Merge branch 'for-3.0' into for-3.1

93 files changed, 9620 insertions, 7854 deletions

diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c
index 58804c7acfc..fd2188c3df2 100644
--- a/sound/aoa/codecs/tas.c
+++ b/sound/aoa/codecs/tas.c
@@ -170,7 +170,7 @@ static void tas_set_volume(struct tas *tas)
 	/* analysing the volume and mixer tables shows
 	 * that they are similar enough when we shift
 	 * the mixer table down by 4 bits. The error
-	 * is minuscule, in just one item the error
+	 * is miniscule, in just one item the error
 	 * is 1, at a value of 0x07f17b (mixer table
 	 * value is 0x07f17a) */
 	tmp = tas_gaintable[left];
diff --git a/sound/core/control.c b/sound/core/control.c
index a08ad57c49b..f8c5be46451 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -366,6 +366,70 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol)
 EXPORT_SYMBOL(snd_ctl_add);
 
 /**
+ * snd_ctl_replace - replace the control instance of the card
+ * @card: the card instance
+ * @kcontrol: the control instance to replace
+ * @add_on_replace: add the control if not already added
+ *
+ * Replaces the given control.  If the given control does not exist
+ * and the add_on_replace flag is set, the control is added.  If the
+ * control exists, it is destroyed first.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ *
+ * It frees automatically the control which cannot be added or replaced.
+ */
+int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol,
+		    bool add_on_replace)
+{
+	struct snd_ctl_elem_id id;
+	unsigned int idx;
+	struct snd_kcontrol *old;
+	int ret;
+
+	if (!kcontrol)
+		return -EINVAL;
+	if (snd_BUG_ON(!card || !kcontrol->info)) {
+		ret = -EINVAL;
+		goto error;
+	}
+	id = kcontrol->id;
+	down_write(&card->controls_rwsem);
+	old = snd_ctl_find_id(card, &id);
+	if (!old) {
+		if (add_on_replace)
+			goto add;
+		up_write(&card->controls_rwsem);
+		ret = -EINVAL;
+		goto error;
+	}
+	ret = snd_ctl_remove(card, old);
+	if (ret < 0) {
+		up_write(&card->controls_rwsem);
+		goto error;
+	}
+add:
+	if (snd_ctl_find_hole(card, kcontrol->count) < 0) {
+		up_write(&card->controls_rwsem);
+		ret = -ENOMEM;
+		goto error;
+	}
+	list_add_tail(&kcontrol->list, &card->controls);
+	card->controls_count += kcontrol->count;
+	kcontrol->id.numid = card->last_numid + 1;
+	card->last_numid += kcontrol->count;
+	up_write(&card->controls_rwsem);
+	for (idx = 0; idx < kcontrol->count; idx++, id.index++, id.numid++)
+		snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id);
+	return 0;
+
+error:
+	snd_ctl_free_one(kcontrol);
+	return ret;
+}
+EXPORT_SYMBOL(snd_ctl_replace);
+
+/**
  * snd_ctl_remove - remove the control from the card and release it
  * @card: the card instance
  * @kcontrol: the control instance to remove
@@ -640,13 +704,12 @@ static int snd_ctl_elem_list(struct snd_card *card,
 	struct snd_ctl_elem_list list;
 	struct snd_kcontrol *kctl;
 	struct snd_ctl_elem_id *dst, *id;
-	unsigned int offset, space, first, jidx;
+	unsigned int offset, space, jidx;
 	
 	if (copy_from_user(&list, _list, sizeof(list)))
 		return -EFAULT;
 	offset = list.offset;
 	space = list.space;
-	first = 0;
 	/* try limit maximum space */
 	if (space > 16384)
 		return -ENOMEM;
diff --git a/sound/core/init.c b/sound/core/init.c
index a0080aa45ae..2c041bb36ab 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -342,7 +342,6 @@ static const struct file_operations snd_shutdown_f_ops =
 int snd_card_disconnect(struct snd_card *card)
 {
 	struct snd_monitor_file *mfile;
-	struct file *file;
 	int err;
 
 	if (!card)
@@ -366,8 +365,6 @@ int snd_card_disconnect(struct snd_card *card)
 	
 	spin_lock(&card->files_lock);
 	list_for_each_entry(mfile, &card->files_list, list) {
-		file = mfile->file;
-
 		/* it's critical part, use endless loop */
 		/* we have no room to fail */
 		mfile->disconnected_f_op = mfile->file->f_op;
@@ -514,7 +511,7 @@ static void snd_card_set_id_no_lock(struct snd_card *card, const char *nid)
 	id = card->id;
 	
 	if (*id == '\0')
-		strcpy(id, "default");
+		strcpy(id, "Default");
 
 	while (1) {
 	      	if (loops-- == 0) {
diff --git a/sound/core/oss/linear.c b/sound/core/oss/linear.c
index 13b3f6f49fa..2045697f449 100644
--- a/sound/core/oss/linear.c
+++ b/sound/core/oss/linear.c
@@ -90,11 +90,8 @@ static snd_pcm_sframes_t linear_transfer(struct snd_pcm_plugin *plugin,
 			       struct snd_pcm_plugin_channel *dst_channels,
 			       snd_pcm_uframes_t frames)
 {
-	struct linear_priv *data;
-
 	if (snd_BUG_ON(!plugin || !src_channels || !dst_channels))
 		return -ENXIO;
-	data = (struct linear_priv *)plugin->extra_data;
 	if (frames == 0)
 		return 0;
 #ifdef CONFIG_SND_DEBUG
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 64449cb8f87..f1341308bed 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -189,6 +189,7 @@ static void xrun(struct snd_pcm_substream *substream)
 #define XRUN_LOG_CNT	10
 
 struct hwptr_log_entry {
+	unsigned int in_interrupt;
 	unsigned long jiffies;
 	snd_pcm_uframes_t pos;
 	snd_pcm_uframes_t period_size;
@@ -204,7 +205,7 @@ struct snd_pcm_hwptr_log {
 };
 
 static void xrun_log(struct snd_pcm_substream *substream,
-		     snd_pcm_uframes_t pos)
+		     snd_pcm_uframes_t pos, int in_interrupt)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_pcm_hwptr_log *log = runtime->hwptr_log;
@@ -220,6 +221,7 @@ static void xrun_log(struct snd_pcm_substream *substream,
 			return;
 	}
 	entry = &log->entries[log->idx];
+	entry->in_interrupt = in_interrupt;
 	entry->jiffies = jiffies;
 	entry->pos = pos;
 	entry->period_size = runtime->period_size;
@@ -246,9 +248,11 @@ static void xrun_log_show(struct snd_pcm_substream *substream)
 		entry = &log->entries[idx];
 		if (entry->period_size == 0)
 			break;
-		snd_printd("hwptr log: %s: j=%lu, pos=%ld/%ld/%ld, "
+		snd_printd("hwptr log: %s: %sj=%lu, pos=%ld/%ld/%ld, "
 			   "hwptr=%ld/%ld\n",
-			   name, entry->jiffies, (unsigned long)entry->pos,
+			   name, entry->in_interrupt ? "[Q] " : "",
+			   entry->jiffies,
+			   (unsigned long)entry->pos,
 			   (unsigned long)entry->period_size,
 			   (unsigned long)entry->buffer_size,
 			   (unsigned long)entry->old_hw_ptr,
@@ -262,7 +266,7 @@ static void xrun_log_show(struct snd_pcm_substream *substream)
 #else /* ! CONFIG_SND_PCM_XRUN_DEBUG */
 
 #define hw_ptr_error(substream, fmt, args...) do { } while (0)
-#define xrun_log(substream, pos)	do { } while (0)
+#define xrun_log(substream, pos, in_interrupt)	do { } while (0)
 #define xrun_log_show(substream)	do { } while (0)
 
 #endif
@@ -326,7 +330,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 	}
 	pos -= pos % runtime->min_align;
 	if (xrun_debug(substream, XRUN_DEBUG_LOG))
-		xrun_log(substream, pos);
+		xrun_log(substream, pos, in_interrupt);
 	hw_base = runtime->hw_ptr_base;
 	new_hw_ptr = hw_base + pos;
 	if (in_interrupt) {
@@ -1752,8 +1756,18 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 	wait_queue_t wait;
 	int err = 0;
 	snd_pcm_uframes_t avail = 0;
-	long tout;
-
+	long wait_time, tout;
+
+	if (runtime->no_period_wakeup)
+		wait_time = MAX_SCHEDULE_TIMEOUT;
+	else {
+		wait_time = 10;
+		if (runtime->rate) {
+			long t = runtime->period_size * 2 / runtime->rate;
+			wait_time = max(t, wait_time);
+		}
+		wait_time = msecs_to_jiffies(wait_time * 1000);
+	}
 	init_waitqueue_entry(&wait, current);
 	add_wait_queue(&runtime->tsleep, &wait);
 	for (;;) {
@@ -1761,9 +1775,8 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 			err = -ERESTARTSYS;
 			break;
 		}
-		set_current_state(TASK_INTERRUPTIBLE);
 		snd_pcm_stream_unlock_irq(substream);
-		tout = schedule_timeout(msecs_to_jiffies(10000));
+		tout = schedule_timeout_interruptible(wait_time);
 		snd_pcm_stream_lock_irq(substream);
 		switch (runtime->status->state) {
 		case SNDRV_PCM_STATE_SUSPENDED:
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 1a07750f383..1c6be91dfb9 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1481,11 +1481,20 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream,
 			break; /* all drained */
 		init_waitqueue_entry(&wait, current);
 		add_wait_queue(&to_check->sleep, &wait);
-		set_current_state(TASK_INTERRUPTIBLE);
 		snd_pcm_stream_unlock_irq(substream);
 		up_read(&snd_pcm_link_rwsem);
 		snd_power_unlock(card);
-		tout = schedule_timeout(10 * HZ);
+		if (runtime->no_period_wakeup)
+			tout = MAX_SCHEDULE_TIMEOUT;
+		else {
+			tout = 10;
+			if (runtime->rate) {
+				long t = runtime->period_size * 2 / runtime->rate;
+				tout = max(t, tout);
+			}
+			tout = msecs_to_jiffies(tout * 1000);
+		}
+		tout = schedule_timeout_interruptible(tout);
 		snd_power_lock(card);
 		down_read(&snd_pcm_link_rwsem);
 		snd_pcm_stream_lock_irq(substream);
@@ -1518,13 +1527,11 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream,
 static int snd_pcm_drop(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime;
-	struct snd_card *card;
 	int result = 0;
 	
 	if (PCM_RUNTIME_CHECK(substream))
 		return -ENXIO;
 	runtime = substream->runtime;
-	card = substream->pcm->card;
 
 	if (runtime->status->state == SNDRV_PCM_STATE_OPEN ||
 	    runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED ||
@@ -2056,7 +2063,6 @@ static int snd_pcm_open_file(struct file *file,
 {
 	struct snd_pcm_file *pcm_file;
 	struct snd_pcm_substream *substream;
-	struct snd_pcm_str *str;
 	int err;
 
 	if (rpcm_file)
@@ -2073,7 +2079,6 @@ static int snd_pcm_open_file(struct file *file,
 	}
 	pcm_file->substream = substream;
 	if (substream->ref_count == 1) {
-		str = substream->pstr;
 		substream->file = pcm_file;
 		substream->pcm_release = pcm_release_private;
 	}
@@ -3015,11 +3020,9 @@ static const struct vm_operations_struct snd_pcm_vm_ops_status =
 static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file *file,
 			       struct vm_area_struct *area)
 {
-	struct snd_pcm_runtime *runtime;
 	long size;
 	if (!(area->vm_flags & VM_READ))
 		return -EINVAL;
-	runtime = substream->runtime;
 	size = area->vm_end - area->vm_start;
 	if (size != PAGE_ALIGN(sizeof(struct snd_pcm_mmap_status)))
 		return -EINVAL;
@@ -3054,11 +3057,9 @@ static const struct vm_operations_struct snd_pcm_vm_ops_control =
 static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file *file,
 				struct vm_area_struct *area)
 {
-	struct snd_pcm_runtime *runtime;
 	long size;
 	if (!(area->vm_flags & VM_READ))
 		return -EINVAL;
-	runtime = substream->runtime;
 	size = area->vm_end - area->vm_start;
 	if (size != PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control)))
 		return -EINVAL;
diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c
index e7a8e9e4edb..f9077361c11 100644
--- a/sound/core/seq/seq_queue.c
+++ b/sound/core/seq/seq_queue.c
@@ -467,13 +467,11 @@ int snd_seq_queue_timer_open(int queueid)
 int snd_seq_queue_timer_close(int queueid)
 {
 	struct snd_seq_queue *queue;
-	struct snd_seq_timer *tmr;
 	int result = 0;
 
 	queue = queueptr(queueid);
 	if (queue == NULL)
 		return -EINVAL;
-	tmr = queue->timer;
 	snd_seq_timer_close(queue);
 	queuefree(queue);
 	return result;
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index e486f48660f..26071489970 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -22,4 +22,15 @@ config SND_FIREWIRE_SPEAKERS
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-firewire-speakers.
 
+config SND_ISIGHT
+	tristate "Apple iSight microphone"
+	select SND_PCM
+	select SND_FIREWIRE_LIB
+	help
+	  Say Y here to include support for the front and rear microphones
+	  of the Apple iSight web camera.
+
+	  To compile this driver as a module, choose M here: the module
+	  will be called snd-isight.
+
 endif # SND_FIREWIRE
diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile
index e5b1634d9ad..d71ed8935f7 100644
--- a/sound/firewire/Makefile
+++ b/sound/firewire/Makefile
@@ -1,6 +1,8 @@
 snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \
 			 fcp.o cmp.o amdtp.o
 snd-firewire-speakers-objs := speakers.o
+snd-isight-objs := isight.o
 
 obj-$(CONFIG_SND_FIREWIRE_LIB) += snd-firewire-lib.o
 obj-$(CONFIG_SND_FIREWIRE_SPEAKERS) += snd-firewire-speakers.o
+obj-$(CONFIG_SND_ISIGHT) += snd-isight.o
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index b18140ff2b9..87657dd7714 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -396,6 +396,7 @@ static void out_packet_callback(struct fw_iso_context *context, u32 cycle,
 
 	for (i = 0; i < packets; ++i)
 		queue_out_packet(s, ++cycle);
+	fw_iso_context_queue_flush(s->context);
 }
 
 static int queue_initial_skip_packets(struct amdtp_out_stream *s)
diff --git a/sound/firewire/cmp.c b/sound/firewire/cmp.c
index 4a37f3a6fab..14cacbc655d 100644
--- a/sound/firewire/cmp.c
+++ b/sound/firewire/cmp.c
@@ -49,10 +49,9 @@ static int pcr_modify(struct cmp_connection *c,
 		      enum bus_reset_handling bus_reset_handling)
 {
 	struct fw_device *device = fw_parent_device(c->resources.unit);
-	__be32 *buffer = c->resources.buffer;
 	int generation = c->resources.generation;
 	int rcode, errors = 0;
-	__be32 old_arg;
+	__be32 old_arg, buffer[2];
 	int err;
 
 	buffer[0] = c->last_pcr_value;
diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
new file mode 100644
index 00000000000..86ee16ca365
--- /dev/null
+++ b/sound/firewire/isight.c
@@ -0,0 +1,755 @@
+/*
+ * Apple iSight audio driver
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <asm/byteorder.h>
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/firewire.h>
+#include <linux/firewire-constants.h>
+#include <linux/module.h>
+#include <linux/mod_devicetable.h>
+#include <linux/mutex.h>
+#include <linux/string.h>
+#include <sound/control.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/tlv.h>
+#include "lib.h"
+#include "iso-resources.h"
+#include "packets-buffer.h"
+
+#define OUI_APPLE		0x000a27
+#define MODEL_APPLE_ISIGHT	0x000008
+#define SW_ISIGHT_AUDIO		0x000010
+
+#define REG_AUDIO_ENABLE	0x000
+#define  AUDIO_ENABLE		0x80000000
+#define REG_DEF_AUDIO_GAIN	0x204
+#define REG_GAIN_RAW_START	0x210
+#define REG_GAIN_RAW_END	0x214
+#define REG_GAIN_DB_START	0x218
+#define REG_GAIN_DB_END		0x21c
+#define REG_SAMPLE_RATE_INQUIRY	0x280
+#define REG_ISO_TX_CONFIG	0x300
+#define  SPEED_SHIFT		16
+#define REG_SAMPLE_RATE		0x400
+#define  RATE_48000		0x80000000
+#define REG_GAIN		0x500
+#define REG_MUTE		0x504
+
+#define MAX_FRAMES_PER_PACKET	475
+
+#define QUEUE_LENGTH		20
+
+struct isight {
+	struct snd_card *card;
+	struct fw_unit *unit;
+	struct fw_device *device;
+	u64 audio_base;
+	struct fw_address_handler iris_handler;
+	struct snd_pcm_substream *pcm;
+	struct mutex mutex;
+	struct iso_packets_buffer buffer;
+	struct fw_iso_resources resources;
+	struct fw_iso_context *context;
+	bool pcm_active;
+	bool pcm_running;
+	bool first_packet;
+	int packet_index;
+	u32 total_samples;
+	unsigned int buffer_pointer;
+	unsigned int period_counter;
+	s32 gain_min, gain_max;
+	unsigned int gain_tlv[4];
+};
+
+struct audio_payload {
+	__be32 sample_count;
+	__be32 signature;
+	__be32 sample_total;
+	__be32 reserved;
+	__be16 samples[2 * MAX_FRAMES_PER_PACKET];
+};
+
+MODULE_DESCRIPTION("iSight audio driver");
+MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
+MODULE_LICENSE("GPL v2");
+
+static struct fw_iso_packet audio_packet = {
+	.payload_length = sizeof(struct audio_payload),
+	.interrupt = 1,
+	.header_length = 4,
+};
+
+static void isight_update_pointers(struct isight *isight, unsigned int count)
+{
+	struct snd_pcm_runtime *runtime = isight->pcm->runtime;
+	unsigned int ptr;
+
+	smp_wmb(); /* update buffer data before buffer pointer */
+
+	ptr = isight->buffer_pointer;
+	ptr += count;
+	if (ptr >= runtime->buffer_size)
+		ptr -= runtime->buffer_size;
+	ACCESS_ONCE(isight->buffer_pointer) = ptr;
+
+	isight->period_counter += count;
+	if (isight->period_counter >= runtime->period_size) {
+		isight->period_counter -= runtime->period_size;
+		snd_pcm_period_elapsed(isight->pcm);
+	}
+}
+
+static void isight_samples(struct isight *isight,
+			   const __be16 *samples, unsigned int count)
+{
+	struct snd_pcm_runtime *runtime;
+	unsigned int count1;
+
+	if (!ACCESS_ONCE(isight->pcm_running))
+		return;
+
+	runtime = isight->pcm->runtime;
+	if (isight->buffer_pointer + count <= runtime->buffer_size) {
+		memcpy(runtime->dma_area + isight->buffer_pointer * 4,
+		       samples, count * 4);
+	} else {
+		count1 = runtime->buffer_size - isight->buffer_pointer;
+		memcpy(runtime->dma_area + isight->buffer_pointer * 4,
+		       samples, count1 * 4);
+		samples += count1 * 2;
+		memcpy(runtime->dma_area, samples, (count - count1) * 4);
+	}
+
+	isight_update_pointers(isight, count);
+}
+
+static void isight_pcm_abort(struct isight *isight)
+{
+	unsigned long flags;
+
+	if (ACCESS_ONCE(isight->pcm_active)) {
+		snd_pcm_stream_lock_irqsave(isight->pcm, flags);
+		if (snd_pcm_running(isight->pcm))
+			snd_pcm_stop(isight->pcm, SNDRV_PCM_STATE_XRUN);
+		snd_pcm_stream_unlock_irqrestore(isight->pcm, flags);
+	}
+}
+
+static void isight_dropped_samples(struct isight *isight, unsigned int total)
+{
+	struct snd_pcm_runtime *runtime;
+	u32 dropped;
+	unsigned int count1;
+
+	if (!ACCESS_ONCE(isight->pcm_running))
+		return;
+
+	runtime = isight->pcm->runtime;
+	dropped = total - isight->total_samples;
+	if (dropped < runtime->buffer_size) {
+		if (isight->buffer_pointer + dropped <= runtime->buffer_size) {
+			memset(runtime->dma_area + isight->buffer_pointer * 4,
+			       0, dropped * 4);
+		} else {
+			count1 = runtime->buffer_size - isight->buffer_pointer;
+			memset(runtime->dma_area + isight->buffer_pointer * 4,
+			       0, count1 * 4);
+			memset(runtime->dma_area, 0, (dropped - count1) * 4);
+		}
+		isight_update_pointers(isight, dropped);
+	} else {
+		isight_pcm_abort(isight);
+	}
+}
+
+static void isight_packet(struct fw_iso_context *context, u32 cycle,
+			  size_t header_length, void *header, void *data)
+{
+	struct isight *isight = data;
+	const struct audio_payload *payload;
+	unsigned int index, length, count, total;
+	int err;
+
+	if (isight->packet_index < 0)
+		return;
+	index = isight->packet_index;
+	payload = isight->buffer.packets[index].buffer;
+	length = be32_to_cpup(header) >> 16;
+
+	if (likely(length >= 16 &&
+		   payload->signature == cpu_to_be32(0x73676874/*"sght"*/))) {
+		count = be32_to_cpu(payload->sample_count);
+		if (likely(count <= (length - 16) / 4)) {
+			total = be32_to_cpu(payload->sample_total);
+			if (unlikely(total != isight->total_samples)) {
+				if (!isight->first_packet)
+					isight_dropped_samples(isight, total);
+				isight->first_packet = false;
+				isight->total_samples = total;
+			}
+
+			isight_samples(isight, payload->samples, count);
+			isight->total_samples += count;
+		}
+	}
+
+	err = fw_iso_context_queue(isight->context, &audio_packet,
+				   &isight->buffer.iso_buffer,
+				   isight->buffer.packets[index].offset);
+	if (err < 0) {
+		dev_err(&isight->unit->device, "queueing error: %d\n", err);
+		isight_pcm_abort(isight);
+		isight->packet_index = -1;
+		return;
+	}
+
+	if (++index >= QUEUE_LENGTH)
+		index = 0;
+	isight->packet_index = index;
+}
+
+static int isight_connect(struct isight *isight)
+{
+	int ch, err, rcode, errors = 0;
+	__be32 value;
+
+retry_after_bus_reset:
+	ch = fw_iso_resources_allocate(&isight->resources,
+				       sizeof(struct audio_payload),
+				       isight->device->max_speed);
+	if (ch < 0) {
+		err = ch;
+		goto error;
+	}
+
+	value = cpu_to_be32(ch | (isight->device->max_speed << SPEED_SHIFT));
+	for (;;) {
+		rcode = fw_run_transaction(
+				isight->device->card,
+				TCODE_WRITE_QUADLET_REQUEST,
+				isight->device->node_id,
+				isight->resources.generation,
+				isight->device->max_speed,
+				isight->audio_base + REG_ISO_TX_CONFIG,
+				&value, 4);
+		if (rcode == RCODE_COMPLETE) {
+			return 0;
+		} else if (rcode == RCODE_GENERATION) {
+			fw_iso_resources_free(&isight->resources);
+			goto retry_after_bus_reset;
+		} else if (rcode_is_permanent_error(rcode) || ++errors >= 3) {
+			err = -EIO;
+			goto err_resources;
+		}
+		msleep(5);
+	}
+
+err_resources:
+	fw_iso_resources_free(&isight->resources);
+error:
+	return err;
+}
+
+static int isight_open(struct snd_pcm_substream *substream)
+{
+	static const struct snd_pcm_hardware hardware = {
+		.info = SNDRV_PCM_INFO_MMAP |
+			SNDRV_PCM_INFO_MMAP_VALID |
+			SNDRV_PCM_INFO_BATCH |
+			SNDRV_PCM_INFO_INTERLEAVED |
+			SNDRV_PCM_INFO_BLOCK_TRANSFER,
+		.formats = SNDRV_PCM_FMTBIT_S16_BE,
+		.rates = SNDRV_PCM_RATE_48000,
+		.rate_min = 48000,
+		.rate_max = 48000,
+		.channels_min = 2,
+		.channels_max = 2,
+		.buffer_bytes_max = 4 * 1024 * 1024,
+		.period_bytes_min = MAX_FRAMES_PER_PACKET * 4,
+		.period_bytes_max = 1024 * 1024,
+		.periods_min = 2,
+		.periods_max = UINT_MAX,
+	};
+	struct isight *isight = substream->private_data;
+
+	substream->runtime->hw = hardware;
+
+	return iso_packets_buffer_init(&isight->buffer, isight->unit,
+				       QUEUE_LENGTH,
+				       sizeof(struct audio_payload),
+				       DMA_FROM_DEVICE);
+}
+
+static int isight_close(struct snd_pcm_substream *substream)
+{
+	struct isight *isight = substream->private_data;
+
+	iso_packets_buffer_destroy(&isight->buffer, isight->unit);
+
+	return 0;
+}
+
+static int isight_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *hw_params)
+{
+	struct isight *isight = substream->private_data;
+	int err;
+
+	err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+					       params_buffer_bytes(hw_params));
+	if (err < 0)
+		return err;
+
+	ACCESS_ONCE(isight->pcm_active) = true;
+
+	return 0;
+}
+
+static int reg_read(struct isight *isight, int offset, __be32 *value)
+{
+	return snd_fw_transaction(isight->unit, TCODE_READ_QUADLET_REQUEST,
+				  isight->audio_base + offset, value, 4);
+}
+
+static int reg_write(struct isight *isight, int offset, __be32 value)
+{
+	return snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST,
+				  isight->audio_base + offset, &value, 4);
+}
+
+static void isight_stop_streaming(struct isight *isight)
+{
+	if (!isight->context)
+		return;
+
+	fw_iso_context_stop(isight->context);
+	fw_iso_context_destroy(isight->context);
+	isight->context = NULL;
+	fw_iso_resources_free(&isight->resources);
+	reg_write(isight, REG_AUDIO_ENABLE, 0);
+}
+
+static int isight_hw_free(struct snd_pcm_substream *substream)
+{
+	struct isight *isight = substream->private_data;
+
+	ACCESS_ONCE(isight->pcm_active) = false;
+
+	mutex_lock(&isight->mutex);
+	isight_stop_streaming(isight);
+	mutex_unlock(&isight->mutex);
+
+	return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int isight_start_streaming(struct isight *isight)
+{
+	unsigned int i;
+	int err;
+
+	if (isight->context) {
+		if (isight->packet_index < 0)
+			isight_stop_streaming(isight);
+		else
+			return 0;
+	}
+
+	err = reg_write(isight, REG_SAMPLE_RATE, cpu_to_be32(RATE_48000));
+	if (err < 0)
+		goto error;
+
+	err = isight_connect(isight);
+	if (err < 0)
+		goto error;
+
+	err = reg_write(isight, REG_AUDIO_ENABLE, cpu_to_be32(AUDIO_ENABLE));
+	if (err < 0)
+		goto err_resources;
+
+	isight->context = fw_iso_context_create(isight->device->card,
+						FW_ISO_CONTEXT_RECEIVE,
+						isight->resources.channel,
+						isight->device->max_speed,
+						4, isight_packet, isight);
+	if (IS_ERR(isight->context)) {
+		err = PTR_ERR(isight->context);
+		isight->context = NULL;
+		goto err_resources;
+	}
+
+	for (i = 0; i < QUEUE_LENGTH; ++i) {
+		err = fw_iso_context_queue(isight->context, &audio_packet,
+					   &isight->buffer.iso_buffer,
+					   isight->buffer.packets[i].offset);
+		if (err < 0)
+			goto err_context;
+	}
+
+	isight->first_packet = true;
+	isight->packet_index = 0;
+
+	err = fw_iso_context_start(isight->context, -1, 0,
+				   FW_ISO_CONTEXT_MATCH_ALL_TAGS/*?*/);
+	if (err < 0)
+		goto err_context;
+
+	return 0;
+
+err_context:
+	fw_iso_context_destroy(isight->context);
+	isight->context = NULL;
+err_resources:
+	fw_iso_resources_free(&isight->resources);
+	reg_write(isight, REG_AUDIO_ENABLE, 0);
+error:
+	return err;
+}
+
+static int isight_prepare(struct snd_pcm_substream *substream)
+{
+	struct isight *isight = substream->private_data;
+	int err;
+
+	isight->buffer_pointer = 0;
+	isight->period_counter = 0;
+
+	mutex_lock(&isight->mutex);
+	err = isight_start_streaming(isight);
+	mutex_unlock(&isight->mutex);
+
+	return err;
+}
+
+static int isight_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct isight *isight = substream->private_data;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		ACCESS_ONCE(isight->pcm_running) = true;
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		ACCESS_ONCE(isight->pcm_running) = false;
+		break;
+	default:
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static snd_pcm_uframes_t isight_pointer(struct snd_pcm_substream *substream)
+{
+	struct isight *isight = substream->private_data;
+
+	return ACCESS_ONCE(isight->buffer_pointer);
+}
+
+static int isight_create_pcm(struct isight *isight)
+{
+	static struct snd_pcm_ops ops = {
+		.open      = isight_open,
+		.close     = isight_close,
+		.ioctl     = snd_pcm_lib_ioctl,
+		.hw_params = isight_hw_params,
+		.hw_free   = isight_hw_free,
+		.prepare   = isight_prepare,
+		.trigger   = isight_trigger,
+		.pointer   = isight_pointer,
+		.page      = snd_pcm_lib_get_vmalloc_page,
+		.mmap      = snd_pcm_lib_mmap_vmalloc,
+	};
+	struct snd_pcm *pcm;
+	int err;
+
+	err = snd_pcm_new(isight->card, "iSight", 0, 0, 1, &pcm);
+	if (err < 0)
+		return err;
+	pcm->private_data = isight;
+	strcpy(pcm->name, "iSight");
+	isight->pcm = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+	isight->pcm->ops = &ops;
+
+	return 0;
+}
+
+static int isight_gain_info(struct snd_kcontrol *ctl,
+			    struct snd_ctl_elem_info *info)
+{
+	struct isight *isight = ctl->private_data;
+
+	info->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	info->count = 1;
+	info->value.integer.min = isight->gain_min;
+	info->value.integer.max = isight->gain_max;
+
+	return 0;
+}
+
+static int isight_gain_get(struct snd_kcontrol *ctl,
+			   struct snd_ctl_elem_value *value)
+{
+	struct isight *isight = ctl->private_data;
+	__be32 gain;
+	int err;
+
+	err = reg_read(isight, REG_GAIN, &gain);
+	if (err < 0)
+		return err;
+
+	value->value.integer.value[0] = (s32)be32_to_cpu(gain);
+
+	return 0;
+}
+
+static int isight_gain_put(struct snd_kcontrol *ctl,
+			   struct snd_ctl_elem_value *value)
+{
+	struct isight *isight = ctl->private_data;
+
+	if (value->value.integer.value[0] < isight->gain_min ||
+	    value->value.integer.value[0] > isight->gain_max)
+		return -EINVAL;
+
+	return reg_write(isight, REG_GAIN,
+			 cpu_to_be32(value->value.integer.value[0]));
+}
+
+static int isight_mute_get(struct snd_kcontrol *ctl,
+			   struct snd_ctl_elem_value *value)
+{
+	struct isight *isight = ctl->private_data;
+	__be32 mute;
+	int err;
+
+	err = reg_read(isight, REG_MUTE, &mute);
+	if (err < 0)
+		return err;
+
+	value->value.integer.value[0] = !mute;
+
+	return 0;
+}
+
+static int isight_mute_put(struct snd_kcontrol *ctl,
+			   struct snd_ctl_elem_value *value)
+{
+	struct isight *isight = ctl->private_data;
+
+	return reg_write(isight, REG_MUTE,
+			 (__force __be32)!value->value.integer.value[0]);
+}
+
+static int isight_create_mixer(struct isight *isight)
+{
+	static const struct snd_kcontrol_new gain_control = {
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Mic Capture Volume",
+		.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+			  SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+		.info = isight_gain_info,
+		.get = isight_gain_get,
+		.put = isight_gain_put,
+	};
+	static const struct snd_kcontrol_new mute_control = {
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Mic Capture Switch",
+		.info = snd_ctl_boolean_mono_info,
+		.get = isight_mute_get,
+		.put = isight_mute_put,
+	};
+	__be32 value;
+	struct snd_kcontrol *ctl;
+	int err;
+
+	err = reg_read(isight, REG_GAIN_RAW_START, &value);
+	if (err < 0)
+		return err;
+	isight->gain_min = be32_to_cpu(value);
+
+	err = reg_read(isight, REG_GAIN_RAW_END, &value);
+	if (err < 0)
+		return err;
+	isight->gain_max = be32_to_cpu(value);
+
+	isight->gain_tlv[0] = SNDRV_CTL_TLVT_DB_MINMAX;
+	isight->gain_tlv[1] = 2 * sizeof(unsigned int);
+
+	err = reg_read(isight, REG_GAIN_DB_START, &value);
+	if (err < 0)
+		return err;
+	isight->gain_tlv[2] = (s32)be32_to_cpu(value) * 100;
+
+	err = reg_read(isight, REG_GAIN_DB_END, &value);
+	if (err < 0)
+		return err;
+	isight->gain_tlv[3] = (s32)be32_to_cpu(value) * 100;
+
+	ctl = snd_ctl_new1(&gain_control, isight);
+	if (ctl)
+		ctl->tlv.p = isight->gain_tlv;
+	err = snd_ctl_add(isight->card, ctl);
+	if (err < 0)
+		return err;
+
+	err = snd_ctl_add(isight->card, snd_ctl_new1(&mute_control, isight));
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
+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);
+	fw_device_put(isight->device);
+	mutex_destroy(&isight->mutex);
+}
+
+static u64 get_unit_base(struct fw_unit *unit)
+{
+	struct fw_csr_iterator i;
+	int key, value;
+
+	fw_csr_iterator_init(&i, unit->directory);
+	while (fw_csr_iterator_next(&i, &key, &value))
+		if (key == CSR_OFFSET)
+			return CSR_REGISTER_BASE + value * 4;
+	return 0;
+}
+
+static int isight_probe(struct device *unit_dev)
+{
+	struct fw_unit *unit = fw_unit(unit_dev);
+	struct fw_device *fw_dev = fw_parent_device(unit);
+	struct snd_card *card;
+	struct isight *isight;
+	int err;
+
+	err = snd_card_create(-1, NULL, THIS_MODULE, sizeof(*isight), &card);
+	if (err < 0)
+		return err;
+	snd_card_set_dev(card, unit_dev);
+
+	isight = card->private_data;
+	isight->card = card;
+	mutex_init(&isight->mutex);
+	isight->unit = fw_unit_get(unit);
+	isight->device = fw_device_get(fw_dev);
+	isight->audio_base = get_unit_base(unit);
+	if (!isight->audio_base) {
+		dev_err(&unit->device, "audio unit base not found\n");
+		err = -ENXIO;
+		goto err_unit;
+	}
+	fw_iso_resources_init(&isight->resources, unit);
+
+	card->private_free = isight_card_free;
+
+	strcpy(card->driver, "iSight");
+	strcpy(card->shortname, "Apple iSight");
+	snprintf(card->longname, sizeof(card->longname),
+		 "Apple iSight (GUID %08x%08x) at %s, S%d",
+		 fw_dev->config_rom[3], fw_dev->config_rom[4],
+		 dev_name(&unit->device), 100 << fw_dev->max_speed);
+	strcpy(card->mixername, "iSight");
+
+	err = isight_create_pcm(isight);
+	if (err < 0)
+		goto error;
+
+	err = isight_create_mixer(isight);
+	if (err < 0)
+		goto error;
+
+	err = snd_card_register(card);
+	if (err < 0)
+		goto error;
+
+	dev_set_drvdata(unit_dev, isight);
+
+	return 0;
+
+err_unit:
+	fw_unit_put(isight->unit);
+	fw_device_put(isight->device);
+	mutex_destroy(&isight->mutex);
+error:
+	snd_card_free(card);
+	return err;
+}
+
+static int isight_remove(struct device *dev)
+{
+	struct isight *isight = dev_get_drvdata(dev);
+
+	isight_pcm_abort(isight);
+
+	snd_card_disconnect(isight->card);
+
+	mutex_lock(&isight->mutex);
+	isight_stop_streaming(isight);
+	mutex_unlock(&isight->mutex);
+
+	snd_card_free_when_closed(isight->card);
+
+	return 0;
+}
+
+static void isight_bus_reset(struct fw_unit *unit)
+{
+	struct isight *isight = dev_get_drvdata(&unit->device);
+
+	if (fw_iso_resources_update(&isight->resources) < 0) {
+		isight_pcm_abort(isight);
+
+		mutex_lock(&isight->mutex);
+		isight_stop_streaming(isight);
+		mutex_unlock(&isight->mutex);
+	}
+}
+
+static const struct ieee1394_device_id isight_id_table[] = {
+	{
+		.match_flags  = IEEE1394_MATCH_SPECIFIER_ID |
+				IEEE1394_MATCH_VERSION,
+		.specifier_id = OUI_APPLE,
+		.version      = SW_ISIGHT_AUDIO,
+	},
+	{ }
+};
+MODULE_DEVICE_TABLE(ieee1394, isight_id_table);
+
+static struct fw_driver isight_driver = {
+	.driver   = {
+		.owner	= THIS_MODULE,
+		.name	= KBUILD_MODNAME,
+		.bus	= &fw_bus_type,
+		.probe	= isight_probe,
+		.remove	= isight_remove,
+	},
+	.update   = isight_bus_reset,
+	.id_table = isight_id_table,
+};
+
+static int __init alsa_isight_init(void)
+{
+	return driver_register(&isight_driver.driver);
+}
+
+static void __exit alsa_isight_exit(void)
+{
+	driver_unregister(&isight_driver.driver);
+}
+
+module_init(alsa_isight_init);
+module_exit(alsa_isight_exit);
diff --git a/sound/firewire/iso-resources.c b/sound/firewire/iso-resources.c
index 775dbd5f344..ffe20b877e9 100644
--- a/sound/firewire/iso-resources.c
+++ b/sound/firewire/iso-resources.c
@@ -11,7 +11,6 @@
 #include <linux/jiffies.h>
 #include <linux/mutex.h>
 #include <linux/sched.h>
-#include <linux/slab.h>
 #include <linux/spinlock.h>
 #include "iso-resources.h"
 
@@ -25,10 +24,6 @@
  */
 int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit)
 {
-	r->buffer = kmalloc(2 * 4, GFP_KERNEL);
-	if (!r->buffer)
-		return -ENOMEM;
-
 	r->channels_mask = ~0uLL;
 	r->unit = fw_unit_get(unit);
 	mutex_init(&r->mutex);
@@ -36,6 +31,7 @@ int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit)
 
 	return 0;
 }
+EXPORT_SYMBOL(fw_iso_resources_init);
 
 /**
  * fw_iso_resources_destroy - destroy a resource manager
@@ -44,10 +40,10 @@ int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit)
 void fw_iso_resources_destroy(struct fw_iso_resources *r)
 {
 	WARN_ON(r->allocated);
-	kfree(r->buffer);
 	mutex_destroy(&r->mutex);
 	fw_unit_put(r->unit);
 }
+EXPORT_SYMBOL(fw_iso_resources_destroy);
 
 static unsigned int packet_bandwidth(unsigned int max_payload_bytes, int speed)
 {
@@ -131,7 +127,7 @@ retry_after_bus_reset:
 
 	bandwidth = r->bandwidth + r->bandwidth_overhead;
 	fw_iso_resource_manage(card, r->generation, r->channels_mask,
-			       &channel, &bandwidth, true, r->buffer);
+			       &channel, &bandwidth, true);
 	if (channel == -EAGAIN) {
 		mutex_unlock(&r->mutex);
 		goto retry_after_bus_reset;
@@ -152,6 +148,7 @@ retry_after_bus_reset:
 
 	return channel;
 }
+EXPORT_SYMBOL(fw_iso_resources_allocate);
 
 /**
  * fw_iso_resources_update - update resource allocations after a bus reset
@@ -184,7 +181,7 @@ int fw_iso_resources_update(struct fw_iso_resources *r)
 	bandwidth = r->bandwidth + r->bandwidth_overhead;
 
 	fw_iso_resource_manage(card, r->generation, 1uLL << r->channel,
-			       &channel, &bandwidth, true, r->buffer);
+			       &channel, &bandwidth, true);
 	/*
 	 * When another bus reset happens, pretend that the allocation
 	 * succeeded; we will try again for the new generation later.
@@ -203,6 +200,7 @@ int fw_iso_resources_update(struct fw_iso_resources *r)
 
 	return channel;
 }
+EXPORT_SYMBOL(fw_iso_resources_update);
 
 /**
  * fw_iso_resources_free - frees allocated resources
@@ -220,7 +218,7 @@ void fw_iso_resources_free(struct fw_iso_resources *r)
 	if (r->allocated) {
 		bandwidth = r->bandwidth + r->bandwidth_overhead;
 		fw_iso_resource_manage(card, r->generation, 1uLL << r->channel,
-				       &channel, &bandwidth, false, r->buffer);
+				       &channel, &bandwidth, false);
 		if (channel < 0)
 			dev_err(&r->unit->device,
 				"isochronous resource deallocation failed\n");
@@ -230,3 +228,4 @@ void fw_iso_resources_free(struct fw_iso_resources *r)
 
 	mutex_unlock(&r->mutex);
 }
+EXPORT_SYMBOL(fw_iso_resources_free);
diff --git a/sound/firewire/iso-resources.h b/sound/firewire/iso-resources.h
index 3f0730e4d84..5a9af7c6165 100644
--- a/sound/firewire/iso-resources.h
+++ b/sound/firewire/iso-resources.h
@@ -24,7 +24,6 @@ struct fw_iso_resources {
 	unsigned int bandwidth_overhead;
 	int generation; /* in which allocation is valid */
 	bool allocated;
-	__be32 *buffer;
 };
 
 int fw_iso_resources_init(struct fw_iso_resources *r,
diff --git a/sound/firewire/packets-buffer.c b/sound/firewire/packets-buffer.c
index 1e20e60ba6a..3c61ca2e615 100644
--- a/sound/firewire/packets-buffer.c
+++ b/sound/firewire/packets-buffer.c
@@ -60,6 +60,7 @@ err_packets:
 error:
 	return err;
 }
+EXPORT_SYMBOL(iso_packets_buffer_init);
 
 /**
  * iso_packets_buffer_destroy - frees packet buffer resources
@@ -72,3 +73,4 @@ void iso_packets_buffer_destroy(struct iso_packets_buffer *b,
 	fw_iso_buffer_destroy(&b->iso_buffer, fw_parent_device(unit)->card);
 	kfree(b->packets);
 }
+EXPORT_SYMBOL(iso_packets_buffer_destroy);
diff --git a/sound/i2c/other/Makefile b/sound/i2c/other/Makefile
index 2dad40f3f62..c95d8f1aae8 100644
--- a/sound/i2c/other/Makefile
+++ b/sound/i2c/other/Makefile
@@ -14,4 +14,4 @@ snd-tea575x-tuner-objs := tea575x-tuner.o
 obj-$(CONFIG_SND_PDAUDIOCF) += snd-ak4117.o
 obj-$(CONFIG_SND_ICE1712) += snd-ak4xxx-adda.o
 obj-$(CONFIG_SND_ICE1724) += snd-ak4114.o snd-ak4113.o snd-ak4xxx-adda.o snd-pt2258.o
-obj-$(CONFIG_SND_FM801_TEA575X) += snd-tea575x-tuner.o
+obj-$(CONFIG_SND_TEA575X) += snd-tea575x-tuner.o
diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c
index ee538f1ae84..4831800239d 100644
--- a/sound/i2c/other/tea575x-tuner.c
+++ b/sound/i2c/other/tea575x-tuner.c
@@ -37,8 +37,8 @@ static int radio_nr = -1;
 module_param(radio_nr, int, 0);
 
 #define RADIO_VERSION KERNEL_VERSION(0, 0, 2)
-#define FREQ_LO		 (87 * 16000)
-#define FREQ_HI		(108 * 16000)
+#define FREQ_LO		 (50UL * 16000)
+#define FREQ_HI		(150UL * 16000)
 
 /*
  * definitions
@@ -77,27 +77,95 @@ static struct v4l2_queryctrl radio_qctrl[] = {
  * lowlevel part
  */
 
+static void snd_tea575x_write(struct snd_tea575x *tea, unsigned int val)
+{
+	u16 l;
+	u8 data;
+
+	tea->ops->set_direction(tea, 1);
+	udelay(16);
+
+	for (l = 25; l > 0; l--) {
+		data = (val >> 24) & TEA575X_DATA;
+		val <<= 1;			/* shift data */
+		tea->ops->set_pins(tea, data | TEA575X_WREN);
+		udelay(2);
+		tea->ops->set_pins(tea, data | TEA575X_WREN | TEA575X_CLK);
+		udelay(2);
+		tea->ops->set_pins(tea, data | TEA575X_WREN);
+		udelay(2);
+	}
+
+	if (!tea->mute)
+		tea->ops->set_pins(tea, 0);
+}
+
+static unsigned int snd_tea575x_read(struct snd_tea575x *tea)
+{
+	u16 l, rdata;
+	u32 data = 0;
+
+	tea->ops->set_direction(tea, 0);
+	tea->ops->set_pins(tea, 0);
+	udelay(16);
+
+	for (l = 24; l--;) {
+		tea->ops->set_pins(tea, TEA575X_CLK);
+		udelay(2);
+		if (!l)
+			tea->tuned = tea->ops->get_pins(tea) & TEA575X_MOST ? 0 : 1;
+		tea->ops->set_pins(tea, 0);
+		udelay(2);
+		data <<= 1;			/* shift data */
+		rdata = tea->ops->get_pins(tea);
+		if (!l)
+			tea->stereo = (rdata & TEA575X_MOST) ?  0 : 1;
+		if (rdata & TEA575X_DATA)
+			data++;
+		udelay(2);
+	}
+
+	if (tea->mute)
+		tea->ops->set_pins(tea, TEA575X_WREN);
+
+	return data;
+}
+
+static void snd_tea575x_get_freq(struct snd_tea575x *tea)
+{
+	unsigned long freq;
+
+	freq = snd_tea575x_read(tea) & TEA575X_BIT_FREQ_MASK;
+	/* freq *= 12.5 */
+	freq *= 125;
+	freq /= 10;
+	/* crystal fixup */
+	if (tea->tea5759)
+		freq += TEA575X_FMIF;
+	else
+		freq -= TEA575X_FMIF;
+
+	tea->freq = freq * 16;		/* from kHz */
+}
+
 static void snd_tea575x_set_freq(struct snd_tea575x *tea)
 {
 	unsigned long freq;
 
-	freq = tea->freq / 16;		/* to kHz */
-	if (freq > 108000)
-		freq = 108000;
-	if (freq < 87000)
-		freq = 87000;
+	freq = clamp(tea->freq, FREQ_LO, FREQ_HI);
+	freq /= 16;		/* to kHz */
 	/* crystal fixup */
 	if (tea->tea5759)
-		freq -= tea->freq_fixup;
+		freq -= TEA575X_FMIF;
 	else
-		freq += tea->freq_fixup;
+		freq += TEA575X_FMIF;
 	/* freq /= 12.5 */
 	freq *= 10;
 	freq /= 125;
 
 	tea->val &= ~TEA575X_BIT_FREQ_MASK;
 	tea->val |= freq & TEA575X_BIT_FREQ_MASK;
-	tea->ops->write(tea, tea->val);
+	snd_tea575x_write(tea, tea->val);
 }
 
 /*
@@ -109,29 +177,34 @@ static int vidioc_querycap(struct file *file, void  *priv,
 {
 	struct snd_tea575x *tea = video_drvdata(file);
 
-	strcpy(v->card, tea->tea5759 ? "TEA5759" : "TEA5757");
 	strlcpy(v->driver, "tea575x-tuner", sizeof(v->driver));
-	strlcpy(v->card, "Maestro Radio", sizeof(v->card));
-	sprintf(v->bus_info, "PCI");
+	strlcpy(v->card, tea->card, sizeof(v->card));
+	strlcat(v->card, tea->tea5759 ? " TEA5759" : " TEA5757", sizeof(v->card));
+	strlcpy(v->bus_info, tea->bus_info, sizeof(v->bus_info));
 	v->version = RADIO_VERSION;
-	v->capabilities = V4L2_CAP_TUNER;
+	v->capabilities = V4L2_CAP_TUNER | V4L2_CAP_RADIO;
 	return 0;
 }
 
 static int vidioc_g_tuner(struct file *file, void *priv,
 					struct v4l2_tuner *v)
 {
+	struct snd_tea575x *tea = video_drvdata(file);
+
 	if (v->index > 0)
 		return -EINVAL;
 
+	snd_tea575x_read(tea);
+
 	strcpy(v->name, "FM");
 	v->type = V4L2_TUNER_RADIO;
+	v->capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO;
 	v->rangelow = FREQ_LO;
 	v->rangehigh = FREQ_HI;
-	v->rxsubchans = V4L2_TUNER_SUB_MONO|V4L2_TUNER_SUB_STEREO;
-	v->capability = V4L2_TUNER_CAP_LOW;
-	v->audmode = V4L2_TUNER_MODE_MONO;
-	v->signal = 0xffff;
+	v->rxsubchans = V4L2_TUNER_SUB_MONO | V4L2_TUNER_SUB_STEREO;
+	v->audmode = tea->stereo ? V4L2_TUNER_MODE_STEREO : V4L2_TUNER_MODE_MONO;
+	v->signal = tea->tuned ? 0xffff : 0;
+
 	return 0;
 }
 
@@ -148,7 +221,10 @@ static int vidioc_g_frequency(struct file *file, void *priv,
 {
 	struct snd_tea575x *tea = video_drvdata(file);
 
+	if (f->tuner != 0)
+		return -EINVAL;
 	f->type = V4L2_TUNER_RADIO;
+	snd_tea575x_get_freq(tea);
 	f->frequency = tea->freq;
 	return 0;
 }
@@ -158,6 +234,9 @@ static int vidioc_s_frequency(struct file *file, void *priv,
 {
 	struct snd_tea575x *tea = video_drvdata(file);
 
+	if (f->tuner != 0 || f->type != V4L2_TUNER_RADIO)
+		return -EINVAL;
+
 	if (f->frequency < FREQ_LO || f->frequency > FREQ_HI)
 		return -EINVAL;
 
@@ -209,10 +288,8 @@ static int vidioc_g_ctrl(struct file *file, void *priv,
 
 	switch (ctrl->id) {
 	case V4L2_CID_AUDIO_MUTE:
-		if (tea->ops->mute) {
-			ctrl->value = tea->mute;
-			return 0;
-		}
+		ctrl->value = tea->mute;
+		return 0;
 	}
 	return -EINVAL;
 }
@@ -224,11 +301,11 @@ static int vidioc_s_ctrl(struct file *file, void *priv,
 
 	switch (ctrl->id) {
 	case V4L2_CID_AUDIO_MUTE:
-		if (tea->ops->mute) {
-			tea->ops->mute(tea, ctrl->value);
+		if (tea->mute != ctrl->value) {
 			tea->mute = ctrl->value;
-			return 0;
+			snd_tea575x_set_freq(tea);
 		}
+		return 0;
 	}
 	return -EINVAL;
 }
@@ -293,18 +370,16 @@ static struct video_device tea575x_radio = {
 /*
  * initialize all the tea575x chips
  */
-void snd_tea575x_init(struct snd_tea575x *tea)
+int snd_tea575x_init(struct snd_tea575x *tea)
 {
 	int retval;
-	unsigned int val;
 	struct video_device *tea575x_radio_inst;
 
-	val = tea->ops->read(tea);
-	if (val == 0x1ffffff || val == 0) {
-		snd_printk(KERN_ERR
-			   "tea575x-tuner: Cannot find TEA575x chip\n");
-		return;
-	}
+	tea->mute = 1;
+
+	snd_tea575x_write(tea, 0x55AA);
+	if (snd_tea575x_read(tea) != 0x55AA)
+		return -ENODEV;
 
 	tea->in_use = 0;
 	tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40;
@@ -313,7 +388,7 @@ void snd_tea575x_init(struct snd_tea575x *tea)
 	tea575x_radio_inst = video_device_alloc();
 	if (tea575x_radio_inst == NULL) {
 		printk(KERN_ERR "tea575x-tuner: not enough memory\n");
-		return;
+		return -ENOMEM;
 	}
 
 	memcpy(tea575x_radio_inst, &tea575x_radio, sizeof(tea575x_radio));
@@ -328,17 +403,13 @@ void snd_tea575x_init(struct snd_tea575x *tea)
 	if (retval) {
 		printk(KERN_ERR "tea575x-tuner: can't register video device!\n");
 		kfree(tea575x_radio_inst);
-		return;
+		return retval;
 	}
 
 	snd_tea575x_set_freq(tea);
-
-	/* mute on init */
-	if (tea->ops->mute) {
-		tea->ops->mute(tea, 1);
-		tea->mute = 1;
-	}
 	tea->vd = tea575x_radio_inst;
+
+	return 0;
 }
 
 void snd_tea575x_exit(struct snd_tea575x *tea)
diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig
index 76c09021807..6c93e051f9a 100644
--- a/sound/oss/Kconfig
+++ b/sound/oss/Kconfig
@@ -22,10 +22,6 @@ config SOUND_VWSND
 	  <file:Documentation/sound/oss/vwsnd> for more info on this driver's
 	  capabilities.
 
