Merge tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There have been little changes in ALSA core stuff, but ASoC core still
  kept rolling for the continued restructuring. The rest are lots of
  small driver-specific changes and some minor API updates. Here are
  highlights:

  General:
  - Appropriate fall-through annotations everywhere
  - Some code cleanup in memalloc code, handling non-cacahed pages more
    commonly in the helper
  - Deployment of SNDRV_PCM_INFO_SYNC_APPLPTR flag consistently

  Drivers:
  - More HD-audio CA0132 codec improvement for supporting other Creative
    boards
  - Plumbing legacy HD-audio codecs as ASoC BE on Intel SST; this will
    give move support of existing HD-audio devices with DSP
  - A few device-specific HD-audio quirks as usual
  - New quirk for RME CC devices and correction for B&W PX for USB-audio
  - FireWire: code refactoring including devres usages

  ASoC Core:
  - Continued componentization works; it's almost done!
  - A bunch of new for_each_foo macros
  - Cleanups and fixes in DAPM code

  ASoC Drivers:
  - MCLK support for several different devices, including CS42L51, STM32
    SAI, and MAX98373
  - Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
    MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
    TI PCM3060"

* tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (299 commits)
  ASoC: stm32: sai: fix master clock naming
  ASoC: stm32: add clock dependency for sai
  ALSA: hda/ca0132 - Actually fix microphone issue
  ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks
  ASoC: wm2000: Remove wm2000_read helper function
  ASoC: cs42l51: fix mclk support
  ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate
  ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer
  ASoC: cs42l51: add mclk support
  ASoC: stm32: sai: set sai as mclk clock provider
  ASoC: dt-bindings: add mclk support to cs42l51
  ASoC: dt-bindings: add mclk provider support to stm32 sai
  ASoC: soc-core: fix trivial checkpatch issues
  ASoC: dapm: Add support for hw_free on CODEC to CODEC links
  ASoC: Intel: kbl_da7219_max98927: minor white space clean up
  ALSA: i2c/cs8427: Fix int to char conversion
  ALSA: doc: Brush up the old writing-an-alsa-driver
  ASoC: rsnd: tidyup SSICR::SWSP for TDM
  ASoC: rsnd: enable TDM settings for SSI parent
  ASoC: pcm3168a: add hw constraint for capture channel
  ...

216 files changed, 10302 insertions, 2971 deletions

diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c
index 000b58522106..bd7c5029fc59 100644
--- a/sound/aoa/soundbus/i2sbus/core.c
+++ b/sound/aoa/soundbus/i2sbus/core.c
@@ -157,18 +157,19 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 	struct device_node *child = NULL, *sound = NULL;
 	struct resource *r;
 	int i, layout = 0, rlen, ok = force;
-	static const char *rnames[] = { "i2sbus: %s (control)",
-					"i2sbus: %s (tx)",
-					"i2sbus: %s (rx)" };
+	char node_name[6];
+	static const char *rnames[] = { "i2sbus: %pOFn (control)",
+					"i2sbus: %pOFn (tx)",
+					"i2sbus: %pOFn (rx)" };
 	static irq_handler_t ints[] = {
 		i2sbus_bus_intr,
 		i2sbus_tx_intr,
 		i2sbus_rx_intr
 	};
 
-	if (strlen(np->name) != 5)
+	if (snprintf(node_name, sizeof(node_name), "%pOFn", np) != 5)
 		return 0;
-	if (strncmp(np->name, "i2s-", 4))
+	if (strncmp(node_name, "i2s-", 4))
 		return 0;
 
 	dev = kzalloc(sizeof(struct i2sbus_dev), GFP_KERNEL);
@@ -228,13 +229,13 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 	dev->sound.pcmid = -1;
 	dev->macio = macio;
 	dev->control = control;
-	dev->bus_number = np->name[4] - 'a';
+	dev->bus_number = node_name[4] - 'a';
 	INIT_LIST_HEAD(&dev->sound.codec_list);
 
 	for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) {
 		dev->interrupts[i] = -1;
 		snprintf(dev->rnames[i], sizeof(dev->rnames[i]),
-			 rnames[i], np->name);
+			 rnames[i], np);
 	}
 	for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) {
 		int irq = irq_of_parse_and_map(np, i);
diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index 5fbd47a9177e..28867732a318 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -31,7 +31,6 @@ endif	# SND_ARM
 
 config SND_PXA2XX_LIB
 	tristate
-	select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
 	select SND_DMAENGINE_PCM
 
 config SND_PXA2XX_LIB_AC97
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 753d5fc4b284..59a4adc286ed 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -25,6 +25,9 @@
 #include <linux/mm.h>
 #include <linux/dma-mapping.h>
 #include <linux/genalloc.h>
+#ifdef CONFIG_X86
+#include <asm/set_memory.h>
+#endif
 #include <sound/memalloc.h>
 
 /*
@@ -82,31 +85,32 @@ EXPORT_SYMBOL(snd_free_pages);
 
 #ifdef CONFIG_HAS_DMA
 /* allocate the coherent DMA pages */
-static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma)
+static void snd_malloc_dev_pages(struct snd_dma_buffer *dmab, size_t size)
 {
-	int pg;
 	gfp_t gfp_flags;
 
-	if (WARN_ON(!dma))
-		return NULL;
-	pg = get_order(size);
 	gfp_flags = GFP_KERNEL
 		| __GFP_COMP	/* compound page lets parts be mapped */
 		| __GFP_NORETRY /* don't trigger OOM-killer */
 		| __GFP_NOWARN; /* no stack trace print - this call is non-critical */
-	return dma_alloc_coherent(dev, PAGE_SIZE << pg, dma, gfp_flags);
+	dmab->area = dma_alloc_coherent(dmab->dev.dev, size, &dmab->addr,
+					gfp_flags);
+#ifdef CONFIG_X86
+	if (dmab->area && dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC)
+		set_memory_wc((unsigned long)dmab->area,
+			      PAGE_ALIGN(size) >> PAGE_SHIFT);
+#endif
 }
 
 /* free the coherent DMA pages */
-static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr,
-			       dma_addr_t dma)
+static void snd_free_dev_pages(struct snd_dma_buffer *dmab)
 {
-	int pg;
-
-	if (ptr == NULL)
-		return;
-	pg = get_order(size);
-	dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma);
+#ifdef CONFIG_X86
+	if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC)
+		set_memory_wb((unsigned long)dmab->area,
+			      PAGE_ALIGN(dmab->bytes) >> PAGE_SHIFT);
+#endif
+	dma_free_coherent(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr);
 }
 
 #ifdef CONFIG_GENERIC_ALLOCATOR
@@ -199,12 +203,15 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
 		 */
 		dmab->dev.type = SNDRV_DMA_TYPE_DEV;
 #endif /* CONFIG_GENERIC_ALLOCATOR */
+		/* fall through */
 	case SNDRV_DMA_TYPE_DEV:
-		dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr);
+	case SNDRV_DMA_TYPE_DEV_UC:
+		snd_malloc_dev_pages(dmab, size);
 		break;
 #endif
 #ifdef CONFIG_SND_DMA_SGBUF
 	case SNDRV_DMA_TYPE_DEV_SG:
+	case SNDRV_DMA_TYPE_DEV_UC_SG:
 		snd_malloc_sgbuf_pages(device, size, dmab, NULL);
 		break;
 #endif
@@ -275,11 +282,13 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab)
 		break;
 #endif /* CONFIG_GENERIC_ALLOCATOR */
 	case SNDRV_DMA_TYPE_DEV:
-		snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr);
+	case SNDRV_DMA_TYPE_DEV_UC:
+		snd_free_dev_pages(dmab);
 		break;
 #endif
 #ifdef CONFIG_SND_DMA_SGBUF
 	case SNDRV_DMA_TYPE_DEV_SG:
+	case SNDRV_DMA_TYPE_DEV_UC_SG:
 		snd_free_sgbuf_pages(dmab);
 		break;
 #endif
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index 0391cb1a4f19..141c5f3a9575 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -111,7 +111,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames)
 		while (plugin->next) {
 			if (plugin->dst_frames)
 				frames = plugin->dst_frames(plugin, frames);
-			if (snd_BUG_ON(frames <= 0))
+			if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0))
 				return -ENXIO;
 			plugin = plugin->next;
 			err = snd_pcm_plugin_alloc(plugin, frames);
@@ -123,7 +123,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames)
 		while (plugin->prev) {
 			if (plugin->src_frames)
 				frames = plugin->src_frames(plugin, frames);
-			if (snd_BUG_ON(frames <= 0))
+			if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0))
 				return -ENXIO;
 			plugin = plugin->prev;
 			err = snd_pcm_plugin_alloc(plugin, frames);
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 4e6110d778bd..40013b26f671 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -2172,18 +2172,25 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream,
 	if (err < 0)
 		goto _end_unlock;
 
+	runtime->twake = runtime->control->avail_min ? : 1;
+	if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
+		snd_pcm_update_hw_ptr(substream);
+
 	if (!is_playback &&
-	    runtime->status->state == SNDRV_PCM_STATE_PREPARED &&
-	    size >= runtime->start_threshold) {
-		err = snd_pcm_start(substream);
-		if (err < 0)
+	    runtime->status->state == SNDRV_PCM_STATE_PREPARED) {
+		if (size >= runtime->start_threshold) {
+			err = snd_pcm_start(substream);
+			if (err < 0)
+				goto _end_unlock;
+		} else {
+			/* nothing to do */
+			err = 0;
 			goto _end_unlock;
+		}
 	}
 
-	runtime->twake = runtime->control->avail_min ? : 1;
-	if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
-		snd_pcm_update_hw_ptr(substream);
 	avail = snd_pcm_avail(substream);
+
 	while (size > 0) {
 		snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
 		snd_pcm_uframes_t cont;
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index 08d5662039e3..ee601d7f0926 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -1236,6 +1236,28 @@ int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream,
 }
 EXPORT_SYMBOL(snd_rawmidi_transmit);
 
+/**
+ * snd_rawmidi_proceed - Discard the all pending bytes and proceed
+ * @substream: rawmidi substream
+ *
+ * Return: the number of discarded bytes
+ */
+int snd_rawmidi_proceed(struct snd_rawmidi_substream *substream)
+{
+	struct snd_rawmidi_runtime *runtime = substream->runtime;
+	unsigned long flags;
+	int count = 0;
+
+	spin_lock_irqsave(&runtime->lock, flags);
+	if (runtime->avail < runtime->buffer_size) {
+		count = runtime->buffer_size - runtime->avail;
+		__snd_rawmidi_transmit_ack(substream, count);
+	}
+	spin_unlock_irqrestore(&runtime->lock, flags);
+	return count;
+}
+EXPORT_SYMBOL(snd_rawmidi_proceed);
+
 static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream,
 				      const unsigned char __user *userbuf,
 				      const unsigned char *kernelbuf,
diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c
index ba127c22539a..0778d28421da 100644
--- a/sound/core/seq/oss/seq_oss_timer.c
+++ b/sound/core/seq/oss/seq_oss_timer.c
@@ -92,7 +92,7 @@ snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *ev)
 		case TMR_WAIT_REL:
 			parm += rec->cur_tick;
 			rec->realtime = 0;
-			/* fall through and continue to next */
+			/* fall through */
 		case TMR_WAIT_ABS:
 			if (parm == 0) {
 				rec->realtime = 1;
diff --git a/sound/core/seq/seq_system.c b/sound/core/seq/seq_system.c
index 8ce1d0b40dce..0dc5d5a45ecc 100644
--- a/sound/core/seq/seq_system.c
+++ b/sound/core/seq/seq_system.c
@@ -123,6 +123,7 @@ int __init snd_seq_system_client_init(void)
 {
 	struct snd_seq_port_callback pcallbacks;
 	struct snd_seq_port_info *port;
+	int err;
 
 	port = kzalloc(sizeof(*port), GFP_KERNEL);
 	if (!port)
@@ -134,6 +135,10 @@ int __init snd_seq_system_client_init(void)
 
 	/* register client */
 	sysclient = snd_seq_create_kernel_client(NULL, 0, "System");
+	if (sysclient < 0) {
+		kfree(port);
+		return sysclient;
+	}
 
 	/* register timer */
 	strcpy(port->name, "Timer");
@@ -144,7 +149,10 @@ int __init snd_seq_system_client_init(void)
 	port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT;
 	port->addr.client = sysclient;
 	port->addr.port = SNDRV_SEQ_PORT_SYSTEM_TIMER;
-	snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port);
+	err = snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT,
+					port);
+	if (err < 0)
+		goto error_port;
 
 	/* register announcement port */
 	strcpy(port->name, "Announce");
@@ -154,16 +162,24 @@ int __init snd_seq_system_client_init(void)
 	port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT;
 	port->addr.client = sysclient;
 	port->addr.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE;
-	snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port);
+	err = snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT,
+					port);
+	if (err < 0)
+		goto error_port;
 	announce_port = port->addr.port;
 
 	kfree(port);
 	return 0;
+
+ error_port:
+	snd_seq_system_client_done();
+	kfree(port);
+	return err;
 }
 
 
 /* unregister our internal client */
-void __exit snd_seq_system_client_done(void)
+void snd_seq_system_client_done(void)
 {
 	int oldsysclient = sysclient;
 
diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c
index cb988efd1ed0..e5a40795914a 100644
--- a/sound/core/seq/seq_virmidi.c
+++ b/sound/core/seq/seq_virmidi.c
@@ -149,9 +149,7 @@ static void snd_vmidi_output_work(struct work_struct *work)
 	/* discard the outputs in dispatch mode unless subscribed */
 	if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH &&
 	    !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) {
-		char buf[32];
-		while (snd_rawmidi_transmit(substream, buf, sizeof(buf)) > 0)
-			; /* ignored */
+		snd_rawmidi_proceed(substream);
 		return;
 	}
 
diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c
index 84fffabdd129..c1cfaa01a5cb 100644
--- a/sound/core/sgbuf.c
+++ b/sound/core/sgbuf.c
@@ -23,6 +23,7 @@
 #include <linux/mm.h>
 #include <linux/vmalloc.h>
 #include <linux/export.h>
+#include <asm/pgtable.h>
 #include <sound/memalloc.h>
 
 
@@ -43,6 +44,8 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab)
 	dmab->area = NULL;
 
 	tmpb.dev.type = SNDRV_DMA_TYPE_DEV;
+	if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC_SG)
+		tmpb.dev.type = SNDRV_DMA_TYPE_DEV_UC;
 	tmpb.dev.dev = sgbuf->dev;
 	for (i = 0; i < sgbuf->pages; i++) {
 		if (!(sgbuf->table[i].addr & ~PAGE_MASK))
@@ -72,12 +75,20 @@ void *snd_malloc_sgbuf_pages(struct device *device,
 	struct snd_dma_buffer tmpb;
 	struct snd_sg_page *table;
 	struct page **pgtable;
+	int type = SNDRV_DMA_TYPE_DEV;
+	pgprot_t prot = PAGE_KERNEL;
 
 	dmab->area = NULL;
 	dmab->addr = 0;
 	dmab->private_data = sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL);
 	if (! sgbuf)
 		return NULL;
+	if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_UC_SG) {
+		type = SNDRV_DMA_TYPE_DEV_UC;
+#ifdef pgprot_noncached
+		prot = pgprot_noncached(PAGE_KERNEL);
+#endif
+	}
 	sgbuf->dev = device;
 	pages = snd_sgbuf_aligned_pages(size);
 	sgbuf->tblsize = sgbuf_align_table(pages);
@@ -98,7 +109,7 @@ void *snd_malloc_sgbuf_pages(struct device *device,
 		if (chunk > maxpages)
 			chunk = maxpages;
 		chunk <<= PAGE_SHIFT;
-		if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, device,
+		if (snd_dma_alloc_pages_fallback(type, device,
 						 chunk, &tmpb) < 0) {
 			if (!sgbuf->pages)
 				goto _failed;
@@ -125,7 +136,7 @@ void *snd_malloc_sgbuf_pages(struct device *device,
 	}
 
 	sgbuf->size = size;
-	dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, PAGE_KERNEL);
+	dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, prot);
 	if (! dmab->area)
 		goto _failed;
 	if (res_size)
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index 529d9f405fa9..8a146b039276 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -147,7 +147,9 @@ config SND_FIREWIRE_MOTU
 	help
 	 Say Y here to enable support for FireWire devices which MOTU produced:
 	  * 828mk2
+	  * Traveler
 	  * 828mk3
+	  * Audio Express
 
 	 To compile this driver as a module, choose M here: the module
 	 will be called snd-firewire-motu.
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index cb9acfe60f6a..fcd965f1d69e 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -140,6 +140,59 @@ const unsigned int amdtp_rate_table[CIP_SFC_COUNT] = {
 };
 EXPORT_SYMBOL(amdtp_rate_table);
 
+static int apply_constraint_to_size(struct snd_pcm_hw_params *params,
+				    struct snd_pcm_hw_rule *rule)
+{
+	struct snd_interval *s = hw_param_interval(params, rule->var);
+	const struct snd_interval *r =
+		hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval t = {
+		.min = s->min, .max = s->max, .integer = 1,
+	};
+	int i;
+
+	for (i = 0; i < CIP_SFC_COUNT; ++i) {
+		unsigned int rate = amdtp_rate_table[i];
+		unsigned int step = amdtp_syt_intervals[i];
+
+		if (!snd_interval_test(r, rate))
+			continue;
+
+		t.min = roundup(t.min, step);
+		t.max = rounddown(t.max, step);
+	}
+
+	if (snd_interval_checkempty(&t))
+		return -EINVAL;
+
+	return snd_interval_refine(s, &t);
+}
+
+static int apply_constraint_to_rate(struct snd_pcm_hw_params *params,
+				    struct snd_pcm_hw_rule *rule)
+{
+	struct snd_interval *r =
+			hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	const struct snd_interval *s = hw_param_interval_c(params, rule->deps[0]);
+	struct snd_interval t = {
+		.min = UINT_MAX, .max = 0, .integer = 1,
+	};
+	int i;
+
+	for (i = 0; i < CIP_SFC_COUNT; ++i) {
+		unsigned int step = amdtp_syt_intervals[i];
+		unsigned int rate = amdtp_rate_table[i];
+
+		if (s->min % step || s->max % step)
+			continue;
+
+		t.min = min(t.min, rate);
+		t.max = max(t.max, rate);
+	}
+
+	return snd_interval_refine(r, &t);
+}
+
 /**
  * amdtp_stream_add_pcm_hw_constraints - add hw constraints for PCM substream
  * @s:		the AMDTP stream, which must be initialized.
@@ -194,16 +247,27 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
 	 * number equals to SYT_INTERVAL. So the number is 8, 16 or 32,
 	 * depending on its sampling rate. For accurate period interrupt, it's
 	 * preferrable to align period/buffer sizes to current SYT_INTERVAL.
-	 *
-	 * TODO: These constraints can be improved with proper rules.
-	 * Currently apply LCM of SYT_INTERVALs.
 	 */
-	err = snd_pcm_hw_constraint_step(runtime, 0,
-					 SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32);
+	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+				  apply_constraint_to_size, NULL,
+				  SNDRV_PCM_HW_PARAM_RATE, -1);
+	if (err < 0)
+		goto end;
+	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				  apply_constraint_to_rate, NULL,
+				  SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1);
+	if (err < 0)
+		goto end;
+	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+				  apply_constraint_to_size, NULL,
+				  SNDRV_PCM_HW_PARAM_RATE, -1);
+	if (err < 0)
+		goto end;
+	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				  apply_constraint_to_rate, NULL,
+				  SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1);
 	if (err < 0)
 		goto end;
-	err = snd_pcm_hw_constraint_step(runtime, 0,
-					 SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32);
 end:
 	return err;
 }
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 93676354f87f..672d13488454 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -126,23 +126,6 @@ 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
- * transactions safely. Thus at returning from .remove(), this module still keep
- * references for the unit.
- */
 static void
 bebob_card_free(struct snd_card *card)
 {
@@ -152,7 +135,7 @@ bebob_card_free(struct snd_card *card)
 	clear_bit(bebob->card_index, devices_used);
 	mutex_unlock(&devices_mutex);
 
-	bebob_free(card->private_data);
+	snd_bebob_stream_destroy_duplex(bebob);
 }
 
 static const struct snd_bebob_spec *
@@ -192,7 +175,6 @@ do_registration(struct work_struct *work)
 		return;
 
 	mutex_lock(&devices_mutex);
-
 	for (card_index = 0; card_index < SNDRV_CARDS; card_index++) {
 		if (!test_bit(card_index, devices_used) && enable[card_index])
 			break;
@@ -208,6 +190,11 @@ do_registration(struct work_struct *work)
 		mutex_unlock(&devices_mutex);
 		return;
 	}
+	set_bit(card_index, devices_used);
+	mutex_unlock(&devices_mutex);
+
+	bebob->card->private_free = bebob_card_free;
+	bebob->card->private_data = bebob;
 
 	err = name_device(bebob);
 	if (err < 0)
@@ -248,23 +235,10 @@ do_registration(struct work_struct *work)
 	if (err < 0)
 		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);
-	kfree(bebob->maudio_special_quirk);
-	bebob->maudio_special_quirk = NULL;
 	snd_card_free(bebob->card);
 	dev_info(&bebob->unit->device,
 		 "Sound card registration failed: %d\n", err);
@@ -295,15 +269,15 @@ bebob_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry)
 	}
 
 	/* Allocate this independent of sound card instance. */
-	bebob = kzalloc(sizeof(struct snd_bebob), GFP_KERNEL);
-	if (bebob == NULL)
+	bebob = devm_kzalloc(&unit->device, sizeof(struct snd_bebob),
+			     GFP_KERNEL);
+	if (!bebob)
 		return -ENOMEM;
-
 	bebob->unit = fw_unit_get(unit);
-	bebob->entry = entry;
-	bebob->spec = spec;
 	dev_set_drvdata(&unit->device, bebob);
 
+	bebob->entry = entry;
+	bebob->spec = spec;
 	mutex_init(&bebob->mutex);
 	spin_lock_init(&bebob->lock);
 	init_waitqueue_head(&bebob->hwdep_wait);
@@ -379,12 +353,12 @@ static void bebob_remove(struct fw_unit *unit)
 	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);
+		// Block till all of ALSA character devices are released.
+		snd_card_free(bebob->card);
 	}
+
+	mutex_destroy(&bebob->mutex);
+	fw_unit_put(bebob->unit);
 }
 
 static const struct snd_bebob_rate_spec normal_rate_spec = {
diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c
index c266997ad299..51152ca4af57 100644
--- a/sound/firewire/bebob/bebob_maudio.c
+++ b/sound/firewire/bebob/bebob_maudio.c
@@ -261,8 +261,9 @@ snd_bebob_maudio_special_discover(struct snd_bebob *bebob, bool is1814)
 	struct special_params *params;
 	int err;
 
-	params = kzalloc(sizeof(struct special_params), GFP_KERNEL);
-	if (params == NULL)
+	params = devm_kzalloc(&bebob->card->card_dev,
+			      sizeof(struct special_params), GFP_KERNEL);
+	if (!params)
 		return -ENOMEM;
 
 	mutex_lock(&bebob->mutex);
diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 774eb2205668..0f6dbcffe711 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -122,25 +122,12 @@ static void dice_card_strings(struct snd_dice *dice)
 	strcpy(card->mixername, "DICE");
 }
 
-static void dice_free(struct snd_dice *dice)
+static void dice_card_free(struct snd_card *card)
 {
+	struct snd_dice *dice = card->private_data;
+
 	snd_dice_stream_destroy_duplex(dice);
 	snd_dice_transaction_destroy(dice);
-	fw_unit_put(dice->unit);
-
-	mutex_destroy(&dice->mutex);
-	kfree(dice);
-}
-
-/*
- * 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 dice_card_free(struct snd_card *card)
-{
-	dice_free(card->private_data);
 }
 
 static void do_registration(struct work_struct *work)
@@ -155,6 +142,8 @@ static void do_registration(struct work_struct *work)
 			   &dice->card);
 	if (err < 0)
 		return;
+	dice->card->private_free = dice_card_free;
+	dice->card->private_data = dice;
 
 	err = snd_dice_transaction_init(dice);
 	if (err < 0)
@@ -192,19 +181,10 @@ static void do_registration(struct work_struct *work)
 	if (err < 0)
 		goto error;
 
-	/*
-	 * After registered, dice instance can be released corresponding to
-	 * releasing the sound card instance.
-	 */
-	dice->card->private_free = dice_card_free;
-	dice->card->private_data = dice;
 	dice->registered = true;
 
 	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);
@@ -223,10 +203,9 @@ static int dice_probe(struct fw_unit *unit,
 	}
 
 	/* Allocate this independent of sound card instance. */
-	dice = kzalloc(sizeof(struct snd_dice), GFP_KERNEL);
-	if (dice == NULL)
+	dice = devm_kzalloc(&unit->device, sizeof(struct snd_dice), GFP_KERNEL);
+	if (!dice)
 		return -ENOMEM;
-
 	dice->unit = fw_unit_get(unit);
 	dev_set_drvdata(&unit->device, dice);
 
@@ -263,10 +242,10 @@ static void dice_remove(struct fw_unit *unit)
 	if (dice->registered) {
 		/* No need to wait for releasing card object in this context. */
 		snd_card_free_when_closed(dice->card);
-	} else {
-		/* Don't forget this case. */
-		dice_free(dice);
 	}
+
+	mutex_destroy(&dice->mutex);
+	fw_unit_put(dice->unit);
 }
 
 static void dice_bus_reset(struct fw_unit *unit)
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
index ef689997d6a5..6c6ea149ef6b 100644
--- a/sound/firewire/digi00x/digi00x.c
+++ b/sound/firewire/digi00x/digi00x.c
@@ -41,20 +41,12 @@ static int name_card(struct snd_dg00x *dg00x)
 	return 0;
 }
 
-static void dg00x_free(struct snd_dg00x *dg00x)
+static void dg00x_card_free(struct snd_card *card)
 {
+	struct snd_dg00x *dg00x = card->private_data;
+
 	snd_dg00x_stream_destroy_duplex(dg00x);
 	snd_dg00x_transaction_unregister(dg00x);
-
-	fw_unit_put(dg00x->unit);
-
-	mutex_destroy(&dg00x->mutex);
-	kfree(dg00x);
-}
-
-static void dg00x_card_free(struct snd_card *card)
-{
-	dg00x_free(card->private_data);
 }
 
 static void do_registration(struct work_struct *work)
@@ -70,6 +62,8 @@ static void do_registration(struct work_struct *work)
 			   &dg00x->card);
 	if (err < 0)
 		return;
+	dg00x->card->private_free = dg00x_card_free;
+	dg00x->card->private_data = dg00x;
 
 	err = name_card(dg00x);
 	if (err < 0)
@@ -101,14 +95,10 @@ static void do_registration(struct work_struct *work)
 	if (err < 0)
 		goto error;
 
-	dg00x->card->private_free = dg00x_card_free;
-	dg00x->card->private_data = dg00x;
 	dg00x->registered = true;
 
 	return;
 error:
-	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);
@@ -120,8 +110,9 @@ static int snd_dg00x_probe(struct fw_unit *unit,
 	struct snd_dg00x *dg00x;
 
 	/* Allocate this independent of sound card instance. */
-	dg00x = kzalloc(sizeof(struct snd_dg00x), GFP_KERNEL);
-	if (dg00x == NULL)
+	dg00x = devm_kzalloc(&unit->device, sizeof(struct snd_dg00x),
+			     GFP_KERNEL);
+	if (!dg00x)
 		return -ENOMEM;
 
 	dg00x->unit = fw_unit_get(unit);
@@ -173,12 +164,12 @@ static void snd_dg00x_remove(struct fw_unit *unit)
 	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);
+		// Block till all of ALSA character devices are released.
+		snd_card_free(dg00x->card);
 	}
+
+	mutex_destroy(&dg00x->mutex);
+	fw_unit_put(dg00x->unit);
 }
 
 static const struct ieee1394_device_id snd_dg00x_id_table[] = {
diff --git a/sound/firewire/fireface/ff.c b/sound/firewire/fireface/ff.c
index 4974bc7980e9..3f61cfeace69 100644
--- a/sound/firewire/fireface/ff.c
+++ b/sound/firewire/fireface/ff.c
@@ -27,20 +27,12 @@ static void name_card(struct snd_ff *ff)
 		 dev_name(&ff->unit->device), 100 << fw_dev->max_speed);
 }
 
-static void ff_free(struct snd_ff *ff)
+static void ff_card_free(struct snd_card *card)
 {
+	struct snd_ff *ff = card->private_data;
+
 	snd_ff_stream_destroy_duplex(ff);
 	snd_ff_transaction_unregister(ff);
-
-	fw_unit_put(ff->unit);
-
-	mutex_destroy(&ff->mutex);
-	kfree(ff);
-}
-
-static void ff_card_free(struct snd_card *card)
-{
-	ff_free(card->private_data);
 }
 
 static void do_registration(struct work_struct *work)
@@ -55,6 +47,8 @@ static void do_registration(struct work_struct *work)
 			   &ff->card);
 	if (err < 0)
 		return;
+	ff->card->private_free = ff_card_free;
+	ff->card->private_data = ff;
 
 	err = snd_ff_transaction_register(ff);
 	if (err < 0)
@@ -84,14 +78,10 @@ static void do_registration(struct work_struct *work)
 	if (err < 0)
 		goto error;
 
-	ff->card->private_free = ff_card_free;
-	ff->card->private_data = ff;
 	ff->registered = true;
 
 	return;
 error:
-	snd_ff_transaction_unregister(ff);
-	snd_ff_stream_destroy_duplex(ff);
 	snd_card_free(ff->card);
 	dev_info(&ff->unit->device,
 		 "Sound card registration failed: %d\n", err);
@@ -102,11 +92,9 @@ static int snd_ff_probe(struct fw_unit *unit,
 {
 	struct snd_ff *ff;
 
-	ff = kzalloc(sizeof(struct snd_ff), GFP_KERNEL);
-	if (ff == NULL)
+	ff = devm_kzalloc(&unit->device, sizeof(struct snd_ff), GFP_KERNEL);
+	if (!ff)
 		return -ENOMEM;
-
-	/* initialize myself */
 	ff->unit = fw_unit_get(unit);
 	dev_set_drvdata(&unit->device, ff);
 
@@ -149,12 +137,12 @@ static void snd_ff_remove(struct fw_unit *unit)
 	cancel_work_sync(&ff->dwork.work);
 
 	if (ff->registered) {
-		/* No need to wait for releasing card object in this context. */
-		snd_card_free_when_closed(ff->card);
-	} else {
-		/* Don't forget this case. */
-		ff_free(ff);
+		// Block till all of ALSA character devices are released.
+		snd_card_free(ff->card);
 	}
+
+	mutex_destroy(&ff->mutex);
+	fw_unit_put(ff->unit);
 }
 
 static const struct snd_ff_spec spec_ff400 = {
diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c
index f2d073365cf6..faf0e001c4c5 100644
--- a/sound/firewire/fireworks/fireworks.c
+++ b/sound/firewire/fireworks/fireworks.c
@@ -184,36 +184,17 @@ end:
 	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
- * transactions safely. Thus at returning from .remove(), this module still keep
- * references for the unit.
- */
 static void
 efw_card_free(struct snd_card *card)
 {
 	struct snd_efw *efw = card->private_data;
 
-	if (efw->card_index >= 0) {
-		mutex_lock(&devices_mutex);
-		clear_bit(efw->card_index, devices_used);
-		mutex_unlock(&devices_mutex);
-	}
+	mutex_lock(&devices_mutex);
+	clear_bit(efw->card_index, devices_used);
+	mutex_unlock(&devices_mutex);
 
-	efw_free(card->private_data);
+	snd_efw_stream_destroy_duplex(efw);
+	snd_efw_transaction_remove_instance(efw);
 }
 
 static void
@@ -226,9 +207,8 @@ do_registration(struct work_struct *work)
 	if (efw->registered)
 		return;
 
-	mutex_lock(&devices_mutex);
-
 	/* check registered cards */
+	mutex_lock(&devices_mutex);
 	for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) {
 		if (!test_bit(card_index, devices_used) && enable[card_index])
 			break;
@@ -244,12 +224,18 @@ do_registration(struct work_struct *work)
 		mutex_unlock(&devices_mutex);
 		return;
 	}
+	set_bit(card_index, devices_used);
+	mutex_unlock(&devices_mutex);
+
+	efw->card->private_free = efw_card_free;
+	efw->card->private_data = efw;
 
 	/* prepare response buffer */
 	snd_efw_resp_buf_size = clamp(snd_efw_resp_buf_size,
 				      SND_EFW_RESPONSE_MAXIMUM_BYTES, 4096U);
-	efw->resp_buf = kzalloc(snd_efw_resp_buf_size, GFP_KERNEL);
-	if (efw->resp_buf == NULL) {
+	efw->resp_buf = devm_kzalloc(&efw->card->card_dev,
+				     snd_efw_resp_buf_size, GFP_KERNEL);
+	if (!efw->resp_buf) {
 		err = -ENOMEM;
 		goto error;
 	}
@@ -284,25 +270,11 @@ do_registration(struct work_struct *work)
 	if (err < 0)
 		goto error;
 
-	set_bit(card_index, devices_used);
-	mutex_unlock(&devices_mutex);
-
-	/*
-	 * 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:
-	mutex_unlock(&devices_mutex);
-	snd_efw_transaction_remove_instance(efw);
-	snd_efw_stream_destroy_duplex(efw);
 	snd_card_free(efw->card);
-	kfree(efw->resp_buf);
-	efw->resp_buf = NULL;
 	dev_info(&efw->unit->device,
 		 "Sound card registration failed: %d\n", err);
 }
@@ -312,10 +284,9 @@ efw_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry)
 {
 	struct snd_efw *efw;
 
-	efw = kzalloc(sizeof(struct snd_efw), GFP_KERNEL);
+	efw = devm_kzalloc(&unit->device, sizeof(struct snd_efw), GFP_KERNEL);
 	if (efw == NULL)
 		return -ENOMEM;
-
 	efw->unit = fw_unit_get(unit);
 	dev_set_drvdata(&unit->device, efw);
 
@@ -363,12 +334,12 @@ static void efw_remove(struct fw_unit *unit)
 	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);
+		// Block till all of ALSA character devices are released.
+		snd_card_free(efw->card);
 	}
+
+	mutex_destroy(&efw->mutex);
+	fw_unit_put(efw->unit);
 }
 
 static const struct ieee1394_device_id efw_id_table[] = {
diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
index 30957477e005..9ebe510ea26b 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -602,8 +602,6 @@ static void isight_card_free(struct snd_card *card)
 	struct isight *isight = card->private_data;
 
 	fw_iso_resources_destroy(&isight->resources);
-	fw_unit_put(isight->unit);
-	mutex_destroy(&isight->mutex);
 }
 
 static u64 get_unit_base(struct fw_unit *unit)
@@ -640,7 +638,7 @@ static int isight_probe(struct fw_unit *unit,
 	if (!isight->audio_base) {
 		dev_err(&unit->device, "audio unit base not found\n");
 		err = -ENXIO;
-		goto err_unit;
+		goto error;
 	}
 	fw_iso_resources_init(&isight->resources, unit);
 
@@ -669,12 +667,12 @@ static int isight_probe(struct fw_unit *unit,
 	dev_set_drvdata(&unit->device, isight);
 
 	return 0;
-
-err_unit:
-	fw_unit_put(isight->unit);
-	mutex_destroy(&isight->mutex);
 error:
 	snd_card_free(card);
+
+	mutex_destroy(&isight->mutex);
+	fw_unit_put(isight->unit);
+
 	return err;
 }
 
@@ -703,7 +701,11 @@ static void isight_remove(struct fw_unit *unit)
 	isight_stop_streaming(isight);
 	mutex_unlock(&isight->mutex);
 
-	snd_card_free_when_closed(isight->card);
+	// Block till all of ALSA character devices are released.
+	snd_card_free(isight->card);
+
+	mutex_destroy(&isight->mutex);
+	fw_unit_put(isight->unit);
 }
 
 static const struct ieee1394_device_id isight_id_table[] = {
diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c
index 300d31b6f191..220e61926ea4 100644
--- a/sound/firewire/motu/motu.c
+++ b/sound/firewire/motu/motu.c
@@ -52,26 +52,12 @@ static void name_card(struct snd_motu *motu)
 		 dev_name(&motu->unit->device), 100 << fw_dev->max_speed);
 }
 
-static void motu_free(struct snd_motu *motu)
+static void motu_card_free(struct snd_card *card)
 {
-	snd_motu_transaction_unregister(motu);
+	struct snd_motu *motu = card->private_data;
 
+	snd_motu_transaction_unregister(motu);
 	snd_motu_stream_destroy_duplex(motu);
-	fw_unit_put(motu->unit);
-
-	mutex_destroy(&motu->mutex);
-	kfree(motu);
-}
-
-/*
- * 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 motu_card_free(struct snd_card *card)
-{
-	motu_free(card->private_data);
 }
 
 static void do_registration(struct work_struct *work)
@@ -86,6 +72,8 @@ static void do_registration(struct work_struct *work)
 			   &motu->card);
 	if (err < 0)
 		return;
+	motu->card->private_free = motu_card_free;
+	motu->card->private_data = motu;
 
 	name_card(motu);
 
@@ -120,18 +108,10 @@ static void do_registration(struct work_struct *work)
 	if (err < 0)
 		goto error;
 
-	/*
-	 * After registered, motu instance can be released corresponding to
-	 * releasing the sound card instance.
-	 */
-	motu->card->private_free = motu_card_free;
-	motu->card->private_data = motu;
 	motu->registered = true;
 
 	return;
 error:
-	snd_motu_transaction_unregister(motu);
-	snd_motu_stream_destroy_duplex(motu);
 	snd_card_free(motu->card);
 	dev_info(&motu->unit->device,
 		 "Sound card registration failed: %d\n", err);
@@ -143,14 +123,13 @@ static int motu_probe(struct fw_unit *unit,
 	struct snd_motu *motu;
 
 	/* Allocate this independently of sound card instance. */
-	motu = kzalloc(sizeof(struct snd_motu), GFP_KERNEL);
-	if (motu == NULL)
+	motu = devm_kzalloc(&unit->device, sizeof(struct snd_motu), GFP_KERNEL);
+	if (!motu)
 		return -ENOMEM;
-
-	motu->spec = (const struct snd_motu_spec *)entry->driver_data;
 	motu->unit = fw_unit_get(unit);
 	dev_set_drvdata(&unit->device, motu);
 
+	motu->spec = (const struct snd_motu_spec *)entry->driver_data;
 	mutex_init(&motu->mutex);
 	spin_lock_init(&motu->lock);
 	init_waitqueue_head(&motu->hwdep_wait);
@@ -174,12 +153,12 @@ static void motu_remove(struct fw_unit *unit)
 	cancel_delayed_work_sync(&motu->dwork);
 
 	if (motu->registered) {
-		/* No need to wait for releasing card object in this context. */
-		snd_card_free_when_closed(motu->card);
-	} else {
-		/* Don't forget this case. */
-		motu_free(motu);
+		// Block till all of ALSA character devices are released.
+		snd_card_free(motu->card);
 	}
+
+	mutex_destroy(&motu->mutex);
+	fw_unit_put(motu->unit);
 }
 
 static void motu_bus_update(struct fw_unit *unit)
diff --git a/sound/firewire/oxfw/oxfw-scs1x.c b/sound/firewire/oxfw/oxfw-scs1x.c
index f33497cdc706..9d9545880a28 100644
--- a/sound/firewire/oxfw/oxfw-scs1x.c
+++ b/sound/firewire/oxfw/oxfw-scs1x.c
@@ -372,8 +372,9 @@ int snd_oxfw_scs1x_add(struct snd_oxfw *oxfw)
 	struct fw_scs1x *scs;
 	int err;
 
-	scs = kzalloc(sizeof(struct fw_scs1x), GFP_KERNEL);
-	if (scs == NULL)
+	scs = devm_kzalloc(&oxfw->card->card_dev, sizeof(struct fw_scs1x),
+			   GFP_KERNEL);
+	if (!scs)
 		return -ENOMEM;
 	scs->fw_dev = fw_parent_device(oxfw->unit);
 	oxfw->spec = scs;
diff --git a/sound/firewire/oxfw/oxfw-spkr.c b/sound/firewire/oxfw/oxfw-spkr.c
index cb905af0660d..66d4b1f73f0f 100644
--- a/sound/firewire/oxfw/oxfw-spkr.c
+++ b/sound/firewire/oxfw/oxfw-spkr.c
@@ -270,8 +270,9 @@ int snd_oxfw_add_spkr(struct snd_oxfw *oxfw, bool is_lacie)
 	unsigned int i, first_ch;
 	int err;
 
-	spkr = kzalloc(sizeof(struct fw_spkr), GFP_KERNEL);
-	if (spkr == NULL)
+	spkr = devm_kzalloc(&oxfw->card->card_dev, sizeof(struct fw_spkr),
+			    GFP_KERNEL);
+	if (!spkr)
 		return -ENOMEM;
 	oxfw->spec = spkr;
 
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index d9361f352133..f230a9e44c3c 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -517,8 +517,9 @@ assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir,
 	if (err < 0)
 		goto end;
 
-	formats[eid] = kmemdup(buf, *len, GFP_KERNEL);
-	if (formats[eid] == NULL) {
+	formats[eid] = devm_kmemdup(&oxfw->card->card_dev, buf, *len,
+				    GFP_KERNEL);
+	if (!formats[eid]) {
 		err = -ENOMEM;
 		goto end;
 	}
@@ -535,7 +536,8 @@ assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir,
 			continue;
 
 		eid++;
-		formats[eid] = kmemdup(buf, *len, GFP_KERNEL);
+		formats[eid] = devm_kmemdup(&oxfw->card->card_dev, buf, *len,
+					    GFP_KERNEL);
 		if (formats[eid] == NULL) {
 			err = -ENOMEM;
 			goto end;
@@ -597,8 +599,9 @@ static int fill_stream_formats(struct snd_oxfw *oxfw,
 		if (err < 0)
 			break;
 
-		formats[eid] = kmemdup(buf, len, GFP_KERNEL);
-		if (formats[eid] == NULL) {
+		formats[eid] = devm_kmemdup(&oxfw->card->card_dev, buf, len,
+					    GFP_KERNEL);
+		if (!formats[eid]) {
 			err = -ENOMEM;
 			break;
 		}
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 2ea8be6c8584..afb78d90384b 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -113,35 +113,13 @@ end:
 	return err;
 }
 
-static void oxfw_free(struct snd_oxfw *oxfw)
+static void oxfw_card_free(struct snd_card *card)
 {
-	unsigned int i;
+	struct snd_oxfw *oxfw = card->private_data;
 
 	snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream);
 	if (oxfw->has_output)
 		snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream);
-
-	fw_unit_put(oxfw->unit);
-
-	for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
-		kfree(oxfw->tx_stream_formats[i]);
-		kfree(oxfw->rx_stream_formats[i]);
-	}
-
-	kfree(oxfw->spec);
-	mutex_destroy(&oxfw->mutex);
-	kfree(oxfw);
-}
-
-/*
- * 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)
@@ -208,7 +186,6 @@ static int detect_quirks(struct snd_oxfw *oxfw)
 static void do_registration(struct work_struct *work)
 {
 	struct snd_oxfw *oxfw = container_of(work, struct snd_oxfw, dwork.work);
-	int i;
 	int err;
 
 	if (oxfw->registered)
@@ -218,6 +195,8 @@ static void do_registration(struct work_struct *work)
 			   &oxfw->card);
 	if (err < 0)
 		return;
+	oxfw->card->private_free = oxfw_card_free;
+	oxfw->card->private_data = oxfw;
 
 	err = name_card(oxfw);
 	if (err < 0)
@@ -258,28 +237,11 @@ static void do_registration(struct work_struct *work)
 	if (err < 0)
 		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);
-	for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; ++i) {
-		kfree(oxfw->tx_stream_formats[i]);
-		oxfw->tx_stream_formats[i] = NULL;
-		kfree(oxfw->rx_stream_formats[i]);
-		oxfw->rx_stream_formats[i] = NULL;
-	}
 	snd_card_free(oxfw->card);
-	kfree(oxfw->spec);
-	oxfw->spec = NULL;
 	dev_info(&oxfw->unit->device,
 		 "Sound card registration failed: %d\n", err);
 }
@@ -293,14 +255,13 @@ static int oxfw_probe(struct fw_unit *unit,
 		return -ENODEV;
 
 	/* Allocate this independent of sound card instance. */
-	oxfw = kzalloc(sizeof(struct snd_oxfw), GFP_KERNEL);
-	if (oxfw == NULL)
+	oxfw = devm_kzalloc(&unit->device, sizeof(struct snd_oxfw), GFP_KERNEL);
+	if (!oxfw)
 		return -ENOMEM;
-
-	oxfw->entry = entry;
 	oxfw->unit = fw_unit_get(unit);
 	dev_set_drvdata(&unit->device, oxfw);
 
+	oxfw->entry = entry;
 	mutex_init(&oxfw->mutex);
 	spin_lock_init(&oxfw->lock);
 	init_waitqueue_head(&oxfw->hwdep_wait);
@@ -347,12 +308,12 @@ static void oxfw_remove(struct fw_unit *unit)
 	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);
+		// Block till all of ALSA character devices are released.
+		snd_card_free(oxfw->card);
 	}
+
+	mutex_destroy(&oxfw->mutex);
+	fw_unit_put(oxfw->unit);
 }
 
 static const struct compat_info griffin_firewave = {
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
index d3fdc463a884..ef57fa4db323 100644
--- a/sound/firewire/tascam/tascam.c
+++ b/sound/firewire/tascam/tascam.c
@@ -85,20 +85,12 @@ static int identify_model(struct snd_tscm *tscm)
 	return 0;
 }
 
-static void tscm_free(struct snd_tscm *tscm)
+static void tscm_card_free(struct snd_card *card)
 {
+	struct snd_tscm *tscm = card->private_data;
+
 	snd_tscm_transaction_unregister(tscm);
 	snd_tscm_stream_destroy_duplex(tscm);
-
-	fw_unit_put(tscm->unit);
-
-	mutex_destroy(&tscm->mutex);
-	kfree(tscm);
-}
-
-static void tscm_card_free(struct snd_card *card)
-{
-	tscm_free(card->private_data);
 }
 
 static void do_registration(struct work_struct *work)
@@ -110,6 +102,8 @@ static void do_registration(struct work_struct *work)
 			   &tscm->card);
 	if (err < 0)
 		return;
+	tscm->card->private_free = tscm_card_free;
+	tscm->card->private_data = tscm;
 
 	err = identify_model(tscm);
 	if (err < 0)
@@ -141,18 +135,10 @@ static void do_registration(struct work_struct *work)
 	if (err < 0)
 		goto error;
 
-	/*
-	 * 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;
 error:
-	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);
@@ -164,11 +150,9 @@ static int snd_tscm_probe(struct fw_unit *unit,
 	struct snd_tscm *tscm;
 
 	/* Allocate this independent of sound card instance. */
-	tscm = kzalloc(sizeof(struct snd_tscm), GFP_KERNEL);
-	if (tscm == NULL)
+	tscm = devm_kzalloc(&unit->device, sizeof(struct snd_tscm), GFP_KERNEL);
+	if (!tscm)
 		return -ENOMEM;
-
-	/* initialize myself */
 	tscm->unit = fw_unit_get(unit);
 	dev_set_drvdata(&unit->device, tscm);
 
@@ -216,12 +200,12 @@ static void snd_tscm_remove(struct fw_unit *unit)
 	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);
+		// Block till all of ALSA character devices are released.
+		snd_card_free(tscm->card);
 	}
+
+	mutex_destroy(&tscm->mutex);
+	fw_unit_put(tscm->unit);
 }
 
 static const struct ieee1394_device_id snd_tscm_id_table[] = {
diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c
index 5bc4a1d587d4..60cb00fd0c69 100644
--- a/sound/hda/ext/hdac_ext_controller.c
+++ b/sound/hda/ext/hdac_ext_controller.c
@@ -48,9 +48,11 @@ void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *bus, bool enable)
 	}
 
 	if (enable)
-		snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_GPROCEN);
+		snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL,
+				 AZX_PPCTL_GPROCEN, AZX_PPCTL_GPROCEN);
 	else
-		snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_GPROCEN, 0);
+		snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL,
+				 AZX_PPCTL_GPROCEN, 0);
 }
 EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_enable);
 
@@ -68,9 +70,11 @@ void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *bus, bool enable)
 	}
 
 	if (enable)
-		snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_PIE);
+		snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL,
+				 AZX_PPCTL_PIE, AZX_PPCTL_PIE);
 	else
-		snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_PIE, 0);
+		snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL,
+				 AZX_PPCTL_PIE, 0);
 }
 EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_int_enable);
 
@@ -194,7 +198,8 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable)
  */
 int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link)
 {
-	snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA);
+	snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL,
+			 AZX_MLCTL_SPA, AZX_MLCTL_SPA);
 
 	return check_hdac_link_power_active(link, true);
 }
@@ -222,8 +227,8 @@ int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus)
 	int ret;
 
 	list_for_each_entry(hlink, &bus->hlink_list, list) {
-		snd_hdac_updatel(hlink->ml_addr,
-				AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA);
+		snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL,
+				 AZX_MLCTL_SPA, AZX_MLCTL_SPA);
 		ret = check_hdac_link_power_active(hlink, true);
 		if (ret < 0)
 			return ret;
@@ -243,7 +248,8 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus)
 	int ret;
 
 	list_for_each_entry(hlink, &bus->hlink_list, list) {
-		snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, AZX_MLCTL_SPA, 0);
+		snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL,
+				 AZX_MLCTL_SPA, 0);
 		ret = check_hdac_link_power_active(hlink, false);
 		if (ret < 0)
 			return ret;
diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c
index 2647309bc675..8afa2f888466 100644
--- a/sound/i2c/cs8427.c
+++ b/sound/i2c/cs8427.c
@@ -118,7 +118,7 @@ static int snd_cs8427_send_corudata(struct snd_i2c_device *device,
 	struct cs8427 *chip = device->private_data;
 	char *hw_data = udata ?
 		chip->playback.hw_udata : chip->playback.hw_status;
-	char data[32];
+	unsigned char data[32];
 	int err, idx;
 
 	if (!memcmp(hw_data, ndata, count))
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index ac0ab6eb40f0..47e0b2820ace 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -389,7 +389,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip,
 	case OPTi9XX_HW_82C931:
 		/* disable 3D sound (set GPIO1 as output, low) */
 		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c);
-	case OPTi9XX_HW_82C933: /* FALL THROUGH */
+		/* fall through */
+	case OPTi9XX_HW_82C933:
 		/*
 		 * The BTC 1817DW has QS1000 wavetable which is connected
 		 * to the serial digital input of the OPTI931.
@@ -400,7 +401,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip,
 		 * or digital input signal.
 		 */
 		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01);
-	case OPTi9XX_HW_82C930: /* FALL THROUGH */
+		/* fall through */
+	case OPTi9XX_HW_82C930:
 		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03);
 		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff);
 		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x10 |
diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c
index 481797744b3c..8288fae90085 100644
--- a/sound/isa/sb/sb8_main.c
+++ b/sound/isa/sb/sb8_main.c
@@ -130,13 +130,13 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream)
 			chip->playback_format = SB_DSP_HI_OUTPUT_AUTO;
 			break;
 		}
-		/* fallthru */
+		/* fall through */
 	case SB_HW_201:
 		if (rate > 23000) {
 			chip->playback_format = SB_DSP_HI_OUTPUT_AUTO;
 			break;
 		}
-		/* fallthru */
+		/* fall through */
 	case SB_HW_20:
 		chip->playback_format = SB_DSP_LO_OUTPUT_AUTO;
 		break;
@@ -287,7 +287,7 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream)
 			chip->capture_format = SB_DSP_HI_INPUT_AUTO;
 			break;
 		}
-		/* fallthru */
+		/* fall through */
 	case SB_HW_20:
 		chip->capture_format = SB_DSP_LO_INPUT_AUTO;
 		break;
@@ -387,7 +387,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip)
 	case SB_MODE_PLAYBACK_16:	/* ok.. playback is active */
 		if (chip->hardware != SB_HW_JAZZ16)
 			break;
-		/* fallthru */
+		/* fall through */
 	case SB_MODE_PLAYBACK_8:
 		substream = chip->playback_substream;
 		if (chip->playback_format == SB_DSP_OUTPUT)
@@ -397,7 +397,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip)
 	case SB_MODE_CAPTURE_16:
 		if (chip->hardware != SB_HW_JAZZ16)
 			break;
-		/* fallthru */
+		/* fall through */
 	case SB_MODE_CAPTURE_8:
 		substream = chip->capture_substream;
 		if (chip->capture_format == SB_DSP_INPUT)
diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c
index c8904e732aaa..a4ed54aeaf1d 100644
--- a/sound/mips/hal2.c
+++ b/sound/mips/hal2.c
@@ -500,7 +500,8 @@ static const struct snd_pcm_hardware hal2_pcm_hw = {
 	.info = (SNDRV_PCM_INFO_MMAP |
 		 SNDRV_PCM_INFO_MMAP_VALID |
 		 SNDRV_PCM_INFO_INTERLEAVED |
-		 SNDRV_PCM_INFO_BLOCK_TRANSFER),
+		 SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		 SNDRV_PCM_INFO_SYNC_APPLPTR),
 	.formats =          SNDRV_PCM_FMTBIT_S16_BE,
 	.rates =            SNDRV_PCM_RATE_8000_48000,
 	.rate_min =         8000,
@@ -563,6 +564,8 @@ static int hal2_playback_prepare(struct snd_pcm_substream *substream)
 	dac->sample_rate = hal2_compute_rate(dac, runtime->rate);
 	memset(&dac->pcm_indirect, 0, sizeof(dac->pcm_indirect));
 	dac->pcm_indirect.hw_buffer_size = H2_BUF_SIZE;
+	dac->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2;
+	dac->pcm_indirect.hw_io = dac->buffer_dma;
 	dac->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
 	dac->substream = substream;
 	hal2_setup_dac(hal2);
@@ -575,9 +578,6 @@ static int hal2_playback_trigger(struct snd_pcm_substream *substream, int cmd)
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-		hal2->dac.pcm_indirect.hw_io = hal2->dac.buffer_dma;
-		hal2->dac.pcm_indirect.hw_data = 0;
-		substream->ops->ack(substream);
 		hal2_start_dac(hal2);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
@@ -615,7 +615,6 @@ static int hal2_playback_ack(struct snd_pcm_substream *substream)
 	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
 	struct hal2_codec *dac = &hal2->dac;
 
-	dac->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2;
 	return snd_pcm_indirect_playback_transfer(substream,
 						  &dac->pcm_indirect,
 						  hal2_playback_transfer);
@@ -655,6 +654,7 @@ static int hal2_capture_prepare(struct snd_pcm_substream *substream)
 	memset(&adc->pcm_indirect, 0, sizeof(adc->pcm_indirect));
 	adc->pcm_indirect.hw_buffer_size = H2_BUF_SIZE;
 	adc->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2;
+	adc->pcm_indirect.hw_io = adc->buffer_dma;
 	adc->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
 	adc->substream = substream;
 	hal2_setup_adc(hal2);
@@ -667,9 +667,6 @@ static int hal2_capture_trigger(struct snd_pcm_substream *substream, int cmd)
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-		hal2->adc.pcm_indirect.hw_io = hal2->adc.buffer_dma;
-		hal2->adc.pcm_indirect.hw_data = 0;
-		printk(KERN_DEBUG "buffer_dma %x\n", hal2->adc.buffer_dma);
 		hal2_start_adc(hal2);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c
index 5ef4fe964366..7c91330af719 100644
--- a/sound/pci/asihpi/hpios.c
+++ b/sound/pci/asihpi/hpios.c
@@ -49,7 +49,7 @@ u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
 	/*?? any benefit in using managed dmam_alloc_coherent? */
 	p_mem_area->vaddr =
 		dma_alloc_coherent(&pdev->dev, size, &p_mem_area->dma_handle,
-		GFP_DMA32 | GFP_KERNEL);
+		GFP_KERNEL);
 
 	if (p_mem_area->vaddr) {
 		HPI_DEBUG_LOG(DEBUG, "allocated %d bytes, dma 0x%x vma %p\n",
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index a1e4944dcfe8..1a41f8c80243 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -903,15 +903,15 @@ static int snd_atiixp_playback_prepare(struct snd_pcm_substream *substream)
 	case 8:
 		data |= ATI_REG_OUT_DMA_SLOT_BIT(10) |
 			ATI_REG_OUT_DMA_SLOT_BIT(11);
-		/* fallthru */
+		/* fall through */
 	case 6:
 		data |= ATI_REG_OUT_DMA_SLOT_BIT(7) |
 			ATI_REG_OUT_DMA_SLOT_BIT(8);
-		/* fallthru */
+		/* fall through */
 	case 4:
 		data |= ATI_REG_OUT_DMA_SLOT_BIT(6) |
 			ATI_REG_OUT_DMA_SLOT_BIT(9);
-		/* fallthru */
+		/* fall through */
 	default:
 		data |= ATI_REG_OUT_DMA_SLOT_BIT(3) |
 			ATI_REG_OUT_DMA_SLOT_BIT(4);
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 2e5b460a847c..96ece1a71cf1 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1115,6 +1115,7 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma,
 		hwwrite(vortex->mmio,
 			VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0xc,
 			snd_pcm_sgbuf_get_addr(dma->substream, psize * 3));
+		/* fall through */
 		/* 3 pages */
 	case 3:
 		dma->cfg0 |= 0x12000000;
@@ -1122,12 +1123,14 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma,
 		hwwrite(vortex->mmio,
 			VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0x8,
 			snd_pcm_sgbuf_get_addr(dma->substream, psize * 2));
+		/* fall through */
 		/* 2 pages */
 	case 2:
 		dma->cfg0 |= 0x88000000 | 0x44000000 | 0x10000000 | (psize - 1);
 		hwwrite(vortex->mmio,
 			VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0x4,
 			snd_pcm_sgbuf_get_addr(dma->substream, psize));
+		/* fall through */
 		/* 1 page */
 	case 1:
 		dma->cfg0 |= 0x80000000 | 0x40000000 | ((psize - 1) << 0xc);
@@ -1390,17 +1393,20 @@ vortex_wtdma_setbuffers(vortex_t * vortex, int wtdma,
 		dma->cfg1 |= 0x88000000 | 0x44000000 | 0x30000000 | (psize-1);
 		hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0xc,
 			snd_pcm_sgbuf_get_addr(dma->substream, psize * 3));
+		/* fall through */
 		/* 3 pages */
 	case 3:
 		dma->cfg0 |= 0x12000000;
 		dma->cfg1 |= 0x80000000 | 0x40000000 | ((psize-1) << 0xc);
 		hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4)  + 0x8,
 			snd_pcm_sgbuf_get_addr(dma->substream, psize * 2));
+		/* fall through */
 		/* 2 pages */
 	case 2:
 		dma->cfg0 |= 0x88000000 | 0x44000000 | 0x10000000 | (psize-1);
 		hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0x4,
 			snd_pcm_sgbuf_get_addr(dma->substream, psize));
+		/* fall through */
 		/* 1 page */
 	case 1:
 		dma->cfg0 |= 0x80000000 | 0x40000000 | ((psize-1) << 0xc);
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 146e1a3498c7..750eec437a79 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -1443,7 +1443,8 @@ static const struct snd_pcm_hardware snd_cs46xx_playback =
 	.info =			(SNDRV_PCM_INFO_MMAP |
 				 SNDRV_PCM_INFO_INTERLEAVED | 
 				 SNDRV_PCM_INFO_BLOCK_TRANSFER /*|*/
-				 /*SNDRV_PCM_INFO_RESUME*/),
+				 /*SNDRV_PCM_INFO_RESUME*/ |
+				 SNDRV_PCM_INFO_SYNC_APPLPTR),
 	.formats =		(SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |
 				 SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
 				 SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE),
@@ -1465,7 +1466,8 @@ static const struct snd_pcm_hardware snd_cs46xx_capture =
 	.info =			(SNDRV_PCM_INFO_MMAP |
 				 SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_BLOCK_TRANSFER /*|*/
-				 /*SNDRV_PCM_INFO_RESUME*/),
+				 /*SNDRV_PCM_INFO_RESUME*/ |
+				 SNDRV_PCM_INFO_SYNC_APPLPTR),
 	.formats =		SNDRV_PCM_FMTBIT_S16_LE,
 	.rates =		SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
 	.rate_min =		5500,
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index 9f2b6097f486..30b3472d0b75 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -1753,7 +1753,8 @@ static const struct snd_pcm_hardware snd_emu10k1_fx8010_playback =
 {
 	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_RESUME |
-				 /* SNDRV_PCM_INFO_MMAP_VALID | */ SNDRV_PCM_INFO_PAUSE),
+				 /* SNDRV_PCM_INFO_MMAP_VALID | */ SNDRV_PCM_INFO_PAUSE |
+				 SNDRV_PCM_INFO_SYNC_APPLPTR),
 	.formats =		SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
 	.rates =		SNDRV_PCM_RATE_48000,
 	.rate_min =		48000,
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index b9a6b66aeb0e..df0d636145f8 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -13,7 +13,7 @@
 #include <linux/export.h>
 #include <linux/sort.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index d1a6a9c1329a..f1457c6b3969 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -9,7 +9,7 @@
 #ifndef __SOUND_HDA_BEEP_H
 #define __SOUND_HDA_BEEP_H
 
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 
 #define HDA_BEEP_MODE_OFF	0
 #define HDA_BEEP_MODE_ON	1
diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index d361bb77ca00..9174f1b3a987 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -11,7 +11,7 @@
 #include <linux/pm.h>
 #include <linux/pm_runtime.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 
 /*
@@ -81,6 +81,12 @@ static int hda_codec_driver_probe(struct device *dev)
 	hda_codec_patch_t patch;
 	int err;
 
+	if (codec->bus->core.ext_ops) {
+		if (WARN_ON(!codec->bus->core.ext_ops->hdev_attach))
+			return -EINVAL;
+		return codec->bus->core.ext_ops->hdev_attach(&codec->core);
+	}
+
 	if (WARN_ON(!codec->preset))
 		return -EINVAL;
 
@@ -134,6 +140,12 @@ static int hda_codec_driver_remove(struct device *dev)
 {
 	struct hda_codec *codec = dev_to_hda_codec(dev);
 
+	if (codec->bus->core.ext_ops) {
+		if (WARN_ON(!codec->bus->core.ext_ops->hdev_detach))
+			return -EINVAL;
+		return codec->bus->core.ext_ops->hdev_detach(&codec->core);
+	}
+
 	if (codec->patch_ops.free)
 		codec->patch_ops.free(codec);
 	snd_hda_codec_cleanup_for_unbind(codec);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 26d348b47867..0957813939e5 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -27,7 +27,7 @@
 #include <linux/pm.h>
 #include <linux/pm_runtime.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include <sound/asoundef.h>
 #include <sound/tlv.h>
 #include <sound/initval.h>
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
deleted file mode 100644
index 0d98bb9068b1..000000000000
--- a/sound/pci/hda/hda_codec.h
+++ /dev/null
@@ -1,534 +0,0 @@
-/*
- * Universal Interface for Intel High Definition Audio Codec
- *
- * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
- *
- *  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.
- */
-
-#ifndef __SOUND_HDA_CODEC_H
-#define __SOUND_HDA_CODEC_H
-
-#include <linux/kref.h>
-#include <linux/mod_devicetable.h>
-#include <sound/info.h>
-#include <sound/control.h>
-#include <sound/pcm.h>
-#include <sound/hwdep.h>
-#include <sound/hdaudio.h>
-#include <sound/hda_verbs.h>
-#include <sound/hda_regmap.h>
-
-/*
- * Structures
- */
-
-struct hda_bus;
-struct hda_beep;
-struct hda_codec;
-struct hda_pcm;
-struct hda_pcm_stream;
-
-/*
- * codec bus
- *
- * each controller needs to creata a hda_bus to assign the accessor.
- * A hda_bus contains several codecs in the list codec_list.
- */
-struct hda_bus {
-	struct hdac_bus core;
-
-	struct snd_card *card;
-
-	struct pci_dev *pci;
-	const char *modelname;
-
-	struct mutex prepare_mutex;
-
-	/* assigned PCMs */
-	DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES);
-
-	/* misc op flags */
-	unsigned int needs_damn_long_delay :1;
-	unsigned int allow_bus_reset:1;	/* allow bus reset at fatal error */
-	/* status for codec/controller */
-	unsigned int shutdown :1;	/* being unloaded */
-	unsigned int response_reset:1;	/* controller was reset */
-	unsigned int in_reset:1;	/* during reset operation */
-	unsigned int no_response_fallback:1; /* don't fallback at RIRB error */
-
-	int primary_dig_out_type;	/* primary digital out PCM type */
-	unsigned int mixer_assigned;	/* codec addr for mixer name */
-};
-
-/* from hdac_bus to hda_bus */
-#define to_hda_bus(bus)		container_of(bus, struct hda_bus, core)
-
-/*
- * codec preset
- *
- * Known codecs have the patch to build and set up the controls/PCMs
- * better than the generic parser.
- */
-typedef int (*hda_codec_patch_t)(struct hda_codec *);
-	
-#define HDA_CODEC_ID_SKIP_PROBE		0x00000001
-#define HDA_CODEC_ID_GENERIC_HDMI	0x00000101
-#define HDA_CODEC_ID_GENERIC		0x00000201
-
-#define HDA_CODEC_REV_ENTRY(_vid, _rev, _name, _patch) \
-	{ .vendor_id = (_vid), .rev_id = (_rev), .name = (_name), \
-	  .api_version = HDA_DEV_LEGACY, \
-	  .driver_data = (unsigned long)(_patch) }
-#define HDA_CODEC_ENTRY(_vid, _name, _patch) \
-	HDA_CODEC_REV_ENTRY(_vid, 0, _name, _patch)
-
-struct hda_codec_driver {
-	struct hdac_driver core;
-	const struct hda_device_id *id;
-};
-
-int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name,
-			       struct module *owner);
-#define hda_codec_driver_register(drv) \
-	__hda_codec_driver_register(drv, KBUILD_MODNAME, THIS_MODULE)
-void hda_codec_driver_unregister(struct hda_codec_driver *drv);
-#define module_hda_codec_driver(drv) \
-	module_driver(drv, hda_codec_driver_register, \
-		      hda_codec_driver_unregister)
-
-/* ops set by the preset patch */
-struct hda_codec_ops {
-	int (*build_controls)(struct hda_codec *codec);
-	int (*build_pcms)(struct hda_codec *codec);
-	int (*init)(struct hda_codec *codec);
-	void (*free)(struct hda_codec *codec);
-	void (*unsol_event)(struct hda_codec *codec, unsigned int res);
-	void (*set_power_state)(struct hda_codec *codec, hda_nid_t fg,
-				unsigned int power_state);
-#ifdef CONFIG_PM
-	int (*suspend)(struct hda_codec *codec);
-	int (*resume)(struct hda_codec *codec);
-	int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid);
-#endif
-	void (*reboot_notify)(struct hda_codec *codec);
-	void (*stream_pm)(struct hda_codec *codec, hda_nid_t nid, bool on);
-};
-
-/* PCM callbacks */
-struct hda_pcm_ops {
-	int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec,
-		    struct snd_pcm_substream *substream);
-	int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec,
-		     struct snd_pcm_substream *substream);
-	int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec,
-		       unsigned int stream_tag, unsigned int format,
-		       struct snd_pcm_substream *substream);
-	int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec,
-		       struct snd_pcm_substream *substream);
-	unsigned int (*get_delay)(struct hda_pcm_stream *info,
-				  struct hda_codec *codec,
-				  struct snd_pcm_substream *substream);
-};
-
-/* PCM information for each substream */
-struct hda_pcm_stream {
-	unsigned int substreams;	/* number of substreams, 0 = not exist*/
-	unsigned int channels_min;	/* min. number of channels */
-	unsigned int channels_max;	/* max. number of channels */
-	hda_nid_t nid;	/* default NID to query rates/formats/bps, or set up */
-	u32 rates;	/* supported rates */
-	u64 formats;	/* supported formats (SNDRV_PCM_FMTBIT_) */
-	unsigned int maxbps;	/* supported max. bit per sample */
-	const struct snd_pcm_chmap_elem *chmap; /* chmap to override */
-	struct hda_pcm_ops ops;
-};
-
-/* PCM types */
-enum {
-	HDA_PCM_TYPE_AUDIO,
-	HDA_PCM_TYPE_SPDIF,
-	HDA_PCM_TYPE_HDMI,
-	HDA_PCM_TYPE_MODEM,
-	HDA_PCM_NTYPES
-};
-
-#define SNDRV_PCM_INVALID_DEVICE	(-1)
-/* for PCM creation */
-struct hda_pcm {
-	char *name;
-	struct hda_pcm_stream stream[2];
-	unsigned int pcm_type;	/* HDA_PCM_TYPE_XXX */
-	int device;		/* device number to assign */
-	struct snd_pcm *pcm;	/* assigned PCM instance */
-	bool own_chmap;		/* codec driver provides own channel maps */
-	/* private: */
-	struct hda_codec *codec;
-	struct kref kref;
-	struct list_head list;
-};
-
-/* codec information */
-struct hda_codec {
-	struct hdac_device core;
-	struct hda_bus *bus;
-	struct snd_card *card;
-	unsigned int addr;	/* codec addr*/
-	u32 probe_id; /* overridden id for probing */
-
-	/* detected preset */
-	const struct hda_device_id *preset;
-	const char *modelname;	/* model name for preset */
-
-	/* set by patch */
-	struct hda_codec_ops patch_ops;
-
-	/* PCM to create, set by patch_ops.build_pcms callback */
-	struct list_head pcm_list_head;
-
-	/* codec specific info */
-	void *spec;
-
-	/* beep device */
-	struct hda_beep *beep;
-	unsigned int beep_mode;
-
-	/* widget capabilities cache */
-	u32 *wcaps;
-
-	struct snd_array mixers;	/* list of assigned mixer elements */
-	struct snd_array nids;		/* list of mapped mixer elements */
-
-	struct list_head conn_list;	/* linked-list of connection-list */
-
-	struct mutex spdif_mutex;
-	struct mutex control_mutex;
-	struct snd_array spdif_out;
-	unsigned int spdif_in_enable;	/* SPDIF input enable? */
-	const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
-	struct snd_array init_pins;	/* initial (BIOS) pin configurations */
-	struct snd_array driver_pins;	/* pin configs set by codec parser */
-	struct snd_array cvt_setups;	/* audio convert setups */
-
-	struct mutex user_mutex;
-#ifdef CONFIG_SND_HDA_RECONFIG
-	struct snd_array init_verbs;	/* additional init verbs */
-	struct snd_array hints;		/* additional hints */
-	struct snd_array user_pins;	/* default pin configs to override */
-#endif
-
-#ifdef CONFIG_SND_HDA_HWDEP
-	struct snd_hwdep *hwdep;	/* assigned hwdep device */
-#endif
-
-	/* misc flags */
-	unsigned int in_freeing:1; /* being released */
-	unsigned int registered:1; /* codec was registered */
-	unsigned int spdif_status_reset :1; /* needs to toggle SPDIF for each
-					     * status change
-					     * (e.g. Realtek codecs)
-					     */
-	unsigned int pin_amp_workaround:1; /* pin out-amp takes index
-					    * (e.g. Conexant codecs)
-					    */
-	unsigned int single_adc_amp:1; /* adc in-amp takes no index
-					* (e.g. CX20549 codec)
-					*/
-	unsigned int no_sticky_stream:1; /* no sticky-PCM stream assignment */
-	unsigned int pins_shutup:1;	/* pins are shut up */
-	unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
-	unsigned int no_jack_detect:1;	/* Machine has no jack-detection */
-	unsigned int inv_eapd:1; /* broken h/w: inverted EAPD control */
-	unsigned int inv_jack_detect:1;	/* broken h/w: inverted detection bit */
-	unsigned int pcm_format_first:1; /* PCM format must be set first */
-	unsigned int cached_write:1;	/* write only to caches */
-	unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */
-	unsigned int dump_coef:1; /* dump processing coefs in codec proc file */
-	unsigned int power_save_node:1; /* advanced PM for each widget */
-	unsigned int auto_runtime_pm:1; /* enable automatic codec runtime pm */
-	unsigned int force_pin_prefix:1; /* Add location prefix */
-	unsigned int link_down_at_suspend:1; /* link down at runtime suspend */
-#ifdef CONFIG_PM
-	unsigned long power_on_acct;
-	unsigned long power_off_acct;
-	unsigned long power_jiffies;
-#endif
-
-	/* filter the requested power state per nid */
-	unsigned int (*power_filter)(struct hda_codec *codec, hda_nid_t nid,
-				     unsigned int power_state);
-
-	/* codec-specific additional proc output */
-	void (*proc_widget_hook)(struct snd_info_buffer *buffer,
-				 struct hda_codec *codec, hda_nid_t nid);
-
-	/* jack detection */
-	struct snd_array jacktbl;
-	unsigned long jackpoll_interval; /* In jiffies. Zero means no poll, rely on unsol events */
-	struct delayed_work jackpoll_work;
-
-	/* jack detection */
-	struct snd_array jacks;
-
-	int depop_delay; /* depop delay in ms, -1 for default delay time */
-
-	/* fix-up list */
-	int fixup_id;
-	const struct hda_fixup *fixup_list;
-	const char *fixup_name;
-
-	/* additional init verbs */
-	struct snd_array verbs;
-};
-
-#define dev_to_hda_codec(_dev)	container_of(_dev, struct hda_codec, core.dev)
-#define hda_codec_dev(_dev)	(&(_dev)->core.dev)
-
-#define list_for_each_codec(c, bus) \
-	list_for_each_entry(c, &(bus)->core.codec_list, core.list)
-#define list_for_each_codec_safe(c, n, bus)				\
-	list_for_each_entry_safe(c, n, &(bus)->core.codec_list, core.list)
-
-/* snd_hda_codec_read/write optional flags */
-#define HDA_RW_NO_RESPONSE_FALLBACK	(1 << 0)
-
-/*
- * constructors
- */
-int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card,
-		      unsigned int codec_addr, struct hda_codec **codecp);
-int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card,
-		      unsigned int codec_addr, struct hda_codec *codec);
-int snd_hda_codec_configure(struct hda_codec *codec);
-int snd_hda_codec_update_widgets(struct hda_codec *codec);
-
-/*
- * low level functions
- */
-static inline unsigned int
-snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
-				int flags,
-				unsigned int verb, unsigned int parm)
-{
-	return snd_hdac_codec_read(&codec->core, nid, flags, verb, parm);
-}
-
-static inline int
-snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags,
-			unsigned int verb, unsigned int parm)
-{
-	return snd_hdac_codec_write(&codec->core, nid, flags, verb, parm);
-}
-
-#define snd_hda_param_read(codec, nid, param) \
-	snd_hdac_read_parm(&(codec)->core, nid, param)
-#define snd_hda_get_sub_nodes(codec, nid, start_nid) \
-	snd_hdac_get_sub_nodes(&(codec)->core, nid, start_nid)
-int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
-			    hda_nid_t *conn_list, int max_conns);
-static inline int
-snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid)
-{
-	return snd_hda_get_connections(codec, nid, NULL, 0);
-}
-
-#define snd_hda_get_raw_connections(codec, nid, list, max_conns) \
-	snd_hdac_get_connections(&(codec)->core, nid, list, max_conns)
-#define snd_hda_get_num_raw_conns(codec, nid) \
-	snd_hdac_get_connections(&(codec)->core, nid, NULL, 0);
-
-int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
-			  const hda_nid_t **listp);
-int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums,
-			  const hda_nid_t *list);
-int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
-			   hda_nid_t nid, int recursive);
-unsigned int snd_hda_get_num_devices(struct hda_codec *codec, hda_nid_t nid);
-int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid,
-			u8 *dev_list, int max_devices);
-int snd_hda_get_dev_select(struct hda_codec *codec, hda_nid_t nid);
-int snd_hda_set_dev_select(struct hda_codec *codec, hda_nid_t nid, int dev_id);
-
-struct hda_verb {
-	hda_nid_t nid;
-	u32 verb;
-	u32 param;
-};
-
-void snd_hda_sequence_write(struct hda_codec *codec,
-			    const struct hda_verb *seq);
-
-/* unsolicited event */
-static inline void
-snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex)
-{
-	snd_hdac_bus_queue_event(&bus->core, res, res_ex);
-}
-
-/* cached write */
-static inline int
-snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
-			  int flags, unsigned int verb, unsigned int parm)
-{
-	return snd_hdac_regmap_write(&codec->core, nid, verb, parm);
-}
-
-/* the struct for codec->pin_configs */
-struct hda_pincfg {
-	hda_nid_t nid;
-	unsigned char ctrl;	/* original pin control value */
-	unsigned char target;	/* target pin control value */
-	unsigned int cfg;	/* default configuration */
-};
-
-unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid);
-int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid,
-			     unsigned int cfg);
-int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list,
-		       hda_nid_t nid, unsigned int cfg); /* for hwdep */
-void snd_hda_shutup_pins(struct hda_codec *codec);
-
-/* SPDIF controls */
-struct hda_spdif_out {
-	hda_nid_t nid;		/* Converter nid values relate to */
-	unsigned int status;	/* IEC958 status bits */
-	unsigned short ctls;	/* SPDIF control bits */
-};
-struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
-					       hda_nid_t nid);
-void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx);
-void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid);
-
-/*
- * Mixer
- */
-int snd_hda_codec_build_controls(struct hda_codec *codec);
-
-/*
- * PCM
- */
-int snd_hda_codec_parse_pcms(struct hda_codec *codec);
-int snd_hda_codec_build_pcms(struct hda_codec *codec);
-
-__printf(2, 3)
-struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec,
-				      const char *fmt, ...);
-
-static inline void snd_hda_codec_pcm_get(struct hda_pcm *pcm)
-{
-	kref_get(&pcm->kref);
-}
-void snd_hda_codec_pcm_put(struct hda_pcm *pcm);
-
-int snd_hda_codec_prepare(struct hda_codec *codec,
-			  struct hda_pcm_stream *hinfo,
-			  unsigned int stream,
-			  unsigned int format,
-			  struct snd_pcm_substream *substream);
-void snd_hda_codec_cleanup(struct hda_codec *codec,
-			   struct hda_pcm_stream *hinfo,
-			   struct snd_pcm_substream *substream);
-
-void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
-				u32 stream_tag,
-				int channel_id, int format);
-void __snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid,
-				    int do_now);
-#define snd_hda_codec_cleanup_stream(codec, nid) \
-	__snd_hda_codec_cleanup_stream(codec, nid, 0)
-
-#define snd_hda_query_supported_pcm(codec, nid, ratesp, fmtsp, bpsp) \
-	snd_hdac_query_supported_pcm(&(codec)->core, nid, ratesp, fmtsp, bpsp)
-#define snd_hda_is_supported_format(codec, nid, fmt) \
-	snd_hdac_is_supported_format(&(codec)->core, nid, fmt)
-
-extern const struct snd_pcm_chmap_elem snd_pcm_2_1_chmaps[];
-
-int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec,
-			      struct hda_pcm *cpcm);
-
-/*
- * Misc
- */
-void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen);
-void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg,
-				    unsigned int power_state);
-
-int snd_hda_lock_devices(struct hda_bus *bus);
-void snd_hda_unlock_devices(struct hda_bus *bus);
-void snd_hda_bus_reset(struct hda_bus *bus);
-void snd_hda_bus_reset_codecs(struct hda_bus *bus);
-
-int snd_hda_codec_set_name(struct hda_codec *codec, const char *name);
-
-/*
- * power management
- */
-extern const struct dev_pm_ops hda_codec_driver_pm;
-
-static inline
-int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid)
-{
-#ifdef CONFIG_PM
-	if (codec->patch_ops.check_power_status)
-		return codec->patch_ops.check_power_status(codec, nid);
-#endif
-	return 0;
-}
-
-/*
- * power saving
- */
-#define snd_hda_power_up(codec)		snd_hdac_power_up(&(codec)->core)
-#define snd_hda_power_up_pm(codec)	snd_hdac_power_up_pm(&(codec)->core)
-#define snd_hda_power_down(codec)	snd_hdac_power_down(&(codec)->core)
-#define snd_hda_power_down_pm(codec)	snd_hdac_power_down_pm(&(codec)->core)
-#ifdef CONFIG_PM
-void snd_hda_set_power_save(struct hda_bus *bus, int delay);
-void snd_hda_update_power_acct(struct hda_codec *codec);
-#else
-static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {}
-#endif
-
-#ifdef CONFIG_SND_HDA_PATCH_LOADER
-/*
- * patch firmware
- */
-int snd_hda_load_patch(struct hda_bus *bus, size_t size, const void *buf);
-#endif
-
-#ifdef CONFIG_SND_HDA_DSP_LOADER
-int snd_hda_codec_load_dsp_prepare(struct hda_codec *codec, unsigned int format,
-				   unsigned int size,
-				   struct snd_dma_buffer *bufp);
-void snd_hda_codec_load_dsp_trigger(struct hda_codec *codec, bool start);
-void snd_hda_codec_load_dsp_cleanup(struct hda_codec *codec,
-				    struct snd_dma_buffer *dmab);
-#else
-static inline int
-snd_hda_codec_load_dsp_prepare(struct hda_codec *codec, unsigned int format,
-				unsigned int size,
-				struct snd_dma_buffer *bufp)
-{
-	return -ENOSYS;
-}
-static inline void
-snd_hda_codec_load_dsp_trigger(struct hda_codec *codec, bool start) {}
-static inline void
-snd_hda_codec_load_dsp_cleanup(struct hda_codec *codec,
-				struct snd_dma_buffer *dmab) {}
-#endif
-
-#endif /* __SOUND_HDA_CODEC_H */
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index a12e594d4e3b..fe2506672a72 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -130,8 +130,9 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream,
 	azx_dev->core.bufsize = 0;
 	azx_dev->core.period_bytes = 0;
 	azx_dev->core.format_val = 0;
-	ret = chip->ops->substream_alloc_pages(chip, substream,
-					  params_buffer_bytes(hw_params));
+	ret = snd_pcm_lib_malloc_pages(substream,
+				       params_buffer_bytes(hw_params));
+
 unlock:
 	dsp_unlock(azx_dev);
 	return ret;
@@ -141,7 +142,6 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
 	struct azx_dev *azx_dev = get_azx_dev(substream);
-	struct azx *chip = apcm->chip;
 	struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream);
 	int err;
 
@@ -152,7 +152,7 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
 
 	snd_hda_codec_cleanup(apcm->codec, hinfo, substream);
 
-	err = chip->ops->substream_free_pages(chip, substream);
+	err = snd_pcm_lib_free_pages(substream);
 	azx_stream(azx_dev)->prepared = 0;
 	dsp_unlock(azx_dev);
 	return err;
@@ -732,6 +732,7 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec,
 	int pcm_dev = cpcm->device;
 	unsigned int size;
 	int s, err;
+	int type = SNDRV_DMA_TYPE_DEV_SG;
 
 	list_for_each_entry(apcm, &chip->pcm_list, list) {
 		if (apcm->pcm->device == pcm_dev) {
@@ -770,7 +771,9 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec,
 	size = CONFIG_SND_HDA_PREALLOC_SIZE * 1024;
 	if (size > MAX_PREALLOC_SIZE)
 		size = MAX_PREALLOC_SIZE;
-	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
+	if (chip->uc_buffer)
+		type = SNDRV_DMA_TYPE_DEV_UC_SG;
+	snd_pcm_lib_preallocate_pages_for_all(pcm, type,
 					      chip->card->dev,
 					      size, MAX_PREALLOC_SIZE);
 	return 0;
@@ -1220,27 +1223,6 @@ void snd_hda_bus_reset(struct hda_bus *bus)
 	bus->in_reset = 0;
 }
 
-static int get_jackpoll_interval(struct azx *chip)
-{
-	int i;
-	unsigned int j;
-
-	if (!chip->jackpoll_ms)
-		return 0;
-
-	i = chip->jackpoll_ms[chip->dev_index];
-	if (i == 0)
-		return 0;
-	if (i < 50 || i > 60000)
-		j = 0;
-	else
-		j = msecs_to_jiffies(i);
-	if (j == 0)
-		dev_warn(chip->card->dev,
-			 "jackpoll_ms value out of range: %d\n", i);
-	return j;
-}
-
 /* HD-audio bus initialization */
 int azx_bus_init(struct azx *chip, const char *model,
 		 const struct hdac_io_ops *io_ops)
@@ -1323,7 +1305,7 @@ int azx_probe_codecs(struct azx *chip, unsigned int max_slots)
 			err = snd_hda_codec_new(&chip->bus, chip->card, c, &codec);
 			if (err < 0)
 				continue;
-			codec->jackpoll_interval = get_jackpoll_interval(chip);
+			codec->jackpoll_interval = chip->jackpoll_interval;
 			codec->beep_mode = chip->beep_mode;
 			codecs++;
 		}
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index a68e75b00ea3..c95097bb5a0c 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -20,7 +20,7 @@
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/initval.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include <sound/hda_register.h>
 
 #define AZX_MAX_CODECS		HDA_MAX_CODECS
@@ -76,7 +76,6 @@ struct azx_dev {
 	 *  when link position is not greater than FIFO size
 	 */
 	unsigned int insufficient:1;
-	unsigned int wc_marked:1;
 };
 
 #define azx_stream(dev)		(&(dev)->core)
@@ -88,11 +87,6 @@ struct azx;
 struct hda_controller_ops {
 	/* Disable msi if supported, PCI only */
 	int (*disable_msi_reset_irq)(struct azx *);
-	int (*substream_alloc_pages)(struct azx *chip,
-				     struct snd_pcm_substream *substream,
-				     size_t size);
-	int (*substream_free_pages)(struct azx *chip,
-				    struct snd_pcm_substream *substream);
 	void (*pcm_mmap_prepare)(struct snd_pcm_substream *substream,
 				 struct vm_area_struct *area);
 	/* Check if current position is acceptable */
@@ -127,7 +121,7 @@ struct azx {
 	int capture_streams;
 	int capture_index_offset;
 	int num_streams;
-	const int *jackpoll_ms; /* per-card jack poll interval */
+	int jackpoll_interval; /* jack poll interval in jiffies */
 
 	/* Register interaction. */
 	const struct hda_controller_ops *ops;
@@ -160,6 +154,7 @@ struct azx {
 	unsigned int msi:1;
 	unsigned int probing:1; /* codec probing phase */
 	unsigned int snoop:1;
+	unsigned int uc_buffer:1; /* non-cached pages for stream buffers */
 	unsigned int align_buffer_size:1;
 	unsigned int region_requested:1;
 	unsigned int disabled:1; /* disabled by vga_switcheroo */
@@ -175,11 +170,10 @@ struct azx {
 #define azx_bus(chip)	(&(chip)->bus.core)
 #define bus_to_azx(_bus)	container_of(_bus, struct azx, bus.core)
 
-#ifdef CONFIG_X86
-#define azx_snoop(chip)		((chip)->snoop)
-#else
-#define azx_snoop(chip)		true
-#endif
+static inline bool azx_snoop(struct azx *chip)
+{
+	return !IS_ENABLED(CONFIG_X86) || chip->snoop;
+}
 
 /*
  * macros for easy use
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index ba7fe9b6655c..806b12ed44a2 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -27,7 +27,7 @@
 #include <sound/core.h>
 #include <asm/unaligned.h>
 #include <sound/hda_chmap.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 
 enum eld_versions {
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 579984ecdec3..276150f29cda 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -32,7 +32,7 @@
 #include <sound/core.h>
 #include <sound/jack.h>
 #include <sound/tlv.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index cc009a4a3d1d..268bba6ec985 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -23,7 +23,7 @@
 #include <linux/compat.h>
 #include <linux/nospec.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include <sound/hda_hwdep.h>
 #include <sound/minors.h>
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index aa4c672dbaf7..d8eb2b5f51ae 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -63,7 +63,7 @@
 #include <linux/vgaarb.h>
 #include <linux/vga_switcheroo.h>
 #include <linux/firmware.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_controller.h"
 #include "hda_intel.h"
 
@@ -399,61 +399,6 @@ static char *driver_short_names[] = {
 	[AZX_DRIVER_GENERIC] = "HD-Audio Generic",
 };
 
-#ifdef CONFIG_X86
-static void __mark_pages_wc(struct azx *chip, struct snd_dma_buffer *dmab, bool on)
-{
-	int pages;
-
-	if (azx_snoop(chip))
-		return;
-	if (!dmab || !dmab->area || !dmab->bytes)
-		return;
-
-#ifdef CONFIG_SND_DMA_SGBUF
-	if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_SG) {
-		struct snd_sg_buf *sgbuf = dmab->private_data;
-		if (chip->driver_type == AZX_DRIVER_CMEDIA)
-			return; /* deal with only CORB/RIRB buffers */
-		if (on)
-			set_pages_array_wc(sgbuf->page_table, sgbuf->pages);
-		else
-			set_pages_array_wb(sgbuf->page_table, sgbuf->pages);
-		return;
-	}
-#endif
-
-	pages = (dmab->bytes + PAGE_SIZE - 1) >> PAGE_SHIFT;
-	if (on)
-		set_memory_wc((unsigned long)dmab->area, pages);
-	else
-		set_memory_wb((unsigned long)dmab->area, pages);
-}
-
-static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf,
-				 bool on)
-{
-	__mark_pages_wc(chip, buf, on);
-}
-static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev,
-				   struct snd_pcm_substream *substream, bool on)
-{
-	if (azx_dev->wc_marked != on) {
-		__mark_pages_wc(chip, snd_pcm_get_dma_buf(substream), on);
-		azx_dev->wc_marked = on;
-	}
-}
-#else
-/* NOP for other archs */
-static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf,
-				 bool on)
-{
-}
-static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev,
-				   struct snd_pcm_substream *substream, bool on)
-{
-}
-#endif
-
 static int azx_acquire_irq(struct azx *chip, int do_disconnect);
 static void set_default_power_save(struct azx *chip);
 
@@ -1678,6 +1623,7 @@ static void azx_check_snoop_available(struct azx *chip)
 		dev_info(chip->card->dev, "Force to %s mode by module option\n",
 			 snoop ? "snoop" : "non-snoop");
 		chip->snoop = snoop;
+		chip->uc_buffer = !snoop;
 		return;
 	}
 
@@ -1698,8 +1644,12 @@ static void azx_check_snoop_available(struct azx *chip)
 		snoop = false;
 
 	chip->snoop = snoop;
-	if (!snoop)
+	if (!snoop) {
 		dev_info(chip->card->dev, "Force to non-snoop mode\n");
+		/* C-Media requires non-cached pages only for CORB/RIRB */
+		if (chip->driver_type != AZX_DRIVER_CMEDIA)
+			chip->uc_buffer = true;
+	}
 }
 
 static void azx_probe_work(struct work_struct *work)
@@ -1767,7 +1717,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci,
 	chip->driver_type = driver_caps & 0xff;
 	check_msi(chip);
 	chip->dev_index = dev;
-	chip->jackpoll_ms = jackpoll_ms;
+	if (jackpoll_ms[dev] >= 50 && jackpoll_ms[dev] <= 60000)
+		chip->jackpoll_interval = msecs_to_jiffies(jackpoll_ms[dev]);
 	INIT_LIST_HEAD(&chip->pcm_list);
 	INIT_WORK(&hda->irq_pending_work, azx_irq_pending_work);
 	INIT_LIST_HEAD(&hda->list);
@@ -2090,55 +2041,24 @@ static int dma_alloc_pages(struct hdac_bus *bus,
 			   struct snd_dma_buffer *buf)
 {
 	struct azx *chip = bus_to_azx(bus);
-	int err;
 
