Linux-libre 4.7.1-gnupck-4.7.1-gnu

178 files changed, 7265 insertions, 1896 deletions

diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index 6d12ca9bc..9749f9e8b 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -141,35 +141,6 @@ config SND_SEQ_HRTIMER_DEFAULT
 	  Say Y here to use the HR-timer backend as the default sequencer
 	  timer.
 
-config SND_RTCTIMER
-	tristate "RTC Timer support"
-	depends on RTC
-	select SND_TIMER
-	help
-	  Say Y here to enable RTC timer support for ALSA.  ALSA uses
-	  the RTC timer as a precise timing source and maps the RTC
-	  timer to ALSA's timer interface.  The ALSA sequencer code also
-	  can use this timing source.
-
-	  To compile this driver as a module, choose M here: the module
-	  will be called snd-rtctimer.
-
-	  Note that this option is exclusive with the new RTC drivers
-	  (CONFIG_RTC_CLASS) since this requires the old API.
-
-config SND_SEQ_RTCTIMER_DEFAULT
-	bool "Use RTC as default sequencer timer"
-	depends on SND_RTCTIMER && SND_SEQUENCER
-	depends on !SND_SEQ_HRTIMER_DEFAULT
-	default y
-	help
-	  Say Y here to use the RTC timer as the default sequencer
-	  timer.  This is strongly recommended because it ensures
-	  precise MIDI timing even when the system timer runs at less
-	  than 1000 Hz.
-
-	  If in doubt, say Y.
-
 config SND_DYNAMIC_MINORS
 	bool "Dynamic device file minor numbers"
 	help
diff --git a/sound/core/Makefile b/sound/core/Makefile
index 48ab4b8f8..e85d9dd12 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -37,7 +37,6 @@ obj-$(CONFIG_SND) 		+= snd.o
 obj-$(CONFIG_SND_HWDEP)		+= snd-hwdep.o
 obj-$(CONFIG_SND_TIMER)		+= snd-timer.o
 obj-$(CONFIG_SND_HRTIMER)	+= snd-hrtimer.o
-obj-$(CONFIG_SND_RTCTIMER)	+= snd-rtctimer.o
 obj-$(CONFIG_SND_PCM)		+= snd-pcm.o
 obj-$(CONFIG_SND_DMAENGINE_PCM)	+= snd-pcm-dmaengine.o
 obj-$(CONFIG_SND_RAWMIDI)	+= snd-rawmidi.o
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index a9933c07a..9b3334be9 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -288,9 +288,12 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf,
 	stream = &data->stream;
 	mutex_lock(&stream->device->lock);
 	/* write is allowed when stream is running or has been steup */
-	if (stream->runtime->state != SNDRV_PCM_STATE_SETUP &&
-	    stream->runtime->state != SNDRV_PCM_STATE_PREPARED &&
-			stream->runtime->state != SNDRV_PCM_STATE_RUNNING) {
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_SETUP:
+	case SNDRV_PCM_STATE_PREPARED:
+	case SNDRV_PCM_STATE_RUNNING:
+		break;
+	default:
 		mutex_unlock(&stream->device->lock);
 		return -EBADFD;
 	}
@@ -391,14 +394,13 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait)
 	int retval = 0;
 
 	if (snd_BUG_ON(!data))
-		return -EFAULT;
+		return POLLERR;
+
 	stream = &data->stream;
-	if (snd_BUG_ON(!stream))
-		return -EFAULT;
 
 	mutex_lock(&stream->device->lock);
 	if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) {
-		retval = -EBADFD;
+		retval = snd_compr_get_poll(stream) | POLLERR;
 		goto out;
 	}
 	poll_wait(f, &stream->runtime->sleep, wait);
@@ -421,10 +423,7 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait)
 			retval = snd_compr_get_poll(stream);
 		break;
 	default:
-		if (stream->direction == SND_COMPRESS_PLAYBACK)
-			retval = POLLOUT | POLLWRNORM | POLLERR;
-		else
-			retval = POLLIN | POLLRDNORM | POLLERR;
+		retval = snd_compr_get_poll(stream) | POLLERR;
 		break;
 	}
 out:
@@ -802,9 +801,9 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg)
 
 	if (snd_BUG_ON(!data))
 		return -EFAULT;
+
 	stream = &data->stream;
-	if (snd_BUG_ON(!stream))
-		return -EFAULT;
+
 	mutex_lock(&stream->device->lock);
 	switch (_IOC_NR(cmd)) {
 	case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION):
diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c
index 656d9a903..e2f27022b 100644
--- a/sound/core/hrtimer.c
+++ b/sound/core/hrtimer.c
@@ -38,37 +38,53 @@ static unsigned int resolution;
 struct snd_hrtimer {
 	struct snd_timer *timer;
 	struct hrtimer hrt;
-	atomic_t running;
+	bool in_callback;
 };
 
 static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
 {
 	struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt);
 	struct snd_timer *t = stime->timer;
-	unsigned long oruns;
-
-	if (!atomic_read(&stime->running))
-		return HRTIMER_NORESTART;
-
-	oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
-	snd_timer_interrupt(stime->timer, t->sticks * oruns);
+	ktime_t delta;
+	unsigned long ticks;
+	enum hrtimer_restart ret = HRTIMER_NORESTART;
+
+	spin_lock(&t->lock);
+	if (!t->running)
+		goto out; /* fast path */
+	stime->in_callback = true;
+	ticks = t->sticks;
+	spin_unlock(&t->lock);
+
+	/* calculate the drift */
+	delta = ktime_sub(hrt->base->get_time(), hrtimer_get_expires(hrt));
+	if (delta.tv64 > 0)
+		ticks += ktime_divns(delta, ticks * resolution);
+
+	snd_timer_interrupt(stime->timer, ticks);
+
+	spin_lock(&t->lock);
+	if (t->running) {
+		hrtimer_add_expires_ns(hrt, t->sticks * resolution);
+		ret = HRTIMER_RESTART;
+	}
 
-	if (!atomic_read(&stime->running))
-		return HRTIMER_NORESTART;
-	return HRTIMER_RESTART;
+	stime->in_callback = false;
+ out:
+	spin_unlock(&t->lock);
+	return ret;
 }
 
 static int snd_hrtimer_open(struct snd_timer *t)
 {
 	struct snd_hrtimer *stime;
 
-	stime = kmalloc(sizeof(*stime), GFP_KERNEL);
+	stime = kzalloc(sizeof(*stime), GFP_KERNEL);
 	if (!stime)
 		return -ENOMEM;
 	hrtimer_init(&stime->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
 	stime->timer = t;
 	stime->hrt.function = snd_hrtimer_callback;
-	atomic_set(&stime->running, 0);
 	t->private_data = stime;
 	return 0;
 }
@@ -78,6 +94,11 @@ static int snd_hrtimer_close(struct snd_timer *t)
 	struct snd_hrtimer *stime = t->private_data;
 
 	if (stime) {
+		spin_lock_irq(&t->lock);
+		t->running = 0; /* just to be sure */
+		stime->in_callback = 1; /* skip start/stop */
+		spin_unlock_irq(&t->lock);
+
 		hrtimer_cancel(&stime->hrt);
 		kfree(stime);
 		t->private_data = NULL;
@@ -89,18 +110,19 @@ static int snd_hrtimer_start(struct snd_timer *t)
 {
 	struct snd_hrtimer *stime = t->private_data;
 
-	atomic_set(&stime->running, 0);
-	hrtimer_try_to_cancel(&stime->hrt);
+	if (stime->in_callback)
+		return 0;
 	hrtimer_start(&stime->hrt, ns_to_ktime(t->sticks * resolution),
 		      HRTIMER_MODE_REL);
-	atomic_set(&stime->running, 1);
 	return 0;
 }
 
 static int snd_hrtimer_stop(struct snd_timer *t)
 {
 	struct snd_hrtimer *stime = t->private_data;
-	atomic_set(&stime->running, 0);
+
+	if (stime->in_callback)
+		return 0;
 	hrtimer_try_to_cancel(&stime->hrt);
 	return 0;
 }
diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c
index 697c166ac..8eb58c709 100644
--- a/sound/core/pcm_dmaengine.c
+++ b/sound/core/pcm_dmaengine.c
@@ -106,8 +106,9 @@ EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config);
  * direction of the substream. If the substream is a playback stream the dst
  * fields will be initialized, if it is a capture stream the src fields will be
  * initialized. The {dst,src}_addr_width field will only be initialized if the
- * addr_width field of the DAI DMA data struct is not equal to
- * DMA_SLAVE_BUSWIDTH_UNDEFINED.
+ * SND_DMAENGINE_PCM_DAI_FLAG_PACK flag is set or if the addr_width field of
+ * the DAI DMA data struct is not equal to DMA_SLAVE_BUSWIDTH_UNDEFINED. If
+ * both conditions are met the latter takes priority.
  */
 void snd_dmaengine_pcm_set_config_from_dai_data(
 	const struct snd_pcm_substream *substream,
@@ -117,11 +118,17 @@ void snd_dmaengine_pcm_set_config_from_dai_data(
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		slave_config->dst_addr = dma_data->addr;
 		slave_config->dst_maxburst = dma_data->maxburst;
+		if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)
+			slave_config->dst_addr_width =
+				DMA_SLAVE_BUSWIDTH_UNDEFINED;
 		if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED)
 			slave_config->dst_addr_width = dma_data->addr_width;
 	} else {
 		slave_config->src_addr = dma_data->addr;
 		slave_config->src_maxburst = dma_data->maxburst;
+		if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)
+			slave_config->src_addr_width =
+				DMA_SLAVE_BUSWIDTH_UNDEFINED;
 		if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED)
 			slave_config->src_addr_width = dma_data->addr_width;
 	}
diff --git a/sound/core/pcm_iec958.c b/sound/core/pcm_iec958.c
index 36b2d7aca..5e6aed64f 100644
--- a/sound/core/pcm_iec958.c
+++ b/sound/core/pcm_iec958.c
@@ -9,30 +9,18 @@
 #include <linux/types.h>
 #include <sound/asoundef.h>
 #include <sound/pcm.h>
+#include <sound/pcm_params.h>
 #include <sound/pcm_iec958.h>
 
-/**
- * snd_pcm_create_iec958_consumer - create consumer format IEC958 channel status
- * @runtime: pcm runtime structure with ->rate filled in
- * @cs: channel status buffer, at least four bytes
- * @len: length of channel status buffer
- *
- * Create the consumer format channel status data in @cs of maximum size
- * @len corresponding to the parameters of the PCM runtime @runtime.
- *
- * Drivers may wish to tweak the contents of the buffer after creation.
- *
- * Returns: length of buffer, or negative error code if something failed.
- */
-int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
-	size_t len)
+static int create_iec958_consumer(uint rate, uint sample_width,
+				  u8 *cs, size_t len)
 {
 	unsigned int fs, ws;
 
 	if (len < 4)
 		return -EINVAL;
 
-	switch (runtime->rate) {
+	switch (rate) {
 	case 32000:
 		fs = IEC958_AES3_CON_FS_32000;
 		break;
@@ -59,7 +47,7 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
 	}
 
 	if (len > 4) {
-		switch (snd_pcm_format_width(runtime->format)) {
+		switch (sample_width) {
 		case 16:
 			ws = IEC958_AES4_CON_WORDLEN_20_16;
 			break;
@@ -71,6 +59,7 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
 			     IEC958_AES4_CON_MAX_WORDLEN_24;
 			break;
 		case 24:
+		case 32: /* Assume 24-bit width for 32-bit samples. */
 			ws = IEC958_AES4_CON_WORDLEN_24_20 |
 			     IEC958_AES4_CON_MAX_WORDLEN_24;
 			break;
@@ -92,4 +81,46 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
 
 	return len;
 }
+
+/**
+ * snd_pcm_create_iec958_consumer - create consumer format IEC958 channel status
+ * @runtime: pcm runtime structure with ->rate filled in
+ * @cs: channel status buffer, at least four bytes
+ * @len: length of channel status buffer
+ *
+ * Create the consumer format channel status data in @cs of maximum size
+ * @len corresponding to the parameters of the PCM runtime @runtime.
+ *
+ * Drivers may wish to tweak the contents of the buffer after creation.
+ *
+ * Returns: length of buffer, or negative error code if something failed.
+ */
+int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
+	size_t len)
+{
+	return create_iec958_consumer(runtime->rate,
+				      snd_pcm_format_width(runtime->format),
+				      cs, len);
+}
 EXPORT_SYMBOL(snd_pcm_create_iec958_consumer);
+
+/**
+ * snd_pcm_create_iec958_consumer_hw_params - create IEC958 channel status
+ * @hw_params: the hw_params instance for extracting rate and sample format
+ * @cs: channel status buffer, at least four bytes
+ * @len: length of channel status buffer
+ *
+ * Create the consumer format channel status data in @cs of maximum size
+ * @len corresponding to the parameters of the PCM runtime @runtime.
+ *
+ * Drivers may wish to tweak the contents of the buffer after creation.
+ *
+ * Returns: length of buffer, or negative error code if something failed.
+ */
+int snd_pcm_create_iec958_consumer_hw_params(struct snd_pcm_hw_params *params,
+					     u8 *cs, size_t len)
+{
+	return create_iec958_consumer(params_rate(params), params_width(params),
+				      cs, len);
+}
+EXPORT_SYMBOL(snd_pcm_create_iec958_consumer_hw_params);
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 3a9b66c6e..bb1261591 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1886,8 +1886,8 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream)
 		snd_timer_interrupt(substream->timer, 1);
 #endif
  _end:
-	snd_pcm_stream_unlock_irqrestore(substream, flags);
 	kill_fasync(&runtime->fasync, SIGIO, POLL_IN);
+	snd_pcm_stream_unlock_irqrestore(substream, flags);
 }
 
 EXPORT_SYMBOL(snd_pcm_period_elapsed);
@@ -2595,6 +2595,8 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream,
 	};
 	int err;
 
+	if (WARN_ON(pcm->streams[stream].chmap_kctl))
+		return -EBUSY;
 	info = kzalloc(sizeof(*info), GFP_KERNEL);
 	if (!info)
 		return -ENOMEM;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 9106d8e23..c61fd50f7 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -3161,7 +3161,7 @@ static unsigned int snd_pcm_playback_poll(struct file *file, poll_table * wait)
 
 	substream = pcm_file->substream;
 	if (PCM_RUNTIME_CHECK(substream))
-		return -ENXIO;
+		return POLLOUT | POLLWRNORM | POLLERR;
 	runtime = substream->runtime;
 
 	poll_wait(file, &runtime->sleep, wait);
@@ -3200,7 +3200,7 @@ static unsigned int snd_pcm_capture_poll(struct file *file, poll_table * wait)
 
 	substream = pcm_file->substream;
 	if (PCM_RUNTIME_CHECK(substream))
-		return -ENXIO;
+		return POLLIN | POLLRDNORM | POLLERR;
 	runtime = substream->runtime;
 
 	poll_wait(file, &runtime->sleep, wait);
diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c
deleted file mode 100644
index f3420d11a..000000000
--- a/sound/core/rtctimer.c
+++ /dev/null
@@ -1,187 +0,0 @@
-/*
- *  RTC based high-frequency timer
- *
- *  Copyright (C) 2000 Takashi Iwai
- *	based on rtctimer.c by Steve Ratcliffe
- *
- *   This program is free software; you can redistribute it and/or modify
- *   it under the terms of the GNU General Public License as published by
- *   the Free Software Foundation; either version 2 of the License, or
- *   (at your option) any later version.
- *
- *   This program is distributed in the hope that it will be useful,
- *   but WITHOUT ANY WARRANTY; without even the implied warranty of
- *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- *   GNU General Public License for more details.
- *
- *   You should have received a copy of the GNU General Public License
- *   along with this program; if not, write to the Free Software
- *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
- *
- */
-
-#include <linux/init.h>
-#include <linux/interrupt.h>
-#include <linux/module.h>
-#include <linux/log2.h>
-#include <sound/core.h>
-#include <sound/timer.h>
-
-#if IS_ENABLED(CONFIG_RTC)
-
-#include <linux/mc146818rtc.h>
-
-#define RTC_FREQ	1024		/* default frequency */
-#define NANO_SEC	1000000000L	/* 10^9 in sec */
-
-/*
- * prototypes
- */
-static int rtctimer_open(struct snd_timer *t);
-static int rtctimer_close(struct snd_timer *t);
-static int rtctimer_start(struct snd_timer *t);
-static int rtctimer_stop(struct snd_timer *t);
-
-
-/*
- * The hardware dependent description for this timer.
- */
-static struct snd_timer_hardware rtc_hw = {
-	.flags =	SNDRV_TIMER_HW_AUTO |
-			SNDRV_TIMER_HW_FIRST |
-			SNDRV_TIMER_HW_TASKLET,
-	.ticks =	100000000L,		/* FIXME: XXX */
-	.open =		rtctimer_open,
-	.close =	rtctimer_close,
-	.start =	rtctimer_start,
-	.stop =		rtctimer_stop,
-};
-
-static int rtctimer_freq = RTC_FREQ;		/* frequency */
-static struct snd_timer *rtctimer;
-static struct tasklet_struct rtc_tasklet;
-static rtc_task_t rtc_task;
-
-
-static int
-rtctimer_open(struct snd_timer *t)
-{
-	int err;
-
-	err = rtc_register(&rtc_task);
-	if (err < 0)
-		return err;
-	t->private_data = &rtc_task;
-	return 0;
-}
-
-static int
-rtctimer_close(struct snd_timer *t)
-{
-	rtc_task_t *rtc = t->private_data;
-	if (rtc) {
-		rtc_unregister(rtc);
-		tasklet_kill(&rtc_tasklet);
-		t->private_data = NULL;
-	}
-	return 0;
-}
-
-static int
-rtctimer_start(struct snd_timer *timer)
-{
-	rtc_task_t *rtc = timer->private_data;
-	if (snd_BUG_ON(!rtc))
-		return -EINVAL;
-	rtc_control(rtc, RTC_IRQP_SET, rtctimer_freq);
-	rtc_control(rtc, RTC_PIE_ON, 0);
-	return 0;
-}
-
-static int
-rtctimer_stop(struct snd_timer *timer)
-{
-	rtc_task_t *rtc = timer->private_data;
-	if (snd_BUG_ON(!rtc))
-		return -EINVAL;
-	rtc_control(rtc, RTC_PIE_OFF, 0);
-	return 0;
-}
-
-static void rtctimer_tasklet(unsigned long data)
-{
-	snd_timer_interrupt((struct snd_timer *)data, 1);
-}
-
-/*
- * interrupt
- */
-static void rtctimer_interrupt(void *private_data)
-{
-	tasklet_schedule(private_data);
-}
-
-
-/*
- *  ENTRY functions
- */
-static int __init rtctimer_init(void)
-{
-	int err;
-	struct snd_timer *timer;
-
-	if (rtctimer_freq < 2 || rtctimer_freq > 8192 ||
-	    !is_power_of_2(rtctimer_freq)) {
-		pr_err("ALSA: rtctimer: invalid frequency %d\n", rtctimer_freq);
-		return -EINVAL;
-	}
-
-	/* Create a new timer and set up the fields */
-	err = snd_timer_global_new("rtc", SNDRV_TIMER_GLOBAL_RTC, &timer);
-	if (err < 0)
-		return err;
-
-	timer->module = THIS_MODULE;
-	strcpy(timer->name, "RTC timer");
-	timer->hw = rtc_hw;
-	timer->hw.resolution = NANO_SEC / rtctimer_freq;
-
-	tasklet_init(&rtc_tasklet, rtctimer_tasklet, (unsigned long)timer);
-
-	/* set up RTC callback */
-	rtc_task.func = rtctimer_interrupt;
-	rtc_task.private_data = &rtc_tasklet;
-
-	err = snd_timer_global_register(timer);
-	if (err < 0) {
-		snd_timer_global_free(timer);
-		return err;
-	}
-	rtctimer = timer; /* remember this */
-
-	return 0;
-}
-
-static void __exit rtctimer_exit(void)
-{
-	if (rtctimer) {
-		snd_timer_global_free(rtctimer);
-		rtctimer = NULL;
-	}
-}
-
-
-/*
- * exported stuff
- */
-module_init(rtctimer_init)
-module_exit(rtctimer_exit)
-
-module_param(rtctimer_freq, int, 0444);
-MODULE_PARM_DESC(rtctimer_freq, "timer frequency in Hz");
-
-MODULE_LICENSE("GPL");
-
-MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_RTC));
-
-#endif /* IS_ENABLED(CONFIG_RTC) */
diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c
index 7e0aabb80..639544b4f 100644
--- a/sound/core/seq/seq.c
+++ b/sound/core/seq/seq.c
@@ -47,8 +47,6 @@ int seq_default_timer_card = -1;
 int seq_default_timer_device =
 #ifdef CONFIG_SND_SEQ_HRTIMER_DEFAULT
 	SNDRV_TIMER_GLOBAL_HRTIMER
-#elif defined(CONFIG_SND_SEQ_RTCTIMER_DEFAULT)
-	SNDRV_TIMER_GLOBAL_RTC
 #else
 	SNDRV_TIMER_GLOBAL_SYSTEM
 #endif
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 23b73f6ac..9a6157ea6 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -37,8 +37,6 @@
 
 #if IS_ENABLED(CONFIG_SND_HRTIMER)
 #define DEFAULT_TIMER_LIMIT 4
-#elif IS_ENABLED(CONFIG_SND_RTCTIMER)
-#define DEFAULT_TIMER_LIMIT 2
 #else
 #define DEFAULT_TIMER_LIMIT 1
 #endif
@@ -1225,6 +1223,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri,
 		tu->tstamp = *tstamp;
 	if ((tu->filter & (1 << event)) == 0 || !tu->tread)
 		return;
+	memset(&r1, 0, sizeof(r1));
 	r1.event = event;
 	r1.tstamp = *tstamp;
 	r1.val = resolution;
@@ -1267,6 +1266,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri,
 	}
 	if ((tu->filter & (1 << SNDRV_TIMER_EVENT_RESOLUTION)) &&
 	    tu->last_resolution != resolution) {
+		memset(&r1, 0, sizeof(r1));
 		r1.event = SNDRV_TIMER_EVENT_RESOLUTION;
 		r1.tstamp = tstamp;
 		r1.val = resolution;
@@ -1739,6 +1739,7 @@ static int snd_timer_user_params(struct file *file,
 	if (tu->timeri->flags & SNDRV_TIMER_IFLG_EARLY_EVENT) {
 		if (tu->tread) {
 			struct snd_timer_tread tread;
+			memset(&tread, 0, sizeof(tread));
 			tread.event = SNDRV_TIMER_EVENT_EARLY;
 			tread.tstamp.tv_sec = 0;
 			tread.tstamp.tv_nsec = 0;
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index 2a779c2f6..ab894ed1f 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -134,6 +134,7 @@ config SND_FIREWIRE_TASCAM
 	 Say Y here to include support for TASCAM.
 	  * FW-1884
 	  * FW-1082
+	  * FW-1804
 
 	 To compile this driver as a module, choose M here: the module
 	 will be called snd-firewire-tascam.
diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile
index 003c09029..0ee1fb115 100644
--- a/sound/firewire/Makefile
+++ b/sound/firewire/Makefile
@@ -1,3 +1,6 @@
+# To find a header included by define_trace.h.
+CFLAGS_amdtp-stream.o	:= -I$(src)
+
 snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \
 			 fcp.o cmp.o amdtp-stream.o amdtp-am824.o
 snd-isight-objs := isight.o
diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h
new file mode 100644
index 000000000..9c04faf20
--- /dev/null
+++ b/sound/firewire/amdtp-stream-trace.h
@@ -0,0 +1,110 @@
+/*
+ * amdtp-stream-trace.h - tracepoint definitions to dump a part of packet data
+ *
+ * Copyright (c) 2016 Takashi Sakamoto
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#undef TRACE_SYSTEM
+#define TRACE_SYSTEM		snd_firewire_lib
+
+#if !defined(_AMDTP_STREAM_TRACE_H) || defined(TRACE_HEADER_MULTI_READ)
+#define _AMDTP_STREAM_TRACE_H
+
+#include <linux/tracepoint.h>
+
+TRACE_EVENT(in_packet,
+	TP_PROTO(const struct amdtp_stream *s, u32 cycles, u32 cip_header[2], unsigned int payload_quadlets, unsigned int index),
+	TP_ARGS(s, cycles, cip_header, payload_quadlets, index),
+	TP_STRUCT__entry(
+		__field(unsigned int, second)
+		__field(unsigned int, cycle)
+		__field(int, channel)
+		__field(int, src)
+		__field(int, dest)
+		__field(u32, cip_header0)
+		__field(u32, cip_header1)
+		__field(unsigned int, payload_quadlets)
+		__field(unsigned int, packet_index)
+		__field(unsigned int, irq)
+		__field(unsigned int, index)
+	),
+	TP_fast_assign(
+		__entry->second = cycles / CYCLES_PER_SECOND;
+		__entry->cycle = cycles % CYCLES_PER_SECOND;
+		__entry->channel = s->context->channel;
+		__entry->src = fw_parent_device(s->unit)->node_id;
+		__entry->dest = fw_parent_device(s->unit)->card->node_id;
+		__entry->cip_header0 = cip_header[0];
+		__entry->cip_header1 = cip_header[1];
+		__entry->payload_quadlets = payload_quadlets;
+		__entry->packet_index = s->packet_index;
+		__entry->irq = !!in_interrupt();
+		__entry->index = index;
+	),
+	TP_printk(
+		"%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u",
+		__entry->second,
+		__entry->cycle,
+		__entry->src,
+		__entry->dest,
+		__entry->channel,
+		__entry->cip_header0,
+		__entry->cip_header1,
+		__entry->payload_quadlets,
+		__entry->packet_index,
+		__entry->irq,
+		__entry->index)
+);
+
+TRACE_EVENT(out_packet,
+	TP_PROTO(const struct amdtp_stream *s, u32 cycles, __be32 *cip_header, unsigned int payload_length, unsigned int index),
+	TP_ARGS(s, cycles, cip_header, payload_length, index),
+	TP_STRUCT__entry(
+		__field(unsigned int, second)
+		__field(unsigned int, cycle)
+		__field(int, channel)
+		__field(int, src)
+		__field(int, dest)
+		__field(u32, cip_header0)
+		__field(u32, cip_header1)
+		__field(unsigned int, payload_quadlets)
+		__field(unsigned int, packet_index)
+		__field(unsigned int, irq)
+		__field(unsigned int, index)
+	),
+	TP_fast_assign(
+		__entry->second = cycles / CYCLES_PER_SECOND;
+		__entry->cycle = cycles % CYCLES_PER_SECOND;
+		__entry->channel = s->context->channel;
+		__entry->src = fw_parent_device(s->unit)->card->node_id;
+		__entry->dest = fw_parent_device(s->unit)->node_id;
+		__entry->cip_header0 = be32_to_cpu(cip_header[0]);
+		__entry->cip_header1 = be32_to_cpu(cip_header[1]);
+		__entry->payload_quadlets = payload_length / 4;
+		__entry->packet_index = s->packet_index;
+		__entry->irq = !!in_interrupt();
+		__entry->index = index;
+	),
+	TP_printk(
+		"%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u",
+		__entry->second,
+		__entry->cycle,
+		__entry->src,
+		__entry->dest,
+		__entry->channel,
+		__entry->cip_header0,
+		__entry->cip_header1,
+		__entry->payload_quadlets,
+		__entry->packet_index,
+		__entry->irq,
+		__entry->index)
+);
+
+#endif
+
+#undef TRACE_INCLUDE_PATH
+#define TRACE_INCLUDE_PATH	.
+#undef TRACE_INCLUDE_FILE
+#define TRACE_INCLUDE_FILE	amdtp-stream-trace
+#include <trace/define_trace.h>
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index ed2902609..00060c4a9 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -19,6 +19,10 @@
 #define CYCLES_PER_SECOND	8000
 #define TICKS_PER_SECOND	(TICKS_PER_CYCLE * CYCLES_PER_SECOND)
 
+/* Always support Linux tracing subsystem. */
+#define CREATE_TRACE_POINTS
+#include "amdtp-stream-trace.h"
+
 #define TRANSFER_DELAY_TICKS	0x2e00 /* 479.17 microseconds */
 
 /* isochronous header parameters */
@@ -87,7 +91,6 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
 
 	init_waitqueue_head(&s->callback_wait);
 	s->callbacked = false;
-	s->sync_slave = NULL;
 
 	s->fmt = fmt;
 	s->process_data_blocks = process_data_blocks;
@@ -102,6 +105,10 @@ EXPORT_SYMBOL(amdtp_stream_init);
  */
 void amdtp_stream_destroy(struct amdtp_stream *s)
 {
+	/* Not initialized. */
+	if (s->protocol == NULL)
+		return;
+
 	WARN_ON(amdtp_stream_running(s));
 	kfree(s->protocol);
 	mutex_destroy(&s->mutex);
@@ -244,7 +251,6 @@ void amdtp_stream_pcm_prepare(struct amdtp_stream *s)
 	tasklet_kill(&s->period_tasklet);
 	s->pcm_buffer_pointer = 0;
 	s->pcm_period_pointer = 0;
-	s->pointer_flush = true;
 }
 EXPORT_SYMBOL(amdtp_stream_pcm_prepare);
 
@@ -349,7 +355,6 @@ static void update_pcm_pointers(struct amdtp_stream *s,
 	s->pcm_period_pointer += frames;
 	if (s->pcm_period_pointer >= pcm->runtime->period_size) {
 		s->pcm_period_pointer -= pcm->runtime->period_size;
-		s->pointer_flush = false;
 		tasklet_hi_schedule(&s->period_tasklet);
 	}
 }
@@ -363,9 +368,8 @@ static void pcm_period_tasklet(unsigned long data)
 		snd_pcm_period_elapsed(pcm);
 }
 
-static int queue_packet(struct amdtp_stream *s,
-			unsigned int header_length,
-			unsigned int payload_length, bool skip)
+static int queue_packet(struct amdtp_stream *s, unsigned int header_length,
+			unsigned int payload_length)
 {
 	struct fw_iso_packet p = {0};
 	int err = 0;
@@ -376,8 +380,10 @@ static int queue_packet(struct amdtp_stream *s,
 	p.interrupt = IS_ALIGNED(s->packet_index + 1, INTERRUPT_INTERVAL);
 	p.tag = TAG_CIP;
 	p.header_length = header_length;
-	p.payload_length = (!skip) ? payload_length : 0;
-	p.skip = skip;
+	if (payload_length > 0)
+		p.payload_length = payload_length;
+	else
+		p.skip = true;
 	err = fw_iso_context_queue(s->context, &p, &s->buffer.iso_buffer,
 				   s->buffer.packets[s->packet_index].offset);
 	if (err < 0) {
@@ -392,27 +398,30 @@ end:
 }
 
 static inline int queue_out_packet(struct amdtp_stream *s,
-				   unsigned int payload_length, bool skip)
+				   unsigned int payload_length)
 {
-	return queue_packet(s, OUT_PACKET_HEADER_SIZE,
-			    payload_length, skip);
+	return queue_packet(s, OUT_PACKET_HEADER_SIZE, payload_length);
 }
 
 static inline int queue_in_packet(struct amdtp_stream *s)
 {
 	return queue_packet(s, IN_PACKET_HEADER_SIZE,
-			    amdtp_stream_get_max_payload(s), false);
+			    amdtp_stream_get_max_payload(s));
 }
 
-static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks,
-			     unsigned int syt)
+static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle,
+			     unsigned int index)
 {
 	__be32 *buffer;
+	unsigned int syt;
+	unsigned int data_blocks;
 	unsigned int payload_length;
 	unsigned int pcm_frames;
 	struct snd_pcm_substream *pcm;
 
 	buffer = s->buffer.packets[s->packet_index].buffer;
+	syt = calculate_syt(s, cycle);
+	data_blocks = calculate_data_blocks(s, syt);
 	pcm_frames = s->process_data_blocks(s, buffer + 2, data_blocks, &syt);
 
 	buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) |
@@ -424,9 +433,11 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks,
 				(syt & CIP_SYT_MASK));
 
 	s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff;
-
 	payload_length = 8 + data_blocks * 4 * s->data_block_quadlets;
-	if (queue_out_packet(s, payload_length, false) < 0)
+
+	trace_out_packet(s, cycle, buffer, payload_length, index);
+
+	if (queue_out_packet(s, payload_length) < 0)
 		return -EIO;
 
 	pcm = ACCESS_ONCE(s->pcm);
@@ -438,19 +449,24 @@ 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 syt)
+			    unsigned int payload_quadlets, unsigned int cycle,
+			    unsigned int index)
 {
+	__be32 *buffer;
 	u32 cip_header[2];
-	unsigned int fmt, fdf;
+	unsigned int fmt, fdf, syt;
 	unsigned int data_block_quadlets, data_block_counter, dbc_interval;
+	unsigned int data_blocks;
 	struct snd_pcm_substream *pcm;
 	unsigned int pcm_frames;
 	bool lost;
 
+	buffer = s->buffer.packets[s->packet_index].buffer;
 	cip_header[0] = be32_to_cpu(buffer[0]);
 	cip_header[1] = be32_to_cpu(buffer[1]);
 
+	trace_in_packet(s, cycle, cip_header, payload_quadlets, index);
+
 	/*
 	 * This module supports 'Two-quadlet CIP header with SYT field'.
 	 * For convenience, also check FMT field is AM824 or not.
@@ -460,7 +476,7 @@ static int handle_in_packet(struct amdtp_stream *s,
 		dev_info_ratelimited(&s->unit->device,
 				"Invalid CIP header for AMDTP: %08X:%08X\n",
 				cip_header[0], cip_header[1]);
-		*data_blocks = 0;
+		data_blocks = 0;
 		pcm_frames = 0;
 		goto end;
 	}
@@ -471,7 +487,7 @@ static int handle_in_packet(struct amdtp_stream *s,
 		dev_info_ratelimited(&s->unit->device,
 				     "Detect unexpected protocol: %08x %08x\n",
 				     cip_header[0], cip_header[1]);
-		*data_blocks = 0;
+		data_blocks = 0;
 		pcm_frames = 0;
 		goto end;
 	}
@@ -480,7 +496,7 @@ static int handle_in_packet(struct amdtp_stream *s,
 	fdf = (cip_header[1] & CIP_FDF_MASK) >> CIP_FDF_SHIFT;
 	if (payload_quadlets < 3 ||
 	    (fmt == CIP_FMT_AM && fdf == AMDTP_FDF_NO_DATA)) {
-		*data_blocks = 0;
+		data_blocks = 0;
 	} else {
 		data_block_quadlets =
 			(cip_header[0] & CIP_DBS_MASK) >> CIP_DBS_SHIFT;
@@ -494,12 +510,12 @@ static int handle_in_packet(struct amdtp_stream *s,
 		if (s->flags & CIP_WRONG_DBS)
 			data_block_quadlets = s->data_block_quadlets;
 
-		*data_blocks = (payload_quadlets - 2) / data_block_quadlets;
+		data_blocks = (payload_quadlets - 2) / data_block_quadlets;
 	}
 
 	/* Check data block counter continuity */
 	data_block_counter = cip_header[0] & CIP_DBC_MASK;
-	if (*data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) &&
+	if (data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) &&
 	    s->data_block_counter != UINT_MAX)
 		data_block_counter = s->data_block_counter;
 
@@ -510,10 +526,10 @@ static int handle_in_packet(struct amdtp_stream *s,
 	} else if (!(s->flags & CIP_DBC_IS_END_EVENT)) {
 		lost = data_block_counter != s->data_block_counter;
 	} else {
-		if ((*data_blocks > 0) && (s->tx_dbc_interval > 0))
+		if (data_blocks > 0 && s->tx_dbc_interval > 0)
 			dbc_interval = s->tx_dbc_interval;
 		else
-			dbc_interval = *data_blocks;
+			dbc_interval = data_blocks;
 
 		lost = data_block_counter !=
 		       ((s->data_block_counter + dbc_interval) & 0xff);
@@ -526,13 +542,14 @@ static int handle_in_packet(struct amdtp_stream *s,
 		return -EIO;
 	}
 
-	pcm_frames = s->process_data_blocks(s, buffer + 2, *data_blocks, &syt);
+	syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK;
+	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;
 	else
 		s->data_block_counter =
-				(data_block_counter + *data_blocks) & 0xff;
+				(data_block_counter + data_blocks) & 0xff;
 end:
 	if (queue_in_packet(s) < 0)
 		return -EIO;
@@ -544,29 +561,50 @@ end:
 	return 0;
 }
 
-static void out_stream_callback(struct fw_iso_context *context, u32 cycle,
+/*
+ * In CYCLE_TIMER register of IEEE 1394, 7 bits are used to represent second. On
+ * the other hand, in DMA descriptors of 1394 OHCI, 3 bits are used to represent
+ * it. Thus, via Linux firewire subsystem, we can get the 3 bits for second.
+ */
+static inline u32 compute_cycle_count(u32 tstamp)
+{
+	return (((tstamp >> 13) & 0x07) * 8000) + (tstamp & 0x1fff);
+}
+
+static inline u32 increment_cycle_count(u32 cycle, unsigned int addend)
+{
+	cycle += addend;
+	if (cycle >= 8 * CYCLES_PER_SECOND)
+		cycle -= 8 * CYCLES_PER_SECOND;
+	return cycle;
+}
+
+static inline u32 decrement_cycle_count(u32 cycle, unsigned int subtrahend)
+{
+	if (cycle < subtrahend)
+		cycle += 8 * CYCLES_PER_SECOND;
+	return cycle - subtrahend;
+}
+
+static void out_stream_callback(struct fw_iso_context *context, u32 tstamp,
 				size_t header_length, void *header,
 				void *private_data)
 {
 	struct amdtp_stream *s = private_data;
-	unsigned int i, syt, packets = header_length / 4;
-	unsigned int data_blocks;
+	unsigned int i, packets = header_length / 4;
+	u32 cycle;
 
 	if (s->packet_index < 0)
 		return;
 
-	/*
-	 * Compute the cycle of the last queued packet.
-	 * (We need only the four lowest bits for the SYT, so we can ignore
-	 * that bits 0-11 must wrap around at 3072.)
-	 */
-	cycle += QUEUE_LENGTH - packets;
+	cycle = compute_cycle_count(tstamp);
 
-	for (i = 0; i < packets; ++i) {
-		syt = calculate_syt(s, ++cycle);
-		data_blocks = calculate_data_blocks(s, syt);
+	/* Align to actual cycle count for the last packet. */
+	cycle = increment_cycle_count(cycle, QUEUE_LENGTH - packets);
 
-		if (handle_out_packet(s, data_blocks, syt) < 0) {
+	for (i = 0; i < packets; ++i) {
+		cycle = increment_cycle_count(cycle, 1);
+		if (handle_out_packet(s, cycle, i) < 0) {
 			s->packet_index = -1;
 			amdtp_stream_pcm_abort(s);
 			return;
@@ -576,15 +614,15 @@ static void out_stream_callback(struct fw_iso_context *context, u32 cycle,
 	fw_iso_context_queue_flush(s->context);
 }
 
-static void in_stream_callback(struct fw_iso_context *context, u32 cycle,
+static void in_stream_callback(struct fw_iso_context *context, u32 tstamp,
 			       size_t header_length, void *header,
 			       void *private_data)
 {
 	struct amdtp_stream *s = private_data;
-	unsigned int p, syt, packets;
+	unsigned int i, packets;
 	unsigned int payload_quadlets, max_payload_quadlets;
-	unsigned int data_blocks;
-	__be32 *buffer, *headers = header;
+	__be32 *headers = header;
+	u32 cycle;
 
 	if (s->packet_index < 0)
 		return;
@@ -592,70 +630,44 @@ static void in_stream_callback(struct fw_iso_context *context, u32 cycle,
 	/* The number of packets in buffer */
 	packets = header_length / IN_PACKET_HEADER_SIZE;
 
+	cycle = compute_cycle_count(tstamp);
+
+	/* Align to actual cycle count for the last packet. */
+	cycle = decrement_cycle_count(cycle, packets);
+
 	/* For buffer-over-run prevention. */
 	max_payload_quadlets = amdtp_stream_get_max_payload(s) / 4;
 
-	for (p = 0; p < packets; p++) {
-		buffer = s->buffer.packets[s->packet_index].buffer;
+	for (i = 0; i < packets; i++) {
+		cycle = increment_cycle_count(cycle, 1);
 
 		/* The number of quadlets in this packet */
 		payload_quadlets =
-			(be32_to_cpu(headers[p]) >> ISO_DATA_LENGTH_SHIFT) / 4;
+			(be32_to_cpu(headers[i]) >> ISO_DATA_LENGTH_SHIFT) / 4;
 		if (payload_quadlets > max_payload_quadlets) {
 			dev_err(&s->unit->device,
 				"Detect jumbo payload: %02x %02x\n",
 				payload_quadlets, max_payload_quadlets);
-			s->packet_index = -1;
 			break;
 		}
 
-		syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK;
-		if (handle_in_packet(s, payload_quadlets, buffer,
-						&data_blocks, syt) < 0) {
-			s->packet_index = -1;
+		if (handle_in_packet(s, payload_quadlets, cycle, i) < 0)
 			break;
-		}
-
-		/* Process sync slave stream */
-		if (s->sync_slave && s->sync_slave->callbacked) {
-			if (handle_out_packet(s->sync_slave,
-					      data_blocks, syt) < 0) {
-				s->packet_index = -1;
-				break;
-			}
-		}
 	}
 
-	/* Queueing error or detecting discontinuity */
-	if (s->packet_index < 0) {
+	/* Queueing error or detecting invalid payload. */
+	if (i < packets) {
+		s->packet_index = -1;
 		amdtp_stream_pcm_abort(s);
-
-		/* Abort sync slave. */
-		if (s->sync_slave) {
-			s->sync_slave->packet_index = -1;
-			amdtp_stream_pcm_abort(s->sync_slave);
-		}
 		return;
 	}
 
-	/* when sync to device, flush the packets for slave stream */
-	if (s->sync_slave && s->sync_slave->callbacked)
-		fw_iso_context_queue_flush(s->sync_slave->context);
-
 	fw_iso_context_queue_flush(s->context);
 }
 
-/* processing is done by master callback */
-static void slave_stream_callback(struct fw_iso_context *context, u32 cycle,
-				  size_t header_length, void *header,
-				  void *private_data)
-{
-	return;
-}
-
 /* this is executed one time */
 static void amdtp_stream_first_callback(struct fw_iso_context *context,
-					u32 cycle, size_t header_length,
+					u32 tstamp, size_t header_length,
 					void *header, void *private_data)
 {
 	struct amdtp_stream *s = private_data;
@@ -669,12 +681,10 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context,
 
 	if (s->direction == AMDTP_IN_STREAM)
 		context->callback.sc = in_stream_callback;
-	else if (s->flags & CIP_SYNC_TO_DEVICE)
-		context->callback.sc = slave_stream_callback;
 	else
 		context->callback.sc = out_stream_callback;
 
-	context->callback.sc(context, cycle, header_length, header, s);
+	context->callback.sc(context, tstamp, header_length, header, s);
 }
 
 /**
@@ -713,8 +723,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed)
 		goto err_unlock;
 	}
 
-	if (s->direction == AMDTP_IN_STREAM &&
-	    s->flags & CIP_SKIP_INIT_DBC_CHECK)
+	if (s->direction == AMDTP_IN_STREAM)
 		s->data_block_counter = UINT_MAX;
 	else
 		s->data_block_counter = 0;
@@ -755,7 +764,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed)
 		if (s->direction == AMDTP_IN_STREAM)
 			err = queue_in_packet(s);
 		else
-			err = queue_out_packet(s, 0, true);
+			err = queue_out_packet(s, 0);
 		if (err < 0)
 			goto err_context;
 	} while (s->packet_index > 0);
@@ -794,11 +803,24 @@ EXPORT_SYMBOL(amdtp_stream_start);
  */
 unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s)
 {
-	/* this optimization is allowed to be racy */
-	if (s->pointer_flush && amdtp_stream_running(s))
+	/*
+	 * This function is called in software IRQ context of period_tasklet or
+	 * process context.
+	 *
+	 * When the software IRQ context was scheduled by software IRQ context
+	 * of IR/IT contexts, queued packets were already handled. Therefore,
+	 * no need to flush the queue in buffer anymore.
+	 *
+	 * When the process context reach here, some packets will be already
+	 * queued in the buffer. These packets should be handled immediately
+	 * to keep better granularity of PCM pointer.
+	 *
+	 * Later, the process context will sometimes schedules software IRQ
+	 * context of the period_tasklet. Then, no need to flush the queue by
+	 * the same reason as described for IR/IT contexts.
+	 */
+	if (!in_interrupt() && amdtp_stream_running(s))
 		fw_iso_context_flush_completions(s->context);
-	else
-		s->pointer_flush = true;
 
 	return ACCESS_ONCE(s->pcm_buffer_pointer);
 }
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 8775704a3..c1bc7fad0 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -17,8 +17,6 @@
  * @CIP_BLOCKING: In blocking mode, each packet contains either zero or
  *	SYT_INTERVAL samples, with these two types alternating so that
  *	the overall sample rate comes out right.
- * @CIP_SYNC_TO_DEVICE: In sync to device mode, time stamp in out packets is
- *	generated by in packets. Defaultly this driver generates timestamp.
  * @CIP_EMPTY_WITH_TAG0: Only for in-stream. Empty in-packets have TAG0.
  * @CIP_DBC_IS_END_EVENT: Only for in-stream. The value of dbc in an in-packet
  *	corresponds to the end of event in the packet. Out of IEC 61883.
@@ -26,8 +24,6 @@
  *	The value of data_block_quadlets is used instead of reported value.
  * @CIP_SKIP_DBC_ZERO_CHECK: Only for in-stream.  Packets with zero in dbc is
  *	skipped for detecting discontinuity.
- * @CIP_SKIP_INIT_DBC_CHECK: Only for in-stream. The value of dbc in first
- *	packet is not continuous from an initial value.
  * @CIP_EMPTY_HAS_WRONG_DBC: Only for in-stream. The value of dbc in empty
  *	packet is wrong but the others are correct.
  * @CIP_JUMBO_PAYLOAD: Only for in-stream. The number of data blocks in an
@@ -37,14 +33,12 @@
 enum cip_flags {
 	CIP_NONBLOCKING		= 0x00,
 	CIP_BLOCKING		= 0x01,
-	CIP_SYNC_TO_DEVICE	= 0x02,
-	CIP_EMPTY_WITH_TAG0	= 0x04,
-	CIP_DBC_IS_END_EVENT	= 0x08,
-	CIP_WRONG_DBS		= 0x10,
-	CIP_SKIP_DBC_ZERO_CHECK	= 0x20,
-	CIP_SKIP_INIT_DBC_CHECK	= 0x40,
-	CIP_EMPTY_HAS_WRONG_DBC	= 0x80,
-	CIP_JUMBO_PAYLOAD	= 0x100,
+	CIP_EMPTY_WITH_TAG0	= 0x02,
+	CIP_DBC_IS_END_EVENT	= 0x04,
+	CIP_WRONG_DBS		= 0x08,
+	CIP_SKIP_DBC_ZERO_CHECK	= 0x10,
+	CIP_EMPTY_HAS_WRONG_DBC	= 0x20,
+	CIP_JUMBO_PAYLOAD	= 0x40,
 };
 
 /**
@@ -132,12 +126,10 @@ struct amdtp_stream {
 	struct tasklet_struct period_tasklet;
 	unsigned int pcm_buffer_pointer;
 	unsigned int pcm_period_pointer;
-	bool pointer_flush;
 
 	/* 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;
@@ -223,23 +215,6 @@ static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc)
 	return sfc & 1;
 }
 
-static inline void amdtp_stream_set_sync(enum cip_flags sync_mode,
-					 struct amdtp_stream *master,
-					 struct amdtp_stream *slave)
-{
-	if (sync_mode == CIP_SYNC_TO_DEVICE) {
-		master->flags |= CIP_SYNC_TO_DEVICE;
-		slave->flags |= CIP_SYNC_TO_DEVICE;
-		master->sync_slave = slave;
-	} else {
-		master->flags &= ~CIP_SYNC_TO_DEVICE;
-		slave->flags &= ~CIP_SYNC_TO_DEVICE;
-		master->sync_slave = NULL;
-	}
-
-	slave->sync_slave = NULL;
-}
-
 /**
  * amdtp_stream_wait_callback - sleep till callbacked or timeout
  * @s: the AMDTP stream
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 3e4e0756e..f7e2cbd2a 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -67,7 +67,7 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS);
 #define MODEL_MAUDIO_PROJECTMIX		0x00010091
 
 static int
-name_device(struct snd_bebob *bebob, unsigned int vendor_id)
+name_device(struct snd_bebob *bebob)
 {
 	struct fw_device *fw_dev = fw_parent_device(bebob->unit);
 	char vendor[24] = {0};
@@ -126,6 +126,17 @@ end:
 	return err;
 }
 
+static void bebob_free(struct snd_bebob *bebob)
+{
+	snd_bebob_stream_destroy_duplex(bebob);
+	fw_unit_put(bebob->unit);
+
+	kfree(bebob->maudio_special_quirk);
+
+	mutex_destroy(&bebob->mutex);
+	kfree(bebob);
+}
+
 /*
  * This module releases the FireWire unit data after all ALSA character devices
  * are released by applications. This is for releasing stream data or finishing
@@ -137,18 +148,11 @@ bebob_card_free(struct snd_card *card)
 {
 	struct snd_bebob *bebob = card->private_data;
 
-	snd_bebob_stream_destroy_duplex(bebob);
-	fw_unit_put(bebob->unit);
-
-	kfree(bebob->maudio_special_quirk);
-
-	if (bebob->card_index >= 0) {
-		mutex_lock(&devices_mutex);
-		clear_bit(bebob->card_index, devices_used);
-		mutex_unlock(&devices_mutex);
-	}
+	mutex_lock(&devices_mutex);
+	clear_bit(bebob->card_index, devices_used);
+	mutex_unlock(&devices_mutex);
 
-	mutex_destroy(&bebob->mutex);
+	bebob_free(card->private_data);
 }
 
 static const struct snd_bebob_spec *
@@ -176,16 +180,17 @@ check_audiophile_booted(struct fw_unit *unit)
 	return strncmp(name, "FW Audiophile Bootloader", 15) != 0;
 }
 
-static int
-bebob_probe(struct fw_unit *unit,
-	    const struct ieee1394_device_id *entry)
+static void
+do_registration(struct work_struct *work)
 {
-	struct snd_card *card;
-	struct snd_bebob *bebob;
-	const struct snd_bebob_spec *spec;
+	struct snd_bebob *bebob =
+			container_of(work, struct snd_bebob, dwork.work);
 	unsigned int card_index;
 	int err;
 
+	if (bebob->registered)
+		return;
+
 	mutex_lock(&devices_mutex);
 
 	for (card_index = 0; card_index < SNDRV_CARDS; card_index++) {
@@ -193,64 +198,39 @@ bebob_probe(struct fw_unit *unit,
 			break;
 	}
 	if (card_index >= SNDRV_CARDS) {
-		err = -ENOENT;
-		goto end;
+		mutex_unlock(&devices_mutex);
+		return;
 	}
 
-	if ((entry->vendor_id == VEN_FOCUSRITE) &&
-	    (entry->model_id == MODEL_FOCUSRITE_SAFFIRE_BOTH))
-		spec = get_saffire_spec(unit);
-	else if ((entry->vendor_id == VEN_MAUDIO1) &&
-		 (entry->model_id == MODEL_MAUDIO_AUDIOPHILE_BOTH) &&
-		 !check_audiophile_booted(unit))
-		spec = NULL;
-	else
-		spec = (const struct snd_bebob_spec *)entry->driver_data;
-
-	if (spec == NULL) {
-		if ((entry->vendor_id == VEN_MAUDIO1) ||
-		    (entry->vendor_id == VEN_MAUDIO2))
-			err = snd_bebob_maudio_load_firmware(unit);
-		else
-			err = -ENOSYS;
-		goto end;
+	err = snd_card_new(&bebob->unit->device, index[card_index],
+			   id[card_index], THIS_MODULE, 0, &bebob->card);
+	if (err < 0) {
+		mutex_unlock(&devices_mutex);
+		return;
 	}
 
-	err = snd_card_new(&unit->device, index[card_index], id[card_index],
-			   THIS_MODULE, sizeof(struct snd_bebob), &card);
+	err = name_device(bebob);
 	if (err < 0)
-		goto end;
-	bebob = card->private_data;
-	bebob->card_index = card_index;
-	set_bit(card_index, devices_used);
-	card->private_free = bebob_card_free;
-
-	bebob->card = card;
-	bebob->unit = fw_unit_get(unit);
-	bebob->spec = spec;
-	mutex_init(&bebob->mutex);
-	spin_lock_init(&bebob->lock);
-	init_waitqueue_head(&bebob->hwdep_wait);
+		goto error;
 
-	err = name_device(bebob, entry->vendor_id);
+	if (bebob->spec == &maudio_special_spec) {
+		if (bebob->entry->model_id == MODEL_MAUDIO_FW1814)
+			err = snd_bebob_maudio_special_discover(bebob, true);
+		else
+			err = snd_bebob_maudio_special_discover(bebob, false);
+	} else {
+		err = snd_bebob_stream_discover(bebob);
+	}
 	if (err < 0)
 		goto error;
 
-	if ((entry->vendor_id == VEN_MAUDIO1) &&
-	    (entry->model_id == MODEL_MAUDIO_FW1814))
-		err = snd_bebob_maudio_special_discover(bebob, true);
-	else if ((entry->vendor_id == VEN_MAUDIO1) &&
-		 (entry->model_id == MODEL_MAUDIO_PROJECTMIX))
-		err = snd_bebob_maudio_special_discover(bebob, false);
-	else
-		err = snd_bebob_stream_discover(bebob);
+	err = snd_bebob_stream_init_duplex(bebob);
 	if (err < 0)
 		goto error;
 
 	snd_bebob_proc_init(bebob);
 
-	if ((bebob->midi_input_ports > 0) ||
-	    (bebob->midi_output_ports > 0)) {
+	if (bebob->midi_input_ports > 0 || bebob->midi_output_ports > 0) {
 		err = snd_bebob_create_midi_devices(bebob);
 		if (err < 0)
 			goto error;
@@ -264,16 +244,75 @@ bebob_probe(struct fw_unit *unit,
 	if (err < 0)
 		goto error;
 
-	err = snd_bebob_stream_init_duplex(bebob);
+	err = snd_card_register(bebob->card);
 	if (err < 0)
 		goto error;
 
-	if (!bebob->maudio_special_quirk) {
-		err = snd_card_register(card);
-		if (err < 0) {
-			snd_bebob_stream_destroy_duplex(bebob);
-			goto error;
-		}
+	set_bit(card_index, devices_used);
+	mutex_unlock(&devices_mutex);
+
+	/*
+	 * After registered, bebob instance can be released corresponding to
+	 * releasing the sound card instance.
+	 */
+	bebob->card->private_free = bebob_card_free;
+	bebob->card->private_data = bebob;
+	bebob->registered = true;
+
+	return;
+error:
+	mutex_unlock(&devices_mutex);
+	snd_bebob_stream_destroy_duplex(bebob);
+	snd_card_free(bebob->card);
+	dev_info(&bebob->unit->device,
+		 "Sound card registration failed: %d\n", err);
+}
+
+static int
+bebob_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry)
+{
+	struct snd_bebob *bebob;
+	const struct snd_bebob_spec *spec;
+
+	if (entry->vendor_id == VEN_FOCUSRITE &&
+	    entry->model_id == MODEL_FOCUSRITE_SAFFIRE_BOTH)
+		spec = get_saffire_spec(unit);
+	else if (entry->vendor_id == VEN_MAUDIO1 &&
+		 entry->model_id == MODEL_MAUDIO_AUDIOPHILE_BOTH &&
+		 !check_audiophile_booted(unit))
+		spec = NULL;
+	else
+		spec = (const struct snd_bebob_spec *)entry->driver_data;
+
+	if (spec == NULL) {
+		if (entry->vendor_id == VEN_MAUDIO1 ||
+		    entry->vendor_id == VEN_MAUDIO2)
+			return snd_bebob_maudio_load_firmware(unit);
+		else
+			return -ENODEV;
+	}
+
+	/* Allocate this independent of sound card instance. */
+	bebob = kzalloc(sizeof(struct snd_bebob), GFP_KERNEL);
+	if (bebob == NULL)
+		return -ENOMEM;
+
+	bebob->unit = fw_unit_get(unit);
+	bebob->entry = entry;
+	bebob->spec = spec;
+	dev_set_drvdata(&unit->device, bebob);
+
+	mutex_init(&bebob->mutex);
+	spin_lock_init(&bebob->lock);
+	init_waitqueue_head(&bebob->hwdep_wait);
+
+	/* Allocate and register this sound card later. */
+	INIT_DEFERRABLE_WORK(&bebob->dwork, do_registration);
+
+	if (entry->vendor_id != VEN_MAUDIO1 ||
+	    (entry->model_id != MODEL_MAUDIO_FW1814 &&
+	     entry->model_id != MODEL_MAUDIO_PROJECTMIX)) {
+		snd_fw_schedule_registration(unit, &bebob->dwork);
 	} else {
 		/*
 		 * This is a workaround. This bus reset seems to have an effect
@@ -285,19 +324,11 @@ bebob_probe(struct fw_unit *unit,
 		 * signals from dbus and starts I/Os. To avoid I/Os till the
 		 * future bus reset, registration is done in next update().
 		 */
-		bebob->deferred_registration = true;
 		fw_schedule_bus_reset(fw_parent_device(bebob->unit)->card,
 				      false, true);
 	}
 
-	dev_set_drvdata(&unit->device, bebob);
-end:
-	mutex_unlock(&devices_mutex);
-	return err;
-error:
-	mutex_unlock(&devices_mutex);
-	snd_card_free(card);
-	return err;
+	return 0;
 }
 
 /*
@@ -324,15 +355,11 @@ bebob_update(struct fw_unit *unit)
 	if (bebob == NULL)
 		return;
 
-	fcp_bus_reset(bebob->unit);
-
-	if (bebob->deferred_registration) {
-		if (snd_card_register(bebob->card) < 0) {
-			snd_bebob_stream_destroy_duplex(bebob);
-			snd_card_free(bebob->card);
-		}
-		bebob->deferred_registration = false;
-	}
+	/* Postpone a workqueue for deferred registration. */
+	if (!bebob->registered)
+		snd_fw_schedule_registration(unit, &bebob->dwork);
+	else
+		fcp_bus_reset(bebob->unit);
 }
 
 static void bebob_remove(struct fw_unit *unit)
@@ -342,8 +369,20 @@ static void bebob_remove(struct fw_unit *unit)
 	if (bebob == NULL)
 		return;
 
-	/* No need to wait for releasing card object in this context. */
-	snd_card_free_when_closed(bebob->card);
+	/*
+	 * Confirm to stop the work for registration before the sound card is
+	 * going to be released. The work is not scheduled again because bus
+	 * reset handler is not called anymore.
+	 */
+	cancel_delayed_work_sync(&bebob->dwork);
+
+	if (bebob->registered) {
+		/* No need to wait for releasing card object in this context. */
+		snd_card_free_when_closed(bebob->card);
+	} else {
+		/* Don't forget this case. */
+		bebob_free(bebob);
+	}
 }
 
 static const struct snd_bebob_rate_spec normal_rate_spec = {
diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h
index b50bb33d9..e7f1bb925 100644
--- a/sound/firewire/bebob/bebob.h
+++ b/sound/firewire/bebob/bebob.h
@@ -83,6 +83,10 @@ struct snd_bebob {
 	struct mutex mutex;
 	spinlock_t lock;
 
+	bool registered;
+	struct delayed_work dwork;
+
+	const struct ieee1394_device_id *entry;
 	const struct snd_bebob_spec *spec;
 
 	unsigned int midi_input_ports;
@@ -90,7 +94,6 @@ struct snd_bebob {
 
 	bool connected;
 
-	struct amdtp_stream *master;
 	struct amdtp_stream tx_stream;
 	struct amdtp_stream rx_stream;
 	struct cmp_connection out_conn;
@@ -111,7 +114,6 @@ struct snd_bebob {
 
 	/* for M-Audio special devices */
 	void *maudio_special_quirk;
-	bool deferred_registration;
 
 	/* For BeBoB version quirk. */
 	unsigned int version;
diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c
index 77cbb02bf..4d3034a68 100644
--- a/sound/firewire/bebob/bebob_stream.c
+++ b/sound/firewire/bebob/bebob_stream.c
@@ -484,30 +484,6 @@ destroy_both_connections(struct snd_bebob *bebob)
 }
 
 static int
-get_sync_mode(struct snd_bebob *bebob, enum cip_flags *sync_mode)
-{
-	enum snd_bebob_clock_type src;
-	int err;
-
-	err = snd_bebob_stream_get_clock_src(bebob, &src);
-	if (err < 0)
-		return err;
-
-	switch (src) {
-	case SND_BEBOB_CLOCK_TYPE_INTERNAL:
-	case SND_BEBOB_CLOCK_TYPE_EXTERNAL:
-		*sync_mode = CIP_SYNC_TO_DEVICE;
-		break;
-	default:
-	case SND_BEBOB_CLOCK_TYPE_SYT:
-		*sync_mode = 0;
-		break;
-	}
-
-	return 0;
-}
-
-static int
 start_stream(struct snd_bebob *bebob, struct amdtp_stream *stream,
 	     unsigned int rate)
 {
@@ -550,8 +526,6 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob)
 		goto end;
 	}
 
-	bebob->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK;
-
 	/*
 	 * BeBoB v3 transfers packets with these qurks:
 	 *  - In the beginning of streaming, the value of dbc is incremented
@@ -584,8 +558,6 @@ end:
 int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int 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;
 	int err = 0;
 
@@ -593,22 +565,11 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate)
 	if (bebob->substreams_counter == 0)
 		goto end;
 
-	err = get_sync_mode(bebob, &sync_mode);
-	if (err < 0)
-		goto end;
-	if (sync_mode == CIP_SYNC_TO_DEVICE) {
-		master = &bebob->tx_stream;
-		slave  = &bebob->rx_stream;
-	} else {
-		master = &bebob->rx_stream;
-		slave  = &bebob->tx_stream;
-	}
-
 	/*
 	 * Considering JACK/FFADO streaming:
 	 * TODO: This can be removed hwdep functionality becomes popular.
 	 */
-	err = check_connection_used_by_others(bebob, master);
+	err = check_connection_used_by_others(bebob, &bebob->rx_stream);
 	if (err < 0)
 		goto end;
 
@@ -618,11 +579,12 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate)
 	 * At bus reset, connections should not be broken here. So streams need
 	 * to be re-started. This is a reason to use SKIP_INIT_DBC_CHECK flag.
 	 */
-	if (amdtp_streaming_error(master))
-		amdtp_stream_stop(master);
-	if (amdtp_streaming_error(slave))
-		amdtp_stream_stop(slave);
-	if (!amdtp_stream_running(master) && !amdtp_stream_running(slave))
+	if (amdtp_streaming_error(&bebob->rx_stream))
+		amdtp_stream_stop(&bebob->rx_stream);
+	if (amdtp_streaming_error(&bebob->tx_stream))
+		amdtp_stream_stop(&bebob->tx_stream);
+	if (!amdtp_stream_running(&bebob->rx_stream) &&
+	    !amdtp_stream_running(&bebob->tx_stream))
 		break_both_connections(bebob);
 
 	/* stop streams if rate is different */
@@ -635,16 +597,13 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate)
 	if (rate == 0)
 		rate = curr_rate;
 	if (rate != curr_rate) {
-		amdtp_stream_stop(master);
-		amdtp_stream_stop(slave);
+		amdtp_stream_stop(&bebob->rx_stream);
+		amdtp_stream_stop(&bebob->tx_stream);
 		break_both_connections(bebob);
 	}
 
 	/* master should be always running */
-	if (!amdtp_stream_running(master)) {
-		amdtp_stream_set_sync(sync_mode, master, slave);
-		bebob->master = master;
-
+	if (!amdtp_stream_running(&bebob->rx_stream)) {
 		/*
 		 * NOTE:
 		 * If establishing connections at first, Yamaha GO46
@@ -666,7 +625,7 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate)
 		if (err < 0)
 			goto end;
 
-		err = start_stream(bebob, master, rate);
+		err = start_stream(bebob, &bebob->rx_stream, rate);
 		if (err < 0) {
 			dev_err(&bebob->unit->device,
 				"fail to run AMDTP master stream:%d\n", err);
@@ -685,15 +644,16 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate)
 				dev_err(&bebob->unit->device,
 					"fail to ensure sampling rate: %d\n",
 					err);
-				amdtp_stream_stop(master);
+				amdtp_stream_stop(&bebob->rx_stream);
 				break_both_connections(bebob);
 				goto end;
 			}
 		}
 
 		/* wait first callback */
-		if (!amdtp_stream_wait_callback(master, CALLBACK_TIMEOUT)) {
-			amdtp_stream_stop(master);
+		if (!amdtp_stream_wait_callback(&bebob->rx_stream,
+						CALLBACK_TIMEOUT)) {
+			amdtp_stream_stop(&bebob->rx_stream);
 			break_both_connections(bebob);
 			err = -ETIMEDOUT;
 			goto end;
@@ -701,20 +661,21 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate)
 	}
 
 	/* start slave if needed */
-	if (!amdtp_stream_running(slave)) {
-		err = start_stream(bebob, slave, rate);
+	if (!amdtp_stream_running(&bebob->tx_stream)) {
+		err = start_stream(bebob, &bebob->tx_stream, rate);
 		if (err < 0) {
 			dev_err(&bebob->unit->device,
 				"fail to run AMDTP slave stream:%d\n", err);
-			amdtp_stream_stop(master);
+			amdtp_stream_stop(&bebob->rx_stream);
 			break_both_connections(bebob);
 			goto end;
 		}
 
 		/* wait first callback */
-		if (!amdtp_stream_wait_callback(slave, CALLBACK_TIMEOUT)) {
-			amdtp_stream_stop(slave);
-			amdtp_stream_stop(master);
+		if (!amdtp_stream_wait_callback(&bebob->tx_stream,
+						CALLBACK_TIMEOUT)) {
+			amdtp_stream_stop(&bebob->tx_stream);
+			amdtp_stream_stop(&bebob->rx_stream);
 			break_both_connections(bebob);
 			err = -ETIMEDOUT;
 		}
@@ -725,22 +686,12 @@ end:
 
 void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob)
 {
-	struct amdtp_stream *master, *slave;
-
-	if (bebob->master == &bebob->rx_stream) {
-		slave  = &bebob->tx_stream;
-		master = &bebob->rx_stream;
-	} else {
-		slave  = &bebob->rx_stream;
-		master = &bebob->tx_stream;
-	}
-
 	if (bebob->substreams_counter == 0) {
-		amdtp_stream_pcm_abort(master);
-		amdtp_stream_stop(master);
+		amdtp_stream_pcm_abort(&bebob->rx_stream);
+		amdtp_stream_stop(&bebob->rx_stream);
 