-config SOUND_AU1550_AC97
-	tristate "Au1550/Au1200 AC97 Sound"
-	depends on SOC_AU1550 || SOC_AU1200
-
 config SOUND_MSNDCLAS
 	tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey"
 	depends on (m || !STANDALONE) && ISA
diff --git a/sound/oss/Makefile b/sound/oss/Makefile
index 90ffb99c6b1..77f21b68bf0 100644
--- a/sound/oss/Makefile
+++ b/sound/oss/Makefile
@@ -25,7 +25,6 @@ obj-$(CONFIG_SOUND_WAVEARTIST)	+= waveartist.o
 obj-$(CONFIG_SOUND_MSNDCLAS)	+= msnd.o msnd_classic.o
 obj-$(CONFIG_SOUND_MSNDPIN)	+= msnd.o msnd_pinnacle.o
 obj-$(CONFIG_SOUND_VWSND)	+= vwsnd.o
-obj-$(CONFIG_SOUND_AU1550_AC97)	+= au1550_ac97.o ac97_codec.o
 obj-$(CONFIG_SOUND_BCM_CS4297A)	+= swarm_cs4297a.o
 
 obj-$(CONFIG_DMASOUND)		+= dmasound/
diff --git a/sound/oss/ac97_codec.c b/sound/oss/ac97_codec.c
deleted file mode 100644
index 0cd23d94888..00000000000
--- a/sound/oss/ac97_codec.c
+++ /dev/null
@@ -1,1203 +0,0 @@
-/*
- * ac97_codec.c: Generic AC97 mixer/modem module
- *
- * Derived from ac97 mixer in maestro and trident driver.
- *
- * Copyright 2000 Silicon Integrated System Corporation
- *
- *	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., 675 Mass Ave, Cambridge, MA 02139, USA.
- *
- **************************************************************************
- *
- * The Intel Audio Codec '97 specification is available at:
- * http://download.intel.com/support/motherboards/desktop/sb/ac97_r23.pdf
- *
- **************************************************************************
- *
- * History
- * May 02, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
- *	Removed non existent WM9700
- *	Added support for WM9705, WM9708, WM9709, WM9710, WM9711
- *	WM9712 and WM9717
- * Mar 28, 2002 Randolph Bentson <bentson@holmsjoen.com>
- *	corrections to support WM9707 in ViewPad 1000
- * v0.4 Mar 15 2000 Ollie Lho
- *	dual codecs support verified with 4 channels output
- * v0.3 Feb 22 2000 Ollie Lho
- *	bug fix for record mask setting
- * v0.2 Feb 10 2000 Ollie Lho
- *	add ac97_read_proc for /proc/driver/{vendor}/ac97
- * v0.1 Jan 14 2000 Ollie Lho <ollie@sis.com.tw> 
- *	Isolated from trident.c to support multiple ac97 codec
- */
-#include <linux/module.h>
-#include <linux/kernel.h>
-#include <linux/slab.h>
-#include <linux/string.h>
-#include <linux/errno.h>
-#include <linux/bitops.h>
-#include <linux/delay.h>
-#include <linux/pci.h>
-#include <linux/ac97_codec.h>
-#include <asm/uaccess.h>
-#include <linux/mutex.h>
-
-#define CODEC_ID_BUFSZ 14
-
-static int ac97_read_mixer(struct ac97_codec *codec, int oss_channel);
-static void ac97_write_mixer(struct ac97_codec *codec, int oss_channel, 
-			     unsigned int left, unsigned int right);
-static void ac97_set_mixer(struct ac97_codec *codec, unsigned int oss_mixer, unsigned int val );
-static int ac97_recmask_io(struct ac97_codec *codec, int rw, int mask);
-static int ac97_mixer_ioctl(struct ac97_codec *codec, unsigned int cmd, unsigned long arg);
-
-static int ac97_init_mixer(struct ac97_codec *codec);
-
-static int wolfson_init03(struct ac97_codec * codec);
-static int wolfson_init04(struct ac97_codec * codec);
-static int wolfson_init05(struct ac97_codec * codec);
-static int wolfson_init11(struct ac97_codec * codec);
-static int wolfson_init13(struct ac97_codec * codec);
-static int tritech_init(struct ac97_codec * codec);
-static int tritech_maestro_init(struct ac97_codec * codec);
-static int sigmatel_9708_init(struct ac97_codec *codec);
-static int sigmatel_9721_init(struct ac97_codec *codec);
-static int sigmatel_9744_init(struct ac97_codec *codec);
-static int ad1886_init(struct ac97_codec *codec);
-static int eapd_control(struct ac97_codec *codec, int);
-static int crystal_digital_control(struct ac97_codec *codec, int slots, int rate, int mode);
-static int cmedia_init(struct ac97_codec * codec);
-static int cmedia_digital_control(struct ac97_codec *codec, int slots, int rate, int mode);
-static int generic_digital_control(struct ac97_codec *codec, int slots, int rate, int mode);
-
-
-/*
- *	AC97 operations.
- *
- *	If you are adding a codec then you should be able to use
- *		eapd_ops - any codec that supports EAPD amp control (most)
- *		null_ops - any ancient codec that supports nothing
- *
- *	The three functions are
- *		init - used for non AC97 standard initialisation
- *		amplifier - used to do amplifier control (1=on 0=off)
- *		digital - switch to digital modes (0 = analog)
- *
- *	Not all codecs support all features, not all drivers use all the
- *	operations yet
- */
- 
-static struct ac97_ops null_ops = { NULL, NULL, NULL };
-static struct ac97_ops default_ops = { NULL, eapd_control, NULL };
-static struct ac97_ops default_digital_ops = { NULL, eapd_control, generic_digital_control};
-static struct ac97_ops wolfson_ops03 = { wolfson_init03, NULL, NULL };
-static struct ac97_ops wolfson_ops04 = { wolfson_init04, NULL, NULL };
-static struct ac97_ops wolfson_ops05 = { wolfson_init05, NULL, NULL };
-static struct ac97_ops wolfson_ops11 = { wolfson_init11, NULL, NULL };
-static struct ac97_ops wolfson_ops13 = { wolfson_init13, NULL, NULL };
-static struct ac97_ops tritech_ops = { tritech_init, NULL, NULL };
-static struct ac97_ops tritech_m_ops = { tritech_maestro_init, NULL, NULL };
-static struct ac97_ops sigmatel_9708_ops = { sigmatel_9708_init, NULL, NULL };
-static struct ac97_ops sigmatel_9721_ops = { sigmatel_9721_init, NULL, NULL };
-static struct ac97_ops sigmatel_9744_ops = { sigmatel_9744_init, NULL, NULL };
-static struct ac97_ops crystal_digital_ops = { NULL, eapd_control, crystal_digital_control };
-static struct ac97_ops ad1886_ops = { ad1886_init, eapd_control, NULL };
-static struct ac97_ops cmedia_ops = { NULL, eapd_control, NULL};
-static struct ac97_ops cmedia_digital_ops = { cmedia_init, eapd_control, cmedia_digital_control};
-
-/* sorted by vendor/device id */
-static const struct {
-	u32 id;
-	char *name;
-	struct ac97_ops *ops;
-	int flags;
-} ac97_codec_ids[] = {
-	{0x41445303, "Analog Devices AD1819",	&null_ops},
-	{0x41445340, "Analog Devices AD1881",	&null_ops},
-	{0x41445348, "Analog Devices AD1881A",	&null_ops},
-	{0x41445360, "Analog Devices AD1885",	&default_ops},
-	{0x41445361, "Analog Devices AD1886",	&ad1886_ops},
-	{0x41445370, "Analog Devices AD1981",	&null_ops},
-	{0x41445372, "Analog Devices AD1981A",	&null_ops},
-	{0x41445374, "Analog Devices AD1981B",	&null_ops},
-	{0x41445460, "Analog Devices AD1885",	&default_ops},
-	{0x41445461, "Analog Devices AD1886",	&ad1886_ops},
-	{0x414B4D00, "Asahi Kasei AK4540",	&null_ops},
-	{0x414B4D01, "Asahi Kasei AK4542",	&null_ops},
-	{0x414B4D02, "Asahi Kasei AK4543",	&null_ops},
-	{0x414C4326, "ALC100P",			&null_ops},
-	{0x414C4710, "ALC200/200P",		&null_ops},
-	{0x414C4720, "ALC650",			&default_digital_ops},
-	{0x434D4941, "CMedia",			&cmedia_ops,		AC97_NO_PCM_VOLUME },
-	{0x434D4942, "CMedia",			&cmedia_ops,		AC97_NO_PCM_VOLUME },
-	{0x434D4961, "CMedia",			&cmedia_digital_ops,	AC97_NO_PCM_VOLUME },
-	{0x43525900, "Cirrus Logic CS4297",	&default_ops},
-	{0x43525903, "Cirrus Logic CS4297",	&default_ops},
-	{0x43525913, "Cirrus Logic CS4297A rev A", &default_ops},
-	{0x43525914, "Cirrus Logic CS4297A rev B", &default_ops},
-	{0x43525923, "Cirrus Logic CS4298",	&null_ops},
-	{0x4352592B, "Cirrus Logic CS4294",	&null_ops},
-	{0x4352592D, "Cirrus Logic CS4294",	&null_ops},
-	{0x43525931, "Cirrus Logic CS4299 rev A", &crystal_digital_ops},
-	{0x43525933, "Cirrus Logic CS4299 rev C", &crystal_digital_ops},
-	{0x43525934, "Cirrus Logic CS4299 rev D", &crystal_digital_ops},
-	{0x43585430, "CXT48",			&default_ops,		AC97_DELUDED_MODEM },
-	{0x43585442, "CXT66",			&default_ops,		AC97_DELUDED_MODEM },
-	{0x44543031, "Diamond Technology DT0893", &default_ops},
-	{0x45838308, "ESS Allegro ES1988",	&null_ops},
-	{0x49434511, "ICE1232",			&null_ops}, /* I hope --jk */
-	{0x4e534331, "National Semiconductor LM4549", &null_ops},
-	{0x53494c22, "Silicon Laboratory Si3036", &null_ops},
-	{0x53494c23, "Silicon Laboratory Si3038", &null_ops},
-	{0x545200FF, "TriTech TR?????",		&tritech_m_ops},
-	{0x54524102, "TriTech TR28022",		&null_ops},
-	{0x54524103, "TriTech TR28023",		&null_ops},
-	{0x54524106, "TriTech TR28026",		&null_ops},
-	{0x54524108, "TriTech TR28028",		&tritech_ops},
-	{0x54524123, "TriTech TR A5",		&null_ops},
-	{0x574D4C03, "Wolfson WM9703/07/08/17",	&wolfson_ops03},
-	{0x574D4C04, "Wolfson WM9704M/WM9704Q",	&wolfson_ops04},
-	{0x574D4C05, "Wolfson WM9705/WM9710",   &wolfson_ops05},
-	{0x574D4C09, "Wolfson WM9709",		&null_ops},
-	{0x574D4C12, "Wolfson WM9711/9712",	&wolfson_ops11},
-	{0x574D4C13, "Wolfson WM9713",	&wolfson_ops13, AC97_DEFAULT_POWER_OFF},
-	{0x83847600, "SigmaTel STAC????",	&null_ops},
-	{0x83847604, "SigmaTel STAC9701/3/4/5", &null_ops},
-	{0x83847605, "SigmaTel STAC9704",	&null_ops},
-	{0x83847608, "SigmaTel STAC9708",	&sigmatel_9708_ops},
-	{0x83847609, "SigmaTel STAC9721/23",	&sigmatel_9721_ops},
-	{0x83847644, "SigmaTel STAC9744/45",	&sigmatel_9744_ops},
-	{0x83847652, "SigmaTel STAC9752/53",	&default_ops},
-	{0x83847656, "SigmaTel STAC9756/57",	&sigmatel_9744_ops},
-	{0x83847666, "SigmaTel STAC9750T",	&sigmatel_9744_ops},
-	{0x83847684, "SigmaTel STAC9783/84?",	&null_ops},
-	{0x57454301, "Winbond 83971D",		&null_ops},
-};
-
-/* this table has default mixer values for all OSS mixers. */
-static struct mixer_defaults {
-	int mixer;
-	unsigned int value;
-} mixer_defaults[SOUND_MIXER_NRDEVICES] = {
-	/* all values 0 -> 100 in bytes */
-	{SOUND_MIXER_VOLUME,	0x4343},
-	{SOUND_MIXER_BASS,	0x4343},
-	{SOUND_MIXER_TREBLE,	0x4343},
-	{SOUND_MIXER_PCM,	0x4343},
-	{SOUND_MIXER_SPEAKER,	0x4343},
-	{SOUND_MIXER_LINE,	0x4343},
-	{SOUND_MIXER_MIC,	0x0000},
-	{SOUND_MIXER_CD,	0x4343},
-	{SOUND_MIXER_ALTPCM,	0x4343},
-	{SOUND_MIXER_IGAIN,	0x4343},
-	{SOUND_MIXER_LINE1,	0x4343},
-	{SOUND_MIXER_PHONEIN,	0x4343},
-	{SOUND_MIXER_PHONEOUT,	0x4343},
-	{SOUND_MIXER_VIDEO,	0x4343},
-	{-1,0}
-};
-
-/* table to scale scale from OSS mixer value to AC97 mixer register value */	
-static struct ac97_mixer_hw {
-	unsigned char offset;
-	int scale;
-} ac97_hw[SOUND_MIXER_NRDEVICES]= {
-	[SOUND_MIXER_VOLUME]	=	{AC97_MASTER_VOL_STEREO,64},
-	[SOUND_MIXER_BASS]	=	{AC97_MASTER_TONE,	16},
-	[SOUND_MIXER_TREBLE]	=	{AC97_MASTER_TONE,	16},
-	[SOUND_MIXER_PCM]	=	{AC97_PCMOUT_VOL,	32},
-	[SOUND_MIXER_SPEAKER]	=	{AC97_PCBEEP_VOL,	16},
-	[SOUND_MIXER_LINE]	=	{AC97_LINEIN_VOL,	32},
-	[SOUND_MIXER_MIC]	=	{AC97_MIC_VOL,		32},
-	[SOUND_MIXER_CD]	=	{AC97_CD_VOL,		32},
-	[SOUND_MIXER_ALTPCM]	=	{AC97_HEADPHONE_VOL,	64},
-	[SOUND_MIXER_IGAIN]	=	{AC97_RECORD_GAIN,	16},
-	[SOUND_MIXER_LINE1]	=	{AC97_AUX_VOL,		32},
-	[SOUND_MIXER_PHONEIN]	= 	{AC97_PHONE_VOL,	32},
-	[SOUND_MIXER_PHONEOUT]	= 	{AC97_MASTER_VOL_MONO,	64},
-	[SOUND_MIXER_VIDEO]	=	{AC97_VIDEO_VOL,	32},
-};
-
-/* the following tables allow us to go from OSS <-> ac97 quickly. */
-enum ac97_recsettings {
-	AC97_REC_MIC=0,
-	AC97_REC_CD,
-	AC97_REC_VIDEO,
-	AC97_REC_AUX,
-	AC97_REC_LINE,
-	AC97_REC_STEREO, /* combination of all enabled outputs..  */
-	AC97_REC_MONO,	      /*.. or the mono equivalent */
-	AC97_REC_PHONE
-};
-
-static const unsigned int ac97_rm2oss[] = {
-	[AC97_REC_MIC] 	 = SOUND_MIXER_MIC,
-	[AC97_REC_CD] 	 = SOUND_MIXER_CD,
-	[AC97_REC_VIDEO] = SOUND_MIXER_VIDEO,
-	[AC97_REC_AUX] 	 = SOUND_MIXER_LINE1,
-	[AC97_REC_LINE]  = SOUND_MIXER_LINE,
-	[AC97_REC_STEREO]= SOUND_MIXER_IGAIN,
-	[AC97_REC_PHONE] = SOUND_MIXER_PHONEIN
-};
-
-/* indexed by bit position */
-static const unsigned int ac97_oss_rm[] = {
-	[SOUND_MIXER_MIC] 	= AC97_REC_MIC,
-	[SOUND_MIXER_CD] 	= AC97_REC_CD,
-	[SOUND_MIXER_VIDEO] 	= AC97_REC_VIDEO,
-	[SOUND_MIXER_LINE1] 	= AC97_REC_AUX,
-	[SOUND_MIXER_LINE] 	= AC97_REC_LINE,
-	[SOUND_MIXER_IGAIN]	= AC97_REC_STEREO,
-	[SOUND_MIXER_PHONEIN] 	= AC97_REC_PHONE
-};
-
-static LIST_HEAD(codecs);
-static LIST_HEAD(codec_drivers);
-static DEFINE_MUTEX(codec_mutex);
-
-/* reads the given OSS mixer from the ac97 the caller must have insured that the ac97 knows
-   about that given mixer, and should be holding a spinlock for the card */
-static int ac97_read_mixer(struct ac97_codec *codec, int oss_channel) 
-{
-	u16 val;
-	int ret = 0;
-	int scale;
-	struct ac97_mixer_hw *mh = &ac97_hw[oss_channel];
-
-	val = codec->codec_read(codec , mh->offset);
-
-	if (val & AC97_MUTE) {
-		ret = 0;
-	} else if (AC97_STEREO_MASK & (1 << oss_channel)) {
-		/* nice stereo mixers .. */
-		int left,right;
-
-		left = (val >> 8)  & 0x7f;
-		right = val  & 0x7f;
-
-		if (oss_channel == SOUND_MIXER_IGAIN) {
-			right = (right * 100) / mh->scale;
-			left = (left * 100) / mh->scale;
-		} else {
-			/* these may have 5 or 6 bit resolution */
-			if(oss_channel == SOUND_MIXER_VOLUME || oss_channel == SOUND_MIXER_ALTPCM)
-				scale = (1 << codec->bit_resolution);
-			else
-				scale = mh->scale;
-
-			right = 100 - ((right * 100) / scale);
-			left = 100 - ((left * 100) / scale);
-		}
-		ret = left | (right << 8);
-	} else if (oss_channel == SOUND_MIXER_SPEAKER) {
-		ret = 100 - ((((val & 0x1e)>>1) * 100) / mh->scale);
-	} else if (oss_channel == SOUND_MIXER_PHONEIN) {
-		ret = 100 - (((val & 0x1f) * 100) / mh->scale);
-	} else if (oss_channel == SOUND_MIXER_PHONEOUT) {
-		scale = (1 << codec->bit_resolution);
-		ret = 100 - (((val & 0x1f) * 100) / scale);
-	} else if (oss_channel == SOUND_MIXER_MIC) {
-		ret = 100 - (((val & 0x1f) * 100) / mh->scale);
-		/*  the low bit is optional in the tone sliders and masking
-		    it lets us avoid the 0xf 'bypass'.. */
-	} else if (oss_channel == SOUND_MIXER_BASS) {
-		ret = 100 - ((((val >> 8) & 0xe) * 100) / mh->scale);
-	} else if (oss_channel == SOUND_MIXER_TREBLE) {
-		ret = 100 - (((val & 0xe) * 100) / mh->scale);
-	}
-
-#ifdef DEBUG
-	printk("ac97_codec: read OSS mixer %2d (%s ac97 register 0x%02x), "
-	       "0x%04x -> 0x%04x\n",
-	       oss_channel, codec->id ? "Secondary" : "Primary",
-	       mh->offset, val, ret);
-#endif
-
-	return ret;
-}
-
-/* write the OSS encoded volume to the given OSS encoded mixer, again caller's job to
-   make sure all is well in arg land, call with spinlock held */
-static void ac97_write_mixer(struct ac97_codec *codec, int oss_channel,
-		      unsigned int left, unsigned int right)
-{
-	u16 val = 0;
-	int scale;
-	struct ac97_mixer_hw *mh = &ac97_hw[oss_channel];
-
-#ifdef DEBUG
-	printk("ac97_codec: wrote OSS mixer %2d (%s ac97 register 0x%02x), "
-	       "left vol:%2d, right vol:%2d:",
-	       oss_channel, codec->id ? "Secondary" : "Primary",
-	       mh->offset, left, right);
-#endif
-
-	if (AC97_STEREO_MASK & (1 << oss_channel)) {
-		/* stereo mixers */
-		if (left == 0 && right == 0) {
-			val = AC97_MUTE;
-		} else {
-			if (oss_channel == SOUND_MIXER_IGAIN) {
-				right = (right * mh->scale) / 100;
-				left = (left * mh->scale) / 100;
-				if (right >= mh->scale)
-					right = mh->scale-1;
-				if (left >= mh->scale)
-					left = mh->scale-1;
-			} else {
-				/* these may have 5 or 6 bit resolution */
-				if (oss_channel == SOUND_MIXER_VOLUME ||
-				    oss_channel == SOUND_MIXER_ALTPCM)
-					scale = (1 << codec->bit_resolution);
-				else
-					scale = mh->scale;
-
-				right = ((100 - right) * scale) / 100;
-				left = ((100 - left) * scale) / 100;
-				if (right >= scale)
-					right = scale-1;
-				if (left >= scale)
-					left = scale-1;
-			}
-			val = (left << 8) | right;
-		}
-	} else if (oss_channel == SOUND_MIXER_BASS) {
-		val = codec->codec_read(codec , mh->offset) & ~0x0f00;
-		left = ((100 - left) * mh->scale) / 100;
-		if (left >= mh->scale)
-			left = mh->scale-1;
-		val |= (left << 8) & 0x0e00;
-	} else if (oss_channel == SOUND_MIXER_TREBLE) {
-		val = codec->codec_read(codec , mh->offset) & ~0x000f;
-		left = ((100 - left) * mh->scale) / 100;
-		if (left >= mh->scale)
-			left = mh->scale-1;
-		val |= left & 0x000e;
-	} else if(left == 0) {
-		val = AC97_MUTE;
-	} else if (oss_channel == SOUND_MIXER_SPEAKER) {
-		left = ((100 - left) * mh->scale) / 100;
-		if (left >= mh->scale)
-			left = mh->scale-1;
-		val = left << 1;
-	} else if (oss_channel == SOUND_MIXER_PHONEIN) {
-		left = ((100 - left) * mh->scale) / 100;
-		if (left >= mh->scale)
-			left = mh->scale-1;
-		val = left;
-	} else if (oss_channel == SOUND_MIXER_PHONEOUT) {
-		scale = (1 << codec->bit_resolution);
-		left = ((100 - left) * scale) / 100;
-		if (left >= mh->scale)
-			left = mh->scale-1;
-		val = left;
-	} else if (oss_channel == SOUND_MIXER_MIC) {
-		val = codec->codec_read(codec , mh->offset) & ~0x801f;
-		left = ((100 - left) * mh->scale) / 100;
-		if (left >= mh->scale)
-			left = mh->scale-1;
-		val |= left;
-		/*  the low bit is optional in the tone sliders and masking
-		    it lets us avoid the 0xf 'bypass'.. */
-	}
-#ifdef DEBUG
-	printk(" 0x%04x", val);
-#endif
-
-	codec->codec_write(codec, mh->offset, val);
-
-#ifdef DEBUG
-	val = codec->codec_read(codec, mh->offset);
-	printk(" -> 0x%04x\n", val);
-#endif
-}
-
-/* a thin wrapper for write_mixer */
-static void ac97_set_mixer(struct ac97_codec *codec, unsigned int oss_mixer, unsigned int val ) 
-{
-	unsigned int left,right;
-
-	/* cleanse input a little */
-	right = ((val >> 8)  & 0xff) ;
-	left = (val  & 0xff) ;
-
-	if (right > 100) right = 100;
-	if (left > 100) left = 100;
-
-	codec->mixer_state[oss_mixer] = (right << 8) | left;
-	codec->write_mixer(codec, oss_mixer, left, right);
-}
-
-/* read or write the recmask, the ac97 can really have left and right recording
-   inputs independently set, but OSS doesn't seem to want us to express that to
-   the user. the caller guarantees that we have a supported bit set, and they
-   must be holding the card's spinlock */
-static int ac97_recmask_io(struct ac97_codec *codec, int rw, int mask) 
-{
-	unsigned int val;
-
-	if (rw) {
-		/* read it from the card */
-		val = codec->codec_read(codec, AC97_RECORD_SELECT);
-#ifdef DEBUG
-		printk("ac97_codec: ac97 recmask to set to 0x%04x\n", val);
-#endif
-		return (1 << ac97_rm2oss[val & 0x07]);
-	}
-
-	/* else, write the first set in the mask as the
-	   output */	
-	/* clear out current set value first (AC97 supports only 1 input!) */
-	val = (1 << ac97_rm2oss[codec->codec_read(codec, AC97_RECORD_SELECT) & 0x07]);
-	if (mask != val)
-	    mask &= ~val;
-       
-	val = ffs(mask); 
-	val = ac97_oss_rm[val-1];
-	val |= val << 8;  /* set both channels */
-
-#ifdef DEBUG
-	printk("ac97_codec: setting ac97 recmask to 0x%04x\n", val);
-#endif
-
-	codec->codec_write(codec, AC97_RECORD_SELECT, val);
-
-	return 0;
-};
-
-static int ac97_mixer_ioctl(struct ac97_codec *codec, unsigned int cmd, unsigned long arg)
-{
-	int i, val = 0;
-
-	if (cmd == SOUND_MIXER_INFO) {
-		mixer_info info;
-		memset(&info, 0, sizeof(info));
-		strlcpy(info.id, codec->name, sizeof(info.id));
-		strlcpy(info.name, codec->name, sizeof(info.name));
-		info.modify_counter = codec->modcnt;
-		if (copy_to_user((void __user *)arg, &info, sizeof(info)))
-			return -EFAULT;
-		return 0;
-	}
-	if (cmd == SOUND_OLD_MIXER_INFO) {
-		_old_mixer_info info;
-		memset(&info, 0, sizeof(info));
-		strlcpy(info.id, codec->name, sizeof(info.id));
-		strlcpy(info.name, codec->name, sizeof(info.name));
-		if (copy_to_user((void __user *)arg, &info, sizeof(info)))
-			return -EFAULT;
-		return 0;
-	}
-
-	if (_IOC_TYPE(cmd) != 'M' || _SIOC_SIZE(cmd) != sizeof(int))
-		return -EINVAL;
-
-	if (cmd == OSS_GETVERSION)
-		return put_user(SOUND_VERSION, (int __user *)arg);
-
-	if (_SIOC_DIR(cmd) == _SIOC_READ) {
-		switch (_IOC_NR(cmd)) {
-		case SOUND_MIXER_RECSRC: /* give them the current record source */
-			if (!codec->recmask_io) {
-				val = 0;
-			} else {
-				val = codec->recmask_io(codec, 1, 0);
-			}
-			break;
-
-		case SOUND_MIXER_DEVMASK: /* give them the supported mixers */
-			val = codec->supported_mixers;
-			break;
-
-		case SOUND_MIXER_RECMASK: /* Arg contains a bit for each supported recording source */
-			val = codec->record_sources;
-			break;
-
-		case SOUND_MIXER_STEREODEVS: /* Mixer channels supporting stereo */
-			val = codec->stereo_mixers;
-			break;
-
-		case SOUND_MIXER_CAPS:
-			val = SOUND_CAP_EXCL_INPUT;
-			break;
-
-		default: /* read a specific mixer */
-			i = _IOC_NR(cmd);
-
-			if (!supported_mixer(codec, i)) 
-				return -EINVAL;
-
-			/* do we ever want to touch the hardware? */
-		        /* val = codec->read_mixer(codec, i); */
-			val = codec->mixer_state[i];
- 			break;
-		}
-		return put_user(val, (int __user *)arg);
-	}
-
-	if (_SIOC_DIR(cmd) == (_SIOC_WRITE|_SIOC_READ)) {
-		codec->modcnt++;
-		if (get_user(val, (int __user *)arg))
-			return -EFAULT;
-
-		switch (_IOC_NR(cmd)) {
-		case SOUND_MIXER_RECSRC: /* Arg contains a bit for each recording source */
-			if (!codec->recmask_io) return -EINVAL;
-			if (!val) return 0;
-			if (!(val &= codec->record_sources)) return -EINVAL;
-
-			codec->recmask_io(codec, 0, val);
-
-			return 0;
-		default: /* write a specific mixer */
-			i = _IOC_NR(cmd);
-
-			if (!supported_mixer(codec, i)) 
-				return -EINVAL;
-
-			ac97_set_mixer(codec, i, val);
-
-			return 0;
-		}
-	}
-	return -EINVAL;
-}
-
-/**
- *	codec_id	-  Turn id1/id2 into a PnP string
- *	@id1: Vendor ID1
- *	@id2: Vendor ID2
- *	@buf: CODEC_ID_BUFSZ byte buffer
- *
- *	Fills buf with a zero terminated PnP ident string for the id1/id2
- *	pair. For convenience the return is the passed in buffer pointer.
- */
- 
-static char *codec_id(u16 id1, u16 id2, char *buf)
-{
-	if(id1&0x8080) {
-		snprintf(buf, CODEC_ID_BUFSZ, "0x%04x:0x%04x", id1, id2);
-	} else {
-		buf[0] = (id1 >> 8);
-		buf[1] = (id1 & 0xFF);
-		buf[2] = (id2 >> 8);
-		snprintf(buf+3, CODEC_ID_BUFSZ - 3, "%d", id2&0xFF);
-	}
-	return buf;
-}
- 
-/**
- *	ac97_check_modem - Check if the Codec is a modem
- *	@codec: codec to check
- *
- *	Return true if the device is an AC97 1.0 or AC97 2.0 modem
- */
- 
-static int ac97_check_modem(struct ac97_codec *codec)
-{
-	/* Check for an AC97 1.0 soft modem (ID1) */
-	if(codec->codec_read(codec, AC97_RESET) & 2)
-		return 1;
-	/* Check for an AC97 2.x soft modem */
-	codec->codec_write(codec, AC97_EXTENDED_MODEM_ID, 0L);
-	if(codec->codec_read(codec, AC97_EXTENDED_MODEM_ID) & 1)
-		return 1;
-	return 0;
-}
-
-
-/**
- *	ac97_alloc_codec - Allocate an AC97 codec
- *
- *	Returns a new AC97 codec structure. AC97 codecs may become
- *	refcounted soon so this interface is needed. Returns with
- *	one reference taken.
- */
- 
-struct ac97_codec *ac97_alloc_codec(void)
-{
-	struct ac97_codec *codec = kzalloc(sizeof(struct ac97_codec), GFP_KERNEL);
-	if(!codec)
-		return NULL;
-
-	spin_lock_init(&codec->lock);
-	INIT_LIST_HEAD(&codec->list);
-	return codec;
-}
-
-EXPORT_SYMBOL(ac97_alloc_codec);
-
-/**
- *	ac97_release_codec -	Release an AC97 codec
- *	@codec: codec to release
- *
- *	Release an allocated AC97 codec. This will be refcounted in
- *	time but for the moment is trivial. Calls the unregister
- *	handler if the codec is now defunct.
- */
- 
-void ac97_release_codec(struct ac97_codec *codec)
-{
-	/* Remove from the list first, we don't want to be
-	   "rediscovered" */
-	mutex_lock(&codec_mutex);
-	list_del(&codec->list);
-	mutex_unlock(&codec_mutex);
-	/*
-	 *	The driver needs to deal with internal
-	 *	locking to avoid accidents here. 
-	 */
-	if(codec->driver)
-		codec->driver->remove(codec, codec->driver);
-	kfree(codec);
-}
-
-EXPORT_SYMBOL(ac97_release_codec);
-
-/**
- *	ac97_probe_codec - Initialize and setup AC97-compatible codec
- *	@codec: (in/out) Kernel info for a single AC97 codec
- *
- *	Reset the AC97 codec, then initialize the mixer and
- *	the rest of the @codec structure.
- *
- *	The codec_read and codec_write fields of @codec are
- *	required to be setup and working when this function
- *	is called.  All other fields are set by this function.
- *
- *	codec_wait field of @codec can optionally be provided
- *	when calling this function.  If codec_wait is not %NULL,
- *	this function will call codec_wait any time it is
- *	necessary to wait for the audio chip to reach the
- *	codec-ready state.  If codec_wait is %NULL, then
- *	the default behavior is to call schedule_timeout.
- *	Currently codec_wait is used to wait for AC97 codec
- *	reset to complete. 
- *
- *     Some codecs will power down when a register reset is
- *     performed. We now check for such codecs.
- *
- *	Returns 1 (true) on success, or 0 (false) on failure.
- */
- 
-int ac97_probe_codec(struct ac97_codec *codec)
-{
-	u16 id1, id2;
-	u16 audio;
-	int i;
-	char cidbuf[CODEC_ID_BUFSZ];
-	u16 f;
-	struct list_head *l;
-	struct ac97_driver *d;
-	
-	/* wait for codec-ready state */
-	if (codec->codec_wait)
-		codec->codec_wait(codec);
-	else
-		udelay(10);
-
-	/* will the codec power down if register reset ? */
-	id1 = codec->codec_read(codec, AC97_VENDOR_ID1);
-	id2 = codec->codec_read(codec, AC97_VENDOR_ID2);
-	codec->name = NULL;
-	codec->codec_ops = &null_ops;
-	for (i = 0; i < ARRAY_SIZE(ac97_codec_ids); i++) {
-		if (ac97_codec_ids[i].id == ((id1 << 16) | id2)) {
-			codec->type = ac97_codec_ids[i].id;
-			codec->name = ac97_codec_ids[i].name;
-			codec->codec_ops = ac97_codec_ids[i].ops;
-			codec->flags = ac97_codec_ids[i].flags;
-			break;
-		}
-	}
-
-	codec->model = (id1 << 16) | id2;
-	if ((codec->flags & AC97_DEFAULT_POWER_OFF) == 0) {
-		/* reset codec and wait for the ready bit before we continue */
-		codec->codec_write(codec, AC97_RESET, 0L);
-		if (codec->codec_wait)
-			codec->codec_wait(codec);
-		else
-			udelay(10);
-	}
-
-	/* probing AC97 codec, AC97 2.0 says that bit 15 of register 0x00 (reset) should
-	 * be read zero.
-	 *
-	 * FIXME: is the following comment outdated?  -jgarzik
-	 * Probing of AC97 in this way is not reliable, it is not even SAFE !!
-	 */
-	if ((audio = codec->codec_read(codec, AC97_RESET)) & 0x8000) {
-		printk(KERN_ERR "ac97_codec: %s ac97 codec not present\n",
-		       (codec->id & 0x2) ? (codec->id&1 ? "4th" : "Tertiary")
-		       : (codec->id&1 ? "Secondary":  "Primary"));
-		return 0;
-	}
-	
-	/* probe for Modem Codec */
-	codec->modem = ac97_check_modem(codec);
-
-	/* enable SPDIF */
-	f = codec->codec_read(codec, AC97_EXTENDED_STATUS);
-	if((codec->codec_ops == &null_ops) && (f & 4))
-		codec->codec_ops = &default_digital_ops;
-	
-	/* A device which thinks its a modem but isn't */
-	if(codec->flags & AC97_DELUDED_MODEM)
-		codec->modem = 0;
-		
-	if (codec->name == NULL)
-		codec->name = "Unknown";
-	printk(KERN_INFO "ac97_codec: AC97 %s codec, id: %s (%s)\n", 
-		codec->modem ? "Modem" : (audio ? "Audio" : ""),
-	       codec_id(id1, id2, cidbuf), codec->name);
-
-	if(!ac97_init_mixer(codec))
-		return 0;
-		
-	/* 
-	 *	Attach last so the caller can override the mixer
-	 *	callbacks.
-	 */
-	 
-	mutex_lock(&codec_mutex);
-	list_add(&codec->list, &codecs);
-
-	list_for_each(l, &codec_drivers) {
-		d = list_entry(l, struct ac97_driver, list);
-		if ((codec->model ^ d->codec_id) & d->codec_mask)
-			continue;
-		if(d->probe(codec, d) == 0)
-		{
-			codec->driver = d;
-			break;
-		}
-	}
-
-	mutex_unlock(&codec_mutex);
-	return 1;
-}
-
-static int ac97_init_mixer(struct ac97_codec *codec)
-{
-	u16 cap;
-	int i;
-
-	cap = codec->codec_read(codec, AC97_RESET);
-
-	/* mixer masks */
-	codec->supported_mixers = AC97_SUPPORTED_MASK;
-	codec->stereo_mixers = AC97_STEREO_MASK;
-	codec->record_sources = AC97_RECORD_MASK;
-	if (!(cap & 0x04))
-		codec->supported_mixers &= ~(SOUND_MASK_BASS|SOUND_MASK_TREBLE);
-	if (!(cap & 0x10))
-		codec->supported_mixers &= ~SOUND_MASK_ALTPCM;
-
-
-	/* detect bit resolution */
-	codec->codec_write(codec, AC97_MASTER_VOL_STEREO, 0x2020);
-	if(codec->codec_read(codec, AC97_MASTER_VOL_STEREO) == 0x2020)
-		codec->bit_resolution = 6;
-	else
-		codec->bit_resolution = 5;
-
-	/* generic OSS to AC97 wrapper */
-	codec->read_mixer = ac97_read_mixer;
-	codec->write_mixer = ac97_write_mixer;
-	codec->recmask_io = ac97_recmask_io;
-	codec->mixer_ioctl = ac97_mixer_ioctl;
-
-	/* initialize mixer channel volumes */
-	for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
-		struct mixer_defaults *md = &mixer_defaults[i];
-		if (md->mixer == -1) 
-			break;
-		if (!supported_mixer(codec, md->mixer)) 
-			continue;
-		ac97_set_mixer(codec, md->mixer, md->value);
-	}
-
-	/* codec specific initialization for 4-6 channel output or secondary codec stuff */
-	if (codec->codec_ops->init != NULL) {
-		codec->codec_ops->init(codec);
-	}
-
-	/*
-	 *	Volume is MUTE only on this device. We have to initialise
-	 *	it but its useless beyond that.
-	 */
-	if(codec->flags & AC97_NO_PCM_VOLUME)
-	{
-		codec->supported_mixers &= ~SOUND_MASK_PCM;
-		printk(KERN_WARNING "AC97 codec does not have proper volume support.\n");
-	}
-	return 1;
-}
-
-#define AC97_SIGMATEL_ANALOG    0x6c	/* Analog Special */
-#define AC97_SIGMATEL_DAC2INVERT 0x6e
-#define AC97_SIGMATEL_BIAS1     0x70
-#define AC97_SIGMATEL_BIAS2     0x72
-#define AC97_SIGMATEL_MULTICHN  0x74	/* Multi-Channel programming */
-#define AC97_SIGMATEL_CIC1      0x76
-#define AC97_SIGMATEL_CIC2      0x78
-
-
-static int sigmatel_9708_init(struct ac97_codec * codec)
-{
-	u16 codec72, codec6c;
-
-	codec72 = codec->codec_read(codec, AC97_SIGMATEL_BIAS2) & 0x8000;
-	codec6c = codec->codec_read(codec, AC97_SIGMATEL_ANALOG);
-
-	if ((codec72==0) && (codec6c==0)) {
-		codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba);
-		codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x1000);
-		codec->codec_write(codec, AC97_SIGMATEL_BIAS1, 0xabba);
-		codec->codec_write(codec, AC97_SIGMATEL_BIAS2, 0x0007);
-	} else if ((codec72==0x8000) && (codec6c==0)) {
-		codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba);
-		codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x1001);
-		codec->codec_write(codec, AC97_SIGMATEL_DAC2INVERT, 0x0008);
-	} else if ((codec72==0x8000) && (codec6c==0x0080)) {
-		/* nothing */
-	}
-	codec->codec_write(codec, AC97_SIGMATEL_MULTICHN, 0x0000);
-	return 0;
-}
-
-
-static int sigmatel_9721_init(struct ac97_codec * codec)
-{
-	/* Only set up secondary codec */
-	if (codec->id == 0)
-		return 0;
-
-	codec->codec_write(codec, AC97_SURROUND_MASTER, 0L);
-
-	/* initialize SigmaTel STAC9721/23 as secondary codec, decoding AC link
-	   sloc 3,4 = 0x01, slot 7,8 = 0x00, */
-	codec->codec_write(codec, AC97_SIGMATEL_MULTICHN, 0x00);
-
-	/* we don't have the crystal when we are on an AMR card, so use
-	   BIT_CLK as our clock source. Write the magic word ABBA and read
-	   back to enable register 0x78 */
-	codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba);
-	codec->codec_read(codec, AC97_SIGMATEL_CIC1);
-
-	/* sync all the clocks*/
-	codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x3802);
-
-	return 0;
-}
-
-
-static int sigmatel_9744_init(struct ac97_codec * codec)
-{
-	// patch for SigmaTel
-	codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba);
-	codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x0000); // is this correct? --jk
-	codec->codec_write(codec, AC97_SIGMATEL_BIAS1, 0xabba);
-	codec->codec_write(codec, AC97_SIGMATEL_BIAS2, 0x0002);
-	codec->codec_write(codec, AC97_SIGMATEL_MULTICHN, 0x0000);
-	return 0;
-}
-
-static int cmedia_init(struct ac97_codec *codec)
-{
-	/* Initialise the CMedia 9739 */
-	/*
-		We could set various options here
-		Register 0x20 bit 0x100 sets mic as center bass
-		Also do multi_channel_ctrl &=~0x3000 |=0x1000
-		
-		For now we set up the GPIO and PC beep 
-	*/
-	
-	u16 v;
-	
-	/* MIC */
-	codec->codec_write(codec, 0x64, 0x3000);
-	v = codec->codec_read(codec, 0x64);
-	v &= ~0x8000;
-	codec->codec_write(codec, 0x64, v);
-	codec->codec_write(codec, 0x70, 0x0100);
-	codec->codec_write(codec, 0x72, 0x0020);
-	return 0;
-}
-	
-#define AC97_WM97XX_FMIXER_VOL 0x72
-#define AC97_WM97XX_RMIXER_VOL 0x74
-#define AC97_WM97XX_TEST 0x5a
-#define AC97_WM9704_RPCM_VOL 0x70
-#define AC97_WM9711_OUT3VOL 0x16
-
-static int wolfson_init03(struct ac97_codec * codec)
-{
-	/* this is known to work for the ViewSonic ViewPad 1000 */
-	codec->codec_write(codec, AC97_WM97XX_FMIXER_VOL, 0x0808);
-	codec->codec_write(codec, AC97_GENERAL_PURPOSE, 0x8000);
-	return 0;
-}
-
-static int wolfson_init04(struct ac97_codec * codec)
-{
-	codec->codec_write(codec, AC97_WM97XX_FMIXER_VOL, 0x0808);
-	codec->codec_write(codec, AC97_WM97XX_RMIXER_VOL, 0x0808);
-
-	// patch for DVD noise
-	codec->codec_write(codec, AC97_WM97XX_TEST, 0x0200);
-
-	// init vol as PCM vol
-	codec->codec_write(codec, AC97_WM9704_RPCM_VOL,
-		codec->codec_read(codec, AC97_PCMOUT_VOL));
-
-	/* set rear surround volume */
-	codec->codec_write(codec, AC97_SURROUND_MASTER, 0x0000);
-	return 0;
-}
-
-/* WM9705, WM9710 */
-static int wolfson_init05(struct ac97_codec * codec)
-{
-	/* set front mixer volume */
-	codec->codec_write(codec, AC97_WM97XX_FMIXER_VOL, 0x0808);
-	return 0;
-}
-
-/* WM9711, WM9712 */
-static int wolfson_init11(struct ac97_codec * codec)
-{
-	/* stop pop's during suspend/resume */
-	codec->codec_write(codec, AC97_WM97XX_TEST,
-		codec->codec_read(codec, AC97_WM97XX_TEST) & 0xffbf);
-
-	/* set out3 volume */
-	codec->codec_write(codec, AC97_WM9711_OUT3VOL, 0x0808);
-	return 0;
-}
-
-/* WM9713 */
-static int wolfson_init13(struct ac97_codec * codec)
-{
-	codec->codec_write(codec, AC97_RECORD_GAIN, 0x00a0);
-	codec->codec_write(codec, AC97_POWER_CONTROL, 0x0000);
-	codec->codec_write(codec, AC97_EXTENDED_MODEM_ID, 0xDA00);
-	codec->codec_write(codec, AC97_EXTEND_MODEM_STAT, 0x3810);
-	codec->codec_write(codec, AC97_PHONE_VOL, 0x0808);
-	codec->codec_write(codec, AC97_PCBEEP_VOL, 0x0808);
-
-	return 0;
-}
-
-static int tritech_init(struct ac97_codec * codec)
-{
-	codec->codec_write(codec, 0x26, 0x0300);
-	codec->codec_write(codec, 0x26, 0x0000);
-	codec->codec_write(codec, AC97_SURROUND_MASTER, 0x0000);
-	codec->codec_write(codec, AC97_RESERVED_3A, 0x0000);
-	return 0;
-}
-
-
-/* copied from drivers/sound/maestro.c */
-static int tritech_maestro_init(struct ac97_codec * codec)
-{
-	/* no idea what this does */
-	codec->codec_write(codec, 0x2A, 0x0001);
-	codec->codec_write(codec, 0x2C, 0x0000);
-	codec->codec_write(codec, 0x2C, 0XFFFF);
-	return 0;
-}
-
-
-
-/* 
- *	Presario700 workaround 
- * 	for Jack Sense/SPDIF Register mis-setting causing
- *	no audible output
- *	by Santiago Nullo 04/05/2002
- */
-
-#define AC97_AD1886_JACK_SENSE 0x72
-
-static int ad1886_init(struct ac97_codec * codec)
-{
-	/* from AD1886 Specs */
-	codec->codec_write(codec, AC97_AD1886_JACK_SENSE, 0x0010);
-	return 0;
-}
-
-
-
-
-/*
- *	This is basically standard AC97. It should work as a default for
- *	almost all modern codecs. Note that some cards wire EAPD *backwards*
- *	That side of it is up to the card driver not us to cope with.
- *
- */
-
-static int eapd_control(struct ac97_codec * codec, int on)
-{
-	if(on)
-		codec->codec_write(codec, AC97_POWER_CONTROL,
-			codec->codec_read(codec, AC97_POWER_CONTROL)|0x8000);
-	else
-		codec->codec_write(codec, AC97_POWER_CONTROL,
-			codec->codec_read(codec, AC97_POWER_CONTROL)&~0x8000);
-	return 0;
-}
-
-static int generic_digital_control(struct ac97_codec *codec, int slots, int rate, int mode)
-{
-	u16 reg;
-	
-	reg = codec->codec_read(codec, AC97_SPDIF_CONTROL);
-	
-	switch(rate)
-	{
-		/* Off by default */
-		default:
-		case 0:
-			reg = codec->codec_read(codec, AC97_EXTENDED_STATUS);
-			codec->codec_write(codec, AC97_EXTENDED_STATUS, (reg & ~AC97_EA_SPDIF));
-			if(rate == 0)
-				return 0;
-			return -EINVAL;
-		case 1:
-			reg = (reg & AC97_SC_SPSR_MASK) | AC97_SC_SPSR_48K;
-			break;
-		case 2:
-			reg = (reg & AC97_SC_SPSR_MASK) | AC97_SC_SPSR_44K;
-			break;
-		case 3:
-			reg = (reg & AC97_SC_SPSR_MASK) | AC97_SC_SPSR_32K;
-			break;
-	}
-	
-	reg &= ~AC97_SC_CC_MASK;
-	reg |= (mode & AUDIO_CCMASK) << 6;
-	
-	if(mode & AUDIO_DIGITAL)
-		reg |= 2;
-	if(mode & AUDIO_PRO)
-		reg |= 1;
-	if(mode & AUDIO_DRS)
-		reg |= 0x4000;
-
-	codec->codec_write(codec, AC97_SPDIF_CONTROL, reg);
-
-	reg = codec->codec_read(codec, AC97_EXTENDED_STATUS);
-	reg &= (AC97_EA_SLOT_MASK);
-	reg |= AC97_EA_VRA | AC97_EA_SPDIF | slots;
-	codec->codec_write(codec, AC97_EXTENDED_STATUS, reg);
-	
-	reg = codec->codec_read(codec, AC97_EXTENDED_STATUS);
-	if(!(reg & 0x0400))
-	{
-		codec->codec_write(codec, AC97_EXTENDED_STATUS, reg & ~ AC97_EA_SPDIF);
-		return -EINVAL;
-	}
-	return 0;
-}
-
-/*
- *	Crystal digital audio control (CS4299)
- */
- 
-static int crystal_digital_control(struct ac97_codec *codec, int slots, int rate, int mode)
-{
-	u16 cv;
-
-	if(mode & AUDIO_DIGITAL)
-		return -EINVAL;
-		
-	switch(rate)
-	{
-		case 0: cv = 0x0; break;	/* SPEN off */
-		case 48000: cv = 0x8004; break;	/* 48KHz digital */
-		case 44100: cv = 0x8104; break;	/* 44.1KHz digital */
-		case 32768: 			/* 32Khz */
-		default:
-			return -EINVAL;
-	}
-	codec->codec_write(codec, 0x68, cv);
-	return 0;
-}
-
-/*
- *	CMedia digital audio control
- *	Needs more work.
- */
- 
-static int cmedia_digital_control(struct ac97_codec *codec, int slots, int rate, int mode)
-{
-	u16 cv;
-
-	if(mode & AUDIO_DIGITAL)
-		return -EINVAL;
-		
-	switch(rate)
-	{
-		case 0:		cv = 0x0001; break;	/* SPEN off */
-		case 48000:	cv = 0x0009; break;	/* 48KHz digital */
-		default:
-			return -EINVAL;
-	}
-	codec->codec_write(codec, 0x2A, 0x05c4);
-	codec->codec_write(codec, 0x6C, cv);
-	
-	/* Switch on mix to surround */
-	cv = codec->codec_read(codec, 0x64);
-	cv &= ~0x0200;
-	if(mode)
-		cv |= 0x0200;
-	codec->codec_write(codec, 0x64, cv);
-	return 0;
-}
-
-
-/* copied from drivers/sound/maestro.c */
-#if 0  /* there has been 1 person on the planet with a pt101 that we
-        know of.  If they care, they can put this back in :) */
-static int pt101_init(struct ac97_codec * codec)
-{
-	printk(KERN_INFO "ac97_codec: PT101 Codec detected, initializing but _not_ installing mixer device.\n");
-	/* who knows.. */
-	codec->codec_write(codec, 0x2A, 0x0001);
-	codec->codec_write(codec, 0x2C, 0x0000);
-	codec->codec_write(codec, 0x2C, 0xFFFF);
-	codec->codec_write(codec, 0x10, 0x9F1F);
-	codec->codec_write(codec, 0x12, 0x0808);
-	codec->codec_write(codec, 0x14, 0x9F1F);
-	codec->codec_write(codec, 0x16, 0x9F1F);
-	codec->codec_write(codec, 0x18, 0x0404);
-	codec->codec_write(codec, 0x1A, 0x0000);
-	codec->codec_write(codec, 0x1C, 0x0000);
-	codec->codec_write(codec, 0x02, 0x0404);
-	codec->codec_write(codec, 0x04, 0x0808);
-	codec->codec_write(codec, 0x0C, 0x801F);
-	codec->codec_write(codec, 0x0E, 0x801F);
-	return 0;
-}
-#endif
-	
-
-EXPORT_SYMBOL(ac97_probe_codec);
-
-MODULE_LICENSE("GPL");
-
diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c
deleted file mode 100644
index a8f626d99c5..00000000000
--- a/sound/oss/au1550_ac97.c
+++ /dev/null
@@ -1,2147 +0,0 @@
-/*
- * au1550_ac97.c  --  Sound driver for Alchemy Au1550 MIPS Internet Edge
- *                    Processor.
- *
- * Copyright 2004 Embedded Edge, LLC
- *	dan@embeddededge.com
- *
- * Mostly copied from the au1000.c driver and some from the
- * PowerMac dbdma driver.
- * We assume the processor can do memory coherent DMA.
- *
- * Ported to 2.6 by Matt Porter <mporter@kernel.crashing.org>
- *
- *  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  SOFTWARE  IS PROVIDED   ``AS  IS'' AND   ANY  EXPRESS OR IMPLIED
- *  WARRANTIES,   INCLUDING, BUT NOT  LIMITED  TO, THE IMPLIED WARRANTIES OF
- *  MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.  IN
- *  NO  EVENT  SHALL   THE AUTHOR  BE    LIABLE FOR ANY   DIRECT, INDIRECT,
- *  INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
- *  NOT LIMITED   TO, PROCUREMENT OF  SUBSTITUTE GOODS  OR SERVICES; LOSS OF
- *  USE, DATA,  OR PROFITS; OR  BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
- *  ANY THEORY OF LIABILITY, WHETHER IN  CONTRACT, STRICT LIABILITY, OR TORT
- *  (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
- *  THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- *
- *  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.,
- *  675 Mass Ave, Cambridge, MA 02139, USA.
- *
- */
-
-#undef DEBUG
-
-#include <linux/module.h>
-#include <linux/string.h>
-#include <linux/ioport.h>
-#include <linux/sched.h>
-#include <linux/delay.h>
-#include <linux/sound.h>
-#include <linux/slab.h>
-#include <linux/soundcard.h>
-#include <linux/init.h>
-#include <linux/interrupt.h>
-#include <linux/kernel.h>
-#include <linux/poll.h>
-#include <linux/bitops.h>
-#include <linux/spinlock.h>
-#include <linux/ac97_codec.h>
-#include <linux/mutex.h>
-
-#include <asm/io.h>
-#include <asm/uaccess.h>
-#include <asm/hardirq.h>
-#include <asm/mach-au1x00/au1xxx_psc.h>
-#include <asm/mach-au1x00/au1xxx_dbdma.h>
-#include <asm/mach-au1x00/au1xxx.h>
-
-#undef OSS_DOCUMENTED_MIXER_SEMANTICS
-
-/* misc stuff */
-#define POLL_COUNT   0x50000
-#define AC97_EXT_DACS (AC97_EXTID_SDAC | AC97_EXTID_CDAC | AC97_EXTID_LDAC)
-
-/* The number of DBDMA ring descriptors to allocate.  No sense making
- * this too large....if you can't keep up with a few you aren't likely
- * to be able to with lots of them, either.
- */
-#define NUM_DBDMA_DESCRIPTORS 4
-
-#define err(format, arg...) printk(KERN_ERR format "\n" , ## arg)
-
-/* Boot options
- * 0 = no VRA, 1 = use VRA if codec supports it
- */
-static DEFINE_MUTEX(au1550_ac97_mutex);
-static int      vra = 1;
-module_param(vra, bool, 0);
-MODULE_PARM_DESC(vra, "if 1 use VRA if codec supports it");
-
-static struct au1550_state {
-	/* soundcore stuff */
-	int             dev_audio;
-
-	struct ac97_codec *codec;
-	unsigned        codec_base_caps; /* AC'97 reg 00h, "Reset Register" */
-	unsigned        codec_ext_caps;  /* AC'97 reg 28h, "Extended Audio ID" */
-	int             no_vra;		/* do not use VRA */
-
-	spinlock_t      lock;
-	struct mutex open_mutex;
-	struct mutex sem;
-	fmode_t          open_mode;
-	wait_queue_head_t open_wait;
-
-	struct dmabuf {
-		u32		dmanr;
-		unsigned        sample_rate;
-		unsigned	src_factor;
-		unsigned        sample_size;
-		int             num_channels;
-		int		dma_bytes_per_sample;
-		int		user_bytes_per_sample;
-		int		cnt_factor;
-
-		void		*rawbuf;
-		unsigned        buforder;
-		unsigned	numfrag;
-		unsigned        fragshift;
-		void		*nextIn;
-		void		*nextOut;
-		int		count;
-		unsigned        total_bytes;
-		unsigned        error;
-		wait_queue_head_t wait;
-
-		/* redundant, but makes calculations easier */
-		unsigned	fragsize;
-		unsigned	dma_fragsize;
-		unsigned	dmasize;
-		unsigned	dma_qcount;
-
-		/* OSS stuff */
-		unsigned        mapped:1;
-		unsigned        ready:1;
-		unsigned        stopped:1;
-		unsigned        ossfragshift;
-		int             ossmaxfrags;
-		unsigned        subdivision;
-	} dma_dac, dma_adc;
-} au1550_state;
-
-static unsigned
-ld2(unsigned int x)
-{
-	unsigned        r = 0;
-
-	if (x >= 0x10000) {
-		x >>= 16;
-		r += 16;
-	}
-	if (x >= 0x100) {
-		x >>= 8;
-		r += 8;
-	}
-	if (x >= 0x10) {
-		x >>= 4;
-		r += 4;
-	}
-	if (x >= 4) {
-		x >>= 2;
-		r += 2;
-	}
-	if (x >= 2)
-		r++;
-	return r;
-}
-
-static void
-au1550_delay(int msec)
-{
-	if (in_interrupt())
-		return;
-
-	schedule_timeout_uninterruptible(msecs_to_jiffies(msec));
-}
-
-static u16
-rdcodec(struct ac97_codec *codec, u8 addr)
-{
-	struct au1550_state *s = codec->private_data;
-	unsigned long   flags;
-	u32             cmd, val;
-	u16             data;
-	int             i;
-
-	spin_lock_irqsave(&s->lock, flags);
-
-	for (i = 0; i < POLL_COUNT; i++) {
-		val = au_readl(PSC_AC97STAT);
-		au_sync();
-		if (!(val & PSC_AC97STAT_CP))
-			break;
-	}
-	if (i == POLL_COUNT)
-		err("rdcodec: codec cmd pending expired!");
-
-	cmd = (u32)PSC_AC97CDC_INDX(addr);
-	cmd |= PSC_AC97CDC_RD;	/* read command */
-	au_writel(cmd, PSC_AC97CDC);
-	au_sync();
-
-	/* now wait for the data
-	*/
-	for (i = 0; i < POLL_COUNT; i++) {
-		val = au_readl(PSC_AC97STAT);
-		au_sync();
-		if (!(val & PSC_AC97STAT_CP))
-			break;
-	}
-	if (i == POLL_COUNT) {
-		err("rdcodec: read poll expired!");
-		data = 0;
-		goto out;
-	}
-
-	/* wait for command done?
-	*/
-	for (i = 0; i < POLL_COUNT; i++) {
-		val = au_readl(PSC_AC97EVNT);
-		au_sync();
-		if (val & PSC_AC97EVNT_CD)
-			break;
-	}
-	if (i == POLL_COUNT) {
-		err("rdcodec: read cmdwait expired!");
-		data = 0;
-		goto out;
-	}
-
-	data = au_readl(PSC_AC97CDC) & 0xffff;
-	au_sync();
-
-	/* Clear command done event.
-	*/
-	au_writel(PSC_AC97EVNT_CD, PSC_AC97EVNT);
-	au_sync();
-
- out:
-	spin_unlock_irqrestore(&s->lock, flags);
-
-	return data;
-}
-
-
-static void
-wrcodec(struct ac97_codec *codec, u8 addr, u16 data)
-{
-	struct au1550_state *s = codec->private_data;
-	unsigned long   flags;
-	u32             cmd, val;
-	int             i;
-
-	spin_lock_irqsave(&s->lock, flags);
-
-	for (i = 0; i < POLL_COUNT; i++) {
-		val = au_readl(PSC_AC97STAT);
-		au_sync();
-		if (!(val & PSC_AC97STAT_CP))
-			break;
-	}
-	if (i == POLL_COUNT)
-		err("wrcodec: codec cmd pending expired!");
-
-	cmd = (u32)PSC_AC97CDC_INDX(addr);
-	cmd |= (u32)data;
-	au_writel(cmd, PSC_AC97CDC);
-	au_sync();
-
-	for (i = 0; i < POLL_COUNT; i++) {
-		val = au_readl(PSC_AC97STAT);
-		au_sync();
-		if (!(val & PSC_AC97STAT_CP))
-			break;
-	}
-	if (i == POLL_COUNT)
-		err("wrcodec: codec cmd pending expired!");
-
-	for (i = 0; i < POLL_COUNT; i++) {
-		val = au_readl(PSC_AC97EVNT);
-		au_sync();
-		if (val & PSC_AC97EVNT_CD)
-			break;
-	}
-	if (i == POLL_COUNT)
-		err("wrcodec: read cmdwait expired!");
-
-	/* Clear command done event.
-	*/
-	au_writel(PSC_AC97EVNT_CD, PSC_AC97EVNT);
-	au_sync();
-
-	spin_unlock_irqrestore(&s->lock, flags);
-}
-
-static void
-waitcodec(struct ac97_codec *codec)
-{
-	u16	temp;
-	u32	val;
-	int	i;
-
-	/* codec_wait is used to wait for a ready state after
-	 * an AC97C_RESET.
-	 */
-	au1550_delay(10);
-
-	/* first poll the CODEC_READY tag bit
-	*/
-	for (i = 0; i < POLL_COUNT; i++) {
-		val = au_readl(PSC_AC97STAT);
-		au_sync();
-		if (val & PSC_AC97STAT_CR)
-			break;
-	}
-	if (i == POLL_COUNT) {
-		err("waitcodec: CODEC_READY poll expired!");
-		return;
-	}
-
-	/* get AC'97 powerdown control/status register
-	*/
-	temp = rdcodec(codec, AC97_POWER_CONTROL);
-
-	/* If anything is powered down, power'em up
-	*/
-	if (temp & 0x7f00) {
-		/* Power on
-		*/
-		wrcodec(codec, AC97_POWER_CONTROL, 0);
-		au1550_delay(100);
-
-		/* Reread
-		*/
-		temp = rdcodec(codec, AC97_POWER_CONTROL);
-	}
-
-	/* Check if Codec REF,ANL,DAC,ADC ready
-	*/
-	if ((temp & 0x7f0f) != 0x000f)
-		err("codec reg 26 status (0x%x) not ready!!", temp);
-}
-
-/* stop the ADC before calling */
-static void