-	err = snd_dma_alloc_pages(type,
-				  bus->dev,
-				  size, buf);
-	if (err < 0)
-		return err;
-	mark_pages_wc(chip, buf, true);
-	return 0;
+	if (!azx_snoop(chip) && type == SNDRV_DMA_TYPE_DEV)
+		type = SNDRV_DMA_TYPE_DEV_UC;
+	return snd_dma_alloc_pages(type, bus->dev, size, buf);
 }
 
 static void dma_free_pages(struct hdac_bus *bus, struct snd_dma_buffer *buf)
 {
-	struct azx *chip = bus_to_azx(bus);
-
-	mark_pages_wc(chip, buf, false);
 	snd_dma_free_pages(buf);
 }
 
-static int substream_alloc_pages(struct azx *chip,
-				 struct snd_pcm_substream *substream,
-				 size_t size)
-{
-	struct azx_dev *azx_dev = get_azx_dev(substream);
-	int ret;
-
-	mark_runtime_wc(chip, azx_dev, substream, false);
-	ret = snd_pcm_lib_malloc_pages(substream, size);
-	if (ret < 0)
-		return ret;
-	mark_runtime_wc(chip, azx_dev, substream, true);
-	return 0;
-}
-
-static int substream_free_pages(struct azx *chip,
-				struct snd_pcm_substream *substream)
-{
-	struct azx_dev *azx_dev = get_azx_dev(substream);
-	mark_runtime_wc(chip, azx_dev, substream, false);
-	return snd_pcm_lib_free_pages(substream);
-}
-
 static void pcm_mmap_prepare(struct snd_pcm_substream *substream,
 			     struct vm_area_struct *area)
 {
 #ifdef CONFIG_X86
 	struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
 	struct azx *chip = apcm->chip;
-	if (!azx_snoop(chip) && chip->driver_type != AZX_DRIVER_CMEDIA)
+	if (chip->uc_buffer)
 		area->vm_page_prot = pgprot_writecombine(area->vm_page_prot);
 #endif
 }
@@ -2156,8 +2076,6 @@ static const struct hdac_io_ops pci_hda_io_ops = {
 
 static const struct hda_controller_ops pci_hda_ops = {
 	.disable_msi_reset_irq = disable_msi_reset_irq,
-	.substream_alloc_pages = substream_alloc_pages,
-	.substream_free_pages = substream_free_pages,
 	.pcm_mmap_prepare = pcm_mmap_prepare,
 	.position_check = azx_position_check,
 	.link_power = azx_intel_link_power,
@@ -2257,8 +2175,12 @@ static struct snd_pci_quirk power_save_blacklist[] = {
 	/* https://bugzilla.redhat.com/show_bug.cgi?id=1581607 */
 	SND_PCI_QUIRK(0x1558, 0x3501, "Clevo W35xSS_370SS", 0),
 	/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
+	SND_PCI_QUIRK(0x1028, 0x0497, "Dell Precision T3600", 0),
+	/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
 	/* Note the P55A-UD3 and Z87-D3HP share the subsys id for the HDA dev */
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P55A-UD3 / Z87-D3HP", 0),
+	/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
+	SND_PCI_QUIRK(0x8086, 0x2040, "Intel DZ77BH-55K", 0),
 	/* https://bugzilla.kernel.org/show_bug.cgi?id=199607 */
 	SND_PCI_QUIRK(0x8086, 0x2057, "Intel NUC5i7RYB", 0),
 	/* https://bugzilla.redhat.com/show_bug.cgi?id=1520902 */
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index a33234e04d4f..c499727920e6 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -15,7 +15,7 @@
 #include <sound/core.h>
 #include <sound/control.h>
 #include <sound/jack.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index c6b778b2580c..a65740419650 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -25,7 +25,7 @@
 #include <linux/slab.h>
 #include <sound/core.h>
 #include <linux/module.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 
 static int dump_coef = -1;
diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c
index 6ec79c58d48d..c154b19a0c45 100644
--- a/sound/pci/hda/hda_sysfs.c
+++ b/sound/pci/hda/hda_sysfs.c
@@ -14,7 +14,7 @@
 #include <linux/string.h>
 #include <linux/export.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include <sound/hda_hwdep.h>
 #include <sound/minors.h>
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 0621920f7617..dd7d4242d6d2 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -35,7 +35,7 @@
 #include <sound/core.h>
 #include <sound/initval.h>
 
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_controller.h"
 
 /* Defines for Nvidia Tegra HDA support */
@@ -99,19 +99,6 @@ static void dma_free_pages(struct hdac_bus *bus, struct snd_dma_buffer *buf)
 	snd_dma_free_pages(buf);
 }
 
-static int substream_alloc_pages(struct azx *chip,
-				 struct snd_pcm_substream *substream,
-				 size_t size)
-{
-	return snd_pcm_lib_malloc_pages(substream, size);
-}
-
-static int substream_free_pages(struct azx *chip,
-				struct snd_pcm_substream *substream)
-{
-	return snd_pcm_lib_free_pages(substream);
-}
-
 /*
  * Register access ops. Tegra HDA register access is DWORD only.
  */
@@ -180,10 +167,7 @@ static const struct hdac_io_ops hda_tegra_io_ops = {
 	.dma_free_pages = dma_free_pages,
 };
 
-static const struct hda_controller_ops hda_tegra_ops = {
-	.substream_alloc_pages = substream_alloc_pages,
-	.substream_free_pages = substream_free_pages,
-};
+static const struct hda_controller_ops hda_tegra_ops; /* nothing special */
 
 static void hda_tegra_init(struct hda_tegra *hda)
 {
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index fd476fb40e1b..ebfd0be885b3 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -24,7 +24,7 @@
 #include <linux/module.h>
 
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_beep.h"
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index c2d9ee9cfdc0..21d0f0610913 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -22,7 +22,7 @@
 #include <linux/slab.h>
 #include <linux/module.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 0166a3d7cd55..0a24037184c3 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -31,8 +31,9 @@
 #include <linux/types.h>
 #include <linux/io.h>
 #include <linux/pci.h>
+#include <asm/io.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
@@ -81,12 +82,12 @@
 #define SCP_GET    1
 
 #define EFX_FILE   "ctefx.bin"
-#define SBZ_EFX_FILE   "ctefx-sbz.bin"
+#define DESKTOP_EFX_FILE   "ctefx-desktop.bin"
 #define R3DI_EFX_FILE  "ctefx-r3di.bin"
 
 #ifdef CONFIG_SND_HDA_CODEC_CA0132_DSP
 MODULE_FIRMWARE(EFX_FILE);
-MODULE_FIRMWARE(SBZ_EFX_FILE);
+MODULE_FIRMWARE(DESKTOP_EFX_FILE);
 MODULE_FIRMWARE(R3DI_EFX_FILE);
 #endif
 
@@ -152,7 +153,10 @@ enum {
 	XBASS_XOVER,
 	EQ_PRESET_ENUM,
 	SMART_VOLUME_ENUM,
-	MIC_BOOST_ENUM
+	MIC_BOOST_ENUM,
+	AE5_HEADPHONE_GAIN_ENUM,
+	AE5_SOUND_FILTER_ENUM,
+	ZXR_HEADPHONE_GAIN
 #define EFFECTS_COUNT  (EFFECT_END_NID - EFFECT_START_NID)
 };
 
@@ -666,6 +670,65 @@ static const struct ct_dsp_volume_ctl ca0132_alt_vol_ctls[] = {
 	}
 };
 
+/* Values for ca0113_mmio_command_set for selecting output. */
+#define AE5_CA0113_OUT_SET_COMMANDS 6
+struct ae5_ca0113_output_set {
+	unsigned int group[AE5_CA0113_OUT_SET_COMMANDS];
+	unsigned int target[AE5_CA0113_OUT_SET_COMMANDS];
+	unsigned int vals[AE5_CA0113_OUT_SET_COMMANDS];
+};
+
+static const struct ae5_ca0113_output_set ae5_ca0113_output_presets[] = {
+	{ .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
+	  .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
+	  .vals =   { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f }
+	},
+	{ .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
+	  .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
+	  .vals =   { 0x3f, 0x3f, 0x00, 0x00, 0x00, 0x00 }
+	},
+	{ .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
+	  .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
+	  .vals =   { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f }
+	}
+};
+
+/* ae5 ca0113 command sequences to set headphone gain levels. */
+#define AE5_HEADPHONE_GAIN_PRESET_MAX_COMMANDS 4
+struct ae5_headphone_gain_set {
+	char *name;
+	unsigned int vals[AE5_HEADPHONE_GAIN_PRESET_MAX_COMMANDS];
+};
+
+static const struct ae5_headphone_gain_set ae5_headphone_gain_presets[] = {
+	{ .name = "Low (16-31",
+	  .vals = { 0xff, 0x2c, 0xf5, 0x32 }
+	},
+	{ .name = "Medium (32-149",
+	  .vals = { 0x38, 0xa8, 0x3e, 0x4c }
+	},
+	{ .name = "High (150-600",
+	  .vals = { 0xff, 0xff, 0xff, 0x7f }
+	}
+};
+
+struct ae5_filter_set {
+	char *name;
+	unsigned int val;
+};
+
+static const struct ae5_filter_set ae5_filter_presets[] = {
+	{ .name = "Slow Roll Off",
+	  .val = 0xa0
+	},
+	{ .name = "Minimum Phase",
+	  .val = 0xc0
+	},
+	{ .name = "Fast Roll Off",
+	  .val = 0x80
+	}
+};
+
 enum hda_cmd_vendor_io {
 	/* for DspIO node */
 	VENDOR_DSPIO_SCP_WRITE_DATA_LOW      = 0x000,
@@ -685,6 +748,9 @@ enum hda_cmd_vendor_io {
 	VENDOR_CHIPIO_DATA_LOW               = 0x300,
 	VENDOR_CHIPIO_DATA_HIGH              = 0x400,
 
+	VENDOR_CHIPIO_8051_WRITE_DIRECT      = 0x500,
+	VENDOR_CHIPIO_8051_READ_DIRECT       = 0xD00,
+
 	VENDOR_CHIPIO_GET_PARAMETER          = 0xF00,
 	VENDOR_CHIPIO_STATUS                 = 0xF01,
 	VENDOR_CHIPIO_HIC_POST_READ          = 0x702,
@@ -692,6 +758,9 @@ enum hda_cmd_vendor_io {
 
 	VENDOR_CHIPIO_8051_DATA_WRITE        = 0x707,
 	VENDOR_CHIPIO_8051_DATA_READ         = 0xF07,
+	VENDOR_CHIPIO_8051_PMEM_READ         = 0xF08,
+	VENDOR_CHIPIO_8051_IRAM_WRITE        = 0x709,
+	VENDOR_CHIPIO_8051_IRAM_READ         = 0xF09,
 
 	VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE   = 0x70A,
 	VENDOR_CHIPIO_CT_EXTENSIONS_GET      = 0xF0A,
@@ -798,6 +867,12 @@ enum control_param_id {
 	 * impedance is selected*/
 	CONTROL_PARAM_PORTD_160OHM_GAIN        = 10,
 
+	/*
+	 * This control param name was found in the 8051 memory, and makes
+	 * sense given the fact the AE-5 uses it and has the ASI flag set.
+	 */
+	CONTROL_PARAM_ASI                      = 23,
+
 	/* Stream Control */
 
 	/* Select stream with the given ID */
@@ -955,7 +1030,11 @@ struct ca0132_spec {
 	long eq_preset_val;
 	unsigned int tlv[4];
 	struct hda_vmaster_mute_hook vmaster_mute;
-
+	/* AE-5 Control values */
+	unsigned char ae5_headphone_gain_val;
+	unsigned char ae5_filter_val;
+	/* ZxR Control Values */
+	unsigned char zxr_gain_set;
 
 	struct hda_codec *codec;
 	struct delayed_work unsol_hp_work;
@@ -995,8 +1074,11 @@ enum {
 	QUIRK_ALIENWARE,
 	QUIRK_ALIENWARE_M17XR4,
 	QUIRK_SBZ,
+	QUIRK_ZXR,
+	QUIRK_ZXR_DBPRO,
 	QUIRK_R3DI,
 	QUIRK_R3D,
+	QUIRK_AE5,
 };
 
 static const struct hda_pintbl alienware_pincfgs[] = {
@@ -1028,6 +1110,21 @@ static const struct hda_pintbl sbz_pincfgs[] = {
 	{}
 };
 
+/* Sound Blaster ZxR pin configs taken from Windows Driver */
+static const struct hda_pintbl zxr_pincfgs[] = {
+	{ 0x0b, 0x01047110 }, /* Port G -- Lineout FRONT L/R */
+	{ 0x0c, 0x414510f0 }, /* SPDIF Out 1 - Disabled*/
+	{ 0x0d, 0x014510f0 }, /* Digital Out */
+	{ 0x0e, 0x41c520f0 }, /* SPDIF In - Disabled*/
+	{ 0x0f, 0x0122711f }, /* Port A -- BackPanel HP */
+	{ 0x10, 0x01017111 }, /* Port D -- Center/LFE */
+	{ 0x11, 0x01017114 }, /* Port B -- LineMicIn2 / Rear L/R */
+	{ 0x12, 0x01a271f0 }, /* Port C -- LineIn1 */
+	{ 0x13, 0x908700f0 }, /* What U Hear In*/
+	{ 0x18, 0x50d000f0 }, /* N/A */
+	{}
+};
+
 /* Recon3D pin configs taken from Windows Driver */
 static const struct hda_pintbl r3d_pincfgs[] = {
 	{ 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */
@@ -1043,6 +1140,21 @@ static const struct hda_pintbl r3d_pincfgs[] = {
 	{}
 };
 
+/* Sound Blaster AE-5 pin configs taken from Windows Driver */
+static const struct hda_pintbl ae5_pincfgs[] = {
+	{ 0x0b, 0x01017010 }, /* Port G -- Lineout FRONT L/R */
+	{ 0x0c, 0x014510f0 }, /* SPDIF Out 1 */
+	{ 0x0d, 0x014510f0 }, /* Digital Out */
+	{ 0x0e, 0x01c510f0 }, /* SPDIF In */
+	{ 0x0f, 0x01017114 }, /* Port A -- Rear L/R. */
+	{ 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */
+	{ 0x11, 0x01a170ff }, /* Port B -- LineMicIn2 / Rear Headphone */
+	{ 0x12, 0x01a170f0 }, /* Port C -- LineIn1 */
+	{ 0x13, 0x908700f0 }, /* What U Hear In*/
+	{ 0x18, 0x50d000f0 }, /* N/A */
+	{}
+};
+
 /* Recon3D integrated pin configs taken from Windows Driver */
 static const struct hda_pintbl r3di_pincfgs[] = {
 	{ 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */
@@ -1069,6 +1181,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
 	SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI),
 	SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI),
 	SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D),
+	SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5),
 	{}
 };
 
@@ -1454,6 +1567,20 @@ static void chipio_set_conn_rate(struct hda_codec *codec,
 }
 
 /*
+ * Writes to the 8051's internal address space directly instead of indirectly,
+ * giving access to the special function registers located at addresses
+ * 0x80-0xFF.
+ */
+static void chipio_8051_write_direct(struct hda_codec *codec,
+		unsigned int addr, unsigned int data)
+{
+	unsigned int verb;
+
+	verb = VENDOR_CHIPIO_8051_WRITE_DIRECT | data;
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, verb, addr);
+}
+
+/*
  * Enable clocks.
  */
 static void chipio_enable_clocks(struct hda_codec *codec)
@@ -3088,7 +3215,9 @@ static bool dspload_wait_loaded(struct hda_codec *codec)
 }
 
 /*
- * Setup GPIO for the other variants of Core3D.
+ * ca0113 related functions. The ca0113 acts as the HDA bus for the pci-e
+ * based cards, and has a second mmio region, region2, that's used for special
+ * commands.
  */
 
 /*
@@ -3096,8 +3225,11 @@ static bool dspload_wait_loaded(struct hda_codec *codec)
  * the mmio address 0x320 is used to set GPIO pins. The format for the data
  * The first eight bits are just the number of the pin. So far, I've only seen
  * this number go to 7.
+ * AE-5 note: The AE-5 seems to use pins 2 and 3 to somehow set the color value
+ * of the on-card LED. It seems to use pin 2 for data, then toggles 3 to on and
+ * then off to send that bit.
  */
-static void ca0132_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin,
+static void ca0113_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin,
 		bool enable)
 {
 	struct ca0132_spec *spec = codec->spec;
@@ -3110,6 +3242,89 @@ static void ca0132_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin,
 }
 
 /*
+ * Special pci region2 commands that are only used by the AE-5. They follow
+ * a set format, and require reads at certain points to seemingly 'clear'
+ * the response data. My first tests didn't do these reads, and would cause
+ * the card to get locked up until the memory was read. These commands
+ * seem to work with three distinct values that I've taken to calling group,
+ * target-id, and value.
+ */
+static void ca0113_mmio_command_set(struct hda_codec *codec, unsigned int group,
+		unsigned int target, unsigned int value)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int write_val;
+
+	writel(0x0000007e, spec->mem_base + 0x210);
+	readl(spec->mem_base + 0x210);
+	writel(0x0000005a, spec->mem_base + 0x210);
+	readl(spec->mem_base + 0x210);
+	readl(spec->mem_base + 0x210);
+
+	writel(0x00800005, spec->mem_base + 0x20c);
+	writel(group, spec->mem_base + 0x804);
+
+	writel(0x00800005, spec->mem_base + 0x20c);
+	write_val = (target & 0xff);
+	write_val |= (value << 8);
+
+
+	writel(write_val, spec->mem_base + 0x204);
+	/*
+	 * Need delay here or else it goes too fast and works inconsistently.
+	 */
+	msleep(20);
+
+	readl(spec->mem_base + 0x860);
+	readl(spec->mem_base + 0x854);
+	readl(spec->mem_base + 0x840);
+
+	writel(0x00800004, spec->mem_base + 0x20c);
+	writel(0x00000000, spec->mem_base + 0x210);
+	readl(spec->mem_base + 0x210);
+	readl(spec->mem_base + 0x210);
+}
+
+/*
+ * This second type of command is used for setting the sound filter type.
+ */
+static void ca0113_mmio_command_set_type2(struct hda_codec *codec,
+		unsigned int group, unsigned int target, unsigned int value)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int write_val;
+
+	writel(0x0000007e, spec->mem_base + 0x210);
+	readl(spec->mem_base + 0x210);
+	writel(0x0000005a, spec->mem_base + 0x210);
+	readl(spec->mem_base + 0x210);
+	readl(spec->mem_base + 0x210);
+
+	writel(0x00800003, spec->mem_base + 0x20c);
+	writel(group, spec->mem_base + 0x804);
+
+	writel(0x00800005, spec->mem_base + 0x20c);
+	write_val = (target & 0xff);
+	write_val |= (value << 8);
+
+
+	writel(write_val, spec->mem_base + 0x204);
+	msleep(20);
+	readl(spec->mem_base + 0x860);
+	readl(spec->mem_base + 0x854);
+	readl(spec->mem_base + 0x840);
+
+	writel(0x00800004, spec->mem_base + 0x20c);
+	writel(0x00000000, spec->mem_base + 0x210);
+	readl(spec->mem_base + 0x210);
+	readl(spec->mem_base + 0x210);
+}
+
+/*
+ * Setup GPIO for the other variants of Core3D.
+ */
+
+/*
  * Sets up the GPIO pins so that they are discoverable. If this isn't done,
  * the card shows as having no GPIO pins.
  */
@@ -3119,6 +3334,7 @@ static void ca0132_gpio_init(struct hda_codec *codec)
 
 	switch (spec->quirk) {
 	case QUIRK_SBZ:
+	case QUIRK_AE5:
 		snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
 		snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
 		snd_hda_codec_write(codec, 0x01, 0, 0x790, 0x23);
@@ -3928,6 +4144,138 @@ exit:
 	return err < 0 ? err : 0;
 }
 
+static int ae5_headphone_gain_set(struct hda_codec *codec, long val);
+static int zxr_headphone_gain_set(struct hda_codec *codec, long val);
+static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val);
+
+static void ae5_mmio_select_out(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int i;
+
+	for (i = 0; i < AE5_CA0113_OUT_SET_COMMANDS; i++)
+		ca0113_mmio_command_set(codec,
+			ae5_ca0113_output_presets[spec->cur_out_type].group[i],
+			ae5_ca0113_output_presets[spec->cur_out_type].target[i],
+			ae5_ca0113_output_presets[spec->cur_out_type].vals[i]);
+}
+
+/*
+ * These are the commands needed to setup output on each of the different card
+ * types.
+ */
+static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp;
+
+	switch (spec->cur_out_type) {
+	case SPEAKER_OUT:
+		switch (spec->quirk) {
+		case QUIRK_SBZ:
+			ca0113_mmio_gpio_set(codec, 7, false);
+			ca0113_mmio_gpio_set(codec, 4, true);
+			ca0113_mmio_gpio_set(codec, 1, true);
+			chipio_set_control_param(codec, 0x0d, 0x18);
+			break;
+		case QUIRK_ZXR:
+			ca0113_mmio_gpio_set(codec, 2, true);
+			ca0113_mmio_gpio_set(codec, 3, true);
+			ca0113_mmio_gpio_set(codec, 5, false);
+			zxr_headphone_gain_set(codec, 0);
+			chipio_set_control_param(codec, 0x0d, 0x24);
+			break;
+		case QUIRK_R3DI:
+			chipio_set_control_param(codec, 0x0d, 0x24);
+			r3di_gpio_out_set(codec, R3DI_LINE_OUT);
+			break;
+		case QUIRK_R3D:
+			chipio_set_control_param(codec, 0x0d, 0x24);
+			ca0113_mmio_gpio_set(codec, 1, true);
+			break;
+		case QUIRK_AE5:
+			ae5_mmio_select_out(codec);
+			ae5_headphone_gain_set(codec, 2);
+			tmp = FLOAT_ZERO;
+			dspio_set_uint_param(codec, 0x96, 0x29, tmp);
+			dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
+			chipio_set_control_param(codec, 0x0d, 0xa4);
+			chipio_write(codec, 0x18b03c, 0x00000012);
+			break;
+		}
+		break;
+	case HEADPHONE_OUT:
+		switch (spec->quirk) {
+		case QUIRK_SBZ:
+			ca0113_mmio_gpio_set(codec, 7, true);
+			ca0113_mmio_gpio_set(codec, 4, true);
+			ca0113_mmio_gpio_set(codec, 1, false);
+			chipio_set_control_param(codec, 0x0d, 0x12);
+			break;
+		case QUIRK_ZXR:
+			ca0113_mmio_gpio_set(codec, 2, false);
+			ca0113_mmio_gpio_set(codec, 3, false);
+			ca0113_mmio_gpio_set(codec, 5, true);
+			zxr_headphone_gain_set(codec, spec->zxr_gain_set);
+			chipio_set_control_param(codec, 0x0d, 0x21);
+			break;
+		case QUIRK_R3DI:
+			chipio_set_control_param(codec, 0x0d, 0x21);
+			r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT);
+			break;
+		case QUIRK_R3D:
+			chipio_set_control_param(codec, 0x0d, 0x21);
+			ca0113_mmio_gpio_set(codec, 0x1, false);
+			break;
+		case QUIRK_AE5:
+			ae5_mmio_select_out(codec);
+			ae5_headphone_gain_set(codec,
+					spec->ae5_headphone_gain_val);
+			tmp = FLOAT_ONE;
+			dspio_set_uint_param(codec, 0x96, 0x29, tmp);
+			dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
+			chipio_set_control_param(codec, 0x0d, 0xa1);
+			chipio_write(codec, 0x18b03c, 0x00000012);
+			break;
+		}
+		break;
+	case SURROUND_OUT:
+		switch (spec->quirk) {
+		case QUIRK_SBZ:
+			ca0113_mmio_gpio_set(codec, 7, false);
+			ca0113_mmio_gpio_set(codec, 4, true);
+			ca0113_mmio_gpio_set(codec, 1, true);
+			chipio_set_control_param(codec, 0x0d, 0x18);
+			break;
+		case QUIRK_ZXR:
+			ca0113_mmio_gpio_set(codec, 2, true);
+			ca0113_mmio_gpio_set(codec, 3, true);
+			ca0113_mmio_gpio_set(codec, 5, false);
+			zxr_headphone_gain_set(codec, 0);
+			chipio_set_control_param(codec, 0x0d, 0x24);
+			break;
+		case QUIRK_R3DI:
+			chipio_set_control_param(codec, 0x0d, 0x24);
+			r3di_gpio_out_set(codec, R3DI_LINE_OUT);
+			break;
+		case QUIRK_R3D:
+			ca0113_mmio_gpio_set(codec, 1, true);
+			chipio_set_control_param(codec, 0x0d, 0x24);
+			break;
+		case QUIRK_AE5:
+			ae5_mmio_select_out(codec);
+			ae5_headphone_gain_set(codec, 2);
+			tmp = FLOAT_ZERO;
+			dspio_set_uint_param(codec, 0x96, 0x29, tmp);
+			dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
+			chipio_set_control_param(codec, 0x0d, 0xa4);
+			chipio_write(codec, 0x18b03c, 0x00000012);
+			break;
+		}
+		break;
+	}
+}
+
 /*
  * This function behaves similarly to the ca0132_select_out funciton above,
  * except with a few differences. It adds the ability to select the current
@@ -3978,26 +4326,11 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 	if (err < 0)
 		goto exit;
 
+	ca0132_alt_select_out_quirk_handler(codec);
+
 	switch (spec->cur_out_type) {
 	case SPEAKER_OUT:
 		codec_dbg(codec, "%s speaker\n", __func__);
-		/*speaker out config*/
-		switch (spec->quirk) {
-		case QUIRK_SBZ:
-			ca0132_mmio_gpio_set(codec, 7, false);
-			ca0132_mmio_gpio_set(codec, 4, true);
-			ca0132_mmio_gpio_set(codec, 1, true);
-			chipio_set_control_param(codec, 0x0D, 0x18);
-			break;
-		case QUIRK_R3DI:
-			chipio_set_control_param(codec, 0x0D, 0x24);
-			r3di_gpio_out_set(codec, R3DI_LINE_OUT);
-			break;
-		case QUIRK_R3D:
-			chipio_set_control_param(codec, 0x0D, 0x24);
-			ca0132_mmio_gpio_set(codec, 1, true);
-			break;
-		}
 
 		/* disable headphone node */
 		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0,
@@ -4021,23 +4354,6 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		break;
 	case HEADPHONE_OUT:
 		codec_dbg(codec, "%s hp\n", __func__);
-		/* Headphone out config*/
-		switch (spec->quirk) {
-		case QUIRK_SBZ:
-			ca0132_mmio_gpio_set(codec, 7, true);
-			ca0132_mmio_gpio_set(codec, 4, true);
-			ca0132_mmio_gpio_set(codec, 1, false);
-			chipio_set_control_param(codec, 0x0D, 0x12);
-			break;
-		case QUIRK_R3DI:
-			chipio_set_control_param(codec, 0x0D, 0x21);
-			r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT);
-			break;
-		case QUIRK_R3D:
-			chipio_set_control_param(codec, 0x0D, 0x21);
-			ca0132_mmio_gpio_set(codec, 0x1, false);
-			break;
-		}
 
 		snd_hda_codec_write(codec, spec->out_pins[0], 0,
 			AC_VERB_SET_EAPD_BTLENABLE, 0x00);
@@ -4067,23 +4383,7 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		break;
 	case SURROUND_OUT:
 		codec_dbg(codec, "%s surround\n", __func__);
-		/* Surround out config*/
-		switch (spec->quirk) {
-		case QUIRK_SBZ:
-			ca0132_mmio_gpio_set(codec, 7, false);
-			ca0132_mmio_gpio_set(codec, 4, true);
-			ca0132_mmio_gpio_set(codec, 1, true);
-			chipio_set_control_param(codec, 0x0D, 0x18);
-			break;
-		case QUIRK_R3DI:
-			chipio_set_control_param(codec, 0x0D, 0x24);
-			r3di_gpio_out_set(codec, R3DI_LINE_OUT);
-			break;
-		case QUIRK_R3D:
-			ca0132_mmio_gpio_set(codec, 1, true);
-			chipio_set_control_param(codec, 0x0D, 0x24);
-			break;
-		}
+
 		/* enable line out node */
 		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0,
 				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
@@ -4108,14 +4408,21 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		snd_hda_set_pin_ctl(codec, spec->out_pins[3],
 				    pin_ctl | PIN_OUT);
 
-		if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
-			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
-		else
-			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
+		dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
 		break;
 	}
+	/*
+	 * Surround always sets it's scp command to req 0x04 to FLOAT_EIGHT.
+	 * With this set though, X_BASS cannot be enabled. So, if we have OutFX
+	 * enabled, we need to make sure X_BASS is off, otherwise everything
+	 * sounds all muffled. Running ca0132_effects_set with X_BASS as the
+	 * effect should sort this out.
+	 */
+	if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
+		ca0132_effects_set(codec, X_BASS,
+			spec->effects_switch[X_BASS - EFFECT_START_NID]);
 
-	/* run through the output dsp commands for line-out */
+	/* run through the output dsp commands for the selected output. */
 	for (i = 0; i < alt_out_presets[spec->cur_out_type].commands; i++) {
 		err = dspio_set_uint_param(codec,
 		alt_out_presets[spec->cur_out_type].mids[i],
@@ -4152,7 +4459,6 @@ static void ca0132_unsol_hp_delayed(struct work_struct *work)
 
 static void ca0132_set_dmic(struct hda_codec *codec, int enable);
 static int ca0132_mic_boost_set(struct hda_codec *codec, long val);
-static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val);
 static void resume_mic1(struct hda_codec *codec, unsigned int oldval);
 static int stop_mic1(struct hda_codec *codec);
 static int ca0132_cvoice_switch_set(struct hda_codec *codec);
@@ -4341,13 +4647,20 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 		switch (spec->quirk) {
 		case QUIRK_SBZ:
 		case QUIRK_R3D:
-			ca0132_mmio_gpio_set(codec, 0, false);
+			ca0113_mmio_gpio_set(codec, 0, false);
+			tmp = FLOAT_THREE;
+			break;
+		case QUIRK_ZXR:
 			tmp = FLOAT_THREE;
 			break;
 		case QUIRK_R3DI:
 			r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
 			tmp = FLOAT_ONE;
 			break;
+		case QUIRK_AE5:
+			ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00);
+			tmp = FLOAT_THREE;
+			break;
 		default:
 			tmp = FLOAT_ONE;
 			break;
@@ -4362,10 +4675,19 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 
 		chipio_set_stream_control(codec, 0x03, 1);
 		chipio_set_stream_control(codec, 0x04, 1);
-
-		if (spec->quirk == QUIRK_SBZ) {
+		switch (spec->quirk) {
+		case QUIRK_SBZ:
 			chipio_write(codec, 0x18B098, 0x0000000C);
 			chipio_write(codec, 0x18B09C, 0x0000000C);
+			break;
+		case QUIRK_ZXR:
+			chipio_write(codec, 0x18B098, 0x0000000C);
+			chipio_write(codec, 0x18B09C, 0x000000CC);
+			break;
+		case QUIRK_AE5:
+			chipio_write(codec, 0x18B098, 0x0000000C);
+			chipio_write(codec, 0x18B09C, 0x0000004C);
+			break;
 		}
 		ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val);
 		break;
@@ -4374,11 +4696,14 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 		switch (spec->quirk) {
 		case QUIRK_SBZ:
 		case QUIRK_R3D:
-			ca0132_mmio_gpio_set(codec, 0, false);
+			ca0113_mmio_gpio_set(codec, 0, false);
 			break;
 		case QUIRK_R3DI:
 			r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
 			break;
+		case QUIRK_AE5:
+			ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00);
+			break;
 		}
 
 		chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
@@ -4389,11 +4714,13 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 		tmp = FLOAT_ZERO;
 		dspio_set_uint_param(codec, 0x80, 0x00, tmp);
 
-		if (spec->quirk == QUIRK_SBZ) {
+		switch (spec->quirk) {
+		case QUIRK_SBZ:
+		case QUIRK_AE5:
 			chipio_write(codec, 0x18B098, 0x00000000);
 			chipio_write(codec, 0x18B09C, 0x00000000);
+			break;
 		}
-
 		chipio_set_stream_control(codec, 0x03, 1);
 		chipio_set_stream_control(codec, 0x04, 1);
 		break;
@@ -4401,14 +4728,18 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 		switch (spec->quirk) {
 		case QUIRK_SBZ:
 		case QUIRK_R3D:
-			ca0132_mmio_gpio_set(codec, 0, true);
-			ca0132_mmio_gpio_set(codec, 5, false);
+			ca0113_mmio_gpio_set(codec, 0, true);
+			ca0113_mmio_gpio_set(codec, 5, false);
 			tmp = FLOAT_THREE;
 			break;
 		case QUIRK_R3DI:
 			r3di_gpio_mic_set(codec, R3DI_FRONT_MIC);
 			tmp = FLOAT_ONE;
 			break;
+		case QUIRK_AE5:
+			ca0113_mmio_command_set(codec, 0x48, 0x28, 0x3f);
+			tmp = FLOAT_THREE;
+			break;
 		default:
 			tmp = FLOAT_ONE;
 			break;
@@ -4424,9 +4755,15 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 		chipio_set_stream_control(codec, 0x03, 1);
 		chipio_set_stream_control(codec, 0x04, 1);
 
-		if (spec->quirk == QUIRK_SBZ) {
+		switch (spec->quirk) {
+		case QUIRK_SBZ:
 			chipio_write(codec, 0x18B098, 0x0000000C);
 			chipio_write(codec, 0x18B09C, 0x000000CC);
+			break;
+		case QUIRK_AE5:
+			chipio_write(codec, 0x18B098, 0x0000000C);
+			chipio_write(codec, 0x18B09C, 0x0000004C);
+			break;
 		}
 		ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val);
 		break;
@@ -4435,7 +4772,6 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 
 	snd_hda_power_down_pm(codec);
 	return 0;
-
 }
 
 /*
@@ -4507,6 +4843,8 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
 		/* if PE if off, turn off out effects. */
 		if (!spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
 			val = 0;
+		if (spec->cur_out_type == SURROUND_OUT && nid == X_BASS)
+			val = 0;
 	}
 
 	/* for in effect, qualify with CrystalVoice */
@@ -4520,7 +4858,7 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
 			val = 0;
 
 		/* If Voice Focus on SBZ, set to two channel. */
-		if ((nid == VOICE_FOCUS) && (spec->quirk == QUIRK_SBZ)
+		if ((nid == VOICE_FOCUS) && (spec->use_pci_mmio)
 				&& (spec->cur_mic_type != REAR_LINE_IN)) {
 			if (spec->effects_switch[CRYSTAL_VOICE -
 						 EFFECT_START_NID]) {
@@ -4539,7 +4877,7 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
 		 * For SBZ noise reduction, there's an extra command
 		 * to module ID 0x47. No clue why.
 		 */
-		if ((nid == NOISE_REDUCTION) && (spec->quirk == QUIRK_SBZ)
+		if ((nid == NOISE_REDUCTION) && (spec->use_pci_mmio)
 				&& (spec->cur_mic_type != REAR_LINE_IN)) {
 			if (spec->effects_switch[CRYSTAL_VOICE -
 						 EFFECT_START_NID]) {
@@ -4678,6 +5016,27 @@ static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val)
 	return ret;
 }
 
+static int ae5_headphone_gain_set(struct hda_codec *codec, long val)
+{
+	unsigned int i;
+
+	for (i = 0; i < 4; i++)
+		ca0113_mmio_command_set(codec, 0x48, 0x11 + i,
+				ae5_headphone_gain_presets[val].vals[i]);
+	return 0;
+}
+
+/*
+ * gpio pin 1 is a relay that switches on/off, apparently setting the headphone
+ * amplifier to handle a 600 ohm load.
+ */
+static int zxr_headphone_gain_set(struct hda_codec *codec, long val)
+{
+	ca0113_mmio_gpio_set(codec, 1, val);
+
+	return 0;
+}
+
 static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_value *ucontrol)
 {
@@ -4942,6 +5301,112 @@ static int ca0132_alt_mic_boost_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
+/*
+ * Sound BlasterX AE-5 Headphone Gain Controls.
+ */
+#define AE5_HEADPHONE_GAIN_MAX 3
+static int ae5_headphone_gain_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	char *sfx = " Ohms)";
+	char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = AE5_HEADPHONE_GAIN_MAX;
+	if (uinfo->value.enumerated.item >= AE5_HEADPHONE_GAIN_MAX)
+		uinfo->value.enumerated.item = AE5_HEADPHONE_GAIN_MAX - 1;
+	sprintf(namestr, "%s %s",
+		ae5_headphone_gain_presets[uinfo->value.enumerated.item].name,
+		sfx);
+	strcpy(uinfo->value.enumerated.name, namestr);
+	return 0;
+}
+
+static int ae5_headphone_gain_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+
+	ucontrol->value.enumerated.item[0] = spec->ae5_headphone_gain_val;
+	return 0;
+}
+
+static int ae5_headphone_gain_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+	int sel = ucontrol->value.enumerated.item[0];
+	unsigned int items = AE5_HEADPHONE_GAIN_MAX;
+
+	if (sel >= items)
+		return 0;
+
+	codec_dbg(codec, "ae5_headphone_gain: boost=%d\n",
+		    sel);
+
+	spec->ae5_headphone_gain_val = sel;
+
+	if (spec->out_enum_val == HEADPHONE_OUT)
+		ae5_headphone_gain_set(codec, spec->ae5_headphone_gain_val);
+
+	return 1;
+}
+
+/*
+ * Sound BlasterX AE-5 sound filter enumerated control.
+ */
+#define AE5_SOUND_FILTER_MAX 3
+
+static int ae5_sound_filter_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = AE5_SOUND_FILTER_MAX;
+	if (uinfo->value.enumerated.item >= AE5_SOUND_FILTER_MAX)
+		uinfo->value.enumerated.item = AE5_SOUND_FILTER_MAX - 1;
+	sprintf(namestr, "%s",
+			ae5_filter_presets[uinfo->value.enumerated.item].name);
+	strcpy(uinfo->value.enumerated.name, namestr);
+	return 0;
+}
+
+static int ae5_sound_filter_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+
+	ucontrol->value.enumerated.item[0] = spec->ae5_filter_val;
+	return 0;
+}
+
+static int ae5_sound_filter_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+	int sel = ucontrol->value.enumerated.item[0];
+	unsigned int items = AE5_SOUND_FILTER_MAX;
+
+	if (sel >= items)
+		return 0;
+
+	codec_dbg(codec, "ae5_sound_filter: %s\n",
+			ae5_filter_presets[sel].name);
+
+	spec->ae5_filter_val = sel;
+
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07,
+			ae5_filter_presets[sel].val);
+
+	return 1;
+}
 
 /*
  * Input Select Control for alternative ca0132 codecs. This exists because
@@ -5330,6 +5795,16 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
 		goto exit;
 	}
 
+	if (nid == ZXR_HEADPHONE_GAIN) {
+		spec->zxr_gain_set = *valp;
+		if (spec->cur_out_type == HEADPHONE_OUT)
+			changed = zxr_headphone_gain_set(codec, *valp);
+		else
+			changed = 0;
+
+		goto exit;
+	}
+
 exit:
 	snd_hda_power_down(codec);
 	return changed;
@@ -5705,6 +6180,50 @@ static int ca0132_alt_add_mic_boost_enum(struct hda_codec *codec)
 }
 
 /*
+ * Add headphone gain enumerated control for the AE-5. This switches between
+ * three modes, low, medium, and high. When non-headphone outputs are selected,
+ * it is automatically set to high. This is the same behavior as Windows.
+ */
+static int ae5_add_headphone_gain_enum(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_MUTE_MONO("AE-5: Headphone Gain",
+				    AE5_HEADPHONE_GAIN_ENUM, 1, 0, HDA_OUTPUT);
+	knew.info = ae5_headphone_gain_info;
+	knew.get = ae5_headphone_gain_get;
+	knew.put = ae5_headphone_gain_put;
+	return snd_hda_ctl_add(codec, AE5_HEADPHONE_GAIN_ENUM,
+				snd_ctl_new1(&knew, codec));
+}
+
+/*
+ * Add sound filter enumerated control for the AE-5. This adds three different
+ * settings: Slow Roll Off, Minimum Phase, and Fast Roll Off. From what I've
+ * read into it, it changes the DAC's interpolation filter.
+ */
+static int ae5_add_sound_filter_enum(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_MUTE_MONO("AE-5: Sound Filter",
+				    AE5_SOUND_FILTER_ENUM, 1, 0, HDA_OUTPUT);
+	knew.info = ae5_sound_filter_info;
+	knew.get = ae5_sound_filter_get;
+	knew.put = ae5_sound_filter_put;
+	return snd_hda_ctl_add(codec, AE5_SOUND_FILTER_ENUM,
+				snd_ctl_new1(&knew, codec));
+}
+
+static int zxr_add_headphone_gain_switch(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		CA0132_CODEC_MUTE_MONO("ZxR: 600 Ohm Gain",
+				    ZXR_HEADPHONE_GAIN, 1, HDA_OUTPUT);
+
+	return snd_hda_ctl_add(codec, ZXR_HEADPHONE_GAIN,
+				snd_ctl_new1(&knew, codec));
+}
+
+/*
  * Need to create slave controls for the alternate codecs that have surround
  * capabilities.
  */
@@ -5847,7 +6366,8 @@ static int ca0132_build_controls(struct hda_codec *codec)
 					    NULL, ca0132_alt_slave_pfxs,
 					    "Playback Switch",
 					    true, &spec->vmaster_mute.sw_kctl);
-
+		if (err < 0)
+			return err;
 	}
 
 	/* Add in and out effects controls.
@@ -5855,8 +6375,8 @@ static int ca0132_build_controls(struct hda_codec *codec)
 	 */
 	num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT;
 	for (i = 0; i < num_fx; i++) {
-		/* SBZ and R3D break if Echo Cancellation is used. */
-		if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3D) {
+		/* Desktop cards break if Echo Cancellation is used. */
+		if (spec->use_pci_mmio) {
 			if (i == (ECHO_CANCELLATION - IN_EFFECT_START_NID +
 						OUT_EFFECTS_COUNT))
 				continue;
@@ -5874,8 +6394,14 @@ static int ca0132_build_controls(struct hda_codec *codec)
 	 * prefix, and change PlayEnhancement and CrystalVoice to match.
 	 */
 	if (spec->use_alt_controls) {
-		ca0132_alt_add_svm_enum(codec);
-		add_ca0132_alt_eq_presets(codec);
+		err = ca0132_alt_add_svm_enum(codec);
+		if (err < 0)
+			return err;
+
+		err = add_ca0132_alt_eq_presets(codec);
+		if (err < 0)
+			return err;
+
 		err = add_fx_switch(codec, PLAY_ENHANCEMENT,
 					"Enable OutFX", 0);
 		if (err < 0)
@@ -5912,7 +6438,9 @@ static int ca0132_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 	}
-	add_voicefx(codec);
+	err = add_voicefx(codec);
+	if (err < 0)
+		return err;
 
 	/*
 	 * If the codec uses alt_functions, you need the enumerated controls
@@ -5920,9 +6448,36 @@ static int ca0132_build_controls(struct hda_codec *codec)
 	 * setting control.
 	 */
 	if (spec->use_alt_functions) {
-		ca0132_alt_add_output_enum(codec);
-		ca0132_alt_add_input_enum(codec);
-		ca0132_alt_add_mic_boost_enum(codec);
+		err = ca0132_alt_add_output_enum(codec);
+		if (err < 0)
+			return err;
+		err = ca0132_alt_add_mic_boost_enum(codec);
+		if (err < 0)
+			return err;
+		/*
+		 * ZxR only has microphone input, there is no front panel
+		 * header on the card, and aux-in is handled by the DBPro board.
+		 */
+		if (spec->quirk != QUIRK_ZXR) {
+			err = ca0132_alt_add_input_enum(codec);
+			if (err < 0)
+				return err;
+		}
+	}
+
+	if (spec->quirk == QUIRK_AE5) {
+		err = ae5_add_headphone_gain_enum(codec);
+		if (err < 0)
+			return err;
+		err = ae5_add_sound_filter_enum(codec);
+		if (err < 0)
+			return err;
+	}
+
+	if (spec->quirk == QUIRK_ZXR) {
+		err = zxr_add_headphone_gain_switch(codec);
+		if (err < 0)
+			return err;
 	}
 #ifdef ENABLE_TUNING_CONTROLS
 	add_tuning_ctls(codec);
@@ -5955,6 +6510,27 @@ static int ca0132_build_controls(struct hda_codec *codec)
 	return 0;
 }
 
+static int dbpro_build_controls(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	int err = 0;
+
+	if (spec->dig_out) {
+		err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out,
+				spec->dig_out);
+		if (err < 0)
+			return err;
+	}
+
+	if (spec->dig_in) {
+		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
 /*
  * PCM
  */
@@ -6058,6 +6634,40 @@ static int ca0132_build_pcms(struct hda_codec *codec)
 	return 0;
 }
 
+static int dbpro_build_pcms(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	struct hda_pcm *info;
+
+	info = snd_hda_codec_pcm_new(codec, "CA0132 Alt Analog");
+	if (!info)
+		return -ENOMEM;
+	info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture;
+	info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1;
+	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0];
+
+
+	if (!spec->dig_out && !spec->dig_in)
+		return 0;
+
+	info = snd_hda_codec_pcm_new(codec, "CA0132 Digital");
+	if (!info)
+		return -ENOMEM;
+	info->pcm_type = HDA_PCM_TYPE_SPDIF;
+	if (spec->dig_out) {
+		info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+			ca0132_pcm_digital_playback;
+		info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dig_out;
+	}
+	if (spec->dig_in) {
+		info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+			ca0132_pcm_digital_capture;
+		info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in;
+	}
+
+	return 0;
+}
+
 static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
 {
 	if (pin) {
@@ -6238,69 +6848,48 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec)
 }
 
 /*
- * Recon3D r3d_setup_defaults sub functions.
+ * Creates a dummy stream to bind the output to. This seems to have to be done
+ * after changing the main outputs source and destination streams.
  */
-
-static void r3d_dsp_scp_startup(struct hda_codec *codec)
+static void ca0132_alt_create_dummy_stream(struct hda_codec *codec)
 {
-	unsigned int tmp;
-
-	tmp = 0x00000000;
-	dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
-
-	tmp = 0x00000001;
-	dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
-
-	tmp = 0x00000004;
-	dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
-	tmp = 0x00000005;
-	dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
-	tmp = 0x00000000;
-	dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
-}
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int stream_format;
 
-static void r3d_dsp_initial_mic_setup(struct hda_codec *codec)
-{
-	unsigned int tmp;
+	stream_format = snd_hdac_calc_stream_format(48000, 2,
+			SNDRV_PCM_FORMAT_S32_LE, 32, 0);
 
-	/* Mic 1 Setup */
-	chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
-	chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
-	/* This ConnPointID is unique to Recon3Di. Haven't seen it elsewhere */
-	chipio_set_conn_rate(codec, 0x0F, SR_96_000);
-	tmp = FLOAT_ONE;
-	dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+	snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id,
+					0, stream_format);
 
-	/* Mic 2 Setup, even though it isn't connected on SBZ */
-	chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000);
-	chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000);
-	chipio_set_conn_rate(codec, 0x0F, SR_96_000);
-	tmp = FLOAT_ZERO;
-	dspio_set_uint_param(codec, 0x80, 0x01, tmp);
+	snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
 }
 
 /*
- * Initialize Sound Blaster Z analog microphones.
+ * Initialize mic for non-chromebook ca0132 implementations.
  */
-static void sbz_init_analog_mics(struct hda_codec *codec)
+static void ca0132_alt_init_analog_mics(struct hda_codec *codec)
 {
+	struct ca0132_spec *spec = codec->spec;
 	unsigned int tmp;
 
 	/* Mic 1 Setup */
 	chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
 	chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
-	tmp = FLOAT_THREE;
+	if (spec->quirk == QUIRK_R3DI) {
+		chipio_set_conn_rate(codec, 0x0F, SR_96_000);
+		tmp = FLOAT_ONE;
+	} else
+		tmp = FLOAT_THREE;
 	dspio_set_uint_param(codec, 0x80, 0x00, tmp);
 
-	/* Mic 2 Setup, even though it isn't connected on SBZ */
+	/* Mic 2 setup (not present on desktop cards) */
 	chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000);
 	chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000);
+	if (spec->quirk == QUIRK_R3DI)
+		chipio_set_conn_rate(codec, 0x0F, SR_96_000);
 	tmp = FLOAT_ZERO;
 	dspio_set_uint_param(codec, 0x80, 0x01, tmp);
-
 }
 
 /*
@@ -6333,7 +6922,6 @@ static void sbz_connect_streams(struct hda_codec *codec)
 	codec_dbg(codec, "Connect Streams exited, mutex released.\n");
 
 	mutex_unlock(&spec->chipio_mutex);
-
 }
 
 /*
@@ -6360,19 +6948,29 @@ static void sbz_chipio_startup_data(struct hda_codec *codec)
 	chipio_set_stream_channels(codec, 0x0C, 6);
 	chipio_set_stream_control(codec, 0x0C, 1);
 	/* No clue what these control */
-	chipio_write_no_mutex(codec, 0x190030, 0x0001e0c0);
-	chipio_write_no_mutex(codec, 0x190034, 0x0001e1c1);
-	chipio_write_no_mutex(codec, 0x190038, 0x0001e4c2);
-	chipio_write_no_mutex(codec, 0x19003c, 0x0001e5c3);
-	chipio_write_no_mutex(codec, 0x190040, 0x0001e2c4);
-	chipio_write_no_mutex(codec, 0x190044, 0x0001e3c5);
-	chipio_write_no_mutex(codec, 0x190048, 0x0001e8c6);
-	chipio_write_no_mutex(codec, 0x19004c, 0x0001e9c7);
-	chipio_write_no_mutex(codec, 0x190050, 0x0001ecc8);
-	chipio_write_no_mutex(codec, 0x190054, 0x0001edc9);
-	chipio_write_no_mutex(codec, 0x190058, 0x0001eaca);
-	chipio_write_no_mutex(codec, 0x19005c, 0x0001ebcb);
-
+	if (spec->quirk == QUIRK_SBZ) {
+		chipio_write_no_mutex(codec, 0x190030, 0x0001e0c0);
+		chipio_write_no_mutex(codec, 0x190034, 0x0001e1c1);
+		chipio_write_no_mutex(codec, 0x190038, 0x0001e4c2);
+		chipio_write_no_mutex(codec, 0x19003c, 0x0001e5c3);
+		chipio_write_no_mutex(codec, 0x190040, 0x0001e2c4);
+		chipio_write_no_mutex(codec, 0x190044, 0x0001e3c5);
+		chipio_write_no_mutex(codec, 0x190048, 0x0001e8c6);
+		chipio_write_no_mutex(codec, 0x19004c, 0x0001e9c7);
+		chipio_write_no_mutex(codec, 0x190050, 0x0001ecc8);
+		chipio_write_no_mutex(codec, 0x190054, 0x0001edc9);
+		chipio_write_no_mutex(codec, 0x190058, 0x0001eaca);
+		chipio_write_no_mutex(codec, 0x19005c, 0x0001ebcb);
+	} else if (spec->quirk == QUIRK_ZXR) {
+		chipio_write_no_mutex(codec, 0x190038, 0x000140c2);
+		chipio_write_no_mutex(codec, 0x19003c, 0x000141c3);
+		chipio_write_no_mutex(codec, 0x190040, 0x000150c4);
+		chipio_write_no_mutex(codec, 0x190044, 0x000151c5);
+		chipio_write_no_mutex(codec, 0x190050, 0x000142c8);
+		chipio_write_no_mutex(codec, 0x190054, 0x000143c9);
+		chipio_write_no_mutex(codec, 0x190058, 0x000152ca);
+		chipio_write_no_mutex(codec, 0x19005c, 0x000153cb);
+	}
 	chipio_write_no_mutex(codec, 0x19042c, 0x00000001);
 
 	codec_dbg(codec, "Startup Data exited, mutex released.\n");
@@ -6380,35 +6978,56 @@ static void sbz_chipio_startup_data(struct hda_codec *codec)
 }
 
 /*
- * Sound Blaster Z uses these after DSP is loaded. Weird SCP commands
- * without a 0x20 source like normal.
+ * Custom DSP SCP commands where the src value is 0x00 instead of 0x20. This is
+ * done after the DSP is loaded.
  */
-static void sbz_dsp_scp_startup(struct hda_codec *codec)
+static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec)
 {
-	unsigned int tmp;
-
-	tmp = 0x00000003;
-	dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
-	tmp = 0x00000000;
-	dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
-
-	tmp = 0x00000001;
-	dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
-
-	tmp = 0x00000004;
-	dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
-	tmp = 0x00000005;
-	dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
-
-	tmp = 0x00000000;
-	dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp, i;
 
+	/*
+	 * Gotta run these twice, or else mic works inconsistently. Not clear
+	 * why this is, but multiple tests have confirmed it.
+	 */
+	for (i = 0; i < 2; i++) {
+		switch (spec->quirk) {
+		case QUIRK_SBZ:
+		case QUIRK_AE5:
+			tmp = 0x00000003;
+			dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+			tmp = 0x00000000;
+			dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
+			tmp = 0x00000001;
+			dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
+			tmp = 0x00000004;
+			dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+			tmp = 0x00000005;
+			dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+			tmp = 0x00000000;
+			dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+			break;
+		case QUIRK_R3D:
+		case QUIRK_R3DI:
+			tmp = 0x00000000;
+			dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
+			tmp = 0x00000001;
+			dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
+			tmp = 0x00000004;
+			dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+			tmp = 0x00000005;
+			dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+			tmp = 0x00000000;
+			dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+			break;
+		}
+		msleep(100);
+	}
 }
 
-static void sbz_dsp_initial_mic_setup(struct hda_codec *codec)
+static void ca0132_alt_dsp_initial_mic_setup(struct hda_codec *codec)
 {
+	struct ca0132_spec *spec = codec->spec;
 	unsigned int tmp;
 
 	chipio_set_stream_control(codec, 0x03, 0);
@@ -6423,8 +7042,161 @@ static void sbz_dsp_initial_mic_setup(struct hda_codec *codec)
 	chipio_set_stream_control(codec, 0x03, 1);
 	chipio_set_stream_control(codec, 0x04, 1);
 
-	chipio_write(codec, 0x18b098, 0x0000000c);
-	chipio_write(codec, 0x18b09C, 0x0000000c);
+	switch (spec->quirk) {
+	case QUIRK_SBZ:
+		chipio_write(codec, 0x18b098, 0x0000000c);
+		chipio_write(codec, 0x18b09C, 0x0000000c);
+		break;
+	case QUIRK_AE5:
+		chipio_write(codec, 0x18b098, 0x0000000c);
+		chipio_write(codec, 0x18b09c, 0x0000004c);
+		break;
+	}
+}
+
+static void ae5_post_dsp_register_set(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	chipio_8051_write_direct(codec, 0x93, 0x10);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2);
+
+	writeb(0xff, spec->mem_base + 0x304);
+	writeb(0xff, spec->mem_base + 0x304);
+	writeb(0xff, spec->mem_base + 0x304);
+	writeb(0xff, spec->mem_base + 0x304);
+	writeb(0x00, spec->mem_base + 0x100);
+	writeb(0xff, spec->mem_base + 0x304);
+	writeb(0x00, spec->mem_base + 0x100);
+	writeb(0xff, spec->mem_base + 0x304);
+	writeb(0x00, spec->mem_base + 0x100);
+	writeb(0xff, spec->mem_base + 0x304);
+	writeb(0x00, spec->mem_base + 0x100);
+	writeb(0xff, spec->mem_base + 0x304);
+
+	ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x3f);
+	ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f);
+	ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+}
+
+static void ae5_post_dsp_param_setup(struct hda_codec *codec)
+{
+	/*
+	 * Param3 in the 8051's memory is represented by the ascii string 'mch'
+	 * which seems to be 'multichannel'. This is also mentioned in the
+	 * AE-5's registry values in Windows.
+	 */
+	chipio_set_control_param(codec, 3, 0);
+	/*
+	 * I believe ASI is 'audio serial interface' and that it's used to
+	 * change colors on the external LED strip connected to the AE-5.
+	 */
+	chipio_set_control_flag(codec, CONTROL_FLAG_ASI_96KHZ, 1);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x724, 0x83);
+	chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x92);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0xfa);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_DATA_WRITE, 0x22);
+}
+
+static void ae5_post_dsp_pll_setup(struct hda_codec *codec)
+{
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x41);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc8);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x45);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xcc);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x40);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xcb);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x51);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0x8d);
+}
+
+static void ae5_post_dsp_stream_setup(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	mutex_lock(&spec->chipio_mutex);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x725, 0x81);
+
+	chipio_set_conn_rate_no_mutex(codec, 0x70, SR_96_000);
+
+	chipio_set_stream_channels(codec, 0x0C, 6);
+	chipio_set_stream_control(codec, 0x0C, 1);
+
+	chipio_set_stream_source_dest(codec, 0x5, 0x43, 0x0);
+
+	chipio_set_stream_source_dest(codec, 0x18, 0x9, 0xd0);
+	chipio_set_conn_rate_no_mutex(codec, 0xd0, SR_96_000);
+	chipio_set_stream_channels(codec, 0x18, 6);
+	chipio_set_stream_control(codec, 0x18, 1);
+
+	chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 4);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7);
+
+	ca0113_mmio_command_set(codec, 0x48, 0x01, 0x80);
+
+	mutex_unlock(&spec->chipio_mutex);
+}
+
+static void ae5_post_dsp_startup_data(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	mutex_lock(&spec->chipio_mutex);
+
+	chipio_write_no_mutex(codec, 0x189000, 0x0001f101);
+	chipio_write_no_mutex(codec, 0x189004, 0x0001f101);
+	chipio_write_no_mutex(codec, 0x189024, 0x00014004);
+	chipio_write_no_mutex(codec, 0x189028, 0x0002000f);
+
+	ca0113_mmio_command_set(codec, 0x48, 0x0a, 0x05);
+	chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 7);
+	ca0113_mmio_command_set(codec, 0x48, 0x0b, 0x12);
+	ca0113_mmio_command_set(codec, 0x48, 0x04, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x06, 0x48);
+	ca0113_mmio_command_set(codec, 0x48, 0x0a, 0x05);
+	ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+	ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00);
+	ca0113_mmio_gpio_set(codec, 0, true);
+	ca0113_mmio_gpio_set(codec, 1, true);
+	ca0113_mmio_command_set(codec, 0x48, 0x07, 0x80);
+
+	chipio_write_no_mutex(codec, 0x18b03c, 0x00000012);
+
+	ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00);
+
+	mutex_unlock(&spec->chipio_mutex);
 }
 
 /*
@@ -6485,9 +7257,8 @@ static void r3d_setup_defaults(struct hda_codec *codec)
 	if (spec->dsp_state != DSP_DOWNLOADED)
 		return;
 
-	r3d_dsp_scp_startup(codec);
-
-	r3d_dsp_initial_mic_setup(codec);
+	ca0132_alt_dsp_scp_startup(codec);
+	ca0132_alt_init_analog_mics(codec);
 
 	/*remove DSP headroom*/
 	tmp = FLOAT_ZERO;
@@ -6523,19 +7294,16 @@ static void r3d_setup_defaults(struct hda_codec *codec)
 static void sbz_setup_defaults(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
-	unsigned int tmp, stream_format;
+	unsigned int tmp;
 	int num_fx;
 	int idx, i;
 
 	if (spec->dsp_state != DSP_DOWNLOADED)
 		return;
 
-	sbz_dsp_scp_startup(codec);
-
-	sbz_init_analog_mics(codec);
-
+	ca0132_alt_dsp_scp_startup(codec);
+	ca0132_alt_init_analog_mics(codec);
 	sbz_connect_streams(codec);
-
 	sbz_chipio_startup_data(codec);
 
 	chipio_set_stream_control(codec, 0x03, 1);
@@ -6561,8 +7329,7 @@ static void sbz_setup_defaults(struct hda_codec *codec)
 	/* Set speaker source? */
 	dspio_set_uint_param(codec, 0x32, 0x00, tmp);
 
-	sbz_dsp_initial_mic_setup(codec);
-
+	ca0132_alt_dsp_initial_mic_setup(codec);
 
 	/* out, in effects + voicefx */
 	num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1;
@@ -6575,23 +7342,74 @@ static void sbz_setup_defaults(struct hda_codec *codec)
 		}
 	}
 
-	/*
-	 * Have to make a stream to bind the sound output to, otherwise
-	 * you'll get dead audio. Before I did this, it would bind to an
-	 * audio input, and would never work
-	 */
-	stream_format = snd_hdac_calc_stream_format(48000, 2,
-			SNDRV_PCM_FORMAT_S32_LE, 32, 0);
+	ca0132_alt_create_dummy_stream(codec);
+}
 
-	snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id,
-					0, stream_format);
+/*
+ * Setup default parameters for the Sound BlasterX AE-5 DSP.
+ */
+static void ae5_setup_defaults(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp;
+	int num_fx;
+	int idx, i;
 
-	snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
+	if (spec->dsp_state != DSP_DOWNLOADED)
+		return;
 
-	snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id,
-					0, stream_format);
+	ca0132_alt_dsp_scp_startup(codec);
+	ca0132_alt_init_analog_mics(codec);
+	chipio_set_stream_control(codec, 0x03, 1);
+	chipio_set_stream_control(codec, 0x04, 1);
 
-	snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
+	/* New, unknown SCP req's */
+	tmp = FLOAT_ZERO;
+	dspio_set_uint_param(codec, 0x96, 0x29, tmp);
+	dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
+	dspio_set_uint_param(codec, 0x80, 0x0d, tmp);
+	dspio_set_uint_param(codec, 0x80, 0x0e, tmp);
+
+	ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f);
+	ca0113_mmio_gpio_set(codec, 0, false);
+	ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
+
+	/* Internal loopback off */
+	tmp = FLOAT_ONE;
+	dspio_set_uint_param(codec, 0x37, 0x08, tmp);
+	dspio_set_uint_param(codec, 0x37, 0x10, tmp);
+
+	/*remove DSP headroom*/
+	tmp = FLOAT_ZERO;
+	dspio_set_uint_param(codec, 0x96, 0x3C, tmp);
+
+	/* set WUH source */
+	tmp = FLOAT_TWO;
+	dspio_set_uint_param(codec, 0x31, 0x00, tmp);
+	chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000);
+
+	/* Set speaker source? */
+	dspio_set_uint_param(codec, 0x32, 0x00, tmp);
+
+	ca0132_alt_dsp_initial_mic_setup(codec);
+	ae5_post_dsp_register_set(codec);
+	ae5_post_dsp_param_setup(codec);
+	ae5_post_dsp_pll_setup(codec);
+	ae5_post_dsp_stream_setup(codec);
+	ae5_post_dsp_startup_data(codec);
+
+	/* out, in effects + voicefx */
+	num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1;
+	for (idx = 0; idx < num_fx; idx++) {
+		for (i = 0; i <= ca0132_effects[idx].params; i++) {
+			dspio_set_uint_param(codec,
+					ca0132_effects[idx].mid,
+					ca0132_effects[idx].reqs[i],
+					ca0132_effects[idx].def_vals[i]);
+		}
+	}
+
+	ca0132_alt_create_dummy_stream(codec);
 }
 
 /*
@@ -6673,12 +7491,14 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec)
 	 */
 	switch (spec->quirk) {
 	case QUIRK_SBZ:
-		if (request_firmware(&fw_entry, SBZ_EFX_FILE,
+	case QUIRK_R3D:
+	case QUIRK_AE5:
+		if (request_firmware(&fw_entry, DESKTOP_EFX_FILE,
 					codec->card->dev) != 0) {
-			codec_dbg(codec, "SBZ alt firmware not detected. ");
+			codec_dbg(codec, "Desktop firmware not found.");
 			spec->alt_firmware_present = false;
 		} else {
-			codec_dbg(codec, "Sound Blaster Z firmware selected.");
+			codec_dbg(codec, "Desktop firmware selected.");
 			spec->alt_firmware_present = true;
 		}
 		break;
@@ -6921,6 +7741,14 @@ static void ca0132_init_chip(struct hda_codec *codec)
 	spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID] = 1;
 	spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID] = 0;
 
+	/*
+	 * The ZxR doesn't have a front panel header, and it's line-in is on
+	 * the daughter board. So, there is no input enum control, and we need
+	 * to make sure that spec->in_enum_val is set properly.
+	 */
+	if (spec->quirk == QUIRK_ZXR)
+		spec->in_enum_val = REAR_MIC;
+
 #ifdef ENABLE_TUNING_CONTROLS
 	ca0132_init_tuning_defaults(codec);
 #endif
@@ -6948,11 +7776,11 @@ static void sbz_region2_exit(struct hda_codec *codec)
 	for (i = 0; i < 8; i++)
 		writeb(0xb3, spec->mem_base + 0x304);
 
-	ca0132_mmio_gpio_set(codec, 0, false);
-	ca0132_mmio_gpio_set(codec, 1, false);
-	ca0132_mmio_gpio_set(codec, 4, true);
-	ca0132_mmio_gpio_set(codec, 5, false);
-	ca0132_mmio_gpio_set(codec, 7, false);
+	ca0113_mmio_gpio_set(codec, 0, false);
+	ca0113_mmio_gpio_set(codec, 1, false);
+	ca0113_mmio_gpio_set(codec, 4, true);
+	ca0113_mmio_gpio_set(codec, 5, false);
+	ca0113_mmio_gpio_set(codec, 7, false);
 }
 
 static void sbz_set_pin_ctl_default(struct hda_codec *codec)
@@ -6995,6 +7823,16 @@ static void sbz_gpio_shutdown_commands(struct hda_codec *codec, int dir,
 				AC_VERB_SET_GPIO_DATA, data);
 }
 
+static void zxr_dbpro_power_state_shutdown(struct hda_codec *codec)
+{
+	hda_nid_t pins[7] = {0x05, 0x0c, 0x09, 0x0e, 0x08, 0x11, 0x01};
+	unsigned int i;
+
+	for (i = 0; i < 7; i++)
+		snd_hda_codec_write(codec, pins[i], 0,
+				AC_VERB_SET_POWER_STATE, 0x03);
+}
+
 static void sbz_exit_chip(struct hda_codec *codec)
 {
 	chipio_set_stream_control(codec, 0x03, 0);
@@ -7037,6 +7875,61 @@ static void r3d_exit_chip(struct hda_codec *codec)
 	snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5b);
 }
 
+static void ae5_exit_chip(struct hda_codec *codec)
+{
+	chipio_set_stream_control(codec, 0x03, 0);
+	chipio_set_stream_control(codec, 0x04, 0);
+
+	ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
+	ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+	ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+	ca0113_mmio_command_set(codec, 0x30, 0x30, 0x00);
+	ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x00);
+	ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x00);
+	ca0113_mmio_gpio_set(codec, 0, false);
+	ca0113_mmio_gpio_set(codec, 1, false);
+
+	snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
+	snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
+
+	chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0);
+
+	chipio_set_stream_control(codec, 0x18, 0);
+	chipio_set_stream_control(codec, 0x0c, 0);
+
+	snd_hda_codec_write(codec, 0x01, 0, 0x724, 0x83);
+}
+
+static void zxr_exit_chip(struct hda_codec *codec)
+{
+	chipio_set_stream_control(codec, 0x03, 0);
+	chipio_set_stream_control(codec, 0x04, 0);
+	chipio_set_stream_control(codec, 0x14, 0);
+	chipio_set_stream_control(codec, 0x0C, 0);
+
+	chipio_set_conn_rate(codec, 0x41, SR_192_000);
+	chipio_set_conn_rate(codec, 0x91, SR_192_000);
+
+	chipio_write(codec, 0x18a020, 0x00000083);
+
+	snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
+	snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
+
+	ca0132_clear_unsolicited(codec);
+	sbz_set_pin_ctl_default(codec);
+	snd_hda_codec_write(codec, 0x0B, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x00);
+
+	ca0113_mmio_gpio_set(codec, 5, false);
+	ca0113_mmio_gpio_set(codec, 2, false);
+	ca0113_mmio_gpio_set(codec, 3, false);
+	ca0113_mmio_gpio_set(codec, 0, false);
+	ca0113_mmio_gpio_set(codec, 4, true);
+	ca0113_mmio_gpio_set(codec, 0, true);
+	ca0113_mmio_gpio_set(codec, 5, true);
+	ca0113_mmio_gpio_set(codec, 2, false);
+	ca0113_mmio_gpio_set(codec, 3, false);
+}
+
 static void ca0132_exit_chip(struct hda_codec *codec)
 {
 	/* put any chip cleanup stuffs here. */
@@ -7140,11 +8033,6 @@ static void sbz_pre_dsp_setup(struct hda_codec *codec)
 	writel(0x00820680, spec->mem_base + 0x01C);
 	writel(0x00820680, spec->mem_base + 0x01C);
 
-	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfc);
-	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfd);
-	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfe);
-	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xff);
-
 	chipio_write(codec, 0x18b0a4, 0x000000c2);
 
 	snd_hda_codec_write(codec, 0x11, 0,
@@ -7153,12 +8041,6 @@ static void sbz_pre_dsp_setup(struct hda_codec *codec)
 
 static void r3d_pre_dsp_setup(struct hda_codec *codec)
 {
-
-	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfc);
-	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfd);
-	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfe);
-	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xff);
-
 	chipio_write(codec, 0x18b0a4, 0x000000c2);
 
 	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
@@ -7205,23 +8087,116 @@ static void ca0132_mmio_init(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
 
-	writel(0x00000000, spec->mem_base + 0x400);
-	writel(0x00000000, spec->mem_base + 0x408);
-	writel(0x00000000, spec->mem_base + 0x40C);
-	writel(0x00880680, spec->mem_base + 0x01C);
-	writel(0x00000083, spec->mem_base + 0xC0C);
+	if (spec->quirk == QUIRK_AE5)
+		writel(0x00000001, spec->mem_base + 0x400);
+	else
+		writel(0x00000000, spec->mem_base + 0x400);
+
+	if (spec->quirk == QUIRK_AE5)
+		writel(0x00000001, spec->mem_base + 0x408);
+	else
+		writel(0x00000000, spec->mem_base + 0x408);
+
+	if (spec->quirk == QUIRK_AE5)
+		writel(0x00000001, spec->mem_base + 0x40c);
+	else
+		writel(0x00000000, spec->mem_base + 0x40C);
+
+	if (spec->quirk == QUIRK_ZXR)
+		writel(0x00880640, spec->mem_base + 0x01C);
+	else
+		writel(0x00880680, spec->mem_base + 0x01C);
+
+	if (spec->quirk == QUIRK_AE5)
+		writel(0x00000080, spec->mem_base + 0xC0C);
+	else
+		writel(0x00000083, spec->mem_base + 0xC0C);
+
 	writel(0x00000030, spec->mem_base + 0xC00);
 	writel(0x00000000, spec->mem_base + 0xC04);
+
+	if (spec->quirk == QUIRK_AE5)
+		writel(0x00000000, spec->mem_base + 0xC0C);
+	else
+		writel(0x00000003, spec->mem_base + 0xC0C);
+
 	writel(0x00000003, spec->mem_base + 0xC0C);
 	writel(0x00000003, spec->mem_base + 0xC0C);
 	writel(0x00000003, spec->mem_base + 0xC0C);
-	writel(0x00000003, spec->mem_base + 0xC0C);
-	writel(0x000000C1, spec->mem_base + 0xC08);
+
+	if (spec->quirk == QUIRK_AE5)
+		writel(0x00000001, spec->mem_base + 0xC08);
+	else
+		writel(0x000000C1, spec->mem_base + 0xC08);
+
 	writel(0x000000F1, spec->mem_base + 0xC08);
 	writel(0x00000001, spec->mem_base + 0xC08);
 	writel(0x000000C7, spec->mem_base + 0xC08);
 	writel(0x000000C1, spec->mem_base + 0xC08);
 	writel(0x00000080, spec->mem_base + 0xC04);
+
+	if (spec->quirk == QUIRK_AE5) {
+		writel(0x00000000, spec->mem_base + 0x42c);
+		writel(0x00000000, spec->mem_base + 0x46c);
+		writel(0x00000000, spec->mem_base + 0x4ac);
+		writel(0x00000000, spec->mem_base + 0x4ec);
+		writel(0x00000000, spec->mem_base + 0x43c);
+		writel(0x00000000, spec->mem_base + 0x47c);
+		writel(0x00000000, spec->mem_base + 0x4bc);
+		writel(0x00000000, spec->mem_base + 0x4fc);
+		writel(0x00000600, spec->mem_base + 0x100);
+		writel(0x00000014, spec->mem_base + 0x410);
+		writel(0x0000060f, spec->mem_base + 0x100);
+		writel(0x0000070f, spec->mem_base + 0x100);
+		writel(0x00000aff, spec->mem_base + 0x830);
+		writel(0x00000000, spec->mem_base + 0x86c);
+		writel(0x0000006b, spec->mem_base + 0x800);
+		writel(0x00000001, spec->mem_base + 0x86c);
+		writel(0x0000006b, spec->mem_base + 0x800);
+		writel(0x00000057, spec->mem_base + 0x804);
+		writel(0x00800000, spec->mem_base + 0x20c);
+	}
+}
+
+/*
+ * This function writes to some SFR's, does some region2 writes, and then
+ * eventually resets the codec with the 0x7ff verb. Not quite sure why it does
+ * what it does.
+ */
+static void ae5_register_set(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	chipio_8051_write_direct(codec, 0x93, 0x10);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2);
+
+	writeb(0x0f, spec->mem_base + 0x304);
+	writeb(0x0f, spec->mem_base + 0x304);
+	writeb(0x0f, spec->mem_base + 0x304);
+	writeb(0x0f, spec->mem_base + 0x304);
+	writeb(0x0e, spec->mem_base + 0x100);
+	writeb(0x1f, spec->mem_base + 0x304);
+	writeb(0x0c, spec->mem_base + 0x100);
+	writeb(0x3f, spec->mem_base + 0x304);
+	writeb(0x08, spec->mem_base + 0x100);
+	writeb(0x7f, spec->mem_base + 0x304);
+	writeb(0x00, spec->mem_base + 0x100);
+	writeb(0xff, spec->mem_base + 0x304);
+
+	ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f);
+
+	chipio_8051_write_direct(codec, 0x90, 0x00);
+	chipio_8051_write_direct(codec, 0x90, 0x10);
+
+	ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+
+	chipio_write(codec, 0x18b0a4, 0x000000c2);
+
+	snd_hda_codec_write(codec, 0x01, 0, 0x7ff, 0x00);
+	snd_hda_codec_write(codec, 0x01, 0, 0x7ff, 0x00);
 }
 
 /*
@@ -7257,6 +8232,21 @@ static void ca0132_alt_init(struct hda_codec *codec)
 		snd_hda_sequence_write(codec, spec->chip_init_verbs);
 		snd_hda_sequence_write(codec, spec->desktop_init_verbs);
 		break;
+	case QUIRK_AE5:
+		ca0132_gpio_init(codec);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x49);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				VENDOR_CHIPIO_PLL_PMU_WRITE, 0x88);
+		chipio_write(codec, 0x18b030, 0x00000020);
+		snd_hda_sequence_write(codec, spec->chip_init_verbs);
+		snd_hda_sequence_write(codec, spec->desktop_init_verbs);
+		ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
+		break;
+	case QUIRK_ZXR:
+		snd_hda_sequence_write(codec, spec->chip_init_verbs);
+		snd_hda_sequence_write(codec, spec->desktop_init_verbs);
+		break;
 	}
 }
 
@@ -7298,6 +8288,9 @@ static int ca0132_init(struct hda_codec *codec)
 
 	snd_hda_power_up_pm(codec);
 
+	if (spec->quirk == QUIRK_AE5)
+		ae5_register_set(codec);
+
 	ca0132_init_unsol(codec);
 	ca0132_init_params(codec);
 	ca0132_init_flags(codec);
@@ -7317,8 +8310,12 @@ static int ca0132_init(struct hda_codec *codec)
 		r3d_setup_defaults(codec);
 		break;
 	case QUIRK_SBZ:
+	case QUIRK_ZXR:
 		sbz_setup_defaults(codec);
 		break;
+	case QUIRK_AE5:
+		ae5_setup_defaults(codec);
+		break;
 	default:
 		ca0132_setup_defaults(codec);
 		ca0132_init_analog_mic2(codec);
@@ -7372,6 +8369,21 @@ static int ca0132_init(struct hda_codec *codec)
 	return 0;
 }
 
+static int dbpro_init(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	unsigned int i;
+
+	init_output(codec, cfg->dig_out_pins[0], spec->dig_out);
+	init_input(codec, cfg->dig_in_pin, spec->dig_in);
+
+	for (i = 0; i < spec->num_inputs; i++)
+		init_input(codec, spec->input_pins[i], spec->adcs[i]);
+
+	return 0;
+}
+
 static void ca0132_free(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
@@ -7382,9 +8394,15 @@ static void ca0132_free(struct hda_codec *codec)
 	case QUIRK_SBZ:
 		sbz_exit_chip(codec);
 		break;
+	case QUIRK_ZXR:
+		zxr_exit_chip(codec);
+		break;
 	case QUIRK_R3D:
 		r3d_exit_chip(codec);
 		break;
+	case QUIRK_AE5:
+		ae5_exit_chip(codec);
+		break;
 	case QUIRK_R3DI:
 		r3di_gpio_shutdown(codec);
 		break;
@@ -7400,6 +8418,16 @@ static void ca0132_free(struct hda_codec *codec)
 	kfree(codec->spec);
 }
 
+static void dbpro_free(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	zxr_dbpro_power_state_shutdown(codec);
+
+	kfree(spec->spec_init_verbs);
+	kfree(codec->spec);
+}
+
 static void ca0132_reboot_notify(struct hda_codec *codec)
 {
 	codec->patch_ops.free(codec);
@@ -7414,6 +8442,13 @@ static const struct hda_codec_ops ca0132_patch_ops = {
 	.reboot_notify = ca0132_reboot_notify,
 };
 
+static const struct hda_codec_ops dbpro_patch_ops = {
+	.build_controls = dbpro_build_controls,
+	.build_pcms = dbpro_build_pcms,
+	.init = dbpro_init,
+	.free = dbpro_free,
+};
+
 static void ca0132_config(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
@@ -7432,9 +8467,33 @@ static void ca0132_config(struct hda_codec *codec)
 
 	switch (spec->quirk) {
 	case QUIRK_ALIENWARE:
-		codec_dbg(codec, "ca0132_config: QUIRK_ALIENWARE applied.\n");
+		codec_dbg(codec, "%s: QUIRK_ALIENWARE applied.\n", __func__);
 		snd_hda_apply_pincfgs(codec, alienware_pincfgs);
+		break;
+	case QUIRK_SBZ:
+		codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__);
+		snd_hda_apply_pincfgs(codec, sbz_pincfgs);
+		break;
+	case QUIRK_ZXR:
+		codec_dbg(codec, "%s: QUIRK_ZXR applied.\n", __func__);
+		snd_hda_apply_pincfgs(codec, zxr_pincfgs);
+		break;
+	case QUIRK_R3D:
+		codec_dbg(codec, "%s: QUIRK_R3D applied.\n", __func__);
+		snd_hda_apply_pincfgs(codec, r3d_pincfgs);
+		break;
+	case QUIRK_R3DI:
+		codec_dbg(codec, "%s: QUIRK_R3DI applied.\n", __func__);
+		snd_hda_apply_pincfgs(codec, r3di_pincfgs);
+		break;
+	case QUIRK_AE5:
+		codec_dbg(codec, "%s: QUIRK_AE5 applied.\n", __func__);
+		snd_hda_apply_pincfgs(codec, r3di_pincfgs);
+		break;
+	}
 
+	switch (spec->quirk) {
+	case QUIRK_ALIENWARE:
 		spec->num_outputs = 2;
 		spec->out_pins[0] = 0x0b; /* speaker out */
 		spec->out_pins[1] = 0x0f;
@@ -7454,15 +8513,6 @@ static void ca0132_config(struct hda_codec *codec)
 		break;
 	case QUIRK_SBZ:
 	case QUIRK_R3D:
-		if (spec->quirk == QUIRK_SBZ) {
-			codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__);
-			snd_hda_apply_pincfgs(codec, sbz_pincfgs);
-		}
-		if (spec->quirk == QUIRK_R3D) {
-			codec_dbg(codec, "%s: QUIRK_R3D applied.\n", __func__);
-			snd_hda_apply_pincfgs(codec, r3d_pincfgs);
-		}
-
 		spec->num_outputs = 2;
 		spec->out_pins[0] = 0x0B; /* Line out */
 		spec->out_pins[1] = 0x0F; /* Rear headphone out */
@@ -7487,10 +8537,62 @@ static void ca0132_config(struct hda_codec *codec)
 		spec->multiout.dig_out_nid = spec->dig_out;
 		spec->dig_in = 0x09;
 		break;
-	case QUIRK_R3DI:
-		codec_dbg(codec, "%s: QUIRK_R3DI applied.\n", __func__);
-		snd_hda_apply_pincfgs(codec, r3di_pincfgs);
+	case QUIRK_ZXR:
+		spec->num_outputs = 2;
+		spec->out_pins[0] = 0x0B; /* Line out */
+		spec->out_pins[1] = 0x0F; /* Rear headphone out */
+		spec->out_pins[2] = 0x10; /* Center/LFE */
+		spec->out_pins[3] = 0x11; /* Rear surround */
+		spec->shared_out_nid = 0x2;
+		spec->unsol_tag_hp = spec->out_pins[1];
+		spec->unsol_tag_front_hp = spec->out_pins[2];
 
+		spec->adcs[0] = 0x7; /* Rear Mic / Line-in */
+		spec->adcs[1] = 0x8; /* Not connected, no front mic */
+		spec->adcs[2] = 0xa; /* what u hear */
+
+		spec->num_inputs = 2;
+		spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */
+		spec->input_pins[1] = 0x13; /* What U Hear */
+		spec->shared_mic_nid = 0x7;
+		spec->unsol_tag_amic1 = spec->input_pins[0];
+		break;
+	case QUIRK_ZXR_DBPRO:
+		spec->adcs[0] = 0x8; /* ZxR DBPro Aux In */
+
+		spec->num_inputs = 1;
+		spec->input_pins[0] = 0x11; /* RCA Line-in */
+
+		spec->dig_out = 0x05;
+		spec->multiout.dig_out_nid = spec->dig_out;
+
+		spec->dig_in = 0x09;
+		break;
+	case QUIRK_AE5:
+		spec->num_outputs = 2;
+		spec->out_pins[0] = 0x0B; /* Line out */
+		spec->out_pins[1] = 0x11; /* Rear headphone out */
+		spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/
+		spec->out_pins[3] = 0x0F; /* Rear surround */
+		spec->shared_out_nid = 0x2;
+		spec->unsol_tag_hp = spec->out_pins[1];
+		spec->unsol_tag_front_hp = spec->out_pins[2];
+
+		spec->adcs[0] = 0x7; /* Rear Mic / Line-in */
+		spec->adcs[1] = 0x8; /* Front Mic, but only if no DSP */
+		spec->adcs[2] = 0xa; /* what u hear */
+
+		spec->num_inputs = 2;
+		spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */
+		spec->input_pins[1] = 0x13; /* What U Hear */
+		spec->shared_mic_nid = 0x7;
+		spec->unsol_tag_amic1 = spec->input_pins[0];
+
+		/* SPDIF I/O */
+		spec->dig_out = 0x05;
+		spec->multiout.dig_out_nid = spec->dig_out;
+		break;
+	case QUIRK_R3DI:
 		spec->num_outputs = 2;
 		spec->out_pins[0] = 0x0B; /* Line out */
 		spec->out_pins[1] = 0x0F; /* Rear headphone out */
@@ -7547,7 +8649,11 @@ static int ca0132_prepare_verbs(struct hda_codec *codec)
 	struct ca0132_spec *spec = codec->spec;
 
 	spec->chip_init_verbs = ca0132_init_verbs0;
-	if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3D)
+	/*
+	 * Since desktop cards use pci_mmio, this can be used to determine
+	 * whether or not to use these verbs instead of a separate bool.
+	 */
+	if (spec->use_pci_mmio)
 		spec->desktop_init_verbs = ca0132_init_verbs1;
 	spec->spec_init_verbs = kcalloc(NUM_SPEC_VERBS,
 					sizeof(struct hda_verb),
@@ -7579,6 +8685,29 @@ static int ca0132_prepare_verbs(struct hda_codec *codec)
 	return 0;
 }
 
+/*
+ * The Sound Blaster ZxR shares the same PCI subsystem ID as some regular
+ * Sound Blaster Z cards. However, they have different HDA codec subsystem
+ * ID's. So, we check for the ZxR's subsystem ID, as well as the DBPro
+ * daughter boards ID.
+ */
+static void sbz_detect_quirk(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	switch (codec->core.subsystem_id) {
+	case 0x11020033:
+		spec->quirk = QUIRK_ZXR;
+		break;
+	case 0x1102003f:
+		spec->quirk = QUIRK_ZXR_DBPRO;
+		break;
+	default:
+		spec->quirk = QUIRK_SBZ;
+		break;
+	}
+}
+
 static int patch_ca0132(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec;
@@ -7593,10 +8722,6 @@ static int patch_ca0132(struct hda_codec *codec)
 	codec->spec = spec;
 	spec->codec = codec;
 
-	codec->patch_ops = ca0132_patch_ops;
-	codec->pcm_format_first = 1;
-	codec->no_sticky_stream = 1;
-
 	/* Detect codec quirk */
 	quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks);
 	if (quirk)
@@ -7604,6 +8729,18 @@ static int patch_ca0132(struct hda_codec *codec)
 	else
 		spec->quirk = QUIRK_NONE;
 
+	if (spec->quirk == QUIRK_SBZ)
+		sbz_detect_quirk(codec);
+
+	if (spec->quirk == QUIRK_ZXR_DBPRO)
+		codec->patch_ops = dbpro_patch_ops;
+	else
+		codec->patch_ops = ca0132_patch_ops;
+
+	codec->pcm_format_first = 1;
+	codec->no_sticky_stream = 1;
+
+
 	spec->dsp_state = DSP_DOWNLOAD_INIT;
 	spec->num_mixers = 1;
 
@@ -7613,6 +8750,12 @@ static int patch_ca0132(struct hda_codec *codec)
 		spec->mixers[0] = desktop_mixer;
 		snd_hda_codec_set_name(codec, "Sound Blaster Z");
 		break;
+	case QUIRK_ZXR:
+		spec->mixers[0] = desktop_mixer;
+		snd_hda_codec_set_name(codec, "Sound Blaster ZxR");
+		break;
+	case QUIRK_ZXR_DBPRO:
+		break;
 	case QUIRK_R3D:
 		spec->mixers[0] = desktop_mixer;
 		snd_hda_codec_set_name(codec, "Recon3D");
@@ -7621,6 +8764,10 @@ static int patch_ca0132(struct hda_codec *codec)
 		spec->mixers[0] = r3di_mixer;
 		snd_hda_codec_set_name(codec, "Recon3Di");
 		break;
+	case QUIRK_AE5:
+		spec->mixers[0] = desktop_mixer;
+		snd_hda_codec_set_name(codec, "Sound BlasterX AE-5");
+		break;
 	default:
 		spec->mixers[0] = ca0132_mixer;
 		break;
@@ -7630,6 +8777,8 @@ static int patch_ca0132(struct hda_codec *codec)
 	switch (spec->quirk) {
 	case QUIRK_SBZ:
 	case QUIRK_R3D:
+	case QUIRK_AE5:
+	case QUIRK_ZXR:
 		spec->use_alt_controls = true;
 		spec->use_alt_functions = true;
 		spec->use_pci_mmio = true;
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index a7f91be45194..64fa5a82bb9f 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -23,7 +23,7 @@
 #include <linux/module.h>
 #include <sound/core.h>
 #include <sound/tlv.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 1b2195dd2b26..52642ba3e2c0 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -25,7 +25,7 @@
 #include <linux/slab.h>
 #include <linux/module.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index cfd4e4f97f8f..950e02e71766 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -27,7 +27,7 @@
 #include <sound/core.h>
 #include <sound/jack.h>
 
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_beep.h"
@@ -943,6 +943,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD),
+	SND_PCI_QUIRK(0x17aa, 0x3905, "Lenovo G50-30", CXT_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC),
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index cb587dce67a9..67099cbb6be2 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -41,7 +41,7 @@
 #include <sound/hdaudio.h>
 #include <sound/hda_i915.h>
 #include <sound/hda_chmap.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_jack.h"
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3ac7ba9b342d..fa61674a5605 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -32,7 +32,7 @@
 #include <linux/input.h>
 #include <sound/core.h>
 #include <sound/jack.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
@@ -6843,6 +6843,12 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x21, 0x0221101f}),
 	SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC294_FIXUP_LENOVO_MIC_LOCATION,
 		{0x14, 0x90170110},
+		{0x19, 0x02a11030},
+		{0x1a, 0x02a11040},
+		{0x1b, 0x01011020},
+		{0x21, 0x0221101f}),
+	SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC294_FIXUP_LENOVO_MIC_LOCATION,
+		{0x14, 0x90170110},
 		{0x19, 0x02a11020},
 		{0x1a, 0x02a11030},
 		{0x21, 0x0221101f}),
@@ -7738,6 +7744,8 @@ enum {
 	ALC662_FIXUP_ASUS_Nx50,
 	ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE,
 	ALC668_FIXUP_ASUS_Nx51,
+	ALC668_FIXUP_MIC_COEF,
+	ALC668_FIXUP_ASUS_G751,
 	ALC891_FIXUP_HEADSET_MODE,
 	ALC891_FIXUP_DELL_MIC_NO_PRESENCE,
 	ALC662_FIXUP_ACER_VERITON,
@@ -8007,6 +8015,23 @@ static const struct hda_fixup alc662_fixups[] = {
 		.chained = true,
 		.chain_id = ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE,
 	},
+	[ALC668_FIXUP_MIC_COEF] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0xc3 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x4000 },
+			{}
+		},
+	},
+	[ALC668_FIXUP_ASUS_G751] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x16, 0x0421101f }, /* HP */
+			{}
+		},
+		.chained = true,
+		.chain_id = ALC668_FIXUP_MIC_COEF
+	},
 	[ALC891_FIXUP_HEADSET_MODE] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_headset_mode,
@@ -8080,6 +8105,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50),
 	SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A),
 	SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50),
+	SND_PCI_QUIRK(0x1043, 0x12ff, "ASUS G751", ALC668_FIXUP_ASUS_G751),
 	SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP),
 	SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16),
 	SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51),
@@ -8184,6 +8210,7 @@ static const struct hda_model_fixup alc662_fixup_models[] = {
 	{.id = ALC668_FIXUP_DELL_XPS13, .name = "dell-xps13"},
 	{.id = ALC662_FIXUP_ASUS_Nx50, .name = "asus-nx50"},
 	{.id = ALC668_FIXUP_ASUS_Nx51, .name = "asus-nx51"},
+	{.id = ALC668_FIXUP_ASUS_G751, .name = "asus-g751"},
 	{.id = ALC891_FIXUP_HEADSET_MODE, .name = "alc891-headset"},
 	{.id = ALC891_FIXUP_DELL_MIC_NO_PRESENCE, .name = "alc891-headset-multi"},
 	{.id = ALC662_FIXUP_ACER_VERITON, .name = "acer-veriton"},
diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c
index f63acb1b965c..c49d25bcd7f2 100644
--- a/sound/pci/hda/patch_si3054.c
+++ b/sound/pci/hda/patch_si3054.c
@@ -27,7 +27,7 @@
 #include <linux/slab.h>
 #include <linux/module.h>
 #include <sound/core.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 
 /* si3054 verbs */
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 046705b4691a..1b6ecfb01759 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -32,7 +32,7 @@
 #include <linux/module.h>
 #include <sound/core.h>
 #include <sound/jack.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_beep.h"
@@ -77,6 +77,7 @@ enum {
 	STAC_DELL_M6_BOTH,
 	STAC_DELL_EQ,
 	STAC_ALIENWARE_M17X,
+	STAC_ELO_VUPOINT_15MX,
 	STAC_92HD89XX_HP_FRONT_JACK,
 	STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK,
 	STAC_92HD73XX_ASUS_MOBO,
@@ -1879,6 +1880,18 @@ static void stac92hd73xx_fixup_no_jd(struct hda_codec *codec,
 		codec->no_jack_detect = 1;
 }
 