-		amdtp_stream_pcm_abort(slave);
-		amdtp_stream_stop(slave);
+		amdtp_stream_pcm_abort(&bebob->tx_stream);
+		amdtp_stream_stop(&bebob->tx_stream);
 
 		break_both_connections(bebob);
 	}
diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 8b64aef31..25e9f7727 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -20,8 +20,6 @@ MODULE_LICENSE("GPL v2");
 #define WEISS_CATEGORY_ID	0x00
 #define LOUD_CATEGORY_ID	0x10
 
-#define PROBE_DELAY_MS		(2 * MSEC_PER_SEC)
-
 /*
  * Some models support several isochronous channels, while these streams are not
  * always available. In this case, add the model name to this list.
@@ -201,6 +199,10 @@ static void do_registration(struct work_struct *work)
 
 	dice_card_strings(dice);
 
+	err = snd_dice_stream_init_duplex(dice);
+	if (err < 0)
+		goto error;
+
 	snd_dice_create_proc(dice);
 
 	err = snd_dice_create_pcm(dice);
@@ -229,28 +231,14 @@ static void do_registration(struct work_struct *work)
 
 	return;
 error:
+	snd_dice_stream_destroy_duplex(dice);
 	snd_dice_transaction_destroy(dice);
+	snd_dice_stream_destroy_duplex(dice);
 	snd_card_free(dice->card);
 	dev_info(&dice->unit->device,
 		 "Sound card registration failed: %d\n", err);
 }
 
-static void schedule_registration(struct snd_dice *dice)
-{
-	struct fw_card *fw_card = fw_parent_device(dice->unit)->card;
-	u64 now, delay;
-
-	now = get_jiffies_64();
-	delay = fw_card->reset_jiffies + msecs_to_jiffies(PROBE_DELAY_MS);
-
-	if (time_after64(delay, now))
-		delay -= now;
-	else
-		delay = 0;
-
-	mod_delayed_work(system_wq, &dice->dwork, delay);
-}
-
 static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id)
 {
 	struct snd_dice *dice;
@@ -273,15 +261,9 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id)
 	init_completion(&dice->clock_accepted);
 	init_waitqueue_head(&dice->hwdep_wait);
 
-	err = snd_dice_stream_init_duplex(dice);
-	if (err < 0) {
-		dice_free(dice);
-		return err;
-	}
-
 	/* Allocate and register this sound card later. */
 	INIT_DEFERRABLE_WORK(&dice->dwork, do_registration);
-	schedule_registration(dice);
+	snd_fw_schedule_registration(unit, &dice->dwork);
 
 	return 0;
 }
@@ -312,7 +294,7 @@ static void dice_bus_reset(struct fw_unit *unit)
 
 	/* Postpone a workqueue for deferred registration. */
 	if (!dice->registered)
-		schedule_registration(dice);
+		snd_fw_schedule_registration(unit, &dice->dwork);
 
 	/* The handler address register becomes initialized. */
 	snd_dice_transaction_reinit(dice);
@@ -335,6 +317,13 @@ static const struct ieee1394_device_id dice_id_table[] = {
 		.match_flags = IEEE1394_MATCH_VERSION,
 		.version     = DICE_INTERFACE,
 	},
+	/* M-Audio Profire 610/2626 has a different value in version field. */
+	{
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_SPECIFIER_ID,
+		.vendor_id	= 0x000d6c,
+		.specifier_id	= 0x000d6c,
+	},
 	{ }
 };
 MODULE_DEVICE_TABLE(ieee1394, dice_id_table);
diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c
index 0ac92aba5..b3cffd01a 100644
--- a/sound/firewire/digi00x/amdtp-dot.c
+++ b/sound/firewire/digi00x/amdtp-dot.c
@@ -421,7 +421,7 @@ int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit,
 
 	/* Use different mode between incoming/outgoing. */
 	if (dir == AMDTP_IN_STREAM) {
-		flags = CIP_NONBLOCKING | CIP_SKIP_INIT_DBC_CHECK;
+		flags = CIP_NONBLOCKING;
 		process_data_blocks = process_tx_data_blocks;
 	} else {
 		flags = CIP_BLOCKING;
diff --git a/sound/firewire/digi00x/digi00x-transaction.c b/sound/firewire/digi00x/digi00x-transaction.c
index 554324d8c..735d35640 100644
--- a/sound/firewire/digi00x/digi00x-transaction.c
+++ b/sound/firewire/digi00x/digi00x-transaction.c
@@ -126,12 +126,17 @@ int snd_dg00x_transaction_register(struct snd_dg00x *dg00x)
 	return err;
 error:
 	fw_core_remove_address_handler(&dg00x->async_handler);
-	dg00x->async_handler.address_callback = NULL;
+	dg00x->async_handler.callback_data = NULL;
 	return err;
 }
 
 void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x)
 {
+	if (dg00x->async_handler.callback_data == NULL)
+		return;
+
 	snd_fw_async_midi_port_destroy(&dg00x->out_control);
 	fw_core_remove_address_handler(&dg00x->async_handler);
+
+	dg00x->async_handler.callback_data = NULL;
 }
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
index 1f33b7a1f..cc4776c6d 100644
--- a/sound/firewire/digi00x/digi00x.c
+++ b/sound/firewire/digi00x/digi00x.c
@@ -40,10 +40,8 @@ static int name_card(struct snd_dg00x *dg00x)
 	return 0;
 }
 
-static void dg00x_card_free(struct snd_card *card)
+static void dg00x_free(struct snd_dg00x *dg00x)
 {
-	struct snd_dg00x *dg00x = card->private_data;
-
 	snd_dg00x_stream_destroy_duplex(dg00x);
 	snd_dg00x_transaction_unregister(dg00x);
 
@@ -52,28 +50,24 @@ static void dg00x_card_free(struct snd_card *card)
 	mutex_destroy(&dg00x->mutex);
 }
 
-static int snd_dg00x_probe(struct fw_unit *unit,
-			   const struct ieee1394_device_id *entry)
+static void dg00x_card_free(struct snd_card *card)
 {
-	struct snd_card *card;
-	struct snd_dg00x *dg00x;
-	int err;
+	dg00x_free(card->private_data);
+}
 
-	/* 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;
+static void do_registration(struct work_struct *work)
+{
+	struct snd_dg00x *dg00x =
+			container_of(work, struct snd_dg00x, dwork.work);
+	int err;
 
-	/* initialize myself */
-	dg00x = card->private_data;
-	dg00x->card = card;
-	dg00x->unit = fw_unit_get(unit);
+	if (dg00x->registered)
+		return;
 
-	mutex_init(&dg00x->mutex);
-	spin_lock_init(&dg00x->lock);
-	init_waitqueue_head(&dg00x->hwdep_wait);
+	err = snd_card_new(&dg00x->unit->device, -1, NULL, THIS_MODULE, 0,
+			   &dg00x->card);
+	if (err < 0)
+		return;
 
 	err = name_card(dg00x);
 	if (err < 0)
@@ -101,35 +95,86 @@ static int snd_dg00x_probe(struct fw_unit *unit,
 	if (err < 0)
 		goto error;
 
-	err = snd_card_register(card);
+	err = snd_card_register(dg00x->card);
 	if (err < 0)
 		goto error;
 
-	dev_set_drvdata(&unit->device, dg00x);
+	dg00x->card->private_free = dg00x_card_free;
+	dg00x->card->private_data = dg00x;
+	dg00x->registered = true;
 
-	return err;
+	return;
 error:
-	snd_card_free(card);
-	return err;
+	snd_dg00x_transaction_unregister(dg00x);
+	snd_dg00x_stream_destroy_duplex(dg00x);
+	snd_card_free(dg00x->card);
+	dev_info(&dg00x->unit->device,
+		 "Sound card registration failed: %d\n", err);
+}
+
+static int snd_dg00x_probe(struct fw_unit *unit,
+			   const struct ieee1394_device_id *entry)
+{
+	struct snd_dg00x *dg00x;
+
+	/* Allocate this independent of sound card instance. */
+	dg00x = kzalloc(sizeof(struct snd_dg00x), GFP_KERNEL);
+	if (dg00x == NULL)
+		return -ENOMEM;
+
+	dg00x->unit = fw_unit_get(unit);
+	dev_set_drvdata(&unit->device, dg00x);
+
+	mutex_init(&dg00x->mutex);
+	spin_lock_init(&dg00x->lock);
+	init_waitqueue_head(&dg00x->hwdep_wait);
+
+	/* Allocate and register this sound card later. */
+	INIT_DEFERRABLE_WORK(&dg00x->dwork, do_registration);
+	snd_fw_schedule_registration(unit, &dg00x->dwork);
+
+	return 0;
 }
 
 static void snd_dg00x_update(struct fw_unit *unit)
 {
 	struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device);
 
+	/* Postpone a workqueue for deferred registration. */
+	if (!dg00x->registered)
+		snd_fw_schedule_registration(unit, &dg00x->dwork);
+
 	snd_dg00x_transaction_reregister(dg00x);
 
-	mutex_lock(&dg00x->mutex);
-	snd_dg00x_stream_update_duplex(dg00x);
-	mutex_unlock(&dg00x->mutex);
+	/*
+	 * After registration, userspace can start packet streaming, then this
+	 * code block works fine.
+	 */
+	if (dg00x->registered) {
+		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);
+	/*
+	 * Confirm to stop the work for registration before the sound card is
+	 * going to be released. The work is not scheduled again because bus
+	 * reset handler is not called anymore.
+	 */
+	cancel_delayed_work_sync(&dg00x->dwork);
+
+	if (dg00x->registered) {
+		/* No need to wait for releasing card object in this context. */
+		snd_card_free_when_closed(dg00x->card);
+	} else {
+		/* Don't forget this case. */
+		dg00x_free(dg00x);
+	}
 }
 
 static const struct ieee1394_device_id snd_dg00x_id_table[] = {
diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h
index 907e73993..2cd465c0c 100644
--- a/sound/firewire/digi00x/digi00x.h
+++ b/sound/firewire/digi00x/digi00x.h
@@ -37,6 +37,9 @@ struct snd_dg00x {
 	struct mutex mutex;
 	spinlock_t lock;
 
+	bool registered;
+	struct delayed_work dwork;
+
 	struct amdtp_stream tx_stream;
 	struct fw_iso_resources tx_resources;
 
diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c
index 8f27b6750..71a0613d3 100644
--- a/sound/firewire/fireworks/fireworks.c
+++ b/sound/firewire/fireworks/fireworks.c
@@ -168,11 +168,34 @@ get_hardware_info(struct snd_efw *efw)
 	       sizeof(struct snd_efw_phys_grp) * hwinfo->phys_in_grp_count);
 	memcpy(&efw->phys_out_grps, hwinfo->phys_out_grps,
 	       sizeof(struct snd_efw_phys_grp) * hwinfo->phys_out_grp_count);
+
+	/* AudioFire8 (since 2009) and AudioFirePre8 */
+	if (hwinfo->type == MODEL_ECHO_AUDIOFIRE_9)
+		efw->is_af9 = true;
+	/* These models uses the same firmware. */
+	if (hwinfo->type == MODEL_ECHO_AUDIOFIRE_2 ||
+	    hwinfo->type == MODEL_ECHO_AUDIOFIRE_4 ||
+	    hwinfo->type == MODEL_ECHO_AUDIOFIRE_9 ||
+	    hwinfo->type == MODEL_GIBSON_RIP ||
+	    hwinfo->type == MODEL_GIBSON_GOLDTOP)
+		efw->is_fireworks3 = true;
 end:
 	kfree(hwinfo);
 	return err;
 }
 
+static void efw_free(struct snd_efw *efw)
+{
+	snd_efw_stream_destroy_duplex(efw);
+	snd_efw_transaction_remove_instance(efw);
+	fw_unit_put(efw->unit);
+
+	kfree(efw->resp_buf);
+
+	mutex_destroy(&efw->mutex);
+	kfree(efw);
+}
+
 /*
  * This module releases the FireWire unit data after all ALSA character devices
  * are released by applications. This is for releasing stream data or finishing
@@ -184,28 +207,24 @@ efw_card_free(struct snd_card *card)
 {
 	struct snd_efw *efw = card->private_data;
 
-	snd_efw_stream_destroy_duplex(efw);
-	snd_efw_transaction_remove_instance(efw);
-	fw_unit_put(efw->unit);
-
-	kfree(efw->resp_buf);
-
 	if (efw->card_index >= 0) {
 		mutex_lock(&devices_mutex);
 		clear_bit(efw->card_index, devices_used);
 		mutex_unlock(&devices_mutex);
 	}
 
-	mutex_destroy(&efw->mutex);
+	efw_free(card->private_data);
 }
 
-static int
-efw_probe(struct fw_unit *unit,
-	  const struct ieee1394_device_id *entry)
+static void
+do_registration(struct work_struct *work)
 {
-	struct snd_card *card;
-	struct snd_efw *efw;
-	int card_index, err;
+	struct snd_efw *efw = container_of(work, struct snd_efw, dwork.work);
+	unsigned int card_index;
+	int err;
+
+	if (efw->registered)
+		return;
 
 	mutex_lock(&devices_mutex);
 
@@ -215,24 +234,16 @@ efw_probe(struct fw_unit *unit,
 			break;
 	}
 	if (card_index >= SNDRV_CARDS) {
-		err = -ENOENT;
-		goto end;
+		mutex_unlock(&devices_mutex);
+		return;
 	}
 
-	err = snd_card_new(&unit->device, index[card_index], id[card_index],
-			   THIS_MODULE, sizeof(struct snd_efw), &card);
-	if (err < 0)
-		goto end;
-	efw = card->private_data;
-	efw->card_index = card_index;
-	set_bit(card_index, devices_used);
-	card->private_free = efw_card_free;
-
-	efw->card = card;
-	efw->unit = fw_unit_get(unit);
-	mutex_init(&efw->mutex);
-	spin_lock_init(&efw->lock);
-	init_waitqueue_head(&efw->hwdep_wait);
+	err = snd_card_new(&efw->unit->device, index[card_index],
+			   id[card_index], THIS_MODULE, 0, &efw->card);
+	if (err < 0) {
+		mutex_unlock(&devices_mutex);
+		return;
+	}
 
 	/* prepare response buffer */
 	snd_efw_resp_buf_size = clamp(snd_efw_resp_buf_size,
@@ -248,16 +259,10 @@ efw_probe(struct fw_unit *unit,
 	err = get_hardware_info(efw);
 	if (err < 0)
 		goto error;
-	/* AudioFire8 (since 2009) and AudioFirePre8 */
-	if (entry->model_id == MODEL_ECHO_AUDIOFIRE_9)
-		efw->is_af9 = true;
-	/* These models uses the same firmware. */
-	if (entry->model_id == MODEL_ECHO_AUDIOFIRE_2 ||
-	    entry->model_id == MODEL_ECHO_AUDIOFIRE_4 ||
-	    entry->model_id == MODEL_ECHO_AUDIOFIRE_9 ||
-	    entry->model_id == MODEL_GIBSON_RIP ||
-	    entry->model_id == MODEL_GIBSON_GOLDTOP)
-		efw->is_fireworks3 = true;
+
+	err = snd_efw_stream_init_duplex(efw);
+	if (err < 0)
+		goto error;
 
 	snd_efw_proc_init(efw);
 
@@ -275,44 +280,93 @@ efw_probe(struct fw_unit *unit,
 	if (err < 0)
 		goto error;
 
-	err = snd_efw_stream_init_duplex(efw);
+	err = snd_card_register(efw->card);
 	if (err < 0)
 		goto error;
 
-	err = snd_card_register(card);
-	if (err < 0) {
-		snd_efw_stream_destroy_duplex(efw);
-		goto error;
-	}
-
-	dev_set_drvdata(&unit->device, efw);
-end:
+	set_bit(card_index, devices_used);
 	mutex_unlock(&devices_mutex);
-	return err;
+
+	/*
+	 * After registered, efw instance can be released corresponding to
+	 * releasing the sound card instance.
+	 */
+	efw->card->private_free = efw_card_free;
+	efw->card->private_data = efw;
+	efw->registered = true;
+
+	return;
 error:
-	snd_efw_transaction_remove_instance(efw);
 	mutex_unlock(&devices_mutex);
-	snd_card_free(card);
-	return err;
+	snd_efw_transaction_remove_instance(efw);
+	snd_efw_stream_destroy_duplex(efw);
+	snd_card_free(efw->card);
+	dev_info(&efw->unit->device,
+		 "Sound card registration failed: %d\n", err);
+}
+
+static int
+efw_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry)
+{
+	struct snd_efw *efw;
+
+	efw = kzalloc(sizeof(struct snd_efw), GFP_KERNEL);
+	if (efw == NULL)
+		return -ENOMEM;
+
+	efw->unit = fw_unit_get(unit);
+	dev_set_drvdata(&unit->device, efw);
+
+	mutex_init(&efw->mutex);
+	spin_lock_init(&efw->lock);
+	init_waitqueue_head(&efw->hwdep_wait);
+
+	/* Allocate and register this sound card later. */
+	INIT_DEFERRABLE_WORK(&efw->dwork, do_registration);
+	snd_fw_schedule_registration(unit, &efw->dwork);
+
+	return 0;
 }
 
 static void efw_update(struct fw_unit *unit)
 {
 	struct snd_efw *efw = dev_get_drvdata(&unit->device);
 
+	/* Postpone a workqueue for deferred registration. */
+	if (!efw->registered)
+		snd_fw_schedule_registration(unit, &efw->dwork);
+
 	snd_efw_transaction_bus_reset(efw->unit);
 
-	mutex_lock(&efw->mutex);
-	snd_efw_stream_update_duplex(efw);
-	mutex_unlock(&efw->mutex);
+	/*
+	 * After registration, userspace can start packet streaming, then this
+	 * code block works fine.
+	 */
+	if (efw->registered) {
+		mutex_lock(&efw->mutex);
+		snd_efw_stream_update_duplex(efw);
+		mutex_unlock(&efw->mutex);
+	}
 }
 
 static void efw_remove(struct fw_unit *unit)
 {
 	struct snd_efw *efw = dev_get_drvdata(&unit->device);
 
-	/* No need to wait for releasing card object in this context. */
-	snd_card_free_when_closed(efw->card);
+	/*
+	 * Confirm to stop the work for registration before the sound card is
+	 * going to be released. The work is not scheduled again because bus
+	 * reset handler is not called anymore.
+	 */
+	cancel_delayed_work_sync(&efw->dwork);
+
+	if (efw->registered) {
+		/* No need to wait for releasing card object in this context. */
+		snd_card_free_when_closed(efw->card);
+	} else {
+		/* Don't forget this case. */
+		efw_free(efw);
+	}
 }
 
 static const struct ieee1394_device_id efw_id_table[] = {
diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h
index 96c4e0c6a..03ed35237 100644
--- a/sound/firewire/fireworks/fireworks.h
+++ b/sound/firewire/fireworks/fireworks.h
@@ -65,6 +65,9 @@ struct snd_efw {
 	struct mutex mutex;
 	spinlock_t lock;
 
+	bool registered;
+	struct delayed_work dwork;
+
 	/* for transaction */
 	u32 seqnum;
 	bool resp_addr_changable;
@@ -81,7 +84,6 @@ struct snd_efw {
 	unsigned int pcm_capture_channels[SND_EFW_MULTIPLIER_MODES];
 	unsigned int pcm_playback_channels[SND_EFW_MULTIPLIER_MODES];
 
-	struct amdtp_stream *master;
 	struct amdtp_stream tx_stream;
 	struct amdtp_stream rx_stream;
 	struct cmp_connection out_conn;
diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c
index 425db8d88..ee47924ae 100644
--- a/sound/firewire/fireworks/fireworks_stream.c
+++ b/sound/firewire/fireworks/fireworks_stream.c
@@ -121,23 +121,6 @@ destroy_stream(struct snd_efw *efw, struct amdtp_stream *stream)
 }
 
 static int
-get_sync_mode(struct snd_efw *efw, enum cip_flags *sync_mode)
-{
-	enum snd_efw_clock_source clock_source;
-	int err;
-
-	err = snd_efw_command_get_clock_source(efw, &clock_source);
-	if (err < 0)
-		return err;
-
-	if (clock_source == SND_EFW_CLOCK_SOURCE_SYTMATCH)
-		return -ENOSYS;
-
-	*sync_mode = CIP_SYNC_TO_DEVICE;
-	return 0;
-}
-
-static int
 check_connection_used_by_others(struct snd_efw *efw, struct amdtp_stream *s)
 {
 	struct cmp_connection *conn;
@@ -208,9 +191,6 @@ end:
 
 int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate)
 {
-	struct amdtp_stream *master, *slave;
-	unsigned int slave_substreams;
-	enum cip_flags sync_mode;
 	unsigned int curr_rate;
 	int err = 0;
 
@@ -218,32 +198,19 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate)
 	if (efw->playback_substreams == 0 && efw->capture_substreams  == 0)
 		goto end;
 
-	err = get_sync_mode(efw, &sync_mode);
-	if (err < 0)
-		goto end;
-	if (sync_mode == CIP_SYNC_TO_DEVICE) {
-		master = &efw->tx_stream;
-		slave  = &efw->rx_stream;
-		slave_substreams  = efw->playback_substreams;
-	} else {
-		master = &efw->rx_stream;
-		slave  = &efw->tx_stream;
-		slave_substreams = efw->capture_substreams;
-	}
-
 	/*
 	 * Considering JACK/FFADO streaming:
 	 * TODO: This can be removed hwdep functionality becomes popular.
 	 */
-	err = check_connection_used_by_others(efw, master);
+	err = check_connection_used_by_others(efw, &efw->rx_stream);
 	if (err < 0)
 		goto end;
 
 	/* packet queueing error */
-	if (amdtp_streaming_error(slave))
-		stop_stream(efw, slave);
-	if (amdtp_streaming_error(master))
-		stop_stream(efw, master);
+	if (amdtp_streaming_error(&efw->tx_stream))
+		stop_stream(efw, &efw->tx_stream);
+	if (amdtp_streaming_error(&efw->rx_stream))
+		stop_stream(efw, &efw->rx_stream);
 
 	/* stop streams if rate is different */
 	err = snd_efw_command_get_sampling_rate(efw, &curr_rate);
@@ -252,20 +219,17 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate)
 	if (rate == 0)
 		rate = curr_rate;
 	if (rate != curr_rate) {
-		stop_stream(efw, slave);
-		stop_stream(efw, master);
+		stop_stream(efw, &efw->tx_stream);
+		stop_stream(efw, &efw->rx_stream);
 	}
 
 	/* master should be always running */
-	if (!amdtp_stream_running(master)) {
-		amdtp_stream_set_sync(sync_mode, master, slave);
-		efw->master = master;
-
+	if (!amdtp_stream_running(&efw->rx_stream)) {
 		err = snd_efw_command_set_sampling_rate(efw, rate);
 		if (err < 0)
 			goto end;
 
-		err = start_stream(efw, master, rate);
+		err = start_stream(efw, &efw->rx_stream, rate);
 		if (err < 0) {
 			dev_err(&efw->unit->device,
 				"fail to start AMDTP master stream:%d\n", err);
@@ -274,12 +238,13 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate)
 	}
 
 	/* start slave if needed */
-	if (slave_substreams > 0 && !amdtp_stream_running(slave)) {
-		err = start_stream(efw, slave, rate);
+	if (efw->capture_substreams > 0 &&
+	    !amdtp_stream_running(&efw->tx_stream)) {
+		err = start_stream(efw, &efw->tx_stream, rate);
 		if (err < 0) {
 			dev_err(&efw->unit->device,
 				"fail to start AMDTP slave stream:%d\n", err);
-			stop_stream(efw, master);
+			stop_stream(efw, &efw->rx_stream);
 		}
 	}
 end:
@@ -288,26 +253,11 @@ end:
 
 void snd_efw_stream_stop_duplex(struct snd_efw *efw)
 {
-	struct amdtp_stream *master, *slave;
-	unsigned int master_substreams, slave_substreams;
-
-	if (efw->master == &efw->rx_stream) {
-		slave  = &efw->tx_stream;
-		master = &efw->rx_stream;
-		slave_substreams  = efw->capture_substreams;
-		master_substreams = efw->playback_substreams;
-	} else {
-		slave  = &efw->rx_stream;
-		master = &efw->tx_stream;
-		slave_substreams  = efw->playback_substreams;
-		master_substreams = efw->capture_substreams;
-	}
-
-	if (slave_substreams == 0) {
-		stop_stream(efw, slave);
+	if (efw->capture_substreams == 0) {
+		stop_stream(efw, &efw->tx_stream);
 
-		if (master_substreams == 0)
-			stop_stream(efw, master);
+		if (efw->playback_substreams == 0)
+			stop_stream(efw, &efw->rx_stream);
 	}
 }
 
diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c
index f80aafa44..ca4dfcf43 100644
--- a/sound/firewire/lib.c
+++ b/sound/firewire/lib.c
@@ -67,6 +67,38 @@ int snd_fw_transaction(struct fw_unit *unit, int tcode,
 }
 EXPORT_SYMBOL(snd_fw_transaction);
 
+#define PROBE_DELAY_MS		(2 * MSEC_PER_SEC)
+
+/**
+ * snd_fw_schedule_registration - schedule work for sound card registration
+ * @unit: an instance for unit on IEEE 1394 bus
+ * @dwork: delayed work with callback function
+ *
+ * This function is not designed for general purposes. When new unit is
+ * connected to IEEE 1394 bus, the bus is under bus-reset state because of
+ * topological change. In this state, units tend to fail both of asynchronous
+ * and isochronous communication. To avoid this problem, this function is used
+ * to postpone sound card registration after the state. The callers must
+ * set up instance of delayed work in advance.
+ */
+void snd_fw_schedule_registration(struct fw_unit *unit,
+				  struct delayed_work *dwork)
+{
+	u64 now, delay;
+
+	now = get_jiffies_64();
+	delay = fw_parent_device(unit)->card->reset_jiffies
+					+ msecs_to_jiffies(PROBE_DELAY_MS);
+
+	if (time_after64(delay, now))
+		delay -= now;
+	else
+		delay = 0;
+
+	mod_delayed_work(system_wq, dwork, delay);
+}
+EXPORT_SYMBOL(snd_fw_schedule_registration);
+
 static void async_midi_port_callback(struct fw_card *card, int rcode,
 				     void *data, size_t length,
 				     void *callback_data)
diff --git a/sound/firewire/lib.h b/sound/firewire/lib.h
index f3f6f84c4..f6769312e 100644
--- a/sound/firewire/lib.h
+++ b/sound/firewire/lib.h
@@ -22,6 +22,9 @@ static inline bool rcode_is_permanent_error(int rcode)
 	return rcode == RCODE_TYPE_ERROR || rcode == RCODE_ADDRESS_ERROR;
 }
 
+void snd_fw_schedule_registration(struct fw_unit *unit,
+				  struct delayed_work *dwork);
+
 struct snd_fw_async_midi_port;
 typedef int (*snd_fw_async_midi_port_fill)(
 				struct snd_rawmidi_substream *substream,
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index 7cb5743c0..d9361f352 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -242,8 +242,7 @@ int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw,
 	 * blocks than IEC 61883-6 defines.
 	 */
 	if (stream == &oxfw->tx_stream) {
-		oxfw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK |
-					 CIP_JUMBO_PAYLOAD;
+		oxfw->tx_stream.flags |= CIP_JUMBO_PAYLOAD;
 		if (oxfw->wrong_dbs)
 			oxfw->tx_stream.flags |= CIP_WRONG_DBS;
 	}
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index abedc2207..e629b88f7 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -118,15 +118,8 @@ end:
 	return err;
 }
 
-/*
- * This module releases the FireWire unit data after all ALSA character devices
- * are released by applications. This is for releasing stream data or finishing
- * transactions safely. Thus at returning from .remove(), this module still keep
- * references for the unit.
- */
-static void oxfw_card_free(struct snd_card *card)
+static void oxfw_free(struct snd_oxfw *oxfw)
 {
-	struct snd_oxfw *oxfw = card->private_data;
 	unsigned int i;
 
 	snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream);
@@ -144,6 +137,17 @@ static void oxfw_card_free(struct snd_card *card)
 	mutex_destroy(&oxfw->mutex);
 }
 
+/*
+ * This module releases the FireWire unit data after all ALSA character devices
+ * are released by applications. This is for releasing stream data or finishing
+ * transactions safely. Thus at returning from .remove(), this module still keep
+ * references for the unit.
+ */
+static void oxfw_card_free(struct snd_card *card)
+{
+	oxfw_free(card->private_data);
+}
+
 static int detect_quirks(struct snd_oxfw *oxfw)
 {
 	struct fw_device *fw_dev = fw_parent_device(oxfw->unit);
@@ -205,41 +209,39 @@ static int detect_quirks(struct snd_oxfw *oxfw)
 	return 0;
 }
 
-static int oxfw_probe(struct fw_unit *unit,
-		      const struct ieee1394_device_id *entry)
+static void do_registration(struct work_struct *work)
 {
-	struct snd_card *card;
-	struct snd_oxfw *oxfw;
+	struct snd_oxfw *oxfw = container_of(work, struct snd_oxfw, dwork.work);
 	int err;
 
-	if (entry->vendor_id == VENDOR_LOUD && !detect_loud_models(unit))
-		return -ENODEV;
+	if (oxfw->registered)
+		return;
 
-	err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE,
-			   sizeof(*oxfw), &card);
+	err = snd_card_new(&oxfw->unit->device, -1, NULL, THIS_MODULE, 0,
+			   &oxfw->card);
 	if (err < 0)
-		return err;
+		return;
 
-	card->private_free = oxfw_card_free;
-	oxfw = card->private_data;
-	oxfw->card = card;
-	mutex_init(&oxfw->mutex);
-	oxfw->unit = fw_unit_get(unit);
-	oxfw->entry = entry;
-	spin_lock_init(&oxfw->lock);
-	init_waitqueue_head(&oxfw->hwdep_wait);
+	err = name_card(oxfw);
+	if (err < 0)
+		goto error;
 
-	err = snd_oxfw_stream_discover(oxfw);
+	err = detect_quirks(oxfw);
 	if (err < 0)
 		goto error;
 
-	err = name_card(oxfw);
+	err = snd_oxfw_stream_discover(oxfw);
 	if (err < 0)
 		goto error;
 
-	err = detect_quirks(oxfw);
+	err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->rx_stream);
 	if (err < 0)
 		goto error;
+	if (oxfw->has_output) {
+		err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->tx_stream);
+		if (err < 0)
+			goto error;
+	}
 
 	err = snd_oxfw_create_pcm(oxfw);
 	if (err < 0)
@@ -255,54 +257,97 @@ static int oxfw_probe(struct fw_unit *unit,
 	if (err < 0)
 		goto error;
 
-	err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->rx_stream);
+	err = snd_card_register(oxfw->card);
 	if (err < 0)
 		goto error;
-	if (oxfw->has_output) {
-		err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->tx_stream);
-		if (err < 0)
-			goto error;
-	}
 
-	err = snd_card_register(card);
-	if (err < 0) {
-		snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream);
-		if (oxfw->has_output)
-			snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream);
-		goto error;
-	}
+	/*
+	 * After registered, oxfw instance can be released corresponding to
+	 * releasing the sound card instance.
+	 */
+	oxfw->card->private_free = oxfw_card_free;
+	oxfw->card->private_data = oxfw;
+	oxfw->registered = true;
+
+	return;
+error:
+	snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream);
+	if (oxfw->has_output)
+		snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream);
+	snd_card_free(oxfw->card);
+	dev_info(&oxfw->unit->device,
+		 "Sound card registration failed: %d\n", err);
+}
+
+static int oxfw_probe(struct fw_unit *unit,
+		      const struct ieee1394_device_id *entry)
+{
+	struct snd_oxfw *oxfw;
+
+	if (entry->vendor_id == VENDOR_LOUD && !detect_loud_models(unit))
+		return -ENODEV;
+
+	/* Allocate this independent of sound card instance. */
+	oxfw = kzalloc(sizeof(struct snd_oxfw), GFP_KERNEL);
+	if (oxfw == NULL)
+		return -ENOMEM;
+
+	oxfw->entry = entry;
+	oxfw->unit = fw_unit_get(unit);
 	dev_set_drvdata(&unit->device, oxfw);
 
+	mutex_init(&oxfw->mutex);
+	spin_lock_init(&oxfw->lock);
+	init_waitqueue_head(&oxfw->hwdep_wait);
+
+	/* Allocate and register this sound card later. */
+	INIT_DEFERRABLE_WORK(&oxfw->dwork, do_registration);
+	snd_fw_schedule_registration(unit, &oxfw->dwork);
+
 	return 0;
-error:
-	snd_card_free(card);
-	return err;
 }
 
 static void oxfw_bus_reset(struct fw_unit *unit)
 {
 	struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device);
 
+	if (!oxfw->registered)
+		snd_fw_schedule_registration(unit, &oxfw->dwork);
+
 	fcp_bus_reset(oxfw->unit);
 
-	mutex_lock(&oxfw->mutex);
+	if (oxfw->registered) {
+		mutex_lock(&oxfw->mutex);
 
-	snd_oxfw_stream_update_simplex(oxfw, &oxfw->rx_stream);
-	if (oxfw->has_output)
-		snd_oxfw_stream_update_simplex(oxfw, &oxfw->tx_stream);
+		snd_oxfw_stream_update_simplex(oxfw, &oxfw->rx_stream);
+		if (oxfw->has_output)
+			snd_oxfw_stream_update_simplex(oxfw, &oxfw->tx_stream);
 
-	mutex_unlock(&oxfw->mutex);
+		mutex_unlock(&oxfw->mutex);
 
-	if (oxfw->entry->vendor_id == OUI_STANTON)
-		snd_oxfw_scs1x_update(oxfw);
+		if (oxfw->entry->vendor_id == OUI_STANTON)
+			snd_oxfw_scs1x_update(oxfw);
+	}
 }
 
 static void oxfw_remove(struct fw_unit *unit)
 {
 	struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device);
 
-	/* No need to wait for releasing card object in this context. */
-	snd_card_free_when_closed(oxfw->card);
+	/*
+	 * Confirm to stop the work for registration before the sound card is
+	 * going to be released. The work is not scheduled again because bus
+	 * reset handler is not called anymore.
+	 */
+	cancel_delayed_work_sync(&oxfw->dwork);
+
+	if (oxfw->registered) {
+		/* No need to wait for releasing card object in this context. */
+		snd_card_free_when_closed(oxfw->card);
+	} else {
+		/* Don't forget this case. */
+		oxfw_free(oxfw);
+	}
 }
 
 static const struct compat_info griffin_firewave = {
diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h
index 9beecc214..2047dcb27 100644
--- a/sound/firewire/oxfw/oxfw.h
+++ b/sound/firewire/oxfw/oxfw.h
@@ -36,10 +36,12 @@
 struct snd_oxfw {
 	struct snd_card *card;
 	struct fw_unit *unit;
-	const struct device_info *device_info;
 	struct mutex mutex;
 	spinlock_t lock;
 
+	bool registered;
+	struct delayed_work dwork;
+
 	bool wrong_dbs;
 	bool has_output;
 	u8 *tx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES];
diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c
index 0e6dd5c61..4ad3bd7fd 100644
--- a/sound/firewire/tascam/tascam-stream.c
+++ b/sound/firewire/tascam/tascam-stream.c
@@ -381,19 +381,17 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate)
 	if (err < 0)
 		return err;
 	if (curr_rate != rate ||
-	    amdtp_streaming_error(&tscm->tx_stream) ||
-	    amdtp_streaming_error(&tscm->rx_stream)) {
+	    amdtp_streaming_error(&tscm->rx_stream) ||
+	    amdtp_streaming_error(&tscm->tx_stream)) {
 		finish_session(tscm);
 
-		amdtp_stream_stop(&tscm->tx_stream);
 		amdtp_stream_stop(&tscm->rx_stream);
+		amdtp_stream_stop(&tscm->tx_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);
+	if (!amdtp_stream_running(&tscm->rx_stream)) {
 		err = keep_resources(tscm, rate);
 		if (err < 0)
 			goto error;
@@ -406,27 +404,27 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate)
 		if (err < 0)
 			goto error;
 
-		err = amdtp_stream_start(&tscm->tx_stream,
-				tscm->tx_resources.channel,
+		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->tx_stream,
+		if (!amdtp_stream_wait_callback(&tscm->rx_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,
+	if (!amdtp_stream_running(&tscm->tx_stream)) {
+		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->rx_stream,
+		if (!amdtp_stream_wait_callback(&tscm->tx_stream,
 						CALLBACK_TIMEOUT)) {
 			err = -ETIMEDOUT;
 			goto error;
@@ -435,8 +433,8 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate)
 
 	return 0;
 error:
-	amdtp_stream_stop(&tscm->tx_stream);
 	amdtp_stream_stop(&tscm->rx_stream);
+	amdtp_stream_stop(&tscm->tx_stream);
 
 	finish_session(tscm);
 	release_resources(tscm);
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
index e281c338e..9dc93a7eb 100644
--- a/sound/firewire/tascam/tascam.c
+++ b/sound/firewire/tascam/tascam.c
@@ -85,10 +85,8 @@ static int identify_model(struct snd_tscm *tscm)
 	return 0;
 }
 
-static void tscm_card_free(struct snd_card *card)
+static void tscm_free(struct snd_tscm *tscm)
 {
-	struct snd_tscm *tscm = card->private_data;
-
 	snd_tscm_transaction_unregister(tscm);
 	snd_tscm_stream_destroy_duplex(tscm);
 
@@ -97,44 +95,36 @@ static void tscm_card_free(struct snd_card *card)
 	mutex_destroy(&tscm->mutex);
 }
 
-static int snd_tscm_probe(struct fw_unit *unit,
-			   const struct ieee1394_device_id *entry)
+static void tscm_card_free(struct snd_card *card)
 {
-	struct snd_card *card;
-	struct snd_tscm *tscm;
+	tscm_free(card->private_data);
+}
+
+static void do_registration(struct work_struct *work)
+{
+	struct snd_tscm *tscm = container_of(work, struct snd_tscm, dwork.work);
 	int err;
 
-	/* create card */
-	err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE,
-			   sizeof(struct snd_tscm), &card);
+	err = snd_card_new(&tscm->unit->device, -1, NULL, THIS_MODULE, 0,
+			   &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);
+		return;
 
 	err = identify_model(tscm);
 	if (err < 0)
 		goto error;
 
-	snd_tscm_proc_init(tscm);
-
-	err = snd_tscm_stream_init_duplex(tscm);
+	err = snd_tscm_transaction_register(tscm);
 	if (err < 0)
 		goto error;
 
-	err = snd_tscm_create_pcm_devices(tscm);
+	err = snd_tscm_stream_init_duplex(tscm);
 	if (err < 0)
 		goto error;
 
-	err = snd_tscm_transaction_register(tscm);
+	snd_tscm_proc_init(tscm);
+
+	err = snd_tscm_create_pcm_devices(tscm);
 	if (err < 0)
 		goto error;
 
@@ -146,35 +136,91 @@ static int snd_tscm_probe(struct fw_unit *unit,
 	if (err < 0)
 		goto error;
 
-	err = snd_card_register(card);
+	err = snd_card_register(tscm->card);
 	if (err < 0)
 		goto error;
 
-	dev_set_drvdata(&unit->device, tscm);
+	/*
+	 * After registered, tscm instance can be released corresponding to
+	 * releasing the sound card instance.
+	 */
+	tscm->card->private_free = tscm_card_free;
+	tscm->card->private_data = tscm;
+	tscm->registered = true;
 
-	return err;
+	return;
 error:
-	snd_card_free(card);
-	return err;
+	snd_tscm_transaction_unregister(tscm);
+	snd_tscm_stream_destroy_duplex(tscm);
+	snd_card_free(tscm->card);
+	dev_info(&tscm->unit->device,
+		 "Sound card registration failed: %d\n", err);
+}
+
+static int snd_tscm_probe(struct fw_unit *unit,
+			   const struct ieee1394_device_id *entry)
+{
+	struct snd_tscm *tscm;
+
+	/* Allocate this independent of sound card instance. */
+	tscm = kzalloc(sizeof(struct snd_tscm), GFP_KERNEL);
+	if (tscm == NULL)
+		return -ENOMEM;
+
+	/* initialize myself */
+	tscm->unit = fw_unit_get(unit);
+	dev_set_drvdata(&unit->device, tscm);
+
+	mutex_init(&tscm->mutex);
+	spin_lock_init(&tscm->lock);
+	init_waitqueue_head(&tscm->hwdep_wait);
+
+	/* Allocate and register this sound card later. */
+	INIT_DEFERRABLE_WORK(&tscm->dwork, do_registration);
+	snd_fw_schedule_registration(unit, &tscm->dwork);
+
+	return 0;
 }
 
 static void snd_tscm_update(struct fw_unit *unit)
 {
 	struct snd_tscm *tscm = dev_get_drvdata(&unit->device);
 
+	/* Postpone a workqueue for deferred registration. */
+	if (!tscm->registered)
+		snd_fw_schedule_registration(unit, &tscm->dwork);
+
 	snd_tscm_transaction_reregister(tscm);
 
-	mutex_lock(&tscm->mutex);
-	snd_tscm_stream_update_duplex(tscm);
-	mutex_unlock(&tscm->mutex);
+	/*
+	 * After registration, userspace can start packet streaming, then this
+	 * code block works fine.
+	 */
+	if (tscm->registered) {
+		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);
+	/*
+	 * Confirm to stop the work for registration before the sound card is
+	 * going to be released. The work is not scheduled again because bus
+	 * reset handler is not called anymore.
+	 */
+	cancel_delayed_work_sync(&tscm->dwork);
+
+	if (tscm->registered) {
+		/* No need to wait for releasing card object in this context. */
+		snd_card_free_when_closed(tscm->card);
+	} else {
+		/* Don't forget this case. */
+		tscm_free(tscm);
+	}
 }
 
 static const struct ieee1394_device_id snd_tscm_id_table[] = {
diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h
index 30ab77e92..1f6101157 100644
--- a/sound/firewire/tascam/tascam.h
+++ b/sound/firewire/tascam/tascam.h
@@ -51,6 +51,8 @@ struct snd_tscm {
 	struct mutex mutex;
 	spinlock_t lock;
 
+	bool registered;
+	struct delayed_work dwork;
 	const struct snd_tscm_spec *spec;
 
 	struct fw_iso_resources tx_resources;
diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c
index 2433f7c81..31b510c5c 100644
--- a/sound/hda/ext/hdac_ext_bus.c
+++ b/sound/hda/ext/hdac_ext_bus.c
@@ -105,6 +105,9 @@ int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev,
 	INIT_LIST_HEAD(&ebus->hlink_list);
 	ebus->idx = idx++;
 
+	mutex_init(&ebus->lock);
+	ebus->cmd_dma_state = true;
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init);
@@ -144,6 +147,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr)
 	if (!edev)
 		return -ENOMEM;
 	hdev = &edev->hdac;
+	edev->ebus = ebus;
 
 	snprintf(name, sizeof(name), "ehdaudio%dD%d", ebus->idx, addr);
 
diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c
index 548cc1e41..860f8cad6 100644
--- a/sound/hda/ext/hdac_ext_controller.c
+++ b/sound/hda/ext/hdac_ext_controller.c
@@ -186,6 +186,9 @@ int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus)
 		hlink->lcaps  = readl(hlink->ml_addr + AZX_REG_ML_LCAP);
 		hlink->lsdiid = readw(hlink->ml_addr + AZX_REG_ML_LSDIID);
 
+		/* since link in On, update the ref */
+		hlink->ref_count = 1;
+
 		list_add_tail(&hlink->list, &ebus->hlink_list);
 	}
 
@@ -327,3 +330,66 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus)
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all);
+
+int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus,
+				struct hdac_ext_link *link)
+{
+	int ret = 0;
+
+	mutex_lock(&ebus->lock);
+
+	/*
+	 * if we move from 0 to 1, count will be 1 so power up this link
+	 * as well, also check the dma status and trigger that
+	 */
+	if (++link->ref_count == 1) {
+		if (!ebus->cmd_dma_state) {
+			snd_hdac_bus_init_cmd_io(&ebus->bus);
+			ebus->cmd_dma_state = true;
+		}
+
+		ret = snd_hdac_ext_bus_link_power_up(link);
+	}
+
+	mutex_unlock(&ebus->lock);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_get);
+
+int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus,
+				struct hdac_ext_link *link)
+{
+	int ret = 0;
+	struct hdac_ext_link *hlink;
+	bool link_up = false;
+
+	mutex_lock(&ebus->lock);
+
+	/*
+	 * if we move from 1 to 0, count will be 0
+	 * so power down this link as well
+	 */
+	if (--link->ref_count == 0) {
+		ret = snd_hdac_ext_bus_link_power_down(link);
+
+		/*
+		 * now check if all links are off, if so turn off
+		 * cmd dma as well
+		 */
+		list_for_each_entry(hlink, &ebus->hlink_list, list) {
+			if (hlink->ref_count) {
+				link_up = true;
+				break;
+			}
+		}
+
+		if (!link_up) {
+			snd_hdac_bus_stop_cmd_io(&ebus->bus);
+			ebus->cmd_dma_state = false;
+		}
+	}
+
+	mutex_unlock(&ebus->lock);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_put);
diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index 8c486235c..9fee464e5 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -80,6 +80,22 @@ void snd_hdac_bus_init_cmd_io(struct hdac_bus *bus)
 }
 EXPORT_SYMBOL_GPL(snd_hdac_bus_init_cmd_io);
 
+/* wait for cmd dmas till they are stopped */
+static void hdac_wait_for_cmd_dmas(struct hdac_bus *bus)
+{
+	unsigned long timeout;
+
+	timeout = jiffies + msecs_to_jiffies(100);
+	while ((snd_hdac_chip_readb(bus, RIRBCTL) & AZX_RBCTL_DMA_EN)
+		&& time_before(jiffies, timeout))
+		udelay(10);
+
+	timeout = jiffies + msecs_to_jiffies(100);
+	while ((snd_hdac_chip_readb(bus, CORBCTL) & AZX_CORBCTL_RUN)
+		&& time_before(jiffies, timeout))
+		udelay(10);
+}
+
 /**
  * snd_hdac_bus_stop_cmd_io - clean up CORB/RIRB buffers
  * @bus: HD-audio core bus
@@ -90,6 +106,7 @@ void snd_hdac_bus_stop_cmd_io(struct hdac_bus *bus)
 	/* disable ringbuffer DMAs */
 	snd_hdac_chip_writeb(bus, RIRBCTL, 0);
 	snd_hdac_chip_writeb(bus, CORBCTL, 0);
+	hdac_wait_for_cmd_dmas(bus);
 	/* disable unsolicited responses */
 	snd_hdac_chip_updatel(bus, GCTL, AZX_GCTL_UNSOL, 0);
 	spin_unlock_irq(&bus->reg_lock);
diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index 607bbeaeb..c9af02267 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -158,22 +158,40 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus)
 }
 EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk);
 
-/* There is a fixed mapping between audio pin node and display port
- * on current Intel platforms:
+/* There is a fixed mapping between audio pin node and display port.
+ * on SNB, IVY, HSW, BSW, SKL, BXT, KBL:
  * 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)
+ *
+ * on VLV, ILK:
+ * Pin Widget 4 - PORT B (port = 1 in i915 driver)
+ * Pin Widget 5 - PORT C (port = 2 in i915 driver)
+ * Pin Widget 6 - PORT D (port = 3 in i915 driver)
  */
-static int pin2port(hda_nid_t pin_nid)
+static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid)
 {
-	if (WARN_ON(pin_nid < 5 || pin_nid > 7))
+	int base_nid;
+
+	switch (codec->vendor_id) {
+	case 0x80860054: /* ILK */
+	case 0x80862804: /* ILK */
+	case 0x80862882: /* VLV */
+		base_nid = 3;
+		break;
+	default:
+		base_nid = 4;
+		break;
+	}
+
+	if (WARN_ON(pin_nid <= base_nid || pin_nid > base_nid + 3))
 		return -1;
-	return pin_nid - 4;
+	return pin_nid - base_nid;
 }
 
 /**
  * snd_hdac_sync_audio_rate - Set N/CTS based on the sample rate
- * @bus: HDA core bus
+ * @codec: HDA codec
  * @nid: the pin widget NID
  * @rate: the sample rate to set
  *
@@ -183,14 +201,15 @@ static int pin2port(hda_nid_t pin_nid)
  * This function sets N/CTS value based on the given sample rate.
  * Returns zero for success, or a negative error code.
  */
-int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate)
+int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate)
 {
+	struct hdac_bus *bus = codec->bus;
 	struct i915_audio_component *acomp = bus->audio_component;
 	int port;
 
 	if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate)
 		return -ENODEV;
-	port = pin2port(nid);
+	port = pin2port(codec, nid);
 	if (port < 0)
 		return -EINVAL;
 	return acomp->ops->sync_audio_rate(acomp->dev, port, rate);
@@ -199,7 +218,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate);
 
 /**
  * snd_hdac_acomp_get_eld - Get the audio state and ELD via component
- * @bus: HDA core bus
+ * @codec: HDA codec
  * @nid: the pin widget NID
  * @audio_enabled: the pointer to store the current audio state
  * @buffer: the buffer pointer to store ELD bytes
@@ -217,16 +236,17 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate);
  * thus it may be over @max_bytes.  If it's over @max_bytes, it implies
  * that only a part of ELD bytes have been fetched.
  */
-int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid,
+int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid,
 			   bool *audio_enabled, char *buffer, int max_bytes)
 {
+	struct hdac_bus *bus = codec->bus;
 	struct i915_audio_component *acomp = bus->audio_component;
 	int port;
 
 	if (!acomp || !acomp->ops || !acomp->ops->get_eld)
 		return -ENODEV;
 
-	port = pin2port(nid);
+	port = pin2port(codec, nid);
 	if (port < 0)
 		return -EINVAL;
 	return acomp->ops->get_eld(acomp->dev, port, audio_enabled,
@@ -338,6 +358,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus)
 	struct i915_audio_component *acomp;
 	int ret;
 
+	if (WARN_ON(hdac_acomp))
+		return -EBUSY;
+
 	if (!i915_gfx_present())
 		return -ENODEV;
 
@@ -371,6 +394,7 @@ out_master_del:
 out_err:
 	kfree(acomp);
 	bus->audio_component = NULL;
+	hdac_acomp = NULL;
 	dev_info(dev, "failed to add i915 component master (%d)\n", ret);
 
 	return ret;
@@ -404,6 +428,7 @@ int snd_hdac_i915_exit(struct hdac_bus *bus)
 
 	kfree(acomp);
 	bus->audio_component = NULL;
+	hdac_acomp = NULL;
 
 	return 0;
 }
diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c
index d7ec86263..c6c75e7e0 100644
--- a/sound/hda/hdmi_chmap.c
+++ b/sound/hda/hdmi_chmap.c
@@ -625,13 +625,30 @@ static void hdmi_cea_alloc_to_tlv_chmap(struct hdac_chmap *hchmap,
 	WARN_ON(count != channels);
 }
 
+static int spk_mask_from_spk_alloc(int spk_alloc)
+{
+	int i;
+	int spk_mask = eld_speaker_allocation_bits[0];
+
+	for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) {
+		if (spk_alloc & (1 << i))
+			spk_mask |= eld_speaker_allocation_bits[i];
+	}
+
+	return spk_mask;
+}
+
 static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 			      unsigned int size, unsigned int __user *tlv)
 {
 	struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol);
 	struct hdac_chmap *chmap = info->private_data;
+	int pcm_idx = kcontrol->private_value;
 	unsigned int __user *dst;
 	int chs, count = 0;
+	unsigned long max_chs;
+	int type;
+	int spk_alloc, spk_mask;
 
 	if (size < 8)
 		return -ENOMEM;
@@ -639,40 +656,59 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 		return -EFAULT;
 	size -= 8;
 	dst = tlv + 2;
-	for (chs = 2; chs <= chmap->channels_max; chs++) {
+
+	spk_alloc = chmap->ops.get_spk_alloc(chmap->hdac, pcm_idx);
+	spk_mask = spk_mask_from_spk_alloc(spk_alloc);
+
+	max_chs = hweight_long(spk_mask);
+
+	for (chs = 2; chs <= max_chs; chs++) {
 		int i;
 		struct hdac_cea_channel_speaker_allocation *cap;
 
 		cap = channel_allocations;
 		for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) {
 			int chs_bytes = chs * 4;
-			int type = chmap->ops.chmap_cea_alloc_validate_get_type(
-								chmap, cap, chs);
 			unsigned int tlv_chmap[8];
 
-			if (type < 0)
+			if (cap->channels != chs)
+				continue;
+
+			if (!(cap->spk_mask == (spk_mask & cap->spk_mask)))
 				continue;
+
+			type = chmap->ops.chmap_cea_alloc_validate_get_type(
+							chmap, cap, chs);
+			if (type < 0)
+				return -ENODEV;
 			if (size < 8)
 				return -ENOMEM;
+
 			if (put_user(type, dst) ||
 			    put_user(chs_bytes, dst + 1))
 				return -EFAULT;
+
 			dst += 2;
 			size -= 8;
 			count += 8;
+
 			if (size < chs_bytes)
 				return -ENOMEM;
+
 			size -= chs_bytes;
 			count += chs_bytes;
 			chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap,
 						tlv_chmap, chs);
+
 			if (copy_to_user(dst, tlv_chmap, chs_bytes))
 				return -EFAULT;
 			dst += chs;
 		}
 	}
+
 	if (put_user(count, tlv + 1))
 		return -EFAULT;
+
 	return 0;
 }
 
diff --git a/sound/hda/local.h b/sound/hda/local.h
index d692f417d..0d5bb159d 100644
--- a/sound/hda/local.h
+++ b/sound/hda/local.h
@@ -16,6 +16,16 @@ static inline int get_wcaps_type(unsigned int wcaps)
 	return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
 }
 
+static inline unsigned int get_wcaps_channels(u32 wcaps)
+{
+	unsigned int chans;
+
+	chans = (wcaps & AC_WCAP_CHAN_CNT_EXT) >> 13;
+	chans = (chans + 1) * 2;
+
+	return chans;
+}
+
 extern const struct attribute_group *hdac_dev_attr_groups[];
 int hda_widget_sysfs_init(struct hdac_device *codec);
 void hda_widget_sysfs_exit(struct hdac_device *codec);
diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c
index 1000ffa7f..7c9dafd3c 100644
--- a/sound/isa/wavefront/wavefront_synth.c
+++ b/sound/isa/wavefront/wavefront_synth.c
@@ -785,6 +785,9 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header)
 	DPRINT (WF_DEBUG_LOAD_PATCH, "downloading patch %d\n",
 				      header->number);
 
+	if (header->number >= ARRAY_SIZE(dev->patch_status))
+		return -EINVAL;
+
 	dev->patch_status[header->number] |= WF_SLOT_FILLED;
 
 	bptr = buf;
@@ -809,6 +812,9 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header)
 	DPRINT (WF_DEBUG_LOAD_PATCH, "downloading program %d\n",
 		header->number);
 
+	if (header->number >= ARRAY_SIZE(dev->prog_status))
+		return -EINVAL;
+
 	dev->prog_status[header->number] = WF_SLOT_USED;
 
 	/* XXX need to zero existing SLOT_USED bit for program_status[i]
@@ -898,6 +904,9 @@ wavefront_send_sample (snd_wavefront_t *dev,
 		header->number = x;
 	}
 
+	if (header->number >= WF_MAX_SAMPLE)
+		return -EINVAL;
+
 	if (header->size) {
 
 		/* XXX it's a debatable point whether or not RDONLY semantics
diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c
index b36ea4752..0b8d0de87 100644
--- a/sound/oss/waveartist.c
+++ b/sound/oss/waveartist.c
@@ -1414,11 +1414,9 @@ attach_waveartist(struct address_info *hw, const struct waveartist_mixer_info *m
 	else {
 #ifdef CONFIG_ARCH_NETWINDER
 		if (machine_is_netwinder()) {
-			init_timer(&vnc_timer);
-			vnc_timer.function = vnc_slider_tick;
-			vnc_timer.expires  = jiffies;
-			vnc_timer.data     = nr_waveartist_devs;
-			add_timer(&vnc_timer);
+			setup_timer(&vnc_timer, vnc_slider_tick,
+				    nr_waveartist_devs);
+			mod_timer(&vnc_timer, jiffies);
 
 			vnc_configure_mixer(devc, 0);
 
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 741771890..d3125c169 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -2150,8 +2150,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir,
 							   stream->resources, en,
 							   VORTEX_RESOURCE_SRC)) < 0) {
 					memset(stream->resources, 0,
-					       sizeof(unsigned char) *
-					       VORTEX_RESOURCE_LAST);
+					       sizeof(stream->resources));
 					return -EBUSY;
 				}
 				if (stream->type != VORTEX_PCM_A3D) {
@@ -2161,7 +2160,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir,
 								   VORTEX_RESOURCE_MIXIN)) < 0) {
 						memset(stream->resources,
 						       0,
-						       sizeof(unsigned char) * VORTEX_RESOURCE_LAST);
+						       sizeof(stream->resources));
 						return -EBUSY;
 					}
 				}
@@ -2174,8 +2173,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir,
 						   stream->resources, en,
 						   VORTEX_RESOURCE_A3D)) < 0) {
 				memset(stream->resources, 0,
-				       sizeof(unsigned char) *
-				       VORTEX_RESOURCE_LAST);
+				       sizeof(stream->resources));
 				dev_err(vortex->card->dev,
 					"out of A3D sources. Sorry\n");
 				return -EBUSY;
@@ -2289,8 +2287,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir,
 						   VORTEX_RESOURCE_MIXOUT))
 			    < 0) {
 				memset(stream->resources, 0,
-				       sizeof(unsigned char) *
-				       VORTEX_RESOURCE_LAST);
+				       sizeof(stream->resources));
 				return -EBUSY;
 			}
 			if ((src[i] =
@@ -2298,8 +2295,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir,
 						   stream->resources, en,
 						   VORTEX_RESOURCE_SRC)) < 0) {
 				memset(stream->resources, 0,
-				       sizeof(unsigned char) *
-				       VORTEX_RESOURCE_LAST);
+				       sizeof(stream->resources));
 				return -EBUSY;
 			}
 		}
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index a6d6d8d08..df5741a78 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -432,7 +432,10 @@ static snd_pcm_uframes_t snd_vortex_pcm_pointer(struct snd_pcm_substream *substr
 #endif
 	//printk(KERN_INFO "vortex: pointer = 0x%x\n", current_ptr);
 	spin_unlock(&chip->lock);
-	return (bytes_to_frames(substream->runtime, current_ptr));
+	current_ptr = bytes_to_frames(substream->runtime, current_ptr);
+	if (current_ptr >= substream->runtime->buffer_size)
+		current_ptr = 0;
+	return current_ptr;
 }
 
 /* operators */
diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c
index a5d460453..8f9453417 100644
--- a/sound/pci/ctxfi/cttimer.c
+++ b/sound/pci/ctxfi/cttimer.c
@@ -49,7 +49,7 @@ struct ct_timer {
 	spinlock_t lock;		/* global timer lock (for xfitimer) */
 	spinlock_t list_lock;		/* lock for instance list */
 	struct ct_atc *atc;
-	struct ct_timer_ops *ops;
+	const struct ct_timer_ops *ops;
 	struct list_head instance_head;
 	struct list_head running_head;
 	unsigned int wc;		/* current wallclock */
@@ -128,7 +128,7 @@ static void ct_systimer_prepare(struct ct_timer_instance *ti)
 
 #define ct_systimer_free	ct_systimer_prepare
 
-static struct ct_timer_ops ct_systimer_ops = {
+static const struct ct_timer_ops ct_systimer_ops = {
 	.init = ct_systimer_init,
 	.free_instance = ct_systimer_free,
 	.prepare = ct_systimer_prepare,
@@ -322,7 +322,7 @@ static void ct_xfitimer_free_global(struct ct_timer *atimer)
 	ct_xfitimer_irq_stop(atimer);
 }
 
-static struct ct_timer_ops ct_xfitimer_ops = {
+static const struct ct_timer_ops ct_xfitimer_ops = {
 	.prepare = ct_xfitimer_prepare,
 	.start = ct_xfitimer_start,
 	.stop = ct_xfitimer_stop,
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 0dc44ebb0..626cd2167 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -1548,7 +1548,7 @@ static int snd_es1373_line_get(struct snd_kcontrol *kcontrol,
 	int val = 0;
 	
 	spin_lock_irq(&ensoniq->reg_lock);
-	if ((ensoniq->ctrl & ES_1371_GPIO_OUTM) >= 4)
+	if (ensoniq->ctrl & ES_1371_GPIO_OUT(4))
 	    	val = 1;
 	ucontrol->value.integer.value[0] = val;
 	spin_unlock_irq(&ensoniq->reg_lock);
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 4cddac50f..f9a670d35 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -50,9 +50,13 @@ config SND_HDA_RECONFIG
 	bool "Allow dynamic codec reconfiguration"
 	help
 	  Say Y here to enable the HD-audio codec re-configuration feature.
-	  This adds the sysfs interfaces to allow user to clear the whole
-	  codec configuration, change the codec setup, add extra verbs,
-	  and re-configure the codec dynamically.
+	  It allows user to clear the whole codec configuration, change the
+	  codec setup, add extra verbs, and re-configure the codec dynamically.
+
+	  Note that this item alone doesn't provide the sysfs interface, but
+	  enables the feature just for the patch loader below.
+	  If you need the traditional sysfs entries for the manual interaction,
+	  turn on CONFIG_SND_HDA_HWDEP as well.
 
 config SND_HDA_INPUT_BEEP
 	bool "Support digital beep via input layer"
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index d77cc76aa..79c7b340a 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -5436,6 +5436,7 @@ static int dyn_adc_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
 	spec->cur_adc_stream_tag = stream_tag;
 	spec->cur_adc_format = format;
 	snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format);
+	call_pcm_capture_hook(hinfo, codec, substream, HDA_GEN_PCM_ACT_PREPARE);
 	return 0;
 }
 
@@ -5446,6 +5447,7 @@ static int dyn_adc_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 	struct hda_gen_spec *spec = codec->spec;
 	snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
 	spec->cur_adc = 0;
+	call_pcm_capture_hook(hinfo, codec, substream, HDA_GEN_PCM_ACT_CLEANUP);
 	return 0;
 }
 
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 17fd81736..0621920f7 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -115,20 +115,20 @@ static int substream_free_pages(struct azx *chip,
 /*
  * Register access ops. Tegra HDA register access is DWORD only.
  */
-static void hda_tegra_writel(u32 value, u32 *addr)
+static void hda_tegra_writel(u32 value, u32 __iomem *addr)
 {
 	writel(value, addr);
 }
 
-static u32 hda_tegra_readl(u32 *addr)
+static u32 hda_tegra_readl(u32 __iomem *addr)
 {
 	return readl(addr);
 }
 
-static void hda_tegra_writew(u16 value, u16 *addr)
+static void hda_tegra_writew(u16 value, u16 __iomem  *addr)
 {
 	unsigned int shift = ((unsigned long)(addr) & 0x3) << 3;
-	void *dword_addr = (void *)((unsigned long)(addr) & ~0x3);
+	void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3);
 	u32 v;
 
 	v = readl(dword_addr);
@@ -137,20 +137,20 @@ static void hda_tegra_writew(u16 value, u16 *addr)
 	writel(v, dword_addr);
 }
 
-static u16 hda_tegra_readw(u16 *addr)
+static u16 hda_tegra_readw(u16 __iomem *addr)
 {
 	unsigned int shift = ((unsigned long)(addr) & 0x3) << 3;
-	void *dword_addr = (void *)((unsigned long)(addr) & ~0x3);
+	void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3);
 	u32 v;
 
 	v = readl(dword_addr);
 	return (v >> shift) & 0xffff;
 }
 
-static void hda_tegra_writeb(u8 value, u8 *addr)
+static void hda_tegra_writeb(u8 value, u8 __iomem *addr)
 {
 	unsigned int shift = ((unsigned long)(addr) & 0x3) << 3;
-	void *dword_addr = (void *)((unsigned long)(addr) & ~0x3);
+	void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3);
 	u32 v;
 
 	v = readl(dword_addr);
@@ -159,10 +159,10 @@ static void hda_tegra_writeb(u8 value, u8 *addr)
 	writel(v, dword_addr);
 }
 
-static u8 hda_tegra_readb(u8 *addr)
+static u8 hda_tegra_readb(u8 __iomem *addr)
 {
 	unsigned int shift = ((unsigned long)(addr) & 0x3) << 3;
-	void *dword_addr = (void *)((unsigned long)(addr) & ~0x3);
+	void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3);
 	u32 v;
 
 	v = readl(dword_addr);
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index a010d704e..d0d5ad8be 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -114,6 +114,9 @@ struct hdmi_ops {
 	int (*setup_stream)(struct hda_codec *codec, hda_nid_t cvt_nid,
 			    hda_nid_t pin_nid, u32 stream_tag, int format);
 
+	void (*pin_cvt_fixup)(struct hda_codec *codec,
+			      struct hdmi_spec_per_pin *per_pin,
+			      hda_nid_t cvt_nid);
 };
 
 struct hdmi_pcm {
@@ -684,7 +687,8 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
 	if (!channels)
 		return;
 
-	if (is_haswell_plus(codec))
+	/* some HW (e.g. HSW+) needs reprogramming the amp at each time */
+	if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP)
 		snd_hda_codec_write(codec, pin_nid, 0,
 					    AC_VERB_SET_AMP_GAIN_MUTE,
 					    AMP_OUT_UNMUTE);
@@ -864,9 +868,6 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid,
 	struct hdmi_spec *spec = codec->spec;
 	int err;
 
-	if (is_haswell_plus(codec))
-		haswell_verify_D0(codec, cvt_nid, pin_nid);
-
 	err = spec->ops.pin_hbr_setup(codec, pin_nid, is_hbr_format(format));
 
 	if (err) {
@@ -884,7 +885,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid,
  * of the pin.
  */
 static int hdmi_choose_cvt(struct hda_codec *codec,
-			int pin_idx, int *cvt_id, int *mux_id)
+			   int pin_idx, int *cvt_id)
 {
 	struct hdmi_spec *spec = codec->spec;
 	struct hdmi_spec_per_pin *per_pin;
@@ -925,8 +926,6 @@ static int hdmi_choose_cvt(struct hda_codec *codec,
 
 	if (cvt_id)
 		*cvt_id = cvt_idx;
-	if (mux_id)
-		*mux_id = mux_idx;
 
 	return 0;
 }
@@ -1019,9 +1018,6 @@ static void intel_not_share_assigned_cvt_nid(struct hda_codec *codec,
 	int mux_idx;
 	struct hdmi_spec *spec = codec->spec;
 
