0 << 8)
+#define NAU8825_ADC_VREFSEL_VMID	(1 << 8)
+#define NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB	(2 << 8)
+#define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB	(3 << 8)
+#define NAU8825_POWERUP_ADCL	(1 << 6)
+
+/* MIC_BIAS (0x74) */
+#define NAU8825_MICBIAS_JKSLV	(1 << 14)
+#define NAU8825_MICBIAS_JKR2	(1 << 12)
+#define NAU8825_MICBIAS_POWERUP_SFT	8
+#define NAU8825_MICBIAS_VOLTAGE_SFT	0
+#define NAU8825_MICBIAS_VOLTAGE_MASK	0x7
+
+/* BOOST (0x76) */
+#define NAU8825_PRECHARGE_DIS	(1 << 13)
+#define NAU8825_GLOBAL_BIAS_EN	(1 << 12)
+#define NAU8825_HP_BOOST_G_DIS	(1 << 8)
+#define NAU8825_SHORT_SHUTDOWN_EN	(1 << 6)
+
+/* POWER_UP_CONTROL (0x7f) */
+#define NAU8825_POWERUP_INTEGR_R	(1 << 5)
+#define NAU8825_POWERUP_INTEGR_L	(1 << 4)
+#define NAU8825_POWERUP_DRV_IN_R	(1 << 3)
+#define NAU8825_POWERUP_DRV_IN_L	(1 << 2)
+#define NAU8825_POWERUP_HP_DRV_R	(1 << 1)
+#define NAU8825_POWERUP_HP_DRV_L	(1 << 0)
+
+/* CHARGE_PUMP (0x80) */
+#define NAU8825_JAMNODCLOW	(1 << 10)
+#define NAU8825_POWER_DOWN_DACR	(1 << 9)
+#define NAU8825_POWER_DOWN_DACL	(1 << 8)
+#define NAU8825_CHANRGE_PUMP_EN	(1 << 5)
+
+
+/* System Clock Source */
+enum {
+	NAU8825_CLK_MCLK = 0,
+	NAU8825_CLK_INTERNAL,
+};
+
+struct nau8825 {
+	struct device *dev;
+	struct regmap *regmap;
+	struct snd_soc_dapm_context *dapm;
+	struct snd_soc_jack *jack;
+	struct clk *mclk;
+	int irq;
+	int mclk_freq; /* 0 - mclk is disabled */
+	int button_pressed;
+	int micbias_voltage;
+	int vref_impedance;
+	bool jkdet_enable;
+	bool jkdet_pull_enable;
+	bool jkdet_pull_up;
+	int jkdet_polarity;
+	int sar_threshold_num;
+	int sar_threshold[8];
+	int sar_hysteresis;
+	int sar_voltage;
+	int sar_compare_time;
+	int sar_sampling_time;
+	int key_debounce;
+	int jack_insert_debounce;
+	int jack_eject_debounce;
+};
+
+int nau8825_enable_jack_detect(struct snd_soc_codec *codec,
+				struct snd_soc_jack *jack);
+
+
+#endif  /* __NAU8825_H__ */
diff --git a/sound/soc/codecs/rl6347a.c b/sound/soc/codecs/rl6347a.c
index 91d5166bd3a1..a4b910efbd45 100644
--- a/sound/soc/codecs/rl6347a.c
+++ b/sound/soc/codecs/rl6347a.c
@@ -11,25 +11,8 @@
  */
 
 #include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/init.h>
-#include <linux/delay.h>
-#include <linux/pm.h>
 #include <linux/i2c.h>
-#include <linux/platform_device.h>
-#include <linux/spi/spi.h>
-#include <linux/dmi.h>
-#include <linux/acpi.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-#include <sound/soc-dapm.h>
-#include <sound/initval.h>
-#include <sound/tlv.h>
-#include <sound/jack.h>
-#include <linux/workqueue.h>
-#include <sound/hda_verbs.h>
+#include <linux/regmap.h>
 
 #include "rl6347a.h"
 
diff --git a/sound/soc/codecs/rl6347a.h b/sound/soc/codecs/rl6347a.h
index 1cb56e50b7f3..e127919cb36b 100644
--- a/sound/soc/codecs/rl6347a.h
+++ b/sound/soc/codecs/rl6347a.h
@@ -12,6 +12,8 @@
 #ifndef __RL6347A_H__
 #define __RL6347A_H__
 
+#include <sound/hda_verbs.h>
+
 #define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D)
 
 #define RL6347A_VENDOR_REGISTERS	0x20
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index bd9365885f73..af2ed774b552 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -29,7 +29,6 @@
 #include <sound/jack.h>
 #include <linux/workqueue.h>
 #include <sound/rt286.h>
-#include <sound/hda_verbs.h>
 
 #include "rl6347a.h"
 #include "rt286.h"
@@ -38,7 +37,7 @@
 #define RT288_VENDOR_ID 0x10ec0288
 
 struct rt286_priv {
-	const struct reg_default *index_cache;
+	struct reg_default *index_cache;
 	int index_cache_size;
 	struct regmap *regmap;
 	struct snd_soc_codec *codec;
@@ -1161,7 +1160,11 @@ static int rt286_i2c_probe(struct i2c_client *i2c,
 		return -ENODEV;
 	}
 
-	rt286->index_cache = rt286_index_def;
+	rt286->index_cache = devm_kmemdup(&i2c->dev, rt286_index_def,
+					  sizeof(rt286_index_def), GFP_KERNEL);
+	if (!rt286->index_cache)
+		return -ENOMEM;
+
 	rt286->index_cache_size = INDEX_CACHE_SIZE;
 	rt286->i2c = i2c;
 	i2c_set_clientdata(i2c, rt286);
diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c
index f823eb502367..b3f795c60749 100644
--- a/sound/soc/codecs/rt298.c
+++ b/sound/soc/codecs/rt298.c
@@ -28,7 +28,6 @@
 #include <sound/jack.h>
 #include <linux/workqueue.h>
 #include <sound/rt298.h>
-#include <sound/hda_verbs.h>
 
 #include "rl6347a.h"
 #include "rt298.h"
@@ -49,7 +48,7 @@ struct rt298_priv {
 	int is_hp_in;
 };
 
-static struct reg_default rt298_index_def[] = {
+static const struct reg_default rt298_index_def[] = {
 	{ 0x01, 0xa5a8 },
 	{ 0x02, 0x8e95 },
 	{ 0x03, 0x0002 },
@@ -129,7 +128,7 @@ static bool rt298_volatile_register(struct device *dev, unsigned int reg)
 	case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_HP_OUT, 0):
 		return true;
 	default:
-		return true;
+		return false;
 	}
 
 
@@ -1165,7 +1164,11 @@ static int rt298_i2c_probe(struct i2c_client *i2c,
 		return -ENODEV;
 	}
 
-	rt298->index_cache = rt298_index_def;
+	rt298->index_cache = devm_kmemdup(&i2c->dev, rt298_index_def,
+					  sizeof(rt298_index_def), GFP_KERNEL);
+	if (!rt298->index_cache)
+		return -ENOMEM;
+
 	rt298->index_cache_size = INDEX_CACHE_SIZE;
 	rt298->i2c = i2c;
 	i2c_set_clientdata(i2c, rt298);
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index e1ceeb885f7d..f2beb1aa5763 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -405,11 +405,14 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = {
 	SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL,
 			RT5640_L_VOL_SFT, RT5640_R_VOL_SFT,
 			175, 0, dac_vol_tlv),
-	/* IN1/IN2 Control */
+	/* IN1/IN2/IN3 Control */
 	SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2,
 		RT5640_BST_SFT1, 8, 0, bst_tlv),
 	SOC_SINGLE_TLV("IN2 Boost", RT5640_IN3_IN4,
 		RT5640_BST_SFT2, 8, 0, bst_tlv),
+	SOC_SINGLE_TLV("IN3 Boost", RT5640_IN1_IN2,
+		RT5640_BST_SFT2, 8, 0, bst_tlv),
+
 	/* INL/INR Volume Control */
 	SOC_DOUBLE_TLV("IN Capture Volume", RT5640_INL_INR_VOL,
 			RT5640_INL_VOL_SFT, RT5640_INR_VOL_SFT,
@@ -598,6 +601,8 @@ static const struct snd_kcontrol_new rt5640_rec_l_mix[] = {
 			RT5640_M_HP_L_RM_L_SFT, 1, 1),
 	SOC_DAPM_SINGLE("INL Switch", RT5640_REC_L2_MIXER,
 			RT5640_M_IN_L_RM_L_SFT, 1, 1),
+	SOC_DAPM_SINGLE("BST3 Switch", RT5640_REC_L2_MIXER,
+			RT5640_M_BST2_RM_L_SFT, 1, 1),
 	SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_L2_MIXER,
 			RT5640_M_BST4_RM_L_SFT, 1, 1),
 	SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_L2_MIXER,
@@ -611,6 +616,8 @@ static const struct snd_kcontrol_new rt5640_rec_r_mix[] = {
 			RT5640_M_HP_R_RM_R_SFT, 1, 1),
 	SOC_DAPM_SINGLE("INR Switch", RT5640_REC_R2_MIXER,
 			RT5640_M_IN_R_RM_R_SFT, 1, 1),
+	SOC_DAPM_SINGLE("BST3 Switch", RT5640_REC_R2_MIXER,
+			RT5640_M_BST2_RM_R_SFT, 1, 1),
 	SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_R2_MIXER,
 			RT5640_M_BST4_RM_R_SFT, 1, 1),
 	SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_R2_MIXER,
@@ -1065,6 +1072,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("IN1N"),
 	SND_SOC_DAPM_INPUT("IN2P"),
 	SND_SOC_DAPM_INPUT("IN2N"),
+	SND_SOC_DAPM_INPUT("IN3P"),
+	SND_SOC_DAPM_INPUT("IN3N"),
 	SND_SOC_DAPM_PGA("DMIC L1", SND_SOC_NOPM, 0, 0, NULL, 0),
 	SND_SOC_DAPM_PGA("DMIC R1", SND_SOC_NOPM, 0, 0, NULL, 0),
 	SND_SOC_DAPM_PGA("DMIC L2", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -1081,6 +1090,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
 		RT5640_PWR_BST1_BIT, 0, NULL, 0),
 	SND_SOC_DAPM_PGA("BST2", RT5640_PWR_ANLG2,
 		RT5640_PWR_BST4_BIT, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("BST3", RT5640_PWR_ANLG2,
+		RT5640_PWR_BST2_BIT, 0, NULL, 0),
 	/* Input Volume */
 	SND_SOC_DAPM_PGA("INL VOL", RT5640_PWR_VOL,
 		RT5640_PWR_IN_L_BIT, 0, NULL, 0),
@@ -1310,6 +1321,7 @@ static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = {
 static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
 	{"IN1P", NULL, "LDO2"},
 	{"IN2P", NULL, "LDO2"},
+	{"IN3P", NULL, "LDO2"},
 
 	{"DMIC L1", NULL, "DMIC1"},
 	{"DMIC R1", NULL, "DMIC1"},
@@ -1320,18 +1332,22 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
 	{"BST1", NULL, "IN1N"},
 	{"BST2", NULL, "IN2P"},
 	{"BST2", NULL, "IN2N"},
+	{"BST3", NULL, "IN3P"},
+	{"BST3", NULL, "IN3N"},
 
 	{"INL VOL", NULL, "IN2P"},
 	{"INR VOL", NULL, "IN2N"},
 
 	{"RECMIXL", "HPOL Switch", "HPOL"},
 	{"RECMIXL", "INL Switch", "INL VOL"},
+	{"RECMIXL", "BST3 Switch", "BST3"},
 	{"RECMIXL", "BST2 Switch", "BST2"},
 	{"RECMIXL", "BST1 Switch", "BST1"},
 	{"RECMIXL", "OUT MIXL Switch", "OUT MIXL"},
 
 	{"RECMIXR", "HPOR Switch", "HPOR"},
 	{"RECMIXR", "INR Switch", "INR VOL"},
+	{"RECMIXR", "BST3 Switch", "BST3"},
 	{"RECMIXR", "BST2 Switch", "BST2"},
 	{"RECMIXR", "BST1 Switch", "BST1"},
 	{"RECMIXR", "OUT MIXR Switch", "OUT MIXR"},
@@ -2260,6 +2276,10 @@ static int rt5640_i2c_probe(struct i2c_client *i2c,
 		regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4,
 					RT5640_IN_DF2, RT5640_IN_DF2);
 
+	if (rt5640->pdata.in3_diff)
+		regmap_update_bits(rt5640->regmap, RT5640_IN1_IN2,
+					RT5640_IN_DF2, RT5640_IN_DF2);
+
 	rt5640->hp_mute = 1;
 
 	return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640,
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 5c101af0ac63..28132375e427 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -42,6 +42,8 @@
 
 #define RT5645_PR_BASE (RT5645_PR_RANGE_BASE + (0 * RT5645_PR_SPACING))
 
+#define RT5645_HWEQ_NUM 57
+
 static const struct regmap_range_cfg rt5645_ranges[] = {
 	{
 		.name = "PR",
@@ -224,6 +226,11 @@ static const struct reg_default rt5645_reg[] = {
 	{ 0xff, 0x6308 },
 };
 
+struct rt5645_eq_param_s {
+	unsigned short reg;
+	unsigned short val;
+};
+
 static const char *const rt5645_supply_names[] = {
 	"avdd",
 	"cpvdd",
@@ -240,6 +247,7 @@ struct rt5645_priv {
 	struct snd_soc_jack *btn_jack;
 	struct delayed_work jack_detect_work;
 	struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)];
+	struct rt5645_eq_param_s *eq_param;
 
 	int codec_type;
 	int sysclk;
@@ -469,6 +477,94 @@ static const DECLARE_TLV_DB_RANGE(bst_tlv,
 	8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0)
 );
 
+/* {-6, -4.5, -3, -1.5, 0, 0.82, 1.58, 2.28} dB */
+static const DECLARE_TLV_DB_RANGE(spk_clsd_tlv,
+	0, 4, TLV_DB_SCALE_ITEM(-600, 150, 0),
+	5, 5, TLV_DB_SCALE_ITEM(82, 0, 0),
+	6, 6, TLV_DB_SCALE_ITEM(158, 0, 0),
+	7, 7, TLV_DB_SCALE_ITEM(228, 0, 0)
+);
+
+static int rt5645_hweq_info(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+	uinfo->count = RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s);
+
+	return 0;
+}
+
+static int rt5645_hweq_get(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+	struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component);
+	struct rt5645_eq_param_s *eq_param =
+		(struct rt5645_eq_param_s *)ucontrol->value.bytes.data;
+	int i;
+
+	for (i = 0; i < RT5645_HWEQ_NUM; i++) {
+		eq_param[i].reg = cpu_to_be16(rt5645->eq_param[i].reg);
+		eq_param[i].val = cpu_to_be16(rt5645->eq_param[i].val);
+	}
+
+	return 0;
+}
+
+static bool rt5645_validate_hweq(unsigned short reg)
+{
+	if ((reg >= 0x1a4 && reg <= 0x1cd) | (reg >= 0x1e5 && reg <= 0x1f8) |
+		(reg == RT5645_EQ_CTRL2))
+		return true;
+
+	return false;
+}
+
+static int rt5645_hweq_put(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+	struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component);
+	struct rt5645_eq_param_s *eq_param =
+		(struct rt5645_eq_param_s *)ucontrol->value.bytes.data;
+	int i;
+
+	for (i = 0; i < RT5645_HWEQ_NUM; i++) {
+		eq_param[i].reg = be16_to_cpu(eq_param[i].reg);
+		eq_param[i].val = be16_to_cpu(eq_param[i].val);
+	}
+
+	/* The final setting of the table should be RT5645_EQ_CTRL2 */
+	for (i = RT5645_HWEQ_NUM - 1; i >= 0; i--) {
+		if (eq_param[i].reg == 0)
+			continue;
+		else if (eq_param[i].reg != RT5645_EQ_CTRL2)
+			return 0;
+		else
+			break;
+	}
+
+	for (i = 0; i < RT5645_HWEQ_NUM; i++) {
+		if (!rt5645_validate_hweq(eq_param[i].reg) &&
+			eq_param[i].reg != 0)
+			return 0;
+		else if (eq_param[i].reg == 0)
+			break;
+	}
+
+	memcpy(rt5645->eq_param, eq_param,
+		RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s));
+
+	return 0;
+}
+
+#define RT5645_HWEQ(xname) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.info = rt5645_hweq_info, \
+	.get = rt5645_hweq_get, \
+	.put = rt5645_hweq_put \
+}
+
 static const struct snd_kcontrol_new rt5645_snd_controls[] = {
 	/* Speaker Output Volume */
 	SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL,
@@ -476,6 +572,10 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = {
 	SOC_DOUBLE_TLV("Speaker Playback Volume", RT5645_SPK_VOL,
 		RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv),
 
+	/* ClassD modulator Speaker Gain Ratio */
+	SOC_SINGLE_TLV("Speaker ClassD Playback Volume", RT5645_SPO_CLSD_RATIO,
+		RT5645_SPK_G_CLSD_SFT, 7, 0, spk_clsd_tlv),
+
 	/* Headphone Output Volume */
 	SOC_DOUBLE("Headphone Channel Switch", RT5645_HP_VOL,
 		RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1),
@@ -529,6 +629,7 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = {
 	/* I2S2 function select */
 	SOC_SINGLE("I2S2 Func Switch", RT5645_GPIO_CTRL1, RT5645_I2S2_SEL_SFT,
 		1, 1),
+	RT5645_HWEQ("Speaker HWEQ"),
 };
 
 /**
@@ -619,6 +720,22 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source,
 
 }
 
+static int rt5645_enable_hweq(struct snd_soc_codec *codec)
+{
+	struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec);
+	int i;
+
+	for (i = 0; i < RT5645_HWEQ_NUM; i++) {
+		if (rt5645_validate_hweq(rt5645->eq_param[i].reg))
+			regmap_write(rt5645->regmap, rt5645->eq_param[i].reg,
+					rt5645->eq_param[i].val);
+		else
+			break;
+	}
+
+	return 0;
+}
+
 /**
  * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters
  * @codec: SoC audio codec device.
@@ -1523,6 +1640,7 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w,
 
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
+		rt5645_enable_hweq(codec);
 		snd_soc_update_bits(codec, RT5645_PWR_DIG1,
 			RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R |
 			RT5645_PWR_CLS_D_L,
@@ -1531,6 +1649,7 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w,
 		break;
 
 	case SND_SOC_DAPM_PRE_PMD:
+		snd_soc_write(codec, RT5645_EQ_CTRL2, 0);
 		snd_soc_update_bits(codec, RT5645_PWR_DIG1,
 			RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R |
 			RT5645_PWR_CLS_D_L, 0);
@@ -2733,6 +2852,10 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_update_bits(codec, RT5645_PWR_ANLG1,
 			RT5645_PWR_FV1 | RT5645_PWR_FV2,
 			RT5645_PWR_FV1 | RT5645_PWR_FV2);
+		if (rt5645->en_button_func &&
+			snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
+			queue_delayed_work(system_power_efficient_wq,
+				&rt5645->jack_detect_work, msecs_to_jiffies(0));
 		break;
 
 	case SND_SOC_BIAS_OFF:
@@ -2829,6 +2952,9 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert)
 			snd_soc_dapm_sync(dapm);
 			rt5645->jack_type = SND_JACK_HEADPHONE;
 		}
+		if (rt5645->pdata.jd_invert)
+			regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
+				RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV);
 	} else { /* jack out */
 		rt5645->jack_type = 0;
 
@@ -2847,6 +2973,9 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert)
 			snd_soc_dapm_disable_pin(dapm, "LDO2");
 		snd_soc_dapm_disable_pin(dapm, "Mic Det Power");
 		snd_soc_dapm_sync(dapm);
+		if (rt5645->pdata.jd_invert)
+			regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
+				RT5645_JD_1_1_MASK, RT5645_JD_1_1_NOR);
 	}
 
 	return rt5645->jack_type;
@@ -3038,6 +3167,9 @@ static int rt5645_probe(struct snd_soc_codec *codec)
 		snd_soc_dapm_sync(dapm);
 	}
 
+	rt5645->eq_param = devm_kzalloc(codec->dev,
+		RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s), GFP_KERNEL);
+
 	return 0;
 }
 
@@ -3098,7 +3230,7 @@ static struct snd_soc_dai_driver rt5645_dai[] = {
 		.capture = {
 			.stream_name = "AIF1 Capture",
 			.channels_min = 1,
-			.channels_max = 2,
+			.channels_max = 4,
 			.rates = RT5645_STEREO_RATES,
 			.formats = RT5645_FORMATS,
 		},
@@ -3209,9 +3341,42 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"),
 		},
 	},
+	{
+		.ident = "Google Reks",
+		.callback = strago_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Reks"),
+		},
+	},
 	{ }
 };
 
+static struct rt5645_platform_data buddy_platform_data = {
+	.dmic1_data_pin = RT5645_DMIC_DATA_GPIO5,
+	.dmic2_data_pin = RT5645_DMIC_DATA_IN2P,
+	.jd_mode = 3,
+	.jd_invert = true,
+};
+
+static int buddy_quirk_cb(const struct dmi_system_id *id)
+{
+	rt5645_pdata = &buddy_platform_data;
+
+	return 1;
+}
+
+static struct dmi_system_id dmi_platform_intel_broadwell[] = {
+	{
+		.ident = "Chrome Buddy",
+		.callback = buddy_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"),
+		},
+	},
+	{ }
+};
+
+
 static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev)
 {
 	rt5645->pdata.in2_diff = device_property_read_bool(dev,
@@ -3244,7 +3409,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
 
 	if (pdata)
 		rt5645->pdata = *pdata;
-	else if (dmi_check_system(dmi_platform_intel_braswell))
+	else if (dmi_check_system(dmi_platform_intel_braswell) ||
+			dmi_check_system(dmi_platform_intel_broadwell))
 		rt5645->pdata = *rt5645_pdata;
 	else
 		rt5645_parse_dt(rt5645, &i2c->dev);
@@ -3472,6 +3638,8 @@ static void rt5645_i2c_shutdown(struct i2c_client *i2c)
 		RT5645_CBJ_MN_JD);
 	regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, RT5645_CBJ_BST1_EN,
 		0);
+	msleep(20);
+	regmap_write(rt5645->regmap, RT5645_RESET, 0);
 }
 
 static struct i2c_driver rt5645_i2c_driver = {
diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h
index 8c964cfb120d..093e46d559fb 100644
--- a/sound/soc/codecs/rt5645.h
+++ b/sound/soc/codecs/rt5645.h
@@ -621,14 +621,14 @@
 #define RT5645_G_OM_L_SM_L_SFT			6
 #define RT5645_M_BST1_L_SM_L			(0x1 << 5)
 #define RT5645_M_BST1_L_SM_L_SFT		5
+#define RT5645_M_BST3_L_SM_L			(0x1 << 4)
+#define RT5645_M_BST3_L_SM_L_SFT		4
 #define RT5645_M_IN_L_SM_L			(0x1 << 3)
 #define RT5645_M_IN_L_SM_L_SFT			3
-#define RT5645_M_DAC_L1_SM_L			(0x1 << 1)
-#define RT5645_M_DAC_L1_SM_L_SFT		1
 #define RT5645_M_DAC_L2_SM_L			(0x1 << 2)
 #define RT5645_M_DAC_L2_SM_L_SFT		2
-#define RT5645_M_BST3_L_SM_L			(0x1 << 4)
-#define RT5645_M_BST3_L_SM_L_SFT		4
+#define RT5645_M_DAC_L1_SM_L			(0x1 << 1)
+#define RT5645_M_DAC_L1_SM_L_SFT		1
 
 /* SPK Right Mixer Control (0x47) */
 #define RT5645_G_RM_R_SM_R_MASK			(0x3 << 14)
@@ -643,14 +643,14 @@
 #define RT5645_G_OM_R_SM_R_SFT			6
 #define RT5645_M_BST2_R_SM_R			(0x1 << 5)
 #define RT5645_M_BST2_R_SM_R_SFT		5
+#define RT5645_M_BST3_R_SM_R			(0x1 << 4)
+#define RT5645_M_BST3_R_SM_R_SFT		4
 #define RT5645_M_IN_R_SM_R			(0x1 << 3)
 #define RT5645_M_IN_R_SM_R_SFT			3
-#define RT5645_M_DAC_R1_SM_R			(0x1 << 1)
-#define RT5645_M_DAC_R1_SM_R_SFT		1
 #define RT5645_M_DAC_R2_SM_R			(0x1 << 2)
 #define RT5645_M_DAC_R2_SM_R_SFT		2
-#define RT5645_M_BST3_R_SM_R			(0x1 << 4)
-#define RT5645_M_BST3_R_SM_R_SFT		4
+#define RT5645_M_DAC_R1_SM_R			(0x1 << 1)
+#define RT5645_M_DAC_R1_SM_R_SFT		1
 
 /* SPOLMIX Control (0x48) */
 #define RT5645_M_DAC_L1_SPM_L			(0x1 << 15)
@@ -670,13 +670,17 @@
 #define RT5645_M_SV_R_SPM_R			(0x1 << 0)
 #define RT5645_M_SV_R_SPM_R_SFT			0
 
+/* SPOMIX Ratio Control (0x4a) */
+#define RT5645_SPK_G_CLSD_MASK			(0x7 << 0)
+#define RT5645_SPK_G_CLSD_SFT			0
+
 /* Mono Output Mixer Control (0x4c) */
+#define RT5645_G_MONOMIX_MASK			(0x1 << 10)
+#define RT5645_G_MONOMIX_SFT			10
 #define RT5645_M_OV_L_MM			(0x1 << 9)
 #define RT5645_M_OV_L_MM_SFT			9
 #define RT5645_M_DAC_L2_MA			(0x1 << 8)
 #define RT5645_M_DAC_L2_MA_SFT			8
-#define RT5645_G_MONOMIX_MASK			(0x1 << 10)
-#define RT5645_G_MONOMIX_SFT			10
 #define RT5645_M_BST2_MM			(0x1 << 4)
 #define RT5645_M_BST2_MM_SFT			4
 #define RT5645_M_DAC_R1_MM			(0x1 << 3)
@@ -779,8 +783,6 @@
 #define RT5645_PWR_CLS_D_R_BIT			9
 #define RT5645_PWR_CLS_D_L			(0x1 << 8)
 #define RT5645_PWR_CLS_D_L_BIT			8
-#define RT5645_PWR_ADC_R			(0x1 << 1)
-#define RT5645_PWR_ADC_R_BIT			1
 #define RT5645_PWR_DAC_L2			(0x1 << 7)
 #define RT5645_PWR_DAC_L2_BIT			7
 #define RT5645_PWR_DAC_R2			(0x1 << 6)
@@ -1628,6 +1630,10 @@
 #define RT5645_OT_P_NOR				(0x0 << 10)
 #define RT5645_OT_P_INV				(0x1 << 10)
 #define RT5645_IRQ_JD_1_1_EN			(0x1 << 9)
+#define RT5645_JD_1_1_MASK			(0x1 << 7)
+#define RT5645_JD_1_1_SFT			7
+#define RT5645_JD_1_1_NOR			(0x0 << 7)
+#define RT5645_JD_1_1_INV			(0x1 << 7)
 
 /* IRQ Control 2 (0xbe) */
 #define RT5645_IRQ_MB1_OC_MASK			(0x1 << 15)
diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index ddb0203fc649..86b81a60ac52 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -723,17 +723,11 @@ static struct snd_soc_codec_driver ssm2518_codec_driver = {
 	.num_dapm_routes = ARRAY_SIZE(ssm2518_routes),
 };
 
-static bool ssm2518_register_volatile(struct device *dev, unsigned int reg)
-{
-	return false;
-}
-
 static const struct regmap_config ssm2518_regmap_config = {
 	.val_bits = 8,
 	.reg_bits = 8,
 
 	.max_register = SSM2518_REG_DRC_9,
-	.volatile_reg = ssm2518_register_volatile,
 
 	.cache_type = REGCACHE_RBTREE,
 	.reg_defaults = ssm2518_reg_defaults,
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 8739126a1f6f..a564759845f9 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -80,6 +80,7 @@ struct aic3x_priv {
 	unsigned int sysclk;
 	unsigned int dai_fmt;
 	unsigned int tdm_delay;
+	unsigned int slot_width;
 	struct list_head list;
 	int master;
 	int gpio_reset;
@@ -1025,10 +1026,14 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
 	u16 d, pll_d = 1;
 	int clk;
+	int width = aic3x->slot_width;
+
+	if (!width)
+		width = params_width(params);
 
 	/* select data word length */
 	data = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
-	switch (params_width(params)) {
+	switch (width) {
 	case 16:
 		break;
 	case 20:
@@ -1170,12 +1175,16 @@ static int aic3x_prepare(struct snd_pcm_substream *substream,
 	struct snd_soc_codec *codec = dai->codec;
 	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
 	int delay = 0;
+	int width = aic3x->slot_width;
+
+	if (!width)
+		width = substream->runtime->sample_bits;
 
 	/* TDM slot selection only valid in DSP_A/_B mode */
 	if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_A)
-		delay += (aic3x->tdm_delay + 1);
+		delay += (aic3x->tdm_delay*width + 1);
 	else if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_B)
-		delay += aic3x->tdm_delay;
+		delay += aic3x->tdm_delay*width;
 
 	/* Configure data delay */
 	snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay);
@@ -1296,7 +1305,20 @@ static int aic3x_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
 		return -EINVAL;
 	}
 
-	aic3x->tdm_delay = lsb * slot_width;
+	switch (slot_width) {
+	case 16:
+	case 20:
+	case 24:
+	case 32:
+		break;
+	default:
+		dev_err(codec->dev, "Unsupported slot width %d\n", slot_width);
+		return -EINVAL;
+	}
+
+
+	aic3x->tdm_delay = lsb;
+	aic3x->slot_width = slot_width;
 
 	/* DOUT in high-impedance on inactive bit clocks */
 	snd_soc_update_bits(codec, AIC3X_ASD_INTF_CTRLA,
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 2713e1845cbc..a5a4e9f75c57 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -1612,19 +1612,16 @@ static void twl4030_constraints(struct twl4030_priv *twl4030,
 		return;
 
 	/* Set the constraints according to the already configured stream */
-	snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+	snd_pcm_hw_constraint_single(slv_substream->runtime,
 				SNDRV_PCM_HW_PARAM_RATE,
-				twl4030->rate,
 				twl4030->rate);
 
-	snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+	snd_pcm_hw_constraint_single(slv_substream->runtime,
 				SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-				twl4030->sample_bits,
 				twl4030->sample_bits);
 
-	snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+	snd_pcm_hw_constraint_single(slv_substream->runtime,
 				SNDRV_PCM_HW_PARAM_CHANNELS,
-				twl4030->channels,
 				twl4030->channels);
 }
 
@@ -1669,9 +1666,9 @@ static int twl4030_startup(struct snd_pcm_substream *substream,
 			/* In option2 4 channel is not supported, set the
 			 * constraint for the first stream for channels, the
 			 * second stream will 'inherit' this cosntraint */
-			snd_pcm_hw_constraint_minmax(substream->runtime,
+			snd_pcm_hw_constraint_single(substream->runtime,
 						     SNDRV_PCM_HW_PARAM_CHANNELS,
-						     2, 2);
+						     2);
 		}
 		twl4030->master_substream = substream;
 	}
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index e19026380534..e4c694c758b8 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -150,14 +150,12 @@ static int uda134x_startup(struct snd_pcm_substream *substream,
 			 master_runtime->sample_bits,
 			 master_runtime->rate);
 
-		snd_pcm_hw_constraint_minmax(substream->runtime,
+		snd_pcm_hw_constraint_single(substream->runtime,
 					     SNDRV_PCM_HW_PARAM_RATE,
-					     master_runtime->rate,
 					     master_runtime->rate);
 
-		snd_pcm_hw_constraint_minmax(substream->runtime,
+		snd_pcm_hw_constraint_single(substream->runtime,
 					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-					     master_runtime->sample_bits,
 					     master_runtime->sample_bits);
 
 		uda134x->slave_substream = substream;
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index 80fb1dc81f6c..7693c1129bab 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -307,11 +307,10 @@ static int wl1273_startup(struct snd_pcm_substream *substream,
 
 	switch (wl1273->mode) {
 	case WL1273_MODE_BT:
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_RATE,
-					     8000, 8000);
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_CHANNELS, 1, 1);
+		snd_pcm_hw_constraint_single(substream->runtime,
+					     SNDRV_PCM_HW_PARAM_RATE, 8000);
+		snd_pcm_hw_constraint_single(substream->runtime,
+					     SNDRV_PCM_HW_PARAM_CHANNELS, 1);
 		break;
 	case WL1273_MODE_FM_RX:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 786abd02b140..a67ea10f41a1 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -620,7 +620,7 @@ static int wm2000_anc_mode_put(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev);
-	int anc_active = ucontrol->value.integer.value[0];
+	unsigned int anc_active = ucontrol->value.integer.value[0];
 	int ret;
 
 	if (anc_active > 1)
@@ -653,7 +653,7 @@ static int wm2000_speaker_put(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev);
-	int val = ucontrol->value.integer.value[0];
+	unsigned int val = ucontrol->value.integer.value[0];
 	int ret;
 
 	if (val > 1)
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 9756578fc752..c04c0bc6f58a 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -38,6 +38,12 @@
 struct wm5110_priv {
 	struct arizona_priv core;
 	struct arizona_fll fll[2];
+
+	unsigned int in_value;
+	int in_pre_pending;
+	int in_post_pending;
+
+	unsigned int in_pga_cache[6];
 };
 
 static const struct wm_adsp_region wm5110_dsp1_regions[] = {
@@ -428,6 +434,127 @@ err:
 	return ret;
 }
 
+static int wm5110_in_pga_get(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+	struct snd_soc_card *card = dapm->card;
+	int ret;
+
+	/*
+	 * PGA Volume is also used as part of the enable sequence, so
+	 * usage of it should be avoided whilst that is running.
+	 */
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+	ret = snd_soc_get_volsw_range(kcontrol, ucontrol);
+
+	mutex_unlock(&card->dapm_mutex);
+
+	return ret;
+}
+
+static int wm5110_in_pga_put(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+	struct snd_soc_card *card = dapm->card;
+	int ret;
+
+	/*
+	 * PGA Volume is also used as part of the enable sequence, so
+	 * usage of it should be avoided whilst that is running.
+	 */
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+	ret = snd_soc_put_volsw_range(kcontrol, ucontrol);
+
+	mutex_unlock(&card->dapm_mutex);
+
+	return ret;
+}
+
+static int wm5110_in_analog_ev(struct snd_soc_dapm_widget *w,
+			       struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct wm5110_priv *wm5110 = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->arizona;
+	unsigned int reg, mask;
+	struct reg_sequence analog_seq[] = {
+		{ 0x80, 0x3 },
+		{ 0x35d, 0 },
+		{ 0x80, 0x0 },
+	};
+
+	reg = ARIZONA_IN1L_CONTROL + ((w->shift ^ 0x1) * 4);
+	mask = ARIZONA_IN1L_PGA_VOL_MASK;
+
+	switch (event) {
+	case SND_SOC_DAPM_WILL_PMU:
+		wm5110->in_value |= 0x3 << ((w->shift ^ 0x1) * 2);
+		wm5110->in_pre_pending++;
+		wm5110->in_post_pending++;
+		return 0;
+	case SND_SOC_DAPM_PRE_PMU:
+		wm5110->in_pga_cache[w->shift] = snd_soc_read(codec, reg);
+
+		snd_soc_update_bits(codec, reg, mask,
+				    0x40 << ARIZONA_IN1L_PGA_VOL_SHIFT);
+
+		wm5110->in_pre_pending--;
+		if (wm5110->in_pre_pending == 0) {
+			analog_seq[1].def = wm5110->in_value;
+			regmap_multi_reg_write_bypassed(arizona->regmap,
+							analog_seq,
+							ARRAY_SIZE(analog_seq));
+
+			msleep(55);
+
+			wm5110->in_value = 0;
+		}
+
+		break;
+	case SND_SOC_DAPM_POST_PMU:
+		snd_soc_update_bits(codec, reg, mask,
+				    wm5110->in_pga_cache[w->shift]);
+
+		wm5110->in_post_pending--;
+		if (wm5110->in_post_pending == 0)
+			regmap_multi_reg_write_bypassed(arizona->regmap,
+							analog_seq,
+							ARRAY_SIZE(analog_seq));
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int wm5110_in_ev(struct snd_soc_dapm_widget *w,
+			struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->arizona;
+
+	switch (arizona->rev) {
+	case 0 ... 4:
+		if (arizona_input_analog(codec, w->shift))
+			wm5110_in_analog_ev(w, kcontrol, event);
+
+		break;
+	default:
+		break;
+	}
+
+	return arizona_in_ev(w, kcontrol, event);
+}
+
 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);
@@ -454,18 +581,24 @@ 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),
-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_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL,
-		     ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL,
-		     ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL,
+			 ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+			 wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL,
+			 ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+			 wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL,
+			 ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+			 wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL,
+			 ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+			 wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL,
+			 ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+			 wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL,
+			 ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+			 wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
 
 SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum),
 
@@ -896,29 +1029,35 @@ 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,
+		   0, NULL, 0, wm5110_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_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+		   SND_SOC_DAPM_WILL_PMU),
 SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
-		   0, NULL, 0, arizona_in_ev,
+		   0, NULL, 0, wm5110_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_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+		   SND_SOC_DAPM_WILL_PMU),
 SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
-		   0, NULL, 0, arizona_in_ev,
+		   0, NULL, 0, wm5110_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_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+		   SND_SOC_DAPM_WILL_PMU),
 SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
-		   0, NULL, 0, arizona_in_ev,
+		   0, NULL, 0, wm5110_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_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+		   SND_SOC_DAPM_WILL_PMU),
 SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT,
-		   0, NULL, 0, arizona_in_ev,
+		   0, NULL, 0, wm5110_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_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+		   SND_SOC_DAPM_WILL_PMU),
 SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT,
-		   0, NULL, 0, arizona_in_ev,
+		   0, NULL, 0, wm5110_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_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+		   SND_SOC_DAPM_WILL_PMU),
 SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index ace8645245a0..07cf1bd7913a 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -132,7 +132,7 @@ static int wm8731_put_deemph(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
-	int deemph = ucontrol->value.integer.value[0];
+	unsigned int deemph = ucontrol->value.integer.value[0];
 	int ret = 0;
 
 	if (deemph > 1)
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index b011253459af..e4cc41e6c23e 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -452,7 +452,7 @@ static int wm8903_put_deemph(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-	int deemph = ucontrol->value.integer.value[0];
+	unsigned int deemph = ucontrol->value.integer.value[0];
 	int ret = 0;
 
 	if (deemph > 1)
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index b783743dc97e..2aa23f1b9e3c 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -534,7 +534,7 @@ static int wm8904_put_deemph(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
-	int deemph = ucontrol->value.integer.value[0];
+	unsigned int deemph = ucontrol->value.integer.value[0];
 
 	if (deemph > 1)
 		return -EINVAL;
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 12e4435f00f8..9db00d53abe7 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -402,7 +402,7 @@ static int wm8955_put_deemph(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
-	int deemph = ucontrol->value.integer.value[0];
+	unsigned int deemph = ucontrol->value.integer.value[0];
 
 	if (deemph > 1)
 		return -EINVAL;
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index dbd88408861a..056375339ea3 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -201,7 +201,7 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
-	int deemph = ucontrol->value.integer.value[0];
+	unsigned int deemph = ucontrol->value.integer.value[0];
 