+
+static void stac92hd73xx_disable_automute(struct hda_codec *codec,
+				     const struct hda_fixup *fix, int action)
+{
+	struct sigmatel_spec *spec = codec->spec;
+
+	if (action != HDA_FIXUP_ACT_PRE_PROBE)
+		return;
+
+	spec->gen.suppress_auto_mute = 1;
+}
+
 static const struct hda_fixup stac92hd73xx_fixups[] = {
 	[STAC_92HD73XX_REF] = {
 		.type = HDA_FIXUP_FUNC,
@@ -1904,6 +1917,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = stac92hd73xx_fixup_alienware_m17x,
 	},
+	[STAC_ELO_VUPOINT_15MX] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = stac92hd73xx_disable_automute,
+	},
 	[STAC_92HD73XX_INTEL] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = intel_dg45id_pin_configs,
@@ -1942,6 +1959,7 @@ static const struct hda_model_fixup stac92hd73xx_models[] = {
 	{ .id = STAC_DELL_M6_BOTH, .name = "dell-m6" },
 	{ .id = STAC_DELL_EQ, .name = "dell-eq" },
 	{ .id = STAC_ALIENWARE_M17X, .name = "alienware" },
+	{ .id = STAC_ELO_VUPOINT_15MX, .name = "elo-vupoint-15mx" },
 	{ .id = STAC_92HD73XX_ASUS_MOBO, .name = "asus-mobo" },
 	{}
 };
@@ -1991,6 +2009,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = {
 		      "Alienware M17x", STAC_ALIENWARE_M17X),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490,
 		      "Alienware M17x R3", STAC_DELL_EQ),
+	SND_PCI_QUIRK(0x1059, 0x1011,
+		      "ELO VuPoint 15MX", STAC_ELO_VUPOINT_15MX),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1927,
 				"HP Z1 G2", STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17,
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 6b9617aee0e6..9f6f13e25145 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -52,7 +52,7 @@
 #include <linux/module.h>
 #include <sound/core.h>
 #include <sound/asoundef.h>
-#include "hda_codec.h"
+#include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
 #include "hda_jack.h"
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 5ee468d1aefe..ffddcdfe0c66 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -38,11 +38,6 @@
 #include <sound/ac97_codec.h>
 #include <sound/info.h>
 #include <sound/initval.h>
-/* for 440MX workaround */
-#include <asm/pgtable.h>
-#ifdef CONFIG_X86
-#include <asm/set_memory.h>
-#endif
 
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455");
@@ -374,7 +369,6 @@ struct ichdev {
 	unsigned int ali_slot;			/* ALI DMA slot */
 	struct ac97_pcm *pcm;
 	int pcm_open_flag;
-	unsigned int page_attr_changed: 1;
 	unsigned int suspended: 1;
 };
 
@@ -724,25 +718,6 @@ static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ich
 	iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI);
 }
 
-#ifdef __i386__
-/*
- * Intel 82443MX running a 100MHz processor system bus has a hardware bug,
- * which aborts PCI busmaster for audio transfer.  A workaround is to set
- * the pages as non-cached.  For details, see the errata in
- *	http://download.intel.com/design/chipsets/specupdt/24505108.pdf
- */
-static void fill_nocache(void *buf, int size, int nocache)
-{
-	size = (size + PAGE_SIZE - 1) >> PAGE_SHIFT;
-	if (nocache)
-		set_pages_uc(virt_to_page(buf), size);
-	else
-		set_pages_wb(virt_to_page(buf), size);
-}
-#else
-#define fill_nocache(buf, size, nocache) do { ; } while (0)
-#endif
-
 /*
  *  Interrupt handler
  */
@@ -850,7 +825,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_RESUME:
 		ichdev->suspended = 0;
-		/* fallthru */
+		/* fall through */
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		val = ICH_IOCE | ICH_STARTBM;
@@ -858,7 +833,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 		ichdev->suspended = 1;
-		/* fallthru */
+		/* fall through */
 	case SNDRV_PCM_TRIGGER_STOP:
 		val = 0;
 		break;
@@ -892,7 +867,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_RESUME:
 		ichdev->suspended = 0;
-		/* fallthru */
+		/* fall through */
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -909,7 +884,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 		ichdev->suspended = 1;
-		/* fallthru */
+		/* fall through */
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		/* pause */
@@ -938,23 +913,12 @@ static int snd_intel8x0_hw_params(struct snd_pcm_substream *substream,
 {
 	struct intel8x0 *chip = snd_pcm_substream_chip(substream);
 	struct ichdev *ichdev = get_ichdev(substream);
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	int dbl = params_rate(hw_params) > 48000;
 	int err;
 
-	if (chip->fix_nocache && ichdev->page_attr_changed) {
-		fill_nocache(runtime->dma_area, runtime->dma_bytes, 0); /* clear */
-		ichdev->page_attr_changed = 0;
-	}
 	err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
 	if (err < 0)
 		return err;
-	if (chip->fix_nocache) {
-		if (runtime->dma_area && ! ichdev->page_attr_changed) {
-			fill_nocache(runtime->dma_area, runtime->dma_bytes, 1);
-			ichdev->page_attr_changed = 1;
-		}
-	}
 	if (ichdev->pcm_open_flag) {
 		snd_ac97_pcm_close(ichdev->pcm);
 		ichdev->pcm_open_flag = 0;
@@ -974,17 +938,12 @@ static int snd_intel8x0_hw_params(struct snd_pcm_substream *substream,
 
 static int snd_intel8x0_hw_free(struct snd_pcm_substream *substream)
 {
-	struct intel8x0 *chip = snd_pcm_substream_chip(substream);
 	struct ichdev *ichdev = get_ichdev(substream);
 
 	if (ichdev->pcm_open_flag) {
 		snd_ac97_pcm_close(ichdev->pcm);
 		ichdev->pcm_open_flag = 0;
 	}
-	if (chip->fix_nocache && ichdev->page_attr_changed) {
-		fill_nocache(substream->runtime->dma_area, substream->runtime->dma_bytes, 0);
-		ichdev->page_attr_changed = 0;
-	}
 	return snd_pcm_lib_free_pages(substream);
 }
 
@@ -1510,6 +1469,9 @@ struct ich_pcm_table {
 	int ac97_idx;
 };
 
+#define intel8x0_dma_type(chip) \
+	((chip)->fix_nocache ? SNDRV_DMA_TYPE_DEV_UC : SNDRV_DMA_TYPE_DEV)
+
 static int snd_intel8x0_pcm1(struct intel8x0 *chip, int device,
 			     struct ich_pcm_table *rec)
 {
@@ -1540,7 +1502,7 @@ static int snd_intel8x0_pcm1(struct intel8x0 *chip, int device,
 		strcpy(pcm->name, chip->card->shortname);
 	chip->pcm[device] = pcm;
 
-	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+	snd_pcm_lib_preallocate_pages_for_all(pcm, intel8x0_dma_type(chip),
 					      snd_dma_pci_data(chip->pci),
 					      rec->prealloc_size, rec->prealloc_max_size);
 
@@ -2629,11 +2591,8 @@ static int snd_intel8x0_free(struct intel8x0 *chip)
       __hw_end:
 	if (chip->irq >= 0)
 		free_irq(chip->irq, chip);
-	if (chip->bdbars.area) {
-		if (chip->fix_nocache)
-			fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 0);
+	if (chip->bdbars.area)
 		snd_dma_free_pages(&chip->bdbars);
-	}
 	if (chip->addr)
 		pci_iounmap(chip->pci, chip->addr);
 	if (chip->bmaddr)
@@ -2657,17 +2616,6 @@ static int intel8x0_suspend(struct device *dev)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
 	for (i = 0; i < chip->pcm_devs; i++)
 		snd_pcm_suspend_all(chip->pcm[i]);
-	/* clear nocache */
-	if (chip->fix_nocache) {
-		for (i = 0; i < chip->bdbars_count; i++) {
-			struct ichdev *ichdev = &chip->ichd[i];
-			if (ichdev->substream && ichdev->page_attr_changed) {
-				struct snd_pcm_runtime *runtime = ichdev->substream->runtime;
-				if (runtime->dma_area)
-					fill_nocache(runtime->dma_area, runtime->dma_bytes, 0);
-			}
-		}
-	}
 	for (i = 0; i < chip->ncodecs; i++)
 		snd_ac97_suspend(chip->ac97[i]);
 	if (chip->device_type == DEVICE_INTEL_ICH4)
@@ -2708,25 +2656,9 @@ static int intel8x0_resume(struct device *dev)
 			  ICH_PCM_SPDIF_1011);
 	}
 
-	/* refill nocache */
-	if (chip->fix_nocache)
-		fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 1);
-
 	for (i = 0; i < chip->ncodecs; i++)
 		snd_ac97_resume(chip->ac97[i]);
 
-	/* refill nocache */
-	if (chip->fix_nocache) {
-		for (i = 0; i < chip->bdbars_count; i++) {
-			struct ichdev *ichdev = &chip->ichd[i];
-			if (ichdev->substream && ichdev->page_attr_changed) {
-				struct snd_pcm_runtime *runtime = ichdev->substream->runtime;
-				if (runtime->dma_area)
-					fill_nocache(runtime->dma_area, runtime->dma_bytes, 1);
-			}
-		}
-	}
-
 	/* resume status */
 	for (i = 0; i < chip->bdbars_count; i++) {
 		struct ichdev *ichdev = &chip->ichd[i];
@@ -3057,6 +2989,12 @@ static int snd_intel8x0_create(struct snd_card *card,
 
 	chip->inside_vm = snd_intel8x0_inside_vm(pci);
 
+	/*
+	 * Intel 82443MX running a 100MHz processor system bus has a hardware
+	 * bug, which aborts PCI busmaster for audio transfer.  A workaround
+	 * is to set the pages as non-cached.  For details, see the errata in
+	 *     http://download.intel.com/design/chipsets/specupdt/24505108.pdf
+	 */
 	if (pci->vendor == PCI_VENDOR_ID_INTEL &&
 	    pci->device == PCI_DEVICE_ID_INTEL_440MX)
 		chip->fix_nocache = 1; /* enable workaround */
@@ -3128,7 +3066,7 @@ static int snd_intel8x0_create(struct snd_card *card,
 
 	/* allocate buffer descriptor lists */
 	/* the start of each lists must be aligned to 8 bytes */
-	if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+	if (snd_dma_alloc_pages(intel8x0_dma_type(chip), snd_dma_pci_data(pci),
 				chip->bdbars_count * sizeof(u32) * ICH_MAX_FRAGS * 2,
 				&chip->bdbars) < 0) {
 		snd_intel8x0_free(chip);
@@ -3137,9 +3075,6 @@ static int snd_intel8x0_create(struct snd_card *card,
 	}
 	/* tables must be aligned to 8 bytes here, but the kernel pages
 	   are much bigger, so we don't care (on i386) */
-	/* workaround for 440MX */
-	if (chip->fix_nocache)
-		fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 1);
 	int_sta_masks = 0;
 	for (i = 0; i < chip->bdbars_count; i++) {
 		ichdev = &chip->ichd[i];
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 943a726b1c1b..c84629190cba 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -1171,16 +1171,6 @@ static int snd_intel8x0m_create(struct snd_card *card,
 	}
 
  port_inited:
-	if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED,
-			KBUILD_MODNAME, chip)) {
-		dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq);
-		snd_intel8x0m_free(chip);
-		return -EBUSY;
-	}
-	chip->irq = pci->irq;
-	pci_set_master(pci);
-	synchronize_irq(chip->irq);
-
 	/* initialize offsets */
 	chip->bdbars_count = 2;
 	tbl = intel_regs;
@@ -1224,11 +1214,21 @@ static int snd_intel8x0m_create(struct snd_card *card,
 	chip->int_sta_reg = ICH_REG_GLOB_STA;
 	chip->int_sta_mask = int_sta_masks;
 
+	pci_set_master(pci);
+
 	if ((err = snd_intel8x0m_chip_init(chip, 1)) < 0) {
 		snd_intel8x0m_free(chip);
 		return err;
 	}
 
+	if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED,
+			KBUILD_MODNAME, chip)) {
+		dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq);
+		snd_intel8x0m_free(chip);
+		return -EBUSY;
+	}
+	chip->irq = pci->irq;
+
 	if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
 		snd_intel8x0m_free(chip);
 		return err;
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index f0906ba416d4..3ac8c71d567c 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -319,7 +319,8 @@ static const struct snd_pcm_hardware snd_rme32_spdif_info = {
 			 SNDRV_PCM_INFO_MMAP_VALID |
 			 SNDRV_PCM_INFO_INTERLEAVED | 
 			 SNDRV_PCM_INFO_PAUSE |
-			 SNDRV_PCM_INFO_SYNC_START),
+			 SNDRV_PCM_INFO_SYNC_START |
+			 SNDRV_PCM_INFO_SYNC_APPLPTR),
 	.formats =	(SNDRV_PCM_FMTBIT_S16_LE | 
 			 SNDRV_PCM_FMTBIT_S32_LE),
 	.rates =	(SNDRV_PCM_RATE_32000 |
@@ -346,7 +347,8 @@ static const struct snd_pcm_hardware snd_rme32_adat_info =
 			      SNDRV_PCM_INFO_MMAP_VALID |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE |
-			      SNDRV_PCM_INFO_SYNC_START),
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_SYNC_APPLPTR),
 	.formats=            SNDRV_PCM_FMTBIT_S16_LE,
 	.rates =             (SNDRV_PCM_RATE_44100 | 
 			      SNDRV_PCM_RATE_48000),
@@ -370,7 +372,8 @@ static const struct snd_pcm_hardware snd_rme32_spdif_fd_info = {
 			 SNDRV_PCM_INFO_MMAP_VALID |
 			 SNDRV_PCM_INFO_INTERLEAVED | 
 			 SNDRV_PCM_INFO_PAUSE |
-			 SNDRV_PCM_INFO_SYNC_START),
+			 SNDRV_PCM_INFO_SYNC_START |
+			 SNDRV_PCM_INFO_SYNC_APPLPTR),
 	.formats =	(SNDRV_PCM_FMTBIT_S16_LE | 
 			 SNDRV_PCM_FMTBIT_S32_LE),
 	.rates =	(SNDRV_PCM_RATE_32000 |
@@ -397,7 +400,8 @@ static const struct snd_pcm_hardware snd_rme32_adat_fd_info =
 			      SNDRV_PCM_INFO_MMAP_VALID |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE |
-			      SNDRV_PCM_INFO_SYNC_START),
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_SYNC_APPLPTR),
 	.formats=            SNDRV_PCM_FMTBIT_S16_LE,
 	.rates =             (SNDRV_PCM_RATE_44100 | 
 			      SNDRV_PCM_RATE_48000),
@@ -1104,16 +1108,6 @@ snd_rme32_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 		snd_pcm_trigger_done(s, substream);
 	}
 	
-	/* prefill playback buffer */
-	if (cmd == SNDRV_PCM_TRIGGER_START && rme32->fullduplex_mode) {
-		snd_pcm_group_for_each_entry(s, substream) {
-			if (s == rme32->playback_substream) {
-				s->ops->ack(s);
-				break;
-			}
-		}
-	}
-
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		if (rme32->running && ! RME32_ISWORKING(rme32))
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 11b5b5e0e058..679ad0415e3b 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6534,7 +6534,7 @@ static int snd_hdspm_create_alsa_devices(struct snd_card *card,
 	dev_dbg(card->dev, "Update mixer controls...\n");
 	hdspm_update_simple_mixer_controls(hdspm);
 
-	dev_dbg(card->dev, "Initializeing complete ???\n");
+	dev_dbg(card->dev, "Initializing complete?\n");
 
 	err = snd_card_register(card);
 	if (err < 0) {
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index 8e3275a96a82..3f813ea5210a 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -42,7 +42,7 @@
 #include "../codecs/da7219.h"
 #include "../codecs/da7219-aad.h"
 
-#define CZ_PLAT_CLK 25000000
+#define CZ_PLAT_CLK 48000000
 #define DUAL_CHANNEL		2
 
 static struct snd_soc_jack cz_jack;
@@ -75,7 +75,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd)
 	da7219_dai_clk = clk_get(component->dev, "da7219-dai-clks");
 
 	ret = snd_soc_card_jack_new(card, "Headset Jack",
-				SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
+				SND_JACK_HEADSET | SND_JACK_LINEOUT |
 				SND_JACK_BTN_0 | SND_JACK_BTN_1 |
 				SND_JACK_BTN_2 | SND_JACK_BTN_3,
 				&cz_jack, NULL, 0);
@@ -133,7 +133,7 @@ static const struct snd_pcm_hw_constraint_list constraints_channels = {
 	.mask = 0,
 };
 
-static int cz_da7219_startup(struct snd_pcm_substream *substream)
+static int cz_da7219_play_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -150,7 +150,28 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream)
 	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
 				   &constraints_rates);
 
-	machine->i2s_instance = I2S_SP_INSTANCE;
+	machine->play_i2s_instance = I2S_SP_INSTANCE;
+	return da7219_clk_enable(substream);
+}
+
+static int cz_da7219_cap_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+	/*
+	 * On this platform for PCM device we support stereo
+	 */
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_channels);
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				   &constraints_rates);
+
+	machine->cap_i2s_instance = I2S_SP_INSTANCE;
 	machine->capture_channel = CAP_CHANNEL1;
 	return da7219_clk_enable(substream);
 }
@@ -162,11 +183,22 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream)
 
 static int cz_max_startup(struct snd_pcm_substream *substream)
 {
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
-	machine->i2s_instance = I2S_BT_INSTANCE;
+	/*
+	 * On this platform for PCM device we support stereo
+	 */
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_channels);
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				   &constraints_rates);
+
+	machine->play_i2s_instance = I2S_BT_INSTANCE;
 	return da7219_clk_enable(substream);
 }
 
@@ -177,21 +209,43 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream)
 
 static int cz_dmic0_startup(struct snd_pcm_substream *substream)
 {
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
-	machine->i2s_instance = I2S_BT_INSTANCE;
+	/*
+	 * On this platform for PCM device we support stereo
+	 */
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_channels);
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				   &constraints_rates);
+
+	machine->cap_i2s_instance = I2S_BT_INSTANCE;
 	return da7219_clk_enable(substream);
 }
 
 static int cz_dmic1_startup(struct snd_pcm_substream *substream)
 {
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
-	machine->i2s_instance = I2S_SP_INSTANCE;
+	/*
+	 * On this platform for PCM device we support stereo
+	 */
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_channels);
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				   &constraints_rates);
+
+	machine->cap_i2s_instance = I2S_SP_INSTANCE;
 	machine->capture_channel = CAP_CHANNEL0;
 	return da7219_clk_enable(substream);
 }
@@ -201,8 +255,13 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream)
 	da7219_clk_disable();
 }
 
+static const struct snd_soc_ops cz_da7219_play_ops = {
+	.startup = cz_da7219_play_startup,
+	.shutdown = cz_da7219_shutdown,
+};
+
 static const struct snd_soc_ops cz_da7219_cap_ops = {
-	.startup = cz_da7219_startup,
+	.startup = cz_da7219_cap_startup,
 	.shutdown = cz_da7219_shutdown,
 };
 
@@ -233,7 +292,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.init = cz_da7219_init,
 		.dpcm_playback = 1,
-		.ops = &cz_da7219_cap_ops,
+		.ops = &cz_da7219_play_ops,
 	},
 	{
 		.name = "amd-da7219-cap",
diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c
index 77b265bd0505..cdebab2f8ce5 100644
--- a/sound/soc/amd/acp-pcm-dma.c
+++ b/sound/soc/amd/acp-pcm-dma.c
@@ -867,8 +867,12 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 
 	if (pinfo) {
-		rtd->i2s_instance = pinfo->i2s_instance;
-		rtd->capture_channel = pinfo->capture_channel;
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			rtd->i2s_instance = pinfo->play_i2s_instance;
+		} else {
+			rtd->i2s_instance = pinfo->cap_i2s_instance;
+			rtd->capture_channel = pinfo->capture_channel;
+		}
 	}
 	if (adata->asic_type == CHIP_STONEY) {
 		val = acp_reg_read(adata->acp_mmio,
@@ -1036,16 +1040,22 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream)
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		period_bytes = frames_to_bytes(runtime, runtime->period_size);
-		dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr);
-		if (dscr == rtd->dma_dscr_idx_1)
-			pos = period_bytes;
-		else
-			pos = 0;
 		bytescount = acp_get_byte_count(rtd);
-		if (bytescount > rtd->bytescount)
+		if (bytescount >= rtd->bytescount)
 			bytescount -= rtd->bytescount;
-		delay = do_div(bytescount, period_bytes);
-		runtime->delay = bytes_to_frames(runtime, delay);
+		if (bytescount < period_bytes) {
+			pos = 0;
+		} else {
+			dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr);
+			if (dscr == rtd->dma_dscr_idx_1)
+				pos = period_bytes;
+			else
+				pos = 0;
+		}
+		if (bytescount > 0) {
+			delay = do_div(bytescount, period_bytes);
+			runtime->delay = bytes_to_frames(runtime, delay);
+		}
 	} else {
 		buffersize = frames_to_bytes(runtime, runtime->buffer_size);
 		bytescount = acp_get_byte_count(rtd);
diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h
index be3963e8f4fa..dbbb1a85638d 100644
--- a/sound/soc/amd/acp.h
+++ b/sound/soc/amd/acp.h
@@ -158,7 +158,8 @@ struct audio_drv_data {
  * and dma driver
  */
 struct acp_platform_info {
-	u16 i2s_instance;
+	u16 play_i2s_instance;
+	u16 cap_i2s_instance;
 	u16 capture_channel;
 };
 
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index 64b784e96f84..64f86f0b87e5 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -97,4 +97,16 @@ config SND_ATMEL_SOC_I2S
 	help
 	  Say Y or M if you want to add support for Atmel ASoc driver for boards
 	  using I2S.
+
+config SND_SOC_MIKROE_PROTO
+	tristate "Support for Mikroe-PROTO board"
+	depends on OF
+	depends on SND_SOC_I2C_AND_SPI
+	select SND_SOC_WM8731
+	help
+	  Say Y or M if you want to add support for MikroElektronika PROTO Audio
+	  Board. This board contains the WM8731 codec, which can be configured
+	  using I2C over SDA (MPU Data Input) and SCL (MPU Clock Input) pins.
+	  Both playback and capture are supported.
+
 endif
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index cd87cb4bcff5..9f41bfa0fea3 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -17,6 +17,7 @@ snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
 snd-atmel-soc-classd-objs := atmel-classd.o
 snd-atmel-soc-pdmic-objs := atmel-pdmic.o
 snd-atmel-soc-tse850-pcm5142-objs := tse850-pcm5142.o
+snd-soc-mikroe-proto-objs := mikroe-proto.o
 
 obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
 obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o
@@ -24,3 +25,4 @@ obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o
 obj-$(CONFIG_SND_ATMEL_SOC_CLASSD) += snd-atmel-soc-classd.o
 obj-$(CONFIG_SND_ATMEL_SOC_PDMIC) += snd-atmel-soc-pdmic.o
 obj-$(CONFIG_SND_ATMEL_SOC_TSE850_PCM5142) += snd-atmel-soc-tse850-pcm5142.o
+obj-$(CONFIG_SND_SOC_MIKROE_PROTO) += snd-soc-mikroe-proto.o
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index d3b69682d9c2..6291ec7f9dd6 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -1005,11 +1005,11 @@ static int asoc_ssc_init(struct device *dev)
 	struct ssc_device *ssc = dev_get_drvdata(dev);
 	int ret;
 
-	ret = snd_soc_register_component(dev, &atmel_ssc_component,
+	ret = devm_snd_soc_register_component(dev, &atmel_ssc_component,
 					 &atmel_ssc_dai, 1);
 	if (ret) {
 		dev_err(dev, "Could not register DAI: %d\n", ret);
-		goto err;
+		return ret;
 	}
 
 	if (ssc->pdata->use_dma)
@@ -1019,15 +1019,10 @@ static int asoc_ssc_init(struct device *dev)
 
 	if (ret) {
 		dev_err(dev, "Could not register PCM: %d\n", ret);
-		goto err_unregister_dai;
+		return ret;
 	}
 
 	return 0;
-
-err_unregister_dai:
-	snd_soc_unregister_component(dev);
-err:
-	return ret;
 }
 
 static void asoc_ssc_exit(struct device *dev)
@@ -1038,8 +1033,6 @@ static void asoc_ssc_exit(struct device *dev)
 		atmel_pcm_dma_platform_unregister(dev);
 	else
 		atmel_pcm_pdc_platform_unregister(dev);
-
-	snd_soc_unregister_component(dev);
 }
 
 /**
diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c
new file mode 100644
index 000000000000..d47aaa5bf75a
--- /dev/null
+++ b/sound/soc/atmel/mikroe-proto.c
@@ -0,0 +1,165 @@
+/*
+ * ASoC driver for PROTO AudioCODEC (with a WM8731)
+ *
+ * Author:      Florian Meier, <koalo@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.
+ */
+
+#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 "../codecs/wm8731.h"
+
+#define XTAL_RATE 12288000	/* This is fixed on this board */
+
+static int snd_proto_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+	/* Set proto sysclk */
+	int ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
+					 XTAL_RATE, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(card->dev, "Failed to set WM8731 SYSCLK: %d\n",
+			ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget snd_proto_widget[] = {
+	SND_SOC_DAPM_MIC("Microphone Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route snd_proto_route[] = {
+	/* speaker connected to LHPOUT/RHPOUT */
+	{"Headphone Jack", NULL, "LHPOUT"},
+	{"Headphone Jack", NULL, "RHPOUT"},
+
+	/* mic is connected to Mic Jack, with WM8731 Mic Bias */
+	{"MICIN", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Microphone Jack"},
+};
+
+/* audio machine driver */
+static struct snd_soc_card snd_proto = {
+	.name		= "snd_mikroe_proto",
+	.owner		= THIS_MODULE,
+	.dapm_widgets	= snd_proto_widget,
+	.num_dapm_widgets = ARRAY_SIZE(snd_proto_widget),
+	.dapm_routes	= snd_proto_route,
+	.num_dapm_routes = ARRAY_SIZE(snd_proto_route),
+};
+
+static int snd_proto_probe(struct platform_device *pdev)
+{
+	struct snd_soc_dai_link *dai;
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *codec_np, *cpu_np;
+	struct device_node *bitclkmaster = NULL;
+	struct device_node *framemaster = NULL;
+	unsigned int dai_fmt;
+	int ret = 0;
+
+	if (!np) {
+		dev_err(&pdev->dev, "No device node supplied\n");
+		return -EINVAL;
+	}
+
+	snd_proto.dev = &pdev->dev;
+	ret = snd_soc_of_parse_card_name(&snd_proto, "model");
+	if (ret)
+		return ret;
+
+	dai = devm_kzalloc(&pdev->dev, sizeof(*dai), GFP_KERNEL);
+	if (!dai)
+		return -ENOMEM;
+
+	snd_proto.dai_link = dai;
+	snd_proto.num_links = 1;
+
+	dai->name = "WM8731";
+	dai->stream_name = "WM8731 HiFi";
+	dai->codec_dai_name = "wm8731-hifi";
+	dai->init = &snd_proto_init;
+
+	codec_np = of_parse_phandle(np, "audio-codec", 0);
+	if (!codec_np) {
+		dev_err(&pdev->dev, "audio-codec node missing\n");
+		return -EINVAL;
+	}
+	dai->codec_of_node = codec_np;
+
+	cpu_np = of_parse_phandle(np, "i2s-controller", 0);
+	if (!cpu_np) {
+		dev_err(&pdev->dev, "i2s-controller missing\n");
+		return -EINVAL;
+	}
+	dai->cpu_of_node = cpu_np;
+	dai->platform_of_node = cpu_np;
+
+	dai_fmt = snd_soc_of_parse_daifmt(np, NULL,
+					  &bitclkmaster, &framemaster);
+	if (bitclkmaster != framemaster) {
+		dev_err(&pdev->dev, "Must be the same bitclock and frame master\n");
+		return -EINVAL;
+	}
+	if (bitclkmaster) {
+		dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+		if (codec_np == bitclkmaster)
+			dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+		else
+			dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+	}
+	of_node_put(bitclkmaster);
+	of_node_put(framemaster);
+	dai->dai_fmt = dai_fmt;
+
+	of_node_put(codec_np);
+	of_node_put(cpu_np);
+
+	ret = snd_soc_register_card(&snd_proto);
+	if (ret && ret != -EPROBE_DEFER)
+		dev_err(&pdev->dev,
+			"snd_soc_register_card() failed: %d\n", ret);
+
+	return ret;
+}
+
+static int snd_proto_remove(struct platform_device *pdev)
+{
+	return snd_soc_unregister_card(&snd_proto);
+}
+
+static const struct of_device_id snd_proto_of_match[] = {
+	{ .compatible = "mikroe,mikroe-proto", },
+	{},
+};
+MODULE_DEVICE_TABLE(of, snd_proto_of_match);
+
+static struct platform_driver snd_proto_driver = {
+	.driver = {
+		.name   = "snd-mikroe-proto",
+		.of_match_table = snd_proto_of_match,
+	},
+	.probe	  = snd_proto_probe,
+	.remove	 = snd_proto_remove,
+};
+
+module_platform_driver(snd_proto_driver);
+
+MODULE_AUTHOR("Florian Meier");
+MODULE_DESCRIPTION("ASoC Driver for PROTO board (WM8731)");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c
index 3a1393283156..214adcad5419 100644
--- a/sound/soc/atmel/tse850-pcm5142.c
+++ b/sound/soc/atmel/tse850-pcm5142.c
@@ -1,44 +1,38 @@
-/*
- * TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec
- *
- * Copyright (C) 2016 Axentia Technologies AB
- *
- * Author: Peter Rosin <peda@axentia.se>
- *
- * 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.
- */
-
-/*
- *               loop1 relays
- *   IN1 +---o  +------------+  o---+ OUT1
- *            \                /
- *             +              +
- *             |   /          |
- *             +--o  +--.     |
- *             |  add   |     |
- *             |        V     |
- *             |      .---.   |
- *   DAC +----------->|Sum|---+
- *             |      '---'   |
- *             |              |
- *             +              +
- *
- *   IN2 +---o--+------------+--o---+ OUT2
- *               loop2 relays
- *
- * The 'loop1' gpio pin controlls two relays, which are either in loop
- * position, meaning that input and output are directly connected, or
- * they are in mixer position, meaning that the signal is passed through
- * the 'Sum' mixer. Similarly for 'loop2'.
- *
- * In the above, the 'loop1' relays are inactive, thus feeding IN1 to the
- * mixer (if 'add' is active) and feeding the mixer output to OUT1. The
- * 'loop2' relays are active, short-cutting the TSE-850 from channel 2.
- * IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name
- * of the (filtered) output from the PCM5142 codec.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec
+//
+// Copyright (C) 2016 Axentia Technologies AB
+//
+// Author: Peter Rosin <peda@axentia.se>
+//
+//               loop1 relays
+//   IN1 +---o  +------------+  o---+ OUT1
+//            \                /
+//             +              +
+//             |   /          |
+//             +--o  +--.     |
+//             |  add   |     |
+//             |        V     |
+//             |      .---.   |
+//   DAC +----------->|Sum|---+
+//             |      '---'   |
+//             |              |
+//             +              +
+//
+//   IN2 +---o--+------------+--o---+ OUT2
+//               loop2 relays
+//
+// The 'loop1' gpio pin controlls two relays, which are either in loop
+// position, meaning that input and output are directly connected, or
+// they are in mixer position, meaning that the signal is passed through
+// the 'Sum' mixer. Similarly for 'loop2'.
+//
+// In the above, the 'loop1' relays are inactive, thus feeding IN1 to the
+// mixer (if 'add' is active) and feeding the mixer output to OUT1. The
+// 'loop2' relays are active, short-cutting the TSE-850 from channel 2.
+// IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name
+// of the (filtered) output from the PCM5142 codec.
 
 #include <linux/clk.h>
 #include <linux/gpio.h>
@@ -452,4 +446,4 @@ module_platform_driver(tse850_driver);
 /* Module information */
 MODULE_AUTHOR("Peter Rosin <peda@axentia.se>");
 MODULE_DESCRIPTION("ALSA SoC driver for TSE-850 with PCM5142 codec");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c
index b733f1446353..b7c358b48d8d 100644
--- a/sound/soc/bcm/cygnus-ssp.c
+++ b/sound/soc/bcm/cygnus-ssp.c
@@ -1334,7 +1334,7 @@ static int cygnus_ssp_probe(struct platform_device *pdev)
 	cygaud->active_ports = 0;
 
 	dev_dbg(dev, "Registering %d DAIs\n", active_port_count);
-	err = snd_soc_register_component(dev, &cygnus_ssp_component,
+	err = devm_snd_soc_register_component(dev, &cygnus_ssp_component,
 				cygnus_ssp_dai, active_port_count);
 	if (err) {
 		dev_err(dev, "snd_soc_register_dai failed\n");
@@ -1345,32 +1345,27 @@ static int cygnus_ssp_probe(struct platform_device *pdev)
 	if (cygaud->irq_num <= 0) {
 		dev_err(dev, "platform_get_irq failed\n");
 		err = cygaud->irq_num;
-		goto err_irq;
+		return err;
 	}
 
 	err = audio_clk_init(pdev, cygaud);
 	if (err) {
 		dev_err(dev, "audio clock initialization failed\n");
-		goto err_irq;
+		return err;
 	}
 
 	err = cygnus_soc_platform_register(dev, cygaud);
 	if (err) {
 		dev_err(dev, "platform reg error %d\n", err);
-		goto err_irq;
+		return err;
 	}
 
 	return 0;
-
-err_irq:
-	snd_soc_unregister_component(dev);
-	return err;
 }
 
 static int cygnus_ssp_remove(struct platform_device *pdev)
 {
 	cygnus_soc_platform_unregister(&pdev->dev);
-	snd_soc_unregister_component(&pdev->dev);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index efb095dbcd71..9cc4f1848c9b 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -82,6 +82,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_ES7241
 	select SND_SOC_GTM601
 	select SND_SOC_HDAC_HDMI
+	select SND_SOC_HDAC_HDA
 	select SND_SOC_ICS43432
 	select SND_SOC_INNO_RK3036
 	select SND_SOC_ISABELLE if I2C
@@ -109,6 +110,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MT6351 if MTK_PMIC_WRAP
 	select SND_SOC_NAU8540 if I2C
 	select SND_SOC_NAU8810 if I2C
+	select SND_SOC_NAU8822 if I2C
 	select SND_SOC_NAU8824 if I2C
 	select SND_SOC_NAU8825 if I2C
 	select SND_SOC_HDMI_CODEC
@@ -119,6 +121,8 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_PCM186X_I2C if I2C
 	select SND_SOC_PCM186X_SPI if SPI_MASTER
 	select SND_SOC_PCM3008
+	select SND_SOC_PCM3060_I2C if I2C
+	select SND_SOC_PCM3060_SPI if SPI_MASTER
 	select SND_SOC_PCM3168A_I2C if I2C
 	select SND_SOC_PCM3168A_SPI if SPI_MASTER
 	select SND_SOC_PCM5102A
@@ -575,7 +579,11 @@ config SND_SOC_DA9055
 	tristate
 
 config SND_SOC_DMIC
-	tristate
+	tristate "Generic Digital Microphone CODEC"
+	depends on GPIOLIB
+	help
+	  Enable support for the Generic Digital Microphone CODEC.
+	  Select this if your sound card has DMICs.
 
 config SND_SOC_HDMI_CODEC
 	tristate
@@ -615,6 +623,10 @@ config SND_SOC_HDAC_HDMI
 	select SND_PCM_ELD
 	select HDMI
 
+config SND_SOC_HDAC_HDA
+	tristate
+	select SND_HDA
+
 config SND_SOC_ICS43432
 	tristate
 
@@ -629,7 +641,8 @@ config SND_SOC_LM49453
 	tristate
 
 config SND_SOC_MAX98088
-       tristate
+	tristate "Maxim MAX98088/9 Low-Power, Stereo Audio Codec"
+	depends on I2C
 
 config SND_SOC_MAX98090
        tristate
@@ -732,6 +745,21 @@ config SND_SOC_PCM186X_SPI
 config SND_SOC_PCM3008
        tristate
 
+config SND_SOC_PCM3060
+       tristate
+
+config SND_SOC_PCM3060_I2C
+	tristate "Texas Instruments PCM3060 CODEC - I2C"
+	depends on I2C
+	select SND_SOC_PCM3060
+	select REGMAP_I2C
+
+config SND_SOC_PCM3060_SPI
+	tristate "Texas Instruments PCM3060 CODEC - SPI"
+	depends on SPI_MASTER
+	select SND_SOC_PCM3060
+	select REGMAP_SPI
+
 config SND_SOC_PCM3168A
 	tristate
 
@@ -1299,6 +1327,10 @@ config SND_SOC_NAU8810
 	tristate "Nuvoton Technology Corporation NAU88C10 CODEC"
 	depends on I2C
 
+config SND_SOC_NAU8822
+	tristate "Nuvoton Technology Corporation NAU88C22 CODEC"
+	depends on I2C
+
 config SND_SOC_NAU8824
 	tristate "Nuvoton Technology Corporation NAU88L24 CODEC"
 	depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 7ae7c85e8219..8ffab8c8dbfa 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -78,6 +78,7 @@ snd-soc-es8328-i2c-objs := es8328-i2c.o
 snd-soc-es8328-spi-objs := es8328-spi.o
 snd-soc-gtm601-objs := gtm601.o
 snd-soc-hdac-hdmi-objs := hdac_hdmi.o
+snd-soc-hdac-hda-objs := hdac_hda.o
 snd-soc-ics43432-objs := ics43432.o
 snd-soc-inno-rk3036-objs := inno_rk3036.o
 snd-soc-isabelle-objs := isabelle.o
@@ -106,6 +107,7 @@ snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o
 snd-soc-mt6351-objs := mt6351.o
 snd-soc-nau8540-objs := nau8540.o
 snd-soc-nau8810-objs := nau8810.o
+snd-soc-nau8822-objs := nau8822.o
 snd-soc-nau8824-objs := nau8824.o
 snd-soc-nau8825-objs := nau8825.o
 snd-soc-hdmi-codec-objs := hdmi-codec.o
@@ -119,6 +121,9 @@ snd-soc-pcm186x-objs := pcm186x.o
 snd-soc-pcm186x-i2c-objs := pcm186x-i2c.o
 snd-soc-pcm186x-spi-objs := pcm186x-spi.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-pcm3060-objs := pcm3060.o
+snd-soc-pcm3060-i2c-objs := pcm3060-i2c.o
+snd-soc-pcm3060-spi-objs := pcm3060-spi.o
 snd-soc-pcm3168a-objs := pcm3168a.o
 snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o
 snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o
@@ -338,6 +343,7 @@ obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
 obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o
 obj-$(CONFIG_SND_SOC_GTM601)    += snd-soc-gtm601.o
 obj-$(CONFIG_SND_SOC_HDAC_HDMI) += snd-soc-hdac-hdmi.o
+obj-$(CONFIG_SND_SOC_HDAC_HDA) += snd-soc-hdac-hda.o
 obj-$(CONFIG_SND_SOC_ICS43432)	+= snd-soc-ics43432.o
 obj-$(CONFIG_SND_SOC_INNO_RK3036)	+= snd-soc-inno-rk3036.o
 obj-$(CONFIG_SND_SOC_ISABELLE)	+= snd-soc-isabelle.o
@@ -366,6 +372,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o
 obj-$(CONFIG_SND_SOC_MT6351)	+= snd-soc-mt6351.o
 obj-$(CONFIG_SND_SOC_NAU8540)   += snd-soc-nau8540.o
 obj-$(CONFIG_SND_SOC_NAU8810)   += snd-soc-nau8810.o
+obj-$(CONFIG_SND_SOC_NAU8822)   += snd-soc-nau8822.o
 obj-$(CONFIG_SND_SOC_NAU8824)   += snd-soc-nau8824.o
 obj-$(CONFIG_SND_SOC_NAU8825)   += snd-soc-nau8825.o
 obj-$(CONFIG_SND_SOC_HDMI_CODEC)	+= snd-soc-hdmi-codec.o
@@ -379,6 +386,9 @@ obj-$(CONFIG_SND_SOC_PCM186X)	+= snd-soc-pcm186x.o
 obj-$(CONFIG_SND_SOC_PCM186X_I2C)	+= snd-soc-pcm186x-i2c.o
 obj-$(CONFIG_SND_SOC_PCM186X_SPI)	+= snd-soc-pcm186x-spi.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_PCM3060)	+= snd-soc-pcm3060.o
+obj-$(CONFIG_SND_SOC_PCM3060_I2C)	+= snd-soc-pcm3060-i2c.o
+obj-$(CONFIG_SND_SOC_PCM3060_SPI)	+= snd-soc-pcm3060-spi.o
 obj-$(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
diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c
index be136e981653..bef3e9e74c26 100644
--- a/sound/soc/codecs/adau1761.c
+++ b/sound/soc/codecs/adau1761.c
@@ -518,7 +518,8 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_component *component)
 			ARRAY_SIZE(adau1761_jack_detect_controls));
 		if (ret)
 			return ret;
-	case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */
+		/* fall through */
+	case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE:
 		ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes,
 			ARRAY_SIZE(adau1761_no_dmic_routes));
 		if (ret)
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index 57169b8ff14e..3959e6ad113d 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -21,11 +21,18 @@
 #include <linux/i2c.h>
 #include <linux/spi/spi.h>
 #include <linux/regmap.h>
+#include <asm/unaligned.h>
 
 #include "sigmadsp.h"
 #include "adau17x1.h"
 #include "adau-utils.h"
 
+#define ADAU17X1_SAFELOAD_TARGET_ADDRESS 0x0006
+#define ADAU17X1_SAFELOAD_TRIGGER 0x0007
+#define ADAU17X1_SAFELOAD_DATA 0x0001
+#define ADAU17X1_SAFELOAD_DATA_SIZE 20
+#define ADAU17X1_WORD_SIZE 4
+
 static const char * const adau17x1_capture_mixer_boost_text[] = {
 	"Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3",
 };
@@ -60,6 +67,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = {
 	SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum),
 };
 
+static int adau17x1_setup_firmware(struct snd_soc_component *component,
+	unsigned int rate);
+
 static int adau17x1_pll_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -313,7 +323,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = {
 	{ "Capture", NULL, "Right Decimator" },
 };
 
-bool adau17x1_has_dsp(struct adau *adau)
+static bool adau17x1_has_dsp(struct adau *adau)
 {
 	switch (adau->type) {
 	case ADAU1761:
@@ -324,7 +334,17 @@ bool adau17x1_has_dsp(struct adau *adau)
 		return false;
 	}
 }
-EXPORT_SYMBOL_GPL(adau17x1_has_dsp);
+
+static bool adau17x1_has_safeload(struct adau *adau)
+{
+	switch (adau->type) {
+	case ADAU1761:
+	case ADAU1781:
+		return true;
+	default:
+		return false;
+	}
+}
 
 static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id,
 	int source, unsigned int freq_in, unsigned int freq_out)
@@ -836,7 +856,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg)
 }
 EXPORT_SYMBOL_GPL(adau17x1_volatile_register);
 
-int adau17x1_setup_firmware(struct snd_soc_component *component,
+static int adau17x1_setup_firmware(struct snd_soc_component *component,
 	unsigned int rate)
 {
 	int ret;
@@ -880,7 +900,6 @@ err:
 
 	return ret;
 }
-EXPORT_SYMBOL_GPL(adau17x1_setup_firmware);
 
 int adau17x1_add_widgets(struct snd_soc_component *component)
 {
@@ -957,6 +976,56 @@ int adau17x1_resume(struct snd_soc_component *component)
 }
 EXPORT_SYMBOL_GPL(adau17x1_resume);
 
+static int adau17x1_safeload(struct sigmadsp *sigmadsp, unsigned int addr,
+	const uint8_t bytes[], size_t len)
+{
+	uint8_t buf[ADAU17X1_WORD_SIZE];
+	uint8_t data[ADAU17X1_SAFELOAD_DATA_SIZE];
+	unsigned int addr_offset;
+	unsigned int nbr_words;
+	int ret;
+
+	/* write data to safeload addresses. Check if len is not a multiple of
+	 * 4 bytes, if so we need to zero pad.
+	 */
+	nbr_words = len / ADAU17X1_WORD_SIZE;
+	if ((len - nbr_words * ADAU17X1_WORD_SIZE) == 0) {
+		ret = regmap_raw_write(sigmadsp->control_data,
+			ADAU17X1_SAFELOAD_DATA, bytes, len);
+	} else {
+		nbr_words++;
+		memset(data, 0, ADAU17X1_SAFELOAD_DATA_SIZE);
+		memcpy(data, bytes, len);
+		ret = regmap_raw_write(sigmadsp->control_data,
+			ADAU17X1_SAFELOAD_DATA, data,
+			nbr_words * ADAU17X1_WORD_SIZE);
+	}
+
+	if (ret < 0)
+		return ret;
+
+	/* Write target address, target address is offset by 1 */
+	addr_offset = addr - 1;
+	put_unaligned_be32(addr_offset, buf);
+	ret = regmap_raw_write(sigmadsp->control_data,
+		ADAU17X1_SAFELOAD_TARGET_ADDRESS, buf, ADAU17X1_WORD_SIZE);
+	if (ret < 0)
+		return ret;
+
+	/* write nbr of words to trigger address */
+	put_unaligned_be32(nbr_words, buf);
+	ret = regmap_raw_write(sigmadsp->control_data,
+		ADAU17X1_SAFELOAD_TRIGGER, buf, ADAU17X1_WORD_SIZE);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static const struct sigmadsp_ops adau17x1_sigmadsp_ops = {
+	.safeload = adau17x1_safeload,
+};
+
 int adau17x1_probe(struct device *dev, struct regmap *regmap,
 	enum adau17x1_type type, void (*switch_mode)(struct device *dev),
 	const char *firmware_name)
@@ -1002,8 +1071,13 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap,
 	dev_set_drvdata(dev, adau);
 
 	if (firmware_name) {
-		adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, NULL,
-			firmware_name);
+		if (adau17x1_has_safeload(adau)) {
+			adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap,
+				&adau17x1_sigmadsp_ops, firmware_name);
+		} else {
+			adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap,
+				NULL, firmware_name);
+		}
 		if (IS_ERR(adau->sigmadsp)) {
 			dev_warn(dev, "Could not find firmware file: %ld\n",
 				PTR_ERR(adau->sigmadsp));
diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h
index e6fe87beec07..98a3b6f5bc96 100644
--- a/sound/soc/codecs/adau17x1.h
+++ b/sound/soc/codecs/adau17x1.h
@@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component);
 
 extern const struct snd_soc_dai_ops adau17x1_dai_ops;
 
-int adau17x1_setup_firmware(struct snd_soc_component *component,
-	unsigned int rate);
-bool adau17x1_has_dsp(struct adau *adau);
-
 #define ADAU17X1_CLOCK_CONTROL			0x4000
 #define ADAU17X1_PLL_CONTROL			0x4002
 #define ADAU17X1_REC_POWER_MGMT			0x4009
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index 407554175282..ab27d2b94d02 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -154,11 +154,11 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = {
 	SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1,
 				6, 1, 0),
 	SOC_ENUM("C Data Access", cam_mode_enum),
+	SOC_SINGLE("SPDIF Switch", CS4265_SPDIF_CTL2, 5, 1, 1),
 	SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2,
 				3, 1, 0),
 	SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum),
-	SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2,
-				0, 1, 0),
+	SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, 0, 1, 0),
 	SOC_ENUM("Mono Channel Select", spdif_mono_select_enum),
 	SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24),
 };
@@ -221,10 +221,11 @@ static const struct snd_soc_dapm_route cs4265_audio_map[] = {
 	{"LINEOUTR", NULL, "DAC"},
 	{"SPDIFOUT", NULL, "SPDIF"},
 
+	{"Pre-amp MIC", NULL, "MICL"},
+	{"Pre-amp MIC", NULL, "MICR"},
+	{"ADC Mux", "MIC", "Pre-amp MIC"},
 	{"ADC Mux", "LINEIN", "LINEINL"},
 	{"ADC Mux", "LINEIN", "LINEINR"},
-	{"ADC Mux", "MIC", "MICL"},
-	{"ADC Mux", "MIC", "MICR"},
 	{"ADC", NULL, "ADC Mux"},
 	{"DOUT", NULL, "ADC"},
 	{"DAI1 Capture", NULL, "DOUT"},
@@ -496,7 +497,8 @@ static int cs4265_set_bias_level(struct snd_soc_component *component,
 			SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
 
 #define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \
-			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE)
+			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE | \
+			SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE)
 
 static const struct snd_soc_dai_ops cs4265_ops = {
 	.hw_params	= cs4265_pcm_hw_params,
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 5080d7a3c279..fd2bd74024c1 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -21,6 +21,7 @@
  *  - master mode *NOT* supported
  */
 
+#include <linux/clk.h>
 #include <linux/module.h>
 #include <linux/slab.h>
 #include <sound/core.h>
@@ -41,6 +42,7 @@ enum master_slave_mode {
 
 struct cs42l51_private {
 	unsigned int mclk;
+	struct clk *mclk_handle;
 	unsigned int audio_mode;	/* The mode (I2S or left-justified) */
 	enum master_slave_mode func;
 };
@@ -237,6 +239,10 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = {
 		&cs42l51_adcr_mux_controls),
 };
 
+static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = {
+	SND_SOC_DAPM_CLOCK_SUPPLY("MCLK")
+};
+
 static const struct snd_soc_dapm_route cs42l51_routes[] = {
 	{"HPL", NULL, "Left DAC"},
 	{"HPR", NULL, "Right DAC"},
@@ -487,6 +493,14 @@ static struct snd_soc_dai_driver cs42l51_dai = {
 static int cs42l51_component_probe(struct snd_soc_component *component)
 {
 	int ret, reg;
+	struct snd_soc_dapm_context *dapm;
+	struct cs42l51_private *cs42l51;
+
+	cs42l51 = snd_soc_component_get_drvdata(component);
+	dapm = snd_soc_component_get_dapm(component);
+
+	if (cs42l51->mclk_handle)
+		snd_soc_dapm_new_controls(dapm, cs42l51_dapm_mclk_widgets, 1);
 
 	/*
 	 * DAC configuration
@@ -540,6 +554,13 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap)
 
 	dev_set_drvdata(dev, cs42l51);
 
+	cs42l51->mclk_handle = devm_clk_get(dev, "MCLK");
+	if (IS_ERR(cs42l51->mclk_handle)) {
+		if (PTR_ERR(cs42l51->mclk_handle) != -ENOENT)
+			return PTR_ERR(cs42l51->mclk_handle);
+		cs42l51->mclk_handle = NULL;
+	}
+
 	/* Verify that we have a CS42L51 */
 	ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
index 8c4926df9286..71322e0410ee 100644
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@@ -148,6 +148,7 @@ static const struct of_device_id dmic_dev_match[] = {
 	{.compatible = "dmic-codec"},
 	{}
 };
+MODULE_DEVICE_TABLE(of, dmic_dev_match);
 
 static struct platform_driver dmic_driver = {
 	.driver = {
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 4b5827dc23aa..04a3aa770722 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -566,14 +566,14 @@ static int es8328_set_sysclk(struct snd_soc_dai *codec_dai,
 		break;
 	case 22579200:
 		mclkdiv2 = 1;
-		/* fallthru */
+		/* fall through */
 	case 11289600:
 		es8328->sysclk_constraints = &constraints_11289;
 		es8328->mclk_ratios = ratios_11289;
 		break;
 	case 24576000:
 		mclkdiv2 = 1;
-		/* fallthru */
+		/* fall through */
 	case 12288000:
 		es8328->sysclk_constraints = &constraints_12288;
 		es8328->mclk_ratios = ratios_12288;
diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c
new file mode 100644
index 000000000000..2aaa83028e55
--- /dev/null
+++ b/sound/soc/codecs/hdac_hda.c
@@ -0,0 +1,483 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2015-18 Intel Corporation.
+
+/*
+ * hdac_hda.c - ASoC extensions to reuse the legacy HDA codec drivers
+ * with ASoC platform drivers. These APIs are called by the legacy HDA
+ * codec drivers using hdac_ext_bus_ops ops.
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/pm_runtime.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/hdaudio_ext.h>
+#include <sound/hda_codec.h>
+#include <sound/hda_register.h>
+#include "hdac_hda.h"
+
+#define HDAC_ANALOG_DAI_ID		0
+#define HDAC_DIGITAL_DAI_ID		1
+#define HDAC_ALT_ANALOG_DAI_ID		2
+
+#define STUB_FORMATS	(SNDRV_PCM_FMTBIT_S8 | \
+			SNDRV_PCM_FMTBIT_U8 | \
+			SNDRV_PCM_FMTBIT_S16_LE | \
+			SNDRV_PCM_FMTBIT_U16_LE | \
+			SNDRV_PCM_FMTBIT_S24_LE | \
+			SNDRV_PCM_FMTBIT_U24_LE | \
+			SNDRV_PCM_FMTBIT_S32_LE | \
+			SNDRV_PCM_FMTBIT_U32_LE | \
+			SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
+
+static int hdac_hda_dai_open(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai);
+static void hdac_hda_dai_close(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai);
+static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai);
+static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai);
+static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai,
+				     unsigned int tx_mask, unsigned int rx_mask,
+				     int slots, int slot_width);
+static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt,
+						 struct snd_soc_dai *dai);
+
+static struct snd_soc_dai_ops hdac_hda_dai_ops = {
+	.startup = hdac_hda_dai_open,
+	.shutdown = hdac_hda_dai_close,
+	.prepare = hdac_hda_dai_prepare,
+	.hw_free = hdac_hda_dai_hw_free,
+	.set_tdm_slot = hdac_hda_dai_set_tdm_slot,
+};
+
+static struct snd_soc_dai_driver hdac_hda_dais[] = {
+{
+	.id = HDAC_ANALOG_DAI_ID,
+	.name = "Analog Codec DAI",
+	.ops = &hdac_hda_dai_ops,
+	.playback = {
+		.stream_name	= "Analog Codec Playback",
+		.channels_min	= 1,
+		.channels_max	= 16,
+		.rates		= SNDRV_PCM_RATE_8000_192000,
+		.formats	= STUB_FORMATS,
+		.sig_bits	= 24,
+	},
+	.capture = {
+		.stream_name    = "Analog Codec Capture",
+		.channels_min   = 1,
+		.channels_max   = 16,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = STUB_FORMATS,
+		.sig_bits = 24,
+	},
+},
+{
+	.id = HDAC_DIGITAL_DAI_ID,
+	.name = "Digital Codec DAI",
+	.ops = &hdac_hda_dai_ops,
+	.playback = {
+		.stream_name    = "Digital Codec Playback",
+		.channels_min   = 1,
+		.channels_max   = 16,
+		.rates          = SNDRV_PCM_RATE_8000_192000,
+		.formats        = STUB_FORMATS,
+		.sig_bits = 24,
+	},
+	.capture = {
+		.stream_name    = "Digital Codec Capture",
+		.channels_min   = 1,
+		.channels_max   = 16,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = STUB_FORMATS,
+		.sig_bits = 24,
+	},
+},
+{
+	.id = HDAC_ALT_ANALOG_DAI_ID,
+	.name = "Alt Analog Codec DAI",
+	.ops = &hdac_hda_dai_ops,
+	.playback = {
+		.stream_name	= "Alt Analog Codec Playback",
+		.channels_min	= 1,
+		.channels_max	= 16,
+		.rates		= SNDRV_PCM_RATE_8000_192000,
+		.formats	= STUB_FORMATS,
+		.sig_bits	= 24,
+	},
+	.capture = {
+		.stream_name    = "Alt Analog Codec Capture",
+		.channels_min   = 1,
+		.channels_max   = 16,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = STUB_FORMATS,
+		.sig_bits = 24,
+	},
+}
+
+};
+
+static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai,
+				     unsigned int tx_mask, unsigned int rx_mask,
+				     int slots, int slot_width)
+{
+	struct snd_soc_component *component = dai->component;
+	struct hdac_hda_priv *hda_pvt;
+	struct hdac_hda_pcm *pcm;
+
+	hda_pvt = snd_soc_component_get_drvdata(component);
+	pcm = &hda_pvt->pcm[dai->id];
+	if (tx_mask)
+		pcm[dai->id].stream_tag[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask;
+	else
+		pcm[dai->id].stream_tag[SNDRV_PCM_STREAM_CAPTURE] = rx_mask;
+
+	return 0;
+}
+
+static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct hdac_hda_priv *hda_pvt;
+	struct hda_pcm_stream *hda_stream;
+	struct hda_pcm *pcm;
+
+	hda_pvt = snd_soc_component_get_drvdata(component);
+	pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+	if (!pcm)
+		return -EINVAL;
+
+	hda_stream = &pcm->stream[substream->stream];
+	snd_hda_codec_cleanup(&hda_pvt->codec, hda_stream, substream);
+
+	return 0;
+}
+
+static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct hdac_hda_priv *hda_pvt;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct hdac_device *hdev;
+	struct hda_pcm_stream *hda_stream;
+	unsigned int format_val;
+	struct hda_pcm *pcm;
+	unsigned int stream;
+	int ret = 0;
+
+	hda_pvt = snd_soc_component_get_drvdata(component);
+	hdev = &hda_pvt->codec.core;
+	pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+	if (!pcm)
+		return -EINVAL;
+
+	hda_stream = &pcm->stream[substream->stream];
+
+	format_val = snd_hdac_calc_stream_format(runtime->rate,
+						 runtime->channels,
+						 runtime->format,
+						 hda_stream->maxbps,
+						 0);
+	if (!format_val) {
+		dev_err(&hdev->dev,
+			"invalid format_val, rate=%d, ch=%d, format=%d\n",
+			runtime->rate, runtime->channels, runtime->format);
+		return -EINVAL;
+	}
+
+	stream = hda_pvt->pcm[dai->id].stream_tag[substream->stream];
+
+	ret = snd_hda_codec_prepare(&hda_pvt->codec, hda_stream,
+				    stream, format_val, substream);
+	if (ret < 0)
+		dev_err(&hdev->dev, "codec prepare failed %d\n", ret);
+
+	return ret;
+}
+
+static int hdac_hda_dai_open(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct hdac_hda_priv *hda_pvt;
+	struct hda_pcm_stream *hda_stream;
+	struct hda_pcm *pcm;
+	int ret;
+
+	hda_pvt = snd_soc_component_get_drvdata(component);
+	pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+	if (!pcm)
+		return -EINVAL;
+
+	snd_hda_codec_pcm_get(pcm);
+
+	hda_stream = &pcm->stream[substream->stream];
+
+	ret = hda_stream->ops.open(hda_stream, &hda_pvt->codec, substream);
+	if (ret < 0)
+		snd_hda_codec_pcm_put(pcm);
+
+	return ret;
+}
+
+static void hdac_hda_dai_close(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct hdac_hda_priv *hda_pvt;
+	struct hda_pcm_stream *hda_stream;
+	struct hda_pcm *pcm;
+
+	hda_pvt = snd_soc_component_get_drvdata(component);
+	pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+	if (!pcm)
+		return;
+
+	hda_stream = &pcm->stream[substream->stream];
+
+	hda_stream->ops.close(hda_stream, &hda_pvt->codec, substream);
+
+	snd_hda_codec_pcm_put(pcm);
+}
+
+static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt,
+						 struct snd_soc_dai *dai)
+{
+	struct hda_codec *hcodec = &hda_pvt->codec;
+	struct hda_pcm *cpcm;
+	const char *pcm_name;
+
+	switch (dai->id) {
+	case HDAC_ANALOG_DAI_ID:
+		pcm_name = "Analog";
+		break;
+	case HDAC_DIGITAL_DAI_ID:
+		pcm_name = "Digital";
+		break;
+	case HDAC_ALT_ANALOG_DAI_ID:
+		pcm_name = "Alt Analog";
+		break;
+	default:
+		dev_err(&hcodec->core.dev, "invalid dai id %d\n", dai->id);
+		return NULL;
+	}
+
+	list_for_each_entry(cpcm, &hcodec->pcm_list_head, list) {
+		if (strpbrk(cpcm->name, pcm_name))
+			return cpcm;
+	}
+
+	dev_err(&hcodec->core.dev, "didn't find PCM for DAI %s\n", dai->name);
+	return NULL;
+}
+
+static int hdac_hda_codec_probe(struct snd_soc_component *component)
+{
+	struct hdac_hda_priv *hda_pvt =
+			snd_soc_component_get_drvdata(component);
+	struct snd_soc_dapm_context *dapm =
+			snd_soc_component_get_dapm(component);
+	struct hdac_device *hdev = &hda_pvt->codec.core;
+	struct hda_codec *hcodec = &hda_pvt->codec;
+	struct hdac_ext_link *hlink;
+	hda_codec_patch_t patch;
+	int ret;
+
+	hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
+	if (!hlink) {
+		dev_err(&hdev->dev, "hdac link not found\n");
+		return -EIO;
+	}
+
+	snd_hdac_ext_bus_link_get(hdev->bus, hlink);
+
+	ret = snd_hda_codec_device_new(hcodec->bus, component->card->snd_card,
+				       hdev->addr, hcodec);
+	if (ret < 0) {
+		dev_err(&hdev->dev, "failed to create hda codec %d\n", ret);
+		goto error_no_pm;
+	}
+
+	/*
+	 * snd_hda_codec_device_new decrements the usage count so call get pm
+	 * else the device will be powered off
+	 */
+	pm_runtime_get_noresume(&hdev->dev);
+
+	hcodec->bus->card = dapm->card->snd_card;
+
+	ret = snd_hda_codec_set_name(hcodec, hcodec->preset->name);
+	if (ret < 0) {
+		dev_err(&hdev->dev, "name failed %s\n", hcodec->preset->name);
+		goto error;
+	}
+
+	ret = snd_hdac_regmap_init(&hcodec->core);
+	if (ret < 0) {
+		dev_err(&hdev->dev, "regmap init failed\n");
+		goto error;
+	}
+
+	patch = (hda_codec_patch_t)hcodec->preset->driver_data;
+	if (patch) {
+		ret = patch(hcodec);
+		if (ret < 0) {
+			dev_err(&hdev->dev, "patch failed %d\n", ret);
+			goto error;
+		}
+	} else {
+		dev_dbg(&hdev->dev, "no patch file found\n");
+	}
+
+	ret = snd_hda_codec_parse_pcms(hcodec);
+	if (ret < 0) {
+		dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret);
+		goto error;
+	}
+
+	ret = snd_hda_codec_build_controls(hcodec);
+	if (ret < 0) {
+		dev_err(&hdev->dev, "unable to create controls %d\n", ret);
+		goto error;
+	}
+
+	hcodec->core.lazy_cache = true;
+
+	/*
+	 * hdac_device core already sets the state to active and calls
+	 * get_noresume. So enable runtime and set the device to suspend.
+	 * pm_runtime_enable is also called during codec registeration
+	 */
+	pm_runtime_put(&hdev->dev);
+	pm_runtime_suspend(&hdev->dev);
+
+	return 0;
+
+error:
+	pm_runtime_put(&hdev->dev);
+error_no_pm:
+	snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+	return ret;
+}
+
+static void hdac_hda_codec_remove(struct snd_soc_component *component)
+{
+	struct hdac_hda_priv *hda_pvt =
+		      snd_soc_component_get_drvdata(component);
+	struct hdac_device *hdev = &hda_pvt->codec.core;
+	struct hdac_ext_link *hlink = NULL;
+
+	hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
+	if (!hlink) {
+		dev_err(&hdev->dev, "hdac link not found\n");
+		return;
+	}
+
+	snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+	pm_runtime_disable(&hdev->dev);
+}
+
+static const struct snd_soc_dapm_route hdac_hda_dapm_routes[] = {
+	{"AIF1TX", NULL, "Codec Input Pin1"},
+	{"AIF2TX", NULL, "Codec Input Pin2"},
+	{"AIF3TX", NULL, "Codec Input Pin3"},
+
+	{"Codec Output Pin1", NULL, "AIF1RX"},
+	{"Codec Output Pin2", NULL, "AIF2RX"},
+	{"Codec Output Pin3", NULL, "AIF3RX"},
+};
+
+static const struct snd_soc_dapm_widget hdac_hda_dapm_widgets[] = {
+	/* Audio Interface */
+	SND_SOC_DAPM_AIF_IN("AIF1RX", "Analog Codec Playback", 0,
+			    SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIF2RX", "Digital Codec Playback", 0,
+			    SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIF3RX", "Alt Analog Codec Playback", 0,
+			    SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF1TX", "Analog Codec Capture", 0,
+			     SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF2TX", "Digital Codec Capture", 0,
+			     SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF3TX", "Alt Analog Codec Capture", 0,
+			     SND_SOC_NOPM, 0, 0),
+
+	/* Input Pins */
+	SND_SOC_DAPM_INPUT("Codec Input Pin1"),
+	SND_SOC_DAPM_INPUT("Codec Input Pin2"),
+	SND_SOC_DAPM_INPUT("Codec Input Pin3"),
+
+	/* Output Pins */
+	SND_SOC_DAPM_OUTPUT("Codec Output Pin1"),
+	SND_SOC_DAPM_OUTPUT("Codec Output Pin2"),
+	SND_SOC_DAPM_OUTPUT("Codec Output Pin3"),
+};
+
+static const struct snd_soc_component_driver hdac_hda_codec = {
+	.probe		= hdac_hda_codec_probe,
+	.remove		= hdac_hda_codec_remove,
+	.idle_bias_on	= false,
+	.dapm_widgets           = hdac_hda_dapm_widgets,
+	.num_dapm_widgets       = ARRAY_SIZE(hdac_hda_dapm_widgets),
+	.dapm_routes            = hdac_hda_dapm_routes,
+	.num_dapm_routes        = ARRAY_SIZE(hdac_hda_dapm_routes),
+};
+
+static int hdac_hda_dev_probe(struct hdac_device *hdev)
+{
+	struct hdac_ext_link *hlink;
+	struct hdac_hda_priv *hda_pvt;
+	int ret;
+
+	/* hold the ref while we probe */
+	hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
+	if (!hlink) {
+		dev_err(&hdev->dev, "hdac link not found\n");
+		return -EIO;
+	}
+	snd_hdac_ext_bus_link_get(hdev->bus, hlink);
+
+	hda_pvt = hdac_to_hda_priv(hdev);
+	if (!hda_pvt)
+		return -ENOMEM;
+
+	/* ASoC specific initialization */
+	ret = devm_snd_soc_register_component(&hdev->dev,
+					 &hdac_hda_codec, hdac_hda_dais,
+					 ARRAY_SIZE(hdac_hda_dais));
+	if (ret < 0) {
+		dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret);
+		return ret;
+	}
+
+	dev_set_drvdata(&hdev->dev, hda_pvt);
+	snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+
+	return ret;
+}
+
+static int hdac_hda_dev_remove(struct hdac_device *hdev)
+{
+	return 0;
+}
+
+static struct hdac_ext_bus_ops hdac_ops = {
+	.hdev_attach = hdac_hda_dev_probe,
+	.hdev_detach = hdac_hda_dev_remove,
+};
+
+struct hdac_ext_bus_ops *snd_soc_hdac_hda_get_ops(void)
+{
+	return &hdac_ops;
+}
+EXPORT_SYMBOL_GPL(snd_soc_hdac_hda_get_ops);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("ASoC Extensions for legacy HDA Drivers");
+MODULE_AUTHOR("Rakesh Ughreja<rakesh.a.ughreja@intel.com>");
diff --git a/sound/soc/codecs/hdac_hda.h b/sound/soc/codecs/hdac_hda.h
new file mode 100644
index 000000000000..e444ef593360
--- /dev/null
+++ b/sound/soc/codecs/hdac_hda.h
@@ -0,0 +1,24 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright(c) 2015-18 Intel Corporation.
+ */
+
+#ifndef __HDAC_HDA_H__
+#define __HDAC_HDA_H__
+
+struct hdac_hda_pcm {
+	int stream_tag[2];
+};
+
+struct hdac_hda_priv {
+	struct hda_codec codec;
+	struct hdac_hda_pcm pcm[2];
+};
+
+#define hdac_to_hda_priv(_hdac) \
+			container_of(_hdac, struct hdac_hda_priv, codec.core)
+#define hdac_to_hda_codec(_hdac) container_of(_hdac, struct hda_codec, core)
+
+struct hdac_ext_bus_ops *snd_soc_hdac_hda_get_ops(void);
+
+#endif /* __HDAC_HDA_H__ */
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 7b8533abf637..4e9854889a95 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -1410,6 +1410,12 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdev,
 		if (ret)
 			return ret;
 
+		/* Filter out 44.1, 88.2 and 176.4Khz */
+		rates &= ~(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 |
+			   SNDRV_PCM_RATE_176400);
+		if (!rates)
+			return -EINVAL;
+
 		sprintf(dai_name, "intel-hdmi-hifi%d", i+1);
 		hdmi_dais[i].name = devm_kstrdup(&hdev->dev,
 					dai_name, GFP_KERNEL);
@@ -1598,7 +1604,7 @@ static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card,
 {
 	struct snd_soc_pcm_runtime *rtd;
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		if (rtd->pcm && (rtd->pcm->device == device))
 			return rtd->pcm;
 	}
@@ -1961,9 +1967,6 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdev, int pcm_idx)
 
 	port = list_first_entry(&pcm->port_list, struct hdac_hdmi_port, head);
 
-	if (!port)
-		return 0;
-
 	if (!port || !port->eld.eld_valid)
 		return 0;
 
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index fb515aaa54fc..ca172a4b6849 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -16,6 +16,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/regmap.h>
+#include <linux/clk.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -42,6 +43,7 @@ struct max98088_priv {
 	struct regmap *regmap;
 	enum max98088_type devtype;
 	struct max98088_pdata *pdata;
+	struct clk *mclk;
 	unsigned int sysclk;
 	struct max98088_cdata dai[2];
 	int eq_textcnt;
@@ -1103,6 +1105,11 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai,
        if (freq == max98088->sysclk)
                return 0;
 
+	if (!IS_ERR(max98088->mclk)) {
+		freq = clk_round_rate(max98088->mclk, freq);
+		clk_set_rate(max98088->mclk, freq);
+	}
+
        /* Setup clocks for slave mode, and using the PLL
         * PSCLK = 0x01 (when master clk is 10MHz to 20MHz)
         *         0x02 (when master clk is 20MHz to 30MHz)..
@@ -1310,6 +1317,20 @@ static int max98088_set_bias_level(struct snd_soc_component *component,
 		break;
 
 	case SND_SOC_BIAS_PREPARE:
+		/*
+		 * SND_SOC_BIAS_PREPARE is called while preparing for a
+		 * transition to ON or away from ON. If current bias_level
+		 * is SND_SOC_BIAS_ON, then it is preparing for a transition
+		 * away from ON. Disable the clock in that case, otherwise
+		 * enable it.
+		 */
+		if (!IS_ERR(max98088->mclk)) {
+			if (snd_soc_component_get_bias_level(component) ==
+			    SND_SOC_BIAS_ON)
+				clk_disable_unprepare(max98088->mclk);
+			else
+				clk_prepare_enable(max98088->mclk);
+		}
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
@@ -1725,6 +1746,11 @@ static int max98088_i2c_probe(struct i2c_client *i2c,
        if (IS_ERR(max98088->regmap))
 	       return PTR_ERR(max98088->regmap);
 
+	max98088->mclk = devm_clk_get(&i2c->dev, "mclk");
+	if (IS_ERR(max98088->mclk))
+		if (PTR_ERR(max98088->mclk) == -EPROBE_DEFER)
+			return PTR_ERR(max98088->mclk);
+
        max98088->devtype = id->driver_data;
 
        i2c_set_clientdata(i2c, max98088);
@@ -1742,9 +1768,19 @@ static const struct i2c_device_id max98088_i2c_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, max98088_i2c_id);
 
+#if defined(CONFIG_OF)
+static const struct of_device_id max98088_of_match[] = {
+	{ .compatible = "maxim,max98088" },
+	{ .compatible = "maxim,max98089" },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, max98088_of_match);
+#endif
+
 static struct i2c_driver max98088_i2c_driver = {
 	.driver = {
 		.name = "max98088",
+		.of_match_table = of_match_ptr(max98088_of_match),
 	},
 	.probe  = max98088_i2c_probe,
 	.id_table = max98088_i2c_id,
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index 1093f766d0d2..a09d01318f79 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -2,6 +2,7 @@
 // Copyright (c) 2017, Maxim Integrated
 
 #include <linux/acpi.h>
+#include <linux/delay.h>
 #include <linux/i2c.h>
 #include <linux/module.h>
 #include <linux/regmap.h>
@@ -454,7 +455,7 @@ SND_SOC_DAPM_SIGGEN("IMON"),
 SND_SOC_DAPM_SIGGEN("FBMON"),
 };
 
-static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, 0, -50, 0);
+static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, -6350, 50, 1);
 static const DECLARE_TLV_DB_RANGE(max98373_spk_tlv,
 	0, 8, TLV_DB_SCALE_ITEM(0, 50, 0),
 	9, 10, TLV_DB_SCALE_ITEM(500, 100, 0),
@@ -470,19 +471,19 @@ static const DECLARE_TLV_DB_RANGE(max98373_dht_spkgain_min_tlv,
 	0, 9, TLV_DB_SCALE_ITEM(800, 100, 0),
 );
 static const DECLARE_TLV_DB_RANGE(max98373_dht_rotation_point_tlv,
-	0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0),
-	2, 7, TLV_DB_SCALE_ITEM(-200, -100, 0),
-	8, 9, TLV_DB_SCALE_ITEM(-1000, -200, 0),
-	10, 11, TLV_DB_SCALE_ITEM(-1500, -300, 0),
-	12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0),
-	14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0),
+	0, 1, TLV_DB_SCALE_ITEM(-3000, 500, 0),
+	2, 4, TLV_DB_SCALE_ITEM(-2200, 200, 0),
+	5, 6, TLV_DB_SCALE_ITEM(-1500, 300, 0),
+	7, 9, TLV_DB_SCALE_ITEM(-1000, 200, 0),
+	10, 13, TLV_DB_SCALE_ITEM(-500, 100, 0),
+	14, 15, TLV_DB_SCALE_ITEM(-100, 50, 0),
 );
 static const DECLARE_TLV_DB_RANGE(max98373_limiter_thresh_tlv,
-	0, 15, TLV_DB_SCALE_ITEM(0, -100, 0),
+	0, 15, TLV_DB_SCALE_ITEM(-1500, 100, 0),
 );
 
 static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv,
-	0, 60, TLV_DB_SCALE_ITEM(0, -25, 0),
+	0, 60, TLV_DB_SCALE_ITEM(-1500, 25, 0),
 );
 