-	if (!is_haswell_plus(codec) && !is_valleyview_plus(codec))
-		return;
-
 	/* On Intel platform, the mapping of converter nid to
 	 * mux index of the pins are always the same.
 	 * The pin nid may be 0, this means all pins will not
@@ -1032,6 +1028,17 @@ static void intel_not_share_assigned_cvt_nid(struct hda_codec *codec,
 		intel_not_share_assigned_cvt(codec, pin_nid, mux_idx);
 }
 
+/* skeleton caller of pin_cvt_fixup ops */
+static void pin_cvt_fixup(struct hda_codec *codec,
+			  struct hdmi_spec_per_pin *per_pin,
+			  hda_nid_t cvt_nid)
+{
+	struct hdmi_spec *spec = codec->spec;
+
+	if (spec->ops.pin_cvt_fixup)
+		spec->ops.pin_cvt_fixup(codec, per_pin, cvt_nid);
+}
+
 /* called in hdmi_pcm_open when no pin is assigned to the PCM
  * in dyn_pcm_assign mode.
  */
@@ -1049,7 +1056,7 @@ static int hdmi_pcm_open_no_pin(struct hda_pcm_stream *hinfo,
 	if (pcm_idx < 0)
 		return -EINVAL;
 
-	err = hdmi_choose_cvt(codec, -1, &cvt_idx, NULL);
+	err = hdmi_choose_cvt(codec, -1, &cvt_idx);
 	if (err)
 		return err;
 
@@ -1057,7 +1064,7 @@ static int hdmi_pcm_open_no_pin(struct hda_pcm_stream *hinfo,
 	per_cvt->assigned = 1;
 	hinfo->nid = per_cvt->cvt_nid;
 
-	intel_not_share_assigned_cvt_nid(codec, 0, per_cvt->cvt_nid);
+	pin_cvt_fixup(codec, NULL, per_cvt->cvt_nid);
 
 	set_bit(pcm_idx, &spec->pcm_in_use);
 	/* todo: setup spdif ctls assign */
@@ -1089,7 +1096,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 {
 	struct hdmi_spec *spec = codec->spec;
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	int pin_idx, cvt_idx, pcm_idx, mux_idx = 0;
+	int pin_idx, cvt_idx, pcm_idx;
 	struct hdmi_spec_per_pin *per_pin;
 	struct hdmi_eld *eld;
 	struct hdmi_spec_per_cvt *per_cvt = NULL;
@@ -1118,7 +1125,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 		}
 	}
 
-	err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx, &mux_idx);
+	err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx);
 	if (err < 0) {
 		mutex_unlock(&spec->pcm_lock);
 		return err;
@@ -1135,11 +1142,10 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 
 	snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0,
 			    AC_VERB_SET_CONNECT_SEL,
-			    mux_idx);
+			    per_pin->mux_idx);
 
 	/* configure unused pins to choose other converters */
-	if (is_haswell_plus(codec) || is_valleyview_plus(codec))
-		intel_not_share_assigned_cvt(codec, per_pin->pin_nid, mux_idx);
+	pin_cvt_fixup(codec, per_pin, 0);
 
 	snd_hda_spdif_ctls_assign(codec, pcm_idx, per_cvt->cvt_nid);
 
@@ -1372,12 +1378,7 @@ static void update_eld(struct hda_codec *codec,
 	 *   and this can make HW reset converter selection on a pin.
 	 */
 	if (eld->eld_valid && !old_eld_valid && per_pin->setup) {
-		if (is_haswell_plus(codec) || is_valleyview_plus(codec)) {
-			intel_verify_pin_cvt_connect(codec, per_pin);
-			intel_not_share_assigned_cvt(codec, per_pin->pin_nid,
-						     per_pin->mux_idx);
-		}
-
+		pin_cvt_fixup(codec, per_pin, 0);
 		hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm);
 	}
 
@@ -1484,7 +1485,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec,
 
 	mutex_lock(&per_pin->lock);
 	eld->monitor_present = false;
-	size = snd_hdac_acomp_get_eld(&codec->bus->core, per_pin->pin_nid,
+	size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid,
 				      &eld->monitor_present, eld->eld_buffer,
 				      ELD_MAX_SIZE);
 	if (size > 0) {
@@ -1711,7 +1712,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 		 * skip pin setup and return 0 to make audio playback
 		 * be ongoing
 		 */
-		intel_not_share_assigned_cvt_nid(codec, 0, cvt_nid);
+		pin_cvt_fixup(codec, NULL, cvt_nid);
 		snd_hda_codec_setup_stream(codec, cvt_nid,
 					stream_tag, 0, format);
 		mutex_unlock(&spec->pcm_lock);
@@ -1724,23 +1725,21 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 	}
 	per_pin = get_pin(spec, pin_idx);
 	pin_nid = per_pin->pin_nid;
-	if (is_haswell_plus(codec) || is_valleyview_plus(codec)) {
-		/* Verify pin:cvt selections to avoid silent audio after S3.
-		 * After S3, the audio driver restores pin:cvt selections
-		 * but this can happen before gfx is ready and such selection
-		 * is overlooked by HW. Thus multiple pins can share a same
-		 * default convertor and mute control will affect each other,
-		 * which can cause a resumed audio playback become silent
-		 * after S3.
-		 */
-		intel_verify_pin_cvt_connect(codec, per_pin);
-		intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx);
-	}
+
+	/* Verify pin:cvt selections to avoid silent audio after S3.
+	 * After S3, the audio driver restores pin:cvt selections
+	 * but this can happen before gfx is ready and such selection
+	 * is overlooked by HW. Thus multiple pins can share a same
+	 * default convertor and mute control will affect each other,
+	 * which can cause a resumed audio playback become silent
+	 * after S3.
+	 */
+	pin_cvt_fixup(codec, per_pin, 0);
 
 	/* Call sync_audio_rate to set the N/CTS/M manually if necessary */
 	/* Todo: add DP1.2 MST audio support later */
 	if (codec_has_acomp(codec))
-		snd_hdac_sync_audio_rate(&codec->bus->core, pin_nid, runtime->rate);
+		snd_hdac_sync_audio_rate(&codec->core, pin_nid, runtime->rate);
 
 	non_pcm = check_non_pcm_per_cvt(codec, cvt_nid);
 	mutex_lock(&per_pin->lock);
@@ -1837,6 +1836,18 @@ static const struct hda_pcm_ops generic_ops = {
 	.cleanup = generic_hdmi_playback_pcm_cleanup,
 };
 
+static int hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx)
+{
+	struct hda_codec *codec = container_of(hdac, struct hda_codec, core);
+	struct hdmi_spec *spec = codec->spec;
+	struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx);
+
+	if (!per_pin)
+		return 0;
+
+	return per_pin->sink_eld.info.spk_alloc;
+}
+
 static void hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx,
 					unsigned char *chmap)
 {
@@ -2075,6 +2086,20 @@ static void hdmi_array_free(struct hdmi_spec *spec)
 	snd_array_free(&spec->cvts);
 }
 
+static void generic_spec_free(struct hda_codec *codec)
+{
+	struct hdmi_spec *spec = codec->spec;
+
+	if (spec) {
+		if (spec->i915_bound)
+			snd_hdac_i915_exit(&codec->bus->core);
+		hdmi_array_free(spec);
+		kfree(spec);
+		codec->spec = NULL;
+	}
+	codec->dp_mst = false;
+}
+
 static void generic_hdmi_free(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec = codec->spec;
@@ -2099,10 +2124,7 @@ static void generic_hdmi_free(struct hda_codec *codec)
 			spec->pcm_rec[pcm_idx].jack = NULL;
 	}
 
-	if (spec->i915_bound)
-		snd_hdac_i915_exit(&codec->bus->core);
-	hdmi_array_free(spec);
-	kfree(spec);
+	generic_spec_free(codec);
 }
 
 #ifdef CONFIG_PM
@@ -2140,6 +2162,55 @@ static const struct hdmi_ops generic_standard_hdmi_ops = {
 	.setup_stream				= hdmi_setup_stream,
 };
 
+/* allocate codec->spec and assign/initialize generic parser ops */
+static int alloc_generic_hdmi(struct hda_codec *codec)
+{
+	struct hdmi_spec *spec;
+
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (!spec)
+		return -ENOMEM;
+
+	spec->ops = generic_standard_hdmi_ops;
+	mutex_init(&spec->pcm_lock);
+	snd_hdac_register_chmap_ops(&codec->core, &spec->chmap);
+
+	spec->chmap.ops.get_chmap = hdmi_get_chmap;
+	spec->chmap.ops.set_chmap = hdmi_set_chmap;
+	spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached;
+	spec->chmap.ops.get_spk_alloc = hdmi_get_spk_alloc,
+
+	codec->spec = spec;
+	hdmi_array_init(spec, 4);
+
+	codec->patch_ops = generic_hdmi_patch_ops;
+
+	return 0;
+}
+
+/* generic HDMI parser */
+static int patch_generic_hdmi(struct hda_codec *codec)
+{
+	int err;
+
+	err = alloc_generic_hdmi(codec);
+	if (err < 0)
+		return err;
+
+	err = hdmi_parse_codec(codec);
+	if (err < 0) {
+		generic_spec_free(codec);
+		return err;
+	}
+
+	generic_hdmi_init_per_pins(codec);
+	return 0;
+}
+
+/*
+ * Intel codec parsers and helpers
+ */
+
 static void intel_haswell_fixup_connect_list(struct hda_codec *codec,
 					     hda_nid_t nid)
 {
@@ -2217,12 +2288,23 @@ static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 static void intel_pin_eld_notify(void *audio_ptr, int port)
 {
 	struct hda_codec *codec = audio_ptr;
-	int pin_nid = port + 0x04;
+	int pin_nid;
 
 	/* we assume only from port-B to port-D */
 	if (port < 1 || port > 3)
 		return;
 
+	switch (codec->core.vendor_id) {
+	case 0x80860054: /* ILK */
+	case 0x80862804: /* ILK */
+	case 0x80862882: /* VLV */
+		pin_nid = port + 0x03;
+		break;
+	default:
+		pin_nid = port + 0x04;
+		break;
+	}
+
 	/* skip notification during system suspend (but not in runtime PM);
 	 * the state will be updated at resume
 	 */
@@ -2236,93 +2318,159 @@ static void intel_pin_eld_notify(void *audio_ptr, int port)
 	check_presence_and_report(codec, pin_nid);
 }
 
-static int patch_generic_hdmi(struct hda_codec *codec)
+/* register i915 component pin_eld_notify callback */
+static void register_i915_notifier(struct hda_codec *codec)
 {
-	struct hdmi_spec *spec;
+	struct hdmi_spec *spec = codec->spec;
 
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
+	spec->use_acomp_notifier = true;
+	spec->i915_audio_ops.audio_ptr = codec;
+	/* intel_audio_codec_enable() or intel_audio_codec_disable()
+	 * will call pin_eld_notify with using audio_ptr pointer
+	 * We need make sure audio_ptr is really setup
+	 */
+	wmb();
+	spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify;
+	snd_hdac_i915_register_notifier(&spec->i915_audio_ops);
+}
 
-	spec->ops = generic_standard_hdmi_ops;
-	mutex_init(&spec->pcm_lock);
-	snd_hdac_register_chmap_ops(&codec->core, &spec->chmap);
+/* setup_stream ops override for HSW+ */
+static int i915_hsw_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid,
+				 hda_nid_t pin_nid, u32 stream_tag, int format)
+{
+	haswell_verify_D0(codec, cvt_nid, pin_nid);
+	return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format);
+}
 
-	spec->chmap.ops.get_chmap = hdmi_get_chmap;
-	spec->chmap.ops.set_chmap = hdmi_set_chmap;
-	spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached;
+/* pin_cvt_fixup ops override for HSW+ and VLV+ */
+static void i915_pin_cvt_fixup(struct hda_codec *codec,
+			       struct hdmi_spec_per_pin *per_pin,
+			       hda_nid_t cvt_nid)
+{
+	if (per_pin) {
+		intel_verify_pin_cvt_connect(codec, per_pin);
+		intel_not_share_assigned_cvt(codec, per_pin->pin_nid,
+					     per_pin->mux_idx);
+	} else {
+		intel_not_share_assigned_cvt_nid(codec, 0, cvt_nid);
+	}
+}
 
-	codec->spec = spec;
-	hdmi_array_init(spec, 4);
+/* Intel Haswell and onwards; audio component with eld notifier */
+static int patch_i915_hsw_hdmi(struct hda_codec *codec)
+{
+	struct hdmi_spec *spec;
+	int err;
 
-#ifdef CONFIG_SND_HDA_I915
-	/* Try to bind with i915 for Intel HSW+ codecs (if not done yet) */
-	if ((codec->core.vendor_id >> 16) == 0x8086 &&
-	    is_haswell_plus(codec)) {
-#if 0
-		/* on-demand binding leads to an unbalanced refcount when
-		 * both i915 and hda drivers are probed concurrently;
-		 * disabled temporarily for now
-		 */
-		if (!codec->bus->core.audio_component)
-			if (!snd_hdac_i915_init(&codec->bus->core))
-				spec->i915_bound = true;
-#endif
-		/* use i915 audio component notifier for hotplug */
-		if (codec->bus->core.audio_component)
-			spec->use_acomp_notifier = true;
+	/* HSW+ requires i915 binding */
+	if (!codec->bus->core.audio_component) {
+		codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n");
+		return -ENODEV;
 	}
-#endif
 
-	if (is_haswell_plus(codec)) {
-		intel_haswell_enable_all_pins(codec, true);
-		intel_haswell_fixup_enable_dp12(codec);
-	}
+	err = alloc_generic_hdmi(codec);
+	if (err < 0)
+		return err;
+	spec = codec->spec;
 
-	/* For Valleyview/Cherryview, only the display codec is in the display
-	 * power well and can use link_power ops to request/release the power.
-	 * For Haswell/Broadwell, the controller is also in the power well and
+	intel_haswell_enable_all_pins(codec, true);
+	intel_haswell_fixup_enable_dp12(codec);
+
+	/* For Haswell/Broadwell, the controller is also in the power well and
 	 * can cover the codec power request, and so need not set this flag.
-	 * For previous platforms, there is no such power well feature.
 	 */
-	if (is_valleyview_plus(codec) || is_skylake(codec) ||
-			is_broxton(codec))
+	if (!is_haswell(codec) && !is_broadwell(codec))
 		codec->core.link_power_control = 1;
 
-	if (hdmi_parse_codec(codec) < 0) {
-		if (spec->i915_bound)
-			snd_hdac_i915_exit(&codec->bus->core);
-		codec->spec = NULL;
-		kfree(spec);
-		return -EINVAL;
+	codec->patch_ops.set_power_state = haswell_set_power_state;
+	codec->dp_mst = true;
+	codec->depop_delay = 0;
+	codec->auto_runtime_pm = 1;
+
+	spec->ops.setup_stream = i915_hsw_setup_stream;
+	spec->ops.pin_cvt_fixup = i915_pin_cvt_fixup;
+
+	err = hdmi_parse_codec(codec);
+	if (err < 0) {
+		generic_spec_free(codec);
+		return err;
 	}
-	codec->patch_ops = generic_hdmi_patch_ops;
-	if (is_haswell_plus(codec)) {
-		codec->patch_ops.set_power_state = haswell_set_power_state;
-		codec->dp_mst = true;
+
+	generic_hdmi_init_per_pins(codec);
+	register_i915_notifier(codec);
+	return 0;
+}
+
+/* Intel Baytrail and Braswell; with eld notifier */
+static int patch_i915_byt_hdmi(struct hda_codec *codec)
+{
+	struct hdmi_spec *spec;
+	int err;
+
+	/* requires i915 binding */
+	if (!codec->bus->core.audio_component) {
+		codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n");
+		return -ENODEV;
 	}
 
-	/* Enable runtime pm for HDMI audio codec of HSW/BDW/SKL/BYT/BSW */
-	if (is_haswell_plus(codec) || is_valleyview_plus(codec))
-		codec->auto_runtime_pm = 1;
+	err = alloc_generic_hdmi(codec);
+	if (err < 0)
+		return err;
+	spec = codec->spec;
 
-	generic_hdmi_init_per_pins(codec);
+	/* For Valleyview/Cherryview, only the display codec is in the display
+	 * power well and can use link_power ops to request/release the power.
+	 */
+	codec->core.link_power_control = 1;
 
+	codec->depop_delay = 0;
+	codec->auto_runtime_pm = 1;
 
-	if (codec_has_acomp(codec)) {
-		codec->depop_delay = 0;
-		spec->i915_audio_ops.audio_ptr = codec;
-		/* intel_audio_codec_enable() or intel_audio_codec_disable()
-		 * will call pin_eld_notify with using audio_ptr pointer
-		 * We need make sure audio_ptr is really setup
-		 */
-		wmb();
-		spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify;
-		snd_hdac_i915_register_notifier(&spec->i915_audio_ops);
+	spec->ops.pin_cvt_fixup = i915_pin_cvt_fixup;
+
+	err = hdmi_parse_codec(codec);
+	if (err < 0) {
+		generic_spec_free(codec);
+		return err;
 	}
 
-	WARN_ON(spec->dyn_pcm_assign && !codec_has_acomp(codec));
+	generic_hdmi_init_per_pins(codec);
+	register_i915_notifier(codec);
+	return 0;
+}
+
+/* Intel IronLake, SandyBridge and IvyBridge; with eld notifier */
+static int patch_i915_cpt_hdmi(struct hda_codec *codec)
+{
+	struct hdmi_spec *spec;
+	int err;
+
+	/* no i915 component should have been bound before this */
+	if (WARN_ON(codec->bus->core.audio_component))
+		return -EBUSY;
+
+	err = alloc_generic_hdmi(codec);
+	if (err < 0)
+		return err;
+	spec = codec->spec;
+
+	/* Try to bind with i915 now */
+	err = snd_hdac_i915_init(&codec->bus->core);
+	if (err < 0)
+		goto error;
+	spec->i915_bound = true;
+
+	err = hdmi_parse_codec(codec);
+	if (err < 0)
+		goto error;
+
+	generic_hdmi_init_per_pins(codec);
+	register_i915_notifier(codec);
 	return 0;
+
+ error:
+	generic_spec_free(codec);
+	return err;
 }
 
 /*
@@ -3492,21 +3640,21 @@ 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(0x80860054, "IbexPeak HDMI",	patch_i915_cpt_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(0x8086280b, "Kabylake HDMI",	patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI",	patch_i915_cpt_hdmi),
+HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI",	patch_i915_cpt_hdmi),
+HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_i915_cpt_hdmi),
+HDA_CODEC_ENTRY(0x80862807, "Haswell HDMI",	patch_i915_hsw_hdmi),
+HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI",	patch_i915_hsw_hdmi),
+HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI",	patch_i915_hsw_hdmi),
+HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI",	patch_i915_hsw_hdmi),
+HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI",	patch_i915_hsw_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(0x80862882, "Valleyview2 HDMI",	patch_i915_byt_hdmi),
+HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI",	patch_i915_byt_hdmi),
 HDA_CODEC_ENTRY(0x808629fb, "Crestline HDMI",	patch_generic_hdmi),
 /* special ID for generic HDMI */
 HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC_HDMI, "Generic HDMI", patch_generic_hdmi),
diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c
index 59ab6cee1..f0955fd7a 100644
--- a/sound/pci/hda/thinkpad_helper.c
+++ b/sound/pci/hda/thinkpad_helper.c
@@ -13,7 +13,7 @@ static void (*old_vmaster_hook)(void *, int);
 static bool is_thinkpad(struct hda_codec *codec)
 {
 	return (codec->core.subsystem_id >> 16 == 0x17aa) &&
-	       (acpi_dev_present("LEN0068") || acpi_dev_present("IBM0068"));
+	       (acpi_dev_found("LEN0068") || acpi_dev_found("IBM0068"));
 }
 
 static void update_tpacpi_mute_led(void *private_data, int enabled)
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 8151318a6..9720a30db 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -42,12 +42,6 @@
 #include <asm/pgtable.h>
 #include <asm/cacheflush.h>
 
-#ifdef CONFIG_KVM_GUEST
-#include <linux/kvm_para.h>
-#else
-#define kvm_para_available() (0)
-#endif
-
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455");
 MODULE_LICENSE("GPL");
@@ -2972,25 +2966,17 @@ static int snd_intel8x0_inside_vm(struct pci_dev *pci)
 		goto fini;
 	}
 
-	/* detect KVM and Parallels virtual environments */
-	result = kvm_para_available();
-#ifdef X86_FEATURE_HYPERVISOR
-	result = result || boot_cpu_has(X86_FEATURE_HYPERVISOR);
-#endif
-	if (!result)
-		goto fini;
-
 	/* check for known (emulated) devices */
+	result = 0;
 	if (pci->subsystem_vendor == PCI_SUBVENDOR_ID_REDHAT_QUMRANET &&
 	    pci->subsystem_device == PCI_SUBDEVICE_ID_QEMU) {
 		/* KVM emulated sound, PCI SSID: 1af4:1100 */
 		msg = "enable KVM";
+		result = 1;
 	} else if (pci->subsystem_vendor == 0x1ab8) {
 		/* Parallels VM emulated sound, PCI SSID: 1ab8:xxxx */
 		msg = "enable Parallels VM";
-	} else {
-		msg = "disable (unknown or VT-d) VM";
-		result = 0;
+		result = 1;
 	}
 
 fini:
diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c
index f3d62020e..a80684bdc 100644
--- a/sound/pci/lx6464es/lx_core.c
+++ b/sound/pci/lx6464es/lx_core.c
@@ -644,7 +644,7 @@ static int lx_pipe_wait_for_state(struct lx6464es *chip, u32 pipe,
 		if (err < 0)
 			return err;
 
-		if (current_state == state)
+		if (!err && current_state == state)
 			return 0;
 
 		mdelay(1);
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 276897033..1267e1af0 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -652,7 +652,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 		rcmr =	  SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period)
 			| SSC_BF(RCMR_STTDLY, 1)
 			| SSC_BF(RCMR_START, SSC_START_RISING_RF)
-			| SSC_BF(RCMR_CKI, SSC_CKI_FALLING)
+			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
 			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
 			| SSC_BF(RCMR_CKS, SSC_CKS_DIV);
 
@@ -692,7 +692,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 		rcmr =	  SSC_BF(RCMR_PERIOD, 0)
 			| SSC_BF(RCMR_STTDLY, START_DELAY)
 			| SSC_BF(RCMR_START, SSC_START_RISING_RF)
-			| SSC_BF(RCMR_CKI, SSC_CKI_FALLING)
+			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
 			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
 			| SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
 					   SSC_CKS_PIN : SSC_CKS_CLOCK);
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 5741c0aa6..b5d1caa04 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -206,8 +206,8 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
 	stype = substream->stream;
 	pcd = to_dmadata(substream);
 
-	DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d "
-	    "runtime->min_align %d\n",
+	DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %zu "
+	    "runtime->min_align %lu\n",
 		(unsigned long)runtime->dma_area,
 		(unsigned long)runtime->dma_addr, runtime->dma_bytes,
 		runtime->min_align);
diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c
index 1c1f22103..6ba204982 100644
--- a/sound/soc/bcm/bcm2835-i2s.c
+++ b/sound/soc/bcm/bcm2835-i2s.c
@@ -259,6 +259,9 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_FORMAT_S16_LE:
 		data_length = 16;
 		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		data_length = 24;
+		break;
 	case SNDRV_PCM_FORMAT_S32_LE:
 		data_length = 32;
 		break;
@@ -273,13 +276,20 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream,
 		/* otherwise calculate a fitting block ratio */
 		bclk_ratio = 2 * data_length;
 
-	/* set target clock rate*/
-	clk_set_rate(dev->clk, sampling_rate * bclk_ratio);
+	/* Clock should only be set up here if CPU is clock master */
+	switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+	case SND_SOC_DAIFMT_CBS_CFM:
+		clk_set_rate(dev->clk, sampling_rate * bclk_ratio);
+		break;
+	default:
+		break;
+	}
 
 	/* Setup the frame format */
 	format = BCM2835_I2S_CHEN;
 
-	if (data_length > 24)
+	if (data_length >= 24)
 		format |= BCM2835_I2S_CHWEX;
 
 	format |= BCM2835_I2S_CHWID((data_length-8)&0xf);
@@ -570,6 +580,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = {
 		.channels_max = 2,
 		.rates =	SNDRV_PCM_RATE_8000_192000,
 		.formats =	SNDRV_PCM_FMTBIT_S16_LE
+				| SNDRV_PCM_FMTBIT_S24_LE
 				| SNDRV_PCM_FMTBIT_S32_LE
 		},
 	.capture = {
@@ -577,6 +588,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = {
 		.channels_max = 2,
 		.rates =	SNDRV_PCM_RATE_8000_192000,
 		.formats =	SNDRV_PCM_FMTBIT_S16_LE
+				| SNDRV_PCM_FMTBIT_S24_LE
 				| SNDRV_PCM_FMTBIT_S32_LE
 		},
 	.ops = &bcm2835_i2s_dai_ops,
@@ -678,6 +690,15 @@ static int bcm2835_i2s_probe(struct platform_device *pdev)
 	dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].maxburst = 2;
 	dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].maxburst = 2;
 
+	/*
+	 * Set the PACK flag to enable S16_LE support (2 S16_LE values
+	 * packed into 32-bit transfers).
+	 */
+	dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].flags =
+		SND_DMAENGINE_PCM_DAI_FLAG_PACK;
+	dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].flags =
+		SND_DMAENGINE_PCM_DAI_FLAG_PACK;
+
 	/* BCLK ratio - use default */
 	dev->bclk_ratio = 0;
 
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 7ef3a0c16..f3fb98f0a 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -43,6 +43,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AK5386
 	select SND_SOC_ALC5623 if I2C
 	select SND_SOC_ALC5632 if I2C
+	select SND_SOC_BT_SCO
 	select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
 	select SND_SOC_CS35L32 if I2C
 	select SND_SOC_CS42L51_I2C if I2C
@@ -64,7 +65,6 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_DA732X if I2C
 	select SND_SOC_DA9055 if I2C
 	select SND_SOC_DMIC
-	select SND_SOC_BT_SCO
 	select SND_SOC_ES8328_SPI if SPI_MASTER
 	select SND_SOC_ES8328_I2C if I2C
 	select SND_SOC_GTM601
@@ -79,6 +79,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MAX98090 if I2C
 	select SND_SOC_MAX98095 if I2C
 	select SND_SOC_MAX98357A if GPIOLIB
+	select SND_SOC_MAX98371 if I2C
 	select SND_SOC_MAX9867 if I2C
 	select SND_SOC_MAX98925 if I2C
 	select SND_SOC_MAX98926 if I2C
@@ -88,12 +89,14 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MC13783 if MFD_MC13XXX
 	select SND_SOC_ML26124 if I2C
 	select SND_SOC_NAU8825 if I2C
+	select SND_SOC_HDMI_CODEC
 	select SND_SOC_PCM1681 if I2C
 	select SND_SOC_PCM179X_I2C if I2C
 	select SND_SOC_PCM179X_SPI if SPI_MASTER
 	select SND_SOC_PCM3008
 	select SND_SOC_PCM3168A_I2C if I2C
 	select SND_SOC_PCM3168A_SPI if SPI_MASTER
+	select SND_SOC_PCM5102A
 	select SND_SOC_PCM512x_I2C if I2C
 	select SND_SOC_PCM512x_SPI if SPI_MASTER
 	select SND_SOC_RT286 if I2C
@@ -124,12 +127,14 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_TAS2552 if I2C
 	select SND_SOC_TAS5086 if I2C
 	select SND_SOC_TAS571X if I2C
+	select SND_SOC_TAS5720 if I2C
 	select SND_SOC_TFA9879 if I2C
 	select SND_SOC_TLV320AIC23_I2C if I2C
 	select SND_SOC_TLV320AIC23_SPI if SPI_MASTER
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
 	select SND_SOC_TLV320AIC31XX if I2C
-	select SND_SOC_TLV320AIC32X4 if I2C
+	select SND_SOC_TLV320AIC32X4_I2C if I2C
+	select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER
 	select SND_SOC_TLV320AIC3X if I2C
 	select SND_SOC_TPA6130A2 if I2C
 	select SND_SOC_TLV320DAC33 if I2C
@@ -365,6 +370,9 @@ config SND_SOC_ALC5623
 config SND_SOC_ALC5632
 	tristate
 
+config SND_SOC_BT_SCO
+	tristate
+
 config SND_SOC_CQ0093VC
 	tristate
 
@@ -471,11 +479,14 @@ config SND_SOC_DA732X
 config SND_SOC_DA9055
 	tristate
 
-config SND_SOC_BT_SCO
+config SND_SOC_DMIC
 	tristate
 
-config SND_SOC_DMIC
+config SND_SOC_HDMI_CODEC
 	tristate
+	select SND_PCM_ELD
+	select SND_PCM_IEC958
+	select HDMI
 
 config SND_SOC_ES8328
 	tristate "Everest Semi ES8328 CODEC"
@@ -522,6 +533,9 @@ config SND_SOC_MAX98095
 config SND_SOC_MAX98357A
        tristate
 
+config SND_SOC_MAX98371
+       tristate
+
 config SND_SOC_MAX9867
 	tristate
 
@@ -575,6 +589,9 @@ config SND_SOC_PCM3168A_SPI
 	select SND_SOC_PCM3168A
 	select REGMAP_SPI
 
+config SND_SOC_PCM5102A
+	tristate
+
 config SND_SOC_PCM512x
 	tristate
 
@@ -738,8 +755,15 @@ config SND_SOC_TAS5086
 	depends on I2C
 
 config SND_SOC_TAS571X
-	tristate "Texas Instruments TAS5711/TAS5717/TAS5719 power amplifiers"
+	tristate "Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 power amplifiers"
+	depends on I2C
+
+config SND_SOC_TAS5720
+	tristate "Texas Instruments TAS5720 Mono Audio amplifier"
 	depends on I2C
+	help
+	  Enable support for Texas Instruments TAS5720L/M high-efficiency mono
+	  Class-D audio power amplifiers.
 
 config SND_SOC_TFA9879
 	tristate "NXP Semiconductors TFA9879 amplifier"
@@ -770,6 +794,16 @@ config SND_SOC_TLV320AIC31XX
 config SND_SOC_TLV320AIC32X4
 	tristate
 
+config SND_SOC_TLV320AIC32X4_I2C
+	tristate
+	depends on I2C
+	select SND_SOC_TLV320AIC32X4
+
+config SND_SOC_TLV320AIC32X4_SPI
+	tristate
+	depends on SPI_MASTER
+	select SND_SOC_TLV320AIC32X4
+
 config SND_SOC_TLV320AIC3X
 	tristate "Texas Instruments TLV320AIC3x CODECs"
 	depends on I2C
@@ -910,7 +944,8 @@ config SND_SOC_WM8955
 	tristate
 
 config SND_SOC_WM8960
-	tristate
+	tristate "Wolfson Microelectronics WM8960 CODEC"
+	depends on I2C
 
 config SND_SOC_WM8961
 	tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 185a712a7..0f548fd34 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -32,6 +32,7 @@ snd-soc-ak4642-objs := ak4642.o
 snd-soc-ak4671-objs := ak4671.o
 snd-soc-ak5386-objs := ak5386.o
 snd-soc-arizona-objs := arizona.o
+snd-soc-bt-sco-objs := bt-sco.o
 snd-soc-cq93vc-objs := cq93vc.o
 snd-soc-cs35l32-objs := cs35l32.o
 snd-soc-cs42l51-objs := cs42l51.o
@@ -55,7 +56,6 @@ snd-soc-da7218-objs := da7218.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
 snd-soc-dmic-objs := dmic.o
 snd-soc-es8328-objs := es8328.o
 snd-soc-es8328-i2c-objs := es8328-i2c.o
@@ -74,6 +74,7 @@ snd-soc-max98088-objs := max98088.o
 snd-soc-max98090-objs := max98090.o
 snd-soc-max98095-objs := max98095.o
 snd-soc-max98357a-objs := max98357a.o
+snd-soc-max98371-objs := max98371.o
 snd-soc-max9867-objs := max9867.o
 snd-soc-max98925-objs := max98925.o
 snd-soc-max98926-objs := max98926.o
@@ -81,6 +82,7 @@ snd-soc-max9850-objs := max9850.o
 snd-soc-mc13783-objs := mc13783.o
 snd-soc-ml26124-objs := ml26124.o
 snd-soc-nau8825-objs := nau8825.o
+snd-soc-hdmi-codec-objs := hdmi-codec.o
 snd-soc-pcm1681-objs := pcm1681.o
 snd-soc-pcm179x-codec-objs := pcm179x.o
 snd-soc-pcm179x-i2c-objs := pcm179x-i2c.o
@@ -89,6 +91,7 @@ snd-soc-pcm3008-objs := pcm3008.o
 snd-soc-pcm3168a-objs := pcm3168a.o
 snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o
 snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o
+snd-soc-pcm5102a-objs := pcm5102a.o
 snd-soc-pcm512x-objs := pcm512x.o
 snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
 snd-soc-pcm512x-spi-objs := pcm512x-spi.o
@@ -129,6 +132,7 @@ snd-soc-stac9766-objs := stac9766.o
 snd-soc-sti-sas-objs := sti-sas.o
 snd-soc-tas5086-objs := tas5086.o
 snd-soc-tas571x-objs := tas571x.o
+snd-soc-tas5720-objs := tas5720.o
 snd-soc-tfa9879-objs := tfa9879.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o
@@ -136,6 +140,8 @@ snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o
 snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o
+snd-soc-tlv320aic32x4-i2c-objs := tlv320aic32x4-i2c.o
+snd-soc-tlv320aic32x4-spi-objs := tlv320aic32x4-spi.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 snd-soc-tlv320dac33-objs := tlv320dac33.o
 snd-soc-ts3a227e-objs := ts3a227e.o
@@ -241,6 +247,7 @@ obj-$(CONFIG_SND_SOC_AK5386)	+= snd-soc-ak5386.o
 obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
 obj-$(CONFIG_SND_SOC_ALC5632)	+= snd-soc-alc5632.o
 obj-$(CONFIG_SND_SOC_ARIZONA)	+= snd-soc-arizona.o
+obj-$(CONFIG_SND_SOC_BT_SCO)	+= snd-soc-bt-sco.o
 obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
 obj-$(CONFIG_SND_SOC_CS35L32)	+= snd-soc-cs35l32.o
 obj-$(CONFIG_SND_SOC_CS42L51)	+= snd-soc-cs42l51.o
@@ -264,7 +271,6 @@ obj-$(CONFIG_SND_SOC_DA7218)	+= snd-soc-da7218.o
 obj-$(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-$(CONFIG_SND_SOC_BT_SCO)	+= snd-soc-bt-sco.o
 obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
 obj-$(CONFIG_SND_SOC_ES8328)	+= snd-soc-es8328.o
 obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
@@ -290,6 +296,7 @@ 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_NAU8825)   += snd-soc-nau8825.o
+obj-$(CONFIG_SND_SOC_HDMI_CODEC)	+= snd-soc-hdmi-codec.o
 obj-$(CONFIG_SND_SOC_PCM1681)	+= snd-soc-pcm1681.o
 obj-$(CONFIG_SND_SOC_PCM179X)	+= snd-soc-pcm179x-codec.o
 obj-$(CONFIG_SND_SOC_PCM179X_I2C)	+= snd-soc-pcm179x-i2c.o
@@ -298,6 +305,7 @@ obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
 obj-$(CONFIG_SND_SOC_PCM3168A)	+= snd-soc-pcm3168a.o
 obj-$(CONFIG_SND_SOC_PCM3168A_I2C)	+= snd-soc-pcm3168a-i2c.o
 obj-$(CONFIG_SND_SOC_PCM3168A_SPI)	+= snd-soc-pcm3168a-spi.o
+obj-$(CONFIG_SND_SOC_PCM5102A)	+= snd-soc-pcm5102a.o
 obj-$(CONFIG_SND_SOC_PCM512x)	+= snd-soc-pcm512x.o
 obj-$(CONFIG_SND_SOC_PCM512x_I2C)	+= snd-soc-pcm512x-i2c.o
 obj-$(CONFIG_SND_SOC_PCM512x_SPI)	+= snd-soc-pcm512x-spi.o
@@ -335,6 +343,7 @@ obj-$(CONFIG_SND_SOC_STI_SAS)	+= snd-soc-sti-sas.o
 obj-$(CONFIG_SND_SOC_TAS2552)	+= snd-soc-tas2552.o
 obj-$(CONFIG_SND_SOC_TAS5086)	+= snd-soc-tas5086.o
 obj-$(CONFIG_SND_SOC_TAS571X)	+= snd-soc-tas571x.o
+obj-$(CONFIG_SND_SOC_TAS5720)	+= snd-soc-tas5720.o
 obj-$(CONFIG_SND_SOC_TFA9879)	+= snd-soc-tfa9879.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C)	+= snd-soc-tlv320aic23-i2c.o
@@ -342,6 +351,8 @@ obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI)	+= snd-soc-tlv320aic23-spi.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC31XX)     += snd-soc-tlv320aic31xx.o
 obj-$(CONFIG_SND_SOC_TLV320AIC32X4)     += snd-soc-tlv320aic32x4.o
+obj-$(CONFIG_SND_SOC_TLV320AIC32X4_I2C)	+= snd-soc-tlv320aic32x4-i2c.o
+obj-$(CONFIG_SND_SOC_TLV320AIC32X4_SPI)	+= snd-soc-tlv320aic32x4-spi.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
 obj-$(CONFIG_SND_SOC_TLV320DAC33)	+= snd-soc-tlv320dac33.o
 obj-$(CONFIG_SND_SOC_TS3A227E)	+= snd-soc-ts3a227e.o
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c
index 647f69de6..5013d2ba0 100644
--- a/sound/soc/codecs/ak4613.c
+++ b/sound/soc/codecs/ak4613.c
@@ -146,6 +146,7 @@ static const struct regmap_config ak4613_regmap_cfg = {
 	.max_register		= 0x16,
 	.reg_defaults		= ak4613_reg,
 	.num_reg_defaults	= ARRAY_SIZE(ak4613_reg),
+	.cache_type		= REGCACHE_RBTREE,
 };
 
 static const struct of_device_id ak4613_of_match[] = {
@@ -530,7 +531,6 @@ static int ak4613_i2c_remove(struct i2c_client *client)
 static struct i2c_driver ak4613_i2c_driver = {
 	.driver = {
 		.name = "ak4613-codec",
-		.owner = THIS_MODULE,
 		.of_match_table = ak4613_of_match,
 	},
 	.probe		= ak4613_i2c_probe,
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index eb8fe212e..4d8b9e49e 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -611,9 +611,7 @@ static struct clk *ak4642_of_parse_mcko(struct device *dev)
 
 	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);
+	clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, 0, rate);
 	if (!IS_ERR(clk))
 		of_clk_add_provider(np, of_clk_src_simple_get, clk);
 
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 83959312f..664a8c044 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -221,6 +221,8 @@ int arizona_init_spk(struct snd_soc_codec *codec)
 
 	switch (arizona->type) {
 	case WM8997:
+	case CS47L24:
+	case WM1831:
 		break;
 	default:
 		ret = snd_soc_dapm_new_controls(dapm, &arizona_spkr, 1);
@@ -1134,7 +1136,6 @@ int arizona_anc_ev(struct snd_soc_dapm_widget *w,
 		   int event)
 {
 	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
-	unsigned int mask = 0x3 << w->shift;
 	unsigned int val;
 
 	switch (event) {
@@ -1148,7 +1149,7 @@ int arizona_anc_ev(struct snd_soc_dapm_widget *w,
 		return 0;
 	}
 
-	snd_soc_update_bits(codec, ARIZONA_CLOCK_CONTROL, mask, val);
+	snd_soc_write(codec, ARIZONA_CLOCK_CONTROL, val);
 
 	return 0;
 }
@@ -2047,7 +2048,21 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
 			init_ratio, Fref, refdiv);
 
 	while (div <= ARIZONA_FLL_MAX_REFDIV) {
-		for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO;
+		/* start from init_ratio because this may already give a
+		 * fractional N.K
+		 */
+		for (ratio = init_ratio; ratio > 0; ratio--) {
+			if (target % (ratio * Fref)) {
+				cfg->refdiv = refdiv;
+				cfg->fratio = ratio - 1;
+				arizona_fll_dbg(fll,
+					"pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n",
+					Fref, refdiv, div, ratio);
+				return ratio;
+			}
+		}
+
+		for (ratio = init_ratio + 1; ratio <= ARIZONA_FLL_MAX_FRATIO;
 		     ratio++) {
 			if ((ARIZONA_FLL_VCO_CORNER / 2) /
 			    (fll->vco_mult * ratio) < Fref) {
@@ -2073,17 +2088,6 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
 			}
 		}
 
-		for (ratio = init_ratio - 1; ratio > 0; ratio--) {
-			if (target % (ratio * Fref)) {
-				cfg->refdiv = refdiv;
-				cfg->fratio = ratio - 1;
-				arizona_fll_dbg(fll,
-					"pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n",
-					Fref, refdiv, div, ratio);
-				return ratio;
-			}
-		}
-
 		div *= 2;
 		Fref /= 2;
 		refdiv++;
diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index 7cd5f769b..eec1ff853 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -56,7 +56,7 @@ struct  cs42l56_private {
 	u8 iface;
 	u8 iface_fmt;
 	u8 iface_inv;
-#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE)
+#if IS_ENABLED(CONFIG_INPUT)
 	struct input_dev *beep;
 	struct work_struct beep_work;
 	int beep_rate;
diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index 00e9b6fc1..5ec5a682d 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -807,6 +807,9 @@ static const struct snd_soc_dapm_route cs47l24_dapm_routes[] = {
 	{ "IN2L PGA", NULL, "IN2L" },
 	{ "IN2R PGA", NULL, "IN2R" },
 
+	{ "Audio Trace DSP", NULL, "DSP2" },
+	{ "Audio Trace DSP", NULL, "SYSCLK" },
+
 	ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
 	ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
 
@@ -1016,6 +1019,27 @@ static struct snd_soc_dai_driver cs47l24_dai[] = {
 			.formats = CS47L24_FORMATS,
 		},
 	},
+	{
+		.name = "cs47l24-cpu-trace",
+		.capture = {
+			.stream_name = "Audio Trace CPU",
+			.channels_min = 1,
+			.channels_max = 6,
+			.rates = CS47L24_RATES,
+			.formats = CS47L24_FORMATS,
+		},
+		.compress_new = snd_soc_new_compress,
+	},
+	{
+		.name = "cs47l24-dsp-trace",
+		.capture = {
+			.stream_name = "Audio Trace DSP",
+			.channels_min = 1,
+			.channels_max = 6,
+			.rates = CS47L24_RATES,
+			.formats = CS47L24_FORMATS,
+		},
+	},
 };
 
 static int cs47l24_open(struct snd_compr_stream *stream)
@@ -1027,6 +1051,8 @@ static int cs47l24_open(struct snd_compr_stream *stream)
 
 	if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-voicectrl") == 0) {
 		n_adsp = 2;
+	} else if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-trace") == 0) {
+		n_adsp = 1;
 	} else {
 		dev_err(arizona->dev,
 			"No suitable compressed stream for DAI '%s'\n",
@@ -1041,10 +1067,16 @@ static irqreturn_t cs47l24_adsp2_irq(int irq, void *data)
 {
 	struct cs47l24_priv *priv = data;
 	struct arizona *arizona = priv->core.arizona;
-	int ret;
+	int serviced = 0;
+	int i, ret;
 
-	ret = wm_adsp_compr_handle_irq(&priv->core.adsp[2]);
-	if (ret == -ENODEV) {
+	for (i = 1; i <= 2; ++i) {
+		ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]);
+		if (ret != -ENODEV)
+			serviced++;
+	}
+
+	if (!serviced) {
 		dev_err(arizona->dev, "Spurious compressed data IRQ\n");
 		return IRQ_NONE;
 	}
@@ -1160,6 +1192,7 @@ static struct snd_compr_ops cs47l24_compr_ops = {
 static struct snd_soc_platform_driver cs47l24_compr_platform = {
 	.compr_ops = &cs47l24_compr_ops,
 };
+
 static int cs47l24_probe(struct platform_device *pdev)
 {
 	struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
@@ -1228,9 +1261,9 @@ static int cs47l24_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "Failed to register platform: %d\n", ret);
 		return ret;
 	}
+
 	ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cs47l24,
 				      cs47l24_dai, ARRAY_SIZE(cs47l24_dai));
-
 	if (ret < 0) {
 		dev_err(&pdev->dev, "Failed to register codec: %d\n", ret);
 		snd_soc_unregister_platform(&pdev->dev);
@@ -1241,10 +1274,15 @@ static int cs47l24_probe(struct platform_device *pdev)
 
 static int cs47l24_remove(struct platform_device *pdev)
 {
+	struct cs47l24_priv *cs47l24 = platform_get_drvdata(pdev);
+
 	snd_soc_unregister_platform(&pdev->dev);
 	snd_soc_unregister_codec(&pdev->dev);
 	pm_runtime_disable(&pdev->dev);
 
+	wm_adsp2_remove(&cs47l24->core.adsp[1]);
+	wm_adsp2_remove(&cs47l24->core.adsp[2]);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index d6f4abbbf..fb3885fe0 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -226,6 +226,7 @@ static int v253_open(struct tty_struct *tty)
 	if (!tty->disc_data)
 		return -ENODEV;
 
+	tty->receive_room = 16;
 	if (tty->ops->write(tty, v253_init, len) != len) {
 		ret = -EIO;
 		goto err;
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 7278f9346..e5527bc57 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -726,6 +726,68 @@ static const struct snd_kcontrol_new da7213_dapm_mixoutr_controls[] = {
 
 
 /*
+ * DAPM Events
+ */
+
+static int da7213_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 da7213_priv *da7213 = 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:
+		/* Enable DAI clks for master mode */
+		if (da7213->master)
+			snd_soc_update_bits(codec, DA7213_DAI_CLK_MODE,
+					    DA7213_DAI_CLK_EN_MASK,
+					    DA7213_DAI_CLK_EN_MASK);
+
+		/* PC synchronised to DAI */
+		snd_soc_update_bits(codec, DA7213_PC_COUNT,
+				    DA7213_PC_FREERUN_MASK, 0);
+
+		/* Slave mode, if SRM not enabled no need for status checks */
+		pll_ctrl = snd_soc_read(codec, DA7213_PLL_CTRL);
+		if (!(pll_ctrl & DA7213_PLL_SRM_EN))
+			return 0;
+
+		/* Check SRM has locked */
+		do {
+			pll_status = snd_soc_read(codec, DA7213_PLL_STATUS);
+			if (pll_status & DA7219_PLL_SRM_LOCK) {
+				srm_lock = true;
+			} else {
+				++i;
+				msleep(50);
+			}
+		} while ((i < DA7213_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, DA7213_PC_COUNT,
+				    DA7213_PC_FREERUN_MASK,
+				    DA7213_PC_FREERUN_MASK);
+
+		/* Disable DAI clks if in master mode */
+		if (da7213->master)
+			snd_soc_update_bits(codec, DA7213_DAI_CLK_MODE,
+					    DA7213_DAI_CLK_EN_MASK, 0);
+		return 0;
+	default:
+		return -EINVAL;
+	}
+}
+
+
+/*
  * DAPM widgets
  */
 
@@ -736,7 +798,8 @@ static const struct snd_soc_dapm_widget da7213_dapm_widgets[] = {
 
 	/* Use a supply here as this controls both input & output DAIs */
 	SND_SOC_DAPM_SUPPLY("DAI", DA7213_DAI_CTRL, DA7213_DAI_EN_SHIFT,
-			    DA7213_NO_INVERT, NULL, 0),
+			    DA7213_NO_INVERT, da7213_dai_event,
+			    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 
 	/*
 	 * Input
@@ -1143,11 +1206,9 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	/* Set master/slave mode */
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBM_CFM:
-		dai_clk_mode |= DA7213_DAI_CLK_EN_MASTER_MODE;
 		da7213->master = true;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFS:
-		dai_clk_mode |= DA7213_DAI_CLK_EN_SLAVE_MODE;
 		da7213->master = false;
 		break;
 	default:
@@ -1281,28 +1342,28 @@ static int da7213_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 	pll_ctrl = 0;
 
 	/* Workout input divider based on MCLK rate */
-	if ((da7213->mclk_rate == 32768) && (source == DA7213_SYSCLK_PLL)) {
+	if (da7213->mclk_rate == 32768) {
 		/* 32KHz PLL Mode */
-		indiv_bits = DA7213_PLL_INDIV_10_20_MHZ;
-		indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL;
+		indiv_bits = DA7213_PLL_INDIV_9_TO_18_MHZ;
+		indiv = DA7213_PLL_INDIV_9_TO_18_MHZ_VAL;
 		freq_ref = 3750000;
 		pll_ctrl |= DA7213_PLL_32K_MODE;
 	} else {
 		/* 5 - 54MHz MCLK */
 		if (da7213->mclk_rate < 5000000) {
 			goto pll_err;
-		} else if (da7213->mclk_rate <= 10000000) {
-			indiv_bits = DA7213_PLL_INDIV_5_10_MHZ;
-			indiv = DA7213_PLL_INDIV_5_10_MHZ_VAL;
-		} else if (da7213->mclk_rate <= 20000000) {
-			indiv_bits = DA7213_PLL_INDIV_10_20_MHZ;
-			indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL;
-		} else if (da7213->mclk_rate <= 40000000) {
-			indiv_bits = DA7213_PLL_INDIV_20_40_MHZ;
-			indiv = DA7213_PLL_INDIV_20_40_MHZ_VAL;
+		} else if (da7213->mclk_rate <= 9000000) {
+			indiv_bits = DA7213_PLL_INDIV_5_TO_9_MHZ;
+			indiv = DA7213_PLL_INDIV_5_TO_9_MHZ_VAL;
+		} else if (da7213->mclk_rate <= 18000000) {
+			indiv_bits = DA7213_PLL_INDIV_9_TO_18_MHZ;
+			indiv = DA7213_PLL_INDIV_9_TO_18_MHZ_VAL;
+		} else if (da7213->mclk_rate <= 36000000) {
+			indiv_bits = DA7213_PLL_INDIV_18_TO_36_MHZ;
+			indiv = DA7213_PLL_INDIV_18_TO_36_MHZ_VAL;
 		} else if (da7213->mclk_rate <= 54000000) {
-			indiv_bits = DA7213_PLL_INDIV_40_54_MHZ;
-			indiv = DA7213_PLL_INDIV_40_54_MHZ_VAL;
+			indiv_bits = DA7213_PLL_INDIV_36_TO_54_MHZ;
+			indiv = DA7213_PLL_INDIV_36_TO_54_MHZ_VAL;
 		} else {
 			goto pll_err;
 		}
@@ -1547,6 +1608,10 @@ static int da7213_probe(struct snd_soc_codec *codec)
 	/* Default to using SRM for slave mode */
 	da7213->srm_en = true;
 
+	/* Default PC counter to free-running */
+	snd_soc_update_bits(codec, DA7213_PC_COUNT, DA7213_PC_FREERUN_MASK,
+			    DA7213_PC_FREERUN_MASK);
+
 	/* Enable all Gain Ramps */
 	snd_soc_update_bits(codec, DA7213_AUX_L_CTRL,
 			    DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN);
diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h
index 030fd691b..fbb7a356a 100644
--- a/sound/soc/codecs/da7213.h
+++ b/sound/soc/codecs/da7213.h
@@ -142,6 +142,9 @@
  * Bit fields
  */
 
+/* DA7213_PLL_STATUS = 0x03 */
+#define DA7219_PLL_SRM_LOCK					(0x1 << 1)
+
 /* DA7213_SR = 0x22 */
 #define DA7213_SR_8000						(0x1 << 0)
 #define DA7213_SR_11025						(0x2 << 0)
@@ -160,10 +163,10 @@
 #define DA7213_VMID_EN						(0x1 << 7)
 
 /* DA7213_PLL_CTRL = 0x27 */
-#define DA7213_PLL_INDIV_5_10_MHZ				(0x0 << 2)
-#define DA7213_PLL_INDIV_10_20_MHZ				(0x1 << 2)
-#define DA7213_PLL_INDIV_20_40_MHZ				(0x2 << 2)
-#define DA7213_PLL_INDIV_40_54_MHZ				(0x3 << 2)
+#define DA7213_PLL_INDIV_5_TO_9_MHZ				(0x0 << 2)
+#define DA7213_PLL_INDIV_9_TO_18_MHZ				(0x1 << 2)
+#define DA7213_PLL_INDIV_18_TO_36_MHZ				(0x2 << 2)
+#define DA7213_PLL_INDIV_36_TO_54_MHZ				(0x3 << 2)
 #define DA7213_PLL_INDIV_MASK					(0x3 << 2)
 #define DA7213_PLL_MCLK_SQR_EN					(0x1 << 4)
 #define DA7213_PLL_32K_MODE					(0x1 << 5)
@@ -178,8 +181,6 @@
 #define DA7213_DAI_BCLKS_PER_WCLK_MASK				(0x3 << 0)
 #define DA7213_DAI_CLK_POL_INV					(0x1 << 2)
 #define DA7213_DAI_WCLK_POL_INV					(0x1 << 3)
-#define DA7213_DAI_CLK_EN_SLAVE_MODE				(0x0 << 7)
-#define DA7213_DAI_CLK_EN_MASTER_MODE				(0x1 << 7)
 #define DA7213_DAI_CLK_EN_MASK					(0x1 << 7)
 
 /* DA7213_DAI_CTRL = 0x29 */
@@ -412,6 +413,9 @@
 #define DA7213_DMIC_CLK_RATE_SHIFT				2
 #define DA7213_DMIC_CLK_RATE_MASK				(0x1 << 2)
 
+/* DA7213_PC_COUNT = 0x94 */
+#define DA7213_PC_FREERUN_MASK					(0x1 << 0)
+
 /* DA7213_DIG_CTRL = 0x99 */
 #define DA7213_DAC_L_INV_SHIFT					3
 #define DA7213_DAC_R_INV_SHIFT					7
@@ -495,15 +499,16 @@
 #define DA7213_ALC_AVG_ITERATIONS	5
 
 /* PLL related */
-#define DA7213_SYSCLK_MCLK		0
-#define DA7213_SYSCLK_PLL		1
-#define DA7213_PLL_FREQ_OUT_90316800	90316800
-#define DA7213_PLL_FREQ_OUT_98304000	98304000
-#define DA7213_PLL_FREQ_OUT_94310400	94310400
-#define DA7213_PLL_INDIV_5_10_MHZ_VAL	2
-#define DA7213_PLL_INDIV_10_20_MHZ_VAL	4
-#define DA7213_PLL_INDIV_20_40_MHZ_VAL	8
-#define DA7213_PLL_INDIV_40_54_MHZ_VAL	16
+#define DA7213_SYSCLK_MCLK			0
+#define DA7213_SYSCLK_PLL			1
+#define DA7213_PLL_FREQ_OUT_90316800		90316800
+#define DA7213_PLL_FREQ_OUT_98304000		98304000
+#define DA7213_PLL_FREQ_OUT_94310400		94310400
+#define DA7213_PLL_INDIV_5_TO_9_MHZ_VAL		2
+#define DA7213_PLL_INDIV_9_TO_18_MHZ_VAL	4
+#define DA7213_PLL_INDIV_18_TO_36_MHZ_VAL	8
+#define DA7213_PLL_INDIV_36_TO_54_MHZ_VAL	16
+#define DA7213_SRM_CHECK_RETRIES		8
 
 enum da7213_clk_src {
 	DA7213_CLKSRC_MCLK = 0,
diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c
index 93575f251..99ce23e11 100644
--- a/sound/soc/codecs/da7218.c
+++ b/sound/soc/codecs/da7218.c
@@ -1868,27 +1868,27 @@ static int da7218_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 
 	/* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */
 	if (da7218->mclk_rate == 32768) {
-		indiv_bits = DA7218_PLL_INDIV_2_5_MHZ;
-		indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL;
+		indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ;
+		indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL;
 	} else if (da7218->mclk_rate < 2000000) {
 		dev_err(codec->dev, "PLL input clock %d below valid range\n",
 			da7218->mclk_rate);
 		return -EINVAL;
-	} else if (da7218->mclk_rate <= 5000000) {
-		indiv_bits = DA7218_PLL_INDIV_2_5_MHZ;
-		indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL;
-	} else if (da7218->mclk_rate <= 10000000) {
-		indiv_bits = DA7218_PLL_INDIV_5_10_MHZ;
-		indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL;
-	} else if (da7218->mclk_rate <= 20000000) {
-		indiv_bits = DA7218_PLL_INDIV_10_20_MHZ;
-		indiv = DA7218_PLL_INDIV_10_20_MHZ_VAL;
-	} else if (da7218->mclk_rate <= 40000000) {
-		indiv_bits = DA7218_PLL_INDIV_20_40_MHZ;
-		indiv = DA7218_PLL_INDIV_20_40_MHZ_VAL;
+	} else if (da7218->mclk_rate <= 4500000) {
+		indiv_bits = DA7218_PLL_INDIV_2_TO_4_5_MHZ;
+		indiv = DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL;
+	} else if (da7218->mclk_rate <= 9000000) {
+		indiv_bits = DA7218_PLL_INDIV_4_5_TO_9_MHZ;
+		indiv = DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL;
+	} else if (da7218->mclk_rate <= 18000000) {
+		indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ;
+		indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL;
+	} else if (da7218->mclk_rate <= 36000000) {
+		indiv_bits = DA7218_PLL_INDIV_18_TO_36_MHZ;
+		indiv = DA7218_PLL_INDIV_18_TO_36_MHZ_VAL;
 	} else if (da7218->mclk_rate <= 54000000) {
-		indiv_bits = DA7218_PLL_INDIV_40_54_MHZ;
-		indiv = DA7218_PLL_INDIV_40_54_MHZ_VAL;
+		indiv_bits = DA7218_PLL_INDIV_36_TO_54_MHZ;
+		indiv = DA7218_PLL_INDIV_36_TO_54_MHZ_VAL;
 	} else {
 		dev_err(codec->dev, "PLL input clock %d above valid range\n",
 			da7218->mclk_rate);
diff --git a/sound/soc/codecs/da7218.h b/sound/soc/codecs/da7218.h
index c2c59049a..477cd3772 100644
--- a/sound/soc/codecs/da7218.h
+++ b/sound/soc/codecs/da7218.h
@@ -876,15 +876,11 @@
 /* DA7218_PLL_CTRL = 0x91 */
 #define DA7218_PLL_INDIV_SHIFT		0
 #define DA7218_PLL_INDIV_MASK		(0x7 << 0)
-#define DA7218_PLL_INDIV_2_5_MHZ	(0x0 << 0)
-#define DA7218_PLL_INDIV_5_10_MHZ	(0x1 << 0)
-#define DA7218_PLL_INDIV_10_20_MHZ	(0x2 << 0)
-#define DA7218_PLL_INDIV_20_40_MHZ	(0x3 << 0)
-#define DA7218_PLL_INDIV_40_54_MHZ	(0x4 << 0)
-#define DA7218_PLL_INDIV_2_10_MHZ_VAL	2
-#define DA7218_PLL_INDIV_10_20_MHZ_VAL	4
-#define DA7218_PLL_INDIV_20_40_MHZ_VAL	8
-#define DA7218_PLL_INDIV_40_54_MHZ_VAL	16
+#define DA7218_PLL_INDIV_2_TO_4_5_MHZ	(0x0 << 0)
+#define DA7218_PLL_INDIV_4_5_TO_9_MHZ	(0x1 << 0)
+#define DA7218_PLL_INDIV_9_TO_18_MHZ	(0x2 << 0)
+#define DA7218_PLL_INDIV_18_TO_36_MHZ	(0x3 << 0)
+#define DA7218_PLL_INDIV_36_TO_54_MHZ	(0x4 << 0)
 #define DA7218_PLL_MCLK_SQR_EN_SHIFT	4
 #define DA7218_PLL_MCLK_SQR_EN_MASK	(0x1 << 4)
 #define DA7218_PLL_MODE_SHIFT		6
@@ -1336,6 +1332,13 @@
 #define DA7218_PLL_FREQ_OUT_90316	90316800
 #define DA7218_PLL_FREQ_OUT_98304	98304000
 
+/* PLL Frequency Dividers */
+#define DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL	1
+#define DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL	2
+#define DA7218_PLL_INDIV_9_TO_18_MHZ_VAL	4
+#define DA7218_PLL_INDIV_18_TO_36_MHZ_VAL	8
+#define DA7218_PLL_INDIV_36_TO_54_MHZ_VAL	16
+
 /* ALC Calibration */
 #define DA7218_ALC_CALIB_DELAY_MIN	2500
 #define DA7218_ALC_CALIB_DELAY_MAX	5000
diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c
index 81c0708b8..5c93899f1 100644
--- a/sound/soc/codecs/da7219.c
+++ b/sound/soc/codecs/da7219.c
@@ -11,6 +11,7 @@
  * option) any later version.
  */
 
+#include <linux/acpi.h>
 #include <linux/clk.h>
 #include <linux/i2c.h>
 #include <linux/of_device.h>
@@ -1025,7 +1026,7 @@ static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq))
 		return 0;
 
-	if (((freq < 2000000) && (freq != 32768)) || (freq > 54000000)) {
+	if ((freq < 2000000) || (freq > 54000000)) {
 		dev_err(codec_dai->dev, "Unsupported MCLK value %d\n",
 			freq);
 		return -EINVAL;
@@ -1079,21 +1080,21 @@ static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 		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 <= 4500000) {
+		indiv_bits = DA7219_PLL_INDIV_2_TO_4_5_MHZ;
+		indiv = DA7219_PLL_INDIV_2_TO_4_5_MHZ_VAL;
+	} else if (da7219->mclk_rate <= 9000000) {
+		indiv_bits = DA7219_PLL_INDIV_4_5_TO_9_MHZ;
+		indiv = DA7219_PLL_INDIV_4_5_TO_9_MHZ_VAL;
+	} else if (da7219->mclk_rate <= 18000000) {
+		indiv_bits = DA7219_PLL_INDIV_9_TO_18_MHZ;
+		indiv = DA7219_PLL_INDIV_9_TO_18_MHZ_VAL;
+	} else if (da7219->mclk_rate <= 36000000) {
+		indiv_bits = DA7219_PLL_INDIV_18_TO_36_MHZ;
+		indiv = DA7219_PLL_INDIV_18_TO_36_MHZ_VAL;
 	} else if (da7219->mclk_rate <= 54000000) {
-		indiv_bits = DA7219_PLL_INDIV_40_54_MHZ;
-		indiv = DA7219_PLL_INDIV_40_54_MHZ_VAL;
+		indiv_bits = DA7219_PLL_INDIV_36_TO_54_MHZ;
+		indiv = DA7219_PLL_INDIV_36_TO_54_MHZ_VAL;
 	} else {
 		dev_err(codec->dev, "PLL input clock %d above valid range\n",
 			da7219->mclk_rate);
@@ -1426,6 +1427,12 @@ static const struct of_device_id da7219_of_match[] = {
 };
 MODULE_DEVICE_TABLE(of, da7219_of_match);
 
+static const struct acpi_device_id da7219_acpi_match[] = {
+	{ .id = "DLGS7219", },
+	{ }
+};
+MODULE_DEVICE_TABLE(acpi, da7219_acpi_match);
+
 static enum da7219_micbias_voltage
 	da7219_of_micbias_lvl(struct snd_soc_codec *codec, u32 val)
 {
@@ -1955,6 +1962,7 @@ static struct i2c_driver da7219_i2c_driver = {
 	.driver = {
 		.name = "da7219",
 		.of_match_table = of_match_ptr(da7219_of_match),
+		.acpi_match_table = ACPI_PTR(da7219_acpi_match),
 	},
 	.probe		= da7219_i2c_probe,
 	.remove		= da7219_i2c_remove,
diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h
index 5a787e738..ff2a2f02c 100644
--- a/sound/soc/codecs/da7219.h
+++ b/sound/soc/codecs/da7219.h
@@ -194,11 +194,11 @@
 /* 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_INDIV_2_TO_4_5_MHZ	(0x0 << 2)
+#define DA7219_PLL_INDIV_4_5_TO_9_MHZ	(0x1 << 2)
+#define DA7219_PLL_INDIV_9_TO_18_MHZ	(0x2 << 2)
+#define DA7219_PLL_INDIV_18_TO_36_MHZ	(0x3 << 2)
+#define DA7219_PLL_INDIV_36_TO_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
@@ -761,11 +761,11 @@
 #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
+#define DA7219_PLL_INDIV_2_TO_4_5_MHZ_VAL	1
+#define DA7219_PLL_INDIV_4_5_TO_9_MHZ_VAL	2
+#define DA7219_PLL_INDIV_9_TO_18_MHZ_VAL	4
+#define DA7219_PLL_INDIV_18_TO_36_MHZ_VAL	8
+#define DA7219_PLL_INDIV_36_TO_54_MHZ_VAL	16
 
 /* SRM */
 #define DA7219_SRM_CHECK_RETRIES	8
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index afa6c5db9..2086d7107 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -26,18 +26,30 @@
 #include <sound/tlv.h>
 #include "es8328.h"
 
-#define ES8328_SYSCLK_RATE_1X 11289600
-#define ES8328_SYSCLK_RATE_2X 22579200
+static const unsigned int rates_12288[] = {
+	8000, 12000, 16000, 24000, 32000, 48000, 96000,
+};
 
-/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */
-static struct {
-	int rate;
-	u8 ratio;
-} mclk_ratios[] = {
-	{ 8000, 9 },
-	{11025, 7 },
-	{22050, 4 },
-	{44100, 2 },
+static const int ratios_12288[] = {
+	10, 7, 6, 4, 3, 2, 0,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_12288 = {
+	.count	= ARRAY_SIZE(rates_12288),
+	.list	= rates_12288,
+};
+
+static const unsigned int rates_11289[] = {
+	8018, 11025, 22050, 44100, 88200,
+};
+
+static const int ratios_11289[] = {
+	9, 7, 4, 2, 0,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_11289 = {
+	.count	= ARRAY_SIZE(rates_11289),
+	.list	= rates_11289,
 };
 
 /* regulator supplies for sgtl5000, VDDD is an optional external supply */
@@ -57,16 +69,28 @@ static const char * const supply_names[ES8328_SUPPLY_NUM] = {
 	"HPVDD",
 };
 
-#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \
+#define ES8328_RATES (SNDRV_PCM_RATE_96000 | \
+		SNDRV_PCM_RATE_48000 | \
+		SNDRV_PCM_RATE_44100 | \
+		SNDRV_PCM_RATE_32000 | \
 		SNDRV_PCM_RATE_22050 | \
-		SNDRV_PCM_RATE_11025)
-#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+		SNDRV_PCM_RATE_16000 | \
+		SNDRV_PCM_RATE_11025 | \
+		SNDRV_PCM_RATE_8000)
+#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+		SNDRV_PCM_FMTBIT_S18_3LE | \
+		SNDRV_PCM_FMTBIT_S20_3LE | \
+		SNDRV_PCM_FMTBIT_S24_LE | \
+		SNDRV_PCM_FMTBIT_S32_LE)
 
 struct es8328_priv {
 	struct regmap *regmap;
 	struct clk *clk;
 	int playback_fs;
 	bool deemph;
+	int mclkdiv2;
+	const struct snd_pcm_hw_constraint_list *sysclk_constraints;
+	const int *mclk_ratios;
 	struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM];
 };
 