-set_adc_rate(struct au1550_state *s, unsigned rate)
-{
-	struct dmabuf  *adc = &s->dma_adc;
-	struct dmabuf  *dac = &s->dma_dac;
-	unsigned        adc_rate, dac_rate;
-	u16             ac97_extstat;
-
-	if (s->no_vra) {
-		/* calc SRC factor
-		*/
-		adc->src_factor = ((96000 / rate) + 1) >> 1;
-		adc->sample_rate = 48000 / adc->src_factor;
-		return;
-	}
-
-	adc->src_factor = 1;
-
-	ac97_extstat = rdcodec(s->codec, AC97_EXTENDED_STATUS);
-
-	rate = rate > 48000 ? 48000 : rate;
-
-	/* enable VRA
-	*/
-	wrcodec(s->codec, AC97_EXTENDED_STATUS,
-		ac97_extstat | AC97_EXTSTAT_VRA);
-
-	/* now write the sample rate
-	*/
-	wrcodec(s->codec, AC97_PCM_LR_ADC_RATE, (u16) rate);
-
-	/* read it back for actual supported rate
-	*/
-	adc_rate = rdcodec(s->codec, AC97_PCM_LR_ADC_RATE);
-
-	pr_debug("set_adc_rate: set to %d Hz\n", adc_rate);
-
-	/* some codec's don't allow unequal DAC and ADC rates, in which case
-	 * writing one rate reg actually changes both.
-	 */
-	dac_rate = rdcodec(s->codec, AC97_PCM_FRONT_DAC_RATE);
-	if (dac->num_channels > 2)
-		wrcodec(s->codec, AC97_PCM_SURR_DAC_RATE, dac_rate);
-	if (dac->num_channels > 4)
-		wrcodec(s->codec, AC97_PCM_LFE_DAC_RATE, dac_rate);
-
-	adc->sample_rate = adc_rate;
-	dac->sample_rate = dac_rate;
-}
-
-/* stop the DAC before calling */
-static void
-set_dac_rate(struct au1550_state *s, unsigned rate)
-{
-	struct dmabuf  *dac = &s->dma_dac;
-	struct dmabuf  *adc = &s->dma_adc;
-	unsigned        adc_rate, dac_rate;
-	u16             ac97_extstat;
-
-	if (s->no_vra) {
-		/* calc SRC factor
-		*/
-		dac->src_factor = ((96000 / rate) + 1) >> 1;
-		dac->sample_rate = 48000 / dac->src_factor;
-		return;
-	}
-
-	dac->src_factor = 1;
-
-	ac97_extstat = rdcodec(s->codec, AC97_EXTENDED_STATUS);
-
-	rate = rate > 48000 ? 48000 : rate;
-
-	/* enable VRA
-	*/
-	wrcodec(s->codec, AC97_EXTENDED_STATUS,
-		ac97_extstat | AC97_EXTSTAT_VRA);
-
-	/* now write the sample rate
-	*/
-	wrcodec(s->codec, AC97_PCM_FRONT_DAC_RATE, (u16) rate);
-
-	/* I don't support different sample rates for multichannel,
-	 * so make these channels the same.
-	 */
-	if (dac->num_channels > 2)
-		wrcodec(s->codec, AC97_PCM_SURR_DAC_RATE, (u16) rate);
-	if (dac->num_channels > 4)
-		wrcodec(s->codec, AC97_PCM_LFE_DAC_RATE, (u16) rate);
-	/* read it back for actual supported rate
-	*/
-	dac_rate = rdcodec(s->codec, AC97_PCM_FRONT_DAC_RATE);
-
-	pr_debug("set_dac_rate: set to %d Hz\n", dac_rate);
-
-	/* some codec's don't allow unequal DAC and ADC rates, in which case
-	 * writing one rate reg actually changes both.
-	 */
-	adc_rate = rdcodec(s->codec, AC97_PCM_LR_ADC_RATE);
-
-	dac->sample_rate = dac_rate;
-	adc->sample_rate = adc_rate;
-}
-
-static void
-stop_dac(struct au1550_state *s)
-{
-	struct dmabuf  *db = &s->dma_dac;
-	u32		stat;
-	unsigned long   flags;
-
-	if (db->stopped)
-		return;
-
-	spin_lock_irqsave(&s->lock, flags);
-
-	au_writel(PSC_AC97PCR_TP, PSC_AC97PCR);
-	au_sync();
-
-	/* Wait for Transmit Busy to show disabled.
-	*/
-	do {
-		stat = au_readl(PSC_AC97STAT);
-		au_sync();
-	} while ((stat & PSC_AC97STAT_TB) != 0);
-
-	au1xxx_dbdma_reset(db->dmanr);
-
-	db->stopped = 1;
-
-	spin_unlock_irqrestore(&s->lock, flags);
-}
-
-static void
-stop_adc(struct au1550_state *s)
-{
-	struct dmabuf  *db = &s->dma_adc;
-	unsigned long   flags;
-	u32		stat;
-
-	if (db->stopped)
-		return;
-
-	spin_lock_irqsave(&s->lock, flags);
-
-	au_writel(PSC_AC97PCR_RP, PSC_AC97PCR);
-	au_sync();
-
-	/* Wait for Receive Busy to show disabled.
-	*/
-	do {
-		stat = au_readl(PSC_AC97STAT);
-		au_sync();
-	} while ((stat & PSC_AC97STAT_RB) != 0);
-
-	au1xxx_dbdma_reset(db->dmanr);
-
-	db->stopped = 1;
-
-	spin_unlock_irqrestore(&s->lock, flags);
-}
-
-
-static void
-set_xmit_slots(int num_channels)
-{
-	u32	ac97_config, stat;
-
-	ac97_config = au_readl(PSC_AC97CFG);
-	au_sync();
-	ac97_config &= ~(PSC_AC97CFG_TXSLOT_MASK | PSC_AC97CFG_DE_ENABLE);
-	au_writel(ac97_config, PSC_AC97CFG);
-	au_sync();
-
-	switch (num_channels) {
-	case 6:		/* stereo with surround and center/LFE,
-			 * slots 3,4,6,7,8,9
-			 */
-		ac97_config |= PSC_AC97CFG_TXSLOT_ENA(6);
-		ac97_config |= PSC_AC97CFG_TXSLOT_ENA(9);
-
-	case 4:		/* stereo with surround, slots 3,4,7,8 */
-		ac97_config |= PSC_AC97CFG_TXSLOT_ENA(7);
-		ac97_config |= PSC_AC97CFG_TXSLOT_ENA(8);
-
-	case 2:		/* stereo, slots 3,4 */
-	case 1:		/* mono */
-		ac97_config |= PSC_AC97CFG_TXSLOT_ENA(3);
-		ac97_config |= PSC_AC97CFG_TXSLOT_ENA(4);
-	}
-
-	au_writel(ac97_config, PSC_AC97CFG);
-	au_sync();
-
-	ac97_config |= PSC_AC97CFG_DE_ENABLE;
-	au_writel(ac97_config, PSC_AC97CFG);
-	au_sync();
-
-	/* Wait for Device ready.
-	*/
-	do {
-		stat = au_readl(PSC_AC97STAT);
-		au_sync();
-	} while ((stat & PSC_AC97STAT_DR) == 0);
-}
-
-static void
-set_recv_slots(int num_channels)
-{
-	u32	ac97_config, stat;
-
-	ac97_config = au_readl(PSC_AC97CFG);
-	au_sync();
-	ac97_config &= ~(PSC_AC97CFG_RXSLOT_MASK | PSC_AC97CFG_DE_ENABLE);
-	au_writel(ac97_config, PSC_AC97CFG);
-	au_sync();
-
-	/* Always enable slots 3 and 4 (stereo). Slot 6 is
-	 * optional Mic ADC, which we don't support yet.
-	 */
-	ac97_config |= PSC_AC97CFG_RXSLOT_ENA(3);
-	ac97_config |= PSC_AC97CFG_RXSLOT_ENA(4);
-
-	au_writel(ac97_config, PSC_AC97CFG);
-	au_sync();
-
-	ac97_config |= PSC_AC97CFG_DE_ENABLE;
-	au_writel(ac97_config, PSC_AC97CFG);
-	au_sync();
-
-	/* Wait for Device ready.
-	*/
-	do {
-		stat = au_readl(PSC_AC97STAT);
-		au_sync();
-	} while ((stat & PSC_AC97STAT_DR) == 0);
-}
-
-/* Hold spinlock for both start_dac() and start_adc() calls */
-static void
-start_dac(struct au1550_state *s)
-{
-	struct dmabuf  *db = &s->dma_dac;
-
-	if (!db->stopped)
-		return;
-
-	set_xmit_slots(db->num_channels);
-	au_writel(PSC_AC97PCR_TC, PSC_AC97PCR);
-	au_sync();
-	au_writel(PSC_AC97PCR_TS, PSC_AC97PCR);
-	au_sync();
-
-	au1xxx_dbdma_start(db->dmanr);
-
-	db->stopped = 0;
-}
-
-static void
-start_adc(struct au1550_state *s)
-{
-	struct dmabuf  *db = &s->dma_adc;
-	int	i;
-
-	if (!db->stopped)
-		return;
-
-	/* Put two buffers on the ring to get things started.
-	*/
-	for (i=0; i<2; i++) {
-		au1xxx_dbdma_put_dest(db->dmanr, virt_to_phys(db->nextIn),
-				db->dma_fragsize, DDMA_FLAGS_IE);
-
-		db->nextIn += db->dma_fragsize;
-		if (db->nextIn >= db->rawbuf + db->dmasize)
-			db->nextIn -= db->dmasize;
-	}
-
-	set_recv_slots(db->num_channels);
-	au1xxx_dbdma_start(db->dmanr);
-	au_writel(PSC_AC97PCR_RC, PSC_AC97PCR);
-	au_sync();
-	au_writel(PSC_AC97PCR_RS, PSC_AC97PCR);
-	au_sync();
-
-	db->stopped = 0;
-}
-
-static int
-prog_dmabuf(struct au1550_state *s, struct dmabuf *db)
-{
-	unsigned user_bytes_per_sec;
-	unsigned        bufs;
-	unsigned        rate = db->sample_rate;
-
-	if (!db->rawbuf) {
-		db->ready = db->mapped = 0;
-		db->buforder = 5;	/* 32 * PAGE_SIZE */
-		db->rawbuf = kmalloc((PAGE_SIZE << db->buforder), GFP_KERNEL);
-		if (!db->rawbuf)
-			return -ENOMEM;
-	}
-
-	db->cnt_factor = 1;
-	if (db->sample_size == 8)
-		db->cnt_factor *= 2;
-	if (db->num_channels == 1)
-		db->cnt_factor *= 2;
-	db->cnt_factor *= db->src_factor;
-
-	db->count = 0;
-	db->dma_qcount = 0;
-	db->nextIn = db->nextOut = db->rawbuf;
-
-	db->user_bytes_per_sample = (db->sample_size>>3) * db->num_channels;
-	db->dma_bytes_per_sample = 2 * ((db->num_channels == 1) ?
-					2 : db->num_channels);
-
-	user_bytes_per_sec = rate * db->user_bytes_per_sample;
-	bufs = PAGE_SIZE << db->buforder;
-	if (db->ossfragshift) {
-		if ((1000 << db->ossfragshift) < user_bytes_per_sec)
-			db->fragshift = ld2(user_bytes_per_sec/1000);
-		else
-			db->fragshift = db->ossfragshift;
-	} else {
-		db->fragshift = ld2(user_bytes_per_sec / 100 /
-				    (db->subdivision ? db->subdivision : 1));
-		if (db->fragshift < 3)
-			db->fragshift = 3;
-	}
-
-	db->fragsize = 1 << db->fragshift;
-	db->dma_fragsize = db->fragsize * db->cnt_factor;
-	db->numfrag = bufs / db->dma_fragsize;
-
-	while (db->numfrag < 4 && db->fragshift > 3) {
-		db->fragshift--;
-		db->fragsize = 1 << db->fragshift;
-		db->dma_fragsize = db->fragsize * db->cnt_factor;
-		db->numfrag = bufs / db->dma_fragsize;
-	}
-
-	if (db->ossmaxfrags >= 4 && db->ossmaxfrags < db->numfrag)
-		db->numfrag = db->ossmaxfrags;
-
-	db->dmasize = db->dma_fragsize * db->numfrag;
-	memset(db->rawbuf, 0, bufs);
-
-	pr_debug("prog_dmabuf: rate=%d, samplesize=%d, channels=%d\n",
-	    rate, db->sample_size, db->num_channels);
-	pr_debug("prog_dmabuf: fragsize=%d, cnt_factor=%d, dma_fragsize=%d\n",
-	    db->fragsize, db->cnt_factor, db->dma_fragsize);
-	pr_debug("prog_dmabuf: numfrag=%d, dmasize=%d\n", db->numfrag, db->dmasize);
-
-	db->ready = 1;
-	return 0;
-}
-
-static int
-prog_dmabuf_adc(struct au1550_state *s)
-{
-	stop_adc(s);
-	return prog_dmabuf(s, &s->dma_adc);
-
-}
-
-static int
-prog_dmabuf_dac(struct au1550_state *s)
-{
-	stop_dac(s);
-	return prog_dmabuf(s, &s->dma_dac);
-}
-
-
-static void dac_dma_interrupt(int irq, void *dev_id)
-{
-	struct au1550_state *s = (struct au1550_state *) dev_id;
-	struct dmabuf  *db = &s->dma_dac;
-	u32	ac97c_stat;
-
-	spin_lock(&s->lock);
-
-	ac97c_stat = au_readl(PSC_AC97STAT);
-	if (ac97c_stat & (AC97C_XU | AC97C_XO | AC97C_TE))
-		pr_debug("AC97C status = 0x%08x\n", ac97c_stat);
-	db->dma_qcount--;
-
-	if (db->count >= db->fragsize) {
-		if (au1xxx_dbdma_put_source(db->dmanr,
-				virt_to_phys(db->nextOut), db->fragsize,
-				DDMA_FLAGS_IE) == 0) {
-			err("qcount < 2 and no ring room!");
-		}
-		db->nextOut += db->fragsize;
-		if (db->nextOut >= db->rawbuf + db->dmasize)
-			db->nextOut -= db->dmasize;
-		db->count -= db->fragsize;
-		db->total_bytes += db->dma_fragsize;
-		db->dma_qcount++;
-	}
-
-	/* wake up anybody listening */
-	if (waitqueue_active(&db->wait))
-		wake_up(&db->wait);
-
-	spin_unlock(&s->lock);
-}
-
-
-static void adc_dma_interrupt(int irq, void *dev_id)
-{
-	struct	au1550_state *s = (struct au1550_state *)dev_id;
-	struct	dmabuf  *dp = &s->dma_adc;
-	u32	obytes;
-	char	*obuf;
-
-	spin_lock(&s->lock);
-
-	/* Pull the buffer from the dma queue.
-	*/
-	au1xxx_dbdma_get_dest(dp->dmanr, (void *)(&obuf), &obytes);
-
-	if ((dp->count + obytes) > dp->dmasize) {
-		/* Overrun. Stop ADC and log the error
-		*/
-		spin_unlock(&s->lock);
-		stop_adc(s);
-		dp->error++;
-		err("adc overrun");
-		return;
-	}
-
-	/* Put a new empty buffer on the destination DMA.
-	*/
-	au1xxx_dbdma_put_dest(dp->dmanr, virt_to_phys(dp->nextIn),
-			      dp->dma_fragsize, DDMA_FLAGS_IE);
-
-	dp->nextIn += dp->dma_fragsize;
-	if (dp->nextIn >= dp->rawbuf + dp->dmasize)
-		dp->nextIn -= dp->dmasize;
-
-	dp->count += obytes;
-	dp->total_bytes += obytes;
-
-	/* wake up anybody listening
-	*/
-	if (waitqueue_active(&dp->wait))
-		wake_up(&dp->wait);
-
-	spin_unlock(&s->lock);
-}
-
-static loff_t
-au1550_llseek(struct file *file, loff_t offset, int origin)
-{
-	return -ESPIPE;
-}
-
-
-static int
-au1550_open_mixdev(struct inode *inode, struct file *file)
-{
-	mutex_lock(&au1550_ac97_mutex);
-	file->private_data = &au1550_state;
-	mutex_unlock(&au1550_ac97_mutex);
-	return 0;
-}
-
-static int
-au1550_release_mixdev(struct inode *inode, struct file *file)
-{
-	return 0;
-}
-
-static int
-mixdev_ioctl(struct ac97_codec *codec, unsigned int cmd,
-                        unsigned long arg)
-{
-	return codec->mixer_ioctl(codec, cmd, arg);
-}
-
-static long
-au1550_ioctl_mixdev(struct file *file, unsigned int cmd, unsigned long arg)
-{
-	struct au1550_state *s = file->private_data;
-	struct ac97_codec *codec = s->codec;
-	int ret;
-
-	mutex_lock(&au1550_ac97_mutex);
-	ret = mixdev_ioctl(codec, cmd, arg);
-	mutex_unlock(&au1550_ac97_mutex);
-
-	return ret;
-}
-
-static /*const */ struct file_operations au1550_mixer_fops = {
-	.owner		= THIS_MODULE,
-	.llseek		= au1550_llseek,
-	.unlocked_ioctl	= au1550_ioctl_mixdev,
-	.open		= au1550_open_mixdev,
-	.release	= au1550_release_mixdev,
-};
-
-static int
-drain_dac(struct au1550_state *s, int nonblock)
-{
-	unsigned long   flags;
-	int             count, tmo;
-
-	if (s->dma_dac.mapped || !s->dma_dac.ready || s->dma_dac.stopped)
-		return 0;
-
-	for (;;) {
-		spin_lock_irqsave(&s->lock, flags);
-		count = s->dma_dac.count;
-		spin_unlock_irqrestore(&s->lock, flags);
-		if (count <= s->dma_dac.fragsize)
-			break;
-		if (signal_pending(current))
-			break;
-		if (nonblock)
-			return -EBUSY;
-		tmo = 1000 * count / (s->no_vra ?
-				      48000 : s->dma_dac.sample_rate);
-		tmo /= s->dma_dac.dma_bytes_per_sample;
-		au1550_delay(tmo);
-	}
-	if (signal_pending(current))
-		return -ERESTARTSYS;
-	return 0;
-}
-
-static inline u8 S16_TO_U8(s16 ch)
-{
-	return (u8) (ch >> 8) + 0x80;
-}
-static inline s16 U8_TO_S16(u8 ch)
-{
-	return (s16) (ch - 0x80) << 8;
-}
-
-/*
- * Translates user samples to dma buffer suitable for AC'97 DAC data:
- *     If mono, copy left channel to right channel in dma buffer.
- *     If 8 bit samples, cvt to 16-bit before writing to dma buffer.
- *     If interpolating (no VRA), duplicate every audio frame src_factor times.
- */
-static int
-translate_from_user(struct dmabuf *db, char* dmabuf, char* userbuf,
-							       int dmacount)
-{
-	int             sample, i;
-	int             interp_bytes_per_sample;
-	int             num_samples;
-	int             mono = (db->num_channels == 1);
-	char            usersample[12];
-	s16             ch, dmasample[6];
-
-	if (db->sample_size == 16 && !mono && db->src_factor == 1) {
-		/* no translation necessary, just copy
-		*/
-		if (copy_from_user(dmabuf, userbuf, dmacount))
-			return -EFAULT;
-		return dmacount;
-	}
-
-	interp_bytes_per_sample = db->dma_bytes_per_sample * db->src_factor;
-	num_samples = dmacount / interp_bytes_per_sample;
-
-	for (sample = 0; sample < num_samples; sample++) {
-		if (copy_from_user(usersample, userbuf,
-				   db->user_bytes_per_sample)) {
-			return -EFAULT;
-		}
-
-		for (i = 0; i < db->num_channels; i++) {
-			if (db->sample_size == 8)
-				ch = U8_TO_S16(usersample[i]);
-			else
-				ch = *((s16 *) (&usersample[i * 2]));
-			dmasample[i] = ch;
-			if (mono)
-				dmasample[i + 1] = ch;	/* right channel */
-		}
-
-		/* duplicate every audio frame src_factor times
-		*/
-		for (i = 0; i < db->src_factor; i++)
-			memcpy(dmabuf, dmasample, db->dma_bytes_per_sample);
-
-		userbuf += db->user_bytes_per_sample;
-		dmabuf += interp_bytes_per_sample;
-	}
-
-	return num_samples * interp_bytes_per_sample;
-}
-
-/*
- * Translates AC'97 ADC samples to user buffer:
- *     If mono, send only left channel to user buffer.
- *     If 8 bit samples, cvt from 16 to 8 bit before writing to user buffer.
- *     If decimating (no VRA), skip over src_factor audio frames.
- */
-static int
-translate_to_user(struct dmabuf *db, char* userbuf, char* dmabuf,
-							     int dmacount)
-{
-	int             sample, i;
-	int             interp_bytes_per_sample;
-	int             num_samples;
-	int             mono = (db->num_channels == 1);
-	char            usersample[12];
-
-	if (db->sample_size == 16 && !mono && db->src_factor == 1) {
-		/* no translation necessary, just copy
-		*/
-		if (copy_to_user(userbuf, dmabuf, dmacount))
-			return -EFAULT;
-		return dmacount;
-	}
-
-	interp_bytes_per_sample = db->dma_bytes_per_sample * db->src_factor;
-	num_samples = dmacount / interp_bytes_per_sample;
-
-	for (sample = 0; sample < num_samples; sample++) {
-		for (i = 0; i < db->num_channels; i++) {
-			if (db->sample_size == 8)
-				usersample[i] =
-					S16_TO_U8(*((s16 *) (&dmabuf[i * 2])));
-			else
-				*((s16 *) (&usersample[i * 2])) =
-					*((s16 *) (&dmabuf[i * 2]));
-		}
-
-		if (copy_to_user(userbuf, usersample,
-				 db->user_bytes_per_sample)) {
-			return -EFAULT;
-		}
-
-		userbuf += db->user_bytes_per_sample;
-		dmabuf += interp_bytes_per_sample;
-	}
-
-	return num_samples * interp_bytes_per_sample;
-}
-
-/*
- * Copy audio data to/from user buffer from/to dma buffer, taking care
- * that we wrap when reading/writing the dma buffer. Returns actual byte
- * count written to or read from the dma buffer.
- */
-static int
-copy_dmabuf_user(struct dmabuf *db, char* userbuf, int count, int to_user)
-{
-	char           *bufptr = to_user ? db->nextOut : db->nextIn;
-	char           *bufend = db->rawbuf + db->dmasize;
-	int             cnt, ret;
-
-	if (bufptr + count > bufend) {
-		int             partial = (int) (bufend - bufptr);
-		if (to_user) {
-			if ((cnt = translate_to_user(db, userbuf,
-						     bufptr, partial)) < 0)
-				return cnt;
-			ret = cnt;
-			if ((cnt = translate_to_user(db, userbuf + partial,
-						     db->rawbuf,
-						     count - partial)) < 0)
-				return cnt;
-			ret += cnt;
-		} else {
-			if ((cnt = translate_from_user(db, bufptr, userbuf,
-						       partial)) < 0)
-				return cnt;
-			ret = cnt;
-			if ((cnt = translate_from_user(db, db->rawbuf,
-						       userbuf + partial,
-						       count - partial)) < 0)
-				return cnt;
-			ret += cnt;
-		}
-	} else {
-		if (to_user)
-			ret = translate_to_user(db, userbuf, bufptr, count);
-		else
-			ret = translate_from_user(db, bufptr, userbuf, count);
-	}
-
-	return ret;
-}
-
-
-static ssize_t
-au1550_read(struct file *file, char *buffer, size_t count, loff_t *ppos)
-{
-	struct au1550_state *s = file->private_data;
-	struct dmabuf  *db = &s->dma_adc;
-	DECLARE_WAITQUEUE(wait, current);
-	ssize_t         ret;
-	unsigned long   flags;
-	int             cnt, usercnt, avail;
-
-	if (db->mapped)
-		return -ENXIO;
-	if (!access_ok(VERIFY_WRITE, buffer, count))
-		return -EFAULT;
-	ret = 0;
-
-	count *= db->cnt_factor;
-
-	mutex_lock(&s->sem);
-	add_wait_queue(&db->wait, &wait);
-
-	while (count > 0) {
-		/* wait for samples in ADC dma buffer
-		*/
-		do {
-			spin_lock_irqsave(&s->lock, flags);
-			if (db->stopped)
-				start_adc(s);
-			avail = db->count;
-			if (avail <= 0)
-				__set_current_state(TASK_INTERRUPTIBLE);
-			spin_unlock_irqrestore(&s->lock, flags);
-			if (avail <= 0) {
-				if (file->f_flags & O_NONBLOCK) {
-					if (!ret)
-						ret = -EAGAIN;
-					goto out;
-				}
-				mutex_unlock(&s->sem);
-				schedule();
-				if (signal_pending(current)) {
-					if (!ret)
-						ret = -ERESTARTSYS;
-					goto out2;
-				}
-				mutex_lock(&s->sem);
-			}
-		} while (avail <= 0);
-
-		/* copy from nextOut to user
-		*/
-		if ((cnt = copy_dmabuf_user(db, buffer,
-					    count > avail ?
-					    avail : count, 1)) < 0) {
-			if (!ret)
-				ret = -EFAULT;
-			goto out;
-		}
-
-		spin_lock_irqsave(&s->lock, flags);
-		db->count -= cnt;
-		db->nextOut += cnt;
-		if (db->nextOut >= db->rawbuf + db->dmasize)
-			db->nextOut -= db->dmasize;
-		spin_unlock_irqrestore(&s->lock, flags);
-
-		count -= cnt;
-		usercnt = cnt / db->cnt_factor;
-		buffer += usercnt;
-		ret += usercnt;
-	}			/* while (count > 0) */
-
-out:
-	mutex_unlock(&s->sem);
-out2:
-	remove_wait_queue(&db->wait, &wait);
-	set_current_state(TASK_RUNNING);
-	return ret;
-}
-
-static ssize_t
-au1550_write(struct file *file, const char *buffer, size_t count, loff_t * ppos)
-{
-	struct au1550_state *s = file->private_data;
-	struct dmabuf  *db = &s->dma_dac;
-	DECLARE_WAITQUEUE(wait, current);
-	ssize_t         ret = 0;
-	unsigned long   flags;
-	int             cnt, usercnt, avail;
-
-	pr_debug("write: count=%d\n", count);
-
-	if (db->mapped)
-		return -ENXIO;
-	if (!access_ok(VERIFY_READ, buffer, count))
-		return -EFAULT;
-
-	count *= db->cnt_factor;
-
-	mutex_lock(&s->sem);
-	add_wait_queue(&db->wait, &wait);
-
-	while (count > 0) {
-		/* wait for space in playback buffer
-		*/
-		do {
-			spin_lock_irqsave(&s->lock, flags);
-			avail = (int) db->dmasize - db->count;
-			if (avail <= 0)
-				__set_current_state(TASK_INTERRUPTIBLE);
-			spin_unlock_irqrestore(&s->lock, flags);
-			if (avail <= 0) {
-				if (file->f_flags & O_NONBLOCK) {
-					if (!ret)
-						ret = -EAGAIN;
-					goto out;
-				}
-				mutex_unlock(&s->sem);
-				schedule();
-				if (signal_pending(current)) {
-					if (!ret)
-						ret = -ERESTARTSYS;
-					goto out2;
-				}
-				mutex_lock(&s->sem);
-			}
-		} while (avail <= 0);
-
-		/* copy from user to nextIn
-		*/
-		if ((cnt = copy_dmabuf_user(db, (char *) buffer,
-					    count > avail ?
-					    avail : count, 0)) < 0) {
-			if (!ret)
-				ret = -EFAULT;
-			goto out;
-		}
-
-		spin_lock_irqsave(&s->lock, flags);
-		db->count += cnt;
-		db->nextIn += cnt;
-		if (db->nextIn >= db->rawbuf + db->dmasize)
-			db->nextIn -= db->dmasize;
-
-		/* If the data is available, we want to keep two buffers
-		 * on the dma queue.  If the queue count reaches zero,
-		 * we know the dma has stopped.
-		 */
-		while ((db->dma_qcount < 2) && (db->count >= db->fragsize)) {
-			if (au1xxx_dbdma_put_source(db->dmanr,
-				virt_to_phys(db->nextOut), db->fragsize,
-				DDMA_FLAGS_IE) == 0) {
-				err("qcount < 2 and no ring room!");
-			}
-			db->nextOut += db->fragsize;
-			if (db->nextOut >= db->rawbuf + db->dmasize)
-				db->nextOut -= db->dmasize;
-			db->total_bytes += db->dma_fragsize;
-			if (db->dma_qcount == 0)
-				start_dac(s);
-			db->dma_qcount++;
-		}
-		spin_unlock_irqrestore(&s->lock, flags);
-
-		count -= cnt;
-		usercnt = cnt / db->cnt_factor;
-		buffer += usercnt;
-		ret += usercnt;
-	}			/* while (count > 0) */
-
-out:
-	mutex_unlock(&s->sem);
-out2:
-	remove_wait_queue(&db->wait, &wait);
-	set_current_state(TASK_RUNNING);
-	return ret;
-}
-
-
-/* No kernel lock - we have our own spinlock */
-static unsigned int
-au1550_poll(struct file *file, struct poll_table_struct *wait)
-{
-	struct au1550_state *s = file->private_data;
-	unsigned long   flags;
-	unsigned int    mask = 0;
-
-	if (file->f_mode & FMODE_WRITE) {
-		if (!s->dma_dac.ready)
-			return 0;
-		poll_wait(file, &s->dma_dac.wait, wait);
-	}
-	if (file->f_mode & FMODE_READ) {
-		if (!s->dma_adc.ready)
-			return 0;
-		poll_wait(file, &s->dma_adc.wait, wait);
-	}
-
-	spin_lock_irqsave(&s->lock, flags);
-
-	if (file->f_mode & FMODE_READ) {
-		if (s->dma_adc.count >= (signed)s->dma_adc.dma_fragsize)
-			mask |= POLLIN | POLLRDNORM;
-	}
-	if (file->f_mode & FMODE_WRITE) {
-		if (s->dma_dac.mapped) {
-			if (s->dma_dac.count >=
-			    (signed)s->dma_dac.dma_fragsize)
-				mask |= POLLOUT | POLLWRNORM;
-		} else {
-			if ((signed) s->dma_dac.dmasize >=
-			    s->dma_dac.count + (signed)s->dma_dac.dma_fragsize)
-				mask |= POLLOUT | POLLWRNORM;
-		}
-	}
-	spin_unlock_irqrestore(&s->lock, flags);
-	return mask;
-}
-
-static int
-au1550_mmap(struct file *file, struct vm_area_struct *vma)
-{
-	struct au1550_state *s = file->private_data;
-	struct dmabuf  *db;
-	unsigned long   size;
-	int ret = 0;
-
-	mutex_lock(&au1550_ac97_mutex);
-	mutex_lock(&s->sem);
-	if (vma->vm_flags & VM_WRITE)
-		db = &s->dma_dac;
-	else if (vma->vm_flags & VM_READ)
-		db = &s->dma_adc;
-	else {
-		ret = -EINVAL;
-		goto out;
-	}
-	if (vma->vm_pgoff != 0) {
-		ret = -EINVAL;
-		goto out;
-	}
-	size = vma->vm_end - vma->vm_start;
-	if (size > (PAGE_SIZE << db->buforder)) {
-		ret = -EINVAL;
-		goto out;
-	}
-	if (remap_pfn_range(vma, vma->vm_start, page_to_pfn(virt_to_page(db->rawbuf)),
-			     size, vma->vm_page_prot)) {
-		ret = -EAGAIN;
-		goto out;
-	}
-	vma->vm_flags &= ~VM_IO;
-	db->mapped = 1;
-out:
-	mutex_unlock(&s->sem);
-	mutex_unlock(&au1550_ac97_mutex);
-	return ret;
-}
-
-#ifdef DEBUG
-static struct ioctl_str_t {
-	unsigned int    cmd;
-	const char     *str;
-} ioctl_str[] = {
-	{SNDCTL_DSP_RESET, "SNDCTL_DSP_RESET"},
-	{SNDCTL_DSP_SYNC, "SNDCTL_DSP_SYNC"},
-	{SNDCTL_DSP_SPEED, "SNDCTL_DSP_SPEED"},
-	{SNDCTL_DSP_STEREO, "SNDCTL_DSP_STEREO"},
-	{SNDCTL_DSP_GETBLKSIZE, "SNDCTL_DSP_GETBLKSIZE"},
-	{SNDCTL_DSP_SAMPLESIZE, "SNDCTL_DSP_SAMPLESIZE"},
-	{SNDCTL_DSP_CHANNELS, "SNDCTL_DSP_CHANNELS"},
-	{SOUND_PCM_WRITE_CHANNELS, "SOUND_PCM_WRITE_CHANNELS"},
-	{SOUND_PCM_WRITE_FILTER, "SOUND_PCM_WRITE_FILTER"},
-	{SNDCTL_DSP_POST, "SNDCTL_DSP_POST"},
-	{SNDCTL_DSP_SUBDIVIDE, "SNDCTL_DSP_SUBDIVIDE"},
-	{SNDCTL_DSP_SETFRAGMENT, "SNDCTL_DSP_SETFRAGMENT"},
-	{SNDCTL_DSP_GETFMTS, "SNDCTL_DSP_GETFMTS"},
-	{SNDCTL_DSP_SETFMT, "SNDCTL_DSP_SETFMT"},
-	{SNDCTL_DSP_GETOSPACE, "SNDCTL_DSP_GETOSPACE"},
-	{SNDCTL_DSP_GETISPACE, "SNDCTL_DSP_GETISPACE"},
-	{SNDCTL_DSP_NONBLOCK, "SNDCTL_DSP_NONBLOCK"},
-	{SNDCTL_DSP_GETCAPS, "SNDCTL_DSP_GETCAPS"},
-	{SNDCTL_DSP_GETTRIGGER, "SNDCTL_DSP_GETTRIGGER"},
-	{SNDCTL_DSP_SETTRIGGER, "SNDCTL_DSP_SETTRIGGER"},
-	{SNDCTL_DSP_GETIPTR, "SNDCTL_DSP_GETIPTR"},
-	{SNDCTL_DSP_GETOPTR, "SNDCTL_DSP_GETOPTR"},
-	{SNDCTL_DSP_MAPINBUF, "SNDCTL_DSP_MAPINBUF"},
-	{SNDCTL_DSP_MAPOUTBUF, "SNDCTL_DSP_MAPOUTBUF"},
-	{SNDCTL_DSP_SETSYNCRO, "SNDCTL_DSP_SETSYNCRO"},
-	{SNDCTL_DSP_SETDUPLEX, "SNDCTL_DSP_SETDUPLEX"},
-	{SNDCTL_DSP_GETODELAY, "SNDCTL_DSP_GETODELAY"},
-	{SNDCTL_DSP_GETCHANNELMASK, "SNDCTL_DSP_GETCHANNELMASK"},
-	{SNDCTL_DSP_BIND_CHANNEL, "SNDCTL_DSP_BIND_CHANNEL"},
-	{OSS_GETVERSION, "OSS_GETVERSION"},
-	{SOUND_PCM_READ_RATE, "SOUND_PCM_READ_RATE"},
-	{SOUND_PCM_READ_CHANNELS, "SOUND_PCM_READ_CHANNELS"},
-	{SOUND_PCM_READ_BITS, "SOUND_PCM_READ_BITS"},
-	{SOUND_PCM_READ_FILTER, "SOUND_PCM_READ_FILTER"}
-};
-#endif
-
-static int
-dma_count_done(struct dmabuf *db)
-{
-	if (db->stopped)
-		return 0;
-
-	return db->dma_fragsize - au1xxx_get_dma_residue(db->dmanr);
-}
-
-
-static int
-au1550_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
-{
-	struct au1550_state *s = file->private_data;
-	unsigned long   flags;
-	audio_buf_info  abinfo;
-	count_info      cinfo;
-	int             count;
-	int             val, mapped, ret, diff;
-
-	mapped = ((file->f_mode & FMODE_WRITE) && s->dma_dac.mapped) ||
-		((file->f_mode & FMODE_READ) && s->dma_adc.mapped);
-
-#ifdef DEBUG
-	for (count = 0; count < ARRAY_SIZE(ioctl_str); count++) {
-		if (ioctl_str[count].cmd == cmd)
-			break;
-	}
-	if (count < ARRAY_SIZE(ioctl_str))
-		pr_debug("ioctl %s, arg=0x%lxn", ioctl_str[count].str, arg);
-	else
-		pr_debug("ioctl 0x%x unknown, arg=0x%lx\n", cmd, arg);
-#endif
-
-	switch (cmd) {
-	case OSS_GETVERSION:
-		return put_user(SOUND_VERSION, (int *) arg);
-
-	case SNDCTL_DSP_SYNC:
-		if (file->f_mode & FMODE_WRITE)
-			return drain_dac(s, file->f_flags & O_NONBLOCK);
-		return 0;
-
-	case SNDCTL_DSP_SETDUPLEX:
-		return 0;
-
-	case SNDCTL_DSP_GETCAPS:
-		return put_user(DSP_CAP_DUPLEX | DSP_CAP_REALTIME |
-				DSP_CAP_TRIGGER | DSP_CAP_MMAP, (int *)arg);
-
-	case SNDCTL_DSP_RESET:
-		if (file->f_mode & FMODE_WRITE) {
-			stop_dac(s);
-			synchronize_irq();
-			s->dma_dac.count = s->dma_dac.total_bytes = 0;
-			s->dma_dac.nextIn = s->dma_dac.nextOut =
-				s->dma_dac.rawbuf;
-		}
-		if (file->f_mode & FMODE_READ) {
-			stop_adc(s);
-			synchronize_irq();
-			s->dma_adc.count = s->dma_adc.total_bytes = 0;
-			s->dma_adc.nextIn = s->dma_adc.nextOut =
-				s->dma_adc.rawbuf;
-		}
-		return 0;
-
-	case SNDCTL_DSP_SPEED:
-		if (get_user(val, (int *) arg))
-			return -EFAULT;
-		if (val >= 0) {
-			if (file->f_mode & FMODE_READ) {
-				stop_adc(s);
-				set_adc_rate(s, val);
-			}
-			if (file->f_mode & FMODE_WRITE) {
-				stop_dac(s);
-				set_dac_rate(s, val);
-			}
-			if (s->open_mode & FMODE_READ)
-				if ((ret = prog_dmabuf_adc(s)))
-					return ret;
-			if (s->open_mode & FMODE_WRITE)
-				if ((ret = prog_dmabuf_dac(s)))
-					return ret;
-		}
-		return put_user((file->f_mode & FMODE_READ) ?
-				s->dma_adc.sample_rate :
-				s->dma_dac.sample_rate,
-				(int *)arg);
-
-	case SNDCTL_DSP_STEREO:
-		if (get_user(val, (int *) arg))
-			return -EFAULT;
-		if (file->f_mode & FMODE_READ) {
-			stop_adc(s);
-			s->dma_adc.num_channels = val ? 2 : 1;
-			if ((ret = prog_dmabuf_adc(s)))
-				return ret;
-		}
-		if (file->f_mode & FMODE_WRITE) {
-			stop_dac(s);
-			s->dma_dac.num_channels = val ? 2 : 1;
-			if (s->codec_ext_caps & AC97_EXT_DACS) {
-				/* disable surround and center/lfe in AC'97
-				*/
-				u16 ext_stat = rdcodec(s->codec,
-						       AC97_EXTENDED_STATUS);
-				wrcodec(s->codec, AC97_EXTENDED_STATUS,
-					ext_stat | (AC97_EXTSTAT_PRI |
-						    AC97_EXTSTAT_PRJ |
-						    AC97_EXTSTAT_PRK));
-			}
-			if ((ret = prog_dmabuf_dac(s)))
-				return ret;
-		}
-		return 0;
-
-	case SNDCTL_DSP_CHANNELS:
-		if (get_user(val, (int *) arg))
-			return -EFAULT;
-		if (val != 0) {
-			if (file->f_mode & FMODE_READ) {
-				if (val < 0 || val > 2)
-					return -EINVAL;
-				stop_adc(s);
-				s->dma_adc.num_channels = val;
-				if ((ret = prog_dmabuf_adc(s)))
-					return ret;
-			}
-			if (file->f_mode & FMODE_WRITE) {
-				switch (val) {
-				case 1:
-				case 2:
-					break;
-				case 3:
-				case 5:
-					return -EINVAL;
-				case 4:
-					if (!(s->codec_ext_caps &
-					      AC97_EXTID_SDAC))
-						return -EINVAL;
-					break;
-				case 6:
-					if ((s->codec_ext_caps &
-					     AC97_EXT_DACS) != AC97_EXT_DACS)
-						return -EINVAL;
-					break;
-				default:
-					return -EINVAL;
-				}
-
-				stop_dac(s);
-				if (val <= 2 &&
-				    (s->codec_ext_caps & AC97_EXT_DACS)) {
-					/* disable surround and center/lfe
-					 * channels in AC'97
-					 */
-					u16             ext_stat =
-						rdcodec(s->codec,
-							AC97_EXTENDED_STATUS);
-					wrcodec(s->codec,
-						AC97_EXTENDED_STATUS,
-						ext_stat | (AC97_EXTSTAT_PRI |
-							    AC97_EXTSTAT_PRJ |
-							    AC97_EXTSTAT_PRK));
-				} else if (val >= 4) {
-					/* enable surround, center/lfe
-					 * channels in AC'97
-					 */
-					u16             ext_stat =
-						rdcodec(s->codec,
-							AC97_EXTENDED_STATUS);
-					ext_stat &= ~AC97_EXTSTAT_PRJ;
-					if (val == 6)
-						ext_stat &=
-							~(AC97_EXTSTAT_PRI |
-							  AC97_EXTSTAT_PRK);
-					wrcodec(s->codec,
-						AC97_EXTENDED_STATUS,
-						ext_stat);
-				}
-
-				s->dma_dac.num_channels = val;
-				if ((ret = prog_dmabuf_dac(s)))
-					return ret;
-			}
-		}
-		return put_user(val, (int *) arg);
-
-	case SNDCTL_DSP_GETFMTS:	/* Returns a mask */
-		return put_user(AFMT_S16_LE | AFMT_U8, (int *) arg);
-
-	case SNDCTL_DSP_SETFMT:	/* Selects ONE fmt */
-		if (get_user(val, (int *) arg))
-			return -EFAULT;
-		if (val != AFMT_QUERY) {
-			if (file->f_mode & FMODE_READ) {
-				stop_adc(s);
-				if (val == AFMT_S16_LE)
-					s->dma_adc.sample_size = 16;
-				else {
-					val = AFMT_U8;
-					s->dma_adc.sample_size = 8;
-				}
-				if ((ret = prog_dmabuf_adc(s)))
-					return ret;
-			}
-			if (file->f_mode & FMODE_WRITE) {
-				stop_dac(s);
-				if (val == AFMT_S16_LE)
-					s->dma_dac.sample_size = 16;
-				else {
-					val = AFMT_U8;
-					s->dma_dac.sample_size = 8;
-				}
-				if ((ret = prog_dmabuf_dac(s)))
-					return ret;
-			}
-		} else {
-			if (file->f_mode & FMODE_READ)
-				val = (s->dma_adc.sample_size == 16) ?
-					AFMT_S16_LE : AFMT_U8;
-			else
-				val = (s->dma_dac.sample_size == 16) ?
-					AFMT_S16_LE : AFMT_U8;
-		}
-		return put_user(val, (int *) arg);
-
-	case SNDCTL_DSP_POST:
-		return 0;
-
-	case SNDCTL_DSP_GETTRIGGER:
-		val = 0;
-		spin_lock_irqsave(&s->lock, flags);
-		if (file->f_mode & FMODE_READ && !s->dma_adc.stopped)
-			val |= PCM_ENABLE_INPUT;
-		if (file->f_mode & FMODE_WRITE && !s->dma_dac.stopped)
-			val |= PCM_ENABLE_OUTPUT;
-		spin_unlock_irqrestore(&s->lock, flags);
-		return put_user(val, (int *) arg);
-
-	case SNDCTL_DSP_SETTRIGGER:
-		if (get_user(val, (int *) arg))
-			return -EFAULT;
-		if (file->f_mode & FMODE_READ) {
-			if (val & PCM_ENABLE_INPUT) {
-				spin_lock_irqsave(&s->lock, flags);
-				start_adc(s);
-				spin_unlock_irqrestore(&s->lock, flags);
-			} else
-				stop_adc(s);
-		}
-		if (file->f_mode & FMODE_WRITE) {
-			if (val & PCM_ENABLE_OUTPUT) {
-				spin_lock_irqsave(&s->lock, flags);
-				start_dac(s);
-				spin_unlock_irqrestore(&s->lock, flags);
-			} else
-				stop_dac(s);
-		}
-		return 0;
-
-	case SNDCTL_DSP_GETOSPACE:
-		if (!(file->f_mode & FMODE_WRITE))
-			return -EINVAL;
-		abinfo.fragsize = s->dma_dac.fragsize;
-		spin_lock_irqsave(&s->lock, flags);
-		count = s->dma_dac.count;
-		count -= dma_count_done(&s->dma_dac);
-		spin_unlock_irqrestore(&s->lock, flags);
-		if (count < 0)
-			count = 0;
-		abinfo.bytes = (s->dma_dac.dmasize - count) /
-			s->dma_dac.cnt_factor;
-		abinfo.fragstotal = s->dma_dac.numfrag;
-		abinfo.fragments = abinfo.bytes >> s->dma_dac.fragshift;
-		pr_debug("ioctl SNDCTL_DSP_GETOSPACE: bytes=%d, fragments=%d\n", abinfo.bytes, abinfo.fragments);
-		return copy_to_user((void *) arg, &abinfo,
-				    sizeof(abinfo)) ? -EFAULT : 0;
-
-	case SNDCTL_DSP_GETISPACE:
-		if (!(file->f_mode & FMODE_READ))
-			return -EINVAL;
-		abinfo.fragsize = s->dma_adc.fragsize;
-		spin_lock_irqsave(&s->lock, flags);
-		count = s->dma_adc.count;
-		count += dma_count_done(&s->dma_adc);
-		spin_unlock_irqrestore(&s->lock, flags);
-		if (count < 0)
-			count = 0;
-		abinfo.bytes = count / s->dma_adc.cnt_factor;
-		abinfo.fragstotal = s->dma_adc.numfrag;
-		abinfo.fragments = abinfo.bytes >> s->dma_adc.fragshift;
-		return copy_to_user((void *) arg, &abinfo,
-				    sizeof(abinfo)) ? -EFAULT : 0;
-
-	case SNDCTL_DSP_NONBLOCK:
-		spin_lock(&file->f_lock);
-		file->f_flags |= O_NONBLOCK;
-		spin_unlock(&file->f_lock);
-		return 0;
-
-	case SNDCTL_DSP_GETODELAY:
-		if (!(file->f_mode & FMODE_WRITE))
-			return -EINVAL;
-		spin_lock_irqsave(&s->lock, flags);
-		count = s->dma_dac.count;
-		count -= dma_count_done(&s->dma_dac);
-		spin_unlock_irqrestore(&s->lock, flags);
-		if (count < 0)
-			count = 0;
-		count /= s->dma_dac.cnt_factor;
-		return put_user(count, (int *) arg);
-
-	case SNDCTL_DSP_GETIPTR:
-		if (!(file->f_mode & FMODE_READ))
-			return -EINVAL;
-		spin_lock_irqsave(&s->lock, flags);
-		cinfo.bytes = s->dma_adc.total_bytes;
-		count = s->dma_adc.count;
-		if (!s->dma_adc.stopped) {
-			diff = dma_count_done(&s->dma_adc);
-			count += diff;
-			cinfo.bytes += diff;
-			cinfo.ptr =  virt_to_phys(s->dma_adc.nextIn) + diff -
-				virt_to_phys(s->dma_adc.rawbuf);
-		} else
-			cinfo.ptr = virt_to_phys(s->dma_adc.nextIn) -
-				virt_to_phys(s->dma_adc.rawbuf);
-		if (s->dma_adc.mapped)
-			s->dma_adc.count &= (s->dma_adc.dma_fragsize-1);
-		spin_unlock_irqrestore(&s->lock, flags);
-		if (count < 0)
-			count = 0;
-		cinfo.blocks = count >> s->dma_adc.fragshift;
-		return copy_to_user((void *) arg, &cinfo, sizeof(cinfo));
-
-	case SNDCTL_DSP_GETOPTR:
-		if (!(file->f_mode & FMODE_READ))
-			return -EINVAL;
-		spin_lock_irqsave(&s->lock, flags);
-		cinfo.bytes = s->dma_dac.total_bytes;
-		count = s->dma_dac.count;
-		if (!s->dma_dac.stopped) {
-			diff = dma_count_done(&s->dma_dac);
-			count -= diff;
-			cinfo.bytes += diff;
-			cinfo.ptr = virt_to_phys(s->dma_dac.nextOut) + diff -
-				virt_to_phys(s->dma_dac.rawbuf);
-		} else
-			cinfo.ptr = virt_to_phys(s->dma_dac.nextOut) -
-				virt_to_phys(s->dma_dac.rawbuf);
-		if (s->dma_dac.mapped)
-			s->dma_dac.count &= (s->dma_dac.dma_fragsize-1);
-		spin_unlock_irqrestore(&s->lock, flags);
-		if (count < 0)
-			count = 0;
-		cinfo.blocks = count >> s->dma_dac.fragshift;
-		return copy_to_user((void *) arg, &cinfo, sizeof(cinfo));
-
-	case SNDCTL_DSP_GETBLKSIZE:
-		if (file->f_mode & FMODE_WRITE)
-			return put_user(s->dma_dac.fragsize, (int *) arg);
-		else
-			return put_user(s->dma_adc.fragsize, (int *) arg);
-
-	case SNDCTL_DSP_SETFRAGMENT:
-		if (get_user(val, (int *) arg))
-			return -EFAULT;
-		if (file->f_mode & FMODE_READ) {
-			stop_adc(s);
-			s->dma_adc.ossfragshift = val & 0xffff;
-			s->dma_adc.ossmaxfrags = (val >> 16) & 0xffff;
-			if (s->dma_adc.ossfragshift < 4)
-				s->dma_adc.ossfragshift = 4;
-			if (s->dma_adc.ossfragshift > 15)
-				s->dma_adc.ossfragshift = 15;
-			if (s->dma_adc.ossmaxfrags < 4)
-				s->dma_adc.ossmaxfrags = 4;
-			if ((ret = prog_dmabuf_adc(s)))
-				return ret;
-		}
-		if (file->f_mode & FMODE_WRITE) {
-			stop_dac(s);
-			s->dma_dac.ossfragshift = val & 0xffff;
-			s->dma_dac.ossmaxfrags = (val >> 16) & 0xffff;
-			if (s->dma_dac.ossfragshift < 4)
-				s->dma_dac.ossfragshift = 4;
-			if (s->dma_dac.ossfragshift > 15)
-				s->dma_dac.ossfragshift = 15;
-			if (s->dma_dac.ossmaxfrags < 4)
-				s->dma_dac.ossmaxfrags = 4;
-			if ((ret = prog_dmabuf_dac(s)))
-				return ret;
-		}
-		return 0;
-
-	case SNDCTL_DSP_SUBDIVIDE:
-		if ((file->f_mode & FMODE_READ && s->dma_adc.subdivision) ||
-		    (file->f_mode & FMODE_WRITE && s->dma_dac.subdivision))
-			return -EINVAL;
-		if (get_user(val, (int *) arg))
-			return -EFAULT;
-		if (val != 1 && val != 2 && val != 4)
-			return -EINVAL;
-		if (file->f_mode & FMODE_READ) {
-			stop_adc(s);
-			s->dma_adc.subdivision = val;
-			if ((ret = prog_dmabuf_adc(s)))
-				return ret;
-		}
-		if (file->f_mode & FMODE_WRITE) {
-			stop_dac(s);
-			s->dma_dac.subdivision = val;
-			if ((ret = prog_dmabuf_dac(s)))
-				return ret;
-		}
-		return 0;
-
-	case SOUND_PCM_READ_RATE:
-		return put_user((file->f_mode & FMODE_READ) ?
-				s->dma_adc.sample_rate :
-				s->dma_dac.sample_rate,
-				(int *)arg);
-
-	case SOUND_PCM_READ_CHANNELS:
-		if (file->f_mode & FMODE_READ)
-			return put_user(s->dma_adc.num_channels, (int *)arg);
-		else
-			return put_user(s->dma_dac.num_channels, (int *)arg);
-
-	case SOUND_PCM_READ_BITS:
-		if (file->f_mode & FMODE_READ)
-			return put_user(s->dma_adc.sample_size, (int *)arg);
-		else
-			return put_user(s->dma_dac.sample_size, (int *)arg);
-
-	case SOUND_PCM_WRITE_FILTER:
-	case SNDCTL_DSP_SETSYNCRO:
-	case SOUND_PCM_READ_FILTER:
-		return -EINVAL;
-	}
-
-	return mixdev_ioctl(s->codec, cmd, arg);
-}
-
-static long
-au1550_unlocked_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
-{
-	int ret;
-
-	mutex_lock(&au1550_ac97_mutex);
-	ret = au1550_ioctl(file, cmd, arg);
-	mutex_unlock(&au1550_ac97_mutex);
-
-	return ret;
-}
-
-static int
-au1550_open(struct inode *inode, struct file *file)
-{
-	int             minor = MINOR(inode->i_rdev);
-	DECLARE_WAITQUEUE(wait, current);
-	struct au1550_state *s = &au1550_state;
-	int             ret;
-
-#ifdef DEBUG
-	if (file->f_flags & O_NONBLOCK)
-		pr_debug("open: non-blocking\n");
-	else
-		pr_debug("open: blocking\n");
-#endif
-
-	file->private_data = s;
-	mutex_lock(&au1550_ac97_mutex);
-	/* wait for device to become free */
-	mutex_lock(&s->open_mutex);
-	while (s->open_mode & file->f_mode) {
-		ret = -EBUSY;
-		if (file->f_flags & O_NONBLOCK)
-			goto out;
-		add_wait_queue(&s->open_wait, &wait);
-		__set_current_state(TASK_INTERRUPTIBLE);
-		mutex_unlock(&s->open_mutex);
-		schedule();
-		remove_wait_queue(&s->open_wait, &wait);
-		set_current_state(TASK_RUNNING);
-		ret = -ERESTARTSYS;
-		if (signal_pending(current))
-			goto out2;
-		mutex_lock(&s->open_mutex);
-	}
-
-	stop_dac(s);
-	stop_adc(s);
-
-	if (file->f_mode & FMODE_READ) {
-		s->dma_adc.ossfragshift = s->dma_adc.ossmaxfrags =
-			s->dma_adc.subdivision = s->dma_adc.total_bytes = 0;
-		s->dma_adc.num_channels = 1;
-		s->dma_adc.sample_size = 8;
-		set_adc_rate(s, 8000);
-		if ((minor & 0xf) == SND_DEV_DSP16)
-			s->dma_adc.sample_size = 16;
-	}
-
-	if (file->f_mode & FMODE_WRITE) {
-		s->dma_dac.ossfragshift = s->dma_dac.ossmaxfrags =
-			s->dma_dac.subdivision = s->dma_dac.total_bytes = 0;
-		s->dma_dac.num_channels = 1;
-		s->dma_dac.sample_size = 8;
-		set_dac_rate(s, 8000);
-		if ((minor & 0xf) == SND_DEV_DSP16)
-			s->dma_dac.sample_size = 16;
-	}
-
-	if (file->f_mode & FMODE_READ) {
-		if ((ret = prog_dmabuf_adc(s)))
-			goto out;
-	}
-	if (file->f_mode & FMODE_WRITE) {
-		if ((ret = prog_dmabuf_dac(s)))
-			goto out;
-	}
-
-	s->open_mode |= file->f_mode & (FMODE_READ | FMODE_WRITE);
-	mutex_init(&s->sem);
-	ret = 0;
-out:
-	mutex_unlock(&s->open_mutex);
-out2:
-	mutex_unlock(&au1550_ac97_mutex);
-	return ret;
-}
-
-static int
-au1550_release(struct inode *inode, struct file *file)
-{
-	struct au1550_state *s = file->private_data;
-
-	mutex_lock(&au1550_ac97_mutex);
-
-	if (file->f_mode & FMODE_WRITE) {
-		mutex_unlock(&au1550_ac97_mutex);
-		drain_dac(s, file->f_flags & O_NONBLOCK);
-		mutex_lock(&au1550_ac97_mutex);
-	}
-
-	mutex_lock(&s->open_mutex);
-	if (file->f_mode & FMODE_WRITE) {
-		stop_dac(s);
-		kfree(s->dma_dac.rawbuf);
-		s->dma_dac.rawbuf = NULL;
-	}
-	if (file->f_mode & FMODE_READ) {
-		stop_adc(s);
-		kfree(s->dma_adc.rawbuf);
-		s->dma_adc.rawbuf = NULL;
-	}
-	s->open_mode &= ((~file->f_mode) & (FMODE_READ|FMODE_WRITE));
-	mutex_unlock(&s->open_mutex);
-	wake_up(&s->open_wait);
-	mutex_unlock(&au1550_ac97_mutex);
-	return 0;
-}
-
-static /*const */ struct file_operations au1550_audio_fops = {
-	.owner		= THIS_MODULE,
-	.llseek		= au1550_llseek,
-	.read		= au1550_read,
-	.write		= au1550_write,
-	.poll		= au1550_poll,
-	.unlocked_ioctl	= au1550_unlocked_ioctl,
-	.mmap		= au1550_mmap,
-	.open		= au1550_open,
-	.release	= au1550_release,
-};
-
-MODULE_AUTHOR("Advanced Micro Devices (AMD), dan@embeddededge.com");
-MODULE_DESCRIPTION("Au1550 AC97 Audio Driver");
-MODULE_LICENSE("GPL");
-
-
-static int __devinit
-au1550_probe(void)
-{
-	struct au1550_state *s = &au1550_state;
-	int             val;
-
-	memset(s, 0, sizeof(struct au1550_state));
-
-	init_waitqueue_head(&s->dma_adc.wait);
-	init_waitqueue_head(&s->dma_dac.wait);
-	init_waitqueue_head(&s->open_wait);
-	mutex_init(&s->open_mutex);
-	spin_lock_init(&s->lock);
-
-	s->codec = ac97_alloc_codec();
-	if(s->codec == NULL) {
-		err("Out of memory");
-		return -1;
-	}
-	s->codec->private_data = s;
-	s->codec->id = 0;
-	s->codec->codec_read = rdcodec;
-	s->codec->codec_write = wrcodec;
-	s->codec->codec_wait = waitcodec;
-
-	if (!request_mem_region(CPHYSADDR(AC97_PSC_SEL),
-			    0x30, "Au1550 AC97")) {
-		err("AC'97 ports in use");
-	}
-
-	/* Allocate the DMA Channels
-	*/
-	if ((s->dma_dac.dmanr = au1xxx_dbdma_chan_alloc(DBDMA_MEM_CHAN,
-	    DBDMA_AC97_TX_CHAN, dac_dma_interrupt, (void *)s)) == 0) {
-		err("Can't get DAC DMA");
-		goto err_dma1;
-	}
-	au1xxx_dbdma_set_devwidth(s->dma_dac.dmanr, 16);
-	if (au1xxx_dbdma_ring_alloc(s->dma_dac.dmanr,
-					NUM_DBDMA_DESCRIPTORS) == 0) {
-		err("Can't get DAC DMA descriptors");
-		goto err_dma1;
-	}
-
-	if ((s->dma_adc.dmanr = au1xxx_dbdma_chan_alloc(DBDMA_AC97_RX_CHAN,
-	    DBDMA_MEM_CHAN, adc_dma_interrupt, (void *)s)) == 0) {
-		err("Can't get ADC DMA");
-		goto err_dma2;
-	}
-	au1xxx_dbdma_set_devwidth(s->dma_adc.dmanr, 16);
-	if (au1xxx_dbdma_ring_alloc(s->dma_adc.dmanr,
-					NUM_DBDMA_DESCRIPTORS) == 0) {
-		err("Can't get ADC DMA descriptors");
-		goto err_dma2;
-	}
-
-	pr_info("DAC: DMA%d, ADC: DMA%d", DBDMA_AC97_TX_CHAN, DBDMA_AC97_RX_CHAN);
-
-	/* register devices */
-
-	if ((s->dev_audio = register_sound_dsp(&au1550_audio_fops, -1)) < 0)
-		goto err_dev1;
-	if ((s->codec->dev_mixer =
-	     register_sound_mixer(&au1550_mixer_fops, -1)) < 0)
-		goto err_dev2;
-
-	/* The GPIO for the appropriate PSC was configured by the
-	 * board specific start up.
-	 *
-	 * configure PSC for AC'97
-	 */
-	au_writel(0, AC97_PSC_CTRL);	/* Disable PSC */
-	au_sync();
-	au_writel((PSC_SEL_CLK_SERCLK | PSC_SEL_PS_AC97MODE), AC97_PSC_SEL);
-	au_sync();
-
-	/* cold reset the AC'97
-	*/
-	au_writel(PSC_AC97RST_RST, PSC_AC97RST);
-	au_sync();
-	au1550_delay(10);
-	au_writel(0, PSC_AC97RST);
-	au_sync();
-
-	/* need to delay around 500msec(bleech) to give
-	   some CODECs enough time to wakeup */
-	au1550_delay(500);
-
-	/* warm reset the AC'97 to start the bitclk
-	*/
-	au_writel(PSC_AC97RST_SNC, PSC_AC97RST);
-	au_sync();
-	udelay(100);
-	au_writel(0, PSC_AC97RST);
-	au_sync();
-
-	/* Enable PSC
-	*/
-	au_writel(PSC_CTRL_ENABLE, AC97_PSC_CTRL);
-	au_sync();
-
-	/* Wait for PSC ready.
-	*/
-	do {
-		val = au_readl(PSC_AC97STAT);
-		au_sync();
-	} while ((val & PSC_AC97STAT_SR) == 0);
-
-	/* Configure AC97 controller.
-	 * Deep FIFO, 16-bit sample, DMA, make sure DMA matches fifo size.
-	 */
-	val = PSC_AC97CFG_SET_LEN(16);
-	val |= PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8;
-
-	/* Enable device so we can at least
-	 * talk over the AC-link.
-	 */
-	au_writel(val, PSC_AC97CFG);
-	au_writel(PSC_AC97MSK_ALLMASK, PSC_AC97MSK);
-	au_sync();
-	val |= PSC_AC97CFG_DE_ENABLE;
-	au_writel(val, PSC_AC97CFG);
-	au_sync();
-
-	/* Wait for Device ready.
-	*/
-	do {
-		val = au_readl(PSC_AC97STAT);
-		au_sync();
-	} while ((val & PSC_AC97STAT_DR) == 0);
-
-	/* codec init */
-	if (!ac97_probe_codec(s->codec))
-		goto err_dev3;
-
-	s->codec_base_caps = rdcodec(s->codec, AC97_RESET);
-	s->codec_ext_caps = rdcodec(s->codec, AC97_EXTENDED_ID);
-	pr_info("AC'97 Base/Extended ID = %04x/%04x",
-	     s->codec_base_caps, s->codec_ext_caps);
-
-	if (!(s->codec_ext_caps & AC97_EXTID_VRA)) {
-		/* codec does not support VRA
-		*/
-		s->no_vra = 1;
-	} else if (!vra) {
-		/* Boot option says disable VRA
-		*/
-		u16 ac97_extstat = rdcodec(s->codec, AC97_EXTENDED_STATUS);
-		wrcodec(s->codec, AC97_EXTENDED_STATUS,
-			ac97_extstat & ~AC97_EXTSTAT_VRA);
-		s->no_vra = 1;
-	}
-	if (s->no_vra)
-		pr_info("no VRA, interpolating and decimating");
-
-	/* set mic to be the recording source */
-	val = SOUND_MASK_MIC;
-	mixdev_ioctl(s->codec, SOUND_MIXER_WRITE_RECSRC,
-		     (unsigned long) &val);
-
-	return 0;
-
- err_dev3:
-	unregister_sound_mixer(s->codec->dev_mixer);
- err_dev2:
-	unregister_sound_dsp(s->dev_audio);
- err_dev1:
-	au1xxx_dbdma_chan_free(s->dma_adc.dmanr);
- err_dma2:
-	au1xxx_dbdma_chan_free(s->dma_dac.dmanr);
- err_dma1:
-	release_mem_region(CPHYSADDR(AC97_PSC_SEL), 0x30);
-
-	ac97_release_codec(s->codec);
-	return -1;
-}
-
-static void __devinit
-au1550_remove(void)
-{
-	struct au1550_state *s = &au1550_state;
-
-	if (!s)
-		return;
-	synchronize_irq();
-	au1xxx_dbdma_chan_free(s->dma_adc.dmanr);
-	au1xxx_dbdma_chan_free(s->dma_dac.dmanr);
-	release_mem_region(CPHYSADDR(AC97_PSC_SEL), 0x30);
-	unregister_sound_dsp(s->dev_audio);
-	unregister_sound_mixer(s->codec->dev_mixer);
-	ac97_release_codec(s->codec);
-}
-
-static int __init
-init_au1550(void)
-{
-	return au1550_probe();
-}
-
-static void __exit
-cleanup_au1550(void)
-{
-	au1550_remove();
-}
-
-module_init(init_au1550);
-module_exit(cleanup_au1550);
-
-#ifndef MODULE
-
-static int __init
-au1550_setup(char *options)
-{
-	char           *this_opt;
-
-	if (!options || !*options)
-		return 0;
-
-	while ((this_opt = strsep(&options, ","))) {
-		if (!*this_opt)
-			continue;
-		if (!strncmp(this_opt, "vra", 3)) {
-			vra = 1;
-		}
-	}
-
-	return 1;
-}
-
-__setup("au1550_audio=", au1550_setup);
-
-#endif /* MODULE */
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 389cd793166..e90d103e177 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -534,6 +534,14 @@ config SND_ES1968_INPUT
 	  If you say N the buttons will directly control the master volume.
 	  It is recommended to say Y.
 