 static bool max98373_readable_register(struct device *dev, unsigned int reg)
@@ -604,7 +605,7 @@ SOC_SINGLE("Dither Switch", MAX98373_R203F_AMP_DSP_CFG,
 SOC_SINGLE("DC Blocker Switch", MAX98373_R203F_AMP_DSP_CFG,
 	MAX98373_AMP_DSP_CFG_DCBLK_SHIFT, 1, 0),
 SOC_SINGLE_TLV("Digital Volume", MAX98373_R203D_AMP_DIG_VOL_CTRL,
-	0, 0x7F, 0, max98373_digital_tlv),
+	0, 0x7F, 1, max98373_digital_tlv),
 SOC_SINGLE_TLV("Speaker Volume", MAX98373_R203E_AMP_PATH_GAIN,
 	MAX98373_SPK_DIGI_GAIN_SHIFT, 10, 0, max98373_spk_tlv),
 SOC_SINGLE_TLV("FS Max Volume", MAX98373_R203E_AMP_PATH_GAIN,
@@ -616,7 +617,7 @@ SOC_SINGLE("DHT Switch", MAX98373_R20D4_DHT_EN,
 SOC_SINGLE_TLV("DHT Min Volume", MAX98373_R20D1_DHT_CFG,
 	MAX98373_DHT_SPK_GAIN_MIN_SHIFT, 9, 0, max98373_dht_spkgain_min_tlv),
 SOC_SINGLE_TLV("DHT Rot Pnt Volume", MAX98373_R20D1_DHT_CFG,
-	MAX98373_DHT_ROT_PNT_SHIFT, 15, 0, max98373_dht_rotation_point_tlv),
+	MAX98373_DHT_ROT_PNT_SHIFT, 15, 1, max98373_dht_rotation_point_tlv),
 SOC_SINGLE_TLV("DHT Attack Step Volume", MAX98373_R20D2_DHT_ATTACK_CFG,
 	MAX98373_DHT_ATTACK_STEP_SHIFT, 4, 0, max98373_dht_step_size_tlv),
 SOC_SINGLE_TLV("DHT Release Step Volume", MAX98373_R20D3_DHT_RELEASE_CFG,
@@ -653,29 +654,29 @@ SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0),
 SOC_SINGLE("BDE Attack Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 4, 0xF, 0),
 SOC_SINGLE("BDE Release Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 0, 0xF, 0),
 SOC_SINGLE_TLV("BDE LVL1 Clip Thresh Volume", MAX98373_R20A9_BDE_L1_CFG_2,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL2 Clip Thresh Volume", MAX98373_R20AC_BDE_L2_CFG_2,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL3 Clip Thresh Volume", MAX98373_R20AF_BDE_L3_CFG_2,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL4 Clip Thresh Volume", MAX98373_R20B2_BDE_L4_CFG_2,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL1 Clip Reduction Volume", MAX98373_R20AA_BDE_L1_CFG_3,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL2 Clip Reduction Volume", MAX98373_R20AD_BDE_L2_CFG_3,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL3 Clip Reduction Volume", MAX98373_R20B0_BDE_L3_CFG_3,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL4 Clip Reduction Volume", MAX98373_R20B3_BDE_L4_CFG_3,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL1 Limiter Thresh Volume", MAX98373_R20A8_BDE_L1_CFG_1,
-	0, 0xF, 0, max98373_limiter_thresh_tlv),
+	0, 0xF, 1, max98373_limiter_thresh_tlv),
 SOC_SINGLE_TLV("BDE LVL2 Limiter Thresh Volume", MAX98373_R20AB_BDE_L2_CFG_1,
-	0, 0xF, 0, max98373_limiter_thresh_tlv),
+	0, 0xF, 1, max98373_limiter_thresh_tlv),
 SOC_SINGLE_TLV("BDE LVL3 Limiter Thresh Volume", MAX98373_R20AE_BDE_L3_CFG_1,
-	0, 0xF, 0, max98373_limiter_thresh_tlv),
+	0, 0xF, 1, max98373_limiter_thresh_tlv),
 SOC_SINGLE_TLV("BDE LVL4 Limiter Thresh Volume", MAX98373_R20B1_BDE_L4_CFG_1,
-	0, 0xF, 0, max98373_limiter_thresh_tlv),
+	0, 0xF, 1, max98373_limiter_thresh_tlv),
 /* Limiter */
 SOC_SINGLE("Limiter Switch", MAX98373_R20E2_LIMITER_EN,
 	MAX98373_LIMITER_EN_SHIFT, 1, 0),
diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c
new file mode 100644
index 000000000000..622ce947f134
--- /dev/null
+++ b/sound/soc/codecs/nau8822.c
@@ -0,0 +1,1136 @@
+/*
+ * nau8822.c  --  NAU8822 ALSA Soc Audio Codec driver
+ *
+ * Copyright 2017 Nuvoton Technology Corp.
+ *
+ * Author: David Lin <ctlin0@nuvoton.com>
+ * Co-author: John Hsu <kchsu0@nuvoton.com>
+ * Co-author: Seven Li <wtli@nuvoton.com>
+ *
+ * Based on WM8974.c
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <asm/div64.h>
+#include "nau8822.h"
+
+#define NAU_PLL_FREQ_MAX 100000000
+#define NAU_PLL_FREQ_MIN 90000000
+#define NAU_PLL_REF_MAX 33000000
+#define NAU_PLL_REF_MIN 8000000
+#define NAU_PLL_OPTOP_MIN 6
+
+static const int nau8822_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 };
+
+static const struct reg_default nau8822_reg_defaults[] = {
+	{ NAU8822_REG_POWER_MANAGEMENT_1, 0x0000 },
+	{ NAU8822_REG_POWER_MANAGEMENT_2, 0x0000 },
+	{ NAU8822_REG_POWER_MANAGEMENT_3, 0x0000 },
+	{ NAU8822_REG_AUDIO_INTERFACE, 0x0050 },
+	{ NAU8822_REG_COMPANDING_CONTROL, 0x0000 },
+	{ NAU8822_REG_CLOCKING, 0x0140 },
+	{ NAU8822_REG_ADDITIONAL_CONTROL, 0x0000 },
+	{ NAU8822_REG_GPIO_CONTROL, 0x0000 },
+	{ NAU8822_REG_JACK_DETECT_CONTROL_1, 0x0000 },
+	{ NAU8822_REG_DAC_CONTROL, 0x0000 },
+	{ NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, 0x00ff },
+	{ NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0x00ff },
+	{ NAU8822_REG_JACK_DETECT_CONTROL_2, 0x0000 },
+	{ NAU8822_REG_ADC_CONTROL, 0x0100 },
+	{ NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, 0x00ff },
+	{ NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0x00ff },
+	{ NAU8822_REG_EQ1, 0x012c },
+	{ NAU8822_REG_EQ2, 0x002c },
+	{ NAU8822_REG_EQ3, 0x002c },
+	{ NAU8822_REG_EQ4, 0x002c },
+	{ NAU8822_REG_EQ5, 0x002c },
+	{ NAU8822_REG_DAC_LIMITER_1, 0x0032 },
+	{ NAU8822_REG_DAC_LIMITER_2, 0x0000 },
+	{ NAU8822_REG_NOTCH_FILTER_1, 0x0000 },
+	{ NAU8822_REG_NOTCH_FILTER_2, 0x0000 },
+	{ NAU8822_REG_NOTCH_FILTER_3, 0x0000 },
+	{ NAU8822_REG_NOTCH_FILTER_4, 0x0000 },
+	{ NAU8822_REG_ALC_CONTROL_1, 0x0038 },
+	{ NAU8822_REG_ALC_CONTROL_2, 0x000b },
+	{ NAU8822_REG_ALC_CONTROL_3, 0x0032 },
+	{ NAU8822_REG_NOISE_GATE, 0x0010 },
+	{ NAU8822_REG_PLL_N, 0x0008 },
+	{ NAU8822_REG_PLL_K1, 0x000c },
+	{ NAU8822_REG_PLL_K2, 0x0093 },
+	{ NAU8822_REG_PLL_K3, 0x00e9 },
+	{ NAU8822_REG_3D_CONTROL, 0x0000 },
+	{ NAU8822_REG_RIGHT_SPEAKER_CONTROL, 0x0000 },
+	{ NAU8822_REG_INPUT_CONTROL, 0x0033 },
+	{ NAU8822_REG_LEFT_INP_PGA_CONTROL, 0x0010 },
+	{ NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0x0010 },
+	{ NAU8822_REG_LEFT_ADC_BOOST_CONTROL, 0x0100 },
+	{ NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0x0100 },
+	{ NAU8822_REG_OUTPUT_CONTROL, 0x0002 },
+	{ NAU8822_REG_LEFT_MIXER_CONTROL, 0x0001 },
+	{ NAU8822_REG_RIGHT_MIXER_CONTROL, 0x0001 },
+	{ NAU8822_REG_LHP_VOLUME, 0x0039 },
+	{ NAU8822_REG_RHP_VOLUME, 0x0039 },
+	{ NAU8822_REG_LSPKOUT_VOLUME, 0x0039 },
+	{ NAU8822_REG_RSPKOUT_VOLUME, 0x0039 },
+	{ NAU8822_REG_AUX2_MIXER, 0x0001 },
+	{ NAU8822_REG_AUX1_MIXER, 0x0001 },
+	{ NAU8822_REG_POWER_MANAGEMENT_4, 0x0000 },
+	{ NAU8822_REG_LEFT_TIME_SLOT, 0x0000 },
+	{ NAU8822_REG_MISC, 0x0020 },
+	{ NAU8822_REG_RIGHT_TIME_SLOT, 0x0000 },
+	{ NAU8822_REG_DEVICE_REVISION, 0x007f },
+	{ NAU8822_REG_DEVICE_ID, 0x001a },
+	{ NAU8822_REG_DAC_DITHER, 0x0114 },
+	{ NAU8822_REG_ALC_ENHANCE_1, 0x0000 },
+	{ NAU8822_REG_ALC_ENHANCE_2, 0x0000 },
+	{ NAU8822_REG_192KHZ_SAMPLING, 0x0008 },
+	{ NAU8822_REG_MISC_CONTROL, 0x0000 },
+	{ NAU8822_REG_INPUT_TIEOFF, 0x0000 },
+	{ NAU8822_REG_POWER_REDUCTION, 0x0000 },
+	{ NAU8822_REG_AGC_PEAK2PEAK, 0x0000 },
+	{ NAU8822_REG_AGC_PEAK_DETECT, 0x0000 },
+	{ NAU8822_REG_AUTOMUTE_CONTROL, 0x0000 },
+	{ NAU8822_REG_OUTPUT_TIEOFF, 0x0000 },
+};
+
+static bool nau8822_readable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1:
+	case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME:
+	case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME:
+	case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5:
+	case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2:
+	case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4:
+	case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3:
+	case NAU8822_REG_3D_CONTROL:
+	case NAU8822_REG_RIGHT_SPEAKER_CONTROL:
+	case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL:
+	case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER:
+	case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID:
+	case NAU8822_REG_DAC_DITHER:
+	case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL:
+	case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool nau8822_writeable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1:
+	case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME:
+	case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME:
+	case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5:
+	case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2:
+	case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4:
+	case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3:
+	case NAU8822_REG_3D_CONTROL:
+	case NAU8822_REG_RIGHT_SPEAKER_CONTROL:
+	case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL:
+	case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER:
+	case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID:
+	case NAU8822_REG_DAC_DITHER:
+	case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL:
+	case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool nau8822_volatile(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case NAU8822_REG_RESET:
+	case NAU8822_REG_DEVICE_REVISION:
+	case NAU8822_REG_DEVICE_ID:
+	case NAU8822_REG_AGC_PEAK2PEAK:
+	case NAU8822_REG_AGC_PEAK_DETECT:
+	case NAU8822_REG_AUTOMUTE_CONTROL:
+		return true;
+	default:
+		return false;
+	}
+}
+
+/* The EQ parameters get function is to get the 5 band equalizer control.
+ * The regmap raw read can't work here because regmap doesn't provide
+ * value format for value width of 9 bits. Therefore, the driver reads data
+ * from cache and makes value format according to the endianness of
+ * bytes type control element.
+ */
+static int nau8822_eq_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component =
+			snd_soc_kcontrol_component(kcontrol);
+	struct soc_bytes_ext *params = (void *)kcontrol->private_value;
+	int i, reg;
+	u16 reg_val, *val;
+
+	val = (u16 *)ucontrol->value.bytes.data;
+	reg = NAU8822_REG_EQ1;
+	for (i = 0; i < params->max / sizeof(u16); i++) {
+		reg_val = snd_soc_component_read32(component, reg + i);
+		/* conversion of 16-bit integers between native CPU format
+		 * and big endian format
+		 */
+		reg_val = cpu_to_be16(reg_val);
+		memcpy(val + i, &reg_val, sizeof(reg_val));
+	}
+
+	return 0;
+}
+
+/* The EQ parameters put function is to make configuration of 5 band equalizer
+ * control. These configuration includes central frequency, equalizer gain,
+ * cut-off frequency, bandwidth control, and equalizer path.
+ * The regmap raw write can't work here because regmap doesn't provide
+ * register and value format for register with address 7 bits and value 9 bits.
+ * Therefore, the driver makes value format according to the endianness of
+ * bytes type control element and writes data to codec.
+ */
+static int nau8822_eq_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component =
+			snd_soc_kcontrol_component(kcontrol);
+	struct soc_bytes_ext *params = (void *)kcontrol->private_value;
+	void *data;
+	u16 *val, value;
+	int i, reg, ret;
+
+	data = kmemdup(ucontrol->value.bytes.data,
+		params->max, GFP_KERNEL | GFP_DMA);
+	if (!data)
+		return -ENOMEM;
+
+	val = (u16 *)data;
+	reg = NAU8822_REG_EQ1;
+	for (i = 0; i < params->max / sizeof(u16); i++) {
+		/* conversion of 16-bit integers between native CPU format
+		 * and big endian format
+		 */
+		value = be16_to_cpu(*(val + i));
+		ret = snd_soc_component_write(component, reg + i, value);
+		if (ret) {
+			dev_err(component->dev,
+			    "EQ configuration fail, register: %x ret: %d\n",
+			    reg + i, ret);
+			kfree(data);
+			return ret;
+		}
+	}
+	kfree(data);
+
+	return 0;
+}
+
+static const char * const nau8822_companding[] = {
+	"Off", "NC", "u-law", "A-law"};
+
+static const struct soc_enum nau8822_companding_adc_enum =
+	SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_ADCCM_SFT,
+		ARRAY_SIZE(nau8822_companding), nau8822_companding);
+
+static const struct soc_enum nau8822_companding_dac_enum =
+	SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_DACCM_SFT,
+		ARRAY_SIZE(nau8822_companding), nau8822_companding);
+
+static const char * const nau8822_eqmode[] = {"Capture", "Playback"};
+
+static const struct soc_enum nau8822_eqmode_enum =
+	SOC_ENUM_SINGLE(NAU8822_REG_EQ1, NAU8822_EQM_SFT,
+		ARRAY_SIZE(nau8822_eqmode), nau8822_eqmode);
+
+static const char * const nau8822_alc1[] = {"Off", "Right", "Left", "Both"};
+static const char * const nau8822_alc3[] = {"Normal", "Limiter"};
+
+static const struct soc_enum nau8822_alc_enable_enum =
+	SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_1, NAU8822_ALCEN_SFT,
+		ARRAY_SIZE(nau8822_alc1), nau8822_alc1);
+
+static const struct soc_enum nau8822_alc_mode_enum =
+	SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_3, NAU8822_ALCM_SFT,
+		ARRAY_SIZE(nau8822_alc3), nau8822_alc3);
+
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0);
+static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0);
+static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0);
+static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1);
+static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0);
+
+static const struct snd_kcontrol_new nau8822_snd_controls[] = {
+	SOC_ENUM("ADC Companding", nau8822_companding_adc_enum),
+	SOC_ENUM("DAC Companding", nau8822_companding_dac_enum),
+
+	SOC_ENUM("EQ Function", nau8822_eqmode_enum),
+	SND_SOC_BYTES_EXT("EQ Parameters", 10,
+		  nau8822_eq_get, nau8822_eq_put),
+
+	SOC_DOUBLE("DAC Inversion Switch",
+		NAU8822_REG_DAC_CONTROL, 0, 1, 1, 0),
+	SOC_DOUBLE_R_TLV("PCM Volume",
+		NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME,
+		NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv),
+
+	SOC_SINGLE("High Pass Filter Switch",
+		NAU8822_REG_ADC_CONTROL, 8, 1, 0),
+	SOC_SINGLE("High Pass Cut Off",
+		NAU8822_REG_ADC_CONTROL, 4, 7, 0),
+
+	SOC_DOUBLE("ADC Inversion Switch",
+		NAU8822_REG_ADC_CONTROL, 0, 1, 1, 0),
+	SOC_DOUBLE_R_TLV("ADC Volume",
+		NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME,
+		NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv),
+
+	SOC_SINGLE("DAC Limiter Switch",
+		NAU8822_REG_DAC_LIMITER_1, 8, 1, 0),
+	SOC_SINGLE("DAC Limiter Decay",
+		NAU8822_REG_DAC_LIMITER_1, 4, 15, 0),
+	SOC_SINGLE("DAC Limiter Attack",
+		NAU8822_REG_DAC_LIMITER_1, 0, 15, 0),
+	SOC_SINGLE("DAC Limiter Threshold",
+		NAU8822_REG_DAC_LIMITER_2, 4, 7, 0),
+	SOC_SINGLE_TLV("DAC Limiter Volume",
+		NAU8822_REG_DAC_LIMITER_2, 0, 12, 0, limiter_tlv),
+
+	SOC_ENUM("ALC Mode", nau8822_alc_mode_enum),
+	SOC_ENUM("ALC Enable Switch", nau8822_alc_enable_enum),
+	SOC_SINGLE("ALC Min Gain",
+		NAU8822_REG_ALC_CONTROL_1, 0, 7, 0),
+	SOC_SINGLE("ALC Max Gain",
+		NAU8822_REG_ALC_CONTROL_1, 3, 7, 0),
+	SOC_SINGLE("ALC Hold",
+		NAU8822_REG_ALC_CONTROL_2, 4, 10, 0),
+	SOC_SINGLE("ALC Target",
+		NAU8822_REG_ALC_CONTROL_2, 0, 15, 0),
+	SOC_SINGLE("ALC Decay",
+		NAU8822_REG_ALC_CONTROL_3, 4, 10, 0),
+	SOC_SINGLE("ALC Attack",
+		NAU8822_REG_ALC_CONTROL_3, 0, 10, 0),
+	SOC_SINGLE("ALC Noise Gate Switch",
+		NAU8822_REG_NOISE_GATE, 3, 1, 0),
+	SOC_SINGLE("ALC Noise Gate Threshold",
+		NAU8822_REG_NOISE_GATE, 0, 7, 0),
+
+	SOC_DOUBLE_R("PGA ZC Switch",
+		NAU8822_REG_LEFT_INP_PGA_CONTROL,
+		NAU8822_REG_RIGHT_INP_PGA_CONTROL,
+		7, 1, 0),
+	SOC_DOUBLE_R_TLV("PGA Volume",
+		NAU8822_REG_LEFT_INP_PGA_CONTROL,
+		NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0, 63, 0, inpga_tlv),
+
+	SOC_DOUBLE_R("Headphone ZC Switch",
+		NAU8822_REG_LHP_VOLUME,
+		NAU8822_REG_RHP_VOLUME, 7, 1, 0),
+	SOC_DOUBLE_R("Headphone Playback Switch",
+		NAU8822_REG_LHP_VOLUME,
+		NAU8822_REG_RHP_VOLUME, 6, 1, 1),
+	SOC_DOUBLE_R_TLV("Headphone Volume",
+		NAU8822_REG_LHP_VOLUME,
+		NAU8822_REG_RHP_VOLUME,	0, 63, 0, spk_tlv),
+
+	SOC_DOUBLE_R("Speaker ZC Switch",
+		NAU8822_REG_LSPKOUT_VOLUME,
+		NAU8822_REG_RSPKOUT_VOLUME, 7, 1, 0),
+	SOC_DOUBLE_R("Speaker Playback Switch",
+		NAU8822_REG_LSPKOUT_VOLUME,
+		NAU8822_REG_RSPKOUT_VOLUME, 6, 1, 1),
+	SOC_DOUBLE_R_TLV("Speaker Volume",
+		NAU8822_REG_LSPKOUT_VOLUME,
+		NAU8822_REG_RSPKOUT_VOLUME, 0, 63, 0, spk_tlv),
+
+	SOC_DOUBLE_R("AUXOUT Playback Switch",
+		NAU8822_REG_AUX2_MIXER,
+		NAU8822_REG_AUX1_MIXER, 6, 1, 1),
+
+	SOC_DOUBLE_R_TLV("PGA Boost Volume",
+		NAU8822_REG_LEFT_ADC_BOOST_CONTROL,
+		NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 8, 1, 0, pga_boost_tlv),
+	SOC_DOUBLE_R_TLV("L2/R2 Boost Volume",
+		NAU8822_REG_LEFT_ADC_BOOST_CONTROL,
+		NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 4, 7, 0, boost_tlv),
+	SOC_DOUBLE_R_TLV("Aux Boost Volume",
+		NAU8822_REG_LEFT_ADC_BOOST_CONTROL,
+		NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0, 7, 0, boost_tlv),
+
+	SOC_SINGLE("DAC 128x Oversampling Switch",
+		NAU8822_REG_DAC_CONTROL, 5, 1, 0),
+	SOC_SINGLE("ADC 128x Oversampling Switch",
+		NAU8822_REG_ADC_CONTROL, 5, 1, 0),
+};
+
+/* LMAIN and RMAIN Mixer */
+static const struct snd_kcontrol_new nau8822_left_out_mixer[] = {
+	SOC_DAPM_SINGLE("LINMIX Switch",
+		NAU8822_REG_LEFT_MIXER_CONTROL, 1, 1, 0),
+	SOC_DAPM_SINGLE("LAUX Switch",
+		NAU8822_REG_LEFT_MIXER_CONTROL, 5, 1, 0),
+	SOC_DAPM_SINGLE("LDAC Switch",
+		NAU8822_REG_LEFT_MIXER_CONTROL, 0, 1, 0),
+	SOC_DAPM_SINGLE("RDAC Switch",
+		NAU8822_REG_OUTPUT_CONTROL, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new nau8822_right_out_mixer[] = {
+	SOC_DAPM_SINGLE("RINMIX Switch",
+		NAU8822_REG_RIGHT_MIXER_CONTROL, 1, 1, 0),
+	SOC_DAPM_SINGLE("RAUX Switch",
+		NAU8822_REG_RIGHT_MIXER_CONTROL, 5, 1, 0),
+	SOC_DAPM_SINGLE("RDAC Switch",
+		NAU8822_REG_RIGHT_MIXER_CONTROL, 0, 1, 0),
+	SOC_DAPM_SINGLE("LDAC Switch",
+		NAU8822_REG_OUTPUT_CONTROL, 6, 1, 0),
+};
+
+/* AUX1 and AUX2 Mixer */
+static const struct snd_kcontrol_new nau8822_auxout1_mixer[] = {
+	SOC_DAPM_SINGLE("RDAC Switch", NAU8822_REG_AUX1_MIXER, 0, 1, 0),
+	SOC_DAPM_SINGLE("RMIX Switch", NAU8822_REG_AUX1_MIXER, 1, 1, 0),
+	SOC_DAPM_SINGLE("RINMIX Switch", NAU8822_REG_AUX1_MIXER, 2, 1, 0),
+	SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX1_MIXER, 3, 1, 0),
+	SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX1_MIXER, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new nau8822_auxout2_mixer[] = {
+	SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX2_MIXER, 0, 1, 0),
+	SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX2_MIXER, 1, 1, 0),
+	SOC_DAPM_SINGLE("LINMIX Switch", NAU8822_REG_AUX2_MIXER, 2, 1, 0),
+	SOC_DAPM_SINGLE("AUX1MIX Output Switch",
+		NAU8822_REG_AUX2_MIXER, 3, 1, 0),
+};
+
+/* Input PGA */
+static const struct snd_kcontrol_new nau8822_left_input_mixer[] = {
+	SOC_DAPM_SINGLE("L2 Switch", NAU8822_REG_INPUT_CONTROL, 2, 1, 0),
+	SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 1, 1, 0),
+	SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 0, 1, 0),
+};
+static const struct snd_kcontrol_new nau8822_right_input_mixer[] = {
+	SOC_DAPM_SINGLE("R2 Switch", NAU8822_REG_INPUT_CONTROL, 6, 1, 0),
+	SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 5, 1, 0),
+	SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 4, 1, 0),
+};
+
+/* Loopback Switch */
+static const struct snd_kcontrol_new nau8822_loopback =
+	SOC_DAPM_SINGLE("Switch", NAU8822_REG_COMPANDING_CONTROL,
+		NAU8822_ADDAP_SFT, 1, 0);
+
+static int check_mclk_select_pll(struct snd_soc_dapm_widget *source,
+			 struct snd_soc_dapm_widget *sink)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(source->dapm);
+	unsigned int value;
+
+	value = snd_soc_component_read32(component, NAU8822_REG_CLOCKING);
+
+	return (value & NAU8822_CLKM_MASK);
+}
+
+static const struct snd_soc_dapm_widget nau8822_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
+		NAU8822_REG_POWER_MANAGEMENT_3, 0, 0),
+	SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
+		NAU8822_REG_POWER_MANAGEMENT_3, 1, 0),
+	SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
+		NAU8822_REG_POWER_MANAGEMENT_2, 0, 0),
+	SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
+		NAU8822_REG_POWER_MANAGEMENT_2, 1, 0),
+
+	SOC_MIXER_ARRAY("Left Output Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_3, 2, 0, nau8822_left_out_mixer),
+	SOC_MIXER_ARRAY("Right Output Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_3,	3, 0, nau8822_right_out_mixer),
+	SOC_MIXER_ARRAY("AUX1 Output Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_1, 7, 0, nau8822_auxout1_mixer),
+	SOC_MIXER_ARRAY("AUX2 Output Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_1,	6, 0, nau8822_auxout2_mixer),
+
+	SOC_MIXER_ARRAY("Left Input Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_2,
+		2, 0, nau8822_left_input_mixer),
+	SOC_MIXER_ARRAY("Right Input Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_2,
+		3, 0, nau8822_right_input_mixer),
+
+	SND_SOC_DAPM_PGA("Left Boost Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_2, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Boost Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_2, 5, 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA("Left Capture PGA",
+		NAU8822_REG_LEFT_INP_PGA_CONTROL, 6, 1, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Capture PGA",
+		NAU8822_REG_RIGHT_INP_PGA_CONTROL, 6, 1, NULL, 0),
+
+	SND_SOC_DAPM_PGA("Left Headphone Out",
+		NAU8822_REG_POWER_MANAGEMENT_2, 7, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Headphone Out",
+		NAU8822_REG_POWER_MANAGEMENT_2, 8, 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA("Left Speaker Out",
+		 NAU8822_REG_POWER_MANAGEMENT_3, 6, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Speaker Out",
+		NAU8822_REG_POWER_MANAGEMENT_3,	5, 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA("AUX1 Out",
+		NAU8822_REG_POWER_MANAGEMENT_3,	8, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("AUX2 Out",
+		NAU8822_REG_POWER_MANAGEMENT_3,	7, 0, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("Mic Bias",
+		NAU8822_REG_POWER_MANAGEMENT_1,	4, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLL",
+		NAU8822_REG_POWER_MANAGEMENT_1,	5, 0, NULL, 0),
+
+	SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0,
+		&nau8822_loopback),
+
+	SND_SOC_DAPM_INPUT("LMICN"),
+	SND_SOC_DAPM_INPUT("LMICP"),
+	SND_SOC_DAPM_INPUT("RMICN"),
+	SND_SOC_DAPM_INPUT("RMICP"),
+	SND_SOC_DAPM_INPUT("LAUX"),
+	SND_SOC_DAPM_INPUT("RAUX"),
+	SND_SOC_DAPM_INPUT("L2"),
+	SND_SOC_DAPM_INPUT("R2"),
+	SND_SOC_DAPM_OUTPUT("LHP"),
+	SND_SOC_DAPM_OUTPUT("RHP"),
+	SND_SOC_DAPM_OUTPUT("LSPK"),
+	SND_SOC_DAPM_OUTPUT("RSPK"),
+	SND_SOC_DAPM_OUTPUT("AUXOUT1"),
+	SND_SOC_DAPM_OUTPUT("AUXOUT2"),
+};
+
+static const struct snd_soc_dapm_route nau8822_dapm_routes[] = {
+	{"Right DAC", NULL, "PLL", check_mclk_select_pll},
+	{"Left DAC", NULL, "PLL", check_mclk_select_pll},
+
+	/* LMAIN and RMAIN Mixer */
+	{"Right Output Mixer", "LDAC Switch", "Left DAC"},
+	{"Right Output Mixer", "RDAC Switch", "Right DAC"},
+	{"Right Output Mixer", "RAUX Switch", "RAUX"},
+	{"Right Output Mixer", "RINMIX Switch", "Right Boost Mixer"},
+
+	{"Left Output Mixer", "LDAC Switch", "Left DAC"},
+	{"Left Output Mixer", "RDAC Switch", "Right DAC"},
+	{"Left Output Mixer", "LAUX Switch", "LAUX"},
+	{"Left Output Mixer", "LINMIX Switch", "Left Boost Mixer"},
+
+	/* AUX1 and AUX2 Mixer */
+	{"AUX1 Output Mixer", "RDAC Switch", "Right DAC"},
+	{"AUX1 Output Mixer", "RMIX Switch", "Right Output Mixer"},
+	{"AUX1 Output Mixer", "RINMIX Switch", "Right Boost Mixer"},
+	{"AUX1 Output Mixer", "LDAC Switch", "Left DAC"},
+	{"AUX1 Output Mixer", "LMIX Switch", "Left Output Mixer"},
+
+	{"AUX2 Output Mixer", "LDAC Switch", "Left DAC"},
+	{"AUX2 Output Mixer", "LMIX Switch", "Left Output Mixer"},
+	{"AUX2 Output Mixer", "LINMIX Switch", "Left Boost Mixer"},
+	{"AUX2 Output Mixer", "AUX1MIX Output Switch", "AUX1 Output Mixer"},
+
+	/* Outputs */
+	{"Right Headphone Out", NULL, "Right Output Mixer"},
+	{"RHP", NULL, "Right Headphone Out"},
+
+	{"Left Headphone Out", NULL, "Left Output Mixer"},
+	{"LHP", NULL, "Left Headphone Out"},
+
+	{"Right Speaker Out", NULL, "Right Output Mixer"},
+	{"RSPK", NULL, "Right Speaker Out"},
+
+	{"Left Speaker Out", NULL, "Left Output Mixer"},
+	{"LSPK", NULL, "Left Speaker Out"},
+
+	{"AUX1 Out", NULL, "AUX1 Output Mixer"},
+	{"AUX2 Out", NULL, "AUX2 Output Mixer"},
+	{"AUXOUT1", NULL, "AUX1 Out"},
+	{"AUXOUT2", NULL, "AUX2 Out"},
+
+	/* Boost Mixer */
+	{"Right ADC", NULL, "PLL", check_mclk_select_pll},
+	{"Left ADC", NULL, "PLL", check_mclk_select_pll},
+
+	{"Right ADC", NULL, "Right Boost Mixer"},
+
+	{"Right Boost Mixer", NULL, "RAUX"},
+	{"Right Boost Mixer", NULL, "Right Capture PGA"},
+	{"Right Boost Mixer", NULL, "R2"},
+
+	{"Left ADC", NULL, "Left Boost Mixer"},
+
+	{"Left Boost Mixer", NULL, "LAUX"},
+	{"Left Boost Mixer", NULL, "Left Capture PGA"},
+	{"Left Boost Mixer", NULL, "L2"},
+
+	/* Input PGA */
+	{"Right Capture PGA", NULL, "Right Input Mixer"},
+	{"Left Capture PGA", NULL, "Left Input Mixer"},
+
+	/* Enable Microphone Power */
+	{"Right Capture PGA", NULL, "Mic Bias"},
+	{"Left Capture PGA", NULL, "Mic Bias"},
+
+	{"Right Input Mixer", "R2 Switch", "R2"},
+	{"Right Input Mixer", "MicN Switch", "RMICN"},
+	{"Right Input Mixer", "MicP Switch", "RMICP"},
+
+	{"Left Input Mixer", "L2 Switch", "L2"},
+	{"Left Input Mixer", "MicN Switch", "LMICN"},
+	{"Left Input Mixer", "MicP Switch", "LMICP"},
+
+	/* Digital Loopback */
+	{"Digital Loopback", "Switch", "Left ADC"},
+	{"Digital Loopback", "Switch", "Right ADC"},
+	{"Left DAC", NULL, "Digital Loopback"},
+	{"Right DAC", NULL, "Digital Loopback"},
+};
+
+static int nau8822_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+				 unsigned int freq, int dir)
+{
+	struct snd_soc_component *component = dai->component;
+	struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+
+	nau8822->div_id = clk_id;
+	nau8822->sysclk = freq;
+	dev_dbg(component->dev, "master sysclk %dHz, source %s\n", freq,
+		clk_id == NAU8822_CLK_PLL ? "PLL" : "MCLK");
+
+	return 0;
+}
+
+static int nau8822_calc_pll(unsigned int pll_in, unsigned int fs,
+				struct nau8822_pll *pll_param)
+{
+	u64 f2, f2_max, pll_ratio;
+	int i, scal_sel;
+
+	if (pll_in > NAU_PLL_REF_MAX || pll_in < NAU_PLL_REF_MIN)
+		return -EINVAL;
+	f2_max = 0;
+	scal_sel = ARRAY_SIZE(nau8822_mclk_scaler);
+
+	for (i = 0; i < scal_sel; i++) {
+		f2 = 256 * fs * 4 * nau8822_mclk_scaler[i] / 10;
+		if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX &&
+			f2_max < f2) {
+			f2_max = f2;
+			scal_sel = i;
+		}
+	}
+
+	if (ARRAY_SIZE(nau8822_mclk_scaler) == scal_sel)
+		return -EINVAL;
+	pll_param->mclk_scaler = scal_sel;
+	f2 = f2_max;
+
+	/* Calculate the PLL 4-bit integer input and the PLL 24-bit fractional
+	 * input; round up the 24+4bit.
+	 */
+	pll_ratio = div_u64(f2 << 28, pll_in);
+	pll_param->pre_factor = 0;
+	if (((pll_ratio >> 28) & 0xF) < NAU_PLL_OPTOP_MIN) {
+		pll_ratio <<= 1;
+		pll_param->pre_factor = 1;
+	}
+	pll_param->pll_int = (pll_ratio >> 28) & 0xF;
+	pll_param->pll_frac = ((pll_ratio & 0xFFFFFFF) >> 4);
+
+	return 0;
+}
+
+static int nau8822_config_clkdiv(struct snd_soc_dai *dai, int div, int rate)
+{
+	struct snd_soc_component *component = dai->component;
+	struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+	struct nau8822_pll *pll = &nau8822->pll;
+	int i, sclk, imclk;
+
+	switch (nau8822->div_id) {
+	case NAU8822_CLK_MCLK:
+		/* Configure the master clock prescaler div to make system
+		 * clock to approximate the internal master clock (IMCLK);
+		 * and large or equal to IMCLK.
+		 */
+		div = 0;
+		imclk = rate * 256;
+		for (i = 1; i < ARRAY_SIZE(nau8822_mclk_scaler); i++) {
+			sclk = (nau8822->sysclk * 10) /	nau8822_mclk_scaler[i];
+			if (sclk < imclk)
+				break;
+			div = i;
+		}
+		dev_dbg(component->dev, "master clock prescaler %x for fs %d\n",
+			div, rate);
+
+		/* master clock from MCLK and disable PLL */
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK,
+			(div << NAU8822_MCLKSEL_SFT));
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK,
+			NAU8822_CLKM_MCLK);
+		break;
+
+	case NAU8822_CLK_PLL:
+		/* master clock from PLL and enable PLL */
+		if (pll->mclk_scaler != div) {
+			dev_err(component->dev,
+			"master clock prescaler not meet PLL parameters\n");
+			return -EINVAL;
+		}
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK,
+			(div << NAU8822_MCLKSEL_SFT));
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK,
+			NAU8822_CLKM_PLL);
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source,
+				unsigned int freq_in, unsigned int freq_out)
+{
+	struct snd_soc_component *component = dai->component;
+	struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+	struct nau8822_pll *pll_param = &nau8822->pll;
+	int ret, fs;
+
+	fs = freq_out / 256;
+
+	ret = nau8822_calc_pll(freq_in, fs, pll_param);
+	if (ret < 0) {
+		dev_err(component->dev, "Unsupported input clock %d\n",
+			freq_in);
+		return ret;
+	}
+
+	dev_info(component->dev,
+		"pll_int=%x pll_frac=%x mclk_scaler=%x pre_factor=%x\n",
+		pll_param->pll_int, pll_param->pll_frac,
+		pll_param->mclk_scaler, pll_param->pre_factor);
+
+	snd_soc_component_update_bits(component,
+		NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK,
+		(pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) |
+		pll_param->pll_int);
+	snd_soc_component_write(component,
+		NAU8822_REG_PLL_K1, (pll_param->pll_frac >> NAU8822_PLLK1_SFT) &
+		NAU8822_PLLK1_MASK);
+	snd_soc_component_write(component,
+		NAU8822_REG_PLL_K2, (pll_param->pll_frac >> NAU8822_PLLK2_SFT) &
+		NAU8822_PLLK2_MASK);
+	snd_soc_component_write(component,
+		NAU8822_REG_PLL_K3, pll_param->pll_frac & NAU8822_PLLK3_MASK);
+	snd_soc_component_update_bits(component,
+		NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK,
+		pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT);
+	snd_soc_component_update_bits(component,
+		NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL);
+
+	return 0;
+}
+
+static int nau8822_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_component *component = dai->component;
+	u16 ctrl1_val = 0, ctrl2_val = 0;
+
+	dev_dbg(component->dev, "%s\n", __func__);
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		ctrl2_val |= 1;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		ctrl2_val &= ~1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		ctrl1_val |= 0x10;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		ctrl1_val |= 0x8;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		ctrl1_val |= 0x18;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		ctrl1_val |= 0x180;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		ctrl1_val |= 0x100;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		ctrl1_val |= 0x80;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_component_update_bits(component,
+		NAU8822_REG_AUDIO_INTERFACE,
+		NAU8822_AIFMT_MASK | NAU8822_LRP_MASK | NAU8822_BCLKP_MASK,
+		ctrl1_val);
+	snd_soc_component_update_bits(component,
+		NAU8822_REG_CLOCKING, NAU8822_CLKIOEN_MASK, ctrl2_val);
+
+	return 0;
+}
+
+static int nau8822_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+	int val_len = 0, val_rate = 0;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		val_len |= NAU8822_WLEN_20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		val_len |= NAU8822_WLEN_24;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		val_len |= NAU8822_WLEN_32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (params_rate(params)) {
+	case 8000:
+		val_rate |= NAU8822_SMPLR_8K;
+		break;
+	case 11025:
+		val_rate |= NAU8822_SMPLR_12K;
+		break;
+	case 16000:
+		val_rate |= NAU8822_SMPLR_16K;
+		break;
+	case 22050:
+		val_rate |= NAU8822_SMPLR_24K;
+		break;
+	case 32000:
+		val_rate |= NAU8822_SMPLR_32K;
+		break;
+	case 44100:
+	case 48000:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_component_update_bits(component,
+		NAU8822_REG_AUDIO_INTERFACE, NAU8822_WLEN_MASK, val_len);
+	snd_soc_component_update_bits(component,
+		NAU8822_REG_ADDITIONAL_CONTROL, NAU8822_SMPLR_MASK, val_rate);
+
+	/* If the master clock is from MCLK, provide the runtime FS for driver
+	 * to get the master clock prescaler configuration.
+	 */
+	if (nau8822->div_id == NAU8822_CLK_MCLK)
+		nau8822_config_clkdiv(dai, 0, params_rate(params));
+
+	return 0;
+}
+
+static int nau8822_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_component *component = dai->component;
+
+	dev_dbg(component->dev, "%s: %d\n", __func__, mute);
+
+	if (mute)
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_DAC_CONTROL, 0x40, 0x40);
+	else
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_DAC_CONTROL, 0x40, 0);
+
+	return 0;
+}
+
+static int nau8822_set_bias_level(struct snd_soc_component *component,
+				 enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_POWER_MANAGEMENT_1,
+			NAU8822_REFIMP_MASK, NAU8822_REFIMP_80K);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_POWER_MANAGEMENT_1,
+			NAU8822_IOBUF_EN | NAU8822_ABIAS_EN,
+			NAU8822_IOBUF_EN | NAU8822_ABIAS_EN);
+
+		if (snd_soc_component_get_bias_level(component) ==
+			SND_SOC_BIAS_OFF) {
+			snd_soc_component_update_bits(component,
+				NAU8822_REG_POWER_MANAGEMENT_1,
+				NAU8822_REFIMP_MASK, NAU8822_REFIMP_3K);
+			mdelay(100);
+		}
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_POWER_MANAGEMENT_1,
+			NAU8822_REFIMP_MASK, NAU8822_REFIMP_300K);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		snd_soc_component_write(component,
+			NAU8822_REG_POWER_MANAGEMENT_1, 0);
+		snd_soc_component_write(component,
+			NAU8822_REG_POWER_MANAGEMENT_2, 0);
+		snd_soc_component_write(component,
+			NAU8822_REG_POWER_MANAGEMENT_3, 0);
+		break;
+	}
+
+	dev_dbg(component->dev, "%s: %d\n", __func__, level);
+
+	return 0;
+}
+
+#define NAU8822_RATES (SNDRV_PCM_RATE_8000_48000)
+
+#define NAU8822_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static const struct snd_soc_dai_ops nau8822_dai_ops = {
+	.hw_params	= nau8822_hw_params,
+	.digital_mute	= nau8822_mute,
+	.set_fmt	= nau8822_set_dai_fmt,
+	.set_sysclk	= nau8822_set_dai_sysclk,
+	.set_pll	= nau8822_set_pll,
+};
+
+static struct snd_soc_dai_driver nau8822_dai = {
+	.name = "nau8822-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = NAU8822_RATES,
+		.formats = NAU8822_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = NAU8822_RATES,
+		.formats = NAU8822_FORMATS,
+	},
+	.ops = &nau8822_dai_ops,
+	.symmetric_rates = 1,
+};
+
+static int nau8822_suspend(struct snd_soc_component *component)
+{
+	struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+
+	snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF);
+
+	regcache_mark_dirty(nau8822->regmap);
+
+	return 0;
+}
+
+static int nau8822_resume(struct snd_soc_component *component)
+{
+	struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+
+	regcache_sync(nau8822->regmap);
+
+	snd_soc_component_force_bias_level(component, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+
+/*
+ * These registers contain an "update" bit - bit 8. This means, for example,
+ * that one can write new DAC digital volume for both channels, but only when
+ * the update bit is set, will also the volume be updated - simultaneously for
+ * both channels.
+ */
+static const int update_reg[] = {
+	NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME,
+	NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME,
+	NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME,
+	NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME,
+	NAU8822_REG_LEFT_INP_PGA_CONTROL,
+	NAU8822_REG_RIGHT_INP_PGA_CONTROL,
+	NAU8822_REG_LHP_VOLUME,
+	NAU8822_REG_RHP_VOLUME,
+	NAU8822_REG_LSPKOUT_VOLUME,
+	NAU8822_REG_RSPKOUT_VOLUME,
+};
+
+static int nau8822_probe(struct snd_soc_component *component)
+{
+	int i;
+
+	/*
+	 * Set the update bit in all registers, that have one. This way all
+	 * writes to those registers will also cause the update bit to be
+	 * written.
+	 */
+	for (i = 0; i < ARRAY_SIZE(update_reg); i++)
+		snd_soc_component_update_bits(component,
+			update_reg[i], 0x100, 0x100);
+
+	return 0;
+}
+
+static const struct snd_soc_component_driver soc_component_dev_nau8822 = {
+	.probe				= nau8822_probe,
+	.suspend			= nau8822_suspend,
+	.resume				= nau8822_resume,
+	.set_bias_level			= nau8822_set_bias_level,
+	.controls			= nau8822_snd_controls,
+	.num_controls			= ARRAY_SIZE(nau8822_snd_controls),
+	.dapm_widgets			= nau8822_dapm_widgets,
+	.num_dapm_widgets		= ARRAY_SIZE(nau8822_dapm_widgets),
+	.dapm_routes			= nau8822_dapm_routes,
+	.num_dapm_routes		= ARRAY_SIZE(nau8822_dapm_routes),
+	.idle_bias_on			= 1,
+	.use_pmdown_time		= 1,
+	.endianness			= 1,
+	.non_legacy_dai_naming		= 1,
+};
+
+static const struct regmap_config nau8822_regmap_config = {
+	.reg_bits = 7,
+	.val_bits = 9,
+
+	.max_register = NAU8822_REG_MAX_REGISTER,
+	.volatile_reg = nau8822_volatile,
+
+	.readable_reg = nau8822_readable_reg,
+	.writeable_reg = nau8822_writeable_reg,
+
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = nau8822_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(nau8822_reg_defaults),
+};
+
+static int nau8822_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	struct device *dev = &i2c->dev;
+	struct nau8822 *nau8822 = dev_get_platdata(dev);
+	int ret;
+
+	if (!nau8822) {
+		nau8822 = devm_kzalloc(dev, sizeof(*nau8822), GFP_KERNEL);
+		if (nau8822 == NULL)
+			return -ENOMEM;
+	}
+	i2c_set_clientdata(i2c, nau8822);
+
+	nau8822->regmap = devm_regmap_init_i2c(i2c, &nau8822_regmap_config);
+	if (IS_ERR(nau8822->regmap)) {
+		ret = PTR_ERR(nau8822->regmap);
+		dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret);
+		return ret;
+	}
+	nau8822->dev = dev;
+
+	/* Reset the codec */
+	ret = regmap_write(nau8822->regmap, NAU8822_REG_RESET, 0x00);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
+		return ret;
+	}
+
+	ret = devm_snd_soc_register_component(dev, &soc_component_dev_nau8822,
+						&nau8822_dai, 1);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct i2c_device_id nau8822_i2c_id[] = {
+	{ "nau8822", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, nau8822_i2c_id);
+
+#ifdef CONFIG_OF
+static const struct of_device_id nau8822_of_match[] = {
+	{ .compatible = "nuvoton,nau8822", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, nau8822_of_match);
+#endif
+
+static struct i2c_driver nau8822_i2c_driver = {
+	.driver = {
+		.name = "nau8822",
+		.of_match_table = of_match_ptr(nau8822_of_match),
+	},
+	.probe =    nau8822_i2c_probe,
+	.id_table = nau8822_i2c_id,
+};
+module_i2c_driver(nau8822_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC NAU8822 codec driver");
+MODULE_AUTHOR("David Lin <ctlin0@nuvoton.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h
new file mode 100644
index 000000000000..aa79c969cd44
--- /dev/null
+++ b/sound/soc/codecs/nau8822.h
@@ -0,0 +1,204 @@
+/*
+ * nau8822.h  --  NAU8822 Soc Audio Codec driver
+ *
+ * Author: David Lin <ctlin0@nuvoton.com>
+ * Co-author: John Hsu <kchsu0@nuvoton.com>
+ * Co-author: Seven Li <wtli@nuvoton.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __NAU8822_H__
+#define __NAU8822_H__
+
+#define NAU8822_REG_RESET			0x00
+#define NAU8822_REG_POWER_MANAGEMENT_1		0x01
+#define NAU8822_REG_POWER_MANAGEMENT_2		0x02
+#define NAU8822_REG_POWER_MANAGEMENT_3		0x03
+#define NAU8822_REG_AUDIO_INTERFACE		0x04
+#define NAU8822_REG_COMPANDING_CONTROL		0x05
+#define NAU8822_REG_CLOCKING			0x06
+#define NAU8822_REG_ADDITIONAL_CONTROL		0x07
+#define NAU8822_REG_GPIO_CONTROL		0x08
+#define NAU8822_REG_JACK_DETECT_CONTROL_1	0x09
+#define NAU8822_REG_DAC_CONTROL			0x0A
+#define NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME	0x0B
+#define NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME	0x0C
+#define NAU8822_REG_JACK_DETECT_CONTROL_2	0x0D
+#define NAU8822_REG_ADC_CONTROL			0x0E
+#define NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME	0x0F
+#define NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME	0x10
+#define NAU8822_REG_EQ1				0x12
+#define NAU8822_REG_EQ2				0x13
+#define NAU8822_REG_EQ3				0x14
+#define NAU8822_REG_EQ4				0x15
+#define NAU8822_REG_EQ5				0x16
+#define NAU8822_REG_DAC_LIMITER_1		0x18
+#define NAU8822_REG_DAC_LIMITER_2		0x19
+#define NAU8822_REG_NOTCH_FILTER_1		0x1B
+#define NAU8822_REG_NOTCH_FILTER_2		0x1C
+#define NAU8822_REG_NOTCH_FILTER_3		0x1D
+#define NAU8822_REG_NOTCH_FILTER_4		0x1E
+#define NAU8822_REG_ALC_CONTROL_1		0x20
+#define NAU8822_REG_ALC_CONTROL_2		0x21
+#define NAU8822_REG_ALC_CONTROL_3		0x22
+#define NAU8822_REG_NOISE_GATE			0x23
+#define NAU8822_REG_PLL_N			0x24
+#define NAU8822_REG_PLL_K1			0x25
+#define NAU8822_REG_PLL_K2			0x26
+#define NAU8822_REG_PLL_K3			0x27
+#define NAU8822_REG_3D_CONTROL			0x29
+#define NAU8822_REG_RIGHT_SPEAKER_CONTROL	0x2B
+#define NAU8822_REG_INPUT_CONTROL		0x2C
+#define NAU8822_REG_LEFT_INP_PGA_CONTROL	0x2D
+#define NAU8822_REG_RIGHT_INP_PGA_CONTROL	0x2E
+#define NAU8822_REG_LEFT_ADC_BOOST_CONTROL	0x2F
+#define NAU8822_REG_RIGHT_ADC_BOOST_CONTROL	0x30
+#define NAU8822_REG_OUTPUT_CONTROL		0x31
+#define NAU8822_REG_LEFT_MIXER_CONTROL		0x32
+#define NAU8822_REG_RIGHT_MIXER_CONTROL		0x33
+#define NAU8822_REG_LHP_VOLUME			0x34
+#define NAU8822_REG_RHP_VOLUME			0x35
+#define NAU8822_REG_LSPKOUT_VOLUME		0x36
+#define NAU8822_REG_RSPKOUT_VOLUME		0x37
+#define NAU8822_REG_AUX2_MIXER			0x38
+#define NAU8822_REG_AUX1_MIXER			0x39
+#define NAU8822_REG_POWER_MANAGEMENT_4		0x3A
+#define NAU8822_REG_LEFT_TIME_SLOT		0x3B
+#define NAU8822_REG_MISC			0x3C
+#define NAU8822_REG_RIGHT_TIME_SLOT		0x3D
+#define NAU8822_REG_DEVICE_REVISION		0x3E
+#define NAU8822_REG_DEVICE_ID			0x3F
+#define NAU8822_REG_DAC_DITHER			0x41
+#define NAU8822_REG_ALC_ENHANCE_1		0x46
+#define NAU8822_REG_ALC_ENHANCE_2		0x47
+#define NAU8822_REG_192KHZ_SAMPLING		0x48
+#define NAU8822_REG_MISC_CONTROL		0x49
+#define NAU8822_REG_INPUT_TIEOFF		0x4A
+#define NAU8822_REG_POWER_REDUCTION		0x4B
+#define NAU8822_REG_AGC_PEAK2PEAK		0x4C
+#define NAU8822_REG_AGC_PEAK_DETECT		0x4D
+#define NAU8822_REG_AUTOMUTE_CONTROL		0x4E
+#define NAU8822_REG_OUTPUT_TIEOFF		0x4F
+#define NAU8822_REG_MAX_REGISTER		NAU8822_REG_OUTPUT_TIEOFF
+
+/* NAU8822_REG_POWER_MANAGEMENT_1 (0x1) */
+#define NAU8822_REFIMP_MASK			0x3
+#define NAU8822_REFIMP_80K			0x1
+#define NAU8822_REFIMP_300K			0x2
+#define NAU8822_REFIMP_3K			0x3
+#define NAU8822_IOBUF_EN			(0x1 << 2)
+#define NAU8822_ABIAS_EN			(0x1 << 3)
+
+/* NAU8822_REG_AUDIO_INTERFACE (0x4) */
+#define NAU8822_AIFMT_MASK			(0x3 << 3)
+#define NAU8822_WLEN_MASK			(0x3 << 5)
+#define NAU8822_WLEN_20				(0x1 << 5)
+#define NAU8822_WLEN_24				(0x2 << 5)
+#define NAU8822_WLEN_32				(0x3 << 5)
+#define NAU8822_LRP_MASK			(0x1 << 7)
+#define NAU8822_BCLKP_MASK			(0x1 << 8)
+
+/* NAU8822_REG_COMPANDING_CONTROL (0x5) */
+#define NAU8822_ADDAP_SFT			0
+#define NAU8822_ADCCM_SFT			1
+#define NAU8822_DACCM_SFT			3
+
+/* NAU8822_REG_CLOCKING (0x6) */
+#define NAU8822_CLKIOEN_MASK			0x1
+#define NAU8822_MCLKSEL_SFT			5
+#define NAU8822_MCLKSEL_MASK			(0x7 << 5)
+#define NAU8822_BCLKSEL_SFT			2
+#define NAU8822_BCLKSEL_MASK			(0x7 << 2)
+#define NAU8822_CLKM_MASK			(0x1 << 8)
+#define NAU8822_CLKM_MCLK			(0x0 << 8)
+#define NAU8822_CLKM_PLL			(0x1 << 8)
+
+/* NAU8822_REG_ADDITIONAL_CONTROL (0x08) */
+#define NAU8822_SMPLR_SFT			1
+#define NAU8822_SMPLR_MASK			(0x7 << 1)
+#define NAU8822_SMPLR_48K			(0x0 << 1)
+#define NAU8822_SMPLR_32K			(0x1 << 1)
+#define NAU8822_SMPLR_24K			(0x2 << 1)
+#define NAU8822_SMPLR_16K			(0x3 << 1)
+#define NAU8822_SMPLR_12K			(0x4 << 1)
+#define NAU8822_SMPLR_8K			(0x5 << 1)
+
+/* NAU8822_REG_EQ1 (0x12) */
+#define NAU8822_EQ1GC_SFT			0
+#define NAU8822_EQ1CF_SFT			5
+#define NAU8822_EQM_SFT				8
+
+/* NAU8822_REG_EQ2 (0x13) */
+#define NAU8822_EQ2GC_SFT			0
+#define NAU8822_EQ2CF_SFT			5
+#define NAU8822_EQ2BW_SFT			8
+
+/* NAU8822_REG_EQ3 (0x14) */
+#define NAU8822_EQ3GC_SFT			0
+#define NAU8822_EQ3CF_SFT			5
+#define NAU8822_EQ3BW_SFT			8
+
+/* NAU8822_REG_EQ4 (0x15) */
+#define NAU8822_EQ4GC_SFT			0
+#define NAU8822_EQ4CF_SFT			5
+#define NAU8822_EQ4BW_SFT			8
+
+/* NAU8822_REG_EQ5 (0x16) */
+#define NAU8822_EQ5GC_SFT			0
+#define NAU8822_EQ5CF_SFT			5
+
+/* NAU8822_REG_ALC_CONTROL_1 (0x20) */
+#define NAU8822_ALCMINGAIN_SFT			0
+#define NAU8822_ALCMXGAIN_SFT			3
+#define NAU8822_ALCEN_SFT			7
+
+/* NAU8822_REG_ALC_CONTROL_2 (0x21) */
+#define NAU8822_ALCSL_SFT			0
+#define NAU8822_ALCHT_SFT			4
+
+/* NAU8822_REG_ALC_CONTROL_3 (0x22) */
+#define NAU8822_ALCATK_SFT			0
+#define NAU8822_ALCDCY_SFT			4
+#define NAU8822_ALCM_SFT			8
+
+/* NAU8822_REG_PLL_N (0x24) */
+#define NAU8822_PLLMCLK_DIV2			(0x1 << 4)
+#define NAU8822_PLLN_MASK			0xF
+
+#define NAU8822_PLLK1_SFT			18
+#define NAU8822_PLLK1_MASK			0x3F
+
+/* NAU8822_REG_PLL_K2 (0x26) */
+#define NAU8822_PLLK2_SFT			9
+#define NAU8822_PLLK2_MASK			0x1FF
+
+/* NAU8822_REG_PLL_K3 (0x27) */
+#define NAU8822_PLLK3_MASK			0x1FF
+
+/* System Clock Source */
+enum {
+	NAU8822_CLK_MCLK,
+	NAU8822_CLK_PLL,
+};
+
+struct nau8822_pll {
+	int pre_factor;
+	int mclk_scaler;
+	int pll_frac;
+	int pll_int;
+};
+
+/* Codec Private Data */
+struct nau8822 {
+	struct device *dev;
+	struct regmap *regmap;
+	int mclk_idx;
+	struct nau8822_pll pll;
+	int sysclk;
+	int div_id;
+};
+
+#endif	/* __NAU8822_H__ */
diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c
index 690c26e7389e..809b7e9f03ca 100644
--- a/sound/soc/codecs/pcm186x.c
+++ b/sound/soc/codecs/pcm186x.c
@@ -401,7 +401,8 @@ static int pcm186x_set_fmt(struct snd_soc_dai *dai, unsigned int format)
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 		priv->tdm_offset += 1;
-		/* Fall through... DSP_A uses the same basic config as DSP_B
+		/* fall through */
+		/* DSP_A uses the same basic config as DSP_B
 		 * except we need to shift the TDM output by one BCK cycle
 		 */
 	case SND_SOC_DAIFMT_DSP_B:
diff --git a/sound/soc/codecs/pcm3060-i2c.c b/sound/soc/codecs/pcm3060-i2c.c
new file mode 100644
index 000000000000..cdc8314882bc
--- /dev/null
+++ b/sound/soc/codecs/pcm3060-i2c.c
@@ -0,0 +1,60 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// PCM3060 I2C driver
+//
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <sound/soc.h>
+
+#include "pcm3060.h"
+
+static int pcm3060_i2c_probe(struct i2c_client *i2c,
+			     const struct i2c_device_id *id)
+{
+	struct pcm3060_priv *priv;
+
+	priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, priv);
+
+	priv->regmap = devm_regmap_init_i2c(i2c, &pcm3060_regmap);
+	if (IS_ERR(priv->regmap))
+		return PTR_ERR(priv->regmap);
+
+	return pcm3060_probe(&i2c->dev);
+}
+
+static const struct i2c_device_id pcm3060_i2c_id[] = {
+	{ .name = "pcm3060" },
+	{ },
+};
+MODULE_DEVICE_TABLE(i2c, pcm3060_i2c_id);
+
+#ifdef CONFIG_OF
+static const struct of_device_id pcm3060_of_match[] = {
+	{ .compatible = "ti,pcm3060" },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, pcm3060_of_match);
+#endif /* CONFIG_OF */
+
+static struct i2c_driver pcm3060_i2c_driver = {
+	.driver = {
+		.name = "pcm3060",
+#ifdef CONFIG_OF
+		.of_match_table = pcm3060_of_match,
+#endif /* CONFIG_OF */
+	},
+	.id_table = pcm3060_i2c_id,
+	.probe = pcm3060_i2c_probe,
+};
+
+module_i2c_driver(pcm3060_i2c_driver);
+
+MODULE_DESCRIPTION("PCM3060 I2C driver");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060-spi.c b/sound/soc/codecs/pcm3060-spi.c
new file mode 100644
index 000000000000..f6f19fa80932
--- /dev/null
+++ b/sound/soc/codecs/pcm3060-spi.c
@@ -0,0 +1,59 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// PCM3060 SPI driver
+//
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+
+#include <linux/module.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+
+#include "pcm3060.h"
+
+static int pcm3060_spi_probe(struct spi_device *spi)
+{
+	struct pcm3060_priv *priv;
+
+	priv = devm_kzalloc(&spi->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	spi_set_drvdata(spi, priv);
+
+	priv->regmap = devm_regmap_init_spi(spi, &pcm3060_regmap);
+	if (IS_ERR(priv->regmap))
+		return PTR_ERR(priv->regmap);
+
+	return pcm3060_probe(&spi->dev);
+}
+
+static const struct spi_device_id pcm3060_spi_id[] = {
+	{ .name = "pcm3060" },
+	{ },
+};
+MODULE_DEVICE_TABLE(spi, pcm3060_spi_id);
+
+#ifdef CONFIG_OF
+static const struct of_device_id pcm3060_of_match[] = {
+	{ .compatible = "ti,pcm3060" },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, pcm3060_of_match);
+#endif /* CONFIG_OF */
+
+static struct spi_driver pcm3060_spi_driver = {
+	.driver = {
+		.name = "pcm3060",
+#ifdef CONFIG_OF
+		.of_match_table = pcm3060_of_match,
+#endif /* CONFIG_OF */
+	},
+	.id_table = pcm3060_spi_id,
+	.probe = pcm3060_spi_probe,
+};
+
+module_spi_driver(pcm3060_spi_driver);
+
+MODULE_DESCRIPTION("PCM3060 SPI driver");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c
new file mode 100644
index 000000000000..494d9d662be8
--- /dev/null
+++ b/sound/soc/codecs/pcm3060.c
@@ -0,0 +1,295 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// PCM3060 codec driver
+//
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+
+#include <linux/module.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "pcm3060.h"
+
+/* dai */
+
+static int pcm3060_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+			      unsigned int freq, int dir)
+{
+	struct snd_soc_component *comp = dai->component;
+	struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp);
+
+	if (dir != SND_SOC_CLOCK_IN) {
+		dev_err(comp->dev, "unsupported sysclock dir: %d\n", dir);
+		return -EINVAL;
+	}
+
+	priv->dai[dai->id].sclk_freq = freq;
+
+	return 0;
+}
+
+static int pcm3060_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_component *comp = dai->component;
+	struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp);
+	unsigned int reg;
+	unsigned int val;
+
+	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) {
+		dev_err(comp->dev, "unsupported DAI polarity: 0x%x\n", fmt);
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		priv->dai[dai->id].is_master = true;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		priv->dai[dai->id].is_master = false;
+		break;
+	default:
+		dev_err(comp->dev, "unsupported DAI master mode: 0x%x\n", fmt);
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		val = PCM3060_REG_FMT_I2S;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		val = PCM3060_REG_FMT_RJ;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		val = PCM3060_REG_FMT_LJ;
+		break;
+	default:
+		dev_err(comp->dev, "unsupported DAI format: 0x%x\n", fmt);
+		return -EINVAL;
+	}
+
+	if (dai->id == PCM3060_DAI_ID_DAC)
+		reg = PCM3060_REG67;
+	else
+		reg = PCM3060_REG72;
+
+	regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_FMT, val);
+
+	return 0;
+}
+
+static int pcm3060_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *comp = dai->component;
+	struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp);
+	unsigned int rate;
+	unsigned int ratio;
+	unsigned int reg;
+	unsigned int val;
+
+	if (!priv->dai[dai->id].is_master) {
+		val = PCM3060_REG_MS_S;
+		goto val_ready;
+	}
+
+	rate = params_rate(params);
+	if (!rate) {
+		dev_err(comp->dev, "rate is not configured\n");
+		return -EINVAL;
+	}
+
+	ratio = priv->dai[dai->id].sclk_freq / rate;
+
+	switch (ratio) {
+	case 768:
+		val = PCM3060_REG_MS_M768;
+		break;
+	case 512:
+		val = PCM3060_REG_MS_M512;
+		break;
+	case 384:
+		val = PCM3060_REG_MS_M384;
+		break;
+	case 256:
+		val = PCM3060_REG_MS_M256;
+		break;
+	case 192:
+		val = PCM3060_REG_MS_M192;
+		break;
+	case 128:
+		val = PCM3060_REG_MS_M128;
+		break;
+	default:
+		dev_err(comp->dev, "unsupported ratio: %d\n", ratio);
+		return -EINVAL;
+	}
+
+val_ready:
+	if (dai->id == PCM3060_DAI_ID_DAC)
+		reg = PCM3060_REG67;
+	else
+		reg = PCM3060_REG72;
+
+	regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_MS, val);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops pcm3060_dai_ops = {
+	.set_sysclk = pcm3060_set_sysclk,
+	.set_fmt = pcm3060_set_fmt,
+	.hw_params = pcm3060_hw_params,
+};
+
+#define PCM3060_DAI_RATES_ADC	(SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 | \
+				 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \
+				 SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define PCM3060_DAI_RATES_DAC	(PCM3060_DAI_RATES_ADC | \
+				 SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
+static struct snd_soc_dai_driver pcm3060_dai[] = {
+	{
+		.name = "pcm3060-dac",
+		.id = PCM3060_DAI_ID_DAC,
+		.playback = {
+			.stream_name = "Playback",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = PCM3060_DAI_RATES_DAC,
+			.formats = SNDRV_PCM_FMTBIT_S24_LE,
+		},
+		.ops = &pcm3060_dai_ops,
+	},
+	{
+		.name = "pcm3060-adc",
+		.id = PCM3060_DAI_ID_ADC,
+		.capture = {
+			.stream_name = "Capture",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = PCM3060_DAI_RATES_ADC,
+			.formats = SNDRV_PCM_FMTBIT_S24_LE,
+		},
+		.ops = &pcm3060_dai_ops,
+	},
+};
+
+/* dapm */
+
+static DECLARE_TLV_DB_SCALE(pcm3060_dapm_tlv, -10050, 50, 1);
+
+static const struct snd_kcontrol_new pcm3060_dapm_controls[] = {
+	SOC_DOUBLE_R_RANGE_TLV("Master Playback Volume",
+			       PCM3060_REG65, PCM3060_REG66, 0,
+			       PCM3060_REG_AT2_MIN, PCM3060_REG_AT2_MAX,
+			       0, pcm3060_dapm_tlv),
+	SOC_DOUBLE("Master Playback Switch", PCM3060_REG68,
+		   PCM3060_REG_SHIFT_MUT21, PCM3060_REG_SHIFT_MUT22, 1, 1),
+
+	SOC_DOUBLE_R_RANGE_TLV("Master Capture Volume",
+			       PCM3060_REG70, PCM3060_REG71, 0,
+			       PCM3060_REG_AT1_MIN, PCM3060_REG_AT1_MAX,
+			       0, pcm3060_dapm_tlv),
+	SOC_DOUBLE("Master Capture Switch", PCM3060_REG73,
+		   PCM3060_REG_SHIFT_MUT11, PCM3060_REG_SHIFT_MUT12, 1, 1),
+};
+
+static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = {
+	SND_SOC_DAPM_OUTPUT("OUTL+"),
+	SND_SOC_DAPM_OUTPUT("OUTR+"),
+	SND_SOC_DAPM_OUTPUT("OUTL-"),
+	SND_SOC_DAPM_OUTPUT("OUTR-"),
+
+	SND_SOC_DAPM_INPUT("INL"),
+	SND_SOC_DAPM_INPUT("INR"),
+};
+
+static const struct snd_soc_dapm_route pcm3060_dapm_map[] = {
+	{ "OUTL+", NULL, "Playback" },
+	{ "OUTR+", NULL, "Playback" },
+	{ "OUTL-", NULL, "Playback" },
+	{ "OUTR-", NULL, "Playback" },
+
+	{ "Capture", NULL, "INL" },
+	{ "Capture", NULL, "INR" },
+};
+
+/* soc component */
+
+static const struct snd_soc_component_driver pcm3060_soc_comp_driver = {
+	.controls = pcm3060_dapm_controls,
+	.num_controls = ARRAY_SIZE(pcm3060_dapm_controls),
+	.dapm_widgets = pcm3060_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(pcm3060_dapm_widgets),
+	.dapm_routes = pcm3060_dapm_map,
+	.num_dapm_routes = ARRAY_SIZE(pcm3060_dapm_map),
+};
+
+/* regmap */
+
+static bool pcm3060_reg_writeable(struct device *dev, unsigned int reg)
+{
+	return (reg >= PCM3060_REG64);
+}
+
+static bool pcm3060_reg_readable(struct device *dev, unsigned int reg)
+{
+	return (reg >= PCM3060_REG64);
+}
+
+static bool pcm3060_reg_volatile(struct device *dev, unsigned int reg)
+{
+	/* PCM3060_REG64 is volatile */
+	return (reg == PCM3060_REG64);
+}
+
+static const struct reg_default pcm3060_reg_defaults[] = {
+	{ PCM3060_REG64,  0xF0 },
+	{ PCM3060_REG65,  0xFF },
+	{ PCM3060_REG66,  0xFF },
+	{ PCM3060_REG67,  0x00 },
+	{ PCM3060_REG68,  0x00 },
+	{ PCM3060_REG69,  0x00 },
+	{ PCM3060_REG70,  0xD7 },
+	{ PCM3060_REG71,  0xD7 },
+	{ PCM3060_REG72,  0x00 },
+	{ PCM3060_REG73,  0x00 },
+};
+
+const struct regmap_config pcm3060_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+	.writeable_reg = pcm3060_reg_writeable,
+	.readable_reg = pcm3060_reg_readable,
+	.volatile_reg = pcm3060_reg_volatile,
+	.max_register = PCM3060_REG73,
+	.reg_defaults = pcm3060_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(pcm3060_reg_defaults),
+	.cache_type = REGCACHE_RBTREE,
+};
+EXPORT_SYMBOL(pcm3060_regmap);
+
+/* device */
+
+int pcm3060_probe(struct device *dev)
+{
+	int rc;
+
+	rc = devm_snd_soc_register_component(dev, &pcm3060_soc_comp_driver,
+					     pcm3060_dai,
+					     ARRAY_SIZE(pcm3060_dai));
+	if (rc) {
+		dev_err(dev, "failed to register component, rc=%d\n", rc);
+		return rc;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL(pcm3060_probe);
+
+MODULE_DESCRIPTION("PCM3060 codec driver");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h
new file mode 100644
index 000000000000..fd89a68aa8a7
--- /dev/null
+++ b/sound/soc/codecs/pcm3060.h
@@ -0,0 +1,88 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * PCM3060 codec driver
+ *
+ * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+ */
+
+#ifndef _SND_SOC_PCM3060_H
+#define _SND_SOC_PCM3060_H
+
+#include <linux/device.h>
+#include <linux/regmap.h>
+
+extern const struct regmap_config pcm3060_regmap;
+
+#define PCM3060_DAI_ID_DAC	0
+#define PCM3060_DAI_ID_ADC	1
+#define PCM3060_DAI_IDS_NUM	2
+
+struct pcm3060_priv_dai {
+	bool is_master;
+	unsigned int sclk_freq;
+};
+
+struct pcm3060_priv {
+	struct regmap *regmap;
+	struct pcm3060_priv_dai dai[PCM3060_DAI_IDS_NUM];
+};
+
+int pcm3060_probe(struct device *dev);
+int pcm3060_remove(struct device *dev);
+
+/* registers */
+
+#define PCM3060_REG64			0x40
+#define PCM3060_REG_MRST		0x80
+#define PCM3060_REG_SRST		0x40
+#define PCM3060_REG_ADPSV		0x20
+#define PCM3060_REG_DAPSV		0x10
+#define PCM3060_REG_SE			0x01
+
+#define PCM3060_REG65			0x41
+#define PCM3060_REG66			0x42
+#define PCM3060_REG_AT2_MIN		0x36
+#define PCM3060_REG_AT2_MAX		0xFF
+
+#define PCM3060_REG67			0x43
+#define PCM3060_REG72			0x48
+#define PCM3060_REG_CSEL		0x80
+#define PCM3060_REG_MASK_MS		0x70
+#define PCM3060_REG_MS_S		0x00
+#define PCM3060_REG_MS_M768		(0x01 << 4)
+#define PCM3060_REG_MS_M512		(0x02 << 4)
+#define PCM3060_REG_MS_M384		(0x03 << 4)
+#define PCM3060_REG_MS_M256		(0x04 << 4)
+#define PCM3060_REG_MS_M192		(0x05 << 4)
+#define PCM3060_REG_MS_M128		(0x06 << 4)
+#define PCM3060_REG_MASK_FMT		0x03
+#define PCM3060_REG_FMT_I2S		0x00
+#define PCM3060_REG_FMT_LJ		0x01
+#define PCM3060_REG_FMT_RJ		0x02
+
+#define PCM3060_REG68			0x44
+#define PCM3060_REG_OVER		0x40
+#define PCM3060_REG_DREV2		0x04
+#define PCM3060_REG_SHIFT_MUT21	0x00
+#define PCM3060_REG_SHIFT_MUT22	0x01
+
+#define PCM3060_REG69			0x45
+#define PCM3060_REG_FLT		0x80
+#define PCM3060_REG_MASK_DMF		0x60
+#define PCM3060_REG_DMC		0x10
+#define PCM3060_REG_ZREV		0x02
+#define PCM3060_REG_AZRO		0x01
+
+#define PCM3060_REG70			0x46
+#define PCM3060_REG71			0x47
+#define PCM3060_REG_AT1_MIN		0x0E
+#define PCM3060_REG_AT1_MAX		0xFF
+
+#define PCM3060_REG73			0x49
+#define PCM3060_REG_ZCDD		0x10
+#define PCM3060_REG_BYP		0x08
+#define PCM3060_REG_DREV1		0x04
+#define PCM3060_REG_SHIFT_MUT11	0x00
+#define PCM3060_REG_SHIFT_MUT12	0x01
+
+#endif /* _SND_SOC_PCM3060_H */
diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 3356c91f55b0..52cc950c9fd1 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -33,6 +33,8 @@
 #define PCM3168A_FMT_RIGHT_J_16		0x3
 #define PCM3168A_FMT_DSP_A		0x4
 #define PCM3168A_FMT_DSP_B		0x5
+#define PCM3168A_FMT_I2S_TDM		0x6
+#define PCM3168A_FMT_LEFT_J_TDM		0x7
 #define PCM3168A_FMT_DSP_MASK		0x4
 
 #define PCM3168A_NUM_SUPPLIES 6
@@ -401,9 +403,11 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 	bool tx, master_mode;
 	u32 val, mask, shift, reg;
 	unsigned int rate, fmt, ratio, max_ratio;
+	unsigned int chan;
 	int i, min_frame_size;
 
 	rate = params_rate(params);
+	chan = params_channels(params);
 
 	ratio = pcm3168a->sysclk / rate;
 
@@ -456,6 +460,21 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
+	/* for TDM */
+	if (chan > 2) {
+		switch (fmt) {
+		case PCM3168A_FMT_I2S:
+			fmt = PCM3168A_FMT_I2S_TDM;
+			break;
+		case PCM3168A_FMT_LEFT_J:
+			fmt = PCM3168A_FMT_LEFT_J_TDM;
+			break;
+		default:
+			dev_err(component->dev, "TDM is supported under I2S/Left_J only\n");
+			return -EINVAL;
+		}
+	}
+
 	if (master_mode)
 		val = ((i + 1) << shift);
 	else
@@ -476,7 +495,69 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int pcm3168a_startup(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	unsigned int fmt;
+	unsigned int sample_min;
+	unsigned int channel_max;
+
+	if (tx)
+		fmt = pcm3168a->dac_fmt;
+	else
+		fmt = pcm3168a->adc_fmt;
+
+	/*
+	 * Available Data Bits
+	 *
+	 * RIGHT_J : 24 / 16
+	 * LEFT_J  : 24
+	 * I2S     : 24
+	 *
+	 * TDM available
+	 *
+	 * I2S
+	 * LEFT_J
+	 */
+	switch (fmt) {
+	case PCM3168A_FMT_RIGHT_J:
+		sample_min  = 16;
+		channel_max =  2;
+		break;
+	case PCM3168A_FMT_LEFT_J:
+		sample_min  = 24;
+		if (tx)
+			channel_max = 8;
+		else
+			channel_max = 6;
+		break;
+	case PCM3168A_FMT_I2S:
+		sample_min  = 24;
+		if (tx)
+			channel_max = 8;
+		else
+			channel_max = 6;
+		break;
+	default:
+		sample_min  = 24;
+		channel_max =  2;
+	}
+
+	snd_pcm_hw_constraint_minmax(substream->runtime,
+				     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+				     sample_min, 32);
+
+	snd_pcm_hw_constraint_minmax(substream->runtime,
+				     SNDRV_PCM_HW_PARAM_CHANNELS,
+				     2, channel_max);
+
+	return 0;
+}
 static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = {
+	.startup	= pcm3168a_startup,
 	.set_fmt	= pcm3168a_set_dai_fmt_dac,
 	.set_sysclk	= pcm3168a_set_dai_sysclk,
 	.hw_params	= pcm3168a_hw_params,
@@ -484,6 +565,7 @@ static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = {
 };
 
 static const struct snd_soc_dai_ops pcm3168a_adc_dai_ops = {
+	.startup	= pcm3168a_startup,
 	.set_fmt	= pcm3168a_set_dai_fmt_adc,
 	.set_sysclk	= pcm3168a_set_dai_sysclk,
 	.hw_params	= pcm3168a_hw_params
diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c
index d88e67341083..0ef966d56bac 100644
--- a/sound/soc/codecs/rt274.c
+++ b/sound/soc/codecs/rt274.c
@@ -755,6 +755,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
 		break;
 	default:
 		dev_warn(component->dev, "invalid pll source, use BCLK\n");
+		/* fall through */
 	case RT274_PLL2_S_BCLK:
 		snd_soc_component_update_bits(component, RT274_PLL2_CTRL,
 				RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_BCLK);
@@ -782,6 +783,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
 			break;
 		default:
 			dev_warn(component->dev, "invalid freq_in, assume 4.8M\n");
+			/* fall through */
 		case 100:
 			snd_soc_component_write(component, 0x7a, 0xaab6);
 			snd_soc_component_write(component, 0x7b, 0x0301);
diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c
index 6478d10c4f4a..4d46f4567c3a 100644
--- a/sound/soc/codecs/rt5514-spi.c
+++ b/sound/soc/codecs/rt5514-spi.c
@@ -91,6 +91,14 @@ static void rt5514_spi_copy_work(struct work_struct *work)
 
 	runtime = rt5514_dsp->substream->runtime;
 	period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream);
+	if (!period_bytes) {
+		schedule_delayed_work(&rt5514_dsp->copy_work, 5);
+		goto done;
+	}
+
+	if (rt5514_dsp->buf_size % period_bytes)
+		rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) *
+			period_bytes;
 
 	if (rt5514_dsp->get_size >= rt5514_dsp->buf_size) {
 		rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf,
@@ -149,13 +157,11 @@ done:
 
 static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp)
 {
-	size_t period_bytes;
 	u8 buf[8];
 
 	if (!rt5514_dsp->substream)
 		return;
 
-	period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream);
 	rt5514_dsp->get_size = 0;
 
 	/**
@@ -183,10 +189,6 @@ static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp)
 
 	rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base;
 
-	if (rt5514_dsp->buf_size % period_bytes)
-		rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) *
-			period_bytes;
-
 	if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit &&
 		rt5514_dsp->buf_rp && rt5514_dsp->buf_size)
 		schedule_delayed_work(&rt5514_dsp->copy_work, 0);
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index 5bcedbc7eb4a..b7ba64350a07 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -10,7 +10,6 @@
  */
 
 #include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/init.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index 70441661ea4a..7eb2cbd39d6e 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -72,6 +72,7 @@ struct rt5663_priv {
 static const struct reg_sequence rt5663_patch_list[] = {
 	{ 0x002a, 0x8020 },
 	{ 0x0086, 0x0028 },
+	{ 0x0100, 0xa020 },
 	{ 0x0117, 0x0f28 },
 	{ 0x02fb, 0x8089 },
 };
@@ -580,7 +581,7 @@ static const struct reg_default rt5663_reg[] = {
 	{ 0x00fd, 0x0001 },
 	{ 0x00fe, 0x10ec },
 	{ 0x00ff, 0x6406 },
-	{ 0x0100, 0xa0a0 },
+	{ 0x0100, 0xa020 },
 	{ 0x0108, 0x4444 },
 	{ 0x0109, 0x4444 },
 	{ 0x010a, 0xaaaa },
@@ -2337,6 +2338,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w,
 				0x8000);
 			snd_soc_component_update_bits(component, RT5663_DEPOP_1, 0x3000,
 				0x3000);
+			snd_soc_component_update_bits(component,
+				RT5663_DIG_VOL_ZCD, 0x00c0, 0x0080);
 		}
 		break;
 
@@ -2351,6 +2354,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w,
 				RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN);
 			snd_soc_component_update_bits(component,
 				RT5663_DACREF_LDO, 0x3e0e, 0);
+			snd_soc_component_update_bits(component,
+				RT5663_DIG_VOL_ZCD, 0x00c0, 0);
 		}
 		break;
 
diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c
index 3f6046a66b56..230a21c93b6b 100644
--- a/sound/soc/codecs/rt5668.c
+++ b/sound/soc/codecs/rt5668.c
@@ -2588,17 +2588,10 @@ static int rt5668_i2c_probe(struct i2c_client *i2c,
 
 	}
 
-	return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668,
+	return devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668,
 			rt5668_dai, ARRAY_SIZE(rt5668_dai));
 }
 
-static int rt5668_i2c_remove(struct i2c_client *i2c)
-{
-	snd_soc_unregister_component(&i2c->dev);
-
-	return 0;
-}
-
 static void rt5668_i2c_shutdown(struct i2c_client *client)
 {
 	struct rt5668_priv *rt5668 = i2c_get_clientdata(client);
@@ -2629,7 +2622,6 @@ static struct i2c_driver rt5668_i2c_driver = {
 		.acpi_match_table = ACPI_PTR(rt5668_acpi_match),
 	},
 	.probe = rt5668_i2c_probe,
-	.remove = rt5668_i2c_remove,
 	.shutdown = rt5668_i2c_shutdown,
 	.id_table = rt5668_i2c_id,
 };
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index f0f8debc2829..453328c988c0 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -2878,6 +2878,18 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = {
 	},
 	{
 		.callback = rt5670_quirk_cb,
+		.ident = "Lenovo Thinkpad Tablet 8",
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 8"),
+		},
+		.driver_data = (unsigned long *)(RT5670_DMIC_EN |
+						 RT5670_DMIC2_INR |
+						 RT5670_DEV_GPIO |
+						 RT5670_JD_MODE1),
+	},
+	{
+		.callback = rt5670_quirk_cb,
 		.ident = "Lenovo Thinkpad Tablet 10",
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c
index bd51f3655ee3..84501c2020c7 100644
--- a/sound/soc/codecs/rt5677-spi.c
+++ b/sound/soc/codecs/rt5677-spi.c
@@ -18,7 +18,6 @@
 #include <linux/interrupt.h>
 #include <linux/irq.h>
 #include <linux/slab.h>
-#include <linux/gpio.h>
 #include <linux/sched.h>
 #include <linux/uaccess.h>
 #include <linux/regulator/consumer.h>
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index 8a3052874c29..34cfaf8f6f34 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -67,7 +67,8 @@ struct rt5682_priv {
 };
 
 static const struct reg_sequence patch_list[] = {
-	{0x01c1, 0x1000},
+	{RT5682_HP_IMP_SENS_CTRL_19, 0x1000},
+	{RT5682_DAC_ADC_DIG_VOL1, 0xa020},
 };
 
 static const struct reg_default rt5682_reg[] = {
@@ -749,7 +750,6 @@ static bool rt5682_readable_register(struct device *dev, unsigned int reg)
 	}
 }
 
-static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0);
 static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0);
 static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0);
 static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
@@ -1108,10 +1108,6 @@ static void rt5682_jack_detect_handler(struct work_struct *work)
 }
 
 static const struct snd_kcontrol_new rt5682_snd_controls[] = {
-	/* Headphone Output Volume */
-	SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5682_HPL_GAIN,
-		RT5682_HPR_GAIN, RT5682_G_HP_SFT, 15, 1, hp_vol_tlv),
-
 	/* DAC Digital Volume */
 	SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL,
 		RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv),
@@ -1437,6 +1433,28 @@ static const struct snd_kcontrol_new hpor_switch =
 	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1,
 					RT5682_R_MUTE_SFT, 1, 1);
 
+static int rt5682_charge_pump_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		snd_soc_component_update_bits(component,
+			RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_HV);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		snd_soc_component_update_bits(component,
+			RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_LV);
+		break;
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
 static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -1449,10 +1467,10 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
 			RT5682_HP_LOGIC_CTRL_2, 0x0012);
 		snd_soc_component_write(component,
 			RT5682_HP_CTRL_2, 0x6000);
-		snd_soc_component_update_bits(component, RT5682_STO_NG2_CTRL_1,
-			RT5682_NG2_EN_MASK, RT5682_NG2_EN);
 		snd_soc_component_update_bits(component,
 			RT5682_DEPOP_1, 0x60, 0x60);
+		snd_soc_component_update_bits(component,
+			RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080);
 		break;
 
 	case SND_SOC_DAPM_POST_PMD:
@@ -1460,6 +1478,8 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
 			RT5682_DEPOP_1, 0x60, 0x0);
 		snd_soc_component_write(component,
 			RT5682_HP_CTRL_2, 0x0000);
+		snd_soc_component_update_bits(component,
+			RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0000);
 		break;
 
 	default:
@@ -1723,7 +1743,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1,
 		RT5682_PWR_HA_R_BIT, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1,
-		RT5682_PUMP_EN_SFT, 0, NULL, 0),
+		RT5682_PUMP_EN_SFT, 0, rt5682_charge_pump_event,
+		SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1,
 		RT5682_CAPLESS_EN_SFT, 0, NULL, 0),
 
@@ -1884,6 +1905,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = {
 	{"HP Amp", NULL, "Charge Pump"},
 	{"HP Amp", NULL, "CLKDET SYS"},
 	{"HP Amp", NULL, "CBJ Power"},
+	{"HP Amp", NULL, "Vref1"},
 	{"HP Amp", NULL, "Vref2"},
 	{"HPOL Playback", "Switch", "HP Amp"},
 	{"HPOR Playback", "Switch", "HP Amp"},
@@ -2452,30 +2474,23 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
 	mutex_lock(&rt5682->calibrate_mutex);
 
 	rt5682_reset(rt5682->regmap);
-	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf);
+	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af);
 	usleep_range(15000, 20000);
-	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf);
-	regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380);
-	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8001);
-	regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000);
-	regmap_write(rt5682->regmap, RT5682_STO1_DAC_MIXER, 0x2080);
-	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x4040);
-	regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0069);
+	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af);
+	regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0300);
+	regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x8000);
+	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100);
+	regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800);
 	regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000);
-	regmap_write(rt5682->regmap, RT5682_HP_CTRL_2, 0x6000);
-	regmap_write(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, 0x0f26);
-	regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7f05);
+	regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005);
 	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c);
 	regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d);
-	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_9, 0x000f);
-	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8d01);
 	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321);
 	regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004);
 	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00);
 	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1);
 	regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311);
-	regmap_write(rt5682->regmap, RT5682_RESET_HPF_CTRL, 0x0000);
-	regmap_write(rt5682->regmap, RT5682_ADC_STO1_HP_CTRL_1, 0x3320);
+	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00);
 
 	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00);
 
@@ -2491,8 +2506,12 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
 		pr_err("HP Calibration Failure\n");
 
 	/* restore settings */
-	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4);
+	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0x02af);
+	regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0080);
+	regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000);
 	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000);
+	regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000);
+	regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005);
 
 	mutex_unlock(&rt5682->calibrate_mutex);
 
@@ -2566,7 +2585,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
 
 	rt5682_calibrate(rt5682);
 
-	ret = regmap_register_patch(rt5682->regmap, patch_list,
+	ret = regmap_multi_reg_write(rt5682->regmap, patch_list,
 				    ARRAY_SIZE(patch_list));
 	if (ret != 0)
 		dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret);
@@ -2620,6 +2639,10 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
 			RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK,
 			RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1);
 	regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000);
+	regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8,
+			RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA);
+	regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1,
+			RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ);
 
 	INIT_DELAYED_WORK(&rt5682->jack_detect_work,
 				rt5682_jack_detect_handler);
@@ -2637,11 +2660,17 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
 
 	}
 
-	return devm_snd_soc_register_component(&i2c->dev,
-			&soc_component_dev_rt5682,
+	return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5682,
 			rt5682_dai, ARRAY_SIZE(rt5682_dai));
 }
 
+static int rt5682_i2c_remove(struct i2c_client *i2c)
+{
+	snd_soc_unregister_component(&i2c->dev);
+
+	return 0;
+}
+
 static void rt5682_i2c_shutdown(struct i2c_client *client)
 {
 	struct rt5682_priv *rt5682 = i2c_get_clientdata(client);
@@ -2672,6 +2701,7 @@ static struct i2c_driver rt5682_i2c_driver = {
 		.acpi_match_table = ACPI_PTR(rt5682_acpi_match),
 	},
 	.probe = rt5682_i2c_probe,
+	.remove = rt5682_i2c_remove,
 	.shutdown = rt5682_i2c_shutdown,
 	.id_table = rt5682_i2c_id,
 };
diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h
index 8068140ebe3f..d82a8301fd74 100644
--- a/sound/soc/codecs/rt5682.h
+++ b/sound/soc/codecs/rt5682.h
@@ -1214,6 +1214,20 @@
 #define RT5682_JDH_NO_PLUG			(0x1 << 4)
 #define RT5682_JDH_PLUG				(0x0 << 4)
 
+/* Bias current control 8 (0x0111) */
+#define RT5682_HPA_CP_BIAS_CTRL_MASK			(0x3 << 2)
+#define RT5682_HPA_CP_BIAS_2UA			(0x0 << 2)
+#define RT5682_HPA_CP_BIAS_3UA			(0x1 << 2)
+#define RT5682_HPA_CP_BIAS_4UA			(0x2 << 2)
+#define RT5682_HPA_CP_BIAS_6UA			(0x3 << 2)
+
+/* Charge Pump Internal Register1 (0x0125) */
+#define RT5682_CP_CLK_HP_MASK			(0x3 << 4)
+#define RT5682_CP_CLK_HP_100KHZ			(0x0 << 4)
+#define RT5682_CP_CLK_HP_200KHZ			(0x1 << 4)
+#define RT5682_CP_CLK_HP_300KHZ			(0x2 << 4)
+#define RT5682_CP_CLK_HP_600KHZ			(0x3 << 4)
+
 /* Chopper and Clock control for DAC (0x013a)*/
 #define RT5682_CKXEN_DAC1_MASK			(0x1 << 13)
 #define RT5682_CKXEN_DAC1_SFT			13
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 60764f6201b1..add18d6d77da 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1218,7 +1218,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_component *component)
 	 * Searching for a suitable index solving this formula:
 	 * idx = 40 * log10(vag_val / lo_cagcntrl) + 15
 	 */
-	vol_quot = (vag * 100) / lo_vag;
+	vol_quot = lo_vag ? (vag * 100) / lo_vag : 0;
 	lo_vol = 0;
 	for (i = 0; i < ARRAY_SIZE(vol_quot_table); i++) {
 		if (vol_quot >= vol_quot_table[i])
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index d5035f2f2b2b..f753d2db0a5a 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -21,6 +21,7 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/init.h>
+#include <linux/clk.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
@@ -142,6 +143,7 @@ static const char *sta32x_supply_names[] = {
 /* codec private data */
 struct sta32x_priv {
 	struct regmap *regmap;
+	struct clk *xti_clk;
 	struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)];
 	struct snd_soc_component *component;
 	struct sta32x_platform_data *pdata;
@@ -879,6 +881,18 @@ static int sta32x_probe(struct snd_soc_component *component)
 	struct sta32x_priv *sta32x = snd_soc_component_get_drvdata(component);
 	struct sta32x_platform_data *pdata = sta32x->pdata;
 	int i, ret = 0, thermal = 0;
+
+	sta32x->component = component;
+
+	if (sta32x->xti_clk) {
+		ret = clk_prepare_enable(sta32x->xti_clk);
+		if (ret != 0) {
+			dev_err(component->dev,
+				"Failed to enable clock: %d\n", ret);
+			return ret;
+		}
+	}
+
 	ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies),
 				    sta32x->supplies);
 	if (ret != 0) {
@@ -981,6 +995,9 @@ static void sta32x_remove(struct snd_soc_component *component)
 
 	sta32x_watchdog_stop(sta32x);
 	regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+
+	if (sta32x->xti_clk)
+		clk_disable_unprepare(sta32x->xti_clk);
 }
 
 static const struct snd_soc_component_driver sta32x_component = {
@@ -1038,6 +1055,8 @@ static int sta32x_probe_dt(struct device *dev, struct sta32x_priv *sta32x)
 	of_property_read_u8(np, "st,ch3-output-mapping",
 			    &pdata->ch3_output_mapping);
 
+	if (of_get_property(np, "st,fault-detect-recovery", NULL))
+		pdata->fault_detect_recovery = 1;
 	if (of_get_property(np, "st,thermal-warning-recovery", NULL))
 		pdata->thermal_warning_recovery = 1;
 	if (of_get_property(np, "st,thermal-warning-adjustment", NULL))
@@ -1095,6 +1114,17 @@ static int sta32x_i2c_probe(struct i2c_client *i2c,
 	}
 #endif
 
+	/* Clock */
+	sta32x->xti_clk = devm_clk_get(dev, "xti");
+	if (IS_ERR(sta32x->xti_clk)) {
+		ret = PTR_ERR(sta32x->xti_clk);
+
+		if (ret == -EPROBE_DEFER)
+			return ret;
+
+		sta32x->xti_clk = NULL;
+	}
+
 	/* GPIOs */
 	sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset",
 						       GPIOD_OUT_LOW);
diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c
index ae3d032ac35a..6bd0e5d5347f 100644
--- a/sound/soc/codecs/tas5720.c
+++ b/sound/soc/codecs/tas5720.c
@@ -152,6 +152,7 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai,
 				    int slots, int slot_width)
 {
 	struct snd_soc_component *component = dai->component;
+	struct tas5720_data *tas5720 = snd_soc_component_get_drvdata(component);
 	unsigned int first_slot;
 	int ret;
 
@@ -185,6 +186,20 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai,
 	if (ret < 0)
 		goto error_snd_soc_component_update_bits;
 
+	/* Configure TDM slot width. This is only applicable to TAS5722. */
+	switch (tas5720->devtype) {
+	case TAS5722:
+		ret = snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG,
+						    TAS5722_TDM_SLOT_16B,
+						    slot_width == 16 ?
+						    TAS5722_TDM_SLOT_16B : 0);
+		if (ret < 0)
+			goto error_snd_soc_component_update_bits;
+		break;
+	default:
+		break;
+	}
+
 	return 0;
 
 error_snd_soc_component_update_bits:
@@ -485,15 +500,56 @@ static const DECLARE_TLV_DB_RANGE(dac_analog_tlv,
 );
 
 /*
- * 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.
+ * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB or 0.25 dB steps
+ * depending on the device. Note that setting the gain below -100 dB
+ * (register value <0x7) is effectively a MUTE as per device datasheet.
+ *
+ * Note that for the TAS5722 the digital volume controls are actually split
+ * over two registers, so we need custom getters/setters for access.
  */
-static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas5720_dac_tlv, -10350, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas5722_dac_tlv, -10350, 25, 0);
+
+static int tas5722_volume_get(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+	unsigned int val;
+
+	snd_soc_component_read(component, TAS5720_VOLUME_CTRL_REG, &val);
+	ucontrol->value.integer.value[0] = val << 1;
+
+	snd_soc_component_read(component, TAS5722_DIGITAL_CTRL2_REG, &val);
+	ucontrol->value.integer.value[0] |= val & TAS5722_VOL_CONTROL_LSB;
+
+	return 0;
+}
+
+static int tas5722_volume_set(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+	unsigned int sel = ucontrol->value.integer.value[0];
+
+	snd_soc_component_write(component, TAS5720_VOLUME_CTRL_REG, sel >> 1);
+	snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG,
+				      TAS5722_VOL_CONTROL_LSB, sel);
+
+	return 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),
+		       TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, tas5720_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_kcontrol_new tas5722_snd_controls[] = {
+	SOC_SINGLE_EXT_TLV("Speaker Driver Playback Volume",
+			   0, 0, 511, 0,
+			   tas5722_volume_get, tas5722_volume_set,
+			   tas5722_dac_tlv),
 	SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG,
 		       TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv),
 };
@@ -527,6 +583,23 @@ static const struct snd_soc_component_driver soc_component_dev_tas5720 = {
 	.non_legacy_dai_naming	= 1,
 };
 
+static const struct snd_soc_component_driver soc_component_dev_tas5722 = {
+	.probe = tas5720_codec_probe,
+	.remove = tas5720_codec_remove,
+	.suspend = tas5720_suspend,
+	.resume = tas5720_resume,
+	.controls = tas5722_snd_controls,
+	.num_controls = ARRAY_SIZE(tas5722_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),
+	.idle_bias_on		= 1,
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
 /* 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)
@@ -613,9 +686,23 @@ static int tas5720_probe(struct i2c_client *client,
 
 	dev_set_drvdata(dev, data);
 
-	ret = devm_snd_soc_register_component(&client->dev,
-				     &soc_component_dev_tas5720,
-				     tas5720_dai, ARRAY_SIZE(tas5720_dai));
+	switch (id->driver_data) {
+	case TAS5720:
+		ret = devm_snd_soc_register_component(&client->dev,
+					&soc_component_dev_tas5720,
+					tas5720_dai,
+					ARRAY_SIZE(tas5720_dai));
+		break;
+	case TAS5722:
+		ret = devm_snd_soc_register_component(&client->dev,
+					&soc_component_dev_tas5722,
+					tas5720_dai,
+					ARRAY_SIZE(tas5720_dai));
+		break;
+	default:
+		dev_err(dev, "unexpected private driver data\n");
+		return -EINVAL;
+	}
 	if (ret < 0) {
 		dev_err(dev, "failed to register component: %d\n", ret);
 		return ret;
diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c
index 0d6145549a98..36aebdb8f55c 100644
--- a/sound/soc/codecs/tas6424.c
+++ b/sound/soc/codecs/tas6424.c
@@ -41,6 +41,7 @@ struct tas6424_data {
 	struct regmap *regmap;
 	struct regulator_bulk_data supplies[TAS6424_NUM_SUPPLIES];
 	struct delayed_work fault_check_work;
+	unsigned int last_cfault;
 	unsigned int last_fault1;
 	unsigned int last_fault2;
 	unsigned int last_warn;
@@ -406,9 +407,54 @@ static void tas6424_fault_check_work(struct work_struct *work)
 	unsigned int reg;
 	int ret;
 
+	ret = regmap_read(tas6424->regmap, TAS6424_CHANNEL_FAULT, &reg);
+	if (ret < 0) {
+		dev_err(dev, "failed to read CHANNEL_FAULT register: %d\n", ret);
+		goto out;
+	}
+
+	if (!reg) {
+		tas6424->last_cfault = reg;
+		goto check_global_fault1_reg;
+	}
+
+	/*
+	 * Only flag errors once for a given occurrence. This is needed as
+	 * the TAS6424 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 ((reg & TAS6424_FAULT_OC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH1))
+		dev_crit(dev, "experienced a channel 1 overcurrent fault\n");
+
+	if ((reg & TAS6424_FAULT_OC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH2))
+		dev_crit(dev, "experienced a channel 2 overcurrent fault\n");
+
+	if ((reg & TAS6424_FAULT_OC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH3))
+		dev_crit(dev, "experienced a channel 3 overcurrent fault\n");
+
+	if ((reg & TAS6424_FAULT_OC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH4))
+		dev_crit(dev, "experienced a channel 4 overcurrent fault\n");
+
+	if ((reg & TAS6424_FAULT_DC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH1))
+		dev_crit(dev, "experienced a channel 1 DC fault\n");
+
+	if ((reg & TAS6424_FAULT_DC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH2))
+		dev_crit(dev, "experienced a channel 2 DC fault\n");
+
+	if ((reg & TAS6424_FAULT_DC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH3))
+		dev_crit(dev, "experienced a channel 3 DC fault\n");
+
+	if ((reg & TAS6424_FAULT_DC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH4))
+		dev_crit(dev, "experienced a channel 4 DC fault\n");
+
+	/* Store current fault1 value so we can detect any changes next time */
+	tas6424->last_cfault = reg;
+
+check_global_fault1_reg:
 	ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT1, &reg);
 	if (ret < 0) {
-		dev_err(dev, "failed to read FAULT1 register: %d\n", ret);
+		dev_err(dev, "failed to read GLOB_FAULT1 register: %d\n", ret);
 		goto out;
 	}
 
@@ -429,12 +475,6 @@ static void tas6424_fault_check_work(struct work_struct *work)
 		goto check_global_fault2_reg;
 	}
 
-	/*
-	 * Only flag errors once for a given occurrence. This is needed as
-	 * the TAS6424 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 ((reg & TAS6424_FAULT_PVDD_OV) && !(tas6424->last_fault1 & TAS6424_FAULT_PVDD_OV))
 		dev_crit(dev, "experienced a PVDD overvoltage fault\n");
 
@@ -453,7 +493,7 @@ static void tas6424_fault_check_work(struct work_struct *work)
 check_global_fault2_reg:
 	ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT2, &reg);
 	if (ret < 0) {
-		dev_err(dev, "failed to read FAULT2 register: %d\n", ret);
+		dev_err(dev, "failed to read GLOB_FAULT2 register: %d\n", ret);
 		goto out;
 	}
 
@@ -530,7 +570,7 @@ check_warn_reg:
 	/* Store current warn value so we can detect any changes next time */
 	tas6424->last_warn = reg;
 
-	/* Clear any faults by toggling the CLEAR_FAULT control bit */
+	/* Clear any warnings by toggling the CLEAR_FAULT control bit */
 	ret = regmap_write_bits(tas6424->regmap, TAS6424_MISC_CTRL3,
 				TAS6424_CLEAR_FAULT, TAS6424_CLEAR_FAULT);
 	if (ret < 0)
diff --git a/sound/soc/codecs/tas6424.h b/sound/soc/codecs/tas6424.h
index b5958c45ed0e..c67a7835ca66 100644
--- a/sound/soc/codecs/tas6424.h
+++ b/sound/soc/codecs/tas6424.h
@@ -116,6 +116,16 @@
 #define TAS6424_LDGBYPASS_MASK		BIT(TAS6424_LDGBYPASS_SHIFT)
 
 /* TAS6424_GLOB_FAULT1_REG */
+#define TAS6424_FAULT_OC_CH1		BIT(7)
+#define TAS6424_FAULT_OC_CH2		BIT(6)
+#define TAS6424_FAULT_OC_CH3		BIT(5)
+#define TAS6424_FAULT_OC_CH4		BIT(4)
+#define TAS6424_FAULT_DC_CH1		BIT(3)
+#define TAS6424_FAULT_DC_CH2		BIT(2)
+#define TAS6424_FAULT_DC_CH3		BIT(1)
+#define TAS6424_FAULT_DC_CH4		BIT(0)
+
+/* TAS6424_GLOB_FAULT1_REG */
 #define TAS6424_FAULT_CLOCK		BIT(4)
 #define TAS6424_FAULT_PVDD_OV		BIT(3)
 #define TAS6424_FAULT_VBAT_OV		BIT(2)
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index bf92d36b8f8a..608ad49ad978 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -167,6 +167,7 @@ struct aic31xx_priv {
 	u8 p_div;
 	int rate_div_line;
 	bool master_dapm_route_applied;
+	int irq;
 };
 
 struct aic31xx_rate_divs {
@@ -1391,6 +1392,69 @@ static const struct acpi_device_id aic31xx_acpi_match[] = {
 MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match);
 #endif
 
+static irqreturn_t aic31xx_irq(int irq, void *data)
+{
+	struct aic31xx_priv *aic31xx = data;
+	struct device *dev = aic31xx->dev;
+	unsigned int value;
+	bool handled = false;
+	int ret;
+
+	ret = regmap_read(aic31xx->regmap, AIC31XX_INTRDACFLAG, &value);
+	if (ret) {
+		dev_err(dev, "Failed to read interrupt mask: %d\n", ret);
+		goto exit;
+	}
+
+	if (value)
+		handled = true;
+	else
+		goto read_overflow;
+
+	if (value & AIC31XX_HPLSCDETECT)
+		dev_err(dev, "Short circuit on Left output is detected\n");
+	if (value & AIC31XX_HPRSCDETECT)
+		dev_err(dev, "Short circuit on Right output is detected\n");
+	if (value & ~(AIC31XX_HPLSCDETECT |
+		      AIC31XX_HPRSCDETECT))
+		dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value);
+
+read_overflow:
+	ret = regmap_read(aic31xx->regmap, AIC31XX_OFFLAG, &value);
+	if (ret) {
+		dev_err(dev, "Failed to read overflow flag: %d\n", ret);
+		goto exit;
+	}
+
+	if (value)
+		handled = true;
+	else
+		goto exit;
+
+	if (value & AIC31XX_DAC_OF_LEFT)
+		dev_warn(dev, "Left-channel DAC overflow has occurred\n");
+	if (value & AIC31XX_DAC_OF_RIGHT)
+		dev_warn(dev, "Right-channel DAC overflow has occurred\n");
+	if (value & AIC31XX_DAC_OF_SHIFTER)
+		dev_warn(dev, "DAC barrel shifter overflow has occurred\n");
+	if (value & AIC31XX_ADC_OF)
+		dev_warn(dev, "ADC overflow has occurred\n");
+	if (value & AIC31XX_ADC_OF_SHIFTER)
+		dev_warn(dev, "ADC barrel shifter overflow has occurred\n");
+	if (value & ~(AIC31XX_DAC_OF_LEFT |
+		      AIC31XX_DAC_OF_RIGHT |
+		      AIC31XX_DAC_OF_SHIFTER |
+		      AIC31XX_ADC_OF |
+		      AIC31XX_ADC_OF_SHIFTER))
+		dev_warn(dev, "Unknown overflow interrupt flags: 0x%08x\n", value);
+
+exit:
+	if (handled)
+		return IRQ_HANDLED;
+	else
+		return IRQ_NONE;
+}
+
 static int aic31xx_i2c_probe(struct i2c_client *i2c,
 			     const struct i2c_device_id *id)
 {
@@ -1413,6 +1477,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
 		return ret;
 	}
 	aic31xx->dev = &i2c->dev;
+	aic31xx->irq = i2c->irq;
 
 	aic31xx->codec_type = id->driver_data;
 
@@ -1456,6 +1521,26 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
 		return ret;
 	}
 
+	if (aic31xx->irq > 0) {
+		regmap_update_bits(aic31xx->regmap, AIC31XX_GPIO1,
+				   AIC31XX_GPIO1_FUNC_MASK,
+				   AIC31XX_GPIO1_INT1 <<
+				   AIC31XX_GPIO1_FUNC_SHIFT);
+
+		regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL,
+			     AIC31XX_SC |
+			     AIC31XX_ENGINE);
+
+		ret = devm_request_threaded_irq(aic31xx->dev, aic31xx->irq,
+						NULL, aic31xx_irq,
+						IRQF_ONESHOT, "aic31xx-irq",
+						aic31xx);
+		if (ret) {
+			dev_err(aic31xx->dev, "Unable to request IRQ\n");
+			return ret;
+		}
+	}
+
 	if (aic31xx->codec_type & DAC31XX_BIT)
 		return devm_snd_soc_register_component(&i2c->dev,
 				&soc_codec_driver_aic31xx,
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
index 0b587585b38b..2636f2c6bc79 100644
--- a/sound/soc/codecs/tlv320aic31xx.h
+++ b/sound/soc/codecs/tlv320aic31xx.h
@@ -173,6 +173,13 @@ struct aic31xx_pdata {
 #define AIC31XX_HPRDRVPWRSTATUS_MASK	BIT(1)
 #define AIC31XX_SPRDRVPWRSTATUS_MASK	BIT(0)
 
+/* AIC31XX_OFFLAG */
+#define AIC31XX_DAC_OF_LEFT		BIT(7)
+#define AIC31XX_DAC_OF_RIGHT		BIT(6)
+#define AIC31XX_DAC_OF_SHIFTER		BIT(5)
+#define AIC31XX_ADC_OF			BIT(3)
+#define AIC31XX_ADC_OF_SHIFTER		BIT(1)
+
 /* AIC31XX_INTRDACFLAG */
 #define AIC31XX_HPLSCDETECT		BIT(7)
 #define AIC31XX_HPRSCDETECT		BIT(6)
@@ -191,6 +198,22 @@ struct aic31xx_pdata {
 #define AIC31XX_SC			BIT(3)
 #define AIC31XX_ENGINE			BIT(2)
 