@@ -439,54 +463,131 @@ static int es8328_mute(struct snd_soc_dai *dai, int mute)
 			mute ? ES8328_DACCONTROL3_DACMUTE : 0);
 }
 
+static int es8328_startup(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+
+	if (es8328->sysclk_constraints)
+		snd_pcm_hw_constraint_list(substream->runtime, 0,
+				SNDRV_PCM_HW_PARAM_RATE,
+				es8328->sysclk_constraints);
+
+	return 0;
+}
+
 static int es8328_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 es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
-	int clk_rate;
 	int i;
 	int reg;
-	u8 ratio;
+	int wl;
+	int ratio;
+
+	if (!es8328->sysclk_constraints) {
+		dev_err(codec->dev, "No MCLK configured\n");
+		return -EINVAL;
+	}
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		reg = ES8328_DACCONTROL2;
 	else
 		reg = ES8328_ADCCONTROL5;
 
-	clk_rate = clk_get_rate(es8328->clk);
+	for (i = 0; i < es8328->sysclk_constraints->count; i++)
+		if (es8328->sysclk_constraints->list[i] == params_rate(params))
+			break;
 
-	if ((clk_rate != ES8328_SYSCLK_RATE_1X) &&
-		(clk_rate != ES8328_SYSCLK_RATE_2X)) {
-		dev_err(codec->dev,
-			"%s: clock is running at %d Hz, not %d or %d Hz\n",
-			 __func__, clk_rate,
-			 ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X);
+	if (i == es8328->sysclk_constraints->count) {
+		dev_err(codec->dev, "LRCLK %d unsupported with current clock\n",
+			params_rate(params));
 		return -EINVAL;
 	}
 
-	/* find master mode MCLK to sampling frequency ratio */
-	ratio = mclk_ratios[0].rate;
-	for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++)
-		if (params_rate(params) <= mclk_ratios[i].rate)
-			ratio = mclk_ratios[i].ratio;
+	ratio = es8328->mclk_ratios[i];
+	snd_soc_update_bits(codec, ES8328_MASTERMODE,
+			ES8328_MASTERMODE_MCLKDIV2,
+			es8328->mclkdiv2 ? ES8328_MASTERMODE_MCLKDIV2 : 0);
+
+	switch (params_width(params)) {
+	case 16:
+		wl = 3;
+		break;
+	case 18:
+		wl = 2;
+		break;
+	case 20:
+		wl = 1;
+		break;
+	case 24:
+		wl = 0;
+		break;
+	case 32:
+		wl = 4;
+		break;
+	default:
+		return -EINVAL;
+	}
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		snd_soc_update_bits(codec, ES8328_DACCONTROL1,
+				ES8328_DACCONTROL1_DACWL_MASK,
+				wl << ES8328_DACCONTROL1_DACWL_SHIFT);
+
 		es8328->playback_fs = params_rate(params);
 		es8328_set_deemph(codec);
-	}
+	} else
+		snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
+				ES8328_ADCCONTROL4_ADCWL_MASK,
+				wl << ES8328_ADCCONTROL4_ADCWL_SHIFT);
 
 	return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio);
 }
 
+static int es8328_set_sysclk(struct snd_soc_dai *codec_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int mclkdiv2 = 0;
+
+	switch (freq) {
+	case 0:
+		es8328->sysclk_constraints = NULL;
+		es8328->mclk_ratios = NULL;
+		break;
+	case 22579200:
+		mclkdiv2 = 1;
+		/* fallthru */
+	case 11289600:
+		es8328->sysclk_constraints = &constraints_11289;
+		es8328->mclk_ratios = ratios_11289;
+		break;
+	case 24576000:
+		mclkdiv2 = 1;
+		/* fallthru */
+	case 12288000:
+		es8328->sysclk_constraints = &constraints_12288;
+		es8328->mclk_ratios = ratios_12288;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	es8328->mclkdiv2 = mclkdiv2;
+	return 0;
+}
+
 static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
-	int clk_rate;
-	u8 mode = ES8328_DACCONTROL1_DACWL_16;
+	u8 dac_mode = 0;
+	u8 adc_mode = 0;
 
 	/* set master/slave audio interface */
 	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM)
@@ -495,13 +596,16 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	/* interface format */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
-		mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+		dac_mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+		adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_I2S;
 		break;
 	case SND_SOC_DAIFMT_RIGHT_J:
-		mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+		dac_mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+		adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_RJUST;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
-		mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+		dac_mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+		adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_LJUST;
 		break;
 	default:
 		return -EINVAL;
@@ -511,18 +615,14 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF)
 		return -EINVAL;
 
-	snd_soc_write(codec, ES8328_DACCONTROL1, mode);
-	snd_soc_write(codec, ES8328_ADCCONTROL4, mode);
+	snd_soc_update_bits(codec, ES8328_DACCONTROL1,
+			ES8328_DACCONTROL1_DACFORMAT_MASK, dac_mode);
+	snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
+			ES8328_ADCCONTROL4_ADCFORMAT_MASK, adc_mode);
 
 	/* Master serial port mode, with BCLK generated automatically */
-	clk_rate = clk_get_rate(es8328->clk);
-	if (clk_rate == ES8328_SYSCLK_RATE_1X)
-		snd_soc_write(codec, ES8328_MASTERMODE,
-				ES8328_MASTERMODE_MSC);
-	else
-		snd_soc_write(codec, ES8328_MASTERMODE,
-				ES8328_MASTERMODE_MCLKDIV2 |
-				ES8328_MASTERMODE_MSC);
+	snd_soc_update_bits(codec, ES8328_MASTERMODE,
+			ES8328_MASTERMODE_MSC, ES8328_MASTERMODE_MSC);
 
 	return 0;
 }
@@ -579,8 +679,10 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec,
 }
 
 static const struct snd_soc_dai_ops es8328_dai_ops = {
+	.startup	= es8328_startup,
 	.hw_params	= es8328_hw_params,
 	.digital_mute	= es8328_mute,
+	.set_sysclk	= es8328_set_sysclk,
 	.set_fmt	= es8328_set_dai_fmt,
 };
 
@@ -601,6 +703,7 @@ static struct snd_soc_dai_driver es8328_dai = {
 		.formats = ES8328_FORMATS,
 	},
 	.ops = &es8328_dai_ops,
+	.symmetric_rates = 1,
 };
 
 static int es8328_suspend(struct snd_soc_codec *codec)
@@ -708,6 +811,7 @@ const struct regmap_config es8328_regmap_config = {
 	.val_bits	= 8,
 	.max_register	= ES8328_REG_MAX,
 	.cache_type	= REGCACHE_RBTREE,
+	.use_single_rw	= true,
 };
 EXPORT_SYMBOL_GPL(es8328_regmap_config);
 
diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h
index 156c748c8..1a736e72a 100644
--- a/sound/soc/codecs/es8328.h
+++ b/sound/soc/codecs/es8328.h
@@ -22,7 +22,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_CONTROL1_VMIDSEL_50k (1 << 0)
 #define ES8328_CONTROL1_VMIDSEL_500k (2 << 0)
 #define ES8328_CONTROL1_VMIDSEL_5k (3 << 0)
-#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0)
+#define ES8328_CONTROL1_VMIDSEL_MASK (3 << 0)
 #define ES8328_CONTROL1_ENREF (1 << 2)
 #define ES8328_CONTROL1_SEQEN (1 << 3)
 #define ES8328_CONTROL1_SAMEFS (1 << 4)
@@ -84,7 +84,20 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_ADCCONTROL1	0x09
 #define ES8328_ADCCONTROL2	0x0a
 #define ES8328_ADCCONTROL3	0x0b
+
 #define ES8328_ADCCONTROL4	0x0c
+#define ES8328_ADCCONTROL4_ADCFORMAT_MASK (3 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_I2S (0 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_PCM (3 << 0)
+#define ES8328_ADCCONTROL4_ADCWL_SHIFT 2
+#define ES8328_ADCCONTROL4_ADCWL_MASK (7 << 2)
+#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK2 (0 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK1 (1 << 5)
+
 #define ES8328_ADCCONTROL5	0x0d
 #define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0)
 
@@ -109,15 +122,13 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_ADCCONTROL14	0x16
 
 #define ES8328_DACCONTROL1	0x17
+#define ES8328_DACCONTROL1_DACFORMAT_MASK (3 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1)
-#define ES8328_DACCONTROL1_DACWL_24 (0 << 3)
-#define ES8328_DACCONTROL1_DACWL_20 (1 << 3)
-#define ES8328_DACCONTROL1_DACWL_18 (2 << 3)
-#define ES8328_DACCONTROL1_DACWL_16 (3 << 3)
-#define ES8328_DACCONTROL1_DACWL_32 (4 << 3)
+#define ES8328_DACCONTROL1_DACWL_SHIFT 3
+#define ES8328_DACCONTROL1_DACWL_MASK (7 << 3)
 #define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6)
 #define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6)
 #define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6)
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index aaa038ffc..2abb742fc 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -29,6 +29,7 @@
 #include <sound/hdaudio_ext.h>
 #include <sound/hda_i915.h>
 #include <sound/pcm_drm_eld.h>
+#include <sound/hda_chmap.h>
 #include "../../hda/local.h"
 #include "hdac_hdmi.h"
 
@@ -60,11 +61,17 @@ struct hdac_hdmi_cvt {
 	struct hdac_hdmi_cvt_params params;
 };
 
+/* Currently only spk_alloc, more to be added */
+struct hdac_hdmi_parsed_eld {
+	u8 spk_alloc;
+};
+
 struct hdac_hdmi_eld {
 	bool	monitor_present;
 	bool	eld_valid;
 	int	eld_size;
 	char    eld_buffer[ELD_MAX_SIZE];
+	struct	hdac_hdmi_parsed_eld info;
 };
 
 struct hdac_hdmi_pin {
@@ -76,6 +83,10 @@ struct hdac_hdmi_pin {
 	struct hdac_ext_device *edev;
 	int repoll_count;
 	struct delayed_work work;
+	struct mutex lock;
+	bool chmap_set;
+	unsigned char chmap[8]; /* ALSA API channel-map */
+	int channels; /* current number of channels */
 };
 
 struct hdac_hdmi_pcm {
@@ -100,8 +111,22 @@ struct hdac_hdmi_priv {
 	int num_pin;
 	int num_cvt;
 	struct mutex pin_mutex;
+	struct hdac_chmap chmap;
 };
 
+static struct hdac_hdmi_pcm *get_hdmi_pcm_from_id(struct hdac_hdmi_priv *hdmi,
+						int pcm_idx)
+{
+	struct hdac_hdmi_pcm *pcm;
+
+	list_for_each_entry(pcm, &hdmi->pcm_list, head) {
+		if (pcm->pcm_id == pcm_idx)
+			return pcm;
+	}
+
+	return NULL;
+}
+
 static inline struct hdac_ext_device *to_hda_ext_device(struct device *dev)
 {
 	struct hdac_device *hdac = dev_to_hdac_dev(dev);
@@ -278,26 +303,31 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac,
 	int i;
 	const u8 *eld_buf;
 	u8 conn_type;
-	int channels = 2;
+	int channels, ca;
 
 	list_for_each_entry(pin, &hdmi->pin_list, head) {
 		if (pin->nid == pin_nid)
 			break;
 	}
 
+	ca = snd_hdac_channel_allocation(&hdac->hdac, pin->eld.info.spk_alloc,
+			pin->channels, pin->chmap_set, true, pin->chmap);
+
+	channels = snd_hdac_get_active_channels(ca);
+	hdmi->chmap.ops.set_channel_count(&hdac->hdac, cvt_nid, channels);
+
+	snd_hdac_setup_channel_mapping(&hdmi->chmap, pin->nid, false, ca,
+				pin->channels, pin->chmap, pin->chmap_set);
+
 	eld_buf = pin->eld.eld_buffer;
 	conn_type = drm_eld_get_conn_type(eld_buf);
 
-	/* setup channel count */
-	snd_hdac_codec_write(&hdac->hdac, cvt_nid, 0,
-			    AC_VERB_SET_CVT_CHAN_COUNT, channels - 1);
-
 	switch (conn_type) {
 	case DRM_ELD_CONN_TYPE_HDMI:
 		hdmi_audio_infoframe_init(&frame);
 
-		/* Default stereo for now */
 		frame.channels = channels;
+		frame.channel_allocation = ca;
 
 		ret = hdmi_audio_infoframe_pack(&frame, buffer, sizeof(buffer));
 		if (ret < 0)
@@ -311,7 +341,7 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac,
 		dp_ai.len	= 0x1b;
 		dp_ai.ver	= 0x11 << 2;
 		dp_ai.CC02_CT47	= channels - 1;
-		dp_ai.CA	= 0;
+		dp_ai.CA	= ca;
 
 		dip = (u8 *)&dp_ai;
 		break;
@@ -370,17 +400,23 @@ static int hdac_hdmi_playback_prepare(struct snd_pcm_substream *substream,
 	struct hdac_ext_device *hdac = snd_soc_dai_get_drvdata(dai);
 	struct hdac_hdmi_priv *hdmi = hdac->private_data;
 	struct hdac_hdmi_dai_pin_map *dai_map;
+	struct hdac_hdmi_pin *pin;
 	struct hdac_ext_dma_params *dd;
 	int ret;
 
 	dai_map = &hdmi->dai_map[dai->id];
+	pin = dai_map->pin;
 
 	dd = (struct hdac_ext_dma_params *)snd_soc_dai_get_dma_data(dai, substream);
 	dev_dbg(&hdac->hdac.dev, "stream tag from cpu dai %d format in cvt 0x%x\n",
 			dd->stream_tag,	dd->format);
 
+	mutex_lock(&pin->lock);
+	pin->channels = substream->runtime->channels;
+
 	ret = hdac_hdmi_setup_audio_infoframe(hdac, dai_map->cvt->nid,
 						dai_map->pin->nid);
+	mutex_unlock(&pin->lock);
 	if (ret < 0)
 		return ret;
 
@@ -640,6 +676,12 @@ static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream,
 		snd_hdac_codec_write(&hdac->hdac, dai_map->pin->nid, 0,
 			AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
 
+		mutex_lock(&dai_map->pin->lock);
+		dai_map->pin->chmap_set = false;
+		memset(dai_map->pin->chmap, 0, sizeof(dai_map->pin->chmap));
+		dai_map->pin->channels = 0;
+		mutex_unlock(&dai_map->pin->lock);
+
 		dai_map->pin = NULL;
 	}
 }
@@ -647,10 +689,19 @@ static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream,
 static int
 hdac_hdmi_query_cvt_params(struct hdac_device *hdac, struct hdac_hdmi_cvt *cvt)
 {
+	unsigned int chans;
+	struct hdac_ext_device *edev = to_ehdac_device(hdac);
+	struct hdac_hdmi_priv *hdmi = edev->private_data;
 	int err;
 
-	/* Only stereo supported as of now */
-	cvt->params.channels_min = cvt->params.channels_max = 2;
+	chans = get_wcaps(hdac, cvt->nid);
+	chans = get_wcaps_channels(chans);
+
+	cvt->params.channels_min = 2;
+
+	cvt->params.channels_max = chans;
+	if (chans > hdmi->chmap.channels_max)
+		hdmi->chmap.channels_max = chans;
 
 	err = snd_hdac_query_supported_pcm(hdac, cvt->nid,
 			&cvt->params.rates,
@@ -1008,6 +1059,12 @@ static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid)
 	return hdac_hdmi_query_cvt_params(&edev->hdac, cvt);
 }
 
+static void hdac_hdmi_parse_eld(struct hdac_ext_device *edev,
+			struct hdac_hdmi_pin *pin)
+{
+	pin->eld.info.spk_alloc = pin->eld.eld_buffer[DRM_ELD_SPEAKER];
+}
+
 static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll)
 {
 	struct hdac_ext_device *edev = pin->edev;
@@ -1065,6 +1122,7 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll)
 
 				snd_jack_report(pcm->jack, SND_JACK_AVOUT);
 			}
+			hdac_hdmi_parse_eld(edev, pin);
 
 			print_hex_dump_bytes("ELD: ", DUMP_PREFIX_OFFSET,
 					pin->eld.eld_buffer, pin->eld.eld_size);
@@ -1123,6 +1181,7 @@ static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid)
 	hdmi->num_pin++;
 
 	pin->edev = edev;
+	mutex_init(&pin->lock);
 	INIT_DELAYED_WORK(&pin->work, hdac_hdmi_repoll_eld);
 
 	return 0;
@@ -1342,6 +1401,19 @@ static struct i915_audio_component_audio_ops aops = {
 	.pin_eld_notify	= hdac_hdmi_eld_notify_cb,
 };
 
+static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card,
+						int device)
+{
+	struct snd_soc_pcm_runtime *rtd;
+
+	list_for_each_entry(rtd, &card->rtd_list, list) {
+		if (rtd->pcm && (rtd->pcm->device == device))
+			return rtd->pcm;
+	}
+
+	return NULL;
+}
+
 int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device)
 {
 	char jack_name[NAME_SIZE];
@@ -1351,6 +1423,8 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device)
 		snd_soc_component_get_dapm(&codec->component);
 	struct hdac_hdmi_priv *hdmi = edev->private_data;
 	struct hdac_hdmi_pcm *pcm;
+	struct snd_pcm *snd_pcm;
+	int err;
 
 	/*
 	 * this is a new PCM device, create new pcm and
@@ -1362,6 +1436,18 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device)
 	pcm->pcm_id = device;
 	pcm->cvt = hdmi->dai_map[dai->id].cvt;
 
+	snd_pcm = hdac_hdmi_get_pcm_from_id(dai->component->card, device);
+	if (snd_pcm) {
+		err = snd_hdac_add_chmap_ctls(snd_pcm, device, &hdmi->chmap);
+		if (err < 0) {
+			dev_err(&edev->hdac.dev,
+				"chmap control add failed with err: %d for pcm: %d\n",
+				err, device);
+			kfree(pcm);
+			return err;
+		}
+	}
+
 	list_add_tail(&pcm->head, &hdmi->pcm_list);
 
 	sprintf(jack_name, "HDMI/DP, pcm=%d Jack", device);
@@ -1378,10 +1464,23 @@ static int hdmi_codec_probe(struct snd_soc_codec *codec)
 	struct snd_soc_dapm_context *dapm =
 		snd_soc_component_get_dapm(&codec->component);
 	struct hdac_hdmi_pin *pin;
+	struct hdac_ext_link *hlink = NULL;
 	int ret;
 
 	edev->scodec = codec;
 
+	/*
+	 * hold the ref while we probe, also no need to drop the ref on
+	 * exit, we call pm_runtime_suspend() so that will do for us
+	 */
+	hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev));
+	if (!hlink) {
+		dev_err(&edev->hdac.dev, "hdac link not found\n");
+		return -EIO;
+	}
+
+	snd_hdac_ext_bus_link_get(edev->ebus, hlink);
+
 	ret = create_fill_widget_route_map(dapm);
 	if (ret < 0)
 		return ret;
@@ -1475,19 +1574,88 @@ static struct snd_soc_codec_driver hdmi_hda_codec = {
 	.idle_bias_off	= true,
 };
 
+static void hdac_hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx,
+					unsigned char *chmap)
+{
+	struct hdac_ext_device *edev = to_ehdac_device(hdac);
+	struct hdac_hdmi_priv *hdmi = edev->private_data;
+	struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx);
+	struct hdac_hdmi_pin *pin = pcm->pin;
+
+	/* chmap is already set to 0 in caller */
+	if (!pin)
+		return;
+
+	memcpy(chmap, pin->chmap, ARRAY_SIZE(pin->chmap));
+}
+
+static void hdac_hdmi_set_chmap(struct hdac_device *hdac, int pcm_idx,
+				unsigned char *chmap, int prepared)
+{
+	struct hdac_ext_device *edev = to_ehdac_device(hdac);
+	struct hdac_hdmi_priv *hdmi = edev->private_data;
+	struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx);
+	struct hdac_hdmi_pin *pin = pcm->pin;
+
+	mutex_lock(&pin->lock);
+	pin->chmap_set = true;
+	memcpy(pin->chmap, chmap, ARRAY_SIZE(pin->chmap));
+	if (prepared)
+		hdac_hdmi_setup_audio_infoframe(edev, pcm->cvt->nid, pin->nid);
+	mutex_unlock(&pin->lock);
+}
+
+static bool is_hdac_hdmi_pcm_attached(struct hdac_device *hdac, int pcm_idx)
+{
+	struct hdac_ext_device *edev = to_ehdac_device(hdac);
+	struct hdac_hdmi_priv *hdmi = edev->private_data;
+	struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx);
+	struct hdac_hdmi_pin *pin = pcm->pin;
+
+	return pin ? true:false;
+}
+
+static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx)
+{
+	struct hdac_ext_device *edev = to_ehdac_device(hdac);
+	struct hdac_hdmi_priv *hdmi = edev->private_data;
+	struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx);
+	struct hdac_hdmi_pin *pin = pcm->pin;
+
+	if (!pin || !pin->eld.eld_valid)
+		return 0;
+
+	return pin->eld.info.spk_alloc;
+}
+
 static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev)
 {
 	struct hdac_device *codec = &edev->hdac;
 	struct hdac_hdmi_priv *hdmi_priv;
 	struct snd_soc_dai_driver *hdmi_dais = NULL;
+	struct hdac_ext_link *hlink = NULL;
 	int num_dais = 0;
 	int ret = 0;
 
+	/* hold the ref while we probe */
+	hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev));
+	if (!hlink) {
+		dev_err(&edev->hdac.dev, "hdac link not found\n");
+		return -EIO;
+	}
+
+	snd_hdac_ext_bus_link_get(edev->ebus, hlink);
+
 	hdmi_priv = devm_kzalloc(&codec->dev, sizeof(*hdmi_priv), GFP_KERNEL);
 	if (hdmi_priv == NULL)
 		return -ENOMEM;
 
 	edev->private_data = hdmi_priv;
+	snd_hdac_register_chmap_ops(codec, &hdmi_priv->chmap);
+	hdmi_priv->chmap.ops.get_chmap = hdac_hdmi_get_chmap;
+	hdmi_priv->chmap.ops.set_chmap = hdac_hdmi_set_chmap;
+	hdmi_priv->chmap.ops.is_pcm_attached = is_hdac_hdmi_pcm_attached;
+	hdmi_priv->chmap.ops.get_spk_alloc = hdac_hdmi_get_spk_alloc;
 
 	dev_set_drvdata(&codec->dev, edev);
 
@@ -1516,8 +1684,12 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev)
 	}
 
 	/* ASoC specific initialization */
-	return snd_soc_register_codec(&codec->dev, &hdmi_hda_codec,
-			hdmi_dais, num_dais);
+	ret = snd_soc_register_codec(&codec->dev, &hdmi_hda_codec,
+					hdmi_dais, num_dais);
+
+	snd_hdac_ext_bus_link_put(edev->ebus, hlink);
+
+	return ret;
 }
 
 static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev)
@@ -1556,6 +1728,8 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
 	struct hdac_ext_device *edev = to_hda_ext_device(dev);
 	struct hdac_device *hdac = &edev->hdac;
 	struct hdac_bus *bus = hdac->bus;
+	struct hdac_ext_bus *ebus = hbus_to_ebus(bus);
+	struct hdac_ext_link *hlink = NULL;
 	int err;
 
 	dev_dbg(dev, "Enter: %s\n", __func__);
@@ -1579,6 +1753,14 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
 		return err;
 	}
 
+	hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev));
+	if (!hlink) {
+		dev_err(dev, "hdac link not found\n");
+		return -EIO;
+	}
+
+	snd_hdac_ext_bus_link_put(ebus, hlink);
+
 	return 0;
 }
 
@@ -1587,6 +1769,8 @@ static int hdac_hdmi_runtime_resume(struct device *dev)
 	struct hdac_ext_device *edev = to_hda_ext_device(dev);
 	struct hdac_device *hdac = &edev->hdac;
 	struct hdac_bus *bus = hdac->bus;
+	struct hdac_ext_bus *ebus = hbus_to_ebus(bus);
+	struct hdac_ext_link *hlink = NULL;
 	int err;
 
 	dev_dbg(dev, "Enter: %s\n", __func__);
@@ -1595,6 +1779,14 @@ static int hdac_hdmi_runtime_resume(struct device *dev)
 	if (!bus)
 		return 0;
 
+	hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev));
+	if (!hlink) {
+		dev_err(dev, "hdac link not found\n");
+		return -EIO;
+	}
+
+	snd_hdac_ext_bus_link_get(ebus, hlink);
+
 	err = snd_hdac_display_power(bus, true);
 	if (err < 0) {
 		dev_err(bus->dev, "Cannot turn on display power on i915\n");
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
new file mode 100644
index 000000000..8e36e883e
--- /dev/null
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -0,0 +1,432 @@
+/*
+ * ALSA SoC codec for HDMI encoder drivers
+ * Copyright (C) 2015 Texas Instruments Incorporated - http://www.ti.com/
+ * Author: Jyri Sarha <jsarha@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.
+ */
+#include <linux/module.h>
+#include <linux/string.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pcm_drm_eld.h>
+#include <sound/hdmi-codec.h>
+#include <sound/pcm_iec958.h>
+
+#include <drm/drm_crtc.h> /* This is only to get MAX_ELD_BYTES */
+
+struct hdmi_codec_priv {
+	struct hdmi_codec_pdata hcd;
+	struct snd_soc_dai_driver *daidrv;
+	struct hdmi_codec_daifmt daifmt[2];
+	struct mutex current_stream_lock;
+	struct snd_pcm_substream *current_stream;
+	struct snd_pcm_hw_constraint_list ratec;
+	uint8_t eld[MAX_ELD_BYTES];
+};
+
+static const struct snd_soc_dapm_widget hdmi_widgets[] = {
+	SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route hdmi_routes[] = {
+	{ "TX", NULL, "Playback" },
+};
+
+enum {
+	DAI_ID_I2S = 0,
+	DAI_ID_SPDIF,
+};
+
+static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_info *uinfo)
+{
+	struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+	struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component);
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+	uinfo->count = sizeof(hcp->eld);
+
+	return 0;
+}
+
+static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+	struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component);
+
+	memcpy(ucontrol->value.bytes.data, hcp->eld, sizeof(hcp->eld));
+
+	return 0;
+}
+
+static const struct snd_kcontrol_new hdmi_controls[] = {
+	{
+		.access = SNDRV_CTL_ELEM_ACCESS_READ |
+			  SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+		.name = "ELD",
+		.info = hdmi_eld_ctl_info,
+		.get = hdmi_eld_ctl_get,
+	},
+};
+
+static int hdmi_codec_new_stream(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+	int ret = 0;
+
+	mutex_lock(&hcp->current_stream_lock);
+	if (!hcp->current_stream) {
+		hcp->current_stream = substream;
+	} else if (hcp->current_stream != substream) {
+		dev_err(dai->dev, "Only one simultaneous stream supported!\n");
+		ret = -EINVAL;
+	}
+	mutex_unlock(&hcp->current_stream_lock);
+
+	return ret;
+}
+
+static int hdmi_codec_startup(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
+{
+	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+	int ret = 0;
+
+	dev_dbg(dai->dev, "%s()\n", __func__);
+
+	ret = hdmi_codec_new_stream(substream, dai);
+	if (ret)
+		return ret;
+
+	if (hcp->hcd.ops->audio_startup) {
+		ret = hcp->hcd.ops->audio_startup(dai->dev->parent);
+		if (ret) {
+			mutex_lock(&hcp->current_stream_lock);
+			hcp->current_stream = NULL;
+			mutex_unlock(&hcp->current_stream_lock);
+			return ret;
+		}
+	}
+
+	if (hcp->hcd.ops->get_eld) {
+		ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->eld,
+					    sizeof(hcp->eld));
+
+		if (!ret) {
+			ret = snd_pcm_hw_constraint_eld(substream->runtime,
+							hcp->eld);
+			if (ret)
+				return ret;
+		}
+	}
+	return 0;
+}
+
+static void hdmi_codec_shutdown(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+
+	dev_dbg(dai->dev, "%s()\n", __func__);
+
+	WARN_ON(hcp->current_stream != substream);
+
+	hcp->hcd.ops->audio_shutdown(dai->dev->parent);
+
+	mutex_lock(&hcp->current_stream_lock);
+	hcp->current_stream = NULL;
+	mutex_unlock(&hcp->current_stream_lock);
+}
+
+static int hdmi_codec_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+	struct hdmi_codec_params hp = {
+		.iec = {
+			.status = { 0 },
+			.subcode = { 0 },
+			.pad = 0,
+			.dig_subframe = { 0 },
+		}
+	};
+	int ret;
+
+	dev_dbg(dai->dev, "%s() width %d rate %d channels %d\n", __func__,
+		params_width(params), params_rate(params),
+		params_channels(params));
+
+	if (params_width(params) > 24)
+		params->msbits = 24;
+
+	ret = snd_pcm_create_iec958_consumer_hw_params(params, hp.iec.status,
+						       sizeof(hp.iec.status));
+	if (ret < 0) {
+		dev_err(dai->dev, "Creating IEC958 channel status failed %d\n",
+			ret);
+		return ret;
+	}
+
+	ret = hdmi_codec_new_stream(substream, dai);
+	if (ret)
+		return ret;
+
+	hdmi_audio_infoframe_init(&hp.cea);
+	hp.cea.channels = params_channels(params);
+	hp.cea.coding_type = HDMI_AUDIO_CODING_TYPE_STREAM;
+	hp.cea.sample_size = HDMI_AUDIO_SAMPLE_SIZE_STREAM;
+	hp.cea.sample_frequency = HDMI_AUDIO_SAMPLE_FREQUENCY_STREAM;
+
+	hp.sample_width = params_width(params);
+	hp.sample_rate = params_rate(params);
+	hp.channels = params_channels(params);
+
+	return hcp->hcd.ops->hw_params(dai->dev->parent, &hcp->daifmt[dai->id],
+				       &hp);
+}
+
+static int hdmi_codec_set_fmt(struct snd_soc_dai *dai,
+			      unsigned int fmt)
+{
+	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+	struct hdmi_codec_daifmt cf = { 0 };
+	int ret = 0;
+
+	dev_dbg(dai->dev, "%s()\n", __func__);
+
+	if (dai->id == DAI_ID_SPDIF) {
+		cf.fmt = HDMI_SPDIF;
+	} else {
+		switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+		case SND_SOC_DAIFMT_CBM_CFM:
+			cf.bit_clk_master = 1;
+			cf.frame_clk_master = 1;
+			break;
+		case SND_SOC_DAIFMT_CBS_CFM:
+			cf.frame_clk_master = 1;
+			break;
+		case SND_SOC_DAIFMT_CBM_CFS:
+			cf.bit_clk_master = 1;
+			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_NB_IF:
+			cf.frame_clk_inv = 1;
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			cf.bit_clk_inv = 1;
+			break;
+		case SND_SOC_DAIFMT_IB_IF:
+			cf.frame_clk_inv = 1;
+			cf.bit_clk_inv = 1;
+			break;
+		}
+
+		switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+		case SND_SOC_DAIFMT_I2S:
+			cf.fmt = HDMI_I2S;
+			break;
+		case SND_SOC_DAIFMT_DSP_A:
+			cf.fmt = HDMI_DSP_A;
+			break;
+		case SND_SOC_DAIFMT_DSP_B:
+			cf.fmt = HDMI_DSP_B;
+			break;
+		case SND_SOC_DAIFMT_RIGHT_J:
+			cf.fmt = HDMI_RIGHT_J;
+			break;
+		case SND_SOC_DAIFMT_LEFT_J:
+			cf.fmt = HDMI_LEFT_J;
+			break;
+		case SND_SOC_DAIFMT_AC97:
+			cf.fmt = HDMI_AC97;
+			break;
+		default:
+			dev_err(dai->dev, "Invalid DAI interface format\n");
+			return -EINVAL;
+		}
+	}
+
+	hcp->daifmt[dai->id] = cf;
+
+	return ret;
+}
+
+static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+
+	dev_dbg(dai->dev, "%s()\n", __func__);
+
+	if (hcp->hcd.ops->digital_mute)
+		return hcp->hcd.ops->digital_mute(dai->dev->parent, mute);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops hdmi_dai_ops = {
+	.startup	= hdmi_codec_startup,
+	.shutdown	= hdmi_codec_shutdown,
+	.hw_params	= hdmi_codec_hw_params,
+	.set_fmt	= hdmi_codec_set_fmt,
+	.digital_mute	= hdmi_codec_digital_mute,
+};
+
+
+#define HDMI_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)
+
+#define SPDIF_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\
+			 SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\
+			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\
+			 SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE)
+
+/*
+ * This list is only for formats allowed on the I2S bus. So there is
+ * some formats listed that are not supported by HDMI interface. For
+ * instance allowing the 32-bit formats enables 24-precision with CPU
+ * DAIs that do not support 24-bit formats. If the extra formats cause
+ * problems, we should add the video side driver an option to disable
+ * them.
+ */
+#define I2S_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\
+			 SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\
+			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\
+			 SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE |\
+			 SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE)
+
+static struct snd_soc_dai_driver hdmi_i2s_dai = {
+	.name = "i2s-hifi",
+	.id = DAI_ID_I2S,
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 8,
+		.rates = HDMI_RATES,
+		.formats = I2S_FORMATS,
+		.sig_bits = 24,
+	},
+	.ops = &hdmi_dai_ops,
+};
+
+static const struct snd_soc_dai_driver hdmi_spdif_dai = {
+	.name = "spdif-hifi",
+	.id = DAI_ID_SPDIF,
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = HDMI_RATES,
+		.formats = SPDIF_FORMATS,
+	},
+	.ops = &hdmi_dai_ops,
+};
+
+static struct snd_soc_codec_driver hdmi_codec = {
+	.controls = hdmi_controls,
+	.num_controls = ARRAY_SIZE(hdmi_controls),
+	.dapm_widgets = hdmi_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(hdmi_widgets),
+	.dapm_routes = hdmi_routes,
+	.num_dapm_routes = ARRAY_SIZE(hdmi_routes),
+};
+
+static int hdmi_codec_probe(struct platform_device *pdev)
+{
+	struct hdmi_codec_pdata *hcd = pdev->dev.platform_data;
+	struct device *dev = &pdev->dev;
+	struct hdmi_codec_priv *hcp;
+	int dai_count, i = 0;
+	int ret;
+
+	dev_dbg(dev, "%s()\n", __func__);
+
+	if (!hcd) {
+		dev_err(dev, "%s: No plalform data\n", __func__);
+		return -EINVAL;
+	}
+
+	dai_count = hcd->i2s + hcd->spdif;
+	if (dai_count < 1 || !hcd->ops || !hcd->ops->hw_params ||
+	    !hcd->ops->audio_shutdown) {
+		dev_err(dev, "%s: Invalid parameters\n", __func__);
+		return -EINVAL;
+	}
+
+	hcp = devm_kzalloc(dev, sizeof(*hcp), GFP_KERNEL);
+	if (!hcp)
+		return -ENOMEM;
+
+	hcp->hcd = *hcd;
+	mutex_init(&hcp->current_stream_lock);
+
+	hcp->daidrv = devm_kzalloc(dev, dai_count * sizeof(*hcp->daidrv),
+				   GFP_KERNEL);
+	if (!hcp->daidrv)
+		return -ENOMEM;
+
+	if (hcd->i2s) {
+		hcp->daidrv[i] = hdmi_i2s_dai;
+		hcp->daidrv[i].playback.channels_max =
+			hcd->max_i2s_channels;
+		i++;
+	}
+
+	if (hcd->spdif)
+		hcp->daidrv[i] = hdmi_spdif_dai;
+
+	ret = snd_soc_register_codec(dev, &hdmi_codec, hcp->daidrv,
+				     dai_count);
+	if (ret) {
+		dev_err(dev, "%s: snd_soc_register_codec() failed (%d)\n",
+			__func__, ret);
+		return ret;
+	}
+
+	dev_set_drvdata(dev, hcp);
+	return 0;
+}
+
+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 = HDMI_CODEC_DRV_NAME,
+	},
+	.probe = hdmi_codec_probe,
+	.remove = hdmi_codec_remove,
+};
+
+module_platform_driver(hdmi_codec_driver);
+
+MODULE_AUTHOR("Jyri Sarha <jsarha@ti.com>");
+MODULE_DESCRIPTION("HDMI Audio Codec Driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" HDMI_CODEC_DRV_NAME);
diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c
new file mode 100644
index 000000000..cf0a39bb6
--- /dev/null
+++ b/sound/soc/codecs/max98371.c
@@ -0,0 +1,441 @@
+/*
+ * max98371.c -- ALSA SoC Stereo MAX98371 driver
+ *
+ * Copyright 2015-16 Maxim Integrated Products
+ *
+ * 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/i2c.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include "max98371.h"
+
+static const char *const monomix_text[] = {
+	"Left", "Right", "LeftRightDiv2",
+};
+
+static const char *const hpf_cutoff_txt[] = {
+	"Disable", "DC Block", "50Hz",
+	"100Hz", "200Hz", "400Hz", "800Hz",
+};
+
+static SOC_ENUM_SINGLE_DECL(max98371_monomix, MAX98371_MONOMIX_CFG, 0,
+		monomix_text);
+
+static SOC_ENUM_SINGLE_DECL(max98371_hpf_cutoff, MAX98371_HPF, 0,
+		hpf_cutoff_txt);
+
+static const DECLARE_TLV_DB_RANGE(max98371_dht_min_gain,
+	0, 1, TLV_DB_SCALE_ITEM(537, 66, 0),
+	2, 3, TLV_DB_SCALE_ITEM(677, 82, 0),
+	4, 5, TLV_DB_SCALE_ITEM(852, 104, 0),
+	6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0),
+	8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0),
+	10, 11, TLV_DB_SCALE_ITEM(1699, 101, 0),
+);
+
+static const DECLARE_TLV_DB_RANGE(max98371_dht_max_gain,
+	0, 1, TLV_DB_SCALE_ITEM(537, 66, 0),
+	2, 3, TLV_DB_SCALE_ITEM(677, 82, 0),
+	4, 5, TLV_DB_SCALE_ITEM(852, 104, 0),
+	6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0),
+	8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0),
+	10, 11, TLV_DB_SCALE_ITEM(1699, 208, 0),
+);
+
+static const DECLARE_TLV_DB_RANGE(max98371_dht_rot_gain,
+	0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0),
+	2, 6, TLV_DB_SCALE_ITEM(-100, -100, 0),
+	7, 8, TLV_DB_SCALE_ITEM(-800, -200, 0),
+	9, 11, TLV_DB_SCALE_ITEM(-1200, -300, 0),
+	12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0),
+	14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0),
+);
+
+static const struct reg_default max98371_reg[] = {
+	{ 0x01, 0x00 },
+	{ 0x02, 0x00 },
+	{ 0x03, 0x00 },
+	{ 0x04, 0x00 },
+	{ 0x05, 0x00 },
+	{ 0x06, 0x00 },
+	{ 0x07, 0x00 },
+	{ 0x08, 0x00 },
+	{ 0x09, 0x00 },
+	{ 0x0A, 0x00 },
+	{ 0x10, 0x06 },
+	{ 0x11, 0x08 },
+	{ 0x14, 0x80 },
+	{ 0x15, 0x00 },
+	{ 0x16, 0x00 },
+	{ 0x18, 0x00 },
+	{ 0x19, 0x00 },
+	{ 0x1C, 0x00 },
+	{ 0x1D, 0x00 },
+	{ 0x1E, 0x00 },
+	{ 0x1F, 0x00 },
+	{ 0x20, 0x00 },
+	{ 0x21, 0x00 },
+	{ 0x22, 0x00 },
+	{ 0x23, 0x00 },
+	{ 0x24, 0x00 },
+	{ 0x25, 0x00 },
+	{ 0x26, 0x00 },
+	{ 0x27, 0x00 },
+	{ 0x28, 0x00 },
+	{ 0x29, 0x00 },
+	{ 0x2A, 0x00 },
+	{ 0x2B, 0x00 },
+	{ 0x2C, 0x00 },
+	{ 0x2D, 0x00 },
+	{ 0x2E, 0x0B },
+	{ 0x31, 0x00 },
+	{ 0x32, 0x18 },
+	{ 0x33, 0x00 },
+	{ 0x34, 0x00 },
+	{ 0x36, 0x00 },
+	{ 0x37, 0x00 },
+	{ 0x38, 0x00 },
+	{ 0x39, 0x00 },
+	{ 0x3A, 0x00 },
+	{ 0x3B, 0x00 },
+	{ 0x3C, 0x00 },
+	{ 0x3D, 0x00 },
+	{ 0x3E, 0x00 },
+	{ 0x3F, 0x00 },
+	{ 0x40, 0x00 },
+	{ 0x41, 0x00 },
+	{ 0x42, 0x00 },
+	{ 0x43, 0x00 },
+	{ 0x4A, 0x00 },
+	{ 0x4B, 0x00 },
+	{ 0x4C, 0x00 },
+	{ 0x4D, 0x00 },
+	{ 0x4E, 0x00 },
+	{ 0x50, 0x00 },
+	{ 0x51, 0x00 },
+	{ 0x55, 0x00 },
+	{ 0x58, 0x00 },
+	{ 0x59, 0x00 },
+	{ 0x5C, 0x00 },
+	{ 0xFF, 0x43 },
+};
+
+static bool max98371_volatile_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case MAX98371_IRQ_CLEAR1:
+	case MAX98371_IRQ_CLEAR2:
+	case MAX98371_IRQ_CLEAR3:
+	case MAX98371_VERSION:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool max98371_readable_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case MAX98371_SOFT_RESET:
+		return false;
+	default:
+		return true;
+	}
+};
+
+static const DECLARE_TLV_DB_RANGE(max98371_gain_tlv,
+	0, 7, TLV_DB_SCALE_ITEM(0, 50, 0),
+	8, 10, TLV_DB_SCALE_ITEM(400, 100, 0)
+);
+
+static const DECLARE_TLV_DB_RANGE(max98371_noload_gain_tlv,
+	0, 11, TLV_DB_SCALE_ITEM(950, 100, 0),
+);
+
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1);
+
+static const struct snd_kcontrol_new max98371_snd_controls[] = {
+	SOC_SINGLE_TLV("Speaker Volume", MAX98371_GAIN,
+			MAX98371_GAIN_SHIFT, (1<<MAX98371_GAIN_WIDTH)-1, 0,
+			max98371_gain_tlv),
+	SOC_SINGLE_TLV("Digital Volume", MAX98371_DIGITAL_GAIN, 0,
+			(1<<MAX98371_DIGITAL_GAIN_WIDTH)-1, 1, digital_tlv),
+	SOC_SINGLE_TLV("Speaker DHT Max Volume", MAX98371_GAIN,
+			0, (1<<MAX98371_DHT_MAX_WIDTH)-1, 0,
+			max98371_dht_max_gain),
+	SOC_SINGLE_TLV("Speaker DHT Min Volume", MAX98371_DHT_GAIN,
+			0, (1<<MAX98371_DHT_GAIN_WIDTH)-1, 0,
+			max98371_dht_min_gain),
+	SOC_SINGLE_TLV("Speaker DHT Rotation Volume", MAX98371_DHT_GAIN,
+			0, (1<<MAX98371_DHT_ROT_WIDTH)-1, 0,
+			max98371_dht_rot_gain),
+	SOC_SINGLE("DHT Attack Step", MAX98371_DHT, MAX98371_DHT_STEP, 3, 0),
+	SOC_SINGLE("DHT Attack Rate", MAX98371_DHT, 0, 7, 0),
+	SOC_ENUM("Monomix Select", max98371_monomix),
+	SOC_ENUM("HPF Cutoff", max98371_hpf_cutoff),
+};
+
+static int max98371_dai_set_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec);
+	unsigned int val = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		dev_err(codec->dev, "DAI clock mode unsupported");
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		val |= 0;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		val |= MAX98371_DAI_RIGHT;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		val |= MAX98371_DAI_LEFT;
+		break;
+	default:
+		dev_err(codec->dev, "DAI wrong mode unsupported");
+		return -EINVAL;
+	}
+	regmap_update_bits(max98371->regmap, MAX98371_FMT,
+			MAX98371_FMT_MODE_MASK, val);
+	return 0;
+}
+
+static int max98371_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 max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec);
+	int blr_clk_ratio, ch_size, channels = params_channels(params);
+	int rate = params_rate(params);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S8:
+		regmap_update_bits(max98371->regmap, MAX98371_FMT,
+				MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16);
+		ch_size = 8;
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		regmap_update_bits(max98371->regmap, MAX98371_FMT,
+				MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16);
+		ch_size = 16;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		regmap_update_bits(max98371->regmap, MAX98371_FMT,
+				MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32);
+		ch_size = 24;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		regmap_update_bits(max98371->regmap, MAX98371_FMT,
+				MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32);
+		ch_size = 32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* BCLK/LRCLK ratio calculation */
+	blr_clk_ratio = channels * ch_size;
+	switch (blr_clk_ratio) {
+	case 32:
+		regmap_update_bits(max98371->regmap,
+			MAX98371_DAI_CLK,
+			MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_32);
+		break;
+	case 48:
+		regmap_update_bits(max98371->regmap,
+			MAX98371_DAI_CLK,
+			MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_48);
+		break;
+	case 64:
+		regmap_update_bits(max98371->regmap,
+			MAX98371_DAI_CLK,
+			MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_64);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (rate) {
+	case 32000:
+		regmap_update_bits(max98371->regmap,
+			MAX98371_SPK_SR,
+			MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_32);
+		break;
+	case 44100:
+		regmap_update_bits(max98371->regmap,
+			MAX98371_SPK_SR,
+			MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_44);
+		break;
+	case 48000:
+		regmap_update_bits(max98371->regmap,
+			MAX98371_SPK_SR,
+			MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_48);
+		break;
+	case 88200:
+		regmap_update_bits(max98371->regmap,
+			MAX98371_SPK_SR,
+			MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_88);
+		break;
+	case 96000:
+		regmap_update_bits(max98371->regmap,
+			MAX98371_SPK_SR,
+			MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_96);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* enabling both the RX channels*/
+	regmap_update_bits(max98371->regmap, MAX98371_MONOMIX_SRC,
+			MAX98371_MONOMIX_SRC_MASK, MONOMIX_RX_0_1);
+	regmap_update_bits(max98371->regmap, MAX98371_DAI_CHANNEL,
+			MAX98371_CHANNEL_MASK, MAX98371_CHANNEL_MASK);
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget max98371_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC", NULL, MAX98371_SPK_ENABLE, 0, 0),
+	SND_SOC_DAPM_SUPPLY("Global Enable", MAX98371_GLOBAL_ENABLE,
+		0, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("SPK_OUT"),
+};
+
+static const struct snd_soc_dapm_route max98371_audio_map[] = {
+	{"DAC", NULL, "HiFi Playback"},
+	{"SPK_OUT", NULL, "DAC"},
+	{"SPK_OUT", NULL, "Global Enable"},
+};
+
+#define MAX98371_RATES SNDRV_PCM_RATE_8000_48000
+#define MAX98371_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
+		SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
+
+static const struct snd_soc_dai_ops max98371_dai_ops = {
+	.set_fmt = max98371_dai_set_fmt,
+	.hw_params = max98371_dai_hw_params,
+};
+
+static struct snd_soc_dai_driver max98371_dai[] = {
+	{
+		.name = "max98371-aif1",
+		.playback = {
+			.stream_name = "HiFi Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_48000,
+			.formats = MAX98371_FORMATS,
+		},
+		.ops = &max98371_dai_ops,
+	}
+};
+
+static const struct snd_soc_codec_driver max98371_codec = {
+	.controls = max98371_snd_controls,
+	.num_controls = ARRAY_SIZE(max98371_snd_controls),
+	.dapm_routes = max98371_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(max98371_audio_map),
+	.dapm_widgets = max98371_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets),
+};
+
+static const struct regmap_config max98371_regmap = {
+	.reg_bits         = 8,
+	.val_bits         = 8,
+	.max_register     = MAX98371_VERSION,
+	.reg_defaults     = max98371_reg,
+	.num_reg_defaults = ARRAY_SIZE(max98371_reg),
+	.volatile_reg     = max98371_volatile_register,
+	.readable_reg     = max98371_readable_register,
+	.cache_type       = REGCACHE_RBTREE,
+};
+
+static int max98371_i2c_probe(struct i2c_client *i2c,
+		const struct i2c_device_id *id)
+{
+	struct max98371_priv *max98371;
+	int ret, reg;
+
+	max98371 = devm_kzalloc(&i2c->dev,
+			sizeof(*max98371), GFP_KERNEL);
+	if (!max98371)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, max98371);
+	max98371->regmap = devm_regmap_init_i2c(i2c, &max98371_regmap);
+	if (IS_ERR(max98371->regmap)) {
+		ret = PTR_ERR(max98371->regmap);
+		dev_err(&i2c->dev,
+				"Failed to allocate regmap: %d\n", ret);
+		return ret;
+	}
+
+	ret = regmap_read(max98371->regmap, MAX98371_VERSION, &reg);
+	if (ret < 0) {
+		dev_info(&i2c->dev, "device error %d\n", ret);
+		return ret;
+	}
+	dev_info(&i2c->dev, "device version %x\n", reg);
+
+	ret = snd_soc_register_codec(&i2c->dev, &max98371_codec,
+			max98371_dai, ARRAY_SIZE(max98371_dai));
+	if (ret < 0) {
+		dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+		return ret;
+	}
+	return ret;
+}
+
+static int max98371_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	return 0;
+}
+
+static const struct i2c_device_id max98371_i2c_id[] = {
+	{ "max98371", 0 },
+};
+
+MODULE_DEVICE_TABLE(i2c, max98371_i2c_id);
+
+static const struct of_device_id max98371_of_match[] = {
+	{ .compatible = "maxim,max98371", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, max98371_of_match);
+
+static struct i2c_driver max98371_i2c_driver = {
+	.driver = {
+		.name = "max98371",
+		.owner = THIS_MODULE,
+		.pm = NULL,
+		.of_match_table = of_match_ptr(max98371_of_match),
+	},
+	.probe  = max98371_i2c_probe,
+	.remove = max98371_i2c_remove,
+	.id_table = max98371_i2c_id,
+};
+
+module_i2c_driver(max98371_i2c_driver);
+
+MODULE_AUTHOR("anish kumar <yesanishhere@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC MAX98371 driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/max98371.h b/sound/soc/codecs/max98371.h
new file mode 100644
index 000000000..9f6330964
--- /dev/null
+++ b/sound/soc/codecs/max98371.h
@@ -0,0 +1,67 @@
+/*
+ * max98371.h -- MAX98371 ALSA SoC Audio driver
+ *
+ * Copyright 2011-2012 Maxim Integrated Products
+ *
+ * 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 _MAX98371_H
+#define _MAX98371_H
+
+#define MAX98371_IRQ_CLEAR1			0x01
+#define MAX98371_IRQ_CLEAR2			0x02
+#define MAX98371_IRQ_CLEAR3			0x03
+#define MAX98371_DAI_CLK			0x10
+#define MAX98371_DAI_BSEL_MASK			0xF
+#define MAX98371_DAI_BSEL_32			2
+#define MAX98371_DAI_BSEL_48			3
+#define MAX98371_DAI_BSEL_64			4
+#define MAX98371_SPK_SR				0x11
+#define MAX98371_SPK_SR_MASK			0xF
+#define MAX98371_SPK_SR_32			6
+#define MAX98371_SPK_SR_44			7
+#define MAX98371_SPK_SR_48			8
+#define MAX98371_SPK_SR_88			10
+#define MAX98371_SPK_SR_96			11
+#define MAX98371_DAI_CHANNEL			0x15
+#define MAX98371_CHANNEL_MASK			0x3
+#define MAX98371_MONOMIX_SRC			0x18
+#define MAX98371_MONOMIX_CFG			0x19
+#define MAX98371_HPF				0x1C
+#define MAX98371_MONOMIX_SRC_MASK		0xFF
+#define MONOMIX_RX_0_1				((0x1)<<(4))
+#define M98371_DAI_CHANNEL_I2S			0x3
+#define MAX98371_DIGITAL_GAIN			0x2D
+#define MAX98371_DIGITAL_GAIN_WIDTH		0x7
+#define MAX98371_GAIN				0x2E
+#define MAX98371_GAIN_SHIFT			0x4
+#define MAX98371_GAIN_WIDTH			0x4
+#define MAX98371_DHT_MAX_WIDTH			4
+#define MAX98371_FMT				0x14
+#define MAX98371_CHANSZ_WIDTH			6
+#define MAX98371_FMT_MASK		        ((0x3)<<(MAX98371_CHANSZ_WIDTH))
+#define MAX98371_FMT_MODE_MASK		        ((0x7)<<(3))
+#define MAX98371_DAI_LEFT		        ((0x1)<<(3))
+#define MAX98371_DAI_RIGHT		        ((0x2)<<(3))
+#define MAX98371_DAI_CHANSZ_16                  ((1)<<(MAX98371_CHANSZ_WIDTH))
+#define MAX98371_DAI_CHANSZ_24                  ((2)<<(MAX98371_CHANSZ_WIDTH))
+#define MAX98371_DAI_CHANSZ_32                  ((3)<<(MAX98371_CHANSZ_WIDTH))
+#define MAX98371_DHT  0x32
+#define MAX98371_DHT_STEP			0x3
+#define MAX98371_DHT_GAIN			0x31
+#define MAX98371_DHT_GAIN_WIDTH			0x4
+#define MAX98371_DHT_ROT_WIDTH			0x4
+#define MAX98371_SPK_ENABLE			0x4A
+#define MAX98371_GLOBAL_ENABLE			0x50
+#define MAX98371_SOFT_RESET			0x51
+#define MAX98371_VERSION			0xFF
+
+
+struct max98371_priv {
+	struct regmap *regmap;
+	struct snd_soc_codec *codec;
+};
+#endif
diff --git a/sound/soc/codecs/pcm5102a.c b/sound/soc/codecs/pcm5102a.c
new file mode 100644
index 000000000..ed5156774
--- /dev/null
+++ b/sound/soc/codecs/pcm5102a.c
@@ -0,0 +1,69 @@
+/*
+ * Driver for the PCM5102A codec
+ *
+ * Author:	Florian Meier <florian.meier@koalo.de>
+ *		Copyright 2013
+ *
+ * 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/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+
+#include <sound/soc.h>
+
+static struct snd_soc_dai_driver pcm5102a_dai = {
+	.name = "pcm5102a-hifi",
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE |
+			   SNDRV_PCM_FMTBIT_S24_LE |
+			   SNDRV_PCM_FMTBIT_S32_LE
+	},
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_pcm5102a;
+
+static int pcm5102a_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm5102a,
+			&pcm5102a_dai, 1);
+}
+
+static int pcm5102a_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+static const struct of_device_id pcm5102a_of_match[] = {
+	{ .compatible = "ti,pcm5102a", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, pcm5102a_of_match);
+
+static struct platform_driver pcm5102a_codec_driver = {
+	.probe		= pcm5102a_probe,
+	.remove		= pcm5102a_remove,
+	.driver		= {
+		.name	= "pcm5102a-codec",
+		.owner	= THIS_MODULE,
+		.of_match_table = pcm5102a_of_match,
+	},
+};
+
+module_platform_driver(pcm5102a_codec_driver);
+
+MODULE_DESCRIPTION("ASoC PCM5102A codec driver");
+MODULE_AUTHOR("Florian Meier <florian.meier@koalo.de>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c
index f0e6c06e8..f80cfe4d2 100644
--- a/sound/soc/codecs/rt298.c
+++ b/sound/soc/codecs/rt298.c
@@ -17,6 +17,7 @@
 #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>
@@ -275,6 +276,8 @@ static int rt298_jack_detect(struct rt298_priv *rt298, bool *hp, bool *mic)
 		} else {
 			*mic = false;
 			regmap_write(rt298->regmap, RT298_SET_MIC1, 0x20);
+			regmap_update_bits(rt298->regmap,
+				RT298_CBJ_CTRL1, 0x0400, 0x0000);
 		}
 	} else {
 		regmap_read(rt298->regmap, RT298_GET_HP_SENSE, &buf);
@@ -481,6 +484,26 @@ static int rt298_adc_event(struct snd_soc_dapm_widget *w,
 		snd_soc_update_bits(codec,
 			VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0),
 			0x7080, 0x7000);
+		 /* If MCLK doesn't exist, reset AD filter */
+		if (!(snd_soc_read(codec, RT298_VAD_CTRL) & 0x200)) {
+			pr_info("NO MCLK\n");
+			switch (nid) {
+			case RT298_ADC_IN1:
+				snd_soc_update_bits(codec,
+					RT298_D_FILTER_CTRL, 0x2, 0x2);
+				mdelay(10);
+				snd_soc_update_bits(codec,
+					RT298_D_FILTER_CTRL, 0x2, 0x0);
+				break;
+			case RT298_ADC_IN2:
+				snd_soc_update_bits(codec,
+					RT298_D_FILTER_CTRL, 0x4, 0x4);
+				mdelay(10);
+				snd_soc_update_bits(codec,
+					RT298_D_FILTER_CTRL, 0x4, 0x0);
+				break;
+			}
+		}
 		break;
 	case SND_SOC_DAPM_PRE_PMD:
 		snd_soc_update_bits(codec,
@@ -519,30 +542,12 @@ static int rt298_mic1_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static int rt298_vref_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);
-
-	switch (event) {
-	case SND_SOC_DAPM_PRE_PMU:
-		snd_soc_update_bits(codec,
-			RT298_CBJ_CTRL1, 0x0400, 0x0000);
-		mdelay(50);
-		break;
-	default:
-		return 0;
-	}
-
-	return 0;
-}
-
 static const struct snd_soc_dapm_widget rt298_dapm_widgets[] = {
 
 	SND_SOC_DAPM_SUPPLY_S("HV", 1, RT298_POWER_CTRL1,
 		12, 1, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("VREF", RT298_POWER_CTRL1,
-		0, 1, rt298_vref_event, SND_SOC_DAPM_PRE_PMU),
+		0, 1, NULL, 0),
 	SND_SOC_DAPM_SUPPLY_S("BG_MBIAS", 1, RT298_POWER_CTRL2,
 		1, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT298_POWER_CTRL2,
@@ -933,18 +938,9 @@ static int rt298_set_bias_level(struct snd_soc_codec *codec,
 		}
 		break;
 
-	case SND_SOC_BIAS_ON:
-		mdelay(30);
-		snd_soc_update_bits(codec,
-			RT298_CBJ_CTRL1, 0x0400, 0x0400);
-
-		break;
-
 	case SND_SOC_BIAS_STANDBY:
 		snd_soc_write(codec,
 			RT298_SET_AUDIO_POWER, AC_PWRST_D3);
-		snd_soc_update_bits(codec,
-			RT298_CBJ_CTRL1, 0x0400, 0x0000);
 		break;
 
 	default:
@@ -1132,6 +1128,17 @@ static const struct acpi_device_id rt298_acpi_match[] = {
 };
 MODULE_DEVICE_TABLE(acpi, rt298_acpi_match);
 
+static const struct dmi_system_id force_combo_jack_table[] = {
+	{
+		.ident = "Intel Broxton P",
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Intel Corp"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Broxton P")
+		}
+	},
+	{ }
+};
+
 static int rt298_i2c_probe(struct i2c_client *i2c,
 			   const struct i2c_device_id *id)
 {
@@ -1184,11 +1191,16 @@ static int rt298_i2c_probe(struct i2c_client *i2c,
 
 	/* enable jack combo mode on supported devices */
 	acpiid = acpi_match_device(dev->driver->acpi_match_table, dev);
-	if (acpiid) {
+	if (acpiid && acpiid->driver_data) {
 		rt298->pdata = *(struct rt298_platform_data *)
 				acpiid->driver_data;
 	}
 
+	if (dmi_check_system(force_combo_jack_table)) {
+		rt298->pdata.cbj_en = true;
+		rt298->pdata.gpio2_en = false;
+	}
+
 	/* VREF Charging */
 	regmap_update_bits(rt298->regmap, 0x04, 0x80, 0x80);
 	regmap_update_bits(rt298->regmap, 0x1b, 0x860, 0x860);
diff --git a/sound/soc/codecs/rt298.h b/sound/soc/codecs/rt298.h
index d66f8847b..3638f3d61 100644
--- a/sound/soc/codecs/rt298.h
+++ b/sound/soc/codecs/rt298.h
@@ -137,6 +137,7 @@
 #define RT298_A_BIAS_CTRL2	0x02
 #define RT298_POWER_CTRL1	0x03
 #define RT298_A_BIAS_CTRL3	0x04
+#define RT298_D_FILTER_CTRL	0x05
 #define RT298_POWER_CTRL2	0x08
 #define RT298_I2S_CTRL1		0x09
 #define RT298_I2S_CTRL2		0x0a
@@ -148,6 +149,7 @@
 #define RT298_IRQ_CTRL		0x33
 #define RT298_WIND_FILTER_CTRL	0x46
 #define RT298_PLL_CTRL1		0x49
+#define RT298_VAD_CTRL		0x4e
 #define RT298_CBJ_CTRL1		0x4f
 #define RT298_CBJ_CTRL2		0x50
 #define RT298_PLL_CTRL		0x63
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 7af5e7380..d70847c9e 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -253,7 +253,7 @@ static const struct reg_default rt5650_reg[] = {
 	{ 0x2b, 0x5454 },
 	{ 0x2c, 0xaaa0 },
 	{ 0x2d, 0x0000 },
-	{ 0x2f, 0x1002 },
+	{ 0x2f, 0x5002 },
 	{ 0x31, 0x5000 },
 	{ 0x32, 0x0000 },
 	{ 0x33, 0x0000 },
@@ -3286,10 +3286,8 @@ static void rt5645_jack_detect_work(struct work_struct *work)
 		if (btn_type == 0)/* button release */
 			report =  rt5645->jack_type;
 		else {
-			if (rt5645->pdata.jd_invert) {
-				mod_timer(&rt5645->btn_check_timer,
-					msecs_to_jiffies(100));
-			}
+			mod_timer(&rt5645->btn_check_timer,
+				msecs_to_jiffies(100));
 		}
 
 		break;
@@ -3557,6 +3555,12 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = {
 			DMI_MATCH(DMI_SYS_VENDOR, "GOOGLE"),
 		},
 	},
+	{
+		.ident = "Google Setzer",
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Setzer"),
+		},
+	},
 	{ }
 };
 
@@ -3810,9 +3814,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
 	if (rt5645->pdata.jd_invert) {
 		regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
 			RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV);
-		setup_timer(&rt5645->btn_check_timer,
-			rt5645_btn_check_callback, (unsigned long)rt5645);
 	}
+	setup_timer(&rt5645->btn_check_timer,
+		rt5645_btn_check_callback, (unsigned long)rt5645);
 
 	INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work);
 	INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work);
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 49a9e7049..0af5ddbef 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -619,7 +619,7 @@ static const struct snd_kcontrol_new rt5670_snd_controls[] = {
 		RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1),
 	SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL,
 		RT5670_L_VOL_SFT, RT5670_R_VOL_SFT,
-		39, 0, out_vol_tlv),
+		39, 1, out_vol_tlv),
 	/* OUTPUT Control */
 	SOC_DOUBLE("OUT Channel Switch", RT5670_LOUT1,
 		RT5670_VOL_L_SFT, RT5670_VOL_R_SFT, 1, 1),
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index adec5cc59..2d83bb52e 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -1241,60 +1241,46 @@ static int rt5677_dmic_use_asrc(struct snd_soc_dapm_widget *source,
 		regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting);
 		asrc_setting = (asrc_setting & RT5677_AD_STO1_CLK_SEL_MASK) >>
 				RT5677_AD_STO1_CLK_SEL_SFT;
-		if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC &&
-			asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC)
-			return 1;
 		break;
 
 	case 10:
 		regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting);
 		asrc_setting = (asrc_setting & RT5677_AD_STO2_CLK_SEL_MASK) >>
 				RT5677_AD_STO2_CLK_SEL_SFT;
-		if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC &&
-			asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC)
-			return 1;
 		break;
 
 	case 9:
 		regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting);
 		asrc_setting = (asrc_setting & RT5677_AD_STO3_CLK_SEL_MASK) >>
 				RT5677_AD_STO3_CLK_SEL_SFT;
-		if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC &&
-			asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC)
-			return 1;
 		break;
 
 	case 8:
 		regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting);
 		asrc_setting = (asrc_setting & RT5677_AD_STO4_CLK_SEL_MASK) >>
 			RT5677_AD_STO4_CLK_SEL_SFT;
-		if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC &&
-			asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC)
-			return 1;
 		break;
 
 	case 7:
 		regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting);
 		asrc_setting = (asrc_setting & RT5677_AD_MONOL_CLK_SEL_MASK) >>
 			RT5677_AD_MONOL_CLK_SEL_SFT;
-		if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC &&
-			asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC)
-			return 1;
 		break;
 
 	case 6:
 		regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting);
 		asrc_setting = (asrc_setting & RT5677_AD_MONOR_CLK_SEL_MASK) >>
 			RT5677_AD_MONOR_CLK_SEL_SFT;
-		if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC &&
-			asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC)
-			return 1;
 		break;
 
 	default:
-		break;
+		return 0;
 	}
 
+	if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC &&
+	    asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC)
+		return 1;
+
 	return 0;
 }
 
@@ -4520,14 +4506,9 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec,
 }
 
 #ifdef CONFIG_GPIOLIB
-static inline struct rt5677_priv *gpio_to_rt5677(struct gpio_chip *chip)
-{
-	return container_of(chip, struct rt5677_priv, gpio_chip);
-}
-
 static void rt5677_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
 {
-	struct rt5677_priv *rt5677 = gpio_to_rt5677(chip);
+	struct rt5677_priv *rt5677 = gpiochip_get_data(chip);
 
 	switch (offset) {
 	case RT5677_GPIO1 ... RT5677_GPIO5:
@@ -4548,7 +4529,7 @@ static void rt5677_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
 static int rt5677_gpio_direction_out(struct gpio_chip *chip,
 				     unsigned offset, int value)
 {
-	struct rt5677_priv *rt5677 = gpio_to_rt5677(chip);
+	struct rt5677_priv *rt5677 = gpiochip_get_data(chip);
 
 	switch (offset) {
 	case RT5677_GPIO1 ... RT5677_GPIO5:
@@ -4572,7 +4553,7 @@ static int rt5677_gpio_direction_out(struct gpio_chip *chip,
 
 static int rt5677_gpio_get(struct gpio_chip *chip, unsigned offset)
 {
-	struct rt5677_priv *rt5677 = gpio_to_rt5677(chip);
+	struct rt5677_priv *rt5677 = gpiochip_get_data(chip);
 	int value, ret;
 
 	ret = regmap_read(rt5677->regmap, RT5677_GPIO_ST, &value);
@@ -4584,7 +4565,7 @@ static int rt5677_gpio_get(struct gpio_chip *chip, unsigned offset)
 
 static int rt5677_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
 {
-	struct rt5677_priv *rt5677 = gpio_to_rt5677(chip);
+	struct rt5677_priv *rt5677 = gpiochip_get_data(chip);
 
 	switch (offset) {
 	case RT5677_GPIO1 ... RT5677_GPIO5:
@@ -4638,7 +4619,7 @@ static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset,
 
 static int rt5677_to_irq(struct gpio_chip *chip, unsigned offset)
 {
-	struct rt5677_priv *rt5677 = gpio_to_rt5677(chip);
+	struct rt5677_priv *rt5677 = gpiochip_get_data(chip);
 	struct regmap_irq_chip_data *data = rt5677->irq_data;
 	int irq;
 