 	if (deemph > 1)
 		return -EINVAL;
diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c
new file mode 100644
index 000000000000..8782dfb628ab
--- /dev/null
+++ b/sound/soc/codecs/wm8998.c
@@ -0,0 +1,1430 @@
+/*
+ * wm8998.c -- ALSA SoC Audio driver for WM8998 codecs
+ *
+ * Copyright 2015 Cirrus Logic, Inc.
+ *
+ * Author: Richard Fitzgerald <rf@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 "wm8998.h"
+
+struct wm8998_priv {
+	struct arizona_priv core;
+	struct arizona_fll fll[2];
+};
+
+static int wm8998_asrc_ev(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *kcontrol,
+			  int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	unsigned int val;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		val = snd_soc_read(codec, ARIZONA_ASRC_RATE1);
+		val &= ARIZONA_ASRC_RATE1_MASK;
+		val >>= ARIZONA_ASRC_RATE1_SHIFT;
+
+		switch (val) {
+		case 0:
+		case 1:
+		case 2:
+			val = snd_soc_read(codec,
+					   ARIZONA_SAMPLE_RATE_1 + val);
+			if (val >= 0x11) {
+				dev_warn(codec->dev,
+					 "Unsupported ASRC rate1 (%s)\n",
+					 arizona_sample_rate_val_to_name(val));
+			return -EINVAL;
+			}
+			break;
+		default:
+			dev_err(codec->dev,
+				"Illegal ASRC rate1 selector (0x%x)\n",
+				val);
+			return -EINVAL;
+		}
+
+		val = snd_soc_read(codec, ARIZONA_ASRC_RATE2);
+		val &= ARIZONA_ASRC_RATE2_MASK;
+		val >>= ARIZONA_ASRC_RATE2_SHIFT;
+
+		switch (val) {
+		case 8:
+		case 9:
+			val -= 0x8;
+			val = snd_soc_read(codec,
+					   ARIZONA_ASYNC_SAMPLE_RATE_1 + val);
+			if (val >= 0x11) {
+				dev_warn(codec->dev,
+					 "Unsupported ASRC rate2 (%s)\n",
+					 arizona_sample_rate_val_to_name(val));
+				return -EINVAL;
+			}
+			break;
+		default:
+			dev_err(codec->dev,
+				"Illegal ASRC rate2 selector (0x%x)\n",
+				val);
+			return -EINVAL;
+		}
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int wm8998_in1mux_put(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+	struct wm8998_priv *wm8998 = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = wm8998->core.arizona;
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	unsigned int mux, inmode;
+	unsigned int mode_val, src_val;
+
+	mux = ucontrol->value.enumerated.item[0];
+	if (mux > 1)
+		return -EINVAL;
+
+	/* L and R registers have same shift and mask */
+	inmode = arizona->pdata.inmode[2 * mux];
+	src_val = mux << ARIZONA_IN1L_SRC_SHIFT;
+	if (inmode & ARIZONA_INMODE_SE)
+		src_val |= 1 << ARIZONA_IN1L_SRC_SE_SHIFT;
+
+	switch (arizona->pdata.inmode[0]) {
+	case ARIZONA_INMODE_DMIC:
+		if (mux)
+			mode_val = 0;	/* B always analogue */
+		else
+			mode_val = 1 << ARIZONA_IN1_MODE_SHIFT;
+
+		snd_soc_update_bits(codec, ARIZONA_IN1L_CONTROL,
+				    ARIZONA_IN1_MODE_MASK, mode_val);
+
+		/* IN1A is digital so L and R must change together */
+		/* src_val setting same for both registers */
+		snd_soc_update_bits(codec,
+				    ARIZONA_ADC_DIGITAL_VOLUME_1L,
+				    ARIZONA_IN1L_SRC_MASK |
+				    ARIZONA_IN1L_SRC_SE_MASK, src_val);
+		snd_soc_update_bits(codec,
+				    ARIZONA_ADC_DIGITAL_VOLUME_1R,
+				    ARIZONA_IN1R_SRC_MASK |
+				    ARIZONA_IN1R_SRC_SE_MASK, src_val);
+		break;
+	default:
+		/* both analogue */
+		snd_soc_update_bits(codec,
+				    e->reg,
+				    ARIZONA_IN1L_SRC_MASK |
+				    ARIZONA_IN1L_SRC_SE_MASK,
+				    src_val);
+		break;
+	}
+
+	return snd_soc_dapm_mux_update_power(dapm, kcontrol,
+					     ucontrol->value.enumerated.item[0],
+					     e, NULL);
+}
+
+static int wm8998_in2mux_put(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+	struct wm8998_priv *wm8998 = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = wm8998->core.arizona;
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	unsigned int mux, inmode, src_val, mode_val;
+
+	mux = ucontrol->value.enumerated.item[0];
+	if (mux > 1)
+		return -EINVAL;
+
+	inmode = arizona->pdata.inmode[1 + (2 * mux)];
+	if (inmode & ARIZONA_INMODE_DMIC)
+		mode_val = 1 << ARIZONA_IN2_MODE_SHIFT;
+	else
+		mode_val = 0;
+
+	src_val = mux << ARIZONA_IN2L_SRC_SHIFT;
+	if (inmode & ARIZONA_INMODE_SE)
+		src_val |= 1 << ARIZONA_IN2L_SRC_SE_SHIFT;
+
+	snd_soc_update_bits(codec, ARIZONA_IN2L_CONTROL,
+			    ARIZONA_IN2_MODE_MASK, mode_val);
+
+	snd_soc_update_bits(codec, ARIZONA_ADC_DIGITAL_VOLUME_2L,
+			    ARIZONA_IN2L_SRC_MASK | ARIZONA_IN2L_SRC_SE_MASK,
+			    src_val);
+
+	return snd_soc_dapm_mux_update_power(dapm, kcontrol,
+					     ucontrol->value.enumerated.item[0],
+					     e, NULL);
+}
+
+static const char * const wm8998_inmux_texts[] = {
+	"A",
+	"B",
+};
+
+static const SOC_ENUM_SINGLE_DECL(wm8998_in1muxl_enum,
+				  ARIZONA_ADC_DIGITAL_VOLUME_1L,
+				  ARIZONA_IN1L_SRC_SHIFT,
+				  wm8998_inmux_texts);
+
+static const SOC_ENUM_SINGLE_DECL(wm8998_in1muxr_enum,
+				  ARIZONA_ADC_DIGITAL_VOLUME_1R,
+				  ARIZONA_IN1R_SRC_SHIFT,
+				  wm8998_inmux_texts);
+
+static const SOC_ENUM_SINGLE_DECL(wm8998_in2mux_enum,
+				  ARIZONA_ADC_DIGITAL_VOLUME_2L,
+				  ARIZONA_IN2L_SRC_SHIFT,
+				  wm8998_inmux_texts);
+
+static const struct snd_kcontrol_new wm8998_in1mux[2] = {
+	SOC_DAPM_ENUM_EXT("IN1L Mux", wm8998_in1muxl_enum,
+			  snd_soc_dapm_get_enum_double, wm8998_in1mux_put),
+	SOC_DAPM_ENUM_EXT("IN1R Mux", wm8998_in1muxr_enum,
+			  snd_soc_dapm_get_enum_double, wm8998_in1mux_put),
+};
+
+static const struct snd_kcontrol_new wm8998_in2mux =
+	SOC_DAPM_ENUM_EXT("IN2 Mux", wm8998_in2mux_enum,
+			  snd_soc_dapm_get_enum_double, wm8998_in2mux_put);
+
+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(ng_tlv, -10200, 600, 0);
+
+#define WM8998_NG_SRC(name, base) \
+	SOC_SINGLE(name " NG HPOUTL Switch",  base,  0, 1, 0), \
+	SOC_SINGLE(name " NG HPOUTR Switch",  base,  1, 1, 0), \
+	SOC_SINGLE(name " NG LINEOUTL Switch",  base,  2, 1, 0), \
+	SOC_SINGLE(name " NG LINEOUTR Switch",  base,  3, 1, 0), \
+	SOC_SINGLE(name " NG EPOUT Switch",   base,  4, 1, 0), \
+	SOC_SINGLE(name " NG SPKOUTL Switch",  base,  6, 1, 0), \
+	SOC_SINGLE(name " NG SPKOUTR Switch",  base,  7, 1, 0)
+
+static const struct snd_kcontrol_new wm8998_snd_controls[] = {
+SOC_ENUM("IN1 OSR", arizona_in_dmic_osr[0]),
+SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]),
+
+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("IN2 Volume", ARIZONA_IN2L_CONTROL,
+		     ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+
+SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum),
+
+SOC_SINGLE("IN1L HPF Switch", ARIZONA_IN1L_CONTROL,
+	   ARIZONA_IN1L_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN1R HPF Switch", ARIZONA_IN1R_CONTROL,
+	   ARIZONA_IN1R_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN2 HPF Switch", ARIZONA_IN2L_CONTROL,
+	   ARIZONA_IN2L_HPF_SHIFT, 1, 0),
+
+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("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+	       ARIZONA_IN2L_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_GAINMUX_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19),
+SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE_SHIFT, 1, 0),
+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),
+
+SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19),
+SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE_SHIFT, 1, 0),
+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),
+
+SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19),
+SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE_SHIFT, 1, 0),
+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),
+
+SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19),
+SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE_SHIFT, 1, 0),
+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_GAINMUX_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_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),
+
+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_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),
+
+SOC_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]),
+SOC_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]),
+SOC_ENUM("ISRC1 FSH", arizona_isrc_fsh[0]),
+SOC_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]),
+SOC_ENUM("ASRC RATE 1", arizona_asrc_rate1),
+
+ARIZONA_MIXER_CONTROLS("HPOUTL", ARIZONA_OUT1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUTR", ARIZONA_OUT1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LINEOUTL", ARIZONA_OUT2LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LINEOUTR", ARIZONA_OUT2RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTL", ARIZONA_OUT4LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDATL", ARIZONA_OUT5LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDATR", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
+
+SOC_DOUBLE_R("HPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("LINEOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1),
+SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	     ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("SPKDAT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("HPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("LINEOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_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_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("SPKDAT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+
+SOC_DOUBLE("SPKDAT Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
+	   ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
+
+SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp),
+SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp),
+
+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),
+
+WM8998_NG_SRC("HPOUTL", ARIZONA_NOISE_GATE_SELECT_1L),
+WM8998_NG_SRC("HPOUTR", ARIZONA_NOISE_GATE_SELECT_1R),
+WM8998_NG_SRC("LINEOUTL", ARIZONA_NOISE_GATE_SELECT_2L),
+WM8998_NG_SRC("LINEOUTR", ARIZONA_NOISE_GATE_SELECT_2R),
+WM8998_NG_SRC("EPOUT",  ARIZONA_NOISE_GATE_SELECT_3L),
+WM8998_NG_SRC("SPKOUTL", ARIZONA_NOISE_GATE_SELECT_4L),
+WM8998_NG_SRC("SPKOUTR", ARIZONA_NOISE_GATE_SELECT_4R),
+WM8998_NG_SRC("SPKDATL", ARIZONA_NOISE_GATE_SELECT_5L),
+WM8998_NG_SRC("SPKDATR", 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("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX3", ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX4", ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX5", ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
+
+ARIZONA_GAINMUX_CONTROLS("SLIMTX1", ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("SLIMTX2", ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("SLIMTX3", ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("SLIMTX4", ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE),
+
+ARIZONA_GAINMUX_CONTROLS("SPDIFTX1", ARIZONA_SPDIFTX1MIX_INPUT_1_SOURCE),
+ARIZONA_GAINMUX_CONTROLS("SPDIFTX2", ARIZONA_SPDIFTX2MIX_INPUT_1_SOURCE),
+};
+
+ARIZONA_MUX_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MUX_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(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(OUT2L, ARIZONA_OUT2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT2R, ARIZONA_OUT2RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT3,  ARIZONA_OUT3LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTL, ARIZONA_OUT4LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTR, ARIZONA_OUT4RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDATL, ARIZONA_OUT5LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDATR, 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(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX3, ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX4, ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX5, ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX6, ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(SLIMTX1, ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(SLIMTX2, ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(SLIMTX3, ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(SLIMTX4, ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(SLIMTX5, ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(SLIMTX6, ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(SPD1TX1, ARIZONA_SPDIFTX1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(SPD1TX2, ARIZONA_SPDIFTX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1INT3, ARIZONA_ISRC1INT3MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1INT4, ARIZONA_ISRC1INT4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1DEC3, ARIZONA_ISRC1DEC3MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1DEC4, ARIZONA_ISRC1DEC4MIX_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 * const wm8998_aec_loopback_texts[] = {
+	"HPOUTL", "HPOUTR", "LINEOUTL", "LINEOUTR", "EPOUT",
+	"SPKOUTL", "SPKOUTR", "SPKDATL", "SPKDATR",
+};
+
+static const unsigned int wm8998_aec_loopback_values[] = {
+	0, 1, 2, 3, 4, 6, 7, 8, 9,
+};
+
+static const SOC_VALUE_ENUM_SINGLE_DECL(wm8998_aec1_loopback,
+					ARIZONA_DAC_AEC_CONTROL_1,
+					ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf,
+					wm8998_aec_loopback_texts,
+					wm8998_aec_loopback_values);
+
+static const SOC_VALUE_ENUM_SINGLE_DECL(wm8998_aec2_loopback,
+					ARIZONA_DAC_AEC_CONTROL_2,
+					ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf,
+					wm8998_aec_loopback_texts,
+					wm8998_aec_loopback_values);
+
+static const struct snd_kcontrol_new wm8998_aec_loopback_mux[] = {
+	SOC_DAPM_ENUM("AEC1 Loopback", wm8998_aec1_loopback),
+	SOC_DAPM_ENUM("AEC2 Loopback", wm8998_aec2_loopback),
+};
+
+static const struct snd_soc_dapm_widget wm8998_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1,
+		    ARIZONA_SYSCLK_ENA_SHIFT, 0, NULL, 0),
+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("DBVDD3", 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("SPKVDDL", 0, 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0, 0),
+
+SND_SOC_DAPM_SIGGEN("TONE"),
+SND_SOC_DAPM_SIGGEN("HAPTICS"),
+
+SND_SOC_DAPM_INPUT("IN1AL"),
+SND_SOC_DAPM_INPUT("IN1AR"),
+SND_SOC_DAPM_INPUT("IN1BL"),
+SND_SOC_DAPM_INPUT("IN1BR"),
+SND_SOC_DAPM_INPUT("IN2A"),
+SND_SOC_DAPM_INPUT("IN2B"),
+
+SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &wm8998_in1mux[0]),
+SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &wm8998_in1mux[1]),
+SND_SOC_DAPM_MUX("IN2 Mux", SND_SOC_NOPM, 0, 0, &wm8998_in2mux),
+
+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 |
+		   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("IN2 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_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_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3,
+		    ARIZONA_MICB1_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("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_E("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0,
+		   NULL, 0, wm8998_asrc_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0,
+		   NULL, 0, wm8998_asrc_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0,
+		   NULL, 0, wm8998_asrc_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0,
+		   NULL, 0, wm8998_asrc_ev, SND_SOC_DAPM_PRE_PMU),
+
+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("ISRC1INT3", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_INT2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1INT4", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_INT3_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("ISRC1DEC3", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_DEC2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1DEC4", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_DEC3_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_MUX("AEC1 Loopback", ARIZONA_DAC_AEC_CONTROL_1,
+		       ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+		       &wm8998_aec_loopback_mux[0]),
+
+SND_SOC_DAPM_MUX("AEC2 Loopback", ARIZONA_DAC_AEC_CONTROL_2,
+		       ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+		       &wm8998_aec_loopback_mux[1]),
+
+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_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_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_OUT("AIF2TX3", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX4", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX5", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX6", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX6_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_IN("AIF2RX3", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX4", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX5", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX6", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX6_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_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("AIF3TX1", NULL, 0,
+		     ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0,
+		     ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0,
+		    ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
+		    ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0),
+
+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("OUT2L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT3", 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),
+
+SND_SOC_DAPM_PGA("SPD1TX1", ARIZONA_SPD1_TX_CONTROL,
+		   ARIZONA_SPD1_VAL1_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("SPD1TX2", ARIZONA_SPD1_TX_CONTROL,
+		   ARIZONA_SPD1_VAL2_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_OUT_DRV("SPD1", ARIZONA_SPD1_TX_CONTROL,
+		     ARIZONA_SPD1_ENA_SHIFT, 0, NULL, 0),
+
+ARIZONA_MUX_WIDGETS(EQ1, "EQ1"),
+ARIZONA_MUX_WIDGETS(EQ2, "EQ2"),
+ARIZONA_MUX_WIDGETS(EQ3, "EQ3"),
+ARIZONA_MUX_WIDGETS(EQ4, "EQ4"),
+
+ARIZONA_MUX_WIDGETS(DRC1L, "DRC1L"),
+ARIZONA_MUX_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(PWM1, "PWM1"),
+ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"),
+
+ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUTL"),
+ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUTR"),
+ARIZONA_MIXER_WIDGETS(OUT2L, "LINEOUTL"),
+ARIZONA_MIXER_WIDGETS(OUT2R, "LINEOUTR"),
+ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"),
+ARIZONA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"),
+ARIZONA_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"),
+ARIZONA_MIXER_WIDGETS(SPKDATL, "SPKDATL"),
+ARIZONA_MIXER_WIDGETS(SPKDATR, "SPKDATR"),
+
+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(AIF2TX1, "AIF2TX1"),
+ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"),
+ARIZONA_MIXER_WIDGETS(AIF2TX3, "AIF2TX3"),
+ARIZONA_MIXER_WIDGETS(AIF2TX4, "AIF2TX4"),
+ARIZONA_MIXER_WIDGETS(AIF2TX5, "AIF2TX5"),
+ARIZONA_MIXER_WIDGETS(AIF2TX6, "AIF2TX6"),
+
+ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"),
+ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"),
+
+ARIZONA_MUX_WIDGETS(SLIMTX1, "SLIMTX1"),
+ARIZONA_MUX_WIDGETS(SLIMTX2, "SLIMTX2"),
+ARIZONA_MUX_WIDGETS(SLIMTX3, "SLIMTX3"),
+ARIZONA_MUX_WIDGETS(SLIMTX4, "SLIMTX4"),
+ARIZONA_MUX_WIDGETS(SLIMTX5, "SLIMTX5"),
+ARIZONA_MUX_WIDGETS(SLIMTX6, "SLIMTX6"),
+
+ARIZONA_MUX_WIDGETS(SPD1TX1, "SPDIFTX1"),
+ARIZONA_MUX_WIDGETS(SPD1TX2, "SPDIFTX2"),
+
+ARIZONA_MUX_WIDGETS(ASRC1L, "ASRC1L"),
+ARIZONA_MUX_WIDGETS(ASRC1R, "ASRC1R"),
+ARIZONA_MUX_WIDGETS(ASRC2L, "ASRC2L"),
+ARIZONA_MUX_WIDGETS(ASRC2R, "ASRC2R"),
+
+ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"),
+ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"),
+ARIZONA_MUX_WIDGETS(ISRC1DEC3, "ISRC1DEC3"),
+ARIZONA_MUX_WIDGETS(ISRC1DEC4, "ISRC1DEC4"),
+
+ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"),
+ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"),
+ARIZONA_MUX_WIDGETS(ISRC1INT3, "ISRC1INT3"),
+ARIZONA_MUX_WIDGETS(ISRC1INT4, "ISRC1INT4"),
+
+ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"),
+ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"),
+
+ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"),
+ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"),
+
+SND_SOC_DAPM_OUTPUT("HPOUTL"),
+SND_SOC_DAPM_OUTPUT("HPOUTR"),
+SND_SOC_DAPM_OUTPUT("LINEOUTL"),
+SND_SOC_DAPM_OUTPUT("LINEOUTR"),
+SND_SOC_DAPM_OUTPUT("EPOUT"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRP"),
+SND_SOC_DAPM_OUTPUT("SPKDATL"),
+SND_SOC_DAPM_OUTPUT("SPKDATR"),
+SND_SOC_DAPM_OUTPUT("SPDIF"),
+
+SND_SOC_DAPM_OUTPUT("MICSUPP"),
+};
+
+#define ARIZONA_MIXER_INPUT_ROUTES(name)	\
+	{ name, "Tone Generator 1", "Tone Generator 1" }, \
+	{ name, "Tone Generator 2", "Tone Generator 2" }, \
+	{ name, "Haptics", "HAPTICS" }, \
+	{ name, "AEC", "AEC1 Loopback" }, \
+	{ name, "AEC2", "AEC2 Loopback" }, \
+	{ name, "IN1L", "IN1L PGA" }, \
+	{ name, "IN1R", "IN1R PGA" }, \
+	{ name, "IN2L", "IN2 PGA" }, \
+	{ name, "AIF1RX1", "AIF1RX1" }, \
+	{ name, "AIF1RX2", "AIF1RX2" }, \
+	{ name, "AIF1RX3", "AIF1RX3" }, \
+	{ name, "AIF1RX4", "AIF1RX4" }, \
+	{ name, "AIF1RX5", "AIF1RX5" }, \
+	{ name, "AIF1RX6", "AIF1RX6" }, \
+	{ name, "AIF2RX1", "AIF2RX1" }, \
+	{ name, "AIF2RX2", "AIF2RX2" }, \
+	{ name, "AIF2RX3", "AIF2RX3" }, \
+	{ name, "AIF2RX4", "AIF2RX4" }, \
+	{ name, "AIF2RX5", "AIF2RX5" }, \
+	{ name, "AIF2RX6", "AIF2RX6" }, \
+	{ name, "AIF3RX1", "AIF3RX1" }, \
+	{ name, "AIF3RX2", "AIF3RX2" }, \
+	{ name, "SLIMRX1", "SLIMRX1" }, \
+	{ name, "SLIMRX2", "SLIMRX2" }, \
+	{ name, "SLIMRX3", "SLIMRX3" }, \
+	{ name, "SLIMRX4", "SLIMRX4" }, \
+	{ 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, "ASRC1L", "ASRC1L" }, \
+	{ name, "ASRC1R", "ASRC1R" }, \
+	{ name, "ASRC2L", "ASRC2L" }, \
+	{ name, "ASRC2R", "ASRC2R" }, \
+	{ name, "ISRC1DEC1", "ISRC1DEC1" }, \
+	{ name, "ISRC1DEC2", "ISRC1DEC2" }, \
+	{ name, "ISRC1DEC3", "ISRC1DEC3" }, \
+	{ name, "ISRC1DEC4", "ISRC1DEC4" }, \
+	{ name, "ISRC1INT1", "ISRC1INT1" }, \
+	{ name, "ISRC1INT2", "ISRC1INT2" }, \
+	{ name, "ISRC1INT3", "ISRC1INT3" }, \
+	{ name, "ISRC1INT4", "ISRC1INT4" }, \
+	{ name, "ISRC2DEC1", "ISRC2DEC1" }, \
+	{ name, "ISRC2DEC2", "ISRC2DEC2" }, \
+	{ name, "ISRC2INT1", "ISRC2INT1" }, \
+	{ name, "ISRC2INT2", "ISRC2INT2" }
+
+static const struct snd_soc_dapm_route wm8998_dapm_routes[] = {
+	{ "AIF2 Capture", NULL, "DBVDD2" },
+	{ "AIF2 Playback", NULL, "DBVDD2" },
+
+	{ "AIF3 Capture", NULL, "DBVDD3" },
+	{ "AIF3 Playback", NULL, "DBVDD3" },
+
+	{ "OUT1L", NULL, "CPVDD" },
+	{ "OUT1R", NULL, "CPVDD" },
+	{ "OUT2L", NULL, "CPVDD" },
+	{ "OUT2R", NULL, "CPVDD" },
+	{ "OUT3",  NULL, "CPVDD" },
+
+	{ "OUT4L", NULL, "SPKVDDL" },
+	{ "OUT4R", NULL, "SPKVDDR" },
+
+	{ "OUT1L", NULL, "SYSCLK" },
+	{ "OUT1R", NULL, "SYSCLK" },
+	{ "OUT2L", NULL, "SYSCLK" },
+	{ "OUT2R", NULL, "SYSCLK" },
+	{ "OUT3",  NULL, "SYSCLK" },
+	{ "OUT4L", NULL, "SYSCLK" },
+	{ "OUT4R", NULL, "SYSCLK" },
+	{ "OUT5L", NULL, "SYSCLK" },
+	{ "OUT5R", NULL, "SYSCLK" },
+
+	{ "IN1AL", NULL, "SYSCLK" },
+	{ "IN1AR", NULL, "SYSCLK" },
+	{ "IN1BL", NULL, "SYSCLK" },
+	{ "IN1BR", NULL, "SYSCLK" },
+	{ "IN2A", NULL, "SYSCLK" },
+	{ "IN2B", NULL, "SYSCLK" },
+
+	{ "SPD1", NULL, "SYSCLK" },
+	{ "SPD1", NULL, "SPD1TX1" },
+	{ "SPD1", NULL, "SPD1TX2" },
+
+	{ "MICBIAS1", NULL, "MICVDD" },
+	{ "MICBIAS2", NULL, "MICVDD" },
+	{ "MICBIAS3", NULL, "MICVDD" },
+
+	{ "Tone Generator 1", NULL, "SYSCLK" },
+	{ "Tone Generator 2", NULL, "SYSCLK" },
+
+	{ "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" },
+
+	{ "AIF1RX1", NULL, "AIF1 Playback" },
+	{ "AIF1RX2", NULL, "AIF1 Playback" },
+	{ "AIF1RX3", NULL, "AIF1 Playback" },
+	{ "AIF1RX4", NULL, "AIF1 Playback" },
+	{ "AIF1RX5", NULL, "AIF1 Playback" },
+	{ "AIF1RX6", NULL, "AIF1 Playback" },
+
+	{ "AIF2 Capture", NULL, "AIF2TX1" },
+	{ "AIF2 Capture", NULL, "AIF2TX2" },
+	{ "AIF2 Capture", NULL, "AIF2TX3" },
+	{ "AIF2 Capture", NULL, "AIF2TX4" },
+	{ "AIF2 Capture", NULL, "AIF2TX5" },
+	{ "AIF2 Capture", NULL, "AIF2TX6" },
+
+	{ "AIF2RX1", NULL, "AIF2 Playback" },
+	{ "AIF2RX2", NULL, "AIF2 Playback" },
+	{ "AIF2RX3", NULL, "AIF2 Playback" },
+	{ "AIF2RX4", NULL, "AIF2 Playback" },
+	{ "AIF2RX5", NULL, "AIF2 Playback" },
+	{ "AIF2RX6", NULL, "AIF2 Playback" },
+
+	{ "AIF3 Capture", NULL, "AIF3TX1" },
+	{ "AIF3 Capture", NULL, "AIF3TX2" },
+
+	{ "AIF3RX1", NULL, "AIF3 Playback" },
+	{ "AIF3RX2", NULL, "AIF3 Playback" },
+
+	{ "Slim1 Capture", NULL, "SLIMTX1" },
+	{ "Slim1 Capture", NULL, "SLIMTX2" },
+	{ "Slim1 Capture", NULL, "SLIMTX3" },
+	{ "Slim1 Capture", NULL, "SLIMTX4" },
+
+	{ "Slim2 Capture", NULL, "SLIMTX5" },
+	{ "Slim2 Capture", NULL, "SLIMTX6" },
+
+	{ "SLIMRX1", NULL, "Slim1 Playback" },
+	{ "SLIMRX2", NULL, "Slim1 Playback" },
+
+	{ "SLIMRX3", NULL, "Slim2 Playback" },
+	{ "SLIMRX4", NULL, "Slim2 Playback" },
+
+	{ "AIF1 Playback", NULL, "SYSCLK" },
+	{ "AIF2 Playback", NULL, "SYSCLK" },
+	{ "AIF3 Playback", NULL, "SYSCLK" },
+	{ "Slim1 Playback", NULL, "SYSCLK" },
+	{ "Slim2 Playback", NULL, "SYSCLK" },
+
+	{ "AIF1 Capture", NULL, "SYSCLK" },
+	{ "AIF2 Capture", NULL, "SYSCLK" },
+	{ "AIF3 Capture", NULL, "SYSCLK" },
+	{ "Slim1 Capture", NULL, "SYSCLK" },
+	{ "Slim2 Capture", NULL, "SYSCLK" },
+
+	{ "IN1L Mux", "A", "IN1AL" },
+	{ "IN1R Mux", "A", "IN1AR" },
+	{ "IN1L Mux", "B", "IN1BL" },
+	{ "IN1R Mux", "B", "IN1BR" },
+
+	{ "IN2 Mux", "A", "IN2A" },
+	{ "IN2 Mux", "B", "IN2B" },
+
+	{ "IN1L PGA", NULL, "IN1L Mux" },
+	{ "IN1R PGA", NULL, "IN1R Mux" },
+	{ "IN2 PGA",  NULL, "IN2 Mux" },
+
+	ARIZONA_MIXER_ROUTES("OUT1L", "HPOUTL"),
+	ARIZONA_MIXER_ROUTES("OUT1R", "HPOUTR"),
+	ARIZONA_MIXER_ROUTES("OUT2L", "LINEOUTL"),
+	ARIZONA_MIXER_ROUTES("OUT2R", "LINEOUTR"),
+	ARIZONA_MIXER_ROUTES("OUT3",  "EPOUT"),
+
+	ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUTL"),
+	ARIZONA_MIXER_ROUTES("OUT4R", "SPKOUTR"),
+	ARIZONA_MIXER_ROUTES("OUT5L", "SPKDATL"),
+	ARIZONA_MIXER_ROUTES("OUT5R", "SPKDATR"),
+
+	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("AIF2TX1", "AIF2TX1"),
+	ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"),
+	ARIZONA_MIXER_ROUTES("AIF2TX3", "AIF2TX3"),
+	ARIZONA_MIXER_ROUTES("AIF2TX4", "AIF2TX4"),
+	ARIZONA_MIXER_ROUTES("AIF2TX5", "AIF2TX5"),
+	ARIZONA_MIXER_ROUTES("AIF2TX6", "AIF2TX6"),
+
+	ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"),
+	ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"),
+
+	ARIZONA_MUX_ROUTES("SLIMTX1", "SLIMTX1"),
+	ARIZONA_MUX_ROUTES("SLIMTX2", "SLIMTX2"),
+	ARIZONA_MUX_ROUTES("SLIMTX3", "SLIMTX3"),
+	ARIZONA_MUX_ROUTES("SLIMTX4", "SLIMTX4"),
+	ARIZONA_MUX_ROUTES("SLIMTX5", "SLIMTX5"),
+	ARIZONA_MUX_ROUTES("SLIMTX6", "SLIMTX6"),
+
+	ARIZONA_MUX_ROUTES("SPD1TX1", "SPDIFTX1"),
+	ARIZONA_MUX_ROUTES("SPD1TX2", "SPDIFTX2"),
+
+	ARIZONA_MUX_ROUTES("EQ1", "EQ1"),
+	ARIZONA_MUX_ROUTES("EQ2", "EQ2"),
+	ARIZONA_MUX_ROUTES("EQ3", "EQ3"),
+	ARIZONA_MUX_ROUTES("EQ4", "EQ4"),
+
+	ARIZONA_MUX_ROUTES("DRC1L", "DRC1L"),
+	ARIZONA_MUX_ROUTES("DRC1R", "DRC1R"),
+
+	ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"),
+	ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"),
+	ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"),
+	ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"),
+
+	ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"),
+	ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"),
+	ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"),
+	ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"),
+
+	ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"),
+	ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"),
+	ARIZONA_MUX_ROUTES("ISRC1INT3", "ISRC1INT3"),
+	ARIZONA_MUX_ROUTES("ISRC1INT4", "ISRC1INT4"),
+
+	ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"),
+	ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"),
+	ARIZONA_MUX_ROUTES("ISRC1DEC3", "ISRC1DEC3"),
+	ARIZONA_MUX_ROUTES("ISRC1DEC4", "ISRC1DEC4"),
+
+	ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"),
+	ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"),
+
+	ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"),
+	ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"),
+
+	{ "AEC1 Loopback", "HPOUTL", "OUT1L" },
+	{ "AEC1 Loopback", "HPOUTR", "OUT1R" },
+	{ "AEC2 Loopback", "HPOUTL", "OUT1L" },
+	{ "AEC2 Loopback", "HPOUTR", "OUT1R" },
+	{ "HPOUTL", NULL, "OUT1L" },
+	{ "HPOUTR", NULL, "OUT1R" },
+
+	{ "AEC1 Loopback", "LINEOUTL", "OUT2L" },
+	{ "AEC1 Loopback", "LINEOUTR", "OUT2R" },
+	{ "AEC2 Loopback", "LINEOUTL", "OUT2L" },
+	{ "AEC2 Loopback", "LINEOUTR", "OUT2R" },
+	{ "LINEOUTL", NULL, "OUT2L" },
+	{ "LINEOUTR", NULL, "OUT2R" },
+
+	{ "AEC1 Loopback", "EPOUT", "OUT3" },
+	{ "AEC2 Loopback", "EPOUT", "OUT3" },
+	{ "EPOUT", NULL, "OUT3" },
+
+	{ "AEC1 Loopback", "SPKOUTL", "OUT4L" },
+	{ "AEC2 Loopback", "SPKOUTL", "OUT4L" },
+	{ "SPKOUTLN", NULL, "OUT4L" },
+	{ "SPKOUTLP", NULL, "OUT4L" },
+
+	{ "AEC1 Loopback", "SPKOUTR", "OUT4R" },
+	{ "AEC2 Loopback", "SPKOUTR", "OUT4R" },
+	{ "SPKOUTRN", NULL, "OUT4R" },
+	{ "SPKOUTRP", NULL, "OUT4R" },
+
+	{ "SPDIF", NULL, "SPD1" },
+
+	{ "AEC1 Loopback", "SPKDATL", "OUT5L" },
+	{ "AEC1 Loopback", "SPKDATR", "OUT5R" },
+	{ "AEC2 Loopback", "SPKDATL", "OUT5L" },
+	{ "AEC2 Loopback", "SPKDATR", "OUT5R" },
+	{ "SPKDATL", NULL, "OUT5L" },
+	{ "SPKDATR", NULL, "OUT5R" },
+
+	{ "MICSUPP", NULL, "SYSCLK" },
+
+	{ "DRC1 Signal Activity", NULL, "DRC1L" },
+	{ "DRC1 Signal Activity", NULL, "DRC1R" },
+};
+
+#define WM8998_RATES SNDRV_PCM_RATE_8000_192000
+
+#define WM8998_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 wm8998_dai[] = {
+	{
+		.name = "wm8998-aif1",
+		.id = 1,
+		.base = ARIZONA_AIF1_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF1 Playback",
+			.channels_min = 1,
+			.channels_max = 6,
+			.rates = WM8998_RATES,
+			.formats = WM8998_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF1 Capture",
+			 .channels_min = 1,
+			 .channels_max = 6,
+			 .rates = WM8998_RATES,
+			 .formats = WM8998_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+		.symmetric_samplebits = 1,
+	},
+	{
+		.name = "wm8998-aif2",
+		.id = 2,
+		.base = ARIZONA_AIF2_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF2 Playback",
+			.channels_min = 1,
+			.channels_max = 6,
+			.rates = WM8998_RATES,
+			.formats = WM8998_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF2 Capture",
+			 .channels_min = 1,
+			 .channels_max = 6,
+			 .rates = WM8998_RATES,
+			 .formats = WM8998_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+		.symmetric_samplebits = 1,
+	},
+	{
+		.name = "wm8998-aif3",
+		.id = 3,
+		.base = ARIZONA_AIF3_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF3 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM8998_RATES,
+			.formats = WM8998_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF3 Capture",
+			 .channels_min = 1,
+			 .channels_max = 2,
+			 .rates = WM8998_RATES,
+			 .formats = WM8998_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+		.symmetric_samplebits = 1,
+	},
+	{
+		.name = "wm8998-slim1",
+		.id = 4,
+		.playback = {
+			.stream_name = "Slim1 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM8998_RATES,
+			.formats = WM8998_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "Slim1 Capture",
+			 .channels_min = 1,
+			 .channels_max = 4,
+			 .rates = WM8998_RATES,
+			 .formats = WM8998_FORMATS,
+		 },
+		.ops = &arizona_simple_dai_ops,
+	},
+	{
+		.name = "wm8998-slim2",
+		.id = 5,
+		.playback = {
+			.stream_name = "Slim2 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM8998_RATES,
+			.formats = WM8998_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "Slim2 Capture",
+			 .channels_min = 1,
+			 .channels_max = 2,
+			 .rates = WM8998_RATES,
+			 .formats = WM8998_FORMATS,
+		 },
+		.ops = &arizona_simple_dai_ops,
+	},
+};
+
+static int wm8998_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
+			  unsigned int Fref, unsigned int Fout)
+{
+	struct wm8998_priv *wm8998 = snd_soc_codec_get_drvdata(codec);
+
+	switch (fll_id) {
+	case WM8998_FLL1:
+		return arizona_set_fll(&wm8998->fll[0], source, Fref, Fout);
+	case WM8998_FLL2:
+		return arizona_set_fll(&wm8998->fll[1], source, Fref, Fout);
+	case WM8998_FLL1_REFCLK:
+		return arizona_set_fll_refclk(&wm8998->fll[0], source, Fref,
+					      Fout);
+	case WM8998_FLL2_REFCLK:
+		return arizona_set_fll_refclk(&wm8998->fll[1], source, Fref,
+					      Fout);
+	default:
+		return -EINVAL;
+	}
+}
+
+static int wm8998_codec_probe(struct snd_soc_codec *codec)
+{
+	struct wm8998_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+
+	priv->core.arizona->dapm = dapm;
+
+	arizona_init_spk(codec);
+	arizona_init_gpio(codec);
+
+	snd_soc_dapm_disable_pin(dapm, "HAPTICS");
+
+	return 0;
+}
+
+static int wm8998_codec_remove(struct snd_soc_codec *codec)
+{
+	struct wm8998_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	priv->core.arizona->dapm = NULL;
+
+	return 0;
+}
+
+#define WM8998_DIG_VU 0x0200
+
+static unsigned int wm8998_digital_vu[] = {
+	ARIZONA_DAC_DIGITAL_VOLUME_1L,
+	ARIZONA_DAC_DIGITAL_VOLUME_1R,
+	ARIZONA_DAC_DIGITAL_VOLUME_2L,
+	ARIZONA_DAC_DIGITAL_VOLUME_2R,
+	ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	ARIZONA_DAC_DIGITAL_VOLUME_4L,
+	ARIZONA_DAC_DIGITAL_VOLUME_4R,
+	ARIZONA_DAC_DIGITAL_VOLUME_5L,
+	ARIZONA_DAC_DIGITAL_VOLUME_5R,
+};
+
+static struct regmap *wm8998_get_regmap(struct device *dev)
+{
+	struct wm8998_priv *priv = dev_get_drvdata(dev);
+
+	return priv->core.arizona->regmap;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_wm8998 = {
+	.probe = wm8998_codec_probe,
+	.remove = wm8998_codec_remove,
+	.get_regmap = wm8998_get_regmap,
+
+	.idle_bias_off = true,
+
+	.set_sysclk = arizona_set_sysclk,
+	.set_pll = wm8998_set_fll,
+
+	.controls = wm8998_snd_controls,
+	.num_controls = ARRAY_SIZE(wm8998_snd_controls),
+	.dapm_widgets = wm8998_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm8998_dapm_widgets),
+	.dapm_routes = wm8998_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(wm8998_dapm_routes),
+};
+
+static int wm8998_probe(struct platform_device *pdev)
+{
+	struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
+	struct wm8998_priv *wm8998;
+	int i;
+
+	wm8998 = devm_kzalloc(&pdev->dev, sizeof(struct wm8998_priv),
+			      GFP_KERNEL);
+	if (!wm8998)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, wm8998);
+
+	wm8998->core.arizona = arizona;
+	wm8998->core.num_inputs = 3;	/* IN1L, IN1R, IN2 */
+
+	for (i = 0; i < ARRAY_SIZE(wm8998->fll); i++)
+		wm8998->fll[i].vco_mult = 1;
+
+	arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1,
+			 ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK,
+			 &wm8998->fll[0]);
+	arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1,
+			 ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK,
+			 &wm8998->fll[1]);
+
+	for (i = 0; i < ARRAY_SIZE(wm8998_dai); i++)
+		arizona_init_dai(&wm8998->core, i);
+
+	/* Latch volume update bits */
+	for (i = 0; i < ARRAY_SIZE(wm8998_digital_vu); i++)
+		regmap_update_bits(arizona->regmap, wm8998_digital_vu[i],
+				   WM8998_DIG_VU, WM8998_DIG_VU);
+
+	pm_runtime_enable(&pdev->dev);
+	pm_runtime_idle(&pdev->dev);
+
+	return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8998,
+				      wm8998_dai, ARRAY_SIZE(wm8998_dai));
+}
+
+static int wm8998_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	pm_runtime_disable(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver wm8998_codec_driver = {
+	.driver = {
+		.name = "wm8998-codec",
+	},
+	.probe = wm8998_probe,
+	.remove = wm8998_remove,
+};
+
+module_platform_driver(wm8998_codec_driver);
+
+MODULE_DESCRIPTION("ASoC WM8998 driver");
+MODULE_AUTHOR("Richard Fitzgerald <rf@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:wm8998-codec");
diff --git a/sound/soc/codecs/wm8998.h b/sound/soc/codecs/wm8998.h
new file mode 100644
index 000000000000..1e8647252162
--- /dev/null
+++ b/sound/soc/codecs/wm8998.h
@@ -0,0 +1,23 @@
+/*
+ * wm8998.h -- ALSA SoC Audio driver for WM8998 codecs
+ *
+ * Copyright 2015 Cirrus Logic, Inc.
+ *
+ * Author: Richard Fitzgerald <rf@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 _WM8998_H
+#define _WM8998_H
+
+#include "arizona.h"
+
+#define WM8998_FLL1        1
+#define WM8998_FLL2        2
+#define WM8998_FLL1_REFCLK 3
+#define WM8998_FLL2_REFCLK 4
+
+#endif
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 7d45d98a861f..4495a40a9468 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -80,12 +80,13 @@ struct davinci_mcasp {
 
 	/* McASP specific data */
 	int	tdm_slots;
+	u32	tdm_mask[2];
+	int	slot_width;
 	u8	op_mode;
 	u8	num_serializer;
 	u8	*serial_dir;
 	u8	version;
 	u8	bclk_div;
-	u16	bclk_lrclk_ratio;
 	int	streams;
 	u32	irq_request[2];
 	int	dma_request[2];
@@ -556,8 +557,21 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
 			mcasp->bclk_div = div;
 		break;
 
-	case 2:		/* BCLK/LRCLK ratio */
-		mcasp->bclk_lrclk_ratio = div;
+	case 2:	/*
+		 * BCLK/LRCLK ratio descries how many bit-clock cycles
+		 * fit into one frame. The clock ratio is given for a
+		 * full period of data (for I2S format both left and
+		 * right channels), so it has to be divided by number
+		 * of tdm-slots (for I2S - divided by 2).
+		 * Instead of storing this ratio, we calculate a new
+		 * tdm_slot width by dividing the the ratio by the
+		 * number of configured tdm slots.
+		 */
+		mcasp->slot_width = div / mcasp->tdm_slots;
+		if (div % mcasp->tdm_slots)
+			dev_warn(mcasp->dev,
+				 "%s(): BCLK/LRCLK %d is not divisible by %d tdm slots",
+				 __func__, div, mcasp->tdm_slots);
 		break;
 
 	default:
@@ -596,12 +610,92 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	return 0;
 }
 
+/* All serializers must have equal number of channels */
+static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp, int stream,
+				       int serializers)
+{
+	struct snd_pcm_hw_constraint_list *cl = &mcasp->chconstr[stream];
+	unsigned int *list = (unsigned int *) cl->list;
+	int slots = mcasp->tdm_slots;
+	int i, count = 0;
+
+	if (mcasp->tdm_mask[stream])
+		slots = hweight32(mcasp->tdm_mask[stream]);
+
+	for (i = 2; i <= slots; i++)
+		list[count++] = i;
+
+	for (i = 2; i <= serializers; i++)
+		list[count++] = i*slots;
+
+	cl->count = count;
+
+	return 0;
+}
+
+static int davinci_mcasp_set_ch_constraints(struct davinci_mcasp *mcasp)
+{
+	int rx_serializers = 0, tx_serializers = 0, ret, i;
+
+	for (i = 0; i < mcasp->num_serializer; i++)
+		if (mcasp->serial_dir[i] == TX_MODE)
+			tx_serializers++;
+		else if (mcasp->serial_dir[i] == RX_MODE)
+			rx_serializers++;
+
+	ret = davinci_mcasp_ch_constraint(mcasp, SNDRV_PCM_STREAM_PLAYBACK,
+					  tx_serializers);
+	if (ret)
+		return ret;
+
+	ret = davinci_mcasp_ch_constraint(mcasp, SNDRV_PCM_STREAM_CAPTURE,
+					  rx_serializers);
+
+	return ret;
+}
+
+
+static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai,
+				      unsigned int tx_mask,
+				      unsigned int rx_mask,
+				      int slots, int slot_width)
+{
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+
+	dev_dbg(mcasp->dev,
+		 "%s() tx_mask 0x%08x rx_mask 0x%08x slots %d width %d\n",
+		 __func__, tx_mask, rx_mask, slots, slot_width);
+
+	if (tx_mask >= (1<<slots) || rx_mask >= (1<<slots)) {
+		dev_err(mcasp->dev,
+			"Bad tdm mask tx: 0x%08x rx: 0x%08x slots %d\n",
+			tx_mask, rx_mask, slots);
+		return -EINVAL;
+	}
+
+	if (slot_width &&
+	    (slot_width < 8 || slot_width > 32 || slot_width % 4 != 0)) {
+		dev_err(mcasp->dev, "%s: Unsupported slot_width %d\n",
+			__func__, slot_width);
+		return -EINVAL;
+	}
+
+	mcasp->tdm_slots = slots;
+	mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = rx_mask;
+	mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = tx_mask;
+	mcasp->slot_width = slot_width;
+
+	return davinci_mcasp_set_ch_constraints(mcasp);
+}
+
 static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
-				       int word_length)
+				       int sample_width)
 {
 	u32 fmt;
-	u32 tx_rotate = (word_length / 4) & 0x7;
-	u32 mask = (1ULL << word_length) - 1;
+	u32 tx_rotate = (sample_width / 4) & 0x7;
+	u32 mask = (1ULL << sample_width) - 1;
+	u32 slot_width = sample_width;
+
 	/*
 	 * For captured data we should not rotate, inversion and masking is
 	 * enoguh to get the data to the right position:
@@ -614,28 +708,23 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
 	u32 rx_rotate = 0;
 
 	/*
-	 * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv()
-	 * callback, take it into account here. That allows us to for example
-	 * send 32 bits per channel to the codec, while only 16 of them carry
-	 * audio payload.
-	 * The clock ratio is given for a full period of data (for I2S format
-	 * both left and right channels), so it has to be divided by number of
-	 * tdm-slots (for I2S - divided by 2).
+	 * Setting the tdm slot width either with set_clkdiv() or
+	 * set_tdm_slot() allows us to for example send 32 bits per
+	 * channel to the codec, while only 16 of them carry audio
+	 * payload.
 	 */
-	if (mcasp->bclk_lrclk_ratio) {
-		u32 slot_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots;
-
+	if (mcasp->slot_width) {
 		/*
-		 * When we have more bclk then it is needed for the data, we
-		 * need to use the rotation to move the received samples to have
-		 * correct alignment.
+		 * When we have more bclk then it is needed for the
+		 * data, we need to use the rotation to move the
+		 * received samples to have correct alignment.
 		 */
-		rx_rotate = (slot_length - word_length) / 4;
-		word_length = slot_length;
+		slot_width = mcasp->slot_width;
+		rx_rotate = (slot_width - sample_width) / 4;
 	}
 
 	/* mapping of the XSSZ bit-field as described in the datasheet */
-	fmt = (word_length >> 1) - 1;
+	fmt = (slot_width >> 1) - 1;
 
 	if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) {
 		mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt),
@@ -776,33 +865,50 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream,
 
 	/*
 	 * If more than one serializer is needed, then use them with
-	 * their specified tdm_slots count. Otherwise, one serializer
-	 * can cope with the transaction using as many slots as channels
-	 * in the stream, requires channels symmetry
+	 * all the specified tdm_slots. Otherwise, one serializer can
+	 * cope with the transaction using just as many slots as there
+	 * are channels in the stream.
 	 */
-	active_serializers = (channels + total_slots - 1) / total_slots;
-	if (active_serializers == 1)
-		active_slots = channels;
-	else
-		active_slots = total_slots;
-
-	for (i = 0; i < active_slots; i++)
-		mask |= (1 << i);
+	if (mcasp->tdm_mask[stream]) {
+		active_slots = hweight32(mcasp->tdm_mask[stream]);
+		active_serializers = (channels + active_slots - 1) /
+			active_slots;
+		if (active_serializers == 1) {
+			active_slots = channels;
+			for (i = 0; i < total_slots; i++) {
+				if ((1 << i) & mcasp->tdm_mask[stream]) {
+					mask |= (1 << i);
+					if (--active_slots <= 0)
+						break;
+				}
+			}
+		}
+	} else {
+		active_serializers = (channels + total_slots - 1) / total_slots;
+		if (active_serializers == 1)
+			active_slots = channels;
+		else
+			active_slots = total_slots;
 
+		for (i = 0; i < active_slots; i++)
+			mask |= (1 << i);
+	}
 	mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC);
 
 	if (!mcasp->dat_port)
 		busel = TXSEL;
 
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
-	mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
-	mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
-		       FSXMOD(total_slots), FSXMOD(0x1FF));
-
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
-	mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
-	mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
-		       FSRMOD(total_slots), FSRMOD(0x1FF));
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
+			       FSXMOD(total_slots), FSXMOD(0x1FF));
+	} else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
+			       FSRMOD(total_slots), FSRMOD(0x1FF));
+	}
 
 	return 0;
 }
@@ -922,6 +1028,9 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 		int sbits = params_width(params);
 		int ppm, div;
 
+		if (mcasp->slot_width)
+			sbits = mcasp->slot_width;
+
 		div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*slots,
 						 &ppm);
 		if (ppm)
@@ -1027,6 +1136,9 @@ static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params,
 	struct snd_interval range;
 	int i;
 
+	if (rd->mcasp->slot_width)
+		sbits = rd->mcasp->slot_width;
+
 	snd_interval_any(&range);
 	range.empty = 1;
 
@@ -1069,10 +1181,14 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params,
 
 	for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) {
 		if (snd_mask_test(fmt, i)) {
-			uint bclk_freq = snd_pcm_format_width(i)*slots*rate;
+			uint sbits = snd_pcm_format_width(i);
 			int ppm;
 
-			davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm);
+			if (rd->mcasp->slot_width)
+				sbits = rd->mcasp->slot_width;
+
+			davinci_mcasp_calc_clk_div(rd->mcasp, sbits*slots*rate,
+						   &ppm);
 			if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) {
 				snd_mask_set(&nfmt, i);
 				count++;
@@ -1094,6 +1210,10 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 					&mcasp->ruledata[substream->stream];
 	u32 max_channels = 0;
 	int i, dir;
+	int tdm_slots = mcasp->tdm_slots;
+
+	if (mcasp->tdm_mask[substream->stream])
+		tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]);
 
 	mcasp->substreams[substream->stream] = substream;
 
@@ -1114,7 +1234,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 			max_channels++;
 	}
 	ruledata->serializers = max_channels;
-	max_channels *= mcasp->tdm_slots;
+	max_channels *= tdm_slots;
 	/*
 	 * If the already active stream has less channels than the calculated
 	 * limnit based on the seirializers * tdm_slots, we need to use that as
@@ -1124,15 +1244,25 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 	 */
 	if (mcasp->channels && mcasp->channels < max_channels)
 		max_channels = mcasp->channels;
+	/*
+	 * But we can always allow channels upto the amount of
+	 * the available tdm_slots.
+	 */
+	if (max_channels < tdm_slots)
+		max_channels = tdm_slots;
 
 	snd_pcm_hw_constraint_minmax(substream->runtime,
 				     SNDRV_PCM_HW_PARAM_CHANNELS,
 				     2, max_channels);
 
-	if (mcasp->chconstr[substream->stream].count)
-		snd_pcm_hw_constraint_list(substream->runtime,
-					   0, SNDRV_PCM_HW_PARAM_CHANNELS,
-					   &mcasp->chconstr[substream->stream]);
+	snd_pcm_hw_constraint_list(substream->runtime,
+				   0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &mcasp->chconstr[substream->stream]);
+
+	if (mcasp->slot_width)
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+					     8, mcasp->slot_width);
 
 	/*
 	 * If we rely on implicit BCLK divider setting we should
@@ -1184,6 +1314,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
 	.set_fmt	= davinci_mcasp_set_dai_fmt,
 	.set_clkdiv	= davinci_mcasp_set_clkdiv,
 	.set_sysclk	= davinci_mcasp_set_sysclk,
+	.set_tdm_slot	= davinci_mcasp_set_tdm_slot,
 };
 
 static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai)
@@ -1514,59 +1645,6 @@ nodata:
 	return  pdata;
 }
 
-/* All serializers must have equal number of channels */
-static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp,
-				       struct snd_pcm_hw_constraint_list *cl,
-				       int serializers)
-{
-	unsigned int *list;
-	int i, count = 0;
-
-	if (serializers <= 1)
-		return 0;
-
-	list = devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
-			    (mcasp->tdm_slots + serializers - 2),
-			    GFP_KERNEL);
-	if (!list)
-		return -ENOMEM;
-
-	for (i = 2; i <= mcasp->tdm_slots; i++)
-		list[count++] = i;
-
-	for (i = 2; i <= serializers; i++)
-		list[count++] = i*mcasp->tdm_slots;
-
-	cl->count = count;
-	cl->list = list;
-
-	return 0;
-}
-
-
-static int davinci_mcasp_init_ch_constraints(struct davinci_mcasp *mcasp)
-{
-	int rx_serializers = 0, tx_serializers = 0, ret, i;
-
-	for (i = 0; i < mcasp->num_serializer; i++)
-		if (mcasp->serial_dir[i] == TX_MODE)
-			tx_serializers++;
-		else if (mcasp->serial_dir[i] == RX_MODE)
-			rx_serializers++;
-
-	ret = davinci_mcasp_ch_constraint(mcasp, &mcasp->chconstr[
-						  SNDRV_PCM_STREAM_PLAYBACK],
-					  tx_serializers);
-	if (ret)
-		return ret;
-
-	ret = davinci_mcasp_ch_constraint(mcasp, &mcasp->chconstr[
-						  SNDRV_PCM_STREAM_CAPTURE],
-					  rx_serializers);
-
-	return ret;
-}
-
 enum {
 	PCM_EDMA,
 	PCM_SDMA,
@@ -1783,7 +1861,28 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 		mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE;
 	}
 