+/* AIC31XX_GPIO1 */
+#define AIC31XX_GPIO1_FUNC_MASK		GENMASK(5, 2)
+#define AIC31XX_GPIO1_FUNC_SHIFT	2
+#define AIC31XX_GPIO1_DISABLED		0x00
+#define AIC31XX_GPIO1_INPUT		0x01
+#define AIC31XX_GPIO1_GPI		0x02
+#define AIC31XX_GPIO1_GPO		0x03
+#define AIC31XX_GPIO1_CLKOUT		0x04
+#define AIC31XX_GPIO1_INT1		0x05
+#define AIC31XX_GPIO1_INT2		0x06
+#define AIC31XX_GPIO1_ADC_WCLK		0x07
+#define AIC31XX_GPIO1_SBCLK		0x08
+#define AIC31XX_GPIO1_SWCLK		0x09
+#define AIC31XX_GPIO1_ADC_MOD_CLK	0x10
+#define AIC31XX_GPIO1_SDOUT		0x11
+
 /* AIC31XX_DACSETUP */
 #define AIC31XX_SOFTSTEP_MASK		GENMASK(1, 0)
 
diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c
index ff85a0bf6170..93d84e5ae2d5 100644
--- a/sound/soc/codecs/tscs454.c
+++ b/sound/soc/codecs/tscs454.c
@@ -3459,7 +3459,7 @@ static int tscs454_i2c_probe(struct i2c_client *i2c,
 	/* Sync pg sel reg with cache */
 	regmap_write(tscs454->regmap, R_PAGESEL, 0x00);
 
-	ret = snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454,
+	ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454,
 			tscs454_dais, ARRAY_SIZE(tscs454_dais));
 	if (ret) {
 		dev_err(&i2c->dev, "Failed to register component (%d)\n", ret);
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index c5ae07234a00..bba330e30162 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -88,19 +88,6 @@ static int wm2000_write(struct i2c_client *i2c, unsigned int reg,
 	return regmap_write(wm2000->regmap, reg, value);
 }
 
-static unsigned int wm2000_read(struct i2c_client *i2c, unsigned int r)
-{
-	struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c);
-	unsigned int val;
-	int ret;
-
-	ret = regmap_read(wm2000->regmap, r, &val);
-	if (ret < 0)
-		return -1;
-
-	return val;
-}
-
 static void wm2000_reset(struct wm2000_priv *wm2000)
 {
 	struct i2c_client *i2c = wm2000->i2c;
@@ -115,14 +102,15 @@ static void wm2000_reset(struct wm2000_priv *wm2000)
 static int wm2000_poll_bit(struct i2c_client *i2c,
 			   unsigned int reg, u8 mask)
 {
+	struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c);
 	int timeout = 4000;
-	int val;
+	unsigned int val;
 
-	val = wm2000_read(i2c, reg);
+	regmap_read(wm2000->regmap, reg, &val);
 
 	while (!(val & mask) && --timeout) {
 		msleep(1);
-		val = wm2000_read(i2c, reg);
+		regmap_read(wm2000->regmap, reg, &val);
 	}
 
 	if (timeout == 0)
@@ -135,6 +123,7 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
 {
 	struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev);
 	unsigned long rate;
+	unsigned int val;
 	int ret;
 
 	if (WARN_ON(wm2000->anc_mode != ANC_OFF))
@@ -213,12 +202,17 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
 			     WM2000_MODE_THERMAL_ENABLE);
 	}
 
-	ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY);
+	ret = regmap_read(wm2000->regmap, WM2000_REG_SPEECH_CLARITY, &val);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Unable to read Speech Clarity: %d\n", ret);
+		regulator_bulk_disable(WM2000_NUM_SUPPLIES, wm2000->supplies);
+		return ret;
+	}
 	if (wm2000->speech_clarity)
-		ret |= WM2000_SPEECH_CLARITY;
+		val |= WM2000_SPEECH_CLARITY;
 	else
-		ret &= ~WM2000_SPEECH_CLARITY;
-	wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret);
+		val &= ~WM2000_SPEECH_CLARITY;
+	wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, val);
 
 	wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33);
 	wm2000_write(i2c, WM2000_REG_SYS_START1, 0x02);
@@ -824,7 +818,7 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
 	const char *filename;
 	const struct firmware *fw = NULL;
 	int ret, i;
-	int reg;
+	unsigned int reg;
 	u16 id;
 
 	wm2000 = devm_kzalloc(&i2c->dev, sizeof(*wm2000), GFP_KERNEL);
@@ -860,9 +854,17 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
 	}
 
 	/* Verify that this is a WM2000 */
-	reg = wm2000_read(i2c, WM2000_REG_ID1);
+	ret = regmap_read(wm2000->regmap, WM2000_REG_ID1, &reg);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Unable to read ID1: %d\n", ret);
+		return ret;
+	}
 	id = reg << 8;
-	reg = wm2000_read(i2c, WM2000_REG_ID2);
+	ret = regmap_read(wm2000->regmap, WM2000_REG_ID2, &reg);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Unable to read ID2: %d\n", ret);
+		return ret;
+	}
 	id |= reg & 0xff;
 
 	if (id != 0x2000) {
@@ -871,7 +873,11 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
 		goto err_supplies;
 	}
 
-	reg = wm2000_read(i2c, WM2000_REG_REVISON);
+	ret = regmap_read(wm2000->regmap, WM2000_REG_REVISON, &reg);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Unable to read Revision: %d\n", ret);
+		return ret;
+	}
 	dev_info(&i2c->dev, "revision %c\n", reg + 'A');
 
 	wm2000->mclk = devm_clk_get(&i2c->dev, "MCLK");
diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c
index 317db9a149a7..cf2cdbece122 100644
--- a/sound/soc/codecs/wm8782.c
+++ b/sound/soc/codecs/wm8782.c
@@ -20,6 +20,7 @@
 #include <linux/module.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
+#include <linux/regulator/consumer.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/ac97_codec.h>
@@ -50,7 +51,51 @@ static struct snd_soc_dai_driver wm8782_dai = {
 	},
 };
 
+/* regulator power supply names */
+static const char *supply_names[] = {
+	"Vdda", /* analog supply, 2.7V - 3.6V */
+	"Vdd",  /* digital supply, 2.7V - 5.5V */
+};
+
+struct wm8782_priv {
+	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
+};
+
+static int wm8782_soc_probe(struct snd_soc_component *component)
+{
+	struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+	return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+
+static void wm8782_soc_remove(struct snd_soc_component *component)
+{
+	struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+	regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+
+#ifdef CONFIG_PM
+static int wm8782_soc_suspend(struct snd_soc_component *component)
+{
+	struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+	regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+	return 0;
+}
+
+static int wm8782_soc_resume(struct snd_soc_component *component)
+{
+	struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+	return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+#else
+#define wm8782_soc_suspend      NULL
+#define wm8782_soc_resume       NULL
+#endif /* CONFIG_PM */
+
 static const struct snd_soc_component_driver soc_component_dev_wm8782 = {
+	.probe			= wm8782_soc_probe,
+	.remove			= wm8782_soc_remove,
+	.suspend		= wm8782_soc_suspend,
+	.resume			= wm8782_soc_resume,
 	.dapm_widgets		= wm8782_dapm_widgets,
 	.num_dapm_widgets	= ARRAY_SIZE(wm8782_dapm_widgets),
 	.dapm_routes		= wm8782_dapm_routes,
@@ -63,6 +108,24 @@ static const struct snd_soc_component_driver soc_component_dev_wm8782 = {
 
 static int wm8782_probe(struct platform_device *pdev)
 {
+	struct device *dev = &pdev->dev;
+	struct wm8782_priv *priv;
+	int ret, i;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	dev_set_drvdata(dev, priv);
+
+	for (i = 0; i < ARRAY_SIZE(supply_names); i++)
+		priv->supplies[i].supply = supply_names[i];
+
+	ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies),
+				      priv->supplies);
+	if (ret < 0)
+		return ret;
+
 	return devm_snd_soc_register_component(&pdev->dev,
 			&soc_component_dev_wm8782, &wm8782_dai, 1);
 }
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 1965635ec07c..2a3e5fbd04e4 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -13,7 +13,6 @@
 
 #include <linux/clk.h>
 #include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/init.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 43edaf8cd276..593a11960888 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -11,7 +11,6 @@
  */
 
 #include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index ade34c26ad2f..e873baa9e778 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -638,13 +638,14 @@ static int wm9712_soc_probe(struct snd_soc_component *component)
 {
 	struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component);
 	struct regmap *regmap;
-	int ret;
 
 	if (wm9712->mfd_pdata) {
 		wm9712->ac97 = wm9712->mfd_pdata->ac97;
 		regmap = wm9712->mfd_pdata->regmap;
 	} else {
 #ifdef CONFIG_SND_SOC_AC97_BUS
+		int ret;
+
 		wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID,
 						      WM9712_VENDOR_ID_MASK);
 		if (IS_ERR(wm9712->ac97)) {
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index f61656070225..a53dc174bbf0 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -311,12 +311,12 @@ struct wm_adsp_alg_xm_struct {
 };
 
 struct wm_adsp_buffer {
-	__be32 X_buf_base;		/* XM base addr of first X area */
-	__be32 X_buf_size;		/* Size of 1st X area in words */
-	__be32 X_buf_base2;		/* XM base addr of 2nd X area */
-	__be32 X_buf_brk;		/* Total X size in words */
-	__be32 Y_buf_base;		/* YM base addr of Y area */
-	__be32 wrap;			/* Total size X and Y in words */
+	__be32 buf1_base;		/* Base addr of first buffer area */
+	__be32 buf1_size;		/* Size of buf1 area in DSP words */
+	__be32 buf2_base;		/* Base addr of 2nd buffer area */
+	__be32 buf1_buf2_size;		/* Size of buf1+buf2 in DSP words */
+	__be32 buf3_base;		/* Base addr of buf3 area */
+	__be32 buf_total_size;		/* Size of buf1+buf2+buf3 in DSP words */
 	__be32 high_water_mark;		/* Point at which IRQ is asserted */
 	__be32 irq_count;		/* bits 1-31 count IRQ assertions */
 	__be32 irq_ack;			/* acked IRQ count, bit 0 enables IRQ */
@@ -393,18 +393,18 @@ struct wm_adsp_buffer_region_def {
 static const struct wm_adsp_buffer_region_def default_regions[] = {
 	{
 		.mem_type = WMFW_ADSP2_XM,
-		.base_offset = HOST_BUFFER_FIELD(X_buf_base),
-		.size_offset = HOST_BUFFER_FIELD(X_buf_size),
+		.base_offset = HOST_BUFFER_FIELD(buf1_base),
+		.size_offset = HOST_BUFFER_FIELD(buf1_size),
 	},
 	{
 		.mem_type = WMFW_ADSP2_XM,
-		.base_offset = HOST_BUFFER_FIELD(X_buf_base2),
-		.size_offset = HOST_BUFFER_FIELD(X_buf_brk),
+		.base_offset = HOST_BUFFER_FIELD(buf2_base),
+		.size_offset = HOST_BUFFER_FIELD(buf1_buf2_size),
 	},
 	{
 		.mem_type = WMFW_ADSP2_YM,
-		.base_offset = HOST_BUFFER_FIELD(Y_buf_base),
-		.size_offset = HOST_BUFFER_FIELD(wrap),
+		.base_offset = HOST_BUFFER_FIELD(buf3_base),
+		.size_offset = HOST_BUFFER_FIELD(buf_total_size),
 	},
 };
 
@@ -3345,7 +3345,7 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf)
 		region->cumulative_size = offset;
 
 		adsp_dbg(buf->dsp,
-			 "region=%d type=%d base=%04x off=%04x size=%04x\n",
+			 "region=%d type=%d base=%08x off=%08x size=%08x\n",
 			 i, region->mem_type, region->base_addr,
 			 region->offset, region->cumulative_size);
 	}
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index f70db8412c7c..267aee776b2d 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -1041,6 +1041,42 @@ static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp,
 	return error_ppm;
 }
 
+static inline u32 davinci_mcasp_tx_delay(struct davinci_mcasp *mcasp)
+{
+	if (!mcasp->txnumevt)
+		return 0;
+
+	return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_WFIFOSTS_OFFSET);
+}
+
+static inline u32 davinci_mcasp_rx_delay(struct davinci_mcasp *mcasp)
+{
+	if (!mcasp->rxnumevt)
+		return 0;
+
+	return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_RFIFOSTS_OFFSET);
+}
+
+static snd_pcm_sframes_t davinci_mcasp_delay(
+			struct snd_pcm_substream *substream,
+			struct snd_soc_dai *cpu_dai)
+{
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+	u32 fifo_use;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		fifo_use = davinci_mcasp_tx_delay(mcasp);
+	else
+		fifo_use = davinci_mcasp_rx_delay(mcasp);
+
+	/*
+	 * Divide the used locations with the channel count to get the
+	 * FIFO usage in samples (don't care about partial samples in the
+	 * buffer).
+	 */
+	return fifo_use / substream->runtime->channels;
+}
+
 static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 					struct snd_pcm_hw_params *params,
 					struct snd_soc_dai *cpu_dai)
@@ -1365,6 +1401,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
 	.startup	= davinci_mcasp_startup,
 	.shutdown	= davinci_mcasp_shutdown,
 	.trigger	= davinci_mcasp_trigger,
+	.delay		= davinci_mcasp_delay,
 	.hw_params	= davinci_mcasp_hw_params,
 	.set_fmt	= davinci_mcasp_set_dai_fmt,
 	.set_clkdiv	= davinci_mcasp_set_clkdiv,
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 1033ac6631b0..01052a0808b0 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -151,7 +151,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream,
 	int ret;
 
 	/* Fetch the Back-End dma_data from DPCM */
-	list_for_each_entry(dpcm, &rtd->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(rtd, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *substream_be;
 		struct snd_soc_dai *dai = be->cpu_dai;
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index c1d1d06783e5..57b484768a58 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -807,7 +807,7 @@ static int fsl_esai_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	esai_priv->pdev = pdev;
-	strncpy(esai_priv->name, np->name, sizeof(esai_priv->name) - 1);
+	snprintf(esai_priv->name, sizeof(esai_priv->name), "%pOFn", np);
 
 	/* Get the addresses and IRQ */
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
index 7f0fa4b52223..9981668ab590 100644
--- a/sound/soc/fsl/fsl_utils.c
+++ b/sound/soc/fsl/fsl_utils.c
@@ -57,8 +57,8 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np,
 		of_node_put(dma_channel_np);
 		return ret;
 	}
-	snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
-		 (unsigned long long) res.start, dma_channel_np->name);
+	snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%pOFn",
+		 (unsigned long long) res.start, dma_channel_np);
 
 	iprop = of_get_property(dma_channel_np, "cell-index", NULL);
 	if (!iprop) {
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index ec731223cab3..e339f36cea95 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -57,6 +57,7 @@ static int pcm030_fabric_probe(struct platform_device *op)
 	struct device_node *platform_np;
 	struct snd_soc_card *card = &pcm030_card;
 	struct pcm030_audio_data *pdata;
+	struct snd_soc_dai_link *dai_link;
 	int ret;
 	int i;
 
@@ -78,8 +79,8 @@ static int pcm030_fabric_probe(struct platform_device *op)
 		return -ENODEV;
 	}
 
-	for (i = 0; i < card->num_links; i++)
-		card->dai_link[i].platform_of_node = platform_np;
+	for_each_card_prelinks(card, i, dai_link)
+		dai_link->platform_of_node = platform_np;
 
 	ret = request_module("snd-soc-wm9712");
 	if (ret)
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 2094d2c8919f..25c819e402e1 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -25,6 +25,8 @@ struct graph_card_data {
 	struct graph_dai_props {
 		struct asoc_simple_dai cpu_dai;
 		struct asoc_simple_dai codec_dai;
+		struct snd_soc_dai_link_component codecs; /* single codec */
+		struct snd_soc_dai_link_component platform;
 		unsigned int mclk_fs;
 	} *dai_props;
 	unsigned int mclk_fs;
@@ -180,7 +182,8 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port,
 	if (ret < 0)
 		goto dai_link_of_err;
 
-	of_property_read_u32(rcpu_ep, "mclk-fs", &dai_props->mclk_fs);
+	of_property_read_u32(cpu_ep,   "mclk-fs", &dai_props->mclk_fs);
+	of_property_read_u32(codec_ep, "mclk-fs", &dai_props->mclk_fs);
 
 	ret = asoc_simple_card_parse_graph_cpu(cpu_ep, dai_link);
 	if (ret < 0)
@@ -213,7 +216,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port,
 	ret = asoc_simple_card_set_dailink_name(dev, dai_link,
 						"%s-%s",
 						dai_link->cpu_dai_name,
-						dai_link->codec_dai_name);
+						dai_link->codecs->dai_name);
 	if (ret < 0)
 		goto dai_link_of_err;
 
@@ -299,7 +302,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
 	struct graph_dai_props *dai_props;
 	struct device *dev = &pdev->dev;
 	struct snd_soc_card *card;
-	int num, ret;
+	int num, ret, i;
 
 	/* Allocate the private data and the DAI link array */
 	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -315,6 +318,18 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
 	if (!dai_props || !dai_link)
 		return -ENOMEM;
 
+	/*
+	 * Use snd_soc_dai_link_component instead of legacy style
+	 * It is codec only. but cpu/platform will be supported in the future.
+	 * see
+	 *	soc-core.c :: snd_soc_init_multicodec()
+	 */
+	for (i = 0; i < num; i++) {
+		dai_link[i].codecs	= &dai_props[i].codecs;
+		dai_link[i].num_codecs	= 1;
+		dai_link[i].platform	= &dai_props[i].platform;
+	}
+
 	priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW);
 	if (IS_ERR(priv->pa_gpio)) {
 		ret = PTR_ERR(priv->pa_gpio);
diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c
index 92882e392d6c..b83bb31021a9 100644
--- a/sound/soc/generic/audio-graph-scu-card.c
+++ b/sound/soc/generic/audio-graph-scu-card.c
@@ -25,7 +25,11 @@
 struct graph_card_data {
 	struct snd_soc_card snd_card;
 	struct snd_soc_codec_conf codec_conf;
-	struct asoc_simple_dai *dai_props;
+	struct graph_dai_props {
+		struct asoc_simple_dai dai;
+		struct snd_soc_dai_link_component codecs;
+		struct snd_soc_dai_link_component platform;
+	} *dai_props;
 	struct snd_soc_dai_link *dai_link;
 	struct asoc_simple_card_data adata;
 };
@@ -39,18 +43,18 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
-	struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num);
+	struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num);
 
-	return asoc_simple_card_clk_enable(dai_props);
+	return asoc_simple_card_clk_enable(&dai_props->dai);
 }
 
 static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
-	struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num);
+	struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num);
 
-	asoc_simple_card_clk_disable(dai_props);
+	asoc_simple_card_clk_disable(&dai_props->dai);
 }
 
 static const struct snd_soc_ops asoc_graph_card_ops = {
@@ -63,7 +67,7 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd)
 	struct graph_card_data *priv =	snd_soc_card_get_drvdata(rtd->card);
 	struct snd_soc_dai *dai;
 	struct snd_soc_dai_link *dai_link;
-	struct asoc_simple_dai *dai_props;
+	struct graph_dai_props *dai_props;
 	int num = rtd->num;
 
 	dai_link	= graph_priv_to_link(priv, num);
@@ -72,7 +76,7 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd)
 				rtd->cpu_dai :
 				rtd->codec_dai;
 
-	return asoc_simple_card_init_dai(dai, dai_props);
+	return asoc_simple_card_init_dai(dai, &dai_props->dai);
 }
 
 static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
@@ -92,15 +96,18 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep,
 {
 	struct device *dev = graph_priv_to_dev(priv);
 	struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, idx);
-	struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, idx);
+	struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx);
 	struct snd_soc_card *card = graph_priv_to_card(priv);
 	int ret;
 
 	if (is_fe) {
+		struct snd_soc_dai_link_component *codecs;
+
 		/* BE is dummy */
-		dai_link->codec_of_node		= NULL;
-		dai_link->codec_dai_name	= "snd-soc-dummy-dai";
-		dai_link->codec_name		= "snd-soc-dummy";
+		codecs			= dai_link->codecs;
+		codecs->of_node		= NULL;
+		codecs->dai_name	= "snd-soc-dummy-dai";
+		codecs->name		= "snd-soc-dummy";
 
 		/* FE settings */
 		dai_link->dynamic		= 1;
@@ -110,7 +117,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep,
 		if (ret)
 			return ret;
 
-		ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai_props);
+		ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, &dai_props->dai);
 		if (ret < 0)
 			return ret;
 
@@ -137,23 +144,23 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep,
 		if (ret < 0)
 			return ret;
 
-		ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai_props);
+		ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, &dai_props->dai);
 		if (ret < 0)
 			return ret;
 
 		ret = asoc_simple_card_set_dailink_name(dev, dai_link,
 							"be.%s",
-							dai_link->codec_dai_name);
+							dai_link->codecs->dai_name);
 		if (ret < 0)
 			return ret;
 
 		snd_soc_of_parse_audio_prefix(card,
 					      &priv->codec_conf,
-					      dai_link->codec_of_node,
+					      dai_link->codecs->of_node,
 					      "prefix");
 	}
 
-	ret = asoc_simple_card_of_parse_tdm(ep, dai_props);
+	ret = asoc_simple_card_of_parse_tdm(ep, &dai_props->dai);
 	if (ret)
 		return ret;
 
@@ -331,10 +338,10 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
 {
 	struct graph_card_data *priv;
 	struct snd_soc_dai_link *dai_link;
-	struct asoc_simple_dai *dai_props;
+	struct graph_dai_props *dai_props;
 	struct device *dev = &pdev->dev;
 	struct snd_soc_card *card;
-	int num, ret;
+	int num, ret, i;
 
 	/* Allocate the private data and the DAI link array */
 	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -350,6 +357,18 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
 	if (!dai_props || !dai_link)
 		return -ENOMEM;
 
+	/*
+	 * Use snd_soc_dai_link_component instead of legacy style
+	 * It is codec only. but cpu/platform will be supported in the future.
+	 * see
+	 *	soc-core.c :: snd_soc_init_multicodec()
+	 */
+	for (i = 0; i < num; i++) {
+		dai_link[i].codecs	= &dai_props[i].codecs;
+		dai_link[i].num_codecs	= 1;
+		dai_link[i].platform	= &dai_props[i].platform;
+	}
+
 	priv->dai_props			= dai_props;
 	priv->dai_link			= dai_link;
 
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index d3f3f0fec74c..f34cc6cddfa2 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -173,12 +173,24 @@ int asoc_simple_card_parse_clk(struct device *dev,
 			       struct device_node *node,
 			       struct device_node *dai_of_node,
 			       struct asoc_simple_dai *simple_dai,
-			       const char *name)
+			       const char *dai_name,
+			       struct snd_soc_dai_link_component *dlc)
 {
 	struct clk *clk;
 	u32 val;
 
 	/*
+	 * Use snd_soc_dai_link_component instead of legacy style.
+	 * It is only for codec, but cpu will be supported in the future.
+	 * see
+	 *	soc-core.c :: snd_soc_init_multicodec()
+	 */
+	if (dlc) {
+		dai_of_node	= dlc->of_node;
+		dai_name	= dlc->dai_name;
+	}
+
+	/*
 	 * Parse dai->sysclk come from "clocks = <&xxx>"
 	 * (if system has common clock)
 	 *  or "system-clock-frequency = <xxx>"
@@ -200,7 +212,7 @@ int asoc_simple_card_parse_clk(struct device *dev,
 	if (of_property_read_bool(node, "system-clock-direction-out"))
 		simple_dai->clk_direction = SND_SOC_CLOCK_OUT;
 
-	dev_dbg(dev, "%s : sysclk = %d, direction %d\n", name,
+	dev_dbg(dev, "%s : sysclk = %d, direction %d\n", dai_name,
 		simple_dai->sysclk, simple_dai->clk_direction);
 
 	return 0;
@@ -208,6 +220,7 @@ int asoc_simple_card_parse_clk(struct device *dev,
 EXPORT_SYMBOL_GPL(asoc_simple_card_parse_clk);
 
 int asoc_simple_card_parse_dai(struct device_node *node,
+				    struct snd_soc_dai_link_component *dlc,
 				    struct device_node **dai_of_node,
 				    const char **dai_name,
 				    const char *list_name,
@@ -221,6 +234,17 @@ int asoc_simple_card_parse_dai(struct device_node *node,
 		return 0;
 
 	/*
+	 * Use snd_soc_dai_link_component instead of legacy style.
+	 * It is only for codec, but cpu will be supported in the future.
+	 * see
+	 *	soc-core.c :: snd_soc_init_multicodec()
+	 */
+	if (dlc) {
+		dai_name	= &dlc->dai_name;
+		dai_of_node	= &dlc->of_node;
+	}
+
+	/*
 	 * Get node via "sound-dai = <&phandle port>"
 	 * it will be used as xxx_of_node on soc_bind_dai_link()
 	 */
@@ -278,6 +302,7 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep)
 }
 
 int asoc_simple_card_parse_graph_dai(struct device_node *ep,
+				     struct snd_soc_dai_link_component *dlc,
 				     struct device_node **dai_of_node,
 				     const char **dai_name)
 {
@@ -285,6 +310,17 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep,
 	struct of_phandle_args args;
 	int ret;
 
+	/*
+	 * Use snd_soc_dai_link_component instead of legacy style.
+	 * It is only for codec, but cpu will be supported in the future.
+	 * see
+	 *	soc-core.c :: snd_soc_init_multicodec()
+	 */
+	if (dlc) {
+		dai_name	= &dlc->dai_name;
+		dai_of_node	= &dlc->of_node;
+	}
+
 	if (!ep)
 		return 0;
 	if (!dai_name)
@@ -340,10 +376,11 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai);
 int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link)
 {
 	/* Assumes platform == cpu */
-	if (!dai_link->platform_of_node)
-		dai_link->platform_of_node = dai_link->cpu_of_node;
+	if (!dai_link->platform->of_node)
+		dai_link->platform->of_node = dai_link->cpu_of_node;
 
 	return 0;
+
 }
 EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_dailink);
 
@@ -367,13 +404,11 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_cpu);
 int asoc_simple_card_clean_reference(struct snd_soc_card *card)
 {
 	struct snd_soc_dai_link *dai_link;
-	int num_links;
+	int i;
 
-	for (num_links = 0, dai_link = card->dai_link;
-	     num_links < card->num_links;
-	     num_links++, dai_link++) {
+	for_each_card_prelinks(card, i, dai_link) {
 		of_node_put(dai_link->cpu_of_node);
-		of_node_put(dai_link->codec_of_node);
+		of_node_put(dai_link->codecs->of_node);
 	}
 	return 0;
 }
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 64bf3560c1d1..5a3f59aa4ba5 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -20,6 +20,8 @@ struct simple_card_data {
 	struct simple_dai_props {
 		struct asoc_simple_dai cpu_dai;
 		struct asoc_simple_dai codec_dai;
+		struct snd_soc_dai_link_component codecs; /* single codec */
+		struct snd_soc_dai_link_component platform;
 		unsigned int mclk_fs;
 	} *dai_props;
 	unsigned int mclk_fs;
@@ -234,7 +236,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *node,
 	ret = asoc_simple_card_set_dailink_name(dev, dai_link,
 						"%s-%s",
 						dai_link->cpu_dai_name,
-						dai_link->codec_dai_name);
+						dai_link->codecs->dai_name);
 	if (ret < 0)
 		goto dai_link_of_err;
 
@@ -363,7 +365,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 	struct device *dev = &pdev->dev;
 	struct device_node *np = dev->of_node;
 	struct snd_soc_card *card;
-	int num, ret;
+	int num, ret, i;
 
 	/* Get the number of DAI links */
 	if (np && of_get_child_by_name(np, PREFIX "dai-link"))
@@ -381,6 +383,18 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 	if (!dai_props || !dai_link)
 		return -ENOMEM;
 
+	/*
+	 * Use snd_soc_dai_link_component instead of legacy style
+	 * It is codec only. but cpu/platform will be supported in the future.
+	 * see
+	 *	soc-core.c :: snd_soc_init_multicodec()
+	 */
+	for (i = 0; i < num; i++) {
+		dai_link[i].codecs	= &dai_props[i].codecs;
+		dai_link[i].num_codecs	= 1;
+		dai_link[i].platform	= &dai_props[i].platform;
+	}
+
 	priv->dai_props			= dai_props;
 	priv->dai_link			= dai_link;
 
@@ -403,6 +417,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 
 	} else {
 		struct asoc_simple_card_info *cinfo;
+		struct snd_soc_dai_link_component *codecs;
+		struct snd_soc_dai_link_component *platform;
 
 		cinfo = dev->platform_data;
 		if (!cinfo) {
@@ -419,13 +435,17 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 			return -EINVAL;
 		}
 
+		codecs			= dai_link->codecs;
+		codecs->name		= cinfo->codec;
+		codecs->dai_name	= cinfo->codec_dai.name;
+
+		platform		= dai_link->platform;
+		platform->name		= cinfo->platform;
+
 		card->name		= (cinfo->card) ? cinfo->card : cinfo->name;
 		dai_link->name		= cinfo->name;
 		dai_link->stream_name	= cinfo->name;
-		dai_link->platform_name	= cinfo->platform;
-		dai_link->codec_name	= cinfo->codec;
 		dai_link->cpu_dai_name	= cinfo->cpu_dai.name;
-		dai_link->codec_dai_name = cinfo->codec_dai.name;
 		dai_link->dai_fmt	= cinfo->daifmt;
 		dai_link->init		= asoc_simple_card_dai_init;
 		memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai,
diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c
index 16a83bc51e0e..85b46f0eae0f 100644
--- a/sound/soc/generic/simple-scu-card.c
+++ b/sound/soc/generic/simple-scu-card.c
@@ -22,7 +22,11 @@
 struct simple_card_data {
 	struct snd_soc_card snd_card;
 	struct snd_soc_codec_conf codec_conf;
-	struct asoc_simple_dai *dai_props;
+	struct simple_dai_props {
+		struct asoc_simple_dai dai;
+		struct snd_soc_dai_link_component codecs;
+		struct snd_soc_dai_link_component platform;
+	} *dai_props;
 	struct snd_soc_dai_link *dai_link;
 	struct asoc_simple_card_data adata;
 };
@@ -40,20 +44,20 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct simple_card_data *priv =	snd_soc_card_get_drvdata(rtd->card);
-	struct asoc_simple_dai *dai_props =
+	struct simple_dai_props *dai_props =
 		simple_priv_to_props(priv, rtd->num);
 
-	return asoc_simple_card_clk_enable(dai_props);
+	return asoc_simple_card_clk_enable(&dai_props->dai);
 }
 
 static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct simple_card_data *priv =	snd_soc_card_get_drvdata(rtd->card);
-	struct asoc_simple_dai *dai_props =
+	struct simple_dai_props *dai_props =
 		simple_priv_to_props(priv, rtd->num);
 
-	asoc_simple_card_clk_disable(dai_props);
+	asoc_simple_card_clk_disable(&dai_props->dai);
 }
 
 static const struct snd_soc_ops asoc_simple_card_ops = {
@@ -66,7 +70,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
 	struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct snd_soc_dai *dai;
 	struct snd_soc_dai_link *dai_link;
-	struct asoc_simple_dai *dai_props;
+	struct simple_dai_props *dai_props;
 	int num = rtd->num;
 
 	dai_link	= simple_priv_to_link(priv, num);
@@ -75,7 +79,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
 				rtd->cpu_dai :
 				rtd->codec_dai;
 
-	return asoc_simple_card_init_dai(dai, dai_props);
+	return asoc_simple_card_init_dai(dai, &dai_props->dai);
 }
 
 static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
@@ -95,17 +99,19 @@ static int asoc_simple_card_dai_link_of(struct device_node *np,
 {
 	struct device *dev = simple_priv_to_dev(priv);
 	struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx);
-	struct asoc_simple_dai *dai_props = simple_priv_to_props(priv, idx);
+	struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx);
 	struct snd_soc_card *card = simple_priv_to_card(priv);
 	int ret;
 
 	if (is_fe) {
 		int is_single_links = 0;
+		struct snd_soc_dai_link_component *codecs;
 
 		/* BE is dummy */
-		dai_link->codec_of_node		= NULL;
-		dai_link->codec_dai_name	= "snd-soc-dummy-dai";
-		dai_link->codec_name		= "snd-soc-dummy";
+		codecs			= dai_link->codecs;
+		codecs->of_node		= NULL;
+		codecs->dai_name	= "snd-soc-dummy-dai";
+		codecs->name		= "snd-soc-dummy";
 
 		/* FE settings */
 		dai_link->dynamic		= 1;
@@ -116,7 +122,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np,
 		if (ret)
 			return ret;
 
-		ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai_props);
+		ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, &dai_props->dai);
 		if (ret < 0)
 			return ret;
 
@@ -141,23 +147,23 @@ static int asoc_simple_card_dai_link_of(struct device_node *np,
 		if (ret < 0)
 			return ret;
 
-		ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai_props);
+		ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, &dai_props->dai);
 		if (ret < 0)
 			return ret;
 
 		ret = asoc_simple_card_set_dailink_name(dev, dai_link,
 							"be.%s",
-							dai_link->codec_dai_name);
+							dai_link->codecs->dai_name);
 		if (ret < 0)
 			return ret;
 
 		snd_soc_of_parse_audio_prefix(card,
 					      &priv->codec_conf,
-					      dai_link->codec_of_node,
+					      dai_link->codecs->of_node,
 					      PREFIX "prefix");
 	}
 
-	ret = asoc_simple_card_of_parse_tdm(np, dai_props);
+	ret = asoc_simple_card_of_parse_tdm(np, &dai_props->dai);
 	if (ret)
 		return ret;
 
@@ -230,11 +236,11 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 {
 	struct simple_card_data *priv;
 	struct snd_soc_dai_link *dai_link;
-	struct asoc_simple_dai *dai_props;
+	struct simple_dai_props *dai_props;
 	struct snd_soc_card *card;
 	struct device *dev = &pdev->dev;
 	struct device_node *np = dev->of_node;
-	int num, ret;
+	int num, ret, i;
 
 	/* Allocate the private data */
 	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -248,6 +254,18 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 	if (!dai_props || !dai_link)
 		return -ENOMEM;
 
+	/*
+	 * Use snd_soc_dai_link_component instead of legacy style
+	 * It is codec only. but cpu/platform will be supported in the future.
+	 * see
+	 *	soc-core.c :: snd_soc_init_multicodec()
+	 */
+	for (i = 0; i < num; i++) {
+		dai_link[i].codecs	= &dai_props[i].codecs;
+		dai_link[i].num_codecs	= 1;
+		dai_link[i].platform	= &dai_props[i].platform;
+	}
+
 	priv->dai_props				= dai_props;
 	priv->dai_link				= dai_link;
 
diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c
index 53344a3b7a60..a69e5b11b3da 100644
--- a/sound/soc/hisilicon/hi6210-i2s.c
+++ b/sound/soc/hisilicon/hi6210-i2s.c
@@ -269,13 +269,13 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream,
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_U16_LE:
 		signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
-		/* fallthru */
+		/* fall through */
 	case SNDRV_PCM_FORMAT_S16_LE:
 		bits = HII2S_BITS_16;
 		break;
 	case SNDRV_PCM_FORMAT_U24_LE:
 		signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
-		/* fallthru */
+		/* fall through */
 	case SNDRV_PCM_FORMAT_S24_LE:
 		bits = HII2S_BITS_24;
 		break;
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 6c36da560877..afc559866095 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -765,7 +765,7 @@ static int sst_soc_prepare(struct device *dev)
 	snd_soc_poweroff(drv->soc_card->dev);
 
 	/* set the SSPs to idle */
-	list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) {
+	for_each_card_rtds(drv->soc_card, rtd) {
 		struct snd_soc_dai *dai = rtd->cpu_dai;
 
 		if (dai->active) {
@@ -786,7 +786,7 @@ static void sst_soc_complete(struct device *dev)
 		return;
 
 	/* restart SSPs */
-	list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) {
+	for_each_card_rtds(drv->soc_card, rtd) {
 		struct snd_soc_dai *dai = rtd->cpu_dai;
 
 		if (dai->active) {
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index cccda87f4b34..73ca1350aa31 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -279,6 +279,28 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH
 	  This adds support for ASoC Onboard Codec I2S machine driver. This will
 	  create an alsa sound card for DA7219 + MAX98357A I2S audio codec.
 	  Say Y if you have such a device.
+
+config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH
+	tristate "KBL with DA7219 and MAX98927 in I2S Mode"
+	depends on MFD_INTEL_LPSS && I2C && ACPI
+	select SND_SOC_DA7219
+	select SND_SOC_MAX98927
+	select SND_SOC_DMIC
+	select SND_SOC_HDAC_HDMI
+	help
+	  This adds support for ASoC Onboard Codec I2S machine driver. This will
+	  create an alsa sound card for DA7219 + MAX98927 I2S audio codec.
+	  Say Y if you have such a device.
+	  If unsure select "N".
+
+config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
+	tristate "SKL/KBL/BXT/APL with HDA Codecs"
+	select SND_SOC_HDAC_HDMI
+	select SND_SOC_HDAC_HDA
+	help
+	  This adds support for ASoC machine driver for Intel platforms
+	  SKL/KBL/BXT/APL with iDisp, HDA audio codecs.
+          Say Y or m if you have such a device. This is a recommended option.
 	  If unsure select "N".
 
 config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 87ef8b4058e5..5381e27df9cc 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -17,9 +17,11 @@ snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o
 snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o
 snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o
 snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o
+snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o
 snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o
 snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o
 snd-soc-skl_rt286-objs := skl_rt286.o
+snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o
 snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o
 snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o
 
@@ -41,8 +43,10 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o
 obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o
+obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o
 obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH) += snd-soc-kbl_rt5663_max98927.o
 obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt5663_rt5514_max98927.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o
+obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 7b0ee67b4fc8..68e6543e6cb0 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -223,7 +223,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
 static int broadwell_suspend(struct snd_soc_card *card){
 	struct snd_soc_component *component;
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strcmp(component->name, "i2c-INT343A:00")) {
 
 			dev_dbg(component->dev, "disabling jack detect before going to suspend.\n");
@@ -237,7 +237,7 @@ static int broadwell_suspend(struct snd_soc_card *card){
 static int broadwell_resume(struct snd_soc_card *card){
 	struct snd_soc_component *component;
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strcmp(component->name, "i2c-INT343A:00")) {
 
 			dev_dbg(component->dev, "enabling jack detect for resume.\n");
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index b6dc524830b2..8587bd3d1cc1 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -1048,7 +1048,7 @@ static int byt_rt5640_suspend(struct snd_soc_card *card)
 	if (!BYT_RT5640_JDSRC(byt_rt5640_quirk))
 		return 0;
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strcmp(component->name, byt_rt5640_codec_name)) {
 			dev_dbg(component->dev, "disabling jack detect before suspend\n");
 			snd_soc_component_set_jack(component, NULL, NULL);
@@ -1067,7 +1067,7 @@ static int byt_rt5640_resume(struct snd_soc_card *card)
 	if (!BYT_RT5640_JDSRC(byt_rt5640_quirk))
 		return 0;
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strcmp(component->name, byt_rt5640_codec_name)) {
 			dev_dbg(component->dev, "re-enabling jack detect after resume\n");
 			snd_soc_component_set_jack(component, &priv->jack, NULL);
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 6af02bf879ac..c44298130720 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -742,7 +742,7 @@ static int byt_rt5651_suspend(struct snd_soc_card *card)
 	if (!BYT_RT5651_JDSRC(byt_rt5651_quirk))
 		return 0;
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strcmp(component->name, byt_rt5651_codec_name)) {
 			dev_dbg(component->dev, "disabling jack detect before suspend\n");
 			snd_soc_component_set_jack(component, NULL, NULL);
@@ -761,7 +761,7 @@ static int byt_rt5651_resume(struct snd_soc_card *card)
 	if (!BYT_RT5651_JDSRC(byt_rt5651_quirk))
 		return 0;
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strcmp(component->name, byt_rt5651_codec_name)) {
 			dev_dbg(component->dev, "re-enabling jack detect after resume\n");
 			snd_soc_component_set_jack(component, &priv->jack, NULL);
diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c
index e5aa13058dd7..51f0d45d6f8f 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5672.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -16,6 +16,7 @@
  *  General Public License for more details.
  */
 
+#include <linux/input.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
@@ -212,6 +213,10 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
         if (ret)
                 return ret;
 
+	snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+	snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+	snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+
 	rt5670_set_jack_detect(component, &ctx->headset);
 	if (ctx->mclk) {
 		/*
@@ -342,7 +347,7 @@ static int cht_suspend_pre(struct snd_soc_card *card)
 	struct snd_soc_component *component;
 	struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card);
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strncmp(component->name,
 			     ctx->codec_name, sizeof(ctx->codec_name))) {
 
@@ -359,7 +364,7 @@ static int cht_resume_post(struct snd_soc_card *card)
 	struct snd_soc_component *component;
 	struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card);
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strncmp(component->name,
 			     ctx->codec_name, sizeof(ctx->codec_name))) {
 
diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
new file mode 100644
index 000000000000..3fa1c3ca6d37
--- /dev/null
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -0,0 +1,983 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2018 Intel Corporation.
+
+/*
+ * Intel Kabylake I2S Machine Driver with MAX98927 & DA7219 Codecs
+ *
+ * Modified from:
+ *   Intel Kabylake I2S Machine driver supporting MAX98927 and
+ *   RT5663 codecs
+ */
+
+#include <linux/input.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../codecs/da7219.h"
+#include "../../codecs/hdac_hdmi.h"
+#include "../skylake/skl.h"
+#include "../../codecs/da7219-aad.h"
+
+#define KBL_DIALOG_CODEC_DAI	"da7219-hifi"
+#define MAX98927_CODEC_DAI	"max98927-aif1"
+#define MAXIM_DEV0_NAME		"i2c-MX98927:00"
+#define MAXIM_DEV1_NAME		"i2c-MX98927:01"
+#define DUAL_CHANNEL	2
+#define QUAD_CHANNEL	4
+#define NAME_SIZE	32
+
+static struct snd_soc_card *kabylake_audio_card;
+static struct snd_soc_jack kabylake_hdmi[3];
+
+struct kbl_hdmi_pcm {
+	struct list_head head;
+	struct snd_soc_dai *codec_dai;
+	int device;
+};
+
+struct kbl_codec_private {
+	struct snd_soc_jack kabylake_headset;
+	struct list_head hdmi_pcm_list;
+};
+
+enum {
+	KBL_DPCM_AUDIO_PB = 0,
+	KBL_DPCM_AUDIO_CP,
+	KBL_DPCM_AUDIO_ECHO_REF_CP,
+	KBL_DPCM_AUDIO_REF_CP,
+	KBL_DPCM_AUDIO_DMIC_CP,
+	KBL_DPCM_AUDIO_HDMI1_PB,
+	KBL_DPCM_AUDIO_HDMI2_PB,
+	KBL_DPCM_AUDIO_HDMI3_PB,
+	KBL_DPCM_AUDIO_HS_PB,
+};
+
+static int platform_clock_control(struct snd_soc_dapm_widget *w,
+					struct snd_kcontrol *k, int  event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct snd_soc_card *card = dapm->card;
+	struct snd_soc_dai *codec_dai;
+	int ret = 0;
+
+	codec_dai = snd_soc_card_get_codec_dai(card, KBL_DIALOG_CODEC_DAI);
+	if (!codec_dai) {
+		dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n");
+		return -EIO;
+	}
+
+	/* Configure sysclk for codec */
+	ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000,
+				     SND_SOC_CLOCK_IN);
+	if (ret) {
+		dev_err(card->dev, "can't set codec sysclk configuration\n");
+		return ret;
+	}
+
+	if (SND_SOC_DAPM_EVENT_OFF(event)) {
+		ret = snd_soc_dai_set_pll(codec_dai, 0,
+				     DA7219_SYSCLK_MCLK, 0, 0);
+		if (ret)
+			dev_err(card->dev, "failed to stop PLL: %d\n", ret);
+	} else if (SND_SOC_DAPM_EVENT_ON(event)) {
+		ret = snd_soc_dai_set_pll(codec_dai, 0,	DA7219_SYSCLK_PLL_SRM,
+				     0, DA7219_PLL_FREQ_OUT_98304);
+		if (ret)
+			dev_err(card->dev, "failed to start PLL: %d\n", ret);
+	}
+
+	return ret;
+}
+
+static const struct snd_kcontrol_new kabylake_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Left Spk"),
+	SOC_DAPM_PIN_SWITCH("Right Spk"),
+};
+
+static const struct snd_soc_dapm_widget kabylake_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_SPK("Left Spk", NULL),
+	SND_SOC_DAPM_SPK("Right Spk", NULL),
+	SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+	SND_SOC_DAPM_SPK("DP", NULL),
+	SND_SOC_DAPM_SPK("HDMI", NULL),
+	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+			platform_clock_control, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route kabylake_map[] = {
+	/* speaker */
+	{ "Left Spk", NULL, "Left BE_OUT" },
+	{ "Right Spk", NULL, "Right BE_OUT" },
+
+	/* other jacks */
+	{ "DMic", NULL, "SoC DMIC" },
+
+	{ "HDMI", NULL, "hif5 Output" },
+	{ "DP", NULL, "hif6 Output" },
+
+	/* CODEC BE connections */
+	{ "Left HiFi Playback", NULL, "ssp0 Tx" },
+	{ "Right HiFi Playback", NULL, "ssp0 Tx" },
+	{ "ssp0 Tx", NULL, "spk_out" },
+
+	/* IV feedback path */
+	{ "codec0_fb_in", NULL, "ssp0 Rx"},
+	{ "ssp0 Rx", NULL, "Left HiFi Capture" },
+	{ "ssp0 Rx", NULL, "Right HiFi Capture" },
+
+	/* AEC capture path */
+	{ "echo_ref_out", NULL, "ssp0 Rx" },
+
+	/* DMIC */
+	{ "dmic01_hifi", NULL, "DMIC01 Rx" },
+	{ "DMIC01 Rx", NULL, "DMIC AIF" },
+
+	{ "hifi1", NULL, "iDisp1 Tx" },
+	{ "iDisp1 Tx", NULL, "iDisp1_out" },
+	{ "hifi2", NULL, "iDisp2 Tx" },
+	{ "iDisp2 Tx", NULL, "iDisp2_out" },
+	{ "hifi3", NULL, "iDisp3 Tx"},
+	{ "iDisp3 Tx", NULL, "iDisp3_out"},
+};
+
+static const struct snd_soc_dapm_route kabylake_ssp1_map[] = {
+	{ "Headphone Jack", NULL, "HPL" },
+	{ "Headphone Jack", NULL, "HPR" },
+
+	/* other jacks */
+	{ "MIC", NULL, "Headset Mic" },
+
+	/* CODEC BE connections */
+	{ "Playback", NULL, "ssp1 Tx" },
+	{ "ssp1 Tx", NULL, "codec1_out" },
+
+	{ "hs_in", NULL, "ssp1 Rx" },
+	{ "ssp1 Rx", NULL, "Capture" },
+
+	{ "Headphone Jack", NULL, "Platform Clock" },
+	{ "Headset Mic", NULL, "Platform Clock" },
+};
+
+static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *runtime = substream->private_data;
+	int ret = 0, j;
+
+	for (j = 0; j < runtime->num_codecs; j++) {
+		struct snd_soc_dai *codec_dai = runtime->codec_dais[j];
+
+		if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) {
+			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16);
+			if (ret < 0) {
+				dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret);
+				return ret;
+			}
+		}
+		if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) {
+			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16);
+			if (ret < 0) {
+				dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret);
+				return ret;
+			}
+		}
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops kabylake_ssp0_ops = {
+	.hw_params = kabylake_ssp0_hw_params,
+};
+
+static int kabylake_ssp_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);
+	struct snd_soc_dpcm *dpcm = container_of(
+			params, struct snd_soc_dpcm, hw_params);
+	struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link;
+	struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link;
+
+	/*
+	 * The ADSP will convert the FE rate to 48k, stereo, 24 bit
+	 */
+	if (!strcmp(fe_dai_link->name, "Kbl Audio Port") ||
+	    !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
+	    !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
+		rate->min = rate->max = 48000;
+		channels->min = channels->max = 2;
+		snd_mask_none(fmt);
+		snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	}
+
+	/*
+	 * The speaker on the SSP0 supports S16_LE and not S24_LE.
+	 * thus changing the mask here
+	 */
+	if (!strcmp(be_dai_link->name, "SSP0-Codec"))
+		snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE);
+
+	return 0;
+}
+
+static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_jack *jack;
+	struct snd_soc_card *card = rtd->card;
+	int ret;
+
+
+	ret = snd_soc_dapm_add_routes(&card->dapm,
+			kabylake_ssp1_map,
+			ARRAY_SIZE(kabylake_ssp1_map));
+
+	/*
+	 * Headset buttons map to the google Reference headset.
+	 * These can be configured by userspace.
+	 */
+	ret = snd_soc_card_jack_new(kabylake_audio_card, "Headset Jack",
+			SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+			SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT,
+			&ctx->kabylake_headset, NULL, 0);
+	if (ret) {
+		dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret);
+		return ret;
+	}
+
+	jack = &ctx->kabylake_headset;
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
+
+	da7219_aad_jack_det(component, &ctx->kabylake_headset);
+
+	ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC");
+	if (ret)
+		dev_err(rtd->dev, "SoC DMIC - Ignore suspend failed %d\n", ret);
+
+	return ret;
+}
+
+static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device)
+{
+	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct kbl_hdmi_pcm *pcm;
+
+	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
+
+	pcm->device = device;
+	pcm->codec_dai = dai;
+
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+	return 0;
+}
+
+static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
+{
+	return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB);
+}
+
+static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
+{
+	return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB);
+}
+
+static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
+{
+	return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB);
+}
+
+static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dapm_context *dapm;
+	struct snd_soc_component *component = rtd->cpu_dai->component;
+
+	dapm = snd_soc_component_get_dapm(component);
+	snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
+
+	return 0;
+}
+
+static const unsigned int rates[] = {
+	48000,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_rates = {
+	.count = ARRAY_SIZE(rates),
+	.list  = rates,
+	.mask = 0,
+};
+
+static const unsigned int channels[] = {
+	DUAL_CHANNEL,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_channels = {
+	.count = ARRAY_SIZE(channels),
+	.list = channels,
+	.mask = 0,
+};
+
+static unsigned int channels_quad[] = {
+	QUAD_CHANNEL,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_channels_quad = {
+	.count = ARRAY_SIZE(channels_quad),
+	.list = channels_quad,
+	.mask = 0,
+};
+
+static int kbl_fe_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	/*
+	 * On this platform for PCM device we support,
+	 * 48Khz
+	 * stereo
+	 * 16 bit audio
+	 */
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+					   &constraints_channels);
+
+	runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+	snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16);
+
+	snd_pcm_hw_constraint_list(runtime, 0,
+				SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
+
+	return 0;
+}
+
+static const struct snd_soc_ops kabylake_da7219_fe_ops = {
+	.startup = kbl_fe_startup,
+};
+
+static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
+		struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *channels = hw_param_interval(params,
+				SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/*
+	 * set BE channel constraint as user FE channels
+	 */
+
+	if (params_channels(params) == 2)
+		channels->min = channels->max = 2;
+	else
+		channels->min = channels->max = 4;
+
+	return 0;
+}
+
+static int kabylake_dmic_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+			&constraints_channels_quad);
+
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
+}
+
+static struct snd_soc_ops kabylake_dmic_ops = {
+	.startup = kabylake_dmic_startup,
+};
+
+static const unsigned int rates_16000[] = {
+	16000,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_16000 = {
+	.count = ARRAY_SIZE(rates_16000),
+	.list  = rates_16000,
+};
+
+static const unsigned int ch_mono[] = {
+	1,
+};
+static const struct snd_pcm_hw_constraint_list constraints_refcap = {
+	.count = ARRAY_SIZE(ch_mono),
+	.list  = ch_mono,
+};
+
+static int kabylake_refcap_startup(struct snd_pcm_substream *substream)
+{
+	substream->runtime->hw.channels_max = 1;
+	snd_pcm_hw_constraint_list(substream->runtime, 0,
+					SNDRV_PCM_HW_PARAM_CHANNELS,
+					&constraints_refcap);
+
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+				SNDRV_PCM_HW_PARAM_RATE,
+				&constraints_16000);
+}
+
+
+static struct snd_soc_ops skylaye_refcap_ops = {
+	.startup = kabylake_refcap_startup,
+};
+
+static struct snd_soc_codec_conf max98927_codec_conf[] = {
+
+	{
+		.dev_name = MAXIM_DEV0_NAME,
+		.name_prefix = "Right",
+	},
+
+	{
+		.dev_name = MAXIM_DEV1_NAME,
+		.name_prefix = "Left",
+	},
+};
+
+static struct snd_soc_dai_link_component ssp0_codec_components[] = {
+	{ /* Left */
+		.name = MAXIM_DEV0_NAME,
+		.dai_name = MAX98927_CODEC_DAI,
+	},
+
+	{  /* For Right */
+		.name = MAXIM_DEV1_NAME,
+		.dai_name = MAX98927_CODEC_DAI,
+	},
+
+};
+
+/* kabylake digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link kabylake_dais[] = {
+	/* Front End DAI links */
+	[KBL_DPCM_AUDIO_PB] = {
+		.name = "Kbl Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:1f.3",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.nonatomic = 1,
+		.init = kabylake_da7219_fe_init,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+		.ops = &kabylake_da7219_fe_ops,
+	},
+	[KBL_DPCM_AUDIO_CP] = {
+		.name = "Kbl Audio Capture Port",
+		.stream_name = "Audio Record",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:1f.3",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.nonatomic = 1,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+		.ops = &kabylake_da7219_fe_ops,
+	},
+	[KBL_DPCM_AUDIO_ECHO_REF_CP] = {
+		.name = "Kbl Audio Echo Reference cap",
+		.stream_name = "Echoreference Capture",
+		.cpu_dai_name = "Echoref Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.init = NULL,
+		.capture_only = 1,
+		.nonatomic = 1,
+	},
+	[KBL_DPCM_AUDIO_REF_CP] = {
+		.name = "Kbl Audio Reference cap",
+		.stream_name = "Wake on Voice",
+		.cpu_dai_name = "Reference Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.init = NULL,
+		.dpcm_capture = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+		.ops = &skylaye_refcap_ops,
+	},
+	[KBL_DPCM_AUDIO_DMIC_CP] = {
+		.name = "Kbl Audio DMIC cap",
+		.stream_name = "dmiccap",
+		.cpu_dai_name = "DMIC Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.init = NULL,
+		.dpcm_capture = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+		.ops = &kabylake_dmic_ops,
+	},
+	[KBL_DPCM_AUDIO_HDMI1_PB] = {
+		.name = "Kbl HDMI Port1",
+		.stream_name = "Hdmi1",
+		.cpu_dai_name = "HDMI1 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[KBL_DPCM_AUDIO_HDMI2_PB] = {
+		.name = "Kbl HDMI Port2",
+		.stream_name = "Hdmi2",
+		.cpu_dai_name = "HDMI2 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[KBL_DPCM_AUDIO_HDMI3_PB] = {
+		.name = "Kbl HDMI Port3",
+		.stream_name = "Hdmi3",
+		.cpu_dai_name = "HDMI3 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+		.init = NULL,
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[KBL_DPCM_AUDIO_HS_PB] = {
+		.name = "Kbl Audio Headset Playback",
+		.stream_name = "Headset Audio",
+		.cpu_dai_name = "System Pin2",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+		.init = kabylake_da7219_fe_init,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.ops = &kabylake_da7219_fe_ops,
+
+	},
+
+	/* Back End DAI links */
+	{
+		/* SSP0 - Codec */
+		.name = "SSP0-Codec",
+		.id = 0,
+		.cpu_dai_name = "SSP0 Pin",
+		.platform_name = "0000:00:1f.3",
+		.no_pcm = 1,
+		.codecs = ssp0_codec_components,
+		.num_codecs = ARRAY_SIZE(ssp0_codec_components),
+		.dai_fmt = SND_SOC_DAIFMT_DSP_B |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = kabylake_ssp_fixup,
+		.ops = &kabylake_ssp0_ops,
+	},
+	{
+		/* SSP1 - Codec */
+		.name = "SSP1-Codec",
+		.id = 1,
+		.cpu_dai_name = "SSP1 Pin",
+		.platform_name = "0000:00:1f.3",
+		.no_pcm = 1,
+		.codec_name = "i2c-DLGS7219:00",
+		.codec_dai_name = KBL_DIALOG_CODEC_DAI,
+		.init = kabylake_da7219_codec_init,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = kabylake_ssp_fixup,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+	{
+		.name = "dmic01",
+		.id = 2,
+		.cpu_dai_name = "DMIC01 Pin",
+		.codec_name = "dmic-codec",
+		.codec_dai_name = "dmic-hifi",
+		.platform_name = "0000:00:1f.3",
+		.be_hw_params_fixup = kabylake_dmic_fixup,
+		.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 = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.init = kabylake_hdmi1_init,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp2",
+		.id = 4,
+		.cpu_dai_name = "iDisp2 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi2",
+		.platform_name = "0000:00:1f.3",
+		.init = kabylake_hdmi2_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 = "0000:00:1f.3",
+		.init = kabylake_hdmi3_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+};
+
+/* kabylake digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link kabylake_max98927_dais[] = {
+	/* Front End DAI links */
+	[KBL_DPCM_AUDIO_PB] = {
+		.name = "Kbl Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:1f.3",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.nonatomic = 1,
+		.init = kabylake_da7219_fe_init,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+		.ops = &kabylake_da7219_fe_ops,
+	},
+	[KBL_DPCM_AUDIO_CP] = {
+		.name = "Kbl Audio Capture Port",
+		.stream_name = "Audio Record",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:1f.3",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.nonatomic = 1,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+		.ops = &kabylake_da7219_fe_ops,
+	},
+	[KBL_DPCM_AUDIO_ECHO_REF_CP] = {
+		.name = "Kbl Audio Echo Reference cap",
+		.stream_name = "Echoreference Capture",
+		.cpu_dai_name = "Echoref Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.init = NULL,
+		.capture_only = 1,
+		.nonatomic = 1,
+	},
+	[KBL_DPCM_AUDIO_REF_CP] = {
+		.name = "Kbl Audio Reference cap",
+		.stream_name = "Wake on Voice",
+		.cpu_dai_name = "Reference Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.init = NULL,
+		.dpcm_capture = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+		.ops = &skylaye_refcap_ops,
+	},
+	[KBL_DPCM_AUDIO_DMIC_CP] = {
+		.name = "Kbl Audio DMIC cap",
+		.stream_name = "dmiccap",
+		.cpu_dai_name = "DMIC Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.init = NULL,
+		.dpcm_capture = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+		.ops = &kabylake_dmic_ops,
+	},
+	[KBL_DPCM_AUDIO_HDMI1_PB] = {
+		.name = "Kbl HDMI Port1",
+		.stream_name = "Hdmi1",
+		.cpu_dai_name = "HDMI1 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[KBL_DPCM_AUDIO_HDMI2_PB] = {
+		.name = "Kbl HDMI Port2",
+		.stream_name = "Hdmi2",
+		.cpu_dai_name = "HDMI2 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[KBL_DPCM_AUDIO_HDMI3_PB] = {
+		.name = "Kbl HDMI Port3",
+		.stream_name = "Hdmi3",
+		.cpu_dai_name = "HDMI3 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+		.init = NULL,
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+
+	/* Back End DAI links */
+	{
+		/* SSP0 - Codec */
+		.name = "SSP0-Codec",
+		.id = 0,
+		.cpu_dai_name = "SSP0 Pin",
+		.platform_name = "0000:00:1f.3",
+		.no_pcm = 1,
+		.codecs = ssp0_codec_components,
+		.num_codecs = ARRAY_SIZE(ssp0_codec_components),
+		.dai_fmt = SND_SOC_DAIFMT_DSP_B |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = kabylake_ssp_fixup,
+		.ops = &kabylake_ssp0_ops,
+	},
+	{
+		.name = "dmic01",
+		.id = 1,
+		.cpu_dai_name = "DMIC01 Pin",
+		.codec_name = "dmic-codec",
+		.codec_dai_name = "dmic-hifi",
+		.platform_name = "0000:00:1f.3",
+		.be_hw_params_fixup = kabylake_dmic_fixup,
+		.ignore_suspend = 1,
+		.dpcm_capture = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp1",
+		.id = 2,
+		.cpu_dai_name = "iDisp1 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi1",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.init = kabylake_hdmi1_init,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp2",
+		.id = 3,
+		.cpu_dai_name = "iDisp2 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi2",
+		.platform_name = "0000:00:1f.3",
+		.init = kabylake_hdmi2_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp3",
+		.id = 4,
+		.cpu_dai_name = "iDisp3 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi3",
+		.platform_name = "0000:00:1f.3",
+		.init = kabylake_hdmi3_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+};
+
+static int kabylake_card_late_probe(struct snd_soc_card *card)
+{
+	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card);
+	struct kbl_hdmi_pcm *pcm;
+	struct snd_soc_component *component = NULL;
+	int err, i = 0;
+	char jack_name[NAME_SIZE];
+
+	list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+		component = pcm->codec_dai->component;
+		snprintf(jack_name, sizeof(jack_name),
+			"HDMI/DP, pcm=%d Jack", pcm->device);
+		err = snd_soc_card_jack_new(card, jack_name,
+					SND_JACK_AVOUT, &kabylake_hdmi[i],
+					NULL, 0);
+
+		if (err)
+			return err;
+
+		err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
+						&kabylake_hdmi[i]);
+		if (err < 0)
+			return err;
+
+		i++;
+	}
+
+	if (!component)
+		return -EINVAL;
+
+	return hdac_hdmi_jack_port_init(component, &card->dapm);
+
+	return 0;
+}
+
+/* kabylake audio machine driver for SPT + DA7219 */
+static struct snd_soc_card kbl_audio_card_da7219_m98927 = {
+	.name = "kblda7219m98927",
+	.owner = THIS_MODULE,
+	.dai_link = kabylake_dais,
+	.num_links = ARRAY_SIZE(kabylake_dais),
+	.controls = kabylake_controls,
+	.num_controls = ARRAY_SIZE(kabylake_controls),
+	.dapm_widgets = kabylake_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(kabylake_widgets),
+	.dapm_routes = kabylake_map,
+	.num_dapm_routes = ARRAY_SIZE(kabylake_map),
+	.codec_conf = max98927_codec_conf,
+	.num_configs = ARRAY_SIZE(max98927_codec_conf),
+	.fully_routed = true,
+	.late_probe = kabylake_card_late_probe,
+};
+
+/* kabylake audio machine driver for Maxim98927 */
+static struct snd_soc_card kbl_audio_card_max98927 = {
+	.name = "kblmax98927",
+	.owner = THIS_MODULE,
+	.dai_link = kabylake_max98927_dais,
+	.num_links = ARRAY_SIZE(kabylake_max98927_dais),
+	.controls = kabylake_controls,
+	.num_controls = ARRAY_SIZE(kabylake_controls),
+	.dapm_widgets = kabylake_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(kabylake_widgets),
+	.dapm_routes = kabylake_map,
+	.num_dapm_routes = ARRAY_SIZE(kabylake_map),
+	.codec_conf = max98927_codec_conf,
+	.num_configs = ARRAY_SIZE(max98927_codec_conf),
+	.fully_routed = true,
+	.late_probe = kabylake_card_late_probe,
+};
+
+static int kabylake_audio_probe(struct platform_device *pdev)
+{
+	struct kbl_codec_private *ctx;
+
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+	if (!ctx)
+		return -ENOMEM;
+
+	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
+	kabylake_audio_card =
+		(struct snd_soc_card *)pdev->id_entry->driver_data;
+
+	kabylake_audio_card->dev = &pdev->dev;
+	snd_soc_card_set_drvdata(kabylake_audio_card, ctx);
+
+	return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card);
+}
+
+static const struct platform_device_id kbl_board_ids[] = {
+	{
+		.name = "kbl_da7219_max98927",
+		.driver_data =
+			(kernel_ulong_t)&kbl_audio_card_da7219_m98927,
+	},
+	{
+		.name = "kbl_max98927",
+		.driver_data =
+			(kernel_ulong_t)&kbl_audio_card_max98927,
+	},
+	{ }
+};
+
+static struct platform_driver kabylake_audio = {
+	.probe = kabylake_audio_probe,
+	.driver = {
+		.name = "kbl_da7219_max98927",
+		.pm = &snd_soc_pm_ops,
+	},
+	.id_table = kbl_board_ids,
+};
+
+module_platform_driver(kabylake_audio)
+
+/* Module information */
+MODULE_DESCRIPTION("Audio KabyLake Machine driver for MAX98927 & DA7219");
+MODULE_AUTHOR("Mac Chiang <mac.chiang@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:kbl_da7219_max98927");
+MODULE_ALIAS("platform:kbl_max98927");
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 21a6490746a6..99e1320c485f 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -488,11 +488,10 @@ static int kabylake_ssp0_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;
 	int ret = 0, j;
 
-	for (j = 0; j < rtd->num_codecs; j++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
-
+	for_each_rtd_codec_dai(rtd, j, codec_dai) {
 		if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) {
 			/*
 			 * Use channel 4 and 5 for the first amp
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index a892b37eab7c..a737c915d46a 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -353,11 +353,10 @@ static int kabylake_ssp0_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;
 	int ret = 0, j;
 
-	for (j = 0; j < rtd->num_codecs; j++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
-
+	for_each_rtd_codec_dai(rtd, j, codec_dai) {
 		if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) {
 			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16);
 			if (ret < 0) {
diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c
new file mode 100644
index 000000000000..3fdbf239da74
--- /dev/null
+++ b/sound/soc/intel/boards/skl_hda_dsp_common.c
@@ -0,0 +1,127 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2015-18 Intel Corporation.
+
+/*
+ * Common functions used in different Intel machine drivers
+ */
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../codecs/hdac_hdmi.h"
+#include "../skylake/skl.h"
+#include "skl_hda_dsp_common.h"
+
+#define NAME_SIZE	32
+
+int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device)
+{
+	struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card);
+	struct skl_hda_hdmi_pcm *pcm;
+	char dai_name[NAME_SIZE];
+
+	pcm = devm_kzalloc(card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
+
+	snprintf(dai_name, sizeof(dai_name), "intel-hdmi-hifi%d",
+		 ctx->dai_index);
+	pcm->codec_dai = snd_soc_card_get_codec_dai(card, dai_name);
+	if (!pcm->codec_dai)
+		return -EINVAL;
+
+	pcm->device = device;
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+	return 0;
+}
+
+/* skl_hda_digital audio interface glue - connects codec <--> CPU */
+struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = {
+	/* Back End DAI links */
+	{
+		.name = "iDisp1",
+		.id = 1,
+		.cpu_dai_name = "iDisp1 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi1",
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp2",
+		.id = 2,
+		.cpu_dai_name = "iDisp2 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi2",
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp3",
+		.id = 3,
+		.cpu_dai_name = "iDisp3 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi3",
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "Analog Playback and Capture",
+		.id = 4,
+		.cpu_dai_name = "Analog CPU DAI",
+		.codec_name = "ehdaudio0D0",
+		.codec_dai_name = "Analog Codec DAI",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.init = NULL,
+		.no_pcm = 1,
+	},
+	{
+		.name = "Digital Playback and Capture",
+		.id = 5,
+		.cpu_dai_name = "Digital CPU DAI",
+		.codec_name = "ehdaudio0D0",
+		.codec_dai_name = "Digital Codec DAI",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.init = NULL,
+		.no_pcm = 1,
+	},
+};
+
+int skl_hda_hdmi_jack_init(struct snd_soc_card *card)
+{
+	struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card);
+	struct snd_soc_component *component = NULL;
+	struct skl_hda_hdmi_pcm *pcm;
+	char jack_name[NAME_SIZE];
+	int err;
+
+	list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+		component = pcm->codec_dai->component;
+		snprintf(jack_name, sizeof(jack_name),
+			 "HDMI/DP, pcm=%d Jack", pcm->device);
+		err = snd_soc_card_jack_new(card, jack_name,
+					    SND_JACK_AVOUT, &pcm->hdmi_jack,
+					    NULL, 0);
+
+		if (err)
+			return err;
+
+		err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
+					  &pcm->hdmi_jack);
+		if (err < 0)
+			return err;
+	}
+
+	if (!component)
+		return -EINVAL;
+
+	return hdac_hdmi_jack_port_init(component, &card->dapm);
+}
diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h
new file mode 100644
index 000000000000..87c50aff56cd
--- /dev/null
+++ b/sound/soc/intel/boards/skl_hda_dsp_common.h
@@ -0,0 +1,38 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright(c) 2015-18 Intel Corporation.
+ */
+
+/*
+ * This file defines data structures used in Machine Driver for Intel
+ * platforms with HDA Codecs.
+ */
+
+#ifndef __SOUND_SOC_HDA_DSP_COMMON_H
+#define __SOUND_SOC_HDA_DSP_COMMON_H
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+
+#define HDA_DSP_MAX_BE_DAI_LINKS 5
+
+struct skl_hda_hdmi_pcm {
+	struct list_head head;
+	struct snd_soc_dai *codec_dai;
+	struct snd_soc_jack hdmi_jack;
+	int device;
+};
+
+struct skl_hda_private {
+	struct list_head hdmi_pcm_list;
+	int pcm_count;
+	int dai_index;
+	const char *platform_name;
+};
+
+extern struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS];
+int skl_hda_hdmi_jack_init(struct snd_soc_card *card);
+int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device);
+
+#endif /* __SOUND_SOC_HDA_DSP_COMMON_H */
diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
new file mode 100644
index 000000000000..b415dd4c85f5
--- /dev/null
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -0,0 +1,183 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2015-18 Intel Corporation.
+
+/*
+ * Machine Driver for SKL+ platforms with DSP and iDisp, HDA Codecs
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../codecs/hdac_hdmi.h"
+#include "../skylake/skl.h"
+#include "skl_hda_dsp_common.h"
+
+static const struct snd_soc_dapm_widget skl_hda_widgets[] = {
+	SND_SOC_DAPM_HP("Analog Out", NULL),
+	SND_SOC_DAPM_MIC("Analog In", NULL),
+	SND_SOC_DAPM_HP("Alt Analog Out", NULL),
+	SND_SOC_DAPM_MIC("Alt Analog In", NULL),
+	SND_SOC_DAPM_SPK("Digital Out", NULL),
+	SND_SOC_DAPM_MIC("Digital In", NULL),
+};
+
+static const struct snd_soc_dapm_route skl_hda_map[] = {
+	{ "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"},
+
+	{ "Analog Out", NULL, "Codec Output Pin1" },
+	{ "Digital Out", NULL, "Codec Output Pin2" },
+	{ "Alt Analog Out", NULL, "Codec Output Pin3" },
+
+	{ "Codec Input Pin1", NULL, "Analog In" },
+	{ "Codec Input Pin2", NULL, "Digital In" },
+	{ "Codec Input Pin3", NULL, "Alt Analog In" },
+
+	/* CODEC BE connections */
+	{ "Analog Codec Playback", NULL, "Analog CPU Playback" },
+	{ "Analog CPU Playback", NULL, "codec0_out" },
+	{ "Digital Codec Playback", NULL, "Digital CPU Playback" },
+	{ "Digital CPU Playback", NULL, "codec1_out" },
+	{ "Alt Analog Codec Playback", NULL, "Alt Analog CPU Playback" },
+	{ "Alt Analog CPU Playback", NULL, "codec2_out" },
+
+	{ "codec0_in", NULL, "Analog CPU Capture" },
+	{ "Analog CPU Capture", NULL, "Analog Codec Capture" },
+	{ "codec1_in", NULL, "Digital CPU Capture" },
+	{ "Digital CPU Capture", NULL, "Digital Codec Capture" },
+	{ "codec2_in", NULL, "Alt Analog CPU Capture" },
+	{ "Alt Analog CPU Capture", NULL, "Alt Analog Codec Capture" },
+};
+
+static int skl_hda_card_late_probe(struct snd_soc_card *card)
+{
+	return skl_hda_hdmi_jack_init(card);
+}
+
+static int
+skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link)
+{
+	struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card);
+	int ret = 0;
+
+	dev_dbg(card->dev, "%s: dai link name - %s\n", __func__, link->name);
+	link->platform_name = ctx->platform_name;
+	link->nonatomic = 1;
+
+	if (strstr(link->name, "HDMI")) {
+		ret = skl_hda_hdmi_add_pcm(card, ctx->pcm_count);
+
+		if (ret < 0)
+			return ret;
+
+		ctx->dai_index++;
+	}
+
+	ctx->pcm_count++;
+	return ret;
+}
+
+static struct snd_soc_card hda_soc_card = {
+	.name = "skl_hda_card",
+	.owner = THIS_MODULE,
+	.dai_link = skl_hda_be_dai_links,
+	.dapm_widgets = skl_hda_widgets,
+	.dapm_routes = skl_hda_map,
+	.add_dai_link = skl_hda_add_dai_link,
+	.fully_routed = true,
+	.late_probe = skl_hda_card_late_probe,
+};
+
+#define IDISP_DAI_COUNT		3
+/* there are two routes per iDisp output */
+#define IDISP_ROUTE_COUNT	(IDISP_DAI_COUNT * 2)
+#define IDISP_CODEC_MASK	0x4
+
+static int skl_hda_fill_card_info(struct skl_machine_pdata *pdata)
+{
+	struct snd_soc_card *card = &hda_soc_card;
+	struct snd_soc_dai_link *dai_link;
+	u32 codec_count, codec_mask;
+	int i, num_links, num_route;
+
+	codec_mask = pdata->codec_mask;
+	codec_count = hweight_long(codec_mask);
+
+	if (codec_count == 1 && pdata->codec_mask & IDISP_CODEC_MASK) {
+		num_links = IDISP_DAI_COUNT;
+		num_route = IDISP_ROUTE_COUNT;
+	} else if (codec_count == 2 && codec_mask & IDISP_CODEC_MASK) {
+		num_links = ARRAY_SIZE(skl_hda_be_dai_links);
+		num_route = ARRAY_SIZE(skl_hda_map),
+		card->dapm_widgets = skl_hda_widgets;
+		card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets);
+	} else {
+		return -EINVAL;
+	}
+
+	card->num_links = num_links;
+	card->num_dapm_routes = num_route;
+
+	for_each_card_prelinks(card, i, dai_link)
+		dai_link->platform_name = pdata->platform;
+
+	return 0;
+}
+
+static int skl_hda_audio_probe(struct platform_device *pdev)
+{
+	struct skl_machine_pdata *pdata;
+	struct skl_hda_private *ctx;
+	int ret;
+
+	dev_dbg(&pdev->dev, "%s: entry\n", __func__);
+
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+	if (!ctx)
+		return -ENOMEM;
+
+	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
+	pdata = dev_get_drvdata(&pdev->dev);
+	if (!pdata)
+		return -EINVAL;
+
+	ret = skl_hda_fill_card_info(pdata);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n");
+		return ret;
+	}
+
+	ctx->pcm_count = hda_soc_card.num_links;
+	ctx->dai_index = 1; /* hdmi codec dai name starts from index 1 */
+	ctx->platform_name = pdata->platform;
+
+	hda_soc_card.dev = &pdev->dev;
+	snd_soc_card_set_drvdata(&hda_soc_card, ctx);
+
+	return devm_snd_soc_register_card(&pdev->dev, &hda_soc_card);
+}
+
+static struct platform_driver skl_hda_audio = {
+	.probe = skl_hda_audio_probe,
+	.driver = {
+		.name = "skl_hda_dsp_generic",
+		.pm = &snd_soc_pm_ops,
+	},
+};
+
+module_platform_driver(skl_hda_audio)
+
+/* Module information */
+MODULE_DESCRIPTION("SKL/KBL/BXT/APL HDA Generic Machine driver");
+MODULE_AUTHOR("Rakesh Ughreja <rakesh.a.ughreja@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:skl_hda_dsp_generic");
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index 915a34cdc8ac..c1f50a079d34 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -7,7 +7,8 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m
 	soc-acpi-intel-hsw-bdw-match.o \
 	soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \
 	soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \
-	soc-acpi-intel-cnl-match.o
+	soc-acpi-intel-cnl-match.o \
+	soc-acpi-intel-hda-match.o
 
 obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
 obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o
diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
index 4daa8a4f0c0c..097dc06377ba 100644
--- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
@@ -34,6 +34,13 @@ static const struct dmi_system_id byt_table[] = {
 		.callback = byt_thinkpad10_quirk_cb,
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 8"),
+		},
+	},
+	{
+		.callback = byt_thinkpad10_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
 			DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 10"),
 		},
 	},
diff --git a/sound/soc/intel/common/soc-acpi-intel-hda-match.c b/sound/soc/intel/common/soc-acpi-intel-hda-match.c
new file mode 100644
index 000000000000..533c1064f84b
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-hda-match.c
@@ -0,0 +1,40 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2018, Intel Corporation.
+
+/*
+ * soc-apci-intel-hda-match.c - tables and support for HDA+ACPI enumeration.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+#include "../skylake/skl.h"
+
+static struct skl_machine_pdata hda_pdata = {
+	.use_tplg_pcm = true,
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_hda_machines[] = {
+	{
+		/* .id is not used in this file */
+		.drv_name = "skl_hda_dsp_generic",
+
+		/* .fw_filename is dynamically set in skylake driver */
+
+		/* .sof_fw_filename is dynamically set in sof/intel driver */
+
+		.sof_tplg_filename = "intel/sof-hda-generic.tplg",
+
+		/*
+		 * .machine_quirk and .quirk_data are not used here but
+		 * can be used if we need a more complicated machine driver
+		 * combining HDA+other device (e.g. DMIC).
+		 */
+		.pdata = &hda_pdata,
+	},
+	{},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_hda_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
index 0ee173ca437d..a317b7790fce 100644
--- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
@@ -32,6 +32,11 @@ static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = {
 	.codecs = {"MX98357A"}
 };
 
+static struct snd_soc_acpi_codecs kbl_7219_98927_codecs = {
+	.num_codecs = 1,
+	.codecs = {"MX98927"}
+};
+
 struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = {
 	{
 		.id = "INT343A",
@@ -83,6 +88,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = {
 		.quirk_data = &kbl_7219_98357_codecs,
 		.pdata = &skl_dmic_data,
 	},
+	{
+		.id = "DLGS7219",
+		.drv_name = "kbl_da7219_max98927",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &kbl_7219_98927_codecs,
+		.pdata = &skl_dmic_data
+	},
 	{},
 };
 EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines);
diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c
index 11041aedea31..1e067504b604 100644
--- a/sound/soc/intel/common/sst-firmware.c
+++ b/sound/soc/intel/common/sst-firmware.c
@@ -355,7 +355,7 @@ struct sst_fw *sst_fw_new(struct sst_dsp *dsp,
 
 	/* allocate DMA buffer to store FW data */
 	sst_fw->dma_buf = dma_alloc_coherent(dsp->dma_dev, sst_fw->size,
-				&sst_fw->dmable_fw_paddr, GFP_DMA | GFP_KERNEL);
+				&sst_fw->dmable_fw_paddr, GFP_KERNEL);
 	if (!sst_fw->dma_buf) {
 		dev_err(dsp->dev, "error: DMA alloc failed\n");
 		kfree(sst_fw);
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 823e39103edd..557f80c0bfe5 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -32,6 +32,7 @@
 #define HDA_MONO 1
 #define HDA_STEREO 2
 #define HDA_QUAD 4
+#define HDA_MAX 8
 
 static const struct snd_pcm_hardware azx_pcm_hw = {
 	.info =			(SNDRV_PCM_INFO_MMAP |
@@ -494,6 +495,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 							stream->lpib);
 			snd_hdac_ext_stream_set_lpib(stream, stream->lpib);
 		}
+		/* fall through */
 
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
@@ -569,7 +571,10 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream,
 	stream_tag = hdac_stream(link_dev)->stream_tag;
 
 	/* set the stream tag in the codec dai dma params  */
-	snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0);
+	else
+		snd_soc_dai_set_tdm_slot(codec_dai, 0, stream_tag, 0, 0);
 
 	p_params.s_fmt = snd_pcm_format_width(params_format(params));
 	p_params.ch = params_channels(params);
@@ -995,21 +1000,63 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
 	},
 },
 {
-	.name = "HD-Codec Pin",
+	.name = "Analog CPU DAI",
 	.ops = &skl_link_dai_ops,
 	.playback = {
-		.stream_name = "HD-Codec Tx",
-		.channels_min = HDA_STEREO,
-		.channels_max = HDA_STEREO,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		.stream_name = "Analog CPU Playback",
+		.channels_min = HDA_MONO,
+		.channels_max = HDA_MAX,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+			SNDRV_PCM_FMTBIT_S32_LE,
 	},
 	.capture = {
-		.stream_name = "HD-Codec Rx",
-		.channels_min = HDA_STEREO,
-		.channels_max = HDA_STEREO,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		.stream_name = "Analog CPU Capture",
+		.channels_min = HDA_MONO,
+		.channels_max = HDA_MAX,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+			SNDRV_PCM_FMTBIT_S32_LE,
+	},
+},
+{
+	.name = "Alt Analog CPU DAI",
+	.ops = &skl_link_dai_ops,
+	.playback = {
+		.stream_name = "Alt Analog CPU Playback",
+		.channels_min = HDA_MONO,
+		.channels_max = HDA_MAX,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+			SNDRV_PCM_FMTBIT_S32_LE,
+	},
+	.capture = {
+		.stream_name = "Alt Analog CPU Capture",
+		.channels_min = HDA_MONO,
+		.channels_max = HDA_MAX,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+			SNDRV_PCM_FMTBIT_S32_LE,
+	},
+},
+{
+	.name = "Digital CPU DAI",
+	.ops = &skl_link_dai_ops,
+	.playback = {
+		.stream_name = "Digital CPU Playback",
+		.channels_min = HDA_MONO,
+		.channels_max = HDA_MAX,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+			SNDRV_PCM_FMTBIT_S32_LE,
+	},
+	.capture = {
+		.stream_name = "Digital CPU Capture",
+		.channels_min = HDA_MONO,
+		.channels_max = HDA_MAX,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+			SNDRV_PCM_FMTBIT_S32_LE,
 	},
 },
 };
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 2620d77729c5..cf8848b779dc 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -898,11 +898,10 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w,
 			bc = (struct skl_algo_data *)sb->dobj.private;
 
 			if (bc->set_params == SKL_PARAM_BIND) {
-				params = kzalloc(bc->max, GFP_KERNEL);
+				params = kmemdup(bc->params, bc->max, GFP_KERNEL);
 				if (!params)
 					return -ENOMEM;
 
-				memcpy(params, bc->params, bc->max);
 				skl_fill_sink_instance_id(ctx, params, bc->max,
 								mconfig);
 
@@ -2461,6 +2460,7 @@ static int skl_tplg_get_token(struct device *dev,
 
 	case SKL_TKN_U8_CORE_ID:
 		mconfig->core_id = tkn_elem->value;
+		break;
 
 	case SKL_TKN_U8_MOD_TYPE:
 		mconfig->m_type = tkn_elem->value;
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 1d17be0f78a0..29225623b4b4 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -33,9 +33,11 @@
 #include <sound/hda_register.h>
 #include <sound/hdaudio.h>
 #include <sound/hda_i915.h>
+#include <sound/hda_codec.h>
 #include "skl.h"
 #include "skl-sst-dsp.h"
 #include "skl-sst-ipc.h"
+#include "../../../soc/codecs/hdac_hda.h"
 
 /*
  * initialize the PCI registers
@@ -472,6 +474,25 @@ static struct skl_ssp_clk skl_ssp_clks[] = {
 						{.name = "ssp5_sclkfs"},
 };
 
+static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl *skl,
+					struct snd_soc_acpi_mach *machines)
+{
+	struct hdac_bus *bus = skl_to_bus(skl);
+	struct snd_soc_acpi_mach *mach;
+
+	/* check if we have any codecs detected on bus */
+	if (bus->codec_mask == 0)
+		return NULL;
+
+	/* point to common table */
+	mach = snd_soc_acpi_intel_hda_machines;
+
+	/* all entries in the machine table use the same firmware */
+	mach->fw_filename = machines->fw_filename;
+
+	return mach;
+}
+
 static int skl_find_machine(struct skl *skl, void *driver_data)
 {
 	struct hdac_bus *bus = skl_to_bus(skl);
@@ -479,9 +500,13 @@ static int skl_find_machine(struct skl *skl, void *driver_data)
 	struct skl_machine_pdata *pdata;
 
 	mach = snd_soc_acpi_find_machine(mach);
-	if (mach == NULL) {
-		dev_err(bus->dev, "No matching machine driver found\n");
-		return -ENODEV;
+	if (!mach) {
+		dev_dbg(bus->dev, "No matching I2S machine driver found\n");
+		mach = skl_find_hda_machine(skl, driver_data);
+		if (!mach) {
+			dev_err(bus->dev, "No matching machine driver found\n");
+			return -ENODEV;
+		}
 	}
 
 	skl->mach = mach;
@@ -498,8 +523,9 @@ static int skl_find_machine(struct skl *skl, void *driver_data)
 
 static int skl_machine_device_register(struct skl *skl)
 {
-	struct hdac_bus *bus = skl_to_bus(skl);
 	struct snd_soc_acpi_mach *mach = skl->mach;
+	struct hdac_bus *bus = skl_to_bus(skl);
+	struct skl_machine_pdata *pdata;
 	struct platform_device *pdev;
 	int ret;
 
@@ -516,8 +542,12 @@ static int skl_machine_device_register(struct skl *skl)
 		return -EIO;
 	}
 
-	if (mach->pdata)
+	if (mach->pdata) {
+		pdata = (struct skl_machine_pdata *)mach->pdata;
+		pdata->platform = dev_name(bus->dev);
+		pdata->codec_mask = bus->codec_mask;
 		dev_set_drvdata(&pdev->dev, mach->pdata);
+	}
 
 	skl->i2s_dev = pdev;
 
@@ -628,6 +658,24 @@ static void skl_clock_device_unregister(struct skl *skl)
 		platform_device_unregister(skl->clk_dev);
 }
 
+#define IDISP_INTEL_VENDOR_ID	0x80860000
+
+/*
+ * load the legacy codec driver
+ */
+static void load_codec_module(struct hda_codec *codec)
+{
+#ifdef MODULE
+	char modalias[MODULE_NAME_LEN];
+	const char *mod = NULL;
+
+	snd_hdac_codec_modalias(&codec->core, modalias, sizeof(modalias));
+	mod = modalias;
+	dev_dbg(&codec->core.dev, "loading %s codec module\n", mod);
+	request_module(mod);
+#endif
+}
+
 /*
  * Probe the given codec address
  */
@@ -637,7 +685,9 @@ static int probe_codec(struct hdac_bus *bus, int addr)
 		(AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
 	unsigned int res = -1;
 	struct skl *skl = bus_to_skl(bus);
+	struct hdac_hda_priv *hda_codec;
 	struct hdac_device *hdev;
+	int err;
 
 	mutex_lock(&bus->cmd_mutex);
 	snd_hdac_bus_send_cmd(bus, cmd);
@@ -645,13 +695,26 @@ static int probe_codec(struct hdac_bus *bus, int addr)
 	mutex_unlock(&bus->cmd_mutex);
 	if (res == -1)
 		return -EIO;
-	dev_dbg(bus->dev, "codec #%d probed OK\n", addr);
+	dev_dbg(bus->dev, "codec #%d probed OK: %x\n", addr, res);
 
-	hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL);
-	if (!hdev)
+	hda_codec = devm_kzalloc(&skl->pci->dev, sizeof(*hda_codec),
+				 GFP_KERNEL);
+	if (!hda_codec)
 		return -ENOMEM;
 
-	return snd_hdac_ext_bus_device_init(bus, addr, hdev);
+	hda_codec->codec.bus = skl_to_hbus(skl);
+	hdev = &hda_codec->codec.core;
+
+	err = snd_hdac_ext_bus_device_init(bus, addr, hdev);
+	if (err < 0)
+		return err;
+
+	/* use legacy bus only for HDA codecs, idisp uses ext bus */
+	if ((res & 0xFFFF0000) != IDISP_INTEL_VENDOR_ID) {
+		hdev->type = HDA_DEV_LEGACY;
+		load_codec_module(&hda_codec->codec);
+	}
+	return 0;
 }
 
 /* Codec initialization */
@@ -786,9 +849,10 @@ static int skl_create(struct pci_dev *pci,
 		      const struct hdac_io_ops *io_ops,
 		      struct skl **rskl)
 {
+	struct hdac_ext_bus_ops *ext_ops = NULL;
 	struct skl *skl;
 	struct hdac_bus *bus;
-
+	struct hda_bus *hbus;
 	int err;
 
 	*rskl = NULL;
@@ -803,13 +867,23 @@ static int skl_create(struct pci_dev *pci,
 		return -ENOMEM;
 	}
 
+	hbus = skl_to_hbus(skl);
 	bus = skl_to_bus(skl);
-	snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, NULL);
+
+#if IS_ENABLED(CONFIG_SND_SOC_HDAC_HDA)
+	ext_ops = snd_soc_hdac_hda_get_ops();
+#endif
+	snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, ext_ops);
 	bus->use_posbuf = 1;
 	skl->pci = pci;
 	INIT_WORK(&skl->probe_work, skl_probe_work);
 	bus->bdl_pos_adj = 0;
 
+	mutex_init(&hbus->prepare_mutex);
+	hbus->pci = pci;
+	hbus->mixer_assigned = -1;
+	hbus->modelname = "sklbus";
+
 	*rskl = skl;
 
 	return 0;
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index 78aa8bdcb619..8d48cd7c56c8 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -23,6 +23,7 @@
 
 #include <sound/hda_register.h>
 #include <sound/hdaudio_ext.h>
+#include <sound/hda_codec.h>
 #include <sound/soc.h>
 #include "skl-nhlt.h"
 #include "skl-ssp-clk.h"
@@ -71,7 +72,7 @@ struct skl_fw_config {
 };
 
 struct skl {
-	struct hdac_bus hbus;
+	struct hda_bus hbus;
 	struct pci_dev *pci;
 
 	unsigned int init_done:1; /* delayed init status */
@@ -105,8 +106,11 @@ struct skl {
 	struct snd_soc_acpi_mach *mach;
 };
 
-#define skl_to_bus(s)  (&(s)->hbus)
-#define bus_to_skl(bus) container_of(bus, struct skl, hbus)
+#define skl_to_bus(s)  (&(s)->hbus.core)
+#define bus_to_skl(bus) container_of(bus, struct skl, hbus.core)
+
+#define skl_to_hbus(s) (&(s)->hbus)
+#define hbus_to_skl(hbus) container_of((hbus), struct skl, (hbus))
 