+config SND_ES1968_RADIO
+	bool "Enable TEA5757 radio tuner support for es1968"
+	depends on SND_ES1968
+	depends on VIDEO_V4L2=y || VIDEO_V4L2=SND_ES1968
+	help
+	  Say Y here to include support for TEA5757 radio tuner integrated on
+	  some MediaForte cards (e.g. SF64-PCE2).
+
 config SND_FM801
 	tristate "ForteMedia FM801"
 	select SND_OPL3_LIB
@@ -552,13 +560,13 @@ config SND_FM801_TEA575X_BOOL
 	depends on VIDEO_V4L2=y || VIDEO_V4L2=SND_FM801
 	help
 	  Say Y here to include support for soundcards based on the ForteMedia
-	  FM801 chip with a TEA5757 tuner connected to GPIO1-3 pins (Media
-	  Forte SF256-PCS-02) into the snd-fm801 driver.
+	  FM801 chip with a TEA5757 tuner (MediaForte SF256-PCS, SF256-PCP and
+	  SF64-PCR) into the snd-fm801 driver.
 
-config SND_FM801_TEA575X
+config SND_TEA575X
 	tristate
-	depends on SND_FM801_TEA575X_BOOL
-	default SND_FM801
+	depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO
+	default SND_FM801 || SND_ES1968
 
 source "sound/pci/hda/Kconfig"
 
@@ -658,6 +666,15 @@ config SND_KORG1212
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-korg1212.
 
+config SND_LOLA
+	tristate "Digigram Lola"
+	select SND_PCM
+	help
+	  Say Y to include support for Digigram Lola boards.
+
+	  To compile this driver as a module, choose M here: the module
+	  will be called snd-lola.
+
 config SND_LX6464ES
 	tristate "Digigram LX6464ES"
 	select SND_PCM
diff --git a/sound/pci/Makefile b/sound/pci/Makefile
index 9cf4348ec13..54fe325e3aa 100644
--- a/sound/pci/Makefile
+++ b/sound/pci/Makefile
@@ -64,6 +64,7 @@ obj-$(CONFIG_SND) += \
 	ca0106/ \
 	cs46xx/ \
 	cs5535audio/ \
+	lola/ \
 	lx6464es/ \
 	echoaudio/ \
 	emu10k1/ \
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index f8ccc9677c6..2ca6f4f85b4 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -42,10 +42,29 @@
 #include <sound/tlv.h>
 #include <sound/hwdep.h>
 
+
 MODULE_LICENSE("GPL");
 MODULE_AUTHOR("AudioScience inc. <support@audioscience.com>");
 MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx");
 
+#if defined CONFIG_SND_DEBUG
+/* copied from pcm_lib.c, hope later patch will make that version public
+and this copy can be removed */
+static void pcm_debug_name(struct snd_pcm_substream *substream,
+			   char *name, size_t len)
+{
+	snprintf(name, len, "pcmC%dD%d%c:%d",
+		 substream->pcm->card->number,
+		 substream->pcm->device,
+		 substream->stream ? 'c' : 'p',
+		 substream->number);
+}
+#define DEBUG_NAME(substream, name) char name[16]; pcm_debug_name(substream, name, sizeof(name))
+#else
+#define pcm_debug_name(s, n, l) do { } while (0)
+#define DEBUG_NAME(name, substream) do { } while (0)
+#endif
+
 #if defined CONFIG_SND_DEBUG_VERBOSE
 /**
  * snd_printddd - very verbose debug printk
@@ -58,7 +77,7 @@ MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx");
 #define snd_printddd(format, args...) \
 	__snd_printk(3, __FILE__, __LINE__, format, ##args)
 #else
-#define snd_printddd(format, args...)	do { } while (0)
+#define snd_printddd(format, args...) do { } while (0)
 #endif
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* index 0-MAX */
@@ -101,13 +120,6 @@ static int adapter_fs = DEFAULT_SAMPLERATE;
 #define PERIOD_BYTES_MIN  2048
 #define BUFFER_BYTES_MAX (512 * 1024)
 
-/* convert stream to character */
-#define SCHR(s) ((s == SNDRV_PCM_STREAM_PLAYBACK) ? 'P' : 'C')
-
-/*#define TIMER_MILLISECONDS 20
-#define FORCE_TIMER_JIFFIES ((TIMER_MILLISECONDS * HZ + 999)/1000)
-*/
-
 #define MAX_CLOCKSOURCES (HPI_SAMPLECLOCK_SOURCE_LAST + 1 + 7)
 
 struct clk_source {
@@ -136,7 +148,7 @@ struct snd_card_asihpi {
 	u32 h_mixer;
 	struct clk_cache cc;
 
-	u16 support_mmap;
+	u16 can_dma;
 	u16 support_grouping;
 	u16 support_mrx;
 	u16 update_interval_frames;
@@ -155,6 +167,7 @@ struct snd_card_asihpi_pcm {
 	unsigned int pcm_buf_host_rw_ofs; /* Host R/W pos */
 	unsigned int pcm_buf_dma_ofs;	/* DMA R/W offset in buffer */
 	unsigned int pcm_buf_elapsed_dma_ofs;	/* DMA R/W offset in buffer */
+	unsigned int drained_count;
 	struct snd_pcm_substream *substream;
 	u32 h_stream;
 	struct hpi_format format;
@@ -288,19 +301,26 @@ static u16 handle_error(u16 err, int line, char *filename)
 #define hpi_handle_error(x)  handle_error(x, __LINE__, __FILE__)
 
 /***************************** GENERAL PCM ****************/
-static void print_hwparams(struct snd_pcm_hw_params *p)
+
+static void print_hwparams(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *p)
 {
-	snd_printd("HWPARAMS \n");
-	snd_printd("samplerate %d \n", params_rate(p));
-	snd_printd("Channels %d \n", params_channels(p));
-	snd_printd("Format %d \n", params_format(p));
-	snd_printd("subformat %d \n", params_subformat(p));
-	snd_printd("Buffer bytes %d \n", params_buffer_bytes(p));
-	snd_printd("Period bytes %d \n", params_period_bytes(p));
-	snd_printd("access %d \n", params_access(p));
-	snd_printd("period_size %d \n", params_period_size(p));
-	snd_printd("periods %d \n", params_periods(p));
-	snd_printd("buffer_size %d \n", params_buffer_size(p));
+	DEBUG_NAME(substream, name);
+	snd_printd("%s HWPARAMS\n", name);
+	snd_printd(" samplerate %d Hz\n", params_rate(p));
+	snd_printd(" channels %d\n", params_channels(p));
+	snd_printd(" format %d\n", params_format(p));
+	snd_printd(" subformat %d\n", params_subformat(p));
+	snd_printd(" buffer %d B\n", params_buffer_bytes(p));
+	snd_printd(" period %d B\n", params_period_bytes(p));
+	snd_printd(" access %d\n", params_access(p));
+	snd_printd(" period_size %d\n", params_period_size(p));
+	snd_printd(" periods %d\n", params_periods(p));
+	snd_printd(" buffer_size %d\n", params_buffer_size(p));
+	snd_printd(" %d B/s\n", params_rate(p) *
+		params_channels(p) *
+		snd_pcm_format_width(params_format(p)) / 8);
+
 }
 
 static snd_pcm_format_t hpi_to_alsa_formats[] = {
@@ -451,7 +471,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream,
 	int width;
 	unsigned int bytes_per_sec;
 
-	print_hwparams(params);
+	print_hwparams(substream, params);
 	err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
 	if (err < 0)
 		return err;
@@ -459,10 +479,6 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream,
 	if (err)
 		return err;
 
-	snd_printdd("format %d, %d chans, %d_hz\n",
-				format, params_channels(params),
-				params_rate(params));
-
 	hpi_handle_error(hpi_format_create(&dpcm->format,
 			params_channels(params),
 			format, params_rate(params), 0, 0));
@@ -477,8 +493,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	dpcm->hpi_buffer_attached = 0;
-	if (card->support_mmap) {
-
+	if (card->can_dma) {
 		err = hpi_stream_host_buffer_attach(dpcm->h_stream,
 			params_buffer_bytes(params),  runtime->dma_addr);
 		if (err == 0) {
@@ -509,8 +524,6 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream,
 	dpcm->bytes_per_sec = bytes_per_sec;
 	dpcm->buffer_bytes = params_buffer_bytes(params);
 	dpcm->period_bytes = params_period_bytes(params);
-	snd_printdd("buffer_bytes=%d, period_bytes=%d, bps=%d\n",
-			dpcm->buffer_bytes, dpcm->period_bytes, bytes_per_sec);
 
 	return 0;
 }
@@ -564,9 +577,10 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
 	struct snd_card_asihpi *card = snd_pcm_substream_chip(substream);
 	struct snd_pcm_substream *s;
 	u16 e;
+	DEBUG_NAME(substream, name);
+
+	snd_printdd("%s trigger\n", name);
 
-	snd_printdd("%c%d trigger\n",
-			SCHR(substream->stream), substream->number);
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		snd_pcm_group_for_each_entry(s, substream) {
@@ -580,8 +594,8 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
 			if (substream->stream != s->stream)
 				continue;
 
-			if ((s->stream == SNDRV_PCM_STREAM_PLAYBACK) &&
-				(card->support_mmap)) {
+			ds->drained_count = 0;
+			if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 				/* How do I know how much valid data is present
 				* in buffer? Must be at least one period!
 				* Guessing 2 periods, but if
@@ -599,9 +613,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
 			}
 
 			if (card->support_grouping) {
-				snd_printdd("\t%c%d group\n",
-						SCHR(s->stream),
-						s->number);
+				snd_printdd("%d group\n", s->number);
 				e = hpi_stream_group_add(
 					dpcm->h_stream,
 					ds->h_stream);
@@ -618,7 +630,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
 		/* start the master stream */
 		snd_card_asihpi_pcm_timer_start(substream);
 		if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) ||
-			!card->support_mmap)
+			!card->can_dma)
 			hpi_handle_error(hpi_stream_start(dpcm->h_stream));
 		break;
 
@@ -636,9 +648,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
 			s->runtime->status->state = SNDRV_PCM_STATE_SETUP;
 
 			if (card->support_grouping) {
-				snd_printdd("\t%c%d group\n",
-				SCHR(s->stream),
-					s->number);
+				snd_printdd("%d group\n", s->number);
 				snd_pcm_trigger_done(s, substream);
 			} else
 				break;
@@ -732,9 +742,9 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 	int loops = 0;
 	u16 state;
 	u32 buffer_size, bytes_avail, samples_played, on_card_bytes;
+	DEBUG_NAME(substream, name);
 
-	snd_printdd("%c%d snd_card_asihpi_timer_function\n",
-				SCHR(substream->stream), substream->number);
+	snd_printdd("%s snd_card_asihpi_timer_function\n", name);
 
 	/* find minimum newdata and buffer pos in group */
 	snd_pcm_group_for_each_entry(s, substream) {
@@ -756,6 +766,9 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 		/* number of bytes in on-card buffer */
 		runtime->delay = on_card_bytes;
 
+		if (!card->can_dma)
+			on_card_bytes = bytes_avail;
+
 		if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 			pcm_buf_dma_ofs = ds->pcm_buf_host_rw_ofs - bytes_avail;
 			if (state == HPI_STATE_STOPPED) {
@@ -763,12 +776,18 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 				    (on_card_bytes < ds->pcm_buf_host_rw_ofs)) {
 					hpi_handle_error(hpi_stream_start(ds->h_stream));
 					snd_printdd("P%d start\n", s->number);
+					ds->drained_count = 0;
 				}
 			} else if (state == HPI_STATE_DRAINED) {
 				snd_printd(KERN_WARNING "P%d drained\n",
 						s->number);
-				/*snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN);
-				continue; */
+				ds->drained_count++;
+				if (ds->drained_count > 2) {
+					snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN);
+					continue;
+				}
+			} else {
+				ds->drained_count = 0;
 			}
 		} else
 			pcm_buf_dma_ofs = bytes_avail + ds->pcm_buf_host_rw_ofs;
@@ -786,16 +805,18 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 				newdata);
 		}
 
-		snd_printdd("hw_ptr x%04lX, appl_ptr x%04lX\n",
+		snd_printdd("hw_ptr 0x%04lX, appl_ptr 0x%04lX\n",
 			(unsigned long)frames_to_bytes(runtime,
 						runtime->status->hw_ptr),
 			(unsigned long)frames_to_bytes(runtime,
 						runtime->control->appl_ptr));
 
-		snd_printdd("%d %c%d S=%d, rw=%04X, dma=x%04X, left=x%04X,"
-			" aux=x%04X space=x%04X\n",
-			loops, SCHR(s->stream),	s->number,
-			state,	ds->pcm_buf_host_rw_ofs, pcm_buf_dma_ofs, (int)bytes_avail,
+		snd_printdd("%d S=%d, "
+			"rw=0x%04X, dma=0x%04X, left=0x%04X, "
+			"aux=0x%04X space=0x%04X\n",
+			s->number, state,
+			ds->pcm_buf_host_rw_ofs, pcm_buf_dma_ofs,
+			(int)bytes_avail,
 			(int)on_card_bytes, buffer_size-bytes_avail);
 		loops++;
 	}
@@ -814,7 +835,7 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 
 	next_jiffies = max(next_jiffies, 1U);
 	dpcm->timer.expires = jiffies + next_jiffies;
-	snd_printdd("jif %d buf pos x%04X newdata x%04X xfer x%04X\n",
+	snd_printdd("jif %d buf pos 0x%04X newdata 0x%04X xfer 0x%04X\n",
 			next_jiffies, pcm_buf_dma_ofs, newdata, xfercount);
 
 	snd_pcm_group_for_each_entry(s, substream) {
@@ -826,30 +847,63 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 
 		ds->pcm_buf_dma_ofs = pcm_buf_dma_ofs;
 
-		if (xfercount && (on_card_bytes <= ds->period_bytes)) {
-			if (card->support_mmap) {
-				if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-					snd_printddd("P%d write x%04x\n",
+		if (xfercount &&
+			/* Limit use of on card fifo for playback */
+			((on_card_bytes <= ds->period_bytes) ||
+			(s->stream == SNDRV_PCM_STREAM_CAPTURE)))
+
+		{
+
+			unsigned int buf_ofs = ds->pcm_buf_host_rw_ofs % ds->buffer_bytes;
+			unsigned int xfer1, xfer2;
+			char *pd = &s->runtime->dma_area[buf_ofs];
+
+			if (card->can_dma) { /* buffer wrap is handled at lower level */
+				xfer1 = xfercount;
+				xfer2 = 0;
+			} else {
+				xfer1 = min(xfercount, ds->buffer_bytes - buf_ofs);
+				xfer2 = xfercount - xfer1;
+			}
+
+			if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+				snd_printddd("P%d write1 0x%04X 0x%04X\n",
+					s->number, xfer1, buf_ofs);
+				hpi_handle_error(
+					hpi_outstream_write_buf(
+						ds->h_stream, pd, xfer1,
+						&ds->format));
+
+				if (xfer2) {
+					pd = s->runtime->dma_area;
+
+					snd_printddd("P%d write2 0x%04X 0x%04X\n",
 							s->number,
-							ds->period_bytes);
+							xfercount - xfer1, buf_ofs);
 					hpi_handle_error(
 						hpi_outstream_write_buf(
-							ds->h_stream,
-							&s->runtime->
-								dma_area[0],
-							xfercount,
+							ds->h_stream, pd,
+							xfercount - xfer1,
 							&ds->format));
-				} else {
-					snd_printddd("C%d read x%04x\n",
-						s->number,
-						xfercount);
+				}
+			} else {
+				snd_printddd("C%d read1 0x%04x\n",
+					s->number, xfer1);
+				hpi_handle_error(
+					hpi_instream_read_buf(
+						ds->h_stream,
+						pd, xfer1));
+				if (xfer2) {
+					pd = s->runtime->dma_area;
+					snd_printddd("C%d read2 0x%04x\n",
+						s->number, xfer2);
 					hpi_handle_error(
 						hpi_instream_read_buf(
 							ds->h_stream,
-							NULL, xfercount));
+							pd, xfer2));
 				}
-				ds->pcm_buf_host_rw_ofs = ds->pcm_buf_host_rw_ofs + xfercount;
-			} /* else R/W will be handled by read/write callbacks */
+			}
+			ds->pcm_buf_host_rw_ofs = ds->pcm_buf_host_rw_ofs + xfercount;
 			ds->pcm_buf_elapsed_dma_ofs = pcm_buf_dma_ofs;
 			snd_pcm_period_elapsed(s);
 		}
@@ -863,7 +917,7 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 static int snd_card_asihpi_playback_ioctl(struct snd_pcm_substream *substream,
 					  unsigned int cmd, void *arg)
 {
-	snd_printdd(KERN_INFO "Playback ioctl %d\n", cmd);
+	snd_printddd(KERN_INFO "P%d ioctl %d\n", substream->number, cmd);
 	return snd_pcm_lib_ioctl(substream, cmd, arg);
 }
 