-	ret = davinci_mcasp_init_ch_constraints(mcasp);
+	/* Allocate memory for long enough list for all possible
+	 * scenarios. Maximum number tdm slots is 32 and there cannot
+	 * be more serializers than given in the configuration.  The
+	 * serializer directions could be taken into account, but it
+	 * would make code much more complex and save only couple of
+	 * bytes.
+	 */
+	mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list =
+		devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
+			     (32 + mcasp->num_serializer - 2),
+			     GFP_KERNEL);
+
+	mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list =
+		devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
+			     (32 + mcasp->num_serializer - 2),
+			     GFP_KERNEL);
+
+	if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list ||
+	    !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list)
+		return -ENOMEM;
+
+	ret = davinci_mcasp_set_ch_constraints(mcasp);
 	if (ret)
 		goto err;
 
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index ba34252b7bba..6e6a70c5c2bd 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -282,23 +282,25 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream,
 
 	config->sample_rate = params_rate(params);
 
-	if (dev->i2s_clk_cfg) {
-		ret = dev->i2s_clk_cfg(config);
-		if (ret < 0) {
-			dev_err(dev->dev, "runtime audio clk config fail\n");
-			return ret;
-		}
-	} else {
-		u32 bitclk = config->sample_rate * config->data_width * 2;
-
-		ret = clk_set_rate(dev->clk, bitclk);
-		if (ret) {
-			dev_err(dev->dev, "Can't set I2S clock rate: %d\n",
-				ret);
-			return ret;
+	if (dev->capability & DW_I2S_MASTER) {
+		if (dev->i2s_clk_cfg) {
+			ret = dev->i2s_clk_cfg(config);
+			if (ret < 0) {
+				dev_err(dev->dev, "runtime audio clk config fail\n");
+				return ret;
+			}
+		} else {
+			u32 bitclk = config->sample_rate *
+					config->data_width * 2;
+
+			ret = clk_set_rate(dev->clk, bitclk);
+			if (ret) {
+				dev_err(dev->dev, "Can't set I2S clock rate: %d\n",
+					ret);
+				return ret;
+			}
 		}
 	}
-
 	return 0;
 }
 
@@ -348,12 +350,43 @@ static int dw_i2s_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
+	int ret = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		if (dev->capability & DW_I2S_SLAVE)
+			ret = 0;
+		else
+			ret = -EINVAL;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		if (dev->capability & DW_I2S_MASTER)
+			ret = 0;
+		else
+			ret = -EINVAL;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+	case SND_SOC_DAIFMT_CBS_CFM:
+		ret = -EINVAL;
+		break;
+	default:
+		dev_dbg(dev->dev, "dwc : Invalid master/slave format\n");
+		ret = -EINVAL;
+		break;
+	}
+	return ret;
+}
+
 static struct snd_soc_dai_ops dw_i2s_dai_ops = {
 	.startup	= dw_i2s_startup,
 	.shutdown	= dw_i2s_shutdown,
 	.hw_params	= dw_i2s_hw_params,
 	.prepare	= dw_i2s_prepare,
 	.trigger	= dw_i2s_trigger,
+	.set_fmt	= dw_i2s_set_fmt,
 };
 
 static const struct snd_soc_component_driver dw_i2s_component = {
@@ -366,7 +399,8 @@ static int dw_i2s_suspend(struct snd_soc_dai *dai)
 {
 	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
 
-	clk_disable(dev->clk);
+	if (dev->capability & DW_I2S_MASTER)
+		clk_disable(dev->clk);
 	return 0;
 }
 
@@ -374,7 +408,8 @@ static int dw_i2s_resume(struct snd_soc_dai *dai)
 {
 	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
 
-	clk_enable(dev->clk);
+	if (dev->capability & DW_I2S_MASTER)
+		clk_enable(dev->clk);
 	return 0;
 }
 
@@ -452,6 +487,14 @@ static int dw_configure_dai(struct dw_i2s_dev *dev,
 		dw_i2s_dai->capture.rates = rates;
 	}
 
+	if (COMP1_MODE_EN(comp1)) {
+		dev_dbg(dev->dev, "designware: i2s master mode supported\n");
+		dev->capability |= DW_I2S_MASTER;
+	} else {
+		dev_dbg(dev->dev, "designware: i2s slave mode supported\n");
+		dev->capability |= DW_I2S_SLAVE;
+	}
+
 	return 0;
 }
 
@@ -538,6 +581,7 @@ static int dw_i2s_probe(struct platform_device *pdev)
 	struct resource *res;
 	int ret;
 	struct snd_soc_dai_driver *dw_i2s_dai;
+	const char *clk_id;
 
 	dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL);
 	if (!dev) {
@@ -559,32 +603,35 @@ static int dw_i2s_probe(struct platform_device *pdev)
 		return PTR_ERR(dev->i2s_base);
 
 	dev->dev = &pdev->dev;
+
 	if (pdata) {
+		dev->capability = pdata->cap;
+		clk_id = NULL;
 		ret = dw_configure_dai_by_pd(dev, dw_i2s_dai, res, pdata);
-		if (ret < 0)
-			return ret;
+	} else {
+		clk_id = "i2sclk";
+		ret = dw_configure_dai_by_dt(dev, dw_i2s_dai, res);
+	}
+	if (ret < 0)
+		return ret;
 
-		dev->capability = pdata->cap;
-		dev->i2s_clk_cfg = pdata->i2s_clk_cfg;
-		if (!dev->i2s_clk_cfg) {
-			dev_err(&pdev->dev, "no clock configure method\n");
-			return -ENODEV;
+	if (dev->capability & DW_I2S_MASTER) {
+		if (pdata) {
+			dev->i2s_clk_cfg = pdata->i2s_clk_cfg;
+			if (!dev->i2s_clk_cfg) {
+				dev_err(&pdev->dev, "no clock configure method\n");
+				return -ENODEV;
+			}
 		}
+		dev->clk = devm_clk_get(&pdev->dev, clk_id);
 
-		dev->clk = devm_clk_get(&pdev->dev, NULL);
-	} else {
-		ret = dw_configure_dai_by_dt(dev, dw_i2s_dai, res);
+		if (IS_ERR(dev->clk))
+			return PTR_ERR(dev->clk);
+
+		ret = clk_prepare_enable(dev->clk);
 		if (ret < 0)
 			return ret;
-
-		dev->clk = devm_clk_get(&pdev->dev, "i2sclk");
 	}
-	if (IS_ERR(dev->clk))
-		return PTR_ERR(dev->clk);
-
-	ret = clk_prepare_enable(dev->clk);
-	if (ret < 0)
-		return ret;
 
 	dev_set_drvdata(&pdev->dev, dev);
 	ret = devm_snd_soc_register_component(&pdev->dev, &dw_i2s_component,
@@ -606,7 +653,8 @@ static int dw_i2s_probe(struct platform_device *pdev)
 	return 0;
 
 err_clk_disable:
-	clk_disable_unprepare(dev->clk);
+	if (dev->capability & DW_I2S_MASTER)
+		clk_disable_unprepare(dev->clk);
 	return ret;
 }
 
@@ -614,7 +662,8 @@ static int dw_i2s_remove(struct platform_device *pdev)
 {
 	struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
 
-	clk_disable_unprepare(dev->clk);
+	if (dev->capability & DW_I2S_MASTER)
+		clk_disable_unprepare(dev->clk);
 
 	return 0;
 }
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 96f55ae75c71..1b05d1c5d9fd 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -14,6 +14,9 @@
 #include <linux/i2c.h>
 #include <linux/module.h>
 #include <linux/of_platform.h>
+#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
+#include <sound/ac97_codec.h>
+#endif
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
@@ -115,6 +118,11 @@ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
 	SND_SOC_DAPM_MIC("DMIC", NULL),
 };
 
+static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
+{
+	return priv->dai_fmt == SND_SOC_DAIFMT_AC97;
+}
+
 static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
 				   struct snd_pcm_hw_params *params)
 {
@@ -133,7 +141,9 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
 	 * set_bias_level(), bypass the remaining settings in hw_params().
 	 * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS.
 	 */
-	if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM)
+	if ((priv->card.set_bias_level &&
+	     priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) ||
+	    fsl_asoc_card_is_ac97(priv))
 		return 0;
 
 	/* Specific configurations of DAIs starts from here */
@@ -300,7 +310,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np,
 	ext_port--;
 
 	/*
-	 * Use asynchronous mode (6 wires) for all cases.
+	 * Use asynchronous mode (6 wires) for all cases except AC97.
 	 * If only 4 wires are needed, just set SSI into
 	 * synchronous mode and enable 4 PADs in IOMUX.
 	 */
@@ -346,15 +356,30 @@ static int fsl_asoc_card_audmux_init(struct device_node *np,
 			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
 		break;
 	default:
-		return -EINVAL;
+		if (!fsl_asoc_card_is_ac97(priv))
+			return -EINVAL;
+	}
+
+	if (fsl_asoc_card_is_ac97(priv)) {
+		int_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR;
 	}
 
 	/* Asynchronous mode can not be set along with RCLKDIR */
-	ret = imx_audmux_v2_configure_port(int_port, 0,
-					   IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
-	if (ret) {
-		dev_err(dev, "audmux internal port setup failed\n");
-		return ret;
+	if (!fsl_asoc_card_is_ac97(priv)) {
+		unsigned int pdcr =
+				IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port);
+
+		ret = imx_audmux_v2_configure_port(int_port, 0,
+						   pdcr);
+		if (ret) {
+			dev_err(dev, "audmux internal port setup failed\n");
+			return ret;
+		}
 	}
 
 	ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
@@ -364,11 +389,16 @@ static int fsl_asoc_card_audmux_init(struct device_node *np,
 		return ret;
 	}
 
-	ret = imx_audmux_v2_configure_port(ext_port, 0,
-					   IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
-	if (ret) {
-		dev_err(dev, "audmux external port setup failed\n");
-		return ret;
+	if (!fsl_asoc_card_is_ac97(priv)) {
+		unsigned int pdcr =
+				IMX_AUDMUX_V2_PDCR_RXDSEL(int_port);
+
+		ret = imx_audmux_v2_configure_port(ext_port, 0,
+						   pdcr);
+		if (ret) {
+			dev_err(dev, "audmux external port setup failed\n");
+			return ret;
+		}
 	}
 
 	ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
@@ -389,6 +419,23 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
 	struct device *dev = card->dev;
 	int ret;
 
+	if (fsl_asoc_card_is_ac97(priv)) {
+#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
+		struct snd_soc_codec *codec = card->rtd[0].codec;
+		struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
+
+		/*
+		 * Use slots 3/4 for S/PDIF so SSI won't try to enable
+		 * other slots and send some samples there
+		 * due to SLOTREQ bits for S/PDIF received from codec
+		 */
+		snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
+				     AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4);
+#endif
+
+		return 0;
+	}
+
 	ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
 				     codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
 	if (ret) {
@@ -407,7 +454,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	struct platform_device *cpu_pdev;
 	struct fsl_asoc_card_priv *priv;
 	struct i2c_client *codec_dev;
-	struct clk *codec_clk;
 	const char *codec_dai_name;
 	u32 width;
 	int ret;
@@ -420,9 +466,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	/* Give a chance to old DT binding */
 	if (!cpu_np)
 		cpu_np = of_parse_phandle(np, "ssi-controller", 0);
-	codec_np = of_parse_phandle(np, "audio-codec", 0);
-	if (!cpu_np || !codec_np) {
-		dev_err(&pdev->dev, "phandle missing or invalid\n");
+	if (!cpu_np) {
+		dev_err(&pdev->dev, "CPU phandle missing or invalid\n");
 		ret = -EINVAL;
 		goto fail;
 	}
@@ -434,22 +479,24 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 		goto fail;
 	}
 
-	codec_dev = of_find_i2c_device_by_node(codec_np);
-	if (!codec_dev) {
-		dev_err(&pdev->dev, "failed to find codec platform device\n");
-		ret = -EINVAL;
-		goto fail;
-	}
+	codec_np = of_parse_phandle(np, "audio-codec", 0);
+	if (codec_np)
+		codec_dev = of_find_i2c_device_by_node(codec_np);
+	else
+		codec_dev = NULL;
 
 	asrc_np = of_parse_phandle(np, "audio-asrc", 0);
 	if (asrc_np)
 		asrc_pdev = of_find_device_by_node(asrc_np);
 
 	/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
-	codec_clk = clk_get(&codec_dev->dev, NULL);
-	if (!IS_ERR(codec_clk)) {
-		priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
-		clk_put(codec_clk);
+	if (codec_dev) {
+		struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);
+
+		if (!IS_ERR(codec_clk)) {
+			priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
+			clk_put(codec_clk);
+		}
 	}
 
 	/* Default sample rate and format, will be updated in hw_params() */
@@ -486,12 +533,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 		priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
 		priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
 		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+	} else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
+		codec_dai_name = "ac97-hifi";
+		priv->card.set_bias_level = NULL;
+		priv->dai_fmt = SND_SOC_DAIFMT_AC97;
 	} else {
 		dev_err(&pdev->dev, "unknown Device Tree compatible\n");
 		ret = -EINVAL;
 		goto asrc_fail;
 	}
 
+	if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
+		dev_err(&pdev->dev, "failed to find codec device\n");
+		ret = -EINVAL;
+		goto asrc_fail;
+	}
+
 	/* Common settings for corresponding Freescale CPU DAI driver */
 	if (strstr(cpu_np->name, "ssi")) {
 		/* Only SSI needs to configure AUDMUX */
@@ -508,7 +565,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 		priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
 	}
 
-	sprintf(priv->name, "%s-audio", codec_dev->name);
+	snprintf(priv->name, sizeof(priv->name), "%s-audio",
+		 fsl_asoc_card_is_ac97(priv) ? "ac97" :
+		 codec_dev->name);
 
 	/* Initialize sound card */
 	priv->pdev = pdev;
@@ -532,8 +591,26 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 
 	/* Normal DAI Link */
 	priv->dai_link[0].cpu_of_node = cpu_np;
-	priv->dai_link[0].codec_of_node = codec_np;
 	priv->dai_link[0].codec_dai_name = codec_dai_name;
+
+	if (!fsl_asoc_card_is_ac97(priv))
+		priv->dai_link[0].codec_of_node = codec_np;
+	else {
+		u32 idx;
+
+		ret = of_property_read_u32(cpu_np, "cell-index", &idx);
+		if (ret) {
+			dev_err(&pdev->dev,
+				"cannot get CPU index property\n");
+			goto asrc_fail;
+		}
+
+		priv->dai_link[0].codec_name =
+				devm_kasprintf(&pdev->dev, GFP_KERNEL,
+					       "ac97-codec.%u",
+					       (unsigned int)idx);
+	}
+
 	priv->dai_link[0].platform_of_node = cpu_np;
 	priv->dai_link[0].dai_fmt = priv->dai_fmt;
 	priv->card.num_links = 1;
@@ -544,6 +621,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 		priv->dai_link[1].platform_of_node = asrc_np;
 		priv->dai_link[2].codec_dai_name = codec_dai_name;
 		priv->dai_link[2].codec_of_node = codec_np;
+		priv->dai_link[2].codec_name =
+				priv->dai_link[0].codec_name;
 		priv->dai_link[2].cpu_of_node = cpu_np;
 		priv->dai_link[2].dai_fmt = priv->dai_fmt;
 		priv->card.num_links = 3;
@@ -579,20 +658,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 
 asrc_fail:
 	of_node_put(asrc_np);
-fail:
 	of_node_put(codec_np);
+fail:
 	of_node_put(cpu_np);
 
 	return ret;
 }
 
 static const struct of_device_id fsl_asoc_card_dt_ids[] = {
+	{ .compatible = "fsl,imx-audio-ac97", },
 	{ .compatible = "fsl,imx-audio-cs42888", },
 	{ .compatible = "fsl,imx-audio-sgtl5000", },
 	{ .compatible = "fsl,imx-audio-wm8962", },
 	{ .compatible = "fsl,imx-audio-wm8960", },
 	{}
 };
+MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
 
 static struct platform_driver fsl_asoc_card_driver = {
 	.probe = fsl_asoc_card_probe,
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 837979ea5c92..59f234e51971 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -652,6 +652,24 @@ static const struct snd_soc_component_driver fsl_esai_component = {
 	.name		= "fsl-esai",
 };
 
+static const struct reg_default fsl_esai_reg_defaults[] = {
+	{0x8,  0x00000000},
+	{0x10, 0x00000000},
+	{0x18, 0x00000000},
+	{0x98, 0x00000000},
+	{0xd0, 0x00000000},
+	{0xd4, 0x00000000},
+	{0xd8, 0x00000000},
+	{0xdc, 0x00000000},
+	{0xe0, 0x00000000},
+	{0xe4, 0x0000ffff},
+	{0xe8, 0x0000ffff},
+	{0xec, 0x0000ffff},
+	{0xf0, 0x0000ffff},
+	{0xf8, 0x00000000},
+	{0xfc, 0x00000000},
+};
+
 static bool fsl_esai_readable_reg(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
@@ -684,6 +702,31 @@ static bool fsl_esai_readable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
+static bool fsl_esai_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case REG_ESAI_ETDR:
+	case REG_ESAI_ERDR:
+	case REG_ESAI_ESR:
+	case REG_ESAI_TFSR:
+	case REG_ESAI_RFSR:
+	case REG_ESAI_TX0:
+	case REG_ESAI_TX1:
+	case REG_ESAI_TX2:
+	case REG_ESAI_TX3:
+	case REG_ESAI_TX4:
+	case REG_ESAI_TX5:
+	case REG_ESAI_RX0:
+	case REG_ESAI_RX1:
+	case REG_ESAI_RX2:
+	case REG_ESAI_RX3:
+	case REG_ESAI_SAISR:
+		return true;
+	default:
+		return false;
+	}
+}
+
 static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
@@ -721,8 +764,12 @@ static const struct regmap_config fsl_esai_regmap_config = {
 	.val_bits = 32,
 
 	.max_register = REG_ESAI_PCRC,
+	.reg_defaults = fsl_esai_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(fsl_esai_reg_defaults),
 	.readable_reg = fsl_esai_readable_reg,
+	.volatile_reg = fsl_esai_volatile_reg,
 	.writeable_reg = fsl_esai_writeable_reg,
+	.cache_type = REGCACHE_RBTREE,
 };
 
 static int fsl_esai_probe(struct platform_device *pdev)
@@ -853,10 +900,51 @@ static const struct of_device_id fsl_esai_dt_ids[] = {
 };
 MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids);
 
+#ifdef CONFIG_PM_SLEEP
+static int fsl_esai_suspend(struct device *dev)
+{
+	struct fsl_esai *esai = dev_get_drvdata(dev);
+
+	regcache_cache_only(esai->regmap, true);
+	regcache_mark_dirty(esai->regmap);
+
+	return 0;
+}
+
+static int fsl_esai_resume(struct device *dev)
+{
+	struct fsl_esai *esai = dev_get_drvdata(dev);
+	int ret;
+
+	regcache_cache_only(esai->regmap, false);
+
+	/* FIFO reset for safety */
+	regmap_update_bits(esai->regmap, REG_ESAI_TFCR,
+			   ESAI_xFCR_xFR, ESAI_xFCR_xFR);
+	regmap_update_bits(esai->regmap, REG_ESAI_RFCR,
+			   ESAI_xFCR_xFR, ESAI_xFCR_xFR);
+
+	ret = regcache_sync(esai->regmap);
+	if (ret)
+		return ret;
+
+	/* FIFO reset done */
+	regmap_update_bits(esai->regmap, REG_ESAI_TFCR, ESAI_xFCR_xFR, 0);
+	regmap_update_bits(esai->regmap, REG_ESAI_RFCR, ESAI_xFCR_xFR, 0);
+
+	return 0;
+}
+#endif /* CONFIG_PM_SLEEP */
+
+static const struct dev_pm_ops fsl_esai_pm_ops = {
+	SET_SYSTEM_SLEEP_PM_OPS(fsl_esai_suspend, fsl_esai_resume)
+};
+
 static struct platform_driver fsl_esai_driver = {
 	.probe = fsl_esai_probe,
 	.driver = {
 		.name = "fsl-esai-dai",
+		.pm = &fsl_esai_pm_ops,
 		.of_match_table = fsl_esai_dt_ids,
 	},
 };
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index a18fd92c4a85..a4435f5e3be9 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -27,13 +27,13 @@
 #define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\
 		       FSL_SAI_CSR_FEIE)
 
-static u32 fsl_sai_rates[] = {
+static const unsigned int fsl_sai_rates[] = {
 	8000, 11025, 12000, 16000, 22050,
 	24000, 32000, 44100, 48000, 64000,
 	88200, 96000, 176400, 192000
 };
 
-static struct snd_pcm_hw_constraint_list fsl_sai_rate_constraints = {
+static const struct snd_pcm_hw_constraint_list fsl_sai_rate_constraints = {
 	.count = ARRAY_SIZE(fsl_sai_rates),
 	.list = fsl_sai_rates,
 };
@@ -637,6 +637,8 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg)
 static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
+	case FSL_SAI_TCSR:
+	case FSL_SAI_RCSR:
 	case FSL_SAI_TFR:
 	case FSL_SAI_RFR:
 	case FSL_SAI_TDR:
@@ -681,6 +683,7 @@ static const struct regmap_config fsl_sai_regmap_config = {
 	.readable_reg = fsl_sai_readable_reg,
 	.volatile_reg = fsl_sai_volatile_reg,
 	.writeable_reg = fsl_sai_writeable_reg,
+	.cache_type = REGCACHE_FLAT,
 };
 
 static int fsl_sai_probe(struct platform_device *pdev)
@@ -801,11 +804,42 @@ static const struct of_device_id fsl_sai_ids[] = {
 	{ .compatible = "fsl,imx6sx-sai", },
 	{ /* sentinel */ }
 };
+MODULE_DEVICE_TABLE(of, fsl_sai_ids);
+
+#ifdef CONFIG_PM_SLEEP
+static int fsl_sai_suspend(struct device *dev)
+{
+	struct fsl_sai *sai = dev_get_drvdata(dev);
+
+	regcache_cache_only(sai->regmap, true);
+	regcache_mark_dirty(sai->regmap);
+
+	return 0;
+}
+
+static int fsl_sai_resume(struct device *dev)
+{
+	struct fsl_sai *sai = dev_get_drvdata(dev);
+
+	regcache_cache_only(sai->regmap, false);
+	regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR);
+	regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR);
+	msleep(1);
+	regmap_write(sai->regmap, FSL_SAI_TCSR, 0);
+	regmap_write(sai->regmap, FSL_SAI_RCSR, 0);
+	return regcache_sync(sai->regmap);
+}
+#endif /* CONFIG_PM_SLEEP */
+
+static const struct dev_pm_ops fsl_sai_pm_ops = {
+	SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume)
+};
 
 static struct platform_driver fsl_sai_driver = {
 	.probe = fsl_sai_probe,
 	.driver = {
 		.name = "fsl-sai",
+		.pm = &fsl_sai_pm_ops,
 		.of_match_table = fsl_sai_ids,
 	},
 };
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index ab729f2426fe..3d59bb6719f2 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -108,6 +108,8 @@ struct fsl_spdif_priv {
 	struct clk *sysclk;
 	struct snd_dmaengine_dai_dma_data dma_params_tx;
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
+	/* regcache for SRPC */
+	u32 regcache_srpc;
 };
 
 /* DPLL locked and lock loss interrupt handler */
@@ -300,6 +302,8 @@ static int spdif_softreset(struct fsl_spdif_priv *spdif_priv)
 	struct regmap *regmap = spdif_priv->regmap;
 	u32 val, cycle = 1000;
 
+	regcache_cache_bypass(regmap, true);
+
 	regmap_write(regmap, REG_SPDIF_SCR, SCR_SOFT_RESET);
 
 	/*
@@ -310,6 +314,10 @@ static int spdif_softreset(struct fsl_spdif_priv *spdif_priv)
 		regmap_read(regmap, REG_SPDIF_SCR, &val);
 	} while ((val & SCR_SOFT_RESET) && cycle--);
 
+	regcache_cache_bypass(regmap, false);
+	regcache_mark_dirty(regmap);
+	regcache_sync(regmap);
+
 	if (cycle)
 		return 0;
 	else
@@ -997,6 +1005,14 @@ static const struct snd_soc_component_driver fsl_spdif_component = {
 };
 
 /* FSL SPDIF REGMAP */
+static const struct reg_default fsl_spdif_reg_defaults[] = {
+	{0x0,  0x00000400},
+	{0x4,  0x00000000},
+	{0xc,  0x00000000},
+	{0x34, 0x00000000},
+	{0x38, 0x00000000},
+	{0x50, 0x00020f00},
+};
 
 static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg)
 {
@@ -1022,6 +1038,26 @@ static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
+static bool fsl_spdif_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case REG_SPDIF_SRPC:
+	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_STL:
+	case REG_SPDIF_STR:
+	case REG_SPDIF_SRFM:
+		return true;
+	default:
+		return false;
+	}
+}
+
 static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
@@ -1047,8 +1083,12 @@ static const struct regmap_config fsl_spdif_regmap_config = {
 	.val_bits = 32,
 
 	.max_register = REG_SPDIF_STC,
+	.reg_defaults = fsl_spdif_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(fsl_spdif_reg_defaults),
 	.readable_reg = fsl_spdif_readable_reg,
+	.volatile_reg = fsl_spdif_volatile_reg,
 	.writeable_reg = fsl_spdif_writeable_reg,
+	.cache_type = REGCACHE_RBTREE,
 };
 
 static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
@@ -1271,6 +1311,38 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	return ret;
 }
 
+#ifdef CONFIG_PM_SLEEP
+static int fsl_spdif_suspend(struct device *dev)
+{
+	struct fsl_spdif_priv *spdif_priv = dev_get_drvdata(dev);
+
+	regmap_read(spdif_priv->regmap, REG_SPDIF_SRPC,
+			&spdif_priv->regcache_srpc);
+
+	regcache_cache_only(spdif_priv->regmap, true);
+	regcache_mark_dirty(spdif_priv->regmap);
+
+	return 0;
+}
+
+static int fsl_spdif_resume(struct device *dev)
+{
+	struct fsl_spdif_priv *spdif_priv = dev_get_drvdata(dev);
+
+	regcache_cache_only(spdif_priv->regmap, false);
+
+	regmap_update_bits(spdif_priv->regmap, REG_SPDIF_SRPC,
+			SRPC_CLKSRC_SEL_MASK | SRPC_GAINSEL_MASK,
+			spdif_priv->regcache_srpc);
+
+	return regcache_sync(spdif_priv->regmap);
+}
+#endif /* CONFIG_PM_SLEEP */
+
+static const struct dev_pm_ops fsl_spdif_pm = {
+	SET_SYSTEM_SLEEP_PM_OPS(fsl_spdif_suspend, fsl_spdif_resume)
+};
+
 static const struct of_device_id fsl_spdif_dt_ids[] = {
 	{ .compatible = "fsl,imx35-spdif", },
 	{ .compatible = "fsl,vf610-spdif", },
@@ -1282,6 +1354,7 @@ static struct platform_driver fsl_spdif_driver = {
 	.driver = {
 		.name = "fsl-spdif-dai",
 		.of_match_table = fsl_spdif_dt_ids,
+		.pm = &fsl_spdif_pm,
 	},
 	.probe = fsl_spdif_probe,
 };
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 37c5cd4d0e59..95d2392303eb 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -111,12 +111,75 @@ struct fsl_ssi_rxtx_reg_val {
 	struct fsl_ssi_reg_val rx;
 	struct fsl_ssi_reg_val tx;
 };
+
+static const struct reg_default fsl_ssi_reg_defaults[] = {
+	{0x10, 0x00000000},
+	{0x18, 0x00003003},
+	{0x1c, 0x00000200},
+	{0x20, 0x00000200},
+	{0x24, 0x00040000},
+	{0x28, 0x00040000},
+	{0x38, 0x00000000},
+	{0x48, 0x00000000},
+	{0x4c, 0x00000000},
+	{0x54, 0x00000000},
+	{0x58, 0x00000000},
+};
+
+static bool fsl_ssi_readable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CCSR_SSI_SACCEN:
+	case CCSR_SSI_SACCDIS:
+		return false;
+	default:
+		return true;
+	}
+}
+
+static bool fsl_ssi_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CCSR_SSI_STX0:
+	case CCSR_SSI_STX1:
+	case CCSR_SSI_SRX0:
+	case CCSR_SSI_SRX1:
+	case CCSR_SSI_SISR:
+	case CCSR_SSI_SFCSR:
+	case CCSR_SSI_SACADD:
+	case CCSR_SSI_SACDAT:
+	case CCSR_SSI_SATAG:
+	case CCSR_SSI_SACCST:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool fsl_ssi_writeable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CCSR_SSI_SRX0:
+	case CCSR_SSI_SRX1:
+	case CCSR_SSI_SACCST:
+		return false;
+	default:
+		return true;
+	}
+}
+
 static const struct regmap_config fsl_ssi_regconfig = {
 	.max_register = CCSR_SSI_SACCDIS,
 	.reg_bits = 32,
 	.val_bits = 32,
 	.reg_stride = 4,
 	.val_format_endian = REGMAP_ENDIAN_NATIVE,
+	.reg_defaults = fsl_ssi_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(fsl_ssi_reg_defaults),
+	.readable_reg = fsl_ssi_readable_reg,
+	.volatile_reg = fsl_ssi_volatile_reg,
+	.writeable_reg = fsl_ssi_writeable_reg,
+	.cache_type = REGCACHE_RBTREE,
 };
 
 struct fsl_ssi_soc_data {
@@ -176,6 +239,9 @@ struct fsl_ssi_private {
 	unsigned int baudclk_streams;
 	unsigned int bitclk_freq;
 
+	/*regcache for SFCSR*/
+	u32 regcache_sfcsr;
+
 	/* DMA params */
 	struct snd_dmaengine_dai_dma_data dma_params_tx;
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
@@ -1514,10 +1580,46 @@ static int fsl_ssi_remove(struct platform_device *pdev)
 	return 0;
 }
 
+#ifdef CONFIG_PM_SLEEP
+static int fsl_ssi_suspend(struct device *dev)
+{
+	struct fsl_ssi_private *ssi_private = dev_get_drvdata(dev);
+	struct regmap *regs = ssi_private->regs;
+
+	regmap_read(regs, CCSR_SSI_SFCSR,
+			&ssi_private->regcache_sfcsr);
+
+	regcache_cache_only(regs, true);
+	regcache_mark_dirty(regs);
+
+	return 0;
+}
+
+static int fsl_ssi_resume(struct device *dev)
+{
+	struct fsl_ssi_private *ssi_private = dev_get_drvdata(dev);
+	struct regmap *regs = ssi_private->regs;
+
+	regcache_cache_only(regs, false);
+
+	regmap_update_bits(regs, CCSR_SSI_SFCSR,
+			CCSR_SSI_SFCSR_RFWM1_MASK | CCSR_SSI_SFCSR_TFWM1_MASK |
+			CCSR_SSI_SFCSR_RFWM0_MASK | CCSR_SSI_SFCSR_TFWM0_MASK,
+			ssi_private->regcache_sfcsr);
+
+	return regcache_sync(regs);
+}
+#endif /* CONFIG_PM_SLEEP */
+
+static const struct dev_pm_ops fsl_ssi_pm = {
+	SET_SYSTEM_SLEEP_PM_OPS(fsl_ssi_suspend, fsl_ssi_resume)
+};
+
 static struct platform_driver fsl_ssi_driver = {
 	.driver = {
 		.name = "fsl-ssi-dai",
 		.of_match_table = fsl_ssi_ids,
+		.pm = &fsl_ssi_pm,
 	},
 	.probe = fsl_ssi_probe,
 	.remove = fsl_ssi_remove,
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index 33da26a12457..a407e833c612 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -89,6 +89,7 @@ MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids);
 static struct platform_driver imx_spdif_driver = {
 	.driver = {
 		.name = "imx-spdif",
+		.pm = &snd_soc_pm_ops,
 		.of_match_table = imx_spdif_dt_ids,
 	},
 	.probe = imx_spdif_audio_probe,
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 3ff76d419436..54c33204541f 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -151,7 +151,9 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai,
 	}
 
 	if (set->slots) {
-		ret = snd_soc_dai_set_tdm_slot(dai, 0, 0,
+		ret = snd_soc_dai_set_tdm_slot(dai,
+					       set->tx_slot_mask,
+					       set->rx_slot_mask,
 						set->slots,
 						set->slot_width);
 		if (ret && ret != -ENOTSUPP) {
@@ -243,7 +245,9 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
 		return ret;
 
 	/* Parse TDM slot */
-	ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width);
+	ret = snd_soc_of_parse_tdm_slot(np, &dai->tx_slot_mask,
+					&dai->rx_slot_mask,
+					&dai->slots, &dai->slot_width);
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 05fde5e6e257..7b778ab85f8b 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -12,6 +12,7 @@ config SND_MFLD_MACHINE
 
 config SND_SST_MFLD_PLATFORM
 	tristate
+	select SND_SOC_COMPRESS
 
 config SND_SST_IPC
 	tristate
@@ -138,4 +139,18 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH
 config SND_SOC_INTEL_SKYLAKE
 	tristate
 	select SND_HDA_EXT_CORE
+	select SND_SOC_TOPOLOGY
 	select SND_SOC_INTEL_SST
+
+config SND_SOC_INTEL_SKL_RT286_MACH
+	tristate "ASoC Audio driver for SKL with RT286 I2S mode"
+	depends on X86 && ACPI
+	select SND_SOC_INTEL_SST
+	select SND_SOC_INTEL_SKYLAKE
+	select SND_SOC_RT286
+	select SND_SOC_DMIC
+	help
+	   This adds support for ASoC machine driver for Skylake platforms
+	   with RT286 I2S audio codec.
+	   Say Y if you have such a device
+	   If unsure select "N".
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 683e50116152..0487cfaac538 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -368,23 +368,6 @@ static void sst_media_close(struct snd_pcm_substream *substream,
 	kfree(stream);
 }
 
-static inline unsigned int get_current_pipe_id(struct snd_soc_dai *dai,
-					       struct snd_pcm_substream *substream)
-{
-	struct sst_data *sst = snd_soc_dai_get_drvdata(dai);
-	struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map;
-	struct sst_runtime_stream *stream =
-			substream->runtime->private_data;
-	u32 str_id = stream->stream_info.str_id;
-	unsigned int pipe_id;
-
-	pipe_id = map[str_id].device_id;
-
-	dev_dbg(dai->dev, "got pipe_id = %#x for str_id = %d\n",
-			pipe_id, str_id);
-	return pipe_id;
-}
-
 static int sst_media_prepare(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
@@ -529,7 +512,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = {
 },
 {
 	.name = "compress-cpu-dai",
-	.compress_dai = 1,
+	.compress_new = snd_soc_new_compress,
 	.ops = &sst_compr_dai_ops,
 	.playback = {
 		.stream_name = "Compress Playback",
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index cb94895c9edb..371c4565cad8 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -6,6 +6,7 @@ snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o
 snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
 snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o
 snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o
+snd-soc-skl_rt286-objs := skl_rt286.o
 
 obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
@@ -15,3 +16,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o
 obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o
 obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o
 obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o
+obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 8bafaf6ceab1..3f8a1e10bed0 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -266,18 +266,11 @@ static int broadwell_audio_probe(struct platform_device *pdev)
 {
 	broadwell_rt286.dev = &pdev->dev;
 
-	return snd_soc_register_card(&broadwell_rt286);
-}
-
-static int broadwell_audio_remove(struct platform_device *pdev)
-{
-	snd_soc_unregister_card(&broadwell_rt286);
-	return 0;
+	return devm_snd_soc_register_card(&pdev->dev, &broadwell_rt286);
 }
 
 static struct platform_driver broadwell_audio = {
 	.probe = broadwell_audio_probe,
-	.remove = broadwell_audio_remove,
 	.driver = {
 		.name = "broadwell-audio",
 	},
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index c4453120b11a..7a5c9a36c1db 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -117,20 +117,10 @@ static int byt_codec_fixup(struct snd_soc_pcm_runtime *rtd,
 	return 0;
 }
 
-static unsigned int rates_48000[] = {
-	48000,
-};
-
-static struct snd_pcm_hw_constraint_list constraints_48000 = {
-	.count = ARRAY_SIZE(rates_48000),
-	.list  = rates_48000,
-};
-
 static int byt_aif1_startup(struct snd_pcm_substream *substream)
 {
-	return snd_pcm_hw_constraint_list(substream->runtime, 0,
-			SNDRV_PCM_HW_PARAM_RATE,
-			&constraints_48000);
+	return snd_pcm_hw_constraint_single(substream->runtime,
+			SNDRV_PCM_HW_PARAM_RATE, 48000);
 }
 
 static struct snd_soc_ops byt_aif1_ops = {
diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
index 49f4869cec48..4e2fcf188dd1 100644
--- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -193,20 +193,10 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
 	return 0;
 }
 
-static unsigned int rates_48000[] = {
-	48000,
-};
-
-static struct snd_pcm_hw_constraint_list constraints_48000 = {
-	.count = ARRAY_SIZE(rates_48000),
-	.list  = rates_48000,
-};
-
 static int cht_aif1_startup(struct snd_pcm_substream *substream)
 {
-	return snd_pcm_hw_constraint_list(substream->runtime, 0,
-			SNDRV_PCM_HW_PARAM_RATE,
-			&constraints_48000);
+	return snd_pcm_hw_constraint_single(substream->runtime,
+			SNDRV_PCM_HW_PARAM_RATE, 48000);
 }
 
 static int cht_max98090_headset_init(struct snd_soc_component *component)
diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c
index 7be8461e4d3b..38d65a3529c4 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5645.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -235,20 +235,10 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
 	return 0;
 }
 
-static unsigned int rates_48000[] = {
-	48000,
-};
-
-static struct snd_pcm_hw_constraint_list constraints_48000 = {
-	.count = ARRAY_SIZE(rates_48000),
-	.list  = rates_48000,
-};
-
 static int cht_aif1_startup(struct snd_pcm_substream *substream)
 {
-	return snd_pcm_hw_constraint_list(substream->runtime, 0,
-			SNDRV_PCM_HW_PARAM_RATE,
-			&constraints_48000);
+	return snd_pcm_hw_constraint_single(substream->runtime,
+			SNDRV_PCM_HW_PARAM_RATE, 48000);
 }
 
 static struct snd_soc_ops cht_aif1_ops = {
diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c
index 23fe04075142..5621ccd92992 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5672.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -222,20 +222,10 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
 	return 0;
 }
 
-static unsigned int rates_48000[] = {
-	48000,
-};
-
-static struct snd_pcm_hw_constraint_list constraints_48000 = {
-	.count = ARRAY_SIZE(rates_48000),
-	.list  = rates_48000,
-};
-
 static int cht_aif1_startup(struct snd_pcm_substream *substream)
 {
-	return snd_pcm_hw_constraint_list(substream->runtime, 0,
-			SNDRV_PCM_HW_PARAM_RATE,
-			&constraints_48000);
+	return snd_pcm_hw_constraint_single(substream->runtime,
+			SNDRV_PCM_HW_PARAM_RATE, 48000);
 }
 
 static struct snd_soc_ops cht_aif1_ops = {
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
new file mode 100644
index 000000000000..a73a431bd8b7
--- /dev/null
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -0,0 +1,259 @@
+/*
+ * Intel Skylake I2S Machine Driver
+ *
+ * Copyright (C) 2014-2015, Intel Corporation. All rights reserved.
+ *
+ * Modified from:
+ *   Intel Broadwell Wildcatpoint SST Audio
+ *
+ *   Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License version
+ * 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/pcm_params.h>
+#include "../../codecs/rt286.h"
+
+static struct snd_soc_jack skylake_headset;
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin skylake_headset_pins[] = {
+	{
+		.pin = "Mic Jack",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Headphone Jack",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+static const struct snd_kcontrol_new skylake_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Speaker"),
+	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+	SOC_DAPM_PIN_SWITCH("Mic Jack"),
+};
+
+static const struct snd_soc_dapm_widget skylake_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_MIC("DMIC2", NULL),
+	SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+};
+
+static const struct snd_soc_dapm_route skylake_rt286_map[] = {
+	/* speaker */
+	{"Speaker", NULL, "SPOR"},
+	{"Speaker", NULL, "SPOL"},
+
+	/* HP jack connectors - unknown if we have jack deteck */
+	{"Headphone Jack", NULL, "HPO Pin"},
+
+	/* other jacks */
+	{"MIC1", NULL, "Mic Jack"},
+
+	/* digital mics */
+	{"DMIC1 Pin", NULL, "DMIC2"},
+	{"DMIC AIF", NULL, "SoC DMIC"},
+
+	/* CODEC BE connections */
+	{ "AIF1 Playback", NULL, "ssp0 Tx"},
+	{ "ssp0 Tx", NULL, "codec0_out"},
+	{ "ssp0 Tx", NULL, "codec1_out"},
+
+	{ "codec0_in", NULL, "ssp0 Rx" },
+	{ "codec1_in", NULL, "ssp0 Rx" },
+	{ "ssp0 Rx", NULL, "AIF1 Capture" },
+
+	{ "dmic01_hifi", NULL, "DMIC01 Rx" },
+	{ "DMIC01 Rx", NULL, "Capture" },
+
+	{ "hif1", NULL, "iDisp Tx"},
+	{ "iDisp Tx", NULL, "iDisp_out"},
+
+};
+
+static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	int ret;
+
+	ret = snd_soc_card_jack_new(rtd->card, "Headset",
+		SND_JACK_HEADSET | SND_JACK_BTN_0,
+		&skylake_headset,
+		skylake_headset_pins, ARRAY_SIZE(skylake_headset_pins));
+
+	if (ret)
+		return ret;
+
+	rt286_mic_detect(codec, &skylake_headset);
+
+	return 0;
+}
+
+
+static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The output is 48KHz, stereo, 16bits */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+	params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
+
+	return 0;
+}
+
+static int skylake_rt286_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_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		dev_err(rtd->dev, "set codec sysclk failed: %d\n", ret);
+
+	return ret;
+}
+
+static struct snd_soc_ops skylake_rt286_ops = {
+	.hw_params = skylake_rt286_hw_params,
+};
+
+/* skylake digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link skylake_rt286_dais[] = {
+	/* Front End DAI links */
+	{
+		.name = "Skl Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:1f.3",
+		.nonatomic = 1,
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST,
+			SND_SOC_DPCM_TRIGGER_POST
+		},
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "Skl Audio Capture Port",
+		.stream_name = "Audio Record",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:1f.3",
+		.nonatomic = 1,
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST,
+			SND_SOC_DPCM_TRIGGER_POST
+		},
+		.dpcm_capture = 1,
+	},
+	{
+		.name = "Skl Audio Reference cap",
+		.stream_name = "refcap",
+		.cpu_dai_name = "Reference Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.init = NULL,
+		.dpcm_capture = 1,
+		.ignore_suspend = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+
+	/* Back End DAI links */
+	{
+		/* SSP0 - Codec */
+		.name = "SSP0-Codec",
+		.be_id = 0,
+		.cpu_dai_name = "SSP0 Pin",
+		.platform_name = "0000:00:1f.3",
+		.no_pcm = 1,
+		.codec_name = "i2c-INT343A:00",
+		.codec_dai_name = "rt286-aif1",
+		.init = skylake_rt286_codec_init,
+		.dai_fmt = SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_suspend = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = skylake_ssp0_fixup,
+		.ops = &skylake_rt286_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+	{
+		.name = "dmic01",
+		.be_id = 1,
+		.cpu_dai_name = "DMIC01 Pin",
+		.codec_name = "dmic-codec",
+		.codec_dai_name = "dmic-hifi",
+		.platform_name = "0000:00:1f.3",
+		.ignore_suspend = 1,
+		.dpcm_capture = 1,
+		.no_pcm = 1,
+	},
+};
+
+/* skylake audio machine driver for SPT + RT286S */
+static struct snd_soc_card skylake_rt286 = {
+	.name = "skylake-rt286",
+	.owner = THIS_MODULE,
+	.dai_link = skylake_rt286_dais,
+	.num_links = ARRAY_SIZE(skylake_rt286_dais),
+	.controls = skylake_controls,
+	.num_controls = ARRAY_SIZE(skylake_controls),
+	.dapm_widgets = skylake_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(skylake_widgets),
+	.dapm_routes = skylake_rt286_map,
+	.num_dapm_routes = ARRAY_SIZE(skylake_rt286_map),
+	.fully_routed = true,
+};
+
+static int skylake_audio_probe(struct platform_device *pdev)
+{
+	skylake_rt286.dev = &pdev->dev;
+
+	return devm_snd_soc_register_card(&pdev->dev, &skylake_rt286);
+}
+
+static struct platform_driver skylake_audio = {
+	.probe = skylake_audio_probe,
+	.driver = {
+		.name = "skl_alc286s_i2s",
+	},
+};
+
+module_platform_driver(skylake_audio)
+
+/* Module information */
+MODULE_AUTHOR("Omair Mohammed Abdullah <omair.m.abdullah@intel.com>");
+MODULE_DESCRIPTION("Intel SST Audio for Skylake");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:skl_alc286s_i2s");
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index f24154ca4e98..d9105584c51f 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -1,7 +1,11 @@
-snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o
+snd-soc-sst-dsp-objs := sst-dsp.o
 snd-soc-sst-acpi-objs := sst-acpi.o
 snd-soc-sst-ipc-objs := sst-ipc.o
 
+ifneq ($(CONFIG_DW_DMAC_CORE),)
+snd-soc-sst-dsp-objs += sst-firmware.o
+endif
+
 obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
 obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o
 
diff --git a/sound/soc/intel/common/sst-dsp-priv.h b/sound/soc/intel/common/sst-dsp-priv.h
index cbd568eac033..2151652d37b7 100644
--- a/sound/soc/intel/common/sst-dsp-priv.h
+++ b/sound/soc/intel/common/sst-dsp-priv.h
@@ -314,6 +314,7 @@ struct sst_dsp {
 	int sst_state;
 	struct skl_cl_dev cl_dev;
 	u32 intr_status;
+	const struct firmware *fw;
 };
 