 /* to pass dai dma data */
 struct skl_dma_params {
@@ -117,6 +121,8 @@ struct skl_dma_params {
 struct skl_machine_pdata {
 	u32 dmic_num;
 	bool use_tplg_pcm; /* use dais and dai links from topology */
+	const char *platform;
+	u32 codec_mask;
 };
 
 struct skl_dsp_ops {
diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c
index 666282b865a8..97f9f38ce6b3 100644
--- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c
+++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c
@@ -299,6 +299,7 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev)
 		devm_kzalloc(&pdev->dev, sizeof(struct mt2701_cs42448_private),
 			     GFP_KERNEL);
 	struct device *dev = &pdev->dev;
+	struct snd_soc_dai_link *dai_link;
 
 	if (!priv)
 		return -ENOMEM;
@@ -309,10 +310,10 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt2701_cs42448_dai_links[i].platform_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->platform_name)
 			continue;
-		mt2701_cs42448_dai_links[i].platform_of_node = platform_node;
+		dai_link->platform_of_node = platform_node;
 	}
 
 	card->dev = dev;
@@ -324,10 +325,10 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev)
 			"Property 'audio-codec' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt2701_cs42448_dai_links[i].codec_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->codec_name)
 			continue;
-		mt2701_cs42448_dai_links[i].codec_of_node = codec_node;
+		dai_link->codec_of_node = codec_node;
 	}
 
 	codec_node_bt_mrg = of_parse_phandle(pdev->dev.of_node,
diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c
index 89f34efd9747..6bc1d3d58e64 100644
--- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c
+++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c
@@ -97,6 +97,7 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &mt2701_wm8960_card;
 	struct device_node *platform_node, *codec_node;
+	struct snd_soc_dai_link *dai_link;
 	int ret, i;
 
 	platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -105,10 +106,10 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt2701_wm8960_dai_links[i].platform_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->platform_name)
 			continue;
-		mt2701_wm8960_dai_links[i].platform_of_node = platform_node;
+		dai_link->platform_of_node = platform_node;
 	}
 
 	card->dev = &pdev->dev;
@@ -120,10 +121,10 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev)
 			"Property 'audio-codec' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt2701_wm8960_dai_links[i].codec_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->codec_name)
 			continue;
-		mt2701_wm8960_dai_links[i].codec_of_node = codec_node;
+		dai_link->codec_of_node = codec_node;
 	}
 
 	ret = snd_soc_of_parse_audio_routing(card, "audio-routing");
@@ -150,7 +151,6 @@ static const struct of_device_id mt2701_wm8960_machine_dt_match[] = {
 static struct platform_driver mt2701_wm8960_machine = {
 	.driver = {
 		.name = "mt2701-wm8960",
-		.owner = THIS_MODULE,
 #ifdef CONFIG_OF
 		.of_match_table = mt2701_wm8960_machine_dt_match,
 #endif
diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c
index b1558c57b9ca..cc41eb531653 100644
--- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c
+++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c
@@ -158,6 +158,7 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &mt6797_mt6351_card;
 	struct device_node *platform_node, *codec_node;
+	struct snd_soc_dai_link *dai_link;
 	int ret, i;
 
 	card->dev = &pdev->dev;
@@ -168,10 +169,10 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt6797_mt6351_dai_links[i].platform_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->platform_name)
 			continue;
-		mt6797_mt6351_dai_links[i].platform_of_node = platform_node;
+		dai_link->platform_of_node = platform_node;
 	}
 
 	codec_node = of_parse_phandle(pdev->dev.of_node,
@@ -181,10 +182,10 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev)
 			"Property 'audio-codec' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt6797_mt6351_dai_links[i].codec_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->codec_name)
 			continue;
-		mt6797_mt6351_dai_links[i].codec_of_node = codec_node;
+		dai_link->codec_of_node = codec_node;
 	}
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
@@ -205,7 +206,6 @@ static const struct of_device_id mt6797_mt6351_dt_match[] = {
 static struct platform_driver mt6797_mt6351_driver = {
 	.driver = {
 		.name = "mt6797-mt6351",
-		.owner = THIS_MODULE,
 #ifdef CONFIG_OF
 		.of_match_table = mt6797_mt6351_dt_match,
 #endif
diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c
index 902d111016d6..4d6596d5cb07 100644
--- a/sound/soc/mediatek/mt8173/mt8173-max98090.c
+++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c
@@ -137,6 +137,7 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &mt8173_max98090_card;
 	struct device_node *codec_node, *platform_node;
+	struct snd_soc_dai_link *dai_link;
 	int ret, i;
 
 	platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -145,10 +146,10 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt8173_max98090_dais[i].platform_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->platform_name)
 			continue;
-		mt8173_max98090_dais[i].platform_of_node = platform_node;
+		dai_link->platform_of_node = platform_node;
 	}
 
 	codec_node = of_parse_phandle(pdev->dev.of_node,
@@ -158,10 +159,10 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
 			"Property 'audio-codec' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt8173_max98090_dais[i].codec_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->codec_name)
 			continue;
-		mt8173_max98090_dais[i].codec_of_node = codec_node;
+		dai_link->codec_of_node = codec_node;
 	}
 	card->dev = &pdev->dev;
 
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
index 582174d98c6c..da5b58ce791b 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
@@ -44,11 +44,10 @@ static int mt8173_rt5650_rt5514_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;
 	int i, ret;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
-
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		/* pll from mclk 12.288M */
 		ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS,
 					  params_rate(params) * 512);
@@ -179,6 +178,7 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &mt8173_rt5650_rt5514_card;
 	struct device_node *platform_node;
+	struct snd_soc_dai_link *dai_link;
 	int i, ret;
 
 	platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -188,10 +188,10 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev)
 		return -EINVAL;
 	}
 
-	for (i = 0; i < card->num_links; i++) {
-		if (mt8173_rt5650_rt5514_dais[i].platform_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->platform_name)
 			continue;
-		mt8173_rt5650_rt5514_dais[i].platform_of_node = platform_node;
+		dai_link->platform_of_node = platform_node;
 	}
 
 	mt8173_rt5650_rt5514_codecs[0].of_node =
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
index b3670c8a5b8d..d83cd039b413 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
@@ -48,11 +48,10 @@ static int mt8173_rt5650_rt5676_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;
 	int i, ret;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
-
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		/* pll from mclk 12.288M */
 		ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS,
 					  params_rate(params) * 512);
@@ -225,6 +224,7 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &mt8173_rt5650_rt5676_card;
 	struct device_node *platform_node;
+	struct snd_soc_dai_link *dai_link;
 	int i, ret;
 
 	platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -234,10 +234,10 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
 		return -EINVAL;
 	}
 
-	for (i = 0; i < card->num_links; i++) {
-		if (mt8173_rt5650_rt5676_dais[i].platform_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->platform_name)
 			continue;
-		mt8173_rt5650_rt5676_dais[i].platform_of_node = platform_node;
+		dai_link->platform_of_node = platform_node;
 	}
 
 	mt8173_rt5650_rt5676_codecs[0].of_node =
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
index 7a89b4aad182..7edf250c8fb1 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
@@ -59,6 +59,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	unsigned int mclk_clock;
+	struct snd_soc_dai *codec_dai;
 	int i, ret;
 
 	switch (mt8173_rt5650_priv.pll_from) {
@@ -76,9 +77,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream,
 		break;
 	}
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
-
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		/* pll from mclk */
 		ret = snd_soc_dai_set_pll(codec_dai, 0, 0, mclk_clock,
 					  params_rate(params) * 512);
@@ -240,6 +239,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
 	struct device_node *platform_node;
 	struct device_node *np;
 	const char *codec_capture_dai;
+	struct snd_soc_dai_link *dai_link;
 	int i, ret;
 
 	platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -249,10 +249,10 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
 		return -EINVAL;
 	}
 
-	for (i = 0; i < card->num_links; i++) {
-		if (mt8173_rt5650_dais[i].platform_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->platform_name)
 			continue;
-		mt8173_rt5650_dais[i].platform_of_node = platform_node;
+		dai_link->platform_of_node = platform_node;
 	}
 
 	mt8173_rt5650_codecs[0].of_node =
diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig
index 8af8bc358a90..8b8426ed2363 100644
--- a/sound/soc/meson/Kconfig
+++ b/sound/soc/meson/Kconfig
@@ -4,6 +4,8 @@ menu "ASoC support for Amlogic platforms"
 config SND_MESON_AXG_FIFO
 	tristate
 	select REGMAP_MMIO
+	imply COMMON_CLK_AXG_AUDIO
+	imply RESET_MESON_AUDIO_ARB
 
 config SND_MESON_AXG_FRDDR
 	tristate "Amlogic AXG Playback FIFO support"
@@ -22,6 +24,7 @@ config SND_MESON_AXG_TODDR
 config SND_MESON_AXG_TDM_FORMATTER
 	tristate
 	select REGMAP_MMIO
+	imply COMMON_CLK_AXG_AUDIO
 
 config SND_MESON_AXG_TDM_INTERFACE
 	tristate
@@ -51,6 +54,7 @@ config SND_MESON_AXG_SOUND_CARD
 	imply SND_MESON_AXG_TDMIN
 	imply SND_MESON_AXG_TDMOUT
 	imply SND_MESON_AXG_SPDIFOUT
+	imply SND_MESON_AXG_PDM
 	help
 	  Select Y or M to add support for the AXG SoC sound card
 
@@ -58,8 +62,17 @@ config SND_MESON_AXG_SPDIFOUT
 	tristate "Amlogic AXG SPDIF Output Support"
 	select SND_PCM_IEC958
 	imply SND_SOC_SPDIF
+	imply COMMON_CLK_AXG_AUDIO
 	help
 	  Select Y or M to add support for SPDIF output serializer embedded
 	  in the Amlogic AXG SoC family
 
+config SND_MESON_AXG_PDM
+	tristate "Amlogic AXG PDM Input Support"
+	imply SND_SOC_DMIC
+	imply COMMON_CLK_AXG_AUDIO
+	help
+	  Select Y or M to add support for PDM input embedded
+	  in the Amlogic AXG SoC family
+
 endmenu
diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile
index c5e003b093db..4cd25104029d 100644
--- a/sound/soc/meson/Makefile
+++ b/sound/soc/meson/Makefile
@@ -9,6 +9,7 @@ snd-soc-meson-axg-tdmin-objs := axg-tdmin.o
 snd-soc-meson-axg-tdmout-objs := axg-tdmout.o
 snd-soc-meson-axg-sound-card-objs := axg-card.o
 snd-soc-meson-axg-spdifout-objs := axg-spdifout.o
+snd-soc-meson-axg-pdm-objs := axg-pdm.o
 
 obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o
 obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o
@@ -19,3 +20,4 @@ obj-$(CONFIG_SND_MESON_AXG_TDMIN) += snd-soc-meson-axg-tdmin.o
 obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o
 obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o
 obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o
+obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 2914ba0d965b..aa54d2c612c9 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -97,14 +97,14 @@ static void axg_card_clean_references(struct axg_card *priv)
 {
 	struct snd_soc_card *card = &priv->card;
 	struct snd_soc_dai_link *link;
+	struct snd_soc_dai_link_component *codec;
 	int i, j;
 
 	if (card->dai_link) {
-		for (i = 0; i < card->num_links; i++) {
-			link = &card->dai_link[i];
+		for_each_card_prelinks(card, i, link) {
 			of_node_put(link->cpu_of_node);
-			for (j = 0; j < link->num_codecs; j++)
-				of_node_put(link->codecs[j].of_node);
+			for_each_link_codecs(link, j, codec)
+				of_node_put(codec->of_node);
 		}
 	}
 
@@ -167,8 +167,7 @@ static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream,
 	if (be->mclk_fs) {
 		mclk = params_rate(params) * be->mclk_fs;
 
-		for (i = 0; i < rtd->num_codecs; i++) {
-			codec_dai = rtd->codec_dais[i];
+		for_each_rtd_codec_dai(rtd, i, codec_dai) {
 			ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
 						     SND_SOC_CLOCK_IN);
 			if (ret && ret != -ENOTSUPP)
@@ -196,8 +195,7 @@ static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd)
 	struct snd_soc_dai *codec_dai;
 	int ret, i;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		ret = snd_soc_dai_set_tdm_slot(codec_dai,
 					       be->codec_masks[i].tx,
 					       be->codec_masks[i].rx,
@@ -478,7 +476,7 @@ static int axg_card_set_be_link(struct snd_soc_card *card,
 
 	ret = axg_card_set_link_name(card, link, "be");
 	if (ret)
-		dev_err(card->dev, "error setting %s link name\n", np->name);
+		dev_err(card->dev, "error setting %pOFn link name\n", np);
 
 	return ret;
 }
diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
index 30262550e37b..0e4f65e654c4 100644
--- a/sound/soc/meson/axg-fifo.c
+++ b/sound/soc/meson/axg-fifo.c
@@ -203,6 +203,8 @@ static int axg_fifo_pcm_open(struct snd_pcm_substream *ss)
 
 	ret = request_irq(fifo->irq, axg_fifo_pcm_irq_block, 0,
 			  dev_name(dev), ss);
+	if (ret)
+		return ret;
 
 	/* Enable pclk to access registers and clock the fifo ip */
 	ret = clk_prepare_enable(fifo->pclk);
diff --git a/sound/soc/meson/axg-pdm.c b/sound/soc/meson/axg-pdm.c
new file mode 100644
index 000000000000..9d5684493ffc
--- /dev/null
+++ b/sound/soc/meson/axg-pdm.c
@@ -0,0 +1,654 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_irq.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/pcm_params.h>
+
+#define PDM_CTRL			0x00
+#define  PDM_CTRL_EN			BIT(31)
+#define  PDM_CTRL_OUT_MODE		BIT(29)
+#define  PDM_CTRL_BYPASS_MODE		BIT(28)
+#define  PDM_CTRL_RST_FIFO		BIT(16)
+#define  PDM_CTRL_CHAN_RSTN_MASK	GENMASK(15, 8)
+#define  PDM_CTRL_CHAN_RSTN(x)		((x) << 8)
+#define  PDM_CTRL_CHAN_EN_MASK		GENMASK(7, 0)
+#define  PDM_CTRL_CHAN_EN(x)		((x) << 0)
+#define PDM_HCIC_CTRL1			0x04
+#define  PDM_FILTER_EN			BIT(31)
+#define  PDM_HCIC_CTRL1_GAIN_SFT_MASK	GENMASK(29, 24)
+#define  PDM_HCIC_CTRL1_GAIN_SFT(x)	((x) << 24)
+#define  PDM_HCIC_CTRL1_GAIN_MULT_MASK	GENMASK(23, 16)
+#define  PDM_HCIC_CTRL1_GAIN_MULT(x)	((x) << 16)
+#define  PDM_HCIC_CTRL1_DSR_MASK	GENMASK(8, 4)
+#define  PDM_HCIC_CTRL1_DSR(x)		((x) << 4)
+#define  PDM_HCIC_CTRL1_STAGE_NUM_MASK	GENMASK(3, 0)
+#define  PDM_HCIC_CTRL1_STAGE_NUM(x)	((x) << 0)
+#define PDM_HCIC_CTRL2			0x08
+#define PDM_F1_CTRL			0x0c
+#define  PDM_LPF_ROUND_MODE_MASK	GENMASK(17, 16)
+#define  PDM_LPF_ROUND_MODE(x)		((x) << 16)
+#define  PDM_LPF_DSR_MASK		GENMASK(15, 12)
+#define  PDM_LPF_DSR(x)			((x) << 12)
+#define  PDM_LPF_STAGE_NUM_MASK		GENMASK(8, 0)
+#define  PDM_LPF_STAGE_NUM(x)		((x) << 0)
+#define  PDM_LPF_MAX_STAGE		336
+#define  PDM_LPF_NUM			3
+#define PDM_F2_CTRL			0x10
+#define PDM_F3_CTRL			0x14
+#define PDM_HPF_CTRL			0x18
+#define  PDM_HPF_SFT_STEPS_MASK		GENMASK(20, 16)
+#define  PDM_HPF_SFT_STEPS(x)		((x) << 16)
+#define  PDM_HPF_OUT_FACTOR_MASK	GENMASK(15, 0)
+#define  PDM_HPF_OUT_FACTOR(x)		((x) << 0)
+#define PDM_CHAN_CTRL			0x1c
+#define  PDM_CHAN_CTRL_POINTER_WIDTH	8
+#define  PDM_CHAN_CTRL_POINTER_MAX	((1 << PDM_CHAN_CTRL_POINTER_WIDTH) - 1)
+#define  PDM_CHAN_CTRL_NUM		4
+#define PDM_CHAN_CTRL1			0x20
+#define PDM_COEFF_ADDR			0x24
+#define PDM_COEFF_DATA			0x28
+#define PDM_CLKG_CTRL			0x2c
+#define PDM_STS				0x30
+
+struct axg_pdm_lpf {
+	unsigned int ds;
+	unsigned int round_mode;
+	const unsigned int *tap;
+	unsigned int tap_num;
+};
+
+struct axg_pdm_hcic {
+	unsigned int shift;
+	unsigned int mult;
+	unsigned int steps;
+	unsigned int ds;
+};
+
+struct axg_pdm_hpf {
+	unsigned int out_factor;
+	unsigned int steps;
+};
+
+struct axg_pdm_filters {
+	struct axg_pdm_hcic hcic;
+	struct axg_pdm_hpf hpf;
+	struct axg_pdm_lpf lpf[PDM_LPF_NUM];
+};
+
+struct axg_pdm_cfg {
+	const struct axg_pdm_filters *filters;
+	unsigned int sys_rate;
+};
+
+struct axg_pdm {
+	const struct axg_pdm_cfg *cfg;
+	struct regmap *map;
+	struct clk *dclk;
+	struct clk *sysclk;
+	struct clk *pclk;
+};
+
+static void axg_pdm_enable(struct regmap *map)
+{
+	/* Reset AFIFO */
+	regmap_update_bits(map, PDM_CTRL, PDM_CTRL_RST_FIFO, PDM_CTRL_RST_FIFO);
+	regmap_update_bits(map, PDM_CTRL, PDM_CTRL_RST_FIFO, 0);
+
+	/* Enable PDM */
+	regmap_update_bits(map, PDM_CTRL, PDM_CTRL_EN, PDM_CTRL_EN);
+}
+
+static void axg_pdm_disable(struct regmap *map)
+{
+	regmap_update_bits(map, PDM_CTRL, PDM_CTRL_EN, 0);
+}
+
+static void axg_pdm_filters_enable(struct regmap *map, bool enable)
+{
+	unsigned int val = enable ? PDM_FILTER_EN : 0;
+
+	regmap_update_bits(map, PDM_HCIC_CTRL1, PDM_FILTER_EN, val);
+	regmap_update_bits(map, PDM_F1_CTRL, PDM_FILTER_EN, val);
+	regmap_update_bits(map, PDM_F2_CTRL, PDM_FILTER_EN, val);
+	regmap_update_bits(map, PDM_F3_CTRL, PDM_FILTER_EN, val);
+	regmap_update_bits(map, PDM_HPF_CTRL, PDM_FILTER_EN, val);
+}
+
+static int axg_pdm_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai)
+{
+	struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		axg_pdm_enable(priv->map);
+		return 0;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		axg_pdm_disable(priv->map);
+		return 0;
+
+	default:
+		return -EINVAL;
+	}
+}
+
+static unsigned int axg_pdm_get_os(struct axg_pdm *priv)
+{
+	const struct axg_pdm_filters *filters = priv->cfg->filters;
+	unsigned int os = filters->hcic.ds;
+	int i;
+
+	/*
+	 * The global oversampling factor is defined by the down sampling
+	 * factor applied by each filter (HCIC and LPFs)
+	 */
+
+	for (i = 0; i < PDM_LPF_NUM; i++)
+		os *= filters->lpf[i].ds;
+
+	return os;
+}
+
+static int axg_pdm_set_sysclk(struct axg_pdm *priv, unsigned int os,
+			      unsigned int rate)
+{
+	unsigned int sys_rate = os * 2 * rate * PDM_CHAN_CTRL_POINTER_MAX;
+
+	/*
+	 * Set the default system clock rate unless it is too fast for
+	 * for the requested sample rate. In this case, the sample pointer
+	 * counter could overflow so set a lower system clock rate
+	 */
+	if (sys_rate < priv->cfg->sys_rate)
+		return clk_set_rate(priv->sysclk, sys_rate);
+
+	return clk_set_rate(priv->sysclk, priv->cfg->sys_rate);
+}
+
+static int axg_pdm_set_sample_pointer(struct axg_pdm *priv)
+{
+	unsigned int spmax, sp, val;
+	int i;
+
+	/* Max sample counter value per half period of dclk */
+	spmax = DIV_ROUND_UP_ULL((u64)clk_get_rate(priv->sysclk),
+				 clk_get_rate(priv->dclk) * 2);
+
+	/* Check if sysclk is not too fast - should not happen */
+	if (WARN_ON(spmax > PDM_CHAN_CTRL_POINTER_MAX))
+		return -EINVAL;
+
+	/* Capture the data when we are at 75% of the half period */
+	sp = spmax * 3 / 4;
+
+	for (i = 0, val = 0; i < PDM_CHAN_CTRL_NUM; i++)
+		val |= sp << (PDM_CHAN_CTRL_POINTER_WIDTH * i);
+
+	regmap_write(priv->map, PDM_CHAN_CTRL, val);
+	regmap_write(priv->map, PDM_CHAN_CTRL1, val);
+
+	return 0;
+}
+
+static void axg_pdm_set_channel_mask(struct axg_pdm *priv,
+				     unsigned int channels)
+{
+	unsigned int mask = GENMASK(channels - 1, 0);
+
+	/* Put all channel in reset */
+	regmap_update_bits(priv->map, PDM_CTRL,
+			   PDM_CTRL_CHAN_RSTN_MASK, 0);
+
+	/* Take the necessary channels out of reset and enable them */
+	regmap_update_bits(priv->map, PDM_CTRL,
+			   PDM_CTRL_CHAN_RSTN_MASK |
+			   PDM_CTRL_CHAN_EN_MASK,
+			   PDM_CTRL_CHAN_RSTN(mask) |
+			   PDM_CTRL_CHAN_EN(mask));
+}
+
+static int axg_pdm_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
+{
+	struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+	unsigned int os = axg_pdm_get_os(priv);
+	unsigned int rate = params_rate(params);
+	unsigned int val;
+	int ret;
+
+	switch (params_width(params)) {
+	case 24:
+		val = PDM_CTRL_OUT_MODE;
+		break;
+	case 32:
+		val = 0;
+		break;
+	default:
+		dev_err(dai->dev, "unsupported sample width\n");
+		return -EINVAL;
+	}
+
+	regmap_update_bits(priv->map, PDM_CTRL, PDM_CTRL_OUT_MODE, val);
+
+	ret = axg_pdm_set_sysclk(priv, os, rate);
+	if (ret) {
+		dev_err(dai->dev, "failed to set system clock\n");
+		return ret;
+	}
+
+	ret = clk_set_rate(priv->dclk, rate * os);
+	if (ret) {
+		dev_err(dai->dev, "failed to set dclk\n");
+		return ret;
+	}
+
+	ret = axg_pdm_set_sample_pointer(priv);
+	if (ret) {
+		dev_err(dai->dev, "invalid clock setting\n");
+		return ret;
+	}
+
+	axg_pdm_set_channel_mask(priv, params_channels(params));
+
+	return 0;
+}
+
+static int axg_pdm_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+	int ret;
+
+	ret = clk_prepare_enable(priv->dclk);
+	if (ret) {
+		dev_err(dai->dev, "enabling dclk failed\n");
+		return ret;
+	}
+
+	/* Enable the filters */
+	axg_pdm_filters_enable(priv->map, true);
+
+	return ret;
+}
+
+static void axg_pdm_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+
+	axg_pdm_filters_enable(priv->map, false);
+	clk_disable_unprepare(priv->dclk);
+}
+
+static const struct snd_soc_dai_ops axg_pdm_dai_ops = {
+	.trigger	= axg_pdm_trigger,
+	.hw_params	= axg_pdm_hw_params,
+	.startup	= axg_pdm_startup,
+	.shutdown	= axg_pdm_shutdown,
+};
+
+static void axg_pdm_set_hcic_ctrl(struct axg_pdm *priv)
+{
+	const struct axg_pdm_hcic *hcic = &priv->cfg->filters->hcic;
+	unsigned int val;
+
+	val = PDM_HCIC_CTRL1_STAGE_NUM(hcic->steps);
+	val |= PDM_HCIC_CTRL1_DSR(hcic->ds);
+	val |= PDM_HCIC_CTRL1_GAIN_MULT(hcic->mult);
+	val |= PDM_HCIC_CTRL1_GAIN_SFT(hcic->shift);
+
+	regmap_update_bits(priv->map, PDM_HCIC_CTRL1,
+			   PDM_HCIC_CTRL1_STAGE_NUM_MASK |
+			   PDM_HCIC_CTRL1_DSR_MASK |
+			   PDM_HCIC_CTRL1_GAIN_MULT_MASK |
+			   PDM_HCIC_CTRL1_GAIN_SFT_MASK,
+			   val);
+}
+
+static void axg_pdm_set_lpf_ctrl(struct axg_pdm *priv, unsigned int index)
+{
+	const struct axg_pdm_lpf *lpf = &priv->cfg->filters->lpf[index];
+	unsigned int offset = index * regmap_get_reg_stride(priv->map)
+		+ PDM_F1_CTRL;
+	unsigned int val;
+
+	val = PDM_LPF_STAGE_NUM(lpf->tap_num);
+	val |= PDM_LPF_DSR(lpf->ds);
+	val |= PDM_LPF_ROUND_MODE(lpf->round_mode);
+
+	regmap_update_bits(priv->map, offset,
+			   PDM_LPF_STAGE_NUM_MASK |
+			   PDM_LPF_DSR_MASK |
+			   PDM_LPF_ROUND_MODE_MASK,
+			   val);
+}
+
+static void axg_pdm_set_hpf_ctrl(struct axg_pdm *priv)
+{
+	const struct axg_pdm_hpf *hpf = &priv->cfg->filters->hpf;
+	unsigned int val;
+
+	val = PDM_HPF_OUT_FACTOR(hpf->out_factor);
+	val |= PDM_HPF_SFT_STEPS(hpf->steps);
+
+	regmap_update_bits(priv->map, PDM_HPF_CTRL,
+			   PDM_HPF_OUT_FACTOR_MASK |
+			   PDM_HPF_SFT_STEPS_MASK,
+			   val);
+}
+
+static int axg_pdm_set_lpf_filters(struct axg_pdm *priv)
+{
+	const struct axg_pdm_lpf *lpf = priv->cfg->filters->lpf;
+	unsigned int count = 0;
+	int i, j;
+
+	for (i = 0; i < PDM_LPF_NUM; i++)
+		count += lpf[i].tap_num;
+
+	/* Make sure the coeffs fit in the memory */
+	if (count >= PDM_LPF_MAX_STAGE)
+		return -EINVAL;
+
+	/* Set the initial APB bus register address */
+	regmap_write(priv->map, PDM_COEFF_ADDR, 0);
+
+	/* Set the tap filter values of all 3 filters */
+	for (i = 0; i < PDM_LPF_NUM; i++) {
+		axg_pdm_set_lpf_ctrl(priv, i);
+
+		for (j = 0; j < lpf[i].tap_num; j++)
+			regmap_write(priv->map, PDM_COEFF_DATA, lpf[i].tap[j]);
+	}
+
+	return 0;
+}
+
+static int axg_pdm_dai_probe(struct snd_soc_dai *dai)
+{
+	struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+	int ret;
+
+	ret = clk_prepare_enable(priv->pclk);
+	if (ret) {
+		dev_err(dai->dev, "enabling pclk failed\n");
+		return ret;
+	}
+
+	/*
+	 * sysclk must be set and enabled as well to access the pdm registers
+	 * Accessing the register w/o it will give a bus error.
+	 */
+	ret = clk_set_rate(priv->sysclk, priv->cfg->sys_rate);
+	if (ret) {
+		dev_err(dai->dev, "setting sysclk failed\n");
+		goto err_pclk;
+	}
+
+	ret = clk_prepare_enable(priv->sysclk);
+	if (ret) {
+		dev_err(dai->dev, "enabling sysclk failed\n");
+		goto err_pclk;
+	}
+
+	/* Make sure the device is initially disabled */
+	axg_pdm_disable(priv->map);
+
+	/* Make sure filter bypass is disabled */
+	regmap_update_bits(priv->map, PDM_CTRL, PDM_CTRL_BYPASS_MODE, 0);
+
+	/* Load filter settings */
+	axg_pdm_set_hcic_ctrl(priv);
+	axg_pdm_set_hpf_ctrl(priv);
+
+	ret = axg_pdm_set_lpf_filters(priv);
+	if (ret) {
+		dev_err(dai->dev, "invalid filter configuration\n");
+		goto err_sysclk;
+	}
+
+	return 0;
+
+err_sysclk:
+	clk_disable_unprepare(priv->sysclk);
+err_pclk:
+	clk_disable_unprepare(priv->pclk);
+	return ret;
+}
+
+static int axg_pdm_dai_remove(struct snd_soc_dai *dai)
+{
+	struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+
+	clk_disable_unprepare(priv->sysclk);
+	clk_disable_unprepare(priv->pclk);
+
+	return 0;
+}
+
+static struct snd_soc_dai_driver axg_pdm_dai_drv = {
+	.name = "PDM",
+	.capture = {
+		.stream_name	= "Capture",
+		.channels_min	= 1,
+		.channels_max	= 8,
+		.rates		= SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min	= 5512,
+		.rate_max	= 48000,
+		.formats	= (SNDRV_PCM_FMTBIT_S24_LE |
+				   SNDRV_PCM_FMTBIT_S32_LE),
+	},
+	.ops		= &axg_pdm_dai_ops,
+	.probe		= axg_pdm_dai_probe,
+	.remove		= axg_pdm_dai_remove,
+};
+
+static const struct snd_soc_component_driver axg_pdm_component_drv = {};
+
+static const struct regmap_config axg_pdm_regmap_cfg = {
+	.reg_bits	= 32,
+	.val_bits	= 32,
+	.reg_stride	= 4,
+	.max_register	= PDM_STS,
+};
+
+static const unsigned int lpf1_default_tap[] = {
+	0x000014, 0xffffb2, 0xfffed9, 0xfffdce, 0xfffd45,
+	0xfffe32, 0x000147, 0x000645, 0x000b86, 0x000e21,
+	0x000ae3, 0x000000, 0xffeece, 0xffdca8, 0xffd212,
+	0xffd7d1, 0xfff2a7, 0x001f4c, 0x0050c2, 0x0072aa,
+	0x006ff1, 0x003c32, 0xffdc4e, 0xff6a18, 0xff0fef,
+	0xfefbaf, 0xff4c40, 0x000000, 0x00ebc8, 0x01c077,
+	0x02209e, 0x01c1a4, 0x008e60, 0xfebe52, 0xfcd690,
+	0xfb8fa5, 0xfba498, 0xfd9812, 0x0181ce, 0x06f5f3,
+	0x0d112f, 0x12a958, 0x169686, 0x18000e, 0x169686,
+	0x12a958, 0x0d112f, 0x06f5f3, 0x0181ce, 0xfd9812,
+	0xfba498, 0xfb8fa5, 0xfcd690, 0xfebe52, 0x008e60,
+	0x01c1a4, 0x02209e, 0x01c077, 0x00ebc8, 0x000000,
+	0xff4c40, 0xfefbaf, 0xff0fef, 0xff6a18, 0xffdc4e,
+	0x003c32, 0x006ff1, 0x0072aa, 0x0050c2, 0x001f4c,
+	0xfff2a7, 0xffd7d1, 0xffd212, 0xffdca8, 0xffeece,
+	0x000000, 0x000ae3, 0x000e21, 0x000b86, 0x000645,
+	0x000147, 0xfffe32, 0xfffd45, 0xfffdce, 0xfffed9,
+	0xffffb2, 0x000014,
+};
+
+static const unsigned int lpf2_default_tap[] = {
+	0x00050a, 0xfff004, 0x0002c1, 0x003c12, 0xffa818,
+	0xffc87d, 0x010aef, 0xff5223, 0xfebd93, 0x028f41,
+	0xff5c0e, 0xfc63f8, 0x055f81, 0x000000, 0xf478a0,
+	0x11c5e3, 0x2ea74d, 0x11c5e3, 0xf478a0, 0x000000,
+	0x055f81, 0xfc63f8, 0xff5c0e, 0x028f41, 0xfebd93,
+	0xff5223, 0x010aef, 0xffc87d, 0xffa818, 0x003c12,
+	0x0002c1, 0xfff004, 0x00050a,
+};
+
+static const unsigned int lpf3_default_tap[] = {
+	0x000000, 0x000081, 0x000000, 0xfffedb, 0x000000,
+	0x00022d, 0x000000, 0xfffc46, 0x000000, 0x0005f7,
+	0x000000, 0xfff6eb, 0x000000, 0x000d4e, 0x000000,
+	0xffed1e, 0x000000, 0x001a1c, 0x000000, 0xffdcb0,
+	0x000000, 0x002ede, 0x000000, 0xffc2d1, 0x000000,
+	0x004ebe, 0x000000, 0xff9beb, 0x000000, 0x007dd7,
+	0x000000, 0xff633a, 0x000000, 0x00c1d2, 0x000000,
+	0xff11d5, 0x000000, 0x012368, 0x000000, 0xfe9c45,
+	0x000000, 0x01b252, 0x000000, 0xfdebf6, 0x000000,
+	0x0290b8, 0x000000, 0xfcca0d, 0x000000, 0x041d7c,
+	0x000000, 0xfa8152, 0x000000, 0x07e9c6, 0x000000,
+	0xf28fb5, 0x000000, 0x28b216, 0x3fffde, 0x28b216,
+	0x000000, 0xf28fb5, 0x000000, 0x07e9c6, 0x000000,
+	0xfa8152, 0x000000, 0x041d7c, 0x000000, 0xfcca0d,
+	0x000000, 0x0290b8, 0x000000, 0xfdebf6, 0x000000,
+	0x01b252, 0x000000, 0xfe9c45, 0x000000, 0x012368,
+	0x000000, 0xff11d5, 0x000000, 0x00c1d2, 0x000000,
+	0xff633a, 0x000000, 0x007dd7, 0x000000, 0xff9beb,
+	0x000000, 0x004ebe, 0x000000, 0xffc2d1, 0x000000,
+	0x002ede, 0x000000, 0xffdcb0, 0x000000, 0x001a1c,
+	0x000000, 0xffed1e, 0x000000, 0x000d4e, 0x000000,
+	0xfff6eb, 0x000000, 0x0005f7, 0x000000, 0xfffc46,
+	0x000000, 0x00022d, 0x000000, 0xfffedb, 0x000000,
+	0x000081, 0x000000,
+};
+
+/*
+ * These values are sane defaults for the axg platform:
+ * - OS = 64
+ * - Latency = 38700 (?)
+ *
+ * TODO: There is a lot of different HCIC, LPFs and HPF configurations possible.
+ *       the configuration may depend on the dmic used by the platform, the
+ *       expected tradeoff between latency and quality, etc ... If/When other
+ *       settings are required, we should add a fw interface to this driver to
+ *       load new filter settings.
+ */
+static const struct axg_pdm_filters axg_default_filters = {
+	.hcic = {
+		.shift = 0x15,
+		.mult = 0x80,
+		.steps = 7,
+		.ds = 8,
+	},
+	.hpf = {
+		.out_factor = 0x8000,
+		.steps = 13,
+	},
+	.lpf = {
+		[0] = {
+			.ds = 2,
+			.round_mode = 1,
+			.tap = lpf1_default_tap,
+			.tap_num = ARRAY_SIZE(lpf1_default_tap),
+		},
+		[1] = {
+			.ds = 2,
+			.round_mode = 0,
+			.tap = lpf2_default_tap,
+			.tap_num = ARRAY_SIZE(lpf2_default_tap),
+		},
+		[2] = {
+			.ds = 2,
+			.round_mode = 1,
+			.tap = lpf3_default_tap,
+			.tap_num = ARRAY_SIZE(lpf3_default_tap)
+		},
+	},
+};
+
+static const struct axg_pdm_cfg axg_pdm_config = {
+	.filters = &axg_default_filters,
+	.sys_rate = 250000000,
+};
+
+static const struct of_device_id axg_pdm_of_match[] = {
+	{
+		.compatible = "amlogic,axg-pdm",
+		.data = &axg_pdm_config,
+	}, {}
+};
+MODULE_DEVICE_TABLE(of, axg_pdm_of_match);
+
+static int axg_pdm_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct axg_pdm *priv;
+	struct resource *res;
+	void __iomem *regs;
+	int ret;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, priv);
+
+	priv->cfg = of_device_get_match_data(dev);
+	if (!priv->cfg) {
+		dev_err(dev, "failed to match device\n");
+		return -ENODEV;
+	}
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	regs = devm_ioremap_resource(dev, res);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	priv->map = devm_regmap_init_mmio(dev, regs, &axg_pdm_regmap_cfg);
+	if (IS_ERR(priv->map)) {
+		dev_err(dev, "failed to init regmap: %ld\n",
+			PTR_ERR(priv->map));
+		return PTR_ERR(priv->map);
+	}
+
+	priv->pclk = devm_clk_get(dev, "pclk");
+	if (IS_ERR(priv->pclk)) {
+		ret = PTR_ERR(priv->pclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get pclk: %d\n", ret);
+		return ret;
+	}
+
+	priv->dclk = devm_clk_get(dev, "dclk");
+	if (IS_ERR(priv->dclk)) {
+		ret = PTR_ERR(priv->dclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get dclk: %d\n", ret);
+		return ret;
+	}
+
+	priv->sysclk = devm_clk_get(dev, "sysclk");
+	if (IS_ERR(priv->sysclk)) {
+		ret = PTR_ERR(priv->sysclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get dclk: %d\n", ret);
+		return ret;
+	}
+
+	return devm_snd_soc_register_component(dev, &axg_pdm_component_drv,
+					       &axg_pdm_dai_drv, 1);
+}
+
+static struct platform_driver axg_pdm_pdrv = {
+	.probe = axg_pdm_probe,
+	.driver = {
+		.name = "axg-pdm",
+		.of_match_table = axg_pdm_of_match,
+	},
+};
+module_platform_driver(axg_pdm_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG PDM Input driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
index 7b8baf46d968..585ce030b79b 100644
--- a/sound/soc/meson/axg-tdm-interface.c
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -42,6 +42,7 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
 	struct axg_tdm_stream *rx = (struct axg_tdm_stream *)
 		dai->capture_dma_data;
 	unsigned int tx_slots, rx_slots;
+	unsigned int fmt = 0;
 
 	tx_slots = axg_tdm_slots_total(tx_mask);
 	rx_slots = axg_tdm_slots_total(rx_mask);
@@ -52,38 +53,45 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
 		return -EINVAL;
 	}
 
-	/*
-	 * Amend the dai driver channel number and let dpcm channel merge do
-	 * its job
-	 */
-	if (tx) {
-		tx->mask = tx_mask;
-		dai->driver->playback.channels_max = tx_slots;
-	}
-
-	if (rx) {
-		rx->mask = rx_mask;
-		dai->driver->capture.channels_max = rx_slots;
-	}
-
 	iface->slots = slots;
 
 	switch (slot_width) {
 	case 0:
-		/* defaults width to 32 if not provided */
-		iface->slot_width = 32;
-		break;
-	case 8:
-	case 16:
-	case 24:
+		slot_width = 32;
+		/* Fall-through */
 	case 32:
-		iface->slot_width = slot_width;
+		fmt |= SNDRV_PCM_FMTBIT_S32_LE;
+		/* Fall-through */
+	case 24:
+		fmt |= SNDRV_PCM_FMTBIT_S24_LE;
+		fmt |= SNDRV_PCM_FMTBIT_S20_LE;
+		/* Fall-through */
+	case 16:
+		fmt |= SNDRV_PCM_FMTBIT_S16_LE;
+		/* Fall-through */
+	case 8:
+		fmt |= SNDRV_PCM_FMTBIT_S8;
 		break;
 	default:
 		dev_err(dai->dev, "unsupported slot width: %d\n", slot_width);
 		return -EINVAL;
 	}
 
+	iface->slot_width = slot_width;
+
+	/* Amend the dai driver and let dpcm merge do its job */
+	if (tx) {
+		tx->mask = tx_mask;
+		dai->driver->playback.channels_max = tx_slots;
+		dai->driver->playback.formats = fmt;
+	}
+
+	if (rx) {
+		rx->mask = rx_mask;
+		dai->driver->capture.channels_max = rx_slots;
+		dai->driver->capture.formats = fmt;
+	}
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(axg_tdm_set_tdm_slots);
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index 81b09d740ed9..6384bb6dacfd 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -356,7 +356,7 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev)
 	if (ret)
 		goto out;
 
-	ret = snd_soc_register_component(&pdev->dev, &nuc900_ac97_component,
+	ret = devm_snd_soc_register_component(&pdev->dev, &nuc900_ac97_component,
 					 &nuc900_ac97_dai, 1);
 	if (ret)
 		goto out;
@@ -373,8 +373,6 @@ out:
 
 static int nuc900_ac97_drvremove(struct platform_device *pdev)
 {
-	snd_soc_unregister_component(&pdev->dev);
-
 	nuc900_ac97_data = NULL;
 	snd_soc_set_ac97_ops(NULL);
 
diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c
index 8a99a8837dc9..673a9eb153b2 100644
--- a/sound/soc/omap/omap-hdmi-audio.c
+++ b/sound/soc/omap/omap-hdmi-audio.c
@@ -348,7 +348,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
 	default:
 		return -EINVAL;
 	}
-	ret = snd_soc_register_component(ad->dssdev, &omap_hdmi_component,
+	ret = devm_snd_soc_register_component(ad->dssdev, &omap_hdmi_component,
 					 dai_drv, 1);
 	if (ret)
 		return ret;
@@ -383,7 +383,6 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
 	ret = snd_soc_register_card(card);
 	if (ret) {
 		dev_err(dev, "snd_soc_register_card failed (%d)\n", ret);
-		snd_soc_unregister_component(ad->dssdev);
 		return ret;
 	}
 
@@ -400,7 +399,6 @@ static int omap_hdmi_audio_remove(struct platform_device *pdev)
 	struct hdmi_audio_data *ad = platform_get_drvdata(pdev);
 
 	snd_soc_unregister_card(ad->card);
-	snd_soc_unregister_component(ad->dssdev);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 776e148b0aa2..943b44de1464 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -19,14 +19,13 @@ config SND_MMP_SOC
 
 config SND_PXA2XX_AC97
 	tristate
-	select SND_AC97_CODEC
 
 config SND_PXA2XX_SOC_AC97
 	tristate
-	select AC97_BUS
+	select AC97_BUS_NEW
 	select SND_PXA2XX_LIB
 	select SND_PXA2XX_LIB_AC97
-	select SND_SOC_AC97_BUS
+	select SND_SOC_AC97_BUS_NEW
 
 config SND_PXA2XX_SOC_I2S
 	select SND_PXA2XX_LIB
@@ -80,6 +79,7 @@ config SND_PXA2XX_SOC_TOSA
 	tristate "SoC AC97 Audio support for Tosa"
 	depends on SND_PXA2XX_SOC && MACH_TOSA
 	depends on MFD_TC6393XB
+	depends on !AC97_BUS
 	select SND_PXA2XX_SOC_AC97
 	select SND_SOC_WM9712
 	help
@@ -89,6 +89,7 @@ config SND_PXA2XX_SOC_TOSA
 config SND_PXA2XX_SOC_E740
 	tristate "SoC AC97 Audio support for e740"
 	depends on SND_PXA2XX_SOC && MACH_E740
+	depends on !AC97_BUS
 	select SND_SOC_WM9705
 	select SND_PXA2XX_SOC_AC97
 	help
@@ -98,6 +99,7 @@ config SND_PXA2XX_SOC_E740
 config SND_PXA2XX_SOC_E750
 	tristate "SoC AC97 Audio support for e750"
 	depends on SND_PXA2XX_SOC && MACH_E750
+	depends on !AC97_BUS
 	select SND_SOC_WM9705
 	select SND_PXA2XX_SOC_AC97
 	help
@@ -107,6 +109,7 @@ config SND_PXA2XX_SOC_E750
 config SND_PXA2XX_SOC_E800
 	tristate "SoC AC97 Audio support for e800"
 	depends on SND_PXA2XX_SOC && MACH_E800
+	depends on !AC97_BUS
 	select SND_SOC_WM9712
 	select SND_PXA2XX_SOC_AC97
 	help
@@ -117,6 +120,7 @@ config SND_PXA2XX_SOC_EM_X270
 	tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300"
 	depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \
 			MACH_CM_X300)
+	depends on !AC97_BUS
 	select SND_PXA2XX_SOC_AC97
 	select SND_SOC_WM9712
 	help
@@ -127,6 +131,7 @@ config SND_PXA2XX_SOC_PALM27X
 	bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive"
 	depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \
 			MACH_PALMT5 || MACH_PALMTE2)
+	depends on !AC97_BUS
 	select SND_PXA2XX_SOC_AC97
 	select SND_SOC_WM9712
 	help
@@ -156,6 +161,7 @@ config SND_SOC_TTC_DKB
 config SND_SOC_ZYLONITE
 	tristate "SoC Audio support for Marvell Zylonite"
 	depends on SND_PXA2XX_SOC && MACH_ZYLONITE
+	depends on !AC97_BUS
 	select SND_PXA2XX_SOC_AC97
 	select SND_PXA_SOC_SSP
 	select SND_SOC_WM9713
@@ -195,6 +201,7 @@ config SND_PXA2XX_SOC_MAGICIAN
 config SND_PXA2XX_SOC_MIOA701
         tristate "SoC Audio support for MIO A701"
         depends on SND_PXA2XX_SOC && MACH_MIOA701
+	depends on !AC97_BUS
         select SND_PXA2XX_SOC_AC97
         select SND_SOC_WM9713
         help
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 69033e1a84e6..adcf8ba9d287 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -103,6 +103,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 		pxa_ssp_disable(ssp);
 	}
 
+	if (priv->extclk)
+		clk_prepare_enable(priv->extclk);
+
 	dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL);
 	if (!dma)
 		return -ENOMEM;
@@ -125,6 +128,9 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
 		clk_disable_unprepare(ssp->clk);
 	}
 
+	if (priv->extclk)
+		clk_disable_unprepare(priv->extclk);
+
 	kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
 	snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
 }
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 9f779657bc86..f8a3aa6c6d4e 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -17,6 +17,7 @@
 #include <linux/dmaengine.h>
 #include <linux/dma/pxa-dma.h>
 
+#include <sound/ac97/controller.h>
 #include <sound/core.h>
 #include <sound/ac97_codec.h>
 #include <sound/soc.h>
@@ -27,43 +28,35 @@
 #include <mach/regs-ac97.h>
 #include <mach/audio.h>
 
-static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
+static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv)
 {
 	pxa2xx_ac97_try_warm_reset();
 
 	pxa2xx_ac97_finish_reset();
 }
 
-static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97)
+static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv)
 {
 	pxa2xx_ac97_try_cold_reset();
 
 	pxa2xx_ac97_finish_reset();
 }
 
-static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97,
-					      unsigned short reg)
+static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot,
+				  unsigned short reg)
 {
-	int ret;
-
-	ret = pxa2xx_ac97_read(ac97->num, reg);
-	if (ret < 0)
-		return 0;
-	else
-		return (unsigned short)(ret & 0xffff);
+	return pxa2xx_ac97_read(slot, reg);
 }
 
-static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97,
-				     unsigned short reg, unsigned short val)
+static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot,
+				   unsigned short reg, unsigned short val)
 {
-	int ret;
-
-	ret = pxa2xx_ac97_write(ac97->num, reg, val);
+	return pxa2xx_ac97_write(slot, reg, val);
 }
 
-static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
-	.read	= pxa2xx_ac97_legacy_read,
-	.write	= pxa2xx_ac97_legacy_write,
+static struct ac97_controller_ops pxa2xx_ac97_ops = {
+	.read	= pxa2xx_ac97_read_actrl,
+	.write	= pxa2xx_ac97_write_actrl,
 	.warm_reset	= pxa2xx_ac97_warm_reset,
 	.reset	= pxa2xx_ac97_cold_reset,
 };
@@ -233,6 +226,9 @@ MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids);
 static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
 {
 	int ret;
+	struct ac97_controller *ctrl;
+	pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data;
+	void **codecs_pdata;
 
 	if (pdev->id != -1) {
 		dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n");
@@ -245,10 +241,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops);
-	if (ret != 0)
-		return ret;
+	codecs_pdata = pdata ? pdata->codec_pdata : NULL;
+	ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev,
+					    AC97_SLOTS_AVAILABLE_ALL,
+					    codecs_pdata);
+	if (IS_ERR(ctrl))
+		return PTR_ERR(ctrl);
 
+	platform_set_drvdata(pdev, ctrl);
 	/* Punt most of the init to the SoC probe; we may need the machine
 	 * driver to do interesting things with the clocking to get us up
 	 * and running.
@@ -259,8 +259,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
 
 static int pxa2xx_ac97_dev_remove(struct platform_device *pdev)
 {
+	struct ac97_controller *ctrl = platform_get_drvdata(pdev);
+
 	snd_soc_unregister_component(&pdev->dev);
-	snd_soc_set_ac97_ops(NULL);
+	snd_ac97_controller_unregister(ctrl);
 	pxa2xx_ac97_hw_remove(pdev);
 	return 0;
 }
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index 1543e85629f8..fb45f396ab4a 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -25,13 +25,12 @@ static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 
 static void apq8096_add_be_ops(struct snd_soc_card *card)
 {
-	struct snd_soc_dai_link *link = card->dai_link;
-	int i, num_links = card->num_links;
+	struct snd_soc_dai_link *link;
+	int i;
 
-	for (i = 0; i < num_links; i++) {
+	for_each_card_prelinks(card, i, link) {
 		if (link->no_pcm == 1)
 			link->be_hw_params_fixup = apq8096_be_hw_params_fixup;
-		link++;
 	}
 }
 
diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c
index 932c3ebfd252..da242515e146 100644
--- a/sound/soc/qcom/qdsp6/q6adm.c
+++ b/sound/soc/qcom/qdsp6/q6adm.c
@@ -2,25 +2,24 @@
 // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
 // Copyright (c) 2018, Linaro Limited
 
-#include <linux/slab.h>
-#include <linux/wait.h>
-#include <linux/kernel.h>
 #include <linux/device.h>
-#include <linux/module.h>
-#include <linux/sched.h>
 #include <linux/jiffies.h>
+#include <linux/kernel.h>
+#include <linux/kref.h>
+#include <linux/module.h>
 #include <linux/of.h>
 #include <linux/of_platform.h>
-#include <linux/kref.h>
-#include <linux/wait.h>
-#include <linux/soc/qcom/apr.h>
 #include <linux/platform_device.h>
+#include <linux/sched.h>
+#include <linux/slab.h>
+#include <linux/soc/qcom/apr.h>
+#include <linux/wait.h>
 #include <sound/asound.h>
 #include "q6adm.h"
 #include "q6afe.h"
 #include "q6core.h"
-#include "q6dsp-errno.h"
 #include "q6dsp-common.h"
+#include "q6dsp-errno.h"
 
 #define ADM_CMD_DEVICE_OPEN_V5		0x00010326
 #define ADM_CMDRSP_DEVICE_OPEN_V5	0x00010329
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 9db9a2944ef2..a16c71c03058 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -8,7 +8,6 @@
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <sound/soc.h>
-#include <sound/soc.h>
 #include <sound/soc-dapm.h>
 #include <sound/pcm.h>
 #include <asm/dma.h>
@@ -319,10 +318,11 @@ static int q6asm_dai_open(struct snd_pcm_substream *substream)
 	prtd->audio_client = q6asm_audio_client_alloc(dev,
 				(q6asm_cb)event_handler, prtd, stream_id,
 				LEGACY_PCM_MODE);
-	if (!prtd->audio_client) {
+	if (IS_ERR(prtd->audio_client)) {
 		pr_info("%s: Could not allocate memory\n", __func__);
+		ret = PTR_ERR(prtd->audio_client);
 		kfree(prtd);
-		return -ENOMEM;
+		return ret;
 	}
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -493,7 +493,7 @@ static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)
 		}
 	}
 
-	return ret;
+	return 0;
 }
 
 static void q6asm_dai_pcm_free(struct snd_pcm *pcm)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 2b2c7233bb5f..e1cfa846a1dc 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -11,7 +11,6 @@
 #include <linux/spinlock.h>
 #include <linux/kref.h>
 #include <linux/of.h>
-#include <linux/of_platform.h>
 #include <uapi/sound/asound.h>
 #include <linux/delay.h>
 #include <linux/slab.h>
diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c
index 06f03a5fe9bd..cdfc8ab6cfc0 100644
--- a/sound/soc/qcom/qdsp6/q6core.c
+++ b/sound/soc/qcom/qdsp6/q6core.c
@@ -10,7 +10,6 @@
 #include <linux/of.h>
 #include <linux/of_platform.h>
 #include <linux/jiffies.h>
-#include <linux/wait.h>
 #include <linux/soc/qcom/apr.h>
 #include "q6core.h"
 #include "q6dsp-errno.h"
@@ -105,12 +104,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data)
 		bytes = sizeof(*fwk) + fwk->num_services *
 				sizeof(fwk->svc_api_info[0]);
 
-		core->fwk_version = kzalloc(bytes, GFP_ATOMIC);
+		core->fwk_version = kmemdup(data->payload, bytes, GFP_ATOMIC);
 		if (!core->fwk_version)
 			return -ENOMEM;
 
-		memcpy(core->fwk_version, data->payload, bytes);
-
 		core->fwk_version_supported = true;
 		core->resp_received = true;
 
@@ -124,12 +121,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data)
 
 		len = sizeof(*v) + v->num_services * sizeof(v->svc_api_info[0]);
 
-		core->svc_version = kzalloc(len, GFP_ATOMIC);
+		core->svc_version = kmemdup(data->payload, len, GFP_ATOMIC);
 		if (!core->svc_version)
 			return -ENOMEM;
 
-		memcpy(core->svc_version, data->payload, len);
-
 		core->get_version_supported = true;
 		core->resp_received = true;
 
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 2a781d87ee65..9effbecc571f 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -195,15 +195,14 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 
 static void sdm845_add_be_ops(struct snd_soc_card *card)
 {
-	struct snd_soc_dai_link *link = card->dai_link;
-	int i, num_links = card->num_links;
+	struct snd_soc_dai_link *link;
+	int i;
 
-	for (i = 0; i < num_links; i++) {
+	for_each_card_prelinks(card, i, link) {
 		if (link->no_pcm == 1) {
 			link->ops = &sdm845_be_ops;
 			link->be_hw_params_fixup = sdm845_be_hw_params_fixup;
 		}
-		link++;
 	}
 }
 
diff --git a/sound/soc/rockchip/rk3288_hdmi_analog.c b/sound/soc/rockchip/rk3288_hdmi_analog.c
index 929b3fe289b0..a472d5eb2950 100644
--- a/sound/soc/rockchip/rk3288_hdmi_analog.c
+++ b/sound/soc/rockchip/rk3288_hdmi_analog.c
@@ -286,7 +286,6 @@ static struct platform_driver rockchip_sound_driver = {
 	.probe = snd_rk_mc_probe,
 	.driver = {
 		.name = DRV_NAME,
-		.owner = THIS_MODULE,
 		.pm = &snd_soc_pm_ops,
 		.of_match_table = rockchip_sound_of_match,
 	},
diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c
index f77538319221..9e7b5fa4cf59 100644
--- a/sound/soc/rockchip/rockchip_pcm.c
+++ b/sound/soc/rockchip/rockchip_pcm.c
@@ -21,7 +21,8 @@ static const struct snd_pcm_hardware snd_rockchip_hardware = {
 	.info			= SNDRV_PCM_INFO_MMAP |
 				  SNDRV_PCM_INFO_MMAP_VALID |
 				  SNDRV_PCM_INFO_PAUSE |
-				  SNDRV_PCM_INFO_RESUME,
+				  SNDRV_PCM_INFO_RESUME |
+				  SNDRV_PCM_INFO_INTERLEAVED,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 8192,
 	.periods_min		= 1,
diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c
index 43332c32d7e9..dc93941e01c3 100644
--- a/sound/soc/samsung/tm2_wm5110.c
+++ b/sound/soc/samsung/tm2_wm5110.c
@@ -491,6 +491,7 @@ static int tm2_probe(struct platform_device *pdev)
 	struct snd_soc_card *card = &tm2_card;
 	struct tm2_machine_priv *priv;
 	struct of_phandle_args args;
+	struct snd_soc_dai_link *dai_link;
 	int num_codecs, ret, i;
 
 	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -558,18 +559,18 @@ static int tm2_probe(struct platform_device *pdev)
 	}
 
 	/* Initialize WM5110 - I2S and HDMI - I2S1 DAI links */
-	for (i = 0; i < card->num_links; i++) {
+	for_each_card_prelinks(card, i, dai_link) {
 		unsigned int dai_index = 0; /* WM5110 */
 
-		card->dai_link[i].cpu_name = NULL;
-		card->dai_link[i].platform_name = NULL;
+		dai_link->cpu_name = NULL;
+		dai_link->platform_name = NULL;
 
 		if (num_codecs > 1 && i == card->num_links - 1)
 			dai_index = 1; /* HDMI */
 
-		card->dai_link[i].codec_of_node = codec_dai_node[dai_index];
-		card->dai_link[i].cpu_of_node = cpu_dai_node[dai_index];
-		card->dai_link[i].platform_of_node = cpu_dai_node[dai_index];
+		dai_link->codec_of_node = codec_dai_node[dai_index];
+		dai_link->cpu_of_node = cpu_dai_node[dai_index];
+		dai_link->platform_of_node = cpu_dai_node[dai_index];
 	}
 
 	if (num_codecs > 1) {
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index c2b496398e6b..17622ceb98c0 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -319,13 +319,12 @@ static int hac_soc_platform_probe(struct platform_device *pdev)
 	if (ret != 0)
 		return ret;
 
-	return snd_soc_register_component(&pdev->dev, &sh4_hac_component,
+	return devm_snd_soc_register_component(&pdev->dev, &sh4_hac_component,
 					  sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
 }
 
 static int hac_soc_platform_remove(struct platform_device *pdev)
 {
-	snd_soc_unregister_component(&pdev->dev);
 	snd_soc_set_ac97_ops(NULL);
 	return 0;
 }
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 051f96405346..28327dd2c6cb 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -582,7 +582,7 @@ static void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct rsnd_adg *adg)
 	int i;
 
 	for_each_rsnd_clk(clk, adg, i)
-		dev_dbg(dev, "%s    : %p : %ld\n",
+		dev_dbg(dev, "%s    : %pa : %ld\n",
 			clk_name[i], clk, clk_get_rate(clk));
 
 	dev_dbg(dev, "BRGCKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n",
@@ -595,7 +595,7 @@ static void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct rsnd_adg *adg)
 	 * by BRGCKR::BRGCKR_31
 	 */
 	for_each_rsnd_clkout(clk, adg, i)
-		dev_dbg(dev, "clkout %d : %p : %ld\n", i,
+		dev_dbg(dev, "clkout %d : %pa : %ld\n", i,
 			clk, clk_get_rate(clk));
 }
 #else
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index d23c2bbff0cf..f930f51b686f 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -102,7 +102,9 @@
 #include "rsnd.h"
 
 #define RSND_RATES SNDRV_PCM_RATE_8000_192000
-#define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
+#define RSND_FMTS (SNDRV_PCM_FMTBIT_S8 |\
+		   SNDRV_PCM_FMTBIT_S16_LE |\
+		   SNDRV_PCM_FMTBIT_S24_LE)
 
 static const struct of_device_id rsnd_of_match[] = {
 	{ .compatible = "renesas,rcar_sound-gen1", .data = (void *)RSND_GEN1 },
@@ -280,6 +282,8 @@ u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io)
 	struct device *dev = rsnd_priv_to_dev(priv);
 
 	switch (snd_pcm_format_width(runtime->format)) {
+	case 8:
+		return 16 << 16;
 	case 16:
 		return 8 << 16;
 	case 24:
@@ -331,7 +335,7 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io)
 		target = cmd ? cmd : ssiu;
 	}
 
-	/* Non target mod or 24bit data needs normal DALIGN */
+	/* Non target mod or non 16bit needs normal DALIGN */
 	if ((snd_pcm_format_width(runtime->format) != 16) ||
 	    (mod != target))
 		return 0x76543210;
@@ -367,7 +371,7 @@ u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod)
 	 * HW    24bit data is located as 0x******00
 	 *
 	 */
-	if (snd_pcm_format_width(runtime->format) == 16)
+	if (snd_pcm_format_width(runtime->format) != 24)
 		return 0;
 
 	for (i = 0; i < ARRAY_SIZE(playback_mods); i++) {
@@ -540,6 +544,14 @@ int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai,
 	return rdai->ssi_lane;
 }
 
+int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width)
+{
+	if (width > 0)
+		rdai->chan_width = width;
+
+	return rdai->chan_width;
+}
+
 struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id)
 {
 	if ((id < 0) || (id >= rsnd_rdai_nr(priv)))
@@ -681,6 +693,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		rdai->frm_clk_inv = 0;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_DSP_B:
 		rdai->sys_delay = 1;
 		rdai->data_alignment = 0;
 		rdai->frm_clk_inv = 1;
@@ -690,6 +703,11 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		rdai->data_alignment = 1;
 		rdai->frm_clk_inv = 1;
 		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		rdai->sys_delay = 0;
+		rdai->data_alignment = 0;
+		rdai->frm_clk_inv = 1;
+		break;
 	}
 
 	/* set clock inversion */
@@ -720,6 +738,16 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai,
 	struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
 	struct device *dev = rsnd_priv_to_dev(priv);
 
+	switch (slot_width) {
+	case 16:
+	case 24:
+	case 32:
+		break;
+	default:
+		/* use default */
+		slot_width = 32;
+	}
+
 	switch (slots) {
 	case 2:
 	case 6:
@@ -727,6 +755,7 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai,
 		/* TDM Extend Mode */
 		rsnd_rdai_channels_set(rdai, slots);
 		rsnd_rdai_ssi_lane_set(rdai, 1);
+		rsnd_rdai_width_set(rdai, slot_width);
 		break;
 	default:
 		dev_err(dev, "unsupported TDM slots (%d)\n", slots);
@@ -755,7 +784,7 @@ static unsigned int rsnd_soc_hw_rate_list[] = {
 	192000,
 };
 
-static int rsnd_soc_hw_rule(struct rsnd_priv *priv,
+static int rsnd_soc_hw_rule(struct rsnd_dai *rdai,
 			    unsigned int *list, int list_num,
 			    struct snd_interval *baseline, struct snd_interval *iv)
 {
@@ -772,14 +801,14 @@ static int rsnd_soc_hw_rule(struct rsnd_priv *priv,
 		if (!snd_interval_test(iv, list[i]))
 			continue;
 
-		rate = rsnd_ssi_clk_query(priv,
+		rate = rsnd_ssi_clk_query(rdai,
 					  baseline->min, list[i], NULL);
 		if (rate > 0) {
 			p.min = min(p.min, list[i]);
 			p.max = max(p.max, list[i]);
 		}
 
-		rate = rsnd_ssi_clk_query(priv,
+		rate = rsnd_ssi_clk_query(rdai,
 					  baseline->max, list[i], NULL);
 		if (rate > 0) {
 			p.min = min(p.min, list[i]);
@@ -790,17 +819,14 @@ static int rsnd_soc_hw_rule(struct rsnd_priv *priv,
 	return snd_interval_refine(iv, &p);
 }
 
-static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params,
-				   struct snd_pcm_hw_rule *rule,
-				   int is_play)
+static int rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params,
+				 struct snd_pcm_hw_rule *rule)
 {
 	struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
 	struct snd_interval ic;
-	struct snd_soc_dai *dai = rule->private;
-	struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
-	struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
-	struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture;
+	struct rsnd_dai_stream *io = rule->private;
+	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
 
 	/*
 	 * possible sampling rate limitation is same as
@@ -811,34 +837,19 @@ static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params,
 	ic.min =
 	ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params);
 
-	return rsnd_soc_hw_rule(priv, rsnd_soc_hw_rate_list,
+	return rsnd_soc_hw_rule(rdai, rsnd_soc_hw_rate_list,
 				ARRAY_SIZE(rsnd_soc_hw_rate_list),
 				&ic, ir);
 }
 
-static int rsnd_soc_hw_rule_rate_playback(struct snd_pcm_hw_params *params,
-				 struct snd_pcm_hw_rule *rule)
-{
-	return __rsnd_soc_hw_rule_rate(params, rule, 1);
-}
-
-static int rsnd_soc_hw_rule_rate_capture(struct snd_pcm_hw_params *params,
-					  struct snd_pcm_hw_rule *rule)
-{
-	return __rsnd_soc_hw_rule_rate(params, rule, 0);
-}
-
-static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params,
-				       struct snd_pcm_hw_rule *rule,
-				       int is_play)
+static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params,
+				     struct snd_pcm_hw_rule *rule)
 {
 	struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
 	struct snd_interval ic;
-	struct snd_soc_dai *dai = rule->private;
-	struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
-	struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
-	struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture;
+	struct rsnd_dai_stream *io = rule->private;
+	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
 
 	/*
 	 * possible sampling rate limitation is same as
@@ -849,23 +860,11 @@ static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params,
 	ic.min =
 	ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params);
 
-	return rsnd_soc_hw_rule(priv, rsnd_soc_hw_channels_list,
+	return rsnd_soc_hw_rule(rdai, rsnd_soc_hw_channels_list,
 				ARRAY_SIZE(rsnd_soc_hw_channels_list),
 				ir, &ic);
 }
 
-static int rsnd_soc_hw_rule_channels_playback(struct snd_pcm_hw_params *params,
-					      struct snd_pcm_hw_rule *rule)
-{
-	return __rsnd_soc_hw_rule_channels(params, rule, 1);
-}
-
-static int rsnd_soc_hw_rule_channels_capture(struct snd_pcm_hw_params *params,
-					     struct snd_pcm_hw_rule *rule)
-{
-	return __rsnd_soc_hw_rule_channels(params, rule, 0);
-}
-
 static const struct snd_pcm_hardware rsnd_pcm_hardware = {
 	.info =		SNDRV_PCM_INFO_INTERLEAVED	|
 			SNDRV_PCM_INFO_MMAP		|
@@ -882,12 +881,10 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream,
 				struct snd_soc_dai *dai)
 {
 	struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
-	struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
 	struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream);
 	struct snd_pcm_hw_constraint_list *constraint = &rdai->constraint;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	unsigned int max_channels = rsnd_rdai_channels_get(rdai);
-	int ret;
 	int i;
 
 	rsnd_dai_stream_init(io, substream);
@@ -922,25 +919,16 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream,
 		int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 
 		snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
-				    is_play ? rsnd_soc_hw_rule_rate_playback :
-					      rsnd_soc_hw_rule_rate_capture,
-				    dai,
+				    rsnd_soc_hw_rule_rate,
+				    is_play ? &rdai->playback : &rdai->capture,
 				    SNDRV_PCM_HW_PARAM_CHANNELS, -1);
 		snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-				    is_play ? rsnd_soc_hw_rule_channels_playback :
-					      rsnd_soc_hw_rule_channels_capture,
-				    dai,
+				    rsnd_soc_hw_rule_channels,
+				    is_play ? &rdai->playback : &rdai->capture,
 				    SNDRV_PCM_HW_PARAM_RATE, -1);
 	}
 
-	/*
-	 * call rsnd_dai_call without spinlock
-	 */
-	ret = rsnd_dai_call(nolock_start, io, priv);
-	if (ret < 0)
-		rsnd_dai_call(nolock_stop, io, priv);
-
-	return ret;
+	return 0;
 }
 
 static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream,
@@ -953,7 +941,7 @@ static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream,
 	/*
 	 * call rsnd_dai_call without spinlock
 	 */
-	rsnd_dai_call(nolock_stop, io, priv);
+	rsnd_dai_call(cleanup, io, priv);
 
 	rsnd_dai_stream_quit(io);
 }
@@ -1083,6 +1071,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv,
 	rdai->capture.rdai		= rdai;
 	rsnd_rdai_channels_set(rdai, 2); /* default 2ch */
 	rsnd_rdai_ssi_lane_set(rdai, 1); /* default 1lane */
+	rsnd_rdai_width_set(rdai, 32);   /* default 32bit width */
 
 	for (io_i = 0;; io_i++) {
 		playback = of_parse_phandle(dai_np, "playback", io_i);
@@ -1274,8 +1263,15 @@ int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io)
 int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io)
 {
 	struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+	struct rsnd_priv *priv = rsnd_io_to_priv(io);
+	struct device *dev = rsnd_priv_to_dev(priv);
+
+	if (!runtime) {
+		dev_warn(dev, "Can't update kctrl when idle\n");
+		return 0;
+	}
 
-	return !!runtime;
+	return 1;
 }
 
 struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg)
diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c
index 6a55aa753003..ad702377a6c3 100644
--- a/sound/soc/sh/rcar/ctu.c
+++ b/sound/soc/sh/rcar/ctu.c
@@ -258,7 +258,7 @@ static int rsnd_ctu_hw_params(struct rsnd_mod *mod,
 		struct snd_pcm_hw_params *be_params;
 		int stream = substream->stream;
 
-		list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+		for_each_dpcm_be(fe, stream, dpcm) {
 			be_params = &dpcm->hw_params;
 			if (params_channels(fe_params) != params_channels(be_params))
 				ctu->channels = params_channels(be_params);
diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c
index d65ea7bc4dac..6d1947515dc8 100644
--- a/sound/soc/sh/rcar/dma.c
+++ b/sound/soc/sh/rcar/dma.c
@@ -106,9 +106,9 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod,
 	return 0;
 }
 
-static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod,
-				   struct rsnd_dai_stream *io,
-				   struct rsnd_priv *priv)
+static int rsnd_dmaen_cleanup(struct rsnd_mod *mod,
+			      struct rsnd_dai_stream *io,
+			      struct rsnd_priv *priv)
 {
 	struct rsnd_dma *dma = rsnd_mod_to_dma(mod);
 	struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma);
@@ -116,7 +116,7 @@ static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod,
 	/*
 	 * DMAEngine release uses mutex lock.
 	 * Thus, it shouldn't be called under spinlock.
-	 * Let's call it under nolock_start
+	 * Let's call it under prepare
 	 */
 	if (dmaen->chan)
 		dma_release_channel(dmaen->chan);
@@ -126,23 +126,22 @@ static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod,
 	return 0;
 }
 
-static int rsnd_dmaen_nolock_start(struct rsnd_mod *mod,
-			    struct rsnd_dai_stream *io,
-			    struct rsnd_priv *priv)
+static int rsnd_dmaen_prepare(struct rsnd_mod *mod,
+			      struct rsnd_dai_stream *io,
+			      struct rsnd_priv *priv)
 {
 	struct rsnd_dma *dma = rsnd_mod_to_dma(mod);
 	struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma);
 	struct device *dev = rsnd_priv_to_dev(priv);
 
-	if (dmaen->chan) {
-		dev_err(dev, "it already has dma channel\n");
-		return -EIO;
-	}
+	/* maybe suspended */
+	if (dmaen->chan)
+		return 0;
 
 	/*
 	 * DMAEngine request uses mutex lock.
 	 * Thus, it shouldn't be called under spinlock.
-	 * Let's call it under nolock_start
+	 * Let's call it under prepare
 	 */
 	dmaen->chan = rsnd_dmaen_request_channel(io,
 						 dma->mod_from,
@@ -291,8 +290,8 @@ static int rsnd_dmaen_pointer(struct rsnd_mod *mod,
 
 static struct rsnd_mod_ops rsnd_dmaen_ops = {
 	.name	= "audmac",
-	.nolock_start = rsnd_dmaen_nolock_start,
-	.nolock_stop  = rsnd_dmaen_nolock_stop,
+	.prepare = rsnd_dmaen_prepare,
+	.cleanup = rsnd_dmaen_cleanup,
 	.start	= rsnd_dmaen_start,
 	.stop	= rsnd_dmaen_stop,
 	.pointer= rsnd_dmaen_pointer,
@@ -302,16 +301,26 @@ static struct rsnd_mod_ops rsnd_dmaen_ops = {
  *		Audio DMAC peri peri
  */
 static const u8 gen2_id_table_ssiu[] = {
-	0x00, /* SSI00 */
-	0x04, /* SSI10 */
-	0x08, /* SSI20 */
-	0x0c, /* SSI3  */
-	0x0d, /* SSI4  */
-	0x0e, /* SSI5  */
-	0x0f, /* SSI6  */
-	0x10, /* SSI7  */
-	0x11, /* SSI8  */
-	0x12, /* SSI90 */
+	/* SSI00 ~ SSI07 */
+	0x00, 0x01, 0x02, 0x03, 0x39, 0x3a, 0x3b, 0x3c,
+	/* SSI10 ~ SSI17 */
+	0x04, 0x05, 0x06, 0x07, 0x3d, 0x3e, 0x3f, 0x40,
+	/* SSI20 ~ SSI27 */
+	0x08, 0x09, 0x0a, 0x0b, 0x41, 0x42, 0x43, 0x44,
+	/* SSI30 ~ SSI37 */
+	0x0c, 0x45, 0x46, 0x47, 0x48, 0x49, 0x4a, 0x4b,
+	/* SSI40 ~ SSI47 */
+	0x0d, 0x4c, 0x4d, 0x4e, 0x4f, 0x50, 0x51, 0x52,
+	/* SSI5 */
+	0x0e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	/* SSI6 */
+	0x0f, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	/* SSI7 */
+	0x10, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	/* SSI8 */
+	0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	/* SSI90 ~ SSI97 */
+	0x12, 0x13, 0x14, 0x15, 0x53, 0x54, 0x55, 0x56,
 };
 static const u8 gen2_id_table_scu[] = {
 	0x2d, /* SCU_SRCI0 */
@@ -337,18 +346,23 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io,
 	struct rsnd_mod *src = rsnd_io_to_mod_src(io);
 	struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io);
 	const u8 *entry = NULL;
-	int id = rsnd_mod_id(mod);
+	int id = 255;
 	int size = 0;
 
 	if (mod == ssi) {
+		int busif = rsnd_ssi_get_busif(io);
+
 		entry = gen2_id_table_ssiu;
 		size = ARRAY_SIZE(gen2_id_table_ssiu);
+		id = (rsnd_mod_id(mod) * 8) + busif;
 	} else if (mod == src) {
 		entry = gen2_id_table_scu;
 		size = ARRAY_SIZE(gen2_id_table_scu);
+		id = rsnd_mod_id(mod);
 	} else if (mod == dvc) {
 		entry = gen2_id_table_cmd;
 		size = ARRAY_SIZE(gen2_id_table_cmd);
+		id = rsnd_mod_id(mod);
 	}
 
 	if ((!entry) || (size <= id)) {
@@ -382,7 +396,7 @@ static void rsnd_dmapp_write(struct rsnd_dma *dma, u32 data, u32 reg)
 	struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv);
 	struct device *dev = rsnd_priv_to_dev(priv);
 
-	dev_dbg(dev, "w %p : %08x\n", rsnd_dmapp_addr(dmac, dma, reg), data);
+	dev_dbg(dev, "w 0x%px : %08x\n", rsnd_dmapp_addr(dmac, dma, reg), data);
 
 	iowrite32(data, rsnd_dmapp_addr(dmac, dma, reg));
 }
@@ -491,11 +505,11 @@ static struct rsnd_mod_ops rsnd_dmapp_ops = {
 #define RDMA_SSI_I_N(addr, i)	(addr ##_reg - 0x00300000 + (0x40 * i) + 0x8)
 #define RDMA_SSI_O_N(addr, i)	(addr ##_reg - 0x00300000 + (0x40 * i) + 0xc)
 
-#define RDMA_SSIU_I_N(addr, i)	(addr ##_reg - 0x00441000 + (0x1000 * i))
-#define RDMA_SSIU_O_N(addr, i)	(addr ##_reg - 0x00441000 + (0x1000 * i))
+#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400))
+#define RDMA_SSIU_O_N(addr, i, j) RDMA_SSIU_I_N(addr, i, j)
 
-#define RDMA_SSIU_I_P(addr, i)	(addr ##_reg - 0x00141000 + (0x1000 * i))
-#define RDMA_SSIU_O_P(addr, i)	(addr ##_reg - 0x00141000 + (0x1000 * i))
+#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400))
+#define RDMA_SSIU_O_P(addr, i, j) RDMA_SSIU_I_P(addr, i, j)
 
 #define RDMA_SRC_I_N(addr, i)	(addr ##_reg - 0x00500000 + (0x400 * i))
 #define RDMA_SRC_O_N(addr, i)	(addr ##_reg - 0x004fc000 + (0x400 * i))
@@ -521,6 +535,7 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io,
 		      !!rsnd_io_to_mod_mix(io) ||
 		      !!rsnd_io_to_mod_ctu(io);
 	int id = rsnd_mod_id(mod);
+	int busif = rsnd_ssi_get_busif(io);
 	struct dma_addr {
 		dma_addr_t out_addr;
 		dma_addr_t in_addr;
@@ -537,25 +552,35 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io,
 		},
 		/* SSI */
 		/* Capture */
-		{{{ RDMA_SSI_O_N(ssi, id),	0 },
-		  { RDMA_SSIU_O_P(ssi, id),	0 },
-		  { RDMA_SSIU_O_P(ssi, id),	0 } },
+		{{{ RDMA_SSI_O_N(ssi, id),		0 },
+		  { RDMA_SSIU_O_P(ssi, id, busif),	0 },
+		  { RDMA_SSIU_O_P(ssi, id, busif),	0 } },
 		 /* Playback */
-		 {{ 0,				RDMA_SSI_I_N(ssi, id) },
-		  { 0,				RDMA_SSIU_I_P(ssi, id) },
-		  { 0,				RDMA_SSIU_I_P(ssi, id) } }
+		 {{ 0,			RDMA_SSI_I_N(ssi, id) },
+		  { 0,			RDMA_SSIU_I_P(ssi, id, busif) },
+		  { 0,			RDMA_SSIU_I_P(ssi, id, busif) } }
 		},
 		/* SSIU */
 		/* Capture */
-		{{{ RDMA_SSIU_O_N(ssi, id),	0 },
-		  { RDMA_SSIU_O_P(ssi, id),	0 },
-		  { RDMA_SSIU_O_P(ssi, id),	0 } },
+		{{{ RDMA_SSIU_O_N(ssi, id, busif),	0 },
+		  { RDMA_SSIU_O_P(ssi, id, busif),	0 },
+		  { RDMA_SSIU_O_P(ssi, id, busif),	0 } },
 		 /* Playback */
-		 {{ 0,				RDMA_SSIU_I_N(ssi, id) },
-		  { 0,				RDMA_SSIU_I_P(ssi, id) },
-		  { 0,				RDMA_SSIU_I_P(ssi, id) } } },
+		 {{ 0,			RDMA_SSIU_I_N(ssi, id, busif) },
+		  { 0,			RDMA_SSIU_I_P(ssi, id, busif) },
+		  { 0,			RDMA_SSIU_I_P(ssi, id, busif) } } },
 	};
 
+	/*
+	 * FIXME
+	 *
+	 * We can't support SSI9-4/5/6/7, because its address is
+	 * out of calculation rule
+	 */
+	if ((id == 9) && (busif >= 4))
+		dev_err(dev, "This driver doesn't support SSI%d-%d, so far",
+			id, busif);
+
 	/* it shouldn't happen */
 	if (use_cmd && !use_src)
 		dev_err(dev, "DVC is selected without SRC\n");
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 0230301fe078..1f7881cc16b2 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -219,12 +219,33 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv)
 		RSND_GEN_S_REG(HDMI1_SEL,	0x9e4),
 
 		/* FIXME: it needs SSI_MODE2/3 in the future */
-		RSND_GEN_M_REG(SSI_BUSIF_MODE,	0x0,	0x80),
-		RSND_GEN_M_REG(SSI_BUSIF_ADINR,	0x4,	0x80),
-		RSND_GEN_M_REG(SSI_BUSIF_DALIGN,0x8,	0x80),
-		RSND_GEN_M_REG(SSI_MODE,	0xc,	0x80),
-		RSND_GEN_M_REG(SSI_CTRL,	0x10,	0x80),
-		RSND_GEN_M_REG(SSI_INT_ENABLE,	0x18,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF0_MODE,		0x0,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF0_ADINR,	0x4,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF0_DALIGN,	0x8,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF1_MODE,		0x20,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF1_ADINR,	0x24,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF1_DALIGN,	0x28,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF2_MODE,		0x40,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF2_ADINR,	0x44,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF2_DALIGN,	0x48,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF3_MODE,		0x60,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF3_ADINR,	0x64,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF3_DALIGN,	0x68,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF4_MODE,		0x500,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF4_ADINR,	0x504,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF4_DALIGN,	0x508,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF5_MODE,		0x520,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF5_ADINR,	0x524,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF5_DALIGN,	0x528,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF6_MODE,		0x540,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF6_ADINR,	0x544,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF6_DALIGN,	0x548,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF7_MODE,		0x560,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF7_ADINR,	0x564,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF7_DALIGN,	0x568,	0x80),
+		RSND_GEN_M_REG(SSI_MODE,		0xc,	0x80),
+		RSND_GEN_M_REG(SSI_CTRL,		0x10,	0x80),
+		RSND_GEN_M_REG(SSI_INT_ENABLE,		0x18,	0x80),
 	};
 
 	static const struct rsnd_regmap_field_conf conf_scu[] = {
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 8f7a0abfa751..4464d1d0a042 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -156,9 +156,30 @@ enum rsnd_reg {
 	RSND_REG_SSI_MODE2,
 	RSND_REG_SSI_CONTROL,
 	RSND_REG_SSI_CTRL,
-	RSND_REG_SSI_BUSIF_MODE,
-	RSND_REG_SSI_BUSIF_ADINR,
-	RSND_REG_SSI_BUSIF_DALIGN,
+	RSND_REG_SSI_BUSIF0_MODE,
+	RSND_REG_SSI_BUSIF0_ADINR,
+	RSND_REG_SSI_BUSIF0_DALIGN,
+	RSND_REG_SSI_BUSIF1_MODE,
+	RSND_REG_SSI_BUSIF1_ADINR,
+	RSND_REG_SSI_BUSIF1_DALIGN,
+	RSND_REG_SSI_BUSIF2_MODE,
+	RSND_REG_SSI_BUSIF2_ADINR,
+	RSND_REG_SSI_BUSIF2_DALIGN,
+	RSND_REG_SSI_BUSIF3_MODE,
+	RSND_REG_SSI_BUSIF3_ADINR,
+	RSND_REG_SSI_BUSIF3_DALIGN,
+	RSND_REG_SSI_BUSIF4_MODE,
+	RSND_REG_SSI_BUSIF4_ADINR,
+	RSND_REG_SSI_BUSIF4_DALIGN,
+	RSND_REG_SSI_BUSIF5_MODE,
+	RSND_REG_SSI_BUSIF5_ADINR,
+	RSND_REG_SSI_BUSIF5_DALIGN,
+	RSND_REG_SSI_BUSIF6_MODE,
+	RSND_REG_SSI_BUSIF6_ADINR,
+	RSND_REG_SSI_BUSIF6_DALIGN,
+	RSND_REG_SSI_BUSIF7_MODE,
+	RSND_REG_SSI_BUSIF7_ADINR,
+	RSND_REG_SSI_BUSIF7_DALIGN,
 	RSND_REG_SSI_INT_ENABLE,
 	RSND_REG_SSI_SYS_STATUS0,
 	RSND_REG_SSI_SYS_STATUS1,
@@ -274,15 +295,12 @@ struct rsnd_mod_ops {
 	int (*fallback)(struct rsnd_mod *mod,
 			struct rsnd_dai_stream *io,
 			struct rsnd_priv *priv);
-	int (*nolock_start)(struct rsnd_mod *mod,
-		    struct rsnd_dai_stream *io,
-		    struct rsnd_priv *priv);
-	int (*nolock_stop)(struct rsnd_mod *mod,
-		    struct rsnd_dai_stream *io,
-		    struct rsnd_priv *priv);
 	int (*prepare)(struct rsnd_mod *mod,
 		       struct rsnd_dai_stream *io,
 		       struct rsnd_priv *priv);
+	int (*cleanup)(struct rsnd_mod *mod,
+		       struct rsnd_dai_stream *io,
+		       struct rsnd_priv *priv);
 };
 
 struct rsnd_dai_stream;
@@ -300,9 +318,8 @@ struct rsnd_mod {
 /*
  * status
  *
- * 0xH0000CBA
+ * 0xH0000CB0
  *
- * A	0: nolock_start	1: nolock_stop
  * B	0: init		1: quit
  * C	0: start	1: stop
  *
@@ -313,9 +330,8 @@ struct rsnd_mod {
  * H	0: hw_params
  * H	0: pointer
  * H	0: prepare
+ * H	0: cleanup
  */
-#define __rsnd_mod_shift_nolock_start	0
-#define __rsnd_mod_shift_nolock_stop	0
 #define __rsnd_mod_shift_init		4
 #define __rsnd_mod_shift_quit		4
 #define __rsnd_mod_shift_start		8
@@ -328,11 +344,12 @@ struct rsnd_mod {
 #define __rsnd_mod_shift_hw_params	28 /* always called */
 #define __rsnd_mod_shift_pointer	28 /* always called */
 #define __rsnd_mod_shift_prepare	28 /* always called */
+#define __rsnd_mod_shift_cleanup	28 /* always called */
 
 #define __rsnd_mod_add_probe		0
 #define __rsnd_mod_add_remove		0
-#define __rsnd_mod_add_nolock_start	 1
-#define __rsnd_mod_add_nolock_stop	-1
+#define __rsnd_mod_add_prepare		0
+#define __rsnd_mod_add_cleanup		0
 #define __rsnd_mod_add_init		 1
 #define __rsnd_mod_add_quit		-1
 #define __rsnd_mod_add_start		 1
@@ -342,10 +359,11 @@ struct rsnd_mod {
 #define __rsnd_mod_add_fallback		0
 #define __rsnd_mod_add_hw_params	0
 #define __rsnd_mod_add_pointer		0
-#define __rsnd_mod_add_prepare		0
 
 #define __rsnd_mod_call_probe		0
 #define __rsnd_mod_call_remove		0
+#define __rsnd_mod_call_prepare		0
+#define __rsnd_mod_call_cleanup		0
 #define __rsnd_mod_call_init		0
 #define __rsnd_mod_call_quit		1
 #define __rsnd_mod_call_start		0
@@ -355,9 +373,6 @@ struct rsnd_mod {
 #define __rsnd_mod_call_fallback	0
 #define __rsnd_mod_call_hw_params	0
 #define __rsnd_mod_call_pointer		0
-#define __rsnd_mod_call_nolock_start	0
-#define __rsnd_mod_call_nolock_stop	1
-#define __rsnd_mod_call_prepare		0
 
 #define rsnd_mod_to_priv(mod)	((mod)->priv)
 #define rsnd_mod_name(mod)	((mod)->ops->name)
@@ -438,6 +453,7 @@ struct rsnd_dai_stream {
 	char name[RSND_DAI_NAME_SIZE];
 	struct snd_pcm_substream *substream;
 	struct rsnd_mod *mod[RSND_MOD_MAX];
+	struct rsnd_mod *dma;
 	struct rsnd_dai *rdai;
 	struct device *dmac_dev; /* for IPMMU */
 	u32 parent_ssi_status;
@@ -467,6 +483,7 @@ struct rsnd_dai {
 
 	int max_channels;	/* 2ch - 16ch */
 	int ssi_lane;		/* 1lane - 4lane */
+	int chan_width;		/* 16/24/32 bit width */
 
 	unsigned int clk_master:1;
 	unsigned int bit_clk_inv:1;
@@ -500,6 +517,11 @@ int rsnd_rdai_channels_ctrl(struct rsnd_dai *rdai,
 int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai,
 			    int ssi_lane);
 
+#define rsnd_rdai_width_set(rdai, width) \
+	rsnd_rdai_width_ctrl(rdai, width)
+#define rsnd_rdai_width_get(rdai) \
+	rsnd_rdai_width_ctrl(rdai, 0)
+int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width);
 void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io);
 int rsnd_dai_connect(struct rsnd_mod *mod,
 		     struct rsnd_dai_stream *io,
@@ -692,6 +714,7 @@ void rsnd_ssi_remove(struct rsnd_priv *priv);
 struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id);
 int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod);
 int rsnd_ssi_use_busif(struct rsnd_dai_stream *io);
+int rsnd_ssi_get_busif(struct rsnd_dai_stream *io);
 u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io);
 
 #define RSND_SSI_HDMI_PORT0	0xf0
@@ -709,7 +732,7 @@ int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod);
 void rsnd_parse_connect_ssi(struct rsnd_dai *rdai,
 			    struct device_node *playback,
 			    struct device_node *capture);
-unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv,
+unsigned int rsnd_ssi_clk_query(struct rsnd_dai *rdai,
 		       int param1, int param2, int *idx);
 
 /*
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index beccfbac7581..cd38a43b976f 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -158,7 +158,7 @@ static int rsnd_src_hw_params(struct rsnd_mod *mod,
 		struct snd_soc_dpcm *dpcm;
 		struct snd_pcm_hw_params *be_params;
 
-		list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+		for_each_dpcm_be(fe, stream, dpcm) {
 			be_params = &dpcm->hw_params;
 
 			if (params_rate(fe_params) != params_rate(be_params))
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 3f880ec66459..fcb4df23248c 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -42,7 +42,13 @@
 #define	DWL_24		(5 << 19)	/* Data Word Length */
 #define	DWL_32		(6 << 19)	/* Data Word Length */
 
+/*
+ * System word length
+ */
+#define	SWL_16		(1 << 16)	/* R/W System Word Length */
+#define	SWL_24		(2 << 16)	/* R/W System Word Length */
 #define	SWL_32		(3 << 16)	/* R/W System Word Length */
+
 #define	SCKD		(1 << 15)	/* Serial Bit Clock Direction */
 #define	SWSD		(1 << 14)	/* Serial WS Direction */
 #define	SCKP		(1 << 13)	/* Serial Bit Clock Polarity */
@@ -72,7 +78,6 @@
 
 struct rsnd_ssi {
 	struct rsnd_mod mod;
-	struct rsnd_mod *dma;
 
 	u32 flags;
 	u32 cr_own;
@@ -145,6 +150,11 @@ int rsnd_ssi_use_busif(struct rsnd_dai_stream *io)
 	return use_busif;
 }
 
+int rsnd_ssi_get_busif(struct rsnd_dai_stream *io)
+{
+	return 0; /* BUSIF0 only for now */
+}
+
 static void rsnd_ssi_status_clear(struct rsnd_mod *mod)
 {
 	rsnd_mod_write(mod, SSISR, 0);
@@ -220,14 +230,32 @@ u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io)
 	return 0;
 }
 
-unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv,
+static u32 rsnd_rdai_width_to_swl(struct rsnd_dai *rdai)
+{
+	struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
+	struct device *dev = rsnd_priv_to_dev(priv);
+	int width = rsnd_rdai_width_get(rdai);
+
+	switch (width) {
+	case 32: return SWL_32;
+	case 24: return SWL_24;
+	case 16: return SWL_16;
+	}
+
+	dev_err(dev, "unsupported slot width value: %d\n", width);
+	return 0;
+}
+
+unsigned int rsnd_ssi_clk_query(struct rsnd_dai *rdai,
 		       int param1, int param2, int *idx)
 {
+	struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
 	int ssi_clk_mul_table[] = {
 		1, 2, 4, 8, 16, 6, 12,
 	};
 	int j, ret;
 	unsigned int main_rate;
+	int width = rsnd_rdai_width_get(rdai);
 
 	for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) {
 
@@ -240,12 +268,7 @@ unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv,
 		if (j == 0)
 			continue;
 
-		/*
-		 * this driver is assuming that
-		 * system word is 32bit x chan
-		 * see rsnd_ssi_init()
-		 */
-		main_rate = 32 * param1 * param2 * ssi_clk_mul_table[j];
+		main_rate = width * param1 * param2 * ssi_clk_mul_table[j];
 
 		ret = rsnd_adg_clk_query(priv, main_rate);
 		if (ret < 0)
@@ -289,10 +312,15 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
 			return -EINVAL;
 		}
 
+		if (ssi->chan != chan) {
+			dev_err(dev, "SSI parent/child should use same chan\n");
+			return -EINVAL;
+		}
+
 		return 0;
 	}
 