@@ -4697,7 +4678,7 @@ static void rt5677_init_gpio(struct i2c_client *i2c)
 	rt5677->gpio_chip.parent = &i2c->dev;
 	rt5677->gpio_chip.base = -1;
 
-	ret = gpiochip_add(&rt5677->gpio_chip);
+	ret = gpiochip_add_data(&rt5677->gpio_chip, rt5677);
 	if (ret != 0)
 		dev_err(&i2c->dev, "Failed to add GPIOs: %d\n", ret);
 }
diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c
index 39307ad41..b8d19b77b 100644
--- a/sound/soc/codecs/tas571x.c
+++ b/sound/soc/codecs/tas571x.c
@@ -4,6 +4,9 @@
  * Copyright (C) 2015 Google, Inc.
  * Copyright (c) 2013 Daniel Mack <zonque@gmail.com>
  *
+ * TAS5721 support:
+ * Copyright (C) 2016 Petr Kulhavy, Barix AG <petr@barix.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
@@ -57,6 +60,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg)
 	case TAS571X_CH1_VOL_REG:
 	case TAS571X_CH2_VOL_REG:
 		return priv->chip->vol_reg_size;
+	case TAS571X_INPUT_MUX_REG:
+	case TAS571X_CH4_SRC_SELECT_REG:
+	case TAS571X_PWM_MUX_REG:
+		return 4;
 	default:
 		return 1;
 	}
@@ -167,6 +174,23 @@ static int tas571x_hw_params(struct snd_pcm_substream *substream,
 				  TAS571X_SDI_FMT_MASK, val);
 }
 
+static int tas571x_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u8 sysctl2;
+	int ret;
+
+	sysctl2 = mute ? TAS571X_SYS_CTRL_2_SDN_MASK : 0;
+
+	ret = snd_soc_update_bits(codec,
+			    TAS571X_SYS_CTRL_2_REG,
+		     TAS571X_SYS_CTRL_2_SDN_MASK,
+		     sysctl2);
+	usleep_range(1000, 2000);
+
+	return ret;
+}
+
 static int tas571x_set_bias_level(struct snd_soc_codec *codec,
 				  enum snd_soc_bias_level level)
 {
@@ -214,6 +238,7 @@ static int tas571x_set_bias_level(struct snd_soc_codec *codec,
 static const struct snd_soc_dai_ops tas571x_dai_ops = {
 	.set_fmt	= tas571x_set_dai_fmt,
 	.hw_params	= tas571x_hw_params,
+	.digital_mute	= tas571x_mute,
 };
 
 static const char *const tas5711_supply_names[] = {
@@ -241,6 +266,26 @@ static const struct snd_kcontrol_new tas5711_controls[] = {
 		   1, 1),
 };
 
+static const struct regmap_range tas571x_readonly_regs_range[] = {
+	regmap_reg_range(TAS571X_CLK_CTRL_REG,  TAS571X_DEV_ID_REG),
+};
+
+static const struct regmap_range tas571x_volatile_regs_range[] = {
+	regmap_reg_range(TAS571X_CLK_CTRL_REG,  TAS571X_ERR_STATUS_REG),
+	regmap_reg_range(TAS571X_OSC_TRIM_REG,  TAS571X_OSC_TRIM_REG),
+};
+
+static const struct regmap_access_table tas571x_write_regs = {
+	.no_ranges =	tas571x_readonly_regs_range,
+	.n_no_ranges =	ARRAY_SIZE(tas571x_readonly_regs_range),
+};
+
+static const struct regmap_access_table tas571x_volatile_regs = {
+	.yes_ranges =	tas571x_volatile_regs_range,
+	.n_yes_ranges =	ARRAY_SIZE(tas571x_volatile_regs_range),
+
+};
+
 static const struct reg_default tas5711_reg_defaults[] = {
 	{ 0x04, 0x05 },
 	{ 0x05, 0x40 },
@@ -260,6 +305,8 @@ static const struct regmap_config tas5711_regmap_config = {
 	.reg_defaults			= tas5711_reg_defaults,
 	.num_reg_defaults		= ARRAY_SIZE(tas5711_reg_defaults),
 	.cache_type			= REGCACHE_RBTREE,
+	.wr_table			= &tas571x_write_regs,
+	.volatile_table			= &tas571x_volatile_regs,
 };
 
 static const struct tas571x_chip tas5711_chip = {
@@ -314,6 +361,8 @@ static const struct regmap_config tas5717_regmap_config = {
 	.reg_defaults			= tas5717_reg_defaults,
 	.num_reg_defaults		= ARRAY_SIZE(tas5717_reg_defaults),
 	.cache_type			= REGCACHE_RBTREE,
+	.wr_table			= &tas571x_write_regs,
+	.volatile_table			= &tas571x_volatile_regs,
 };
 
 /* This entry is reused for tas5719 as the software interface is identical. */
@@ -326,6 +375,77 @@ static const struct tas571x_chip tas5717_chip = {
 	.vol_reg_size			= 2,
 };
 
+static const char *const tas5721_supply_names[] = {
+	"AVDD",
+	"DVDD",
+	"DRVDD",
+	"PVDD",
+};
+
+static const struct snd_kcontrol_new tas5721_controls[] = {
+	SOC_SINGLE_TLV("Master Volume",
+		       TAS571X_MVOL_REG,
+		       0, 0xff, 1, tas5711_volume_tlv),
+	SOC_DOUBLE_R_TLV("Speaker Volume",
+			 TAS571X_CH1_VOL_REG,
+			 TAS571X_CH2_VOL_REG,
+			 0, 0xff, 1, tas5711_volume_tlv),
+	SOC_DOUBLE("Speaker Switch",
+		   TAS571X_SOFT_MUTE_REG,
+		   TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT,
+		   1, 1),
+};
+
+static const struct reg_default tas5721_reg_defaults[] = {
+	{TAS571X_CLK_CTRL_REG,		0x6c},
+	{TAS571X_DEV_ID_REG,		0x00},
+	{TAS571X_ERR_STATUS_REG,	0x00},
+	{TAS571X_SYS_CTRL_1_REG,	0xa0},
+	{TAS571X_SDI_REG,		0x05},
+	{TAS571X_SYS_CTRL_2_REG,	0x40},
+	{TAS571X_SOFT_MUTE_REG,		0x00},
+	{TAS571X_MVOL_REG,		0xff},
+	{TAS571X_CH1_VOL_REG,		0x30},
+	{TAS571X_CH2_VOL_REG,		0x30},
+	{TAS571X_CH3_VOL_REG,		0x30},
+	{TAS571X_VOL_CFG_REG,		0x91},
+	{TAS571X_MODULATION_LIMIT_REG,	0x02},
+	{TAS571X_IC_DELAY_CH1_REG,	0xac},
+	{TAS571X_IC_DELAY_CH2_REG,	0x54},
+	{TAS571X_IC_DELAY_CH3_REG,	0xac},
+	{TAS571X_IC_DELAY_CH4_REG,	0x54},
+	{TAS571X_PWM_CH_SDN_GROUP_REG,	0x30},
+	{TAS571X_START_STOP_PERIOD_REG,	0x0f},
+	{TAS571X_OSC_TRIM_REG,		0x82},
+	{TAS571X_BKND_ERR_REG,		0x02},
+	{TAS571X_INPUT_MUX_REG,		0x17772},
+	{TAS571X_CH4_SRC_SELECT_REG,	0x4303},
+	{TAS571X_PWM_MUX_REG,		0x1021345},
+};
+
+static const struct regmap_config tas5721_regmap_config = {
+	.reg_bits			= 8,
+	.val_bits			= 32,
+	.max_register			= 0xff,
+	.reg_read			= tas571x_reg_read,
+	.reg_write			= tas571x_reg_write,
+	.reg_defaults			= tas5721_reg_defaults,
+	.num_reg_defaults		= ARRAY_SIZE(tas5721_reg_defaults),
+	.cache_type			= REGCACHE_RBTREE,
+	.wr_table			= &tas571x_write_regs,
+	.volatile_table			= &tas571x_volatile_regs,
+};
+
+
+static const struct tas571x_chip tas5721_chip = {
+	.supply_names			= tas5721_supply_names,
+	.num_supply_names		= ARRAY_SIZE(tas5721_supply_names),
+	.controls			= tas5711_controls,
+	.num_controls			= ARRAY_SIZE(tas5711_controls),
+	.regmap_config			= &tas5721_regmap_config,
+	.vol_reg_size			= 1,
+};
+
 static const struct snd_soc_dapm_widget tas571x_dapm_widgets[] = {
 	SND_SOC_DAPM_DAC("DACL", NULL, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_DAC("DACR", NULL, SND_SOC_NOPM, 0, 0),
@@ -386,11 +506,10 @@ static int tas571x_i2c_probe(struct i2c_client *client,
 	i2c_set_clientdata(client, priv);
 
 	of_id = of_match_device(tas571x_of_match, dev);
-	if (!of_id) {
-		dev_err(dev, "Unknown device type\n");
-		return -EINVAL;
-	}
-	priv->chip = of_id->data;
+	if (of_id)
+		priv->chip = of_id->data;
+	else
+		priv->chip = (void *) id->driver_data;
 
 	priv->mclk = devm_clk_get(dev, "mclk");
 	if (IS_ERR(priv->mclk) && PTR_ERR(priv->mclk) != -ENOENT) {
@@ -445,10 +564,6 @@ static int tas571x_i2c_probe(struct i2c_client *client,
 	if (ret)
 		return ret;
 
-	ret = regmap_update_bits(priv->regmap, TAS571X_SYS_CTRL_2_REG,
-				 TAS571X_SYS_CTRL_2_SDN_MASK, 0);
-	if (ret)
-		return ret;
 
 	memcpy(&priv->codec_driver, &tas571x_codec, sizeof(priv->codec_driver));
 	priv->codec_driver.controls = priv->chip->controls;
@@ -486,14 +601,16 @@ static const struct of_device_id tas571x_of_match[] = {
 	{ .compatible = "ti,tas5711", .data = &tas5711_chip, },
 	{ .compatible = "ti,tas5717", .data = &tas5717_chip, },
 	{ .compatible = "ti,tas5719", .data = &tas5717_chip, },
+	{ .compatible = "ti,tas5721", .data = &tas5721_chip, },
 	{ }
 };
 MODULE_DEVICE_TABLE(of, tas571x_of_match);
 
 static const struct i2c_device_id tas571x_i2c_id[] = {
-	{ "tas5711", 0 },
-	{ "tas5717", 0 },
-	{ "tas5719", 0 },
+	{ "tas5711", (kernel_ulong_t) &tas5711_chip },
+	{ "tas5717", (kernel_ulong_t) &tas5717_chip },
+	{ "tas5719", (kernel_ulong_t) &tas5717_chip },
+	{ "tas5721", (kernel_ulong_t) &tas5721_chip },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, tas571x_i2c_id);
diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h
index 0aee47123..cf800c364 100644
--- a/sound/soc/codecs/tas571x.h
+++ b/sound/soc/codecs/tas571x.h
@@ -13,6 +13,10 @@
 #define _TAS571X_H
 
 /* device registers */
+#define TAS571X_CLK_CTRL_REG		0x00
+#define TAS571X_DEV_ID_REG		0x01
+#define TAS571X_ERR_STATUS_REG		0x02
+#define TAS571X_SYS_CTRL_1_REG		0x03
 #define TAS571X_SDI_REG			0x04
 #define TAS571X_SDI_FMT_MASK		0x0f
 
@@ -27,7 +31,25 @@
 #define TAS571X_MVOL_REG		0x07
 #define TAS571X_CH1_VOL_REG		0x08
 #define TAS571X_CH2_VOL_REG		0x09
+#define TAS571X_CH3_VOL_REG		0x0a
+#define TAS571X_VOL_CFG_REG		0x0e
+#define TAS571X_MODULATION_LIMIT_REG	0x10
+#define TAS571X_IC_DELAY_CH1_REG	0x11
+#define TAS571X_IC_DELAY_CH2_REG	0x12
+#define TAS571X_IC_DELAY_CH3_REG	0x13
+#define TAS571X_IC_DELAY_CH4_REG	0x14
 
+#define TAS571X_PWM_CH_SDN_GROUP_REG	0x19	/* N/A on TAS5717, TAS5719 */
+#define TAS571X_PWM_CH1_SDN_MASK	(1<<0)
+#define TAS571X_PWM_CH2_SDN_SHIFT	(1<<1)
+#define TAS571X_PWM_CH3_SDN_SHIFT	(1<<2)
+#define TAS571X_PWM_CH4_SDN_SHIFT	(1<<3)
+
+#define TAS571X_START_STOP_PERIOD_REG	0x1a
 #define TAS571X_OSC_TRIM_REG		0x1b
+#define TAS571X_BKND_ERR_REG		0x1c
+#define TAS571X_INPUT_MUX_REG		0x20
+#define TAS571X_CH4_SRC_SELECT_REG	0x21
+#define TAS571X_PWM_MUX_REG		0x25
 
 #endif /* _TAS571X_H */
diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c
new file mode 100644
index 000000000..f54fb46b7
--- /dev/null
+++ b/sound/soc/codecs/tas5720.c
@@ -0,0 +1,620 @@
+/*
+ * tas5720.c - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier
+ *
+ * Copyright (C)2015-2016 Texas Instruments Incorporated -  http://www.ti.com
+ *
+ * Author: Andreas Dannenberg <dannenberg@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.
+ */
+
+#include <linux/module.h>
+#include <linux/errno.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/regulator/consumer.h>
+#include <linux/delay.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "tas5720.h"
+
+/* Define how often to check (and clear) the fault status register (in ms) */
+#define TAS5720_FAULT_CHECK_INTERVAL		200
+
+static const char * const tas5720_supply_names[] = {
+	"dvdd",		/* Digital power supply. Connect to 3.3-V supply. */
+	"pvdd",		/* Class-D amp and analog power supply (connected). */
+};
+
+#define TAS5720_NUM_SUPPLIES	ARRAY_SIZE(tas5720_supply_names)
+
+struct tas5720_data {
+	struct snd_soc_codec *codec;
+	struct regmap *regmap;
+	struct i2c_client *tas5720_client;
+	struct regulator_bulk_data supplies[TAS5720_NUM_SUPPLIES];
+	struct delayed_work fault_check_work;
+	unsigned int last_fault;
+};
+
+static int tas5720_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;
+	unsigned int rate = params_rate(params);
+	bool ssz_ds;
+	int ret;
+
+	switch (rate) {
+	case 44100:
+	case 48000:
+		ssz_ds = false;
+		break;
+	case 88200:
+	case 96000:
+		ssz_ds = true;
+		break;
+	default:
+		dev_err(codec->dev, "unsupported sample rate: %u\n", rate);
+		return -EINVAL;
+	}
+
+	ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG,
+				  TAS5720_SSZ_DS, ssz_ds);
+	if (ret < 0) {
+		dev_err(codec->dev, "error setting sample rate: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int tas5720_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u8 serial_format;
+	int ret;
+
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
+		dev_vdbg(codec->dev, "DAI Format master is not found\n");
+		return -EINVAL;
+	}
+
+	switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK |
+		       SND_SOC_DAIFMT_INV_MASK)) {
+	case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
+		/* 1st data bit occur one BCLK cycle after the frame sync */
+		serial_format = TAS5720_SAIF_I2S;
+		break;
+	case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF):
+		/*
+		 * Note that although the TAS5720 does not have a dedicated DSP
+		 * mode it doesn't care about the LRCLK duty cycle during TDM
+		 * operation. Therefore we can use the device's I2S mode with
+		 * its delaying of the 1st data bit to receive DSP_A formatted
+		 * data. See device datasheet for additional details.
+		 */
+		serial_format = TAS5720_SAIF_I2S;
+		break;
+	case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF):
+		/*
+		 * Similar to DSP_A, we can use the fact that the TAS5720 does
+		 * not care about the LRCLK duty cycle during TDM to receive
+		 * DSP_B formatted data in LEFTJ mode (no delaying of the 1st
+		 * data bit).
+		 */
+		serial_format = TAS5720_SAIF_LEFTJ;
+		break;
+	case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF):
+		/* No delay after the frame sync */
+		serial_format = TAS5720_SAIF_LEFTJ;
+		break;
+	default:
+		dev_vdbg(codec->dev, "DAI Format is not found\n");
+		return -EINVAL;
+	}
+
+	ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG,
+				  TAS5720_SAIF_FORMAT_MASK,
+				  serial_format);
+	if (ret < 0) {
+		dev_err(codec->dev, "error setting SAIF format: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int tas5720_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;
+	unsigned int first_slot;
+	int ret;
+
+	if (!tx_mask) {
+		dev_err(codec->dev, "tx masks must not be 0\n");
+		return -EINVAL;
+	}
+
+	/*
+	 * Determine the first slot that is being requested. We will only
+	 * use the first slot that is found since the TAS5720 is a mono
+	 * amplifier.
+	 */
+	first_slot = __ffs(tx_mask);
+
+	if (first_slot > 7) {
+		dev_err(codec->dev, "slot selection out of bounds (%u)\n",
+			first_slot);
+		return -EINVAL;
+	}
+
+	/* Enable manual TDM slot selection (instead of I2C ID based) */
+	ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG,
+				  TAS5720_TDM_CFG_SRC, TAS5720_TDM_CFG_SRC);
+	if (ret < 0)
+		goto error_snd_soc_update_bits;
+
+	/* Configure the TDM slot to process audio from */
+	ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG,
+				  TAS5720_TDM_SLOT_SEL_MASK, first_slot);
+	if (ret < 0)
+		goto error_snd_soc_update_bits;
+
+	return 0;
+
+error_snd_soc_update_bits:
+	dev_err(codec->dev, "error configuring TDM mode: %d\n", ret);
+	return ret;
+}
+
+static int tas5720_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	int ret;
+
+	ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG,
+				  TAS5720_MUTE, mute ? TAS5720_MUTE : 0);
+	if (ret < 0) {
+		dev_err(codec->dev, "error (un-)muting device: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static void tas5720_fault_check_work(struct work_struct *work)
+{
+	struct tas5720_data *tas5720 = container_of(work, struct tas5720_data,
+			fault_check_work.work);
+	struct device *dev = tas5720->codec->dev;
+	unsigned int curr_fault;
+	int ret;
+
+	ret = regmap_read(tas5720->regmap, TAS5720_FAULT_REG, &curr_fault);
+	if (ret < 0) {
+		dev_err(dev, "failed to read FAULT register: %d\n", ret);
+		goto out;
+	}
+
+	/* Check/handle all errors except SAIF clock errors */
+	curr_fault &= TAS5720_OCE | TAS5720_DCE | TAS5720_OTE;
+
+	/*
+	 * Only flag errors once for a given occurrence. This is needed as
+	 * the TAS5720 will take time clearing the fault condition internally
+	 * during which we don't want to bombard the system with the same
+	 * error message over and over.
+	 */
+	if ((curr_fault & TAS5720_OCE) && !(tas5720->last_fault & TAS5720_OCE))
+		dev_crit(dev, "experienced an over current hardware fault\n");
+
+	if ((curr_fault & TAS5720_DCE) && !(tas5720->last_fault & TAS5720_DCE))
+		dev_crit(dev, "experienced a DC detection fault\n");
+
+	if ((curr_fault & TAS5720_OTE) && !(tas5720->last_fault & TAS5720_OTE))
+		dev_crit(dev, "experienced an over temperature fault\n");
+
+	/* Store current fault value so we can detect any changes next time */
+	tas5720->last_fault = curr_fault;
+
+	if (!curr_fault)
+		goto out;
+
+	/*
+	 * Periodically toggle SDZ (shutdown bit) H->L->H to clear any latching
+	 * faults as long as a fault condition persists. Always going through
+	 * the full sequence no matter the first return value to minimizes
+	 * chances for the device to end up in shutdown mode.
+	 */
+	ret = regmap_write_bits(tas5720->regmap, TAS5720_POWER_CTRL_REG,
+				TAS5720_SDZ, 0);
+	if (ret < 0)
+		dev_err(dev, "failed to write POWER_CTRL register: %d\n", ret);
+
+	ret = regmap_write_bits(tas5720->regmap, TAS5720_POWER_CTRL_REG,
+				TAS5720_SDZ, TAS5720_SDZ);
+	if (ret < 0)
+		dev_err(dev, "failed to write POWER_CTRL register: %d\n", ret);
+
+out:
+	/* Schedule the next fault check at the specified interval */
+	schedule_delayed_work(&tas5720->fault_check_work,
+			      msecs_to_jiffies(TAS5720_FAULT_CHECK_INTERVAL));
+}
+
+static int tas5720_codec_probe(struct snd_soc_codec *codec)
+{
+	struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec);
+	unsigned int device_id;
+	int ret;
+
+	tas5720->codec = codec;
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(tas5720->supplies),
+				    tas5720->supplies);
+	if (ret != 0) {
+		dev_err(codec->dev, "failed to enable supplies: %d\n", ret);
+		return ret;
+	}
+
+	ret = regmap_read(tas5720->regmap, TAS5720_DEVICE_ID_REG, &device_id);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to read device ID register: %d\n",
+			ret);
+		goto probe_fail;
+	}
+
+	if (device_id != TAS5720_DEVICE_ID) {
+		dev_err(codec->dev, "wrong device ID. expected: %u read: %u\n",
+			TAS5720_DEVICE_ID, device_id);
+		ret = -ENODEV;
+		goto probe_fail;
+	}
+
+	/* Set device to mute */
+	ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG,
+				  TAS5720_MUTE, TAS5720_MUTE);
+	if (ret < 0)
+		goto error_snd_soc_update_bits;
+
+	/*
+	 * Enter shutdown mode - our default when not playing audio - to
+	 * minimize current consumption. On the TAS5720 there is no real down
+	 * side doing so as all device registers are preserved and the wakeup
+	 * of the codec is rather quick which we do using a dapm widget.
+	 */
+	ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG,
+				  TAS5720_SDZ, 0);
+	if (ret < 0)
+		goto error_snd_soc_update_bits;
+
+	INIT_DELAYED_WORK(&tas5720->fault_check_work, tas5720_fault_check_work);
+
+	return 0;
+
+error_snd_soc_update_bits:
+	dev_err(codec->dev, "error configuring device registers: %d\n", ret);
+
+probe_fail:
+	regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies),
+			       tas5720->supplies);
+	return ret;
+}
+
+static int tas5720_codec_remove(struct snd_soc_codec *codec)
+{
+	struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	cancel_delayed_work_sync(&tas5720->fault_check_work);
+
+	ret = regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies),
+				     tas5720->supplies);
+	if (ret < 0)
+		dev_err(codec->dev, "failed to disable supplies: %d\n", ret);
+
+	return ret;
+};
+
+static int tas5720_dac_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 tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	if (event & SND_SOC_DAPM_POST_PMU) {
+		/* Take TAS5720 out of shutdown mode */
+		ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG,
+					  TAS5720_SDZ, TAS5720_SDZ);
+		if (ret < 0) {
+			dev_err(codec->dev, "error waking codec: %d\n", ret);
+			return ret;
+		}
+
+		/*
+		 * Observe codec shutdown-to-active time. The datasheet only
+		 * lists a nominal value however just use-it as-is without
+		 * additional padding to minimize the delay introduced in
+		 * starting to play audio (actually there is other setup done
+		 * by the ASoC framework that will provide additional delays,
+		 * so we should always be safe).
+		 */
+		msleep(25);
+
+		/* Turn on TAS5720 periodic fault checking/handling */
+		tas5720->last_fault = 0;
+		schedule_delayed_work(&tas5720->fault_check_work,
+				msecs_to_jiffies(TAS5720_FAULT_CHECK_INTERVAL));
+	} else if (event & SND_SOC_DAPM_PRE_PMD) {
+		/* Disable TAS5720 periodic fault checking/handling */
+		cancel_delayed_work_sync(&tas5720->fault_check_work);
+
+		/* Place TAS5720 in shutdown mode to minimize current draw */
+		ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG,
+					  TAS5720_SDZ, 0);
+		if (ret < 0) {
+			dev_err(codec->dev, "error shutting down codec: %d\n",
+				ret);
+			return ret;
+		}
+	}
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int tas5720_suspend(struct snd_soc_codec *codec)
+{
+	struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	regcache_cache_only(tas5720->regmap, true);
+	regcache_mark_dirty(tas5720->regmap);
+
+	ret = regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies),
+				     tas5720->supplies);
+	if (ret < 0)
+		dev_err(codec->dev, "failed to disable supplies: %d\n", ret);
+
+	return ret;
+}
+
+static int tas5720_resume(struct snd_soc_codec *codec)
+{
+	struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(tas5720->supplies),
+				    tas5720->supplies);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to enable supplies: %d\n", ret);
+		return ret;
+	}
+
+	regcache_cache_only(tas5720->regmap, false);
+
+	ret = regcache_sync(tas5720->regmap);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to sync regcache: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+#else
+#define tas5720_suspend NULL
+#define tas5720_resume NULL
+#endif
+
+static bool tas5720_is_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TAS5720_DEVICE_ID_REG:
+	case TAS5720_FAULT_REG:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static const struct regmap_config tas5720_regmap_config = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.max_register = TAS5720_MAX_REG,
+	.cache_type = REGCACHE_RBTREE,
+	.volatile_reg = tas5720_is_volatile_reg,
+};
+
+/*
+ * DAC analog gain. There are four discrete values to select from, ranging
+ * from 19.2 dB to 26.3dB.
+ */
+static const DECLARE_TLV_DB_RANGE(dac_analog_tlv,
+	0x0, 0x0, TLV_DB_SCALE_ITEM(1920, 0, 0),
+	0x1, 0x1, TLV_DB_SCALE_ITEM(2070, 0, 0),
+	0x2, 0x2, TLV_DB_SCALE_ITEM(2350, 0, 0),
+	0x3, 0x3, TLV_DB_SCALE_ITEM(2630, 0, 0),
+);
+
+/*
+ * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that
+ * setting the gain below -100 dB (register value <0x7) is effectively a MUTE
+ * as per device datasheet.
+ */
+static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0);
+
+static const struct snd_kcontrol_new tas5720_snd_controls[] = {
+	SOC_SINGLE_TLV("Speaker Driver Playback Volume",
+		       TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, dac_tlv),
+	SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG,
+		       TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv),
+};
+
+static const struct snd_soc_dapm_widget tas5720_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas5720_dac_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_OUTPUT("OUT")
+};
+
+static const struct snd_soc_dapm_route tas5720_audio_map[] = {
+	{ "DAC", NULL, "DAC IN" },
+	{ "OUT", NULL, "DAC" },
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_tas5720 = {
+	.probe = tas5720_codec_probe,
+	.remove = tas5720_codec_remove,
+	.suspend = tas5720_suspend,
+	.resume = tas5720_resume,
+
+	.controls = tas5720_snd_controls,
+	.num_controls = ARRAY_SIZE(tas5720_snd_controls),
+	.dapm_widgets = tas5720_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets),
+	.dapm_routes = tas5720_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(tas5720_audio_map),
+};
+
+/* PCM rates supported by the TAS5720 driver */
+#define TAS5720_RATES	(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+			 SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+/* Formats supported by TAS5720 driver */
+#define TAS5720_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE |\
+			 SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops tas5720_speaker_dai_ops = {
+	.hw_params	= tas5720_hw_params,
+	.set_fmt	= tas5720_set_dai_fmt,
+	.set_tdm_slot	= tas5720_set_dai_tdm_slot,
+	.digital_mute	= tas5720_mute,
+};
+
+/*
+ * TAS5720 DAI structure
+ *
+ * Note that were are advertising .playback.channels_max = 2 despite this being
+ * a mono amplifier. The reason for that is that some serial ports such as TI's
+ * McASP module have a minimum number of channels (2) that they can output.
+ * Advertising more channels than we have will allow us to interface with such
+ * a serial port without really any negative side effects as the TAS5720 will
+ * simply ignore any extra channel(s) asides from the one channel that is
+ * configured to be played back.
+ */
+static struct snd_soc_dai_driver tas5720_dai[] = {
+	{
+		.name = "tas5720-amplifier",
+		.playback = {
+			.stream_name = "Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = TAS5720_RATES,
+			.formats = TAS5720_FORMATS,
+		},
+		.ops = &tas5720_speaker_dai_ops,
+	},
+};
+
+static int tas5720_probe(struct i2c_client *client,
+			 const struct i2c_device_id *id)
+{
+	struct device *dev = &client->dev;
+	struct tas5720_data *data;
+	int ret;
+	int i;
+
+	data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL);
+	if (!data)
+		return -ENOMEM;
+
+	data->tas5720_client = client;
+	data->regmap = devm_regmap_init_i2c(client, &tas5720_regmap_config);
+	if (IS_ERR(data->regmap)) {
+		ret = PTR_ERR(data->regmap);
+		dev_err(dev, "failed to allocate register map: %d\n", ret);
+		return ret;
+	}
+
+	for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
+		data->supplies[i].supply = tas5720_supply_names[i];
+
+	ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies),
+				      data->supplies);
+	if (ret != 0) {
+		dev_err(dev, "failed to request supplies: %d\n", ret);
+		return ret;
+	}
+
+	dev_set_drvdata(dev, data);
+
+	ret = snd_soc_register_codec(&client->dev,
+				     &soc_codec_dev_tas5720,
+				     tas5720_dai, ARRAY_SIZE(tas5720_dai));
+	if (ret < 0) {
+		dev_err(dev, "failed to register codec: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int tas5720_remove(struct i2c_client *client)
+{
+	struct device *dev = &client->dev;
+
+	snd_soc_unregister_codec(dev);
+
+	return 0;
+}
+
+static const struct i2c_device_id tas5720_id[] = {
+	{ "tas5720", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, tas5720_id);
+
+#if IS_ENABLED(CONFIG_OF)
+static const struct of_device_id tas5720_of_match[] = {
+	{ .compatible = "ti,tas5720", },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, tas5720_of_match);
+#endif
+
+static struct i2c_driver tas5720_i2c_driver = {
+	.driver = {
+		.name = "tas5720",
+		.of_match_table = of_match_ptr(tas5720_of_match),
+	},
+	.probe = tas5720_probe,
+	.remove = tas5720_remove,
+	.id_table = tas5720_id,
+};
+
+module_i2c_driver(tas5720_i2c_driver);
+
+MODULE_AUTHOR("Andreas Dannenberg <dannenberg@ti.com>");
+MODULE_DESCRIPTION("TAS5720 Audio amplifier driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tas5720.h b/sound/soc/codecs/tas5720.h
new file mode 100644
index 000000000..3d077c779
--- /dev/null
+++ b/sound/soc/codecs/tas5720.h
@@ -0,0 +1,90 @@
+/*
+ * tas5720.h - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier
+ *
+ * Copyright (C)2015-2016 Texas Instruments Incorporated -  http://www.ti.com
+ *
+ * Author: Andreas Dannenberg <dannenberg@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.
+ */
+
+#ifndef __TAS5720_H__
+#define __TAS5720_H__
+
+/* Register Address Map */
+#define TAS5720_DEVICE_ID_REG		0x00
+#define TAS5720_POWER_CTRL_REG		0x01
+#define TAS5720_DIGITAL_CTRL1_REG	0x02
+#define TAS5720_DIGITAL_CTRL2_REG	0x03
+#define TAS5720_VOLUME_CTRL_REG		0x04
+#define TAS5720_ANALOG_CTRL_REG		0x06
+#define TAS5720_FAULT_REG		0x08
+#define TAS5720_DIGITAL_CLIP2_REG	0x10
+#define TAS5720_DIGITAL_CLIP1_REG	0x11
+#define TAS5720_MAX_REG			TAS5720_DIGITAL_CLIP1_REG
+
+/* TAS5720_DEVICE_ID_REG */
+#define TAS5720_DEVICE_ID		0x01
+
+/* TAS5720_POWER_CTRL_REG */
+#define TAS5720_DIG_CLIP_MASK		GENMASK(7, 2)
+#define TAS5720_SLEEP			BIT(1)
+#define TAS5720_SDZ			BIT(0)
+
+/* TAS5720_DIGITAL_CTRL1_REG */
+#define TAS5720_HPF_BYPASS		BIT(7)
+#define TAS5720_TDM_CFG_SRC		BIT(6)
+#define TAS5720_SSZ_DS			BIT(3)
+#define TAS5720_SAIF_RIGHTJ_24BIT	(0x0)
+#define TAS5720_SAIF_RIGHTJ_20BIT	(0x1)
+#define TAS5720_SAIF_RIGHTJ_18BIT	(0x2)
+#define TAS5720_SAIF_RIGHTJ_16BIT	(0x3)
+#define TAS5720_SAIF_I2S		(0x4)
+#define TAS5720_SAIF_LEFTJ		(0x5)
+#define TAS5720_SAIF_FORMAT_MASK	GENMASK(2, 0)
+
+/* TAS5720_DIGITAL_CTRL2_REG */
+#define TAS5720_MUTE			BIT(4)
+#define TAS5720_TDM_SLOT_SEL_MASK	GENMASK(2, 0)
+
+/* TAS5720_ANALOG_CTRL_REG */
+#define TAS5720_PWM_RATE_6_3_FSYNC	(0x0 << 4)
+#define TAS5720_PWM_RATE_8_4_FSYNC	(0x1 << 4)
+#define TAS5720_PWM_RATE_10_5_FSYNC	(0x2 << 4)
+#define TAS5720_PWM_RATE_12_6_FSYNC	(0x3 << 4)
+#define TAS5720_PWM_RATE_14_7_FSYNC	(0x4 << 4)
+#define TAS5720_PWM_RATE_16_8_FSYNC	(0x5 << 4)
+#define TAS5720_PWM_RATE_20_10_FSYNC	(0x6 << 4)
+#define TAS5720_PWM_RATE_24_12_FSYNC	(0x7 << 4)
+#define TAS5720_PWM_RATE_MASK		GENMASK(6, 4)
+#define TAS5720_ANALOG_GAIN_19_2DBV	(0x0 << 2)
+#define TAS5720_ANALOG_GAIN_20_7DBV	(0x1 << 2)
+#define TAS5720_ANALOG_GAIN_23_5DBV	(0x2 << 2)
+#define TAS5720_ANALOG_GAIN_26_3DBV	(0x3 << 2)
+#define TAS5720_ANALOG_GAIN_MASK	GENMASK(3, 2)
+#define TAS5720_ANALOG_GAIN_SHIFT	(0x2)
+
+/* TAS5720_FAULT_REG */
+#define TAS5720_OC_THRESH_100PCT	(0x0 << 4)
+#define TAS5720_OC_THRESH_75PCT		(0x1 << 4)
+#define TAS5720_OC_THRESH_50PCT		(0x2 << 4)
+#define TAS5720_OC_THRESH_25PCT		(0x3 << 4)
+#define TAS5720_OC_THRESH_MASK		GENMASK(5, 4)
+#define TAS5720_CLKE			BIT(3)
+#define TAS5720_OCE			BIT(2)
+#define TAS5720_DCE			BIT(1)
+#define TAS5720_OTE			BIT(0)
+#define TAS5720_FAULT_MASK		GENMASK(3, 0)
+
+/* TAS5720_DIGITAL_CLIP1_REG */
+#define TAS5720_CLIP1_MASK		GENMASK(7, 2)
+#define TAS5720_CLIP1_SHIFT		(0x2)
+
+#endif /* __TAS5720_H__ */
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index ee4def4f8..3c5e1df01 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -28,6 +28,7 @@
 #include <linux/i2c.h>
 #include <linux/gpio.h>
 #include <linux/regulator/consumer.h>
+#include <linux/acpi.h>
 #include <linux/of.h>
 #include <linux/of_gpio.h>
 #include <linux/slab.h>
@@ -1280,10 +1281,19 @@ static const struct i2c_device_id aic31xx_i2c_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id);
 
+#ifdef CONFIG_ACPI
+static const struct acpi_device_id aic31xx_acpi_match[] = {
+	{ "10TI3100", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match);
+#endif
+
 static struct i2c_driver aic31xx_i2c_driver = {
 	.driver = {
 		.name	= "tlv320aic31xx-codec",
 		.of_match_table = of_match_ptr(tlv320aic31xx_of_match),
+		.acpi_match_table = ACPI_PTR(aic31xx_acpi_match),
 	},
 	.probe		= aic31xx_i2c_probe,
 	.remove		= aic31xx_i2c_remove,
diff --git a/sound/soc/codecs/tlv320aic32x4-i2c.c b/sound/soc/codecs/tlv320aic32x4-i2c.c
new file mode 100644
index 000000000..59606cf30
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic32x4-i2c.c
@@ -0,0 +1,74 @@
+/*
+ * linux/sound/soc/codecs/tlv320aic32x4-i2c.c
+ *
+ * Copyright 2011 NW Digital Radio
+ *
+ * Author: Jeremy McDermond <nh6z@nh6z.net>
+ *
+ * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27.
+ *
+ * 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/i2c.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+
+#include "tlv320aic32x4.h"
+
+static int aic32x4_i2c_probe(struct i2c_client *i2c,
+			     const struct i2c_device_id *id)
+{
+	struct regmap *regmap;
+	struct regmap_config config;
+
+	config = aic32x4_regmap_config;
+	config.reg_bits = 8;
+	config.val_bits = 8;
+
+	regmap = devm_regmap_init_i2c(i2c, &config);
+	return aic32x4_probe(&i2c->dev, regmap);
+}
+
+static int aic32x4_i2c_remove(struct i2c_client *i2c)
+{
+	return aic32x4_remove(&i2c->dev);
+}
+
+static const struct i2c_device_id aic32x4_i2c_id[] = {
+	{ "tlv320aic32x4", 0 },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
+
+static const struct of_device_id aic32x4_of_id[] = {
+	{ .compatible = "ti,tlv320aic32x4", },
+	{ /* senitel */ }
+};
+MODULE_DEVICE_TABLE(of, aic32x4_of_id);
+
+static struct i2c_driver aic32x4_i2c_driver = {
+	.driver = {
+		.name = "tlv320aic32x4",
+		.of_match_table = aic32x4_of_id,
+	},
+	.probe =    aic32x4_i2c_probe,
+	.remove =   aic32x4_i2c_remove,
+	.id_table = aic32x4_i2c_id,
+};
+
+module_i2c_driver(aic32x4_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver I2C");
+MODULE_AUTHOR("Jeremy McDermond <nh6z@nh6z.net>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c
new file mode 100644
index 000000000..724fcdd49
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic32x4-spi.c
@@ -0,0 +1,76 @@
+/*
+ * linux/sound/soc/codecs/tlv320aic32x4-spi.c
+ *
+ * Copyright 2011 NW Digital Radio
+ *
+ * Author: Jeremy McDermond <nh6z@nh6z.net>
+ *
+ * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27.
+ *
+ * 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/spi/spi.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+
+#include "tlv320aic32x4.h"
+
+static int aic32x4_spi_probe(struct spi_device *spi)
+{
+	struct regmap *regmap;
+	struct regmap_config config;
+
+	config = aic32x4_regmap_config;
+	config.reg_bits = 7;
+	config.pad_bits = 1;
+	config.val_bits = 8;
+	config.read_flag_mask = 0x01;
+
+	regmap = devm_regmap_init_spi(spi, &config);
+	return aic32x4_probe(&spi->dev, regmap);
+}
+
+static int aic32x4_spi_remove(struct spi_device *spi)
+{
+	return aic32x4_remove(&spi->dev);
+}
+
+static const struct spi_device_id aic32x4_spi_id[] = {
+	{ "tlv320aic32x4", 0 },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(spi, aic32x4_spi_id);
+
+static const struct of_device_id aic32x4_of_id[] = {
+	{ .compatible = "ti,tlv320aic32x4", },
+	{ /* senitel */ }
+};
+MODULE_DEVICE_TABLE(of, aic32x4_of_id);
+
+static struct spi_driver aic32x4_spi_driver = {
+	.driver = {
+		.name = "tlv320aic32x4",
+		.owner = THIS_MODULE,
+		.of_match_table = aic32x4_of_id,
+	},
+	.probe =    aic32x4_spi_probe,
+	.remove =   aic32x4_spi_remove,
+	.id_table = aic32x4_spi_id,
+};
+
+module_spi_driver(aic32x4_spi_driver);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver SPI");
+MODULE_AUTHOR("Jeremy McDermond <nh6z@nh6z.net>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index f2d319196..85d4978d0 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -30,7 +30,6 @@
 #include <linux/pm.h>
 #include <linux/gpio.h>
 #include <linux/of_gpio.h>
-#include <linux/i2c.h>
 #include <linux/cdev.h>
 #include <linux/slab.h>
 #include <linux/clk.h>
@@ -160,7 +159,10 @@ static const struct aic32x4_rate_divs aic32x4_divs[] = {
 	/* 48k rate */
 	{AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4},
 	{AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4},
-	{AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4}
+	{AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4},
+
+	/* 96k rate */
+	{AIC32X4_FREQ_25000000, 96000, 2, 7, 8643, 64, 4, 4, 64, 4, 4, 1},
 };
 
 static const struct snd_kcontrol_new hpl_output_mixer_controls[] = {
@@ -181,16 +183,71 @@ static const struct snd_kcontrol_new lor_output_mixer_controls[] = {
 	SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_LORROUTE, 3, 1, 0),
 };
 
-static const struct snd_kcontrol_new left_input_mixer_controls[] = {
-	SOC_DAPM_SINGLE("IN1_L P Switch", AIC32X4_LMICPGAPIN, 6, 1, 0),
-	SOC_DAPM_SINGLE("IN2_L P Switch", AIC32X4_LMICPGAPIN, 4, 1, 0),
-	SOC_DAPM_SINGLE("IN3_L P Switch", AIC32X4_LMICPGAPIN, 2, 1, 0),
+static const char * const resistor_text[] = {
+	"Off", "10 kOhm", "20 kOhm", "40 kOhm",
 };
 
-static const struct snd_kcontrol_new right_input_mixer_controls[] = {
-	SOC_DAPM_SINGLE("IN1_R P Switch", AIC32X4_RMICPGAPIN, 6, 1, 0),
-	SOC_DAPM_SINGLE("IN2_R P Switch", AIC32X4_RMICPGAPIN, 4, 1, 0),
-	SOC_DAPM_SINGLE("IN3_R P Switch", AIC32X4_RMICPGAPIN, 2, 1, 0),
+/* Left mixer pins */
+static SOC_ENUM_SINGLE_DECL(in1l_lpga_p_enum, AIC32X4_LMICPGAPIN, 6, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in2l_lpga_p_enum, AIC32X4_LMICPGAPIN, 4, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in3l_lpga_p_enum, AIC32X4_LMICPGAPIN, 2, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in1r_lpga_p_enum, AIC32X4_LMICPGAPIN, 0, resistor_text);
+
+static SOC_ENUM_SINGLE_DECL(cml_lpga_n_enum, AIC32X4_LMICPGANIN, 6, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in2r_lpga_n_enum, AIC32X4_LMICPGANIN, 4, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in3r_lpga_n_enum, AIC32X4_LMICPGANIN, 2, resistor_text);
+
+static const struct snd_kcontrol_new in1l_to_lmixer_controls[] = {
+	SOC_DAPM_ENUM("IN1_L L+ Switch", in1l_lpga_p_enum),
+};
+static const struct snd_kcontrol_new in2l_to_lmixer_controls[] = {
+	SOC_DAPM_ENUM("IN2_L L+ Switch", in2l_lpga_p_enum),
+};
+static const struct snd_kcontrol_new in3l_to_lmixer_controls[] = {
+	SOC_DAPM_ENUM("IN3_L L+ Switch", in3l_lpga_p_enum),
+};
+static const struct snd_kcontrol_new in1r_to_lmixer_controls[] = {
+	SOC_DAPM_ENUM("IN1_R L+ Switch", in1r_lpga_p_enum),
+};
+static const struct snd_kcontrol_new cml_to_lmixer_controls[] = {
+	SOC_DAPM_ENUM("CM_L L- Switch", cml_lpga_n_enum),
+};
+static const struct snd_kcontrol_new in2r_to_lmixer_controls[] = {
+	SOC_DAPM_ENUM("IN2_R L- Switch", in2r_lpga_n_enum),
+};
+static const struct snd_kcontrol_new in3r_to_lmixer_controls[] = {
+	SOC_DAPM_ENUM("IN3_R L- Switch", in3r_lpga_n_enum),
+};
+
+/*  Right mixer pins */
+static SOC_ENUM_SINGLE_DECL(in1r_rpga_p_enum, AIC32X4_RMICPGAPIN, 6, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in2r_rpga_p_enum, AIC32X4_RMICPGAPIN, 4, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in3r_rpga_p_enum, AIC32X4_RMICPGAPIN, 2, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in2l_rpga_p_enum, AIC32X4_RMICPGAPIN, 0, resistor_text);
+static SOC_ENUM_SINGLE_DECL(cmr_rpga_n_enum, AIC32X4_RMICPGANIN, 6, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in1l_rpga_n_enum, AIC32X4_RMICPGANIN, 4, resistor_text);
+static SOC_ENUM_SINGLE_DECL(in3l_rpga_n_enum, AIC32X4_RMICPGANIN, 2, resistor_text);
+
+static const struct snd_kcontrol_new in1r_to_rmixer_controls[] = {
+	SOC_DAPM_ENUM("IN1_R R+ Switch", in1r_rpga_p_enum),
+};
+static const struct snd_kcontrol_new in2r_to_rmixer_controls[] = {
+	SOC_DAPM_ENUM("IN2_R R+ Switch", in2r_rpga_p_enum),
+};
+static const struct snd_kcontrol_new in3r_to_rmixer_controls[] = {
+	SOC_DAPM_ENUM("IN3_R R+ Switch", in3r_rpga_p_enum),
+};
+static const struct snd_kcontrol_new in2l_to_rmixer_controls[] = {
+	SOC_DAPM_ENUM("IN2_L R+ Switch", in2l_rpga_p_enum),
+};
+static const struct snd_kcontrol_new cmr_to_rmixer_controls[] = {
+	SOC_DAPM_ENUM("CM_R R- Switch", cmr_rpga_n_enum),
+};
+static const struct snd_kcontrol_new in1l_to_rmixer_controls[] = {
+	SOC_DAPM_ENUM("IN1_L R- Switch", in1l_rpga_n_enum),
+};
+static const struct snd_kcontrol_new in3l_to_rmixer_controls[] = {
+	SOC_DAPM_ENUM("IN3_L R- Switch", in3l_rpga_n_enum),
 };
 
 static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = {
@@ -214,14 +271,39 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = {
 			   &lor_output_mixer_controls[0],
 			   ARRAY_SIZE(lor_output_mixer_controls)),
 	SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0),
-	SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0,
-			   &left_input_mixer_controls[0],
-			   ARRAY_SIZE(left_input_mixer_controls)),
-	SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0,
-			   &right_input_mixer_controls[0],
-			   ARRAY_SIZE(right_input_mixer_controls)),
-	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0),
+
 	SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0),
+	SND_SOC_DAPM_MUX("IN1_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+			in1r_to_rmixer_controls),
+	SND_SOC_DAPM_MUX("IN2_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+			in2r_to_rmixer_controls),
+	SND_SOC_DAPM_MUX("IN3_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+			in3r_to_rmixer_controls),
+	SND_SOC_DAPM_MUX("IN2_L to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+			in2l_to_rmixer_controls),
+	SND_SOC_DAPM_MUX("CM_R to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+			cmr_to_rmixer_controls),
+	SND_SOC_DAPM_MUX("IN1_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+			in1l_to_rmixer_controls),
+	SND_SOC_DAPM_MUX("IN3_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+			in3l_to_rmixer_controls),
+
+	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0),
+	SND_SOC_DAPM_MUX("IN1_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+			in1l_to_lmixer_controls),
+	SND_SOC_DAPM_MUX("IN2_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+			in2l_to_lmixer_controls),
+	SND_SOC_DAPM_MUX("IN3_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+			in3l_to_lmixer_controls),
+	SND_SOC_DAPM_MUX("IN1_R to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0,
+			in1r_to_lmixer_controls),
+	SND_SOC_DAPM_MUX("CM_L to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+			cml_to_lmixer_controls),
+	SND_SOC_DAPM_MUX("IN2_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+			in2r_to_lmixer_controls),
+	SND_SOC_DAPM_MUX("IN3_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0,
+			in3r_to_lmixer_controls),
+
 	SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0),
 
 	SND_SOC_DAPM_OUTPUT("HPL"),
@@ -261,19 +343,77 @@ static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = {
 	{"LOR Power", NULL, "LOR Output Mixer"},
 	{"LOR", NULL, "LOR Power"},
 
-	/* Left input */
-	{"Left Input Mixer", "IN1_L P Switch", "IN1_L"},
-	{"Left Input Mixer", "IN2_L P Switch", "IN2_L"},
-	{"Left Input Mixer", "IN3_L P Switch", "IN3_L"},
-
-	{"Left ADC", NULL, "Left Input Mixer"},
-
 	/* Right Input */
-	{"Right Input Mixer", "IN1_R P Switch", "IN1_R"},
-	{"Right Input Mixer", "IN2_R P Switch", "IN2_R"},
-	{"Right Input Mixer", "IN3_R P Switch", "IN3_R"},
-
-	{"Right ADC", NULL, "Right Input Mixer"},
+	{"Right ADC", NULL, "IN1_R to Right Mixer Positive Resistor"},
+	{"IN1_R to Right Mixer Positive Resistor", "10 kOhm", "IN1_R"},
+	{"IN1_R to Right Mixer Positive Resistor", "20 kOhm", "IN1_R"},
+	{"IN1_R to Right Mixer Positive Resistor", "40 kOhm", "IN1_R"},
+
+	{"Right ADC", NULL, "IN2_R to Right Mixer Positive Resistor"},
+	{"IN2_R to Right Mixer Positive Resistor", "10 kOhm", "IN2_R"},
+	{"IN2_R to Right Mixer Positive Resistor", "20 kOhm", "IN2_R"},
+	{"IN2_R to Right Mixer Positive Resistor", "40 kOhm", "IN2_R"},
+
+	{"Right ADC", NULL, "IN3_R to Right Mixer Positive Resistor"},
+	{"IN3_R to Right Mixer Positive Resistor", "10 kOhm", "IN3_R"},
+	{"IN3_R to Right Mixer Positive Resistor", "20 kOhm", "IN3_R"},
+	{"IN3_R to Right Mixer Positive Resistor", "40 kOhm", "IN3_R"},
+
+	{"Right ADC", NULL, "IN2_L to Right Mixer Positive Resistor"},
+	{"IN2_L to Right Mixer Positive Resistor", "10 kOhm", "IN2_L"},
+	{"IN2_L to Right Mixer Positive Resistor", "20 kOhm", "IN2_L"},
+	{"IN2_L to Right Mixer Positive Resistor", "40 kOhm", "IN2_L"},
+
+	{"Right ADC", NULL, "CM_R to Right Mixer Negative Resistor"},
+	{"CM_R to Right Mixer Negative Resistor", "10 kOhm", "CM_R"},
+	{"CM_R to Right Mixer Negative Resistor", "20 kOhm", "CM_R"},
+	{"CM_R to Right Mixer Negative Resistor", "40 kOhm", "CM_R"},
+
+	{"Right ADC", NULL, "IN1_L to Right Mixer Negative Resistor"},
+	{"IN1_L to Right Mixer Negative Resistor", "10 kOhm", "IN1_L"},
+	{"IN1_L to Right Mixer Negative Resistor", "20 kOhm", "IN1_L"},
+	{"IN1_L to Right Mixer Negative Resistor", "40 kOhm", "IN1_L"},
+
+	{"Right ADC", NULL, "IN3_L to Right Mixer Negative Resistor"},
+	{"IN3_L to Right Mixer Negative Resistor", "10 kOhm", "IN3_L"},
+	{"IN3_L to Right Mixer Negative Resistor", "20 kOhm", "IN3_L"},
+	{"IN3_L to Right Mixer Negative Resistor", "40 kOhm", "IN3_L"},
+
+	/* Left Input */
+	{"Left ADC", NULL, "IN1_L to Left Mixer Positive Resistor"},
+	{"IN1_L to Left Mixer Positive Resistor", "10 kOhm", "IN1_L"},
+	{"IN1_L to Left Mixer Positive Resistor", "20 kOhm", "IN1_L"},
+	{"IN1_L to Left Mixer Positive Resistor", "40 kOhm", "IN1_L"},
+
+	{"Left ADC", NULL, "IN2_L to Left Mixer Positive Resistor"},
+	{"IN2_L to Left Mixer Positive Resistor", "10 kOhm", "IN2_L"},
+	{"IN2_L to Left Mixer Positive Resistor", "20 kOhm", "IN2_L"},
+	{"IN2_L to Left Mixer Positive Resistor", "40 kOhm", "IN2_L"},
+
+	{"Left ADC", NULL, "IN3_L to Left Mixer Positive Resistor"},
+	{"IN3_L to Left Mixer Positive Resistor", "10 kOhm", "IN3_L"},
+	{"IN3_L to Left Mixer Positive Resistor", "20 kOhm", "IN3_L"},
+	{"IN3_L to Left Mixer Positive Resistor", "40 kOhm", "IN3_L"},
+
+	{"Left ADC", NULL, "IN1_R to Left Mixer Positive Resistor"},
+	{"IN1_R to Left Mixer Positive Resistor", "10 kOhm", "IN1_R"},
+	{"IN1_R to Left Mixer Positive Resistor", "20 kOhm", "IN1_R"},
+	{"IN1_R to Left Mixer Positive Resistor", "40 kOhm", "IN1_R"},
+
+	{"Left ADC", NULL, "CM_L to Left Mixer Negative Resistor"},
+	{"CM_L to Left Mixer Negative Resistor", "10 kOhm", "CM_L"},
+	{"CM_L to Left Mixer Negative Resistor", "20 kOhm", "CM_L"},
+	{"CM_L to Left Mixer Negative Resistor", "40 kOhm", "CM_L"},
+
+	{"Left ADC", NULL, "IN2_R to Left Mixer Negative Resistor"},
+	{"IN2_R to Left Mixer Negative Resistor", "10 kOhm", "IN2_R"},
+	{"IN2_R to Left Mixer Negative Resistor", "20 kOhm", "IN2_R"},
+	{"IN2_R to Left Mixer Negative Resistor", "40 kOhm", "IN2_R"},
+
+	{"Left ADC", NULL, "IN3_R to Left Mixer Negative Resistor"},
+	{"IN3_R to Left Mixer Negative Resistor", "10 kOhm", "IN3_R"},
+	{"IN3_R to Left Mixer Negative Resistor", "20 kOhm", "IN3_R"},
+	{"IN3_R to Left Mixer Negative Resistor", "40 kOhm", "IN3_R"},
 };
 
 static const struct regmap_range_cfg aic32x4_regmap_pages[] = {
@@ -287,14 +427,12 @@ static const struct regmap_range_cfg aic32x4_regmap_pages[] = {
 	},
 };
 
-static const struct regmap_config aic32x4_regmap = {
-	.reg_bits = 8,
-	.val_bits = 8,
-
+const struct regmap_config aic32x4_regmap_config = {
 	.max_register = AIC32X4_RMICPGAVOL,
 	.ranges = aic32x4_regmap_pages,
 	.num_ranges = ARRAY_SIZE(aic32x4_regmap_pages),
 };
+EXPORT_SYMBOL(aic32x4_regmap_config);
 
 static inline int aic32x4_get_divs(int mclk, int rate)
 {
@@ -567,7 +705,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-#define AIC32X4_RATES	SNDRV_PCM_RATE_8000_48000
+#define AIC32X4_RATES	SNDRV_PCM_RATE_8000_96000
 #define AIC32X4_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
 			 | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
 
@@ -596,7 +734,7 @@ static struct snd_soc_dai_driver aic32x4_dai = {
 	.symmetric_rates = 1,
 };
 
-static int aic32x4_probe(struct snd_soc_codec *codec)
+static int aic32x4_codec_probe(struct snd_soc_codec *codec)
 {
 	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
 	u32 tmp_reg;
@@ -655,7 +793,7 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
 }
 
 static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
-	.probe = aic32x4_probe,
+	.probe = aic32x4_codec_probe,
 	.set_bias_level = aic32x4_set_bias_level,
 	.suspend_bias_off = true,
 
@@ -777,24 +915,22 @@ error_ldo:
 	return ret;
 }
 
-static int aic32x4_i2c_probe(struct i2c_client *i2c,
-			     const struct i2c_device_id *id)
+int aic32x4_probe(struct device *dev, struct regmap *regmap)
 {
-	struct aic32x4_pdata *pdata = i2c->dev.platform_data;
 	struct aic32x4_priv *aic32x4;
-	struct device_node *np = i2c->dev.of_node;
+	struct aic32x4_pdata *pdata = dev->platform_data;
+	struct device_node *np = dev->of_node;
 	int ret;
 
-	aic32x4 = devm_kzalloc(&i2c->dev, sizeof(struct aic32x4_priv),
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
+
+	aic32x4 = devm_kzalloc(dev, sizeof(struct aic32x4_priv),
 			       GFP_KERNEL);
 	if (aic32x4 == NULL)
 		return -ENOMEM;
 
-	aic32x4->regmap = devm_regmap_init_i2c(i2c, &aic32x4_regmap);
-	if (IS_ERR(aic32x4->regmap))
-		return PTR_ERR(aic32x4->regmap);
-
-	i2c_set_clientdata(i2c, aic32x4);
+	dev_set_drvdata(dev, aic32x4);
 
 	if (pdata) {
 		aic32x4->power_cfg = pdata->power_cfg;
@@ -804,7 +940,7 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c,
 	} else if (np) {
 		ret = aic32x4_parse_dt(aic32x4, np);
 		if (ret) {
-			dev_err(&i2c->dev, "Failed to parse DT node\n");
+			dev_err(dev, "Failed to parse DT node\n");
 			return ret;
 		}
 	} else {
@@ -814,71 +950,48 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c,
 		aic32x4->rstn_gpio = -1;
 	}
 
-	aic32x4->mclk = devm_clk_get(&i2c->dev, "mclk");
+	aic32x4->mclk = devm_clk_get(dev, "mclk");
 	if (IS_ERR(aic32x4->mclk)) {
-		dev_err(&i2c->dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n");
+		dev_err(dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n");
 		return PTR_ERR(aic32x4->mclk);
 	}
 
 	if (gpio_is_valid(aic32x4->rstn_gpio)) {
-		ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio,
+		ret = devm_gpio_request_one(dev, aic32x4->rstn_gpio,
 				GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn");
 		if (ret != 0)
 			return ret;
 	}
 
-	ret = aic32x4_setup_regulators(&i2c->dev, aic32x4);
+	ret = aic32x4_setup_regulators(dev, aic32x4);
 	if (ret) {
-		dev_err(&i2c->dev, "Failed to setup regulators\n");
+		dev_err(dev, "Failed to setup regulators\n");
 		return ret;
 	}
 
-	ret = snd_soc_register_codec(&i2c->dev,
+	ret = snd_soc_register_codec(dev,
 			&soc_codec_dev_aic32x4, &aic32x4_dai, 1);
 	if (ret) {
-		dev_err(&i2c->dev, "Failed to register codec\n");
+		dev_err(dev, "Failed to register codec\n");
 		aic32x4_disable_regulators(aic32x4);
 		return ret;
 	}
 
-	i2c_set_clientdata(i2c, aic32x4);
-
 	return 0;
 }
+EXPORT_SYMBOL(aic32x4_probe);
 
-static int aic32x4_i2c_remove(struct i2c_client *client)
+int aic32x4_remove(struct device *dev)
 {
-	struct aic32x4_priv *aic32x4 = i2c_get_clientdata(client);
+	struct aic32x4_priv *aic32x4 = dev_get_drvdata(dev);
 
 	aic32x4_disable_regulators(aic32x4);
 
-	snd_soc_unregister_codec(&client->dev);
+	snd_soc_unregister_codec(dev);
+
 	return 0;
 }
-
-static const struct i2c_device_id aic32x4_i2c_id[] = {
-	{ "tlv320aic32x4", 0 },
-	{ }
-};
-MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
-
-static const struct of_device_id aic32x4_of_id[] = {
-	{ .compatible = "ti,tlv320aic32x4", },
-	{ /* senitel */ }
-};
-MODULE_DEVICE_TABLE(of, aic32x4_of_id);
-
-static struct i2c_driver aic32x4_i2c_driver = {
-	.driver = {
-		.name = "tlv320aic32x4",
-		.of_match_table = aic32x4_of_id,
-	},
-	.probe =    aic32x4_i2c_probe,
-	.remove =   aic32x4_i2c_remove,
-	.id_table = aic32x4_i2c_id,
-};
-
-module_i2c_driver(aic32x4_i2c_driver);
+EXPORT_SYMBOL(aic32x4_remove);
 
 MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver");
 MODULE_AUTHOR("Javier Martin <javier.martin@vista-silicon.com>");
diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h
index 995f033a8..a197dd51a 100644
--- a/sound/soc/codecs/tlv320aic32x4.h
+++ b/sound/soc/codecs/tlv320aic32x4.h
@@ -10,6 +10,13 @@
 #ifndef _TLV320AIC32X4_H
 #define _TLV320AIC32X4_H
 
+struct device;
+struct regmap_config;
+
+extern const struct regmap_config aic32x4_regmap_config;
+int aic32x4_probe(struct device *dev, struct regmap *regmap);
+int aic32x4_remove(struct device *dev);
+
 /* tlv320aic32x4 register space (in decimal to match datasheet) */
 
 #define AIC32X4_PAGE1		128
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index bc3de2e84..1f7081043 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -824,7 +824,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
 {
 	struct twl6040 *twl6040 = codec->control_data;
 	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
-	int ret;
+	int ret = 0;
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
@@ -832,12 +832,16 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (priv->codec_powered)
+		if (priv->codec_powered) {
+			/* Select low power PLL in standby */
+			ret = twl6040_set_pll(twl6040, TWL6040_SYSCLK_SEL_LPPLL,
+					      32768, 19200000);
 			break;
+		}
 
 		ret = twl6040_power(twl6040, 1);
 		if (ret)
-			return ret;
+			break;
 
 		priv->codec_powered = 1;
 
@@ -853,7 +857,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	return 0;
+	return ret;
 }
 
 static int twl6040_startup(struct snd_pcm_substream *substream,
@@ -983,9 +987,9 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i
 		if (mute) {
 			/* Power down drivers and DACs */
 			hflctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA |
-				    TWL6040_HFDRVENA);
+				    TWL6040_HFDRVENA | TWL6040_HFSWENA);
 			hfrctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA |
-				    TWL6040_HFDRVENA);
+				    TWL6040_HFDRVENA | TWL6040_HFSWENA);
 		}
 
 		twl6040_reg_write(twl6040, TWL6040_REG_HFLCTL, hflctl);
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 171a23ddd..512a9d25f 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -17,6 +17,7 @@
 #include <linux/export.h>
 #include <linux/pm.h>
 #include <linux/gcd.h>
+#include <linux/gpio/driver.h>
 #include <linux/gpio.h>
 #include <linux/i2c.h>
 #include <linux/pm_runtime.h>
@@ -2236,14 +2237,9 @@ static irqreturn_t wm5100_edge_irq(int irq, void *data)
 }
 
 #ifdef CONFIG_GPIOLIB
-static inline struct wm5100_priv *gpio_to_wm5100(struct gpio_chip *chip)
-{
-	return container_of(chip, struct wm5100_priv, gpio_chip);
-}
-
 static void wm5100_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
 {
-	struct wm5100_priv *wm5100 = gpio_to_wm5100(chip);
+	struct wm5100_priv *wm5100 = gpiochip_get_data(chip);
 
 	regmap_update_bits(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset,
 			   WM5100_GP1_LVL, !!value << WM5100_GP1_LVL_SHIFT);
@@ -2252,7 +2248,7 @@ static void wm5100_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
 static int wm5100_gpio_direction_out(struct gpio_chip *chip,
 				     unsigned offset, int value)
 {
-	struct wm5100_priv *wm5100 = gpio_to_wm5100(chip);
+	struct wm5100_priv *wm5100 = gpiochip_get_data(chip);
 	int val, ret;
 
 	val = (1 << WM5100_GP1_FN_SHIFT) | (!!value << WM5100_GP1_LVL_SHIFT);
@@ -2268,7 +2264,7 @@ static int wm5100_gpio_direction_out(struct gpio_chip *chip,
 
 static int wm5100_gpio_get(struct gpio_chip *chip, unsigned offset)
 {
-	struct wm5100_priv *wm5100 = gpio_to_wm5100(chip);
+	struct wm5100_priv *wm5100 = gpiochip_get_data(chip);
 	unsigned int reg;
 	int ret;
 
@@ -2281,7 +2277,7 @@ static int wm5100_gpio_get(struct gpio_chip *chip, unsigned offset)
 
 static int wm5100_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
 {
-	struct wm5100_priv *wm5100 = gpio_to_wm5100(chip);
+	struct wm5100_priv *wm5100 = gpiochip_get_data(chip);
 
 	return regmap_update_bits(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset,
 				  WM5100_GP1_FN_MASK | WM5100_GP1_DIR,
@@ -2313,7 +2309,7 @@ static void wm5100_init_gpio(struct i2c_client *i2c)
 	else
 		wm5100->gpio_chip.base = -1;
 
-	ret = gpiochip_add(&wm5100->gpio_chip);
+	ret = gpiochip_add_data(&wm5100->gpio_chip, wm5100);
 	if (ret != 0)
 		dev_err(&i2c->dev, "Failed to add GPIOs: %d\n", ret);
 }
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 1bae17ee8..e7fe6b7b9 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -1872,7 +1872,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = {
 		.capture = {
 			.stream_name = "Audio Trace CPU",
 			.channels_min = 1,
-			.channels_max = 6,
+			.channels_max = 4,
 			.rates = WM5102_RATES,
 			.formats = WM5102_FORMATS,
 		},
@@ -2098,10 +2098,14 @@ static int wm5102_probe(struct platform_device *pdev)
 
 static int wm5102_remove(struct platform_device *pdev)
 {
+	struct wm5102_priv *wm5102 = platform_get_drvdata(pdev);
+
 	snd_soc_unregister_platform(&pdev->dev);
 	snd_soc_unregister_codec(&pdev->dev);
 	pm_runtime_disable(&pdev->dev);
 