 /* Size optimised DRAM/IRAM memcpy */
diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c
index a627236dd1f5..c9452e02e0dd 100644
--- a/sound/soc/intel/common/sst-dsp.c
+++ b/sound/soc/intel/common/sst-dsp.c
@@ -420,6 +420,7 @@ void sst_dsp_inbox_read(struct sst_dsp *sst, void *message, size_t bytes)
 }
 EXPORT_SYMBOL_GPL(sst_dsp_inbox_read);
 
+#if IS_ENABLED(CONFIG_DW_DMAC_CORE)
 struct sst_dsp *sst_dsp_new(struct device *dev,
 	struct sst_dsp_device *sst_dev, struct sst_pdata *pdata)
 {
@@ -484,6 +485,7 @@ void sst_dsp_free(struct sst_dsp *sst)
 	sst_dma_free(sst->dma);
 }
 EXPORT_SYMBOL_GPL(sst_dsp_free);
+#endif
 
 /* Module information */
 MODULE_AUTHOR("Liam Girdwood");
diff --git a/sound/soc/intel/common/sst-dsp.h b/sound/soc/intel/common/sst-dsp.h
index 1f45f18715c0..859f0de00339 100644
--- a/sound/soc/intel/common/sst-dsp.h
+++ b/sound/soc/intel/common/sst-dsp.h
@@ -216,10 +216,12 @@ struct sst_pdata {
 	void *dsp;
 };
 
+#if IS_ENABLED(CONFIG_DW_DMAC_CORE)
 /* Initialization */
 struct sst_dsp *sst_dsp_new(struct device *dev,
 	struct sst_dsp_device *sst_dev, struct sst_pdata *pdata);
 void sst_dsp_free(struct sst_dsp *sst);
+#endif
 
 /* SHIM Read / Write */
 void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value);
diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c
index ebcca6dc48d1..1636a1eeb002 100644
--- a/sound/soc/intel/common/sst-firmware.c
+++ b/sound/soc/intel/common/sst-firmware.c
@@ -26,7 +26,6 @@
 #include <linux/acpi.h>
 
 /* supported DMA engine drivers */
-#include <linux/platform_data/dma-dw.h>
 #include <linux/dma/dw.h>
 
 #include <asm/page.h>
@@ -169,12 +168,6 @@ err:
 	return ret;
 }
 
-static struct dw_dma_platform_data dw_pdata = {
-	.is_private = 1,
-	.chan_allocation_order = CHAN_ALLOCATION_ASCENDING,
-	.chan_priority = CHAN_PRIORITY_ASCENDING,
-};
-
 static struct dw_dma_chip *dw_probe(struct device *dev, struct resource *mem,
 	int irq)
 {
@@ -195,7 +188,8 @@ static struct dw_dma_chip *dw_probe(struct device *dev, struct resource *mem,
 		return ERR_PTR(err);
 
 	chip->dev = dev;
-	err = dw_dma_probe(chip, &dw_pdata);
+
+	err = dw_dma_probe(chip, NULL);
 	if (err)
 		return ERR_PTR(err);
 
diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile
index 27db22178204..914b6dab9bea 100644
--- a/sound/soc/intel/skylake/Makefile
+++ b/sound/soc/intel/skylake/Makefile
@@ -1,4 +1,5 @@
-snd-soc-skl-objs := skl.o skl-pcm.o skl-nhlt.o skl-messages.o
+snd-soc-skl-objs := skl.o skl-pcm.o skl-nhlt.o skl-messages.o \
+skl-topology.o
 
 obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o
 
diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index 826d4fd8930a..50a109503a3f 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -54,6 +54,24 @@ static int skl_free_dma_buf(struct device *dev, struct snd_dma_buffer *dmab)
 	return 0;
 }
 
+#define NOTIFICATION_PARAM_ID 3
+#define NOTIFICATION_MASK 0xf
+
+/* disable notfication for underruns/overruns from firmware module */
+static void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable)
+{
+	struct notification_mask mask;
+	struct skl_ipc_large_config_msg	msg = {0};
+
+	mask.notify = NOTIFICATION_MASK;
+	mask.enable = enable;
+
+	msg.large_param_id = NOTIFICATION_PARAM_ID;
+	msg.param_data_size = sizeof(mask);
+
+	skl_ipc_set_large_config(&ctx->ipc, &msg, (u32 *)&mask);
+}
+
 int skl_init_dsp(struct skl *skl)
 {
 	void __iomem *mmio_base;
@@ -79,7 +97,10 @@ int skl_init_dsp(struct skl *skl)
 
 	ret = skl_sst_dsp_init(bus->dev, mmio_base, irq,
 			loader_ops, &skl->skl_sst);
+	if (ret < 0)
+		return ret;
 
+	skl_dsp_enable_notification(skl->skl_sst, false);
 	dev_dbg(bus->dev, "dsp registration status=%d\n", ret);
 
 	return ret;
@@ -122,6 +143,7 @@ int skl_suspend_dsp(struct skl *skl)
 int skl_resume_dsp(struct skl *skl)
 {
 	struct skl_sst *ctx = skl->skl_sst;
+	int ret;
 
 	/* if ppcap is not supported return 0 */
 	if (!skl->ebus.ppcap)
@@ -131,7 +153,12 @@ int skl_resume_dsp(struct skl *skl)
 	snd_hdac_ext_bus_ppcap_enable(&skl->ebus, true);
 	snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, true);
 
-	return skl_dsp_wake(ctx->dsp);
+	ret = skl_dsp_wake(ctx->dsp);
+	if (ret < 0)
+		return ret;
+
+	skl_dsp_enable_notification(skl->skl_sst, false);
+	return ret;
 }
 
 enum skl_bitdepth skl_get_bit_depth(int params)
@@ -294,6 +321,7 @@ static void skl_copy_copier_caps(struct skl_module_cfg *mconfig,
 			(mconfig->formats_config.caps_size) / 4;
 }
 
+#define SKL_NON_GATEWAY_CPR_NODE_ID 0xFFFFFFFF
 /*
  * Calculate the gatewat settings required for copier module, type of
  * gateway and index of gateway to use
@@ -303,6 +331,7 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx,
 			struct skl_cpr_cfg *cpr_mconfig)
 {
 	union skl_connector_node_id node_id = {0};
+	union skl_ssp_dma_node ssp_node  = {0};
 	struct skl_pipe_params *params = mconfig->pipe->p_params;
 
 	switch (mconfig->dev_type) {
@@ -320,9 +349,9 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx,
 			(SKL_CONN_SOURCE == mconfig->hw_conn_type) ?
 			SKL_DMA_I2S_LINK_OUTPUT_CLASS :
 			SKL_DMA_I2S_LINK_INPUT_CLASS;
-		node_id.node.vindex = params->host_dma_id +
-					 (mconfig->time_slot << 1) +
-					 (mconfig->vbus_id << 3);
+		ssp_node.dma_node.time_slot_index = mconfig->time_slot;
+		ssp_node.dma_node.i2s_instance = mconfig->vbus_id;
+		node_id.node.vindex = ssp_node.val;
 		break;
 
 	case SKL_DEVICE_DMIC:
@@ -339,13 +368,18 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx,
 		node_id.node.vindex = params->link_dma_id;
 		break;
 
-	default:
+	case SKL_DEVICE_HDAHOST:
 		node_id.node.dma_type =
 			(SKL_CONN_SOURCE == mconfig->hw_conn_type) ?
 			SKL_DMA_HDA_HOST_OUTPUT_CLASS :
 			SKL_DMA_HDA_HOST_INPUT_CLASS;
 		node_id.node.vindex = params->host_dma_id;
 		break;
+
+	default:
+		cpr_mconfig->gtw_cfg.node_id = SKL_NON_GATEWAY_CPR_NODE_ID;
+		cpr_mconfig->cpr_feature_mask = 0;
+		return;
 	}
 
 	cpr_mconfig->gtw_cfg.node_id = node_id.val;
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 13036b19d7e5..b0c7bd113aac 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -25,7 +25,7 @@ static u8 OSC_UUID[16] = {0x6E, 0x88, 0x9F, 0xA6, 0xEB, 0x6C, 0x94, 0x45,
 
 #define DSDT_NHLT_PATH "\\_SB.PCI0.HDAS"
 
-void __iomem *skl_nhlt_init(struct device *dev)
+void *skl_nhlt_init(struct device *dev)
 {
 	acpi_handle handle;
 	union acpi_object *obj;
@@ -40,17 +40,17 @@ void __iomem *skl_nhlt_init(struct device *dev)
 	if (obj && obj->type == ACPI_TYPE_BUFFER) {
 		nhlt_ptr = (struct nhlt_resource_desc  *)obj->buffer.pointer;
 
-		return ioremap_cache(nhlt_ptr->min_addr, nhlt_ptr->length);
+		return memremap(nhlt_ptr->min_addr, nhlt_ptr->length,
+				MEMREMAP_WB);
 	}
 
 	dev_err(dev, "device specific method to extract NHLT blob failed\n");
 	return NULL;
 }
 
-void skl_nhlt_free(void __iomem *addr)
+void skl_nhlt_free(void *addr)
 {
-	iounmap(addr);
-	addr = NULL;
+	memunmap(addr);
 }
 
 static struct nhlt_specific_cfg *skl_get_specific_cfg(
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 7d617bf493bc..a2f94ce1679d 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -24,6 +24,7 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include "skl.h"
+#include "skl-topology.h"
 
 #define HDA_MONO 1
 #define HDA_STEREO 2
@@ -115,7 +116,7 @@ static int skl_pcm_open(struct snd_pcm_substream *substream,
 
 	dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
 	ret = pm_runtime_get_sync(dai->dev);
-	if (ret)
+	if (ret < 0)
 		return ret;
 
 	stream = snd_hdac_ext_stream_assign(ebus, substream,
@@ -214,6 +215,8 @@ static int skl_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev);
 	struct hdac_ext_stream *stream = get_hdac_ext_stream(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct skl_pipe_params p_params = {0};
+	struct skl_module_cfg *m_cfg;
 	int ret, dma_id;
 
 	dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
@@ -228,6 +231,16 @@ static int skl_pcm_hw_params(struct snd_pcm_substream *substream,
 	dma_id = hdac_stream(stream)->stream_tag - 1;
 	dev_dbg(dai->dev, "dma_id=%d\n", dma_id);
 
+	p_params.s_fmt = snd_pcm_format_width(params_format(params));
+	p_params.ch = params_channels(params);
+	p_params.s_freq = params_rate(params);
+	p_params.host_dma_id = dma_id;
+	p_params.stream = substream->stream;
+
+	m_cfg = skl_tplg_fe_get_cpr_module(dai, p_params.stream);
+	if (m_cfg)
+		skl_tplg_update_pipe_params(dai->dev, m_cfg, &p_params);
+
 	return 0;
 }
 
@@ -268,6 +281,46 @@ static int skl_pcm_hw_free(struct snd_pcm_substream *substream,
 	return skl_substream_free_pages(ebus_to_hbus(ebus), substream);
 }
 
+static int skl_be_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct skl_pipe_params p_params = {0};
+
+	p_params.s_fmt = snd_pcm_format_width(params_format(params));
+	p_params.ch = params_channels(params);
+	p_params.s_freq = params_rate(params);
+	p_params.stream = substream->stream;
+	skl_tplg_be_update_params(dai, &p_params);
+
+	return 0;
+}
+
+static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
+		struct snd_soc_dai *dai)
+{
+	struct skl *skl = get_skl_ctx(dai->dev);
+	struct skl_sst *ctx = skl->skl_sst;
+	struct skl_module_cfg *mconfig;
+
+	mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream);
+	if (!mconfig)
+		return -EIO;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		return skl_run_pipe(ctx, mconfig->pipe);
+
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		return skl_stop_pipe(ctx, mconfig->pipe);
+
+	default:
+		return 0;
+	}
+}
+
 static int skl_link_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
@@ -277,9 +330,8 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
 	struct skl_dma_params *dma_params;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	int dma_id;
+	struct skl_pipe_params p_params = {0};
 
-	pr_debug("%s\n", __func__);
 	link_dev = snd_hdac_ext_stream_assign(ebus, substream,
 					HDAC_EXT_STREAM_TYPE_LINK);
 	if (!link_dev)
@@ -293,7 +345,14 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream,
 	if (dma_params)
 		dma_params->stream_tag =  hdac_stream(link_dev)->stream_tag;
 	snd_soc_dai_set_dma_data(codec_dai, substream, (void *)dma_params);
-	dma_id = hdac_stream(link_dev)->stream_tag - 1;
+
+	p_params.s_fmt = snd_pcm_format_width(params_format(params));
+	p_params.ch = params_channels(params);
+	p_params.s_freq = params_rate(params);
+	p_params.stream = substream->stream;
+	p_params.link_dma_id = hdac_stream(link_dev)->stream_tag - 1;
+
+	skl_tplg_be_update_params(dai, &p_params);
 
 	return 0;
 }
@@ -308,27 +367,12 @@ static int skl_link_pcm_prepare(struct snd_pcm_substream *substream,
 	unsigned int format_val = 0;
 	struct skl_dma_params *dma_params;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_pcm_hw_params *params;
-	struct snd_interval *channels, *rate;
 	struct hdac_ext_link *link;
 
-	dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
 	if (link_dev->link_prepared) {
 		dev_dbg(dai->dev, "already stream is prepared - returning\n");
 		return 0;
 	}
-	params  = devm_kzalloc(dai->dev, sizeof(*params), GFP_KERNEL);
-	if (params == NULL)
-		return -ENOMEM;
-
-	channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
-	channels->min = channels->max = substream->runtime->channels;
-	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
-	rate->min = rate->max = substream->runtime->rate;
-	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
-					SNDRV_PCM_HW_PARAM_FIRST_MASK],
-					substream->runtime->format);
-
 
 	dma_params  = (struct skl_dma_params *)
 			snd_soc_dai_get_dma_data(codec_dai, substream);
@@ -399,13 +443,13 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int skl_hda_be_startup(struct snd_pcm_substream *substream,
+static int skl_be_startup(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
 	return pm_runtime_get_sync(dai->dev);
 }
 
-static void skl_hda_be_shutdown(struct snd_pcm_substream *substream,
+static void skl_be_shutdown(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
 	pm_runtime_mark_last_busy(dai->dev);
@@ -418,20 +462,28 @@ static struct snd_soc_dai_ops skl_pcm_dai_ops = {
 	.prepare = skl_pcm_prepare,
 	.hw_params = skl_pcm_hw_params,
 	.hw_free = skl_pcm_hw_free,
+	.trigger = skl_pcm_trigger,
 };
 
 static struct snd_soc_dai_ops skl_dmic_dai_ops = {
-	.startup = skl_hda_be_startup,
-	.shutdown = skl_hda_be_shutdown,
+	.startup = skl_be_startup,
+	.hw_params = skl_be_hw_params,
+	.shutdown = skl_be_shutdown,
+};
+
+static struct snd_soc_dai_ops skl_be_ssp_dai_ops = {
+	.startup = skl_be_startup,
+	.hw_params = skl_be_hw_params,
+	.shutdown = skl_be_shutdown,
 };
 
 static struct snd_soc_dai_ops skl_link_dai_ops = {
-	.startup = skl_hda_be_startup,
+	.startup = skl_be_startup,
 	.prepare = skl_link_pcm_prepare,
 	.hw_params = skl_link_hw_params,
 	.hw_free = skl_link_hw_free,
 	.trigger = skl_link_pcm_trigger,
-	.shutdown = skl_hda_be_shutdown,
+	.shutdown = skl_be_shutdown,
 };
 
 static struct snd_soc_dai_driver skl_platform_dai[] = {
@@ -488,6 +540,24 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
 },
 /* BE CPU  Dais */
 {
+	.name = "SSP0 Pin",
+	.ops = &skl_be_ssp_dai_ops,
+	.playback = {
+		.stream_name = "ssp0 Tx",
+		.channels_min = HDA_STEREO,
+		.channels_max = HDA_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "ssp0 Rx",
+		.channels_min = HDA_STEREO,
+		.channels_max = HDA_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+{
 	.name = "iDisp Pin",
 	.ops = &skl_link_dai_ops,
 	.playback = {
@@ -510,17 +580,6 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
 	},
 },
 {
-	.name = "DMIC23 Pin",
-	.ops = &skl_dmic_dai_ops,
-	.capture = {
-		.stream_name = "DMIC23 Rx",
-		.channels_min = HDA_STEREO,
-		.channels_max = HDA_STEREO,
-		.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
-	},
-},
-{
 	.name = "HD-Codec Pin",
 	.ops = &skl_link_dai_ops,
 	.playback = {
@@ -538,28 +597,6 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 },
-{
-	.name = "HD-Codec-SPK Pin",
-	.ops = &skl_link_dai_ops,
-	.playback = {
-		.stream_name = "HD-Codec-SPK Tx",
-		.channels_min = HDA_STEREO,
-		.channels_max = HDA_STEREO,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
-	},
-},
-{
-	.name = "HD-Codec-AMIC Pin",
-	.ops = &skl_link_dai_ops,
-	.capture = {
-		.stream_name = "HD-Codec-AMIC Rx",
-		.channels_min = HDA_STEREO,
-		.channels_max = HDA_STEREO,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
-	},
-},
 };
 
 static int skl_platform_open(struct snd_pcm_substream *substream)
@@ -577,7 +614,7 @@ static int skl_platform_open(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int skl_pcm_trigger(struct snd_pcm_substream *substream,
+static int skl_coupled_trigger(struct snd_pcm_substream *substream,
 					int cmd)
 {
 	struct hdac_ext_bus *ebus = get_bus_ctx(substream);
@@ -651,7 +688,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int skl_dsp_trigger(struct snd_pcm_substream *substream,
+static int skl_decoupled_trigger(struct snd_pcm_substream *substream,
 		int cmd)
 {
 	struct hdac_ext_bus *ebus = get_bus_ctx(substream);
@@ -708,9 +745,9 @@ static int skl_platform_pcm_trigger(struct snd_pcm_substream *substream,
 	struct hdac_ext_bus *ebus = get_bus_ctx(substream);
 
 	if (ebus->ppcap)
-		return skl_dsp_trigger(substream, cmd);
+		return skl_decoupled_trigger(substream, cmd);
 	else
-		return skl_pcm_trigger(substream, cmd);
+		return skl_coupled_trigger(substream, cmd);
 }
 
 /* calculate runtime delay from LPIB */
@@ -877,7 +914,17 @@ static int skl_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	return retval;
 }
 
+static int skl_platform_soc_probe(struct snd_soc_platform *platform)
+{
+	struct hdac_ext_bus *ebus = dev_get_drvdata(platform->dev);
+
+	if (ebus->ppcap)
+		return skl_tplg_init(platform, ebus);
+
+	return 0;
+}
 static struct snd_soc_platform_driver skl_platform_drv  = {
+	.probe		= skl_platform_soc_probe,
 	.ops		= &skl_platform_ops,
 	.pcm_new	= skl_pcm_new,
 	.pcm_free	= skl_pcm_free,
@@ -890,6 +937,11 @@ static const struct snd_soc_component_driver skl_component = {
 int skl_platform_register(struct device *dev)
 {
 	int ret;
+	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+	struct skl *skl = ebus_to_skl(ebus);
+
+	INIT_LIST_HEAD(&skl->ppl_list);
+	INIT_LIST_HEAD(&skl->dapm_path_list);
 
 	ret = snd_soc_register_platform(dev, &skl_platform_drv);
 	if (ret) {
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c
index 94875b008b0b..1bfb7f63b572 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.c
+++ b/sound/soc/intel/skylake/skl-sst-dsp.c
@@ -175,7 +175,7 @@ static int skl_dsp_core_power_down(struct sst_dsp  *ctx)
 	/* poll with timeout to check if operation successful */
 	return sst_dsp_register_poll(ctx,
 			SKL_ADSP_REG_ADSPCS,
-			SKL_ADSPCS_SPA_MASK,
+			SKL_ADSPCS_CPA_MASK,
 			0,
 			SKL_DSP_PD_TO,
 			"Power down");
@@ -262,6 +262,11 @@ irqreturn_t skl_dsp_sst_interrupt(int irq, void *dev_id)
 	val = sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPIS);
 	ctx->intr_status = val;
 
+	if (val == 0xffffffff) {
+		spin_unlock(&ctx->spinlock);
+		return IRQ_NONE;
+	}
+
 	if (val & SKL_ADSPIS_IPC) {
 		skl_ipc_int_disable(ctx);
 		result = IRQ_WAKE_THREAD;
diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c
index 937a0a3a63a0..3345ea0d4414 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.c
+++ b/sound/soc/intel/skylake/skl-sst-ipc.c
@@ -464,6 +464,18 @@ void skl_ipc_op_int_enable(struct sst_dsp *ctx)
 		SKL_ADSP_REG_HIPCCTL_BUSY, SKL_ADSP_REG_HIPCCTL_BUSY);
 }
 
+void skl_ipc_op_int_disable(struct sst_dsp *ctx)
+{
+	/* disable IPC DONE interrupt */
+	sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_HIPCCTL,
+					SKL_ADSP_REG_HIPCCTL_DONE, 0);
+
+	/* Disable IPC BUSY interrupt */
+	sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_HIPCCTL,
+					SKL_ADSP_REG_HIPCCTL_BUSY, 0);
+
+}
+
 bool skl_ipc_int_status(struct sst_dsp *ctx)
 {
 	return sst_dsp_shim_read_unlocked(ctx,
diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h
index 9f5f67202858..f1a154e45dc3 100644
--- a/sound/soc/intel/skylake/skl-sst-ipc.h
+++ b/sound/soc/intel/skylake/skl-sst-ipc.h
@@ -116,6 +116,7 @@ int skl_ipc_set_large_config(struct sst_generic_ipc *ipc,
 
 void skl_ipc_int_enable(struct sst_dsp *dsp);
 void skl_ipc_op_int_enable(struct sst_dsp *ctx);
+void skl_ipc_op_int_disable(struct sst_dsp *ctx);
 void skl_ipc_int_disable(struct sst_dsp *dsp);
 
 bool skl_ipc_int_status(struct sst_dsp *dsp);
diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c
index c18ea51b7484..3b83dc99f1d4 100644
--- a/sound/soc/intel/skylake/skl-sst.c
+++ b/sound/soc/intel/skylake/skl-sst.c
@@ -70,15 +70,31 @@ static int skl_transfer_firmware(struct sst_dsp *ctx,
 static int skl_load_base_firmware(struct sst_dsp *ctx)
 {
 	int ret = 0, i;
-	const struct firmware *fw = NULL;
 	struct skl_sst *skl = ctx->thread_context;
 	u32 reg;
 
-	ret = request_firmware(&fw, "dsp_fw_release.bin", ctx->dev);
+	skl->boot_complete = false;
+	init_waitqueue_head(&skl->boot_wait);
+
+	if (ctx->fw == NULL) {
+		ret = request_firmware(&ctx->fw, "dsp_fw_release.bin", ctx->dev);
+		if (ret < 0) {
+			dev_err(ctx->dev, "Request firmware failed %d\n", ret);
+			skl_dsp_disable_core(ctx);
+			return -EIO;
+		}
+	}
+
+	ret = skl_dsp_boot(ctx);
 	if (ret < 0) {
-		dev_err(ctx->dev, "Request firmware failed %d\n", ret);
-		skl_dsp_disable_core(ctx);
-		return -EIO;
+		dev_err(ctx->dev, "Boot dsp core failed ret: %d", ret);
+		goto skl_load_base_firmware_failed;
+	}
+
+	ret = skl_cldma_prepare(ctx);
+	if (ret < 0) {
+		dev_err(ctx->dev, "CL dma prepare failed : %d", ret);
+		goto skl_load_base_firmware_failed;
 	}
 
 	/* enable Interrupt */
@@ -102,7 +118,7 @@ static int skl_load_base_firmware(struct sst_dsp *ctx)
 		goto skl_load_base_firmware_failed;
 	}
 
-	ret = skl_transfer_firmware(ctx, fw->data, fw->size);
+	ret = skl_transfer_firmware(ctx, ctx->fw->data, ctx->fw->size);
 	if (ret < 0) {
 		dev_err(ctx->dev, "Transfer firmware failed%d\n", ret);
 		goto skl_load_base_firmware_failed;
@@ -118,13 +134,12 @@ static int skl_load_base_firmware(struct sst_dsp *ctx)
 		dev_dbg(ctx->dev, "Download firmware successful%d\n", ret);
 		skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING);
 	}
-	release_firmware(fw);
-
 	return 0;
 
 skl_load_base_firmware_failed:
 	skl_dsp_disable_core(ctx);
-	release_firmware(fw);
+	release_firmware(ctx->fw);
+	ctx->fw = NULL;
 	return ret;
 }
 
@@ -172,6 +187,12 @@ static int skl_set_dsp_D3(struct sst_dsp *ctx)
 	}
 	skl_dsp_set_state_locked(ctx, SKL_DSP_RESET);
 
+	/* disable Interrupt */
+	ctx->cl_dev.ops.cl_cleanup_controller(ctx);
+	skl_cldma_int_disable(ctx);
+	skl_ipc_op_int_disable(ctx);
+	skl_ipc_int_disable(ctx);
+
 	return ret;
 }
 
@@ -235,22 +256,6 @@ int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 	if (ret)
 		return ret;
 
-	skl->boot_complete = false;
-	init_waitqueue_head(&skl->boot_wait);
-
-	ret = skl_dsp_boot(sst);
-	if (ret < 0) {
-		dev_err(skl->dev, "Boot dsp core failed ret: %d", ret);
-		goto free_ipc;
-	}
-
-	ret = skl_cldma_prepare(sst);
-	if (ret < 0) {
-		dev_err(dev, "CL dma prepare failed : %d", ret);
-		goto free_ipc;
-	}
-
-
 	ret = sst->fw_ops.load_fw(sst);
 	if (ret < 0) {
 		dev_err(dev, "Load base fw failed : %d", ret);
@@ -262,7 +267,6 @@ int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 
 	return 0;
 