-	main_rate = rsnd_ssi_clk_query(priv, rate, chan, &idx);
+	main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx);
 	if (!main_rate) {
 		dev_err(dev, "unsupported clock rate\n");
 		return -EIO;
@@ -312,9 +340,11 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
 	 * SSICR  : FORCE, SCKD, SWSD
 	 * SSIWSR : CONT
 	 */
-	ssi->cr_clk = FORCE | SWL_32 | SCKD | SWSD | CKDV(idx);
+	ssi->cr_clk = FORCE | rsnd_rdai_width_to_swl(rdai) |
+			SCKD | SWSD | CKDV(idx);
 	ssi->wsr = CONT;
 	ssi->rate = rate;
+	ssi->chan = chan;
 
 	dev_dbg(dev, "%s[%d] outputs %u Hz\n",
 		rsnd_mod_name(mod),
@@ -340,6 +370,7 @@ static void rsnd_ssi_master_clk_stop(struct rsnd_mod *mod,
 
 	ssi->cr_clk	= 0;
 	ssi->rate	= 0;
+	ssi->chan	= 0;
 
 	rsnd_adg_ssi_clk_stop(mod);
 }
@@ -357,15 +388,11 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 
 	is_tdm = rsnd_runtime_is_ssi_tdm(io);
 
-	/*
-	 * always use 32bit system word.
-	 * see also rsnd_ssi_master_clk_enable()
-	 */
-	cr_own |= FORCE | SWL_32;
+	cr_own |= FORCE | rsnd_rdai_width_to_swl(rdai);
 
 	if (rdai->bit_clk_inv)
 		cr_own |= SCKP;
-	if (rdai->frm_clk_inv ^ is_tdm)
+	if (rdai->frm_clk_inv && !is_tdm)
 		cr_own |= SWSP;
 	if (rdai->data_alignment)
 		cr_own |= SDTA;
@@ -373,6 +400,17 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 		cr_own |= DEL;
 
 	/*
+	 * TDM Mode
+	 * see
+	 *	rsnd_ssiu_init_gen2()
+	 */
+	wsr = ssi->wsr;
+	if (is_tdm) {
+		wsr	|= WS_MODE;
+		cr_own	|= CHNL_8;
+	}
+
+	/*
 	 * We shouldn't exchange SWSP after running.
 	 * This means, parent needs to care it.
 	 */
@@ -384,6 +422,9 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 
 	cr_own &= ~DWL_MASK;
 	switch (snd_pcm_format_width(runtime->format)) {
+	case 8:
+		cr_own |= DWL_8;
+		break;
 	case 16:
 		cr_own |= DWL_16;
 		break;
@@ -399,16 +440,6 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 		cr_mode = DIEN;		/* PIO : enable Data interrupt */
 	}
 
-	/*
-	 * TDM Extend Mode
-	 * see
-	 *	rsnd_ssiu_init_gen2()
-	 */
-	wsr = ssi->wsr;
-	if (is_tdm) {
-		wsr	|= WS_MODE;
-		cr_own	|= CHNL_8;
-	}
 init_end:
 	ssi->cr_own	= cr_own;
 	ssi->cr_mode	= cr_mode;
@@ -488,26 +519,16 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod,
 			      struct snd_pcm_substream *substream,
 			      struct snd_pcm_hw_params *params)
 {
-	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
-	int chan = params_channels(params);
+	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
+	unsigned int fmt_width = snd_pcm_format_width(params_format(params));
 
-	/*
-	 * snd_pcm_ops::hw_params will be called *before*
-	 * snd_soc_dai_ops::trigger. Thus, ssi->usrcnt is 0
-	 * in 1st call.
-	 */
-	if (ssi->usrcnt) {
-		/*
-		 * Already working.
-		 * It will happen if SSI has parent/child connection.
-		 * it is error if child <-> parent SSI uses
-		 * different channels.
-		 */
-		if (ssi->chan != chan)
-			return -EIO;
-	}
+	if (fmt_width > rdai->chan_width) {
+		struct rsnd_priv *priv = rsnd_io_to_priv(io);
+		struct device *dev = rsnd_priv_to_dev(priv);
 
-	ssi->chan = chan;
+		dev_err(dev, "invalid combination of slot-width and format-data-width\n");
+		return -EINVAL;
+	}
 
 	return 0;
 }
@@ -873,7 +894,6 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
 			      struct rsnd_dai_stream *io,
 			      struct rsnd_priv *priv)
 {
-	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
 	int ret;
 
 	/*
@@ -888,7 +908,7 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
 		return ret;
 
 	/* SSI probe might be called many times in MUX multi path */
-	ret = rsnd_dma_attach(io, mod, &ssi->dma);
+	ret = rsnd_dma_attach(io, mod, &io->dma);
 
 	return ret;
 }
diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index 016fbf5ac242..39b67643b5dc 100644
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -10,9 +10,12 @@
 
 struct rsnd_ssiu {
 	struct rsnd_mod mod;
+	u32 busif_status[8]; /* for BUSIF0 - BUSIF7 */
+	unsigned int usrcnt;
 };
 
 #define rsnd_ssiu_nr(priv) ((priv)->ssiu_nr)
+#define rsnd_mod_to_ssiu(_mod) container_of((_mod), struct rsnd_ssiu, mod)
 #define for_each_rsnd_ssiu(pos, priv, i)				\
 	for (i = 0;							\
 	     (i < rsnd_ssiu_nr(priv)) &&				\
@@ -120,6 +123,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod,
 			       struct rsnd_dai_stream *io,
 			       struct rsnd_priv *priv)
 {
+	struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod);
 	int hdmi = rsnd_ssi_hdmi_port(io);
 	int ret;
 	u32 mode = 0;
@@ -128,6 +132,8 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod,
 	if (ret < 0)
 		return ret;
 
+	ssiu->usrcnt++;
+
 	if (rsnd_runtime_is_ssi_tdm(io)) {
 		/*
 		 * TDM Extend Mode
@@ -140,15 +146,59 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod,
 	rsnd_mod_write(mod, SSI_MODE, mode);
 
 	if (rsnd_ssi_use_busif(io)) {
-		rsnd_mod_write(mod, SSI_BUSIF_ADINR,
-			       rsnd_get_adinr_bit(mod, io) |
-			       (rsnd_io_is_play(io) ?
-				rsnd_runtime_channel_after_ctu(io) :
-				rsnd_runtime_channel_original(io)));
-		rsnd_mod_write(mod, SSI_BUSIF_MODE,
-			       rsnd_get_busif_shift(io, mod) | 1);
-		rsnd_mod_write(mod, SSI_BUSIF_DALIGN,
-			       rsnd_get_dalign(mod, io));
+		int id = rsnd_mod_id(mod);
+		int busif = rsnd_ssi_get_busif(io);
+
+		/*
+		 * FIXME
+		 *
+		 * We can't support SSI9-4/5/6/7, because its address is
+		 * out of calculation rule
+		 */
+		if ((id == 9) && (busif >= 4)) {
+			struct device *dev = rsnd_priv_to_dev(priv);
+
+			dev_err(dev, "This driver doesn't support SSI%d-%d, so far",
+				id, busif);
+		}
+
+#define RSND_WRITE_BUSIF(i)						\
+		rsnd_mod_write(mod, SSI_BUSIF##i##_ADINR,		\
+			       rsnd_get_adinr_bit(mod, io) |		\
+			       (rsnd_io_is_play(io) ?			\
+				rsnd_runtime_channel_after_ctu(io) :	\
+				rsnd_runtime_channel_original(io)));	\
+		rsnd_mod_write(mod, SSI_BUSIF##i##_MODE,		\
+			       rsnd_get_busif_shift(io, mod) | 1);	\
+		rsnd_mod_write(mod, SSI_BUSIF##i##_DALIGN,		\
+			       rsnd_get_dalign(mod, io))
+
+		switch (busif) {
+		case 0:
+			RSND_WRITE_BUSIF(0);
+			break;
+		case 1:
+			RSND_WRITE_BUSIF(1);
+			break;
+		case 2:
+			RSND_WRITE_BUSIF(2);
+			break;
+		case 3:
+			RSND_WRITE_BUSIF(3);
+			break;
+		case 4:
+			RSND_WRITE_BUSIF(4);
+			break;
+		case 5:
+			RSND_WRITE_BUSIF(5);
+			break;
+		case 6:
+			RSND_WRITE_BUSIF(6);
+			break;
+		case 7:
+			RSND_WRITE_BUSIF(7);
+			break;
+		}
 	}
 
 	if (hdmi) {
@@ -194,10 +244,12 @@ static int rsnd_ssiu_start_gen2(struct rsnd_mod *mod,
 				struct rsnd_dai_stream *io,
 				struct rsnd_priv *priv)
 {
+	int busif = rsnd_ssi_get_busif(io);
+
 	if (!rsnd_ssi_use_busif(io))
 		return 0;
 
-	rsnd_mod_write(mod, SSI_CTRL, 0x1);
+	rsnd_mod_bset(mod, SSI_CTRL, 1 << (busif * 4), 1 << (busif * 4));
 
 	if (rsnd_ssi_multi_slaves_runtime(io))
 		rsnd_mod_write(mod, SSI_CONTROL, 0x1);
@@ -209,10 +261,16 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod,
 			       struct rsnd_dai_stream *io,
 			       struct rsnd_priv *priv)
 {
+	struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod);
+	int busif = rsnd_ssi_get_busif(io);
+
 	if (!rsnd_ssi_use_busif(io))
 		return 0;
 
-	rsnd_mod_write(mod, SSI_CTRL, 0);
+	rsnd_mod_bset(mod, SSI_CTRL, 1 << (busif * 4), 0);
+
+	if (--ssiu->usrcnt)
+		return 0;
 
 	if (rsnd_ssi_multi_slaves_runtime(io))
 		rsnd_mod_write(mod, SSI_CONTROL, 0);
@@ -246,6 +304,16 @@ int rsnd_ssiu_attach(struct rsnd_dai_stream *io,
 	return rsnd_dai_connect(mod, io, mod->type);
 }
 
+static u32 *rsnd_ssiu_get_status(struct rsnd_dai_stream *io,
+				 struct rsnd_mod *mod,
+				 enum rsnd_mod_type type)
+{
+	struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod);
+	int busif = rsnd_ssi_get_busif(io);
+
+	return &ssiu->busif_status[busif];
+}
+
 int rsnd_ssiu_probe(struct rsnd_priv *priv)
 {
 	struct device *dev = rsnd_priv_to_dev(priv);
@@ -269,7 +337,7 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv)
 
 	for_each_rsnd_ssiu(ssiu, priv, i) {
 		ret = rsnd_mod_init(priv, rsnd_mod_get(ssiu),
-				    ops, NULL, rsnd_mod_get_status,
+				    ops, NULL, rsnd_ssiu_get_status,
 				    RSND_MOD_SSIU, i);
 		if (ret)
 			return ret;
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 409d082e80d1..699397a09167 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -157,7 +157,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
 	ret = dpcm_be_dai_startup(fe, stream);
 	if (ret < 0) {
 		/* clean up all links */
-		list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+		for_each_dpcm_be(fe, stream, dpcm)
 			dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
 
 		dpcm_be_disconnect(fe, stream);
@@ -321,7 +321,7 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream)
 	ret = dpcm_be_dai_shutdown(fe, stream);
 
 	/* mark FE's links ready to prune */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+	for_each_dpcm_be(fe, stream, dpcm)
 		dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
 
 	dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 473eefe8658e..6ddcf12bc030 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -52,6 +52,10 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root);
 
 static DEFINE_MUTEX(client_mutex);
 static LIST_HEAD(component_list);
+static LIST_HEAD(unbind_card_list);
+
+#define for_each_component(component)			\
+	list_for_each_entry(component, &component_list, list)
 
 /*
  * This is a timeout to do a DAPM powerdown after a stream is closed().
@@ -62,8 +66,9 @@ static int pmdown_time = 5000;
 module_param(pmdown_time, int, 0);
 MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
 
-/* If a DMI filed contain strings in this blacklist (e.g.
- * "Type2 - Board Manufacturer" or  "Type1 - TBD by OEM"), it will be taken
+/*
+ * If a DMI filed contain strings in this blacklist (e.g.
+ * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken
  * as invalid and dropped when setting the card long name from DMI info.
  */
 static const char * const dmi_blacklist[] = {
@@ -175,8 +180,8 @@ static int dai_list_show(struct seq_file *m, void *v)
 
 	mutex_lock(&client_mutex);
 
-	list_for_each_entry(component, &component_list, list)
-		list_for_each_entry(dai, &component->dai_list, list)
+	for_each_component(component)
+		for_each_component_dais(component, dai)
 			seq_printf(m, "%s\n", dai->name);
 
 	mutex_unlock(&client_mutex);
@@ -191,7 +196,7 @@ static int component_list_show(struct seq_file *m, void *v)
 
 	mutex_lock(&client_mutex);
 
-	list_for_each_entry(component, &component_list, list)
+	for_each_component(component)
 		seq_printf(m, "%s\n", component->name);
 
 	mutex_unlock(&client_mutex);
@@ -218,7 +223,7 @@ static void soc_init_card_debugfs(struct snd_soc_card *card)
 						    &card->pop_time);
 	if (!card->debugfs_pop_time)
 		dev_warn(card->dev,
-		       "ASoC: Failed to create pop time debugfs file\n");
+			 "ASoC: Failed to create pop time debugfs file\n");
 }
 
 static void soc_cleanup_card_debugfs(struct snd_soc_card *card)
@@ -341,7 +346,7 @@ struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card,
 {
 	struct snd_soc_pcm_runtime *rtd;
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		if (rtd->dai_link->no_pcm &&
 			!strcmp(rtd->dai_link->name, dai_link))
 			return rtd->pcm->streams[stream].substream;
@@ -398,7 +403,7 @@ static void soc_remove_pcm_runtimes(struct snd_soc_card *card)
 {
 	struct snd_soc_pcm_runtime *rtd, *_rtd;
 
-	list_for_each_entry_safe(rtd, _rtd, &card->rtd_list, list) {
+	for_each_card_rtds_safe(card, rtd, _rtd) {
 		list_del(&rtd->list);
 		soc_free_pcm_runtime(rtd);
 	}
@@ -411,7 +416,7 @@ struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card,
 {
 	struct snd_soc_pcm_runtime *rtd;
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		if (!strcmp(rtd->dai_link->name, dai_link))
 			return rtd;
 	}
@@ -422,7 +427,8 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime);
 
 static void codec2codec_close_delayed_work(struct work_struct *work)
 {
-	/* Currently nothing to do for c2c links
+	/*
+	 * Currently nothing to do for c2c links
 	 * Since c2c links are internal nodes in the DAPM graph and
 	 * don't interface with the outside world or application layer
 	 * we don't have to do any special handling on close.
@@ -442,8 +448,9 @@ int snd_soc_suspend(struct device *dev)
 	if (!card->instantiated)
 		return 0;
 
-	/* Due to the resume being scheduled into a workqueue we could
-	* suspend before that's finished - wait for it to complete.
+	/*
+	 * Due to the resume being scheduled into a workqueue we could
+	 * suspend before that's finished - wait for it to complete.
 	 */
 	snd_power_wait(card->snd_card, SNDRV_CTL_POWER_D0);
 
@@ -451,13 +458,13 @@ int snd_soc_suspend(struct device *dev)
 	snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D3hot);
 
 	/* mute any active DACs */
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
+		struct snd_soc_dai *dai;
 
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		for (i = 0; i < rtd->num_codecs; i++) {
-			struct snd_soc_dai *dai = rtd->codec_dais[i];
+		for_each_rtd_codec_dai(rtd, i, dai) {
 			struct snd_soc_dai_driver *drv = dai->driver;
 
 			if (drv->ops->digital_mute && dai->playback_active)
@@ -466,7 +473,7 @@ int snd_soc_suspend(struct device *dev)
 	}
 
 	/* suspend all pcms */
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
@@ -476,7 +483,7 @@ int snd_soc_suspend(struct device *dev)
 	if (card->suspend_pre)
 		card->suspend_pre(card);
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 
 		if (rtd->dai_link->ignore_suspend)
@@ -487,10 +494,10 @@ int snd_soc_suspend(struct device *dev)
 	}
 
 	/* close any waiting streams */
-	list_for_each_entry(rtd, &card->rtd_list, list)
+	for_each_card_rtds(card, rtd)
 		flush_delayed_work(&rtd->delayed_work);
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 
 		if (rtd->dai_link->ignore_suspend)
 			continue;
@@ -509,11 +516,14 @@ int snd_soc_suspend(struct device *dev)
 	snd_soc_dapm_sync(&card->dapm);
 
 	/* suspend all COMPONENTs */
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
-		struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+	for_each_card_components(card, component) {
+		struct snd_soc_dapm_context *dapm =
+				snd_soc_component_get_dapm(component);
 
-		/* If there are paths active then the COMPONENT will be held with
-		 * bias _ON and should not be suspended. */
+		/*
+		 * If there are paths active then the COMPONENT will be held
+		 * with bias _ON and should not be suspended.
+		 */
 		if (!component->suspended) {
 			switch (snd_soc_dapm_get_bias_level(dapm)) {
 			case SND_SOC_BIAS_STANDBY:
@@ -547,7 +557,7 @@ int snd_soc_suspend(struct device *dev)
 		}
 	}
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 
 		if (rtd->dai_link->ignore_suspend)
@@ -567,18 +577,21 @@ int snd_soc_suspend(struct device *dev)
 }
 EXPORT_SYMBOL_GPL(snd_soc_suspend);
 
-/* deferred resume work, so resume can complete before we finished
+/*
+ * deferred resume work, so resume can complete before we finished
  * setting our codec back up, which can be very slow on I2C
  */
 static void soc_resume_deferred(struct work_struct *work)
 {
 	struct snd_soc_card *card =
-			container_of(work, struct snd_soc_card, deferred_resume_work);
+			container_of(work, struct snd_soc_card,
+				     deferred_resume_work);
 	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_component *component;
 	int i;
 
-	/* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
+	/*
+	 * our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
 	 * so userspace apps are blocked from touching us
 	 */
 
@@ -591,7 +604,7 @@ static void soc_resume_deferred(struct work_struct *work)
 		card->resume_pre(card);
 
 	/* resume control bus DAIs */
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 
 		if (rtd->dai_link->ignore_suspend)
@@ -601,7 +614,7 @@ static void soc_resume_deferred(struct work_struct *work)
 			cpu_dai->driver->resume(cpu_dai);
 	}
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (component->suspended) {
 			if (component->driver->resume)
 				component->driver->resume(component);
@@ -609,7 +622,7 @@ static void soc_resume_deferred(struct work_struct *work)
 		}
 	}
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 
 		if (rtd->dai_link->ignore_suspend)
 			continue;
@@ -624,13 +637,13 @@ static void soc_resume_deferred(struct work_struct *work)
 	}
 
 	/* unmute any active DACs */
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
+		struct snd_soc_dai *dai;
 
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		for (i = 0; i < rtd->num_codecs; i++) {
-			struct snd_soc_dai *dai = rtd->codec_dais[i];
+		for_each_rtd_codec_dai(rtd, i, dai) {
 			struct snd_soc_dai_driver *drv = dai->driver;
 
 			if (drv->ops->digital_mute && dai->playback_active)
@@ -638,7 +651,7 @@ static void soc_resume_deferred(struct work_struct *work)
 		}
 	}
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 
 		if (rtd->dai_link->ignore_suspend)
@@ -673,16 +686,15 @@ int snd_soc_resume(struct device *dev)
 		return 0;
 
 	/* activate pins from sleep state */
-	list_for_each_entry(rtd, &card->rtd_list, list) {
-		struct snd_soc_dai **codec_dais = rtd->codec_dais;
+	for_each_card_rtds(card, rtd) {
+		struct snd_soc_dai *codec_dai;
 		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 		int j;
 
 		if (cpu_dai->active)
 			pinctrl_pm_select_default_state(cpu_dai->dev);
 
-		for (j = 0; j < rtd->num_codecs; j++) {
-			struct snd_soc_dai *codec_dai = codec_dais[j];
+		for_each_rtd_codec_dai(rtd, j, codec_dai) {
 			if (codec_dai->active)
 				pinctrl_pm_select_default_state(codec_dai->dev);
 		}
@@ -694,8 +706,9 @@ int snd_soc_resume(struct device *dev)
 	 * have that problem and may take a substantial amount of time to resume
 	 * due to I/O costs and anti-pop so handle them out of line.
 	 */
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
 		bus_control |= cpu_dai->driver->bus_control;
 	}
 	if (bus_control) {
@@ -725,7 +738,7 @@ static struct snd_soc_component *soc_find_component(
 
 	lockdep_assert_held(&client_mutex);
 
-	list_for_each_entry(component, &component_list, list) {
+	for_each_component(component) {
 		if (of_node) {
 			if (component->dev->of_node == of_node)
 				return component;
@@ -737,6 +750,24 @@ static struct snd_soc_component *soc_find_component(
 	return NULL;
 }
 
+static int snd_soc_is_matching_component(
+	const struct snd_soc_dai_link_component *dlc,
+	struct snd_soc_component *component)
+{
+	struct device_node *component_of_node;
+
+	component_of_node = component->dev->of_node;
+	if (!component_of_node && component->dev->parent)
+		component_of_node = component->dev->parent->of_node;
+
+	if (dlc->of_node && component_of_node != dlc->of_node)
+		return 0;
+	if (dlc->name && strcmp(component->name, dlc->name))
+		return 0;
+
+	return 1;
+}
+
 /**
  * snd_soc_find_dai - Find a registered DAI
  *
@@ -753,21 +784,14 @@ struct snd_soc_dai *snd_soc_find_dai(
 {
 	struct snd_soc_component *component;
 	struct snd_soc_dai *dai;
-	struct device_node *component_of_node;
 
 	lockdep_assert_held(&client_mutex);
 
-	/* Find CPU DAI from registered DAIs*/
-	list_for_each_entry(component, &component_list, list) {
-		component_of_node = component->dev->of_node;
-		if (!component_of_node && component->dev->parent)
-			component_of_node = component->dev->parent->of_node;
-
-		if (dlc->of_node && component_of_node != dlc->of_node)
-			continue;
-		if (dlc->name && strcmp(component->name, dlc->name))
+	/* Find CPU DAI from registered DAIs */
+	for_each_component(component) {
+		if (!snd_soc_is_matching_component(dlc, component))
 			continue;
-		list_for_each_entry(dai, &component->dai_list, list) {
+		for_each_component_dais(component, dai) {
 			if (dlc->dai_name && strcmp(dai->name, dlc->dai_name)
 			    && (!dai->driver->name
 				|| strcmp(dai->driver->name, dlc->dai_name)))
@@ -781,7 +805,6 @@ struct snd_soc_dai *snd_soc_find_dai(
 }
 EXPORT_SYMBOL_GPL(snd_soc_find_dai);
 
-
 /**
  * snd_soc_find_dai_link - Find a DAI link
  *
@@ -805,7 +828,7 @@ struct snd_soc_dai_link *snd_soc_find_dai_link(struct snd_soc_card *card,
 
 	lockdep_assert_held(&client_mutex);
 
-	list_for_each_entry_safe(link, _link, &card->dai_link_list, list) {
+	for_each_card_links_safe(card, link, _link) {
 		if (link->id != id)
 			continue;
 
@@ -828,7 +851,7 @@ static bool soc_is_dai_link_bound(struct snd_soc_card *card,
 {
 	struct snd_soc_pcm_runtime *rtd;
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		if (rtd->dai_link == dai_link)
 			return true;
 	}
@@ -844,8 +867,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
 	struct snd_soc_dai_link_component cpu_dai_component;
 	struct snd_soc_component *component;
 	struct snd_soc_dai **codec_dais;
-	struct device_node *platform_of_node;
-	const char *platform_name;
 	int i;
 
 	if (dai_link->ignore)
@@ -877,6 +898,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
 	rtd->num_codecs = dai_link->num_codecs;
 
 	/* Find CODEC from registered CODECs */
+	/* we can use for_each_rtd_codec_dai() after this */
 	codec_dais = rtd->codec_dais;
 	for (i = 0; i < rtd->num_codecs; i++) {
 		codec_dais[i] = snd_soc_find_dai(&codecs[i]);
@@ -891,24 +913,11 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
 	/* Single codec links expect codec and codec_dai in runtime data */
 	rtd->codec_dai = codec_dais[0];
 
-	/* if there's no platform we match on the empty platform */
-	platform_name = dai_link->platform_name;
-	if (!platform_name && !dai_link->platform_of_node)
-		platform_name = "snd-soc-dummy";
-
 	/* find one from the set of registered platforms */
-	list_for_each_entry(component, &component_list, list) {
-		platform_of_node = component->dev->of_node;
-		if (!platform_of_node && component->dev->parent->of_node)
-			platform_of_node = component->dev->parent->of_node;
-
-		if (dai_link->platform_of_node) {
-			if (platform_of_node != dai_link->platform_of_node)
-				continue;
-		} else {
-			if (strcmp(component->name, platform_name))
-				continue;
-		}
+	for_each_component(component) {
+		if (!snd_soc_is_matching_component(dai_link->platform,
+						   component))
+			continue;
 
 		snd_soc_rtdcom_add(rtd, component);
 	}
@@ -918,7 +927,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
 
 _err_defer:
 	soc_free_pcm_runtime(rtd);
-	return  -EPROBE_DEFER;
+	return -EPROBE_DEFER;
 }
 
 static void soc_remove_component(struct snd_soc_component *component)
@@ -942,23 +951,25 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order)
 {
 	int err;
 
-	if (dai && dai->probed &&
-			dai->driver->remove_order == order) {
-		if (dai->driver->remove) {
-			err = dai->driver->remove(dai);
-			if (err < 0)
-				dev_err(dai->dev,
-					"ASoC: failed to remove %s: %d\n",
-					dai->name, err);
-		}
-		dai->probed = 0;
+	if (!dai || !dai->probed ||
+	    dai->driver->remove_order != order)
+		return;
+
+	if (dai->driver->remove) {
+		err = dai->driver->remove(dai);
+		if (err < 0)
+			dev_err(dai->dev,
+				"ASoC: failed to remove %s: %d\n",
+				dai->name, err);
 	}
+	dai->probed = 0;
 }
 
 static void soc_remove_link_dais(struct snd_soc_card *card,
 		struct snd_soc_pcm_runtime *rtd, int order)
 {
 	int i;
+	struct snd_soc_dai *codec_dai;
 
 	/* unregister the rtd device */
 	if (rtd->dev_registered) {
@@ -967,8 +978,8 @@ static void soc_remove_link_dais(struct snd_soc_card *card,
 	}
 
 	/* remove the CODEC DAI */
-	for (i = 0; i < rtd->num_codecs; i++)
-		soc_remove_dai(rtd->codec_dais[i], order);
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		soc_remove_dai(codec_dai, order);
 
 	soc_remove_dai(rtd->cpu_dai, order);
 }
@@ -993,28 +1004,57 @@ static void soc_remove_dai_links(struct snd_soc_card *card)
 	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_dai_link *link, *_link;
 
-	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-			order++) {
-		list_for_each_entry(rtd, &card->rtd_list, list)
+	for_each_comp_order(order) {
+		for_each_card_rtds(card, rtd)
 			soc_remove_link_dais(card, rtd, order);
 	}
 
-	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-			order++) {
-		list_for_each_entry(rtd, &card->rtd_list, list)
+	for_each_comp_order(order) {
+		for_each_card_rtds(card, rtd)
 			soc_remove_link_components(card, rtd, order);
 	}
 
-	list_for_each_entry_safe(link, _link, &card->dai_link_list, list) {
+	for_each_card_links_safe(card, link, _link) {
 		if (link->dobj.type == SND_SOC_DOBJ_DAI_LINK)
 			dev_warn(card->dev, "Topology forgot to remove link %s?\n",
 				link->name);
 
 		list_del(&link->list);
-		card->num_dai_links--;
 	}
 }
 
+static int snd_soc_init_platform(struct snd_soc_card *card,
+				 struct snd_soc_dai_link *dai_link)
+{
+	struct snd_soc_dai_link_component *platform = dai_link->platform;
+
+	/*
+	 * FIXME
+	 *
+	 * this function should be removed in the future
+	 */
+	/* convert Legacy platform link */
+	if (!platform) {
+		platform = devm_kzalloc(card->dev,
+				sizeof(struct snd_soc_dai_link_component),
+				GFP_KERNEL);
+		if (!platform)
+			return -ENOMEM;
+
+		dai_link->platform	= platform;
+		platform->name		= dai_link->platform_name;
+		platform->of_node	= dai_link->platform_of_node;
+		platform->dai_name	= NULL;
+	}
+
+	/* if there's no platform we match on the empty platform */
+	if (!platform->name &&
+	    !platform->of_node)
+		platform->name = "snd-soc-dummy";
+
+	return 0;
+}
+
 static int snd_soc_init_multicodec(struct snd_soc_card *card,
 				   struct snd_soc_dai_link *dai_link)
 {
@@ -1043,9 +1083,16 @@ static int snd_soc_init_multicodec(struct snd_soc_card *card,
 }
 
 static int soc_init_dai_link(struct snd_soc_card *card,
-				   struct snd_soc_dai_link *link)
+			     struct snd_soc_dai_link *link)
 {
 	int i, ret;
+	struct snd_soc_dai_link_component *codec;
+
+	ret = snd_soc_init_platform(card, link);
+	if (ret) {
+		dev_err(card->dev, "ASoC: failed to init multiplatform\n");
+		return ret;
+	}
 
 	ret = snd_soc_init_multicodec(card, link);
 	if (ret) {
@@ -1053,19 +1100,19 @@ static int soc_init_dai_link(struct snd_soc_card *card,
 		return ret;
 	}
 
-	for (i = 0; i < link->num_codecs; i++) {
+	for_each_link_codecs(link, i, codec) {
 		/*
 		 * Codec must be specified by 1 of name or OF node,
 		 * not both or neither.
 		 */
-		if (!!link->codecs[i].name ==
-		    !!link->codecs[i].of_node) {
+		if (!!codec->name ==
+		    !!codec->of_node) {
 			dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n",
 				link->name);
 			return -EINVAL;
 		}
 		/* Codec DAI name must be specified */
-		if (!link->codecs[i].dai_name) {
+		if (!codec->dai_name) {
 			dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n",
 				link->name);
 			return -EINVAL;
@@ -1076,13 +1123,12 @@ static int soc_init_dai_link(struct snd_soc_card *card,
 	 * Platform may be specified by either name or OF node, but
 	 * can be left unspecified, and a dummy platform will be used.
 	 */
-	if (link->platform_name && link->platform_of_node) {
+	if (link->platform->name && link->platform->of_node) {
 		dev_err(card->dev,
 			"ASoC: Both platform name/of_node are set for %s\n",
 			link->name);
 		return -EINVAL;
 	}
-
 	/*
 	 * CPU device may be specified by either name or OF node, but
 	 * can be left unspecified, and will be matched based on DAI
@@ -1111,7 +1157,8 @@ static int soc_init_dai_link(struct snd_soc_card *card,
 
 void snd_soc_disconnect_sync(struct device *dev)
 {
-	struct snd_soc_component *component = snd_soc_lookup_component(dev, NULL);
+	struct snd_soc_component *component =
+			snd_soc_lookup_component(dev, NULL);
 
 	if (!component || !component->card)
 		return;
@@ -1142,14 +1189,14 @@ int snd_soc_add_dai_link(struct snd_soc_card *card,
 	}
 
 	lockdep_assert_held(&client_mutex);
-	/* Notify the machine driver for extra initialization
+	/*
+	 * Notify the machine driver for extra initialization
 	 * on the link created by topology.
 	 */
 	if (dai_link->dobj.type && card->add_dai_link)
 		card->add_dai_link(card, dai_link);
 
 	list_add_tail(&dai_link->list, &card->dai_link_list);
-	card->num_dai_links++;
 
 	return 0;
 }
@@ -1178,16 +1225,16 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card,
 	}
 
 	lockdep_assert_held(&client_mutex);
-	/* Notify the machine driver for extra destruction
+	/*
+	 * Notify the machine driver for extra destruction
 	 * on the link created by topology.
 	 */
 	if (dai_link->dobj.type && card->remove_dai_link)
 		card->remove_dai_link(card, dai_link);
 
-	list_for_each_entry_safe(link, _link, &card->dai_link_list, list) {
+	for_each_card_links_safe(card, link, _link) {
 		if (link == dai_link) {
 			list_del(&link->list);
-			card->num_dai_links--;
 			return;
 		}
 	}
@@ -1239,7 +1286,8 @@ static void soc_set_name_prefix(struct snd_soc_card *card,
 static int soc_probe_component(struct snd_soc_card *card,
 	struct snd_soc_component *component)
 {
-	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+	struct snd_soc_dapm_context *dapm =
+			snd_soc_component_get_dapm(component);
 	struct snd_soc_dai *dai;
 	int ret;
 
@@ -1277,7 +1325,7 @@ static int soc_probe_component(struct snd_soc_card *card,
 		}
 	}
 
-	list_for_each_entry(dai, &component->dai_list, list) {
+	for_each_component_dais(component, dai) {
 		ret = snd_soc_dapm_new_dai_widgets(dapm, dai);
 		if (ret != 0) {
 			dev_err(component->dev,
@@ -1320,6 +1368,7 @@ static int soc_probe_component(struct snd_soc_card *card,
 					component->driver->num_dapm_routes);
 
 	list_add(&dapm->list, &card->dapm_list);
+	/* see for_each_card_components */
 	list_add(&component->card_list, &card->component_dev_list);
 
 	return 0;
@@ -1370,8 +1419,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
 }
 
 static int soc_probe_link_components(struct snd_soc_card *card,
-			struct snd_soc_pcm_runtime *rtd,
-				     int order)
+				     struct snd_soc_pcm_runtime *rtd, int order)
 {
 	struct snd_soc_component *component;
 	struct snd_soc_rtdcom_list *rtdcom;
@@ -1398,6 +1446,7 @@ static int soc_probe_dai(struct snd_soc_dai *dai, int order)
 
 	if (dai->driver->probe) {
 		int ret = dai->driver->probe(dai);
+
 		if (ret < 0) {
 			dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n",
 				dai->name, ret);
@@ -1431,48 +1480,6 @@ static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais,
 	return 0;
 }
 
-static int soc_link_dai_widgets(struct snd_soc_card *card,
-				struct snd_soc_dai_link *dai_link,
-				struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dapm_widget *sink, *source;
-	int ret;
-
-	if (rtd->num_codecs > 1)
-		dev_warn(card->dev, "ASoC: Multiple codecs not supported yet\n");
-
-	/* link the DAI widgets */
-	sink = codec_dai->playback_widget;
-	source = cpu_dai->capture_widget;
-	if (sink && source) {
-		ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params,
-					   dai_link->num_params,
-					   source, sink);
-		if (ret != 0) {
-			dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n",
-				sink->name, source->name, ret);
-			return ret;
-		}
-	}
-
-	sink = cpu_dai->playback_widget;
-	source = codec_dai->capture_widget;
-	if (sink && source) {
-		ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params,
-					   dai_link->num_params,
-					   source, sink);
-		if (ret != 0) {
-			dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n",
-				sink->name, source->name, ret);
-			return ret;
-		}
-	}
-
-	return 0;
-}
-
 static int soc_probe_link_dais(struct snd_soc_card *card,
 		struct snd_soc_pcm_runtime *rtd, int order)
 {
@@ -1480,6 +1487,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_rtdcom_list *rtdcom;
 	struct snd_soc_component *component;
+	struct snd_soc_dai *codec_dai;
 	int i, ret, num;
 
 	dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n",
@@ -1493,8 +1501,8 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 		return ret;
 
 	/* probe the CODEC DAI */
-	for (i = 0; i < rtd->num_codecs; i++) {
-		ret = soc_probe_dai(rtd->codec_dais[i], order);
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		ret = soc_probe_dai(codec_dai, order);
 		if (ret)
 			return ret;
 	}
@@ -1546,7 +1554,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 	}
 
 	if (cpu_dai->driver->compress_new) {
-		/*create compress_device"*/
+		/* create compress_device" */
 		ret = cpu_dai->driver->compress_new(rtd, num);
 		if (ret < 0) {
 			dev_err(card->dev, "ASoC: can't create compress %s\n",
@@ -1560,7 +1568,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 			ret = soc_new_pcm(rtd, num);
 			if (ret < 0) {
 				dev_err(card->dev, "ASoC: can't create pcm %s :%d\n",
-				       dai_link->stream_name, ret);
+					dai_link->stream_name, ret);
 				return ret;
 			}
 			ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd);
@@ -1573,11 +1581,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 		} else {
 			INIT_DELAYED_WORK(&rtd->delayed_work,
 						codec2codec_close_delayed_work);
-
-			/* link the DAI widgets */
-			ret = soc_link_dai_widgets(card, dai_link, rtd);
-			if (ret)
-				return ret;
 		}
 	}
 
@@ -1628,8 +1631,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card)
 	int order;
 	int ret;
 
-	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-		order++) {
+	for_each_comp_order(order) {
 		list_for_each_entry(comp, &card->aux_comp_list, card_aux_list) {
 			if (comp->driver->probe_order == order) {
 				ret = soc_probe_component(card,	comp);
@@ -1651,8 +1653,7 @@ static void soc_remove_aux_devices(struct snd_soc_card *card)
 	struct snd_soc_component *comp, *_comp;
 	int order;
 
-	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-		order++) {
+	for_each_comp_order(order) {
 		list_for_each_entry_safe(comp, _comp,
 			&card->aux_comp_list, card_aux_list) {
 
@@ -1681,14 +1682,12 @@ static void soc_remove_aux_devices(struct snd_soc_card *card)
 int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
 	unsigned int dai_fmt)
 {
-	struct snd_soc_dai **codec_dais = rtd->codec_dais;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai;
 	unsigned int i;
 	int ret;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		struct snd_soc_dai *codec_dai = codec_dais[i];
-
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt);
 		if (ret != 0 && ret != -ENOTSUPP) {
 			dev_warn(codec_dai->dev,
@@ -1697,8 +1696,10 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
 		}
 	}
 
-	/* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */
-	/* the component which has non_legacy_dai_naming is Codec */
+	/*
+	 * Flip the polarity for the "CPU" end of a CODEC<->CODEC link
+	 * the component which has non_legacy_dai_naming is Codec
+	 */
 	if (cpu_dai->component->driver->non_legacy_dai_naming) {
 		unsigned int inv_dai_fmt;
 
@@ -1732,9 +1733,9 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
 }
 EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt);
 
-
 #ifdef CONFIG_DMI
-/* Trim special characters, and replace '-' with '_' since '-' is used to
+/*
+ * Trim special characters, and replace '-' with '_' since '-' is used to
  * separate different DMI fields in the card long name. Only number and
  * alphabet characters and a few separator characters are kept.
  */
@@ -1753,7 +1754,8 @@ static void cleanup_dmi_name(char *name)
 	name[j] = '\0';
 }
 
-/* Check if a DMI field is valid, i.e. not containing any string
+/*
+ * Check if a DMI field is valid, i.e. not containing any string
  * in the black list.
  */
 static int is_dmi_valid(const char *field)
@@ -1816,7 +1818,6 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour)
 		return 0;
 	}
 
-
 	snprintf(card->dmi_longname, sizeof(card->snd_card->longname),
 			 "%s", vendor);
 	cleanup_dmi_name(card->dmi_longname);
@@ -1832,7 +1833,8 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour)
 		if (len < longname_buf_size)
 			cleanup_dmi_name(card->dmi_longname + len);
 
-		/* some vendors like Lenovo may only put a self-explanatory
+		/*
+		 * some vendors like Lenovo may only put a self-explanatory
 		 * name in the product version field
 		 */
 		product_version = dmi_get_system_info(DMI_PRODUCT_VERSION);
@@ -1891,7 +1893,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
 	struct snd_soc_dai_link *dai_link;
 	int i;
 
-	list_for_each_entry(component, &component_list, list) {
+	for_each_component(component) {
 
 		/* does this component override FEs ? */
 		if (!component->driver->ignore_machine)
@@ -1903,9 +1905,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
 			continue;
 
 		/* machine matches, so override the rtd data */
-		for (i = 0; i < card->num_links; i++) {
-
-			dai_link = &card->dai_link[i];
+		for_each_card_prelinks(card, i, dai_link) {
 
 			/* ignore this FE */
 			if (dai_link->dynamic) {
@@ -1917,7 +1917,11 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
 				 card->dai_link[i].name);
 
 			/* override platform component */
-			dai_link->platform_name = component->name;
+			if (snd_soc_init_platform(card, dai_link) < 0) {
+				dev_err(card->dev, "init platform error");
+				continue;
+			}
+			dai_link->platform->name = component->name;
 
 			/* convert non BE into BE */
 			dai_link->no_pcm = 1;
@@ -1926,7 +1930,8 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
 			dai_link->be_hw_params_fixup =
 				component->driver->be_hw_params_fixup;
 
-			/* most BE links don't set stream name, so set it to
+			/*
+			 * most BE links don't set stream name, so set it to
 			 * dai link name if it's NULL to help bind widgets.
 			 */
 			if (!dai_link->stream_name)
@@ -1936,7 +1941,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
 		/* Inform userspace we are using alternate topology */
 		if (component->driver->topology_name_prefix) {
 
-			/* topology shortname created ? */
+			/* topology shortname created? */
 			if (!card->topology_shortname_created) {
 				comp_drv = component->driver;
 
@@ -1965,8 +1970,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	soc_check_tplg_fes(card);
 
 	/* bind DAIs */
-	for (i = 0; i < card->num_links; i++) {
-		ret = soc_bind_dai_link(card, &card->dai_link[i]);
+	for_each_card_prelinks(card, i, dai_link) {
+		ret = soc_bind_dai_link(card, dai_link);
 		if (ret != 0)
 			goto base_error;
 	}
@@ -1979,8 +1984,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	/* add predefined DAI links to the list */
-	for (i = 0; i < card->num_links; i++)
-		snd_soc_add_dai_link(card, card->dai_link+i);
+	for_each_card_prelinks(card, i, dai_link)
+		snd_soc_add_dai_link(card, dai_link);
 
 	/* card bind complete so register a sound card */
 	ret = snd_card_new(card->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
@@ -2024,9 +2029,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	/* probe all components used by DAI links on this card */
-	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-			order++) {
-		list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_comp_order(order) {
+		for_each_card_rtds(card, rtd) {
 			ret = soc_probe_link_components(card, rtd, order);
 			if (ret < 0) {
 				dev_err(card->dev,
@@ -2042,10 +2046,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	if (ret < 0)
 		goto probe_dai_err;
 
-	/* Find new DAI links added during probing components and bind them.
+	/*
+	 * Find new DAI links added during probing components and bind them.
 	 * Components with topology may bring new DAIs and DAI links.
 	 */
-	list_for_each_entry(dai_link, &card->dai_link_list, list) {
+	for_each_card_links(card, dai_link) {
 		if (soc_is_dai_link_bound(card, dai_link))
 			continue;
 
@@ -2058,9 +2063,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	/* probe all DAI links on this card */
-	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-			order++) {
-		list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_comp_order(order) {
+		for_each_card_rtds(card, rtd) {
 			ret = soc_probe_link_dais(card, rtd, order);
 			if (ret < 0) {
 				dev_err(card->dev,
@@ -2075,7 +2079,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	snd_soc_dapm_connect_dai_link_widgets(card);
 
 	if (card->controls)
-		snd_soc_add_card_controls(card, card->controls, card->num_controls);
+		snd_soc_add_card_controls(card, card->controls,
+					  card->num_controls);
 
 	if (card->dapm_routes)
 		snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
@@ -2181,7 +2186,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card)
 	struct snd_soc_pcm_runtime *rtd;
 
 	/* make sure any delayed work runs */
-	list_for_each_entry(rtd, &card->rtd_list, list)
+	for_each_card_rtds(card, rtd)
 		flush_delayed_work(&rtd->delayed_work);
 
 	/* free the ALSA card at first; this syncs with pending operations */
@@ -2221,21 +2226,23 @@ int snd_soc_poweroff(struct device *dev)
 	if (!card->instantiated)
 		return 0;
 
-	/* Flush out pmdown_time work - we actually do want to run it
-	 * now, we're shutting down so no imminent restart. */
-	list_for_each_entry(rtd, &card->rtd_list, list)
+	/*
+	 * Flush out pmdown_time work - we actually do want to run it
+	 * now, we're shutting down so no imminent restart.
+	 */
+	for_each_card_rtds(card, rtd)
 		flush_delayed_work(&rtd->delayed_work);
 
 	snd_soc_dapm_shutdown(card);
 
 	/* deactivate pins to sleep state */
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+		struct snd_soc_dai *codec_dai;
 		int i;
 
 		pinctrl_pm_select_sleep_state(cpu_dai->dev);
-		for (i = 0; i < rtd->num_codecs; i++) {
-			struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+		for_each_rtd_codec_dai(rtd, i, codec_dai) {
 			pinctrl_pm_select_sleep_state(codec_dai->dev);
 		}
 	}
@@ -2315,6 +2322,7 @@ static int snd_soc_add_controls(struct snd_card *card, struct device *dev,
 
 	for (i = 0; i < num_controls; i++) {
 		const struct snd_kcontrol_new *control = &controls[i];
+
 		err = snd_ctl_add(card, snd_soc_cnew(control, data,
 						     control->name, prefix));
 		if (err < 0) {
@@ -2432,8 +2440,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
  *
  * Configures the CODEC master (MCLK) or system (SYSCLK) clocking.
  */
-int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id,
-			     int source, unsigned int freq, int dir)
+int snd_soc_component_set_sysclk(struct snd_soc_component *component,
+				 int clk_id, int source, unsigned int freq,
+				 int dir)
 {
 	if (component->driver->set_sysclk)
 		return component->driver->set_sysclk(component, clk_id, source,
@@ -2501,7 +2510,7 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id,
 {
 	if (component->driver->set_pll)
 		return component->driver->set_pll(component, pll_id, source,
-					      freq_in, freq_out);
+						  freq_in, freq_out);
 
 	return -EINVAL;
 }
@@ -2532,8 +2541,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio);
  */
 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
-	if (dai->driver == NULL)
-		return -EINVAL;
 	if (dai->driver->ops->set_fmt == NULL)
 		return -ENOTSUPP;
 	return dai->driver->ops->set_fmt(dai, fmt);
@@ -2549,8 +2556,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
  * Generates the TDM tx and rx slot default masks for DAI.
  */
 static int snd_soc_xlate_tdm_slot_mask(unsigned int slots,
-					  unsigned int *tx_mask,
-					  unsigned int *rx_mask)
+				       unsigned int *tx_mask,
+				       unsigned int *rx_mask)
 {
 	if (*tx_mask || *rx_mask)
 		return 0;
@@ -2680,9 +2687,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
 			     int direction)
 {
-	if (!dai->driver)
-		return -ENOTSUPP;
-
 	if (dai->driver->ops->mute_stream)
 		return dai->driver->ops->mute_stream(dai, mute, direction);
 	else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
@@ -2693,6 +2697,33 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
 
+static int snd_soc_bind_card(struct snd_soc_card *card)
+{
+	struct snd_soc_pcm_runtime *rtd;
+	int ret;
+
+	ret = snd_soc_instantiate_card(card);
+	if (ret != 0)
+		return ret;
+
+	/* deactivate pins to sleep state */
+	for_each_card_rtds(card, rtd) {
+		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+		struct snd_soc_dai *codec_dai;
+		int j;
+
+		for_each_rtd_codec_dai(rtd, j, codec_dai) {
+			if (!codec_dai->active)
+				pinctrl_pm_select_sleep_state(codec_dai->dev);
+		}
+
+		if (!cpu_dai->active)
+			pinctrl_pm_select_sleep_state(cpu_dai->dev);
+	}
+
+	return ret;
+}
+
 /**
  * snd_soc_register_card - Register a card with the ASoC core
  *
@@ -2702,13 +2733,12 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
 int snd_soc_register_card(struct snd_soc_card *card)
 {
 	int i, ret;
-	struct snd_soc_pcm_runtime *rtd;
+	struct snd_soc_dai_link *link;
 
 	if (!card->name || !card->dev)
 		return -EINVAL;
 
-	for (i = 0; i < card->num_links; i++) {
-		struct snd_soc_dai_link *link = &card->dai_link[i];
+	for_each_card_prelinks(card, i, link) {
 
 		ret = soc_init_dai_link(card, link);
 		if (ret) {
@@ -2723,7 +2753,6 @@ int snd_soc_register_card(struct snd_soc_card *card)
 	snd_soc_initialize_card_lists(card);
 
 	INIT_LIST_HEAD(&card->dai_link_list);
-	card->num_dai_links = 0;
 
 	INIT_LIST_HEAD(&card->rtd_list);
 	card->num_rtd = 0;
@@ -2734,28 +2763,23 @@ int snd_soc_register_card(struct snd_soc_card *card)
 	mutex_init(&card->mutex);
 	mutex_init(&card->dapm_mutex);
 
-	ret = snd_soc_instantiate_card(card);
-	if (ret != 0)
-		return ret;
-
-	/* deactivate pins to sleep state */
-	list_for_each_entry(rtd, &card->rtd_list, list)  {
-		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-		int j;
-
-		for (j = 0; j < rtd->num_codecs; j++) {
-			struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
-			if (!codec_dai->active)
-				pinctrl_pm_select_sleep_state(codec_dai->dev);
-		}
+	return snd_soc_bind_card(card);
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_card);
 
-		if (!cpu_dai->active)
-			pinctrl_pm_select_sleep_state(cpu_dai->dev);
+static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister)
+{
+	if (card->instantiated) {
+		card->instantiated = false;
+		snd_soc_dapm_shutdown(card);
+		soc_cleanup_card_resources(card);
+		if (!unregister)
+			list_add(&card->list, &unbind_card_list);
+	} else {
+		if (unregister)
+			list_del(&card->list);
 	}
-
-	return ret;
 }
-EXPORT_SYMBOL_GPL(snd_soc_register_card);
 
 /**
  * snd_soc_unregister_card - Unregister a card with the ASoC core
@@ -2765,12 +2789,8 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card);
  */
 int snd_soc_unregister_card(struct snd_soc_card *card)
 {
-	if (card->instantiated) {
-		card->instantiated = false;
-		snd_soc_dapm_shutdown(card);
-		soc_cleanup_card_resources(card);
-		dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
-	}
+	snd_soc_unbind_card(card, true);
+	dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
 
 	return 0;
 }
@@ -2802,7 +2822,7 @@ static char *fmt_single_name(struct device *dev, int *id)
 		}
 
 	} else {
-		/* I2C component devices are named "bus-addr"  */
+		/* I2C component devices are named "bus-addr" */
 		if (sscanf(name, "%x-%x", &id1, &id2) == 2) {
 			char tmp[NAME_SIZE];
 
@@ -2810,7 +2830,8 @@ static char *fmt_single_name(struct device *dev, int *id)
 			*id = ((id1 & 0xffff) << 16) + id2;
 
 			/* sanitize component name for DAI link creation */
-			snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name, name);
+			snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name,
+				 name);
 			strlcpy(name, tmp, NAME_SIZE);
 		} else
 			*id = 0;
@@ -2845,7 +2866,7 @@ static void snd_soc_unregister_dais(struct snd_soc_component *component)
 {
 	struct snd_soc_dai *dai, *_dai;
 
-	list_for_each_entry_safe(dai, _dai, &component->dai_list, list) {
+	for_each_component_dais_safe(component, dai, _dai) {
 		dev_dbg(component->dev, "ASoC: Unregistered DAI '%s'\n",
 			dai->name);
 		list_del(&dai->list);
@@ -2877,7 +2898,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component,
 	 * component-less anymore.
 	 */
 	if (legacy_dai_naming &&
-	   (dai_drv->id == 0 || dai_drv->name == NULL)) {
+	    (dai_drv->id == 0 || dai_drv->name == NULL)) {
 		dai->name = fmt_single_name(dev, &dai->id);
 	} else {
 		dai->name = fmt_multiple_name(dev, dai_drv);
@@ -2897,6 +2918,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component,
 	if (!dai->driver->ops)
 		dai->driver->ops = &null_dai_ops;
 
+	/* see for_each_component_dais */
 	list_add_tail(&dai->list, &component->dai_list);
 	component->num_dai++;
 
@@ -2910,11 +2932,10 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component,
  * @component: The component the DAIs are registered for
  * @dai_drv: DAI driver to use for the DAIs
  * @count: Number of DAIs
- * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the
- *                     parent's name.
  */
 static int snd_soc_register_dais(struct snd_soc_component *component,
-				 struct snd_soc_dai_driver *dai_drv, size_t count)
+				 struct snd_soc_dai_driver *dai_drv,
+				 size_t count)
 {
 	struct device *dev = component->dev;
 	struct snd_soc_dai *dai;
@@ -2925,8 +2946,8 @@ static int snd_soc_register_dais(struct snd_soc_component *component,
 
 	for (i = 0; i < count; i++) {
 
-		dai = soc_add_dai(component, dai_drv + i,
-				  count == 1 && !component->driver->non_legacy_dai_naming);
+		dai = soc_add_dai(component, dai_drv + i, count == 1 &&
+				  !component->driver->non_legacy_dai_naming);
 		if (dai == NULL) {
 			ret = -ENOMEM;
 			goto err;
@@ -2970,7 +2991,8 @@ int snd_soc_register_dai(struct snd_soc_component *component,
 	if (!dai)
 		return -ENOMEM;
 
-	/* Create the DAI widgets here. After adding DAIs, topology may
+	/*
+	 * Create the DAI widgets here. After adding DAIs, topology may
 	 * also add routes that need these widgets as source or sink.
 	 */
 	ret = snd_soc_dapm_new_dai_widgets(dapm, dai);
@@ -3052,7 +3074,8 @@ static void snd_soc_component_setup_regmap(struct snd_soc_component *component)
 #ifdef CONFIG_REGMAP
 
 /**
- * snd_soc_component_init_regmap() - Initialize regmap instance for the component
+ * snd_soc_component_init_regmap() - Initialize regmap instance for the
+ *                                   component
  * @component: The component for which to initialize the regmap instance
  * @regmap: The regmap instance that should be used by the component
  *
@@ -3070,7 +3093,8 @@ void snd_soc_component_init_regmap(struct snd_soc_component *component,
 EXPORT_SYMBOL_GPL(snd_soc_component_init_regmap);
 
 /**
- * snd_soc_component_exit_regmap() - De-initialize regmap instance for the component
+ * snd_soc_component_exit_regmap() - De-initialize regmap instance for the
+ *                                   component
  * @component: The component for which to de-initialize the regmap instance
  *
  * Calls regmap_exit() on the regmap instance associated to the component and
@@ -3094,11 +3118,13 @@ static void snd_soc_component_add(struct snd_soc_component *component)
 
 	if (!component->driver->write && !component->driver->read) {
 		if (!component->regmap)
-			component->regmap = dev_get_regmap(component->dev, NULL);
+			component->regmap = dev_get_regmap(component->dev,
+							   NULL);
 		if (component->regmap)
 			snd_soc_component_setup_regmap(component);
 	}
 
+	/* see for_each_component */
 	list_add(&component->list, &component_list);
 	INIT_LIST_HEAD(&component->dobj_list);
 
@@ -3116,7 +3142,7 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component)
 	struct snd_soc_card *card = component->card;
 
 	if (card)
-		snd_soc_unregister_card(card);
+		snd_soc_unbind_card(card, false);
 
 	list_del(&component->list);
 }
@@ -3156,6 +3182,18 @@ static void convert_endianness_formats(struct snd_soc_pcm_stream *stream)
 			stream->formats |= endianness_format_map[i];
 }
 
+static void snd_soc_try_rebind_card(void)
+{
+	struct snd_soc_card *card, *c;
+
+	if (!list_empty(&unbind_card_list)) {
+		list_for_each_entry_safe(card, c, &unbind_card_list, list) {
+			if (!snd_soc_bind_card(card))
+				list_del(&card->list);
+		}
+	}
+}
+
 int snd_soc_add_component(struct device *dev,
 			struct snd_soc_component *component,
 			const struct snd_soc_component_driver *component_driver,
@@ -3183,6 +3221,7 @@ int snd_soc_add_component(struct device *dev,
 	}
 
 	snd_soc_component_add(component);
+	snd_soc_try_rebind_card();
 
 	return 0;
 
@@ -3221,27 +3260,28 @@ static int __snd_soc_unregister_component(struct device *dev)
 	int found = 0;
 
 	mutex_lock(&client_mutex);
-	list_for_each_entry(component, &component_list, list) {
+	for_each_component(component) {
 		if (dev != component->dev)
 			continue;
 
-		snd_soc_tplg_component_remove(component, SND_SOC_TPLG_INDEX_ALL);
+		snd_soc_tplg_component_remove(component,
+					      SND_SOC_TPLG_INDEX_ALL);
 		snd_soc_component_del_unlocked(component);
 		found = 1;
 		break;
 	}
 	mutex_unlock(&client_mutex);
 
-	if (found) {
+	if (found)
 		snd_soc_component_cleanup(component);
-	}
 
 	return found;
 }
 
 void snd_soc_unregister_component(struct device *dev)
 {
-	while (__snd_soc_unregister_component(dev));
+	while (__snd_soc_unregister_component(dev))
+		;
 }
 EXPORT_SYMBOL_GPL(snd_soc_unregister_component);
 
@@ -3253,7 +3293,7 @@ struct snd_soc_component *snd_soc_lookup_component(struct device *dev,
 
 	ret = NULL;
 	mutex_lock(&client_mutex);
-	list_for_each_entry(component, &component_list, list) {
+	for_each_component(component) {
 		if (dev != component->dev)
 			continue;
 
@@ -3653,7 +3693,7 @@ int snd_soc_get_dai_id(struct device_node *ep)
 	 */
 	ret = -ENOTSUPP;
 	mutex_lock(&client_mutex);
-	list_for_each_entry(pos, &component_list, list) {
+	for_each_component(pos) {
 		struct device_node *component_of_node = pos->dev->of_node;
 
 		if (!component_of_node && pos->dev->parent)
@@ -3683,7 +3723,7 @@ int snd_soc_get_dai_name(struct of_phandle_args *args,
 	int ret = -EPROBE_DEFER;
 
 	mutex_lock(&client_mutex);
-	list_for_each_entry(pos, &component_list, list) {
+	for_each_component(pos) {
 		component_of_node = pos->dev->of_node;
 		if (!component_of_node && pos->dev->parent)
 			component_of_node = pos->dev->parent->of_node;
@@ -3719,7 +3759,7 @@ int snd_soc_get_dai_name(struct of_phandle_args *args,
 			ret = 0;
 
 			/* find target DAI */
-			list_for_each_entry(dai, &pos->dai_list, list) {
+			for_each_component_dais(pos, dai) {
 				if (id == 0)
 					break;
 				id--;
@@ -3764,10 +3804,10 @@ EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name);
  */
 void snd_soc_of_put_dai_link_codecs(struct snd_soc_dai_link *dai_link)
 {
-	struct snd_soc_dai_link_component *component = dai_link->codecs;
+	struct snd_soc_dai_link_component *component;
 	int index;
 
-	for (index = 0; index < dai_link->num_codecs; index++, component++) {
+	for_each_link_codecs(dai_link, index, component) {
 		if (!component->of_node)
 			break;
 		of_node_put(component->of_node);
@@ -3819,12 +3859,10 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev,
 	dai_link->num_codecs = num_codecs;
 
 	/* Parse the list */
-	for (index = 0, component = dai_link->codecs;
-	     index < dai_link->num_codecs;
-	     index++, component++) {
+	for_each_link_codecs(dai_link, index, component) {
 		ret = of_parse_phandle_with_args(of_node, name,
 						 "#sound-dai-cells",
-						  index, &args);
+						 index, &args);
 		if (ret)
 			goto err;
 		component->of_node = args.np;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 461d951917c0..a5178845065b 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -18,7 +18,6 @@
 //      device reopen.
 
 #include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/init.h>
 #include <linux/async.h>
 #include <linux/delay.h>
@@ -364,10 +363,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 				ret = PTR_ERR(data->widget);
 				goto err_data;
 			}
-			if (!data->widget) {
-				ret = -ENOMEM;
-				goto err_data;
-			}
 		}
 		break;
 	case snd_soc_dapm_demux:
@@ -402,10 +397,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 				ret = PTR_ERR(data->widget);
 				goto err_data;
 			}
-			if (!data->widget) {
-				ret = -ENOMEM;
-				goto err_data;
-			}
 
 			snd_soc_dapm_add_path(widget->dapm, data->widget,
 					      widget, NULL, NULL);
@@ -1026,9 +1017,10 @@ static int dapm_new_dai_link(struct snd_soc_dapm_widget *w)
 	struct snd_kcontrol *kcontrol;
 	struct snd_soc_dapm_context *dapm = w->dapm;
 	struct snd_card *card = dapm->card->snd_card;
+	struct snd_soc_pcm_runtime *rtd = w->priv;
 
 	/* create control for links with > 1 config */
-	if (w->num_params <= 1)
+	if (rtd->dai_link->num_params <= 1)
 		return 0;
 
 	/* add kcontrol */
@@ -1320,14 +1312,13 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w,
 
 	soc_dapm_async_complete(w->dapm);
 
-#ifdef CONFIG_HAVE_CLK
 	if (SND_SOC_DAPM_EVENT_ON(event)) {
 		return clk_prepare_enable(w->clk);
 	} else {
 		clk_disable_unprepare(w->clk);
 		return 0;
 	}
-#endif
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(dapm_clock_event);
@@ -1953,7 +1944,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
 	dapm_pre_sequence_async(&card->dapm, 0);
 	/* Run other bias changes in parallel */
 	list_for_each_entry(d, &card->dapm_list, list) {
-		if (d != &card->dapm)
+		if (d != &card->dapm && d->bias_level != d->target_bias_level)
 			async_schedule_domain(dapm_pre_sequence_async, d,
 						&async_domain);
 	}
@@ -1977,7 +1968,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
 
 	/* Run all the bias changes in parallel */
 	list_for_each_entry(d, &card->dapm_list, list) {
-		if (d != &card->dapm)
+		if (d != &card->dapm && d->bias_level != d->target_bias_level)
 			async_schedule_domain(dapm_post_sequence_async, d,
 						&async_domain);
 	}
@@ -2371,12 +2362,13 @@ static ssize_t dapm_widget_show(struct device *dev,
 	struct device_attribute *attr, char *buf)
 {
 	struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
+	struct snd_soc_dai *codec_dai;
 	int i, count = 0;
 
 	mutex_lock(&rtd->card->dapm_mutex);
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		struct snd_soc_component *cmpnt = rtd->codec_dais[i]->component;
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		struct snd_soc_component *cmpnt = codec_dai->component;
 
 		count += dapm_widget_show_component(cmpnt, buf + count);
 	}
@@ -3426,35 +3418,6 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch);
 
 struct snd_soc_dapm_widget *
-snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
-	const struct snd_soc_dapm_widget *widget)
-{
-	struct snd_soc_dapm_widget *w;
-
-	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
-	w = snd_soc_dapm_new_control_unlocked(dapm, widget);
-	/* Do not nag about probe deferrals */
-	if (IS_ERR(w)) {
-		int ret = PTR_ERR(w);
-
-		if (ret != -EPROBE_DEFER)
-			dev_err(dapm->dev,
-				"ASoC: Failed to create DAPM control %s (%d)\n",
-				widget->name, ret);
-		goto out_unlock;
-	}
-	if (!w)
-		dev_err(dapm->dev,
-			"ASoC: Failed to create DAPM control %s\n",
-			widget->name);
-
-out_unlock:
-	mutex_unlock(&dapm->card->dapm_mutex);
-	return w;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
-
-struct snd_soc_dapm_widget *
 snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
 			 const struct snd_soc_dapm_widget *widget)
 {
@@ -3464,53 +3427,37 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
 	int ret;
 
 	if ((w = dapm_cnew_widget(widget)) == NULL)
-		return NULL;
+		return ERR_PTR(-ENOMEM);
 
 	switch (w->id) {
 	case snd_soc_dapm_regulator_supply:
 		w->regulator = devm_regulator_get(dapm->dev, w->name);
 		if (IS_ERR(w->regulator)) {
 			ret = PTR_ERR(w->regulator);
-			if (ret == -EPROBE_DEFER)
-				return ERR_PTR(ret);
-			dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
-				w->name, ret);
-			return NULL;
+			goto request_failed;
 		}
 
 		if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) {
 			ret = regulator_allow_bypass(w->regulator, true);
 			if (ret != 0)
-				dev_warn(w->dapm->dev,
+				dev_warn(dapm->dev,
 					 "ASoC: Failed to bypass %s: %d\n",
 					 w->name, ret);
 		}
 		break;
 	case snd_soc_dapm_pinctrl:
 		w->pinctrl = devm_pinctrl_get(dapm->dev);
-		if (IS_ERR_OR_NULL(w->pinctrl)) {
+		if (IS_ERR(w->pinctrl)) {
 			ret = PTR_ERR(w->pinctrl);
-			if (ret == -EPROBE_DEFER)
-				return ERR_PTR(ret);
-			dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
-				w->name, ret);
-			return NULL;
+			goto request_failed;
 		}
 		break;
 	case snd_soc_dapm_clock_supply:
-#ifdef CONFIG_CLKDEV_LOOKUP
 		w->clk = devm_clk_get(dapm->dev, w->name);
 		if (IS_ERR(w->clk)) {
 			ret = PTR_ERR(w->clk);
-			if (ret == -EPROBE_DEFER)
-				return ERR_PTR(ret);
-			dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
-				w->name, ret);
-			return NULL;
+			goto request_failed;
 		}
-#else
-		return NULL;
-#endif
 		break;
 	default:
 		break;
@@ -3523,7 +3470,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
 		w->name = kstrdup_const(widget->name, GFP_KERNEL);
 	if (w->name == NULL) {
 		kfree(w);
-		return NULL;
+		return ERR_PTR(-ENOMEM);
 	}
 
 	switch (w->id) {
@@ -3600,7 +3547,37 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
 	/* machine layer sets up unconnected pins and insertions */
 	w->connected = 1;
 	return w;
+
+request_failed:
+	if (ret != -EPROBE_DEFER)
+		dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
+			w->name, ret);
+
+	return ERR_PTR(ret);
+}
+
+/**
+ * snd_soc_dapm_new_control - create new dapm control
+ * @dapm: DAPM context
+ * @widget: widget template
+ *
+ * Creates new DAPM control based upon a template.
+ *
+ * Returns a widget pointer on success or an error pointer on failure
+ */
+struct snd_soc_dapm_widget *
+snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
+			 const struct snd_soc_dapm_widget *widget)
+{
+	struct snd_soc_dapm_widget *w;
+
+	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+	w = snd_soc_dapm_new_control_unlocked(dapm, widget);
+	mutex_unlock(&dapm->card->dapm_mutex);
+
+	return w;
 }
+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
 
 /**
  * snd_soc_dapm_new_controls - create new dapm controls
@@ -3625,19 +3602,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
 		w = snd_soc_dapm_new_control_unlocked(dapm, widget);
 		if (IS_ERR(w)) {
 			ret = PTR_ERR(w);
-			/* Do not nag about probe deferrals */
-			if (ret == -EPROBE_DEFER)
-				break;
-			dev_err(dapm->dev,
-				"ASoC: Failed to create DAPM control %s (%d)\n",
-				widget->name, ret);
-			break;
-		}
-		if (!w) {
-			dev_err(dapm->dev,
-				"ASoC: Failed to create DAPM control %s\n",
-				widget->name);
-			ret = -ENOMEM;
 			break;
 		}
 		widget++;
@@ -3650,32 +3614,23 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls);
 static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 				  struct snd_kcontrol *kcontrol, int event)
 {
-	struct snd_soc_dapm_path *source_p, *sink_p;
+	struct snd_soc_dapm_path *path;
 	struct snd_soc_dai *source, *sink;
 	struct snd_soc_pcm_runtime *rtd = w->priv;
-	const struct snd_soc_pcm_stream *config = w->params + w->params_select;
+	const struct snd_soc_pcm_stream *config;
 	struct snd_pcm_substream substream;
 	struct snd_pcm_hw_params *params = NULL;
 	struct snd_pcm_runtime *runtime = NULL;
 	unsigned int fmt;
-	int ret;
+	int ret = 0;
+
+	config = rtd->dai_link->params + rtd->params_select;
 
 	if (WARN_ON(!config) ||
 	    WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) ||
 		    list_empty(&w->edges[SND_SOC_DAPM_DIR_IN])))
 		return -EINVAL;
 
-	/* We only support a single source and sink, pick the first */
-	source_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_OUT],
-				    struct snd_soc_dapm_path,
-				    list_node[SND_SOC_DAPM_DIR_OUT]);
-	sink_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_IN],
-				    struct snd_soc_dapm_path,
-				    list_node[SND_SOC_DAPM_DIR_IN]);
-
-	source = source_p->source->priv;
-	sink = sink_p->sink->priv;
-
 	/* Be a little careful as we don't want to overflow the mask array */
 	if (config->formats) {
 		fmt = ffs(config->formats) - 1;
@@ -3717,59 +3672,95 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
 		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
-		if (source->driver->ops->startup) {
-			ret = source->driver->ops->startup(&substream, source);
-			if (ret < 0) {
-				dev_err(source->dev,
-					"ASoC: startup() failed: %d\n", ret);
-				goto out;
+		snd_soc_dapm_widget_for_each_source_path(w, path) {
+			source = path->source->priv;
+
+			if (source->driver->ops->startup) {
+				ret = source->driver->ops->startup(&substream,
+								   source);
+				if (ret < 0) {
+					dev_err(source->dev,
+						"ASoC: startup() failed: %d\n",
+						ret);
+					goto out;
+				}
+				source->active++;
 			}
-			source->active++;
+			ret = soc_dai_hw_params(&substream, params, source);
+			if (ret < 0)
+				goto out;
 		}
-		ret = soc_dai_hw_params(&substream, params, source);
-		if (ret < 0)
-			goto out;
 
 		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
-		if (sink->driver->ops->startup) {
-			ret = sink->driver->ops->startup(&substream, sink);
-			if (ret < 0) {
-				dev_err(sink->dev,
-					"ASoC: startup() failed: %d\n", ret);
-				goto out;
+		snd_soc_dapm_widget_for_each_sink_path(w, path) {
+			sink = path->sink->priv;
+
+			if (sink->driver->ops->startup) {
+				ret = sink->driver->ops->startup(&substream,
+								 sink);
+				if (ret < 0) {
+					dev_err(sink->dev,
+						"ASoC: startup() failed: %d\n",
+						ret);
+					goto out;
+				}
+				sink->active++;
 			}
-			sink->active++;
+			ret = soc_dai_hw_params(&substream, params, sink);
+			if (ret < 0)
+				goto out;
 		}
-		ret = soc_dai_hw_params(&substream, params, sink);
-		if (ret < 0)
-			goto out;
 		break;
 
 	case SND_SOC_DAPM_POST_PMU:
-		ret = snd_soc_dai_digital_mute(sink, 0,
-					       SNDRV_PCM_STREAM_PLAYBACK);
-		if (ret != 0 && ret != -ENOTSUPP)
-			dev_warn(sink->dev, "ASoC: Failed to unmute: %d\n", ret);
-		ret = 0;
+		snd_soc_dapm_widget_for_each_sink_path(w, path) {
+			sink = path->sink->priv;
+
+			ret = snd_soc_dai_digital_mute(sink, 0,
+						       SNDRV_PCM_STREAM_PLAYBACK);
+			if (ret != 0 && ret != -ENOTSUPP)
+				dev_warn(sink->dev,
+					 "ASoC: Failed to unmute: %d\n", ret);
+			ret = 0;
+		}
 		break;
 
 	case SND_SOC_DAPM_PRE_PMD:
-		ret = snd_soc_dai_digital_mute(sink, 1,
-					       SNDRV_PCM_STREAM_PLAYBACK);
-		if (ret != 0 && ret != -ENOTSUPP)
-			dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret);
-		ret = 0;
+		snd_soc_dapm_widget_for_each_sink_path(w, path) {
+			sink = path->sink->priv;
+
+			ret = snd_soc_dai_digital_mute(sink, 1,
+						       SNDRV_PCM_STREAM_PLAYBACK);
+			if (ret != 0 && ret != -ENOTSUPP)
+				dev_warn(sink->dev,
+					 "ASoC: Failed to mute: %d\n", ret);
+			ret = 0;
+		}
 
-		source->active--;
-		if (source->driver->ops->shutdown) {
-			substream.stream = SNDRV_PCM_STREAM_CAPTURE;
-			source->driver->ops->shutdown(&substream, source);
+		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+		snd_soc_dapm_widget_for_each_source_path(w, path) {
+			source = path->source->priv;
+
+			if (source->driver->ops->hw_free)
+				source->driver->ops->hw_free(&substream,
+							     source);
+
+			source->active--;
+			if (source->driver->ops->shutdown)
+				source->driver->ops->shutdown(&substream,
+							      source);
 		}
 
-		sink->active--;
-		if (sink->driver->ops->shutdown) {
-			substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
-			sink->driver->ops->shutdown(&substream, sink);
+		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+		snd_soc_dapm_widget_for_each_sink_path(w, path) {
+			sink = path->sink->priv;
+
+			if (sink->driver->ops->hw_free)
+				sink->driver->ops->hw_free(&substream, sink);
+
+			sink->active--;
+			if (sink->driver->ops->shutdown)
+				sink->driver->ops->shutdown(&substream, sink);
 		}
 		break;
 
@@ -3788,8 +3779,9 @@ static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol,
 			  struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_pcm_runtime *rtd = w->priv;
 
-	ucontrol->value.enumerated.item[0] = w->params_select;
+	ucontrol->value.enumerated.item[0] = rtd->params_select;
 
 	return 0;
 }
@@ -3798,18 +3790,19 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol,
 			  struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_pcm_runtime *rtd = w->priv;
 
 	/* Can't change the config when widget is already powered */
 	if (w->power)
 		return -EBUSY;
 
-	if (ucontrol->value.enumerated.item[0] == w->params_select)
+	if (ucontrol->value.enumerated.item[0] == rtd->params_select)
 		return 0;
 
-	if (ucontrol->value.enumerated.item[0] >= w->num_params)
+	if (ucontrol->value.enumerated.item[0] >= rtd->dai_link->num_params)
 		return -EINVAL;
 
-	w->params_select = ucontrol->value.enumerated.item[0];
+	rtd->params_select = ucontrol->value.enumerated.item[0];
 
 	return 0;
 }
@@ -3896,12 +3889,10 @@ outfree_w_param:
 	return NULL;
 }
 
-int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
-			 struct snd_soc_pcm_runtime *rtd,
-			 const struct snd_soc_pcm_stream *params,
-			 unsigned int num_params,
-			 struct snd_soc_dapm_widget *source,
-			 struct snd_soc_dapm_widget *sink)
+static struct snd_soc_dapm_widget *
+snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
+		     struct snd_soc_dapm_widget *source,
+		     struct snd_soc_dapm_widget *sink)
 {
 	struct snd_soc_dapm_widget template;
 	struct snd_soc_dapm_widget *w;
@@ -3913,7 +3904,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
 	link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s",
 				   source->name, sink->name);
 	if (!link_name)
-		return -ENOMEM;
+		return ERR_PTR(-ENOMEM);
 
 	memset(&template, 0, sizeof(template));
 	template.reg = SND_SOC_NOPM;
@@ -3925,9 +3916,10 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
 	template.kcontrol_news = NULL;
 
 	/* allocate memory for control, only in case of multiple configs */
-	if (num_params > 1) {
-		w_param_text = devm_kcalloc(card->dev, num_params,
-					sizeof(char *), GFP_KERNEL);
+	if (rtd->dai_link->num_params > 1) {
+		w_param_text = devm_kcalloc(card->dev,
+					    rtd->dai_link->num_params,
+					    sizeof(char *), GFP_KERNEL);
 		if (!w_param_text) {
 			ret = -ENOMEM;
 			goto param_fail;
@@ -3936,7 +3928,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
 		template.num_kcontrols = 1;
 		template.kcontrol_news =
 					snd_soc_dapm_alloc_kcontrol(card,
-						link_name, params, num_params,
+						link_name,
+						rtd->dai_link->params,
+						rtd->dai_link->num_params,
 						w_param_text, &private_value);
 		if (!template.kcontrol_news) {
 			ret = -ENOMEM;
@@ -3950,37 +3944,20 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
 	w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template);
 	if (IS_ERR(w)) {
 		ret = PTR_ERR(w);
-		/* Do not nag about probe deferrals */
-		if (ret != -EPROBE_DEFER)
-			dev_err(card->dev,
-				"ASoC: Failed to create %s widget (%d)\n",
-				link_name, ret);
-		goto outfree_kcontrol_news;
-	}
-	if (!w) {
-		dev_err(card->dev, "ASoC: Failed to create %s widget\n",
-			link_name);
-		ret = -ENOMEM;
 		goto outfree_kcontrol_news;
 	}
 
-	w->params = params;
-	w->num_params = num_params;
 	w->priv = rtd;
 
-	ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL);
-	if (ret)
-		goto outfree_w;
-	return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL);
+	return w;
 
-outfree_w:
-	devm_kfree(card->dev, w);
 outfree_kcontrol_news:
 	devm_kfree(card->dev, (void *)template.kcontrol_news);
-	snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text);
+	snd_soc_dapm_free_kcontrol(card, &private_value,
+				   rtd->dai_link->num_params, w_param_text);
 param_fail:
 	devm_kfree(card->dev, link_name);
-	return ret;
+	return ERR_PTR(ret);
 }
 
 int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
@@ -4003,21 +3980,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 			template.name);
 
 		w = snd_soc_dapm_new_control_unlocked(dapm, &template);
-		if (IS_ERR(w)) {
-			int ret = PTR_ERR(w);
-
-			/* Do not nag about probe deferrals */
-			if (ret != -EPROBE_DEFER)
-				dev_err(dapm->dev,
-				"ASoC: Failed to create %s widget (%d)\n",
-				dai->driver->playback.stream_name, ret);
-			return ret;
-		}
-		if (!w) {
-			dev_err(dapm->dev, "ASoC: Failed to create %s widget\n",
-				dai->driver->playback.stream_name);
-			return -ENOMEM;
-		}
+		if (IS_ERR(w))
+			return PTR_ERR(w);
 
 		w->priv = dai;
 		dai->playback_widget = w;
@@ -4032,21 +3996,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 			template.name);
 
 		w = snd_soc_dapm_new_control_unlocked(dapm, &template);
-		if (IS_ERR(w)) {
-			int ret = PTR_ERR(w);
-
-			/* Do not nag about probe deferrals */
-			if (ret != -EPROBE_DEFER)
-				dev_err(dapm->dev,
-				"ASoC: Failed to create %s widget (%d)\n",
-				dai->driver->playback.stream_name, ret);
-			return ret;
-		}
-		if (!w) {
-			dev_err(dapm->dev, "ASoC: Failed to create %s widget\n",
-				dai->driver->capture.stream_name);
-			return -ENOMEM;
-		}
+		if (IS_ERR(w))
+			return PTR_ERR(w);
 
 		w->priv = dai;
 		dai->capture_widget = w;
@@ -4115,34 +4066,79 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 					  struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dapm_widget *sink, *source;
+	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dapm_widget *playback = NULL, *capture = NULL;
+	struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu;
 	int i;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+	if (rtd->dai_link->params) {
+		playback_cpu = cpu_dai->capture_widget;
+		capture_cpu = cpu_dai->playback_widget;
+	} else {
+		playback = cpu_dai->playback_widget;
+		capture = cpu_dai->capture_widget;
+		playback_cpu = playback;
+		capture_cpu = capture;
+	}
+
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 
 		/* connect BE DAI playback if widgets are valid */
-		if (codec_dai->playback_widget && cpu_dai->playback_widget) {
-			source = cpu_dai->playback_widget;
-			sink = codec_dai->playback_widget;
+		codec = codec_dai->playback_widget;
+
+		if (playback_cpu && codec) {
+			if (!playback) {
+				playback = snd_soc_dapm_new_dai(card, rtd,
+								playback_cpu,
+								codec);
+				if (IS_ERR(playback)) {
+					dev_err(rtd->dev,
+						"ASoC: Failed to create DAI %s: %ld\n",
+						codec_dai->name,
+						PTR_ERR(playback));
+					continue;
+				}
+
+				snd_soc_dapm_add_path(&card->dapm, playback_cpu,
+						      playback, NULL, NULL);
+			}
+
 			dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
-				cpu_dai->component->name, source->name,
-				codec_dai->component->name, sink->name);
+				cpu_dai->component->name, playback_cpu->name,
+				codec_dai->component->name, codec->name);
 
-			snd_soc_dapm_add_path(&card->dapm, source, sink,
-				NULL, NULL);
+			snd_soc_dapm_add_path(&card->dapm, playback, codec,
+					      NULL, NULL);
 		}
+	}
 
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		/* connect BE DAI capture if widgets are valid */
-		if (codec_dai->capture_widget && cpu_dai->capture_widget) {
-			source = codec_dai->capture_widget;
-			sink = cpu_dai->capture_widget;
+		codec = codec_dai->capture_widget;
+
+		if (codec && capture_cpu) {
+			if (!capture) {
+				capture = snd_soc_dapm_new_dai(card, rtd,
+							       codec,
+							       capture_cpu);
+				if (IS_ERR(capture)) {
+					dev_err(rtd->dev,
+						"ASoC: Failed to create DAI %s: %ld\n",
+						codec_dai->name,
+						PTR_ERR(capture));
+					continue;
+				}
+
+				snd_soc_dapm_add_path(&card->dapm, capture,
+						      capture_cpu, NULL, NULL);
+			}
+
 			dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
-				codec_dai->component->name, source->name,
-				cpu_dai->component->name, sink->name);
+				codec_dai->component->name, codec->name,
+				cpu_dai->component->name, capture_cpu->name);
 
-			snd_soc_dapm_add_path(&card->dapm, source, sink,
-				NULL, NULL);
+			snd_soc_dapm_add_path(&card->dapm, codec, capture,
+					      NULL, NULL);
 		}
 	}
 }
@@ -4192,12 +4188,12 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
 	struct snd_soc_pcm_runtime *rtd;
 
 	/* for each BE DAI link... */
-	list_for_each_entry(rtd, &card->rtd_list, list)  {
+	for_each_card_rtds(card, rtd)  {
 		/*
 		 * dynamic FE links have no fixed DAI mapping.
 		 * CODEC<->CODEC links have no direct connection.
 		 */
-		if (rtd->dai_link->dynamic || rtd->dai_link->params)
+		if (rtd->dai_link->dynamic)
 			continue;
 
 		dapm_connect_dai_link_widgets(card, rtd);
@@ -4207,11 +4203,12 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
 static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
 	int event)
 {
+	struct snd_soc_dai *codec_dai;
 	int i;
 
 	soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event);
-	for (i = 0; i < rtd->num_codecs; i++)
-		soc_dapm_dai_stream_event(rtd->codec_dais[i], stream, event);
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		soc_dapm_dai_stream_event(codec_dai, stream, event);
 
 	dapm_power_widgets(rtd->card, event);
 }
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 592efb370c44..f4dc3d445aae 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -373,7 +373,7 @@ int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol,
 	unsigned int rshift = mc->rshift;
 	int max = mc->max;
 	int min = mc->min;
-	unsigned int mask = (1 << (fls(min + max) - 1)) - 1;
+	unsigned int mask = (1U << (fls(min + max) - 1)) - 1;
 	unsigned int val;
 	int ret;
 
@@ -418,7 +418,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol,
 	unsigned int rshift = mc->rshift;
 	int max = mc->max;
 	int min = mc->min;
-	unsigned int mask = (1 << (fls(min + max) - 1)) - 1;
+	unsigned int mask = (1U << (fls(min + max) - 1)) - 1;
 	int err = 0;
 	unsigned int val, val_mask, val2 = 0;
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e8b98bfd4cf1..03f36e534050 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -59,25 +59,26 @@ static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream)
 void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
 {
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai;
 	int i;
 
 	lockdep_assert_held(&rtd->pcm_mutex);
 
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		cpu_dai->playback_active++;
-		for (i = 0; i < rtd->num_codecs; i++)
-			rtd->codec_dais[i]->playback_active++;
+		for_each_rtd_codec_dai(rtd, i, codec_dai)
+			codec_dai->playback_active++;
 	} else {
 		cpu_dai->capture_active++;
-		for (i = 0; i < rtd->num_codecs; i++)
-			rtd->codec_dais[i]->capture_active++;
+		for_each_rtd_codec_dai(rtd, i, codec_dai)
+			codec_dai->capture_active++;
 	}
 
 	cpu_dai->active++;
 	cpu_dai->component->active++;
-	for (i = 0; i < rtd->num_codecs; i++) {
-		rtd->codec_dais[i]->active++;
-		rtd->codec_dais[i]->component->active++;
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		codec_dai->active++;
+		codec_dai->component->active++;
 	}
 }
 
@@ -94,25 +95,26 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
 void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream)
 {
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai;
 	int i;
 
 	lockdep_assert_held(&rtd->pcm_mutex);
 
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		cpu_dai->playback_active--;
-		for (i = 0; i < rtd->num_codecs; i++)
-			rtd->codec_dais[i]->playback_active--;
+		for_each_rtd_codec_dai(rtd, i, codec_dai)
+			codec_dai->playback_active--;
 	} else {
 		cpu_dai->capture_active--;
-		for (i = 0; i < rtd->num_codecs; i++)
-			rtd->codec_dais[i]->capture_active--;
+		for_each_rtd_codec_dai(rtd, i, codec_dai)
+			codec_dai->capture_active--;
 	}
 
 	cpu_dai->active--;
 	cpu_dai->component->active--;
-	for (i = 0; i < rtd->num_codecs; i++) {
-		rtd->codec_dais[i]->component->active--;
-		rtd->codec_dais[i]->active--;
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		codec_dai->component->active--;
+		codec_dai->active--;
 	}
 }
 
@@ -172,7 +174,7 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir,
 {
 	struct snd_soc_dpcm *dpcm;
 
-	list_for_each_entry(dpcm, &fe->dpcm[dir].be_clients, list_be) {
+	for_each_dpcm_be(fe, dir, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 
@@ -253,6 +255,7 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai;
 	unsigned int rate, channels, sample_bits, symmetry, i;
 
 	rate = params_rate(params);
@@ -263,8 +266,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
 	symmetry = cpu_dai->driver->symmetric_rates ||
 		rtd->dai_link->symmetric_rates;
 
-	for (i = 0; i < rtd->num_codecs; i++)
-		symmetry |= rtd->codec_dais[i]->driver->symmetric_rates;
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		symmetry |= codec_dai->driver->symmetric_rates;
 
 	if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) {
 		dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n",
@@ -275,8 +278,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
 	symmetry = cpu_dai->driver->symmetric_channels ||
 		rtd->dai_link->symmetric_channels;
 
-	for (i = 0; i < rtd->num_codecs; i++)
-		symmetry |= rtd->codec_dais[i]->driver->symmetric_channels;
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		symmetry |= codec_dai->driver->symmetric_channels;
 
 	if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) {
 		dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n",
@@ -287,8 +290,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
 	symmetry = cpu_dai->driver->symmetric_samplebits ||
 		rtd->dai_link->symmetric_samplebits;
 
-	for (i = 0; i < rtd->num_codecs; i++)
-		symmetry |= rtd->codec_dais[i]->driver->symmetric_samplebits;
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		symmetry |= codec_dai->driver->symmetric_samplebits;
 
 	if (symmetry && cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) {
 		dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n",
@@ -304,17 +307,18 @@ static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver;
 	struct snd_soc_dai_link *link = rtd->dai_link;
+	struct snd_soc_dai *codec_dai;
 	unsigned int symmetry, i;
 
 	symmetry = cpu_driver->symmetric_rates || link->symmetric_rates ||
 		cpu_driver->symmetric_channels || link->symmetric_channels ||
 		cpu_driver->symmetric_samplebits || link->symmetric_samplebits;
 
-	for (i = 0; i < rtd->num_codecs; i++)
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
 		symmetry = symmetry ||
-			rtd->codec_dais[i]->driver->symmetric_rates ||
-			rtd->codec_dais[i]->driver->symmetric_channels ||
-			rtd->codec_dais[i]->driver->symmetric_samplebits;
+			codec_dai->driver->symmetric_rates ||
+			codec_dai->driver->symmetric_channels ||
+			codec_dai->driver->symmetric_samplebits;
 
 	return symmetry;
 }
@@ -342,8 +346,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream)
 	unsigned int bits = 0, cpu_bits;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		for (i = 0; i < rtd->num_codecs; i++) {
-			codec_dai = rtd->codec_dais[i];
+		for_each_rtd_codec_dai(rtd, i, codec_dai) {
 			if (codec_dai->driver->playback.sig_bits == 0) {
 				bits = 0;
 				break;
@@ -352,8 +355,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream)
 		}
 		cpu_bits = cpu_dai->driver->playback.sig_bits;
 	} else {
-		for (i = 0; i < rtd->num_codecs; i++) {
-			codec_dai = rtd->codec_dais[i];
+		for_each_rtd_codec_dai(rtd, i, codec_dai) {
 			if (codec_dai->driver->capture.sig_bits == 0) {
 				bits = 0;
 				break;
@@ -372,6 +374,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_pcm_hardware *hw = &runtime->hw;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai;
 	struct snd_soc_dai_driver *cpu_dai_drv = rtd->cpu_dai->driver;
 	struct snd_soc_dai_driver *codec_dai_drv;
 	struct snd_soc_pcm_stream *codec_stream;
@@ -388,7 +391,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 		cpu_stream = &cpu_dai_drv->capture;
 
 	/* first calculate min/max only for CODECs in the DAI link */
-	for (i = 0; i < rtd->num_codecs; i++) {
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 
 		/*
 		 * Skip CODECs which don't support the current stream type.
@@ -399,11 +402,11 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 		 * bailed out on a higher level, since there would be no
 		 * CODEC to support the transfer direction in that case.
 		 */
-		if (!snd_soc_dai_stream_valid(rtd->codec_dais[i],
+		if (!snd_soc_dai_stream_valid(codec_dai,
 					      substream->stream))
 			continue;
 
-		codec_dai_drv = rtd->codec_dais[i]->driver;
+		codec_dai_drv = codec_dai->driver;
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			codec_stream = &codec_dai_drv->playback;
 		else
@@ -482,8 +485,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	int i, ret = 0;
 
 	pinctrl_pm_select_default_state(cpu_dai->dev);
-	for (i = 0; i < rtd->num_codecs; i++)
-		pinctrl_pm_select_default_state(rtd->codec_dais[i]->dev);
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		pinctrl_pm_select_default_state(codec_dai->dev);
 
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
@@ -520,8 +523,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	}
 	component = NULL;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->startup) {
 			ret = codec_dai->driver->ops->startup(substream,
 							      codec_dai);
@@ -588,10 +590,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 			goto config_err;
 	}
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		if (rtd->codec_dais[i]->active) {
-			ret = soc_pcm_apply_symmetry(substream,
-						     rtd->codec_dais[i]);
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		if (codec_dai->active) {
+			ret = soc_pcm_apply_symmetry(substream, codec_dai);
 			if (ret != 0)
 				goto config_err;
 		}
@@ -620,8 +621,7 @@ machine_err:
 	i = rtd->num_codecs;
 
 codec_dai_err:
-	while (--i >= 0) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->shutdown)
 			codec_dai->driver->ops->shutdown(substream, codec_dai);
 	}
@@ -641,9 +641,9 @@ out:
 		pm_runtime_put_autosuspend(component->dev);
 	}
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		if (!rtd->codec_dais[i]->active)
-			pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev);
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		if (!codec_dai->active)
+			pinctrl_pm_select_sleep_state(codec_dai->dev);
 	}
 	if (!cpu_dai->active)
 		pinctrl_pm_select_sleep_state(cpu_dai->dev);
@@ -701,8 +701,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	if (!cpu_dai->active)
 		cpu_dai->rate = 0;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (!codec_dai->active)
 			codec_dai->rate = 0;
 	}
@@ -712,8 +711,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	if (cpu_dai->driver->ops->shutdown)
 		cpu_dai->driver->ops->shutdown(substream, cpu_dai);
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->shutdown)
 			codec_dai->driver->ops->shutdown(substream, codec_dai);
 	}
@@ -751,9 +749,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 		pm_runtime_put_autosuspend(component->dev);
 	}
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		if (!rtd->codec_dais[i]->active)
-			pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev);
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		if (!codec_dai->active)
+			pinctrl_pm_select_sleep_state(codec_dai->dev);
 	}
 	if (!cpu_dai->active)
 		pinctrl_pm_select_sleep_state(cpu_dai->dev);
@@ -801,8 +799,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 		}
 	}
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->prepare) {
 			ret = codec_dai->driver->ops->prepare(substream,
 							      codec_dai);
@@ -834,8 +831,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	snd_soc_dapm_stream_event(rtd, substream->stream,
 			SND_SOC_DAPM_STREAM_START);
 