@@ -873,7 +927,7 @@ static int snd_card_asihpi_playback_prepare(struct snd_pcm_substream *
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_card_asihpi_pcm *dpcm = runtime->private_data;
 
-	snd_printdd("playback prepare %d\n", substream->number);
+	snd_printdd("P%d prepare\n", substream->number);
 
 	hpi_handle_error(hpi_outstream_reset(dpcm->h_stream));
 	dpcm->pcm_buf_host_rw_ofs = 0;
@@ -890,7 +944,7 @@ snd_card_asihpi_playback_pointer(struct snd_pcm_substream *substream)
 	snd_pcm_uframes_t ptr;
 
 	ptr = bytes_to_frames(runtime, dpcm->pcm_buf_dma_ofs  % dpcm->buffer_bytes);
-	snd_printddd("playback_pointer=x%04lx\n", (unsigned long)ptr);
+	snd_printddd("P%d pointer = 0x%04lx\n", substream->number, (unsigned long)ptr);
 	return ptr;
 }
 
@@ -986,11 +1040,9 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream)
 					SNDRV_PCM_INFO_DOUBLE |
 					SNDRV_PCM_INFO_BATCH |
 					SNDRV_PCM_INFO_BLOCK_TRANSFER |
-					SNDRV_PCM_INFO_PAUSE;
-
-	if (card->support_mmap)
-		snd_card_asihpi_playback.info |= SNDRV_PCM_INFO_MMAP |
-						SNDRV_PCM_INFO_MMAP_VALID;
+					SNDRV_PCM_INFO_PAUSE |
+					SNDRV_PCM_INFO_MMAP |
+					SNDRV_PCM_INFO_MMAP_VALID;
 
 	if (card->support_grouping)
 		snd_card_asihpi_playback.info |= SNDRV_PCM_INFO_SYNC_START;
@@ -998,7 +1050,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream)
 	/* struct is copied, so can create initializer dynamically */
 	runtime->hw = snd_card_asihpi_playback;
 
-	if (card->support_mmap)
+	if (card->can_dma)
 		err = snd_pcm_hw_constraint_pow2(runtime, 0,
 					SNDRV_PCM_HW_PARAM_BUFFER_BYTES);
 	if (err < 0)
@@ -1028,58 +1080,6 @@ static int snd_card_asihpi_playback_close(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int snd_card_asihpi_playback_copy(struct snd_pcm_substream *substream,
-					int channel,
-					snd_pcm_uframes_t pos,
-					void __user *src,
-					snd_pcm_uframes_t count)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_card_asihpi_pcm *dpcm = runtime->private_data;
-	unsigned int len;
-
-	len = frames_to_bytes(runtime, count);
-
-	if (copy_from_user(runtime->dma_area, src, len))
-		return -EFAULT;
-
-	snd_printddd("playback copy%d %u bytes\n",
-			substream->number, len);
-
-	hpi_handle_error(hpi_outstream_write_buf(dpcm->h_stream,
-				runtime->dma_area, len, &dpcm->format));
-
-	dpcm->pcm_buf_host_rw_ofs += len;
-
-	return 0;
-}
-
-static int snd_card_asihpi_playback_silence(struct snd_pcm_substream *
-					    substream, int channel,
-					    snd_pcm_uframes_t pos,
-					    snd_pcm_uframes_t count)
-{
-	/* Usually writes silence to DMA buffer, which should be overwritten
-	by real audio later.  Our fifos cannot be overwritten, and are not
-	free-running DMAs. Silence is output on fifo underflow.
-	This callback is still required to allow the copy callback to be used.
-	*/
-	return 0;
-}
-
-static struct snd_pcm_ops snd_card_asihpi_playback_ops = {
-	.open = snd_card_asihpi_playback_open,
-	.close = snd_card_asihpi_playback_close,
-	.ioctl = snd_card_asihpi_playback_ioctl,
-	.hw_params = snd_card_asihpi_pcm_hw_params,
-	.hw_free = snd_card_asihpi_hw_free,
-	.prepare = snd_card_asihpi_playback_prepare,
-	.trigger = snd_card_asihpi_trigger,
-	.pointer = snd_card_asihpi_playback_pointer,
-	.copy = snd_card_asihpi_playback_copy,
-	.silence = snd_card_asihpi_playback_silence,
-};
-
 static struct snd_pcm_ops snd_card_asihpi_playback_mmap_ops = {
 	.open = snd_card_asihpi_playback_open,
 	.close = snd_card_asihpi_playback_close,
@@ -1211,18 +1211,16 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream)
 	snd_card_asihpi_capture_format(card, dpcm->h_stream,
 				       &snd_card_asihpi_capture);
 	snd_card_asihpi_pcm_samplerates(card,  &snd_card_asihpi_capture);
-	snd_card_asihpi_capture.info = SNDRV_PCM_INFO_INTERLEAVED;
-
-	if (card->support_mmap)
-		snd_card_asihpi_capture.info |= SNDRV_PCM_INFO_MMAP |
-						SNDRV_PCM_INFO_MMAP_VALID;
+	snd_card_asihpi_capture.info = SNDRV_PCM_INFO_INTERLEAVED |
+					SNDRV_PCM_INFO_MMAP |
+					SNDRV_PCM_INFO_MMAP_VALID;
 
 	if (card->support_grouping)
 		snd_card_asihpi_capture.info |= SNDRV_PCM_INFO_SYNC_START;
 
 	runtime->hw = snd_card_asihpi_capture;
 
-	if (card->support_mmap)
+	if (card->can_dma)
 		err = snd_pcm_hw_constraint_pow2(runtime, 0,
 					SNDRV_PCM_HW_PARAM_BUFFER_BYTES);
 	if (err < 0)
@@ -1246,28 +1244,6 @@ static int snd_card_asihpi_capture_close(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int snd_card_asihpi_capture_copy(struct snd_pcm_substream *substream,
-				int channel, snd_pcm_uframes_t pos,
-				void __user *dst, snd_pcm_uframes_t count)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_card_asihpi_pcm *dpcm = runtime->private_data;
-	u32 len;
-
-	len = frames_to_bytes(runtime, count);
-
-	snd_printddd("capture copy%d %d bytes\n", substream->number, len);
-	hpi_handle_error(hpi_instream_read_buf(dpcm->h_stream,
-				runtime->dma_area, len));
-
-	dpcm->pcm_buf_host_rw_ofs = dpcm->pcm_buf_host_rw_ofs + len;
-
-	if (copy_to_user(dst, runtime->dma_area, len))
-		return -EFAULT;
-
-	return 0;
-}
-
 static struct snd_pcm_ops snd_card_asihpi_capture_mmap_ops = {
 	.open = snd_card_asihpi_capture_open,
 	.close = snd_card_asihpi_capture_close,
@@ -1279,18 +1255,6 @@ static struct snd_pcm_ops snd_card_asihpi_capture_mmap_ops = {
 	.pointer = snd_card_asihpi_capture_pointer,
 };
 
-static struct snd_pcm_ops snd_card_asihpi_capture_ops = {
-	.open = snd_card_asihpi_capture_open,
-	.close = snd_card_asihpi_capture_close,
-	.ioctl = snd_card_asihpi_capture_ioctl,
-	.hw_params = snd_card_asihpi_pcm_hw_params,
-	.hw_free = snd_card_asihpi_hw_free,
-	.prepare = snd_card_asihpi_capture_prepare,
-	.trigger = snd_card_asihpi_trigger,
-	.pointer = snd_card_asihpi_capture_pointer,
-	.copy = snd_card_asihpi_capture_copy
-};
-
 static int __devinit snd_card_asihpi_pcm_new(struct snd_card_asihpi *asihpi,
 				      int device, int substreams)
 {
@@ -1303,17 +1267,10 @@ static int __devinit snd_card_asihpi_pcm_new(struct snd_card_asihpi *asihpi,
 	if (err < 0)
 		return err;
 	/* pointer to ops struct is stored, dont change ops afterwards! */
-	if (asihpi->support_mmap) {
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
 				&snd_card_asihpi_playback_mmap_ops);
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
 				&snd_card_asihpi_capture_mmap_ops);
-	} else {
-		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
-				&snd_card_asihpi_playback_ops);
-		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
-				&snd_card_asihpi_capture_ops);
-	}
 
 	pcm->private_data = asihpi;
 	pcm->info_flags = 0;
@@ -1413,14 +1370,16 @@ static void asihpi_ctl_init(struct snd_kcontrol_new *snd_control,
 				struct hpi_control *hpi_ctl,
 				char *name)
 {
-	char *dir = "";
+	char *dir;
 	memset(snd_control, 0, sizeof(*snd_control));
 	snd_control->name = hpi_ctl->name;
 	snd_control->private_value = hpi_ctl->h_control;
 	snd_control->iface = SNDRV_CTL_ELEM_IFACE_MIXER;
 	snd_control->index = 0;
 
-	if (hpi_ctl->dst_node_type + HPI_DESTNODE_NONE == HPI_DESTNODE_ISTREAM)
+	if (hpi_ctl->src_node_type + HPI_SOURCENODE_NONE == HPI_SOURCENODE_CLOCK_SOURCE)
+		dir = ""; /* clock is neither capture nor playback */
+	else if (hpi_ctl->dst_node_type + HPI_DESTNODE_NONE == HPI_DESTNODE_ISTREAM)
 		dir = "Capture ";  /* On or towards a PCM capture destination*/
 	else if ((hpi_ctl->src_node_type + HPI_SOURCENODE_NONE != HPI_SOURCENODE_OSTREAM) &&
 		(!hpi_ctl->dst_node_type))
@@ -1433,7 +1392,7 @@ static void asihpi_ctl_init(struct snd_kcontrol_new *snd_control,
 		dir = "Playback "; /* PCM Playback source, or  output node */
 
 	if (hpi_ctl->src_node_type && hpi_ctl->dst_node_type)
-		sprintf(hpi_ctl->name, "%s%d %s%d %s%s",
+		sprintf(hpi_ctl->name, "%s %d %s %d %s%s",
 			asihpi_src_names[hpi_ctl->src_node_type],
 			hpi_ctl->src_node_index,
 			asihpi_dst_names[hpi_ctl->dst_node_type],
@@ -2875,14 +2834,14 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev,
 	if (err)
 		asihpi->update_interval_frames = 512;
 
-	if (!asihpi->support_mmap)
+	if (!asihpi->can_dma)
 		asihpi->update_interval_frames *= 2;
 
 	hpi_handle_error(hpi_instream_open(asihpi->adapter_index,
 			     0, &h_stream));
 
 	err = hpi_instream_host_buffer_free(h_stream);
-	asihpi->support_mmap = (!err);
+	asihpi->can_dma = (!err);
 
 	hpi_handle_error(hpi_instream_close(h_stream));
 
@@ -2894,8 +2853,8 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev,
 		asihpi->out_max_chans = 2;
 	}
 
-	snd_printk(KERN_INFO "supports mmap:%d grouping:%d mrx:%d\n",
-			asihpi->support_mmap,
+	snd_printk(KERN_INFO "has dma:%d, grouping:%d, mrx:%d\n",
+			asihpi->can_dma,
 			asihpi->support_grouping,
 			asihpi->support_mrx
 	      );
@@ -2925,10 +2884,7 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev,
 	    by enable_hwdep  module param*/
 	snd_asihpi_hpi_new(asihpi, 0, NULL);
 
-	if (asihpi->support_mmap)
-		strcpy(card->driver, "ASIHPI-MMAP");
-	else
-		strcpy(card->driver, "ASIHPI");
+	strcpy(card->driver, "ASIHPI");
 
 	sprintf(card->shortname, "AudioScience ASI%4X", asihpi->type);
 	sprintf(card->longname, "%s %i",
diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c
index 8c8aac4c567..df4aed5295d 100644
--- a/sound/pci/asihpi/hpi6000.c
+++ b/sound/pci/asihpi/hpi6000.c
@@ -200,8 +200,8 @@ static void hpi_read_block(struct dsp_obj *pdo, u32 address, u32 *pdata,
 static void subsys_create_adapter(struct hpi_message *phm,
 	struct hpi_response *phr);
 
-static void subsys_delete_adapter(struct hpi_message *phm,
-	struct hpi_response *phr);
+static void adapter_delete(struct hpi_adapter_obj *pao,
+	struct hpi_message *phm, struct hpi_response *phr);
 
 static void adapter_get_asserts(struct hpi_adapter_obj *pao,
 	struct hpi_message *phm, struct hpi_response *phr);
@@ -222,9 +222,6 @@ static void subsys_message(struct hpi_message *phm, struct hpi_response *phr)
 	case HPI_SUBSYS_CREATE_ADAPTER:
 		subsys_create_adapter(phm, phr);
 		break;
-	case HPI_SUBSYS_DELETE_ADAPTER:
-		subsys_delete_adapter(phm, phr);
-		break;
 	default:
 		phr->error = HPI_ERROR_INVALID_FUNC;
 		break;
@@ -279,6 +276,10 @@ static void adapter_message(struct hpi_adapter_obj *pao,
 		adapter_get_asserts(pao, phm, phr);
 		break;
 
+	case HPI_ADAPTER_DELETE:
+		adapter_delete(pao, phm, phr);
+		break;
+
 	default:
 		hw_message(pao, phm, phr);
 		break;
@@ -333,26 +334,22 @@ void HPI_6000(struct hpi_message *phm, struct hpi_response *phr)
 {
 	struct hpi_adapter_obj *pao = NULL;
 
-	/* subsytem messages get executed by every HPI. */
-	/* All other messages are ignored unless the adapter index matches */
-	/* an adapter in the HPI */
-	/*HPI_DEBUG_LOG(DEBUG, "O %d,F %x\n", phm->wObject, phm->wFunction); */
-
-	/* if Dsp has crashed then do not communicate with it any more */
 	if (phm->object != HPI_OBJ_SUBSYSTEM) {
 		pao = hpi_find_adapter(phm->adapter_index);
 		if (!pao) {
-			HPI_DEBUG_LOG(DEBUG,
-				" %d,%d refused, for another HPI?\n",
-				phm->object, phm->function);
+			hpi_init_response(phr, phm->object, phm->function,
+				HPI_ERROR_BAD_ADAPTER_NUMBER);
+			HPI_DEBUG_LOG(DEBUG, "invalid adapter index: %d \n",
+				phm->adapter_index);
 			return;
 		}
 
+		/* Don't even try to communicate with crashed DSP */
 		if (pao->dsp_crashed >= 10) {
 			hpi_init_response(phr, phm->object, phm->function,
 				HPI_ERROR_DSP_HARDWARE);
-			HPI_DEBUG_LOG(DEBUG, " %d,%d dsp crashed.\n",
-				phm->object, phm->function);
+			HPI_DEBUG_LOG(DEBUG, "adapter %d dsp crashed\n",
+				phm->adapter_index);
 			return;
 		}
 	}
@@ -463,15 +460,9 @@ static void subsys_create_adapter(struct hpi_message *phm,
 	phr->error = 0;
 }
 
-static void subsys_delete_adapter(struct hpi_message *phm,
-	struct hpi_response *phr)
+static void adapter_delete(struct hpi_adapter_obj *pao,
+	struct hpi_message *phm, struct hpi_response *phr)
 {
-	struct hpi_adapter_obj *pao = NULL;
-
-	pao = hpi_find_adapter(phm->obj_index);
-	if (!pao)
-		return;
-
 	delete_adapter_obj(pao);
 	hpi_delete_adapter(pao);
 	phr->error = 0;
diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c
index 22e9f08dea6..9d5df54a6b4 100644
--- a/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -152,8 +152,8 @@ static void hw_message(struct hpi_adapter_obj *pao, struct hpi_message *phm,
 
 static void subsys_create_adapter(struct hpi_message *phm,
 	struct hpi_response *phr);
-static void subsys_delete_adapter(struct hpi_message *phm,
-	struct hpi_response *phr);
+static void adapter_delete(struct hpi_adapter_obj *pao,
+	struct hpi_message *phm, struct hpi_response *phr);
 
 static u16 create_adapter_obj(struct hpi_adapter_obj *pao,
 	u32 *pos_error_code);
@@ -223,15 +223,13 @@ static u16 boot_loader_test_pld(struct hpi_adapter_obj *pao, int dsp_index);
 
 /*****************************************************************************/
 
-static void subsys_message(struct hpi_message *phm, struct hpi_response *phr)
+static void subsys_message(struct hpi_adapter_obj *pao,
+	struct hpi_message *phm, struct hpi_response *phr)
 {
 	switch (phm->function) {
 	case HPI_SUBSYS_CREATE_ADAPTER:
 		subsys_create_adapter(phm, phr);
 		break;
-	case HPI_SUBSYS_DELETE_ADAPTER:
-		subsys_delete_adapter(phm, phr);
-		break;
 	default:
 		phr->error = HPI_ERROR_INVALID_FUNC;
 		break;
@@ -279,6 +277,10 @@ static void adapter_message(struct hpi_adapter_obj *pao,
 	struct hpi_message *phm, struct hpi_response *phr)
 {
 	switch (phm->function) {
+	case HPI_ADAPTER_DELETE:
+		adapter_delete(pao, phm, phr);
+		break;
+
 	default:
 		hw_message(pao, phm, phr);
 		break;
@@ -371,36 +373,17 @@ static void instream_message(struct hpi_adapter_obj *pao,
 /** Entry point to this HPI backend
  * All calls to the HPI start here
  */
-void HPI_6205(struct hpi_message *phm, struct hpi_response *phr)
+void _HPI_6205(struct hpi_adapter_obj *pao, struct hpi_message *phm,
+	struct hpi_response *phr)
 {
-	struct hpi_adapter_obj *pao = NULL;
-
-	/* subsytem messages are processed by every HPI.
-	 * All other messages are ignored unless the adapter index matches
-	 * an adapter in the HPI
-	 */
-	/* HPI_DEBUG_LOG(DEBUG, "HPI Obj=%d, Func=%d\n", phm->wObject,
-	   phm->wFunction); */
-
-	/* if Dsp has crashed then do not communicate with it any more */
-	if (phm->object != HPI_OBJ_SUBSYSTEM) {
-		pao = hpi_find_adapter(phm->adapter_index);
-		if (!pao) {
-			HPI_DEBUG_LOG(DEBUG,
-				" %d,%d refused, for another HPI?\n",
-				phm->object, phm->function);
-			return;
-		}
-
-		if ((pao->dsp_crashed >= 10)
-			&& (phm->function != HPI_ADAPTER_DEBUG_READ)) {
-			/* allow last resort debug read even after crash */
-			hpi_init_response(phr, phm->object, phm->function,
-				HPI_ERROR_DSP_HARDWARE);
-			HPI_DEBUG_LOG(WARNING, " %d,%d dsp crashed.\n",
-				phm->object, phm->function);
-			return;
-		}
+	if (pao && (pao->dsp_crashed >= 10)
+		&& (phm->function != HPI_ADAPTER_DEBUG_READ)) {
+		/* allow last resort debug read even after crash */
+		hpi_init_response(phr, phm->object, phm->function,
+			HPI_ERROR_DSP_HARDWARE);
+		HPI_DEBUG_LOG(WARNING, " %d,%d dsp crashed.\n", phm->object,
+			phm->function);
+		return;
 	}
 
 	/* Init default response  */
@@ -412,7 +395,7 @@ void HPI_6205(struct hpi_message *phm, struct hpi_response *phr)
 	case HPI_TYPE_MESSAGE:
 		switch (phm->object) {
 		case HPI_OBJ_SUBSYSTEM:
-			subsys_message(phm, phr);
+			subsys_message(pao, phm, phr);
 			break;
 
 		case HPI_OBJ_ADAPTER:
@@ -444,6 +427,26 @@ void HPI_6205(struct hpi_message *phm, struct hpi_response *phr)
 	}
 }
 
+void HPI_6205(struct hpi_message *phm, struct hpi_response *phr)
+{
+	struct hpi_adapter_obj *pao = NULL;
+
+	if (phm->object != HPI_OBJ_SUBSYSTEM) {
+		/* normal messages must have valid adapter index */
+		pao = hpi_find_adapter(phm->adapter_index);
+	} else {
+		/* subsys messages don't address an adapter */
+		_HPI_6205(NULL, phm, phr);
+		return;
+	}
+
+	if (pao)
+		_HPI_6205(pao, phm, phr);
+	else
+		hpi_init_response(phr, phm->object, phm->function,
+			HPI_ERROR_BAD_ADAPTER_NUMBER);
+}
+
 /*****************************************************************************/
 /* SUBSYSTEM */
 
@@ -491,13 +494,11 @@ static void subsys_create_adapter(struct hpi_message *phm,
 }
 
 /** delete an adapter - required by WDM driver */
-static void subsys_delete_adapter(struct hpi_message *phm,
-	struct hpi_response *phr)
+static void adapter_delete(struct hpi_adapter_obj *pao,
+	struct hpi_message *phm, struct hpi_response *phr)
 {
-	struct hpi_adapter_obj *pao;
 	struct hpi_hw_obj *phw;
 
-	pao = hpi_find_adapter(phm->obj_index);
 	if (!pao) {
 		phr->error = HPI_ERROR_INVALID_OBJ_INDEX;
 		return;
@@ -563,11 +564,12 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao,
 	}
 
 	err = adapter_boot_load_dsp(pao, pos_error_code);
-	if (err)
+	if (err) {
+		HPI_DEBUG_LOG(ERROR, "DSP code load failed\n");
 		/* no need to clean up as SubSysCreateAdapter */
 		/* calls DeleteAdapter on error. */
 		return err;
-
+	}
 	HPI_DEBUG_LOG(INFO, "load DSP code OK\n");
 
 	/* allow boot load even if mem alloc wont work */
@@ -604,6 +606,7 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao,
 				control_cache.number_of_controls,
 				interface->control_cache.size_in_bytes,
 				p_control_cache_virtual);
+
 			if (!phw->p_cache)
 				err = HPI_ERROR_MEMORY_ALLOC;
 		}
@@ -675,16 +678,14 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao,
 }
 
 /** Free memory areas allocated by adapter
- * this routine is called from SubSysDeleteAdapter,
+ * this routine is called from AdapterDelete,
   * and SubSysCreateAdapter if duplicate index
 */
 static void delete_adapter_obj(struct hpi_adapter_obj *pao)
 {
-	struct hpi_hw_obj *phw;
+	struct hpi_hw_obj *phw = pao->priv;
 	int i;
 
-	phw = pao->priv;
-
 	if (hpios_locked_mem_valid(&phw->h_control_cache)) {
 		hpios_locked_mem_free(&phw->h_control_cache);
 		hpi_free_control_cache(phw->p_cache);
@@ -1275,6 +1276,7 @@ static u16 adapter_boot_load_dsp(struct hpi_adapter_obj *pao,
 	case HPI_ADAPTER_FAMILY_ASI(0x6300):
 		boot_code_id[1] = HPI_ADAPTER_FAMILY_ASI(0x6400);
 		break;
+	case HPI_ADAPTER_FAMILY_ASI(0x5500):
 	case HPI_ADAPTER_FAMILY_ASI(0x5600):
 	case HPI_ADAPTER_FAMILY_ASI(0x6500):
 		boot_code_id[1] = HPI_ADAPTER_FAMILY_ASI(0x6600);
@@ -2059,7 +2061,6 @@ static int wait_dsp_ack(struct hpi_hw_obj *phw, int state, int timeout_us)
 static void send_dsp_command(struct hpi_hw_obj *phw, int cmd)
 {
 	struct bus_master_interface *interface = phw->p_interface_buffer;
-
 	u32 r;
 
 	interface->host_cmd = cmd;
diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index 3b9fd115da3..bf5eced76ba 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -294,7 +294,7 @@ enum HPI_CONTROL_ATTRIBUTES {
 
 /* These defines are used to fill in protocol information for an Ethernet packet
     sent using HMI on CS18102 */
-/** ID supplied by Cirrius for ASI packets. */
+/** ID supplied by Cirrus for ASI packets. */
 #define HPI_ETHERNET_PACKET_ID                  0x85
 /** Simple packet - no special routing required */
 #define HPI_ETHERNET_PACKET_V1                  0x01
@@ -307,7 +307,7 @@ enum HPI_CONTROL_ATTRIBUTES {
 /** This packet must make its way to the host across the HPI interface */
 #define HPI_ETHERNET_PACKET_HOSTED_VIA_HPI_V1   0x41
 
-#define HPI_ETHERNET_UDP_PORT (44600)	/*!< UDP messaging port */
+#define HPI_ETHERNET_UDP_PORT 44600 /**< HPI UDP service */
 
 /** Default network timeout in milli-seconds. */
 #define HPI_ETHERNET_TIMEOUT_MS 500
@@ -397,14 +397,14 @@ enum HPI_FUNCTION_IDS {
 	HPI_SUBSYS_OPEN = HPI_FUNC_ID(SUBSYSTEM, 1),
 	HPI_SUBSYS_GET_VERSION = HPI_FUNC_ID(SUBSYSTEM, 2),
 	HPI_SUBSYS_GET_INFO = HPI_FUNC_ID(SUBSYSTEM, 3),
-	HPI_SUBSYS_FIND_ADAPTERS = HPI_FUNC_ID(SUBSYSTEM, 4),
+	/* HPI_SUBSYS_FIND_ADAPTERS     = HPI_FUNC_ID(SUBSYSTEM, 4), */
 	HPI_SUBSYS_CREATE_ADAPTER = HPI_FUNC_ID(SUBSYSTEM, 5),
 	HPI_SUBSYS_CLOSE = HPI_FUNC_ID(SUBSYSTEM, 6),
-	HPI_SUBSYS_DELETE_ADAPTER = HPI_FUNC_ID(SUBSYSTEM, 7),
+	/* HPI_SUBSYS_DELETE_ADAPTER    = HPI_FUNC_ID(SUBSYSTEM, 7), */
 	HPI_SUBSYS_DRIVER_LOAD = HPI_FUNC_ID(SUBSYSTEM, 8),
 	HPI_SUBSYS_DRIVER_UNLOAD = HPI_FUNC_ID(SUBSYSTEM, 9),
-	HPI_SUBSYS_READ_PORT_8 = HPI_FUNC_ID(SUBSYSTEM, 10),
-	HPI_SUBSYS_WRITE_PORT_8 = HPI_FUNC_ID(SUBSYSTEM, 11),
+	/* HPI_SUBSYS_READ_PORT_8               = HPI_FUNC_ID(SUBSYSTEM, 10), */
+	/* HPI_SUBSYS_WRITE_PORT_8              = HPI_FUNC_ID(SUBSYSTEM, 11), */
 	HPI_SUBSYS_GET_NUM_ADAPTERS = HPI_FUNC_ID(SUBSYSTEM, 12),
 	HPI_SUBSYS_GET_ADAPTER = HPI_FUNC_ID(SUBSYSTEM, 13),
 	HPI_SUBSYS_SET_NETWORK_INTERFACE = HPI_FUNC_ID(SUBSYSTEM, 14),
@@ -433,7 +433,8 @@ enum HPI_FUNCTION_IDS {
 	HPI_ADAPTER_DEBUG_READ = HPI_FUNC_ID(ADAPTER, 18),
 	HPI_ADAPTER_IRQ_QUERY_AND_CLEAR = HPI_FUNC_ID(ADAPTER, 19),
 	HPI_ADAPTER_IRQ_CALLBACK = HPI_FUNC_ID(ADAPTER, 20),
-#define HPI_ADAPTER_FUNCTION_COUNT 20
+	HPI_ADAPTER_DELETE = HPI_FUNC_ID(ADAPTER, 21),
+#define HPI_ADAPTER_FUNCTION_COUNT 21
 
 	HPI_OSTREAM_OPEN = HPI_FUNC_ID(OSTREAM, 1),
 	HPI_OSTREAM_CLOSE = HPI_FUNC_ID(OSTREAM, 2),
@@ -1561,8 +1562,6 @@ void hpi_send_recv(struct hpi_message *phm, struct hpi_response *phr);
 u16 hpi_subsys_create_adapter(const struct hpi_resource *p_resource,
 	u16 *pw_adapter_index);
 
-u16 hpi_subsys_delete_adapter(u16 adapter_index);
-
 u16 hpi_outstream_host_buffer_get_info(u32 h_outstream, u8 **pp_buffer,
 	struct hpi_hostbuffer_status **pp_status);
 
@@ -1584,9 +1583,7 @@ void hpi_stream_response_to_legacy(struct hpi_stream_res *pSR);
 
 /*////////////////////////////////////////////////////////////////////////// */
 /* declarations for individual HPI entry points */
-hpi_handler_func HPI_1000;
 hpi_handler_func HPI_6000;
 hpi_handler_func HPI_6205;
-hpi_handler_func HPI_COMMON;
 
 #endif				/* _HPI_INTERNAL_H_ */
diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index 3e9c5c28976..b15a02e91f8 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -227,8 +227,9 @@ static unsigned int control_cache_alloc_check(struct hpi_control_cache *pC)
 			if (info->control_type) {
 				pC->p_info[info->control_index] = info;
 				cached++;
-			} else	/* dummy cache entry */
+			} else {	/* dummy cache entry */
 				pC->p_info[info->control_index] = NULL;
+			}
 
 			byte_count += info->size_in32bit_words * 4;
 
@@ -298,7 +299,7 @@ struct pad_ofs_size {
 	unsigned int field_size;
 };
 
-static struct pad_ofs_size pad_desc[] = {
+static const struct pad_ofs_size pad_desc[] = {
 	HPICMN_PAD_OFS_AND_SIZE(c_channel),	/* HPI_PAD_CHANNEL_NAME */
 	HPICMN_PAD_OFS_AND_SIZE(c_artist),	/* HPI_PAD_ARTIST */
 	HPICMN_PAD_OFS_AND_SIZE(c_title),	/* HPI_PAD_TITLE */
@@ -617,6 +618,10 @@ void hpi_cmn_control_cache_sync_to_msg(struct hpi_control_cache *p_cache,
 	}
 }
 
+/** Allocate control cache.
+
+\return Cache pointer, or NULL if allocation fails.
+*/
 struct hpi_control_cache *hpi_alloc_control_cache(const u32 control_count,
 	const u32 size_in_bytes, u8 *p_dsp_control_buffer)
 {
@@ -667,7 +672,6 @@ static void subsys_message(struct hpi_message *phm, struct hpi_response *phr)
 		phr->u.s.num_adapters = adapters.gw_num_adapters;
 		break;
 	case HPI_SUBSYS_CREATE_ADAPTER:
-	case HPI_SUBSYS_DELETE_ADAPTER:
 		break;
 	default:
 		phr->error = HPI_ERROR_INVALID_FUNC;
diff --git a/sound/pci/asihpi/hpicmn.h b/sound/pci/asihpi/hpicmn.h
index 590f0b69e65..d53cdf6e535 100644
--- a/sound/pci/asihpi/hpicmn.h
+++ b/sound/pci/asihpi/hpicmn.h
@@ -60,3 +60,5 @@ void hpi_cmn_control_cache_sync_to_msg(struct hpi_control_cache *pC,
 	struct hpi_message *phm, struct hpi_response *phr);
 
 u16 hpi_validate_response(struct hpi_message *phm, struct hpi_response *phr);
+
+hpi_handler_func HPI_COMMON;
diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c
index c38fc948756..7397b169b89 100644
--- a/sound/pci/asihpi/hpifunc.c
+++ b/sound/pci/asihpi/hpifunc.c
@@ -105,33 +105,6 @@ u16 hpi_subsys_get_version_ex(u32 *pversion_ex)
 	return hr.error;
 }
 
-u16 hpi_subsys_create_adapter(const struct hpi_resource *p_resource,
-	u16 *pw_adapter_index)
-{
-	struct hpi_message hm;
-	struct hpi_response hr;
-
-	hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM,
-		HPI_SUBSYS_CREATE_ADAPTER);
-	hm.u.s.resource = *p_resource;
-
-	hpi_send_recv(&hm, &hr);
-
-	*pw_adapter_index = hr.u.s.adapter_index;
-	return hr.error;
-}
-
-u16 hpi_subsys_delete_adapter(u16 adapter_index)
-{
-	struct hpi_message hm;
-	struct hpi_response hr;
-	hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM,
-		HPI_SUBSYS_DELETE_ADAPTER);
-	hm.obj_index = adapter_index;
-	hpi_send_recv(&hm, &hr);
-	return hr.error;
-}
-
 u16 hpi_subsys_get_num_adapters(int *pn_num_adapters)
 {
 	struct hpi_message hm;
diff --git a/sound/pci/asihpi/hpimsgx.c b/sound/pci/asihpi/hpimsgx.c
index 360028b9abf..7352a5f7b4f 100644
--- a/sound/pci/asihpi/hpimsgx.c
+++ b/sound/pci/asihpi/hpimsgx.c
@@ -211,24 +211,6 @@ static void subsys_message(struct hpi_message *phm, struct hpi_response *phr,
 		HPIMSGX__init(phm, phr);
 		break;
 
-	case HPI_SUBSYS_DELETE_ADAPTER:
-		HPIMSGX__cleanup(phm->obj_index, h_owner);
-		{
-			struct hpi_message hm;
-			struct hpi_response hr;
-			hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER,
-				HPI_ADAPTER_CLOSE);
-			hm.adapter_index = phm->obj_index;
-			hw_entry_point(&hm, &hr);
-		}
-		if ((phm->obj_index < HPI_MAX_ADAPTERS)
-			&& hpi_entry_points[phm->obj_index]) {
-			hpi_entry_points[phm->obj_index] (phm, phr);
-			hpi_entry_points[phm->obj_index] = NULL;
-		} else
-			phr->error = HPI_ERROR_INVALID_OBJ_INDEX;
-
-		break;
 	default:
 		/* Must explicitly handle every subsys message in this switch */
 		hpi_init_response(phr, HPI_OBJ_SUBSYSTEM, phm->function,
@@ -247,6 +229,19 @@ static void adapter_message(struct hpi_message *phm, struct hpi_response *phr,
 	case HPI_ADAPTER_CLOSE:
 		adapter_close(phm, phr);
 		break;
+	case HPI_ADAPTER_DELETE:
+		HPIMSGX__cleanup(phm->adapter_index, h_owner);
+		{
+			struct hpi_message hm;
+			struct hpi_response hr;
+			hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER,
+				HPI_ADAPTER_CLOSE);
+			hm.adapter_index = phm->adapter_index;
+			hw_entry_point(&hm, &hr);
+		}
+		hw_entry_point(phm, phr);
+		break;
+
 	default:
 		hw_entry_point(phm, phr);
 		break;
diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index cd624f13ff8..d8e7047512f 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -25,6 +25,7 @@ Common Linux HPI ioctl and module probe/remove functions
 #include "hpidebug.h"
 #include "hpimsgx.h"
 #include "hpioctl.h"
+#include "hpicmn.h"
 
 #include <linux/fs.h>
 #include <linux/slab.h>
@@ -161,26 +162,24 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
 		goto out;
 	}
 
-	pa = &adapters[hm->h.adapter_index];
+	switch (hm->h.function) {
+	case HPI_SUBSYS_CREATE_ADAPTER:
+	case HPI_ADAPTER_DELETE:
+		/* Application must not use these functions! */
+		hr->h.size = sizeof(hr->h);
+		hr->h.error = HPI_ERROR_INVALID_OPERATION;
+		hr->h.function = hm->h.function;
+		uncopied_bytes = copy_to_user(puhr, hr, hr->h.size);
+		if (uncopied_bytes)
+			err = -EFAULT;
+		else
+			err = 0;
+		goto out;
+	}
+
 	hr->h.size = res_max_size;
 	if (hm->h.object == HPI_OBJ_SUBSYSTEM) {
-		switch (hm->h.function) {
-		case HPI_SUBSYS_CREATE_ADAPTER:
-		case HPI_SUBSYS_DELETE_ADAPTER:
-			/* Application must not use these functions! */
-			hr->h.size = sizeof(hr->h);
-			hr->h.error = HPI_ERROR_INVALID_OPERATION;
-			hr->h.function = hm->h.function;
-			uncopied_bytes = copy_to_user(puhr, hr, hr->h.size);
-			if (uncopied_bytes)
-				err = -EFAULT;
-			else
-				err = 0;
-			goto out;
-
-		default:
-			hpi_send_recv_f(&hm->m0, &hr->r0, file);
-		}
+		hpi_send_recv_f(&hm->m0, &hr->r0, file);
 	} else {
 		u16 __user *ptr = NULL;
 		u32 size = 0;
@@ -188,8 +187,9 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
 		/* -1=no data 0=read from user mem, 1=write to user mem */
 		int wrflag = -1;
 		u32 adapter = hm->h.adapter_index;
+		pa = &adapters[adapter];
 
-		if ((hm->h.adapter_index > HPI_MAX_ADAPTERS) || (!pa->type)) {
+		if ((adapter > HPI_MAX_ADAPTERS) || (!pa->type)) {
 			hpi_init_response(&hr->r0, HPI_OBJ_ADAPTER,
 				HPI_ADAPTER_OPEN,
 				HPI_ERROR_BAD_ADAPTER_NUMBER);
@@ -317,7 +317,7 @@ out:
 int __devinit asihpi_adapter_probe(struct pci_dev *pci_dev,
 	const struct pci_device_id *pci_id)
 {
-	int err, idx, nm;
+	int idx, nm;
 	unsigned int memlen;
 	struct hpi_message hm;
 	struct hpi_response hr;
@@ -351,11 +351,8 @@ int __devinit asihpi_adapter_probe(struct pci_dev *pci_dev,
 	nm = HPI_MAX_ADAPTER_MEM_SPACES;
 
 	for (idx = 0; idx < nm; idx++) {
-		HPI_DEBUG_LOG(INFO, "resource %d %s %08llx-%08llx %04llx\n",
-			idx, pci_dev->resource[idx].name,
-			(unsigned long long)pci_resource_start(pci_dev, idx),
-			(unsigned long long)pci_resource_end(pci_dev, idx),
-			(unsigned long long)pci_resource_flags(pci_dev, idx));
+		HPI_DEBUG_LOG(INFO, "resource %d %pR\n", idx,
+			&pci_dev->resource[idx]);
 
 		if (pci_resource_flags(pci_dev, idx) & IORESOURCE_MEM) {
 			memlen = pci_resource_len(pci_dev, idx);
@@ -395,17 +392,20 @@ int __devinit asihpi_adapter_probe(struct pci_dev *pci_dev,
 
 	adapter.index = hr.u.s.adapter_index;
 	adapter.type = hr.u.s.adapter_type;
+
+	hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER,
+		HPI_ADAPTER_OPEN);
 	hm.adapter_index = adapter.index;
+	hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL);
 
-	err = hpi_adapter_open(adapter.index);
-	if (err)
+	if (hr.error)
 		goto err;
 
 	adapter.snd_card_asihpi = NULL;
 	/* WARNING can't init mutex in 'adapter'
 	 * and then copy it to adapters[] ?!?!
 	 */
-	adapters[hr.u.s.adapter_index] = adapter;
+	adapters[adapter.index] = adapter;
 	mutex_init(&adapters[adapter.index].mutex);
 	pci_set_drvdata(pci_dev, &adapters[adapter.index]);
 
@@ -440,10 +440,9 @@ void __devexit asihpi_adapter_remove(struct pci_dev *pci_dev)
 	struct hpi_adapter *pa;
 	pa = pci_get_drvdata(pci_dev);
 
-	hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM,
-		HPI_SUBSYS_DELETE_ADAPTER);
-	hm.obj_index = pa->index;
-	hm.adapter_index = HPI_ADAPTER_INDEX_INVALID;
+	hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER,
+		HPI_ADAPTER_DELETE);
+	hm.adapter_index = pa->index;
 	hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL);
 
 	/* unmap PCI memory space, mapped during device init. */
diff --git a/sound/pci/au88x0/au8810.h b/sound/pci/au88x0/au8810.h
index 5d69c31fe3f..79fbee3845e 100644
--- a/sound/pci/au88x0/au8810.h
+++ b/sound/pci/au88x0/au8810.h
@@ -4,7 +4,7 @@
 
 #define CHIP_AU8810
 
-#define CARD_NAME "Aureal Advantage 3D Sound Processor"
+#define CARD_NAME "Aureal Advantage"
 #define CARD_NAME_SHORT "au8810"
 
 #define NR_ADB		0x10
diff --git a/sound/pci/au88x0/au8820.h b/sound/pci/au88x0/au8820.h
index abbe85e4f7a..cafdb9668a3 100644
--- a/sound/pci/au88x0/au8820.h
+++ b/sound/pci/au88x0/au8820.h
@@ -11,7 +11,7 @@
 
 #define CHIP_AU8820
 
-#define CARD_NAME "Aureal Vortex 3D Sound Processor"
+#define CARD_NAME "Aureal Vortex"
 #define CARD_NAME_SHORT "au8820"
 
 /* Number of ADB and WT channels */
diff --git a/sound/pci/au88x0/au8830.h b/sound/pci/au88x0/au8830.h
index 04ece1b1c21..999b29ab34a 100644
--- a/sound/pci/au88x0/au8830.h
+++ b/sound/pci/au88x0/au8830.h
@@ -11,7 +11,7 @@
 
 #define CHIP_AU8830
 
-#define CARD_NAME "Aureal Vortex 2 3D Sound Processor"
+#define CARD_NAME "Aureal Vortex 2"
 #define CARD_NAME_SHORT "au8830"
 
 #define NR_ADB 0x20
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index 33f0ba5559a..c5f7ae46afe 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -44,10 +44,10 @@ static struct snd_pcm_hardware snd_vortex_playback_hw_adb = {
 	.channels_min = 1,
 	.channels_max = 2,
 	.buffer_bytes_max = 0x10000,
-	.period_bytes_min = 0x1,
+	.period_bytes_min = 0x20,
 	.period_bytes_max = 0x1000,
 	.periods_min = 2,
-	.periods_max = 32,
+	.periods_max = 1024,
 };
 
 #ifndef CHIP_AU8820
@@ -140,6 +140,9 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream)
 					SNDRV_PCM_HW_PARAM_PERIOD_BYTES)) < 0)
 		return err;
 
+	snd_pcm_hw_constraint_step(runtime, 0,
+					SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 64);
+
 	if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) {
 #ifndef CHIP_AU8820
 		if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_A3D) {
@@ -423,11 +426,11 @@ static struct snd_pcm_ops snd_vortex_playback_ops = {
 */
 
 static char *vortex_pcm_prettyname[VORTEX_PCM_LAST] = {
-	"AU88x0 ADB",
-	"AU88x0 SPDIF",
-	"AU88x0 A3D",
-	"AU88x0 WT",
-	"AU88x0 I2S",
+	CARD_NAME " ADB",
+	CARD_NAME " SPDIF",
+	CARD_NAME " A3D",
+	CARD_NAME " WT",
+	CARD_NAME " I2S",
 };
 static char *vortex_pcm_name[VORTEX_PCM_LAST] = {
 	"adb",
@@ -524,7 +527,8 @@ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr)
 			  nr_capt, &pcm);
 	if (err < 0)
 		return err;
-	strcpy(pcm->name, vortex_pcm_name[idx]);
+	snprintf(pcm->name, sizeof(pcm->name),
+		"%s %s", CARD_NAME_SHORT, vortex_pcm_name[idx]);
 	chip->pcm[idx] = pcm;
 	// This is an evil hack, but it saves a lot of duplicated code.
 	VORTEX_PCM_TYPE(pcm) = idx;
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index 7a9401462c1..dae4050ede5 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -303,6 +303,9 @@ static const u32 db_table[101] = {
 static const DECLARE_TLV_DB_SCALE(snd_emu10k1_db_scale1, -4000, 40, 1);
 static const DECLARE_TLV_DB_LINEAR(snd_emu10k1_db_linear, TLV_DB_GAIN_MUTE, 0);
 
+/* EMU10K1 bass/treble db gain */
+static const DECLARE_TLV_DB_SCALE(snd_emu10k1_bass_treble_db_scale, -1200, 60, 0);
+
 static const u32 onoff_table[2] = {
 	0x00000000, 0x00000001
 };
@@ -2163,6 +2166,7 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu)
 	ctl->min = 0;
 	ctl->max = 40;
 	ctl->value[0] = ctl->value[1] = 20;
+	ctl->tlv = snd_emu10k1_bass_treble_db_scale;
 	ctl->translation = EMU10K1_GPR_TRANSLATION_BASS;
 	ctl = &controls[i + 1];
 	ctl->id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
@@ -2172,6 +2176,7 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu)
 	ctl->min = 0;
 	ctl->max = 40;
 	ctl->value[0] = ctl->value[1] = 20;
+	ctl->tlv = snd_emu10k1_bass_treble_db_scale;
 	ctl->translation = EMU10K1_GPR_TRANSLATION_TREBLE;
 
 #define BASS_GPR	0x8c
diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c
index 05afe06e353..9d890a5aec5 100644
--- a/sound/pci/emu10k1/emumixer.c
+++ b/sound/pci/emu10k1/emumixer.c
@@ -1729,8 +1729,6 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu,
 		"Master Mono Playback Volume",
 		"PCM Out Path & Mute",
 		"Mono Output Select",
-		"Front Playback Switch",
-		"Front Playback Volume",
 		"Surround Playback Switch",
 		"Surround Playback Volume",
 		"Center Playback Switch",
@@ -1879,6 +1877,8 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu,
 				emu->rear_ac97 = 1;
 				snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE|AC97SLOT_REAR_LEFT|AC97SLOT_REAR_RIGHT);
 				snd_ac97_write_cache(emu->ac97, AC97_HEADPHONE, 0x0202);
+				remove_ctl(card,"Front Playback Volume");
+				remove_ctl(card,"Front Playback Switch");
 			}
 			/* remove unused AC97 controls */
 			snd_ac97_write_cache(emu->ac97, AC97_SURROUND_MASTER, 0x0202);
@@ -1913,6 +1913,12 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu,
 	for (; *c; c += 2)
 		rename_ctl(card, c[0], c[1]);
 
+	if (emu->card_capabilities->subsystem == 0x80401102) { /* SB Live! Platinum CT4760P */
+		remove_ctl(card, "Center Playback Volume");
+		remove_ctl(card, "LFE Playback Volume");
+		remove_ctl(card, "Wave Center Playback Volume");
+		remove_ctl(card, "Wave LFE Playback Volume");
+	}
 	if (emu->card_capabilities->subsystem == 0x20071102) {  /* Audigy 4 Pro */
 		rename_ctl(card, "Line2 Capture Volume", "Line1/Mic Capture Volume");
 		rename_ctl(card, "Analog Mix Capture Volume", "Line2 Capture Volume");
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 7c17f45d876..ab0a6156a70 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -112,6 +112,10 @@
 #include <sound/ac97_codec.h>
 #include <sound/initval.h>
 
+#ifdef CONFIG_SND_ES1968_RADIO
+#include <sound/tea575x-tuner.h>
+#endif
+
 #define CARD_NAME "ESS Maestro1/2"
 #define DRIVER_NAME "ES1968"
 
@@ -553,6 +557,10 @@ struct es1968 {
 	spinlock_t ac97_lock;
 	struct tasklet_struct hwvol_tq;
 #endif
+
+#ifdef CONFIG_SND_ES1968_RADIO
+	struct snd_tea575x tea;
+#endif
 };
 
 static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id);
@@ -2571,6 +2579,63 @@ static int __devinit snd_es1968_input_register(struct es1968 *chip)
 }
 #endif /* CONFIG_SND_ES1968_INPUT */
 
+#ifdef CONFIG_SND_ES1968_RADIO
+#define GPIO_DATA	0x60
+#define IO_MASK		4      /* mask      register offset from GPIO_DATA
+				bits 1=unmask write to given bit */
+#define IO_DIR		8      /* direction register offset from GPIO_DATA
+				bits 0/1=read/write direction */
+/* mask bits for GPIO lines */
+#define STR_DATA	0x0040 /* GPIO6 */
+#define STR_CLK		0x0080 /* GPIO7 */
+#define STR_WREN	0x0100 /* GPIO8 */
+#define STR_MOST	0x0200 /* GPIO9 */
+
+static void snd_es1968_tea575x_set_pins(struct snd_tea575x *tea, u8 pins)
+{
+	struct es1968 *chip = tea->private_data;
+	unsigned long io = chip->io_port + GPIO_DATA;
+	u16 val = 0;
+
+	val |= (pins & TEA575X_DATA) ? STR_DATA : 0;
+	val |= (pins & TEA575X_CLK)  ? STR_CLK  : 0;
+	val |= (pins & TEA575X_WREN) ? STR_WREN : 0;
+
+	outw(val, io);
+}
+
+static u8 snd_es1968_tea575x_get_pins(struct snd_tea575x *tea)
+{
+	struct es1968 *chip = tea->private_data;
+	unsigned long io = chip->io_port + GPIO_DATA;
+	u16 val = inw(io);
+
+	return  (val & STR_DATA) ? TEA575X_DATA : 0 |
+		(val & STR_MOST) ? TEA575X_MOST : 0;
+}
+
+static void snd_es1968_tea575x_set_direction(struct snd_tea575x *tea, bool output)
+{
+	struct es1968 *chip = tea->private_data;
+	unsigned long io = chip->io_port + GPIO_DATA;
+	u16 odir = inw(io + IO_DIR);
+
+	if (output) {
+		outw(~(STR_DATA | STR_CLK | STR_WREN), io + IO_MASK);
+		outw(odir | STR_DATA | STR_CLK | STR_WREN, io + IO_DIR);
+	} else {
+		outw(~(STR_CLK | STR_WREN | STR_DATA | STR_MOST), io + IO_MASK);
+		outw((odir & ~(STR_DATA | STR_MOST)) | STR_CLK | STR_WREN, io + IO_DIR);
+	}
+}
+
+static struct snd_tea575x_ops snd_es1968_tea_ops = {
+	.set_pins = snd_es1968_tea575x_set_pins,
+	.get_pins = snd_es1968_tea575x_get_pins,
+	.set_direction = snd_es1968_tea575x_set_direction,
+};
+#endif
+
 static int snd_es1968_free(struct es1968 *chip)
 {
 #ifdef CONFIG_SND_ES1968_INPUT
@@ -2585,6 +2650,10 @@ static int snd_es1968_free(struct es1968 *chip)
 		outw(0, chip->io_port + ESM_PORT_HOST_IRQ); /* disable IRQ */
 	}
 