-free_ipc:
 	skl_ipc_free(&skl->ipc);
 	return ret;
 }
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
new file mode 100644
index 000000000000..a7854c8fc523
--- /dev/null
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -0,0 +1,1252 @@
+/*
+ *  skl-topology.c - Implements Platform component ALSA controls/widget
+ *  handlers.
+ *
+ *  Copyright (C) 2014-2015 Intel Corp
+ *  Author: Jeeja KP <jeeja.kp@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/slab.h>
+#include <linux/types.h>
+#include <linux/firmware.h>
+#include <sound/soc.h>
+#include <sound/soc-topology.h>
+#include "skl-sst-dsp.h"
+#include "skl-sst-ipc.h"
+#include "skl-topology.h"
+#include "skl.h"
+#include "skl-tplg-interface.h"
+
+#define SKL_CH_FIXUP_MASK		(1 << 0)
+#define SKL_RATE_FIXUP_MASK		(1 << 1)
+#define SKL_FMT_FIXUP_MASK		(1 << 2)
+
+/*
+ * SKL DSP driver modelling uses only few DAPM widgets so for rest we will
+ * ignore. This helpers checks if the SKL driver handles this widget type
+ */
+static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w)
+{
+	switch (w->id) {
+	case snd_soc_dapm_dai_link:
+	case snd_soc_dapm_dai_in:
+	case snd_soc_dapm_aif_in:
+	case snd_soc_dapm_aif_out:
+	case snd_soc_dapm_dai_out:
+	case snd_soc_dapm_switch:
+		return false;
+	default:
+		return true;
+	}
+}
+
+/*
+ * Each pipelines needs memory to be allocated. Check if we have free memory
+ * from available pool. Then only add this to pool
+ * This is freed when pipe is deleted
+ * Note: DSP does actual memory management we only keep track for complete
+ * pool
+ */
+static bool skl_tplg_alloc_pipe_mem(struct skl *skl,
+				struct skl_module_cfg *mconfig)
+{
+	struct skl_sst *ctx = skl->skl_sst;
+
+	if (skl->resource.mem + mconfig->pipe->memory_pages >
+				skl->resource.max_mem) {
+		dev_err(ctx->dev,
+				"%s: module_id %d instance %d\n", __func__,
+				mconfig->id.module_id,
+				mconfig->id.instance_id);
+		dev_err(ctx->dev,
+				"exceeds ppl memory available %d mem %d\n",
+				skl->resource.max_mem, skl->resource.mem);
+		return false;
+	}
+
+	skl->resource.mem += mconfig->pipe->memory_pages;
+	return true;
+}
+
+/*
+ * Pipeline needs needs DSP CPU resources for computation, this is
+ * quantified in MCPS (Million Clocks Per Second) required for module/pipe
+ *
+ * Each pipelines needs mcps to be allocated. Check if we have mcps for this
+ * pipe. This adds the mcps to driver counter
+ * This is removed on pipeline delete
+ */
+static bool skl_tplg_alloc_pipe_mcps(struct skl *skl,
+				struct skl_module_cfg *mconfig)
+{
+	struct skl_sst *ctx = skl->skl_sst;
+
+	if (skl->resource.mcps + mconfig->mcps > skl->resource.max_mcps) {
+		dev_err(ctx->dev,
+			"%s: module_id %d instance %d\n", __func__,
+			mconfig->id.module_id, mconfig->id.instance_id);
+		dev_err(ctx->dev,
+			"exceeds ppl memory available %d > mem %d\n",
+			skl->resource.max_mcps, skl->resource.mcps);
+		return false;
+	}
+
+	skl->resource.mcps += mconfig->mcps;
+	return true;
+}
+
+/*
+ * Free the mcps when tearing down
+ */
+static void
+skl_tplg_free_pipe_mcps(struct skl *skl, struct skl_module_cfg *mconfig)
+{
+	skl->resource.mcps -= mconfig->mcps;
+}
+
+/*
+ * Free the memory when tearing down
+ */
+static void
+skl_tplg_free_pipe_mem(struct skl *skl, struct skl_module_cfg *mconfig)
+{
+	skl->resource.mem -= mconfig->pipe->memory_pages;
+}
+
+
+static void skl_dump_mconfig(struct skl_sst *ctx,
+					struct skl_module_cfg *mcfg)
+{
+	dev_dbg(ctx->dev, "Dumping config\n");
+	dev_dbg(ctx->dev, "Input Format:\n");
+	dev_dbg(ctx->dev, "channels = %d\n", mcfg->in_fmt.channels);
+	dev_dbg(ctx->dev, "s_freq = %d\n", mcfg->in_fmt.s_freq);
+	dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->in_fmt.ch_cfg);
+	dev_dbg(ctx->dev, "valid bit depth = %d\n",
+			mcfg->in_fmt.valid_bit_depth);
+	dev_dbg(ctx->dev, "Output Format:\n");
+	dev_dbg(ctx->dev, "channels = %d\n", mcfg->out_fmt.channels);
+	dev_dbg(ctx->dev, "s_freq = %d\n", mcfg->out_fmt.s_freq);
+	dev_dbg(ctx->dev, "valid bit depth = %d\n",
+			mcfg->out_fmt.valid_bit_depth);
+	dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->out_fmt.ch_cfg);
+}
+
+static void skl_tplg_update_params(struct skl_module_fmt *fmt,
+			struct skl_pipe_params *params, int fixup)
+{
+	if (fixup & SKL_RATE_FIXUP_MASK)
+		fmt->s_freq = params->s_freq;
+	if (fixup & SKL_CH_FIXUP_MASK)
+		fmt->channels = params->ch;
+	if (fixup & SKL_FMT_FIXUP_MASK)
+		fmt->valid_bit_depth = params->s_fmt;
+}
+
+/*
+ * A pipeline may have modules which impact the pcm parameters, like SRC,
+ * channel converter, format converter.
+ * We need to calculate the output params by applying the 'fixup'
+ * Topology will tell driver which type of fixup is to be applied by
+ * supplying the fixup mask, so based on that we calculate the output
+ *
+ * Now In FE the pcm hw_params is source/target format. Same is applicable
+ * for BE with its hw_params invoked.
+ * here based on FE, BE pipeline and direction we calculate the input and
+ * outfix and then apply that for a module
+ */
+static void skl_tplg_update_params_fixup(struct skl_module_cfg *m_cfg,
+		struct skl_pipe_params *params, bool is_fe)
+{
+	int in_fixup, out_fixup;
+	struct skl_module_fmt *in_fmt, *out_fmt;
+
+	in_fmt = &m_cfg->in_fmt;
+	out_fmt = &m_cfg->out_fmt;
+
+	if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		if (is_fe) {
+			in_fixup = m_cfg->params_fixup;
+			out_fixup = (~m_cfg->converter) &
+					m_cfg->params_fixup;
+		} else {
+			out_fixup = m_cfg->params_fixup;
+			in_fixup = (~m_cfg->converter) &
+					m_cfg->params_fixup;
+		}
+	} else {
+		if (is_fe) {
+			out_fixup = m_cfg->params_fixup;
+			in_fixup = (~m_cfg->converter) &
+					m_cfg->params_fixup;
+		} else {
+			in_fixup = m_cfg->params_fixup;
+			out_fixup = (~m_cfg->converter) &
+					m_cfg->params_fixup;
+		}
+	}
+
+	skl_tplg_update_params(in_fmt, params, in_fixup);
+	skl_tplg_update_params(out_fmt, params, out_fixup);
+}
+
+/*
+ * A module needs input and output buffers, which are dependent upon pcm
+ * params, so once we have calculate params, we need buffer calculation as
+ * well.
+ */
+static void skl_tplg_update_buffer_size(struct skl_sst *ctx,
+				struct skl_module_cfg *mcfg)
+{
+	int multiplier = 1;
+
+	if (mcfg->m_type == SKL_MODULE_TYPE_SRCINT)
+		multiplier = 5;
+
+	mcfg->ibs = (mcfg->in_fmt.s_freq / 1000) *
+				(mcfg->in_fmt.channels) *
+				(mcfg->in_fmt.bit_depth >> 3) *
+				multiplier;
+
+	mcfg->obs = (mcfg->out_fmt.s_freq / 1000) *
+				(mcfg->out_fmt.channels) *
+				(mcfg->out_fmt.bit_depth >> 3) *
+				multiplier;
+}
+
+static void skl_tplg_update_module_params(struct snd_soc_dapm_widget *w,
+							struct skl_sst *ctx)
+{
+	struct skl_module_cfg *m_cfg = w->priv;
+	struct skl_pipe_params *params = m_cfg->pipe->p_params;
+	int p_conn_type = m_cfg->pipe->conn_type;
+	bool is_fe;
+
+	if (!m_cfg->params_fixup)
+		return;
+
+	dev_dbg(ctx->dev, "Mconfig for widget=%s BEFORE updation\n",
+				w->name);
+
+	skl_dump_mconfig(ctx, m_cfg);
+
+	if (p_conn_type == SKL_PIPE_CONN_TYPE_FE)
+		is_fe = true;
+	else
+		is_fe = false;
+
+	skl_tplg_update_params_fixup(m_cfg, params, is_fe);
+	skl_tplg_update_buffer_size(ctx, m_cfg);
+
+	dev_dbg(ctx->dev, "Mconfig for widget=%s AFTER updation\n",
+				w->name);
+
+	skl_dump_mconfig(ctx, m_cfg);
+}
+
+/*
+ * A pipe can have multiple modules, each of them will be a DAPM widget as
+ * well. While managing a pipeline we need to get the list of all the
+ * widgets in a pipelines, so this helper - skl_tplg_get_pipe_widget() helps
+ * to get the SKL type widgets in that pipeline
+ */
+static int skl_tplg_alloc_pipe_widget(struct device *dev,
+	struct snd_soc_dapm_widget *w, struct skl_pipe *pipe)
+{
+	struct skl_module_cfg *src_module = NULL;
+	struct snd_soc_dapm_path *p = NULL;
+	struct skl_pipe_module *p_module = NULL;
+
+	p_module = devm_kzalloc(dev, sizeof(*p_module), GFP_KERNEL);
+	if (!p_module)
+		return -ENOMEM;
+
+	p_module->w = w;
+	list_add_tail(&p_module->node, &pipe->w_list);
+
+	snd_soc_dapm_widget_for_each_sink_path(w, p) {
+		if ((p->sink->priv == NULL)
+				&& (!is_skl_dsp_widget_type(w)))
+			continue;
+
+		if ((p->sink->priv != NULL) && p->connect
+				&& is_skl_dsp_widget_type(p->sink)) {
+
+			src_module = p->sink->priv;
+			if (pipe->ppl_id == src_module->pipe->ppl_id)
+				skl_tplg_alloc_pipe_widget(dev,
+							p->sink, pipe);
+		}
+	}
+	return 0;
+}
+
+/*
+ * Inside a pipe instance, we can have various modules. These modules need
+ * to instantiated in DSP by invoking INIT_MODULE IPC, which is achieved by
+ * skl_init_module() routine, so invoke that for all modules in a pipeline
+ */
+static int
+skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
+{
+	struct skl_pipe_module *w_module;
+	struct snd_soc_dapm_widget *w;
+	struct skl_module_cfg *mconfig;
+	struct skl_sst *ctx = skl->skl_sst;
+	int ret = 0;
+
+	list_for_each_entry(w_module, &pipe->w_list, node) {
+		w = w_module->w;
+		mconfig = w->priv;
+
+		/* check resource available */
+		if (!skl_tplg_alloc_pipe_mcps(skl, mconfig))
+			return -ENOMEM;
+
+		/*
+		 * apply fix/conversion to module params based on
+		 * FE/BE params
+		 */
+		skl_tplg_update_module_params(w, ctx);
+		ret = skl_init_module(ctx, mconfig, NULL);
+		if (ret < 0)
+			return ret;
+	}
+
+	return 0;
+}
+
+/*
+ * Mixer module represents a pipeline. So in the Pre-PMU event of mixer we
+ * need create the pipeline. So we do following:
+ *   - check the resources
+ *   - Create the pipeline
+ *   - Initialize the modules in pipeline
+ *   - finally bind all modules together
+ */
+static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
+							struct skl *skl)
+{
+	int ret;
+	struct skl_module_cfg *mconfig = w->priv;
+	struct skl_pipe_module *w_module;
+	struct skl_pipe *s_pipe = mconfig->pipe;
+	struct skl_module_cfg *src_module = NULL, *dst_module;
+	struct skl_sst *ctx = skl->skl_sst;
+
+	/* check resource available */
+	if (!skl_tplg_alloc_pipe_mcps(skl, mconfig))
+		return -EBUSY;
+
+	if (!skl_tplg_alloc_pipe_mem(skl, mconfig))
+		return -ENOMEM;
+
+	/*
+	 * Create a list of modules for pipe.
+	 * This list contains modules from source to sink
+	 */
+	ret = skl_create_pipeline(ctx, mconfig->pipe);
+	if (ret < 0)
+		return ret;
+
+	/*
+	 * we create a w_list of all widgets in that pipe. This list is not
+	 * freed on PMD event as widgets within a pipe are static. This
+	 * saves us cycles to get widgets in pipe every time.
+	 *
+	 * So if we have already initialized all the widgets of a pipeline
+	 * we skip, so check for list_empty and create the list if empty
+	 */
+	if (list_empty(&s_pipe->w_list)) {
+		ret = skl_tplg_alloc_pipe_widget(ctx->dev, w, s_pipe);
+		if (ret < 0)
+			return ret;
+	}
+
+	/* Init all pipe modules from source to sink */
+	ret = skl_tplg_init_pipe_modules(skl, s_pipe);
+	if (ret < 0)
+		return ret;
+
+	/* Bind modules from source to sink */
+	list_for_each_entry(w_module, &s_pipe->w_list, node) {
+		dst_module = w_module->w->priv;
+
+		if (src_module == NULL) {
+			src_module = dst_module;
+			continue;
+		}
+
+		ret = skl_bind_modules(ctx, src_module, dst_module);
+		if (ret < 0)
+			return ret;
+
+		src_module = dst_module;
+	}
+
+	return 0;
+}
+
+/*
+ * A PGA represents a module in a pipeline. So in the Pre-PMU event of PGA
+ * we need to do following:
+ *   - Bind to sink pipeline
+ *      Since the sink pipes can be running and we don't get mixer event on
+ *      connect for already running mixer, we need to find the sink pipes
+ *      here and bind to them. This way dynamic connect works.
+ *   - Start sink pipeline, if not running
+ *   - Then run current pipe
+ */
+static int skl_tplg_pga_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
+							struct skl *skl)
+{
+	struct snd_soc_dapm_path *p;
+	struct skl_dapm_path_list *path_list;
+	struct snd_soc_dapm_widget *source, *sink;
+	struct skl_module_cfg *src_mconfig, *sink_mconfig;
+	struct skl_sst *ctx = skl->skl_sst;
+	int ret = 0;
+
+	source = w;
+	src_mconfig = source->priv;
+
+	/*
+	 * find which sink it is connected to, bind with the sink,
+	 * if sink is not started, start sink pipe first, then start
+	 * this pipe
+	 */
+	snd_soc_dapm_widget_for_each_source_path(w, p) {
+		if (!p->connect)
+			continue;
+
+		dev_dbg(ctx->dev, "%s: src widget=%s\n", __func__, w->name);
+		dev_dbg(ctx->dev, "%s: sink widget=%s\n", __func__, p->sink->name);
+
+		/*
+		 * here we will check widgets in sink pipelines, so that
+		 * can be any widgets type and we are only interested if
+		 * they are ones used for SKL so check that first
+		 */
+		if ((p->sink->priv != NULL) &&
+					is_skl_dsp_widget_type(p->sink)) {
+
+			sink = p->sink;
+			src_mconfig = source->priv;
+			sink_mconfig = sink->priv;
+
+			/* Bind source to sink, mixin is always source */
+			ret = skl_bind_modules(ctx, src_mconfig, sink_mconfig);
+			if (ret)
+				return ret;
+
+			/* Start sinks pipe first */
+			if (sink_mconfig->pipe->state != SKL_PIPE_STARTED) {
+				ret = skl_run_pipe(ctx, sink_mconfig->pipe);
+				if (ret)
+					return ret;
+			}
+
+			path_list = kzalloc(
+					sizeof(struct skl_dapm_path_list),
+					GFP_KERNEL);
+			if (path_list == NULL)
+				return -ENOMEM;
+
+			/* Add connected path to one global list */
+			path_list->dapm_path = p;
+			list_add_tail(&path_list->node, &skl->dapm_path_list);
+			break;
+		}
+	}
+
+	/* Start source pipe last after starting all sinks */
+	ret = skl_run_pipe(ctx, src_mconfig->pipe);
+	if (ret)
+		return ret;
+
+	return 0;
+}
+
+/*
+ * in the Post-PMU event of mixer we need to do following:
+ *   - Check if this pipe is running
+ *   - if not, then
+ *	- bind this pipeline to its source pipeline
+ *	  if source pipe is already running, this means it is a dynamic
+ *	  connection and we need to bind only to that pipe
+ *	- start this pipeline
+ */
+static int skl_tplg_mixer_dapm_post_pmu_event(struct snd_soc_dapm_widget *w,
+							struct skl *skl)
+{
+	int ret = 0;
+	struct snd_soc_dapm_path *p;
+	struct snd_soc_dapm_widget *source, *sink;
+	struct skl_module_cfg *src_mconfig, *sink_mconfig;
+	struct skl_sst *ctx = skl->skl_sst;
+	int src_pipe_started = 0;
+
+	sink = w;
+	sink_mconfig = sink->priv;
+
+	/*
+	 * If source pipe is already started, that means source is driving
+	 * one more sink before this sink got connected, Since source is
+	 * started, bind this sink to source and start this pipe.
+	 */
+	snd_soc_dapm_widget_for_each_sink_path(w, p) {
+		if (!p->connect)
+			continue;
+
+		dev_dbg(ctx->dev, "sink widget=%s\n", w->name);
+		dev_dbg(ctx->dev, "src widget=%s\n", p->source->name);
+
+		/*
+		 * here we will check widgets in sink pipelines, so that
+		 * can be any widgets type and we are only interested if
+		 * they are ones used for SKL so check that first
+		 */
+		if ((p->source->priv != NULL) &&
+					is_skl_dsp_widget_type(p->source)) {
+			source = p->source;
+			src_mconfig = source->priv;
+			sink_mconfig = sink->priv;
+			src_pipe_started = 1;
+
+			/*
+			 * check pipe state, then no need to bind or start
+			 * the pipe
+			 */
+			if (src_mconfig->pipe->state != SKL_PIPE_STARTED)
+				src_pipe_started = 0;
+		}
+	}
+
+	if (src_pipe_started) {
+		ret = skl_bind_modules(ctx, src_mconfig, sink_mconfig);
+		if (ret)
+			return ret;
+
+		ret = skl_run_pipe(ctx, sink_mconfig->pipe);
+	}
+
+	return ret;
+}
+
+/*
+ * in the Pre-PMD event of mixer we need to do following:
+ *   - Stop the pipe
+ *   - find the source connections and remove that from dapm_path_list
+ *   - unbind with source pipelines if still connected
+ */
+static int skl_tplg_mixer_dapm_pre_pmd_event(struct snd_soc_dapm_widget *w,
+							struct skl *skl)
+{
+	struct snd_soc_dapm_widget *source, *sink;
+	struct skl_module_cfg *src_mconfig, *sink_mconfig;
+	int ret = 0, path_found = 0;
+	struct skl_dapm_path_list *path_list, *tmp_list;
+	struct skl_sst *ctx = skl->skl_sst;
+
+	sink = w;
+	sink_mconfig = sink->priv;
+
+	/* Stop the pipe */
+	ret = skl_stop_pipe(ctx, sink_mconfig->pipe);
+	if (ret)
+		return ret;
+
+	/*
+	 * This list, dapm_path_list handling here does not need any locks
+	 * as we are under dapm lock while handling widget events.
+	 * List can be manipulated safely only under dapm widgets handler
+	 * routines
+	 */
+	list_for_each_entry_safe(path_list, tmp_list,
+				&skl->dapm_path_list, node) {
+		if (path_list->dapm_path->sink == sink) {
+			dev_dbg(ctx->dev, "Path found = %s\n",
+					path_list->dapm_path->name);
+			source = path_list->dapm_path->source;
+			src_mconfig = source->priv;
+			path_found = 1;
+
+			list_del(&path_list->node);
+			kfree(path_list);
+			break;
+		}
+	}
+
+	/*
+	 * If path_found == 1, that means pmd for source pipe has
+	 * not occurred, source is connected to some other sink.
+	 * so its responsibility of sink to unbind itself from source.
+	 */
+	if (path_found) {
+		ret = skl_stop_pipe(ctx, src_mconfig->pipe);
+		if (ret < 0)
+			return ret;
+
+		ret = skl_unbind_modules(ctx, src_mconfig, sink_mconfig);
+	}
+
+	return ret;
+}
+
+/*
+ * in the Post-PMD event of mixer we need to do following:
+ *   - Free the mcps used
+ *   - Free the mem used
+ *   - Unbind the modules within the pipeline
+ *   - Delete the pipeline (modules are not required to be explicitly
+ *     deleted, pipeline delete is enough here
+ */
+static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
+							struct skl *skl)
+{
+	struct skl_module_cfg *mconfig = w->priv;
+	struct skl_pipe_module *w_module;
+	struct skl_module_cfg *src_module = NULL, *dst_module;
+	struct skl_sst *ctx = skl->skl_sst;
+	struct skl_pipe *s_pipe = mconfig->pipe;
+	int ret = 0;
+
+	skl_tplg_free_pipe_mcps(skl, mconfig);
+
+	list_for_each_entry(w_module, &s_pipe->w_list, node) {
+		dst_module = w_module->w->priv;
+
+		if (src_module == NULL) {
+			src_module = dst_module;
+			continue;
+		}
+
+		ret = skl_unbind_modules(ctx, src_module, dst_module);
+		if (ret < 0)
+			return ret;
+
+		src_module = dst_module;
+	}
+
+	ret = skl_delete_pipe(ctx, mconfig->pipe);
+	skl_tplg_free_pipe_mem(skl, mconfig);
+
+	return ret;
+}
+
+/*
+ * in the Post-PMD event of PGA we need to do following:
+ *   - Free the mcps used
+ *   - Stop the pipeline
+ *   - In source pipe is connected, unbind with source pipelines
+ */
+static int skl_tplg_pga_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
+								struct skl *skl)
+{
+	struct snd_soc_dapm_widget *source, *sink;
+	struct skl_module_cfg *src_mconfig, *sink_mconfig;
+	int ret = 0, path_found = 0;
+	struct skl_dapm_path_list *path_list, *tmp_path_list;
+	struct skl_sst *ctx = skl->skl_sst;
+
+	source = w;
+	src_mconfig = source->priv;
+
+	skl_tplg_free_pipe_mcps(skl, src_mconfig);
+	/* Stop the pipe since this is a mixin module */
+	ret = skl_stop_pipe(ctx, src_mconfig->pipe);
+	if (ret)
+		return ret;
+
+	list_for_each_entry_safe(path_list, tmp_path_list, &skl->dapm_path_list, node) {
+		if (path_list->dapm_path->source == source) {
+			dev_dbg(ctx->dev, "Path found = %s\n",
+					path_list->dapm_path->name);
+			sink = path_list->dapm_path->sink;
+			sink_mconfig = sink->priv;
+			path_found = 1;
+
+			list_del(&path_list->node);
+			kfree(path_list);
+			break;
+		}
+	}
+
+	/*
+	 * This is a connector and if path is found that means
+	 * unbind between source and sink has not happened yet
+	 */
+	if (path_found) {
+		ret = skl_stop_pipe(ctx, src_mconfig->pipe);
+		if (ret < 0)
+			return ret;
+
+		ret = skl_unbind_modules(ctx, src_mconfig, sink_mconfig);
+	}
+
+	return ret;
+}
+
+/*
+ * In modelling, we assume there will be ONLY one mixer in a pipeline.  If
+ * mixer is not required then it is treated as static mixer aka vmixer with
+ * a hard path to source module
+ * So we don't need to check if source is started or not as hard path puts
+ * dependency on each other
+ */
+static int skl_tplg_vmixer_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *k, int event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct skl *skl = get_skl_ctx(dapm->dev);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		return skl_tplg_mixer_dapm_pre_pmu_event(w, skl);
+
+	case SND_SOC_DAPM_POST_PMD:
+		return skl_tplg_mixer_dapm_post_pmd_event(w, skl);
+	}
+
+	return 0;
+}
+
+/*
+ * In modelling, we assume there will be ONLY one mixer in a pipeline. If a
+ * second one is required that is created as another pipe entity.
+ * The mixer is responsible for pipe management and represent a pipeline
+ * instance
+ */
+static int skl_tplg_mixer_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *k, int event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct skl *skl = get_skl_ctx(dapm->dev);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		return skl_tplg_mixer_dapm_pre_pmu_event(w, skl);
+
+	case SND_SOC_DAPM_POST_PMU:
+		return skl_tplg_mixer_dapm_post_pmu_event(w, skl);
+
+	case SND_SOC_DAPM_PRE_PMD:
+		return skl_tplg_mixer_dapm_pre_pmd_event(w, skl);
+
+	case SND_SOC_DAPM_POST_PMD:
+		return skl_tplg_mixer_dapm_post_pmd_event(w, skl);
+	}
+
+	return 0;
+}
+
+/*
+ * In modelling, we assumed rest of the modules in pipeline are PGA. But we
+ * are interested in last PGA (leaf PGA) in a pipeline to disconnect with
+ * the sink when it is running (two FE to one BE or one FE to two BE)
+ * scenarios
+ */
+static int skl_tplg_pga_event(struct snd_soc_dapm_widget *w,
+			struct snd_kcontrol *k, int event)
+
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct skl *skl = get_skl_ctx(dapm->dev);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		return skl_tplg_pga_dapm_pre_pmu_event(w, skl);
+
+	case SND_SOC_DAPM_POST_PMD:
+		return skl_tplg_pga_dapm_post_pmd_event(w, skl);
+	}
+
+	return 0;
+}
+
+/*
+ * The FE params are passed by hw_params of the DAI.
+ * On hw_params, the params are stored in Gateway module of the FE and we
+ * need to calculate the format in DSP module configuration, that
+ * conversion is done here
+ */
+int skl_tplg_update_pipe_params(struct device *dev,
+			struct skl_module_cfg *mconfig,
+			struct skl_pipe_params *params)
+{
+	struct skl_pipe *pipe = mconfig->pipe;
+	struct skl_module_fmt *format = NULL;
+
+	memcpy(pipe->p_params, params, sizeof(*params));
+
+	if (params->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		format = &mconfig->in_fmt;
+	else
+		format = &mconfig->out_fmt;
+
+	/* set the hw_params */
+	format->s_freq = params->s_freq;
+	format->channels = params->ch;
+	format->valid_bit_depth = skl_get_bit_depth(params->s_fmt);
+
+	/*
+	 * 16 bit is 16 bit container whereas 24 bit is in 32 bit
+	 * container so update bit depth accordingly
+	 */
+	switch (format->valid_bit_depth) {
+	case SKL_DEPTH_16BIT:
+		format->bit_depth = format->valid_bit_depth;
+		break;
+
+	case SKL_DEPTH_24BIT:
+		format->bit_depth = SKL_DEPTH_32BIT;
+		break;
+
+	default:
+		dev_err(dev, "Invalid bit depth %x for pipe\n",
+				format->valid_bit_depth);
+		return -EINVAL;
+	}
+
+	if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		mconfig->ibs = (format->s_freq / 1000) *
+				(format->channels) *
+				(format->bit_depth >> 3);
+	} else {
+		mconfig->obs = (format->s_freq / 1000) *
+				(format->channels) *
+				(format->bit_depth >> 3);
+	}
+
+	return 0;
+}
+
+/*
+ * Query the module config for the FE DAI
+ * This is used to find the hw_params set for that DAI and apply to FE
+ * pipeline
+ */
+struct skl_module_cfg *
+skl_tplg_fe_get_cpr_module(struct snd_soc_dai *dai, int stream)
+{
+	struct snd_soc_dapm_widget *w;
+	struct snd_soc_dapm_path *p = NULL;
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		w = dai->playback_widget;
+		snd_soc_dapm_widget_for_each_sink_path(w, p) {
+			if (p->connect && p->sink->power &&
+					is_skl_dsp_widget_type(p->sink))
+				continue;
+
+			if (p->sink->priv) {
+				dev_dbg(dai->dev, "set params for %s\n",
+						p->sink->name);
+				return p->sink->priv;
+			}
+		}
+	} else {
+		w = dai->capture_widget;
+		snd_soc_dapm_widget_for_each_source_path(w, p) {
+			if (p->connect && p->source->power &&
+					is_skl_dsp_widget_type(p->source))
+				continue;
+
+			if (p->source->priv) {
+				dev_dbg(dai->dev, "set params for %s\n",
+						p->source->name);
+				return p->source->priv;
+			}
+		}
+	}
+
+	return NULL;
+}
+
+static u8 skl_tplg_be_link_type(int dev_type)
+{
+	int ret;
+
+	switch (dev_type) {
+	case SKL_DEVICE_BT:
+		ret = NHLT_LINK_SSP;
+		break;
+
+	case SKL_DEVICE_DMIC:
+		ret = NHLT_LINK_DMIC;
+		break;
+
+	case SKL_DEVICE_I2S:
+		ret = NHLT_LINK_SSP;
+		break;
+
+	case SKL_DEVICE_HDALINK:
+		ret = NHLT_LINK_HDA;
+		break;
+
+	default:
+		ret = NHLT_LINK_INVALID;
+		break;
+	}
+
+	return ret;
+}
+
+/*
+ * Fill the BE gateway parameters
+ * The BE gateway expects a blob of parameters which are kept in the ACPI
+ * NHLT blob, so query the blob for interface type (i2s/pdm) and instance.
+ * The port can have multiple settings so pick based on the PCM
+ * parameters
+ */
+static int skl_tplg_be_fill_pipe_params(struct snd_soc_dai *dai,
+				struct skl_module_cfg *mconfig,
+				struct skl_pipe_params *params)
+{
+	struct skl_pipe *pipe = mconfig->pipe;
+	struct nhlt_specific_cfg *cfg;
+	struct skl *skl = get_skl_ctx(dai->dev);
+	int link_type = skl_tplg_be_link_type(mconfig->dev_type);
+
+	memcpy(pipe->p_params, params, sizeof(*params));
+
+	/* update the blob based on virtual bus_id*/
+	cfg = skl_get_ep_blob(skl, mconfig->vbus_id, link_type,
+					params->s_fmt, params->ch,
+					params->s_freq, params->stream);
+	if (cfg) {
+		mconfig->formats_config.caps_size = cfg->size;
+		mconfig->formats_config.caps = (u32 *) &cfg->caps;
+	} else {
+		dev_err(dai->dev, "Blob NULL for id %x type %d dirn %d\n",
+					mconfig->vbus_id, link_type,
+					params->stream);
+		dev_err(dai->dev, "PCM: ch %d, freq %d, fmt %d\n",
+				 params->ch, params->s_freq, params->s_fmt);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int skl_tplg_be_set_src_pipe_params(struct snd_soc_dai *dai,
+				struct snd_soc_dapm_widget *w,
+				struct skl_pipe_params *params)
+{
+	struct snd_soc_dapm_path *p;
+	int ret = -EIO;
+
+	snd_soc_dapm_widget_for_each_source_path(w, p) {
+		if (p->connect && is_skl_dsp_widget_type(p->source) &&
+						p->source->priv) {
+
+			if (!p->source->power) {
+				ret = skl_tplg_be_fill_pipe_params(
+						dai, p->source->priv,
+						params);
+				if (ret < 0)
+					return ret;
+			} else {
+				return -EBUSY;
+			}
+		} else {
+			ret = skl_tplg_be_set_src_pipe_params(
+						dai, p->source,	params);
+			if (ret < 0)
+				return ret;
+		}
+	}
+
+	return ret;
+}
+
+static int skl_tplg_be_set_sink_pipe_params(struct snd_soc_dai *dai,
+	struct snd_soc_dapm_widget *w, struct skl_pipe_params *params)
+{
+	struct snd_soc_dapm_path *p = NULL;
+	int ret = -EIO;
+
+	snd_soc_dapm_widget_for_each_sink_path(w, p) {
+		if (p->connect && is_skl_dsp_widget_type(p->sink) &&
+						p->sink->priv) {
+
+			if (!p->sink->power) {
+				ret = skl_tplg_be_fill_pipe_params(
+						dai, p->sink->priv, params);
+				if (ret < 0)
+					return ret;
+			} else {
+				return -EBUSY;
+			}
+
+		} else {
+			ret = skl_tplg_be_set_sink_pipe_params(
+						dai, p->sink, params);
+			if (ret < 0)
+				return ret;
+		}
+	}
+
+	return ret;
+}
+
+/*
+ * BE hw_params can be a source parameters (capture) or sink parameters
+ * (playback). Based on sink and source we need to either find the source
+ * list or the sink list and set the pipeline parameters
+ */
+int skl_tplg_be_update_params(struct snd_soc_dai *dai,
+				struct skl_pipe_params *params)
+{
+	struct snd_soc_dapm_widget *w;
+
+	if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		w = dai->playback_widget;
+
+		return skl_tplg_be_set_src_pipe_params(dai, w, params);
+
+	} else {
+		w = dai->capture_widget;
+
+		return skl_tplg_be_set_sink_pipe_params(dai, w, params);
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_tplg_widget_events skl_tplg_widget_ops[] = {
+	{SKL_MIXER_EVENT, skl_tplg_mixer_event},
+	{SKL_VMIXER_EVENT, skl_tplg_vmixer_event},
+	{SKL_PGA_EVENT, skl_tplg_pga_event},
+};
+
+/*
+ * The topology binary passes the pin info for a module so initialize the pin
+ * info passed into module instance
+ */
+static void skl_fill_module_pin_info(struct skl_dfw_module_pin *dfw_pin,
+						struct skl_module_pin *m_pin,
+						bool is_dynamic, int max_pin)
+{
+	int i;
+
+	for (i = 0; i < max_pin; i++) {
+		m_pin[i].id.module_id = dfw_pin[i].module_id;
+		m_pin[i].id.instance_id = dfw_pin[i].instance_id;
+		m_pin[i].in_use = false;
+		m_pin[i].is_dynamic = is_dynamic;
+	}
+}
+
+/*
+ * Add pipeline from topology binary into driver pipeline list
+ *
+ * If already added we return that instance
+ * Otherwise we create a new instance and add into driver list
+ */
+static struct skl_pipe *skl_tplg_add_pipe(struct device *dev,
+			struct skl *skl, struct skl_dfw_pipe *dfw_pipe)
+{
+	struct skl_pipeline *ppl;
+	struct skl_pipe *pipe;
+	struct skl_pipe_params *params;
+
+	list_for_each_entry(ppl, &skl->ppl_list, node) {
+		if (ppl->pipe->ppl_id == dfw_pipe->pipe_id)
+			return ppl->pipe;
+	}
+
+	ppl = devm_kzalloc(dev, sizeof(*ppl), GFP_KERNEL);
+	if (!ppl)
+		return NULL;
+
+	pipe = devm_kzalloc(dev, sizeof(*pipe), GFP_KERNEL);
+	if (!pipe)
+		return NULL;
+
+	params = devm_kzalloc(dev, sizeof(*params), GFP_KERNEL);
+	if (!params)
+		return NULL;
+
+	pipe->ppl_id = dfw_pipe->pipe_id;
+	pipe->memory_pages = dfw_pipe->memory_pages;
+	pipe->pipe_priority = dfw_pipe->pipe_priority;
+	pipe->conn_type = dfw_pipe->conn_type;
+	pipe->state = SKL_PIPE_INVALID;
+	pipe->p_params = params;
+	INIT_LIST_HEAD(&pipe->w_list);
+
+	ppl->pipe = pipe;
+	list_add(&ppl->node, &skl->ppl_list);
+
+	return ppl->pipe;
+}
+
+/*
+ * Topology core widget load callback
+ *
+ * This is used to save the private data for each widget which gives
+ * information to the driver about module and pipeline parameters which DSP
+ * FW expects like ids, resource values, formats etc
+ */
+static int skl_tplg_widget_load(struct snd_soc_component *cmpnt,
+				struct snd_soc_dapm_widget *w,
+				struct snd_soc_tplg_dapm_widget *tplg_w)
+{
+	int ret;
+	struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt);
+	struct skl *skl = ebus_to_skl(ebus);
+	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct skl_module_cfg *mconfig;
+	struct skl_pipe *pipe;
+	struct skl_dfw_module *dfw_config =
+				(struct skl_dfw_module *)tplg_w->priv.data;
+
+	if (!tplg_w->priv.size)
+		goto bind_event;
+
+	mconfig = devm_kzalloc(bus->dev, sizeof(*mconfig), GFP_KERNEL);
+
+	if (!mconfig)
+		return -ENOMEM;
+
+	w->priv = mconfig;
+	mconfig->id.module_id = dfw_config->module_id;
+	mconfig->id.instance_id = dfw_config->instance_id;
+	mconfig->mcps = dfw_config->max_mcps;
+	mconfig->ibs = dfw_config->ibs;
+	mconfig->obs = dfw_config->obs;
+	mconfig->core_id = dfw_config->core_id;
+	mconfig->max_in_queue = dfw_config->max_in_queue;
+	mconfig->max_out_queue = dfw_config->max_out_queue;
+	mconfig->is_loadable = dfw_config->is_loadable;
+	mconfig->in_fmt.channels = dfw_config->in_fmt.channels;
+	mconfig->in_fmt.s_freq = dfw_config->in_fmt.freq;
+	mconfig->in_fmt.bit_depth = dfw_config->in_fmt.bit_depth;
+	mconfig->in_fmt.valid_bit_depth =
+				dfw_config->in_fmt.valid_bit_depth;
+	mconfig->in_fmt.ch_cfg = dfw_config->in_fmt.ch_cfg;
+	mconfig->out_fmt.channels = dfw_config->out_fmt.channels;
+	mconfig->out_fmt.s_freq = dfw_config->out_fmt.freq;
+	mconfig->out_fmt.bit_depth = dfw_config->out_fmt.bit_depth;
+	mconfig->out_fmt.valid_bit_depth =
+				dfw_config->out_fmt.valid_bit_depth;
+	mconfig->out_fmt.ch_cfg = dfw_config->out_fmt.ch_cfg;
+	mconfig->params_fixup = dfw_config->params_fixup;
+	mconfig->converter = dfw_config->converter;
+	mconfig->m_type = dfw_config->module_type;
+	mconfig->vbus_id = dfw_config->vbus_id;
+
+	pipe = skl_tplg_add_pipe(bus->dev, skl, &dfw_config->pipe);
+	if (pipe)
+		mconfig->pipe = pipe;
+
+	mconfig->dev_type = dfw_config->dev_type;
+	mconfig->hw_conn_type = dfw_config->hw_conn_type;
+	mconfig->time_slot = dfw_config->time_slot;
+	mconfig->formats_config.caps_size = dfw_config->caps.caps_size;
+
+	mconfig->m_in_pin = devm_kzalloc(bus->dev,
+				(mconfig->max_in_queue) *
+					sizeof(*mconfig->m_in_pin),
+				GFP_KERNEL);
+	if (!mconfig->m_in_pin)
+		return -ENOMEM;
+
+	mconfig->m_out_pin = devm_kzalloc(bus->dev, (mconfig->max_out_queue) *
+						sizeof(*mconfig->m_out_pin),
+						GFP_KERNEL);
+	if (!mconfig->m_out_pin)
+		return -ENOMEM;
+
+	skl_fill_module_pin_info(dfw_config->in_pin, mconfig->m_in_pin,
+						dfw_config->is_dynamic_in_pin,
+						mconfig->max_in_queue);
+
+	skl_fill_module_pin_info(dfw_config->out_pin, mconfig->m_out_pin,
+						 dfw_config->is_dynamic_out_pin,
+							mconfig->max_out_queue);
+
+
+	if (mconfig->formats_config.caps_size == 0)
+		goto bind_event;
+
+	mconfig->formats_config.caps = (u32 *)devm_kzalloc(bus->dev,
+			mconfig->formats_config.caps_size, GFP_KERNEL);
+
+	if (mconfig->formats_config.caps == NULL)
+		return -ENOMEM;
+
+	memcpy(mconfig->formats_config.caps, dfw_config->caps.caps,
+					 dfw_config->caps.caps_size);
+
+bind_event:
+	if (tplg_w->event_type == 0) {
+		dev_dbg(bus->dev, "ASoC: No event handler required\n");
+		return 0;
+	}
+
+	ret = snd_soc_tplg_widget_bind_event(w, skl_tplg_widget_ops,
+					ARRAY_SIZE(skl_tplg_widget_ops),
+					tplg_w->event_type);
+
+	if (ret) {
+		dev_err(bus->dev, "%s: No matching event handlers found for %d\n",
+					__func__, tplg_w->event_type);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_tplg_ops skl_tplg_ops  = {
+	.widget_load = skl_tplg_widget_load,
+};
+
+/* This will be read from topology manifest, currently defined here */
+#define SKL_MAX_MCPS 30000000
+#define SKL_FW_MAX_MEM 1000000
+
+/*
+ * SKL topology init routine
+ */
+int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus)
+{
+	int ret;
+	const struct firmware *fw;
+	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct skl *skl = ebus_to_skl(ebus);
+
+	ret = request_firmware(&fw, "dfw_sst.bin", bus->dev);
+	if (ret < 0) {
+		dev_err(bus->dev, "tplg fw %s load failed with %d\n",
+				"dfw_sst.bin", ret);
+		return ret;
+	}
+
+	/*
+	 * The complete tplg for SKL is loaded as index 0, we don't use
+	 * any other index
+	 */
+	ret = snd_soc_tplg_component_load(&platform->component,
+					&skl_tplg_ops, fw, 0);
+	if (ret < 0) {
+		dev_err(bus->dev, "tplg component load failed%d\n", ret);
+		return -EINVAL;
+	}
+
+	skl->resource.max_mcps = SKL_MAX_MCPS;
+	skl->resource.max_mem = SKL_FW_MAX_MEM;
+
+	return 0;
+}
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index 8c7767baa94f..76053a8de41c 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -129,6 +129,11 @@ struct skl_src_module_cfg {
 	enum skl_s_freq src_cfg;
 } __packed;
 
+struct notification_mask {
+	u32 notify;
+	u32 enable;
+} __packed;
+
 struct skl_up_down_mixer_cfg {
 	struct skl_base_cfg base_cfg;
 	enum skl_ch_cfg out_ch_cfg;
@@ -153,8 +158,7 @@ enum skl_dma_type {
 union skl_ssp_dma_node {
 	u8 val;
 	struct {
-		u8 dual_mono:1;
-		u8 time_slot:3;
+		u8 time_slot_index:4;
 		u8 i2s_instance:4;
 	} dma_node;
 };
@@ -263,6 +267,34 @@ struct skl_module_cfg {
 	struct skl_specific_cfg formats_config;
 };
 
+struct skl_pipeline {
+	struct skl_pipe *pipe;
+	struct list_head node;
+};
+
+struct skl_dapm_path_list {
+	struct snd_soc_dapm_path *dapm_path;
+	struct list_head node;
+};
+
+static inline struct skl *get_skl_ctx(struct device *dev)
+{
+	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+
+	return ebus_to_skl(ebus);
+}
+
+int skl_tplg_be_update_params(struct snd_soc_dai *dai,
+	struct skl_pipe_params *params);
+void skl_tplg_set_be_dmic_config(struct snd_soc_dai *dai,
+	struct skl_pipe_params *params, int stream);
+int skl_tplg_init(struct snd_soc_platform *platform,
+				struct hdac_ext_bus *ebus);
+struct skl_module_cfg *skl_tplg_fe_get_cpr_module(
+		struct snd_soc_dai *dai, int stream);
+int skl_tplg_update_pipe_params(struct device *dev,
+		struct skl_module_cfg *mconfig, struct skl_pipe_params *params);
+
 int skl_create_pipeline(struct skl_sst *ctx, struct skl_pipe *pipe);
 
 int skl_run_pipe(struct skl_sst *ctx, struct skl_pipe *pipe);
diff --git a/sound/soc/intel/skylake/skl-tplg-interface.h b/sound/soc/intel/skylake/skl-tplg-interface.h
index a50689825bca..2bc396d54cbe 100644
--- a/sound/soc/intel/skylake/skl-tplg-interface.h
+++ b/sound/soc/intel/skylake/skl-tplg-interface.h
@@ -19,6 +19,29 @@
 #ifndef __HDA_TPLG_INTERFACE_H__
 #define __HDA_TPLG_INTERFACE_H__
 
+/*
+ * Default types range from 0~12. type can range from 0 to 0xff
+ * SST types start at higher to avoid any overlapping in future
+ */
+#define SOC_CONTROL_TYPE_HDA_SST_ALGO_PARAMS	0x100
+#define SOC_CONTROL_TYPE_HDA_SST_MUX		0x101
+#define SOC_CONTROL_TYPE_HDA_SST_MIX		0x101
+#define SOC_CONTROL_TYPE_HDA_SST_BYTE		0x103
+
+#define HDA_SST_CFG_MAX	900 /* size of copier cfg*/
+#define MAX_IN_QUEUE 8
+#define MAX_OUT_QUEUE 8
+
+/* Event types goes here */
+/* Reserve event type 0 for no event handlers */
+enum skl_event_types {
+	SKL_EVENT_NONE = 0,
+	SKL_MIXER_EVENT,
+	SKL_MUX_EVENT,
+	SKL_VMIXER_EVENT,
+	SKL_PGA_EVENT
+};
+
 /**
  * enum skl_ch_cfg - channel configuration
  *
@@ -83,6 +106,66 @@ enum skl_dev_type {
 	SKL_DEVICE_I2S = 0x2,
 	SKL_DEVICE_SLIMBUS = 0x3,
 	SKL_DEVICE_HDALINK = 0x4,
+	SKL_DEVICE_HDAHOST = 0x5,
 	SKL_DEVICE_NONE
 };
+
+struct skl_dfw_module_pin {
+	u16 module_id;
+	u16 instance_id;
+} __packed;
+
+struct skl_dfw_module_fmt {
+	u32 channels;
+	u32 freq;
+	u32 bit_depth;
+	u32 valid_bit_depth;
+	u32 ch_cfg;
+} __packed;
+
+struct skl_dfw_module_caps {
+	u32 caps_size;
+	u32 caps[HDA_SST_CFG_MAX];
+};
+
+struct skl_dfw_pipe {
+	u8 pipe_id;
+	u8 pipe_priority;
+	u16 conn_type;
+	u32 memory_pages;
+} __packed;
+
+struct skl_dfw_module {
+	u16 module_id;
+	u16 instance_id;
+	u32 max_mcps;
+	u8 core_id;
+	u8 max_in_queue;
+	u8 max_out_queue;
+	u8 is_loadable;
+	u8 conn_type;
+	u8 dev_type;
+	u8 hw_conn_type;
+	u8 time_slot;
+	u32 obs;
+	u32 ibs;
+	u32 params_fixup;
+	u32 converter;
+	u32 module_type;
+	u32 vbus_id;
+	u8 is_dynamic_in_pin;
+	u8 is_dynamic_out_pin;
+	struct skl_dfw_pipe pipe;
+	struct skl_dfw_module_fmt in_fmt;
+	struct skl_dfw_module_fmt out_fmt;
+	struct skl_dfw_module_pin in_pin[MAX_IN_QUEUE];
+	struct skl_dfw_module_pin out_pin[MAX_OUT_QUEUE];
+	struct skl_dfw_module_caps caps;
+} __packed;
+
+struct skl_dfw_algo_data {
+	u32 max;
+	char *params;
+} __packed;
+
 #endif
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 348d094e81d6..5319529aedf7 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -166,12 +166,20 @@ static int skl_runtime_suspend(struct device *dev)
 	struct pci_dev *pci = to_pci_dev(dev);
 	struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
 	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct skl *skl = ebus_to_skl(ebus);
+	int ret;
 
 	dev_dbg(bus->dev, "in %s\n", __func__);
 
 	/* enable controller wake up event */
 	snd_hdac_chip_updatew(bus, WAKEEN, 0, STATESTS_INT_MASK);
 
+	snd_hdac_ext_bus_link_power_down_all(ebus);
+
+	ret = skl_suspend_dsp(skl);
+	if (ret < 0)
+		return ret;
+
 	snd_hdac_bus_stop_chip(bus);
 	snd_hdac_bus_enter_link_reset(bus);
 
@@ -183,7 +191,7 @@ static int skl_runtime_resume(struct device *dev)
 	struct pci_dev *pci = to_pci_dev(dev);
 	struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
 	struct hdac_bus *bus = ebus_to_hbus(ebus);
-	struct skl *hda = ebus_to_skl(ebus);
+	struct skl *skl = ebus_to_skl(ebus);
 	int status;
 
 	dev_dbg(bus->dev, "in %s\n", __func__);
@@ -191,12 +199,12 @@ static int skl_runtime_resume(struct device *dev)
 	/* Read STATESTS before controller reset */
 	status = snd_hdac_chip_readw(bus, STATESTS);
 
-	skl_init_pci(hda);
+	skl_init_pci(skl);
 	snd_hdac_bus_init_chip(bus, true);
 	/* disable controller Wake Up event */
 	snd_hdac_chip_updatew(bus, WAKEEN, STATESTS_INT_MASK, 0);
 
-	return 0;
+	return skl_resume_dsp(skl);
 }
 #endif /* CONFIG_PM */
 
@@ -453,21 +461,28 @@ static int skl_probe(struct pci_dev *pci,
 	if (err < 0)
 		goto out_free;
 
+	skl->nhlt = skl_nhlt_init(bus->dev);
+
+	if (skl->nhlt == NULL)
+		goto out_free;
+
 	pci_set_drvdata(skl->pci, ebus);
 
 	/* check if dsp is there */
 	if (ebus->ppcap) {
-		/* TODO register with dsp IPC */
-		dev_dbg(bus->dev, "Register dsp\n");
+		err = skl_init_dsp(skl);
+		if (err < 0) {
+			dev_dbg(bus->dev, "error failed to register dsp\n");
+			goto out_free;
+		}
 	}
-
 	if (ebus->mlcap)
 		snd_hdac_ext_bus_get_ml_capabilities(ebus);
 
 	/* create device for soc dmic */
 	err = skl_dmic_device_register(skl);
 	if (err < 0)
-		goto out_free;
+		goto out_dsp_free;
 
 	/* register platform dai and controls */
 	err = skl_platform_register(bus->dev);
@@ -491,6 +506,8 @@ out_unregister:
 	skl_platform_unregister(bus->dev);
 out_dmic_free:
 	skl_dmic_device_unregister(skl);
+out_dsp_free:
+	skl_free_dsp(skl);
 out_free:
 	skl->init_failed = 1;
 	skl_free(ebus);
@@ -507,6 +524,7 @@ static void skl_remove(struct pci_dev *pci)
 		pm_runtime_get_noresume(&pci->dev);
 	pci_dev_put(pci);
 	skl_platform_unregister(&pci->dev);
+	skl_free_dsp(skl);
 	skl_dmic_device_unregister(skl);
 	skl_free(ebus);
 	dev_set_drvdata(&pci->dev, NULL);
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index f7fdbb02947f..dd2e79ae45a8 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -48,6 +48,13 @@
 #define AZX_REG_VS_SDXEFIFOS_XBASE	0x1094
 #define AZX_REG_VS_SDXEFIFOS_XINTERVAL	0x20
 
+struct skl_dsp_resource {
+	u32 max_mcps;
+	u32 max_mem;
+	u32 mcps;
+	u32 mem;
+};
+
 struct skl {
 	struct hdac_ext_bus ebus;
 	struct pci_dev *pci;
@@ -55,8 +62,12 @@ struct skl {
 	unsigned int init_failed:1; /* delayed init failed */
 	struct platform_device *dmic_dev;
 
-	void __iomem *nhlt; /* nhlt ptr */
+	void *nhlt; /* nhlt ptr */
 	struct skl_sst *skl_sst; /* sst skl ctx */
+
+	struct skl_dsp_resource resource;
+	struct list_head ppl_list;
+	struct list_head dapm_path_list;
 };
 
 #define skl_to_ebus(s)	(&(s)->ebus)
@@ -72,8 +83,8 @@ struct skl_dma_params {
 int skl_platform_unregister(struct device *dev);
 int skl_platform_register(struct device *dev);
 
-void __iomem *skl_nhlt_init(struct device *dev);
-void skl_nhlt_free(void __iomem *addr);
+void *skl_nhlt_init(struct device *dev);
+void skl_nhlt_free(void *addr);
 struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance,
 			u8 link_type, u8 s_fmt, u8 no_ch, u32 s_rate, u8 dirn);
 
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index b05fb1c1a848..794a3499e567 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -485,6 +485,7 @@ static const struct of_device_id jz4740_of_matches[] = {
 	{ .compatible = "ingenic,jz4780-i2s", .data = (void *)JZ_I2S_JZ4780 },
 	{ /* sentinel */ }
 };
+MODULE_DEVICE_TABLE(of, jz4740_of_matches);
 #endif
 
 static int jz4740_i2s_dev_probe(struct platform_device *pdev)
diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c
index de7563bdc5c2..e0304d544f26 100644
--- a/sound/soc/kirkwood/armada-370-db.c
+++ b/sound/soc/kirkwood/armada-370-db.c
@@ -130,6 +130,7 @@ static const struct of_device_id a370db_dt_ids[] = {
 	{ .compatible = "marvell,a370db-audio" },
 	{ },
 };
+MODULE_DEVICE_TABLE(of, a370db_dt_ids);
 
 static struct platform_driver a370db_driver = {
 	.driver		= {
diff --git a/sound/soc/mediatek/mt8173-max98090.c b/sound/soc/mediatek/mt8173-max98090.c
index 684e8a78bed0..71a1a35047ba 100644
--- a/sound/soc/mediatek/mt8173-max98090.c
+++ b/sound/soc/mediatek/mt8173-max98090.c
@@ -179,21 +179,13 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
 	}
 	card->dev = &pdev->dev;
 
-	ret = snd_soc_register_card(card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret)
 		dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
 			__func__, ret);
 	return ret;
 }
 
-static int mt8173_max98090_dev_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-	return 0;
-}
-
 static const struct of_device_id mt8173_max98090_dt_match[] = {
 	{ .compatible = "mediatek,mt8173-max98090", },
 	{ }
@@ -209,7 +201,6 @@ static struct platform_driver mt8173_max98090_driver = {
 #endif
 	},
 	.probe = mt8173_max98090_dev_probe,
-	.remove = mt8173_max98090_dev_remove,
 };
 
 module_platform_driver(mt8173_max98090_driver);
diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
index 86cf9752f18a..50ba538eccb3 100644
--- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
@@ -246,21 +246,13 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
 	card->dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
 
-	ret = snd_soc_register_card(card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret)
 		dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
 			__func__, ret);
 	return ret;
 }
 
-static int mt8173_rt5650_rt5676_dev_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-	return 0;
-}
-
 static const struct of_device_id mt8173_rt5650_rt5676_dt_match[] = {
 	{ .compatible = "mediatek,mt8173-rt5650-rt5676", },
 	{ }
@@ -276,7 +268,6 @@ static struct platform_driver mt8173_rt5650_rt5676_driver = {
 #endif
 	},
 	.probe = mt8173_rt5650_rt5676_dev_probe,
-	.remove = mt8173_rt5650_rt5676_dev_remove,
 };
 
 module_platform_driver(mt8173_rt5650_rt5676_driver);
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 6e6fce6a14ba..2b23ffbac6b1 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -142,7 +142,7 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev)
 	card->dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
 
-	ret = snd_soc_register_card(card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
 			ret);
@@ -154,12 +154,8 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev)
 
 static int mxs_sgtl5000_remove(struct platform_device *pdev)
 {
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
 	mxs_saif_put_mclk(0);
 
-	snd_soc_unregister_card(card);
-
 	return 0;
 }
 
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index dcb5336b5698..190f868e78b2 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -99,8 +99,7 @@ static int n810_startup(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 
-	snd_pcm_hw_constraint_minmax(runtime,
-				     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
+	snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2);
 
 	n810_ext_control(&rtd->card->dapm);
 	return clk_prepare_enable(sys_clkout2);
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 3bebfb1d3a6f..5e21f08579d8 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -107,8 +107,7 @@ static int rx51_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_card *card = rtd->card;
 
-	snd_pcm_hw_constraint_minmax(runtime,
-				     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
+	snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2);
 	rx51_ext_control(&card->dapm);
 
 	return 0;
@@ -297,7 +296,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
 		dev_err(card->dev, "Failed to add TPA6130A2 controls\n");
 		return err;
 	}
-	snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42);
+	snd_soc_limit_volume(card, "TPA6130A2 Headphone Playback Volume", 42);
 
 	err = omap_mcbsp_st_add_controls(rtd, 2);
 	if (err < 0) {
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
index 2b26318bc200..6147e86e9b0f 100644
--- a/sound/soc/pxa/brownstone.c
+++ b/sound/soc/pxa/brownstone.c
@@ -116,26 +116,19 @@ static int brownstone_probe(struct platform_device *pdev)
 	int ret;
 
 	brownstone.dev = &pdev->dev;
-	ret = snd_soc_register_card(&brownstone);
+	ret = devm_snd_soc_register_card(&pdev->dev, &brownstone);
 	if (ret)
 		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
 				ret);
 	return ret;
 }
 
-static int brownstone_remove(struct platform_device *pdev)
-{
-	snd_soc_unregister_card(&brownstone);
-	return 0;
-}
-
 static struct platform_driver mmp_driver = {
 	.driver		= {
 		.name	= "brownstone-audio",
 		.pm     = &snd_soc_pm_ops,
 	},
 	.probe		= brownstone_probe,
-	.remove		= brownstone_remove,
 };
 
 module_platform_driver(mmp_driver);
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 3580d10c9f28..c97dc13d3608 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -295,28 +295,19 @@ static int corgi_probe(struct platform_device *pdev)
 
 	card->dev = &pdev->dev;
 
-	ret = snd_soc_register_card(card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret)
 		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
 			ret);
 	return ret;
 }
 
-static int corgi_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-	return 0;
-}
-
 static struct platform_driver corgi_driver = {
 	.driver		= {
 		.name	= "corgi-audio",
 		.pm     = &snd_soc_pm_ops,
 	},
 	.probe		= corgi_probe,
-	.remove		= corgi_remove,
 };
 
 module_platform_driver(corgi_driver);
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index d72e124a3676..1de876529aa1 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -138,7 +138,7 @@ static int e740_probe(struct platform_device *pdev)
 
 	card->dev = &pdev->dev;
 
-	ret = snd_soc_register_card(card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
 			ret);
@@ -149,10 +149,7 @@ static int e740_probe(struct platform_device *pdev)
 
 static int e740_remove(struct platform_device *pdev)
 {
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
 	gpio_free_array(e740_audio_gpios, ARRAY_SIZE(e740_audio_gpios));
-	snd_soc_unregister_card(card);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index 48f2d7c2e68c..b7eb7cd5df7d 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -120,7 +120,7 @@ static int e750_probe(struct platform_device *pdev)
 
 	card->dev = &pdev->dev;
 
-	ret = snd_soc_register_card(card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
 			ret);
@@ -131,10 +131,7 @@ static int e750_probe(struct platform_device *pdev)
 
 static int e750_remove(struct platform_device *pdev)
 {
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
 	gpio_free_array(e750_audio_gpios, ARRAY_SIZE(e750_audio_gpios));
-	snd_soc_unregister_card(card);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 45d4bd46fff6..41bf71466a7b 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -119,7 +119,7 @@ static int e800_probe(struct platform_device *pdev)
 
 	card->dev = &pdev->dev;
 
-	ret = snd_soc_register_card(card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
 			ret);
@@ -130,10 +130,7 @@ static int e800_probe(struct platform_device *pdev)
 
 static int e800_remove(struct platform_device *pdev)
 {
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
 	gpio_free_array(e800_audio_gpios, ARRAY_SIZE(e800_audio_gpios));
-	snd_soc_unregister_card(card);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c
index 9f8be7cd567e..ecbf2873b7ff 100644
--- a/sound/soc/pxa/hx4700.c
+++ b/sound/soc/pxa/hx4700.c
@@ -193,7 +193,7 @@ static int hx4700_audio_probe(struct platform_device *pdev)
 		return ret;
 
 	snd_soc_card_hx4700.dev = &pdev->dev;
-	ret = snd_soc_register_card(&snd_soc_card_hx4700);
+	ret = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_hx4700);
 	if (ret)
 		gpio_free_array(hx4700_audio_gpios,
 				ARRAY_SIZE(hx4700_audio_gpios));
@@ -203,8 +203,6 @@ static int hx4700_audio_probe(struct platform_device *pdev)
 
 static int hx4700_audio_remove(struct platform_device *pdev)
 {
-	snd_soc_unregister_card(&snd_soc_card_hx4700);
-
 	gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0);
 	gpio_set_value(GPIO107_HX4700_SPK_nSD, 0);
 
diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c
index 29fabbfd21f1..9d0e40771ef5 100644
--- a/sound/soc/pxa/imote2.c
+++ b/sound/soc/pxa/imote2.c
@@ -72,28 +72,19 @@ static int imote2_probe(struct platform_device *pdev)
 
 	card->dev = &pdev->dev;
 
-	ret = snd_soc_register_card(card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret)
 		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
 			ret);
 	return ret;
 }
 
-static int imote2_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-	return 0;
-}
-
 static struct platform_driver imote2_driver = {
 	.driver		= {
 		.name	= "imote2-audio",
 		.pm     = &snd_soc_pm_ops,
 	},
 	.probe		= imote2_probe,
-	.remove		= imote2_remove,
 };
 
 module_platform_driver(imote2_driver);
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index a9615a574546..29bc60e85e92 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -181,7 +181,7 @@ static int mioa701_wm9713_probe(struct platform_device *pdev)
 		return -ENODEV;
 
 	mioa701.dev = &pdev->dev;
-	rc =  snd_soc_register_card(&mioa701);
+	rc = devm_snd_soc_register_card(&pdev->dev, &mioa701);
 	if (!rc)
 		dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
 			 "lead to overheating and possible destruction of your device."
@@ -189,17 +189,8 @@ static int mioa701_wm9713_probe(struct platform_device *pdev)
 	return rc;
 }
 
-static int mioa701_wm9713_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-	return 0;
-}
-
 static struct platform_driver mioa701_wm9713_driver = {
 	.probe		= mioa701_wm9713_probe,
-	.remove		= mioa701_wm9713_remove,
 	.driver		= {
 		.name		= "mioa701-wm9713",
 		.pm     = &snd_soc_pm_ops,
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index c20bbc042425..4e74d9573f03 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -140,22 +140,15 @@ static int palm27x_asoc_probe(struct platform_device *pdev)
 
 	palm27x_asoc.dev = &pdev->dev;
 
-	ret = snd_soc_register_card(&palm27x_asoc);
+	ret = devm_snd_soc_register_card(&pdev->dev, &palm27x_asoc);
 	if (ret)
 		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
 			ret);
 	return ret;
 }
 
-static int palm27x_asoc_remove(struct platform_device *pdev)
-{
-	snd_soc_unregister_card(&palm27x_asoc);
-	return 0;
-}
-
 static struct platform_driver palm27x_wm9712_driver = {
 	.probe		= palm27x_asoc_probe,
-	.remove		= palm27x_asoc_remove,
 	.driver		= {
 		.name		= "palm27x-asoc",
 		.pm     = &snd_soc_pm_ops,
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 80b457ac522a..84d0e2e50808 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -267,28 +267,19 @@ static int poodle_probe(struct platform_device *pdev)
 
 	card->dev = &pdev->dev;
 
-	ret = snd_soc_register_card(card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret)
 		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
 			ret);
 	return ret;
 }
 
-static int poodle_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-	return 0;
-}
-
 static struct platform_driver poodle_driver = {
 	.driver		= {
 		.name	= "poodle-audio",
 		.pm     = &snd_soc_pm_ops,
 	},
 	.probe		= poodle_probe,
-	.remove		= poodle_remove,
 };
 
 module_platform_driver(poodle_driver);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 3da485ec1de7..da03fad1b9cd 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -809,6 +809,7 @@ static const struct of_device_id pxa_ssp_of_ids[] = {
 	{ .compatible = "mrvl,pxa-ssp-dai" },
 	{}
 };
+MODULE_DEVICE_TABLE(of, pxa_ssp_of_ids);
 #endif
 
 static int asoc_ssp_probe(struct platform_device *pdev)
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 9e4b04e0fbd1..f3de615aacd7 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -15,6 +15,7 @@
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/dmaengine.h>
+#include <linux/dma/pxa-dma.h>
 
 #include <sound/core.h>
 #include <sound/ac97_codec.h>
@@ -49,7 +50,11 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
 	.reset	= pxa2xx_ac97_cold_reset,
 };
 
-static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 11;
+static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = {
+	.prio = PXAD_PRIO_LOWEST,
+	.drcmr = 11,
+};
+
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
 	.addr		= __PREG(PCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
@@ -57,7 +62,11 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
 	.filter_data	= &pxa2xx_ac97_pcm_stereo_in_req,
 };
 
-static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 12;
+static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = {
+	.prio = PXAD_PRIO_LOWEST,
+	.drcmr = 12,
+};
+
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
 	.addr		= __PREG(PCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
@@ -65,7 +74,10 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
 	.filter_data	= &pxa2xx_ac97_pcm_stereo_out_req,
 };
 
-static unsigned long pxa2xx_ac97_pcm_aux_mono_out_req = 10;
+static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = {
+	.prio = PXAD_PRIO_LOWEST,
+	.drcmr = 10,
+};
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = {
 	.addr		= __PREG(MODR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_2_BYTES,
@@ -73,7 +85,10 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = {
 	.filter_data	= &pxa2xx_ac97_pcm_aux_mono_out_req,
 };
 
-static unsigned long pxa2xx_ac97_pcm_aux_mono_in_req = 9;
+static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = {
+	.prio = PXAD_PRIO_LOWEST,
+	.drcmr = 9,
+};
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = {
 	.addr		= __PREG(MODR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_2_BYTES,
@@ -81,7 +96,10 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = {
 	.filter_data	= &pxa2xx_ac97_pcm_aux_mono_in_req,
 };
 
-static unsigned long pxa2xx_ac97_pcm_aux_mic_mono_req = 8;
+static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = {
+	.prio = PXAD_PRIO_LOWEST,
+	.drcmr = 8,
+};
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = {
 	.addr		= __PREG(MCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_2_BYTES,
@@ -89,9 +107,8 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = {
 	.filter_data	= &pxa2xx_ac97_pcm_aux_mic_mono_req,
 };
 
-static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params,
-				 struct snd_soc_dai *cpu_dai)
+static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream,
+				    struct snd_soc_dai *cpu_dai)
 {
 	struct snd_dmaengine_dai_dma_data *dma_data;
 
@@ -105,9 +122,8 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
-				     struct snd_pcm_hw_params *params,
-				     struct snd_soc_dai *cpu_dai)
+static int pxa2xx_ac97_aux_startup(struct snd_pcm_substream *substream,
+				   struct snd_soc_dai *cpu_dai)
 {
 	struct snd_dmaengine_dai_dma_data *dma_data;
 
@@ -121,9 +137,8 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
-				     struct snd_pcm_hw_params *params,
-				     struct snd_soc_dai *cpu_dai)
+static int pxa2xx_ac97_mic_startup(struct snd_pcm_substream *substream,
+				   struct snd_soc_dai *cpu_dai)
 {
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		return -ENODEV;
@@ -139,15 +154,15 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_48000)
 
 static const struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = {
-	.hw_params	= pxa2xx_ac97_hw_params,
+	.startup	= pxa2xx_ac97_hifi_startup,
 };
 
 static const struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = {
-	.hw_params	= pxa2xx_ac97_hw_aux_params,
+	.startup	= pxa2xx_ac97_aux_startup,
 };
 
 static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = {
-	.hw_params	= pxa2xx_ac97_hw_mic_params,
+	.startup	= pxa2xx_ac97_mic_startup,
 };
 
 /*
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 6b4e40036910..0389cf7b4b1e 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -319,6 +319,9 @@ static int pxa2xx_i2s_probe(struct snd_soc_dai *dai)
 	/* Along with FIFO servicing */
 	SAIMR &= ~(SAIMR_RFS | SAIMR_TFS);
 
+	snd_soc_dai_init_dma_data(dai, &pxa2xx_i2s_pcm_stereo_out,
+		&pxa2xx_i2s_pcm_stereo_in);
+
 	return 0;
 }
 
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 831ee37d2e3e..9f390398d518 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -15,8 +15,6 @@
 #include <linux/dmaengine.h>
 #include <linux/of.h>
 
-#include <mach/dma.h>
-
 #include <sound/core.h>
 #include <sound/soc.h>
 #include <sound/pxa2xx-lib.h>
@@ -27,11 +25,8 @@
 static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
-	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 snd_dmaengine_dai_dma_data *dma;
-	int ret;
 
 	dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 
@@ -40,40 +35,13 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 	if (!dma)
 		return 0;
 
-	/* this may get called several times by oss emulation
-	 * with different params */
-	if (prtd->params == NULL) {
-		prtd->params = dma;
-		ret = pxa_request_dma("name", DMA_PRIO_LOW,
-			      pxa2xx_pcm_dma_irq, substream);
-		if (ret < 0)
-			return ret;
-		prtd->dma_ch = ret;
-	} else if (prtd->params != dma) {
-		pxa_free_dma(prtd->dma_ch);
-		prtd->params = dma;
-		ret = pxa_request_dma("name", DMA_PRIO_LOW,
-			      pxa2xx_pcm_dma_irq, substream);
-		if (ret < 0)
-			return ret;
-		prtd->dma_ch = ret;
-	}
-
 	return __pxa2xx_pcm_hw_params(substream, params);
 }
 
 static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
-	struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
-
 	__pxa2xx_pcm_hw_free(substream);
 
-	if (prtd->dma_ch >= 0) {
-		pxa_free_dma(prtd->dma_ch);
-		prtd->dma_ch = -1;
-		prtd->params = NULL;
-	}
-
 	return 0;
 }
 
@@ -132,6 +100,7 @@ static const struct of_device_id snd_soc_pxa_audio_match[] = {
 	{ .compatible   = "mrvl,pxa-pcm-audio" },
 	{ }
 };
+MODULE_DEVICE_TABLE(of, snd_soc_pxa_audio_match);
 #endif
 
 static struct platform_driver pxa_pcm_driver = {
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 461123ad5ff2..b00222620fd0 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -305,7 +305,7 @@ static int spitz_probe(struct platform_device *pdev)
 
 	card->dev = &pdev->dev;
 
-	ret = snd_soc_register_card(card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
 			ret);
@@ -322,9 +322,6 @@ err1:
 
 static int spitz_remove(struct platform_device *pdev)
 {
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
 	gpio_free(spitz_mic_gpio);
 	return 0;
 }
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index f59f566551ef..49518dd642aa 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -233,7 +233,7 @@ static int tosa_probe(struct platform_device *pdev)
 
 	card->dev = &pdev->dev;
 
-	ret = snd_soc_register_card(card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
 			ret);
@@ -244,10 +244,7 @@ static int tosa_probe(struct platform_device *pdev)
 
 static int tosa_remove(struct platform_device *pdev)
 {
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
 	gpio_free(TOSA_GPIO_L_MUTE);
-	snd_soc_unregister_card(card);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
index 1753c7d9e760..65c20f779177 100644
--- a/sound/soc/pxa/ttc-dkb.c
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -128,7 +128,7 @@ static int ttc_dkb_probe(struct platform_device *pdev)
 
 	card->dev = &pdev->dev;
 
-	ret = snd_soc_register_card(card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret)
 		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
 			ret);
@@ -136,22 +136,12 @@ static int ttc_dkb_probe(struct platform_device *pdev)
 	return ret;
 }
 
-static int ttc_dkb_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-
-	return 0;
-}
-
 static struct platform_driver ttc_dkb_driver = {
 	.driver		= {
 		.name	= "ttc-dkb-audio",
 		.pm     = &snd_soc_pm_ops,
 	},
 	.probe		= ttc_dkb_probe,
-	.remove		= ttc_dkb_remove,
 };
 
 module_platform_driver(ttc_dkb_driver);
diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index 97bc2023f08a..e5101e0d2d37 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -438,7 +438,8 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev)
 		if (IS_ERR(drvdata->mi2s_bit_clk[dai_id])) {
 			dev_err(&pdev->dev,
 				"%s() error getting mi2s-bit-clk: %ld\n",
-				__func__, PTR_ERR(drvdata->mi2s_bit_clk[i]));
+				__func__,
+				PTR_ERR(drvdata->mi2s_bit_clk[dai_id]));
 			return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]);
 		}
 	}
diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig
index 58bae8e2cf5f..f1e0c703e0d2 100644
--- a/sound/soc/rockchip/Kconfig
+++ b/sound/soc/rockchip/Kconfig
@@ -15,9 +15,17 @@ config SND_SOC_ROCKCHIP_I2S
 	  Rockchip I2S device. The device supports upto maximum of
 	  8 channels each for play and record.
 