-	for (i = 0; i < rtd->num_codecs; i++)
-		snd_soc_dai_digital_mute(rtd->codec_dais[i], 0,
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		snd_soc_dai_digital_mute(codec_dai, 0,
 					 substream->stream);
 	snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream);
 
@@ -920,6 +917,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_component *component;
 	struct snd_soc_rtdcom_list *rtdcom;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai;
 	int i, ret = 0;
 
 	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
@@ -932,8 +930,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		struct snd_pcm_hw_params codec_params;
 
 		/*
@@ -1018,8 +1015,7 @@ interface_err:
 	i = rtd->num_codecs;
 
 codec_err:
-	while (--i >= 0) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->hw_free)
 			codec_dai->driver->ops->hw_free(substream, codec_dai);
 		codec_dai->rate = 0;
@@ -1052,8 +1048,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 		cpu_dai->sample_bits = 0;
 	}
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->active == 1) {
 			codec_dai->rate = 0;
 			codec_dai->channels = 0;
@@ -1062,10 +1057,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	}
 
 	/* apply codec digital mute */
-	for (i = 0; i < rtd->num_codecs; i++) {
-		if ((playback && rtd->codec_dais[i]->playback_active == 1) ||
-		    (!playback && rtd->codec_dais[i]->capture_active == 1))
-			snd_soc_dai_digital_mute(rtd->codec_dais[i], 1,
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		if ((playback && codec_dai->playback_active == 1) ||
+		    (!playback && codec_dai->capture_active == 1))
+			snd_soc_dai_digital_mute(codec_dai, 1,
 						 substream->stream);
 	}
 
@@ -1077,8 +1072,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	soc_pcm_components_hw_free(substream, NULL);
 
 	/* now free hw params for the DAIs  */
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->hw_free)
 			codec_dai->driver->ops->hw_free(substream, codec_dai);
 	}
@@ -1099,8 +1093,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct snd_soc_dai *codec_dai;
 	int i, ret;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->trigger) {
 			ret = codec_dai->driver->ops->trigger(substream,
 							      cmd, codec_dai);
@@ -1144,8 +1137,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *codec_dai;
 	int i, ret;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->bespoke_trigger) {
 			ret = codec_dai->driver->ops->bespoke_trigger(substream,
 								cmd, codec_dai);
@@ -1199,8 +1191,7 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
 	if (cpu_dai->driver->ops->delay)
 		delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->delay)
 			codec_delay = max(codec_delay,
 					codec_dai->driver->ops->delay(substream,
@@ -1220,7 +1211,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe,
 	struct snd_soc_dpcm *dpcm;
 
 	/* only add new dpcms */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		if (dpcm->be == be && dpcm->fe == fe)
 			return 0;
 	}
@@ -1261,7 +1252,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe,
 
 	be_substream = snd_soc_dpcm_get_substream(be, stream);
 
-	list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) {
+	for_each_dpcm_fe(be, stream, dpcm) {
 		if (dpcm->fe == fe)
 			continue;
 
@@ -1281,7 +1272,7 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream)
 {
 	struct snd_soc_dpcm *dpcm, *d;
 
-	list_for_each_entry_safe(dpcm, d, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be_safe(fe, stream, dpcm, d) {
 		dev_dbg(fe->dev, "ASoC: BE %s disconnect check for %s\n",
 				stream ? "capture" : "playback",
 				dpcm->be->dai_link->name);
@@ -1310,12 +1301,13 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
 		struct snd_soc_dapm_widget *widget, int stream)
 {
 	struct snd_soc_pcm_runtime *be;
+	struct snd_soc_dai *dai;
 	int i;
 
 	dev_dbg(card->dev, "ASoC: find BE for widget %s\n", widget->name);
 
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		list_for_each_entry(be, &card->rtd_list, list) {
+		for_each_card_rtds(card, be) {
 
 			if (!be->dai_link->no_pcm)
 				continue;
@@ -1327,15 +1319,14 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
 			if (be->cpu_dai->playback_widget == widget)
 				return be;
 
-			for (i = 0; i < be->num_codecs; i++) {
-				struct snd_soc_dai *dai = be->codec_dais[i];
+			for_each_rtd_codec_dai(be, i, dai) {
 				if (dai->playback_widget == widget)
 					return be;
 			}
 		}
 	} else {
 
-		list_for_each_entry(be, &card->rtd_list, list) {
+		for_each_card_rtds(card, be) {
 
 			if (!be->dai_link->no_pcm)
 				continue;
@@ -1347,8 +1338,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
 			if (be->cpu_dai->capture_widget == widget)
 				return be;
 
-			for (i = 0; i < be->num_codecs; i++) {
-				struct snd_soc_dai *dai = be->codec_dais[i];
+			for_each_rtd_codec_dai(be, i, dai) {
 				if (dai->capture_widget == widget)
 					return be;
 			}
@@ -1388,32 +1378,31 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget,
 {
 	struct snd_soc_card *card = widget->dapm->card;
 	struct snd_soc_pcm_runtime *rtd;
+	struct snd_soc_dai *dai;
 	int i;
 
 	if (dir == SND_SOC_DAPM_DIR_OUT) {
-		list_for_each_entry(rtd, &card->rtd_list, list) {
+		for_each_card_rtds(card, rtd) {
 			if (!rtd->dai_link->no_pcm)
 				continue;
 
 			if (rtd->cpu_dai->playback_widget == widget)
 				return true;
 
-			for (i = 0; i < rtd->num_codecs; ++i) {
-				struct snd_soc_dai *dai = rtd->codec_dais[i];
+			for_each_rtd_codec_dai(rtd, i, dai) {
 				if (dai->playback_widget == widget)
 					return true;
 			}
 		}
 	} else { /* SND_SOC_DAPM_DIR_IN */
-		list_for_each_entry(rtd, &card->rtd_list, list) {
+		for_each_card_rtds(card, rtd) {
 			if (!rtd->dai_link->no_pcm)
 				continue;
 
 			if (rtd->cpu_dai->capture_widget == widget)
 				return true;
 
-			for (i = 0; i < rtd->num_codecs; ++i) {
-				struct snd_soc_dai *dai = rtd->codec_dais[i];
+			for_each_rtd_codec_dai(rtd, i, dai) {
 				if (dai->capture_widget == widget)
 					return true;
 			}
@@ -1445,10 +1434,11 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
 	struct snd_soc_dpcm *dpcm;
 	struct snd_soc_dapm_widget_list *list = *list_;
 	struct snd_soc_dapm_widget *widget;
+	struct snd_soc_dai *dai;
 	int prune = 0;
 
 	/* Destroy any old FE <--> BE connections */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		unsigned int i;
 
 		/* is there a valid CPU DAI widget for this BE */
@@ -1459,8 +1449,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
 			continue;
 
 		/* is there a valid CODEC DAI widget for this BE */
-		for (i = 0; i < dpcm->be->num_codecs; i++) {
-			struct snd_soc_dai *dai = dpcm->be->codec_dais[i];
+		for_each_rtd_codec_dai(dpcm->be, i, dai) {
 			widget = dai_get_widget(dai, stream);
 
 			/* prune the BE if it's no longer in our active list */
@@ -1555,7 +1544,7 @@ void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream)
 {
 	struct snd_soc_dpcm *dpcm;
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+	for_each_dpcm_be(fe, stream, dpcm)
 		dpcm->be->dpcm[stream].runtime_update =
 						SND_SOC_DPCM_UPDATE_NO;
 }
@@ -1566,7 +1555,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe,
 	struct snd_soc_dpcm *dpcm;
 
 	/* disable any enabled and non active backends */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
@@ -1595,7 +1584,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream)
 	int err, count = 0;
 
 	/* only startup BE DAIs that are either sinks or sources to this FE DAI */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
@@ -1649,7 +1638,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream)
 
 unwind:
 	/* disable any enabled and non active backends */
-	list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be_rollback(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
 			snd_soc_dpcm_get_substream(be, stream);
@@ -1680,7 +1669,7 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
 				 struct snd_soc_pcm_stream *stream)
 {
 	runtime->hw.rate_min = stream->rate_min;
-	runtime->hw.rate_max = stream->rate_max;
+	runtime->hw.rate_max = min_not_zero(stream->rate_max, UINT_MAX);
 	runtime->hw.channels_min = stream->channels_min;
 	runtime->hw.channels_max = stream->channels_max;
 	if (runtime->hw.formats)
@@ -1695,6 +1684,7 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *fe = substream->private_data;
 	struct snd_soc_dpcm *dpcm;
+	struct snd_soc_dai *dai;
 	int stream = substream->stream;
 
 	if (!fe->dai_link->dpcm_merged_format)
@@ -1705,22 +1695,21 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream,
 	 * if FE want to use it (= dpcm_merged_format)
 	 */
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_soc_dai_driver *codec_dai_drv;
 		struct snd_soc_pcm_stream *codec_stream;
 		int i;
 
-		for (i = 0; i < be->num_codecs; i++) {
+		for_each_rtd_codec_dai(be, i, dai) {
 			/*
 			 * Skip CODECs which don't support the current stream
 			 * type. See soc_pcm_init_runtime_hw() for more details
 			 */
-			if (!snd_soc_dai_stream_valid(be->codec_dais[i],
-						      stream))
+			if (!snd_soc_dai_stream_valid(dai, stream))
 				continue;
 
-			codec_dai_drv = be->codec_dais[i]->driver;
+			codec_dai_drv = dai->driver;
 			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
 				codec_stream = &codec_dai_drv->playback;
 			else
@@ -1747,7 +1736,7 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream,
 	 * if FE want to use it (= dpcm_merged_chan)
 	 */
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_soc_dai_driver *cpu_dai_drv =  be->cpu_dai->driver;
 		struct snd_soc_dai_driver *codec_dai_drv;
@@ -1799,12 +1788,13 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream,
 	 * if FE want to use it (= dpcm_merged_chan)
 	 */
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_soc_dai_driver *cpu_dai_drv =  be->cpu_dai->driver;
 		struct snd_soc_dai_driver *codec_dai_drv;
 		struct snd_soc_pcm_stream *codec_stream;
 		struct snd_soc_pcm_stream *cpu_stream;
+		struct snd_soc_dai *dai;
 		int i;
 
 		if (stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -1816,16 +1806,15 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream,
 		*rate_max = min_not_zero(*rate_max, cpu_stream->rate_max);
 		*rates = snd_pcm_rate_mask_intersect(*rates, cpu_stream->rates);
 
-		for (i = 0; i < be->num_codecs; i++) {
+		for_each_rtd_codec_dai(be, i, dai) {
 			/*
 			 * Skip CODECs which don't support the current stream
 			 * type. See soc_pcm_init_runtime_hw() for more details
 			 */
-			if (!snd_soc_dai_stream_valid(be->codec_dais[i],
-						      stream))
+			if (!snd_soc_dai_stream_valid(dai, stream))
 				continue;
 
-			codec_dai_drv = be->codec_dais[i]->driver;
+			codec_dai_drv = dai->driver;
 			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
 				codec_stream = &codec_dai_drv->playback;
 			else
@@ -1902,11 +1891,12 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream,
 	}
 
 	/* apply symmetry for BE */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
 			snd_soc_dpcm_get_substream(be, stream);
 		struct snd_soc_pcm_runtime *rtd = be_substream->private_data;
+		struct snd_soc_dai *codec_dai;
 		int i;
 
 		if (rtd->dai_link->be_hw_params_fixup)
@@ -1923,10 +1913,10 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream,
 				return err;
 		}
 
-		for (i = 0; i < rtd->num_codecs; i++) {
-			if (rtd->codec_dais[i]->active) {
+		for_each_rtd_codec_dai(rtd, i, codec_dai) {
+			if (codec_dai->active) {
 				err = soc_pcm_apply_symmetry(fe_substream,
-							     rtd->codec_dais[i]);
+							     codec_dai);
 				if (err < 0)
 					return err;
 			}
@@ -1986,7 +1976,7 @@ int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream)
 	struct snd_soc_dpcm *dpcm;
 
 	/* only shutdown BEs that are either sinks or sources to this FE DAI */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
@@ -2050,7 +2040,7 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
 
 	/* only hw_params backends that are either sinks or sources
 	 * to this frontend DAI */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
@@ -2119,7 +2109,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream)
 	struct snd_soc_dpcm *dpcm;
 	int ret;
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
@@ -2170,7 +2160,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream)
 
 unwind:
 	/* disable any enabled and non active backends */
-	list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be_rollback(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
 			snd_soc_dpcm_get_substream(be, stream);
@@ -2250,7 +2240,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
 	struct snd_soc_dpcm *dpcm;
 	int ret = 0;
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
@@ -2436,7 +2426,7 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream)
 	struct snd_soc_dpcm *dpcm;
 	int ret = 0;
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
@@ -2646,7 +2636,7 @@ close:
 	dpcm_be_dai_shutdown(fe, stream);
 disconnect:
 	/* disconnect any non started BEs */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
 				dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
@@ -2771,14 +2761,14 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
 
 	mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
 	/* shutdown all old paths first */
-	list_for_each_entry(fe, &card->rtd_list, list) {
+	for_each_card_rtds(card, fe) {
 		ret = soc_dpcm_fe_runtime_update(fe, 0);
 		if (ret)
 			goto out;
 	}
 
 	/* bring new paths up */
-	list_for_each_entry(fe, &card->rtd_list, list) {
+	for_each_card_rtds(card, fe) {
 		ret = soc_dpcm_fe_runtime_update(fe, 1);
 		if (ret)
 			goto out;
@@ -2791,10 +2781,9 @@ out:
 int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
 {
 	struct snd_soc_dpcm *dpcm;
-	struct list_head *clients =
-		&fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients;
+	struct snd_soc_dai *dai;
 
-	list_for_each_entry(dpcm, clients, list_be) {
+	for_each_dpcm_be(fe, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		int i;
@@ -2802,8 +2791,7 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
 		if (be->dai_link->ignore_suspend)
 			continue;
 
-		for (i = 0; i < be->num_codecs; i++) {
-			struct snd_soc_dai *dai = be->codec_dais[i];
+		for_each_rtd_codec_dai(be, i, dai) {
 			struct snd_soc_dai_driver *drv = dai->driver;
 
 			dev_dbg(be->dev, "ASoC: BE digital mute %s\n",
@@ -2844,7 +2832,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
 	ret = dpcm_fe_dai_startup(fe_substream);
 	if (ret < 0) {
 		/* clean up all links */
-		list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+		for_each_dpcm_be(fe, stream, dpcm)
 			dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
 
 		dpcm_be_disconnect(fe, stream);
@@ -2867,7 +2855,7 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
 	ret = dpcm_fe_dai_shutdown(fe_substream);
 
 	/* mark FE's links ready to prune */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+	for_each_dpcm_be(fe, stream, dpcm)
 		dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
 
 	dpcm_be_disconnect(fe, stream);
@@ -3041,8 +3029,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 		playback = rtd->dai_link->dpcm_playback;
 		capture = rtd->dai_link->dpcm_capture;
 	} else {
-		for (i = 0; i < rtd->num_codecs; i++) {
-			codec_dai = rtd->codec_dais[i];
+		for_each_rtd_codec_dai(rtd, i, codec_dai) {
 			if (codec_dai->driver->playback.channels_min)
 				playback = 1;
 			if (codec_dai->driver->capture.channels_min)
@@ -3230,7 +3217,7 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
 	struct snd_soc_dpcm *dpcm;
 	int state;
 
-	list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) {
+	for_each_dpcm_fe(be, stream, dpcm) {
 
 		if (dpcm->fe == fe)
 			continue;
@@ -3257,7 +3244,7 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
 	struct snd_soc_dpcm *dpcm;
 	int state;
 
-	list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) {
+	for_each_dpcm_fe(be, stream, dpcm) {
 
 		if (dpcm->fe == fe)
 			continue;
@@ -3337,7 +3324,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe,
 		goto out;
 	}
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		params = &dpcm->hw_params;
 
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 66e77e020745..045ef136903d 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -993,7 +993,7 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count,
 				kfree(se);
 				continue;
 			}
-			/* fall through and create texts */
+			/* fall through */
 		case SND_SOC_TPLG_CTL_ENUM:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
@@ -1310,7 +1310,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create(
 					ec->hdr.name);
 				goto err_se;
 			}
-			/* fall through to create texts */
+			/* fall through */
 		case SND_SOC_TPLG_CTL_ENUM:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
@@ -1565,17 +1565,6 @@ widget:
 		widget = snd_soc_dapm_new_control_unlocked(dapm, &template);
 	if (IS_ERR(widget)) {
 		ret = PTR_ERR(widget);
-		/* Do not nag about probe deferrals */
-		if (ret != -EPROBE_DEFER)
-			dev_err(tplg->dev,
-				"ASoC: failed to create widget %s controls (%d)\n",
-				w->name, ret);
-		goto hdr_err;
-	}
-	if (widget == NULL) {
-		dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n",
-			w->name);
-		ret = -ENOMEM;
 		goto hdr_err;
 	}
 
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index e0c93496c0cd..e3b9dd634c6d 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -273,13 +273,13 @@ static int dummy_dma_open(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static const struct snd_pcm_ops dummy_dma_ops = {
+static const struct snd_pcm_ops snd_dummy_dma_ops = {
 	.open		= dummy_dma_open,
 	.ioctl		= snd_pcm_lib_ioctl,
 };
 
 static const struct snd_soc_component_driver dummy_platform = {
-	.ops = &dummy_dma_ops,
+	.ops = &snd_dummy_dma_ops,
 };
 
 static const struct snd_soc_component_driver dummy_codec = {
diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig
index 9b2681397dba..c66ffa72057e 100644
--- a/sound/soc/stm/Kconfig
+++ b/sound/soc/stm/Kconfig
@@ -3,6 +3,7 @@ menu "STMicroelectronics STM32 SOC audio support"
 config SND_SOC_STM32_SAI
 	tristate "STM32 SAI interface (Serial Audio Interface) support"
 	depends on (ARCH_STM32 && OF) || COMPILE_TEST
+	depends on COMMON_CLK
 	depends on SND_SOC
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 	select REGMAP_MMIO
diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c
index f22654253c43..d597eba61992 100644
--- a/sound/soc/stm/stm32_sai.c
+++ b/sound/soc/stm/stm32_sai.c
@@ -104,7 +104,7 @@ static int stm32_sai_set_sync(struct stm32_sai_data *sai_client,
 
 	if (!pdev) {
 		dev_err(&sai_client->pdev->dev,
-			"Device not found for node %s\n", np_provider->name);
+			"Device not found for node %pOFn\n", np_provider);
 		return -ENODEV;
 	}
 
diff --git a/sound/soc/stm/stm32_sai.h b/sound/soc/stm/stm32_sai.h
index f25422174909..08de899c766b 100644
--- a/sound/soc/stm/stm32_sai.h
+++ b/sound/soc/stm/stm32_sai.h
@@ -91,6 +91,9 @@
 #define SAI_XCR1_OSR_SHIFT	26
 #define SAI_XCR1_OSR		BIT(SAI_XCR1_OSR_SHIFT)
 
+#define SAI_XCR1_MCKEN_SHIFT	27
+#define SAI_XCR1_MCKEN		BIT(SAI_XCR1_MCKEN_SHIFT)
+
 /******************* Bit definition for SAI_XCR2 register *******************/
 #define SAI_XCR2_FTH_SHIFT	0
 #define SAI_XCR2_FTH_MASK	GENMASK(2, SAI_XCR2_FTH_SHIFT)
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index 06fba9650ac4..ea05cc91aa05 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -17,6 +17,7 @@
  */
 
 #include <linux/clk.h>
+#include <linux/clk-provider.h>
 #include <linux/kernel.h>
 #include <linux/module.h>
 #include <linux/of_irq.h>
@@ -68,6 +69,8 @@
 #define SAI_IEC60958_BLOCK_FRAMES	192
 #define SAI_IEC60958_STATUS_BYTES	24
 
+#define SAI_MCLK_NAME_LEN		32
+
 /**
  * struct stm32_sai_sub_data - private data of SAI sub block (block A or B)
  * @pdev: device data pointer
@@ -80,6 +83,7 @@
  * @pdata: SAI block parent data pointer
  * @np_sync_provider: synchronization provider node
  * @sai_ck: kernel clock feeding the SAI clock generator
+ * @sai_mclk: master clock from SAI mclk provider
  * @phys_addr: SAI registers physical base address
  * @mclk_rate: SAI block master clock frequency (Hz). set at init
  * @id: SAI sub block id corresponding to sub-block A or B
@@ -110,6 +114,7 @@ struct stm32_sai_sub_data {
 	struct stm32_sai_data *pdata;
 	struct device_node *np_sync_provider;
 	struct clk *sai_ck;
+	struct clk *sai_mclk;
 	dma_addr_t phys_addr;
 	unsigned int mclk_rate;
 	unsigned int id;
@@ -251,6 +256,176 @@ static const struct snd_kcontrol_new iec958_ctls = {
 	.put = snd_pcm_iec958_put,
 };
 
+struct stm32_sai_mclk_data {
+	struct clk_hw hw;
+	unsigned long freq;
+	struct stm32_sai_sub_data *sai_data;
+};
+
+#define to_mclk_data(_hw) container_of(_hw, struct stm32_sai_mclk_data, hw)
+#define STM32_SAI_MAX_CLKS 1
+
+static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai,
+				 unsigned long input_rate,
+				 unsigned long output_rate)
+{
+	int version = sai->pdata->conf->version;
+	int div;
+
+	div = DIV_ROUND_CLOSEST(input_rate, output_rate);
+	if (div > SAI_XCR1_MCKDIV_MAX(version)) {
+		dev_err(&sai->pdev->dev, "Divider %d out of range\n", div);
+		return -EINVAL;
+	}
+	dev_dbg(&sai->pdev->dev, "SAI divider %d\n", div);
+
+	if (input_rate % div)
+		dev_dbg(&sai->pdev->dev,
+			"Rate not accurate. requested (%ld), actual (%ld)\n",
+			output_rate, input_rate / div);
+
+	return div;
+}
+
+static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai,
+				 unsigned int div)
+{
+	int version = sai->pdata->conf->version;
+	int ret, cr1, mask;
+
+	if (div > SAI_XCR1_MCKDIV_MAX(version)) {
+		dev_err(&sai->pdev->dev, "Divider %d out of range\n", div);
+		return -EINVAL;
+	}
+
+	mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version));
+	cr1 = SAI_XCR1_MCKDIV_SET(div);
+	ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1);
+	if (ret < 0)
+		dev_err(&sai->pdev->dev, "Failed to update CR1 register\n");
+
+	return ret;
+}
+
+static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate,
+				      unsigned long *prate)
+{
+	struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+	struct stm32_sai_sub_data *sai = mclk->sai_data;
+	int div;
+
+	div = stm32_sai_get_clk_div(sai, *prate, rate);
+	if (div < 0)
+		return div;
+
+	mclk->freq = *prate / div;
+
+	return mclk->freq;
+}
+
+static unsigned long stm32_sai_mclk_recalc_rate(struct clk_hw *hw,
+						unsigned long parent_rate)
+{
+	struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+
+	return mclk->freq;
+}
+
+static int stm32_sai_mclk_set_rate(struct clk_hw *hw, unsigned long rate,
+				   unsigned long parent_rate)
+{
+	struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+	struct stm32_sai_sub_data *sai = mclk->sai_data;
+	unsigned int div;
+	int ret;
+
+	div = stm32_sai_get_clk_div(sai, parent_rate, rate);
+	if (div < 0)
+		return div;
+
+	ret = stm32_sai_set_clk_div(sai, div);
+	if (ret)
+		return ret;
+
+	mclk->freq = rate;
+
+	return 0;
+}
+
+static int stm32_sai_mclk_enable(struct clk_hw *hw)
+{
+	struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+	struct stm32_sai_sub_data *sai = mclk->sai_data;
+
+	dev_dbg(&sai->pdev->dev, "Enable master clock\n");
+
+	return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
+				  SAI_XCR1_MCKEN, SAI_XCR1_MCKEN);
+}
+
+static void stm32_sai_mclk_disable(struct clk_hw *hw)
+{
+	struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+	struct stm32_sai_sub_data *sai = mclk->sai_data;
+
+	dev_dbg(&sai->pdev->dev, "Disable master clock\n");
+
+	regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_MCKEN, 0);
+}
+
+static const struct clk_ops mclk_ops = {
+	.enable = stm32_sai_mclk_enable,
+	.disable = stm32_sai_mclk_disable,
+	.recalc_rate = stm32_sai_mclk_recalc_rate,
+	.round_rate = stm32_sai_mclk_round_rate,
+	.set_rate = stm32_sai_mclk_set_rate,
+};
+
+static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai)
+{
+	struct clk_hw *hw;
+	struct stm32_sai_mclk_data *mclk;
+	struct device *dev = &sai->pdev->dev;
+	const char *pname = __clk_get_name(sai->sai_ck);
+	char *mclk_name, *p, *s = (char *)pname;
+	int ret, i = 0;
+
+	mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL);
+	if (!mclk)
+		return -ENOMEM;
+
+	mclk_name = devm_kcalloc(dev, sizeof(char),
+				 SAI_MCLK_NAME_LEN, GFP_KERNEL);
+	if (!mclk_name)
+		return -ENOMEM;
+
+	/*
+	 * Forge mclk clock name from parent clock name and suffix.
+	 * String after "_" char is stripped in parent name.
+	 */
+	p = mclk_name;
+	while (*s && *s != '_' && (i < (SAI_MCLK_NAME_LEN - 7))) {
+		*p++ = *s++;
+		i++;
+	}
+	STM_SAI_IS_SUB_A(sai) ? strcat(p, "a_mclk") : strcat(p, "b_mclk");
+
+	mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0);
+	mclk->sai_data = sai;
+	hw = &mclk->hw;
+
+	dev_dbg(dev, "Register master clock %s\n", mclk_name);
+	ret = devm_clk_hw_register(&sai->pdev->dev, hw);
+	if (ret) {
+		dev_err(dev, "mclk register returned %d\n", ret);
+		return ret;
+	}
+	sai->sai_mclk = hw->clk;
+
+	/* register mclk provider */
+	return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw);
+}
+
 static irqreturn_t stm32_sai_isr(int irq, void *devid)
 {
 	struct stm32_sai_sub_data *sai = (struct stm32_sai_sub_data *)devid;
@@ -312,15 +487,25 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai,
 	struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai);
 	int ret;
 
-	if ((dir == SND_SOC_CLOCK_OUT) && sai->master) {
+	if (dir == SND_SOC_CLOCK_OUT) {
 		ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
 					 SAI_XCR1_NODIV,
 					 (unsigned int)~SAI_XCR1_NODIV);
 		if (ret < 0)
 			return ret;
 
-		sai->mclk_rate = freq;
 		dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq);
+		sai->mclk_rate = freq;
+
+		if (sai->sai_mclk) {
+			ret = clk_set_rate_exclusive(sai->sai_mclk,
+						     sai->mclk_rate);
+			if (ret) {
+				dev_err(cpu_dai->dev,
+					"Could not set mclk rate\n");
+				return ret;
+			}
+		}
 	}
 
 	return 0;
@@ -715,15 +900,9 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
 {
 	struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai);
 	int cr1, mask, div = 0;
-	int sai_clk_rate, mclk_ratio, den, ret;
-	int version = sai->pdata->conf->version;
+	int sai_clk_rate, mclk_ratio, den;
 	unsigned int rate = params_rate(params);
 
-	if (!sai->mclk_rate) {
-		dev_err(cpu_dai->dev, "Mclk rate is null\n");
-		return -EINVAL;
-	}
-
 	if (!(rate % 11025))
 		clk_set_parent(sai->sai_ck, sai->pdata->clk_x11k);
 	else
@@ -731,14 +910,22 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
 	sai_clk_rate = clk_get_rate(sai->sai_ck);
 
 	if (STM_SAI_IS_F4(sai->pdata)) {
-		/*
-		 * mclk_rate = 256 * fs
-		 * MCKDIV = 0 if sai_ck < 3/2 * mclk_rate
-		 * MCKDIV = sai_ck / (2 * mclk_rate) otherwise
+		/* mclk on (NODIV=0)
+		 *   mclk_rate = 256 * fs
+		 *   MCKDIV = 0 if sai_ck < 3/2 * mclk_rate
+		 *   MCKDIV = sai_ck / (2 * mclk_rate) otherwise
+		 * mclk off (NODIV=1)
+		 *   MCKDIV ignored. sck = sai_ck
 		 */
-		if (2 * sai_clk_rate >= 3 * sai->mclk_rate)
-			div = DIV_ROUND_CLOSEST(sai_clk_rate,
-						2 * sai->mclk_rate);
+		if (!sai->mclk_rate)
+			return 0;
+
+		if (2 * sai_clk_rate >= 3 * sai->mclk_rate) {
+			div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+						    2 * sai->mclk_rate);
+			if (div < 0)
+				return div;
+		}
 	} else {
 		/*
 		 * TDM mode :
@@ -750,8 +937,10 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
 		 * Note: NOMCK/NODIV correspond to same bit.
 		 */
 		if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) {
-			div = DIV_ROUND_CLOSEST(sai_clk_rate,
-						(params_rate(params) * 128));
+			div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+						    rate * 128);
+			if (div < 0)
+				return div;
 		} else {
 			if (sai->mclk_rate) {
 				mclk_ratio = sai->mclk_rate / rate;
@@ -764,31 +953,22 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
 						mclk_ratio);
 					return -EINVAL;
 				}
-				div = DIV_ROUND_CLOSEST(sai_clk_rate,
-							sai->mclk_rate);
+				div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+							    sai->mclk_rate);
+				if (div < 0)
+					return div;
 			} else {
 				/* mclk-fs not set, master clock not active */
 				den = sai->fs_length * params_rate(params);
-				div = DIV_ROUND_CLOSEST(sai_clk_rate, den);
+				div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+							    den);
+				if (div < 0)
+					return div;
 			}
 		}
 	}
 
-	if (div > SAI_XCR1_MCKDIV_MAX(version)) {
-		dev_err(cpu_dai->dev, "Divider %d out of range\n", div);
-		return -EINVAL;
-	}
-	dev_dbg(cpu_dai->dev, "SAI clock %d, divider %d\n", sai_clk_rate, div);
-
-	mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version));
-	cr1 = SAI_XCR1_MCKDIV_SET(div);
-	ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1);
-	if (ret < 0) {
-		dev_err(cpu_dai->dev, "Failed to update CR1 register\n");
-		return ret;
-	}
-
-	return 0;
+	return stm32_sai_set_clk_div(sai, div);
 }
 
 static int stm32_sai_hw_params(struct snd_pcm_substream *substream,
@@ -881,6 +1061,9 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream,
 			   SAI_XCR1_NODIV);
 
 	clk_disable_unprepare(sai->sai_ck);
+
+	clk_rate_exclusive_put(sai->sai_mclk);
+
 	sai->substream = NULL;
 }
 
@@ -903,6 +1086,8 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 	struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev);
 	int cr1 = 0, cr1_mask;
 
+	sai->cpu_dai = cpu_dai;
+
 	sai->dma_params.addr = (dma_addr_t)(sai->phys_addr + STM_SAI_DR_REGX);
 	/*
 	 * DMA supports 4, 8 or 16 burst sizes. Burst size 4 is the best choice,
@@ -1124,16 +1309,15 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev,
 	sai->sync = SAI_SYNC_NONE;
 	if (args.np) {
 		if (args.np == np) {
-			dev_err(&pdev->dev, "%s sync own reference\n",
-				np->name);
+			dev_err(&pdev->dev, "%pOFn sync own reference\n", np);
 			of_node_put(args.np);
 			return -EINVAL;
 		}
 
 		sai->np_sync_provider  = of_get_parent(args.np);
 		if (!sai->np_sync_provider) {
-			dev_err(&pdev->dev, "%s parent node not found\n",
-				np->name);
+			dev_err(&pdev->dev, "%pOFn parent node not found\n",
+				np);
 			of_node_put(args.np);
 			return -ENODEV;
 		}
@@ -1182,6 +1366,23 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev,
 		return PTR_ERR(sai->sai_ck);
 	}
 
+	if (STM_SAI_IS_F4(sai->pdata))
+		return 0;
+
+	/* Register mclk provider if requested */
+	if (of_find_property(np, "#clock-cells", NULL)) {
+		ret = stm32_sai_add_mclk_provider(sai);
+		if (ret < 0)
+			return ret;
+	} else {
+		sai->sai_mclk = devm_clk_get(&pdev->dev, "MCLK");
+		if (IS_ERR(sai->sai_mclk)) {
+			if (PTR_ERR(sai->sai_mclk) != -ENOENT)
+				return PTR_ERR(sai->sai_mclk);
+			sai->sai_mclk = NULL;
+		}
+	}
+
 	return 0;
 }
 
diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig
index 22408bc2d6ec..66aad0d3f9c7 100644
--- a/sound/soc/sunxi/Kconfig
+++ b/sound/soc/sunxi/Kconfig
@@ -12,7 +12,7 @@ config SND_SUN4I_CODEC
 config SND_SUN8I_CODEC
 	tristate "Allwinner SUN8I audio codec"
 	depends on OF
-	depends on MACH_SUN8I || COMPILE_TEST
+	depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST
 	select REGMAP_MMIO
 	help
 	  This option enables the digital part of the internal audio codec for
@@ -23,11 +23,19 @@ config SND_SUN8I_CODEC
 config SND_SUN8I_CODEC_ANALOG
 	tristate "Allwinner sun8i Codec Analog Controls Support"
 	depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST
-	select REGMAP
+	select SND_SUN8I_ADDA_PR_REGMAP
 	help
 	  Say Y or M if you want to add support for the analog controls for
 	  the codec embedded in newer Allwinner SoCs.
 
+config SND_SUN50I_CODEC_ANALOG
+	tristate "Allwinner sun50i Codec Analog Controls Support"
+	depends on (ARM64 && ARCH_SUNXI) || COMPILE_TEST
+	select SND_SUNXI_ADDA_PR_REGMAP
+	help
+	  Say Y or M if you want to add support for the analog controls for
+	  the codec embedded in Allwinner A64 SoC.
+
 config SND_SUN4I_I2S
 	tristate "Allwinner A10 I2S Support"
 	select SND_SOC_GENERIC_DMAENGINE_PCM
@@ -45,4 +53,9 @@ config SND_SUN4I_SPDIF
 	help
 	  Say Y or M to add support for the S/PDIF audio block in the Allwinner
 	  A10 and affiliated SoCs.
+
+config SND_SUN8I_ADDA_PR_REGMAP
+	tristate
+	select REGMAP
+
 endmenu
diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile
index 4a9ef67386ca..a86be340a076 100644
--- a/sound/soc/sunxi/Makefile
+++ b/sound/soc/sunxi/Makefile
@@ -3,4 +3,6 @@ obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o
 obj-$(CONFIG_SND_SUN4I_I2S) += sun4i-i2s.o
 obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o
 obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o
+obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o
 obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o
+obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index a4aa931ebfae..d5ec1a20499d 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -644,40 +644,6 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 	return 0;
 }
 
-static int sun4i_i2s_startup(struct snd_pcm_substream *substream,
-			     struct snd_soc_dai *dai)
-{
-	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-
-	/* Enable the whole hardware block */
-	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
-			   SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN);
-
-	/* Enable the first output line */
-	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
-			   SUN4I_I2S_CTRL_SDO_EN_MASK,
-			   SUN4I_I2S_CTRL_SDO_EN(0));
-
-
-	return clk_prepare_enable(i2s->mod_clk);
-}
-
-static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream,
-			       struct snd_soc_dai *dai)
-{
-	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-
-	clk_disable_unprepare(i2s->mod_clk);
-
-	/* Disable our output lines */
-	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
-			   SUN4I_I2S_CTRL_SDO_EN_MASK, 0);
-
-	/* Disable the whole hardware block */
-	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
-			   SUN4I_I2S_CTRL_GL_EN, 0);
-}
-
 static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 				unsigned int freq, int dir)
 {
@@ -695,8 +661,6 @@ static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = {
 	.hw_params	= sun4i_i2s_hw_params,
 	.set_fmt	= sun4i_i2s_set_fmt,
 	.set_sysclk	= sun4i_i2s_set_sysclk,
-	.shutdown	= sun4i_i2s_shutdown,
-	.startup	= sun4i_i2s_startup,
 	.trigger	= sun4i_i2s_trigger,
 };
 
@@ -869,6 +833,21 @@ static int sun4i_i2s_runtime_resume(struct device *dev)
 		goto err_disable_clk;
 	}
 
+	/* Enable the whole hardware block */
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+			   SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN);
+
+	/* Enable the first output line */
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+			   SUN4I_I2S_CTRL_SDO_EN_MASK,
+			   SUN4I_I2S_CTRL_SDO_EN(0));
+
+	ret = clk_prepare_enable(i2s->mod_clk);
+	if (ret) {
+		dev_err(dev, "Failed to enable module clock\n");
+		goto err_disable_clk;
+	}
+
 	return 0;
 
 err_disable_clk:
@@ -880,6 +859,16 @@ static int sun4i_i2s_runtime_suspend(struct device *dev)
 {
 	struct sun4i_i2s *i2s = dev_get_drvdata(dev);
 
+	clk_disable_unprepare(i2s->mod_clk);
+
+	/* Disable our output lines */
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+			   SUN4I_I2S_CTRL_SDO_EN_MASK, 0);
+
+	/* Disable the whole hardware block */
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+			   SUN4I_I2S_CTRL_GL_EN, 0);
+
 	regcache_cache_only(i2s->regmap, true);
 
 	clk_disable_unprepare(i2s->bus_clk);
@@ -961,6 +950,23 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
 	.field_rxchansel	= REG_FIELD(SUN8I_I2S_RX_CHAN_SEL_REG, 0, 2),
 };
 
+static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
+	.has_reset		= true,
+	.reg_offset_txdata	= SUN8I_I2S_FIFO_TX_REG,
+	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
+	.has_slave_select_bit	= true,
+	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
+	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
+	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
+	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
+	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
+	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
+	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
+	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
+	.field_rxchansel	= REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2),
+};
+
 static int sun4i_i2s_init_regmap_fields(struct device *dev,
 					struct sun4i_i2s *i2s)
 {
@@ -1169,6 +1175,10 @@ static const struct of_device_id sun4i_i2s_match[] = {
 		.compatible = "allwinner,sun8i-h3-i2s",
 		.data = &sun8i_h3_i2s_quirks,
 	},
+	{
+		.compatible = "allwinner,sun50i-a64-codec-i2s",
+		.data = &sun50i_a64_codec_i2s_quirks,
+	},
 	{}
 };
 MODULE_DEVICE_TABLE(of, sun4i_i2s_match);
diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c
new file mode 100644
index 000000000000..8f5f999df631
--- /dev/null
+++ b/sound/soc/sunxi/sun50i-codec-analog.c
@@ -0,0 +1,444 @@
+// SPDX-License-Identifier: GPL-2.0+
+/*
+ * This driver supports the analog controls for the internal codec
+ * found in Allwinner's A64 SoC.
+ *
+ * Copyright (C) 2016 Chen-Yu Tsai <wens@csie.org>
+ * Copyright (C) 2017 Marcus Cooper <codekipper@gmail.com>
+ * Copyright (C) 2018 Vasily Khoruzhick <anarsoul@gmail.com>
+ *
+ * Based on sun8i-codec-analog.c
+ *
+ */
+
+#include <linux/io.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <linux/platform_device.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "sun8i-adda-pr-regmap.h"
+
+/* Codec analog control register offsets and bit fields */
+#define SUN50I_ADDA_HP_CTRL		0x00
+#define SUN50I_ADDA_HP_CTRL_PA_CLK_GATE		7
+#define SUN50I_ADDA_HP_CTRL_HPPA_EN		6
+#define SUN50I_ADDA_HP_CTRL_HPVOL		0
+
+#define SUN50I_ADDA_OL_MIX_CTRL		0x01
+#define SUN50I_ADDA_OL_MIX_CTRL_MIC1		6
+#define SUN50I_ADDA_OL_MIX_CTRL_MIC2		5
+#define SUN50I_ADDA_OL_MIX_CTRL_PHONE		4
+#define SUN50I_ADDA_OL_MIX_CTRL_PHONEN		3
+#define SUN50I_ADDA_OL_MIX_CTRL_LINEINL		2
+#define SUN50I_ADDA_OL_MIX_CTRL_DACL		1
+#define SUN50I_ADDA_OL_MIX_CTRL_DACR		0
+
+#define SUN50I_ADDA_OR_MIX_CTRL		0x02
+#define SUN50I_ADDA_OR_MIX_CTRL_MIC1		6
+#define SUN50I_ADDA_OR_MIX_CTRL_MIC2		5
+#define SUN50I_ADDA_OR_MIX_CTRL_PHONE		4
+#define SUN50I_ADDA_OR_MIX_CTRL_PHONEP		3
+#define SUN50I_ADDA_OR_MIX_CTRL_LINEINR		2
+#define SUN50I_ADDA_OR_MIX_CTRL_DACR		1
+#define SUN50I_ADDA_OR_MIX_CTRL_DACL		0
+
+#define SUN50I_ADDA_LINEOUT_CTRL0	0x05
+#define SUN50I_ADDA_LINEOUT_CTRL0_LEN		7
+#define SUN50I_ADDA_LINEOUT_CTRL0_REN		6
+#define SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL	5
+#define SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL	4
+
+#define SUN50I_ADDA_LINEOUT_CTRL1	0x06
+#define SUN50I_ADDA_LINEOUT_CTRL1_VOL		0
+
+#define SUN50I_ADDA_MIC1_CTRL		0x07
+#define SUN50I_ADDA_MIC1_CTRL_MIC1G		4
+#define SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN		3
+#define SUN50I_ADDA_MIC1_CTRL_MIC1BOOST		0
+
+#define SUN50I_ADDA_MIC2_CTRL		0x08
+#define SUN50I_ADDA_MIC2_CTRL_MIC2G		4
+#define SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN		3
+#define SUN50I_ADDA_MIC2_CTRL_MIC2BOOST		0
+
+#define SUN50I_ADDA_LINEIN_CTRL		0x09
+#define SUN50I_ADDA_LINEIN_CTRL_LINEING		0
+
+#define SUN50I_ADDA_MIX_DAC_CTRL	0x0a
+#define SUN50I_ADDA_MIX_DAC_CTRL_DACAREN	7
+#define SUN50I_ADDA_MIX_DAC_CTRL_DACALEN	6
+#define SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN		5
+#define SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN		4
+#define SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE	3
+#define SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE	2
+#define SUN50I_ADDA_MIX_DAC_CTRL_RHPIS		1
+#define SUN50I_ADDA_MIX_DAC_CTRL_LHPIS		0
+
+#define SUN50I_ADDA_L_ADCMIX_SRC	0x0b
+#define SUN50I_ADDA_L_ADCMIX_SRC_MIC1		6
+#define SUN50I_ADDA_L_ADCMIX_SRC_MIC2		5
+#define SUN50I_ADDA_L_ADCMIX_SRC_PHONE		4
+#define SUN50I_ADDA_L_ADCMIX_SRC_PHONEN		3
+#define SUN50I_ADDA_L_ADCMIX_SRC_LINEINL	2
+#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL		1
+#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR		0
+
+#define SUN50I_ADDA_R_ADCMIX_SRC	0x0c
+#define SUN50I_ADDA_R_ADCMIX_SRC_MIC1		6
+#define SUN50I_ADDA_R_ADCMIX_SRC_MIC2		5
+#define SUN50I_ADDA_R_ADCMIX_SRC_PHONE		4
+#define SUN50I_ADDA_R_ADCMIX_SRC_PHONEP		3
+#define SUN50I_ADDA_R_ADCMIX_SRC_LINEINR	2
+#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXR		1
+#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXL		0
+
+#define SUN50I_ADDA_ADC_CTRL		0x0d
+#define SUN50I_ADDA_ADC_CTRL_ADCREN		7
+#define SUN50I_ADDA_ADC_CTRL_ADCLEN		6
+#define SUN50I_ADDA_ADC_CTRL_ADCG		0
+
+#define SUN50I_ADDA_HS_MBIAS_CTRL	0x0e
+#define SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN	7
+
+#define SUN50I_ADDA_JACK_MIC_CTRL	0x1d
+#define SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN	5
+
+/* mixer controls */
+static const struct snd_kcontrol_new sun50i_a64_codec_mixer_controls[] = {
+	SOC_DAPM_DOUBLE_R("DAC Playback Switch",
+			  SUN50I_ADDA_OL_MIX_CTRL,
+			  SUN50I_ADDA_OR_MIX_CTRL,
+			  SUN50I_ADDA_OL_MIX_CTRL_DACL, 1, 0),
+	SOC_DAPM_DOUBLE_R("DAC Reversed Playback Switch",
+			  SUN50I_ADDA_OL_MIX_CTRL,
+			  SUN50I_ADDA_OR_MIX_CTRL,
+			  SUN50I_ADDA_OL_MIX_CTRL_DACR, 1, 0),
+	SOC_DAPM_DOUBLE_R("Line In Playback Switch",
+			  SUN50I_ADDA_OL_MIX_CTRL,
+			  SUN50I_ADDA_OR_MIX_CTRL,
+			  SUN50I_ADDA_OL_MIX_CTRL_LINEINL, 1, 0),
+	SOC_DAPM_DOUBLE_R("Mic1 Playback Switch",
+			  SUN50I_ADDA_OL_MIX_CTRL,
+			  SUN50I_ADDA_OR_MIX_CTRL,
+			  SUN50I_ADDA_OL_MIX_CTRL_MIC1, 1, 0),
+	SOC_DAPM_DOUBLE_R("Mic2 Playback Switch",
+			  SUN50I_ADDA_OL_MIX_CTRL,
+			  SUN50I_ADDA_OR_MIX_CTRL,
+			  SUN50I_ADDA_OL_MIX_CTRL_MIC2, 1, 0),
+};
+
+/* ADC mixer controls */
+static const struct snd_kcontrol_new sun50i_codec_adc_mixer_controls[] = {
+	SOC_DAPM_DOUBLE_R("Mixer Capture Switch",
+			  SUN50I_ADDA_L_ADCMIX_SRC,
+			  SUN50I_ADDA_R_ADCMIX_SRC,
+			  SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL, 1, 0),
+	SOC_DAPM_DOUBLE_R("Mixer Reversed Capture Switch",
+			  SUN50I_ADDA_L_ADCMIX_SRC,
+			  SUN50I_ADDA_R_ADCMIX_SRC,
+			  SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR, 1, 0),
+	SOC_DAPM_DOUBLE_R("Line In Capture Switch",
+			  SUN50I_ADDA_L_ADCMIX_SRC,
+			  SUN50I_ADDA_R_ADCMIX_SRC,
+			  SUN50I_ADDA_L_ADCMIX_SRC_LINEINL, 1, 0),
+	SOC_DAPM_DOUBLE_R("Mic1 Capture Switch",
+			  SUN50I_ADDA_L_ADCMIX_SRC,
+			  SUN50I_ADDA_R_ADCMIX_SRC,
+			  SUN50I_ADDA_L_ADCMIX_SRC_MIC1, 1, 0),
+	SOC_DAPM_DOUBLE_R("Mic2 Capture Switch",
+			  SUN50I_ADDA_L_ADCMIX_SRC,
+			  SUN50I_ADDA_R_ADCMIX_SRC,
+			  SUN50I_ADDA_L_ADCMIX_SRC_MIC2, 1, 0),
+};
+
+static const DECLARE_TLV_DB_SCALE(sun50i_codec_out_mixer_pregain_scale,
+				  -450, 150, 0);
+static const DECLARE_TLV_DB_RANGE(sun50i_codec_mic_gain_scale,
+	0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+	1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0),
+);
+
+static const DECLARE_TLV_DB_SCALE(sun50i_codec_hp_vol_scale, -6300, 100, 1);
+
+static const DECLARE_TLV_DB_RANGE(sun50i_codec_lineout_vol_scale,
+	0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1),
+	2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0),
+);
+
+
+/* volume / mute controls */
+static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = {
+	SOC_SINGLE_TLV("Headphone Playback Volume",
+		       SUN50I_ADDA_HP_CTRL,
+		       SUN50I_ADDA_HP_CTRL_HPVOL, 0x3f, 0,
+		       sun50i_codec_hp_vol_scale),
+
+	SOC_DOUBLE("Headphone Playback Switch",
+		   SUN50I_ADDA_MIX_DAC_CTRL,
+		   SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE,
+		   SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE, 1, 0),
+
+	/* Mixer pre-gain */
+	SOC_SINGLE_TLV("Mic1 Playback Volume", SUN50I_ADDA_MIC1_CTRL,
+		       SUN50I_ADDA_MIC1_CTRL_MIC1G,
+		       0x7, 0, sun50i_codec_out_mixer_pregain_scale),
+
+	/* Microphone Amp boost gain */
+	SOC_SINGLE_TLV("Mic1 Boost Volume", SUN50I_ADDA_MIC1_CTRL,
+		       SUN50I_ADDA_MIC1_CTRL_MIC1BOOST, 0x7, 0,
+		       sun50i_codec_mic_gain_scale),
+
+	/* Mixer pre-gain */
+	SOC_SINGLE_TLV("Mic2 Playback Volume",
+		       SUN50I_ADDA_MIC2_CTRL, SUN50I_ADDA_MIC2_CTRL_MIC2G,
+		       0x7, 0, sun50i_codec_out_mixer_pregain_scale),
+
+	/* Microphone Amp boost gain */
+	SOC_SINGLE_TLV("Mic2 Boost Volume", SUN50I_ADDA_MIC2_CTRL,
+		       SUN50I_ADDA_MIC2_CTRL_MIC2BOOST, 0x7, 0,
+		       sun50i_codec_mic_gain_scale),
+
+	/* ADC */
+	SOC_SINGLE_TLV("ADC Gain Capture Volume", SUN50I_ADDA_ADC_CTRL,
+		       SUN50I_ADDA_ADC_CTRL_ADCG, 0x7, 0,
+		       sun50i_codec_out_mixer_pregain_scale),
+
+	/* Mixer pre-gain */
+	SOC_SINGLE_TLV("Line In Playback Volume", SUN50I_ADDA_LINEIN_CTRL,
+		       SUN50I_ADDA_LINEIN_CTRL_LINEING,
+		       0x7, 0, sun50i_codec_out_mixer_pregain_scale),
+
+	SOC_SINGLE_TLV("Line Out Playback Volume",
+		       SUN50I_ADDA_LINEOUT_CTRL1,
+		       SUN50I_ADDA_LINEOUT_CTRL1_VOL, 0x1f, 0,
+		       sun50i_codec_lineout_vol_scale),
+
+	SOC_DOUBLE("Line Out Playback Switch",
+		   SUN50I_ADDA_LINEOUT_CTRL0,
+		   SUN50I_ADDA_LINEOUT_CTRL0_LEN,
+		   SUN50I_ADDA_LINEOUT_CTRL0_REN, 1, 0),
+
+};
+
+static const char * const sun50i_codec_hp_src_enum_text[] = {
+	"DAC", "Mixer",
+};
+
+static SOC_ENUM_DOUBLE_DECL(sun50i_codec_hp_src_enum,
+			    SUN50I_ADDA_MIX_DAC_CTRL,
+			    SUN50I_ADDA_MIX_DAC_CTRL_LHPIS,
+			    SUN50I_ADDA_MIX_DAC_CTRL_RHPIS,
+			    sun50i_codec_hp_src_enum_text);
+
+static const struct snd_kcontrol_new sun50i_codec_hp_src[] = {
+	SOC_DAPM_ENUM("Headphone Source Playback Route",
+		      sun50i_codec_hp_src_enum),
+};
+
+static const char * const sun50i_codec_lineout_src_enum_text[] = {
+	"Stereo", "Mono Differential",
+};
+
+static SOC_ENUM_DOUBLE_DECL(sun50i_codec_lineout_src_enum,
+			    SUN50I_ADDA_LINEOUT_CTRL0,
+			    SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL,
+			    SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL,
+			    sun50i_codec_lineout_src_enum_text);
+
+static const struct snd_kcontrol_new sun50i_codec_lineout_src[] = {
+	SOC_DAPM_ENUM("Line Out Source Playback Route",
+		      sun50i_codec_lineout_src_enum),
+};
+
+static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = {
+	/* DAC */
+	SND_SOC_DAPM_DAC("Left DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL,
+			 SUN50I_ADDA_MIX_DAC_CTRL_DACALEN, 0),
+	SND_SOC_DAPM_DAC("Right DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL,
+			 SUN50I_ADDA_MIX_DAC_CTRL_DACAREN, 0),
+	/* ADC */
+	SND_SOC_DAPM_ADC("Left ADC", NULL, SUN50I_ADDA_ADC_CTRL,
+			 SUN50I_ADDA_ADC_CTRL_ADCLEN, 0),
+	SND_SOC_DAPM_ADC("Right ADC", NULL, SUN50I_ADDA_ADC_CTRL,
+			 SUN50I_ADDA_ADC_CTRL_ADCREN, 0),
+	/*
+	 * Due to this component and the codec belonging to separate DAPM
+	 * contexts, we need to manually link the above widgets to their
+	 * stream widgets at the card level.
+	 */
+
+	SND_SOC_DAPM_MUX("Headphone Source Playback Route",
+			 SND_SOC_NOPM, 0, 0, sun50i_codec_hp_src),
+	SND_SOC_DAPM_OUT_DRV("Headphone Amp", SUN50I_ADDA_HP_CTRL,
+			     SUN50I_ADDA_HP_CTRL_HPPA_EN, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("HP"),
+
+	SND_SOC_DAPM_MUX("Line Out Source Playback Route",
+			 SND_SOC_NOPM, 0, 0, sun50i_codec_lineout_src),
+	SND_SOC_DAPM_OUTPUT("LINEOUT"),
+
+	/* Microphone inputs */
+	SND_SOC_DAPM_INPUT("MIC1"),
+
+	/* Microphone Bias */
+	SND_SOC_DAPM_SUPPLY("MBIAS", SUN50I_ADDA_HS_MBIAS_CTRL,
+			    SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN,
+			    0, NULL, 0),
+
+	/* Mic input path */
+	SND_SOC_DAPM_PGA("Mic1 Amplifier", SUN50I_ADDA_MIC1_CTRL,
+			 SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN, 0, NULL, 0),
+
+	/* Microphone input */
+	SND_SOC_DAPM_INPUT("MIC2"),
+
+	/* Microphone Bias */
+	SND_SOC_DAPM_SUPPLY("HBIAS", SUN50I_ADDA_JACK_MIC_CTRL,
+			    SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN,
+			    0, NULL, 0),
+
+	/* Mic input path */
+	SND_SOC_DAPM_PGA("Mic2 Amplifier", SUN50I_ADDA_MIC2_CTRL,
+			 SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN, 0, NULL, 0),
+
+	/* Line input */
+	SND_SOC_DAPM_INPUT("LINEIN"),
+
+	/* Mixers */
+	SND_SOC_DAPM_MIXER("Left Mixer", SUN50I_ADDA_MIX_DAC_CTRL,
+			   SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN, 0,
+			   sun50i_a64_codec_mixer_controls,
+			   ARRAY_SIZE(sun50i_a64_codec_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Right Mixer", SUN50I_ADDA_MIX_DAC_CTRL,
+			   SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN, 0,
+			   sun50i_a64_codec_mixer_controls,
+			   ARRAY_SIZE(sun50i_a64_codec_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Left ADC Mixer", SUN50I_ADDA_ADC_CTRL,
+			   SUN50I_ADDA_ADC_CTRL_ADCLEN, 0,
+			   sun50i_codec_adc_mixer_controls,
+			   ARRAY_SIZE(sun50i_codec_adc_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Right ADC Mixer", SUN50I_ADDA_ADC_CTRL,
+			   SUN50I_ADDA_ADC_CTRL_ADCREN, 0,
+			   sun50i_codec_adc_mixer_controls,
+			   ARRAY_SIZE(sun50i_codec_adc_mixer_controls)),
+};
+
+static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = {
+	/* Left Mixer Routes */
+	{ "Left Mixer", "DAC Playback Switch", "Left DAC" },
+	{ "Left Mixer", "DAC Reversed Playback Switch", "Right DAC" },
+	{ "Left Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" },
+
+	/* Right Mixer Routes */
+	{ "Right Mixer", "DAC Playback Switch", "Right DAC" },
+	{ "Right Mixer", "DAC Reversed Playback Switch", "Left DAC" },
+	{ "Right Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" },
+
+	/* Left ADC Mixer Routes */
+	{ "Left ADC Mixer", "Mixer Capture Switch", "Left Mixer" },
+	{ "Left ADC Mixer", "Mixer Reversed Capture Switch", "Right Mixer" },
+	{ "Left ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" },
+
+	/* Right ADC Mixer Routes */
+	{ "Right ADC Mixer", "Mixer Capture Switch", "Right Mixer" },
+	{ "Right ADC Mixer", "Mixer Reversed Capture Switch", "Left Mixer" },
+	{ "Right ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" },
+
+	/* ADC Routes */
+	{ "Left ADC", NULL, "Left ADC Mixer" },
+	{ "Right ADC", NULL, "Right ADC Mixer" },
+
+	/* Headphone Routes */
+	{ "Headphone Source Playback Route", "DAC", "Left DAC" },
+	{ "Headphone Source Playback Route", "DAC", "Right DAC" },
+	{ "Headphone Source Playback Route", "Mixer", "Left Mixer" },
+	{ "Headphone Source Playback Route", "Mixer", "Right Mixer" },
+	{ "Headphone Amp", NULL, "Headphone Source Playback Route" },
+	{ "HP", NULL, "Headphone Amp" },
+
+	/* Microphone Routes */
+	{ "Mic1 Amplifier", NULL, "MIC1"},
+
+	/* Microphone Routes */
+	{ "Mic2 Amplifier", NULL, "MIC2"},
+	{ "Left Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" },
+	{ "Right Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" },
+	{ "Left ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" },
+	{ "Right ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" },
+
+	/* Line-in Routes */
+	{ "Left Mixer", "Line In Playback Switch", "LINEIN" },
+	{ "Right Mixer", "Line In Playback Switch", "LINEIN" },
+	{ "Left ADC Mixer", "Line In Capture Switch", "LINEIN" },
+	{ "Right ADC Mixer", "Line In Capture Switch", "LINEIN" },
+
+	/* Line-out Routes */
+	{ "Line Out Source Playback Route", "Stereo", "Left Mixer" },
+	{ "Line Out Source Playback Route", "Stereo", "Right Mixer" },
+	{ "Line Out Source Playback Route", "Mono Differential", "Left Mixer" },
+	{ "Line Out Source Playback Route", "Mono Differential",
+		"Right Mixer" },
+	{ "LINEOUT", NULL, "Line Out Source Playback Route" },
+};
+
+static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = {
+	.controls		= sun50i_a64_codec_controls,
+	.num_controls		= ARRAY_SIZE(sun50i_a64_codec_controls),
+	.dapm_widgets		= sun50i_a64_codec_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(sun50i_a64_codec_widgets),
+	.dapm_routes		= sun50i_a64_codec_routes,
+	.num_dapm_routes	= ARRAY_SIZE(sun50i_a64_codec_routes),
+};
+
+static const struct of_device_id sun50i_codec_analog_of_match[] = {
+	{
+		.compatible = "allwinner,sun50i-a64-codec-analog",
+	},
+	{}
+};
+MODULE_DEVICE_TABLE(of, sun50i_codec_analog_of_match);
+
+static int sun50i_codec_analog_probe(struct platform_device *pdev)
+{
+	struct resource *res;
+	struct regmap *regmap;
+	void __iomem *base;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	base = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(base)) {
+		dev_err(&pdev->dev, "Failed to map the registers\n");
+		return PTR_ERR(base);
+	}
+
+	regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base);
+	if (IS_ERR(regmap)) {
+		dev_err(&pdev->dev, "Failed to create regmap\n");
+		return PTR_ERR(regmap);
+	}
+
+	return devm_snd_soc_register_component(&pdev->dev,
+					       &sun50i_codec_analog_cmpnt_drv,
+					       NULL, 0);
+}
+
+static struct platform_driver sun50i_codec_analog_driver = {
+	.driver = {
+		.name = "sun50i-codec-analog",
+		.of_match_table = sun50i_codec_analog_of_match,
+	},
+	.probe = sun50i_codec_analog_probe,
+};
+module_platform_driver(sun50i_codec_analog_driver);
+
+MODULE_DESCRIPTION("Allwinner internal codec analog controls driver for A64");
+MODULE_AUTHOR("Vasily Khoruzhick <anarsoul@gmail.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:sun50i-codec-analog");
diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.c b/sound/soc/sunxi/sun8i-adda-pr-regmap.c
new file mode 100644
index 000000000000..e68ce9d2884d
--- /dev/null
+++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.c
@@ -0,0 +1,102 @@
+// SPDX-License-Identifier: GPL-2.0+
+/*
+ * This driver provides regmap to access to analog part of audio codec
+ * found on Allwinner A23, A31s, A33, H3 and A64 Socs
+ *
+ * Copyright 2016 Chen-Yu Tsai <wens@csie.org>
+ * Copyright (C) 2018 Vasily Khoruzhick <anarsoul@gmail.com>
+ */
+
+#include <linux/io.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+
+#include "sun8i-adda-pr-regmap.h"
+
+/* Analog control register access bits */
+#define ADDA_PR			0x0		/* PRCM base + 0x1c0 */
+#define ADDA_PR_RESET			BIT(28)
+#define ADDA_PR_WRITE			BIT(24)
+#define ADDA_PR_ADDR_SHIFT		16
+#define ADDA_PR_ADDR_MASK		GENMASK(4, 0)
+#define ADDA_PR_DATA_IN_SHIFT		8
+#define ADDA_PR_DATA_IN_MASK		GENMASK(7, 0)
+#define ADDA_PR_DATA_OUT_SHIFT		0
+#define ADDA_PR_DATA_OUT_MASK		GENMASK(7, 0)
+
+/* regmap access bits */
+static int adda_reg_read(void *context, unsigned int reg, unsigned int *val)
+{
+	void __iomem *base = (void __iomem *)context;
+	u32 tmp;
+
+	/* De-assert reset */
+	writel(readl(base) | ADDA_PR_RESET, base);
+
+	/* Clear write bit */
+	writel(readl(base) & ~ADDA_PR_WRITE, base);
+
+	/* Set register address */
+	tmp = readl(base);
+	tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
+	tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
+	writel(tmp, base);
+
+	/* Read back value */
+	*val = readl(base) & ADDA_PR_DATA_OUT_MASK;
+
+	return 0;
+}
+
+static int adda_reg_write(void *context, unsigned int reg, unsigned int val)
+{
+	void __iomem *base = (void __iomem *)context;
+	u32 tmp;
+
+	/* De-assert reset */
+	writel(readl(base) | ADDA_PR_RESET, base);
+
+	/* Set register address */
+	tmp = readl(base);
+	tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
+	tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
+	writel(tmp, base);
+
+	/* Set data to write */
+	tmp = readl(base);
+	tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT);
+	tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT;
+	writel(tmp, base);
+
+	/* Set write bit to signal a write */
+	writel(readl(base) | ADDA_PR_WRITE, base);
+
+	/* Clear write bit */
+	writel(readl(base) & ~ADDA_PR_WRITE, base);
+
+	return 0;
+}
+
+static const struct regmap_config adda_pr_regmap_cfg = {
+	.name		= "adda-pr",
+	.reg_bits	= 5,
+	.reg_stride	= 1,
+	.val_bits	= 8,
+	.reg_read	= adda_reg_read,
+	.reg_write	= adda_reg_write,
+	.fast_io	= true,
+	.max_register	= 31,
+};
+
+struct regmap *sun8i_adda_pr_regmap_init(struct device *dev,
+					 void __iomem *base)
+{
+	return devm_regmap_init(dev, NULL, base, &adda_pr_regmap_cfg);
+}
+EXPORT_SYMBOL_GPL(sun8i_adda_pr_regmap_init);
+
+MODULE_DESCRIPTION("Allwinner analog audio codec regmap driver");
+MODULE_AUTHOR("Vasily Khoruzhick <anarsoul@gmail.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:sunxi-adda-pr");
diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.h b/sound/soc/sunxi/sun8i-adda-pr-regmap.h
new file mode 100644
index 000000000000..a5ae95dfebc1
--- /dev/null
+++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.h
@@ -0,0 +1,7 @@
+/* SPDX-License-Identifier: GPL-2.0+ */
+/*
+ * Copyright (C) 2018 Vasily Khoruzhick <anarsoul@gmail.com>
+ */
+
+struct regmap *sun8i_adda_pr_regmap_init(struct device *dev,
+					 void __iomem *base);
diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c
index 485e79f292c4..916a46bbc1c8 100644
--- a/sound/soc/sunxi/sun8i-codec-analog.c
+++ b/sound/soc/sunxi/sun8i-codec-analog.c
@@ -27,6 +27,8 @@
 #include <sound/soc-dapm.h>
 #include <sound/tlv.h>
 
+#include "sun8i-adda-pr-regmap.h"
+
 /* Codec analog control register offsets and bit fields */
 #define SUN8I_ADDA_HP_VOLC		0x00
 #define SUN8I_ADDA_HP_VOLC_PA_CLK_GATE		7
@@ -120,81 +122,6 @@
 #define SUN8I_ADDA_ADC_AP_EN_ADCLEN		6
 #define SUN8I_ADDA_ADC_AP_EN_ADCG		0
 
-/* Analog control register access bits */
-#define ADDA_PR			0x0		/* PRCM base + 0x1c0 */
-#define ADDA_PR_RESET			BIT(28)
-#define ADDA_PR_WRITE			BIT(24)
-#define ADDA_PR_ADDR_SHIFT		16
-#define ADDA_PR_ADDR_MASK		GENMASK(4, 0)
-#define ADDA_PR_DATA_IN_SHIFT		8
-#define ADDA_PR_DATA_IN_MASK		GENMASK(7, 0)
-#define ADDA_PR_DATA_OUT_SHIFT		0
-#define ADDA_PR_DATA_OUT_MASK		GENMASK(7, 0)
-
-/* regmap access bits */
-static int adda_reg_read(void *context, unsigned int reg, unsigned int *val)
-{
-	void __iomem *base = (void __iomem *)context;
-	u32 tmp;
-
-	/* De-assert reset */
-	writel(readl(base) | ADDA_PR_RESET, base);
-
-	/* Clear write bit */
-	writel(readl(base) & ~ADDA_PR_WRITE, base);
-
-	/* Set register address */
-	tmp = readl(base);
-	tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
-	tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
-	writel(tmp, base);
-
-	/* Read back value */
-	*val = readl(base) & ADDA_PR_DATA_OUT_MASK;
-
-	return 0;
-}
-
-static int adda_reg_write(void *context, unsigned int reg, unsigned int val)
-{
-	void __iomem *base = (void __iomem *)context;
-	u32 tmp;
-
-	/* De-assert reset */
-	writel(readl(base) | ADDA_PR_RESET, base);
-
-	/* Set register address */
-	tmp = readl(base);
-	tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
-	tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
-	writel(tmp, base);
-
-	/* Set data to write */
-	tmp = readl(base);
-	tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT);
-	tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT;
-	writel(tmp, base);
-
-	/* Set write bit to signal a write */
-	writel(readl(base) | ADDA_PR_WRITE, base);
-
-	/* Clear write bit */
-	writel(readl(base) & ~ADDA_PR_WRITE, base);
-
-	return 0;
-}
-
-static const struct regmap_config adda_pr_regmap_cfg = {
-	.name		= "adda-pr",
-	.reg_bits	= 5,
-	.reg_stride	= 1,
-	.val_bits	= 8,
-	.reg_read	= adda_reg_read,
-	.reg_write	= adda_reg_write,
-	.fast_io	= true,
-	.max_register	= 24,
-};
-
 /* mixer controls */
 static const struct snd_kcontrol_new sun8i_codec_mixer_controls[] = {
 	SOC_DAPM_DOUBLE_R("DAC Playback Switch",
@@ -912,7 +839,7 @@ static int sun8i_codec_analog_probe(struct platform_device *pdev)
 		return PTR_ERR(base);
 	}
 
-	regmap = devm_regmap_init(&pdev->dev, NULL, base, &adda_pr_regmap_cfg);
+	regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base);
 	if (IS_ERR(regmap)) {
 		dev_err(&pdev->dev, "Failed to create regmap\n");
 		return PTR_ERR(regmap);
diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c
index fb37dd927e33..522a72fde78d 100644
--- a/sound/soc/sunxi/sun8i-codec.c
+++ b/sound/soc/sunxi/sun8i-codec.c
@@ -24,6 +24,7 @@
 #include <linux/io.h>
 #include <linux/pm_runtime.h>
 #include <linux/regmap.h>
+#include <linux/log2.h>
 
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
@@ -52,7 +53,6 @@
 #define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_INV		13
 #define SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV		9
 #define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV		6
-#define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_16		(1 << 6)
 #define SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ		4
 #define SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ_16		(1 << 4)
 #define SUN8I_AIF1CLK_CTRL_AIF1_DATA_FMT		2
@@ -300,12 +300,23 @@ static u8 sun8i_codec_get_bclk_div(struct sun8i_codec *scodec,
 	return best_val;
 }
 
+static int sun8i_codec_get_lrck_div(unsigned int channels,
+				    unsigned int word_size)
+{
+	unsigned int div = word_size * channels;
+
+	if (div < 16 || div > 256)
+		return -EINVAL;
+
+	return ilog2(div) - 4;
+}
+
 static int sun8i_codec_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *params,
 				 struct snd_soc_dai *dai)
 {
 	struct sun8i_codec *scodec = snd_soc_component_get_drvdata(dai->component);
-	int sample_rate;
+	int sample_rate, lrck_div;
 	u8 bclk_div;
 
 	/*
@@ -321,9 +332,14 @@ static int sun8i_codec_hw_params(struct snd_pcm_substream *substream,
 			   SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV_MASK,
 			   bclk_div << SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV);
 
+	lrck_div = sun8i_codec_get_lrck_div(params_channels(params),
+					    params_physical_width(params));
+	if (lrck_div < 0)
+		return lrck_div;
+
 	regmap_update_bits(scodec->regmap, SUN8I_AIF1CLK_CTRL,
 			   SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_MASK,
-			   SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_16);
+			   lrck_div << SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV);
 
 	sample_rate = sun8i_codec_get_hw_rate(params);
 	if (sample_rate < 0)
diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c
index 45a4aa9d2a47..901457da25ec 100644
--- a/sound/soc/tegra/tegra_sgtl5000.c
+++ b/sound/soc/tegra/tegra_sgtl5000.c
@@ -149,14 +149,14 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev,
 			"Property 'nvidia,i2s-controller' missing/invalid\n");
 		ret = -EINVAL;
-		goto err;
+		goto err_put_codec_of_node;
 	}
 
 	tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_dai.cpu_of_node;
 
 	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
 	if (ret)
-		goto err;
+		goto err_put_cpu_of_node;
 
 	ret = snd_soc_register_card(card);
 	if (ret) {
@@ -169,6 +169,13 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev)
 
 err_fini_utils:
 	tegra_asoc_utils_fini(&machine->util_data);
+err_put_cpu_of_node:
+	of_node_put(tegra_sgtl5000_dai.cpu_of_node);
+	tegra_sgtl5000_dai.cpu_of_node = NULL;
+	tegra_sgtl5000_dai.platform_of_node = NULL;
+err_put_codec_of_node:
+	of_node_put(tegra_sgtl5000_dai.codec_of_node);
+	tegra_sgtl5000_dai.codec_of_node = NULL;
 err:
 	return ret;
 }
@@ -183,6 +190,12 @@ static int tegra_sgtl5000_driver_remove(struct platform_device *pdev)
 
 	tegra_asoc_utils_fini(&machine->util_data);
 
+	of_node_put(tegra_sgtl5000_dai.cpu_of_node);
+	tegra_sgtl5000_dai.cpu_of_node = NULL;
+	tegra_sgtl5000_dai.platform_of_node = NULL;
+	of_node_put(tegra_sgtl5000_dai.codec_of_node);
+	tegra_sgtl5000_dai.codec_of_node = NULL;
+
 	return ret;
 }
 
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
index e2ad00e3cae1..1cfca698ae4b 100644
--- a/sound/soc/txx9/txx9aclc-ac97.c
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -208,13 +208,12 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev)
 	if (err < 0)
 		return err;
 
-	return snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component,
+	return devm_snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component,
 					  &txx9aclc_ac97_dai, 1);
 }
 
 static int txx9aclc_ac97_dev_remove(struct platform_device *pdev)
 {
-	snd_soc_unregister_component(&pdev->dev);
 	snd_soc_set_ac97_ops(NULL);
 	return 0;
 }
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index d55ca48de3ea..f4a72e39ffa9 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -200,6 +200,7 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb)
 			break;
 		}
 #ifdef CONFIG_SND_USB_CAIAQ_INPUT
+		/* fall through */
 	case EP1_CMD_READ_ERP:
 	case EP1_CMD_READ_ANALOG:
 		snd_usb_caiaq_input_dispatch(cdev, buf, urb->actual_length);
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index dcfc546d81b9..b737f0ec77d0 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1175,8 +1175,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream,
 		if (port->ep->umidi->disconnected) {
 			/* gobble up remaining bytes to prevent wait in
 			 * snd_rawmidi_drain_output */
-			while (!snd_rawmidi_transmit_empty(substream))
-				snd_rawmidi_transmit_ack(substream, 1);
+			snd_rawmidi_proceed(substream);
 			return;
 		}
 		tasklet_schedule(&port->ep->tasklet);
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index cbfb48bdea51..85ae0ff2382a 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -29,6 +29,7 @@
 
 #include <linux/hid.h>
 #include <linux/init.h>
+#include <linux/math64.h>
 #include <linux/slab.h>
 #include <linux/usb.h>
 #include <linux/usb/audio.h>
@@ -1817,6 +1818,380 @@ static int dell_dock_mixer_init(struct usb_mixer_interface *mixer)
 	return 0;
 }
 
+/* RME Class Compliant device quirks */
+
+#define SND_RME_GET_STATUS1			23
+#define SND_RME_GET_CURRENT_FREQ		17
+#define SND_RME_CLK_SYSTEM_SHIFT		16
+#define SND_RME_CLK_SYSTEM_MASK			0x1f
+#define SND_RME_CLK_AES_SHIFT			8
+#define SND_RME_CLK_SPDIF_SHIFT			12
+#define SND_RME_CLK_AES_SPDIF_MASK		0xf
+#define SND_RME_CLK_SYNC_SHIFT			6
+#define SND_RME_CLK_SYNC_MASK			0x3
+#define SND_RME_CLK_FREQMUL_SHIFT		18
+#define SND_RME_CLK_FREQMUL_MASK		0x7
+#define SND_RME_CLK_SYSTEM(x) \
+	((x >> SND_RME_CLK_SYSTEM_SHIFT) & SND_RME_CLK_SYSTEM_MASK)
+#define SND_RME_CLK_AES(x) \
+	((x >> SND_RME_CLK_AES_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK)
+#define SND_RME_CLK_SPDIF(x) \
+	((x >> SND_RME_CLK_SPDIF_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK)
+#define SND_RME_CLK_SYNC(x) \
+	((x >> SND_RME_CLK_SYNC_SHIFT) & SND_RME_CLK_SYNC_MASK)
+#define SND_RME_CLK_FREQMUL(x) \
+	((x >> SND_RME_CLK_FREQMUL_SHIFT) & SND_RME_CLK_FREQMUL_MASK)
+#define SND_RME_CLK_AES_LOCK			0x1
+#define SND_RME_CLK_AES_SYNC			0x4
+#define SND_RME_CLK_SPDIF_LOCK			0x2
+#define SND_RME_CLK_SPDIF_SYNC			0x8
+#define SND_RME_SPDIF_IF_SHIFT			4
+#define SND_RME_SPDIF_FORMAT_SHIFT		5
+#define SND_RME_BINARY_MASK			0x1
+#define SND_RME_SPDIF_IF(x) \
+	((x >> SND_RME_SPDIF_IF_SHIFT) & SND_RME_BINARY_MASK)
+#define SND_RME_SPDIF_FORMAT(x) \
+	((x >> SND_RME_SPDIF_FORMAT_SHIFT) & SND_RME_BINARY_MASK)
+
+static const u32 snd_rme_rate_table[] = {
+	32000, 44100, 48000, 50000,
+	64000, 88200, 96000, 100000,
+	128000, 176400, 192000, 200000,
+	256000,	352800, 384000, 400000,
+	512000, 705600, 768000, 800000
+};
+/* maximum number of items for AES and S/PDIF rates for above table */
+#define SND_RME_RATE_IDX_AES_SPDIF_NUM		12
+
+enum snd_rme_domain {
+	SND_RME_DOMAIN_SYSTEM,
+	SND_RME_DOMAIN_AES,
+	SND_RME_DOMAIN_SPDIF
+};
+
+enum snd_rme_clock_status {
+	SND_RME_CLOCK_NOLOCK,
+	SND_RME_CLOCK_LOCK,
+	SND_RME_CLOCK_SYNC
+};
+
+static int snd_rme_read_value(struct snd_usb_audio *chip,
+			      unsigned int item,
+			      u32 *value)
+{
+	struct usb_device *dev = chip->dev;
+	int err;
+
+	err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
+			      item,
+			      USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE,
+			      0, 0,
+			      value, sizeof(*value));
+	if (err < 0)
+		dev_err(&dev->dev,
+			"unable to issue vendor read request %d (ret = %d)",
+			item, err);
+	return err;
+}
+
+static int snd_rme_get_status1(struct snd_kcontrol *kcontrol,
+			       u32 *status1)
+{
+	struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol);
+	struct snd_usb_audio *chip = list->mixer->chip;
+	int err;
+
+	err = snd_usb_lock_shutdown(chip);
+	if (err < 0)
+		return err;
+	err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, status1);
+	snd_usb_unlock_shutdown(chip);
+	return err;
+}
+
+static int snd_rme_rate_get(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	u32 status1;
+	u32 rate = 0;
+	int idx;
+	int err;
+
+	err = snd_rme_get_status1(kcontrol, &status1);
+	if (err < 0)
+		return err;
+	switch (kcontrol->private_value) {
+	case SND_RME_DOMAIN_SYSTEM:
+		idx = SND_RME_CLK_SYSTEM(status1);
+		if (idx < ARRAY_SIZE(snd_rme_rate_table))
+			rate = snd_rme_rate_table[idx];
+		break;
+	case SND_RME_DOMAIN_AES:
+		idx = SND_RME_CLK_AES(status1);
+		if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM)
+			rate = snd_rme_rate_table[idx];
+		break;
+	case SND_RME_DOMAIN_SPDIF:
+		idx = SND_RME_CLK_SPDIF(status1);
+		if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM)
+			rate = snd_rme_rate_table[idx];
+		break;
+	default:
+		return -EINVAL;
+	}
+	ucontrol->value.integer.value[0] = rate;
+	return 0;
+}
+
+static int snd_rme_sync_state_get(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	u32 status1;
+	int idx = SND_RME_CLOCK_NOLOCK;
+	int err;
+
+	err = snd_rme_get_status1(kcontrol, &status1);
+	if (err < 0)
+		return err;
+	switch (kcontrol->private_value) {
+	case SND_RME_DOMAIN_AES:  /* AES */
+		if (status1 & SND_RME_CLK_AES_SYNC)
+			idx = SND_RME_CLOCK_SYNC;
+		else if (status1 & SND_RME_CLK_AES_LOCK)
+			idx = SND_RME_CLOCK_LOCK;
+		break;
+	case SND_RME_DOMAIN_SPDIF:  /* SPDIF */
+		if (status1 & SND_RME_CLK_SPDIF_SYNC)
+			idx = SND_RME_CLOCK_SYNC;
+		else if (status1 & SND_RME_CLK_SPDIF_LOCK)
+			idx = SND_RME_CLOCK_LOCK;
+		break;
+	default:
+		return -EINVAL;
+	}
+	ucontrol->value.enumerated.item[0] = idx;
+	return 0;
+}
+
+static int snd_rme_spdif_if_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	u32 status1;
+	int err;
+
+	err = snd_rme_get_status1(kcontrol, &status1);
+	if (err < 0)
+		return err;
+	ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_IF(status1);
+	return 0;
+}
+
+static int snd_rme_spdif_format_get(struct snd_kcontrol *kcontrol,
+				    struct snd_ctl_elem_value *ucontrol)
+{
+	u32 status1;
+	int err;
+
+	err = snd_rme_get_status1(kcontrol, &status1);
+	if (err < 0)
+		return err;
+	ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_FORMAT(status1);
+	return 0;
+}
+
+static int snd_rme_sync_source_get(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	u32 status1;
+	int err;
+
+	err = snd_rme_get_status1(kcontrol, &status1);
+	if (err < 0)
+		return err;
+	ucontrol->value.enumerated.item[0] = SND_RME_CLK_SYNC(status1);
+	return 0;
+}
+
+static int snd_rme_current_freq_get(struct snd_kcontrol *kcontrol,
+				    struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol);
+	struct snd_usb_audio *chip = list->mixer->chip;
+	u32 status1;
+	const u64 num = 104857600000000ULL;
+	u32 den;
+	unsigned int freq;
+	int err;
+
+	err = snd_usb_lock_shutdown(chip);
+	if (err < 0)
+		return err;
+	err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, &status1);
+	if (err < 0)
+		goto end;
+	err = snd_rme_read_value(chip, SND_RME_GET_CURRENT_FREQ, &den);
+	if (err < 0)
+		goto end;
+	freq = (den == 0) ? 0 : div64_u64(num, den);
+	freq <<= SND_RME_CLK_FREQMUL(status1);
+	ucontrol->value.integer.value[0] = freq;
+
+end:
+	snd_usb_unlock_shutdown(chip);
+	return err;
+}
+
+static int snd_rme_rate_info(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	switch (kcontrol->private_value) {
+	case SND_RME_DOMAIN_SYSTEM:
+		uinfo->value.integer.min = 32000;
+		uinfo->value.integer.max = 800000;
+		break;
+	case SND_RME_DOMAIN_AES:
+	case SND_RME_DOMAIN_SPDIF:
+	default:
+		uinfo->value.integer.min = 0;
+		uinfo->value.integer.max = 200000;
+	}
+	uinfo->value.integer.step = 0;
+	return 0;
+}
+
+static int snd_rme_sync_state_info(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_info *uinfo)
+{
+	static const char *const sync_states[] = {
+		"No Lock", "Lock", "Sync"
+	};
+
+	return snd_ctl_enum_info(uinfo, 1,
+				 ARRAY_SIZE(sync_states), sync_states);
+}
+
+static int snd_rme_spdif_if_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	static const char *const spdif_if[] = {
+		"Coaxial", "Optical"
+	};
+
+	return snd_ctl_enum_info(uinfo, 1,
+				 ARRAY_SIZE(spdif_if), spdif_if);
+}
+
+static int snd_rme_spdif_format_info(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_info *uinfo)
+{
+	static const char *const optical_type[] = {
+		"Consumer", "Professional"
+	};
+
+	return snd_ctl_enum_info(uinfo, 1,
+				 ARRAY_SIZE(optical_type), optical_type);
+}
+
+static int snd_rme_sync_source_info(struct snd_kcontrol *kcontrol,
+				    struct snd_ctl_elem_info *uinfo)
+{
+	static const char *const sync_sources[] = {
+		"Internal", "AES", "SPDIF", "Internal"
+	};
+
+	return snd_ctl_enum_info(uinfo, 1,
+				 ARRAY_SIZE(sync_sources), sync_sources);
+}
+
+static struct snd_kcontrol_new snd_rme_controls[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "AES Rate",
+		.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = snd_rme_rate_info,
+		.get = snd_rme_rate_get,
+		.private_value = SND_RME_DOMAIN_AES
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "AES Sync",
+		.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = snd_rme_sync_state_info,
+		.get = snd_rme_sync_state_get,
+		.private_value = SND_RME_DOMAIN_AES
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "SPDIF Rate",
+		.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = snd_rme_rate_info,
+		.get = snd_rme_rate_get,
+		.private_value = SND_RME_DOMAIN_SPDIF
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "SPDIF Sync",
+		.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = snd_rme_sync_state_info,
+		.get = snd_rme_sync_state_get,
+		.private_value = SND_RME_DOMAIN_SPDIF
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "SPDIF Interface",
+		.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = snd_rme_spdif_if_info,
+		.get = snd_rme_spdif_if_get,
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "SPDIF Format",
+		.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = snd_rme_spdif_format_info,
+		.get = snd_rme_spdif_format_get,
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Sync Source",
+		.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = snd_rme_sync_source_info,
+		.get = snd_rme_sync_source_get
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "System Rate",
+		.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = snd_rme_rate_info,
+		.get = snd_rme_rate_get,
+		.private_value = SND_RME_DOMAIN_SYSTEM
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Current Frequency",
+		.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = snd_rme_rate_info,
+		.get = snd_rme_current_freq_get
+	}
+};
+
+static int snd_rme_controls_create(struct usb_mixer_interface *mixer)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(snd_rme_controls); ++i) {
+		err = add_single_ctl_with_resume(mixer, 0,
+						 NULL,
+						 &snd_rme_controls[i],
+						 NULL);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
 int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 {
 	int err = 0;
@@ -1904,6 +2279,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 	case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */
 		err = dell_dock_mixer_init(mixer);
 		break;
+
+	case USB_ID(0x2a39, 0x3fd2): /* RME ADI-2 Pro */
+	case USB_ID(0x2a39, 0x3fd3): /* RME ADI-2 DAC */
+	case USB_ID(0x2a39, 0x3fd4): /* RME */
+		err = snd_rme_controls_create(mixer);
+		break;
 	}
 
 	return err;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 08aa78007020..849953e5775c 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3346,19 +3346,14 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 				.ifnum = 0,
 				.type = QUIRK_AUDIO_STANDARD_MIXER,
 			},
-			/* Capture */
-			{
-				.ifnum = 1,
-				.type = QUIRK_IGNORE_INTERFACE,
-			},
 			/* Playback */
 			{
-				.ifnum = 2,
+				.ifnum = 1,
 				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
 				.data = &(const struct audioformat) {
 					.formats = SNDRV_PCM_FMTBIT_S16_LE,
 					.channels = 2,
-					.iface = 2,
+					.iface = 1,
 					.altsetting = 1,
 					.altset_idx = 1,
 					.attributes = UAC_EP_CS_ATTR_FILL_MAX |
diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c
index fa7dca5a68c8..83d76c345940 100644
--- a/sound/x86/intel_hdmi_audio.c
+++ b/sound/x86/intel_hdmi_audio.c
@@ -30,7 +30,6 @@
 #include <linux/pm_runtime.h>
 #include <linux/dma-mapping.h>
 #include <linux/delay.h>
-#include <asm/set_memory.h>
 #include <sound/core.h>
 #include <sound/asoundef.h>
 #include <sound/pcm.h>
@@ -1141,8 +1140,7 @@ static int had_pcm_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *hw_params)
 {
 	struct snd_intelhad *intelhaddata;
-	unsigned long addr;
-	int pages, buf_size, retval;
+	int buf_size, retval;
 
 	intelhaddata = snd_pcm_substream_chip(substream);
 	buf_size = params_buffer_bytes(hw_params);
@@ -1151,17 +1149,6 @@ static int had_pcm_hw_params(struct snd_pcm_substream *substream,
 		return retval;
 	dev_dbg(intelhaddata->dev, "%s:allocated memory = %d\n",
 		__func__, buf_size);
-	/* mark the pages as uncached region */
-	addr = (unsigned long) substream->runtime->dma_area;
-	pages = (substream->runtime->dma_bytes + PAGE_SIZE - 1) / PAGE_SIZE;
-	retval = set_memory_uc(addr, pages);
-	if (retval) {
-		dev_err(intelhaddata->dev, "set_memory_uc failed.Error:%d\n",
-			retval);
-		return retval;
-	}
-	memset(substream->runtime->dma_area, 0, buf_size);
-
 	return retval;
 }
 
@@ -1171,21 +1158,11 @@ static int had_pcm_hw_params(struct snd_pcm_substream *substream,
 static int had_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_intelhad *intelhaddata;
-	unsigned long addr;
-	u32 pages;
 
 	intelhaddata = snd_pcm_substream_chip(substream);
 	had_do_reset(intelhaddata);
 
-	/* mark back the pages as cached/writeback region before the free */
-	if (substream->runtime->dma_area != NULL) {
-		addr = (unsigned long) substream->runtime->dma_area;
-		pages = (substream->runtime->dma_bytes + PAGE_SIZE - 1) /
-								PAGE_SIZE;
-		set_memory_wb(addr, pages);
-		return snd_pcm_lib_free_pages(substream);
-	}
-	return 0;
+	return snd_pcm_lib_free_pages(substream);
 }
 
 /*
@@ -1860,7 +1837,7 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev)
 		 * try to allocate 600k buffer as default which is large enough
 		 */
 		snd_pcm_lib_preallocate_pages_for_all(pcm,
-						      SNDRV_DMA_TYPE_DEV, NULL,
+						      SNDRV_DMA_TYPE_DEV_UC, NULL,
 						      HAD_DEFAULT_BUFFER, HAD_MAX_BUFFER);
 
 		/* create controls */
diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c
index 129180e17db1..2cbd9679aca1 100644
--- a/sound/xen/xen_snd_front_alsa.c
+++ b/sound/xen/xen_snd_front_alsa.c
@@ -637,31 +637,31 @@ static int alsa_pb_fill_silence(struct snd_pcm_substream *substream,
  * to know when the buffer can be transferred to the backend.
  */
 
-static struct snd_pcm_ops snd_drv_alsa_playback_ops = {
-	.open = alsa_open,
-	.close = alsa_close,
-	.ioctl = snd_pcm_lib_ioctl,
-	.hw_params = alsa_hw_params,
-	.hw_free = alsa_hw_free,
-	.prepare = alsa_prepare,
-	.trigger = alsa_trigger,
-	.pointer = alsa_pointer,
-	.copy_user = alsa_pb_copy_user,
-	.copy_kernel = alsa_pb_copy_kernel,
-	.fill_silence = alsa_pb_fill_silence,
+static const struct snd_pcm_ops snd_drv_alsa_playback_ops = {
+	.open		= alsa_open,
+	.close		= alsa_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= alsa_hw_params,
+	.hw_free	= alsa_hw_free,
+	.prepare	= alsa_prepare,
+	.trigger	= alsa_trigger,
+	.pointer	= alsa_pointer,
+	.copy_user	= alsa_pb_copy_user,
+	.copy_kernel	= alsa_pb_copy_kernel,
+	.fill_silence	= alsa_pb_fill_silence,
 };
 
-static struct snd_pcm_ops snd_drv_alsa_capture_ops = {
-	.open = alsa_open,
-	.close = alsa_close,
-	.ioctl = snd_pcm_lib_ioctl,
-	.hw_params = alsa_hw_params,
-	.hw_free = alsa_hw_free,
-	.prepare = alsa_prepare,
-	.trigger = alsa_trigger,
-	.pointer = alsa_pointer,
-	.copy_user = alsa_cap_copy_user,
-	.copy_kernel = alsa_cap_copy_kernel,
+static const struct snd_pcm_ops snd_drv_alsa_capture_ops = {
+	.open		= alsa_open,
+	.close		= alsa_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= alsa_hw_params,
+	.hw_free	= alsa_hw_free,
+	.prepare	= alsa_prepare,
+	.trigger	= alsa_trigger,
+	.pointer	= alsa_pointer,
+	.copy_user	= alsa_cap_copy_user,
+	.copy_kernel	= alsa_cap_copy_kernel,
 };
 
 static int new_pcm_instance(struct xen_snd_front_card_info *card_info,