+#ifdef CONFIG_SND_ES1968_RADIO
+	snd_tea575x_exit(&chip->tea);
+#endif
+
 	if (chip->irq >= 0)
 		free_irq(chip->irq, chip);
 	snd_es1968_free_gameport(chip);
@@ -2723,6 +2792,15 @@ static int __devinit snd_es1968_create(struct snd_card *card,
 
 	snd_card_set_dev(card, &pci->dev);
 
+#ifdef CONFIG_SND_ES1968_RADIO
+	chip->tea.private_data = chip;
+	chip->tea.ops = &snd_es1968_tea_ops;
+	strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card));
+	sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci));
+	if (!snd_tea575x_init(&chip->tea))
+		printk(KERN_INFO "es1968: detected TEA575x radio\n");
+#endif
+
 	*chip_ret = chip;
 
 	return 0;
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index e1baad74ea4..eacd4901a30 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -38,7 +38,6 @@
 
 #ifdef CONFIG_SND_FM801_TEA575X_BOOL
 #include <sound/tea575x-tuner.h>
-#define TEA575X_RADIO 1
 #endif
 
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
@@ -53,7 +52,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;	/* Enable this card *
 /*
  *  Enable TEA575x tuner
  *    1 = MediaForte 256-PCS
- *    2 = MediaForte 256-PCPR
+ *    2 = MediaForte 256-PCP
  *    3 = MediaForte 64-PCR
  *   16 = setup tuner only (this is additional bit), i.e. SF64-PCR FM card
  *  High 16-bits are video (radio) device number + 1
@@ -67,7 +66,7 @@ MODULE_PARM_DESC(id, "ID string for the FM801 soundcard.");
 module_param_array(enable, bool, NULL, 0444);
 MODULE_PARM_DESC(enable, "Enable FM801 soundcard.");
 module_param_array(tea575x_tuner, int, NULL, 0444);
-MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (1 = SF256-PCS, 2=SF256-PCPR, 3=SF64-PCR, +16=tuner-only).");
+MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only).");
 
 #define TUNER_ONLY		(1<<4)
 #define TUNER_TYPE_MASK		(~TUNER_ONLY & 0xFFFF)
@@ -196,7 +195,7 @@ struct fm801 {
 	spinlock_t reg_lock;
 	struct snd_info_entry *proc_entry;
 
-#ifdef TEA575X_RADIO
+#ifdef CONFIG_SND_FM801_TEA575X_BOOL
 	struct snd_tea575x tea;
 #endif
 
@@ -715,310 +714,89 @@ static int __devinit snd_fm801_pcm(struct fm801 *chip, int device, struct snd_pc
  *  TEA5757 radio
  */
 
-#ifdef TEA575X_RADIO
-
-/* 256PCS GPIO numbers */
-#define TEA_256PCS_DATA			1
-#define TEA_256PCS_WRITE_ENABLE		2	/* inverted */
-#define TEA_256PCS_BUS_CLOCK		3
-
-static void snd_fm801_tea575x_256pcs_write(struct snd_tea575x *tea, unsigned int val)
-{
-	struct fm801 *chip = tea->private_data;
-	unsigned short reg;
-	int i = 25;
+#ifdef CONFIG_SND_FM801_TEA575X_BOOL
 
-	spin_lock_irq(&chip->reg_lock);
-	reg = inw(FM801_REG(chip, GPIO_CTRL));
-	/* use GPIO lines and set write enable bit */
-	reg |= FM801_GPIO_GS(TEA_256PCS_DATA) |
-	       FM801_GPIO_GS(TEA_256PCS_WRITE_ENABLE) |
-	       FM801_GPIO_GS(TEA_256PCS_BUS_CLOCK);
-	/* all of lines are in the write direction */
-	/* clear data and clock lines */
-	reg &= ~(FM801_GPIO_GD(TEA_256PCS_DATA) |
-	         FM801_GPIO_GD(TEA_256PCS_WRITE_ENABLE) |
-	         FM801_GPIO_GD(TEA_256PCS_BUS_CLOCK) |
-	         FM801_GPIO_GP(TEA_256PCS_DATA) |
-	         FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK) |
-		 FM801_GPIO_GP(TEA_256PCS_WRITE_ENABLE));
-	outw(reg, FM801_REG(chip, GPIO_CTRL));
-	udelay(1);
-
-	while (i--) {
-		if (val & (1 << i))
-			reg |= FM801_GPIO_GP(TEA_256PCS_DATA);
-		else
-			reg &= ~FM801_GPIO_GP(TEA_256PCS_DATA);
-		outw(reg, FM801_REG(chip, GPIO_CTRL));
-		udelay(1);
-		reg |= FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK);
-		outw(reg, FM801_REG(chip, GPIO_CTRL));
-		reg &= ~FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK);
-		outw(reg, FM801_REG(chip, GPIO_CTRL));
-		udelay(1);
-	}
+/* GPIO to TEA575x maps */
+struct snd_fm801_tea575x_gpio {
+	u8 data, clk, wren, most;
+	char *name;
+};
 
-	/* and reset the write enable bit */
-	reg |= FM801_GPIO_GP(TEA_256PCS_WRITE_ENABLE) |
-	       FM801_GPIO_GP(TEA_256PCS_DATA);
-	outw(reg, FM801_REG(chip, GPIO_CTRL));
-	spin_unlock_irq(&chip->reg_lock);
-}
+static struct snd_fm801_tea575x_gpio snd_fm801_tea575x_gpios[] = {
+	{ .data = 1, .clk = 3, .wren = 2, .most = 0, .name = "SF256-PCS" },
+	{ .data = 1, .clk = 0, .wren = 2, .most = 3, .name = "SF256-PCP" },
+	{ .data = 2, .clk = 0, .wren = 1, .most = 3, .name = "SF64-PCR" },
+};
 
-static unsigned int snd_fm801_tea575x_256pcs_read(struct snd_tea575x *tea)
+static void snd_fm801_tea575x_set_pins(struct snd_tea575x *tea, u8 pins)
 {
 	struct fm801 *chip = tea->private_data;
-	unsigned short reg;
-	unsigned int val = 0;
-	int i;
-	
-	spin_lock_irq(&chip->reg_lock);
-	reg = inw(FM801_REG(chip, GPIO_CTRL));
-	/* use GPIO lines, set data direction to input */
-	reg |= FM801_GPIO_GS(TEA_256PCS_DATA) |
-	       FM801_GPIO_GS(TEA_256PCS_WRITE_ENABLE) |
-	       FM801_GPIO_GS(TEA_256PCS_BUS_CLOCK) |
-	       FM801_GPIO_GD(TEA_256PCS_DATA) |
-	       FM801_GPIO_GP(TEA_256PCS_DATA) |
-	       FM801_GPIO_GP(TEA_256PCS_WRITE_ENABLE);
-	/* all of lines are in the write direction, except data */
-	/* clear data, write enable and clock lines */
-	reg &= ~(FM801_GPIO_GD(TEA_256PCS_WRITE_ENABLE) |
-	         FM801_GPIO_GD(TEA_256PCS_BUS_CLOCK) |
-	         FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK));
-
-	for (i = 0; i < 24; i++) {
-		reg &= ~FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK);
-		outw(reg, FM801_REG(chip, GPIO_CTRL));
-		udelay(1);
-		reg |= FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK);
-		outw(reg, FM801_REG(chip, GPIO_CTRL));
-		udelay(1);
-		val <<= 1;
-		if (inw(FM801_REG(chip, GPIO_CTRL)) & FM801_GPIO_GP(TEA_256PCS_DATA))
-			val |= 1;
-	}
+	unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL));
+	struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1];
 
-	spin_unlock_irq(&chip->reg_lock);
+	reg &= ~(FM801_GPIO_GP(gpio.data) |
+		 FM801_GPIO_GP(gpio.clk) |
+		 FM801_GPIO_GP(gpio.wren));
 
-	return val;
-}
+	reg |= (pins & TEA575X_DATA) ? FM801_GPIO_GP(gpio.data) : 0;
+	reg |= (pins & TEA575X_CLK)  ? FM801_GPIO_GP(gpio.clk) : 0;
+	/* WRITE_ENABLE is inverted */
+	reg |= (pins & TEA575X_WREN) ? 0 : FM801_GPIO_GP(gpio.wren);
 
-/* 256PCPR GPIO numbers */
-#define TEA_256PCPR_BUS_CLOCK		0
-#define TEA_256PCPR_DATA		1
-#define TEA_256PCPR_WRITE_ENABLE	2	/* inverted */
-
-static void snd_fm801_tea575x_256pcpr_write(struct snd_tea575x *tea, unsigned int val)
-{
-	struct fm801 *chip = tea->private_data;
-	unsigned short reg;
-	int i = 25;
-
-	spin_lock_irq(&chip->reg_lock);
-	reg = inw(FM801_REG(chip, GPIO_CTRL));
-	/* use GPIO lines and set write enable bit */
-	reg |= FM801_GPIO_GS(TEA_256PCPR_DATA) |
-	       FM801_GPIO_GS(TEA_256PCPR_WRITE_ENABLE) |
-	       FM801_GPIO_GS(TEA_256PCPR_BUS_CLOCK);
-	/* all of lines are in the write direction */
-	/* clear data and clock lines */
-	reg &= ~(FM801_GPIO_GD(TEA_256PCPR_DATA) |
-	         FM801_GPIO_GD(TEA_256PCPR_WRITE_ENABLE) |
-	         FM801_GPIO_GD(TEA_256PCPR_BUS_CLOCK) |
-	         FM801_GPIO_GP(TEA_256PCPR_DATA) |
-	         FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK) |
-		 FM801_GPIO_GP(TEA_256PCPR_WRITE_ENABLE));
 	outw(reg, FM801_REG(chip, GPIO_CTRL));
-	udelay(1);
-
-	while (i--) {
-		if (val & (1 << i))
-			reg |= FM801_GPIO_GP(TEA_256PCPR_DATA);
-		else
-			reg &= ~FM801_GPIO_GP(TEA_256PCPR_DATA);
-		outw(reg, FM801_REG(chip, GPIO_CTRL));
-		udelay(1);
-		reg |= FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK);
-		outw(reg, FM801_REG(chip, GPIO_CTRL));
-		reg &= ~FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK);
-		outw(reg, FM801_REG(chip, GPIO_CTRL));
-		udelay(1);
-	}
-
-	/* and reset the write enable bit */
-	reg |= FM801_GPIO_GP(TEA_256PCPR_WRITE_ENABLE) |
-	       FM801_GPIO_GP(TEA_256PCPR_DATA);
-	outw(reg, FM801_REG(chip, GPIO_CTRL));
-	spin_unlock_irq(&chip->reg_lock);
 }
 
-static unsigned int snd_fm801_tea575x_256pcpr_read(struct snd_tea575x *tea)
+static u8 snd_fm801_tea575x_get_pins(struct snd_tea575x *tea)
 {
 	struct fm801 *chip = tea->private_data;
-	unsigned short reg;
-	unsigned int val = 0;
-	int i;
-	
-	spin_lock_irq(&chip->reg_lock);
-	reg = inw(FM801_REG(chip, GPIO_CTRL));
-	/* use GPIO lines, set data direction to input */
-	reg |= FM801_GPIO_GS(TEA_256PCPR_DATA) |
-	       FM801_GPIO_GS(TEA_256PCPR_WRITE_ENABLE) |
-	       FM801_GPIO_GS(TEA_256PCPR_BUS_CLOCK) |
-	       FM801_GPIO_GD(TEA_256PCPR_DATA) |
-	       FM801_GPIO_GP(TEA_256PCPR_DATA) |
-	       FM801_GPIO_GP(TEA_256PCPR_WRITE_ENABLE);
-	/* all of lines are in the write direction, except data */
-	/* clear data, write enable and clock lines */
-	reg &= ~(FM801_GPIO_GD(TEA_256PCPR_WRITE_ENABLE) |
-	         FM801_GPIO_GD(TEA_256PCPR_BUS_CLOCK) |
-	         FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK));
-
-	for (i = 0; i < 24; i++) {
-		reg &= ~FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK);
-		outw(reg, FM801_REG(chip, GPIO_CTRL));
-		udelay(1);
-		reg |= FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK);
-		outw(reg, FM801_REG(chip, GPIO_CTRL));
-		udelay(1);
-		val <<= 1;
-		if (inw(FM801_REG(chip, GPIO_CTRL)) & FM801_GPIO_GP(TEA_256PCPR_DATA))
-			val |= 1;
-	}
+	unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL));
+	struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1];
 
-	spin_unlock_irq(&chip->reg_lock);
-
-	return val;
+	return  (reg & FM801_GPIO_GP(gpio.data)) ? TEA575X_DATA : 0 |
+		(reg & FM801_GPIO_GP(gpio.most)) ? TEA575X_MOST : 0;
 }
 
-/* 64PCR GPIO numbers */
-#define TEA_64PCR_BUS_CLOCK		0
-#define TEA_64PCR_WRITE_ENABLE		1	/* inverted */
-#define TEA_64PCR_DATA			2
-
-static void snd_fm801_tea575x_64pcr_write(struct snd_tea575x *tea, unsigned int val)
+static void snd_fm801_tea575x_set_direction(struct snd_tea575x *tea, bool output)
 {
 	struct fm801 *chip = tea->private_data;
-	unsigned short reg;
-	int i = 25;
+	unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL));
+	struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1];
 
-	spin_lock_irq(&chip->reg_lock);
-	reg = inw(FM801_REG(chip, GPIO_CTRL));
 	/* use GPIO lines and set write enable bit */
-	reg |= FM801_GPIO_GS(TEA_64PCR_DATA) |
-	       FM801_GPIO_GS(TEA_64PCR_WRITE_ENABLE) |
-	       FM801_GPIO_GS(TEA_64PCR_BUS_CLOCK);
-	/* all of lines are in the write direction */
-	/* clear data and clock lines */
-	reg &= ~(FM801_GPIO_GD(TEA_64PCR_DATA) |
-	         FM801_GPIO_GD(TEA_64PCR_WRITE_ENABLE) |
-	         FM801_GPIO_GD(TEA_64PCR_BUS_CLOCK) |
-	         FM801_GPIO_GP(TEA_64PCR_DATA) |
-	         FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK) |
-		 FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE));
-	outw(reg, FM801_REG(chip, GPIO_CTRL));
-	udelay(1);
-
-	while (i--) {
-		if (val & (1 << i))
-			reg |= FM801_GPIO_GP(TEA_64PCR_DATA);
-		else
-			reg &= ~FM801_GPIO_GP(TEA_64PCR_DATA);
-		outw(reg, FM801_REG(chip, GPIO_CTRL));
-		udelay(1);
-		reg |= FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK);
-		outw(reg, FM801_REG(chip, GPIO_CTRL));
-		reg &= ~FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK);
-		outw(reg, FM801_REG(chip, GPIO_CTRL));
-		udelay(1);
+	reg |= FM801_GPIO_GS(gpio.data) |
+	       FM801_GPIO_GS(gpio.wren) |
+	       FM801_GPIO_GS(gpio.clk) |
+	       FM801_GPIO_GS(gpio.most);
+	if (output) {
+		/* all of lines are in the write direction */
+		/* clear data and clock lines */
+		reg &= ~(FM801_GPIO_GD(gpio.data) |
+			 FM801_GPIO_GD(gpio.wren) |
+			 FM801_GPIO_GD(gpio.clk) |
+			 FM801_GPIO_GP(gpio.data) |
+			 FM801_GPIO_GP(gpio.clk) |
+			 FM801_GPIO_GP(gpio.wren));
+	} else {
+		/* use GPIO lines, set data direction to input */
+		reg |= FM801_GPIO_GD(gpio.data) |
+		       FM801_GPIO_GD(gpio.most) |
+		       FM801_GPIO_GP(gpio.data) |
+		       FM801_GPIO_GP(gpio.most) |
+		       FM801_GPIO_GP(gpio.wren);
+		/* all of lines are in the write direction, except data */
+		/* clear data, write enable and clock lines */
+		reg &= ~(FM801_GPIO_GD(gpio.wren) |
+			 FM801_GPIO_GD(gpio.clk) |
+			 FM801_GPIO_GP(gpio.clk));
 	}
 
-	/* and reset the write enable bit */
-	reg |= FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE) |
-	       FM801_GPIO_GP(TEA_64PCR_DATA);
 	outw(reg, FM801_REG(chip, GPIO_CTRL));
-	spin_unlock_irq(&chip->reg_lock);
-}
-
-static unsigned int snd_fm801_tea575x_64pcr_read(struct snd_tea575x *tea)
-{
-	struct fm801 *chip = tea->private_data;
-	unsigned short reg;
-	unsigned int val = 0;
-	int i;
-	
-	spin_lock_irq(&chip->reg_lock);
-	reg = inw(FM801_REG(chip, GPIO_CTRL));
-	/* use GPIO lines, set data direction to input */
-	reg |= FM801_GPIO_GS(TEA_64PCR_DATA) |
-	       FM801_GPIO_GS(TEA_64PCR_WRITE_ENABLE) |
-	       FM801_GPIO_GS(TEA_64PCR_BUS_CLOCK) |
-	       FM801_GPIO_GD(TEA_64PCR_DATA) |
-	       FM801_GPIO_GP(TEA_64PCR_DATA) |
-	       FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE);
-	/* all of lines are in the write direction, except data */
-	/* clear data, write enable and clock lines */
-	reg &= ~(FM801_GPIO_GD(TEA_64PCR_WRITE_ENABLE) |
-	         FM801_GPIO_GD(TEA_64PCR_BUS_CLOCK) |
-	         FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK));
-
-	for (i = 0; i < 24; i++) {
-		reg &= ~FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK);
-		outw(reg, FM801_REG(chip, GPIO_CTRL));
-		udelay(1);
-		reg |= FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK);
-		outw(reg, FM801_REG(chip, GPIO_CTRL));
-		udelay(1);
-		val <<= 1;
-		if (inw(FM801_REG(chip, GPIO_CTRL)) & FM801_GPIO_GP(TEA_64PCR_DATA))
-			val |= 1;
-	}
-
-	spin_unlock_irq(&chip->reg_lock);
-
-	return val;
 }
 
-static void snd_fm801_tea575x_64pcr_mute(struct snd_tea575x *tea,
-					  unsigned int mute)
-{
-	struct fm801 *chip = tea->private_data;
-	unsigned short reg;
-
-	spin_lock_irq(&chip->reg_lock);
-
-	reg = inw(FM801_REG(chip, GPIO_CTRL));
-	if (mute)
-		/* 0xf800 (mute) */
-		reg &= ~FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE);
-	else
-		/* 0xf802 (unmute) */
-		reg |= FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE);
-	outw(reg, FM801_REG(chip, GPIO_CTRL));
-	udelay(1);
-
-	spin_unlock_irq(&chip->reg_lock);
-}
-
-static struct snd_tea575x_ops snd_fm801_tea_ops[3] = {
-	{
-		/* 1 = MediaForte 256-PCS */
-		.write = snd_fm801_tea575x_256pcs_write,
-		.read = snd_fm801_tea575x_256pcs_read,
-	},
-	{
-		/* 2 = MediaForte 256-PCPR */
-		.write = snd_fm801_tea575x_256pcpr_write,
-		.read = snd_fm801_tea575x_256pcpr_read,
-	},
-	{
-		/* 3 = MediaForte 64-PCR */
-		.write = snd_fm801_tea575x_64pcr_write,
-		.read = snd_fm801_tea575x_64pcr_read,
-		.mute = snd_fm801_tea575x_64pcr_mute,
-	}
+static struct snd_tea575x_ops snd_fm801_tea_ops = {
+	.set_pins = snd_fm801_tea575x_set_pins,
+	.get_pins = snd_fm801_tea575x_get_pins,
+	.set_direction = snd_fm801_tea575x_set_direction,
 };
 #endif
 
@@ -1371,7 +1149,7 @@ static int snd_fm801_free(struct fm801 *chip)
 	outw(cmdw, FM801_REG(chip, IRQ_MASK));
 
       __end_hw:
-#ifdef TEA575X_RADIO
+#ifdef CONFIG_SND_FM801_TEA575X_BOOL
 	snd_tea575x_exit(&chip->tea);
 #endif
 	if (chip->irq >= 0)
@@ -1450,16 +1228,25 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 
 	snd_card_set_dev(card, &pci->dev);
 
-#ifdef TEA575X_RADIO
+#ifdef CONFIG_SND_FM801_TEA575X_BOOL
+	chip->tea.private_data = chip;
+	chip->tea.ops = &snd_fm801_tea_ops;
+	sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci));
 	if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 &&
 	    (tea575x_tuner & TUNER_TYPE_MASK) < 4) {
-		chip->tea.dev_nr = tea575x_tuner >> 16;
-		chip->tea.card = card;
-		chip->tea.freq_fixup = 10700;
-		chip->tea.private_data = chip;
-		chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & TUNER_TYPE_MASK) - 1];
-		snd_tea575x_init(&chip->tea);
-	}
+		if (snd_tea575x_init(&chip->tea))
+			snd_printk(KERN_ERR "TEA575x radio not found\n");
+	} else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0)
+		/* autodetect tuner connection */
+		for (tea575x_tuner = 1; tea575x_tuner <= 3; tea575x_tuner++) {
+			chip->tea575x_tuner = tea575x_tuner;
+			if (!snd_tea575x_init(&chip->tea)) {
+				snd_printk(KERN_INFO "detected TEA575x radio type %s\n",
+					snd_fm801_tea575x_gpios[tea575x_tuner - 1].name);
+				break;
+			}
+		}
+	strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card));
 #endif
 
 	*rchip = chip;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 430f41db604..45b4a8d70e0 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -307,6 +307,12 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
 }
 EXPORT_SYMBOL_HDA(snd_hda_get_sub_nodes);
 
+static int _hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
+				hda_nid_t *conn_list, int max_conns);
+static bool add_conn_list(struct snd_array *array, hda_nid_t nid);
+static int copy_conn_list(hda_nid_t nid, hda_nid_t *dst, int max_dst,
+			  hda_nid_t *src, int len);
+
 /**
  * snd_hda_get_connections - get connection list
  * @codec: the HDA codec
@@ -320,7 +326,44 @@ EXPORT_SYMBOL_HDA(snd_hda_get_sub_nodes);
  * Returns the number of connections, or a negative error code.
  */
 int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
-			    hda_nid_t *conn_list, int max_conns)
+			     hda_nid_t *conn_list, int max_conns)
+{
+	struct snd_array *array = &codec->conn_lists;
+	int i, len, old_used;
+	hda_nid_t list[HDA_MAX_CONNECTIONS];
+
+	/* look up the cached results */
+	for (i = 0; i < array->used; ) {
+		hda_nid_t *p = snd_array_elem(array, i);
+		len = p[1];
+		if (nid == *p)
+			return copy_conn_list(nid, conn_list, max_conns,
+					      p + 2, len);
+		i += len + 2;
+	}
+
+	len = _hda_get_connections(codec, nid, list, HDA_MAX_CONNECTIONS);
+	if (len < 0)
+		return len;
+
+	/* add to the cache */
+	old_used = array->used;
+	if (!add_conn_list(array, nid) || !add_conn_list(array, len))
+		goto error_add;
+	for (i = 0; i < len; i++)
+		if (!add_conn_list(array, list[i]))
+			goto error_add;
+
+	return copy_conn_list(nid, conn_list, max_conns, list, len);
+		
+ error_add:
+	array->used = old_used;
+	return -ENOMEM;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_connections);
+
+static int _hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
+			     hda_nid_t *conn_list, int max_conns)
 {
 	unsigned int parm;
 	int i, conn_len, conns;
@@ -417,8 +460,28 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
 	}
 	return conns;
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_connections);
 
+static bool add_conn_list(struct snd_array *array, hda_nid_t nid)
+{
+	hda_nid_t *p = snd_array_new(array);
+	if (!p)
+		return false;
+	*p = nid;
+	return true;
+}
+
+static int copy_conn_list(hda_nid_t nid, hda_nid_t *dst, int max_dst,
+			  hda_nid_t *src, int len)
+{
+	if (len > max_dst) {
+		snd_printk(KERN_ERR "hda_codec: "
+			   "Too many connections %d for NID 0x%x\n",
+			   len, nid);
+		return -EINVAL;
+	}
+	memcpy(dst, src, len * sizeof(hda_nid_t));
+	return len;
+}
 
 /**
  * snd_hda_queue_unsol_event - add an unsolicited event to queue
@@ -937,6 +1000,7 @@ void snd_hda_shutup_pins(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_HDA(snd_hda_shutup_pins);
 
+#ifdef SND_HDA_NEEDS_RESUME
 /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */
 static void restore_shutup_pins(struct hda_codec *codec)
 {
@@ -953,6 +1017,7 @@ static void restore_shutup_pins(struct hda_codec *codec)
 	}
 	codec->pins_shutup = 0;
 }
+#endif
 
 static void init_hda_cache(struct hda_cache_rec *cache,
 			   unsigned int record_size);
@@ -1017,6 +1082,7 @@ static void snd_hda_codec_free(struct hda_codec *codec)
 	list_del(&codec->list);
 	snd_array_free(&codec->mixers);
 	snd_array_free(&codec->nids);
+	snd_array_free(&codec->conn_lists);
 	codec->bus->caddr_tbl[codec->addr] = NULL;
 	if (codec->patch_ops.free)
 		codec->patch_ops.free(codec);
@@ -1077,6 +1143,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus,
 	snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16);
 	snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16);
 	snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8);
+	snd_array_init(&codec->conn_lists, sizeof(hda_nid_t), 64);
 	if (codec->bus->modelname) {
 		codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL);
 		if (!codec->modelname) {
@@ -1329,6 +1396,7 @@ static void purify_inactive_streams(struct hda_codec *codec)
 	}
 }
 
+#ifdef SND_HDA_NEEDS_RESUME
 /* clean up all streams; called from suspend */
 static void hda_cleanup_all_streams(struct hda_codec *codec)
 {
@@ -1340,6 +1408,7 @@ static void hda_cleanup_all_streams(struct hda_codec *codec)
 			really_cleanup_stream(codec, p);
 	}
 }
+#endif
 
 /*
  * amp access functions
@@ -2552,7 +2621,7 @@ static unsigned int convert_to_spdif_status(unsigned short val)
 static void set_dig_out(struct hda_codec *codec, hda_nid_t nid,
 			int verb, int val)
 {
-	hda_nid_t *d;
+	const hda_nid_t *d;
 
 	snd_hda_codec_write_cache(codec, nid, 0, verb, val);
 	d = codec->slave_dig_outs;
@@ -3803,7 +3872,8 @@ EXPORT_SYMBOL_HDA(snd_hda_check_board_codec_sid_config);
  *
  * Returns 0 if successful, or a negative error code.
  */
-int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew)
+int snd_hda_add_new_ctls(struct hda_codec *codec,
+			 const struct snd_kcontrol_new *knew)
 {
 	int err;
 
@@ -3946,7 +4016,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
 				 struct hda_loopback_check *check,
 				 hda_nid_t nid)
 {
-	struct hda_amp_list *p;
+	const struct hda_amp_list *p;
 	int ch, v;
 
 	if (!check->amplist)
@@ -4114,7 +4184,7 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid,
 				    -1);
 	snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
 	if (codec->slave_dig_outs) {
-		hda_nid_t *d;
+		const hda_nid_t *d;
 		for (d = codec->slave_dig_outs; *d; d++)
 			snd_hda_codec_setup_stream(codec, *d, stream_tag, 0,
 						   format);
@@ -4129,7 +4199,7 @@ static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid)
 {
 	snd_hda_codec_cleanup_stream(codec, nid);
 	if (codec->slave_dig_outs) {
-		hda_nid_t *d;
+		const hda_nid_t *d;
 		for (d = codec->slave_dig_outs; *d; d++)
 			snd_hda_codec_cleanup_stream(codec, *d);
 	}
@@ -4276,7 +4346,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
 				     unsigned int format,
 				     struct snd_pcm_substream *substream)
 {
-	hda_nid_t *nids = mout->dac_nids;
+	const hda_nid_t *nids = mout->dac_nids;
 	int chs = substream->runtime->channels;
 	int i;
 
@@ -4331,7 +4401,7 @@ EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_prepare);
 int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
 				     struct hda_multi_out *mout)
 {
-	hda_nid_t *nids = mout->dac_nids;
+	const hda_nid_t *nids = mout->dac_nids;
 	int i;
 
 	for (i = 0; i < mout->num_dacs; i++)
@@ -4356,7 +4426,7 @@ EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_cleanup);
  * Helper for automatic pin configuration
  */
 
-static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list)
+static int is_in_nid_list(hda_nid_t nid, const hda_nid_t *list)
 {
 	for (; *list; list++)
 		if (*list == nid)
@@ -4437,7 +4507,7 @@ static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg)
  */
 int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 				 struct auto_pin_cfg *cfg,
-				 hda_nid_t *ignore_nids)
+				 const hda_nid_t *ignore_nids)
 {
 	hda_nid_t nid, end_nid;
 	short seq, assoc_line_out, assoc_speaker;
@@ -4628,10 +4698,13 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 	/*
 	 * debug prints of the parsed results
 	 */
-	snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
+	snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x) type:%s\n",
 		   cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1],
 		   cfg->line_out_pins[2], cfg->line_out_pins[3],
-		   cfg->line_out_pins[4]);
+		   cfg->line_out_pins[4],
+		   cfg->line_out_type == AUTO_PIN_HP_OUT ? "hp" :
+		   (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT ?
+		    "speaker" : "line"));
 	snd_printd("   speaker_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
 		   cfg->speaker_outs, cfg->speaker_pins[0],
 		   cfg->speaker_pins[1], cfg->speaker_pins[2],
@@ -4646,7 +4719,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 			   cfg->dig_out_pins[0], cfg->dig_out_pins[1]);
 	snd_printd("   inputs:");
 	for (i = 0; i < cfg->num_inputs; i++) {
-		snd_printdd(" %s=0x%x",
+		snd_printd(" %s=0x%x",
 			    hda_get_autocfg_input_label(codec, cfg, i),
 			    cfg->inputs[i].pin);
 	}
@@ -4982,6 +5055,8 @@ static const char *get_jack_default_name(struct hda_codec *codec, hda_nid_t nid,
 		return "Line-out";
 	case SND_JACK_HEADSET:
 		return "Headset";
+	case SND_JACK_VIDEOOUT:
+		return "HDMI/DP";
 	default:
 		return "Misc";
 	}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index e46d5420a9f..59c97306c1d 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -825,12 +825,14 @@ struct hda_codec {
 	struct hda_cache_rec amp_cache;	/* cache for amp access */
 	struct hda_cache_rec cmd_cache;	/* cache for other commands */
 
+	struct snd_array conn_lists;	/* connection-list array */
+
 	struct mutex spdif_mutex;
 	struct mutex control_mutex;
 	unsigned int spdif_status;	/* IEC958 status bits */
 	unsigned short spdif_ctls;	/* SPDIF control bits */
 	unsigned int spdif_in_enable;	/* SPDIF input enable? */
-	hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
+	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 */
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 74b0560289c..b05f7be9dc1 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -312,23 +312,6 @@ out_fail:
 	return -EINVAL;
 }
 
-static int hdmi_eld_valid(struct hda_codec *codec, hda_nid_t nid)
-{
-	int eldv;
-	int present;
-
-	present = snd_hda_pin_sense(codec, nid);
-	eldv    = (present & AC_PINSENSE_ELDV);
-	present = (present & AC_PINSENSE_PRESENCE);
-
-#ifdef CONFIG_SND_DEBUG_VERBOSE
-	printk(KERN_INFO "HDMI: sink_present = %d, eld_valid = %d\n",
-			!!present, !!eldv);
-#endif
-
-	return eldv && present;
-}
-
 int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid)
 {
 	return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_SIZE,
@@ -343,7 +326,7 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
 	int size;
 	unsigned char *buf;
 
-	if (!hdmi_eld_valid(codec, nid))
+	if (!eld->eld_valid)
 		return -ENOENT;
 
 	size = snd_hdmi_get_eld_size(codec, nid);
@@ -477,6 +460,8 @@ static void hdmi_print_eld_info(struct snd_info_entry *entry,
 
 	snd_iprintf(buffer, "monitor_present\t\t%d\n", e->monitor_present);
 	snd_iprintf(buffer, "eld_valid\t\t%d\n", e->eld_valid);
+	if (!e->eld_valid)
+		return;
 	snd_iprintf(buffer, "monitor_name\t\t%s\n", e->monitor_name);
 	snd_iprintf(buffer, "connection_type\t\t%s\n",
 				eld_connection_type_names[e->conn_type]);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 70a9d32f0e9..486f6deb3ee 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -126,6 +126,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
 			 "{Intel, ICH10},"
 			 "{Intel, PCH},"
 			 "{Intel, CPT},"
+			 "{Intel, PPT},"
 			 "{Intel, PBG},"
 			 "{Intel, SCH},"
 			 "{ATI, SB450},"
@@ -390,6 +391,7 @@ struct azx {
 
 	/* chip type specific */
 	int driver_type;
+	unsigned int driver_caps;
 	int playback_streams;
 	int playback_index_offset;
 	int capture_streams;
@@ -463,6 +465,34 @@ enum {
 	AZX_NUM_DRIVERS, /* keep this as last entry */
 };
 
+/* driver quirks (capabilities) */
+/* bits 0-7 are used for indicating driver type */
+#define AZX_DCAPS_NO_TCSEL	(1 << 8)	/* No Intel TCSEL bit */
+#define AZX_DCAPS_NO_MSI	(1 << 9)	/* No MSI support */
+#define AZX_DCAPS_ATI_SNOOP	(1 << 10)	/* ATI snoop enable */
+#define AZX_DCAPS_NVIDIA_SNOOP	(1 << 11)	/* Nvidia snoop enable */
+#define AZX_DCAPS_SCH_SNOOP	(1 << 12)	/* SCH/PCH snoop enable */
+#define AZX_DCAPS_RIRB_DELAY	(1 << 13)	/* Long delay in read loop */
+#define AZX_DCAPS_RIRB_PRE_DELAY (1 << 14)	/* Put a delay before read */
+#define AZX_DCAPS_CTX_WORKAROUND (1 << 15)	/* X-Fi workaround */
+#define AZX_DCAPS_POSFIX_LPIB	(1 << 16)	/* Use LPIB as default */
+#define AZX_DCAPS_POSFIX_VIA	(1 << 17)	/* Use VIACOMBO as default */
+#define AZX_DCAPS_NO_64BIT	(1 << 18)	/* No 64bit address */
+#define AZX_DCAPS_SYNC_WRITE	(1 << 19)	/* sync each cmd write */
+
+/* quirks for ATI SB / AMD Hudson */
+#define AZX_DCAPS_PRESET_ATI_SB \
+	(AZX_DCAPS_ATI_SNOOP | AZX_DCAPS_NO_TCSEL | \
+	 AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB)
+
+/* quirks for ATI/AMD HDMI */
+#define AZX_DCAPS_PRESET_ATI_HDMI \
+	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB)
+
+/* quirks for Nvidia */
+#define AZX_DCAPS_PRESET_NVIDIA \
+	(AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI)
+
 static char *driver_short_names[] __devinitdata = {
 	[AZX_DRIVER_ICH] = "HDA Intel",
 	[AZX_DRIVER_PCH] = "HDA Intel PCH",
@@ -565,7 +595,7 @@ static void azx_init_cmd_io(struct azx *chip)
 	/* reset the rirb hw write pointer */
 	azx_writew(chip, RIRBWP, ICH6_RIRBWP_RST);
 	/* set N=1, get RIRB response interrupt for new entry */
-	if (chip->driver_type == AZX_DRIVER_CTX)
+	if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND)
 		azx_writew(chip, RINTCNT, 0xc0);
 	else
 		azx_writew(chip, RINTCNT, 1);
@@ -1055,19 +1085,24 @@ static void azx_init_pci(struct azx *chip)
 	 * codecs.
 	 * The PCI register TCSEL is defined in the Intel manuals.
 	 */
-	if (chip->driver_type != AZX_DRIVER_ATI &&
-	    chip->driver_type != AZX_DRIVER_ATIHDMI)
+	if (!(chip->driver_caps & AZX_DCAPS_NO_TCSEL)) {
+		snd_printdd(SFX "Clearing TCSEL\n");
 		update_pci_byte(chip->pci, ICH6_PCIREG_TCSEL, 0x07, 0);
+	}
 
-	switch (chip->driver_type) {
-	case AZX_DRIVER_ATI:
-		/* For ATI SB450 azalia HD audio, we need to enable snoop */
+	/* For ATI SB450/600/700/800/900 and AMD Hudson azalia HD audio,
+	 * we need to enable snoop.
+	 */
+	if (chip->driver_caps & AZX_DCAPS_ATI_SNOOP) {
+		snd_printdd(SFX "Enabling ATI snoop\n");
 		update_pci_byte(chip->pci,
 				ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, 
 				0x07, ATI_SB450_HDAUDIO_ENABLE_SNOOP);
-		break;
-	case AZX_DRIVER_NVIDIA:
-		/* For NVIDIA HDA, enable snoop */
+	}
+
+	/* For NVIDIA HDA, enable snoop */
+	if (chip->driver_caps & AZX_DCAPS_NVIDIA_SNOOP) {
+		snd_printdd(SFX "Enabling Nvidia snoop\n");
 		update_pci_byte(chip->pci,
 				NVIDIA_HDA_TRANSREG_ADDR,
 				0x0f, NVIDIA_HDA_ENABLE_COHBITS);
@@ -1077,9 +1112,10 @@ static void azx_init_pci(struct azx *chip)
 		update_pci_byte(chip->pci,
 				NVIDIA_HDA_OSTRM_COH,
 				0x01, NVIDIA_HDA_ENABLE_COHBIT);
-		break;
-	case AZX_DRIVER_SCH:
-	case AZX_DRIVER_PCH:
+	}
+
+	/* Enable SCH/PCH snoop if needed */
+	if (chip->driver_caps & AZX_DCAPS_SCH_SNOOP) {
 		pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop);
 		if (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) {
 			pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC,
@@ -1090,8 +1126,6 @@ static void azx_init_pci(struct azx *chip)
 				(snoop & INTEL_SCH_HDA_DEVC_NOSNOOP)
 				? "Failed" : "OK");
 		}
-		break;
-
         }
 }
 
@@ -1145,7 +1179,7 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
 	status = azx_readb(chip, RIRBSTS);
 	if (status & RIRB_INT_MASK) {
 		if (status & RIRB_INT_RESPONSE) {
-			if (chip->driver_type == AZX_DRIVER_CTX)
+			if (chip->driver_caps & AZX_DCAPS_RIRB_PRE_DELAY)
 				udelay(80);
 			azx_update_rirb(chip);
 		}
@@ -1414,8 +1448,10 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model)
 	if (err < 0)
 		return err;
 
-	if (chip->driver_type == AZX_DRIVER_NVIDIA)
+	if (chip->driver_caps & AZX_DCAPS_RIRB_DELAY) {
+		snd_printd(SFX "Enable delay in RIRB handling\n");
 		chip->bus->needs_damn_long_delay = 1;
+	}
 
 	codecs = 0;
 	max_slots = azx_max_codecs[chip->driver_type];
@@ -1446,6 +1482,16 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model)
 		}
 	}
 
+	/* AMD chipsets often cause the communication stalls upon certain
+	 * sequence like the pin-detection.  It seems that forcing the synced
+	 * access works around the stall.  Grrr...
+	 */
+	if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) {
+		snd_printd(SFX "Enable sync_write for stable communication\n");
+		chip->bus->sync_write = 1;
+		chip->bus->allow_bus_reset = 1;
+	}
+
 	/* Then create codec instances */
 	for (c = 0; c < max_slots; c++) {
 		if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) {
@@ -1702,7 +1748,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
 
 	stream_tag = azx_dev->stream_tag;
 	/* CA-IBG chips need the playback stream starting from 1 */
-	if (chip->driver_type == AZX_DRIVER_CTX &&
+	if ((chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) &&
 	    stream_tag > chip->capture_streams)
 		stream_tag -= chip->capture_streams;
 	return snd_hda_codec_prepare(apcm->codec, hinfo, stream_tag,
@@ -2347,13 +2393,14 @@ static int __devinit check_position_fix(struct azx *chip, int fix)
 	}
 
 	/* Check VIA/ATI HD Audio Controller exist */
-	switch (chip->driver_type) {
-	case AZX_DRIVER_VIA:
-	case AZX_DRIVER_ATI:
-		/* Use link position directly, avoid any transfer problem. */
+	if (chip->driver_caps & AZX_DCAPS_POSFIX_VIA) {
+		snd_printd(SFX "Using VIACOMBO position fix\n");
 		return POS_FIX_VIACOMBO;
 	}
-
+	if (chip->driver_caps & AZX_DCAPS_POSFIX_LPIB) {
+		snd_printd(SFX "Using LPIB position fix\n");
+		return POS_FIX_LPIB;
+	}
 	return POS_FIX_AUTO;
 }
 
@@ -2435,8 +2482,8 @@ static void __devinit check_msi(struct azx *chip)
 	}
 
 	/* NVidia chipsets seem to cause troubles with MSI */
-	if (chip->driver_type == AZX_DRIVER_NVIDIA) {
-		printk(KERN_INFO "hda_intel: Disable MSI for Nvidia chipset\n");
+	if (chip->driver_caps & AZX_DCAPS_NO_MSI) {
+		printk(KERN_INFO "hda_intel: Disabling MSI\n");
 		chip->msi = 0;
 	}
 }
@@ -2446,7 +2493,7 @@ static void __devinit check_msi(struct azx *chip)
  * constructor
  */
 static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
-				int dev, int driver_type,
+				int dev, unsigned int driver_caps,
 				struct azx **rchip)
 {
 	struct azx *chip;
@@ -2474,7 +2521,8 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 	chip->card = card;
 	chip->pci = pci;
 	chip->irq = -1;
-	chip->driver_type = driver_type;
+	chip->driver_caps = driver_caps;
+	chip->driver_type = driver_caps & 0xff;
 	check_msi(chip);
 	chip->dev_index = dev;
 	INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work);
@@ -2538,8 +2586,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 	snd_printdd(SFX "chipset global capabilities = 0x%x\n", gcap);
 
 	/* disable SB600 64bit support for safety */
-	if ((chip->driver_type == AZX_DRIVER_ATI) ||
-	    (chip->driver_type == AZX_DRIVER_ATIHDMI)) {
+	if (chip->pci->vendor == PCI_VENDOR_ID_ATI) {
 		struct pci_dev *p_smbus;
 		p_smbus = pci_get_device(PCI_VENDOR_ID_ATI,
 					 PCI_DEVICE_ID_ATI_SBX00_SMBUS,
@@ -2551,10 +2598,11 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 		}
 	}
 
-	/* disable 64bit DMA address for Teradici */
-	/* it does not work with device 6549:1200 subsys e4a2:040b */
-	if (chip->driver_type == AZX_DRIVER_TERA)
+	/* disable 64bit DMA address on some devices */
+	if (chip->driver_caps & AZX_DCAPS_NO_64BIT) {
+		snd_printd(SFX "Disabling 64bit DMA\n");
 		gcap &= ~ICH6_GCAP_64OK;
+	}
 
 	/* allow 64bit DMA address if supported by H/W */
 	if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
@@ -2756,36 +2804,62 @@ static void __devexit azx_remove(struct pci_dev *pci)
 /* PCI IDs */
 static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	/* CPT */
-	{ PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH },
+	{ PCI_DEVICE(0x8086, 0x1c20),
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP },
 	/* PBG */
-	{ PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH },
+	{ PCI_DEVICE(0x8086, 0x1d20),
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP },
+	/* Panther Point */
+	{ PCI_DEVICE(0x8086, 0x1e20),
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP },
 	/* SCH */
-	{ PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH },
+	{ PCI_DEVICE(0x8086, 0x811b),
+	  .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP },
 	/* Generic Intel */
 	{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID),
 	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
 	  .class_mask = 0xffffff,
 	  .driver_data = AZX_DRIVER_ICH },
-	/* ATI SB 450/600 */
-	{ PCI_DEVICE(0x1002, 0x437b), .driver_data = AZX_DRIVER_ATI },
-	{ PCI_DEVICE(0x1002, 0x4383), .driver_data = AZX_DRIVER_ATI },
+	/* ATI SB 450/600/700/800/900 */
+	{ PCI_DEVICE(0x1002, 0x437b),
+	  .driver_data = AZX_DRIVER_ATI | AZX_DCAPS_PRESET_ATI_SB },
+	{ PCI_DEVICE(0x1002, 0x4383),
+	  .driver_data = AZX_DRIVER_ATI | AZX_DCAPS_PRESET_ATI_SB },
+	/* AMD Hudson */
+	{ PCI_DEVICE(0x1022, 0x780d),
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB },
 	/* ATI HDMI */
-	{ PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI },
-	{ PCI_DEVICE(0x1002, 0x7919), .driver_data = AZX_DRIVER_ATIHDMI },
-	{ PCI_DEVICE(0x1002, 0x960f), .driver_data = AZX_DRIVER_ATIHDMI },
-	{ PCI_DEVICE(0x1002, 0x970f), .driver_data = AZX_DRIVER_ATIHDMI },
-	{ PCI_DEVICE(0x1002, 0xaa00), .driver_data = AZX_DRIVER_ATIHDMI },
-	{ PCI_DEVICE(0x1002, 0xaa08), .driver_data = AZX_DRIVER_ATIHDMI },
-	{ PCI_DEVICE(0x1002, 0xaa10), .driver_data = AZX_DRIVER_ATIHDMI },
-	{ PCI_DEVICE(0x1002, 0xaa18), .driver_data = AZX_DRIVER_ATIHDMI },
-	{ PCI_DEVICE(0x1002, 0xaa20), .driver_data = AZX_DRIVER_ATIHDMI },
-	{ PCI_DEVICE(0x1002, 0xaa28), .driver_data = AZX_DRIVER_ATIHDMI },
-	{ PCI_DEVICE(0x1002, 0xaa30), .driver_data = AZX_DRIVER_ATIHDMI },
-	{ PCI_DEVICE(0x1002, 0xaa38), .driver_data = AZX_DRIVER_ATIHDMI },
-	{ PCI_DEVICE(0x1002, 0xaa40), .driver_data = AZX_DRIVER_ATIHDMI },
-	{ PCI_DEVICE(0x1002, 0xaa48), .driver_data = AZX_DRIVER_ATIHDMI },
+	{ PCI_DEVICE(0x1002, 0x793b),
+	  .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
+	{ PCI_DEVICE(0x1002, 0x7919),
+	  .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
+	{ PCI_DEVICE(0x1002, 0x960f),
+	  .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
+	{ PCI_DEVICE(0x1002, 0x970f),
+	  .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
+	{ PCI_DEVICE(0x1002, 0xaa00),
+	  .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
+	{ PCI_DEVICE(0x1002, 0xaa08),
+	  .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
+	{ PCI_DEVICE(0x1002, 0xaa10),
+	  .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
+	{ PCI_DEVICE(0x1002, 0xaa18),
+	  .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
+	{ PCI_DEVICE(0x1002, 0xaa20),
+	  .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
+	{ PCI_DEVICE(0x1002, 0xaa28),
+	  .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
+	{ PCI_DEVICE(0x1002, 0xaa30),
+	  .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
+	{ PCI_DEVICE(0x1002, 0xaa38),
+	  .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
+	{ PCI_DEVICE(0x1002, 0xaa40),
+	  .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
+	{ PCI_DEVICE(0x1002, 0xaa48),
+	  .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI },
 	/* VIA VT8251/VT8237A */
-	{ PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA },
+	{ PCI_DEVICE(0x1106, 0x3288),
+	  .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA },
 	/* SIS966 */
 	{ PCI_DEVICE(0x1039, 0x7502), .driver_data = AZX_DRIVER_SIS },
 	/* ULI M5461 */
@@ -2794,9 +2868,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	{ PCI_DEVICE(PCI_VENDOR_ID_NVIDIA, PCI_ANY_ID),
 	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
 	  .class_mask = 0xffffff,
-	  .driver_data = AZX_DRIVER_NVIDIA },
+	  .driver_data = AZX_DRIVER_NVIDIA | AZX_DCAPS_PRESET_NVIDIA },
 	/* Teradici */
-	{ PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA },
+	{ PCI_DEVICE(0x6549, 0x1200),
+	  .driver_data = AZX_DRIVER_TERA | AZX_DCAPS_NO_64BIT },
 	/* Creative X-Fi (CA0110-IBG) */
 #if !defined(CONFIG_SND_CTXFI) && !defined(CONFIG_SND_CTXFI_MODULE)
 	/* the following entry conflicts with snd-ctxfi driver,
@@ -2806,10 +2881,13 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	{ PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_ANY_ID),
 	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
 	  .class_mask = 0xffffff,
-	  .driver_data = AZX_DRIVER_CTX },
+	  .driver_data = AZX_DRIVER_CTX | AZX_DCAPS_CTX_WORKAROUND |
+	  AZX_DCAPS_RIRB_PRE_DELAY },
 #else
 	/* this entry seems still valid -- i.e. without emu20kx chip */
-	{ PCI_DEVICE(0x1102, 0x0009), .driver_data = AZX_DRIVER_CTX },
+	{ PCI_DEVICE(0x1102, 0x0009),
+	  .driver_data = AZX_DRIVER_CTX | AZX_DCAPS_CTX_WORKAROUND |
+	  AZX_DCAPS_RIRB_PRE_DELAY },
 #endif
 	/* Vortex86MX */
 	{ PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC },
@@ -2819,11 +2897,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	{ PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID),
 	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
 	  .class_mask = 0xffffff,
-	  .driver_data = AZX_DRIVER_GENERIC },
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI },
 	{ PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_ANY_ID),
 	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
 	  .class_mask = 0xffffff,
-	  .driver_data = AZX_DRIVER_GENERIC },
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI },
 	{ 0, }
 };
 MODULE_DEVICE_TABLE(pci, azx_ids);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index ff5e2ac2239..08ec073444e 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -267,11 +267,11 @@ enum { HDA_DIG_NONE, HDA_DIG_EXCLUSIVE, HDA_DIG_ANALOG_DUP }; /* dig_out_used */
 
 struct hda_multi_out {
 	int num_dacs;		/* # of DACs, must be more than 1 */
-	hda_nid_t *dac_nids;	/* DAC list */
+	const hda_nid_t *dac_nids;	/* DAC list */
 	hda_nid_t hp_nid;	/* optional DAC for HP, 0 when not exists */
 	hda_nid_t extra_out_nid[3];	/* optional DACs, 0 when not exists */
 	hda_nid_t dig_out_nid;	/* digital out audio widget */
-	hda_nid_t *slave_dig_outs;
+	const hda_nid_t *slave_dig_outs;
 	int max_channels;	/* currently supported analog channels */
 	int dig_out_used;	/* current usage of digital out (HDA_DIG_XXX) */
 	int no_share_stream;	/* don't share a stream with multiple pins */
@@ -347,7 +347,7 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec,
                                int num_configs, const char * const *models,
                                const struct snd_pci_quirk *tbl);
 int snd_hda_add_new_ctls(struct hda_codec *codec,
-			 struct snd_kcontrol_new *knew);
+			 const struct snd_kcontrol_new *knew);
 