+config SND_SOC_ROCKCHIP_SPDIF
+	tristate "Rockchip SPDIF Device Driver"
+	depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+	help
+	  Say Y or M if you want to add support for SPDIF driver for
+	  Rockchip SPDIF transceiver device.
+
 config SND_SOC_ROCKCHIP_MAX98090
 	tristate "ASoC support for Rockchip boards using a MAX98090 codec"
-	depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB
+	depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP
 	select SND_SOC_ROCKCHIP_I2S
 	select SND_SOC_MAX98090
 	select SND_SOC_TS3A227E
@@ -27,7 +35,7 @@ config SND_SOC_ROCKCHIP_MAX98090
 
 config SND_SOC_ROCKCHIP_RT5645
 	tristate "ASoC support for Rockchip boards using a RT5645/RT5650 codec"
-	depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB
+	depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP
 	select SND_SOC_ROCKCHIP_I2S
 	select SND_SOC_RT5645
 	help
diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile
index 1bc1dc3c729a..c0bf560125f3 100644
--- a/sound/soc/rockchip/Makefile
+++ b/sound/soc/rockchip/Makefile
@@ -1,7 +1,9 @@
 # ROCKCHIP Platform Support
-snd-soc-i2s-objs := rockchip_i2s.o
+snd-soc-rockchip-i2s-objs := rockchip_i2s.o
+snd-soc-rockchip-spdif-objs := rockchip_spdif.o
 
-obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-i2s.o
+obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o
+obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o
 
 snd-soc-rockchip-max98090-objs := rockchip_max98090.o
 snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index b93610212e3d..58ee64594f07 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -226,6 +226,7 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream,
 				  struct snd_soc_dai *dai)
 {
 	struct rk_i2s_dev *i2s = to_info(dai);
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	unsigned int val = 0;
 
 	switch (params_format(params)) {
@@ -245,13 +246,46 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	regmap_update_bits(i2s->regmap, I2S_TXCR, I2S_TXCR_VDW_MASK, val);
-	regmap_update_bits(i2s->regmap, I2S_RXCR, I2S_RXCR_VDW_MASK, val);
+	switch (params_channels(params)) {
+	case 8:
+		val |= I2S_CHN_8;
+		break;
+	case 6:
+		val |= I2S_CHN_6;
+		break;
+	case 4:
+		val |= I2S_CHN_4;
+		break;
+	case 2:
+		val |= I2S_CHN_2;
+		break;
+	default:
+		dev_err(i2s->dev, "invalid channel: %d\n",
+			params_channels(params));
+		return -EINVAL;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		regmap_update_bits(i2s->regmap, I2S_RXCR,
+				   I2S_RXCR_VDW_MASK | I2S_RXCR_CSR_MASK,
+				   val);
+	else
+		regmap_update_bits(i2s->regmap, I2S_TXCR,
+				   I2S_TXCR_VDW_MASK | I2S_TXCR_CSR_MASK,
+				   val);
+
 	regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK,
 			   I2S_DMACR_TDL(16));
 	regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDL_MASK,
 			   I2S_DMACR_RDL(16));
 
+	val = I2S_CKR_TRCM_TXRX;
+	if (dai->driver->symmetric_rates || rtd->dai_link->symmetric_rates)
+		val = I2S_CKR_TRCM_TXSHARE;
+
+	regmap_update_bits(i2s->regmap, I2S_CKR,
+			   I2S_CKR_TRCM_MASK,
+			   val);
 	return 0;
 }
 
@@ -415,10 +449,12 @@ static const struct regmap_config rockchip_i2s_regmap_config = {
 
 static int rockchip_i2s_probe(struct platform_device *pdev)
 {
+	struct device_node *node = pdev->dev.of_node;
 	struct rk_i2s_dev *i2s;
 	struct resource *res;
 	void __iomem *regs;
 	int ret;
+	int val;
 
 	i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL);
 	if (!i2s) {
@@ -475,6 +511,14 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
 			goto err_pm_disable;
 	}
 
+	/* refine capture channels */
+	if (!of_property_read_u32(node, "rockchip,capture-channels", &val)) {
+		if (val >= 2 && val <= 8)
+			rockchip_i2s_dai.capture.channels_max = val;
+		else
+			rockchip_i2s_dai.capture.channels_max = 2;
+	}
+
 	ret = devm_snd_soc_register_component(&pdev->dev,
 					      &rockchip_i2s_component,
 					      &rockchip_i2s_dai, 1);
diff --git a/sound/soc/rockchip/rockchip_i2s.h b/sound/soc/rockchip/rockchip_i2s.h
index 93f456f518a9..dc6e2c74d088 100644
--- a/sound/soc/rockchip/rockchip_i2s.h
+++ b/sound/soc/rockchip/rockchip_i2s.h
@@ -49,6 +49,9 @@
  * RXCR
  * receive operation control register
 */
+#define I2S_RXCR_CSR_SHIFT	15
+#define I2S_RXCR_CSR(x)		(x << I2S_RXCR_CSR_SHIFT)
+#define I2S_RXCR_CSR_MASK	(3 << I2S_RXCR_CSR_SHIFT)
 #define I2S_RXCR_HWT		BIT(14)
 #define I2S_RXCR_SJM_SHIFT	12
 #define I2S_RXCR_SJM_R		(0 << I2S_RXCR_SJM_SHIFT)
@@ -75,6 +78,12 @@
  * CKR
  * clock generation register
 */
+#define I2S_CKR_TRCM_SHIFT	28
+#define I2S_CKR_TRCM(x)	(x << I2S_CKR_TRCM_SHIFT)
+#define I2S_CKR_TRCM_TXRX	(0 << I2S_CKR_TRCM_SHIFT)
+#define I2S_CKR_TRCM_TXSHARE	(1 << I2S_CKR_TRCM_SHIFT)
+#define I2S_CKR_TRCM_RXSHARE	(2 << I2S_CKR_TRCM_SHIFT)
+#define I2S_CKR_TRCM_MASK	(3 << I2S_CKR_TRCM_SHIFT)
 #define I2S_CKR_MSS_SHIFT	27
 #define I2S_CKR_MSS_MASTER	(0 << I2S_CKR_MSS_SHIFT)
 #define I2S_CKR_MSS_SLAVE	(1 << I2S_CKR_MSS_SHIFT)
@@ -207,6 +216,13 @@ enum {
 	ROCKCHIP_DIV_BCLK,
 };
 
+/* channel select */
+#define I2S_CSR_SHIFT	15
+#define I2S_CHN_2	(0 << I2S_CSR_SHIFT)
+#define I2S_CHN_4	(1 << I2S_CSR_SHIFT)
+#define I2S_CHN_6	(2 << I2S_CSR_SHIFT)
+#define I2S_CHN_8	(3 << I2S_CSR_SHIFT)
+
 /* I2S REGS */
 #define I2S_TXCR	(0x0000)
 #define I2S_RXCR	(0x0004)
diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c
new file mode 100644
index 000000000000..a38a3029062c
--- /dev/null
+++ b/sound/soc/rockchip/rockchip_spdif.c
@@ -0,0 +1,405 @@
+/* sound/soc/rockchip/rk_spdif.c
+ *
+ * ALSA SoC Audio Layer - Rockchip I2S Controller driver
+ *
+ * Copyright (c) 2014 Rockchip Electronics Co. Ltd.
+ * Author: Jianqun <jay.xu@rock-chips.com>
+ * Copyright (c) 2015 Collabora Ltd.
+ * Author: Sjoerd Simons <sjoerd.simons@collabora.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/of_gpio.h>
+#include <linux/clk.h>
+#include <linux/pm_runtime.h>
+#include <linux/mfd/syscon.h>
+#include <linux/regmap.h>
+#include <sound/pcm_params.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "rockchip_spdif.h"
+
+enum rk_spdif_type {
+	RK_SPDIF_RK3066,
+	RK_SPDIF_RK3188,
+	RK_SPDIF_RK3288,
+};
+
+#define RK3288_GRF_SOC_CON2 0x24c
+
+struct rk_spdif_dev {
+	struct device *dev;
+
+	struct clk *mclk;
+	struct clk *hclk;
+
+	struct snd_dmaengine_dai_dma_data playback_dma_data;
+
+	struct regmap *regmap;
+};
+
+static const struct of_device_id rk_spdif_match[] = {
+	{ .compatible = "rockchip,rk3066-spdif",
+	  .data = (void *) RK_SPDIF_RK3066 },
+	{ .compatible = "rockchip,rk3188-spdif",
+	  .data = (void *) RK_SPDIF_RK3188 },
+	{ .compatible = "rockchip,rk3288-spdif",
+	  .data = (void *) RK_SPDIF_RK3288 },
+	{},
+};
+MODULE_DEVICE_TABLE(of, rk_spdif_match);
+
+static int rk_spdif_runtime_suspend(struct device *dev)
+{
+	struct rk_spdif_dev *spdif = dev_get_drvdata(dev);
+
+	clk_disable_unprepare(spdif->mclk);
+	clk_disable_unprepare(spdif->hclk);
+
+	return 0;
+}
+
+static int rk_spdif_runtime_resume(struct device *dev)
+{
+	struct rk_spdif_dev *spdif = dev_get_drvdata(dev);
+	int ret;
+
+	ret = clk_prepare_enable(spdif->mclk);
+	if (ret) {
+		dev_err(spdif->dev, "mclk clock enable failed %d\n", ret);
+		return ret;
+	}
+
+	ret = clk_prepare_enable(spdif->hclk);
+	if (ret) {
+		dev_err(spdif->dev, "hclk clock enable failed %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int rk_spdif_hw_params(struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
+{
+	struct rk_spdif_dev *spdif = snd_soc_dai_get_drvdata(dai);
+	unsigned int val = SPDIF_CFGR_HALFWORD_ENABLE;
+	int srate, mclk;
+	int ret;
+
+	srate = params_rate(params);
+	switch (srate) {
+	case 32000:
+	case 48000:
+	case 96000:
+		mclk = 96000 * 128; /* 12288000 hz */
+		break;
+	case 44100:
+		mclk = 44100 * 256; /* 11289600 hz */
+		break;
+	case 192000:
+		mclk = 192000 * 128; /* 24576000 hz */
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		val |= SPDIF_CFGR_VDW_16;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		val |= SPDIF_CFGR_VDW_20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		val |= SPDIF_CFGR_VDW_24;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* Set clock and calculate divider */
+	ret = clk_set_rate(spdif->mclk, mclk);
+	if (ret != 0) {
+		dev_err(spdif->dev, "Failed to set module clock rate: %d\n",
+			ret);
+		return ret;
+	}
+
+	val |= SPDIF_CFGR_CLK_DIV(mclk/(srate * 256));
+	ret = regmap_update_bits(spdif->regmap, SPDIF_CFGR,
+		SPDIF_CFGR_CLK_DIV_MASK | SPDIF_CFGR_HALFWORD_ENABLE |
+		SDPIF_CFGR_VDW_MASK,
+		val);
+
+	return ret;
+}
+
+static int rk_spdif_trigger(struct snd_pcm_substream *substream,
+				int cmd, struct snd_soc_dai *dai)
+{
+	struct rk_spdif_dev *spdif = snd_soc_dai_get_drvdata(dai);
+	int ret;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		ret = regmap_update_bits(spdif->regmap, SPDIF_DMACR,
+				   SPDIF_DMACR_TDE_ENABLE,
+				   SPDIF_DMACR_TDE_ENABLE);
+
+		if (ret != 0)
+			return ret;
+
+		ret = regmap_update_bits(spdif->regmap, SPDIF_XFER,
+				   SPDIF_XFER_TXS_START,
+				   SPDIF_XFER_TXS_START);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		ret = regmap_update_bits(spdif->regmap, SPDIF_DMACR,
+				   SPDIF_DMACR_TDE_ENABLE,
+				   SPDIF_DMACR_TDE_DISABLE);
+
+		if (ret != 0)
+			return ret;
+
+		ret = regmap_update_bits(spdif->regmap, SPDIF_XFER,
+				   SPDIF_XFER_TXS_START,
+				   SPDIF_XFER_TXS_STOP);
+		break;
+	default:
+		ret = -EINVAL;
+		break;
+	}
+
+	return ret;
+}
+
+static int rk_spdif_dai_probe(struct snd_soc_dai *dai)
+{
+	struct rk_spdif_dev *spdif = snd_soc_dai_get_drvdata(dai);
+
+	dai->playback_dma_data = &spdif->playback_dma_data;
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops rk_spdif_dai_ops = {
+	.hw_params = rk_spdif_hw_params,
+	.trigger = rk_spdif_trigger,
+};
+
+static struct snd_soc_dai_driver rk_spdif_dai = {
+	.probe = rk_spdif_dai_probe,
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = (SNDRV_PCM_RATE_32000 |
+			  SNDRV_PCM_RATE_44100 |
+			  SNDRV_PCM_RATE_48000 |
+			  SNDRV_PCM_RATE_96000 |
+			  SNDRV_PCM_RATE_192000),
+		.formats = (SNDRV_PCM_FMTBIT_S16_LE |
+			    SNDRV_PCM_FMTBIT_S20_3LE |
+			    SNDRV_PCM_FMTBIT_S24_LE),
+	},
+	.ops = &rk_spdif_dai_ops,
+};
+
+static const struct snd_soc_component_driver rk_spdif_component = {
+	.name = "rockchip-spdif",
+};
+
+static bool rk_spdif_wr_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case SPDIF_CFGR:
+	case SPDIF_DMACR:
+	case SPDIF_INTCR:
+	case SPDIF_XFER:
+	case SPDIF_SMPDR:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool rk_spdif_rd_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case SPDIF_CFGR:
+	case SPDIF_SDBLR:
+	case SPDIF_INTCR:
+	case SPDIF_INTSR:
+	case SPDIF_XFER:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool rk_spdif_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case SPDIF_INTSR:
+	case SPDIF_SDBLR:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static const struct regmap_config rk_spdif_regmap_config = {
+	.reg_bits = 32,
+	.reg_stride = 4,
+	.val_bits = 32,
+	.max_register = SPDIF_SMPDR,
+	.writeable_reg = rk_spdif_wr_reg,
+	.readable_reg = rk_spdif_rd_reg,
+	.volatile_reg = rk_spdif_volatile_reg,
+	.cache_type = REGCACHE_FLAT,
+};
+
+static int rk_spdif_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct rk_spdif_dev *spdif;
+	const struct of_device_id *match;
+	struct resource *res;
+	void __iomem *regs;
+	int ret;
+
+	match = of_match_node(rk_spdif_match, np);
+	if ((int) match->data == RK_SPDIF_RK3288) {
+		struct regmap *grf;
+
+		grf = syscon_regmap_lookup_by_phandle(np, "rockchip,grf");
+		if (IS_ERR(grf)) {
+			dev_err(&pdev->dev,
+				"rockchip_spdif missing 'rockchip,grf' \n");
+			return PTR_ERR(grf);
+		}
+
+		/* Select the 8 channel SPDIF solution on RK3288 as
+		 * the 2 channel one does not appear to work
+		 */
+		regmap_write(grf, RK3288_GRF_SOC_CON2, BIT(1) << 16);
+	}
+
+	spdif = devm_kzalloc(&pdev->dev, sizeof(*spdif), GFP_KERNEL);
+	if (!spdif)
+		return -ENOMEM;
+
+	spdif->hclk = devm_clk_get(&pdev->dev, "hclk");
+	if (IS_ERR(spdif->hclk)) {
+		dev_err(&pdev->dev, "Can't retrieve rk_spdif bus clock\n");
+		return PTR_ERR(spdif->hclk);
+	}
+	ret = clk_prepare_enable(spdif->hclk);
+	if (ret) {
+		dev_err(spdif->dev, "hclock enable failed %d\n", ret);
+		return ret;
+	}
+
+	spdif->mclk = devm_clk_get(&pdev->dev, "mclk");
+	if (IS_ERR(spdif->mclk)) {
+		dev_err(&pdev->dev, "Can't retrieve rk_spdif master clock\n");
+		return PTR_ERR(spdif->mclk);
+	}
+
+	ret = clk_prepare_enable(spdif->mclk);
+	if (ret) {
+		dev_err(spdif->dev, "clock enable failed %d\n", ret);
+		return ret;
+	}
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	regs = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	spdif->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "hclk", regs,
+						  &rk_spdif_regmap_config);
+	if (IS_ERR(spdif->regmap)) {
+		dev_err(&pdev->dev,
+			"Failed to initialise managed register map\n");
+		return PTR_ERR(spdif->regmap);
+	}
+
+	spdif->playback_dma_data.addr = res->start + SPDIF_SMPDR;
+	spdif->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+	spdif->playback_dma_data.maxburst = 4;
+
+	spdif->dev = &pdev->dev;
+	dev_set_drvdata(&pdev->dev, spdif);
+
+	pm_runtime_set_active(&pdev->dev);
+	pm_runtime_enable(&pdev->dev);
+	pm_request_idle(&pdev->dev);
+
+	ret = devm_snd_soc_register_component(&pdev->dev,
+					      &rk_spdif_component,
+					      &rk_spdif_dai, 1);
+	if (ret) {
+		dev_err(&pdev->dev, "Could not register DAI\n");
+		goto err_pm_runtime;
+	}
+
+	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+	if (ret) {
+		dev_err(&pdev->dev, "Could not register PCM\n");
+		goto err_pm_runtime;
+	}
+
+	return 0;
+
+err_pm_runtime:
+	pm_runtime_disable(&pdev->dev);
+
+	return ret;
+}
+
+static int rk_spdif_remove(struct platform_device *pdev)
+{
+	struct rk_spdif_dev *spdif = dev_get_drvdata(&pdev->dev);
+
+	pm_runtime_disable(&pdev->dev);
+	if (!pm_runtime_status_suspended(&pdev->dev))
+		rk_spdif_runtime_suspend(&pdev->dev);
+
+	clk_disable_unprepare(spdif->mclk);
+	clk_disable_unprepare(spdif->hclk);
+
+	return 0;
+}
+
+static const struct dev_pm_ops rk_spdif_pm_ops = {
+	SET_RUNTIME_PM_OPS(rk_spdif_runtime_suspend, rk_spdif_runtime_resume,
+			   NULL)
+};
+
+static struct platform_driver rk_spdif_driver = {
+	.probe = rk_spdif_probe,
+	.remove = rk_spdif_remove,
+	.driver = {
+		.name = "rockchip-spdif",
+		.of_match_table = of_match_ptr(rk_spdif_match),
+		.pm = &rk_spdif_pm_ops,
+	},
+};
+module_platform_driver(rk_spdif_driver);
+
+MODULE_ALIAS("platform:rockchip-spdif");
+MODULE_DESCRIPTION("ROCKCHIP SPDIF transceiver Interface");
+MODULE_AUTHOR("Sjoerd Simons <sjoerd.simons@collabora.co.uk>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/rockchip/rockchip_spdif.h b/sound/soc/rockchip/rockchip_spdif.h
new file mode 100644
index 000000000000..07f86a21046a
--- /dev/null
+++ b/sound/soc/rockchip/rockchip_spdif.h
@@ -0,0 +1,63 @@
+/*
+ * ALSA SoC Audio Layer - Rockchip SPDIF transceiver driver
+ *
+ * Copyright (c) 2015 Collabora Ltd.
+ * Author: Sjoerd Simons <sjoerd.simons@collabora.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _ROCKCHIP_SPDIF_H
+#define _ROCKCHIP_SPDIF_H
+
+/*
+ * CFGR
+ * transfer configuration register
+*/
+#define SPDIF_CFGR_CLK_DIV_SHIFT	(16)
+#define SPDIF_CFGR_CLK_DIV_MASK		(0xff << SPDIF_CFGR_CLK_DIV_SHIFT)
+#define SPDIF_CFGR_CLK_DIV(x)		(x << SPDIF_CFGR_CLK_DIV_SHIFT)
+
+#define SPDIF_CFGR_HALFWORD_SHIFT	2
+#define SPDIF_CFGR_HALFWORD_DISABLE	(0 << SPDIF_CFGR_HALFWORD_SHIFT)
+#define SPDIF_CFGR_HALFWORD_ENABLE	(1 << SPDIF_CFGR_HALFWORD_SHIFT)
+
+#define SPDIF_CFGR_VDW_SHIFT	0
+#define SPDIF_CFGR_VDW(x)	(x << SPDIF_CFGR_VDW_SHIFT)
+#define SDPIF_CFGR_VDW_MASK	(0xf << SPDIF_CFGR_VDW_SHIFT)
+
+#define SPDIF_CFGR_VDW_16	SPDIF_CFGR_VDW(0x00)
+#define SPDIF_CFGR_VDW_20	SPDIF_CFGR_VDW(0x01)
+#define SPDIF_CFGR_VDW_24	SPDIF_CFGR_VDW(0x10)
+
+/*
+ * DMACR
+ * DMA control register
+*/
+#define SPDIF_DMACR_TDE_SHIFT	5
+#define SPDIF_DMACR_TDE_DISABLE	(0 << SPDIF_DMACR_TDE_SHIFT)
+#define SPDIF_DMACR_TDE_ENABLE	(1 << SPDIF_DMACR_TDE_SHIFT)
+
+#define SPDIF_DMACR_TDL_SHIFT	0
+#define SPDIF_DMACR_TDL(x)	((x) << SPDIF_DMACR_TDL_SHIFT)
+#define SPDIF_DMACR_TDL_MASK	(0x1f << SDPIF_DMACR_TDL_SHIFT)
+
+/*
+ * XFER
+ * Transfer control register
+*/
+#define SPDIF_XFER_TXS_SHIFT	0
+#define SPDIF_XFER_TXS_STOP	(0 << SPDIF_XFER_TXS_SHIFT)
+#define SPDIF_XFER_TXS_START	(1 << SPDIF_XFER_TXS_SHIFT)
+
+#define SPDIF_CFGR	(0x0000)
+#define SPDIF_SDBLR	(0x0004)
+#define SPDIF_DMACR	(0x0008)
+#define SPDIF_INTCR	(0x000c)
+#define SPDIF_INTSR	(0x0010)
+#define SPDIF_XFER	(0x0018)
+#define SPDIF_SMPDR	(0x0020)
+
+#endif /* _ROCKCHIP_SPDIF_H */
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index c72e9fb26658..5f5825faeb2a 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -26,16 +26,15 @@
 #include <mach/gpio-samsung.h>
 #include "s3c24xx-i2s.h"
 
-static unsigned int rates[] = {
+static const unsigned int rates[] = {
 	11025,
 	22050,
 	44100,
 };
 
-static struct snd_pcm_hw_constraint_list hw_rates = {
+static const struct snd_pcm_hw_constraint_list hw_rates = {
 	.count = ARRAY_SIZE(rates),
 	.list = rates,
-	.mask = 0,
 };
 
 static struct snd_soc_jack hp_jack;
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 35e37c457f1f..fa096abe9e75 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -38,16 +38,15 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream,
 static int rx1950_spk_power(struct snd_soc_dapm_widget *w,
 				struct snd_kcontrol *kcontrol, int event);
 
-static unsigned int rates[] = {
+static const unsigned int rates[] = {
 	16000,
 	44100,
 	48000,
 };
 
-static struct snd_pcm_hw_constraint_list hw_rates = {
+static const struct snd_pcm_hw_constraint_list hw_rates = {
 	.count = ARRAY_SIZE(rates),
 	.list = rates,
-	.mask = 0,
 };
 
 static struct snd_soc_jack hp_jack;
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 07114b0b0dc1..206d1edab07c 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -37,10 +37,11 @@ config SND_SOC_SH4_SIU
 config SND_SOC_RCAR
 	tristate "R-Car series SRU/SCU/SSIU/SSI support"
 	depends on DMA_OF
+	depends on COMMON_CLK
 	select SND_SIMPLE_CARD
 	select REGMAP_MMIO
 	help
-	  This option enables R-Car SUR/SCU/SSIU/SSI sound support
+	  This option enables R-Car SRU/SCU/SSIU/SSI sound support
 
 config SND_SOC_RSRC_CARD
 	tristate "Renesas Sampling Rate Convert Sound Card"
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index fefc881dbac2..2a5b3a293cd2 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -7,7 +7,7 @@
  * License.  See the file "COPYING" in the main directory of this archive
  * for more details.
  */
-#include <linux/sh_clk.h>
+#include <linux/clk-provider.h>
 #include "rsnd.h"
 
 #define CLKA	0
@@ -16,12 +16,26 @@
 #define CLKI	3
 #define CLKMAX	4
 
+#define CLKOUT	0
+#define CLKOUT1	1
+#define CLKOUT2	2
+#define CLKOUT3	3
+#define CLKOUTMAX 4
+
+#define BRRx_MASK(x) (0x3FF & x)
+
+static struct rsnd_mod_ops adg_ops = {
+	.name = "adg",
+};
+
 struct rsnd_adg {
 	struct clk *clk[CLKMAX];
+	struct clk *clkout[CLKOUTMAX];
+	struct clk_onecell_data onecell;
+	struct rsnd_mod mod;
 
-	int rbga_rate_for_441khz_div_6;	/* RBGA */
-	int rbgb_rate_for_48khz_div_6;	/* RBGB */
-	u32 ckr;
+	int rbga_rate_for_441khz; /* RBGA */
+	int rbgb_rate_for_48khz;  /* RBGB */
 };
 
 #define for_each_rsnd_clk(pos, adg, i)		\
@@ -29,17 +43,36 @@ struct rsnd_adg {
 	     (i < CLKMAX) &&			\
 	     ((pos) = adg->clk[i]);		\
 	     i++)
+#define for_each_rsnd_clkout(pos, adg, i)	\
+	for (i = 0;				\
+	     (i < CLKOUTMAX) &&			\
+	     ((pos) = adg->clkout[i]);	\
+	     i++)
 #define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg)
 
+static u32 rsnd_adg_calculate_rbgx(unsigned long div)
+{
+	int i, ratio;
+
+	if (!div)
+		return 0;
+
+	for (i = 3; i >= 0; i--) {
+		ratio = 2 << (i * 2);
+		if (0 == (div % ratio))
+			return (u32)((i << 8) | ((div / ratio) - 1));
+	}
+
+	return ~0;
+}
 
 static u32 rsnd_adg_ssi_ws_timing_gen2(struct rsnd_dai_stream *io)
 {
 	struct rsnd_mod *mod = rsnd_io_to_mod_ssi(io);
-	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
 	int id = rsnd_mod_id(mod);
 	int ws = id;
 
-	if (rsnd_ssi_is_pin_sharing(rsnd_ssi_mod_get(priv, id))) {
+	if (rsnd_ssi_is_pin_sharing(io)) {
 		switch (id) {
 		case 1:
 		case 2:
@@ -60,6 +93,9 @@ static u32 rsnd_adg_ssi_ws_timing_gen2(struct rsnd_dai_stream *io)
 int rsnd_adg_set_cmd_timsel_gen2(struct rsnd_mod *mod,
 				 struct rsnd_dai_stream *io)
 {
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+	struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
 	int id = rsnd_mod_id(mod);
 	int shift = (id % 2) ? 16 : 0;
 	u32 mask, val;
@@ -69,21 +105,26 @@ int rsnd_adg_set_cmd_timsel_gen2(struct rsnd_mod *mod,
 	val  = val	<< shift;
 	mask = 0xffff	<< shift;
 
-	rsnd_mod_bset(mod, CMDOUT_TIMSEL, mask, val);
+	rsnd_mod_bset(adg_mod, CMDOUT_TIMSEL, mask, val);
 
 	return 0;
 }
 
-static int rsnd_adg_set_src_timsel_gen2(struct rsnd_mod *mod,
+static int rsnd_adg_set_src_timsel_gen2(struct rsnd_mod *src_mod,
 					struct rsnd_dai_stream *io,
 					u32 timsel)
 {
+	struct rsnd_priv *priv = rsnd_mod_to_priv(src_mod);
+	struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+	struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
 	int is_play = rsnd_io_is_play(io);
-	int id = rsnd_mod_id(mod);
+	int id = rsnd_mod_id(src_mod);
 	int shift = (id % 2) ? 16 : 0;
 	u32 mask, ws;
 	u32 in, out;
 
+	rsnd_mod_confirm_src(src_mod);
+
 	ws = rsnd_adg_ssi_ws_timing_gen2(io);
 
 	in  = (is_play) ? timsel : ws;
@@ -95,37 +136,38 @@ static int rsnd_adg_set_src_timsel_gen2(struct rsnd_mod *mod,
 
 	switch (id / 2) {
 	case 0:
-		rsnd_mod_bset(mod, SRCIN_TIMSEL0,  mask, in);
-		rsnd_mod_bset(mod, SRCOUT_TIMSEL0, mask, out);
+		rsnd_mod_bset(adg_mod, SRCIN_TIMSEL0,  mask, in);
+		rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL0, mask, out);
 		break;
 	case 1:
-		rsnd_mod_bset(mod, SRCIN_TIMSEL1,  mask, in);
-		rsnd_mod_bset(mod, SRCOUT_TIMSEL1, mask, out);
+		rsnd_mod_bset(adg_mod, SRCIN_TIMSEL1,  mask, in);
+		rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL1, mask, out);
 		break;
 	case 2:
-		rsnd_mod_bset(mod, SRCIN_TIMSEL2,  mask, in);
-		rsnd_mod_bset(mod, SRCOUT_TIMSEL2, mask, out);
+		rsnd_mod_bset(adg_mod, SRCIN_TIMSEL2,  mask, in);
+		rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL2, mask, out);
 		break;
 	case 3:
-		rsnd_mod_bset(mod, SRCIN_TIMSEL3,  mask, in);
-		rsnd_mod_bset(mod, SRCOUT_TIMSEL3, mask, out);
+		rsnd_mod_bset(adg_mod, SRCIN_TIMSEL3,  mask, in);
+		rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL3, mask, out);
 		break;
 	case 4:
-		rsnd_mod_bset(mod, SRCIN_TIMSEL4,  mask, in);
-		rsnd_mod_bset(mod, SRCOUT_TIMSEL4, mask, out);
+		rsnd_mod_bset(adg_mod, SRCIN_TIMSEL4,  mask, in);
+		rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL4, mask, out);
 		break;
 	}
 
 	return 0;
 }
 
-int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod,
+int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *src_mod,
 				  struct rsnd_dai_stream *io,
 				  unsigned int src_rate,
 				  unsigned int dst_rate)
 {
-	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct rsnd_priv *priv = rsnd_mod_to_priv(src_mod);
 	struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+	struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
 	struct device *dev = rsnd_priv_to_dev(priv);
 	int idx, sel, div, step, ret;
 	u32 val, en;
@@ -134,10 +176,12 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod,
 		clk_get_rate(adg->clk[CLKA]),	/* 0000: CLKA */
 		clk_get_rate(adg->clk[CLKB]),	/* 0001: CLKB */
 		clk_get_rate(adg->clk[CLKC]),	/* 0010: CLKC */
-		adg->rbga_rate_for_441khz_div_6,/* 0011: RBGA */
-		adg->rbgb_rate_for_48khz_div_6,	/* 0100: RBGB */
+		adg->rbga_rate_for_441khz,	/* 0011: RBGA */
+		adg->rbgb_rate_for_48khz,	/* 0100: RBGB */
 	};
 
+	rsnd_mod_confirm_src(src_mod);
+
 	min = ~0;
 	val = 0;
 	en = 0;
@@ -175,25 +219,27 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod,
 		return -EIO;
 	}
 
-	ret = rsnd_adg_set_src_timsel_gen2(mod, io, val);
+	ret = rsnd_adg_set_src_timsel_gen2(src_mod, io, val);
 	if (ret < 0) {
 		dev_err(dev, "timsel error\n");
 		return ret;
 	}
 
-	rsnd_mod_bset(mod, DIV_EN, en, en);
+	rsnd_mod_bset(adg_mod, DIV_EN, en, en);
 
 	dev_dbg(dev, "convert rate %d <-> %d\n", src_rate, dst_rate);
 
 	return 0;
 }
 
-int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *mod,
+int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *src_mod,
 				     struct rsnd_dai_stream *io)
 {
 	u32 val = rsnd_adg_ssi_ws_timing_gen2(io);
 
-	return rsnd_adg_set_src_timsel_gen2(mod, io, val);
+	rsnd_mod_confirm_src(src_mod);
+
+	return rsnd_adg_set_src_timsel_gen2(src_mod, io, val);
 }
 
 int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
@@ -202,6 +248,7 @@ int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
 				  unsigned int dst_rate)
 {
 	struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+	struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
 	struct device *dev = rsnd_priv_to_dev(priv);
 	int idx, sel, div, shift;
 	u32 mask, val;
@@ -211,8 +258,8 @@ int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
 		clk_get_rate(adg->clk[CLKB]),	/* 001: CLKB */
 		clk_get_rate(adg->clk[CLKC]),	/* 010: CLKC */
 		0,				/* 011: MLBCLK (not used) */
-		adg->rbga_rate_for_441khz_div_6,/* 100: RBGA */
-		adg->rbgb_rate_for_48khz_div_6,	/* 101: RBGB */
+		adg->rbga_rate_for_441khz,	/* 100: RBGA */
+		adg->rbgb_rate_for_48khz,	/* 101: RBGB */
 	};
 
 	/* find div (= 1/128, 1/256, 1/512, 1/1024, 1/2048 */
@@ -238,13 +285,13 @@ find_rate:
 
 	switch (id / 4) {
 	case 0:
-		rsnd_mod_bset(mod, AUDIO_CLK_SEL3, mask, val);
+		rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL3, mask, val);
 		break;
 	case 1:
-		rsnd_mod_bset(mod, AUDIO_CLK_SEL4, mask, val);
+		rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL4, mask, val);
 		break;
 	case 2:
-		rsnd_mod_bset(mod, AUDIO_CLK_SEL5, mask, val);
+		rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL5, mask, val);
 		break;
 	}
 
@@ -257,12 +304,17 @@ find_rate:
 	return 0;
 }
 
-static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val)
+static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val)
 {
-	int id = rsnd_mod_id(mod);
+	struct rsnd_priv *priv = rsnd_mod_to_priv(ssi_mod);
+	struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+	struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
+	int id = rsnd_mod_id(ssi_mod);
 	int shift = (id % 4) * 8;
 	u32 mask = 0xFF << shift;
 
+	rsnd_mod_confirm_ssi(ssi_mod);
+
 	val = val << shift;
 
 	/*
@@ -274,13 +326,13 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val)
 
 	switch (id / 4) {
 	case 0:
-		rsnd_mod_bset(mod, AUDIO_CLK_SEL0, mask, val);
+		rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL0, mask, val);
 		break;
 	case 1:
-		rsnd_mod_bset(mod, AUDIO_CLK_SEL1, mask, val);
+		rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL1, mask, val);
 		break;
 	case 2:
-		rsnd_mod_bset(mod, AUDIO_CLK_SEL2, mask, val);
+		rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL2, mask, val);
 		break;
 	}
 }
@@ -326,14 +378,14 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate)
 	}
 
 	/*
-	 * find 1/6 clock from BRGA/BRGB
+	 * find divided clock from BRGA/BRGB
 	 */
-	if (rate == adg->rbga_rate_for_441khz_div_6) {
+	if (rate  == adg->rbga_rate_for_441khz) {
 		data = 0x10;
 		goto found_clock;
 	}
 
-	if (rate == adg->rbgb_rate_for_48khz_div_6) {
+	if (rate == adg->rbgb_rate_for_48khz) {
 		data = 0x20;
 		goto found_clock;
 	}
@@ -342,29 +394,60 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate)
 
 found_clock:
 
-	/* see rsnd_adg_ssi_clk_init() */
-	rsnd_mod_bset(mod, SSICKR, 0x00FF0000, adg->ckr);
-	rsnd_mod_write(mod, BRRA,  0x00000002); /* 1/6 */
-	rsnd_mod_write(mod, BRRB,  0x00000002); /* 1/6 */
-
 	/*
 	 * This "mod" = "ssi" here.
 	 * we can get "ssi id" from mod
 	 */
 	rsnd_adg_set_ssi_clk(mod, data);
 
-	dev_dbg(dev, "ADG: ssi%d selects clk%d = %d",
-		rsnd_mod_id(mod), i, rate);
+	dev_dbg(dev, "ADG: %s[%d] selects 0x%x for %d\n",
+		rsnd_mod_name(mod), rsnd_mod_id(mod),
+		data, rate);
 
 	return 0;
 }
 
-static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
+static void rsnd_adg_get_clkin(struct rsnd_priv *priv,
+			       struct rsnd_adg *adg)
+{
+	struct device *dev = rsnd_priv_to_dev(priv);
+	struct clk *clk;
+	static const char * const clk_name[] = {
+		[CLKA]	= "clk_a",
+		[CLKB]	= "clk_b",
+		[CLKC]	= "clk_c",
+		[CLKI]	= "clk_i",
+	};
+	int i;
+
+	for (i = 0; i < CLKMAX; i++) {
+		clk = devm_clk_get(dev, clk_name[i]);
+		adg->clk[i] = IS_ERR(clk) ? NULL : clk;
+	}
+
+	for_each_rsnd_clk(clk, adg, i)
+		dev_dbg(dev, "clk %d : %p : %ld\n", i, clk, clk_get_rate(clk));
+}
+
+static void rsnd_adg_get_clkout(struct rsnd_priv *priv,
+				struct rsnd_adg *adg)
 {
 	struct clk *clk;
-	unsigned long rate;
-	u32 ckr;
+	struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
+	struct device *dev = rsnd_priv_to_dev(priv);
+	struct device_node *np = dev->of_node;
+	u32 ckr, rbgx, rbga, rbgb;
+	u32 rate, req_rate, div;
+	uint32_t count = 0;
+	unsigned long req_48kHz_rate, req_441kHz_rate;
 	int i;
+	const char *parent_clk_name = NULL;
+	static const char * const clkout_name[] = {
+		[CLKOUT]  = "audio_clkout",
+		[CLKOUT1] = "audio_clkout1",
+		[CLKOUT2] = "audio_clkout2",
+		[CLKOUT3] = "audio_clkout3",
+	};
 	int brg_table[] = {
 		[CLKA] = 0x0,
 		[CLKB] = 0x1,
@@ -372,19 +455,34 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
 		[CLKI] = 0x2,
 	};
 
+	of_property_read_u32(np, "#clock-cells", &count);
+
+	/*
+	 * ADG supports BRRA/BRRB output only
+	 * this means all clkout0/1/2/3 will be same rate
+	 */
+	of_property_read_u32(np, "clock-frequency", &req_rate);
+	req_48kHz_rate = 0;
+	req_441kHz_rate = 0;
+	if (0 == (req_rate % 44100))
+		req_441kHz_rate = req_rate;
+	if (0 == (req_rate % 48000))
+		req_48kHz_rate = req_rate;
+
 	/*
 	 * 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()
+	 *	rsnd_ssi_master_clk_start()
 	 */
 	ckr = 0;
-	adg->rbga_rate_for_441khz_div_6 = 0;
-	adg->rbgb_rate_for_48khz_div_6  = 0;
+	rbga = 2; /* default 1/6 */
+	rbgb = 2; /* default 1/6 */
+	adg->rbga_rate_for_441khz	= 0;
+	adg->rbgb_rate_for_48khz	= 0;
 	for_each_rsnd_clk(clk, adg, i) {
 		rate = clk_get_rate(clk);
 
@@ -392,19 +490,86 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
 			continue;
 
 		/* RBGA */
-		if (!adg->rbga_rate_for_441khz_div_6 && (0 == rate % 44100)) {
-			adg->rbga_rate_for_441khz_div_6 = rate / 6;
-			ckr |= brg_table[i] << 20;
+		if (!adg->rbga_rate_for_441khz && (0 == rate % 44100)) {
+			div = 6;
+			if (req_441kHz_rate)
+				div = rate / req_441kHz_rate;
+			rbgx = rsnd_adg_calculate_rbgx(div);
+			if (BRRx_MASK(rbgx) == rbgx) {
+				rbga = rbgx;
+				adg->rbga_rate_for_441khz = rate / div;
+				ckr |= brg_table[i] << 20;
+				if (req_441kHz_rate)
+					parent_clk_name = __clk_get_name(clk);
+			}
 		}
 
 		/* RBGB */
-		if (!adg->rbgb_rate_for_48khz_div_6 && (0 == rate % 48000)) {
-			adg->rbgb_rate_for_48khz_div_6 = rate / 6;
-			ckr |= brg_table[i] << 16;
+		if (!adg->rbgb_rate_for_48khz && (0 == rate % 48000)) {
+			div = 6;
+			if (req_48kHz_rate)
+				div = rate / req_48kHz_rate;
+			rbgx = rsnd_adg_calculate_rbgx(div);
+			if (BRRx_MASK(rbgx) == rbgx) {
+				rbgb = rbgx;
+				adg->rbgb_rate_for_48khz = rate / div;
+				ckr |= brg_table[i] << 16;
+				if (req_48kHz_rate) {
+					parent_clk_name = __clk_get_name(clk);
+					ckr |= 0x80000000;
+				}
+			}
 		}
 	}
 