+	wm_adsp2_remove(&wm5102->core.adsp[0]);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 2728ac545..d54f1b46c 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -1723,6 +1723,7 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
 	{ "OUT2L", NULL, "SYSCLK" },
 	{ "OUT2R", NULL, "SYSCLK" },
 	{ "OUT3L", NULL, "SYSCLK" },
+	{ "OUT3R", NULL, "SYSCLK" },
 	{ "OUT4L", NULL, "SYSCLK" },
 	{ "OUT4R", NULL, "SYSCLK" },
 	{ "OUT5L", NULL, "SYSCLK" },
@@ -2437,10 +2438,16 @@ static int wm5110_probe(struct platform_device *pdev)
 
 static int wm5110_remove(struct platform_device *pdev)
 {
+	struct wm5110_priv *wm5110 = platform_get_drvdata(pdev);
+	int i;
+
 	snd_soc_unregister_platform(&pdev->dev);
 	snd_soc_unregister_codec(&pdev->dev);
 	pm_runtime_disable(&pdev->dev);
 
+	for (i = 0; i < WM5110_NUM_ADSP; i++)
+		wm_adsp2_remove(&wm5110->core.adsp[i]);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index a82b8bc2c..a26ca490c 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -20,7 +20,7 @@
 #include <linux/init.h>
 #include <linux/completion.h>
 #include <linux/delay.h>
-#include <linux/gpio.h>
+#include <linux/gpio/driver.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/regmap.h>
@@ -1766,11 +1766,6 @@ static int wm8903_resume(struct snd_soc_codec *codec)
 }
 
 #ifdef CONFIG_GPIOLIB
-static inline struct wm8903_priv *gpio_to_wm8903(struct gpio_chip *chip)
-{
-	return container_of(chip, struct wm8903_priv, gpio_chip);
-}
-
 static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset)
 {
 	if (offset >= WM8903_NUM_GPIO)
@@ -1781,7 +1776,7 @@ static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset)
 
 static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
 {
-	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
+	struct wm8903_priv *wm8903 = gpiochip_get_data(chip);
 	unsigned int mask, val;
 	int ret;
 
@@ -1799,7 +1794,7 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
 
 static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset)
 {
-	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
+	struct wm8903_priv *wm8903 = gpiochip_get_data(chip);
 	unsigned int reg;
 
 	regmap_read(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, &reg);
@@ -1810,7 +1805,7 @@ static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset)
 static int wm8903_gpio_direction_out(struct gpio_chip *chip,
 				     unsigned offset, int value)
 {
-	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
+	struct wm8903_priv *wm8903 = gpiochip_get_data(chip);
 	unsigned int mask, val;
 	int ret;
 
@@ -1828,7 +1823,7 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip,
 
 static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
 {
-	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
+	struct wm8903_priv *wm8903 = gpiochip_get_data(chip);
 
 	regmap_update_bits(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset,
 			   WM8903_GP1_LVL_MASK,
@@ -1860,7 +1855,7 @@ static void wm8903_init_gpio(struct wm8903_priv *wm8903)
 	else
 		wm8903->gpio_chip.base = -1;
 
-	ret = gpiochip_add(&wm8903->gpio_chip);
+	ret = gpiochip_add_data(&wm8903->gpio_chip, wm8903);
 	if (ret != 0)
 		dev_err(wm8903->dev, "Failed to add GPIOs: %d\n", ret);
 }
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index f6f9395ea..1c600819f 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -743,6 +743,7 @@ static const struct regmap_config wm8940_regmap = {
 	.max_register = WM8940_MONOMIX,
 	.reg_defaults = wm8940_reg_defaults,
 	.num_reg_defaults = ARRAY_SIZE(wm8940_reg_defaults),
+	.cache_type = REGCACHE_RBTREE,
 
 	.readable_reg = wm8940_readable_register,
 	.volatile_reg = wm8940_volatile_register,
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 720a14e06..f3109da24 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -18,7 +18,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/gcd.h>
-#include <linux/gpio.h>
+#include <linux/gpio/driver.h>
 #include <linux/i2c.h>
 #include <linux/input.h>
 #include <linux/pm_runtime.h>
@@ -3307,14 +3307,9 @@ static void wm8962_set_gpio_mode(struct wm8962_priv *wm8962, int gpio)
 }
 
 #ifdef CONFIG_GPIOLIB
-static inline struct wm8962_priv *gpio_to_wm8962(struct gpio_chip *chip)
-{
-	return container_of(chip, struct wm8962_priv, gpio_chip);
-}
-
 static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset)
 {
-	struct wm8962_priv *wm8962 = gpio_to_wm8962(chip);
+	struct wm8962_priv *wm8962 = gpiochip_get_data(chip);
 
 	/* The WM8962 GPIOs aren't linearly numbered.  For simplicity
 	 * we export linear numbers and error out if the unsupported
@@ -3337,7 +3332,7 @@ static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset)
 
 static void wm8962_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
 {
-	struct wm8962_priv *wm8962 = gpio_to_wm8962(chip);
+	struct wm8962_priv *wm8962 = gpiochip_get_data(chip);
 	struct snd_soc_codec *codec = wm8962->codec;
 
 	snd_soc_update_bits(codec, WM8962_GPIO_BASE + offset,
@@ -3347,7 +3342,7 @@ static void wm8962_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
 static int wm8962_gpio_direction_out(struct gpio_chip *chip,
 				     unsigned offset, int value)
 {
-	struct wm8962_priv *wm8962 = gpio_to_wm8962(chip);
+	struct wm8962_priv *wm8962 = gpiochip_get_data(chip);
 	struct snd_soc_codec *codec = wm8962->codec;
 	int ret, val;
 
@@ -3386,7 +3381,7 @@ static void wm8962_init_gpio(struct snd_soc_codec *codec)
 	else
 		wm8962->gpio_chip.base = -1;
 
-	ret = gpiochip_add(&wm8962->gpio_chip);
+	ret = gpiochip_add_data(&wm8962->gpio_chip, wm8962);
 	if (ret != 0)
 		dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret);
 }
@@ -3798,9 +3793,8 @@ static int wm8962_runtime_resume(struct device *dev)
 	ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies),
 				    wm8962->supplies);
 	if (ret != 0) {
-		dev_err(dev,
-			"Failed to enable supplies: %d\n", ret);
-		return ret;
+		dev_err(dev, "Failed to enable supplies: %d\n", ret);
+		goto disable_clock;
 	}
 
 	regcache_cache_only(wm8962->regmap, false);
@@ -3838,6 +3832,10 @@ static int wm8962_runtime_resume(struct device *dev)
 	msleep(5);
 
 	return 0;
+
+disable_clock:
+	clk_disable_unprepare(wm8962->pdata.mclk);
+	return ret;
 }
 
 static int wm8962_runtime_suspend(struct device *dev)
diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h
index 910aafd09..e63a318a3 100644
--- a/sound/soc/codecs/wm8962.h
+++ b/sound/soc/codecs/wm8962.h
@@ -16,9 +16,9 @@
 #include <asm/types.h>
 #include <sound/soc.h>
 
-#define WM8962_SYSCLK_MCLK 1
-#define WM8962_SYSCLK_FLL  2
-#define WM8962_SYSCLK_PLL3 3
+#define WM8962_SYSCLK_MCLK 0
+#define WM8962_SYSCLK_FLL  1
+#define WM8962_SYSCLK_PLL3 2
 
 #define WM8962_FLL  1
 
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index f99b34f76..a73044251 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -17,6 +17,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/gcd.h>
+#include <linux/gpio/driver.h>
 #include <linux/gpio.h>
 #include <linux/i2c.h>
 #include <linux/regmap.h>
@@ -2139,14 +2140,9 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 }
 
 #ifdef CONFIG_GPIOLIB
-static inline struct wm8996_priv *gpio_to_wm8996(struct gpio_chip *chip)
-{
-	return container_of(chip, struct wm8996_priv, gpio_chip);
-}
-
 static void wm8996_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
 {
-	struct wm8996_priv *wm8996 = gpio_to_wm8996(chip);
+	struct wm8996_priv *wm8996 = gpiochip_get_data(chip);
 
 	regmap_update_bits(wm8996->regmap, WM8996_GPIO_1 + offset,
 			   WM8996_GP1_LVL, !!value << WM8996_GP1_LVL_SHIFT);
@@ -2155,7 +2151,7 @@ static void wm8996_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
 static int wm8996_gpio_direction_out(struct gpio_chip *chip,
 				     unsigned offset, int value)
 {
-	struct wm8996_priv *wm8996 = gpio_to_wm8996(chip);
+	struct wm8996_priv *wm8996 = gpiochip_get_data(chip);
 	int val;
 
 	val = (1 << WM8996_GP1_FN_SHIFT) | (!!value << WM8996_GP1_LVL_SHIFT);
@@ -2167,7 +2163,7 @@ static int wm8996_gpio_direction_out(struct gpio_chip *chip,
 
 static int wm8996_gpio_get(struct gpio_chip *chip, unsigned offset)
 {
-	struct wm8996_priv *wm8996 = gpio_to_wm8996(chip);
+	struct wm8996_priv *wm8996 = gpiochip_get_data(chip);
 	unsigned int reg;
 	int ret;
 
@@ -2180,7 +2176,7 @@ static int wm8996_gpio_get(struct gpio_chip *chip, unsigned offset)
 
 static int wm8996_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
 {
-	struct wm8996_priv *wm8996 = gpio_to_wm8996(chip);
+	struct wm8996_priv *wm8996 = gpiochip_get_data(chip);
 
 	return regmap_update_bits(wm8996->regmap, WM8996_GPIO_1 + offset,
 				  WM8996_GP1_FN_MASK | WM8996_GP1_DIR,
@@ -2211,7 +2207,7 @@ static void wm8996_init_gpio(struct wm8996_priv *wm8996)
 	else
 		wm8996->gpio_chip.base = -1;
 
-	ret = gpiochip_add(&wm8996->gpio_chip);
+	ret = gpiochip_add_data(&wm8996->gpio_chip, wm8996);
 	if (ret != 0)
 		dev_err(wm8996->dev, "Failed to add GPIOs: %d\n", ret);
 }
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 781f40a4d..78832c218 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -160,6 +160,8 @@
 #define ADSP2_RAM_RDY_SHIFT                    0
 #define ADSP2_RAM_RDY_WIDTH                    1
 
+#define ADSP_MAX_STD_CTRL_SIZE               512
+
 struct wm_adsp_buf {
 	struct list_head list;
 	void *buf;
@@ -271,8 +273,11 @@ struct wm_adsp_buffer {
 	__be32 words_written[2];	/* total words written (64 bit) */
 };
 
+struct wm_adsp_compr;
+
 struct wm_adsp_compr_buf {
 	struct wm_adsp *dsp;
+	struct wm_adsp_compr *compr;
 
 	struct wm_adsp_buffer_region *regions;
 	u32 host_buf_ptr;
@@ -435,6 +440,7 @@ struct wm_coeff_ctl {
 	size_t len;
 	unsigned int set:1;
 	struct snd_kcontrol *kcontrol;
+	struct soc_bytes_ext bytes_ext;
 	unsigned int flags;
 };
 
@@ -711,10 +717,17 @@ static void wm_adsp2_show_fw_status(struct wm_adsp *dsp)
 		 be16_to_cpu(scratch[3]));
 }
 
+static inline struct wm_coeff_ctl *bytes_ext_to_ctl(struct soc_bytes_ext *ext)
+{
+	return container_of(ext, struct wm_coeff_ctl, bytes_ext);
+}
+
 static int wm_coeff_info(struct snd_kcontrol *kctl,
 			 struct snd_ctl_elem_info *uinfo)
 {
-	struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value;
+	struct soc_bytes_ext *bytes_ext =
+		(struct soc_bytes_ext *)kctl->private_value;
+	struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
 
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
 	uinfo->count = ctl->len;
@@ -763,7 +776,9 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl,
 static int wm_coeff_put(struct snd_kcontrol *kctl,
 			struct snd_ctl_elem_value *ucontrol)
 {
-	struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value;
+	struct soc_bytes_ext *bytes_ext =
+		(struct soc_bytes_ext *)kctl->private_value;
+	struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
 	char *p = ucontrol->value.bytes.data;
 	int ret = 0;
 
@@ -780,6 +795,29 @@ static int wm_coeff_put(struct snd_kcontrol *kctl,
 	return ret;
 }
 
+static int wm_coeff_tlv_put(struct snd_kcontrol *kctl,
+			    const unsigned int __user *bytes, unsigned int size)
+{
+	struct soc_bytes_ext *bytes_ext =
+		(struct soc_bytes_ext *)kctl->private_value;
+	struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
+	int ret = 0;
+
+	mutex_lock(&ctl->dsp->pwr_lock);
+
+	if (copy_from_user(ctl->cache, bytes, size)) {
+		ret = -EFAULT;
+	} else {
+		ctl->set = 1;
+		if (ctl->enabled)
+			ret = wm_coeff_write_control(ctl, ctl->cache, size);
+	}
+
+	mutex_unlock(&ctl->dsp->pwr_lock);
+
+	return ret;
+}
+
 static int wm_coeff_read_control(struct wm_coeff_ctl *ctl,
 				 void *buf, size_t len)
 {
@@ -822,7 +860,9 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl,
 static int wm_coeff_get(struct snd_kcontrol *kctl,
 			struct snd_ctl_elem_value *ucontrol)
 {
-	struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value;
+	struct soc_bytes_ext *bytes_ext =
+		(struct soc_bytes_ext *)kctl->private_value;
+	struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
 	char *p = ucontrol->value.bytes.data;
 	int ret = 0;
 
@@ -845,12 +885,72 @@ static int wm_coeff_get(struct snd_kcontrol *kctl,
 	return ret;
 }
 
+static int wm_coeff_tlv_get(struct snd_kcontrol *kctl,
+			    unsigned int __user *bytes, unsigned int size)
+{
+	struct soc_bytes_ext *bytes_ext =
+		(struct soc_bytes_ext *)kctl->private_value;
+	struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext);
+	int ret = 0;
+
+	mutex_lock(&ctl->dsp->pwr_lock);
+
+	if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) {
+		if (ctl->enabled)
+			ret = wm_coeff_read_control(ctl, ctl->cache, size);
+		else
+			ret = -EPERM;
+	} else {
+		if (!ctl->flags && ctl->enabled)
+			ret = wm_coeff_read_control(ctl, ctl->cache, size);
+	}
+
+	if (!ret && copy_to_user(bytes, ctl->cache, size))
+		ret = -EFAULT;
+
+	mutex_unlock(&ctl->dsp->pwr_lock);
+
+	return ret;
+}
+
 struct wmfw_ctl_work {
 	struct wm_adsp *dsp;
 	struct wm_coeff_ctl *ctl;
 	struct work_struct work;
 };
 
+static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len)
+{
+	unsigned int out, rd, wr, vol;
+
+	if (len > ADSP_MAX_STD_CTRL_SIZE) {
+		rd = SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+		wr = SNDRV_CTL_ELEM_ACCESS_TLV_WRITE;
+		vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE;
+
+		out = SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK;
+	} else {
+		rd = SNDRV_CTL_ELEM_ACCESS_READ;
+		wr = SNDRV_CTL_ELEM_ACCESS_WRITE;
+		vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE;
+
+		out = 0;
+	}
+
+	if (in) {
+		if (in & WMFW_CTL_FLAG_READABLE)
+			out |= rd;
+		if (in & WMFW_CTL_FLAG_WRITEABLE)
+			out |= wr;
+		if (in & WMFW_CTL_FLAG_VOLATILE)
+			out |= vol;
+	} else {
+		out |= rd | wr | vol;
+	}
+
+	return out;
+}
+
 static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl)
 {
 	struct snd_kcontrol_new *kcontrol;
@@ -868,19 +968,15 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl)
 	kcontrol->info = wm_coeff_info;
 	kcontrol->get = wm_coeff_get;
 	kcontrol->put = wm_coeff_put;
-	kcontrol->private_value = (unsigned long)ctl;
+	kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+	kcontrol->tlv.c = snd_soc_bytes_tlv_callback;
+	kcontrol->private_value = (unsigned long)&ctl->bytes_ext;
 
-	if (ctl->flags) {
-		if (ctl->flags & WMFW_CTL_FLAG_WRITEABLE)
-			kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_WRITE;
-		if (ctl->flags & WMFW_CTL_FLAG_READABLE)
-			kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_READ;
-		if (ctl->flags & WMFW_CTL_FLAG_VOLATILE)
-			kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE;
-	} else {
-		kcontrol->access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
-		kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE;
-	}
+	ctl->bytes_ext.max = ctl->len;
+	ctl->bytes_ext.get = wm_coeff_tlv_get;
+	ctl->bytes_ext.put = wm_coeff_tlv_put;
+
+	kcontrol->access = wmfw_convert_flags(ctl->flags, ctl->len);
 
 	ret = snd_soc_add_card_controls(dsp->card, kcontrol, 1);
 	if (ret < 0)
@@ -944,6 +1040,13 @@ static void wm_adsp_ctl_work(struct work_struct *work)
 	kfree(ctl_work);
 }
 
+static void wm_adsp_free_ctl_blk(struct wm_coeff_ctl *ctl)
+{
+	kfree(ctl->cache);
+	kfree(ctl->name);
+	kfree(ctl);
+}
+
 static int wm_adsp_create_control(struct wm_adsp *dsp,
 				  const struct wm_adsp_alg_region *alg_region,
 				  unsigned int offset, unsigned int len,
@@ -1032,11 +1135,6 @@ static int wm_adsp_create_control(struct wm_adsp *dsp,
 
 	ctl->flags = flags;
 	ctl->offset = offset;
-	if (len > 512) {
-		adsp_warn(dsp, "Truncating control %s from %d\n",
-			  ctl->name, len);
-		len = 512;
-	}
 	ctl->len = len;
 	ctl->cache = kzalloc(ctl->len, GFP_KERNEL);
 	if (!ctl->cache) {
@@ -1564,6 +1662,19 @@ static struct wm_adsp_alg_region *wm_adsp_create_region(struct wm_adsp *dsp,
 	return alg_region;
 }
 
+static void wm_adsp_free_alg_regions(struct wm_adsp *dsp)
+{
+	struct wm_adsp_alg_region *alg_region;
+
+	while (!list_empty(&dsp->alg_regions)) {
+		alg_region = list_first_entry(&dsp->alg_regions,
+					      struct wm_adsp_alg_region,
+					      list);
+		list_del(&alg_region->list);
+		kfree(alg_region);
+	}
+}
+
 static int wm_adsp1_setup_algs(struct wm_adsp *dsp)
 {
 	struct wmfw_adsp1_id_hdr adsp1_id;
@@ -1994,7 +2105,6 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
 	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
 	struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec);
 	struct wm_adsp *dsp = &dsps[w->shift];
-	struct wm_adsp_alg_region *alg_region;
 	struct wm_coeff_ctl *ctl;
 	int ret;
 	unsigned int val;
@@ -2074,13 +2184,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w,
 		list_for_each_entry(ctl, &dsp->ctl_list, list)
 			ctl->enabled = 0;
 
-		while (!list_empty(&dsp->alg_regions)) {
-			alg_region = list_first_entry(&dsp->alg_regions,
-						      struct wm_adsp_alg_region,
-						      list);
-			list_del(&alg_region->list);
-			kfree(alg_region);
-		}
+
+		wm_adsp_free_alg_regions(dsp);
 		break;
 
 	default:
@@ -2222,7 +2327,6 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
 	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
 	struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec);
 	struct wm_adsp *dsp = &dsps[w->shift];
-	struct wm_adsp_alg_region *alg_region;
 	struct wm_coeff_ctl *ctl;
 	int ret;
 
@@ -2240,9 +2344,13 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
 		if (ret != 0)
 			goto err;
 
+		mutex_lock(&dsp->pwr_lock);
+
 		if (wm_adsp_fw[dsp->fw].num_caps != 0)
 			ret = wm_adsp_buffer_init(dsp);
 
+		mutex_unlock(&dsp->pwr_lock);
+
 		break;
 
 	case SND_SOC_DAPM_PRE_PMD:
@@ -2269,13 +2377,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
 		list_for_each_entry(ctl, &dsp->ctl_list, list)
 			ctl->enabled = 0;
 
-		while (!list_empty(&dsp->alg_regions)) {
-			alg_region = list_first_entry(&dsp->alg_regions,
-						      struct wm_adsp_alg_region,
-						      list);
-			list_del(&alg_region->list);
-			kfree(alg_region);
-		}
+		wm_adsp_free_alg_regions(dsp);
 
 		if (wm_adsp_fw[dsp->fw].num_caps != 0)
 			wm_adsp_buffer_free(dsp);
@@ -2340,6 +2442,54 @@ int wm_adsp2_init(struct wm_adsp *dsp)
 }
 EXPORT_SYMBOL_GPL(wm_adsp2_init);
 
+void wm_adsp2_remove(struct wm_adsp *dsp)
+{
+	struct wm_coeff_ctl *ctl;
+
+	while (!list_empty(&dsp->ctl_list)) {
+		ctl = list_first_entry(&dsp->ctl_list, struct wm_coeff_ctl,
+					list);
+		list_del(&ctl->list);
+		wm_adsp_free_ctl_blk(ctl);
+	}
+}
+EXPORT_SYMBOL_GPL(wm_adsp2_remove);
+
+static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr)
+{
+	return compr->buf != NULL;
+}
+
+static int wm_adsp_compr_attach(struct wm_adsp_compr *compr)
+{
+	/*
+	 * Note this will be more complex once each DSP can support multiple
+	 * streams
+	 */
+	if (!compr->dsp->buffer)
+		return -EINVAL;
+
+	compr->buf = compr->dsp->buffer;
+	compr->buf->compr = compr;
+
+	return 0;
+}
+
+static void wm_adsp_compr_detach(struct wm_adsp_compr *compr)
+{
+	if (!compr)
+		return;
+
+	/* Wake the poll so it can see buffer is no longer attached */
+	if (compr->stream)
+		snd_compr_fragment_elapsed(compr->stream);
+
+	if (wm_adsp_compr_attached(compr)) {
+		compr->buf->compr = NULL;
+		compr->buf = NULL;
+	}
+}
+
 int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream)
 {
 	struct wm_adsp_compr *compr;
@@ -2393,6 +2543,7 @@ int wm_adsp_compr_free(struct snd_compr_stream *stream)
 
 	mutex_lock(&dsp->pwr_lock);
 
+	wm_adsp_compr_detach(compr);
 	dsp->compr = NULL;
 
 	kfree(compr->raw_buf);
@@ -2689,6 +2840,8 @@ err_buffer:
 static int wm_adsp_buffer_free(struct wm_adsp *dsp)
 {
 	if (dsp->buffer) {
+		wm_adsp_compr_detach(dsp->buffer->compr);
+
 		kfree(dsp->buffer->regions);
 		kfree(dsp->buffer);
 
@@ -2698,25 +2851,6 @@ static int wm_adsp_buffer_free(struct wm_adsp *dsp)
 	return 0;
 }
 
-static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr)
-{
-	return compr->buf != NULL;
-}
-
-static int wm_adsp_compr_attach(struct wm_adsp_compr *compr)
-{
-	/*
-	 * Note this will be more complex once each DSP can support multiple
-	 * streams
-	 */
-	if (!compr->dsp->buffer)
-		return -EINVAL;
-
-	compr->buf = compr->dsp->buffer;
-
-	return 0;
-}
-
 int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd)
 {
 	struct wm_adsp_compr *compr = stream->runtime->private_data;
@@ -2805,21 +2939,41 @@ static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf)
 		avail += wm_adsp_buffer_size(buf);
 
 	adsp_dbg(buf->dsp, "readindex=0x%x, writeindex=0x%x, avail=%d\n",
-		 buf->read_index, write_index, avail);
+		 buf->read_index, write_index, avail * WM_ADSP_DATA_WORD_SIZE);
 
 	buf->avail = avail;
 
 	return 0;
 }
 
+static int wm_adsp_buffer_get_error(struct wm_adsp_compr_buf *buf)
+{
+	int ret;
+
+	ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error);
+	if (ret < 0) {
+		adsp_err(buf->dsp, "Failed to check buffer error: %d\n", ret);
+		return ret;
+	}
+	if (buf->error != 0) {
+		adsp_err(buf->dsp, "Buffer error occurred: %d\n", buf->error);
+		return -EIO;
+	}
+
+	return 0;
+}
+
 int wm_adsp_compr_handle_irq(struct wm_adsp *dsp)
 {
-	struct wm_adsp_compr_buf *buf = dsp->buffer;
-	struct wm_adsp_compr *compr = dsp->compr;
+	struct wm_adsp_compr_buf *buf;
+	struct wm_adsp_compr *compr;
 	int ret = 0;
 
 	mutex_lock(&dsp->pwr_lock);
 
+	buf = dsp->buffer;
+	compr = dsp->compr;
+
 	if (!buf) {
 		ret = -ENODEV;
 		goto out;
@@ -2827,16 +2981,9 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp)
 
 	adsp_dbg(dsp, "Handling buffer IRQ\n");
 
-	ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error);
-	if (ret < 0) {
-		adsp_err(dsp, "Failed to check buffer error: %d\n", ret);
-		goto out;
-	}
-	if (buf->error != 0) {
-		adsp_err(dsp, "Buffer error occurred: %d\n", buf->error);
-		ret = -EIO;
-		goto out;
-	}
+	ret = wm_adsp_buffer_get_error(buf);
+	if (ret < 0)
+		goto out_notify; /* Wake poll to report error */
 
 	ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(irq_count),
 				  &buf->irq_count);
@@ -2851,6 +2998,7 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp)
 		goto out;
 	}
 
+out_notify:
 	if (compr && compr->stream)
 		snd_compr_fragment_elapsed(compr->stream);
 
@@ -2879,14 +3027,16 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream,
 			  struct snd_compr_tstamp *tstamp)
 {
 	struct wm_adsp_compr *compr = stream->runtime->private_data;
-	struct wm_adsp_compr_buf *buf = compr->buf;
 	struct wm_adsp *dsp = compr->dsp;
+	struct wm_adsp_compr_buf *buf;
 	int ret = 0;
 
 	adsp_dbg(dsp, "Pointer request\n");
 
 	mutex_lock(&dsp->pwr_lock);
 
+	buf = compr->buf;
+
 	if (!compr->buf) {
 		ret = -ENXIO;
 		goto out;
@@ -2909,6 +3059,10 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream,
 		 * DSP to inform us once a whole fragment is available.
 		 */
 		if (buf->avail < wm_adsp_compr_frag_words(compr)) {
+			ret = wm_adsp_buffer_get_error(buf);
+			if (ret < 0)
+				goto out;
+
 			ret = wm_adsp_buffer_reenable_irq(buf);
 			if (ret < 0) {
 				adsp_err(dsp,
diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h
index b61cb57e6..feb61e2c4 100644
--- a/sound/soc/codecs/wm_adsp.h
+++ b/sound/soc/codecs/wm_adsp.h
@@ -92,6 +92,7 @@ extern const struct snd_kcontrol_new wm_adsp_fw_controls[];
 
 int wm_adsp1_init(struct wm_adsp *dsp);
 int wm_adsp2_init(struct wm_adsp *dsp);
+void wm_adsp2_remove(struct wm_adsp *dsp);
 int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec);
 int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec);
 int wm_adsp1_event(struct snd_soc_dapm_widget *w,
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 50ca291cc..6b732d8e5 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -16,7 +16,11 @@ config SND_EDMA_SOC
 	  - DRA7xx family
 
 config SND_DAVINCI_SOC_I2S
-	tristate
+	tristate "DaVinci Multichannel Buffered Serial Port (McBSP) support"
+	depends on SND_EDMA_SOC
+	help
+	  Say Y or M here if you want to have support for McBSP IP found in
+	  Texas Instruments DaVinci DA850 SoCs.
 
 config SND_DAVINCI_SOC_MCASP
 	tristate "Multichannel Audio Serial Port (McASP) support"
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index ec98548a5..384961651 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -4,9 +4,15 @@
  * Author:      Vladimir Barinov, <vbarinov@embeddedalley.com>
  * Copyright:   (C) 2007 MontaVista Software, Inc., <source@mvista.com>
  *
+ * DT support	(c) 2016 Petr Kulhavy, Barix AG <petr@barix.com>
+ *		based on davinci-mcasp.c DT support
+ *
  * 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.
+ *
+ * TODO:
+ * on DA850 implement HW FIFOs instead of DMA into DXR and DRR registers
  */
 
 #include <linux/init.h>
@@ -650,13 +656,24 @@ static const struct snd_soc_component_driver davinci_i2s_component = {
 
 static int davinci_i2s_probe(struct platform_device *pdev)
 {
+	struct snd_dmaengine_dai_dma_data *dma_data;
 	struct davinci_mcbsp_dev *dev;
 	struct resource *mem, *res;
 	void __iomem *io_base;
 	int *dma;
 	int ret;
 
-	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
+	if (!mem) {
+		dev_warn(&pdev->dev,
+			 "\"mpu\" mem resource not found, using index 0\n");
+		mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+		if (!mem) {
+			dev_err(&pdev->dev, "no mem resource?\n");
+			return -ENODEV;
+		}
+	}
+
 	io_base = devm_ioremap_resource(&pdev->dev, mem);
 	if (IS_ERR(io_base))
 		return PTR_ERR(io_base);
@@ -666,39 +683,43 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 	if (!dev)
 		return -ENOMEM;
 
-	dev->clk = clk_get(&pdev->dev, NULL);
-	if (IS_ERR(dev->clk))
-		return -ENODEV;
-	clk_enable(dev->clk);
-
 	dev->base = io_base;
 
-	dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr =
-	    (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG);
+	/* setup DMA, first TX, then RX */
+	dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
+	dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG);
 
-	dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr =
-	    (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG);
-
-	/* first TX, then RX */
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!res) {
-		dev_err(&pdev->dev, "no DMA resource\n");
-		ret = -ENXIO;
-		goto err_release_clk;
+	if (res) {
+		dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK];
+		*dma = res->start;
+		dma_data->filter_data = dma;
+	} else if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) {
+		dma_data->filter_data = "tx";
+	} else {
+		dev_err(&pdev->dev, "Missing DMA tx resource\n");
+		return -ENODEV;
 	}
-	dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK];
-	*dma = res->start;
-	dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = dma;
+
+	dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE];
+	dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG);
 
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-	if (!res) {
-		dev_err(&pdev->dev, "no DMA resource\n");
-		ret = -ENXIO;
-		goto err_release_clk;
+	if (res) {
+		dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE];
+		*dma = res->start;
+		dma_data->filter_data = dma;
+	} else if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) {
+		dma_data->filter_data = "rx";
+	} else {
+		dev_err(&pdev->dev, "Missing DMA rx resource\n");
+		return -ENODEV;
 	}
-	dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE];
-	*dma = res->start;
-	dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = dma;
+
+	dev->clk = clk_get(&pdev->dev, NULL);
+	if (IS_ERR(dev->clk))
+		return -ENODEV;
+	clk_enable(dev->clk);
 
 	dev->dev = &pdev->dev;
 	dev_set_drvdata(&pdev->dev, dev);
@@ -737,11 +758,18 @@ static int davinci_i2s_remove(struct platform_device *pdev)
 	return 0;
 }
 
+static const struct of_device_id davinci_i2s_match[] = {
+	{ .compatible = "ti,da850-mcbsp" },
+	{},
+};
+MODULE_DEVICE_TABLE(of, davinci_i2s_match);
+
 static struct platform_driver davinci_mcbsp_driver = {
 	.probe		= davinci_i2s_probe,
 	.remove		= davinci_i2s_remove,
 	.driver		= {
 		.name	= "davinci-mcbsp",
+		.of_match_table = of_match_ptr(davinci_i2s_match),
 	},
 };
 
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index e1324989b..237dc6700 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -489,7 +489,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
 
 		mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG,
-			       ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR);
+			       ACLKX | AFSX | ACLKR | AHCLKR | AFSR);
 		mcasp->bclk_master = 0;
 		break;
 	default:
@@ -540,21 +540,19 @@ out:
 	return ret;
 }
 
-static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
+static int __davinci_mcasp_set_clkdiv(struct davinci_mcasp *mcasp, int div_id,
 				      int div, bool explicit)
 {
-	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
-
 	pm_runtime_get_sync(mcasp->dev);
 	switch (div_id) {
-	case 0:		/* MCLK divider */
+	case MCASP_CLKDIV_AUXCLK:			/* MCLK divider */
 		mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG,
 			       AHCLKXDIV(div - 1), AHCLKXDIV_MASK);
 		mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG,
 			       AHCLKRDIV(div - 1), AHCLKRDIV_MASK);
 		break;
 
-	case 1:		/* BCLK divider */
+	case MCASP_CLKDIV_BCLK:			/* BCLK divider */
 		mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG,
 			       ACLKXDIV(div - 1), ACLKXDIV_MASK);
 		mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG,
@@ -563,7 +561,8 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
 			mcasp->bclk_div = div;
 		break;
 
-	case 2:	/*
+	case MCASP_CLKDIV_BCLK_FS_RATIO:
+		/*
 		 * 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
@@ -591,7 +590,9 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
 static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
 				    int div)
 {
-	return __davinci_mcasp_set_clkdiv(dai, div_id, div, 1);
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+
+	return __davinci_mcasp_set_clkdiv(mcasp, div_id, div, 1);
 }
 
 static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
@@ -999,27 +1000,53 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp,
 }
 
 static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp,
-				      unsigned int bclk_freq,
-				      int *error_ppm)
+				      unsigned int bclk_freq, bool set)
 {
-	int div = mcasp->sysclk_freq / bclk_freq;
-	int rem = mcasp->sysclk_freq % bclk_freq;
+	int error_ppm;
+	unsigned int sysclk_freq = mcasp->sysclk_freq;
+	u32 reg = mcasp_get_reg(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG);
+	int div = sysclk_freq / bclk_freq;
+	int rem = sysclk_freq % bclk_freq;
+	int aux_div = 1;
+
+	if (div > (ACLKXDIV_MASK + 1)) {
+		if (reg & AHCLKXE) {
+			aux_div = div / (ACLKXDIV_MASK + 1);
+			if (div % (ACLKXDIV_MASK + 1))
+				aux_div++;
+
+			sysclk_freq /= aux_div;
+			div = sysclk_freq / bclk_freq;
+			rem = sysclk_freq % bclk_freq;
+		} else if (set) {
+			dev_warn(mcasp->dev, "Too fast reference clock (%u)\n",
+				 sysclk_freq);
+		}
+	}
 
 	if (rem != 0) {
 		if (div == 0 ||
-		    ((mcasp->sysclk_freq / div) - bclk_freq) >
-		    (bclk_freq - (mcasp->sysclk_freq / (div+1)))) {
+		    ((sysclk_freq / div) - bclk_freq) >
+		    (bclk_freq - (sysclk_freq / (div+1)))) {
 			div++;
 			rem = rem - bclk_freq;
 		}
 	}
-	if (error_ppm)
-		*error_ppm =
-			(div*1000000 + (int)div64_long(1000000LL*rem,
-						       (int)bclk_freq))
-			/div - 1000000;
+	error_ppm = (div*1000000 + (int)div64_long(1000000LL*rem,
+		     (int)bclk_freq)) / div - 1000000;
+
+	if (set) {
+		if (error_ppm)
+			dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n",
+				 error_ppm);
+
+		__davinci_mcasp_set_clkdiv(mcasp, MCASP_CLKDIV_BCLK, div, 0);
+		if (reg & AHCLKXE)
+			__davinci_mcasp_set_clkdiv(mcasp, MCASP_CLKDIV_AUXCLK,
+						   aux_div, 0);
+	}
 
-	return div;
+	return error_ppm;
 }
 
 static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
@@ -1044,18 +1071,11 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 		int slots = mcasp->tdm_slots;
 		int rate = params_rate(params);
 		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)
-			dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n",
-				 ppm);
-
-		__davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0);
+		davinci_mcasp_calc_clk_div(mcasp, rate * sbits * slots, true);
 	}
 
 	ret = mcasp_common_hw_param(mcasp, substream->stream,
@@ -1166,7 +1186,8 @@ static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params,
 				davinci_mcasp_dai_rates[i];
 			int ppm;
 
-			davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm);
+			ppm = davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq,
+							 false);
 			if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) {
 				if (range.empty) {
 					range.min = davinci_mcasp_dai_rates[i];
@@ -1205,8 +1226,9 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params,
 			if (rd->mcasp->slot_width)
 				sbits = rd->mcasp->slot_width;
 
-			davinci_mcasp_calc_clk_div(rd->mcasp, sbits*slots*rate,
-						   &ppm);
+			ppm = davinci_mcasp_calc_clk_div(rd->mcasp,
+							 sbits * slots * rate,
+							 false);
 			if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) {
 				snd_mask_set(&nfmt, i);
 				count++;
@@ -1230,11 +1252,15 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 	int i, dir;
 	int tdm_slots = mcasp->tdm_slots;
 
-	if (mcasp->tdm_mask[substream->stream])
-		tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]);
+	/* Do not allow more then one stream per direction */
+	if (mcasp->substreams[substream->stream])
+		return -EBUSY;
 
 	mcasp->substreams[substream->stream] = substream;
 
+	if (mcasp->tdm_mask[substream->stream])
+		tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]);
+
 	if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
 		return 0;
 
@@ -1487,8 +1513,9 @@ static struct davinci_mcasp_pdata am33xx_mcasp_pdata = {
 };
 
 static struct davinci_mcasp_pdata dra7_mcasp_pdata = {
-	.tx_dma_offset = 0x200,
-	.rx_dma_offset = 0x284,
+	/* The CFG port offset will be calculated if it is needed */
+	.tx_dma_offset = 0,
+	.rx_dma_offset = 0,
 	.version = MCASP_VERSION_4,
 };
 
@@ -1708,6 +1735,52 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp)
 	return PCM_EDMA;
 }
 
+static u32 davinci_mcasp_txdma_offset(struct davinci_mcasp_pdata *pdata)
+{
+	int i;
+	u32 offset = 0;
+
+	if (pdata->version != MCASP_VERSION_4)
+		return pdata->tx_dma_offset;
+
+	for (i = 0; i < pdata->num_serializer; i++) {
+		if (pdata->serial_dir[i] == TX_MODE) {
+			if (!offset) {
+				offset = DAVINCI_MCASP_TXBUF_REG(i);
+			} else {
+				pr_err("%s: Only one serializer allowed!\n",
+				       __func__);
+				break;
+			}
+		}
+	}
+
+	return offset;
+}
+
+static u32 davinci_mcasp_rxdma_offset(struct davinci_mcasp_pdata *pdata)
+{
+	int i;
+	u32 offset = 0;
+
+	if (pdata->version != MCASP_VERSION_4)
+		return pdata->rx_dma_offset;
+
+	for (i = 0; i < pdata->num_serializer; i++) {
+		if (pdata->serial_dir[i] == RX_MODE) {
+			if (!offset) {
+				offset = DAVINCI_MCASP_RXBUF_REG(i);
+			} else {
+				pr_err("%s: Only one serializer allowed!\n",
+				       __func__);
+				break;
+			}
+		}
+	}
+
+	return offset;
+}
+
 static int davinci_mcasp_probe(struct platform_device *pdev)
 {
 	struct snd_dmaengine_dai_dma_data *dma_data;
@@ -1836,7 +1909,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	if (dat)
 		dma_data->addr = dat->start;
 	else
-		dma_data->addr = mem->start + pdata->tx_dma_offset;
+		dma_data->addr = mem->start + davinci_mcasp_txdma_offset(pdata);
 
 	dma = &mcasp->dma_request[SNDRV_PCM_STREAM_PLAYBACK];
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
@@ -1857,7 +1930,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 		if (dat)
 			dma_data->addr = dat->start;
 		else
-			dma_data->addr = mem->start + pdata->rx_dma_offset;
+			dma_data->addr =
+				mem->start + davinci_mcasp_rxdma_offset(pdata);
 
 		dma = &mcasp->dma_request[SNDRV_PCM_STREAM_CAPTURE];
 		res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index a3be108a8..afddc8010 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -85,9 +85,9 @@
 						(n << 2))
 
 /* Transmit Buffer for Serializer n */
-#define DAVINCI_MCASP_TXBUF_REG		0x200
+#define DAVINCI_MCASP_TXBUF_REG(n)	(0x200 + (n << 2))
 /* Receive Buffer for Serializer n */
-#define DAVINCI_MCASP_RXBUF_REG		0x280
+#define DAVINCI_MCASP_RXBUF_REG(n)	(0x280 + (n << 2))
 
 /* McASP FIFO Registers */
 #define DAVINCI_MCASP_V2_AFIFO_BASE	(0x1010)
@@ -306,4 +306,9 @@
 #define NUMEVT(x)	(((x) & 0xFF) << 8)
 #define NUMDMA_MASK	(0xFF)
 
+/* clock divider IDs */
+#define MCASP_CLKDIV_AUXCLK		0 /* HCLK divider from AUXCLK */
+#define MCASP_CLKDIV_BCLK		1 /* BCLK divider from HCLK */
+#define MCASP_CLKDIV_BCLK_FS_RATIO	2 /* to set BCLK FS ration */
+
 #endif	/* DAVINCI_MCASP_H */
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index bff258d7b..0db69b7e9 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -100,6 +100,7 @@ struct dw_i2s_dev {
 	struct device *dev;
 	u32 ccr;
 	u32 xfer_resolution;
+	u32 fifo_th;
 
 	/* data related to DMA transfers b/w i2s and DMAC */
 	union dw_i2s_snd_dma_data play_dma_data;
@@ -147,17 +148,18 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream)
 static void i2s_start(struct dw_i2s_dev *dev,
 		      struct snd_pcm_substream *substream)
 {
+	struct i2s_clk_config_data *config = &dev->config;
 	u32 i, irq;
 	i2s_write_reg(dev->i2s_base, IER, 1);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		for (i = 0; i < 4; i++) {
+		for (i = 0; i < (config->chan_nr / 2); i++) {
 			irq = i2s_read_reg(dev->i2s_base, IMR(i));
 			i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30);
 		}
 		i2s_write_reg(dev->i2s_base, ITER, 1);
 	} else {
-		for (i = 0; i < 4; i++) {
+		for (i = 0; i < (config->chan_nr / 2); i++) {
 			irq = i2s_read_reg(dev->i2s_base, IMR(i));
 			i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03);
 		}
@@ -231,14 +233,16 @@ static void dw_i2s_config(struct dw_i2s_dev *dev, int stream)
 		if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 			i2s_write_reg(dev->i2s_base, TCR(ch_reg),
 				      dev->xfer_resolution);
-			i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02);
+			i2s_write_reg(dev->i2s_base, TFCR(ch_reg),
+				      dev->fifo_th - 1);
 			irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg));
 			i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30);
 			i2s_write_reg(dev->i2s_base, TER(ch_reg), 1);
 		} else {
 			i2s_write_reg(dev->i2s_base, RCR(ch_reg),
 				      dev->xfer_resolution);
-			i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07);
+			i2s_write_reg(dev->i2s_base, RFCR(ch_reg),
+				      dev->fifo_th - 1);
 			irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg));
 			i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03);
 			i2s_write_reg(dev->i2s_base, RER(ch_reg), 1);
@@ -498,6 +502,7 @@ static int dw_configure_dai(struct dw_i2s_dev *dev,
 	 */
 	u32 comp1 = i2s_read_reg(dev->i2s_base, dev->i2s_reg_comp1);
 	u32 comp2 = i2s_read_reg(dev->i2s_base, dev->i2s_reg_comp2);
+	u32 fifo_depth = 1 << (1 + COMP1_FIFO_DEPTH_GLOBAL(comp1));
 	u32 idx;
 
 	if (dev->capability & DWC_I2S_RECORD &&
@@ -536,6 +541,7 @@ static int dw_configure_dai(struct dw_i2s_dev *dev,
 		dev->capability |= DW_I2S_SLAVE;
 	}
 
+	dev->fifo_th = fifo_depth / 2;
 	return 0;
 }
 
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 0754df771..2147994ab 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -21,6 +21,8 @@
 #include <sound/core.h>
 #include <sound/dmaengine_pcm.h>
 #include <sound/pcm_params.h>
+#include <linux/mfd/syscon.h>
+#include <linux/mfd/syscon/imx6q-iomuxc-gpr.h>
 
 #include "fsl_sai.h"
 #include "imx-pcm.h"
@@ -786,10 +788,12 @@ static int fsl_sai_probe(struct platform_device *pdev)
 {
 	struct device_node *np = pdev->dev.of_node;
 	struct fsl_sai *sai;
+	struct regmap *gpr;
 	struct resource *res;
 	void __iomem *base;
 	char tmp[8];
 	int irq, ret, i;
+	int index;
 
 	sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
 	if (!sai)
@@ -797,7 +801,8 @@ static int fsl_sai_probe(struct platform_device *pdev)
 
 	sai->pdev = pdev;
 
-	if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai"))
+	if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") ||
+	    of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai"))
 		sai->sai_on_imx = true;
 
 	sai->is_lsb_first = of_property_read_bool(np, "lsb-first");
@@ -877,6 +882,22 @@ static int fsl_sai_probe(struct platform_device *pdev)
 		fsl_sai_dai.symmetric_samplebits = 0;
 	}
 
+	if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) &&
+	    of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) {
+		gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr");
+		if (IS_ERR(gpr)) {
+			dev_err(&pdev->dev, "cannot find iomuxc registers\n");
+			return PTR_ERR(gpr);
+		}
+
+		index = of_alias_get_id(np, "sai");
+		if (index < 0)
+			return index;
+
+		regmap_update_bits(gpr, IOMUXC_GPR1, MCLK_DIR(index),
+				   MCLK_DIR(index));
+	}
+
 	sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
 	sai->dma_params_tx.addr = res->start + FSL_SAI_TDR;
 	sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX;
@@ -898,6 +919,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
 static const struct of_device_id fsl_sai_ids[] = {
 	{ .compatible = "fsl,vf610-sai", },
 	{ .compatible = "fsl,imx6sx-sai", },
+	{ .compatible = "fsl,imx6ul-sai", },
 	{ /* sentinel */ }
 };
 MODULE_DEVICE_TABLE(of, fsl_sai_ids);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index ed8de1035..bedec4a32 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -137,6 +137,7 @@ static bool fsl_ssi_volatile_reg(struct device *dev, unsigned int reg)
 	case CCSR_SSI_SACDAT:
 	case CCSR_SSI_SATAG:
 	case CCSR_SSI_SACCST:
+	case CCSR_SSI_SOR:
 		return true;
 	default:
 		return false;
@@ -261,6 +262,7 @@ struct fsl_ssi_private {
 	struct fsl_ssi_dbg dbg_stats;
 
 	const struct fsl_ssi_soc_data *soc;
+	struct device *dev;
 };
 
 /*
@@ -400,6 +402,26 @@ static void fsl_ssi_rxtx_config(struct fsl_ssi_private *ssi_private,
 }
 
 /*
+ * Clear RX or TX FIFO to remove samples from the previous
+ * stream session which may be still present in the FIFO and
+ * may introduce bad samples and/or channel slipping.
+ *
+ * Note: The SOR is not documented in recent IMX datasheet, but
+ * is described in IMX51 reference manual at section 56.3.3.15.
+ */
+static void fsl_ssi_fifo_clear(struct fsl_ssi_private *ssi_private,
+		bool is_rx)
+{
+	if (is_rx) {
+		regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR,
+			CCSR_SSI_SOR_RX_CLR, CCSR_SSI_SOR_RX_CLR);
+	} else {
+		regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR,
+			CCSR_SSI_SOR_TX_CLR, CCSR_SSI_SOR_TX_CLR);
+	}
+}
+
+/*
  * Calculate the bits that have to be disabled for the current stream that is
  * getting disabled. This keeps the bits enabled that are necessary for the
  * second stream to work if 'stream_active' is true.
@@ -474,9 +496,11 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable,
 	 * (online configuration)
 	 */
 	if (enable) {
-		regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier);
+		fsl_ssi_fifo_clear(ssi_private, vals->scr & CCSR_SSI_SCR_RE);
+
 		regmap_update_bits(regs, CCSR_SSI_SRCR, vals->srcr, vals->srcr);
 		regmap_update_bits(regs, CCSR_SSI_STCR, vals->stcr, vals->stcr);
+		regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier);
 	} else {
 		u32 sier;
 		u32 srcr;
@@ -506,8 +530,40 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable,
 
 config_done:
 	/* Enabling of subunits is done after configuration */
-	if (enable)
+	if (enable) {
+		if (ssi_private->use_dma && (vals->scr & CCSR_SSI_SCR_TE)) {
+			/*
+			 * Be sure the Tx FIFO is filled when TE is set.
+			 * Otherwise, there are some chances to start the
+			 * playback with some void samples inserted first,
+			 * generating a channel slip.
+			 *
+			 * First, SSIEN must be set, to let the FIFO be filled.
+			 *
+			 * Notes:
+			 * - Limit this fix to the DMA case until FIQ cases can
+			 *   be tested.
+			 * - Limit the length of the busy loop to not lock the
+			 *   system too long, even if 1-2 loops are sufficient
+			 *   in general.
+			 */
+			int i;
+			int max_loop = 100;
+			regmap_update_bits(regs, CCSR_SSI_SCR,
+					CCSR_SSI_SCR_SSIEN, CCSR_SSI_SCR_SSIEN);
+			for (i = 0; i < max_loop; i++) {
+				u32 sfcsr;
+				regmap_read(regs, CCSR_SSI_SFCSR, &sfcsr);
+				if (CCSR_SSI_SFCSR_TFCNT0(sfcsr))
+					break;
+			}
+			if (i == max_loop) {
+				dev_err(ssi_private->dev,
+					"Timeout waiting TX FIFO filling\n");
+			}
+		}
 		regmap_update_bits(regs, CCSR_SSI_SCR, vals->scr, vals->scr);
+	}
 }
 
 
@@ -670,6 +726,15 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
 	if (IS_ERR(ssi_private->baudclk))
 		return -EINVAL;
 
+	/*
+	 * Hardware limitation: The bclk rate must be
+	 * never greater than 1/5 IPG clock rate
+	 */
+	if (freq * 5 > clk_get_rate(ssi_private->clk)) {
+		dev_err(cpu_dai->dev, "bitclk > ipgclk/5\n");
+		return -EINVAL;
+	}
+
 	baudclk_is_used = ssi_private->baudclk_streams & ~(BIT(substream->stream));
 
 	/* It should be already enough to divide clock by setting pm alone */
@@ -686,13 +751,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
 		else
 			clkrate = clk_round_rate(ssi_private->baudclk, tmprate);
 
-		/*
-		 * Hardware limitation: The bclk rate must be
-		 * never greater than 1/5 IPG clock rate
-		 */
-		if (clkrate * 5 > clk_get_rate(ssi_private->clk))
-			continue;
-
 		clkrate /= factor;
 		afreq = clkrate / (i + 1);
 
@@ -894,16 +952,16 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev,
 	ssi_private->i2s_mode = CCSR_SSI_SCR_NET;
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
+		regmap_update_bits(regs, CCSR_SSI_STCCR,
+				   CCSR_SSI_SxCCR_DC_MASK,
+				   CCSR_SSI_SxCCR_DC(2));
+		regmap_update_bits(regs, CCSR_SSI_SRCCR,
+				   CCSR_SSI_SxCCR_DC_MASK,
+				   CCSR_SSI_SxCCR_DC(2));
 		switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 		case SND_SOC_DAIFMT_CBM_CFS:
 		case SND_SOC_DAIFMT_CBS_CFS:
 			ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_MASTER;
-			regmap_update_bits(regs, CCSR_SSI_STCCR,
-					CCSR_SSI_SxCCR_DC_MASK,
-					CCSR_SSI_SxCCR_DC(2));
-			regmap_update_bits(regs, CCSR_SSI_SRCCR,
-					CCSR_SSI_SxCCR_DC_MASK,
-					CCSR_SSI_SxCCR_DC(2));
 			break;
 		case SND_SOC_DAIFMT_CBM_CFM:
 			ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_SLAVE;
@@ -1158,14 +1216,14 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = {
 	.playback = {
 		.stream_name = "CPU-Playback",
 		.channels_min = 1,
-		.channels_max = 2,
+		.channels_max = 32,
 		.rates = FSLSSI_I2S_RATES,
 		.formats = FSLSSI_I2S_FORMATS,
 	},
 	.capture = {
 		.stream_name = "CPU-Capture",
 		.channels_min = 1,
-		.channels_max = 2,
+		.channels_max = 32,
 		.rates = FSLSSI_I2S_RATES,
 		.formats = FSLSSI_I2S_FORMATS,
 	},
@@ -1402,6 +1460,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	}
 
 	ssi_private->soc = of_id->data;
+	ssi_private->dev = &pdev->dev;
 
 	sprop = of_get_property(np, "fsl,mode", NULL);
 	if (sprop) {
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index e63cd5ecf..dac668854 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -220,7 +220,7 @@ static int snd_imx_pcm_mmap(struct snd_pcm_substream *substream,
 	ret = dma_mmap_wc(substream->pcm->card->dev, vma, runtime->dma_area,
 			  runtime->dma_addr, runtime->dma_bytes);
 
-	pr_debug("%s: ret: %d %p %pad 0x%08x\n", __func__, ret,
+	pr_debug("%s: ret: %d %p %pad 0x%08zx\n", __func__, ret,
 			runtime->dma_area,
 			&runtime->dma_addr,
 			runtime->dma_bytes);
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 2389ab47e..466492b7d 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -643,6 +643,7 @@ MODULE_DEVICE_TABLE(of, asoc_simple_of_match);
 static struct platform_driver asoc_simple_card = {
 	.driver = {
 		.name = "asoc-simple-card",
+		.pm = &snd_soc_pm_ops,
 		.of_match_table = asoc_simple_of_match,
 	},
 	.probe = asoc_simple_card_probe,
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 1120f4f4d..91c15abb6 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -58,6 +58,21 @@ config SND_SOC_INTEL_HASWELL_MACH
 	  Say Y if you have such a device
 	  If unsure select "N".
 
+config SND_SOC_INTEL_BXT_RT298_MACH
+	tristate "ASoC Audio driver for Broxton with RT298 I2S mode"
+	depends on X86 && ACPI && I2C
+	select SND_SOC_INTEL_SST
+	select SND_SOC_INTEL_SKYLAKE
+	select SND_SOC_RT298
+	select SND_SOC_DMIC
+	select SND_SOC_HDAC_HDMI
+	select SND_HDA_DSP_LOADER
+	help
+	   This adds support for ASoC machine driver for Broxton platforms
+	   with RT286 I2S audio codec.
+	   Say Y if you have such a device
+	   If unsure select "N".
+
 config SND_SOC_INTEL_BYT_RT5640_MACH
 	tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec"
 	depends on X86_INTEL_LPSS && I2C
@@ -162,6 +177,7 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH
 config SND_SOC_INTEL_SKYLAKE
 	tristate
 	select SND_HDA_EXT_CORE
+	select SND_HDA_DSP_LOADER
 	select SND_SOC_TOPOLOGY
 	select SND_SOC_INTEL_SST
 
diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c
index b97e6adcf..98720a93d 100644
--- a/sound/soc/intel/atom/sst-atom-controls.c
+++ b/sound/soc/intel/atom/sst-atom-controls.c
@@ -195,7 +195,7 @@ static int sst_check_and_send_slot_map(struct sst_data *drv, struct snd_kcontrol
 
 	if (e->w && e->w->power)
 		ret = sst_send_slot_map(drv);
-	else
+	else if (!e->w)
 		dev_err(&drv->pdev->dev, "Slot control: %s doesn't have DAPM widget!!!\n",
 				kcontrol->id.name);
 	return ret;
diff --git a/sound/soc/intel/atom/sst-mfld-platform-compress.c b/sound/soc/intel/atom/sst-mfld-platform-compress.c
index 395168986..1bead81bb 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-compress.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-compress.c
@@ -182,24 +182,29 @@ static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd)
 	case SNDRV_PCM_TRIGGER_START:
 		if (stream->compr_ops->stream_start)
 			return stream->compr_ops->stream_start(sst->dev, stream->id);
+		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 		if (stream->compr_ops->stream_drop)
 			return stream->compr_ops->stream_drop(sst->dev, stream->id);
+		break;
 	case SND_COMPR_TRIGGER_DRAIN:
 		if (stream->compr_ops->stream_drain)
 			return stream->compr_ops->stream_drain(sst->dev, stream->id);
+		break;
 	case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
 		if (stream->compr_ops->stream_partial_drain)
 			return stream->compr_ops->stream_partial_drain(sst->dev, stream->id);
+		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		if (stream->compr_ops->stream_pause)
 			return stream->compr_ops->stream_pause(sst->dev, stream->id);
+		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		if (stream->compr_ops->stream_pause_release)
 			return stream->compr_ops->stream_pause_release(sst->dev, stream->id);
-	default:
-		return -EINVAL;
+		break;
 	}
+	return -EINVAL;
 }
 
 static int sst_platform_compr_pointer(struct snd_compr_stream *cstream,
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 3310c0f9c..a8506774f 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -2,6 +2,7 @@ snd-soc-sst-haswell-objs := haswell.o
 snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
 snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
 snd-soc-sst-broadwell-objs := broadwell.o
+snd-soc-sst-bxt-rt298-objs := bxt_rt298.o
 snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o
 snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o
 snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
@@ -14,6 +15,7 @@ snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.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
 obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
+obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o
 obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
 obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o
 obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 3f8a1e10b..7486a0022 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -201,7 +201,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
 	{
 		/* SSP0 - Codec */
 		.name = "Codec",
-		.be_id = 0,
+		.id = 0,
 		.cpu_dai_name = "snd-soc-dummy-dai",
 		.platform_name = "snd-soc-dummy",
 		.no_pcm = 1,
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
new file mode 100644
index 000000000..f4787515c
--- /dev/null
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -0,0 +1,353 @@
+/*
+ * Intel Broxton-P I2S Machine Driver
+ *
+ * Copyright (C) 2014-2016, Intel Corporation. All rights reserved.
+ *
+ * Modified from:
+ *   Intel Skylake I2S Machine driver
+ *
+ * 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/hdac_hdmi.h"
+#include "../../codecs/rt298.h"
+
+static struct snd_soc_jack broxton_headset;
+/* Headset jack detection DAPM pins */
+
+enum {
+	BXT_DPCM_AUDIO_PB = 0,
+	BXT_DPCM_AUDIO_CP,
+	BXT_DPCM_AUDIO_REF_CP,
+	BXT_DPCM_AUDIO_HDMI1_PB,
+	BXT_DPCM_AUDIO_HDMI2_PB,
+	BXT_DPCM_AUDIO_HDMI3_PB,
+};
+
+static struct snd_soc_jack_pin broxton_headset_pins[] = {
+	{
+		.pin = "Mic Jack",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Headphone Jack",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+static const struct snd_kcontrol_new broxton_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 broxton_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),
+	SND_SOC_DAPM_SPK("HDMI1", NULL),
+	SND_SOC_DAPM_SPK("HDMI2", NULL),
+	SND_SOC_DAPM_SPK("HDMI3", NULL),
+};
+
+static const struct snd_soc_dapm_route broxton_rt298_map[] = {
+	/* speaker */
+	{"Speaker", NULL, "SPOR"},
+	{"Speaker", NULL, "SPOL"},
+
+	/* HP jack connectors - unknown if we have jack detect */
+	{"Headphone Jack", NULL, "HPO Pin"},
+
+	/* other jacks */
+	{"MIC1", NULL, "Mic Jack"},
+
+	/* digital mics */
+	{"DMIC1 Pin", NULL, "DMIC2"},
+	{"DMic", NULL, "SoC DMIC"},
+
+	{"HDMI1", NULL, "hif5 Output"},
+	{"HDMI2", NULL, "hif6 Output"},
+	{"HDMI3", NULL, "hif7 Output"},
+
+	/* CODEC BE connections */
+	{ "AIF1 Playback", NULL, "ssp5 Tx"},
+	{ "ssp5 Tx", NULL, "codec0_out"},
+
+	{ "codec0_in", NULL, "ssp5 Rx" },
+	{ "ssp5 Rx", NULL, "AIF1 Capture" },
+
+	{ "dmic01_hifi", NULL, "DMIC01 Rx" },
+	{ "DMIC01 Rx", NULL, "Capture" },
+
+	{ "hifi3", NULL, "iDisp3 Tx"},
+	{ "iDisp3 Tx", NULL, "iDisp3_out"},
+	{ "hifi2", NULL, "iDisp2 Tx"},
+	{ "iDisp2 Tx", NULL, "iDisp2_out"},
+	{ "hifi1", NULL, "iDisp1 Tx"},
+	{ "iDisp1 Tx", NULL, "iDisp1_out"},
+
+};
+
+static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	int ret = 0;
+
+	ret = snd_soc_card_jack_new(rtd->card, "Headset",
+		SND_JACK_HEADSET | SND_JACK_BTN_0,
+		&broxton_headset,
+		broxton_headset_pins, ARRAY_SIZE(broxton_headset_pins));
+
+	if (ret)
+		return ret;
+
+	rt298_mic_detect(codec, &broxton_headset);
+	return 0;
+}
+
+static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *dai = rtd->codec_dai;
+
+	return hdac_hdmi_jack_init(dai, BXT_DPCM_AUDIO_HDMI1_PB + dai->id);
+}
+
+static int broxton_ssp5_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);
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	/* The ADSP will covert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP5 to 24 bit */
+	snd_mask_none(fmt);
+	snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+
+	return 0;
+}
+
+static int broxton_rt298_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, RT298_SCLK_S_PLL,
+					19200000, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+		return ret;
+	}
+
+	return ret;
+}
+
+static struct snd_soc_ops broxton_rt298_ops = {
+	.hw_params = broxton_rt298_hw_params,
+};
+
+/* broxton digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link broxton_rt298_dais[] = {
+	/* Front End DAI links */
+	[BXT_DPCM_AUDIO_PB]
+	{
+		.name = "Bxt Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:0e.0",
+		.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,
+	},
+	[BXT_DPCM_AUDIO_CP]
+	{
+		.name = "Bxt Audio Capture Port",
+		.stream_name = "Audio Record",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:0e.0",
+		.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,
+	},
+	[BXT_DPCM_AUDIO_REF_CP]
+	{
+		.name = "Bxt 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 = "0000:00:0e.0",
+		.init = NULL,
+		.dpcm_capture = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[BXT_DPCM_AUDIO_HDMI1_PB]
+	{
+		.name = "Bxt HDMI Port1",
+		.stream_name = "Hdmi1",
+		.cpu_dai_name = "HDMI1 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:0e.0",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[BXT_DPCM_AUDIO_HDMI2_PB]
+	{
+		.name = "Bxt HDMI Port2",
+		.stream_name = "Hdmi2",
+		.cpu_dai_name = "HDMI2 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:0e.0",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[BXT_DPCM_AUDIO_HDMI3_PB]
+	{
+		.name = "Bxt HDMI Port3",
+		.stream_name = "Hdmi3",
+		.cpu_dai_name = "HDMI3 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:0e.0",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	/* Back End DAI links */
+	{
+		/* SSP5 - Codec */
+		.name = "SSP5-Codec",
+		.id = 0,
+		.cpu_dai_name = "SSP5 Pin",
+		.platform_name = "0000:00:0e.0",
+		.no_pcm = 1,
+		.codec_name = "i2c-INT343A:00",
+		.codec_dai_name = "rt298-aif1",
+		.init = broxton_rt298_codec_init,
+		.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF |
+						SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = broxton_ssp5_fixup,
+		.ops = &broxton_rt298_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+	{
+		.name = "dmic01",
+		.id = 1,
+		.cpu_dai_name = "DMIC01 Pin",
+		.codec_name = "dmic-codec",
+		.codec_dai_name = "dmic-hifi",
+		.platform_name = "0000:00:0e.0",
+		.ignore_suspend = 1,
+		.dpcm_capture = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp1",
+		.id = 3,
+		.cpu_dai_name = "iDisp1 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi1",
+		.platform_name = "0000:00:0e.0",
+		.init = broxton_hdmi_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp2",
+		.id = 4,
+		.cpu_dai_name = "iDisp2 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi2",
+		.platform_name = "0000:00:0e.0",
+		.init = broxton_hdmi_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp3",
+		.id = 5,
+		.cpu_dai_name = "iDisp3 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi3",
+		.platform_name = "0000:00:0e.0",
+		.init = broxton_hdmi_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+};
+
+/* broxton audio machine driver for SPT + RT298S */
+static struct snd_soc_card broxton_rt298 = {
+	.name = "broxton-rt298",
+	.owner = THIS_MODULE,
+	.dai_link = broxton_rt298_dais,
+	.num_links = ARRAY_SIZE(broxton_rt298_dais),
+	.controls = broxton_controls,
+	.num_controls = ARRAY_SIZE(broxton_controls),
+	.dapm_widgets = broxton_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(broxton_widgets),
+	.dapm_routes = broxton_rt298_map,
+	.num_dapm_routes = ARRAY_SIZE(broxton_rt298_map),
+	.fully_routed = true,
+};
+
+static int broxton_audio_probe(struct platform_device *pdev)
+{
+	broxton_rt298.dev = &pdev->dev;
+
+	return devm_snd_soc_register_card(&pdev->dev, &broxton_rt298);
+}
+
+static struct platform_driver broxton_audio = {
+	.probe = broxton_audio_probe,
+	.driver = {
+		.name = "bxt_alc298s_i2s",
+	},
+};
+module_platform_driver(broxton_audio)
+
+/* Module information */
+MODULE_AUTHOR("Ramesh Babu <Ramesh.Babu@intel.com>");
+MODULE_AUTHOR("Senthilnathan Veppur <senthilnathanx.veppur@intel.com>");
+MODULE_DESCRIPTION("Intel SST Audio for Broxton");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:bxt_alc298s_i2s");
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 032a2e753..88efb6243 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -304,7 +304,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = {
 		/* back ends */
 	{
 		.name = "SSP2-Codec",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "ssp2-port",
 		.platform_name = "sst-mfld-platform",
 		.no_pcm = 1,
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 1c95ccc88..35f591eab 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -267,7 +267,7 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = {
 	/* back ends */
 	{
 		.name = "SSP2-Codec",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "ssp2-port",
 		.platform_name = "sst-mfld-platform",
 		.no_pcm = 1,
diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
index e609f0895..cdcced9f3 100644
--- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -255,7 +255,7 @@ static struct snd_soc_dai_link cht_dailink[] = {
 	/* back ends */
 	{
 		.name = "SSP2-Codec",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "ssp2-port",
 		.platform_name = "sst-mfld-platform",
 		.no_pcm = 1,
@@ -296,7 +296,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
 	if (!drv)
 		return -ENOMEM;
 