 /*
  * unsolicited event handler
@@ -443,7 +443,7 @@ struct auto_pin_cfg {
 
 int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 				 struct auto_pin_cfg *cfg,
-				 hda_nid_t *ignore_nids);
+				 const hda_nid_t *ignore_nids);
 
 /* amp values */
 #define AMP_IN_MUTE(idx)	(0x7080 | ((idx)<<8))
@@ -493,6 +493,12 @@ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid);
 u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid);
 int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid);
 
+static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
+{
+	return (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT) &&
+		(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP);
+}
+
 /* flags for hda_nid_item */
 #define HDA_NID_ITEM_AMP	(1<<0)
 
@@ -567,7 +573,7 @@ struct hda_amp_list {
 };
 
 struct hda_loopback_check {
-	struct hda_amp_list *amplist;
+	const struct hda_amp_list *amplist;
 	int power_on;
 };
 
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 2942d2a9ea1..696ac259030 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -30,7 +30,7 @@
 #include "hda_beep.h"
 
 struct ad198x_spec {
-	struct snd_kcontrol_new *mixers[6];
+	const struct snd_kcontrol_new *mixers[6];
 	int num_mixers;
 	unsigned int beep_amp;	/* beep amp value, set via set_beep_amp() */
 	const struct hda_verb *init_verbs[6];	/* initialization verbs
@@ -46,17 +46,17 @@ struct ad198x_spec {
 	unsigned int cur_eapd;
 	unsigned int need_dac_fix;
 
-	hda_nid_t *alt_dac_nid;
-	struct hda_pcm_stream *stream_analog_alt_playback;
+	const hda_nid_t *alt_dac_nid;
+	const struct hda_pcm_stream *stream_analog_alt_playback;
 
 	/* capture */
 	unsigned int num_adc_nids;
-	hda_nid_t *adc_nids;
+	const hda_nid_t *adc_nids;
 	hda_nid_t dig_in_nid;		/* digital-in NID; optional */
 
 	/* capture source */
 	const struct hda_input_mux *input_mux;
-	hda_nid_t *capsrc_nids;
+	const hda_nid_t *capsrc_nids;
 	unsigned int cur_mux[3];
 
 	/* channel model */
@@ -182,13 +182,13 @@ static void ad198x_free_kctls(struct hda_codec *codec);
 
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
 /* additional beep mixers; the actual parameters are overwritten at build */
-static struct snd_kcontrol_new ad_beep_mixer[] = {
+static const struct snd_kcontrol_new ad_beep_mixer[] = {
 	HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_OUTPUT),
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad_beep2_mixer[] = {
+static const struct snd_kcontrol_new ad_beep2_mixer[] = {
 	HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_BEEP("Digital Beep Playback Switch", 0, 0, HDA_OUTPUT),
 	{ } /* end */
@@ -231,7 +231,7 @@ static int ad198x_build_controls(struct hda_codec *codec)
 	/* create beep controls if needed */
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
 	if (spec->beep_amp) {
-		struct snd_kcontrol_new *knew;
+		const struct snd_kcontrol_new *knew;
 		knew = spec->analog_beep ? ad_beep2_mixer : ad_beep_mixer;
 		for ( ; knew->name; knew++) {
 			struct snd_kcontrol *kctl;
@@ -331,7 +331,7 @@ static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
 	return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
 }
 
-static struct hda_pcm_stream ad198x_pcm_analog_alt_playback = {
+static const struct hda_pcm_stream ad198x_pcm_analog_alt_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -403,7 +403,7 @@ static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 
 /*
  */
-static struct hda_pcm_stream ad198x_pcm_analog_playback = {
+static const struct hda_pcm_stream ad198x_pcm_analog_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 6, /* changed later */
@@ -415,7 +415,7 @@ static struct hda_pcm_stream ad198x_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream ad198x_pcm_analog_capture = {
+static const struct hda_pcm_stream ad198x_pcm_analog_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -426,7 +426,7 @@ static struct hda_pcm_stream ad198x_pcm_analog_capture = {
 	},
 };
 
-static struct hda_pcm_stream ad198x_pcm_digital_playback = {
+static const struct hda_pcm_stream ad198x_pcm_digital_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -439,7 +439,7 @@ static struct hda_pcm_stream ad198x_pcm_digital_playback = {
 	},
 };
 
-static struct hda_pcm_stream ad198x_pcm_digital_capture = {
+static const struct hda_pcm_stream ad198x_pcm_digital_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -489,11 +489,6 @@ static int ad198x_build_pcms(struct hda_codec *codec)
 	return 0;
 }
 
-static inline void ad198x_shutup(struct hda_codec *codec)
-{
-	snd_hda_shutup_pins(codec);
-}
-
 static void ad198x_free_kctls(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec = codec->spec;
@@ -547,6 +542,12 @@ static void ad198x_power_eapd(struct hda_codec *codec)
 	}
 }
 
+static void ad198x_shutup(struct hda_codec *codec)
+{
+	snd_hda_shutup_pins(codec);
+	ad198x_power_eapd(codec);
+}
+
 static void ad198x_free(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec = codec->spec;
@@ -564,12 +565,11 @@ static void ad198x_free(struct hda_codec *codec)
 static int ad198x_suspend(struct hda_codec *codec, pm_message_t state)
 {
 	ad198x_shutup(codec);
-	ad198x_power_eapd(codec);
 	return 0;
 }
 #endif
 
-static struct hda_codec_ops ad198x_patch_ops = {
+static const struct hda_codec_ops ad198x_patch_ops = {
 	.build_controls = ad198x_build_controls,
 	.build_pcms = ad198x_build_pcms,
 	.init = ad198x_init,
@@ -639,13 +639,13 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
 #define AD1986A_CLFE_DAC	0x05
 #define AD1986A_ADC		0x06
 
-static hda_nid_t ad1986a_dac_nids[3] = {
+static const hda_nid_t ad1986a_dac_nids[3] = {
 	AD1986A_FRONT_DAC, AD1986A_SURR_DAC, AD1986A_CLFE_DAC
 };
-static hda_nid_t ad1986a_adc_nids[1] = { AD1986A_ADC };
-static hda_nid_t ad1986a_capsrc_nids[1] = { 0x12 };
+static const hda_nid_t ad1986a_adc_nids[1] = { AD1986A_ADC };
+static const hda_nid_t ad1986a_capsrc_nids[1] = { 0x12 };
 
-static struct hda_input_mux ad1986a_capture_source = {
+static const struct hda_input_mux ad1986a_capture_source = {
 	.num_items = 7,
 	.items = {
 		{ "Mic", 0x0 },
@@ -659,7 +659,7 @@ static struct hda_input_mux ad1986a_capture_source = {
 };
 
 
-static struct hda_bind_ctls ad1986a_bind_pcm_vol = {
+static const struct hda_bind_ctls ad1986a_bind_pcm_vol = {
 	.ops = &snd_hda_bind_vol,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
@@ -669,7 +669,7 @@ static struct hda_bind_ctls ad1986a_bind_pcm_vol = {
 	},
 };
 
-static struct hda_bind_ctls ad1986a_bind_pcm_sw = {
+static const struct hda_bind_ctls ad1986a_bind_pcm_sw = {
 	.ops = &snd_hda_bind_sw,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
@@ -682,7 +682,7 @@ static struct hda_bind_ctls ad1986a_bind_pcm_sw = {
 /*
  * mixers
  */
-static struct snd_kcontrol_new ad1986a_mixers[] = {
+static const struct snd_kcontrol_new ad1986a_mixers[] = {
 	/*
 	 * bind volumes/mutes of 3 DACs as a single PCM control for simplicity
 	 */
@@ -723,7 +723,7 @@ static struct snd_kcontrol_new ad1986a_mixers[] = {
 };
 
 /* additional mixers for 3stack mode */
-static struct snd_kcontrol_new ad1986a_3st_mixers[] = {
+static const struct snd_kcontrol_new ad1986a_3st_mixers[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Channel Mode",
@@ -735,10 +735,10 @@ static struct snd_kcontrol_new ad1986a_3st_mixers[] = {
 };
 
 /* laptop model - 2ch only */
-static hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC };
+static const hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC };
 
 /* master controls both pins 0x1a and 0x1b */
-static struct hda_bind_ctls ad1986a_laptop_master_vol = {
+static const struct hda_bind_ctls ad1986a_laptop_master_vol = {
 	.ops = &snd_hda_bind_vol,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
@@ -747,7 +747,7 @@ static struct hda_bind_ctls ad1986a_laptop_master_vol = {
 	},
 };
 
-static struct hda_bind_ctls ad1986a_laptop_master_sw = {
+static const struct hda_bind_ctls ad1986a_laptop_master_sw = {
 	.ops = &snd_hda_bind_sw,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
@@ -756,7 +756,7 @@ static struct hda_bind_ctls ad1986a_laptop_master_sw = {
 	},
 };
 
-static struct snd_kcontrol_new ad1986a_laptop_mixers[] = {
+static const struct snd_kcontrol_new ad1986a_laptop_mixers[] = {
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
 	HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
@@ -787,7 +787,7 @@ static struct snd_kcontrol_new ad1986a_laptop_mixers[] = {
 
 /* laptop-eapd model - 2ch only */
 
-static struct hda_input_mux ad1986a_laptop_eapd_capture_source = {
+static const struct hda_input_mux ad1986a_laptop_eapd_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic", 0x0 },
@@ -796,7 +796,7 @@ static struct hda_input_mux ad1986a_laptop_eapd_capture_source = {
 	},
 };
 
-static struct hda_input_mux ad1986a_automic_capture_source = {
+static const struct hda_input_mux ad1986a_automic_capture_source = {
 	.num_items = 2,
 	.items = {
 		{ "Mic", 0x0 },
@@ -804,13 +804,13 @@ static struct hda_input_mux ad1986a_automic_capture_source = {
 	},
 };
 
-static struct snd_kcontrol_new ad1986a_laptop_master_mixers[] = {
+static const struct snd_kcontrol_new ad1986a_laptop_master_mixers[] = {
 	HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
 	HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
+static const struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
@@ -837,7 +837,7 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = {
+static const struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = {
 	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, HDA_OUTPUT),
 	{ } /* end */
@@ -931,7 +931,7 @@ static int ad1986a_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new ad1986a_automute_master_mixers[] = {
+static const struct snd_kcontrol_new ad1986a_automute_master_mixers[] = {
 	HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -949,7 +949,7 @@ static struct snd_kcontrol_new ad1986a_automute_master_mixers[] = {
 /*
  * initialization verbs
  */
-static struct hda_verb ad1986a_init_verbs[] = {
+static const struct hda_verb ad1986a_init_verbs[] = {
 	/* Front, Surround, CLFE DAC; mute as default */
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
 	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
@@ -1004,7 +1004,7 @@ static struct hda_verb ad1986a_init_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb ad1986a_ch2_init[] = {
+static const struct hda_verb ad1986a_ch2_init[] = {
 	/* Surround out -> Line In */
 	{ 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
  	/* Line-in selectors */
@@ -1016,7 +1016,7 @@ static struct hda_verb ad1986a_ch2_init[] = {
 	{ } /* end */
 };
 
-static struct hda_verb ad1986a_ch4_init[] = {
+static const struct hda_verb ad1986a_ch4_init[] = {
 	/* Surround out -> Surround */
 	{ 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
@@ -1026,7 +1026,7 @@ static struct hda_verb ad1986a_ch4_init[] = {
 	{ } /* end */
 };
 
-static struct hda_verb ad1986a_ch6_init[] = {
+static const struct hda_verb ad1986a_ch6_init[] = {
 	/* Surround out -> Surround out */
 	{ 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
@@ -1036,19 +1036,19 @@ static struct hda_verb ad1986a_ch6_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode ad1986a_modes[3] = {
+static const struct hda_channel_mode ad1986a_modes[3] = {
 	{ 2, ad1986a_ch2_init },
 	{ 4, ad1986a_ch4_init },
 	{ 6, ad1986a_ch6_init },
 };
 
 /* eapd initialization */
-static struct hda_verb ad1986a_eapd_init_verbs[] = {
+static const struct hda_verb ad1986a_eapd_init_verbs[] = {
 	{0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 },
 	{}
 };
 
-static struct hda_verb ad1986a_automic_verbs[] = {
+static const struct hda_verb ad1986a_automic_verbs[] = {
 	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
 	{0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
 	/*{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},*/
@@ -1058,7 +1058,7 @@ static struct hda_verb ad1986a_automic_verbs[] = {
 };
 
 /* Ultra initialization */
-static struct hda_verb ad1986a_ultra_init[] = {
+static const struct hda_verb ad1986a_ultra_init[] = {
 	/* eapd initialization */
 	{ 0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 },
 	/* CLFE -> Mic in */
@@ -1069,7 +1069,7 @@ static struct hda_verb ad1986a_ultra_init[] = {
 };
 
 /* pin sensing on HP jack */
-static struct hda_verb ad1986a_hp_init_verbs[] = {
+static const struct hda_verb ad1986a_hp_init_verbs[] = {
 	{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_HP_EVENT},
 	{}
 };
@@ -1120,7 +1120,7 @@ static const char * const ad1986a_models[AD1986A_MODELS] = {
 	[AD1986A_SAMSUNG_P50]	= "samsung-p50",
 };
 
-static struct snd_pci_quirk ad1986a_cfg_tbl[] = {
+static const struct snd_pci_quirk ad1986a_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_LAPTOP_EAPD),
 	SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9", AD1986A_LAPTOP_EAPD),
 	SND_PCI_QUIRK(0x1043, 0x11f7, "ASUS U5A", AD1986A_LAPTOP_EAPD),
@@ -1152,7 +1152,7 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = {
 };
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list ad1986a_loopbacks[] = {
+static const struct hda_amp_list ad1986a_loopbacks[] = {
 	{ 0x13, HDA_OUTPUT, 0 }, /* Mic */
 	{ 0x14, HDA_OUTPUT, 0 }, /* Phone */
 	{ 0x15, HDA_OUTPUT, 0 }, /* CD */
@@ -1329,11 +1329,11 @@ static int patch_ad1986a(struct hda_codec *codec)
 #define AD1983_DAC		0x03
 #define AD1983_ADC		0x04
 
-static hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC };
-static hda_nid_t ad1983_adc_nids[1] = { AD1983_ADC };
-static hda_nid_t ad1983_capsrc_nids[1] = { 0x15 };
+static const hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC };
+static const hda_nid_t ad1983_adc_nids[1] = { AD1983_ADC };
+static const hda_nid_t ad1983_capsrc_nids[1] = { 0x15 };
 
-static struct hda_input_mux ad1983_capture_source = {
+static const struct hda_input_mux ad1983_capture_source = {
 	.num_items = 4,
 	.items = {
 		{ "Mic", 0x0 },
@@ -1348,7 +1348,7 @@ static struct hda_input_mux ad1983_capture_source = {
  */
 static int ad1983_spdif_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "PCM", "ADC" };
+	static const char * const texts[] = { "PCM", "ADC" };
 
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
@@ -1385,7 +1385,7 @@ static int ad1983_spdif_route_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
 	return 0;
 }
 
-static struct snd_kcontrol_new ad1983_mixers[] = {
+static const struct snd_kcontrol_new ad1983_mixers[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
@@ -1418,7 +1418,7 @@ static struct snd_kcontrol_new ad1983_mixers[] = {
 	{ } /* end */
 };
 
-static struct hda_verb ad1983_init_verbs[] = {
+static const struct hda_verb ad1983_init_verbs[] = {
 	/* Front, HP, Mono; mute as default */
 	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
 	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
@@ -1458,7 +1458,7 @@ static struct hda_verb ad1983_init_verbs[] = {
 };
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list ad1983_loopbacks[] = {
+static const struct hda_amp_list ad1983_loopbacks[] = {
 	{ 0x12, HDA_OUTPUT, 0 }, /* Mic */
 	{ 0x13, HDA_OUTPUT, 0 }, /* Line */
 	{ } /* end */
@@ -1518,12 +1518,12 @@ static int patch_ad1983(struct hda_codec *codec)
 #define AD1981_DAC		0x03
 #define AD1981_ADC		0x04
 
-static hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC };
-static hda_nid_t ad1981_adc_nids[1] = { AD1981_ADC };
-static hda_nid_t ad1981_capsrc_nids[1] = { 0x15 };
+static const hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC };
+static const hda_nid_t ad1981_adc_nids[1] = { AD1981_ADC };
+static const hda_nid_t ad1981_capsrc_nids[1] = { 0x15 };
 
 /* 0x0c, 0x09, 0x0e, 0x0f, 0x19, 0x05, 0x18, 0x17 */
-static struct hda_input_mux ad1981_capture_source = {
+static const struct hda_input_mux ad1981_capture_source = {
 	.num_items = 7,
 	.items = {
 		{ "Front Mic", 0x0 },
@@ -1536,7 +1536,7 @@ static struct hda_input_mux ad1981_capture_source = {
 	},
 };
 
-static struct snd_kcontrol_new ad1981_mixers[] = {
+static const struct snd_kcontrol_new ad1981_mixers[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
@@ -1577,7 +1577,7 @@ static struct snd_kcontrol_new ad1981_mixers[] = {
 	{ } /* end */
 };
 
-static struct hda_verb ad1981_init_verbs[] = {
+static const struct hda_verb ad1981_init_verbs[] = {
 	/* Front, HP, Mono; mute as default */
 	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
 	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
@@ -1625,7 +1625,7 @@ static struct hda_verb ad1981_init_verbs[] = {
 };
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list ad1981_loopbacks[] = {
+static const struct hda_amp_list ad1981_loopbacks[] = {
 	{ 0x12, HDA_OUTPUT, 0 }, /* Front Mic */
 	{ 0x13, HDA_OUTPUT, 0 }, /* Line */
 	{ 0x1b, HDA_OUTPUT, 0 }, /* Aux */
@@ -1645,7 +1645,7 @@ static struct hda_amp_list ad1981_loopbacks[] = {
 #define AD1981_HP_EVENT		0x37
 #define AD1981_MIC_EVENT	0x38
 
-static struct hda_verb ad1981_hp_init_verbs[] = {
+static const struct hda_verb ad1981_hp_init_verbs[] = {
 	{0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, /* default off */
 	/* pin sensing on HP and Mic jacks */
 	{0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT},
@@ -1674,7 +1674,7 @@ static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 }
 
 /* bind volumes of both NID 0x05 and 0x06 */
-static struct hda_bind_ctls ad1981_hp_bind_master_vol = {
+static const struct hda_bind_ctls ad1981_hp_bind_master_vol = {
 	.ops = &snd_hda_bind_vol,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT),
@@ -1696,12 +1696,12 @@ static void ad1981_hp_automute(struct hda_codec *codec)
 /* toggle input of built-in and mic jack appropriately */
 static void ad1981_hp_automic(struct hda_codec *codec)
 {
-	static struct hda_verb mic_jack_on[] = {
+	static const struct hda_verb mic_jack_on[] = {
 		{0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
 		{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
 		{}
 	};
-	static struct hda_verb mic_jack_off[] = {
+	static const struct hda_verb mic_jack_off[] = {
 		{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
 		{0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
 		{}
@@ -1730,7 +1730,7 @@ static void ad1981_hp_unsol_event(struct hda_codec *codec,
 	}
 }
 
-static struct hda_input_mux ad1981_hp_capture_source = {
+static const struct hda_input_mux ad1981_hp_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic", 0x0 },
@@ -1739,7 +1739,7 @@ static struct hda_input_mux ad1981_hp_capture_source = {
 	},
 };
 
-static struct snd_kcontrol_new ad1981_hp_mixers[] = {
+static const struct snd_kcontrol_new ad1981_hp_mixers[] = {
 	HDA_BIND_VOL("Master Playback Volume", &ad1981_hp_bind_master_vol),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1790,7 +1790,7 @@ static int ad1981_hp_init(struct hda_codec *codec)
 }
 
 /* configuration for Toshiba Laptops */
-static struct hda_verb ad1981_toshiba_init_verbs[] = {
+static const struct hda_verb ad1981_toshiba_init_verbs[] = {
 	{0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x01 }, /* default on */
 	/* pin sensing on HP and Mic jacks */
 	{0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT},
@@ -1798,14 +1798,14 @@ static struct hda_verb ad1981_toshiba_init_verbs[] = {
 	{}
 };
 
-static struct snd_kcontrol_new ad1981_toshiba_mixers[] = {
+static const struct snd_kcontrol_new ad1981_toshiba_mixers[] = {
 	HDA_CODEC_VOLUME("Amp Volume", 0x1a, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Amp Switch", 0x1a, 0x0, HDA_OUTPUT),
 	{ }
 };
 
 /* configuration for Lenovo Thinkpad T60 */
-static struct snd_kcontrol_new ad1981_thinkpad_mixers[] = {
+static const struct snd_kcontrol_new ad1981_thinkpad_mixers[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x05, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Master Playback Switch", 0x05, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
@@ -1835,7 +1835,7 @@ static struct snd_kcontrol_new ad1981_thinkpad_mixers[] = {
 	{ } /* end */
 };
 
-static struct hda_input_mux ad1981_thinkpad_capture_source = {
+static const struct hda_input_mux ad1981_thinkpad_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic", 0x0 },
@@ -1860,7 +1860,7 @@ static const char * const ad1981_models[AD1981_MODELS] = {
 	[AD1981_TOSHIBA]	= "toshiba"
 };
 
-static struct snd_pci_quirk ad1981_cfg_tbl[] = {
+static const struct snd_pci_quirk ad1981_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD),
 	SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD),
 	/* All HP models */
@@ -2075,32 +2075,32 @@ enum {
  * mixers
  */
 
-static hda_nid_t ad1988_6stack_dac_nids[4] = {
+static const hda_nid_t ad1988_6stack_dac_nids[4] = {
 	0x04, 0x06, 0x05, 0x0a
 };
 
-static hda_nid_t ad1988_3stack_dac_nids[3] = {
+static const hda_nid_t ad1988_3stack_dac_nids[3] = {
 	0x04, 0x05, 0x0a
 };
 
 /* for AD1988A revision-2, DAC2-4 are swapped */
-static hda_nid_t ad1988_6stack_dac_nids_rev2[4] = {
+static const hda_nid_t ad1988_6stack_dac_nids_rev2[4] = {
 	0x04, 0x05, 0x0a, 0x06
 };
 
-static hda_nid_t ad1988_alt_dac_nid[1] = {
+static const hda_nid_t ad1988_alt_dac_nid[1] = {
 	0x03
 };
 
-static hda_nid_t ad1988_3stack_dac_nids_rev2[3] = {
+static const hda_nid_t ad1988_3stack_dac_nids_rev2[3] = {
 	0x04, 0x0a, 0x06
 };
 
-static hda_nid_t ad1988_adc_nids[3] = {
+static const hda_nid_t ad1988_adc_nids[3] = {
 	0x08, 0x09, 0x0f
 };
 
-static hda_nid_t ad1988_capsrc_nids[3] = {
+static const hda_nid_t ad1988_capsrc_nids[3] = {
 	0x0c, 0x0d, 0x0e
 };
 
@@ -2108,11 +2108,11 @@ static hda_nid_t ad1988_capsrc_nids[3] = {
 #define AD1988_SPDIF_OUT_HDMI	0x0b
 #define AD1988_SPDIF_IN		0x07
 
-static hda_nid_t ad1989b_slave_dig_outs[] = {
+static const hda_nid_t ad1989b_slave_dig_outs[] = {
 	AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0
 };
 
-static struct hda_input_mux ad1988_6stack_capture_source = {
+static const struct hda_input_mux ad1988_6stack_capture_source = {
 	.num_items = 5,
 	.items = {
 		{ "Front Mic", 0x1 },	/* port-B */
@@ -2123,7 +2123,7 @@ static struct hda_input_mux ad1988_6stack_capture_source = {
 	},
 };
 
-static struct hda_input_mux ad1988_laptop_capture_source = {
+static const struct hda_input_mux ad1988_laptop_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic/Line", 0x1 },	/* port-B */
@@ -2166,7 +2166,7 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
 }
 
 /* 6-stack mode */
-static struct snd_kcontrol_new ad1988_6stack_mixers1[] = {
+static const struct snd_kcontrol_new ad1988_6stack_mixers1[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
@@ -2175,7 +2175,7 @@ static struct snd_kcontrol_new ad1988_6stack_mixers1[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = {
+static const struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x05, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
@@ -2184,7 +2184,7 @@ static struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
+static const struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
 	HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
 	HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT),
 	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT),
@@ -2211,14 +2211,14 @@ static struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1988_6stack_fp_mixers[] = {
+static const struct snd_kcontrol_new ad1988_6stack_fp_mixers[] = {
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
 
 	{ } /* end */
 };
 
 /* 3-stack mode */
-static struct snd_kcontrol_new ad1988_3stack_mixers1[] = {
+static const struct snd_kcontrol_new ad1988_3stack_mixers1[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
@@ -2226,7 +2226,7 @@ static struct snd_kcontrol_new ad1988_3stack_mixers1[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = {
+static const struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x06, 1, 0x0, HDA_OUTPUT),
@@ -2234,7 +2234,7 @@ static struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1988_3stack_mixers2[] = {
+static const struct snd_kcontrol_new ad1988_3stack_mixers2[] = {
 	HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
 	HDA_BIND_MUTE("Surround Playback Switch", 0x2c, 2, HDA_INPUT),
 	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x26, 1, 2, HDA_INPUT),
@@ -2268,7 +2268,7 @@ static struct snd_kcontrol_new ad1988_3stack_mixers2[] = {
 };
 
 /* laptop mode */
-static struct snd_kcontrol_new ad1988_laptop_mixers[] = {
+static const struct snd_kcontrol_new ad1988_laptop_mixers[] = {
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x29, 0x0, HDA_INPUT),
 	HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
@@ -2299,7 +2299,7 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = {
 };
 
 /* capture */
-static struct snd_kcontrol_new ad1988_capture_mixers[] = {
+static const struct snd_kcontrol_new ad1988_capture_mixers[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
@@ -2324,7 +2324,7 @@ static struct snd_kcontrol_new ad1988_capture_mixers[] = {
 static int ad1988_spdif_playback_source_info(struct snd_kcontrol *kcontrol,
 					     struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {
+	static const char * const texts[] = {
 		"PCM", "ADC1", "ADC2", "ADC3"
 	};
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
@@ -2405,7 +2405,7 @@ static int ad1988_spdif_playback_source_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new ad1988_spdif_out_mixers[] = {
+static const struct snd_kcontrol_new ad1988_spdif_out_mixers[] = {
 	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -2418,12 +2418,12 @@ static struct snd_kcontrol_new ad1988_spdif_out_mixers[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1988_spdif_in_mixers[] = {
+static const struct snd_kcontrol_new ad1988_spdif_in_mixers[] = {
 	HDA_CODEC_VOLUME("IEC958 Capture Volume", 0x1c, 0x0, HDA_INPUT),
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1989_spdif_out_mixers[] = {
+static const struct snd_kcontrol_new ad1989_spdif_out_mixers[] = {
 	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("HDMI Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
 	{ } /* end */
@@ -2436,7 +2436,7 @@ static struct snd_kcontrol_new ad1989_spdif_out_mixers[] = {
 /*
  * for 6-stack (+dig)
  */
-static struct hda_verb ad1988_6stack_init_verbs[] = {
+static const struct hda_verb ad1988_6stack_init_verbs[] = {
 	/* Front, Surround, CLFE, side DAC; unmute as default */
 	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -2496,7 +2496,7 @@ static struct hda_verb ad1988_6stack_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb ad1988_6stack_fp_init_verbs[] = {
+static const struct hda_verb ad1988_6stack_fp_init_verbs[] = {
 	/* Headphone; unmute as default */
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	/* Port-A front headphon path */
@@ -2509,7 +2509,7 @@ static struct hda_verb ad1988_6stack_fp_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb ad1988_capture_init_verbs[] = {
+static const struct hda_verb ad1988_capture_init_verbs[] = {
 	/* mute analog mix */
 	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
@@ -2527,7 +2527,7 @@ static struct hda_verb ad1988_capture_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb ad1988_spdif_init_verbs[] = {
+static const struct hda_verb ad1988_spdif_init_verbs[] = {
 	/* SPDIF out sel */
 	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
 	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */
@@ -2539,14 +2539,14 @@ static struct hda_verb ad1988_spdif_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb ad1988_spdif_in_init_verbs[] = {
+static const struct hda_verb ad1988_spdif_in_init_verbs[] = {
 	/* unmute SPDIF input pin */
 	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ }
 };
 
 /* AD1989 has no ADC -> SPDIF route */
-static struct hda_verb ad1989_spdif_init_verbs[] = {
+static const struct hda_verb ad1989_spdif_init_verbs[] = {
 	/* SPDIF-1 out pin */
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
@@ -2559,7 +2559,7 @@ static struct hda_verb ad1989_spdif_init_verbs[] = {
 /*
  * verbs for 3stack (+dig)
  */
-static struct hda_verb ad1988_3stack_ch2_init[] = {
+static const struct hda_verb ad1988_3stack_ch2_init[] = {
 	/* set port-C to line-in */
 	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
 	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
@@ -2569,7 +2569,7 @@ static struct hda_verb ad1988_3stack_ch2_init[] = {
 	{ } /* end */
 };
 
-static struct hda_verb ad1988_3stack_ch6_init[] = {
+static const struct hda_verb ad1988_3stack_ch6_init[] = {
 	/* set port-C to surround out */
 	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
@@ -2579,12 +2579,12 @@ static struct hda_verb ad1988_3stack_ch6_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode ad1988_3stack_modes[2] = {
+static const struct hda_channel_mode ad1988_3stack_modes[2] = {
 	{ 2, ad1988_3stack_ch2_init },
 	{ 6, ad1988_3stack_ch6_init },
 };
 
-static struct hda_verb ad1988_3stack_init_verbs[] = {
+static const struct hda_verb ad1988_3stack_init_verbs[] = {
 	/* Front, Surround, CLFE, side DAC; unmute as default */
 	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -2644,13 +2644,13 @@ static struct hda_verb ad1988_3stack_init_verbs[] = {
 /*
  * verbs for laptop mode (+dig)
  */
-static struct hda_verb ad1988_laptop_hp_on[] = {
+static const struct hda_verb ad1988_laptop_hp_on[] = {
 	/* unmute port-A and mute port-D */
 	{ 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
 	{ 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
 	{ } /* end */
 };
-static struct hda_verb ad1988_laptop_hp_off[] = {
+static const struct hda_verb ad1988_laptop_hp_off[] = {
 	/* mute port-A and unmute port-D */
 	{ 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
 	{ 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
@@ -2659,7 +2659,7 @@ static struct hda_verb ad1988_laptop_hp_off[] = {
 
 #define AD1988_HP_EVENT	0x01
 
-static struct hda_verb ad1988_laptop_init_verbs[] = {
+static const struct hda_verb ad1988_laptop_init_verbs[] = {
 	/* Front, Surround, CLFE, side DAC; unmute as default */
 	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -2723,7 +2723,7 @@ static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res)
 } 
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list ad1988_loopbacks[] = {
+static const struct hda_amp_list ad1988_loopbacks[] = {
 	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
 	{ 0x20, HDA_INPUT, 1 }, /* Line */
 	{ 0x20, HDA_INPUT, 4 }, /* Mic */
@@ -2741,7 +2741,7 @@ enum {
 	AD_CTL_WIDGET_MUTE,
 	AD_CTL_BIND_MUTE,
 };
-static struct snd_kcontrol_new ad1988_control_templates[] = {
+static const struct snd_kcontrol_new ad1988_control_templates[] = {
 	HDA_CODEC_VOLUME(NULL, 0, 0, 0),
 	HDA_CODEC_MUTE(NULL, 0, 0, 0),
 	HDA_BIND_MUTE(NULL, 0, 0, 0),
@@ -2770,18 +2770,18 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name,
 #define AD1988_PIN_CD_NID		0x18
 #define AD1988_PIN_BEEP_NID		0x10
 
-static hda_nid_t ad1988_mixer_nids[8] = {
+static const hda_nid_t ad1988_mixer_nids[8] = {
 	/* A     B     C     D     E     F     G     H */
 	0x22, 0x2b, 0x2c, 0x29, 0x26, 0x2a, 0x27, 0x28
 };
 
 static inline hda_nid_t ad1988_idx_to_dac(struct hda_codec *codec, int idx)
 {
-	static hda_nid_t idx_to_dac[8] = {
+	static const hda_nid_t idx_to_dac[8] = {
 		/* A     B     C     D     E     F     G     H */
 		0x04, 0x06, 0x05, 0x04, 0x0a, 0x06, 0x05, 0x0a
 	};
-	static hda_nid_t idx_to_dac_rev2[8] = {
+	static const hda_nid_t idx_to_dac_rev2[8] = {
 		/* A     B     C     D     E     F     G     H */
 		0x04, 0x05, 0x0a, 0x04, 0x06, 0x05, 0x0a, 0x06
 	};
@@ -2791,13 +2791,13 @@ static inline hda_nid_t ad1988_idx_to_dac(struct hda_codec *codec, int idx)
 		return idx_to_dac[idx];
 }
 
-static hda_nid_t ad1988_boost_nids[8] = {
+static const hda_nid_t ad1988_boost_nids[8] = {
 	0x38, 0x39, 0x3a, 0x3d, 0x3c, 0x3b, 0, 0
 };
 
 static int ad1988_pin_idx(hda_nid_t nid)
 {
-	static hda_nid_t ad1988_io_pins[8] = {
+	static const hda_nid_t ad1988_io_pins[8] = {
 		0x11, 0x14, 0x15, 0x12, 0x17, 0x16, 0x24, 0x25
 	};
 	int i;
@@ -2809,7 +2809,7 @@ static int ad1988_pin_idx(hda_nid_t nid)
 
 static int ad1988_pin_to_loopback_idx(hda_nid_t nid)
 {
-	static int loopback_idx[8] = {
+	static const int loopback_idx[8] = {
 		2, 0, 1, 3, 4, 5, 1, 4
 	};
 	switch (nid) {
@@ -2822,7 +2822,7 @@ static int ad1988_pin_to_loopback_idx(hda_nid_t nid)
 
 static int ad1988_pin_to_adc_idx(hda_nid_t nid)
 {
-	static int adc_idx[8] = {
+	static const int adc_idx[8] = {
 		0, 1, 2, 8, 4, 3, 6, 7
 	};
 	switch (nid) {
@@ -2845,7 +2845,7 @@ static int ad1988_auto_fill_dac_nids(struct hda_codec *codec,
 	/* check the pins hardwired to audio widget */
 	for (i = 0; i < cfg->line_outs; i++) {
 		idx = ad1988_pin_idx(cfg->line_out_pins[i]);
-		spec->multiout.dac_nids[i] = ad1988_idx_to_dac(codec, idx);
+		spec->private_dac_nids[i] = ad1988_idx_to_dac(codec, idx);
 	}
 	spec->multiout.num_dacs = cfg->line_outs;
 	return 0;
@@ -3070,6 +3070,7 @@ static void ad1988_auto_init_analog_input(struct hda_codec *codec)
 
 	for (i = 0; i < cfg->num_inputs; i++) {
 		hda_nid_t nid = cfg->inputs[i].pin;
+		int type = cfg->inputs[i].type;
 		switch (nid) {
 		case 0x15: /* port-C */
 			snd_hda_codec_write(codec, 0x33, 0, AC_VERB_SET_CONNECT_SEL, 0x0);
@@ -3079,7 +3080,7 @@ static void ad1988_auto_init_analog_input(struct hda_codec *codec)
 			break;
 		}
 		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-				    i == AUTO_PIN_MIC ? PIN_VREF80 : PIN_IN);
+				    type == AUTO_PIN_MIC ? PIN_VREF80 : PIN_IN);
 		if (nid != AD1988_PIN_CD_NID)
 			snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
 					    AMP_OUT_MUTE);
@@ -3154,10 +3155,11 @@ static const char * const ad1988_models[AD1988_MODEL_LAST] = {
 	[AD1988_AUTO]		= "auto",
 };
 
-static struct snd_pci_quirk ad1988_cfg_tbl[] = {
+static const struct snd_pci_quirk ad1988_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG),
 	SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG),
 	SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG),
+	SND_PCI_QUIRK(0x1043, 0x82c0, "Asus M3N-HT Deluxe", AD1988_6STACK_DIG),
 	SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG),
 	{}
 };
@@ -3342,21 +3344,21 @@ static int patch_ad1988(struct hda_codec *codec)
  * but no build-up framework is given, so far.
  */
 
-static hda_nid_t ad1884_dac_nids[1] = {
+static const hda_nid_t ad1884_dac_nids[1] = {
 	0x04,
 };
 
-static hda_nid_t ad1884_adc_nids[2] = {
+static const hda_nid_t ad1884_adc_nids[2] = {
 	0x08, 0x09,
 };
 
-static hda_nid_t ad1884_capsrc_nids[2] = {
+static const hda_nid_t ad1884_capsrc_nids[2] = {
 	0x0c, 0x0d,
 };
 
 #define AD1884_SPDIF_OUT	0x02
 
-static struct hda_input_mux ad1884_capture_source = {
+static const struct hda_input_mux ad1884_capture_source = {
 	.num_items = 4,
 	.items = {
 		{ "Front Mic", 0x0 },
@@ -3366,7 +3368,7 @@ static struct hda_input_mux ad1884_capture_source = {
 	},
 };
 
-static struct snd_kcontrol_new ad1884_base_mixers[] = {
+static const struct snd_kcontrol_new ad1884_base_mixers[] = {
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
 	/* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
@@ -3410,7 +3412,7 @@ static struct snd_kcontrol_new ad1884_base_mixers[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1984_dmic_mixers[] = {
+static const struct snd_kcontrol_new ad1984_dmic_mixers[] = {
 	HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0,
@@ -3423,7 +3425,7 @@ static struct snd_kcontrol_new ad1984_dmic_mixers[] = {
 /*
  * initialization verbs
  */
-static struct hda_verb ad1884_init_verbs[] = {
+static const struct hda_verb ad1884_init_verbs[] = {
 	/* DACs; mute as default */
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
@@ -3469,7 +3471,7 @@ static struct hda_verb ad1884_init_verbs[] = {
 };
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list ad1884_loopbacks[] = {
+static const struct hda_amp_list ad1884_loopbacks[] = {
 	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
 	{ 0x20, HDA_INPUT, 1 }, /* Mic */
 	{ 0x20, HDA_INPUT, 2 }, /* CD */
@@ -3541,7 +3543,7 @@ static int patch_ad1884(struct hda_codec *codec)
 /*
  * Lenovo Thinkpad T61/X61
  */
-static struct hda_input_mux ad1984_thinkpad_capture_source = {
+static const struct hda_input_mux ad1984_thinkpad_capture_source = {
 	.num_items = 4,
 	.items = {
 		{ "Mic", 0x0 },
@@ -3555,7 +3557,7 @@ static struct hda_input_mux ad1984_thinkpad_capture_source = {
 /*
  * Dell Precision T3400
  */
-static struct hda_input_mux ad1984_dell_desktop_capture_source = {
+static const struct hda_input_mux ad1984_dell_desktop_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Front Mic", 0x0 },
@@ -3565,7 +3567,7 @@ static struct hda_input_mux ad1984_dell_desktop_capture_source = {
 };
 
 
-static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = {
+static const struct snd_kcontrol_new ad1984_thinkpad_mixers[] = {
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
 	/* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
@@ -3611,7 +3613,7 @@ static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = {
 };
 
 /* additional verbs */
-static struct hda_verb ad1984_thinkpad_init_verbs[] = {
+static const struct hda_verb ad1984_thinkpad_init_verbs[] = {
 	/* Port-E (docking station mic) pin */
 	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
 	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
@@ -3629,7 +3631,7 @@ static struct hda_verb ad1984_thinkpad_init_verbs[] = {
 /*
  * Dell Precision T3400
  */
-static struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = {
+static const struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = {
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT),
@@ -3680,7 +3682,7 @@ static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo,
 	return 0;
 }
 
-static struct hda_pcm_stream ad1984_pcm_dmic_capture = {
+static const struct hda_pcm_stream ad1984_pcm_dmic_capture = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -3722,7 +3724,7 @@ static const char * const ad1984_models[AD1984_MODELS] = {
 	[AD1984_DELL_DESKTOP]	= "dell_desktop",
 };
 
-static struct snd_pci_quirk ad1984_cfg_tbl[] = {
+static const struct snd_pci_quirk ad1984_cfg_tbl[] = {
 	/* Lenovo Thinkpad T61/X61 */
 	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD),
 	SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP),
@@ -3787,7 +3789,7 @@ static int patch_ad1984(struct hda_codec *codec)
  * We share the single DAC for both HP and line-outs (see AD1884/1984).
  */
 
-static hda_nid_t ad1884a_dac_nids[1] = {
+static const hda_nid_t ad1884a_dac_nids[1] = {
 	0x03,
 };
 
@@ -3796,7 +3798,7 @@ static hda_nid_t ad1884a_dac_nids[1] = {
 
 #define AD1884A_SPDIF_OUT	0x02
 
-static struct hda_input_mux ad1884a_capture_source = {
+static const struct hda_input_mux ad1884a_capture_source = {
 	.num_items = 5,
 	.items = {
 		{ "Front Mic", 0x0 },
@@ -3807,7 +3809,7 @@ static struct hda_input_mux ad1884a_capture_source = {
 	},
 };
 
-static struct snd_kcontrol_new ad1884a_base_mixers[] = {
+static const struct snd_kcontrol_new ad1884a_base_mixers[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
@@ -3859,7 +3861,7 @@ static struct snd_kcontrol_new ad1884a_base_mixers[] = {
 /*
  * initialization verbs
  */
-static struct hda_verb ad1884a_init_verbs[] = {
+static const struct hda_verb ad1884a_init_verbs[] = {
 	/* DACs; unmute as default */
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
 	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
@@ -3914,7 +3916,7 @@ static struct hda_verb ad1884a_init_verbs[] = {
 };
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list ad1884a_loopbacks[] = {
+static const struct hda_amp_list ad1884a_loopbacks[] = {
 	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
 	{ 0x20, HDA_INPUT, 1 }, /* Mic */
 	{ 0x20, HDA_INPUT, 2 }, /* CD */
@@ -3947,7 +3949,7 @@ static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
 	return ret;
 }
 
-static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
+static const struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -3975,7 +3977,7 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
+static const struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
 	/*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
 	{
@@ -4095,7 +4097,7 @@ static int ad1884a_laptop_init(struct hda_codec *codec)
 }
 
 /* additional verbs for laptop model */
-static struct hda_verb ad1884a_laptop_verbs[] = {
+static const struct hda_verb ad1884a_laptop_verbs[] = {
 	/* Port-A (HP) pin - always unmuted */
 	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	/* Port-F (int speaker) mixer - route only from analog mixer */
@@ -4126,7 +4128,7 @@ static struct hda_verb ad1884a_laptop_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb ad1884a_mobile_verbs[] = {
+static const struct hda_verb ad1884a_mobile_verbs[] = {
 	/* DACs; unmute as default */
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
 	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
@@ -4181,7 +4183,7 @@ static struct hda_verb ad1884a_mobile_verbs[] = {
  * 0x17 - built-in mic
  */
 
-static struct hda_verb ad1984a_thinkpad_verbs[] = {
+static const struct hda_verb ad1984a_thinkpad_verbs[] = {
 	/* HP unmute */
 	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	/* analog mix */
@@ -4198,7 +4200,7 @@ static struct hda_verb ad1984a_thinkpad_verbs[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = {
+static const struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
@@ -4219,7 +4221,7 @@ static struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = {
 	{ } /* end */
 };
 
-static struct hda_input_mux ad1984a_thinkpad_capture_source = {
+static const struct hda_input_mux ad1984a_thinkpad_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic", 0x0 },
@@ -4262,7 +4264,7 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec)
  * 0x15 - mic-in
  */
 
-static struct hda_verb ad1984a_precision_verbs[] = {
+static const struct hda_verb ad1984a_precision_verbs[] = {
 	/* Unmute main output path */
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x1f}, /* 0dB */
@@ -4288,7 +4290,7 @@ static struct hda_verb ad1984a_precision_verbs[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1984a_precision_mixers[] = {
+static const struct snd_kcontrol_new ad1984a_precision_mixers[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
@@ -4344,7 +4346,7 @@ static int ad1984a_precision_init(struct hda_codec *codec)
  * digital-mic (0x17) - Internal mic
  */
 
-static struct hda_verb ad1984a_touchsmart_verbs[] = {
+static const struct hda_verb ad1984a_touchsmart_verbs[] = {
 	/* DACs; unmute as default */
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
 	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
@@ -4396,7 +4398,7 @@ static struct hda_verb ad1984a_touchsmart_verbs[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = {
+static const struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
 /*	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
 	{
@@ -4475,7 +4477,7 @@ static const char * const ad1884a_models[AD1884A_MODELS] = {
 	[AD1984A_PRECISION]	= "precision",
 };
 
-static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
+static const struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x04ac, "Precision R5500", AD1984A_PRECISION),
 	SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
 	SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP),
@@ -4614,22 +4616,22 @@ static int patch_ad1884a(struct hda_codec *codec)
  * port-G - rear clfe-out (6stack)
  */
 
-static hda_nid_t ad1882_dac_nids[3] = {
+static const hda_nid_t ad1882_dac_nids[3] = {
 	0x04, 0x03, 0x05
 };
 
-static hda_nid_t ad1882_adc_nids[2] = {
+static const hda_nid_t ad1882_adc_nids[2] = {
 	0x08, 0x09,
 };
 
-static hda_nid_t ad1882_capsrc_nids[2] = {
+static const hda_nid_t ad1882_capsrc_nids[2] = {
 	0x0c, 0x0d,
 };
 
 #define AD1882_SPDIF_OUT	0x02
 
 /* list: 0x11, 0x39, 0x3a, 0x18, 0x3c, 0x3b, 0x12, 0x20 */
-static struct hda_input_mux ad1882_capture_source = {
+static const struct hda_input_mux ad1882_capture_source = {
 	.num_items = 5,
 	.items = {
 		{ "Front Mic", 0x1 },
@@ -4641,7 +4643,7 @@ static struct hda_input_mux ad1882_capture_source = {
 };
 
 /* list: 0x11, 0x39, 0x3a, 0x3c, 0x18, 0x1f, 0x12, 0x20 */
-static struct hda_input_mux ad1882a_capture_source = {
+static const struct hda_input_mux ad1882a_capture_source = {
 	.num_items = 5,
 	.items = {
 		{ "Front Mic", 0x1 },
@@ -4652,7 +4654,7 @@ static struct hda_input_mux ad1882a_capture_source = {
 	},
 };
 