-	adg->ckr = ckr;
+	/*
+	 * ADG supports BRRA/BRRB output only.
+	 * this means all clkout0/1/2/3 will be * same rate
+	 */
+
+	/*
+	 * for clkout
+	 */
+	if (!count) {
+		clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT],
+					      parent_clk_name,
+					      (parent_clk_name) ?
+					      0 : CLK_IS_ROOT, req_rate);
+		if (!IS_ERR(clk)) {
+			adg->clkout[CLKOUT] = clk;
+			of_clk_add_provider(np, of_clk_src_simple_get, clk);
+		}
+	}
+	/*
+	 * for clkout0/1/2/3
+	 */
+	else {
+		for (i = 0; i < CLKOUTMAX; i++) {
+			clk = clk_register_fixed_rate(dev, clkout_name[i],
+						      parent_clk_name,
+						      (parent_clk_name) ?
+						      0 : CLK_IS_ROOT,
+						      req_rate);
+			if (!IS_ERR(clk)) {
+				adg->onecell.clks	= adg->clkout;
+				adg->onecell.clk_num	= CLKOUTMAX;
+
+				adg->clkout[i] = clk;
+
+				of_clk_add_provider(np, of_clk_src_onecell_get,
+						    &adg->onecell);
+			}
+		}
+	}
+
+	rsnd_mod_bset(adg_mod, SSICKR, 0x00FF0000, ckr);
+	rsnd_mod_write(adg_mod, BRRA,  rbga);
+	rsnd_mod_write(adg_mod, BRRB,  rbgb);
+
+	for_each_rsnd_clkout(clk, adg, i)
+		dev_dbg(dev, "clkout %d : %p : %ld\n", i, clk, clk_get_rate(clk));
+	dev_dbg(dev, "SSICKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n",
+		ckr, rbga, rbgb);
 }
 
 int rsnd_adg_probe(struct platform_device *pdev,
@@ -413,8 +578,6 @@ int rsnd_adg_probe(struct platform_device *pdev,
 {
 	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) {
@@ -422,15 +585,16 @@ int rsnd_adg_probe(struct platform_device *pdev,
 		return -ENOMEM;
 	}
 
-	adg->clk[CLKA]	= devm_clk_get(dev, "clk_a");
-	adg->clk[CLKB]	= devm_clk_get(dev, "clk_b");
-	adg->clk[CLKC]	= devm_clk_get(dev, "clk_c");
-	adg->clk[CLKI]	= devm_clk_get(dev, "clk_i");
-
-	for_each_rsnd_clk(clk, adg, i)
-		dev_dbg(dev, "clk %d : %p : %ld\n", i, clk, clk_get_rate(clk));
+	/*
+	 * ADG is special module.
+	 * Use ADG mod without rsnd_mod_init() to make debug easy
+	 * for rsnd_write/rsnd_read
+	 */
+	adg->mod.ops = &adg_ops;
+	adg->mod.priv = priv;
 
-	rsnd_adg_ssi_clk_init(priv, adg);
+	rsnd_adg_get_clkin(priv, adg);
+	rsnd_adg_get_clkout(priv, adg);
 
 	priv->adg = adg;
 
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index f3feed5ce9b6..deed48ef28b8 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -110,6 +110,7 @@ static const struct rsnd_of_data rsnd_of_data_gen2 = {
 static const struct of_device_id rsnd_of_match[] = {
 	{ .compatible = "renesas,rcar_sound-gen1", .data = &rsnd_of_data_gen1 },
 	{ .compatible = "renesas,rcar_sound-gen2", .data = &rsnd_of_data_gen2 },
+	{ .compatible = "renesas,rcar_sound-gen3", .data = &rsnd_of_data_gen2 }, /* gen2 compatible */
 	{},
 };
 MODULE_DEVICE_TABLE(of, rsnd_of_match);
@@ -126,6 +127,17 @@ MODULE_DEVICE_TABLE(of, rsnd_of_match);
 #define rsnd_info_id(priv, io, name) \
 	((io)->info->name - priv->info->name##_info)
 
+void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type)
+{
+	if (mod->type != type) {
+		struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+		struct device *dev = rsnd_priv_to_dev(priv);
+
+		dev_warn(dev, "%s[%d] is not your expected module\n",
+			 rsnd_mod_name(mod), rsnd_mod_id(mod));
+	}
+}
+
 /*
  *	rsnd_mod functions
  */
@@ -288,7 +300,7 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io)
 /*
  *	rsnd_dai functions
  */
-#define __rsnd_mod_call(mod, io, func, param...)		\
+#define rsnd_mod_call(mod, io, func, param...)			\
 ({								\
 	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);		\
 	struct device *dev = rsnd_priv_to_dev(priv);		\
@@ -296,24 +308,17 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io)
 	u8 val  = (mod->status >> __rsnd_mod_shift_##func) & 0xF;	\
 	u8 add  = ((val + __rsnd_mod_add_##func) & 0xF);		\
 	int ret = 0;							\
-	int called = 0;							\
-	if (val == __rsnd_mod_call_##func) {				\
-		called = 1;						\
-		ret = (mod)->ops->func(mod, io, param);			\
-	}								\
+	int call = (val == __rsnd_mod_call_##func) && (mod)->ops->func;	\
 	mod->status = (mod->status & ~mask) +				\
 		(add << __rsnd_mod_shift_##func);			\
-	dev_dbg(dev, "%s[%d] 0x%08x %s\n",				\
-		rsnd_mod_name(mod), rsnd_mod_id(mod), mod->status,	\
-		called ? #func : "");					\
+	dev_dbg(dev, "%s[%d]\t0x%08x %s\n",				\
+		rsnd_mod_name(mod), rsnd_mod_id(mod),			\
+		mod->status, call ? #func : "");			\
+	if (call)							\
+		ret = (mod)->ops->func(mod, io, param);			\
 	ret;								\
 })
 
-#define rsnd_mod_call(mod, io, func, param...)	\
-	(!(mod) ? -ENODEV :			\
-	 !((mod)->ops->func) ? 0 :		\
-	 __rsnd_mod_call(mod, io, func, param))
-
 #define rsnd_dai_call(fn, io, param...)				\
 ({								\
 	struct rsnd_mod *mod;					\
@@ -322,9 +327,7 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io)
 		mod = (io)->mod[i];				\
 		if (!mod)					\
 			continue;				\
-		ret = rsnd_mod_call(mod, io, fn, param);	\
-		if (ret < 0)					\
-			break;					\
+		ret |= rsnd_mod_call(mod, io, fn, param);	\
 	}							\
 	ret;							\
 })
@@ -490,16 +493,10 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd,
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 		ret = rsnd_dai_call(stop, io, priv);
-		if (ret < 0)
-			goto dai_trigger_end;
 
-		ret = rsnd_dai_call(quit, io, priv);
-		if (ret < 0)
-			goto dai_trigger_end;
+		ret |= rsnd_dai_call(quit, io, priv);
 
-		ret = rsnd_platform_call(priv, dai, stop, ssi_id);
-		if (ret < 0)
-			goto dai_trigger_end;
+		ret |= rsnd_platform_call(priv, dai, stop, ssi_id);
 
 		rsnd_dai_stream_quit(io);
 		break;
@@ -1224,20 +1221,11 @@ static int rsnd_probe(struct platform_device *pdev)
 	};
 	int ret, i;
 
-	info = NULL;
-	of_data = NULL;
-	if (of_id) {
-		info = devm_kzalloc(&pdev->dev,
-				    sizeof(struct rcar_snd_info), GFP_KERNEL);
-		of_data = of_id->data;
-	} else {
-		info = pdev->dev.platform_data;
-	}
-
-	if (!info) {
-		dev_err(dev, "driver needs R-Car sound information\n");
-		return -ENODEV;
-	}
+	info = devm_kzalloc(&pdev->dev, sizeof(struct rcar_snd_info),
+			    GFP_KERNEL);
+	if (!info)
+		return -ENOMEM;
+	of_data = of_id->data;
 
 	/*
 	 *	init priv data
diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c
index 05498bba5874..3cb214ab848b 100644
--- a/sound/soc/sh/rcar/ctu.c
+++ b/sound/soc/sh/rcar/ctu.c
@@ -35,7 +35,7 @@ static int rsnd_ctu_init(struct rsnd_mod *mod,
 			 struct rsnd_dai_stream *io,
 			 struct rsnd_priv *priv)
 {
-	rsnd_mod_hw_start(mod);
+	rsnd_mod_power_on(mod);
 
 	rsnd_ctu_initialize_lock(mod);
 
@@ -50,7 +50,7 @@ static int rsnd_ctu_quit(struct rsnd_mod *mod,
 			 struct rsnd_dai_stream *io,
 			 struct rsnd_priv *priv)
 {
-	rsnd_mod_hw_stop(mod);
+	rsnd_mod_power_off(mod);
 
 	return 0;
 }
@@ -66,7 +66,7 @@ struct rsnd_mod *rsnd_ctu_mod_get(struct rsnd_priv *priv, int id)
 	if (WARN_ON(id < 0 || id >= rsnd_ctu_nr(priv)))
 		id = 0;
 
-	return &((struct rsnd_ctu *)(priv->ctu) + id)->mod;
+	return rsnd_mod_get((struct rsnd_ctu *)(priv->ctu) + id);
 }
 
 static void rsnd_of_parse_ctu(struct platform_device *pdev,
@@ -118,10 +118,8 @@ int rsnd_ctu_probe(struct platform_device *pdev,
 	int i, nr, ret;
 
 	/* This driver doesn't support Gen1 at this point */
-	if (rsnd_is_gen1(priv)) {
-		dev_warn(dev, "CTU is not supported on Gen1\n");
-		return -EINVAL;
-	}
+	if (rsnd_is_gen1(priv))
+		return 0;
 
 	rsnd_of_parse_ctu(pdev, of_data, priv);
 
@@ -150,7 +148,7 @@ int rsnd_ctu_probe(struct platform_device *pdev,
 
 		ctu->info = &info->ctu_info[i];
 
-		ret = rsnd_mod_init(priv, &ctu->mod, &rsnd_ctu_ops,
+		ret = rsnd_mod_init(priv, rsnd_mod_get(ctu), &rsnd_ctu_ops,
 				    clk, RSND_MOD_CTU, i);
 		if (ret)
 			return ret;
@@ -166,6 +164,6 @@ void rsnd_ctu_remove(struct platform_device *pdev,
 	int i;
 
 	for_each_rsnd_ctu(ctu, priv, i) {
-		rsnd_mod_quit(&ctu->mod);
+		rsnd_mod_quit(rsnd_mod_get(ctu));
 	}
 }
diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c
index bfbb8a5e93bd..5d084d040961 100644
--- a/sound/soc/sh/rcar/dma.c
+++ b/sound/soc/sh/rcar/dma.c
@@ -470,7 +470,7 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io,
 		dev_err(dev, "DVC is selected without SRC\n");
 
 	/* use SSIU or SSI ? */
-	if (is_ssi && rsnd_ssi_use_busif(io, mod))
+	if (is_ssi && rsnd_ssi_use_busif(io))
 		is_ssi++;
 
 	return (is_from) ?
diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c
index 57796387d482..58f690900e6d 100644
--- a/sound/soc/sh/rcar/dvc.c
+++ b/sound/soc/sh/rcar/dvc.c
@@ -153,7 +153,7 @@ static int rsnd_dvc_init(struct rsnd_mod *mod,
 			 struct rsnd_dai_stream *io,
 			 struct rsnd_priv *priv)
 {
-	rsnd_mod_hw_start(mod);
+	rsnd_mod_power_on(mod);
 
 	rsnd_dvc_soft_reset(mod);
 
@@ -175,7 +175,7 @@ static int rsnd_dvc_quit(struct rsnd_mod *mod,
 			 struct rsnd_dai_stream *io,
 			 struct rsnd_priv *priv)
 {
-	rsnd_mod_hw_stop(mod);
+	rsnd_mod_power_off(mod);
 
 	return 0;
 }
@@ -282,7 +282,7 @@ struct rsnd_mod *rsnd_dvc_mod_get(struct rsnd_priv *priv, int id)
 	if (WARN_ON(id < 0 || id >= rsnd_dvc_nr(priv)))
 		id = 0;
 
-	return &((struct rsnd_dvc *)(priv->dvc) + id)->mod;
+	return rsnd_mod_get((struct rsnd_dvc *)(priv->dvc) + id);
 }
 
 static void rsnd_of_parse_dvc(struct platform_device *pdev,
@@ -333,10 +333,8 @@ int rsnd_dvc_probe(struct platform_device *pdev,
 	int i, nr, ret;
 
 	/* This driver doesn't support Gen1 at this point */
-	if (rsnd_is_gen1(priv)) {
-		dev_warn(dev, "CMD is not supported on Gen1\n");
-		return -EINVAL;
-	}
+	if (rsnd_is_gen1(priv))
+		return 0;
 
 	rsnd_of_parse_dvc(pdev, of_data, priv);
 
@@ -361,7 +359,7 @@ int rsnd_dvc_probe(struct platform_device *pdev,
 
 		dvc->info = &info->dvc_info[i];
 
-		ret = rsnd_mod_init(priv, &dvc->mod, &rsnd_dvc_ops,
+		ret = rsnd_mod_init(priv, rsnd_mod_get(dvc), &rsnd_dvc_ops,
 			      clk, RSND_MOD_DVC, i);
 		if (ret)
 			return ret;
@@ -377,6 +375,6 @@ void rsnd_dvc_remove(struct platform_device *pdev,
 	int i;
 
 	for_each_rsnd_dvc(dvc, priv, i) {
-		rsnd_mod_quit(&dvc->mod);
+		rsnd_mod_quit(rsnd_mod_get(dvc));
 	}
 }
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index f04d17bc6e3d..76da7620904c 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -22,13 +22,15 @@
 #include "rsnd.h"
 
 struct rsnd_gen {
-	void __iomem *base[RSND_BASE_MAX];
-
 	struct rsnd_gen_ops *ops;
 
+	/* RSND_BASE_MAX base */
+	void __iomem *base[RSND_BASE_MAX];
+	phys_addr_t res[RSND_BASE_MAX];
 	struct regmap *regmap[RSND_BASE_MAX];
+
+	/* RSND_REG_MAX base */
 	struct regmap_field *regs[RSND_REG_MAX];
-	phys_addr_t res[RSND_REG_MAX];
 };
 
 #define rsnd_priv_to_gen(p)	((struct rsnd_gen *)(p)->gen)
@@ -79,11 +81,11 @@ u32 rsnd_read(struct rsnd_priv *priv,
 	if (!rsnd_is_accessible_reg(priv, gen, reg))
 		return 0;
 
+	regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val);
+
 	dev_dbg(dev, "r %s[%d] - %4d : %08x\n",
 		rsnd_mod_name(mod), rsnd_mod_id(mod), reg, val);
 
-	regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val);
-
 	return val;
 }
 
@@ -182,6 +184,7 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv,
 	if (IS_ERR(regmap))
 		return PTR_ERR(regmap);
 
+	/* RSND_BASE_MAX base */
 	gen->base[reg_id] = base;
 	gen->regmap[reg_id] = regmap;
 	gen->res[reg_id] = res->start;
@@ -198,6 +201,7 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv,
 		if (IS_ERR(regs))
 			return PTR_ERR(regs);
 
+		/* RSND_REG_MAX base */
 		gen->regs[conf[i].idx] = regs;
 	}
 
diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c
index 0d5c102db6f5..953dd0be9b60 100644
--- a/sound/soc/sh/rcar/mix.c
+++ b/sound/soc/sh/rcar/mix.c
@@ -58,7 +58,7 @@ static int rsnd_mix_init(struct rsnd_mod *mod,
 			 struct rsnd_dai_stream *io,
 			 struct rsnd_priv *priv)
 {
-	rsnd_mod_hw_start(mod);
+	rsnd_mod_power_on(mod);
 
 	rsnd_mix_soft_reset(mod);
 
@@ -83,7 +83,7 @@ static int rsnd_mix_quit(struct rsnd_mod *mod,
 			 struct rsnd_dai_stream *io,
 			 struct rsnd_priv *priv)
 {
-	rsnd_mod_hw_stop(mod);
+	rsnd_mod_power_off(mod);
 
 	return 0;
 }
@@ -99,7 +99,7 @@ struct rsnd_mod *rsnd_mix_mod_get(struct rsnd_priv *priv, int id)
 	if (WARN_ON(id < 0 || id >= rsnd_mix_nr(priv)))
 		id = 0;
 
-	return &((struct rsnd_mix *)(priv->mix) + id)->mod;
+	return rsnd_mod_get((struct rsnd_mix *)(priv->mix) + id);
 }
 
 static void rsnd_of_parse_mix(struct platform_device *pdev,
@@ -151,10 +151,8 @@ int rsnd_mix_probe(struct platform_device *pdev,
 	int i, nr, ret;
 
 	/* This driver doesn't support Gen1 at this point */
-	if (rsnd_is_gen1(priv)) {
-		dev_warn(dev, "MIX is not supported on Gen1\n");
-		return -EINVAL;
-	}
+	if (rsnd_is_gen1(priv))
+		return 0;
 
 	rsnd_of_parse_mix(pdev, of_data, priv);
 
@@ -179,7 +177,7 @@ int rsnd_mix_probe(struct platform_device *pdev,
 
 		mix->info = &info->mix_info[i];
 
-		ret = rsnd_mod_init(priv, &mix->mod, &rsnd_mix_ops,
+		ret = rsnd_mod_init(priv, rsnd_mod_get(mix), &rsnd_mix_ops,
 				    clk, RSND_MOD_MIX, i);
 		if (ret)
 			return ret;
@@ -195,6 +193,6 @@ void rsnd_mix_remove(struct platform_device *pdev,
 	int i;
 
 	for_each_rsnd_mix(mix, priv, i) {
-		rsnd_mod_quit(&mix->mod);
+		rsnd_mod_quit(rsnd_mod_get(mix));
 	}
 }
diff --git a/sound/soc/sh/rcar/rcar_snd.h b/sound/soc/sh/rcar/rcar_snd.h
new file mode 100644
index 000000000000..d8e33d38da43
--- /dev/null
+++ b/sound/soc/sh/rcar/rcar_snd.h
@@ -0,0 +1,117 @@
+/*
+ * Renesas R-Car SRU/SCU/SSIU/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.
+ */
+
+#ifndef RCAR_SND_H
+#define RCAR_SND_H
+
+
+#define RSND_GEN1_SRU	0
+#define RSND_GEN1_ADG	1
+#define RSND_GEN1_SSI	2
+
+#define RSND_GEN2_SCU	0
+#define RSND_GEN2_ADG	1
+#define RSND_GEN2_SSIU	2
+#define RSND_GEN2_SSI	3
+
+#define RSND_BASE_MAX	4
+
+/*
+ * flags
+ *
+ * 0xAB000000
+ *
+ * A : clock sharing settings
+ * B : SSI direction
+ */
+#define RSND_SSI_CLK_PIN_SHARE		(1 << 31)
+#define RSND_SSI_NO_BUSIF		(1 << 30) /* SSI+DMA without BUSIF */
+
+#define RSND_SSI(_dma_id, _irq, _flags)		\
+{ .dma_id = _dma_id, .irq = _irq, .flags = _flags }
+#define RSND_SSI_UNUSED \
+{ .dma_id = -1, .irq = -1, .flags = 0 }
+
+struct rsnd_ssi_platform_info {
+	int dma_id;
+	int irq;
+	u32 flags;
+};
+
+#define RSND_SRC(rate, _dma_id)						\
+{ .convert_rate = rate, .dma_id = _dma_id, }
+#define RSND_SRC_UNUSED				\
+{ .convert_rate = 0, .dma_id = -1, }
+
+struct rsnd_src_platform_info {
+	u32 convert_rate; /* sampling rate convert */
+	int dma_id; /* for Gen2 SCU */
+	int irq;
+};
+
+/*
+ * flags
+ */
+struct rsnd_ctu_platform_info {
+	u32 flags;
+};
+
+struct rsnd_mix_platform_info {
+	u32 flags;
+};
+
+struct rsnd_dvc_platform_info {
+	u32 flags;
+};
+
+struct rsnd_dai_path_info {
+	struct rsnd_ssi_platform_info *ssi;
+	struct rsnd_src_platform_info *src;
+	struct rsnd_ctu_platform_info *ctu;
+	struct rsnd_mix_platform_info *mix;
+	struct rsnd_dvc_platform_info *dvc;
+};
+
+struct rsnd_dai_platform_info {
+	struct rsnd_dai_path_info playback;
+	struct rsnd_dai_path_info capture;
+};
+
+/*
+ * flags
+ *
+ * 0x0000000A
+ *
+ * A : generation
+ */
+#define RSND_GEN_MASK	(0xF << 0)
+#define RSND_GEN1	(1 << 0) /* fixme */
+#define RSND_GEN2	(2 << 0) /* fixme */
+
+struct rcar_snd_info {
+	u32 flags;
+	struct rsnd_ssi_platform_info *ssi_info;
+	int ssi_info_nr;
+	struct rsnd_src_platform_info *src_info;
+	int src_info_nr;
+	struct rsnd_ctu_platform_info *ctu_info;
+	int ctu_info_nr;
+	struct rsnd_mix_platform_info *mix_info;
+	int mix_info_nr;
+	struct rsnd_dvc_platform_info *dvc_info;
+	int dvc_info_nr;
+	struct rsnd_dai_platform_info *dai_info;
+	int dai_info_nr;
+	int (*start)(int id);
+	int (*stop)(int id);
+};
+
+#endif
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 7a0e52b4640a..085329878525 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -21,10 +21,11 @@
 #include <linux/of_irq.h>
 #include <linux/sh_dma.h>
 #include <linux/workqueue.h>
-#include <sound/rcar_snd.h>
 #include <sound/soc.h>
 #include <sound/pcm_params.h>
 
+#include "rcar_snd.h"
+
 /*
  *	pseudo register
  *
@@ -214,6 +215,7 @@ struct rsnd_dma {
 };
 #define rsnd_dma_to_dmaen(dma)	(&(dma)->dma.en)
 #define rsnd_dma_to_dmapp(dma)	(&(dma)->dma.pp)
+#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma)
 
 void rsnd_dma_start(struct rsnd_dai_stream *io, struct rsnd_dma *dma);
 void rsnd_dma_stop(struct rsnd_dai_stream *io, struct rsnd_dma *dma);
@@ -225,8 +227,6 @@ int rsnd_dma_probe(struct platform_device *pdev,
 struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node,
 					  struct rsnd_mod *mod, char *name);
 
-#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma)
-
 /*
  *	R-Car sound mod
  */
@@ -330,8 +330,9 @@ struct rsnd_mod {
 #define rsnd_mod_to_priv(mod) ((mod)->priv)
 #define rsnd_mod_to_dma(mod) (&(mod)->dma)
 #define rsnd_mod_id(mod) ((mod) ? (mod)->id : -1)
-#define rsnd_mod_hw_start(mod)	clk_enable((mod)->clk)
-#define rsnd_mod_hw_stop(mod)	clk_disable((mod)->clk)
+#define rsnd_mod_power_on(mod)	clk_enable((mod)->clk)
+#define rsnd_mod_power_off(mod)	clk_disable((mod)->clk)
+#define rsnd_mod_get(ip)	(&(ip)->mod)
 
 int rsnd_mod_init(struct rsnd_priv *priv,
 		  struct rsnd_mod *mod,
@@ -571,9 +572,12 @@ int rsnd_ssi_probe(struct platform_device *pdev,
 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);
-int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod);
 int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod);
-int rsnd_ssi_use_busif(struct rsnd_dai_stream *io, struct rsnd_mod *mod);
+int rsnd_ssi_use_busif(struct rsnd_dai_stream *io);
+
+#define rsnd_ssi_is_pin_sharing(io)	\
+	__rsnd_ssi_is_pin_sharing(rsnd_io_to_mod_ssi(io))
+int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod);
 
 /*
  *	R-Car SRC
@@ -627,4 +631,15 @@ void rsnd_dvc_remove(struct platform_device *pdev,
 		     struct rsnd_priv *priv);
 struct rsnd_mod *rsnd_dvc_mod_get(struct rsnd_priv *priv, int id);
 
+#ifdef DEBUG
+void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type);
+#define rsnd_mod_confirm_ssi(mssi)	rsnd_mod_make_sure(mssi, RSND_MOD_SSI)
+#define rsnd_mod_confirm_src(msrc)	rsnd_mod_make_sure(msrc, RSND_MOD_SRC)
+#define rsnd_mod_confirm_dvc(mdvc)	rsnd_mod_make_sure(mdvc, RSND_MOD_DVC)
+#else
+#define rsnd_mod_confirm_ssi(mssi)
+#define rsnd_mod_confirm_src(msrc)
+#define rsnd_mod_confirm_dvc(mdvc)
+#endif
+
 #endif
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 89a18e102feb..261b50217c48 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -159,7 +159,7 @@ int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod,
 	/*
 	 * SSI_MODE1
 	 */
-	if (rsnd_ssi_is_pin_sharing(ssi_mod)) {
+	if (rsnd_ssi_is_pin_sharing(io)) {
 		int shift = -1;
 		switch (ssi_id) {
 		case 1:
@@ -352,7 +352,7 @@ static int rsnd_src_init(struct rsnd_mod *mod,
 {
 	struct rsnd_src *src = rsnd_mod_to_src(mod);
 
-	rsnd_mod_hw_start(mod);
+	rsnd_mod_power_on(mod);
 
 	rsnd_src_soft_reset(mod);
 
@@ -373,7 +373,7 @@ static int rsnd_src_quit(struct rsnd_mod *mod,
 	struct rsnd_src *src = rsnd_mod_to_src(mod);
 	struct device *dev = rsnd_priv_to_dev(priv);
 
-	rsnd_mod_hw_stop(mod);
+	rsnd_mod_power_off(mod);
 
 	if (src->err)
 		dev_warn(dev, "%s[%d] under/over flow err = %d\n",
@@ -918,11 +918,10 @@ static void rsnd_src_reconvert_update(struct rsnd_dai_stream *io,
 	rsnd_mod_write(mod, SRC_IFSVR, fsrate);
 }
 
-static int rsnd_src_pcm_new(struct rsnd_mod *mod,
+static int rsnd_src_pcm_new_gen2(struct rsnd_mod *mod,
 			    struct rsnd_dai_stream *io,
 			    struct snd_soc_pcm_runtime *rtd)
 {
-	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
 	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
 	struct rsnd_src *src = rsnd_mod_to_src(mod);
 	int ret;
@@ -932,12 +931,6 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod,
 	 */
 
 	/*
-	 * Gen1 is not supported
-	 */
-	if (rsnd_is_gen1(priv))
-		return 0;
-
-	/*
 	 * SRC sync convert needs clock master
 	 */
 	if (!rsnd_rdai_is_clk_master(rdai))
@@ -975,7 +968,7 @@ static struct rsnd_mod_ops rsnd_src_gen2_ops = {
 	.start	= rsnd_src_start_gen2,
 	.stop	= rsnd_src_stop_gen2,
 	.hw_params = rsnd_src_hw_params,
-	.pcm_new = rsnd_src_pcm_new,
+	.pcm_new = rsnd_src_pcm_new_gen2,
 };
 
 struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id)
@@ -983,7 +976,7 @@ struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id)
 	if (WARN_ON(id < 0 || id >= rsnd_src_nr(priv)))
 		id = 0;
 
-	return &((struct rsnd_src *)(priv->src) + id)->mod;
+	return rsnd_mod_get((struct rsnd_src *)(priv->src) + id);
 }
 
 static void rsnd_of_parse_src(struct platform_device *pdev,
@@ -1043,8 +1036,10 @@ int rsnd_src_probe(struct platform_device *pdev,
 	int i, nr, ret;
 
 	ops = NULL;
-	if (rsnd_is_gen1(priv))
+	if (rsnd_is_gen1(priv)) {
 		ops = &rsnd_src_gen1_ops;
+		dev_warn(dev, "Gen1 support will be removed soon\n");
+	}
 	if (rsnd_is_gen2(priv))
 		ops = &rsnd_src_gen2_ops;
 	if (!ops) {
@@ -1078,7 +1073,7 @@ int rsnd_src_probe(struct platform_device *pdev,
 
 		src->info = &info->src_info[i];
 
-		ret = rsnd_mod_init(priv, &src->mod, ops, clk, RSND_MOD_SRC, i);
+		ret = rsnd_mod_init(priv, rsnd_mod_get(src), ops, clk, RSND_MOD_SRC, i);
 		if (ret)
 			return ret;
 	}
@@ -1093,6 +1088,6 @@ void rsnd_src_remove(struct platform_device *pdev,
 	int i;
 
 	for_each_rsnd_src(src, priv, i) {
-		rsnd_mod_quit(&src->mod);
+		rsnd_mod_quit(rsnd_mod_get(src));
 	}
 }
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index d45b9a7e324e..1427ec21bd7e 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -79,7 +79,6 @@ struct rsnd_ssi {
 
 #define rsnd_ssi_nr(priv) ((priv)->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->irq > 0)
 #define rsnd_ssi_parent(ssi) ((ssi)->parent)
 #define rsnd_ssi_mode_flags(p) ((p)->info->flags)
@@ -87,8 +86,9 @@ struct rsnd_ssi {
 #define rsnd_ssi_of_node(priv) \
 	of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, "rcar_sound,ssi")
 
-int rsnd_ssi_use_busif(struct rsnd_dai_stream *io, struct rsnd_mod *mod)
+int rsnd_ssi_use_busif(struct rsnd_dai_stream *io)
 {
+	struct rsnd_mod *mod = rsnd_io_to_mod_ssi(io);
 	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
 	int use_busif = 0;
 
@@ -128,10 +128,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
 	struct rsnd_priv *priv = rsnd_io_to_priv(io);
 	struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
 	struct device *dev = rsnd_priv_to_dev(priv);
-	int i, j, ret;
-	int adg_clk_div_table[] = {
-		1, 6, /* see adg.c */
-	};
+	struct rsnd_mod *mod = rsnd_mod_get(ssi);
+	int j, ret;
 	int ssi_clk_mul_table[] = {
 		1, 2, 4, 8, 16, 6, 12,
 	};
@@ -141,28 +139,25 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
 	/*
 	 * 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_init()
-			 */
-			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->cr_clk	= FORCE | SWL_32 |
-						  SCKD | SWSD | CKDV(j);
-
-				dev_dbg(dev, "%s[%d] outputs %u Hz\n",
-					rsnd_mod_name(&ssi->mod),
-					rsnd_mod_id(&ssi->mod), rate);
-
-				return 0;
-			}
+	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_init()
+		 */
+		main_rate = rate * 32 * 2 * ssi_clk_mul_table[j];
+
+		ret = rsnd_adg_ssi_clk_try_start(mod, main_rate);
+		if (0 == ret) {
+			ssi->cr_clk	= FORCE | SWL_32 |
+				SCKD | SWSD | CKDV(j);
+
+			dev_dbg(dev, "%s[%d] outputs %u Hz\n",
+				rsnd_mod_name(mod),
+				rsnd_mod_id(mod), rate);
+
+			return 0;
 		}
 	}
 
@@ -172,8 +167,10 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
 
 static void rsnd_ssi_master_clk_stop(struct rsnd_ssi *ssi)
 {
+	struct rsnd_mod *mod = rsnd_mod_get(ssi);
+
 	ssi->cr_clk = 0;
-	rsnd_adg_ssi_clk_stop(&ssi->mod);
+	rsnd_adg_ssi_clk_stop(mod);
 }
 
 static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
@@ -182,11 +179,12 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
 	struct rsnd_priv *priv = rsnd_io_to_priv(io);
 	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
 	struct device *dev = rsnd_priv_to_dev(priv);
+	struct rsnd_mod *mod = rsnd_mod_get(ssi);
 	u32 cr_mode;
 	u32 cr;
 
 	if (0 == ssi->usrcnt) {
-		rsnd_mod_hw_start(&ssi->mod);
+		rsnd_mod_power_on(mod);
 
 		if (rsnd_rdai_is_clk_master(rdai)) {
 			struct rsnd_ssi *ssi_parent = rsnd_ssi_parent(ssi);
@@ -198,7 +196,7 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
 		}
 	}
 
-	if (rsnd_ssi_is_dma_mode(&ssi->mod)) {
+	if (rsnd_ssi_is_dma_mode(mod)) {
 		cr_mode = UIEN | OIEN |	/* over/under run */
 			  DMEN;		/* DMA : enable DMA */
 	} else {
@@ -210,24 +208,25 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
 		cr_mode		|
 		EN;
 
-	rsnd_mod_write(&ssi->mod, SSICR, cr);
+	rsnd_mod_write(mod, SSICR, cr);
 
 	/* enable WS continue */
 	if (rsnd_rdai_is_clk_master(rdai))
-		rsnd_mod_write(&ssi->mod, SSIWSR, CONT);
+		rsnd_mod_write(mod, SSIWSR, CONT);
 
 	/* clear error status */
-	rsnd_mod_write(&ssi->mod, SSISR, 0);
+	rsnd_mod_write(mod, SSISR, 0);
 
 	ssi->usrcnt++;
 
 	dev_dbg(dev, "%s[%d] hw started\n",
-		rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod));
+		rsnd_mod_name(mod), rsnd_mod_id(mod));
 }
 
 static void rsnd_ssi_hw_stop(struct rsnd_dai_stream *io, struct rsnd_ssi *ssi)
 {
-	struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod);
+	struct rsnd_mod *mod = rsnd_mod_get(ssi);
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
 	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
 	struct device *dev = rsnd_priv_to_dev(priv);
 	u32 cr;
@@ -247,15 +246,15 @@ static void rsnd_ssi_hw_stop(struct rsnd_dai_stream *io, struct rsnd_ssi *ssi)
 		cr  =	ssi->cr_own	|
 			ssi->cr_clk;
 
-		rsnd_mod_write(&ssi->mod, SSICR, cr | EN);
-		rsnd_ssi_status_check(&ssi->mod, DIRQ);
+		rsnd_mod_write(mod, SSICR, cr | EN);
+		rsnd_ssi_status_check(mod, DIRQ);
 
 		/*
 		 * disable SSI,
 		 * and, wait idle state
 		 */
-		rsnd_mod_write(&ssi->mod, SSICR, cr);	/* disabled all */
-		rsnd_ssi_status_check(&ssi->mod, IIRQ);
+		rsnd_mod_write(mod, SSICR, cr);	/* disabled all */
+		rsnd_ssi_status_check(mod, IIRQ);
 
 		if (rsnd_rdai_is_clk_master(rdai)) {
 			struct rsnd_ssi *ssi_parent = rsnd_ssi_parent(ssi);
@@ -266,13 +265,13 @@ static void rsnd_ssi_hw_stop(struct rsnd_dai_stream *io, struct rsnd_ssi *ssi)
 				rsnd_ssi_master_clk_stop(ssi);
 		}
 
-		rsnd_mod_hw_stop(&ssi->mod);
+		rsnd_mod_power_off(mod);
 
 		ssi->chan = 0;
 	}
 
 	dev_dbg(dev, "%s[%d] hw stopped\n",
-		rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod));
+		rsnd_mod_name(mod), rsnd_mod_id(mod));
 }
 
 /*
@@ -371,7 +370,7 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod,
 	/* It will be removed on rsnd_ssi_hw_stop */
 	ssi->chan = chan;
 	if (ssi_parent)
-		return rsnd_ssi_hw_params(&ssi_parent->mod, io,
+		return rsnd_ssi_hw_params(rsnd_mod_get(ssi_parent), io,
 					  substream, params);
 
 	return 0;
@@ -379,12 +378,14 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod,
 
 static void rsnd_ssi_record_error(struct rsnd_ssi *ssi, u32 status)
 {
+	struct rsnd_mod *mod = rsnd_mod_get(ssi);
+
 	/* under/over flow error */
 	if (status & (UIRQ | OIRQ)) {
 		ssi->err++;
 
 		/* clear error status */
-		rsnd_mod_write(&ssi->mod, SSISR, 0);
+		rsnd_mod_write(mod, SSISR, 0);
 	}
 }
 
@@ -394,7 +395,7 @@ static int rsnd_ssi_start(struct rsnd_mod *mod,
 {
 	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
 
-	rsnd_src_ssiu_start(mod, io, rsnd_ssi_use_busif(io, mod));
+	rsnd_src_ssiu_start(mod, io, rsnd_ssi_use_busif(io));
 
 	rsnd_ssi_hw_start(ssi, io);
 
@@ -554,7 +555,7 @@ static int rsnd_ssi_dma_remove(struct rsnd_mod *mod,
 	rsnd_dma_quit(io, rsnd_mod_to_dma(mod));
 
 	/* PIO will request IRQ again */
-	devm_free_irq(dev, irq, ssi);
+	devm_free_irq(dev, irq, mod);
 
 	return 0;
 }
@@ -613,7 +614,7 @@ static struct dma_chan *rsnd_ssi_dma_req(struct rsnd_dai_stream *io,
 	int is_play = rsnd_io_is_play(io);
 	char *name;
 
-	if (rsnd_ssi_use_busif(io, mod))
+	if (rsnd_ssi_use_busif(io))
 		name = is_play ? "rxu" : "txu";
 	else
 		name = is_play ? "rx" : "tx";
@@ -656,10 +657,10 @@ struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id)
 	if (WARN_ON(id < 0 || id >= rsnd_ssi_nr(priv)))
 		id = 0;
 
-	return &((struct rsnd_ssi *)(priv->ssi) + id)->mod;
+	return rsnd_mod_get((struct rsnd_ssi *)(priv->ssi) + id);
 }
 
-int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod)
+int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod)
 {
 	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
 
@@ -668,10 +669,12 @@ int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod)
 
 static void rsnd_ssi_parent_setup(struct rsnd_priv *priv, struct rsnd_ssi *ssi)
 {
-	if (!rsnd_ssi_is_pin_sharing(&ssi->mod))
+	struct rsnd_mod *mod = rsnd_mod_get(ssi);
+
+	if (!__rsnd_ssi_is_pin_sharing(mod))
 		return;
 
-	switch (rsnd_mod_id(&ssi->mod)) {
+	switch (rsnd_mod_id(mod)) {
 	case 1:
 	case 2:
 		ssi->parent = rsnd_mod_to_ssi(rsnd_ssi_mod_get(priv, 0));
@@ -697,9 +700,6 @@ static void rsnd_of_parse_ssi(struct platform_device *pdev,
 	struct device *dev = &pdev->dev;
 	int nr, i;
 
-	if (!of_data)
-		return;
-
 	node = rsnd_ssi_of_node(priv);
 	if (!node)
 		return;
@@ -794,7 +794,8 @@ int rsnd_ssi_probe(struct platform_device *pdev,
 		else if (rsnd_ssi_pio_available(ssi))
 			ops = &rsnd_ssi_pio_ops;
 
-		ret = rsnd_mod_init(priv, &ssi->mod, ops, clk, RSND_MOD_SSI, i);
+		ret = rsnd_mod_init(priv, rsnd_mod_get(ssi), ops, clk,
+				    RSND_MOD_SSI, i);
 		if (ret)
 			return ret;
 
@@ -811,6 +812,6 @@ void rsnd_ssi_remove(struct platform_device *pdev,
 	int i;
 
 	for_each_rsnd_ssi(ssi, priv, i) {
-		rsnd_mod_quit(&ssi->mod);
+		rsnd_mod_quit(rsnd_mod_get(ssi));
 	}
 }
diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c
index abb0d956231c..76b2ab8c2b4a 100644
--- a/sound/soc/sh/siu_dai.c
+++ b/sound/soc/sh/siu_dai.c
@@ -738,7 +738,7 @@ static int siu_probe(struct platform_device *pdev)
 	struct siu_info *info;
 	int ret;
 
-	info = kmalloc(sizeof(*info), GFP_KERNEL);
+	info = devm_kmalloc(&pdev->dev, sizeof(*info), GFP_KERNEL);
 	if (!info)
 		return -ENOMEM;
 	siu_i2s_data = info;
@@ -746,7 +746,7 @@ static int siu_probe(struct platform_device *pdev)
 
 	ret = request_firmware(&fw_entry, "siu_spb.bin", &pdev->dev);
 	if (ret)
-		goto ereqfw;
+		return ret;
 
 	/*
 	 * Loaded firmware is "const" - read only, but we have to modify it in
@@ -757,89 +757,52 @@ static int siu_probe(struct platform_device *pdev)
 	release_firmware(fw_entry);
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!res) {
-		ret = -ENODEV;
-		goto egetres;
-	}
+	if (!res)
+		return -ENODEV;
 
-	region = request_mem_region(res->start, resource_size(res),
-				    pdev->name);
+	region = devm_request_mem_region(&pdev->dev, res->start,
+					 resource_size(res), pdev->name);
 	if (!region) {
 		dev_err(&pdev->dev, "SIU region already claimed\n");
-		ret = -EBUSY;
-		goto ereqmemreg;
+		return -EBUSY;
 	}
 
-	ret = -ENOMEM;
-	info->pram = ioremap(res->start, PRAM_SIZE);
+	info->pram = devm_ioremap(&pdev->dev, res->start, PRAM_SIZE);
 	if (!info->pram)
-		goto emappram;
-	info->xram = ioremap(res->start + XRAM_OFFSET, XRAM_SIZE);
+		return -ENOMEM;
+	info->xram = devm_ioremap(&pdev->dev, res->start + XRAM_OFFSET,
+				  XRAM_SIZE);
 	if (!info->xram)
-		goto emapxram;
-	info->yram = ioremap(res->start + YRAM_OFFSET, YRAM_SIZE);
+		return -ENOMEM;
+	info->yram = devm_ioremap(&pdev->dev, res->start + YRAM_OFFSET,
+				  YRAM_SIZE);
 	if (!info->yram)
-		goto emapyram;
-	info->reg = ioremap(res->start + REG_OFFSET, resource_size(res) -
-			    REG_OFFSET);
+		return -ENOMEM;
+	info->reg = devm_ioremap(&pdev->dev, res->start + REG_OFFSET,
+			    resource_size(res) - REG_OFFSET);
 	if (!info->reg)
-		goto emapreg;
+		return -ENOMEM;
 
 	dev_set_drvdata(&pdev->dev, info);
 
 	/* register using ARRAY version so we can keep dai name */
-	ret = snd_soc_register_component(&pdev->dev, &siu_i2s_component,
-					 &siu_i2s_dai, 1);
+	ret = devm_snd_soc_register_component(&pdev->dev, &siu_i2s_component,
+					      &siu_i2s_dai, 1);
 	if (ret < 0)
-		goto edaiinit;
+		return ret;
 
-	ret = snd_soc_register_platform(&pdev->dev, &siu_platform);
+	ret = devm_snd_soc_register_platform(&pdev->dev, &siu_platform);
 	if (ret < 0)
-		goto esocregp;
+		return ret;
 
 	pm_runtime_enable(&pdev->dev);
 
-	return ret;
-
-esocregp:
-	snd_soc_unregister_component(&pdev->dev);
-edaiinit:
-	iounmap(info->reg);
-emapreg:
-	iounmap(info->yram);
-emapyram:
-	iounmap(info->xram);
-emapxram:
-	iounmap(info->pram);
-emappram:
-	release_mem_region(res->start, resource_size(res));
-ereqmemreg:
-egetres:
-ereqfw:
-	kfree(info);
-
-	return ret;
+	return 0;
 }
 
 static int siu_remove(struct platform_device *pdev)
 {
-	struct siu_info *info = dev_get_drvdata(&pdev->dev);
-	struct resource *res;
-
 	pm_runtime_disable(&pdev->dev);
-
-	snd_soc_unregister_platform(&pdev->dev);
-	snd_soc_unregister_component(&pdev->dev);
-
-	iounmap(info->reg);
-	iounmap(info->yram);
-	iounmap(info->xram);
-	iounmap(info->pram);
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (res)
-		release_mem_region(res->start, resource_size(res));
-	kfree(info);
-
 	return 0;
 }
 
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 025c38fbe3c0..12a9820feac1 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -612,8 +612,15 @@ static struct snd_compr_ops soc_compr_dyn_ops = {
 	.get_codec_caps = soc_compr_get_codec_caps
 };
 
-/* create a new compress */
-int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
+/**
+ * snd_soc_new_compress - create a new compress.
+ *
+ * @rtd: The runtime for which we will create compress
+ * @num: the device index number (zero based - shared with normal PCMs)
+ *
+ * Return: 0 for success, else error.
+ */
+int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_platform *platform = rtd->platform;
@@ -703,3 +710,4 @@ compr_err:
 	kfree(compr);
 	return ret;
 }
+EXPORT_SYMBOL_GPL(snd_soc_new_compress);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6173d15236c3..24b096066a07 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1370,9 +1370,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order)
 		soc_dpcm_debugfs_add(rtd);
 #endif
 
-	if (cpu_dai->driver->compress_dai) {
+	if (cpu_dai->driver->compress_new) {
 		/*create compress_device"*/
-		ret = soc_new_compress(rtd, num);
+		ret = cpu_dai->driver->compress_new(rtd, num);
 		if (ret < 0) {
 			dev_err(card->dev, "ASoC: can't create compress %s\n",
 					 dai_link->stream_name);
@@ -3291,13 +3291,38 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
 }
 EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets);
 
+static int snd_soc_of_get_slot_mask(struct device_node *np,
+				    const char *prop_name,
+				    unsigned int *mask)
+{
+	u32 val;
+	const __be32 *of_slot_mask = of_get_property(np, prop_name, &val);
+	int i;
+
+	if (!of_slot_mask)
+		return 0;
+	val /= sizeof(u32);
+	for (i = 0; i < val; i++)
+		if (be32_to_cpup(&of_slot_mask[i]))
+			*mask |= (1 << i);
+
+	return val;
+}
+
 int snd_soc_of_parse_tdm_slot(struct device_node *np,
+			      unsigned int *tx_mask,
+			      unsigned int *rx_mask,
 			      unsigned int *slots,
 			      unsigned int *slot_width)
 {
 	u32 val;
 	int ret;
 
+	if (tx_mask)
+		snd_soc_of_get_slot_mask(np, "dai-tdm-slot-tx-mask", tx_mask);
+	if (rx_mask)
+		snd_soc_of_get_slot_mask(np, "dai-tdm-slot-rx-mask", rx_mask);
+
 	if (of_property_read_bool(np, "dai-tdm-slot-num")) {
 		ret = of_property_read_u32(np, "dai-tdm-slot-num", &val);
 		if (ret)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index ff8bda471b25..016eba10b1ec 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -509,6 +509,18 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol,
 }
 