-	drv->ts3a227e_present = acpi_dev_present("104C227E");
+	drv->ts3a227e_present = acpi_dev_found("104C227E");
 	if (!drv->ts3a227e_present) {
 		/* no need probe TI jack detection chip */
 		snd_soc_card_cht.aux_dev = NULL;
diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c
index 2a6f80843..d7ef292c4 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5645.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -295,7 +295,7 @@ static struct snd_soc_dai_link cht_dailink[] = {
 	/* back ends */
 	{
 		.name = "SSP2-Codec",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "ssp2-port",
 		.platform_name = "sst-mfld-platform",
 		.no_pcm = 1,
@@ -357,7 +357,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	for (i = 0; i < ARRAY_SIZE(snd_soc_cards); i++) {
-		if (acpi_dev_present(snd_soc_cards[i].codec_id)) {
+		if (acpi_dev_found(snd_soc_cards[i].codec_id)) {
 			dev_dbg(&pdev->dev,
 				"found codec %s\n", snd_soc_cards[i].codec_id);
 			card = snd_soc_cards[i].soc_card;
diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c
index 2e5347f8f..df9d254ba 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5672.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -273,7 +273,7 @@ static struct snd_soc_dai_link cht_dailink[] = {
 	{
 		/* SSP2 - Codec */
 		.name = "SSP2-Codec",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "ssp2-port",
 		.platform_name = "sst-mfld-platform",
 		.no_pcm = 1,
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index 22558572c..863f1d5e2 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -156,7 +156,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
 	{
 		/* SSP0 - Codec */
 		.name = "Codec",
-		.be_id = 0,
+		.id = 0,
 		.cpu_dai_name = "snd-soc-dummy-dai",
 		.platform_name = "snd-soc-dummy",
 		.no_pcm = 1,
diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
index 72176b79a..d2808652b 100644
--- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c
+++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
@@ -30,6 +30,16 @@
 static struct snd_soc_jack skylake_headset;
 static struct snd_soc_card skylake_audio_card;
 
+struct skl_hdmi_pcm {
+	struct list_head head;
+	struct snd_soc_dai *codec_dai;
+	int device;
+};
+
+struct skl_nau8825_private {
+	struct list_head hdmi_pcm_list;
+};
+
 enum {
 	SKL_DPCM_AUDIO_PB = 0,
 	SKL_DPCM_AUDIO_CP,
@@ -192,23 +202,56 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd)
 
 static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card);
 	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct skl_hdmi_pcm *pcm;
+
+	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
+
+	pcm->device = SKL_DPCM_AUDIO_HDMI1_PB;
+	pcm->codec_dai = dai;
 
-	return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB);
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+	return 0;
 }
 
 static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card);
 	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct skl_hdmi_pcm *pcm;
+
+	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
+
+	pcm->device = SKL_DPCM_AUDIO_HDMI2_PB;
+	pcm->codec_dai = dai;
 
-	return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI2_PB);
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+	return 0;
 }
 
 static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card);
 	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct skl_hdmi_pcm *pcm;
+
+	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
 
-	return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI3_PB);
+	pcm->device = SKL_DPCM_AUDIO_HDMI3_PB;
+	pcm->codec_dai = dai;
+
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+	return 0;
 }
 
 static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd)
@@ -391,7 +434,6 @@ static struct snd_soc_dai_link skylake_dais[] = {
 		.platform_name = "0000:00:1f.3",
 		.init = NULL,
 		.dpcm_capture = 1,
-		.ignore_suspend = 1,
 		.nonatomic = 1,
 		.dynamic = 1,
 		.ops = &skylaye_refcap_ops,
@@ -456,7 +498,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	{
 		/* SSP0 - Codec */
 		.name = "SSP0-Codec",
-		.be_id = 0,
+		.id = 0,
 		.cpu_dai_name = "SSP0 Pin",
 		.platform_name = "0000:00:1f.3",
 		.no_pcm = 1,
@@ -472,7 +514,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	{
 		/* SSP1 - Codec */
 		.name = "SSP1-Codec",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "SSP1 Pin",
 		.platform_name = "0000:00:1f.3",
 		.no_pcm = 1,
@@ -489,7 +531,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "dmic01",
-		.be_id = 2,
+		.id = 2,
 		.cpu_dai_name = "DMIC01 Pin",
 		.codec_name = "dmic-codec",
 		.codec_dai_name = "dmic-hifi",
@@ -501,7 +543,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "iDisp1",
-		.be_id = 3,
+		.id = 3,
 		.cpu_dai_name = "iDisp1 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi1",
@@ -512,7 +554,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "iDisp2",
-		.be_id = 4,
+		.id = 4,
 		.cpu_dai_name = "iDisp2 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi2",
@@ -523,7 +565,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "iDisp3",
-		.be_id = 5,
+		.id = 5,
 		.cpu_dai_name = "iDisp3 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi3",
@@ -534,6 +576,21 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 };
 
+static int skylake_card_late_probe(struct snd_soc_card *card)
+{
+	struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(card);
+	struct skl_hdmi_pcm *pcm;
+	int err;
+
+	list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+		err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
 /* skylake audio machine driver for SPT + NAU88L25 */
 static struct snd_soc_card skylake_audio_card = {
 	.name = "sklnau8825max",
@@ -547,11 +604,21 @@ static struct snd_soc_card skylake_audio_card = {
 	.dapm_routes = skylake_map,
 	.num_dapm_routes = ARRAY_SIZE(skylake_map),
 	.fully_routed = true,
+	.late_probe = skylake_card_late_probe,
 };
 
 static int skylake_audio_probe(struct platform_device *pdev)
 {
+	struct skl_nau8825_private *ctx;
+
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+	if (!ctx)
+		return -ENOMEM;
+
+	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
 	skylake_audio_card.dev = &pdev->dev;
+	snd_soc_card_set_drvdata(&skylake_audio_card, ctx);
 
 	return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card);
 }
diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
index 5f1ca99ae..e19aa99c4 100644
--- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
+++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
@@ -34,6 +34,15 @@
 static struct snd_soc_jack skylake_headset;
 static struct snd_soc_card skylake_audio_card;
 
+struct skl_hdmi_pcm {
+	struct list_head head;
+	struct snd_soc_dai *codec_dai;
+	int device;
+};
+
+struct skl_nau88125_private {
+	struct list_head hdmi_pcm_list;
+};
 enum {
 	SKL_DPCM_AUDIO_PB = 0,
 	SKL_DPCM_AUDIO_CP,
@@ -222,24 +231,57 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd)
 
 static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card);
 	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct skl_hdmi_pcm *pcm;
+
+	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
+
+	pcm->device = SKL_DPCM_AUDIO_HDMI1_PB;
+	pcm->codec_dai = dai;
+
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
 
-	return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB);
+	return 0;
 }
 
 static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card);
 	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct skl_hdmi_pcm *pcm;
+
+	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
+
+	pcm->device = SKL_DPCM_AUDIO_HDMI2_PB;
+	pcm->codec_dai = dai;
 
-	return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI2_PB);
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+	return 0;
 }
 
 
 static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card);
 	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct skl_hdmi_pcm *pcm;
+
+	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
 
-	return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI3_PB);
+	pcm->device = SKL_DPCM_AUDIO_HDMI3_PB;
+	pcm->codec_dai = dai;
+
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+	return 0;
 }
 
 static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd)
@@ -440,7 +482,6 @@ static struct snd_soc_dai_link skylake_dais[] = {
 		.platform_name = "0000:00:1f.3",
 		.init = NULL,
 		.dpcm_capture = 1,
-		.ignore_suspend = 1,
 		.nonatomic = 1,
 		.dynamic = 1,
 		.ops = &skylaye_refcap_ops,
@@ -505,7 +546,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	{
 		/* SSP0 - Codec */
 		.name = "SSP0-Codec",
-		.be_id = 0,
+		.id = 0,
 		.cpu_dai_name = "SSP0 Pin",
 		.platform_name = "0000:00:1f.3",
 		.no_pcm = 1,
@@ -523,7 +564,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	{
 		/* SSP1 - Codec */
 		.name = "SSP1-Codec",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "SSP1 Pin",
 		.platform_name = "0000:00:1f.3",
 		.no_pcm = 1,
@@ -540,7 +581,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "dmic01",
-		.be_id = 2,
+		.id = 2,
 		.cpu_dai_name = "DMIC01 Pin",
 		.codec_name = "dmic-codec",
 		.codec_dai_name = "dmic-hifi",
@@ -552,7 +593,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "iDisp1",
-		.be_id = 3,
+		.id = 3,
 		.cpu_dai_name = "iDisp1 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi1",
@@ -563,7 +604,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "iDisp2",
-		.be_id = 4,
+		.id = 4,
 		.cpu_dai_name = "iDisp2 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi2",
@@ -574,7 +615,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "iDisp3",
-		.be_id = 5,
+		.id = 5,
 		.cpu_dai_name = "iDisp3 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi3",
@@ -585,6 +626,21 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 };
 
+static int skylake_card_late_probe(struct snd_soc_card *card)
+{
+	struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(card);
+	struct skl_hdmi_pcm *pcm;
+	int err;
+
+	list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+		err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
 /* skylake audio machine driver for SPT + NAU88L25 */
 static struct snd_soc_card skylake_audio_card = {
 	.name = "sklnau8825adi",
@@ -600,11 +656,21 @@ static struct snd_soc_card skylake_audio_card = {
 	.codec_conf = ssm4567_codec_conf,
 	.num_configs = ARRAY_SIZE(ssm4567_codec_conf),
 	.fully_routed = true,
+	.late_probe = skylake_card_late_probe,
 };
 
 static int skylake_audio_probe(struct platform_device *pdev)
 {
+	struct skl_nau88125_private *ctx;
+
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+	if (!ctx)
+		return -ENOMEM;
+
+	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
 	skylake_audio_card.dev = &pdev->dev;
+	snd_soc_card_set_drvdata(&skylake_audio_card, ctx);
 
 	return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card);
 }
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
index 2016397a8..426b48233 100644
--- a/sound/soc/intel/boards/skl_rt286.c
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -30,6 +30,16 @@
 
 static struct snd_soc_jack skylake_headset;
 
+struct skl_hdmi_pcm {
+	struct list_head head;
+	struct snd_soc_dai *codec_dai;
+	int device;
+};
+
+struct skl_rt286_private {
+	struct list_head hdmi_pcm_list;
+};
+
 enum {
 	SKL_DPCM_AUDIO_PB = 0,
 	SKL_DPCM_AUDIO_CP,
@@ -142,9 +152,20 @@ static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd)
 
 static int skylake_hdmi_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card);
 	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct skl_hdmi_pcm *pcm;
+
+	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
 
-	return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB + dai->id);
+	pcm->device = SKL_DPCM_AUDIO_HDMI1_PB + dai->id;
+	pcm->codec_dai = dai;
+
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+	return 0;
 }
 
 static unsigned int rates[] = {
@@ -317,7 +338,6 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
 		.platform_name = "0000:00:1f.3",
 		.init = NULL,
 		.dpcm_capture = 1,
-		.ignore_suspend = 1,
 		.nonatomic = 1,
 		.dynamic = 1,
 	},
@@ -375,7 +395,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
 	{
 		/* SSP0 - Codec */
 		.name = "SSP0-Codec",
-		.be_id = 0,
+		.id = 0,
 		.cpu_dai_name = "SSP0 Pin",
 		.platform_name = "0000:00:1f.3",
 		.no_pcm = 1,
@@ -393,7 +413,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
 	},
 	{
 		.name = "dmic01",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "DMIC01 Pin",
 		.codec_name = "dmic-codec",
 		.codec_dai_name = "dmic-hifi",
@@ -405,7 +425,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
 	},
 	{
 		.name = "iDisp1",
-		.be_id = 2,
+		.id = 2,
 		.cpu_dai_name = "iDisp1 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi1",
@@ -416,7 +436,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
 	},
 	{
 		.name = "iDisp2",
-		.be_id = 3,
+		.id = 3,
 		.cpu_dai_name = "iDisp2 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi2",
@@ -427,7 +447,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
 	},
 	{
 		.name = "iDisp3",
-		.be_id = 4,
+		.id = 4,
 		.cpu_dai_name = "iDisp3 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi3",
@@ -438,6 +458,21 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
 	},
 };
 
+static int skylake_card_late_probe(struct snd_soc_card *card)
+{
+	struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(card);
+	struct skl_hdmi_pcm *pcm;
+	int err;
+
+	list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+		err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
 /* skylake audio machine driver for SPT + RT286S */
 static struct snd_soc_card skylake_rt286 = {
 	.name = "skylake-rt286",
@@ -451,11 +486,21 @@ static struct snd_soc_card skylake_rt286 = {
 	.dapm_routes = skylake_rt286_map,
 	.num_dapm_routes = ARRAY_SIZE(skylake_rt286_map),
 	.fully_routed = true,
+	.late_probe = skylake_card_late_probe,
 };
 
 static int skylake_audio_probe(struct platform_device *pdev)
 {
+	struct skl_rt286_private *ctx;
+
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+	if (!ctx)
+		return -ENOMEM;
+
+	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
 	skylake_rt286.dev = &pdev->dev;
+	snd_soc_card_set_drvdata(&skylake_rt286, ctx);
 
 	return devm_snd_soc_register_card(&pdev->dev, &skylake_rt286);
 }
diff --git a/sound/soc/intel/common/sst-acpi.h b/sound/soc/intel/common/sst-acpi.h
index 4dcfb7e5e..8398cb227 100644
--- a/sound/soc/intel/common/sst-acpi.h
+++ b/sound/soc/intel/common/sst-acpi.h
@@ -12,10 +12,19 @@
  *
  */
 
+#include <linux/kconfig.h>
+#include <linux/stddef.h>
 #include <linux/acpi.h>
 
 /* translation fron HID to I2C name, needed for DAI codec_name */
+#if IS_ENABLED(CONFIG_ACPI)
 const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]);
+#else
+inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN])
+{
+	return NULL;
+}
+#endif
 
 /* acpi match */
 struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines);
diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c
index ef4881e77..259935273 100644
--- a/sound/soc/intel/common/sst-firmware.c
+++ b/sound/soc/intel/common/sst-firmware.c
@@ -203,7 +203,7 @@ static struct dw_dma_chip *dw_probe(struct device *dev, struct resource *mem,
 
 	chip->dev = dev;
 
-	err = dw_dma_probe(chip, NULL);
+	err = dw_dma_probe(chip);
 	if (err)
 		return ERR_PTR(err);
 
diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c
index 1aa819c7e..994256b39 100644
--- a/sound/soc/intel/haswell/sst-haswell-pcm.c
+++ b/sound/soc/intel/haswell/sst-haswell-pcm.c
@@ -445,7 +445,7 @@ static int create_adsp_page_table(struct snd_pcm_substream *substream,
 
 	pages = snd_sgbuf_aligned_pages(size);
 
-	dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n",
+	dev_dbg(rtd->dev, "generating page table for %p size 0x%zx pages %d\n",
 		dma_area, size, pages);
 
 	for (i = 0; i < pages; i++) {
diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile
index 914b6dab9..c28f5d0e1 100644
--- a/sound/soc/intel/skylake/Makefile
+++ b/sound/soc/intel/skylake/Makefile
@@ -5,6 +5,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o
 
 # Skylake IPC Support
 snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \
-		skl-sst.o
+		skl-sst.o bxt-sst.o
 
 obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o
diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c
new file mode 100644
index 000000000..76ba82a9c
--- /dev/null
+++ b/sound/soc/intel/skylake/bxt-sst.c
@@ -0,0 +1,329 @@
+/*
+ *  bxt-sst.c - DSP library functions for BXT platform
+ *
+ *  Copyright (C) 2015-16 Intel Corp
+ *  Author:Rafal Redzimski <rafal.f.redzimski@intel.com>
+ *	   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 published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  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/delay.h>
+#include <linux/firmware.h>
+#include <linux/device.h>
+
+#include "../common/sst-dsp.h"
+#include "../common/sst-dsp-priv.h"
+#include "skl-sst-ipc.h"
+
+#define BXT_BASEFW_TIMEOUT	3000
+#define BXT_INIT_TIMEOUT	500
+#define BXT_IPC_PURGE_FW	0x01004000
+
+#define BXT_ROM_INIT		0x5
+#define BXT_ADSP_SRAM0_BASE	0x80000
+
+/* Firmware status window */
+#define BXT_ADSP_FW_STATUS	BXT_ADSP_SRAM0_BASE
+#define BXT_ADSP_ERROR_CODE     (BXT_ADSP_FW_STATUS + 0x4)
+
+#define BXT_ADSP_SRAM1_BASE	0xA0000
+
+static unsigned int bxt_get_errorcode(struct sst_dsp *ctx)
+{
+	 return sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE);
+}
+
+static int sst_bxt_prepare_fw(struct sst_dsp *ctx,
+			const void *fwdata, u32 fwsize)
+{
+	int stream_tag, ret, i;
+	u32 reg;
+
+	stream_tag = ctx->dsp_ops.prepare(ctx->dev, 0x40, fwsize, &ctx->dmab);
+	if (stream_tag < 0) {
+		dev_err(ctx->dev, "Failed to prepare DMA FW loading err: %x\n",
+				stream_tag);
+		return stream_tag;
+	}
+
+	ctx->dsp_ops.stream_tag = stream_tag;
+	memcpy(ctx->dmab.area, fwdata, fwsize);
+
+	/* Purge FW request */
+	sst_dsp_shim_write(ctx, SKL_ADSP_REG_HIPCI, SKL_ADSP_REG_HIPCI_BUSY |
+					 BXT_IPC_PURGE_FW | (stream_tag - 1));
+
+	ret = skl_dsp_enable_core(ctx);
+	if (ret < 0) {
+		dev_err(ctx->dev, "Boot dsp core failed ret: %d\n", ret);
+		ret = -EIO;
+		goto base_fw_load_failed;
+	}
+
+	for (i = BXT_INIT_TIMEOUT; i > 0; --i) {
+		reg = sst_dsp_shim_read(ctx, SKL_ADSP_REG_HIPCIE);
+
+		if (reg & SKL_ADSP_REG_HIPCIE_DONE) {
+			sst_dsp_shim_update_bits_forced(ctx,
+					SKL_ADSP_REG_HIPCIE,
+					SKL_ADSP_REG_HIPCIE_DONE,
+					SKL_ADSP_REG_HIPCIE_DONE);
+			break;
+		}
+		mdelay(1);
+	}
+	if (!i) {
+		dev_info(ctx->dev, "Waiting for HIPCIE done, reg: 0x%x\n", reg);
+		sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_HIPCIE,
+				SKL_ADSP_REG_HIPCIE_DONE,
+				SKL_ADSP_REG_HIPCIE_DONE);
+	}
+
+	/* enable Interrupt */
+	skl_ipc_int_enable(ctx);
+	skl_ipc_op_int_enable(ctx);
+
+	for (i = BXT_INIT_TIMEOUT; i > 0; --i) {
+		if (SKL_FW_INIT ==
+				(sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS) &
+				SKL_FW_STS_MASK)) {
+
+			dev_info(ctx->dev, "ROM loaded, continue FW loading\n");
+			break;
+		}
+		mdelay(1);
+	}
+	if (!i) {
+		dev_err(ctx->dev, "Timeout for ROM init, HIPCIE: 0x%x\n", reg);
+		ret = -EIO;
+		goto base_fw_load_failed;
+	}
+
+	return ret;
+
+base_fw_load_failed:
+	ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, stream_tag);
+	skl_dsp_disable_core(ctx);
+	return ret;
+}
+
+static int sst_transfer_fw_host_dma(struct sst_dsp *ctx)
+{
+	int ret;
+
+	ctx->dsp_ops.trigger(ctx->dev, true, ctx->dsp_ops.stream_tag);
+	ret = sst_dsp_register_poll(ctx, BXT_ADSP_FW_STATUS, SKL_FW_STS_MASK,
+			BXT_ROM_INIT, BXT_BASEFW_TIMEOUT, "Firmware boot");
+
+	ctx->dsp_ops.trigger(ctx->dev, false, ctx->dsp_ops.stream_tag);
+	ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, ctx->dsp_ops.stream_tag);
+
+	return ret;
+}
+
+static int bxt_load_base_firmware(struct sst_dsp *ctx)
+{
+	const struct firmware *fw = NULL;
+	struct skl_sst *skl = ctx->thread_context;
+	int ret;
+
+	ret = reject_firmware(&fw, ctx->fw_name, ctx->dev);
+	if (ret < 0) {
+		dev_err(ctx->dev, "Request firmware failed %d\n", ret);
+		goto sst_load_base_firmware_failed;
+	}
+
+	ret = sst_bxt_prepare_fw(ctx, fw->data, fw->size);
+	/* Retry Enabling core and ROM load. Retry seemed to help */
+	if (ret < 0) {
+		ret = sst_bxt_prepare_fw(ctx, fw->data, fw->size);
+		if (ret < 0) {
+			dev_err(ctx->dev, "Core En/ROM load fail:%d\n", ret);
+			goto sst_load_base_firmware_failed;
+		}
+	}
+
+	ret = sst_transfer_fw_host_dma(ctx);
+	if (ret < 0) {
+		dev_err(ctx->dev, "Transfer firmware failed %d\n", ret);
+		dev_info(ctx->dev, "Error code=0x%x: FW status=0x%x\n",
+			sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE),
+			sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS));
+
+		skl_dsp_disable_core(ctx);
+	} else {
+		dev_dbg(ctx->dev, "Firmware download successful\n");
+		ret = wait_event_timeout(skl->boot_wait, skl->boot_complete,
+					msecs_to_jiffies(SKL_IPC_BOOT_MSECS));
+		if (ret == 0) {
+			dev_err(ctx->dev, "DSP boot fail, FW Ready timeout\n");
+			skl_dsp_disable_core(ctx);
+			ret = -EIO;
+		} else {
+			skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING);
+			ret = 0;
+		}
+	}
+
+sst_load_base_firmware_failed:
+	release_firmware(fw);
+	return ret;
+}
+
+static int bxt_set_dsp_D0(struct sst_dsp *ctx)
+{
+	struct skl_sst *skl = ctx->thread_context;
+	int ret;
+
+	skl->boot_complete = false;
+
+	ret = skl_dsp_enable_core(ctx);
+	if (ret < 0) {
+		dev_err(ctx->dev, "enable dsp core failed ret: %d\n", ret);
+		return ret;
+	}
+
+	/* enable interrupt */
+	skl_ipc_int_enable(ctx);
+	skl_ipc_op_int_enable(ctx);
+
+	ret = wait_event_timeout(skl->boot_wait, skl->boot_complete,
+					msecs_to_jiffies(SKL_IPC_BOOT_MSECS));
+	if (ret == 0) {
+		dev_err(ctx->dev, "ipc: error DSP boot timeout\n");
+		dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n",
+			sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE),
+			sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS));
+		return -EIO;
+	}
+
+	skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING);
+	return 0;
+}
+
+static int bxt_set_dsp_D3(struct sst_dsp *ctx)
+{
+	struct skl_ipc_dxstate_info dx;
+	struct skl_sst *skl = ctx->thread_context;
+	int ret = 0;
+
+	if (!is_skl_dsp_running(ctx))
+		return ret;
+
+	dx.core_mask = SKL_DSP_CORE0_MASK;
+	dx.dx_mask = SKL_IPC_D3_MASK;
+
+	ret = skl_ipc_set_dx(&skl->ipc, SKL_INSTANCE_ID,
+				SKL_BASE_FW_MODULE_ID, &dx);
+	if (ret < 0) {
+		dev_err(ctx->dev, "Failed to set DSP to D3 state: %d\n", ret);
+		return ret;
+	}
+
+	ret = skl_dsp_disable_core(ctx);
+	if (ret < 0) {
+		dev_err(ctx->dev, "disbale dsp core failed: %d\n", ret);
+		ret = -EIO;
+	}
+
+	skl_dsp_set_state_locked(ctx, SKL_DSP_RESET);
+	return 0;
+}
+
+static struct skl_dsp_fw_ops bxt_fw_ops = {
+	.set_state_D0 = bxt_set_dsp_D0,
+	.set_state_D3 = bxt_set_dsp_D3,
+	.load_fw = bxt_load_base_firmware,
+	.get_fw_errcode = bxt_get_errorcode,
+};
+
+static struct sst_ops skl_ops = {
+	.irq_handler = skl_dsp_sst_interrupt,
+	.write = sst_shim32_write,
+	.read = sst_shim32_read,
+	.ram_read = sst_memcpy_fromio_32,
+	.ram_write = sst_memcpy_toio_32,
+	.free = skl_dsp_free,
+};
+
+static struct sst_dsp_device skl_dev = {
+	.thread = skl_dsp_irq_thread_handler,
+	.ops = &skl_ops,
+};
+
+int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
+			const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
+			struct skl_sst **dsp)
+{
+	struct skl_sst *skl;
+	struct sst_dsp *sst;
+	int ret;
+
+	skl = devm_kzalloc(dev, sizeof(*skl), GFP_KERNEL);
+	if (skl == NULL)
+		return -ENOMEM;
+
+	skl->dev = dev;
+	skl_dev.thread_context = skl;
+
+	skl->dsp = skl_dsp_ctx_init(dev, &skl_dev, irq);
+	if (!skl->dsp) {
+		dev_err(skl->dev, "skl_dsp_ctx_init failed\n");
+		return -ENODEV;
+	}
+
+	sst = skl->dsp;
+	sst->fw_name = fw_name;
+	sst->dsp_ops = dsp_ops;
+	sst->fw_ops = bxt_fw_ops;
+	sst->addr.lpe = mmio_base;
+	sst->addr.shim = mmio_base;
+
+	sst_dsp_mailbox_init(sst, (BXT_ADSP_SRAM0_BASE + SKL_ADSP_W0_STAT_SZ),
+			SKL_ADSP_W0_UP_SZ, BXT_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ);
+
+	INIT_LIST_HEAD(&sst->module_list);
+	ret = skl_ipc_init(dev, skl);
+	if (ret)
+		return ret;
+
+	skl->boot_complete = false;
+	init_waitqueue_head(&skl->boot_wait);
+
+	ret = sst->fw_ops.load_fw(sst);
+	if (ret < 0) {
+		dev_err(dev, "Load base fw failed: %x", ret);
+		return ret;
+	}
+
+	if (dsp)
+		*dsp = skl;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(bxt_sst_dsp_init);
+
+
+void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx)
+{
+	skl_ipc_free(&ctx->ipc);
+	ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp);
+
+	if (ctx->dsp->addr.lpe)
+		iounmap(ctx->dsp->addr.lpe);
+
+	ctx->dsp->ops->free(ctx->dsp);
+}
+EXPORT_SYMBOL_GPL(bxt_sst_dsp_cleanup);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Broxton IPC driver");
diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index 79c5089b8..226db84ba 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -72,6 +72,105 @@ static void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable)
 	skl_ipc_set_large_config(&ctx->ipc, &msg, (u32 *)&mask);
 }
 
+static int skl_dsp_setup_spib(struct device *dev, unsigned int size,
+				int stream_tag, int enable)
+{
+	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_stream *stream = snd_hdac_get_stream(bus,
+			SNDRV_PCM_STREAM_PLAYBACK, stream_tag);
+	struct hdac_ext_stream *estream;
+
+	if (!stream)
+		return -EINVAL;
+
+	estream = stream_to_hdac_ext_stream(stream);
+	/* enable/disable SPIB for this hdac stream */
+	snd_hdac_ext_stream_spbcap_enable(ebus, enable, stream->index);
+
+	/* set the spib value */
+	snd_hdac_ext_stream_set_spib(ebus, estream, size);
+
+	return 0;
+}
+
+static int skl_dsp_prepare(struct device *dev, unsigned int format,
+			unsigned int size, struct snd_dma_buffer *dmab)
+{
+	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_ext_stream *estream;
+	struct hdac_stream *stream;
+	struct snd_pcm_substream substream;
+	int ret;
+
+	if (!bus)
+		return -ENODEV;
+
+	memset(&substream, 0, sizeof(substream));
+	substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+
+	estream = snd_hdac_ext_stream_assign(ebus, &substream,
+					HDAC_EXT_STREAM_TYPE_HOST);
+	if (!estream)
+		return -ENODEV;
+
+	stream = hdac_stream(estream);
+
+	/* assign decouple host dma channel */
+	ret = snd_hdac_dsp_prepare(stream, format, size, dmab);
+	if (ret < 0)
+		return ret;
+
+	skl_dsp_setup_spib(dev, size, stream->stream_tag, true);
+
+	return stream->stream_tag;
+}
+
+static int skl_dsp_trigger(struct device *dev, bool start, int stream_tag)
+{
+	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+	struct hdac_stream *stream;
+	struct hdac_bus *bus = ebus_to_hbus(ebus);
+
+	if (!bus)
+		return -ENODEV;
+
+	stream = snd_hdac_get_stream(bus,
+		SNDRV_PCM_STREAM_PLAYBACK, stream_tag);
+	if (!stream)
+		return -EINVAL;
+
+	snd_hdac_dsp_trigger(stream, start);
+
+	return 0;
+}
+
+static int skl_dsp_cleanup(struct device *dev,
+		struct snd_dma_buffer *dmab, int stream_tag)
+{
+	struct hdac_ext_bus *ebus = dev_get_drvdata(dev);
+	struct hdac_stream *stream;
+	struct hdac_ext_stream *estream;
+	struct hdac_bus *bus = ebus_to_hbus(ebus);
+
+	if (!bus)
+		return -ENODEV;
+
+	stream = snd_hdac_get_stream(bus,
+		SNDRV_PCM_STREAM_PLAYBACK, stream_tag);
+	if (!stream)
+		return -EINVAL;
+
+	estream = stream_to_hdac_ext_stream(stream);
+	skl_dsp_setup_spib(dev, 0, stream_tag, false);
+	snd_hdac_ext_stream_release(estream, HDAC_EXT_STREAM_TYPE_HOST);
+
+	snd_hdac_dsp_cleanup(stream, dmab);
+
+	return 0;
+}
+
 static struct skl_dsp_loader_ops skl_get_loader_ops(void)
 {
 	struct skl_dsp_loader_ops loader_ops;
@@ -84,6 +183,21 @@ static struct skl_dsp_loader_ops skl_get_loader_ops(void)
 	return loader_ops;
 };
 
+static struct skl_dsp_loader_ops bxt_get_loader_ops(void)
+{
+	struct skl_dsp_loader_ops loader_ops;
+
+	memset(&loader_ops, 0, sizeof(loader_ops));
+
+	loader_ops.alloc_dma_buf = skl_alloc_dma_buf;
+	loader_ops.free_dma_buf = skl_free_dma_buf;
+	loader_ops.prepare = skl_dsp_prepare;
+	loader_ops.trigger = skl_dsp_trigger;
+	loader_ops.cleanup = skl_dsp_cleanup;
+
+	return loader_ops;
+};
+
 static const struct skl_dsp_ops dsp_ops[] = {
 	{
 		.id = 0x9d70,
@@ -91,6 +205,12 @@ static const struct skl_dsp_ops dsp_ops[] = {
 		.init = skl_sst_dsp_init,
 		.cleanup = skl_sst_dsp_cleanup
 	},
+	{
+		.id = 0x5a98,
+		.loader_ops = bxt_get_loader_ops,
+		.init = bxt_sst_dsp_init,
+		.cleanup = bxt_sst_dsp_cleanup
+	},
 };
 
 static int skl_get_dsp_ops(int pci_id)
@@ -744,7 +864,7 @@ int skl_init_module(struct skl_sst *ctx,
 		return ret;
 	}
 	mconfig->m_state = SKL_MODULE_INIT_DONE;
-
+	kfree(param_data);
 	return ret;
 }
 
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index ee7849d4b..c54f54f38 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -25,11 +25,12 @@ static u8 OSC_UUID[16] = {0x6E, 0x88, 0x9F, 0xA6, 0xEB, 0x6C, 0x94, 0x45,
 
 #define DSDT_NHLT_PATH "\\_SB.PCI0.HDAS"
 
-void *skl_nhlt_init(struct device *dev)
+struct nhlt_acpi_table *skl_nhlt_init(struct device *dev)
 {
 	acpi_handle handle;
 	union acpi_object *obj;
 	struct nhlt_resource_desc  *nhlt_ptr = NULL;
+	struct nhlt_acpi_table *nhlt_table = NULL;
 
 	if (ACPI_FAILURE(acpi_get_handle(NULL, DSDT_NHLT_PATH, &handle))) {
 		dev_err(dev, "Requested NHLT device not found\n");
@@ -39,18 +40,20 @@ void *skl_nhlt_init(struct device *dev)
 	obj = acpi_evaluate_dsm(handle, OSC_UUID, 1, 1, NULL);
 	if (obj && obj->type == ACPI_TYPE_BUFFER) {
 		nhlt_ptr = (struct nhlt_resource_desc  *)obj->buffer.pointer;
-
-		return memremap(nhlt_ptr->min_addr, nhlt_ptr->length,
+		nhlt_table = (struct nhlt_acpi_table *)
+				memremap(nhlt_ptr->min_addr, nhlt_ptr->length,
 				MEMREMAP_WB);
+		ACPI_FREE(obj);
+		return nhlt_table;
 	}
 
 	dev_err(dev, "device specific method to extract NHLT blob failed\n");
 	return NULL;
 }
 
-void skl_nhlt_free(void *addr)
+void skl_nhlt_free(struct nhlt_acpi_table *nhlt)
 {
-	memunmap(addr);
+	memunmap((void *) nhlt);
 }
 
 static struct nhlt_specific_cfg *skl_get_specific_cfg(
@@ -120,7 +123,7 @@ struct nhlt_specific_cfg
 	struct hdac_bus *bus = ebus_to_hbus(&skl->ebus);
 	struct device *dev = bus->dev;
 	struct nhlt_specific_cfg *sp_config;
-	struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt;
+	struct nhlt_acpi_table *nhlt = skl->nhlt;
 	u16 bps = (s_fmt == 16) ? 16 : 32;
 	u8 j;
 
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index dab0900ee..7c81b3174 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -51,7 +51,7 @@ static struct snd_pcm_hardware azx_pcm_hw = {
 	.rate_min =		8000,
 	.rate_max =		48000,
 	.channels_min =		1,
-	.channels_max =		HDA_QUAD,
+	.channels_max =		8,
 	.buffer_bytes_max =	AZX_MAX_BUF_SIZE,
 	.period_bytes_min =	128,
 	.period_bytes_max =	AZX_MAX_BUF_SIZE / 2,
@@ -213,7 +213,7 @@ static int skl_be_prepare(struct snd_pcm_substream *substream,
 	struct skl_sst *ctx = skl->skl_sst;
 	struct skl_module_cfg *mconfig;
 
-	if ((dai->playback_active > 1) || (dai->capture_active > 1))
+	if (dai->playback_widget->power || dai->capture_widget->power)
 		return 0;
 
 	mconfig = skl_tplg_be_get_cpr_module(dai, substream->stream);
@@ -402,23 +402,33 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 	struct skl_module_cfg *mconfig;
 	struct hdac_ext_bus *ebus = get_bus_ctx(substream);
 	struct hdac_ext_stream *stream = get_hdac_ext_stream(substream);
+	struct snd_soc_dapm_widget *w;
 	int ret;
 
 	mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream);
 	if (!mconfig)
 		return -EIO;
 
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		w = dai->playback_widget;
+	else
+		w = dai->capture_widget;
+
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_RESUME:
-		skl_pcm_prepare(substream, dai);
-		/*
-		 * enable DMA Resume enable bit for the stream, set the dpib
-		 * & lpib position to resune before starting the DMA
-		 */
-		snd_hdac_ext_stream_drsm_enable(ebus, true,
-					hdac_stream(stream)->index);
-		snd_hdac_ext_stream_set_dpibr(ebus, stream, stream->dpib);
-		snd_hdac_ext_stream_set_lpib(stream, stream->lpib);
+		if (!w->ignore_suspend) {
+			skl_pcm_prepare(substream, dai);
+			/*
+			 * enable DMA Resume enable bit for the stream, set the
+			 * dpib & lpib position to resume before starting the
+			 * DMA
+			 */
+			snd_hdac_ext_stream_drsm_enable(ebus, true,
+						hdac_stream(stream)->index);
+			snd_hdac_ext_stream_set_dpibr(ebus, stream,
+							stream->dpib);
+			snd_hdac_ext_stream_set_lpib(stream, stream->lpib);
+		}
 
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
@@ -448,7 +458,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 			return ret;
 
 		ret = skl_decoupled_trigger(substream, cmd);
-		if (cmd == SNDRV_PCM_TRIGGER_SUSPEND) {
+		if ((cmd == SNDRV_PCM_TRIGGER_SUSPEND) && !w->ignore_suspend) {
 			/* save the dpib and lpib positions */
 			stream->dpib = readl(ebus->bus.remap_addr +
 					AZX_REG_VS_SDXDPIB_XBASE +
@@ -523,7 +533,6 @@ static int skl_link_pcm_prepare(struct snd_pcm_substream *substream,
 	if (!link)
 		return -EINVAL;
 
-	snd_hdac_ext_bus_link_power_up(link);
 	snd_hdac_ext_link_stream_reset(link_dev);
 
 	snd_hdac_ext_link_stream_setup(link_dev, format_val);
@@ -682,7 +691,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
 	.playback = {
 		.stream_name = "HDMI1 Playback",
 		.channels_min = HDA_STEREO,
-		.channels_max = HDA_STEREO,
+		.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 |
@@ -697,7 +706,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
 	.playback = {
 		.stream_name = "HDMI2 Playback",
 		.channels_min = HDA_STEREO,
-		.channels_max = HDA_STEREO,
+		.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 |
@@ -712,7 +721,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
 	.playback = {
 		.stream_name = "HDMI3 Playback",
 		.channels_min = HDA_STEREO,
-		.channels_max = HDA_STEREO,
+		.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 |
@@ -760,12 +769,84 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
 	},
 },
 {
+	.name = "SSP2 Pin",
+	.ops = &skl_be_ssp_dai_ops,
+	.playback = {
+		.stream_name = "ssp2 Tx",
+		.channels_min = HDA_STEREO,
+		.channels_max = HDA_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "ssp2 Rx",
+		.channels_min = HDA_STEREO,
+		.channels_max = HDA_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+{
+	.name = "SSP3 Pin",
+	.ops = &skl_be_ssp_dai_ops,
+	.playback = {
+		.stream_name = "ssp3 Tx",
+		.channels_min = HDA_STEREO,
+		.channels_max = HDA_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "ssp3 Rx",
+		.channels_min = HDA_STEREO,
+		.channels_max = HDA_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+{
+	.name = "SSP4 Pin",
+	.ops = &skl_be_ssp_dai_ops,
+	.playback = {
+		.stream_name = "ssp4 Tx",
+		.channels_min = HDA_STEREO,
+		.channels_max = HDA_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "ssp4 Rx",
+		.channels_min = HDA_STEREO,
+		.channels_max = HDA_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+{
+	.name = "SSP5 Pin",
+	.ops = &skl_be_ssp_dai_ops,
+	.playback = {
+		.stream_name = "ssp5 Tx",
+		.channels_min = HDA_STEREO,
+		.channels_max = HDA_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "ssp5 Rx",
+		.channels_min = HDA_STEREO,
+		.channels_max = HDA_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+{
 	.name = "iDisp1 Pin",
 	.ops = &skl_link_dai_ops,
 	.playback = {
 		.stream_name = "iDisp1 Tx",
 		.channels_min = HDA_STEREO,
-		.channels_max = HDA_STEREO,
+		.channels_max = 8,
 		.rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE |
 			SNDRV_PCM_FMTBIT_S24_LE,
@@ -777,7 +858,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
 	.playback = {
 		.stream_name = "iDisp2 Tx",
 		.channels_min = HDA_STEREO,
-		.channels_max = HDA_STEREO,
+		.channels_max = 8,
 		.rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|
 			SNDRV_PCM_RATE_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE |
@@ -790,7 +871,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
 	.playback = {
 		.stream_name = "iDisp3 Tx",
 		.channels_min = HDA_STEREO,
-		.channels_max = HDA_STEREO,
+		.channels_max = 8,
 		.rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|
 			SNDRV_PCM_RATE_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE |
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c
index 2962ef22f..13c19855e 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.c
+++ b/sound/soc/intel/skylake/skl-sst-dsp.c
@@ -336,8 +336,6 @@ void skl_dsp_free(struct sst_dsp *dsp)
 	skl_ipc_int_disable(dsp);
 
 	free_irq(dsp->irq, dsp);
-	dsp->cl_dev.ops.cl_cleanup_controller(dsp);
-	skl_cldma_int_disable(dsp);
 	skl_ipc_op_int_disable(dsp);
 	skl_ipc_int_disable(dsp);
 
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h
index b6e310d49..deabe7308 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.h
+++ b/sound/soc/intel/skylake/skl-sst-dsp.h
@@ -118,16 +118,25 @@ struct skl_dsp_fw_ops {
 	int (*set_state_D0)(struct sst_dsp *ctx);
 	int (*set_state_D3)(struct sst_dsp *ctx);
 	unsigned int (*get_fw_errcode)(struct sst_dsp *ctx);
-	int (*load_mod)(struct sst_dsp *ctx, u16 mod_id, char *mod_name);
+	int (*load_mod)(struct sst_dsp *ctx, u16 mod_id, u8 *mod_name);
 	int (*unload_mod)(struct sst_dsp *ctx, u16 mod_id);
 
 };
 
 struct skl_dsp_loader_ops {
+	int stream_tag;
+
 	int (*alloc_dma_buf)(struct device *dev,
 		struct snd_dma_buffer *dmab, size_t size);
 	int (*free_dma_buf)(struct device *dev,
 		struct snd_dma_buffer *dmab);
+	int (*prepare)(struct device *dev, unsigned int format,
+				unsigned int byte_size,
+				struct snd_dma_buffer *bufp);
+	int (*trigger)(struct device *dev, bool start, int stream_tag);
+
+	int (*cleanup)(struct device *dev, struct snd_dma_buffer *dmab,
+				 int stream_tag);
 };
 
 struct skl_load_module_info {
@@ -160,6 +169,10 @@ int skl_dsp_boot(struct sst_dsp *ctx);
 int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
 		const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
 		struct skl_sst **dsp);
+int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq,
+		const char *fw_name, struct skl_dsp_loader_ops dsp_ops,
+		struct skl_sst **dsp);
 void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx);
+void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx);
 
 #endif /*__SKL_SST_DSP_H__*/
diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c
index add3d7e00..463e72a35 100644
--- a/sound/soc/intel/skylake/skl-sst.c
+++ b/sound/soc/intel/skylake/skl-sst.c
@@ -20,6 +20,7 @@
 #include <linux/delay.h>
 #include <linux/device.h>
 #include <linux/err.h>
+#include <linux/uuid.h>
 #include "../common/sst-dsp.h"
 #include "../common/sst-dsp-priv.h"
 #include "../common/sst-ipc.h"
@@ -304,12 +305,14 @@ static int skl_transfer_module(struct sst_dsp *ctx,
 	return ret;
 }
 
-static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, char *guid)
+static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid)
 {
 	struct skl_module_table *module_entry = NULL;
 	int ret = 0;
 	char mod_name[64]; /* guid str = 32 chars + 4 hyphens */
+	uuid_le *uuid_mod;
 
+	uuid_mod = (uuid_le *)guid;
 	snprintf(mod_name, sizeof(mod_name), "/*(DEBLOBBED)*/");
 
 	module_entry = skl_module_get_from_id(ctx, mod_id);
@@ -450,6 +453,10 @@ void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx)
 	skl_clear_module_table(ctx->dsp);
 	skl_ipc_free(&ctx->ipc);
 	ctx->dsp->ops->free(ctx->dsp);
+	if (ctx->boot_complete) {
+		ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp);
+		skl_cldma_int_disable(ctx->dsp);
+	}
 }
 EXPORT_SYMBOL_GPL(skl_sst_dsp_cleanup);
 
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 525d00dc1..bebbb3f2f 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -154,13 +154,32 @@ static void skl_dump_mconfig(struct skl_sst *ctx,
 	dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->out_fmt[0].ch_cfg);
 }
 
+static void skl_tplg_update_chmap(struct skl_module_fmt *fmt, int chs)
+{
+	int slot_map = 0xFFFFFFFF;
+	int start_slot = 0;
+	int i;
+
+	for (i = 0; i < chs; i++) {
+		/*
+		 * For 2 channels with starting slot as 0, slot map will
+		 * look like 0xFFFFFF10.
+		 */
+		slot_map &= (~(0xF << (4 * i)) | (start_slot << (4 * i)));
+		start_slot++;
+	}
+	fmt->ch_map = slot_map;
+}
+
 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)
+	if (fixup & SKL_CH_FIXUP_MASK) {
 		fmt->channels = params->ch;
+		skl_tplg_update_chmap(fmt, fmt->channels);
+	}
 	if (fixup & SKL_FMT_FIXUP_MASK) {
 		fmt->valid_bit_depth = skl_get_bit_depth(params->s_fmt);
 
@@ -1564,6 +1583,8 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt,
 		return -ENOMEM;
 
 	w->priv = mconfig;
+	memcpy(&mconfig->guid, &dfw_config->uuid, 16);
+
 	mconfig->id.module_id = dfw_config->module_id;
 	mconfig->id.instance_id = dfw_config->instance_id;
 	mconfig->mcps = dfw_config->max_mcps;
@@ -1593,10 +1614,6 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt,
 	mconfig->time_slot = dfw_config->time_slot;
 	mconfig->formats_config.caps_size = dfw_config->caps.caps_size;
 
-	if (dfw_config->is_loadable)
-		memcpy(mconfig->guid, dfw_config->uuid,
-					ARRAY_SIZE(dfw_config->uuid));
-
 	mconfig->m_in_pin = devm_kzalloc(bus->dev, (mconfig->max_in_queue) *
 						sizeof(*mconfig->m_in_pin),
 						GFP_KERNEL);
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index d2d923002..e4b399cd7 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -281,7 +281,7 @@ enum skl_module_state {
 };
 
 struct skl_module_cfg {
-	char guid[SKL_UUID_STR_SZ];
+	u8 guid[16];
 	struct skl_module_inst_id id;
 	u8 domain;
 	bool homogenous_inputs;
diff --git a/sound/soc/intel/skylake/skl-tplg-interface.h b/sound/soc/intel/skylake/skl-tplg-interface.h
index 1db88a63a..a32e5e9cc 100644
--- a/sound/soc/intel/skylake/skl-tplg-interface.h
+++ b/sound/soc/intel/skylake/skl-tplg-interface.h
@@ -181,7 +181,7 @@ struct skl_dfw_pipe {
 } __packed;
 
 struct skl_dfw_module {
-	char uuid[SKL_UUID_STR_SZ];
+	u8 uuid[16];
 
 	u16 module_id;
 	u16 instance_id;
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 92213d6d6..95bf61ed7 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -229,7 +229,12 @@ static int skl_suspend(struct device *dev)
 	 * running, we need to save the state for these and continue
 	 */
 	if (skl->supend_active) {
+		/* turn off the links and stop the CORB/RIRB DMA if it is On */
 		snd_hdac_ext_bus_link_power_down_all(ebus);
+
+		if (ebus->cmd_dma_state)
+			snd_hdac_bus_stop_cmd_io(&ebus->bus);
+
 		enable_irq_wake(bus->irq);
 		pci_save_state(pci);
 		pci_disable_device(pci);
@@ -255,6 +260,7 @@ static int skl_resume(struct device *dev)
 	struct hdac_ext_bus *ebus = pci_get_drvdata(pci);
 	struct skl *skl  = ebus_to_skl(ebus);
 	struct hdac_bus *bus = ebus_to_hbus(ebus);
+	struct hdac_ext_link *hlink = NULL;
 	int ret;
 
 	/* Turned OFF in HDMI codec driver after codec reconfiguration */
@@ -276,8 +282,29 @@ static int skl_resume(struct device *dev)
 		ret = pci_enable_device(pci);
 		snd_hdac_ext_bus_link_power_up_all(ebus);
 		disable_irq_wake(bus->irq);
+		/*
+		 * turn On the links which are On before active suspend
+		 * and start the CORB/RIRB DMA if On before
+		 * active suspend.
+		 */
+		list_for_each_entry(hlink, &ebus->hlink_list, list) {
+			if (hlink->ref_count)
+				snd_hdac_ext_bus_link_power_up(hlink);
+		}
+
+		if (ebus->cmd_dma_state)
+			snd_hdac_bus_init_cmd_io(&ebus->bus);
 	} else {
 		ret = _skl_resume(ebus);
+
+		/* turn off the links which are off before suspend */
+		list_for_each_entry(hlink, &ebus->hlink_list, list) {
+			if (!hlink->ref_count)
+				snd_hdac_ext_bus_link_power_down(hlink);
+		}
+
+		if (!ebus->cmd_dma_state)
+			snd_hdac_bus_stop_cmd_io(&ebus->bus);
 	}
 
 	return ret;
@@ -613,6 +640,7 @@ static int skl_probe(struct pci_dev *pci,
 	struct skl *skl;
 	struct hdac_ext_bus *ebus = NULL;
 	struct hdac_bus *bus = NULL;
+	struct hdac_ext_link *hlink = NULL;
 	int err;
 
 	/* we use ext core ops, so provide NULL for ops here */
@@ -643,7 +671,7 @@ static int skl_probe(struct pci_dev *pci,
 		err = skl_machine_device_register(skl,
 				  (void *)pci_id->driver_data);
 		if (err < 0)
-			goto out_free;
+			goto out_nhlt_free;
 
 		err = skl_init_dsp(skl);
 		if (err < 0) {
@@ -679,6 +707,12 @@ static int skl_probe(struct pci_dev *pci,
 		}
 	}
 
+	/*
+	 * we are done probling so decrement link counts
+	 */
+	list_for_each_entry(hlink, &ebus->hlink_list, list)
+		snd_hdac_ext_bus_link_put(ebus, hlink);
+
 	/*configure PM */
 	pm_runtime_put_noidle(bus->dev);
 	pm_runtime_allow(bus->dev);
@@ -693,6 +727,8 @@ out_dsp_free:
 	skl_free_dsp(skl);
 out_mach_free:
 	skl_machine_device_unregister(skl);
+out_nhlt_free:
+	skl_nhlt_free(skl->nhlt);
 out_free:
 	skl->init_failed = 1;
 	skl_free(ebus);
@@ -743,6 +779,7 @@ static void skl_remove(struct pci_dev *pci)
 	skl_free_dsp(skl);
 	skl_machine_device_unregister(skl);
 	skl_dmic_device_unregister(skl);
+	skl_nhlt_free(skl->nhlt);
 	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 39e16fa7a..4b4b3876a 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -66,7 +66,7 @@ struct skl {
 	struct platform_device *dmic_dev;
 	struct platform_device *i2s_dev;
 
-	void *nhlt; /* nhlt ptr */
+	struct nhlt_acpi_table *nhlt; /* nhlt ptr */
 	struct skl_sst *skl_sst; /* sst skl ctx */
 
 	struct skl_dsp_resource resource;
@@ -103,8 +103,8 @@ struct skl_dsp_ops {
 int skl_platform_unregister(struct device *dev);
 int skl_platform_register(struct device *dev);
 
-void *skl_nhlt_init(struct device *dev);
-void skl_nhlt_free(void *addr);
+struct nhlt_acpi_table *skl_nhlt_init(struct device *dev);
+void skl_nhlt_free(struct nhlt_acpi_table *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/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 132bb83f8..bc3c7b5ac 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -1,6 +1,7 @@
 config SND_KIRKWOOD_SOC
 	tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
 	depends on ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST
+	depends on HAS_DMA
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Kirkwood I2S interface. You will also need to select the
diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig
index f7e789e97..3abf51c07 100644
--- a/sound/soc/mediatek/Kconfig
+++ b/sound/soc/mediatek/Kconfig
@@ -43,6 +43,7 @@ config SND_SOC_MT8173_RT5650_RT5676
 	depends on SND_SOC_MEDIATEK && I2C
 	select SND_SOC_RT5645
 	select SND_SOC_RT5677
+	select SND_SOC_HDMI_CODEC
 	help
 	  This adds ASoC driver for Mediatek MT8173 boards
 	  with the RT5650 and RT5676 codecs.
diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
index 5c4c58c69..bb593926c 100644
--- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c
@@ -134,7 +134,9 @@ static struct snd_soc_dai_link_component mt8173_rt5650_rt5676_codecs[] = {
 enum {
 	DAI_LINK_PLAYBACK,
 	DAI_LINK_CAPTURE,
+	DAI_LINK_HDMI,
 	DAI_LINK_CODEC_I2S,
+	DAI_LINK_HDMI_I2S,
 	DAI_LINK_INTERCODEC
 };
 
@@ -161,6 +163,16 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = {
 		.dynamic = 1,
 		.dpcm_capture = 1,
 	},
+	[DAI_LINK_HDMI] = {
+		.name = "HDMI",
+		.stream_name = "HDMI PCM",
+		.cpu_dai_name = "HDMI",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dynamic = 1,
+		.dpcm_playback = 1,
+	},
 
 	/* Back End DAI links */
 	[DAI_LINK_CODEC_I2S] = {
@@ -177,6 +189,13 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = {
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
 	},
+	[DAI_LINK_HDMI_I2S] = {
+		.name = "HDMI BE",
+		.cpu_dai_name = "HDMIO",
+		.no_pcm = 1,
+		.codec_dai_name = "i2s-hifi",
+		.dpcm_playback = 1,
+	},
 	/* rt5676 <-> rt5650 intercodec link: Sets rt5676 I2S2 as master */
 	[DAI_LINK_INTERCODEC] = {
 		.name = "rt5650_rt5676 intercodec",
@@ -251,6 +270,14 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
 	mt8173_rt5650_rt5676_dais[DAI_LINK_INTERCODEC].codec_of_node =
 		mt8173_rt5650_rt5676_codecs[1].of_node;
 
+	mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codec_of_node =
+		of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 2);
+	if (!mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codec_of_node) {
+		dev_err(&pdev->dev,
+			"Property 'audio-codec' missing or invalid\n");
+		return -EINVAL;
+	}
+
 	card->dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
 
diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c
index bb09bb1b7..a27a6673d 100644
--- a/sound/soc/mediatek/mt8173-rt5650.c
+++ b/sound/soc/mediatek/mt8173-rt5650.c
@@ -85,12 +85,29 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime)
 {
 	struct snd_soc_card *card = runtime->card;
 	struct snd_soc_codec *codec = runtime->codec_dais[0]->codec;
+	const char *codec_capture_dai = runtime->codec_dais[1]->name;
 	int ret;
 
 	rt5645_sel_asrc_clk_src(codec,
-				RT5645_DA_STEREO_FILTER |
-				RT5645_AD_STEREO_FILTER,
+				RT5645_DA_STEREO_FILTER,
 				RT5645_CLK_SEL_I2S1_ASRC);
+
+	if (!strcmp(codec_capture_dai, "rt5645-aif1")) {
+		rt5645_sel_asrc_clk_src(codec,
+					RT5645_AD_STEREO_FILTER,
+					RT5645_CLK_SEL_I2S1_ASRC);
+	} else if (!strcmp(codec_capture_dai, "rt5645-aif2")) {
+		rt5645_sel_asrc_clk_src(codec,
+					RT5645_AD_STEREO_FILTER,
+					RT5645_CLK_SEL_I2S2_ASRC);
+	} else {
+		dev_warn(card->dev,
+			 "Only one dai codec found in DTS, enabled rt5645 AD filter\n");
+		rt5645_sel_asrc_clk_src(codec,
+					RT5645_AD_STEREO_FILTER,
+					RT5645_CLK_SEL_I2S1_ASRC);
+	}
+
 	/* enable jack detection */
 	ret = snd_soc_card_jack_new(card, "Headset Jack",
 				    SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
@@ -110,6 +127,11 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime)
 
 static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = {
 	{
+		/* Playback */
+		.dai_name = "rt5645-aif1",
+	},
+	{
+		/* Capture */
 		.dai_name = "rt5645-aif1",
 	},
 };
@@ -149,7 +171,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = {
 		.cpu_dai_name = "I2S",
 		.no_pcm = 1,
 		.codecs = mt8173_rt5650_codecs,
-		.num_codecs = 1,
+		.num_codecs = 2,
 		.init = mt8173_rt5650_init,
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			   SND_SOC_DAIFMT_CBS_CFS,
@@ -177,6 +199,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &mt8173_rt5650_card;
 	struct device_node *platform_node;
+	struct device_node *np;
+	const char *codec_capture_dai;
 	int i, ret;
 
 	platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -199,6 +223,26 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
 			"Property 'audio-codec' missing or invalid\n");
 		return -EINVAL;
 	}
+	mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node;
+
+	if (of_find_node_by_name(platform_node, "codec-capture")) {
+		np = of_get_child_by_name(pdev->dev.of_node, "codec-capture");
+		if (!np) {
+			dev_err(&pdev->dev,
+				"%s: Can't find codec-capture DT node\n",
+				__func__);
+			return -EINVAL;
+		}
+		ret = snd_soc_of_get_dai_name(np, &codec_capture_dai);
+		if (ret < 0) {
+			dev_err(&pdev->dev,
+				"%s codec_capture_dai name fail %d\n",
+				__func__, ret);
+			return ret;
+		}
+		mt8173_rt5650_codecs[1].dai_name = codec_capture_dai;
+	}
+
 	card->dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
 
diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c
index f1c58a2c1..2b5df2ef5 100644
--- a/sound/soc/mediatek/mtk-afe-pcm.c
+++ b/sound/soc/mediatek/mtk-afe-pcm.c
@@ -123,6 +123,7 @@
 #define AFE_TDM_CON1_WLEN_32BIT		(0x2 << 8)
 #define AFE_TDM_CON1_MSB_ALIGNED	(0x1 << 4)
 #define AFE_TDM_CON1_1_BCK_DELAY	(0x1 << 3)
+#define AFE_TDM_CON1_LRCK_INV		(0x1 << 2)
 #define AFE_TDM_CON1_BCK_INV		(0x1 << 1)
 #define AFE_TDM_CON1_EN			(0x1 << 0)
 
@@ -449,6 +450,7 @@ static int mtk_afe_hdmi_prepare(struct snd_pcm_substream *substream,
 			      runtime->rate * runtime->channels * 32);
 
 	val = AFE_TDM_CON1_BCK_INV |
+	      AFE_TDM_CON1_LRCK_INV |
 	      AFE_TDM_CON1_1_BCK_DELAY |
 	      AFE_TDM_CON1_MSB_ALIGNED | /* I2S mode */
 	      AFE_TDM_CON1_WLEN_32BIT |
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index c7563e230..4a16e7789 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -260,6 +260,10 @@ static void omap_st_on(struct omap_mcbsp *mcbsp)
 	if (mcbsp->pdata->enable_st_clock)
 		mcbsp->pdata->enable_st_clock(mcbsp->id, 1);
 
+	/* Disable Sidetone clock auto-gating for normal operation */
+	w = MCBSP_ST_READ(mcbsp, SYSCONFIG);
+	MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w & ~(ST_AUTOIDLE));
+
 	/* Enable McBSP Sidetone */
 	w = MCBSP_READ(mcbsp, SSELCR);
 	MCBSP_WRITE(mcbsp, SSELCR, w | SIDETONEEN);
@@ -279,6 +283,10 @@ static void omap_st_off(struct omap_mcbsp *mcbsp)
 	w = MCBSP_READ(mcbsp, SSELCR);
 	MCBSP_WRITE(mcbsp, SSELCR, w & ~(SIDETONEEN));
 
+	/* Enable Sidetone clock auto-gating to reduce power consumption */
+	w = MCBSP_ST_READ(mcbsp, SYSCONFIG);
+	MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w | ST_AUTOIDLE);
+
 	if (mcbsp->pdata->enable_st_clock)
 		mcbsp->pdata->enable_st_clock(mcbsp->id, 0);
 }
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 99381a272..a84f67723 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -82,6 +82,8 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct dma_chan *chan;
 	int err = 0;
 
+	memset(&config, 0x00, sizeof(config));
+
 	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 
 	/* return if this is a bufferless transfer e.g.
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
index ec522e94b..b6cb9950f 100644
--- a/sound/soc/pxa/brownstone.c
+++ b/sound/soc/pxa/brownstone.c
@@ -133,3 +133,4 @@ module_platform_driver(mmp_driver);
 MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
 MODULE_DESCRIPTION("ALSA SoC Brownstone");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:brownstone-audio");
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 5c8f9db50..d1661fa6e 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -207,3 +207,4 @@ module_platform_driver(mioa701_wm9713_driver);
 MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)");
 MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mioa701-wm9713");
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
index 51e790d00..96df9b2d8 100644
--- a/sound/soc/pxa/mmp-pcm.c
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -248,3 +248,4 @@ module_platform_driver(mmp_pcm_driver);
 MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
 MODULE_DESCRIPTION("MMP Soc Audio DMA module");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mmp-pcm-audio");
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
index eca60c297..ca8b23f8c 100644
--- a/sound/soc/pxa/mmp-sspa.c
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -482,3 +482,4 @@ module_platform_driver(asoc_mmp_sspa_driver);
 MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
 MODULE_DESCRIPTION("MMP SSPA SoC Interface");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mmp-sspa-dai");
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 4e74d9573..bcc81e920 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -161,3 +161,4 @@ module_platform_driver(palm27x_wm9712_driver);
 MODULE_AUTHOR("Marek Vasut <marek.vasut@gmail.com>");
 MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:palm27x-asoc");
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index da03fad1b..3cad990da 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -833,3 +833,4 @@ module_platform_driver(asoc_ssp_driver);
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
 MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:pxa-ssp-dai");
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index f3de615aa..9615e6de1 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -287,3 +287,4 @@ module_platform_driver(pxa2xx_ac97_driver);
 MODULE_AUTHOR("Nicolas Pitre");
 MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:pxa2xx-ac97");
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 9f390398d..410d48b93 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -117,3 +117,4 @@ module_platform_driver(pxa_pcm_driver);
 MODULE_AUTHOR("Nicolas Pitre");
 MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:pxa-pcm-audio");
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index 6e8665430..db000c698 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -474,7 +474,7 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime)
 	struct lpass_data *drvdata =
 		snd_soc_platform_get_drvdata(soc_runtime->platform);
 	struct lpass_variant *v = drvdata->variant;
-	int ret;
+	int ret = -EINVAL;
 	struct lpass_pcm_data *data;
 	size_t size = lpass_platform_pcm_hardware.buffer_bytes_max;
 
@@ -491,7 +491,7 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime)
 			data->rdma_ch = v->alloc_dma_channel(drvdata,
 						SNDRV_PCM_STREAM_PLAYBACK);
 
-		if (IS_ERR_VALUE(data->rdma_ch))
+		if (data->rdma_ch < 0)
 			return data->rdma_ch;
 
 		drvdata->substream[data->rdma_ch] = psubstream;
@@ -518,8 +518,10 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime)
 			data->wrdma_ch = v->alloc_dma_channel(drvdata,
 						SNDRV_PCM_STREAM_CAPTURE);
 
-		if (IS_ERR_VALUE(data->wrdma_ch))
+		if (data->wrdma_ch < 0) {
+			ret = data->wrdma_ch;
 			goto capture_alloc_err;
+		}
 
 		drvdata->substream[data->wrdma_ch] = csubstream;
 
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 2f8e20416..574c6af28 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -34,6 +34,13 @@ struct rk_i2s_dev {
 
 	struct regmap *regmap;
 
+/*
+ * Used to indicate the tx/rx status.
+ * I2S controller hopes to start the tx and rx together,
+ * also to stop them when they are both try to stop.
+*/
+	bool tx_start;
+	bool rx_start;
 	bool is_master_mode;
 };
 
@@ -75,29 +82,37 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on)
 				   I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE);
 
 		regmap_update_bits(i2s->regmap, I2S_XFER,
-				   I2S_XFER_TXS_START,
-				   I2S_XFER_TXS_START);
+				   I2S_XFER_TXS_START | I2S_XFER_RXS_START,
+				   I2S_XFER_TXS_START | I2S_XFER_RXS_START);
+
+		i2s->tx_start = true;
 	} else {
+		i2s->tx_start = false;
+
 		regmap_update_bits(i2s->regmap, I2S_DMACR,
 				   I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE);
 
-		regmap_update_bits(i2s->regmap, I2S_XFER,
-				   I2S_XFER_TXS_START,
-				   I2S_XFER_TXS_STOP);
+		if (!i2s->rx_start) {
+			regmap_update_bits(i2s->regmap, I2S_XFER,
+					   I2S_XFER_TXS_START |
+					   I2S_XFER_RXS_START,
+					   I2S_XFER_TXS_STOP |
+					   I2S_XFER_RXS_STOP);
 
-		regmap_update_bits(i2s->regmap, I2S_CLR,
-				   I2S_CLR_TXC,
-				   I2S_CLR_TXC);
+			regmap_update_bits(i2s->regmap, I2S_CLR,
+					   I2S_CLR_TXC | I2S_CLR_RXC,
+					   I2S_CLR_TXC | I2S_CLR_RXC);
 
-		regmap_read(i2s->regmap, I2S_CLR, &val);
-
-		/* Should wait for clear operation to finish */
-		while (val & I2S_CLR_TXC) {
 			regmap_read(i2s->regmap, I2S_CLR, &val);
-			retry--;
-			if (!retry) {
-				dev_warn(i2s->dev, "fail to clear\n");
-				break;
+
+			/* Should wait for clear operation to finish */
+			while (val) {
+				regmap_read(i2s->regmap, I2S_CLR, &val);
+				retry--;
+				if (!retry) {
+					dev_warn(i2s->dev, "fail to clear\n");
+					break;
+				}
 			}
 		}
 	}
@@ -113,29 +128,37 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on)
 				   I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE);
 
 		regmap_update_bits(i2s->regmap, I2S_XFER,
-				   I2S_XFER_RXS_START,
-				   I2S_XFER_RXS_START);
+				   I2S_XFER_TXS_START | I2S_XFER_RXS_START,
+				   I2S_XFER_TXS_START | I2S_XFER_RXS_START);
+
+		i2s->rx_start = true;
 	} else {
+		i2s->rx_start = false;
+
 		regmap_update_bits(i2s->regmap, I2S_DMACR,
 				   I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE);
 
-		regmap_update_bits(i2s->regmap, I2S_XFER,
-				   I2S_XFER_RXS_START,
-				   I2S_XFER_RXS_STOP);
+		if (!i2s->tx_start) {
+			regmap_update_bits(i2s->regmap, I2S_XFER,
+					   I2S_XFER_TXS_START |
+					   I2S_XFER_RXS_START,
+					   I2S_XFER_TXS_STOP |
+					   I2S_XFER_RXS_STOP);
 
-		regmap_update_bits(i2s->regmap, I2S_CLR,
-				   I2S_CLR_RXC,
-				   I2S_CLR_RXC);
+			regmap_update_bits(i2s->regmap, I2S_CLR,
+					   I2S_CLR_TXC | I2S_CLR_RXC,
+					   I2S_CLR_TXC | I2S_CLR_RXC);
 