 /**
+ * snd_soc_dapm_kcontrol_widget() - Returns the widget associated to a
+ *   kcontrol
+ * @kcontrol: The kcontrol
+ */
+struct snd_soc_dapm_widget *snd_soc_dapm_kcontrol_widget(
+				struct snd_kcontrol *kcontrol)
+{
+	return dapm_kcontrol_get_wlist(kcontrol)->widgets[0];
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_widget);
+
+/**
  * snd_soc_dapm_kcontrol_dapm() - Returns the dapm context associated to a
  *  kcontrol
  * @kcontrol: The kcontrol
@@ -779,7 +791,7 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm,
  * 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,
+static int dapm_create_or_share_kcontrol(struct snd_soc_dapm_widget *w,
 	int kci)
 {
 	struct snd_soc_dapm_context *dapm = w->dapm;
@@ -810,6 +822,7 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w,
 			switch (w->id) {
 			case snd_soc_dapm_switch:
 			case snd_soc_dapm_mixer:
+			case snd_soc_dapm_pga:
 				wname_in_long_name = true;
 				kcname_in_long_name = true;
 				break;
@@ -899,7 +912,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w)
 				continue;
 
 			if (!w->kcontrols[i]) {
-				ret = dapm_create_or_share_mixmux_kcontrol(w, i);
+				ret = dapm_create_or_share_kcontrol(w, i);
 				if (ret < 0)
 					return ret;
 			}
@@ -952,7 +965,7 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w)
 		return -EINVAL;
 	}
 
-	ret = dapm_create_or_share_mixmux_kcontrol(w, 0);
+	ret = dapm_create_or_share_kcontrol(w, 0);
 	if (ret < 0)
 		return ret;
 
@@ -967,9 +980,13 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w)
 /* create new dapm volume control */
 static int dapm_new_pga(struct snd_soc_dapm_widget *w)
 {
-	if (w->num_kcontrols)
-		dev_err(w->dapm->dev,
-			"ASoC: PGA controls not supported: '%s'\n", w->name);
+	int i, ret;
+
+	for (i = 0; i < w->num_kcontrols; i++) {
+		ret = dapm_create_or_share_kcontrol(w, i);
+		if (ret < 0)
+			return ret;
+	}
 
 	return 0;
 }
@@ -3473,11 +3490,29 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
 		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+		if (source->driver->ops && source->driver->ops->startup) {
+			ret = source->driver->ops->startup(&substream, source);
+			if (ret < 0) {
+				dev_err(source->dev,
+					"ASoC: startup() failed: %d\n", ret);
+				goto out;
+			}
+			source->active++;
+		}
 		ret = soc_dai_hw_params(&substream, params, source);
 		if (ret < 0)
 			goto out;
 
 		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+		if (sink->driver->ops && sink->driver->ops->startup) {
+			ret = sink->driver->ops->startup(&substream, sink);
+			if (ret < 0) {
+				dev_err(sink->dev,
+					"ASoC: startup() failed: %d\n", ret);
+				goto out;
+			}
+			sink->active++;
+		}
 		ret = soc_dai_hw_params(&substream, params, sink);
 		if (ret < 0)
 			goto out;
@@ -3497,6 +3532,18 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 		if (ret != 0 && ret != -ENOTSUPP)
 			dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret);
 		ret = 0;
+
+		source->active--;
+		if (source->driver->ops && source->driver->ops->shutdown) {
+			substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+			source->driver->ops->shutdown(&substream, source);
+		}
+
+		sink->active--;
+		if (sink->driver->ops && sink->driver->ops->shutdown) {
+			substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+			sink->driver->ops->shutdown(&substream, sink);
+		}
 		break;
 
 	default:
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 05977ae1ff2a..ecd38e52285a 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -588,16 +588,16 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range);
 /**
  * snd_soc_limit_volume - Set new limit to an existing volume control.
  *
- * @codec: where to look for the control
+ * @card: where to look for the control
  * @name: Name of the control
  * @max: new maximum limit
  *
  * Return 0 for success, else error.
  */
-int snd_soc_limit_volume(struct snd_soc_codec *codec,
+int snd_soc_limit_volume(struct snd_soc_card *card,
 	const char *name, int max)
 {
-	struct snd_card *card = codec->component.card->snd_card;
+	struct snd_card *snd_card = card->snd_card;
 	struct snd_kcontrol *kctl;
 	struct soc_mixer_control *mc;
 	int found = 0;
@@ -607,7 +607,7 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec,
 	if (unlikely(!name || max <= 0))
 		return -EINVAL;
 
-	list_for_each_entry(kctl, &card->controls, list) {
+	list_for_each_entry(kctl, &snd_card->controls, list) {
 		if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) {
 			found = 1;
 			break;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 70e4b9d8bdcd..c86dc96e8986 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -34,6 +34,24 @@
 
 #define DPCM_MAX_BE_USERS	8
 
+/*
+ * snd_soc_dai_stream_valid() - check if a DAI supports the given stream
+ *
+ * Returns true if the DAI supports the indicated stream type.
+ */
+static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream)
+{
+	struct snd_soc_pcm_stream *codec_stream;
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+		codec_stream = &dai->driver->playback;
+	else
+		codec_stream = &dai->driver->capture;
+
+	/* If the codec specifies any rate at all, it supports the stream. */
+	return codec_stream->rates;
+}
+
 /**
  * snd_soc_runtime_activate() - Increment active count for PCM runtime components
  * @rtd: ASoC PCM runtime that is activated
@@ -182,9 +200,9 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream,
 		dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %dHz rate\n",
 				soc_dai->rate);
 
-		ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+		ret = snd_pcm_hw_constraint_single(substream->runtime,
 						SNDRV_PCM_HW_PARAM_RATE,
-						soc_dai->rate, soc_dai->rate);
+						soc_dai->rate);
 		if (ret < 0) {
 			dev_err(soc_dai->dev,
 				"ASoC: Unable to apply rate constraint: %d\n",
@@ -198,9 +216,8 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream,
 		dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %d channel(s)\n",
 				soc_dai->channels);
 
-		ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+		ret = snd_pcm_hw_constraint_single(substream->runtime,
 						SNDRV_PCM_HW_PARAM_CHANNELS,
-						soc_dai->channels,
 						soc_dai->channels);
 		if (ret < 0) {
 			dev_err(soc_dai->dev,
@@ -215,9 +232,8 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream,
 		dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %d sample bits\n",
 				soc_dai->sample_bits);
 
-		ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+		ret = snd_pcm_hw_constraint_single(substream->runtime,
 						SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-						soc_dai->sample_bits,
 						soc_dai->sample_bits);
 		if (ret < 0) {
 			dev_err(soc_dai->dev,
@@ -371,6 +387,20 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 
 	/* first calculate min/max only for CODECs in the DAI link */
 	for (i = 0; i < rtd->num_codecs; i++) {
+
+		/*
+		 * Skip CODECs which don't support the current stream type.
+		 * Otherwise, since the rate, channel, and format values will
+		 * zero in that case, we would have no usable settings left,
+		 * causing the resulting setup to fail.
+		 * At least one CODEC should match, otherwise we should have
+		 * bailed out on a higher level, since there would be no
+		 * CODEC to support the transfer direction in that case.
+		 */
+		if (!snd_soc_dai_stream_valid(rtd->codec_dais[i],
+					      substream->stream))
+			continue;
+
 		codec_dai_drv = rtd->codec_dais[i]->driver;
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			codec_stream = &codec_dai_drv->playback;
@@ -827,6 +857,23 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
 		struct snd_pcm_hw_params codec_params;
 
+		/*
+		 * Skip CODECs which don't support the current stream type,
+		 * the idea being that if a CODEC is not used for the currently
+		 * set up transfer direction, it should not need to be
+		 * configured, especially since the configuration used might
+		 * not even be supported by that CODEC. There may be cases
+		 * however where a CODEC needs to be set up although it is
+		 * actually not being used for the transfer, e.g. if a
+		 * capture-only CODEC is acting as an LRCLK and/or BCLK master
+		 * for the DAI link including a playback-only CODEC.
+		 * If this becomes necessary, we will have to augment the
+		 * machine driver setup with information on how to act, so
+		 * we can do the right thing here.
+		 */
+		if (!snd_soc_dai_stream_valid(codec_dai, substream->stream))
+			continue;
+
 		/* copy params for each codec */
 		codec_params = *params;
 
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 69d01cd925ce..8d7ec80af51b 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1558,7 +1558,7 @@ static int soc_tplg_pcm_dai_elems_load(struct soc_tplg *tplg,
 	pcm_dai = (struct snd_soc_tplg_pcm_dai *)tplg->pos;
 
 	if (soc_tplg_check_elem_count(tplg,
-		sizeof(struct snd_soc_tplg_pcm_dai), count,
+		sizeof(struct snd_soc_tplg_pcm), count,
 		hdr->payload_size, "PCM DAI")) {
 		dev_err(tplg->dev, "ASoC: invalid count %d for PCM DAI elems\n",
 			count);
@@ -1566,7 +1566,7 @@ static int soc_tplg_pcm_dai_elems_load(struct soc_tplg *tplg,
 	}
 
 	dev_dbg(tplg->dev, "ASoC: adding %d PCM DAIs\n", count);
-	tplg->pos += sizeof(struct snd_soc_tplg_pcm_dai) * count;
+	tplg->pos += sizeof(struct snd_soc_tplg_pcm) * count;
 
 	dobj = kzalloc(sizeof(struct snd_soc_dobj), GFP_KERNEL);
 	if (dobj == NULL)
diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig
new file mode 100644
index 000000000000..84c72ec6ad73
--- /dev/null
+++ b/sound/soc/sunxi/Kconfig
@@ -0,0 +1,11 @@
+menu "Allwinner SoC Audio support"
+
+config SND_SUN4I_CODEC
+	tristate "Allwinner A10 Codec Support"
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+	select REGMAP_MMIO
+	help
+	  Select Y or M to add support for the Codec embedded in the Allwinner
+	  A10 and affiliated SoCs.
+
+endmenu
diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile
new file mode 100644
index 000000000000..ea8a08c881d6
--- /dev/null
+++ b/sound/soc/sunxi/Makefile
@@ -0,0 +1,2 @@
+obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o
+
diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
new file mode 100644
index 000000000000..bcbf4da168b6
--- /dev/null
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -0,0 +1,712 @@
+/*
+ * Copyright 2014 Emilio López <emilio@elopez.com.ar>
+ * Copyright 2014 Jon Smirl <jonsmirl@gmail.com>
+ * Copyright 2015 Maxime Ripard <maxime.ripard@free-electrons.com>
+ *
+ * Based on the Allwinner SDK driver, released under the GPL.
+ *
+ * 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/init.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/of_address.h>
+#include <linux/clk.h>
+#include <linux/regmap.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include <sound/dmaengine_pcm.h>
+
+/* Codec DAC register offsets and bit fields */
+#define SUN4I_CODEC_DAC_DPC			(0x00)
+#define SUN4I_CODEC_DAC_DPC_EN_DA			(31)
+#define SUN4I_CODEC_DAC_DPC_DVOL			(12)
+#define SUN4I_CODEC_DAC_FIFOC			(0x04)
+#define SUN4I_CODEC_DAC_FIFOC_DAC_FS			(29)
+#define SUN4I_CODEC_DAC_FIFOC_FIR_VERSION		(28)
+#define SUN4I_CODEC_DAC_FIFOC_SEND_LASAT		(26)
+#define SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE		(24)
+#define SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT		(21)
+#define SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL		(8)
+#define SUN4I_CODEC_DAC_FIFOC_MONO_EN			(6)
+#define SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS		(5)
+#define SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN		(4)
+#define SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH		(0)
+#define SUN4I_CODEC_DAC_FIFOS			(0x08)
+#define SUN4I_CODEC_DAC_TXDATA			(0x0c)
+#define SUN4I_CODEC_DAC_ACTL			(0x10)
+#define SUN4I_CODEC_DAC_ACTL_DACAENR			(31)
+#define SUN4I_CODEC_DAC_ACTL_DACAENL			(30)
+#define SUN4I_CODEC_DAC_ACTL_MIXEN			(29)
+#define SUN4I_CODEC_DAC_ACTL_LDACLMIXS			(15)
+#define SUN4I_CODEC_DAC_ACTL_RDACRMIXS			(14)
+#define SUN4I_CODEC_DAC_ACTL_LDACRMIXS			(13)
+#define SUN4I_CODEC_DAC_ACTL_DACPAS			(8)
+#define SUN4I_CODEC_DAC_ACTL_MIXPAS			(7)
+#define SUN4I_CODEC_DAC_ACTL_PA_MUTE			(6)
+#define SUN4I_CODEC_DAC_ACTL_PA_VOL			(0)
+#define SUN4I_CODEC_DAC_TUNE			(0x14)
+#define SUN4I_CODEC_DAC_DEBUG			(0x18)
+
+/* Codec ADC register offsets and bit fields */
+#define SUN4I_CODEC_ADC_FIFOC			(0x1c)
+#define SUN4I_CODEC_ADC_FIFOC_EN_AD			(28)
+#define SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE		(24)
+#define SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL		(8)
+#define SUN4I_CODEC_ADC_FIFOC_MONO_EN			(7)
+#define SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS		(6)
+#define SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN		(4)
+#define SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH		(0)
+#define SUN4I_CODEC_ADC_FIFOS			(0x20)
+#define SUN4I_CODEC_ADC_RXDATA			(0x24)
+#define SUN4I_CODEC_ADC_ACTL			(0x28)
+#define SUN4I_CODEC_ADC_ACTL_ADC_R_EN			(31)
+#define SUN4I_CODEC_ADC_ACTL_ADC_L_EN			(30)
+#define SUN4I_CODEC_ADC_ACTL_PREG1EN			(29)
+#define SUN4I_CODEC_ADC_ACTL_PREG2EN			(28)
+#define SUN4I_CODEC_ADC_ACTL_VMICEN			(27)
+#define SUN4I_CODEC_ADC_ACTL_VADCG			(20)
+#define SUN4I_CODEC_ADC_ACTL_ADCIS			(17)
+#define SUN4I_CODEC_ADC_ACTL_PA_EN			(4)
+#define SUN4I_CODEC_ADC_ACTL_DDE			(3)
+#define SUN4I_CODEC_ADC_DEBUG			(0x2c)
+
+/* Other various ADC registers */
+#define SUN4I_CODEC_DAC_TXCNT			(0x30)
+#define SUN4I_CODEC_ADC_RXCNT			(0x34)
+#define SUN4I_CODEC_AC_SYS_VERI			(0x38)
+#define SUN4I_CODEC_AC_MIC_PHONE_CAL		(0x3c)
+
+struct sun4i_codec {
+	struct device	*dev;
+	struct regmap	*regmap;
+	struct clk	*clk_apb;
+	struct clk	*clk_module;
+
+	struct snd_dmaengine_dai_dma_data	playback_dma_data;
+};
+
+static void sun4i_codec_start_playback(struct sun4i_codec *scodec)
+{
+	/*
+	 * FIXME: according to the BSP, we might need to drive a PA
+	 *        GPIO high here on some boards
+	 */
+
+	/* Flush TX FIFO */
+	regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+			   BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH),
+			   BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH));
+
+	/* Enable DAC DRQ */
+	regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+			   BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN),
+			   BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN));
+}
+
+static void sun4i_codec_stop_playback(struct sun4i_codec *scodec)
+{
+	/*
+	 * FIXME: according to the BSP, we might need to drive a PA
+	 *        GPIO low here on some boards
+	 */
+
+	/* Disable DAC DRQ */
+	regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+			   BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN),
+			   0);
+}
+
+static int sun4i_codec_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		return -ENOTSUPP;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		sun4i_codec_start_playback(scodec);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		sun4i_codec_stop_playback(scodec);
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int sun4i_codec_prepare(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+	u32 val;
+
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		return -ENOTSUPP;
+
+	/* Flush the TX FIFO */
+	regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+			   BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH),
+			   BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH));
+
+	/* Set TX FIFO Empty Trigger Level */
+	regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+			   0x3f << SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL,
+			   0xf << SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL);
+
+	if (substream->runtime->rate > 32000)
+		/* Use 64 bits FIR filter */
+		val = 0;
+	else
+		/* Use 32 bits FIR filter */
+		val = BIT(SUN4I_CODEC_DAC_FIFOC_FIR_VERSION);
+
+	regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+			   BIT(SUN4I_CODEC_DAC_FIFOC_FIR_VERSION),
+			   val);
+
+	/* Send zeros when we have an underrun */
+	regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+			   BIT(SUN4I_CODEC_DAC_FIFOC_SEND_LASAT),
+			   0);
+
+	return 0;
+}
+
+static unsigned long sun4i_codec_get_mod_freq(struct snd_pcm_hw_params *params)
+{
+	unsigned int rate = params_rate(params);
+
+	switch (rate) {
+	case 176400:
+	case 88200:
+	case 44100:
+	case 33075:
+	case 22050:
+	case 14700:
+	case 11025:
+	case 7350:
+		return 22579200;
+
+	case 192000:
+	case 96000:
+	case 48000:
+	case 32000:
+	case 24000:
+	case 16000:
+	case 12000:
+	case 8000:
+		return 24576000;
+
+	default:
+		return 0;
+	}
+}
+
+static int sun4i_codec_get_hw_rate(struct snd_pcm_hw_params *params)
+{
+	unsigned int rate = params_rate(params);
+
+	switch (rate) {
+	case 192000:
+	case 176400:
+		return 6;
+
+	case 96000:
+	case 88200:
+		return 7;
+
+	case 48000:
+	case 44100:
+		return 0;
+
+	case 32000:
+	case 33075:
+		return 1;
+
+	case 24000:
+	case 22050:
+		return 2;
+
+	case 16000:
+	case 14700:
+		return 3;
+
+	case 12000:
+	case 11025:
+		return 4;
+
+	case 8000:
+	case 7350:
+		return 5;
+
+	default:
+		return -EINVAL;
+	}
+}
+
+static int sun4i_codec_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 sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+	unsigned long clk_freq;
+	int ret, hwrate;
+	u32 val;
+
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		return -ENOTSUPP;
+
+	clk_freq = sun4i_codec_get_mod_freq(params);
+	if (!clk_freq)
+		return -EINVAL;
+
+	ret = clk_set_rate(scodec->clk_module, clk_freq);
+	if (ret)
+		return ret;
+
+	hwrate = sun4i_codec_get_hw_rate(params);
+	if (hwrate < 0)
+		return hwrate;
+
+	/* Set DAC sample rate */
+	regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+			   7 << SUN4I_CODEC_DAC_FIFOC_DAC_FS,
+			   hwrate << SUN4I_CODEC_DAC_FIFOC_DAC_FS);
+
+	/* Set the number of channels we want to use */
+	if (params_channels(params) == 1)
+		val = BIT(SUN4I_CODEC_DAC_FIFOC_MONO_EN);
+	else
+		val = 0;
+
+	regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+			   BIT(SUN4I_CODEC_DAC_FIFOC_MONO_EN),
+			   val);
+
+	/* Set the number of sample bits to either 16 or 24 bits */
+	if (hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min == 32) {
+		regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+				   BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS),
+				   BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS));
+
+		/* Set TX FIFO mode to padding the LSBs with 0 */
+		regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+				   BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE),
+				   0);
+
+		scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+	} else {
+		regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+				   BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS),
+				   0);
+
+		/* Set TX FIFO mode to repeat the MSB */
+		regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+				   BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE),
+				   BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE));
+
+		scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+	}
+
+	return 0;
+}
+
+static int sun4i_codec_startup(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+
+	/*
+	 * Stop issuing DRQ when we have room for less than 16 samples
+	 * in our TX FIFO
+	 */
+	regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC,
+			   3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT,
+			   3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT);
+
+	return clk_prepare_enable(scodec->clk_module);
+}
+
+static void sun4i_codec_shutdown(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
+
+	clk_disable_unprepare(scodec->clk_module);
+}
+
+static const struct snd_soc_dai_ops sun4i_codec_dai_ops = {
+	.startup	= sun4i_codec_startup,
+	.shutdown	= sun4i_codec_shutdown,
+	.trigger	= sun4i_codec_trigger,
+	.hw_params	= sun4i_codec_hw_params,
+	.prepare	= sun4i_codec_prepare,
+};
+
+static struct snd_soc_dai_driver sun4i_codec_dai = {
+	.name	= "Codec",
+	.ops	= &sun4i_codec_dai_ops,
+	.playback = {
+		.stream_name	= "Codec Playback",
+		.channels_min	= 1,
+		.channels_max	= 2,
+		.rate_min	= 8000,
+		.rate_max	= 192000,
+		.rates		= SNDRV_PCM_RATE_8000_48000 |
+				  SNDRV_PCM_RATE_96000 |
+				  SNDRV_PCM_RATE_192000,
+		.formats	= SNDRV_PCM_FMTBIT_S16_LE |
+				  SNDRV_PCM_FMTBIT_S32_LE,
+		.sig_bits	= 24,
+	},
+};
+
+/*** Codec ***/
+static const struct snd_kcontrol_new sun4i_codec_pa_mute =
+	SOC_DAPM_SINGLE("Switch", SUN4I_CODEC_DAC_ACTL,
+			SUN4I_CODEC_DAC_ACTL_PA_MUTE, 1, 0);
+
+static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1);
+
+static const struct snd_kcontrol_new sun4i_codec_widgets[] = {
+	SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL,
+		       SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0,
+		       sun4i_codec_pa_volume_scale),
+};
+
+static const struct snd_kcontrol_new sun4i_codec_left_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+			SUN4I_CODEC_DAC_ACTL_LDACLMIXS, 1, 0),
+};
+
+static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Right DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+			SUN4I_CODEC_DAC_ACTL_RDACRMIXS, 1, 0),
+	SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+			SUN4I_CODEC_DAC_ACTL_LDACRMIXS, 1, 0),
+};
+
+static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = {
+	SOC_DAPM_SINGLE("DAC Playback Switch", SUN4I_CODEC_DAC_ACTL,
+			SUN4I_CODEC_DAC_ACTL_DACPAS, 1, 0),
+	SOC_DAPM_SINGLE("Mixer Playback Switch", SUN4I_CODEC_DAC_ACTL,
+			SUN4I_CODEC_DAC_ACTL_MIXPAS, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = {
+	/* Digital parts of the DACs */
+	SND_SOC_DAPM_SUPPLY("DAC", SUN4I_CODEC_DAC_DPC,
+			    SUN4I_CODEC_DAC_DPC_EN_DA, 0,
+			    NULL, 0),
+
+	/* Analog parts of the DACs */
+	SND_SOC_DAPM_DAC("Left DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL,
+			 SUN4I_CODEC_DAC_ACTL_DACAENL, 0),
+	SND_SOC_DAPM_DAC("Right DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL,
+			 SUN4I_CODEC_DAC_ACTL_DACAENR, 0),
+
+	/* Mixers */
+	SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+			   sun4i_codec_left_mixer_controls,
+			   ARRAY_SIZE(sun4i_codec_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+			   sun4i_codec_right_mixer_controls,
+			   ARRAY_SIZE(sun4i_codec_right_mixer_controls)),
+
+	/* Global Mixer Enable */
+	SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL,
+			    SUN4I_CODEC_DAC_ACTL_MIXEN, 0, NULL, 0),
+
+	/* Pre-Amplifier */
+	SND_SOC_DAPM_MIXER("Pre-Amplifier", SUN4I_CODEC_ADC_ACTL,
+			   SUN4I_CODEC_ADC_ACTL_PA_EN, 0,
+			   sun4i_codec_pa_mixer_controls,
+			   ARRAY_SIZE(sun4i_codec_pa_mixer_controls)),
+	SND_SOC_DAPM_SWITCH("Pre-Amplifier Mute", SND_SOC_NOPM, 0, 0,
+			    &sun4i_codec_pa_mute),
+
+	SND_SOC_DAPM_OUTPUT("HP Right"),
+	SND_SOC_DAPM_OUTPUT("HP Left"),
+};
+
+static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = {
+	/* Left DAC Routes */
+	{ "Left DAC", NULL, "DAC" },
+
+	/* Right DAC Routes */
+	{ "Right DAC", NULL, "DAC" },
+
+	/* Right Mixer Routes */
+	{ "Right Mixer", NULL, "Mixer Enable" },
+	{ "Right Mixer", "Left DAC Playback Switch", "Left DAC" },
+	{ "Right Mixer", "Right DAC Playback Switch", "Right DAC" },
+
+	/* Left Mixer Routes */
+	{ "Left Mixer", NULL, "Mixer Enable" },
+	{ "Left Mixer", "Left DAC Playback Switch", "Left DAC" },
+
+	/* Pre-Amplifier Mixer Routes */
+	{ "Pre-Amplifier", "Mixer Playback Switch", "Left Mixer" },
+	{ "Pre-Amplifier", "Mixer Playback Switch", "Right Mixer" },
+	{ "Pre-Amplifier", "DAC Playback Switch", "Left DAC" },
+	{ "Pre-Amplifier", "DAC Playback Switch", "Right DAC" },
+
+	/* PA -> HP path */
+	{ "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" },
+	{ "HP Right", NULL, "Pre-Amplifier Mute" },
+	{ "HP Left", NULL, "Pre-Amplifier Mute" },
+};
+
+static struct snd_soc_codec_driver sun4i_codec_codec = {
+	.controls		= sun4i_codec_widgets,
+	.num_controls		= ARRAY_SIZE(sun4i_codec_widgets),
+	.dapm_widgets		= sun4i_codec_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(sun4i_codec_dapm_widgets),
+	.dapm_routes		= sun4i_codec_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(sun4i_codec_dapm_routes),
+};
+
+static const struct snd_soc_component_driver sun4i_codec_component = {
+	.name = "sun4i-codec",
+};
+
+#define SUN4I_CODEC_RATES	SNDRV_PCM_RATE_8000_192000
+#define SUN4I_CODEC_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | \
+				 SNDRV_PCM_FMTBIT_S32_LE)
+
+static int sun4i_codec_dai_probe(struct snd_soc_dai *dai)
+{
+	struct snd_soc_card *card = snd_soc_dai_get_drvdata(dai);
+	struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card);
+
+	snd_soc_dai_init_dma_data(dai, &scodec->playback_dma_data,
+				  NULL);
+
+	return 0;
+}
+
+static struct snd_soc_dai_driver dummy_cpu_dai = {
+	.name	= "sun4i-codec-cpu-dai",
+	.probe	= sun4i_codec_dai_probe,
+	.playback = {
+		.stream_name	= "Playback",
+		.channels_min	= 1,
+		.channels_max	= 2,
+		.rates		= SUN4I_CODEC_RATES,
+		.formats	= SUN4I_CODEC_FORMATS,
+		.sig_bits	= 24,
+	},
+};
+
+static const struct regmap_config sun4i_codec_regmap_config = {
+	.reg_bits	= 32,
+	.reg_stride	= 4,
+	.val_bits	= 32,
+	.max_register	= SUN4I_CODEC_AC_MIC_PHONE_CAL,
+};
+
+static const struct of_device_id sun4i_codec_of_match[] = {
+	{ .compatible = "allwinner,sun4i-a10-codec" },
+	{ .compatible = "allwinner,sun7i-a20-codec" },
+	{}
+};
+MODULE_DEVICE_TABLE(of, sun4i_codec_of_match);
+
+static struct snd_soc_dai_link *sun4i_codec_create_link(struct device *dev,
+							int *num_links)
+{
+	struct snd_soc_dai_link *link = devm_kzalloc(dev, sizeof(*link),
+						     GFP_KERNEL);
+	if (!link)
+		return NULL;
+
+	link->name		= "cdc";
+	link->stream_name	= "CDC PCM";
+	link->codec_dai_name	= "Codec";
+	link->cpu_dai_name	= dev_name(dev);
+	link->codec_name	= dev_name(dev);
+	link->platform_name	= dev_name(dev);
+	link->dai_fmt		= SND_SOC_DAIFMT_I2S;
+
+	*num_links = 1;
+
+	return link;
+};
+
+static struct snd_soc_card *sun4i_codec_create_card(struct device *dev)
+{
+	struct snd_soc_card *card;
+
+	card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
+	if (!card)
+		return NULL;
+
+	card->dai_link = sun4i_codec_create_link(dev, &card->num_links);
+	if (!card->dai_link)
+		return NULL;
+
+	card->dev		= dev;
+	card->name		= "sun4i-codec";
+
+	return card;
+};
+
+static int sun4i_codec_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card;
+	struct sun4i_codec *scodec;
+	struct resource *res;
+	void __iomem *base;
+	int ret;
+
+	scodec = devm_kzalloc(&pdev->dev, sizeof(*scodec), GFP_KERNEL);
+	if (!scodec)
+		return -ENOMEM;
+
+	scodec->dev = &pdev->dev;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	base = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(base)) {
+		dev_err(&pdev->dev, "Failed to map the registers\n");
+		return PTR_ERR(base);
+	}
+
+	scodec->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+					     &sun4i_codec_regmap_config);
+	if (IS_ERR(scodec->regmap)) {
+		dev_err(&pdev->dev, "Failed to create our regmap\n");
+		return PTR_ERR(scodec->regmap);
+	}
+
+	/* Get the clocks from the DT */
+	scodec->clk_apb = devm_clk_get(&pdev->dev, "apb");
+	if (IS_ERR(scodec->clk_apb)) {
+		dev_err(&pdev->dev, "Failed to get the APB clock\n");
+		return PTR_ERR(scodec->clk_apb);
+	}
+
+	scodec->clk_module = devm_clk_get(&pdev->dev, "codec");
+	if (IS_ERR(scodec->clk_module)) {
+		dev_err(&pdev->dev, "Failed to get the module clock\n");
+		return PTR_ERR(scodec->clk_module);
+	}
+
+	/* Enable the bus clock */
+	if (clk_prepare_enable(scodec->clk_apb)) {
+		dev_err(&pdev->dev, "Failed to enable the APB clock\n");
+		return -EINVAL;
+	}
+
+	/* DMA configuration for TX FIFO */
+	scodec->playback_dma_data.addr = res->start + SUN4I_CODEC_DAC_TXDATA;
+	scodec->playback_dma_data.maxburst = 4;
+	scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+
+	ret = snd_soc_register_codec(&pdev->dev, &sun4i_codec_codec,
+				     &sun4i_codec_dai, 1);
+	if (ret) {
+		dev_err(&pdev->dev, "Failed to register our codec\n");
+		goto err_clk_disable;
+	}
+
+	ret = devm_snd_soc_register_component(&pdev->dev,
+					      &sun4i_codec_component,
+					      &dummy_cpu_dai, 1);
+	if (ret) {
+		dev_err(&pdev->dev, "Failed to register our DAI\n");
+		goto err_unregister_codec;
+	}
+
+	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+	if (ret) {
+		dev_err(&pdev->dev, "Failed to register against DMAEngine\n");
+		goto err_unregister_codec;
+	}
+
+	card = sun4i_codec_create_card(&pdev->dev);
+	if (!card) {
+		dev_err(&pdev->dev, "Failed to create our card\n");
+		goto err_unregister_codec;
+	}
+
+	platform_set_drvdata(pdev, card);
+	snd_soc_card_set_drvdata(card, scodec);
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "Failed to register our card\n");
+		goto err_unregister_codec;
+	}
+
+	return 0;
+
+err_unregister_codec:
+	snd_soc_unregister_codec(&pdev->dev);
+err_clk_disable:
+	clk_disable_unprepare(scodec->clk_apb);
+	return ret;
+}
+
+static int sun4i_codec_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card);
+
+	snd_soc_unregister_card(card);
+	snd_soc_unregister_codec(&pdev->dev);
+	clk_disable_unprepare(scodec->clk_apb);
+
+	return 0;
+}
+
+static struct platform_driver sun4i_codec_driver = {
+	.driver = {
+		.name = "sun4i-codec",
+		.of_match_table = sun4i_codec_of_match,
+	},
+	.probe = sun4i_codec_probe,
+	.remove = sun4i_codec_remove,
+};
+module_platform_driver(sun4i_codec_driver);
+
+MODULE_DESCRIPTION("Allwinner A10 codec driver");
+MODULE_AUTHOR("Emilio López <emilio@elopez.com.ar>");
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_AUTHOR("Maxime Ripard <maxime.ripard@free-electrons.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c
index 4e0c0e502ade..ba9fc099cf67 100644
--- a/sound/soc/ux500/mop500.c
+++ b/sound/soc/ux500/mop500.c
@@ -152,6 +152,7 @@ static const struct of_device_id snd_soc_mop500_match[] = {
 	{ .compatible = "stericsson,snd-soc-mop500", },
 	{},
 };
+MODULE_DEVICE_TABLE(of, snd_soc_mop500_match);
 
 static struct platform_driver snd_soc_mop500_driver = {
 	.driver = {
diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index f5df08ded770..6d5698b25bd4 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -522,9 +522,9 @@ static int ux500_msp_dai_hw_params(struct snd_pcm_substream *substream,
 		slots_active = hweight32(mask);
 		dev_dbg(dai->dev, "TDM-slots active: %d", slots_active);
 
-		snd_pcm_hw_constraint_minmax(runtime,
+		snd_pcm_hw_constraint_single(runtime,
 				SNDRV_PCM_HW_PARAM_CHANNELS,
-				slots_active, slots_active);
+				slots_active);
 		break;
 
 	default:
@@ -843,6 +843,7 @@ static const struct of_device_id ux500_msp_i2s_match[] = {
 	{ .compatible = "stericsson,ux500-msp-i2s", },
 	{},
 };
+MODULE_DEVICE_TABLE(of, ux500_msp_i2s_match);
 
 static struct platform_driver msp_i2s_driver = {
 	.driver = {
diff --git a/sound/usb/card.h b/sound/usb/card.h
index ef580b43f1e3..71778ca4b26a 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -122,6 +122,7 @@ struct snd_usb_substream {
 	unsigned int buffer_periods;	/* current periods per buffer */
 	unsigned int altset_idx;     /* USB data format: index of alternate setting */
 	unsigned int txfr_quirk:1;	/* allow sub-frame alignment */
+	unsigned int tx_length_quirk:1;	/* add length specifier to transfers */
 	unsigned int fmt_type;		/* USB audio format type (1-3) */
 	unsigned int pkt_offset_adj;	/* Bytes to drop from beginning of packets (for non-compliant devices) */
 
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index e6f71894ecdc..7b1cb365ffab 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -183,13 +183,53 @@ static void retire_inbound_urb(struct snd_usb_endpoint *ep,
 		ep->retire_data_urb(ep->data_subs, urb);
 }
 
+static void prepare_silent_urb(struct snd_usb_endpoint *ep,
+			       struct snd_urb_ctx *ctx)
+{
+	struct urb *urb = ctx->urb;
+	unsigned int offs = 0;
+	unsigned int extra = 0;
+	__le32 packet_length;
+	int i;
+
+	/* For tx_length_quirk, put packet length at start of packet */
+	if (ep->chip->tx_length_quirk)
+		extra = sizeof(packet_length);
+
+	for (i = 0; i < ctx->packets; ++i) {
+		unsigned int offset;
+		unsigned int length;
+		int counts;
+
+		if (ctx->packet_size[i])
+			counts = ctx->packet_size[i];
+		else
+			counts = snd_usb_endpoint_next_packet_size(ep);
+
+		length = counts * ep->stride; /* number of silent bytes */
+		offset = offs * ep->stride + extra * i;
+		urb->iso_frame_desc[i].offset = offset;
+		urb->iso_frame_desc[i].length = length + extra;
+		if (extra) {
+			packet_length = cpu_to_le32(length);
+			memcpy(urb->transfer_buffer + offset,
+			       &packet_length, sizeof(packet_length));
+		}
+		memset(urb->transfer_buffer + offset + extra,
+		       ep->silence_value, length);
+		offs += counts;
+	}
+
+	urb->number_of_packets = ctx->packets;
+	urb->transfer_buffer_length = offs * ep->stride + ctx->packets * extra;
+}
+
 /*
  * Prepare a PLAYBACK urb for submission to the bus.
  */
 static void prepare_outbound_urb(struct snd_usb_endpoint *ep,
 				 struct snd_urb_ctx *ctx)
 {
-	int i;
 	struct urb *urb = ctx->urb;
 	unsigned char *cp = urb->transfer_buffer;
 
@@ -201,24 +241,7 @@ static void prepare_outbound_urb(struct snd_usb_endpoint *ep,
 			ep->prepare_data_urb(ep->data_subs, urb);
 		} else {
 			/* no data provider, so send silence */
-			unsigned int offs = 0;
-			for (i = 0; i < ctx->packets; ++i) {
-				int counts;
-
-				if (ctx->packet_size[i])
-					counts = ctx->packet_size[i];
-				else
-					counts = snd_usb_endpoint_next_packet_size(ep);
-
-				urb->iso_frame_desc[i].offset = offs * ep->stride;
-				urb->iso_frame_desc[i].length = counts * ep->stride;
-				offs += counts;
-			}
-
-			urb->number_of_packets = ctx->packets;
-			urb->transfer_buffer_length = offs * ep->stride;
-			memset(urb->transfer_buffer, ep->silence_value,
-			       offs * ep->stride);
+			prepare_silent_urb(ep, ctx);
 		}
 		break;
 
@@ -594,6 +617,8 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep,
 	unsigned int max_packs_per_period, urbs_per_period, urb_packs;
 	unsigned int max_urbs, i;
 	int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels;
+	int tx_length_quirk = (ep->chip->tx_length_quirk &&
+			       usb_pipeout(ep->pipe));
 
 	if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) {
 		/*
@@ -610,13 +635,34 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep,
 
 	/* assume max. frequency is 25% higher than nominal */
 	ep->freqmax = ep->freqn + (ep->freqn >> 2);
-	maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3))
-				>> (16 - ep->datainterval);
+	/* Round up freqmax to nearest integer in order to calculate maximum
+	 * packet size, which must represent a whole number of frames.
+	 * This is accomplished by adding 0x0.ffff before converting the
+	 * Q16.16 format into integer.
+	 * In order to accurately calculate the maximum packet size when
+	 * the data interval is more than 1 (i.e. ep->datainterval > 0),
+	 * multiply by the data interval prior to rounding. For instance,
+	 * a freqmax of 41 kHz will result in a max packet size of 6 (5.125)
+	 * frames with a data interval of 1, but 11 (10.25) frames with a
+	 * data interval of 2.
+	 * (ep->freqmax << ep->datainterval overflows at 8.192 MHz for the
+	 * maximum datainterval value of 3, at USB full speed, higher for
+	 * USB high speed, noting that ep->freqmax is in units of
+	 * frames per packet in Q16.16 format.)
+	 */
+	maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) *
+			 (frame_bits >> 3);
+	if (tx_length_quirk)
+		maxsize += sizeof(__le32); /* Space for length descriptor */
 	/* but wMaxPacketSize might reduce this */
 	if (ep->maxpacksize && ep->maxpacksize < maxsize) {
 		/* whatever fits into a max. size packet */
-		maxsize = ep->maxpacksize;
-		ep->freqmax = (maxsize / (frame_bits >> 3))
+		unsigned int data_maxsize = maxsize = ep->maxpacksize;
+
+		if (tx_length_quirk)
+			/* Need to remove the length descriptor to calc freq */
+			data_maxsize -= sizeof(__le32);
+		ep->freqmax = (data_maxsize / (frame_bits >> 3))
 				<< (16 - ep->datainterval);
 	}
 
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 417ebb11cf48..7661616f3636 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1903,11 +1903,14 @@ static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi *umidi)
 
 	hostif = &intf->altsetting[1];
 	intfd = get_iface_desc(hostif);
+       /* If either or both of the endpoints support interrupt transfer,
+        * then use the alternate setting
+        */
 	if (intfd->bNumEndpoints != 2 ||
-	    (get_endpoint(hostif, 0)->bmAttributes &
-	     USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK ||
-	    (get_endpoint(hostif, 1)->bmAttributes &
-	     USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT)
+	    !((get_endpoint(hostif, 0)->bmAttributes &
+	       USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT ||
+	      (get_endpoint(hostif, 1)->bmAttributes &
+	       USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT))
 		return;
 
 	dev_dbg(&umidi->dev->dev, "switching to altsetting %d with int ep\n",
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index d3608c0a29f3..fe91184ce832 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -338,7 +338,7 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol,
 	struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol);
 	struct usb_mixer_interface *mixer = list->mixer;
 	int index = kcontrol->private_value & 0xff;
-	int value = ucontrol->value.integer.value[0];
+	unsigned int value = ucontrol->value.integer.value[0];
 	int old_value = kcontrol->private_value >> 8;
 	int err;
 
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index cdac5179db3f..9245f52d43bd 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -1383,6 +1383,56 @@ static inline void fill_playback_urb_dsd_dop(struct snd_usb_substream *subs,
 			subs->hwptr_done++;
 		}
 	}
+	if (subs->hwptr_done >= runtime->buffer_size * stride)
+		subs->hwptr_done -= runtime->buffer_size * stride;
+}
+
+static void copy_to_urb(struct snd_usb_substream *subs, struct urb *urb,
+			int offset, int stride, unsigned int bytes)
+{
+	struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+
+	if (subs->hwptr_done + bytes > runtime->buffer_size * stride) {
+		/* err, the transferred area goes over buffer boundary. */
+		unsigned int bytes1 =
+			runtime->buffer_size * stride - subs->hwptr_done;
+		memcpy(urb->transfer_buffer + offset,
+		       runtime->dma_area + subs->hwptr_done, bytes1);
+		memcpy(urb->transfer_buffer + offset + bytes1,
+		       runtime->dma_area, bytes - bytes1);
+	} else {
+		memcpy(urb->transfer_buffer + offset,
+		       runtime->dma_area + subs->hwptr_done, bytes);
+	}
+	subs->hwptr_done += bytes;
+	if (subs->hwptr_done >= runtime->buffer_size * stride)
+		subs->hwptr_done -= runtime->buffer_size * stride;
+}
+
+static unsigned int copy_to_urb_quirk(struct snd_usb_substream *subs,
+				      struct urb *urb, int stride,
+				      unsigned int bytes)
+{
+	__le32 packet_length;
+	int i;
+
+	/* Put __le32 length descriptor at start of each packet. */
+	for (i = 0; i < urb->number_of_packets; i++) {
+		unsigned int length = urb->iso_frame_desc[i].length;
+		unsigned int offset = urb->iso_frame_desc[i].offset;
+
+		packet_length = cpu_to_le32(length);
+		offset += i * sizeof(packet_length);
+		urb->iso_frame_desc[i].offset = offset;
+		urb->iso_frame_desc[i].length += sizeof(packet_length);
+		memcpy(urb->transfer_buffer + offset,
+		       &packet_length, sizeof(packet_length));
+		copy_to_urb(subs, urb, offset + sizeof(packet_length),
+			    stride, length);
+	}
+	/* Adjust transfer size accordingly. */
+	bytes += urb->number_of_packets * sizeof(packet_length);
+	return bytes;
 }
 
 static void prepare_playback_urb(struct snd_usb_substream *subs,
@@ -1460,27 +1510,17 @@ static void prepare_playback_urb(struct snd_usb_substream *subs,
 		}
 
 		subs->hwptr_done += bytes;
+		if (subs->hwptr_done >= runtime->buffer_size * stride)
+			subs->hwptr_done -= runtime->buffer_size * stride;
 	} else {
 		/* usual PCM */
-		if (subs->hwptr_done + bytes > runtime->buffer_size * stride) {
-			/* err, the transferred area goes over buffer boundary. */
-			unsigned int bytes1 =
-				runtime->buffer_size * stride - subs->hwptr_done;
-			memcpy(urb->transfer_buffer,
-			       runtime->dma_area + subs->hwptr_done, bytes1);
-			memcpy(urb->transfer_buffer + bytes1,
-			       runtime->dma_area, bytes - bytes1);
-		} else {
-			memcpy(urb->transfer_buffer,
-			       runtime->dma_area + subs->hwptr_done, bytes);
-		}
-
-		subs->hwptr_done += bytes;
+		if (!subs->tx_length_quirk)
+			copy_to_urb(subs, urb, 0, stride, bytes);
+		else
+			bytes = copy_to_urb_quirk(subs, urb, stride, bytes);
+			/* bytes is now amount of outgoing data */
 	}
 
-	if (subs->hwptr_done >= runtime->buffer_size * stride)
-		subs->hwptr_done -= runtime->buffer_size * stride;
-
 	/* update delay with exact number of samples queued */
 	runtime->delay = subs->last_delay;
 	runtime->delay += frames;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index e4756651a52c..1a1e2e4df35e 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2664,6 +2664,15 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	USB_DEVICE(0x1235, 0x000a),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		/* .vendor_name = "Novation", */
+		/* .product_name = "Nocturn", */
+		.ifnum = 0,
+		.type = QUIRK_MIDI_RAW_BYTES
+	}
+},
+{
 	USB_DEVICE(0x1235, 0x000e),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
 		/* .vendor_name = "Novation", */
@@ -3182,10 +3191,9 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 {
 	/*
 	 * ZOOM R16/24 in audio interface mode.
-	 * Mixer descriptors are garbage, further quirks will be needed
-	 * to make any of it functional, thus disabled for now.
-	 * Playback stream appears to start and run fine but no sound
-	 * is produced, so also disabled for now.
+	 * Playback requires an extra four byte LE length indicator
+	 * at the start of each isochronous packet. This quirk is
+	 * enabled in create_standard_audio_quirk().
 	 */
 	USB_DEVICE(0x1686, 0x00dd),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
@@ -3193,14 +3201,9 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 		.type = QUIRK_COMPOSITE,
 		.data = (const struct snd_usb_audio_quirk[]) {
 			{
-				/* Mixer */
-				.ifnum = 0,
-				.type = QUIRK_IGNORE_INTERFACE,
-			},
-			{
 				/* Playback  */
 				.ifnum = 1,
-				.type = QUIRK_IGNORE_INTERFACE,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE,
 			},
 			{
 				/* Capture */
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 00ebc0ca008e..4897ea171194 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -115,6 +115,9 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip,
 	struct usb_interface_descriptor *altsd;
 	int err;
 
+	if (chip->usb_id == USB_ID(0x1686, 0x00dd)) /* Zoom R16/24 */
+		chip->tx_length_quirk = 1;
+
 	alts = &iface->altsetting[0];
 	altsd = get_iface_desc(alts);
 	err = snd_usb_parse_audio_interface(chip, altsd->bInterfaceNumber);
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 970086015cde..8ee14f2365e7 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -92,6 +92,7 @@ static void snd_usb_init_substream(struct snd_usb_stream *as,
 	subs->direction = stream;
 	subs->dev = as->chip->dev;
 	subs->txfr_quirk = as->chip->txfr_quirk;
+	subs->tx_length_quirk = as->chip->tx_length_quirk;
 	subs->speed = snd_usb_get_speed(subs->dev);
 	subs->pkt_offset_adj = 0;
 
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 33a176437e2e..15a12715bd05 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -43,6 +43,7 @@ struct snd_usb_audio {
 	atomic_t usage_count;
 	wait_queue_head_t shutdown_wait;
 	unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
+	unsigned int tx_length_quirk:1; /* Put length specifier in transfers */
 	
 	int num_interfaces;
 	int num_suspended_intf;