-		regmap_read(i2s->regmap, I2S_CLR, &val);
-
-		/* Should wait for clear operation to finish */
-		while (val & I2S_CLR_RXC) {
 			regmap_read(i2s->regmap, I2S_CLR, &val);
-			retry--;
-			if (!retry) {
-				dev_warn(i2s->dev, "fail to clear\n");
-				break;
+
+			/* Should wait for clear operation to finish */
+			while (val) {
+				regmap_read(i2s->regmap, I2S_CLR, &val);
+				retry--;
+				if (!retry) {
+					dev_warn(i2s->dev, "fail to clear\n");
+					break;
+				}
 			}
 		}
 	}
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 606399de6..c4c51a4d3 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -492,9 +492,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv,
 	 */
 	if (!count) {
 		clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT],
-					      parent_clk_name,
-					      (parent_clk_name) ?
-					      0 : CLK_IS_ROOT, req_rate);
+					      parent_clk_name, 0, req_rate);
 		if (!IS_ERR(clk)) {
 			adg->clkout[CLKOUT] = clk;
 			of_clk_add_provider(np, of_clk_src_simple_get, clk);
@@ -506,9 +504,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv,
 	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,
+						      parent_clk_name, 0,
 						      req_rate);
 			if (!IS_ERR(clk)) {
 				adg->onecell.clks	= adg->clkout;
@@ -522,7 +518,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv,
 		}
 	}
 
-	rsnd_mod_bset(adg_mod, SSICKR, 0x00FF0000, ckr);
+	rsnd_mod_bset(adg_mod, SSICKR, 0x80FF0000, ckr);
 	rsnd_mod_write(adg_mod, BRRA,  rbga);
 	rsnd_mod_write(adg_mod, BRRB,  rbgb);
 
diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c
index 7658e8fd7..6bc93cbb3 100644
--- a/sound/soc/sh/rcar/dma.c
+++ b/sound/soc/sh/rcar/dma.c
@@ -316,11 +316,15 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io,
 		size = ARRAY_SIZE(gen2_id_table_cmd);
 	}
 
-	if (!entry)
-		return 0xFF;
+	if ((!entry) || (size <= id)) {
+		struct device *dev = rsnd_priv_to_dev(rsnd_io_to_priv(io));
 
-	if (size <= id)
-		return 0xFF;
+		dev_err(dev, "unknown connection (%s[%d])\n",
+			rsnd_mod_name(mod), rsnd_mod_id(mod));
+
+		/* use non-prohibited SRS number as error */
+		return 0x00; /* SSI00 */
+	}
 
 	return entry[id];
 }
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index fc89a6725..a8f61d793 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -276,8 +276,9 @@ struct rsnd_mod {
 /*
  * status
  *
- * 0xH0000CB0
+ * 0xH0000CBA
  *
+ * A	0: probe	1: remove
  * B	0: init		1: quit
  * C	0: start	1: stop
  *
@@ -287,19 +288,19 @@ struct rsnd_mod {
  * H	0: fallback
  * H	0: hw_params
  */
+#define __rsnd_mod_shift_probe		0
+#define __rsnd_mod_shift_remove		0
 #define __rsnd_mod_shift_init		4
 #define __rsnd_mod_shift_quit		4
 #define __rsnd_mod_shift_start		8
 #define __rsnd_mod_shift_stop		8
-#define __rsnd_mod_shift_probe		28 /* always called */
-#define __rsnd_mod_shift_remove		28 /* always called */
 #define __rsnd_mod_shift_irq		28 /* always called */
 #define __rsnd_mod_shift_pcm_new	28 /* always called */
 #define __rsnd_mod_shift_fallback	28 /* always called */
 #define __rsnd_mod_shift_hw_params	28 /* always called */
 
-#define __rsnd_mod_add_probe		0
-#define __rsnd_mod_add_remove		0
+#define __rsnd_mod_add_probe		 1
+#define __rsnd_mod_add_remove		-1
 #define __rsnd_mod_add_init		 1
 #define __rsnd_mod_add_quit		-1
 #define __rsnd_mod_add_start		 1
@@ -310,7 +311,7 @@ struct rsnd_mod {
 #define __rsnd_mod_add_hw_params	0
 
 #define __rsnd_mod_call_probe		0
-#define __rsnd_mod_call_remove		0
+#define __rsnd_mod_call_remove		1
 #define __rsnd_mod_call_init		0
 #define __rsnd_mod_call_quit		1
 #define __rsnd_mod_call_start		0
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 15d6ffe8b..e39f916d0 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -572,6 +572,9 @@ int rsnd_src_probe(struct rsnd_priv *priv)
 
 	i = 0;
 	for_each_child_of_node(node, np) {
+		if (!of_device_is_available(np))
+			goto skip;
+
 		src = rsnd_src_get(priv, i);
 
 		snprintf(name, RSND_SRC_NAME_SIZE, "%s.%d",
@@ -595,6 +598,7 @@ int rsnd_src_probe(struct rsnd_priv *priv)
 		if (ret)
 			goto rsnd_src_probe_done;
 
+skip:
 		i++;
 	}
 
diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c
index 7e0acd83b..bc4a55bb3 100644
--- a/sound/soc/soc-ac97.c
+++ b/sound/soc/soc-ac97.c
@@ -59,8 +59,7 @@ static void soc_ac97_device_release(struct device *dev)
 #ifdef CONFIG_GPIOLIB
 static inline struct snd_soc_codec *gpio_to_codec(struct gpio_chip *chip)
 {
-	struct snd_ac97_gpio_priv *gpio_priv =
-		container_of(chip, struct snd_ac97_gpio_priv, gpio_chip);
+	struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip);
 
 	return gpio_priv->codec;
 }
@@ -98,8 +97,7 @@ static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset)
 static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned offset,
 				  int value)
 {
-	struct snd_ac97_gpio_priv *gpio_priv =
-		container_of(chip, struct snd_ac97_gpio_priv, gpio_chip);
+	struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip);
 	struct snd_soc_codec *codec = gpio_to_codec(chip);
 
 	gpio_priv->gpios_set &= ~(1 << offset);
@@ -145,7 +143,7 @@ static int snd_soc_ac97_init_gpio(struct snd_ac97 *ac97,
 	gpio_priv->gpio_chip.parent = codec->dev;
 	gpio_priv->gpio_chip.base = -1;
 
-	ret = gpiochip_add(&gpio_priv->gpio_chip);
+	ret = gpiochip_add_data(&gpio_priv->gpio_chip, gpio_priv);
 	if (ret != 0)
 		dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret);
 	return ret;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d2e62b159..16369cad4 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -930,7 +930,18 @@ static struct snd_soc_component *soc_find_component(
 	return NULL;
 }
 
-static struct snd_soc_dai *snd_soc_find_dai(
+/**
+ * snd_soc_find_dai - Find a registered DAI
+ *
+ * @dlc: name of the DAI and optional component info to match
+ *
+ * This function will search all regsitered components and their DAIs to
+ * find the DAI of the same name. The component's of_node and name
+ * should also match if being specified.
+ *
+ * Return: pointer of DAI, or NULL if not found.
+ */
+struct snd_soc_dai *snd_soc_find_dai(
 	const struct snd_soc_dai_link_component *dlc)
 {
 	struct snd_soc_component *component;
@@ -959,6 +970,7 @@ static struct snd_soc_dai *snd_soc_find_dai(
 
 	return NULL;
 }
+EXPORT_SYMBOL_GPL(snd_soc_find_dai);
 
 static bool soc_is_dai_link_bound(struct snd_soc_card *card,
 		struct snd_soc_dai_link *dai_link)
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 6fd1906af..6cef39775 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -163,31 +163,42 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea
 	}
 
 	/*
-	 * Prepare formats mask for valid/allowed sample types. If the dma does
-	 * not have support for the given physical word size, it needs to be
-	 * masked out so user space can not use the format which produces
-	 * corrupted audio.
-	 * In case the dma driver does not implement the slave_caps the default
-	 * assumption is that it supports 1, 2 and 4 bytes widths.
+	 * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep
+	 * hw.formats set to 0, meaning no restrictions are in place.
+	 * In this case it's the responsibility of the DAI driver to
+	 * provide the supported format information.
 	 */
-	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
-		int bits = snd_pcm_format_physical_width(i);
-
-		/* Enable only samples with DMA supported physical widths */
-		switch (bits) {
-		case 8:
-		case 16:
-		case 24:
-		case 32:
-		case 64:
-			if (addr_widths & (1 << (bits / 8)))
-				hw.formats |= (1LL << i);
-			break;
-		default:
-			/* Unsupported types */
-			break;
+	if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK))
+		/*
+		 * Prepare formats mask for valid/allowed sample types. If the
+		 * dma does not have support for the given physical word size,
+		 * it needs to be masked out so user space can not use the
+		 * format which produces corrupted audio.
+		 * In case the dma driver does not implement the slave_caps the
+		 * default assumption is that it supports 1, 2 and 4 bytes
+		 * widths.
+		 */
+		for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+			int bits = snd_pcm_format_physical_width(i);
+
+			/*
+			 * Enable only samples with DMA supported physical
+			 * widths
+			 */
+			switch (bits) {
+			case 8:
+			case 16:
+			case 24:
+			case 32:
+			case 64:
+				if (addr_widths & (1 << (bits / 8)))
+					hw.formats |= (1LL << i);
+				break;
+			default:
+				/* Unsupported types */
+				break;
+			}
 		}
-	}
 
 	return snd_soc_set_runtime_hwparams(substream, &hw);
 }
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 1cf94d7fb..ee7f15aa4 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1023,6 +1023,11 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg,
 
 		control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos;
 
+		if (control_hdr->size != sizeof(*control_hdr)) {
+			dev_err(tplg->dev, "ASoC: invalid control size\n");
+			return -EINVAL;
+		}
+
 		switch (control_hdr->ops.info) {
 		case SND_SOC_TPLG_CTL_VOLSW:
 		case SND_SOC_TPLG_CTL_STROBE:
@@ -1476,6 +1481,8 @@ widget:
 	widget->dobj.type = SND_SOC_DOBJ_WIDGET;
 	widget->dobj.ops = tplg->ops;
 	widget->dobj.index = tplg->index;
+	kfree(template.sname);
+	kfree(template.name);
 	list_add(&widget->dobj.list, &tplg->comp->dobj_list);
 	return 0;
 
@@ -1499,10 +1506,17 @@ static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg,
 
 	for (i = 0; i < count; i++) {
 		widget = (struct snd_soc_tplg_dapm_widget *) tplg->pos;
+		if (widget->size != sizeof(*widget)) {
+			dev_err(tplg->dev, "ASoC: invalid widget size\n");
+			return -EINVAL;
+		}
+
 		ret = soc_tplg_dapm_widget_create(tplg, widget);
-		if (ret < 0)
+		if (ret < 0) {
 			dev_err(tplg->dev, "ASoC: failed to load widget %s\n",
 				widget->name);
+			return ret;
+		}
 	}
 
 	return 0;
@@ -1586,6 +1600,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
 	return snd_soc_register_dai(tplg->comp, dai_drv);
 }
 
+/* create the FE DAI link */
 static int soc_tplg_link_create(struct soc_tplg *tplg,
 	struct snd_soc_tplg_pcm *pcm)
 {
@@ -1598,6 +1613,16 @@ static int soc_tplg_link_create(struct soc_tplg *tplg,
 
 	link->name = pcm->pcm_name;
 	link->stream_name = pcm->pcm_name;
+	link->id = pcm->pcm_id;
+
+	link->cpu_dai_name = pcm->dai_name;
+	link->codec_name = "snd-soc-dummy";
+	link->codec_dai_name = "snd-soc-dummy-dai";
+
+	/* enable DPCM */
+	link->dynamic = 1;
+	link->dpcm_playback = pcm->playback;
+	link->dpcm_capture = pcm->capture;
 
 	/* pass control to component driver for optional further init */
 	ret = soc_tplg_dai_link_load(tplg, link);
@@ -1639,8 +1664,6 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
 	if (tplg->pass != SOC_TPLG_PASS_PCM_DAI)
 		return 0;
 
-	pcm = (struct snd_soc_tplg_pcm *)tplg->pos;
-
 	if (soc_tplg_check_elem_count(tplg,
 		sizeof(struct snd_soc_tplg_pcm), count,
 		hdr->payload_size, "PCM DAI")) {
@@ -1650,7 +1673,13 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
 	}
 
 	/* create the FE DAIs and DAI links */
+	pcm = (struct snd_soc_tplg_pcm *)tplg->pos;
 	for (i = 0; i < count; i++) {
+		if (pcm->size != sizeof(*pcm)) {
+			dev_err(tplg->dev, "ASoC: invalid pcm size\n");
+			return -EINVAL;
+		}
+
 		soc_tplg_pcm_create(tplg, pcm);
 		pcm++;
 	}
@@ -1670,6 +1699,11 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg,
 		return 0;
 
 	manifest = (struct snd_soc_tplg_manifest *)tplg->pos;
+	if (manifest->size != sizeof(*manifest)) {
+		dev_err(tplg->dev, "ASoC: invalid manifest size\n");
+		return -EINVAL;
+	}
+
 	tplg->pos += sizeof(struct snd_soc_tplg_manifest);
 
 	if (tplg->comp && tplg->ops && tplg->ops->manifest)
@@ -1686,6 +1720,14 @@ static int soc_valid_header(struct soc_tplg *tplg,
 	if (soc_tplg_get_hdr_offset(tplg) >= tplg->fw->size)
 		return 0;
 
+	if (hdr->size != sizeof(*hdr)) {
+		dev_err(tplg->dev,
+			"ASoC: invalid header size for type %d at offset 0x%lx size 0x%zx.\n",
+			hdr->type, soc_tplg_get_hdr_offset(tplg),
+			tplg->fw->size);
+		return -EINVAL;
+	}
+
 	/* big endian firmware objects not supported atm */
 	if (hdr->magic == cpu_to_be32(SND_SOC_TPLG_MAGIC)) {
 		dev_err(tplg->dev,
diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c
index 39bcefe5e..488ef4ed8 100644
--- a/sound/soc/sti/sti_uniperif.c
+++ b/sound/soc/sti/sti_uniperif.c
@@ -11,6 +11,142 @@
 #include "uniperif.h"
 
 /*
+ * User frame size shall be 2, 4, 6 or 8 32-bits words length
+ * (i.e. 8, 16, 24 or 32 bytes)
+ * This constraint comes from allowed values for
+ * UNIPERIF_I2S_FMT_NUM_CH register
+ */
+#define UNIPERIF_MAX_FRAME_SZ 0x20
+#define UNIPERIF_ALLOWED_FRAME_SZ (0x08 | 0x10 | 0x18 | UNIPERIF_MAX_FRAME_SZ)
+
+int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+			       unsigned int rx_mask, int slots,
+			       int slot_width)
+{
+	struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai);
+	struct uniperif *uni = priv->dai_data.uni;
+	int i, frame_size, avail_slots;
+
+	if (!UNIPERIF_TYPE_IS_TDM(uni)) {
+		dev_err(uni->dev, "cpu dai not in tdm mode\n");
+		return -EINVAL;
+	}
+
+	/* store info in unip context */
+	uni->tdm_slot.slots = slots;
+	uni->tdm_slot.slot_width = slot_width;
+	/* unip is unidirectionnal */
+	uni->tdm_slot.mask = (tx_mask != 0) ? tx_mask : rx_mask;
+
+	/* number of available timeslots */
+	for (i = 0, avail_slots = 0; i < uni->tdm_slot.slots; i++) {
+		if ((uni->tdm_slot.mask >> i) & 0x01)
+			avail_slots++;
+	}
+	uni->tdm_slot.avail_slots = avail_slots;
+
+	/* frame size in bytes */
+	frame_size = uni->tdm_slot.avail_slots * uni->tdm_slot.slot_width / 8;
+
+	/* check frame size is allowed */
+	if ((frame_size > UNIPERIF_MAX_FRAME_SZ) ||
+	    (frame_size & ~(int)UNIPERIF_ALLOWED_FRAME_SZ)) {
+		dev_err(uni->dev, "frame size not allowed: %d bytes\n",
+			frame_size);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+int sti_uniperiph_fix_tdm_chan(struct snd_pcm_hw_params *params,
+			       struct snd_pcm_hw_rule *rule)
+{
+	struct uniperif *uni = rule->private;
+	struct snd_interval t;
+
+	t.min = uni->tdm_slot.avail_slots;
+	t.max = uni->tdm_slot.avail_slots;
+	t.openmin = 0;
+	t.openmax = 0;
+	t.integer = 0;
+
+	return snd_interval_refine(hw_param_interval(params, rule->var), &t);
+}
+
+int sti_uniperiph_fix_tdm_format(struct snd_pcm_hw_params *params,
+				 struct snd_pcm_hw_rule *rule)
+{
+	struct uniperif *uni = rule->private;
+	struct snd_mask *maskp = hw_param_mask(params, rule->var);
+	u64 format;
+
+	switch (uni->tdm_slot.slot_width) {
+	case 16:
+		format = SNDRV_PCM_FMTBIT_S16_LE;
+		break;
+	case 32:
+		format = SNDRV_PCM_FMTBIT_S32_LE;
+		break;
+	default:
+		dev_err(uni->dev, "format not supported: %d bits\n",
+			uni->tdm_slot.slot_width);
+		return -EINVAL;
+	}
+
+	maskp->bits[0] &= (u_int32_t)format;
+	maskp->bits[1] &= (u_int32_t)(format >> 32);
+	/* clear remaining indexes */
+	memset(maskp->bits + 2, 0, (SNDRV_MASK_MAX - 64) / 8);
+
+	if (!maskp->bits[0] && !maskp->bits[1])
+		return -EINVAL;
+
+	return 0;
+}
+
+int sti_uniperiph_get_tdm_word_pos(struct uniperif *uni,
+				   unsigned int *word_pos)
+{
+	int slot_width = uni->tdm_slot.slot_width / 8;
+	int slots_num = uni->tdm_slot.slots;
+	unsigned int slots_mask = uni->tdm_slot.mask;
+	int i, j, k;
+	unsigned int word16_pos[4];
+
+	/* word16_pos:
+	 * word16_pos[0] = WORDX_LSB
+	 * word16_pos[1] = WORDX_MSB,
+	 * word16_pos[2] = WORDX+1_LSB
+	 * word16_pos[3] = WORDX+1_MSB
+	 */
+
+	/* set unip word position */
+	for (i = 0, j = 0, k = 0; (i < slots_num) && (k < WORD_MAX); i++) {
+		if ((slots_mask >> i) & 0x01) {
+			word16_pos[j] = i * slot_width;
+
+			if (slot_width == 4) {
+				word16_pos[j + 1] = word16_pos[j] + 2;
+				j++;
+			}
+			j++;
+
+			if (j > 3) {
+				word_pos[k] = word16_pos[1] |
+					      (word16_pos[0] << 8) |
+					      (word16_pos[3] << 16) |
+					      (word16_pos[2] << 24);
+				j = 0;
+				k++;
+			}
+		}
+	}
+
+	return 0;
+}
+
+/*
  * sti_uniperiph_dai_create_ctrl
  * This function is used to create Ctrl associated to DAI but also pcm device.
  * Request is done by front end to associate ctrl with pcm device id
@@ -45,10 +181,16 @@ int sti_uniperiph_dai_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
 {
+	struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai);
+	struct uniperif *uni = priv->dai_data.uni;
 	struct snd_dmaengine_dai_dma_data *dma_data;
 	int transfer_size;
 
-	transfer_size = params_channels(params) * UNIPERIF_FIFO_FRAMES;
+	if (uni->info->type == SND_ST_UNIPERIF_TYPE_TDM)
+		/* transfer size = user frame size (in 32-bits FIFO cell) */
+		transfer_size = snd_soc_params_to_frame_size(params) / 32;
+	else
+		transfer_size = params_channels(params) * UNIPERIF_FIFO_FRAMES;
 
 	dma_data = snd_soc_dai_get_dma_data(dai, substream);
 	dma_data->maxburst = transfer_size;
diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h
index f0fd5a994..eb9933c62 100644
--- a/sound/soc/sti/uniperif.h
+++ b/sound/soc/sti/uniperif.h
@@ -25,7 +25,7 @@
 	writel_relaxed((((value) & mask) << shift), ip->base + offset)
 
 /*
- * AUD_UNIPERIF_SOFT_RST reg
+ * UNIPERIF_SOFT_RST reg
  */
 
 #define UNIPERIF_SOFT_RST_OFFSET(ip) 0x0000
@@ -50,7 +50,7 @@
 		UNIPERIF_SOFT_RST_SOFT_RST_MASK(ip))
 
 /*
- * AUD_UNIPERIF_FIFO_DATA reg
+ * UNIPERIF_FIFO_DATA reg
  */
 
 #define UNIPERIF_FIFO_DATA_OFFSET(ip) 0x0004
@@ -58,7 +58,7 @@
 	writel_relaxed(value, ip->base + UNIPERIF_FIFO_DATA_OFFSET(ip))
 
 /*
- * AUD_UNIPERIF_CHANNEL_STA_REGN reg
+ * UNIPERIF_CHANNEL_STA_REGN reg
  */
 
 #define UNIPERIF_CHANNEL_STA_REGN(ip, n) (0x0060 + (4 * n))
@@ -105,7 +105,7 @@
 	writel_relaxed(value, ip->base + UNIPERIF_CHANNEL_STA_REG5_OFFSET(ip))
 
 /*
- *  AUD_UNIPERIF_ITS reg
+ *  UNIPERIF_ITS reg
  */
 
 #define UNIPERIF_ITS_OFFSET(ip) 0x000C
@@ -143,7 +143,7 @@
 		0 : (BIT(UNIPERIF_ITS_UNDERFLOW_REC_FAILED_SHIFT(ip))))
 
 /*
- *  AUD_UNIPERIF_ITS_BCLR reg
+ *  UNIPERIF_ITS_BCLR reg
  */
 
 /* FIFO_ERROR */
@@ -160,7 +160,7 @@
 	writel_relaxed(value, ip->base + UNIPERIF_ITS_BCLR_OFFSET(ip))
 
 /*
- *  AUD_UNIPERIF_ITM reg
+ *  UNIPERIF_ITM reg
  */
 
 #define UNIPERIF_ITM_OFFSET(ip) 0x0018
@@ -188,7 +188,7 @@
 		0 : (BIT(UNIPERIF_ITM_UNDERFLOW_REC_FAILED_SHIFT(ip))))
 
 /*
- *  AUD_UNIPERIF_ITM_BCLR reg
+ *  UNIPERIF_ITM_BCLR reg
  */
 
 #define UNIPERIF_ITM_BCLR_OFFSET(ip) 0x001c
@@ -213,7 +213,7 @@
 		UNIPERIF_ITM_BCLR_DMA_ERROR_MASK(ip))
 
 /*
- *  AUD_UNIPERIF_ITM_BSET reg
+ *  UNIPERIF_ITM_BSET reg
  */
 
 #define UNIPERIF_ITM_BSET_OFFSET(ip) 0x0020
@@ -767,7 +767,7 @@
 	SET_UNIPERIF_REG(ip, \
 		UNIPERIF_CTRL_OFFSET(ip), \
 		UNIPERIF_CTRL_READER_OUT_SEL_SHIFT(ip), \
-		CORAUD_UNIPERIF_CTRL_READER_OUT_SEL_MASK(ip), 1)
+		UNIPERIF_CTRL_READER_OUT_SEL_MASK(ip), 1)
 
 /* UNDERFLOW_REC_WINDOW */
 #define UNIPERIF_CTRL_UNDERFLOW_REC_WINDOW_SHIFT(ip) 20
@@ -1046,7 +1046,7 @@
 		UNIPERIF_STATUS_1_UNDERFLOW_DURATION_MASK(ip), value)
 
 /*
- * AUD_UNIPERIF_CHANNEL_STA_REGN reg
+ * UNIPERIF_CHANNEL_STA_REGN reg
  */
 
 #define UNIPERIF_CHANNEL_STA_REGN(ip, n) (0x0060 + (4 * n))
@@ -1057,7 +1057,7 @@
 			UNIPERIF_CHANNEL_STA_REGN(ip, n))
 
 /*
- * AUD_UNIPERIF_USER_VALIDITY reg
+ * UNIPERIF_USER_VALIDITY reg
  */
 
 #define UNIPERIF_USER_VALIDITY_OFFSET(ip) 0x0090
@@ -1101,12 +1101,136 @@
 		UNIPERIF_DBG_STANDBY_LEFT_SP_MASK(ip), value)
 
 /*
+ * UNIPERIF_TDM_ENABLE
+ */
+#define UNIPERIF_TDM_ENABLE_OFFSET(ip) 0x0118
+#define GET_UNIPERIF_TDM_ENABLE(ip) \
+	readl_relaxed(ip->base + UNIPERIF_TDM_ENABLE_OFFSET(ip))
+#define SET_UNIPERIF_TDM_ENABLE(ip, value) \
+	writel_relaxed(value, ip->base + UNIPERIF_TDM_ENABLE_OFFSET(ip))
+
+/* TDM_ENABLE */
+#define UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip) 0x0
+#define UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip) 0x1
+#define GET_UNIPERIF_TDM_ENABLE_EN_TDM(ip) \
+		GET_UNIPERIF_REG(ip, \
+		UNIPERIF_TDM_ENABLE_OFFSET(ip), \
+		UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip), \
+		UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip))
+#define SET_UNIPERIF_TDM_ENABLE_TDM_ENABLE(ip) \
+		SET_UNIPERIF_REG(ip, \
+		UNIPERIF_TDM_ENABLE_OFFSET(ip), \
+		UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip), \
+		UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip), 1)
+#define SET_UNIPERIF_TDM_ENABLE_TDM_DISABLE(ip) \
+		SET_UNIPERIF_REG(ip, \
+		UNIPERIF_TDM_ENABLE_OFFSET(ip), \
+		UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip), \
+		UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip), 0)
+
+/*
+ * UNIPERIF_TDM_FS_REF_FREQ
+ */
+#define UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip) 0x011c
+#define GET_UNIPERIF_TDM_FS_REF_FREQ(ip) \
+	readl_relaxed(ip->base + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip))
+#define SET_UNIPERIF_TDM_FS_REF_FREQ(ip, value) \
+	writel_relaxed(value, ip->base + \
+			UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip))
+
+/* REF_FREQ */
+#define UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip) 0x0
+#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_8KHZ(ip) 0
+#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_16KHZ(ip) 1
+#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_32KHZ(ip) 2
+#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_48KHZ(ip) 3
+#define UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip) 0x3
+#define GET_UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ(ip) \
+		GET_UNIPERIF_REG(ip, \
+		UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \
+		UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \
+		UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip))
+#define SET_UNIPERIF_TDM_FS_REF_FREQ_8KHZ(ip) \
+		SET_UNIPERIF_REG(ip, \
+		UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \
+		UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \
+		UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \
+		VALUE_UNIPERIF_TDM_FS_REF_FREQ_8KHZ(ip))
+#define SET_UNIPERIF_TDM_FS_REF_FREQ_16KHZ(ip) \
+		SET_UNIPERIF_REG(ip, \
+		UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \
+		UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \
+		UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \
+		VALUE_UNIPERIF_TDM_FS_REF_FREQ_16KHZ(ip))
+#define SET_UNIPERIF_TDM_FS_REF_FREQ_32KHZ(ip) \
+		SET_UNIPERIF_REG(ip, \
+		UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \
+		UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \
+		UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \
+		VALUE_UNIPERIF_TDM_FS_REF_FREQ_32KHZ(ip))
+#define SET_UNIPERIF_TDM_FS_REF_FREQ_48KHZ(ip) \
+		SET_UNIPERIF_REG(ip, \
+		UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \
+		UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \
+		UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \
+		VALUE_UNIPERIF_TDM_FS_REF_FREQ_48KHZ(ip))
+
+/*
+ * UNIPERIF_TDM_FS_REF_DIV
+ */
+#define UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip) 0x0120
+#define GET_UNIPERIF_TDM_FS_REF_DIV(ip) \
+	readl_relaxed(ip->base + UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip))
+#define SET_UNIPERIF_TDM_FS_REF_DIV(ip, value) \
+		writel_relaxed(value, ip->base + \
+			UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip))
+
+/* NUM_TIMESLOT */
+#define UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_SHIFT(ip) 0x0
+#define UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_MASK(ip) 0xff
+#define GET_UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT(ip) \
+		GET_UNIPERIF_REG(ip, \
+		UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip), \
+		UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_SHIFT(ip), \
+		UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_MASK(ip))
+#define SET_UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT(ip, value) \
+		SET_UNIPERIF_REG(ip, \
+		UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip), \
+		UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_SHIFT(ip), \
+		UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_MASK(ip), value)
+
+/*
+ * UNIPERIF_TDM_WORD_POS_X_Y
+ * 32 bits of UNIPERIF_TDM_WORD_POS_X_Y register shall be set in 1 shot
+ */
+#define UNIPERIF_TDM_WORD_POS_1_2_OFFSET(ip) 0x013c
+#define UNIPERIF_TDM_WORD_POS_3_4_OFFSET(ip) 0x0140
+#define UNIPERIF_TDM_WORD_POS_5_6_OFFSET(ip) 0x0144
+#define UNIPERIF_TDM_WORD_POS_7_8_OFFSET(ip) 0x0148
+#define GET_UNIPERIF_TDM_WORD_POS(ip, words) \
+	readl_relaxed(ip->base + UNIPERIF_TDM_WORD_POS_##words##_OFFSET(ip))
+#define SET_UNIPERIF_TDM_WORD_POS(ip, words, value) \
+		writel_relaxed(value, ip->base + \
+		UNIPERIF_TDM_WORD_POS_##words##_OFFSET(ip))
+/*
  * uniperipheral IP capabilities
  */
 
 #define UNIPERIF_FIFO_SIZE		70 /* FIFO is 70 cells deep */
 #define UNIPERIF_FIFO_FRAMES		4  /* FDMA trigger limit in frames */
 
+#define UNIPERIF_TYPE_IS_HDMI(p) \
+	((p)->info->type == SND_ST_UNIPERIF_TYPE_HDMI)
+#define UNIPERIF_TYPE_IS_PCM(p) \
+	((p)->info->type == SND_ST_UNIPERIF_TYPE_PCM)
+#define UNIPERIF_TYPE_IS_SPDIF(p) \
+	((p)->info->type == SND_ST_UNIPERIF_TYPE_SPDIF)
+#define UNIPERIF_TYPE_IS_IEC958(p) \
+	(UNIPERIF_TYPE_IS_HDMI(p) || \
+		UNIPERIF_TYPE_IS_SPDIF(p))
+#define UNIPERIF_TYPE_IS_TDM(p) \
+	((p)->info->type == SND_ST_UNIPERIF_TYPE_TDM)
+
 /*
  * Uniperipheral IP revisions
  */
@@ -1125,10 +1249,11 @@ enum uniperif_version {
 };
 
 enum uniperif_type {
-	SND_ST_UNIPERIF_PLAYER_TYPE_NONE,
-	SND_ST_UNIPERIF_PLAYER_TYPE_HDMI,
-	SND_ST_UNIPERIF_PLAYER_TYPE_PCM,
-	SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF
+	SND_ST_UNIPERIF_TYPE_NONE,
+	SND_ST_UNIPERIF_TYPE_HDMI,
+	SND_ST_UNIPERIF_TYPE_PCM,
+	SND_ST_UNIPERIF_TYPE_SPDIF,
+	SND_ST_UNIPERIF_TYPE_TDM
 };
 
 enum uniperif_state {
@@ -1145,9 +1270,17 @@ enum uniperif_iec958_encoding_mode {
 	UNIPERIF_IEC958_ENCODING_MODE_ENCODED
 };
 
+enum uniperif_word_pos {
+	WORD_1_2,
+	WORD_3_4,
+	WORD_5_6,
+	WORD_7_8,
+	WORD_MAX
+};
+
 struct uniperif_info {
 	int id; /* instance value of the uniperipheral IP */
-	enum uniperif_type player_type;
+	enum uniperif_type type;
 	int underflow_enabled;		/* Underflow recovery mode */
 };
 
@@ -1156,12 +1289,20 @@ struct uniperif_iec958_settings {
 	struct snd_aes_iec958 iec958;
 };
 
+struct dai_tdm_slot {
+	unsigned int mask;
+	int slots;
+	int slot_width;
+	unsigned int avail_slots;
+};
+
 struct uniperif {
 	/* System information */
 	struct uniperif_info *info;
 	struct device *dev;
 	int ver; /* IP version, used by register access macros */
 	struct regmap_field *clk_sel;
+	struct regmap_field *valid_sel;
 
 	/* capabilities */
 	const struct snd_pcm_hardware *hw;
@@ -1192,6 +1333,7 @@ struct uniperif {
 
 	/* dai properties */
 	unsigned int daifmt;
+	struct dai_tdm_slot tdm_slot;
 
 	/* DAI callbacks */
 	const struct snd_soc_dai_ops *dai_ops;
@@ -1209,6 +1351,28 @@ struct sti_uniperiph_data {
 	struct sti_uniperiph_dai dai_data;
 };
 
+static const struct snd_pcm_hardware uni_tdm_hw = {
+	.info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP |
+		SNDRV_PCM_INFO_MMAP_VALID,
+
+	.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE,
+
+	.rates = SNDRV_PCM_RATE_CONTINUOUS,
+	.rate_min = 8000,
+	.rate_max = 48000,
+
+	.channels_min = 1,
+	.channels_max = 32,
+
+	.periods_min = 2,
+	.periods_max = 10,
+
+	.period_bytes_min = 128,
+	.period_bytes_max = 64 * PAGE_SIZE,
+	.buffer_bytes_max = 256 * PAGE_SIZE
+};
+
 /* uniperiph player*/
 int uni_player_init(struct platform_device *pdev,
 		    struct uniperif *uni_player);
@@ -1226,4 +1390,28 @@ int sti_uniperiph_dai_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai);
 
+static inline int sti_uniperiph_get_user_frame_size(
+	struct snd_pcm_runtime *runtime)
+{
+	return (runtime->channels * snd_pcm_format_width(runtime->format) / 8);
+}
+
+static inline int sti_uniperiph_get_unip_tdm_frame_size(struct uniperif *uni)
+{
+	return (uni->tdm_slot.slots * uni->tdm_slot.slot_width / 8);
+}
+
+int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+			       unsigned int rx_mask, int slots,
+			       int slot_width);
+
+int sti_uniperiph_get_tdm_word_pos(struct uniperif *uni,
+				   unsigned int *word_pos);
+
+int sti_uniperiph_fix_tdm_chan(struct snd_pcm_hw_params *params,
+			       struct snd_pcm_hw_rule *rule);
+
+int sti_uniperiph_fix_tdm_format(struct snd_pcm_hw_params *params,
+				 struct snd_pcm_hw_rule *rule);
+
 #endif
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index 7aca6b92f..ee1c7c245 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -21,23 +21,14 @@
 
 /* sys config registers definitions */
 #define SYS_CFG_AUDIO_GLUE 0xA4
-#define SYS_CFG_AUDI0_GLUE_PCM_CLKX 8
 
 /*
  * Driver specific types.
  */
-#define UNIPERIF_PLAYER_TYPE_IS_HDMI(p) \
-	((p)->info->player_type == SND_ST_UNIPERIF_PLAYER_TYPE_HDMI)
-#define UNIPERIF_PLAYER_TYPE_IS_PCM(p) \
-	((p)->info->player_type == SND_ST_UNIPERIF_PLAYER_TYPE_PCM)
-#define UNIPERIF_PLAYER_TYPE_IS_SPDIF(p) \
-	((p)->info->player_type == SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF)
-#define UNIPERIF_PLAYER_TYPE_IS_IEC958(p) \
-	(UNIPERIF_PLAYER_TYPE_IS_HDMI(p) || \
-		UNIPERIF_PLAYER_TYPE_IS_SPDIF(p))
 
 #define UNIPERIF_PLAYER_CLK_ADJ_MIN  -999999
 #define UNIPERIF_PLAYER_CLK_ADJ_MAX  1000000
+#define UNIPERIF_PLAYER_I2S_OUT 1 /* player id connected to I2S/TDM TX bus */
 
 /*
  * Note: snd_pcm_hardware is linked to DMA controller but is declared here to
@@ -444,18 +435,11 @@ static int uni_player_prepare_pcm(struct uniperif *player,
 
 	/* Force slot width to 32 in I2S mode (HW constraint) */
 	if ((player->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) ==
-		SND_SOC_DAIFMT_I2S) {
+		SND_SOC_DAIFMT_I2S)
 		slot_width = 32;
-	} else {
-		switch (runtime->format) {
-		case SNDRV_PCM_FORMAT_S16_LE:
-			slot_width = 16;
-			break;
-		default:
-			slot_width = 32;
-			break;
-		}
-	}
+	else
+		slot_width = snd_pcm_format_width(runtime->format);
+
 	output_frame_size = slot_width * runtime->channels;
 
 	clk_div = player->mclk / runtime->rate;
@@ -530,7 +514,6 @@ static int uni_player_prepare_pcm(struct uniperif *player,
 	SET_UNIPERIF_CONFIG_ONE_BIT_AUD_DISABLE(player);
 
 	SET_UNIPERIF_I2S_FMT_ORDER_MSB(player);
-	SET_UNIPERIF_I2S_FMT_SCLK_EDGE_FALLING(player);
 
 	/* No iec958 formatting as outputting to DAC  */
 	SET_UNIPERIF_CTRL_SPDIF_FMT_OFF(player);
@@ -538,6 +521,55 @@ static int uni_player_prepare_pcm(struct uniperif *player,
 	return 0;
 }
 
+static int uni_player_prepare_tdm(struct uniperif *player,
+				  struct snd_pcm_runtime *runtime)
+{
+	int tdm_frame_size; /* unip tdm frame size in bytes */
+	int user_frame_size; /* user tdm frame size in bytes */
+	/* default unip TDM_WORD_POS_X_Y */
+	unsigned int word_pos[4] = {
+		0x04060002, 0x0C0E080A, 0x14161012, 0x1C1E181A};
+	int freq, ret;
+
+	tdm_frame_size =
+		sti_uniperiph_get_unip_tdm_frame_size(player);
+	user_frame_size =
+		sti_uniperiph_get_user_frame_size(runtime);
+
+	/* fix 16/0 format */
+	SET_UNIPERIF_CONFIG_MEM_FMT_16_0(player);
+	SET_UNIPERIF_I2S_FMT_DATA_SIZE_32(player);
+
+	/* number of words inserted on the TDM line */
+	SET_UNIPERIF_I2S_FMT_NUM_CH(player, user_frame_size / 4 / 2);
+
+	SET_UNIPERIF_I2S_FMT_ORDER_MSB(player);
+	SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(player);
+
+	/* Enable the tdm functionality */
+	SET_UNIPERIF_TDM_ENABLE_TDM_ENABLE(player);
+
+	/* number of 8 bits timeslots avail in unip tdm frame */
+	SET_UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT(player, tdm_frame_size);
+
+	/* set the timeslot allocation for words in FIFO */
+	sti_uniperiph_get_tdm_word_pos(player, word_pos);
+	SET_UNIPERIF_TDM_WORD_POS(player, 1_2, word_pos[WORD_1_2]);
+	SET_UNIPERIF_TDM_WORD_POS(player, 3_4, word_pos[WORD_3_4]);
+	SET_UNIPERIF_TDM_WORD_POS(player, 5_6, word_pos[WORD_5_6]);
+	SET_UNIPERIF_TDM_WORD_POS(player, 7_8, word_pos[WORD_7_8]);
+
+	/* set unip clk rate (not done vai set_sysclk ops) */
+	freq = runtime->rate * tdm_frame_size * 8;
+	mutex_lock(&player->ctrl_lock);
+	ret = uni_player_clk_set_rate(player, freq);
+	if (!ret)
+		player->mclk = freq;
+	mutex_unlock(&player->ctrl_lock);
+
+	return 0;
+}
+
 /*
  * ALSA uniperipheral iec958 controls
  */
@@ -668,11 +700,29 @@ static int uni_player_startup(struct snd_pcm_substream *substream,
 {
 	struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai);
 	struct uniperif *player = priv->dai_data.uni;
+	int ret;
+
 	player->substream = substream;
 
 	player->clk_adj = 0;
 
-	return 0;
+	if (!UNIPERIF_TYPE_IS_TDM(player))
+		return 0;
+
+	/* refine hw constraint in tdm mode */
+	ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+				  SNDRV_PCM_HW_PARAM_CHANNELS,
+				  sti_uniperiph_fix_tdm_chan,
+				  player, SNDRV_PCM_HW_PARAM_CHANNELS,
+				  -1);
+	if (ret < 0)
+		return ret;
+
+	return snd_pcm_hw_rule_add(substream->runtime, 0,
+				   SNDRV_PCM_HW_PARAM_FORMAT,
+				   sti_uniperiph_fix_tdm_format,
+				   player, SNDRV_PCM_HW_PARAM_FORMAT,
+				   -1);
 }
 
 static int uni_player_set_sysclk(struct snd_soc_dai *dai, int clk_id,
@@ -682,7 +732,7 @@ static int uni_player_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	struct uniperif *player = priv->dai_data.uni;
 	int ret;
 
-	if (dir == SND_SOC_CLOCK_IN)
+	if (UNIPERIF_TYPE_IS_TDM(player) || (dir == SND_SOC_CLOCK_IN))
 		return 0;
 
 	if (clk_id != 0)
@@ -714,7 +764,13 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
 	}
 
 	/* Calculate transfer size (in fifo cells and bytes) for frame count */
-	transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES;
+	if (player->info->type == SND_ST_UNIPERIF_TYPE_TDM) {
+		/* transfer size = user frame size (in 32 bits FIFO cell) */
+		transfer_size =
+			sti_uniperiph_get_user_frame_size(runtime) / 4;
+	} else {
+		transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES;
+	}
 
 	/* Calculate number of empty cells available before asserting DREQ */
 	if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) {
@@ -738,16 +794,19 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
 	SET_UNIPERIF_CONFIG_DMA_TRIG_LIMIT(player, trigger_limit);
 
 	/* Uniperipheral setup depends on player type */
-	switch (player->info->player_type) {
-	case SND_ST_UNIPERIF_PLAYER_TYPE_HDMI:
+	switch (player->info->type) {
+	case SND_ST_UNIPERIF_TYPE_HDMI:
 		ret = uni_player_prepare_iec958(player, runtime);
 		break;
-	case SND_ST_UNIPERIF_PLAYER_TYPE_PCM:
+	case SND_ST_UNIPERIF_TYPE_PCM:
 		ret = uni_player_prepare_pcm(player, runtime);
 		break;
-	case SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF:
+	case SND_ST_UNIPERIF_TYPE_SPDIF:
 		ret = uni_player_prepare_iec958(player, runtime);
 		break;
+	case SND_ST_UNIPERIF_TYPE_TDM:
+		ret = uni_player_prepare_tdm(player, runtime);
+		break;
 	default:
 		dev_err(player->dev, "invalid player type");
 		return -EINVAL;
@@ -852,8 +911,8 @@ static int uni_player_start(struct uniperif *player)
 	 * will not take affect and hang the player.
 	 */
 	if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
-		if (UNIPERIF_PLAYER_TYPE_IS_IEC958(player))
-				SET_UNIPERIF_CTRL_SPDIF_FMT_ON(player);
+		if (UNIPERIF_TYPE_IS_IEC958(player))
+			SET_UNIPERIF_CTRL_SPDIF_FMT_ON(player);
 
 	/* Force channel status update (no update if clk disable) */
 	if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
@@ -954,27 +1013,30 @@ static void uni_player_shutdown(struct snd_pcm_substream *substream,
 	player->substream = NULL;
 }
 
-static int uni_player_parse_dt_clk_glue(struct platform_device *pdev,
-					struct uniperif *player)
+static int uni_player_parse_dt_audio_glue(struct platform_device *pdev,
+					  struct uniperif *player)
 {
-	int bit_offset;
 	struct device_node *node = pdev->dev.of_node;
 	struct regmap *regmap;
-
-	bit_offset = SYS_CFG_AUDI0_GLUE_PCM_CLKX + player->info->id;
+	struct reg_field regfield[2] = {
+		/* PCM_CLK_SEL */
+		REG_FIELD(SYS_CFG_AUDIO_GLUE,
+			  8 + player->info->id,
+			  8 + player->info->id),
+		/* PCMP_VALID_SEL */
+		REG_FIELD(SYS_CFG_AUDIO_GLUE, 0, 1)
+	};
 
 	regmap = syscon_regmap_lookup_by_phandle(node, "st,syscfg");
 
-	if (regmap) {
-		struct reg_field regfield =
-			REG_FIELD(SYS_CFG_AUDIO_GLUE, bit_offset, bit_offset);
-
-		player->clk_sel = regmap_field_alloc(regmap, regfield);
-	} else {
+	if (!regmap) {
 		dev_err(&pdev->dev, "sti-audio-clk-glue syscf not found\n");
 		return -EINVAL;
 	}
 
+	player->clk_sel = regmap_field_alloc(regmap, regfield[0]);
+	player->valid_sel = regmap_field_alloc(regmap, regfield[1]);
+
 	return 0;
 }
 
@@ -1012,19 +1074,21 @@ static int uni_player_parse_dt(struct platform_device *pdev,
 	}
 
 	if (strcasecmp(mode, "hdmi") == 0)
-		info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI;
+		info->type = SND_ST_UNIPERIF_TYPE_HDMI;
 	else if (strcasecmp(mode, "pcm") == 0)
-		info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_PCM;
+		info->type = SND_ST_UNIPERIF_TYPE_PCM;
 	else if (strcasecmp(mode, "spdif") == 0)
-		info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF;
+		info->type = SND_ST_UNIPERIF_TYPE_SPDIF;
+	else if (strcasecmp(mode, "tdm") == 0)
+		info->type = SND_ST_UNIPERIF_TYPE_TDM;
 	else
-		info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_NONE;
+		info->type = SND_ST_UNIPERIF_TYPE_NONE;
 
 	/* Save the info structure */
 	player->info = info;
 
-	/* Get the PCM_CLK_SEL bit from audio-glue-ctrl SoC register */
-	if (uni_player_parse_dt_clk_glue(pdev, player))
+	/* Get PCM_CLK_SEL & PCMP_VALID_SEL from audio-glue-ctrl SoC reg */
+	if (uni_player_parse_dt_audio_glue(pdev, player))
 		return -EINVAL;
 
 	return 0;
@@ -1037,7 +1101,8 @@ static const struct snd_soc_dai_ops uni_player_dai_ops = {
 		.trigger = uni_player_trigger,
 		.hw_params = sti_uniperiph_dai_hw_params,
 		.set_fmt = sti_uniperiph_dai_set_fmt,
-		.set_sysclk = uni_player_set_sysclk
+		.set_sysclk = uni_player_set_sysclk,
+		.set_tdm_slot = sti_uniperiph_set_tdm_slot
 };
 
 int uni_player_init(struct platform_device *pdev,
@@ -1047,7 +1112,6 @@ int uni_player_init(struct platform_device *pdev,
 
 	player->dev = &pdev->dev;
 	player->state = UNIPERIF_STATE_STOPPED;
-	player->hw = &uni_player_pcm_hw;
 	player->dai_ops = &uni_player_dai_ops;
 
 	ret = uni_player_parse_dt(pdev, player);
@@ -1057,6 +1121,11 @@ int uni_player_init(struct platform_device *pdev,
 		return ret;
 	}
 
+	if (UNIPERIF_TYPE_IS_TDM(player))
+		player->hw = &uni_tdm_hw;
+	else
+		player->hw = &uni_player_pcm_hw;
+
 	/* Get uniperif resource */
 	player->clk = of_clk_get(pdev->dev.of_node, 0);
 	if (IS_ERR(player->clk))
@@ -1073,6 +1142,17 @@ int uni_player_init(struct platform_device *pdev,
 		}
 	}
 
+	/* connect to I2S/TDM TX bus */
+	if (player->valid_sel &&
+	    (player->info->id == UNIPERIF_PLAYER_I2S_OUT)) {
+		ret = regmap_field_write(player->valid_sel, player->info->id);
+		if (ret) {
+			dev_err(player->dev,
+				"%s: unable to connect to tdm bus", __func__);
+			return ret;
+		}
+	}
+
 	ret = devm_request_irq(&pdev->dev, player->irq,
 			       uni_player_irq_handler, IRQF_SHARED,
 			       dev_name(&pdev->dev), player);
@@ -1087,7 +1167,7 @@ int uni_player_init(struct platform_device *pdev,
 	SET_UNIPERIF_CTRL_SPDIF_LAT_OFF(player);
 	SET_UNIPERIF_CONFIG_IDLE_MOD_DISABLE(player);
 
-	if (UNIPERIF_PLAYER_TYPE_IS_IEC958(player)) {
+	if (UNIPERIF_TYPE_IS_IEC958(player)) {
 		/* Set default iec958 status bits  */
 
 		/* Consumer, PCM, copyright, 2ch, mode 0 */
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index 8a0eb2050..eb74a328c 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -73,55 +73,10 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id)
 	return ret;
 }
 
-static int uni_reader_prepare(struct snd_pcm_substream *substream,
-			      struct snd_soc_dai *dai)
+static int uni_reader_prepare_pcm(struct snd_pcm_runtime *runtime,
+				  struct uniperif *reader)
 {
-	struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai);
-	struct uniperif *reader = priv->dai_data.uni;
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	int transfer_size, trigger_limit;
 	int slot_width;
-	int count = 10;
-
-	/* The reader should be stopped */
-	if (reader->state != UNIPERIF_STATE_STOPPED) {
-		dev_err(reader->dev, "%s: invalid reader state %d", __func__,
-			reader->state);
-		return -EINVAL;
-	}
-
-	/* Calculate transfer size (in fifo cells and bytes) for frame count */
-	transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES;
-
-	/* Calculate number of empty cells available before asserting DREQ */
-	if (reader->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
-		trigger_limit = UNIPERIF_FIFO_SIZE - transfer_size;
-	else
-		/*
-		 * Since SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0
-		 * FDMA_TRIGGER_LIMIT also controls when the state switches
-		 * from OFF or STANDBY to AUDIO DATA.
-		 */
-		trigger_limit = transfer_size;
-
-	/* Trigger limit must be an even number */
-	if ((!trigger_limit % 2) ||
-	    (trigger_limit != 1 && transfer_size % 2) ||
-	    (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(reader))) {
-		dev_err(reader->dev, "invalid trigger limit %d", trigger_limit);
-		return -EINVAL;
-	}
-
-	SET_UNIPERIF_CONFIG_DMA_TRIG_LIMIT(reader, trigger_limit);
-
-	switch (reader->daifmt & SND_SOC_DAIFMT_INV_MASK) {
-	case SND_SOC_DAIFMT_IB_IF:
-	case SND_SOC_DAIFMT_NB_IF:
-		SET_UNIPERIF_I2S_FMT_LR_POL_HIG(reader);
-		break;
-	default:
-		SET_UNIPERIF_I2S_FMT_LR_POL_LOW(reader);
-	}
 
 	/* Force slot width to 32 in I2S mode */
 	if ((reader->daifmt & SND_SOC_DAIFMT_FORMAT_MASK)
@@ -173,6 +128,109 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
+	/* Number of channels must be even */
+	if ((runtime->channels % 2) || (runtime->channels < 2) ||
+	    (runtime->channels > 10)) {
+		dev_err(reader->dev, "%s: invalid nb of channels", __func__);
+		return -EINVAL;
+	}
+
+	SET_UNIPERIF_I2S_FMT_NUM_CH(reader, runtime->channels / 2);
+	SET_UNIPERIF_I2S_FMT_ORDER_MSB(reader);
+
+	return 0;
+}
+
+static int uni_reader_prepare_tdm(struct snd_pcm_runtime *runtime,
+				  struct uniperif *reader)
+{
+	int frame_size; /* user tdm frame size in bytes */
+	/* default unip TDM_WORD_POS_X_Y */
+	unsigned int word_pos[4] = {
+		0x04060002, 0x0C0E080A, 0x14161012, 0x1C1E181A};
+
+	frame_size = sti_uniperiph_get_user_frame_size(runtime);
+
+	/* fix 16/0 format */
+	SET_UNIPERIF_CONFIG_MEM_FMT_16_0(reader);
+	SET_UNIPERIF_I2S_FMT_DATA_SIZE_32(reader);
+
+	/* number of words inserted on the TDM line */
+	SET_UNIPERIF_I2S_FMT_NUM_CH(reader, frame_size / 4 / 2);
+
+	SET_UNIPERIF_I2S_FMT_ORDER_MSB(reader);
+	SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(reader);
+	SET_UNIPERIF_TDM_ENABLE_TDM_ENABLE(reader);
+
+	/*
+	 * set the timeslots allocation for words in FIFO
+	 *
+	 * HW bug: (LSB word < MSB word) => this config is not possible
+	 *         So if we want (LSB word < MSB) word, then it shall be
+	 *         handled by user
+	 */
+	sti_uniperiph_get_tdm_word_pos(reader, word_pos);
+	SET_UNIPERIF_TDM_WORD_POS(reader, 1_2, word_pos[WORD_1_2]);
+	SET_UNIPERIF_TDM_WORD_POS(reader, 3_4, word_pos[WORD_3_4]);
+	SET_UNIPERIF_TDM_WORD_POS(reader, 5_6, word_pos[WORD_5_6]);
+	SET_UNIPERIF_TDM_WORD_POS(reader, 7_8, word_pos[WORD_7_8]);
+
+	return 0;
+}
+
+static int uni_reader_prepare(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
+{
+	struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai);
+	struct uniperif *reader = priv->dai_data.uni;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	int transfer_size, trigger_limit, ret;
+	int count = 10;
+
+	/* The reader should be stopped */
+	if (reader->state != UNIPERIF_STATE_STOPPED) {
+		dev_err(reader->dev, "%s: invalid reader state %d", __func__,
+			reader->state);
+		return -EINVAL;
+	}
+
+	/* Calculate transfer size (in fifo cells and bytes) for frame count */
+	if (reader->info->type == SND_ST_UNIPERIF_TYPE_TDM) {
+		/* transfer size = unip frame size (in 32 bits FIFO cell) */
+		transfer_size =
+			sti_uniperiph_get_user_frame_size(runtime) / 4;
+	} else {
+		transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES;
+	}
+
+	/* Calculate number of empty cells available before asserting DREQ */
+	if (reader->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
+		trigger_limit = UNIPERIF_FIFO_SIZE - transfer_size;
+	else
+		/*
+		 * Since SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0
+		 * FDMA_TRIGGER_LIMIT also controls when the state switches
+		 * from OFF or STANDBY to AUDIO DATA.
+		 */
+		trigger_limit = transfer_size;
+
+	/* Trigger limit must be an even number */
+	if ((!trigger_limit % 2) ||
+	    (trigger_limit != 1 && transfer_size % 2) ||
+	    (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(reader))) {
+		dev_err(reader->dev, "invalid trigger limit %d", trigger_limit);
+		return -EINVAL;
+	}
+
+	SET_UNIPERIF_CONFIG_DMA_TRIG_LIMIT(reader, trigger_limit);
+
+	if (UNIPERIF_TYPE_IS_TDM(reader))
+		ret = uni_reader_prepare_tdm(runtime, reader);
+	else
+		ret = uni_reader_prepare_pcm(runtime, reader);
+	if (ret)
+		return ret;
+
 	switch (reader->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(reader);
@@ -191,21 +249,26 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	SET_UNIPERIF_I2S_FMT_ORDER_MSB(reader);
-
-	/* Data clocking (changing) on the rising edge */
-	SET_UNIPERIF_I2S_FMT_SCLK_EDGE_RISING(reader);
-
-	/* Number of channels must be even */
-
-	if ((runtime->channels % 2) || (runtime->channels < 2) ||
-	    (runtime->channels > 10)) {
-		dev_err(reader->dev, "%s: invalid nb of channels", __func__);
-		return -EINVAL;
+	/* Data clocking (changing) on the rising/falling edge */
+	switch (reader->daifmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		SET_UNIPERIF_I2S_FMT_LR_POL_LOW(reader);
+		SET_UNIPERIF_I2S_FMT_SCLK_EDGE_RISING(reader);
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		SET_UNIPERIF_I2S_FMT_LR_POL_HIG(reader);
+		SET_UNIPERIF_I2S_FMT_SCLK_EDGE_RISING(reader);
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		SET_UNIPERIF_I2S_FMT_LR_POL_LOW(reader);
+		SET_UNIPERIF_I2S_FMT_SCLK_EDGE_FALLING(reader);
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		SET_UNIPERIF_I2S_FMT_LR_POL_HIG(reader);
+		SET_UNIPERIF_I2S_FMT_SCLK_EDGE_FALLING(reader);
+		break;
 	}
 
-	SET_UNIPERIF_I2S_FMT_NUM_CH(reader, runtime->channels / 2);
-
 	/* Clear any pending interrupts */
 	SET_UNIPERIF_ITS_BCLR(reader, GET_UNIPERIF_ITS(reader));
 
@@ -293,6 +356,32 @@ static int  uni_reader_trigger(struct snd_pcm_substream *substream,
 	}
 }
 
+static int uni_reader_startup(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
+{
+	struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai);
+	struct uniperif *reader = priv->dai_data.uni;
+	int ret;
+
+	if (!UNIPERIF_TYPE_IS_TDM(reader))
+		return 0;
+
+	/* refine hw constraint in tdm mode */
+	ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+				  SNDRV_PCM_HW_PARAM_CHANNELS,
+				  sti_uniperiph_fix_tdm_chan,
+				  reader, SNDRV_PCM_HW_PARAM_CHANNELS,
+				  -1);
+	if (ret < 0)
+		return ret;
+
+	return snd_pcm_hw_rule_add(substream->runtime, 0,
+				   SNDRV_PCM_HW_PARAM_FORMAT,
+				   sti_uniperiph_fix_tdm_format,
+				   reader, SNDRV_PCM_HW_PARAM_FORMAT,
+				   -1);
+}
+
 static void uni_reader_shutdown(struct snd_pcm_substream *substream,
 				struct snd_soc_dai *dai)
 {
@@ -310,6 +399,7 @@ static int uni_reader_parse_dt(struct platform_device *pdev,
 {
 	struct uniperif_info *info;
 	struct device_node *node = pdev->dev.of_node;
+	const char *mode;
 
 	/* Allocate memory for the info structure */
 	info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL);
@@ -322,6 +412,17 @@ static int uni_reader_parse_dt(struct platform_device *pdev,
 		return -EINVAL;
 	}
 
+	/* Read the device mode property */
+	if (of_property_read_string(node, "st,mode", &mode)) {
+		dev_err(&pdev->dev, "uniperipheral mode not defined");
+		return -EINVAL;
+	}
+
+	if (strcasecmp(mode, "tdm") == 0)
+		info->type = SND_ST_UNIPERIF_TYPE_TDM;
+	else
+		info->type = SND_ST_UNIPERIF_TYPE_PCM;
+
 	/* Save the info structure */
 	reader->info = info;
 
@@ -329,11 +430,13 @@ static int uni_reader_parse_dt(struct platform_device *pdev,
 }
 
 static const struct snd_soc_dai_ops uni_reader_dai_ops = {
+		.startup = uni_reader_startup,
 		.shutdown = uni_reader_shutdown,
 		.prepare = uni_reader_prepare,
 		.trigger = uni_reader_trigger,
 		.hw_params = sti_uniperiph_dai_hw_params,
 		.set_fmt = sti_uniperiph_dai_set_fmt,
+		.set_tdm_slot = sti_uniperiph_set_tdm_slot
 };
 
 int uni_reader_init(struct platform_device *pdev,
@@ -343,7 +446,6 @@ int uni_reader_init(struct platform_device *pdev,
 
 	reader->dev = &pdev->dev;
 	reader->state = UNIPERIF_STATE_STOPPED;
-	reader->hw = &uni_reader_pcm_hw;
 	reader->dai_ops = &uni_reader_dai_ops;
 
 	ret = uni_reader_parse_dt(pdev, reader);
@@ -352,6 +454,11 @@ int uni_reader_init(struct platform_device *pdev,
 		return ret;
 	}
 
+	if (UNIPERIF_TYPE_IS_TDM(reader))
+		reader->hw = &uni_tdm_hw;
+	else
+		reader->hw = &uni_reader_pcm_hw;
+
 	ret = devm_request_irq(&pdev->dev, reader->irq,
 			       uni_reader_irq_handler, IRQF_SHARED,
 			       dev_name(&pdev->dev), reader);
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 2d493501b..9e5276d6d 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -350,6 +350,7 @@ static int snd_usb_audio_create(struct usb_interface *intf,
 	case USB_SPEED_HIGH:
 	case USB_SPEED_WIRELESS:
 	case USB_SPEED_SUPER:
+	case USB_SPEED_SUPER_PLUS:
 		break;
 	default:
 		dev_err(&dev->dev, "unknown device speed %d\n", snd_usb_get_speed(dev));
@@ -450,6 +451,9 @@ static int snd_usb_audio_create(struct usb_interface *intf,
 	case USB_SPEED_SUPER:
 		strlcat(card->longname, ", super speed", sizeof(card->longname));
 		break;
+	case USB_SPEED_SUPER_PLUS:
+		strlcat(card->longname, ", super speed plus", sizeof(card->longname));
+		break;
 	default:
 		break;
 	}
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 7ccbcaf6a..26dd5f20f 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -309,6 +309,9 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface,
 	 * support reading */
 	if (snd_usb_get_sample_rate_quirk(chip))
 		return 0;
+	/* the firmware is likely buggy, don't repeat to fail too many times */
+	if (chip->sample_rate_read_error > 2)
+		return 0;
 
 	if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR,
 				   USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN,
@@ -316,6 +319,7 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface,
 				   data, sizeof(data))) < 0) {
 		dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n",
 			iface, fmt->altsetting, ep);
+		chip->sample_rate_read_error++;
 		return 0; /* some devices don't support reading */
 	}
 
diff --git a/sound/usb/helper.c b/sound/usb/helper.c
index 51ed1ac82..7712e2b84 100644
--- a/sound/usb/helper.c
+++ b/sound/usb/helper.c
@@ -120,6 +120,7 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip,
 	case USB_SPEED_HIGH:
 	case USB_SPEED_WIRELESS:
 	case USB_SPEED_SUPER:
+	case USB_SPEED_SUPER_PLUS:
 		if (get_endpoint(alts, 0)->bInterval >= 1 &&
 		    get_endpoint(alts, 0)->bInterval <= 4)
 			return get_endpoint(alts, 0)->bInterval - 1;
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 47de8af42..7ba92921b 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -911,6 +911,7 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep,
 	switch (snd_usb_get_speed(ep->umidi->dev)) {
 	case USB_SPEED_HIGH:
 	case USB_SPEED_SUPER:
+	case USB_SPEED_SUPER_PLUS:
 		count = 1;
 		break;
 	default:
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 4f8575700..2f8c388ef 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -45,6 +45,7 @@
 #include <linux/bitops.h>
 #include <linux/init.h>
 #include <linux/list.h>
+#include <linux/log2.h>
 #include <linux/slab.h>
 #include <linux/string.h>
 #include <linux/usb.h>
@@ -1378,6 +1379,71 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 	snd_usb_mixer_add_control(&cval->head, kctl);
 }
 
+static int parse_clock_source_unit(struct mixer_build *state, int unitid,
+				   void *_ftr)
+{
+	struct uac_clock_source_descriptor *hdr = _ftr;
+	struct usb_mixer_elem_info *cval;
+	struct snd_kcontrol *kctl;
+	char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+	int ret;
+
+	if (state->mixer->protocol != UAC_VERSION_2)
+		return -EINVAL;
+
+	if (hdr->bLength != sizeof(*hdr)) {
+		usb_audio_dbg(state->chip,
+			      "Bogus clock source descriptor length of %d, ignoring.\n",
+			      hdr->bLength);
+		return 0;
+	}
+
+	/*
+	 * The only property of this unit we are interested in is the
+	 * clock source validity. If that isn't readable, just bail out.
+	 */
+	if (!uac2_control_is_readable(hdr->bmControls,
+				      ilog2(UAC2_CS_CONTROL_CLOCK_VALID)))
+		return 0;
+
+	cval = kzalloc(sizeof(*cval), GFP_KERNEL);
+	if (!cval)
+		return -ENOMEM;
+
+	snd_usb_mixer_elem_init_std(&cval->head, state->mixer, hdr->bClockID);
+
+	cval->min = 0;
+	cval->max = 1;
+	cval->channels = 1;
+	cval->val_type = USB_MIXER_BOOLEAN;
+	cval->control = UAC2_CS_CONTROL_CLOCK_VALID;
+
+	if (uac2_control_is_writeable(hdr->bmControls,
+				      ilog2(UAC2_CS_CONTROL_CLOCK_VALID)))
+		kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval);
+	else {
+		cval->master_readonly = 1;
+		kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval);
+	}
+
+	if (!kctl) {
+		kfree(cval);
+		return -ENOMEM;
+	}
+
+	kctl->private_free = snd_usb_mixer_elem_free;
+	ret = snd_usb_copy_string_desc(state, hdr->iClockSource,
+				       name, sizeof(name));
+	if (ret > 0)
+		snprintf(kctl->id.name, sizeof(kctl->id.name),
+			 "%s Validity", name);
+	else
+		snprintf(kctl->id.name, sizeof(kctl->id.name),
+			 "Clock Source %d Validity", hdr->bClockID);
+
+	return snd_usb_mixer_add_control(&cval->head, kctl);
+}
+
 /*
  * parse a feature unit
  *
@@ -2126,10 +2192,11 @@ static int parse_audio_unit(struct mixer_build *state, int unitid)
 
 	switch (p1[2]) {
 	case UAC_INPUT_TERMINAL:
-	case UAC2_CLOCK_SOURCE:
 		return 0; /* NOP */
 	case UAC_MIXER_UNIT:
 		return parse_audio_mixer_unit(state, unitid, p1);
+	case UAC2_CLOCK_SOURCE:
+		return parse_clock_source_unit(state, unitid, p1);
 	case UAC_SELECTOR_UNIT:
 	case UAC2_CLOCK_SELECTOR:
 		return parse_audio_selector_unit(state, unitid, p1);
@@ -2307,6 +2374,7 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer,
 	__u8 unitid = (index >> 8) & 0xff;
 	__u8 control = (value >> 8) & 0xff;
 	__u8 channel = value & 0xff;
+	unsigned int count = 0;
 
 	if (channel >= MAX_CHANNELS) {
 		usb_audio_dbg(mixer->chip,
@@ -2315,6 +2383,12 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer,
 		return;
 	}
 
+	for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem)
+		count++;
+
+	if (count == 0)
+		return;
+
 	for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) {
 		struct usb_mixer_elem_info *info;
 
@@ -2322,7 +2396,7 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer,
 			continue;
 
 		info = (struct usb_mixer_elem_info *)list;
-		if (info->control != control)
+		if (count > 1 && info->control != control)
 			continue;
 
 		switch (attribute) {
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index b665d8555..4d5c89a7b 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -47,6 +47,7 @@ struct snd_usb_audio {
 	
 	int num_interfaces;
 	int num_suspended_intf;
+	int sample_rate_read_error;
 
 	struct list_head pcm_list;	/* list of pcm streams */
 	struct list_head ep_list;	/* list of audio-related endpoints */