-static struct snd_kcontrol_new ad1882_base_mixers[] = {
+static const struct snd_kcontrol_new ad1882_base_mixers[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
@@ -4694,7 +4696,7 @@ static struct snd_kcontrol_new ad1882_base_mixers[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1882_loopback_mixers[] = {
+static const struct snd_kcontrol_new ad1882_loopback_mixers[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
@@ -4706,7 +4708,7 @@ static struct snd_kcontrol_new ad1882_loopback_mixers[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1882a_loopback_mixers[] = {
+static const struct snd_kcontrol_new ad1882a_loopback_mixers[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
@@ -4719,7 +4721,7 @@ static struct snd_kcontrol_new ad1882a_loopback_mixers[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1882_3stack_mixers[] = {
+static const struct snd_kcontrol_new ad1882_3stack_mixers[] = {
 	HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x17, 1, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x17, 2, 0x0, HDA_OUTPUT),
@@ -4733,14 +4735,14 @@ static struct snd_kcontrol_new ad1882_3stack_mixers[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new ad1882_6stack_mixers[] = {
+static const struct snd_kcontrol_new ad1882_6stack_mixers[] = {
 	HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x24, 2, 0x0, HDA_OUTPUT),
 	{ } /* end */
 };
 
-static struct hda_verb ad1882_ch2_init[] = {
+static const struct hda_verb ad1882_ch2_init[] = {
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
@@ -4750,7 +4752,7 @@ static struct hda_verb ad1882_ch2_init[] = {
 	{ } /* end */
 };
 
-static struct hda_verb ad1882_ch4_init[] = {
+static const struct hda_verb ad1882_ch4_init[] = {
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
@@ -4760,7 +4762,7 @@ static struct hda_verb ad1882_ch4_init[] = {
 	{ } /* end */
 };
 
-static struct hda_verb ad1882_ch6_init[] = {
+static const struct hda_verb ad1882_ch6_init[] = {
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
@@ -4770,7 +4772,7 @@ static struct hda_verb ad1882_ch6_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode ad1882_modes[3] = {
+static const struct hda_channel_mode ad1882_modes[3] = {
 	{ 2, ad1882_ch2_init },
 	{ 4, ad1882_ch4_init },
 	{ 6, ad1882_ch6_init },
@@ -4779,7 +4781,7 @@ static struct hda_channel_mode ad1882_modes[3] = {
 /*
  * initialization verbs
  */
-static struct hda_verb ad1882_init_verbs[] = {
+static const struct hda_verb ad1882_init_verbs[] = {
 	/* DACs; mute as default */
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
@@ -4848,7 +4850,7 @@ static struct hda_verb ad1882_init_verbs[] = {
 };
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list ad1882_loopbacks[] = {
+static const struct hda_amp_list ad1882_loopbacks[] = {
 	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
 	{ 0x20, HDA_INPUT, 1 }, /* Mic */
 	{ 0x20, HDA_INPUT, 4 }, /* Line */
@@ -4945,7 +4947,7 @@ static int patch_ad1882(struct hda_codec *codec)
 /*
  * patch entries
  */
-static struct hda_codec_preset snd_hda_preset_analog[] = {
+static const struct hda_codec_preset snd_hda_preset_analog[] = {
 	{ .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a },
 	{ .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 },
 	{ .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a },
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index 46c8bf48c31..61b92634b16 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -134,7 +134,7 @@ static int ca0110_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 /*
  */
 
-static char *dirstr[2] = { "Playback", "Capture" };
+static const char * const dirstr[2] = { "Playback", "Capture" };
 
 static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
 		       int chan, int dir)
@@ -171,7 +171,7 @@ static int ca0110_build_controls(struct hda_codec *codec)
 {
 	struct ca0110_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
-	static char *prefix[AUTO_CFG_MAX_OUTS] = {
+	static const char * const prefix[AUTO_CFG_MAX_OUTS] = {
 		"Front", "Surround", NULL, "Side", "Multi"
 	};
 	hda_nid_t mutenid;
@@ -259,7 +259,7 @@ static int ca0110_build_controls(struct hda_codec *codec)
 
 /*
  */
-static struct hda_pcm_stream ca0110_pcm_analog_playback = {
+static const struct hda_pcm_stream ca0110_pcm_analog_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 8,
@@ -270,7 +270,7 @@ static struct hda_pcm_stream ca0110_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream ca0110_pcm_analog_capture = {
+static const struct hda_pcm_stream ca0110_pcm_analog_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -280,7 +280,7 @@ static struct hda_pcm_stream ca0110_pcm_analog_capture = {
 	},
 };
 
-static struct hda_pcm_stream ca0110_pcm_digital_playback = {
+static const struct hda_pcm_stream ca0110_pcm_digital_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -291,7 +291,7 @@ static struct hda_pcm_stream ca0110_pcm_digital_playback = {
 	},
 };
 
-static struct hda_pcm_stream ca0110_pcm_digital_capture = {
+static const struct hda_pcm_stream ca0110_pcm_digital_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -389,7 +389,7 @@ static void ca0110_free(struct hda_codec *codec)
 	kfree(codec->spec);
 }
 
-static struct hda_codec_ops ca0110_patch_ops = {
+static const struct hda_codec_ops ca0110_patch_ops = {
 	.build_controls = ca0110_build_controls,
 	.build_pcms = ca0110_build_pcms,
 	.init = ca0110_init,
@@ -539,7 +539,7 @@ static int patch_ca0110(struct hda_codec *codec)
 /*
  * patch entries
  */
-static struct hda_codec_preset snd_hda_preset_ca0110[] = {
+static const struct hda_codec_preset snd_hda_preset_ca0110[] = {
 	{ .id = 0x1102000a, .name = "CA0110-IBG", .patch = patch_ca0110 },
 	{ .id = 0x1102000b, .name = "CA0110-IBG", .patch = patch_ca0110 },
 	{ .id = 0x1102000d, .name = "SB0880 X-Fi", .patch = patch_ca0110 },
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 067982f4f18..26a1521045b 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -51,7 +51,7 @@ struct cs_spec {
 	unsigned int cur_adc_format;
 	hda_nid_t dig_in;
 
-	struct hda_bind_ctls *capture_bind[2];
+	const struct hda_bind_ctls *capture_bind[2];
 
 	unsigned int gpio_mask;
 	unsigned int gpio_dir;
@@ -231,7 +231,7 @@ static int cs_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 
 /*
  */
-static struct hda_pcm_stream cs_pcm_analog_playback = {
+static const struct hda_pcm_stream cs_pcm_analog_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -242,7 +242,7 @@ static struct hda_pcm_stream cs_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream cs_pcm_analog_capture = {
+static const struct hda_pcm_stream cs_pcm_analog_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -252,7 +252,7 @@ static struct hda_pcm_stream cs_pcm_analog_capture = {
 	},
 };
 
-static struct hda_pcm_stream cs_pcm_digital_playback = {
+static const struct hda_pcm_stream cs_pcm_digital_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -264,7 +264,7 @@ static struct hda_pcm_stream cs_pcm_digital_playback = {
 	},
 };
 
-static struct hda_pcm_stream cs_pcm_digital_capture = {
+static const struct hda_pcm_stream cs_pcm_digital_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -331,8 +331,8 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
 	struct cs_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 	hda_nid_t pin = cfg->inputs[idx].pin;
-	unsigned int val = snd_hda_query_pin_caps(codec, pin);
-	if (!(val & AC_PINCAP_PRES_DETECT))
+	unsigned int val;
+	if (!is_jack_detectable(codec, pin))
 		return 0;
 	val = snd_hda_codec_get_pincfg(codec, pin);
 	return (snd_hda_get_input_pin_attr(val) != INPUT_PIN_ATTR_INT);
@@ -349,8 +349,7 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
 		hda_nid_t pins[2];
 		unsigned int type;
 		int j, nums;
-		type = (get_wcaps(codec, nid) & AC_WCAP_TYPE)
-			>> AC_WCAP_TYPE_SHIFT;
+		type = get_wcaps_type(get_wcaps(codec, nid));
 		if (type != AC_WID_AUD_IN)
 			continue;
 		nums = snd_hda_get_connections(codec, nid, pins,
@@ -559,10 +558,10 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx,
 	const char *name;
 	int err, index;
 	struct snd_kcontrol *kctl;
-	static char *speakers[] = {
+	static const char * const speakers[] = {
 		"Front Speaker", "Surround Speaker", "Bass Speaker"
 	};
-	static char *line_outs[] = {
+	static const char * const line_outs[] = {
 		"Front Line-Out", "Surround Line-Out", "Bass Line-Out"
 	};
 
@@ -642,7 +641,7 @@ static int build_output(struct hda_codec *codec)
 /*
  */
 
-static struct snd_kcontrol_new cs_capture_ctls[] = {
+static const struct snd_kcontrol_new cs_capture_ctls[] = {
 	HDA_BIND_SW("Capture Switch", 0),
 	HDA_BIND_VOL("Capture Volume", 0),
 };
@@ -710,7 +709,7 @@ static int cs_capture_source_put(struct snd_kcontrol *kcontrol,
 	return change_cur_input(codec, idx, 0);
 }
 
-static struct snd_kcontrol_new cs_capture_source = {
+static const struct snd_kcontrol_new cs_capture_source = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Capture Source",
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
@@ -719,7 +718,7 @@ static struct snd_kcontrol_new cs_capture_source = {
 	.put = cs_capture_source_put,
 };
 
-static struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec,
+static const struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec,
 					       struct hda_ctl_ops *ops)
 {
 	struct cs_spec *spec = codec->spec;
@@ -847,15 +846,14 @@ static void cs_automute(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
-	unsigned int caps, hp_present;
+	unsigned int hp_present;
 	hda_nid_t nid;
 	int i;
 
 	hp_present = 0;
 	for (i = 0; i < cfg->hp_outs; i++) {
 		nid = cfg->hp_pins[i];
-		caps = snd_hda_query_pin_caps(codec, nid);
-		if (!(caps & AC_PINCAP_PRES_DETECT))
+		if (!is_jack_detectable(codec, nid))
 			continue;
 		hp_present = snd_hda_jack_detect(codec, nid);
 		if (hp_present)
@@ -924,7 +922,7 @@ static void init_output(struct hda_codec *codec)
 				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
 		if (!cfg->speaker_outs)
 			continue;
-		if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) {
+		if (is_jack_detectable(codec, nid)) {
 			snd_hda_codec_write(codec, nid, 0,
 					    AC_VERB_SET_UNSOLICITED_ENABLE,
 					    AC_USRSP_EN | HP_EVENT);
@@ -983,7 +981,7 @@ static void init_input(struct hda_codec *codec)
 	cs_vendor_coef_set(codec, IDX_ADC_CFG, coef);
 }
 
-static struct hda_verb cs_coef_init_verbs[] = {
+static const struct hda_verb cs_coef_init_verbs[] = {
 	{0x11, AC_VERB_SET_PROC_STATE, 1},
 	{0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG},
 	{0x11, AC_VERB_SET_PROC_COEF,
@@ -1017,7 +1015,7 @@ static struct hda_verb cs_coef_init_verbs[] = {
  * blocks, which will alleviate the issue.
  */
 
-static struct hda_verb cs_errata_init_verbs[] = {
+static const struct hda_verb cs_errata_init_verbs[] = {
 	{0x01, AC_VERB_SET_POWER_STATE, 0x00}, /* AFG: D0 */
 	{0x11, AC_VERB_SET_PROC_STATE, 0x01},  /* VPW: processing on */
 
@@ -1126,7 +1124,7 @@ static void cs_unsol_event(struct hda_codec *codec, unsigned int res)
 	}
 }
 
-static struct hda_codec_ops cs_patch_ops = {
+static const struct hda_codec_ops cs_patch_ops = {
 	.build_controls = cs_build_controls,
 	.build_pcms = cs_build_pcms,
 	.init = cs_init,
@@ -1166,7 +1164,7 @@ static const char * const cs420x_models[CS420X_MODELS] = {
 };
 
 
-static struct snd_pci_quirk cs420x_cfg_tbl[] = {
+static const struct snd_pci_quirk cs420x_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x10de, 0x0ac0, "MacBookPro 5,3", CS420X_MBP53),
 	SND_PCI_QUIRK(0x10de, 0x0d94, "MacBookAir 3,1(2)", CS420X_MBP55),
 	SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55),
@@ -1180,7 +1178,7 @@ struct cs_pincfg {
 	u32 val;
 };
 
-static struct cs_pincfg mbp53_pincfgs[] = {
+static const struct cs_pincfg mbp53_pincfgs[] = {
 	{ 0x09, 0x012b4050 },
 	{ 0x0a, 0x90100141 },
 	{ 0x0b, 0x90100140 },
@@ -1194,7 +1192,7 @@ static struct cs_pincfg mbp53_pincfgs[] = {
 	{} /* terminator */
 };
 
-static struct cs_pincfg mbp55_pincfgs[] = {
+static const struct cs_pincfg mbp55_pincfgs[] = {
 	{ 0x09, 0x012b4030 },
 	{ 0x0a, 0x90100121 },
 	{ 0x0b, 0x90100120 },
@@ -1208,7 +1206,7 @@ static struct cs_pincfg mbp55_pincfgs[] = {
 	{} /* terminator */
 };
 
-static struct cs_pincfg imac27_pincfgs[] = {
+static const struct cs_pincfg imac27_pincfgs[] = {
 	{ 0x09, 0x012b4050 },
 	{ 0x0a, 0x90100140 },
 	{ 0x0b, 0x90100142 },
@@ -1222,7 +1220,7 @@ static struct cs_pincfg imac27_pincfgs[] = {
 	{} /* terminator */
 };
 
-static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = {
+static const struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = {
 	[CS420X_MBP53] = mbp53_pincfgs,
 	[CS420X_MBP55] = mbp55_pincfgs,
 	[CS420X_IMAC27] = imac27_pincfgs,
@@ -1283,7 +1281,7 @@ static int patch_cs420x(struct hda_codec *codec)
 /*
  * patch entries
  */
-static struct hda_codec_preset snd_hda_preset_cirrus[] = {
+static const struct hda_codec_preset snd_hda_preset_cirrus[] = {
 	{ .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x },
 	{ .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x },
 	{} /* terminator */
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 1f8bbcd0f80..ab3308daa96 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -53,7 +53,7 @@ struct cmi_spec {
 	int num_dacs;
 
 	/* capture */
-	hda_nid_t *adc_nids;
+	const hda_nid_t *adc_nids;
 	hda_nid_t dig_in_nid;
 
 	/* capture source */
@@ -110,7 +110,7 @@ static int cmi_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
  */
 
 /* 3-stack / 2 channel */
-static struct hda_verb cmi9880_ch2_init[] = {
+static const struct hda_verb cmi9880_ch2_init[] = {
 	/* set line-in PIN for input */
 	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
 	/* set mic PIN for input, also enable vref */
@@ -121,7 +121,7 @@ static struct hda_verb cmi9880_ch2_init[] = {
 };
 
 /* 3-stack / 6 channel */
-static struct hda_verb cmi9880_ch6_init[] = {
+static const struct hda_verb cmi9880_ch6_init[] = {
 	/* set line-in PIN for output */
 	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	/* set mic PIN for output */
@@ -132,7 +132,7 @@ static struct hda_verb cmi9880_ch6_init[] = {
 };
 
 /* 3-stack+front / 8 channel */
-static struct hda_verb cmi9880_ch8_init[] = {
+static const struct hda_verb cmi9880_ch8_init[] = {
 	/* set line-in PIN for output */
 	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	/* set mic PIN for output */
@@ -142,7 +142,7 @@ static struct hda_verb cmi9880_ch8_init[] = {
 	{}
 };
 
-static struct hda_channel_mode cmi9880_channel_modes[3] = {
+static const struct hda_channel_mode cmi9880_channel_modes[3] = {
 	{ 2, cmi9880_ch2_init },
 	{ 6, cmi9880_ch6_init },
 	{ 8, cmi9880_ch8_init },
@@ -174,7 +174,7 @@ static int cmi_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va
 
 /*
  */
-static struct snd_kcontrol_new cmi9880_basic_mixer[] = {
+static const struct snd_kcontrol_new cmi9880_basic_mixer[] = {
 	/* CMI9880 has no playback volumes! */
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), /* front */
 	HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0x0, HDA_OUTPUT),
@@ -205,7 +205,7 @@ static struct snd_kcontrol_new cmi9880_basic_mixer[] = {
 /*
  * shared I/O pins
  */
-static struct snd_kcontrol_new cmi9880_ch_mode_mixer[] = {
+static const struct snd_kcontrol_new cmi9880_ch_mode_mixer[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Channel Mode",
@@ -219,7 +219,7 @@ static struct snd_kcontrol_new cmi9880_ch_mode_mixer[] = {
 /* AUD-in selections:
  * 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x1f 0x20
  */
-static struct hda_input_mux cmi9880_basic_mux = {
+static const struct hda_input_mux cmi9880_basic_mux = {
 	.num_items = 4,
 	.items = {
 		{ "Front Mic", 0x5 },
@@ -229,7 +229,7 @@ static struct hda_input_mux cmi9880_basic_mux = {
 	}
 };
 
-static struct hda_input_mux cmi9880_no_line_mux = {
+static const struct hda_input_mux cmi9880_no_line_mux = {
 	.num_items = 3,
 	.items = {
 		{ "Front Mic", 0x5 },
@@ -239,11 +239,11 @@ static struct hda_input_mux cmi9880_no_line_mux = {
 };
 
 /* front, rear, clfe, rear_surr */
-static hda_nid_t cmi9880_dac_nids[4] = {
+static const hda_nid_t cmi9880_dac_nids[4] = {
 	0x03, 0x04, 0x05, 0x06
 };
 /* ADC0, ADC1 */
-static hda_nid_t cmi9880_adc_nids[2] = {
+static const hda_nid_t cmi9880_adc_nids[2] = {
 	0x08, 0x09
 };
 
@@ -252,7 +252,7 @@ static hda_nid_t cmi9880_adc_nids[2] = {
 
 /*
  */
-static struct hda_verb cmi9880_basic_init[] = {
+static const struct hda_verb cmi9880_basic_init[] = {
 	/* port-D for line out (rear panel) */
 	{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
 	/* port-E for HP out (front panel) */
@@ -281,7 +281,7 @@ static struct hda_verb cmi9880_basic_init[] = {
 	{} /* terminator */
 };
 
-static struct hda_verb cmi9880_allout_init[] = {
+static const struct hda_verb cmi9880_allout_init[] = {
 	/* port-D for line out (rear panel) */
 	{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
 	/* port-E for HP out (front panel) */
@@ -528,7 +528,7 @@ static int cmi9880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 
 /*
  */
-static struct hda_pcm_stream cmi9880_pcm_analog_playback = {
+static const struct hda_pcm_stream cmi9880_pcm_analog_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 8,
@@ -540,7 +540,7 @@ static struct hda_pcm_stream cmi9880_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream cmi9880_pcm_analog_capture = {
+static const struct hda_pcm_stream cmi9880_pcm_analog_capture = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -551,7 +551,7 @@ static struct hda_pcm_stream cmi9880_pcm_analog_capture = {
 	},
 };
 
-static struct hda_pcm_stream cmi9880_pcm_digital_playback = {
+static const struct hda_pcm_stream cmi9880_pcm_digital_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -563,7 +563,7 @@ static struct hda_pcm_stream cmi9880_pcm_digital_playback = {
 	},
 };
 
-static struct hda_pcm_stream cmi9880_pcm_digital_capture = {
+static const struct hda_pcm_stream cmi9880_pcm_digital_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -617,14 +617,14 @@ static const char * const cmi9880_models[CMI_MODELS] = {
 	[CMI_AUTO]	= "auto",
 };
 
-static struct snd_pci_quirk cmi9880_cfg_tbl[] = {
+static const struct snd_pci_quirk cmi9880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", CMI_FULL_DIG),
 	SND_PCI_QUIRK(0x1854, 0x002b, "LG LS75", CMI_MINIMAL),
 	SND_PCI_QUIRK(0x1854, 0x0032, "LG", CMI_FULL_DIG),
 	{} /* terminator */
 };
 
-static struct hda_codec_ops cmi9880_patch_ops = {
+static const struct hda_codec_ops cmi9880_patch_ops = {
 	.build_controls = cmi9880_build_controls,
 	.build_pcms = cmi9880_build_pcms,
 	.init = cmi9880_init,
@@ -745,7 +745,7 @@ static int patch_cmi9880(struct hda_codec *codec)
 /*
  * patch entries
  */
-static struct hda_codec_preset snd_hda_preset_cmedia[] = {
+static const struct hda_codec_preset snd_hda_preset_cmedia[] = {
 	{ .id = 0x13f69880, .name = "CMI9880", .patch = patch_cmi9880 },
  	{ .id = 0x434d4980, .name = "CMI9880", .patch = patch_cmi9880 },
 	{} /* terminator */
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index ad97d937d3a..3e6b9a8539c 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -39,6 +39,7 @@
 
 #define CONEXANT_HP_EVENT	0x37
 #define CONEXANT_MIC_EVENT	0x38
+#define CONEXANT_LINE_EVENT	0x39
 
 /* Conexant 5051 specific */
 
@@ -55,9 +56,16 @@ struct pin_dac_pair {
 	int type;
 };
 
+struct imux_info {
+	hda_nid_t pin;		/* input pin NID */
+	hda_nid_t adc;		/* connected ADC NID */	
+	hda_nid_t boost;	/* optional boost volume NID */
+	int index;		/* corresponding to autocfg.input */
+};
+
 struct conexant_spec {
 
-	struct snd_kcontrol_new *mixers[5];
+	const struct snd_kcontrol_new *mixers[5];
 	int num_mixers;
 	hda_nid_t vmaster_nid;
 
@@ -74,14 +82,17 @@ struct conexant_spec {
 					 */
 	unsigned int cur_eapd;
 	unsigned int hp_present;
+	unsigned int line_present;
 	unsigned int auto_mic;
-	int auto_mic_ext;		/* autocfg.inputs[] index for ext mic */
+	int auto_mic_ext;		/* imux_pins[] index for ext mic */
+	int auto_mic_dock;		/* imux_pins[] index for dock mic */
+	int auto_mic_int;		/* imux_pins[] index for int mic */
 	unsigned int need_dac_fix;
 	hda_nid_t slave_dig_outs[2];
 
 	/* capture */
 	unsigned int num_adc_nids;
-	hda_nid_t *adc_nids;
+	const hda_nid_t *adc_nids;
 	hda_nid_t dig_in_nid;		/* digital-in NID; optional */
 
 	unsigned int cur_adc_idx;
@@ -89,9 +100,11 @@ struct conexant_spec {
 	unsigned int cur_adc_stream_tag;
 	unsigned int cur_adc_format;
 
+	const struct hda_pcm_stream *capture_stream;
+
 	/* capture source */
 	const struct hda_input_mux *input_mux;
-	hda_nid_t *capsrc_nids;
+	const hda_nid_t *capsrc_nids;
 	unsigned int cur_mux[3];
 
 	/* channel model */
@@ -106,12 +119,17 @@ struct conexant_spec {
 	/* dynamic controls, init_verbs and input_mux */
 	struct auto_pin_cfg autocfg;
 	struct hda_input_mux private_imux;
+	struct imux_info imux_info[HDA_MAX_NUM_INPUTS];
+	hda_nid_t private_adc_nids[HDA_MAX_NUM_INPUTS];
 	hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
 	struct pin_dac_pair dac_info[8];
 	int dac_info_filled;
 
 	unsigned int port_d_mode;
 	unsigned int auto_mute:1;	/* used in auto-parser */
+	unsigned int detect_line:1;	/* Line-out detection enabled */
+	unsigned int automute_lines:1;	/* automute line-out as well */
+	unsigned int automute_hp_lo:1;	/* both HP and LO available */
 	unsigned int dell_automute:1;
 	unsigned int dell_vostro:1;
 	unsigned int ideapad:1;
@@ -119,6 +137,8 @@ struct conexant_spec {
 	unsigned int hp_laptop:1;
 	unsigned int asus:1;
 
+	unsigned int adc_switching:1;
+
 	unsigned int ext_mic_present;
 	unsigned int recording;
 	void (*capture_prepare)(struct hda_codec *codec);
@@ -227,7 +247,7 @@ static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 
 
 
-static struct hda_pcm_stream conexant_pcm_analog_playback = {
+static const struct hda_pcm_stream conexant_pcm_analog_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -239,7 +259,7 @@ static struct hda_pcm_stream conexant_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream conexant_pcm_analog_capture = {
+static const struct hda_pcm_stream conexant_pcm_analog_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -251,7 +271,7 @@ static struct hda_pcm_stream conexant_pcm_analog_capture = {
 };
 
 
-static struct hda_pcm_stream conexant_pcm_digital_playback = {
+static const struct hda_pcm_stream conexant_pcm_digital_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -263,7 +283,7 @@ static struct hda_pcm_stream conexant_pcm_digital_playback = {
 	},
 };
 
-static struct hda_pcm_stream conexant_pcm_digital_capture = {
+static const struct hda_pcm_stream conexant_pcm_digital_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -294,7 +314,7 @@ static int cx5051_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 	return 0;
 }
 
-static struct hda_pcm_stream cx5051_pcm_analog_capture = {
+static const struct hda_pcm_stream cx5051_pcm_analog_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -319,13 +339,19 @@ static int conexant_build_pcms(struct hda_codec *codec)
 		spec->multiout.max_channels;
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
 		spec->multiout.dac_nids[0];
-	if (codec->vendor_id == 0x14f15051)
-		info->stream[SNDRV_PCM_STREAM_CAPTURE] =
-			cx5051_pcm_analog_capture;
-	else
-		info->stream[SNDRV_PCM_STREAM_CAPTURE] =
-			conexant_pcm_analog_capture;
-	info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids;
+	if (spec->capture_stream)
+		info->stream[SNDRV_PCM_STREAM_CAPTURE] = *spec->capture_stream;
+	else {
+		if (codec->vendor_id == 0x14f15051)
+			info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+				cx5051_pcm_analog_capture;
+		else {
+			info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+				conexant_pcm_analog_capture;
+			info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams =
+				spec->num_adc_nids;
+		}
+	}
 	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
 
 	if (spec->multiout.dig_out_nid) {
@@ -433,7 +459,7 @@ static void conexant_free(struct hda_codec *codec)
 	kfree(codec->spec);
 }
 
-static struct snd_kcontrol_new cxt_capture_mixers[] = {
+static const struct snd_kcontrol_new cxt_capture_mixers[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Capture Source",
@@ -446,7 +472,7 @@ static struct snd_kcontrol_new cxt_capture_mixers[] = {
 
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
 /* additional beep mixers; the actual parameters are overwritten at build */
-static struct snd_kcontrol_new cxt_beep_mixer[] = {
+static const struct snd_kcontrol_new cxt_beep_mixer[] = {
 	HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0, 1, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_BEEP_MONO("Beep Playback Switch", 0, 1, 0, HDA_OUTPUT),
 	{ } /* end */
@@ -456,12 +482,18 @@ static struct snd_kcontrol_new cxt_beep_mixer[] = {
 static const char * const slave_vols[] = {
 	"Headphone Playback Volume",
 	"Speaker Playback Volume",
+	"Front Playback Volume",
+	"Surround Playback Volume",
+	"CLFE Playback Volume",
 	NULL
 };
 
 static const char * const slave_sws[] = {
 	"Headphone Playback Switch",
 	"Speaker Playback Switch",
+	"Front Playback Switch",
+	"Surround Playback Switch",
+	"CLFE Playback Switch",
 	NULL
 };
 
@@ -521,7 +553,7 @@ static int conexant_build_controls(struct hda_codec *codec)
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
 	/* create beep controls if needed */
 	if (spec->beep_amp) {
-		struct snd_kcontrol_new *knew;
+		const struct snd_kcontrol_new *knew;
 		for (knew = cxt_beep_mixer; knew->name; knew++) {
 			struct snd_kcontrol *kctl;
 			kctl = snd_ctl_new1(knew, codec);
@@ -546,7 +578,7 @@ static int conexant_suspend(struct hda_codec *codec, pm_message_t state)
 }
 #endif
 
-static struct hda_codec_ops conexant_patch_ops = {
+static const struct hda_codec_ops conexant_patch_ops = {
 	.build_controls = conexant_build_controls,
 	.build_pcms = conexant_build_pcms,
 	.init = conexant_init,
@@ -564,6 +596,7 @@ static struct hda_codec_ops conexant_patch_ops = {
 #define set_beep_amp(spec, nid, idx, dir) /* NOP */
 #endif
 
+static int patch_conexant_auto(struct hda_codec *codec);
 /*
  * EAPD control
  * the private value = nid | (invert << 8)
@@ -662,16 +695,16 @@ static int conexant_ch_mode_put(struct snd_kcontrol *kcontrol,
 
 /* Conexant 5045 specific */
 
-static hda_nid_t cxt5045_dac_nids[1] = { 0x19 };
-static hda_nid_t cxt5045_adc_nids[1] = { 0x1a };
-static hda_nid_t cxt5045_capsrc_nids[1] = { 0x1a };
+static const hda_nid_t cxt5045_dac_nids[1] = { 0x19 };
+static const hda_nid_t cxt5045_adc_nids[1] = { 0x1a };
+static const hda_nid_t cxt5045_capsrc_nids[1] = { 0x1a };
 #define CXT5045_SPDIF_OUT	0x18
 
-static struct hda_channel_mode cxt5045_modes[1] = {
+static const struct hda_channel_mode cxt5045_modes[1] = {
 	{ 2, NULL },
 };
 
-static struct hda_input_mux cxt5045_capture_source = {
+static const struct hda_input_mux cxt5045_capture_source = {
 	.num_items = 2,
 	.items = {
 		{ "IntMic", 0x1 },
@@ -679,7 +712,7 @@ static struct hda_input_mux cxt5045_capture_source = {
 	}
 };
 
-static struct hda_input_mux cxt5045_capture_source_benq = {
+static const struct hda_input_mux cxt5045_capture_source_benq = {
 	.num_items = 5,
 	.items = {
 		{ "IntMic", 0x1 },
@@ -690,7 +723,7 @@ static struct hda_input_mux cxt5045_capture_source_benq = {
 	}
 };
 
-static struct hda_input_mux cxt5045_capture_source_hp530 = {
+static const struct hda_input_mux cxt5045_capture_source_hp530 = {
 	.num_items = 2,
 	.items = {
 		{ "ExtMic", 0x1 },
@@ -723,7 +756,7 @@ static int cxt5045_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 }
 
 /* bind volumes of both NID 0x10 and 0x11 */
-static struct hda_bind_ctls cxt5045_hp_bind_master_vol = {
+static const struct hda_bind_ctls cxt5045_hp_bind_master_vol = {
 	.ops = &snd_hda_bind_vol,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x10, 3, 0, HDA_OUTPUT),
@@ -735,12 +768,12 @@ static struct hda_bind_ctls cxt5045_hp_bind_master_vol = {
 /* toggle input of built-in and mic jack appropriately */
 static void cxt5045_hp_automic(struct hda_codec *codec)
 {
-	static struct hda_verb mic_jack_on[] = {
+	static const struct hda_verb mic_jack_on[] = {
 		{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
 		{0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
 		{}
 	};
-	static struct hda_verb mic_jack_off[] = {
+	static const struct hda_verb mic_jack_off[] = {
 		{0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
 		{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
 		{}
@@ -784,7 +817,7 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec,
 	}
 }
 
-static struct snd_kcontrol_new cxt5045_mixers[] = {
+static const struct snd_kcontrol_new cxt5045_mixers[] = {
 	HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
@@ -808,7 +841,7 @@ static struct snd_kcontrol_new cxt5045_mixers[] = {
 	{}
 };
 
-static struct snd_kcontrol_new cxt5045_benq_mixers[] = {
+static const struct snd_kcontrol_new cxt5045_benq_mixers[] = {
 	HDA_CODEC_VOLUME("CD Capture Volume", 0x1a, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Capture Switch", 0x1a, 0x04, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT),
@@ -825,7 +858,7 @@ static struct snd_kcontrol_new cxt5045_benq_mixers[] = {
 	{}
 };
 
-static struct snd_kcontrol_new cxt5045_mixers_hp530[] = {
+static const struct snd_kcontrol_new cxt5045_mixers_hp530[] = {
 	HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
@@ -849,7 +882,7 @@ static struct snd_kcontrol_new cxt5045_mixers_hp530[] = {
 	{}
 };
 
-static struct hda_verb cxt5045_init_verbs[] = {
+static const struct hda_verb cxt5045_init_verbs[] = {
 	/* Line in, Mic */
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 },
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 },
@@ -875,7 +908,7 @@ static struct hda_verb cxt5045_init_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb cxt5045_benq_init_verbs[] = {
+static const struct hda_verb cxt5045_benq_init_verbs[] = {
 	/* Internal Mic, Mic */
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 },
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 },
@@ -901,13 +934,13 @@ static struct hda_verb cxt5045_benq_init_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb cxt5045_hp_sense_init_verbs[] = {
+static const struct hda_verb cxt5045_hp_sense_init_verbs[] = {
 	/* pin sensing on HP jack */
 	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
 	{ } /* end */
 };
 
-static struct hda_verb cxt5045_mic_sense_init_verbs[] = {
+static const struct hda_verb cxt5045_mic_sense_init_verbs[] = {
 	/* pin sensing on HP jack */
 	{0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
 	{ } /* end */
@@ -917,7 +950,7 @@ static struct hda_verb cxt5045_mic_sense_init_verbs[] = {
 /* Test configuration for debugging, modelled after the ALC260 test
  * configuration.
  */
-static struct hda_input_mux cxt5045_test_capture_source = {
+static const struct hda_input_mux cxt5045_test_capture_source = {
 	.num_items = 5,
 	.items = {
 		{ "MIXER", 0x0 },
@@ -928,7 +961,7 @@ static struct hda_input_mux cxt5045_test_capture_source = {
         },
 };
 
-static struct snd_kcontrol_new cxt5045_test_mixer[] = {
+static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
 
 	/* Output controls */
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x10, 0x0, HDA_OUTPUT),
@@ -978,7 +1011,7 @@ static struct snd_kcontrol_new cxt5045_test_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb cxt5045_test_init_verbs[] = {
+static const struct hda_verb cxt5045_test_init_verbs[] = {
 	/* Set connections */
 	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
 	{ 0x11, AC_VERB_SET_CONNECT_SEL, 0x0 },
@@ -1047,6 +1080,7 @@ enum {
 #ifdef CONFIG_SND_DEBUG
 	CXT5045_TEST,
 #endif
+	CXT5045_AUTO,
 	CXT5045_MODELS
 };
 
@@ -1059,9 +1093,10 @@ static const char * const cxt5045_models[CXT5045_MODELS] = {
 #ifdef CONFIG_SND_DEBUG
 	[CXT5045_TEST]		= "test",
 #endif
+	[CXT5045_AUTO]			= "auto",
 };
 
-static struct snd_pci_quirk cxt5045_cfg_tbl[] = {
+static const struct snd_pci_quirk cxt5045_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series",
 			   CXT5045_LAPTOP_HPSENSE),
@@ -1085,6 +1120,16 @@ static int patch_cxt5045(struct hda_codec *codec)
 	struct conexant_spec *spec;
 	int board_config;
 
+	board_config = snd_hda_check_board_config(codec, CXT5045_MODELS,
+						  cxt5045_models,
+						  cxt5045_cfg_tbl);
+#if 0 /* use the old method just for safety */
+	if (board_config < 0)
+		board_config = CXT5045_AUTO;
+#endif
+	if (board_config == CXT5045_AUTO)
+		return patch_conexant_auto(codec);
+
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (!spec)
 		return -ENOMEM;
@@ -1111,9 +1156,6 @@ static int patch_cxt5045(struct hda_codec *codec)
 
 	codec->patch_ops = conexant_patch_ops;
 
-	board_config = snd_hda_check_board_config(codec, CXT5045_MODELS,
-						  cxt5045_models,
-						  cxt5045_cfg_tbl);
 	switch (board_config) {
 	case CXT5045_LAPTOP_HPSENSE:
 		codec->patch_ops.unsol_event = cxt5045_hp_unsol_event;
@@ -1196,15 +1238,15 @@ static int patch_cxt5045(struct hda_codec *codec)
 /* Conexant 5047 specific */
 #define CXT5047_SPDIF_OUT	0x11
 
-static hda_nid_t cxt5047_dac_nids[1] = { 0x10 }; /* 0x1c */
-static hda_nid_t cxt5047_adc_nids[1] = { 0x12 };
-static hda_nid_t cxt5047_capsrc_nids[1] = { 0x1a };
+static const hda_nid_t cxt5047_dac_nids[1] = { 0x10 }; /* 0x1c */
+static const hda_nid_t cxt5047_adc_nids[1] = { 0x12 };
+static const hda_nid_t cxt5047_capsrc_nids[1] = { 0x1a };
 
-static struct hda_channel_mode cxt5047_modes[1] = {
+static const struct hda_channel_mode cxt5047_modes[1] = {
 	{ 2, NULL },
 };
 
-static struct hda_input_mux cxt5047_toshiba_capture_source = {
+static const struct hda_input_mux cxt5047_toshiba_capture_source = {
 	.num_items = 2,
 	.items = {
 		{ "ExtMic", 0x2 },
@@ -1256,12 +1298,12 @@ static void cxt5047_hp_automute(struct hda_codec *codec)
 /* toggle input of built-in and mic jack appropriately */
 static void cxt5047_hp_automic(struct hda_codec *codec)
 {
-	static struct hda_verb mic_jack_on[] = {
+	static const struct hda_verb mic_jack_on[] = {
 		{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 		{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 		{}
 	};
-	static struct hda_verb mic_jack_off[] = {
+	static const struct hda_verb mic_jack_off[] = {
 		{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 		{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 		{}
@@ -1289,7 +1331,7 @@ static void cxt5047_hp_unsol_event(struct hda_codec *codec,
 	}
 }
 
-static struct snd_kcontrol_new cxt5047_base_mixers[] = {
+static const struct snd_kcontrol_new cxt5047_base_mixers[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x19, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x19, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost Volume", 0x1a, 0x0, HDA_OUTPUT),
@@ -1309,19 +1351,19 @@ static struct snd_kcontrol_new cxt5047_base_mixers[] = {
 	{}
 };
 
-static struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = {
+static const struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = {
 	/* See the note in cxt5047_hp_master_sw_put */
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x01, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT),
 	{}
 };
 
-static struct snd_kcontrol_new cxt5047_hp_only_mixers[] = {
+static const struct snd_kcontrol_new cxt5047_hp_only_mixers[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x13, 0x00, HDA_OUTPUT),
 	{ } /* end */
 };
 
-static struct hda_verb cxt5047_init_verbs[] = {
+static const struct hda_verb cxt5047_init_verbs[] = {
 	/* Line in, Mic, Built-in Mic */
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 },
@@ -1348,7 +1390,7 @@ static struct hda_verb cxt5047_init_verbs[] = {
 };
 
 /* configuration for Toshiba Laptops */
-static struct hda_verb cxt5047_toshiba_init_verbs[] = {
+static const struct hda_verb cxt5047_toshiba_init_verbs[] = {
 	{0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0}, /* default off */
 	{}
 };
@@ -1357,7 +1399,7 @@ static struct hda_verb cxt5047_toshiba_init_verbs[] = {
  * configuration.
  */
 #ifdef CONFIG_SND_DEBUG
-static struct hda_input_mux cxt5047_test_capture_source = {
+static const struct hda_input_mux cxt5047_test_capture_source = {
 	.num_items = 4,
 	.items = {
 		{ "LINE1 pin", 0x0 },
@@ -1367,7 +1409,7 @@ static struct hda_input_mux cxt5047_test_capture_source = {
         },
 };
 
-static struct snd_kcontrol_new cxt5047_test_mixer[] = {
+static const struct snd_kcontrol_new cxt5047_test_mixer[] = {
 
 	/* Output only controls */
 	HDA_CODEC_VOLUME("OutAmp-1 Volume", 0x10, 0x0, HDA_OUTPUT),
@@ -1420,7 +1462,7 @@ static struct snd_kcontrol_new cxt5047_test_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb cxt5047_test_init_verbs[] = {
+static const struct hda_verb cxt5047_test_init_verbs[] = {
 	/* Enable retasking pins as output, initially without power amp */
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
@@ -1492,6 +1534,7 @@ enum {
 #ifdef CONFIG_SND_DEBUG
 	CXT5047_TEST,
 #endif
+	CXT5047_AUTO,
 	CXT5047_MODELS
 };
 
@@ -1502,9 +1545,10 @@ static const char * const cxt5047_models[CXT5047_MODELS] = {
 #ifdef CONFIG_SND_DEBUG
 	[CXT5047_TEST]		= "test",
 #endif
+	[CXT5047_AUTO]		= "auto",
 };
 
-static struct snd_pci_quirk cxt5047_cfg_tbl[] = {
+static const struct snd_pci_quirk cxt5047_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series",
 			   CXT5047_LAPTOP),
@@ -1517,6 +1561,16 @@ static int patch_cxt5047(struct hda_codec *codec)
 	struct conexant_spec *spec;
 	int board_config;
 
+	board_config = snd_hda_check_board_config(codec, CXT5047_MODELS,
+						  cxt5047_models,
+						  cxt5047_cfg_tbl);
+#if 0 /* not enabled as default, as BIOS often broken for this codec */
+	if (board_config < 0)
+		board_config = CXT5047_AUTO;
+#endif
+	if (board_config == CXT5047_AUTO)
+		return patch_conexant_auto(codec);
+
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (!spec)
 		return -ENOMEM;
@@ -1540,9 +1594,6 @@ static int patch_cxt5047(struct hda_codec *codec)
 
 	codec->patch_ops = conexant_patch_ops;
 
-	board_config = snd_hda_check_board_config(codec, CXT5047_MODELS,
-						  cxt5047_models,
-						  cxt5047_cfg_tbl);
 	switch (board_config) {
 	case CXT5047_LAPTOP:
 		spec->num_mixers = 2;
@@ -1591,10 +1642,10 @@ static int patch_cxt5047(struct hda_codec *codec)
 }
 
 /* Conexant 5051 specific */
-static hda_nid_t cxt5051_dac_nids[1] = { 0x10 };
-static hda_nid_t cxt5051_adc_nids[2] = { 0x14, 0x15 };
+static const hda_nid_t cxt5051_dac_nids[1] = { 0x10 };
+static const hda_nid_t cxt5051_adc_nids[2] = { 0x14, 0x15 };
 
-static struct hda_channel_mode cxt5051_modes[1] = {
+static const struct hda_channel_mode cxt5051_modes[1] = {
 	{ 2, NULL },
 };
 
@@ -1696,7 +1747,7 @@ static void cxt5051_hp_unsol_event(struct hda_codec *codec,
 	snd_hda_input_jack_report(codec, nid);
 }
 
-static struct snd_kcontrol_new cxt5051_playback_mixers[] = {
+static const struct snd_kcontrol_new cxt5051_playback_mixers[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1709,7 +1760,7 @@ static struct snd_kcontrol_new cxt5051_playback_mixers[] = {
 	{}
 };
 
-static struct snd_kcontrol_new cxt5051_capture_mixers[] = {
+static const struct snd_kcontrol_new cxt5051_capture_mixers[] = {
 	HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT),
 	HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT),
@@ -1719,7 +1770,7 @@ static struct snd_kcontrol_new cxt5051_capture_mixers[] = {
 	{}
 };
 
-static struct snd_kcontrol_new cxt5051_hp_mixers[] = {
+static const struct snd_kcontrol_new cxt5051_hp_mixers[] = {
 	HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT),
 	HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Volume", 0x15, 0x00, HDA_INPUT),
@@ -1727,19 +1778,19 @@ static struct snd_kcontrol_new cxt5051_hp_mixers[] = {
 	{}
 };
 
-static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = {
+static const struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x00, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x14, 0x00, HDA_INPUT),
 	{}
 };
 
-static struct snd_kcontrol_new cxt5051_f700_mixers[] = {
+static const struct snd_kcontrol_new cxt5051_f700_mixers[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x14, 0x01, HDA_INPUT),
 	{}
 };
 
-static struct snd_kcontrol_new cxt5051_toshiba_mixers[] = {
+static const struct snd_kcontrol_new cxt5051_toshiba_mixers[] = {
 	HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT),
 	HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT),
@@ -1747,7 +1798,7 @@ static struct snd_kcontrol_new cxt5051_toshiba_mixers[] = {
 	{}
 };
 
-static struct hda_verb cxt5051_init_verbs[] = {
+static const struct hda_verb cxt5051_init_verbs[] = {
 	/* Line in, Mic */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
 	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
@@ -1776,7 +1827,7 @@ static struct hda_verb cxt5051_init_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb cxt5051_hp_dv6736_init_verbs[] = {
+static const struct hda_verb cxt5051_hp_dv6736_init_verbs[] = {
 	/* Line in, Mic */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
 	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
@@ -1801,7 +1852,7 @@ static struct hda_verb cxt5051_hp_dv6736_init_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = {
+static const struct hda_verb cxt5051_lenovo_x200_init_verbs[] = {
 	/* Line in, Mic */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
 	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
@@ -1834,7 +1885,7 @@ static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb cxt5051_f700_init_verbs[] = {
+static const struct hda_verb cxt5051_f700_init_verbs[] = {
 	/* Line in, Mic */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
 	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
@@ -1869,7 +1920,7 @@ static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid,
 	snd_hda_input_jack_report(codec, nid);
 }
 
-static struct hda_verb cxt5051_ideapad_init_verbs[] = {
+static const struct hda_verb cxt5051_ideapad_init_verbs[] = {
 	/* Subwoofer */
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -1906,6 +1957,7 @@ enum {
 	CXT5051_F700,       /* HP Compaq Presario F700 */
 	CXT5051_TOSHIBA,	/* Toshiba M300 & co */
 	CXT5051_IDEAPAD,	/* Lenovo IdeaPad Y430 */
+	CXT5051_AUTO,		/* auto-parser */
 	CXT5051_MODELS
 };
 
@@ -1917,9 +1969,10 @@ static const char *const cxt5051_models[CXT5051_MODELS] = {
 	[CXT5051_F700]          = "hp-700",
 	[CXT5051_TOSHIBA]	= "toshiba",
 	[CXT5051_IDEAPAD]	= "ideapad",
+	[CXT5051_AUTO]		= "auto",
 };
 
-static struct snd_pci_quirk cxt5051_cfg_tbl[] = {
+static const struct snd_pci_quirk cxt5051_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736),
 	SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP),
 	SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700),
@@ -1937,6 +1990,16 @@ static int patch_cxt5051(struct hda_codec *codec)
 	struct conexant_spec *spec;
 	int board_config;
 
+	board_config = snd_hda_check_board_config(codec, CXT5051_MODELS,
+						  cxt5051_models,
+						  cxt5051_cfg_tbl);
+#if 0 /* use the old method just for safety */
+	if (board_config < 0)
+		board_config = CXT5051_AUTO;
+#endif
+	if (board_config == CXT5051_AUTO)
+		return patch_conexant_auto(codec);
+
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (!spec)
 		return -ENOMEM;
@@ -1967,9 +2030,6 @@ static int patch_cxt5051(struct hda_codec *codec)
 
 	codec->patch_ops.unsol_event = cxt5051_hp_unsol_event;
 
-	board_config = snd_hda_check_board_config(codec, CXT5051_MODELS,
-						  cxt5051_models,
-						  cxt5051_cfg_tbl);
 	spec->auto_mic = AUTO_MIC_PORTB | AUTO_MIC_PORTC;
 	switch (board_config) {
 	case CXT5051_HP:
@@ -2011,17 +2071,17 @@ static int patch_cxt5051(struct hda_codec *codec)
 
 /* Conexant 5066 specific */
 
-static hda_nid_t cxt5066_dac_nids[1] = { 0x10 };
-static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
-static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
-static hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 };
+static const hda_nid_t cxt5066_dac_nids[1] = { 0x10 };
+static const hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
+static const hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
+static const hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 };
 
 /* OLPC's microphone port is DC coupled for use with external sensors,
  * therefore we use a 50% mic bias in order to center the input signal with
  * the DC input range of the codec. */
 #define CXT5066_OLPC_EXT_MIC_BIAS PIN_VREF50
 
-static struct hda_channel_mode cxt5066_modes[1] = {
+static const struct hda_channel_mode cxt5066_modes[1] = {
 	{ 2, NULL },
 };
 
@@ -2176,7 +2236,7 @@ static void cxt5066_vostro_automic(struct hda_codec *codec)
 		{0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
 		{}
 	};
-	static struct hda_verb ext_mic_absent[] = {
+	static const struct hda_verb ext_mic_absent[] = {
 		/* enable internal mic, port C */
 		{0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 
@@ -2209,7 +2269,7 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec)
 		{0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
 		{}
 	};
-	static struct hda_verb ext_mic_absent[] = {
+	static const struct hda_verb ext_mic_absent[] = {
 		{0x14, AC_VERB_SET_CONNECT_SEL, 2},
 		{0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 		{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
@@ -2257,7 +2317,7 @@ static void cxt5066_thinkpad_automic(struct hda_codec *codec)
 {
 	unsigned int ext_present, dock_present;
 
-	static struct hda_verb ext_mic_present[] = {
+	static const struct hda_verb ext_mic_present[] = {
 		{0x14, AC_VERB_SET_CONNECT_SEL, 0},
 		{0x17, AC_VERB_SET_CONNECT_SEL, 1},
 		{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
@@ -2265,7 +2325,7 @@ static void cxt5066_thinkpad_automic(struct hda_codec *codec)
 		{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
 		{}
 	};
-	static struct hda_verb dock_mic_present[] = {
+	static const struct hda_verb dock_mic_present[] = {
 		{0x14, AC_VERB_SET_CONNECT_SEL, 0},
 		{0x17, AC_VERB_SET_CONNECT_SEL, 0},
 		{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
@@ -2273,7 +2333,7 @@ static void cxt5066_thinkpad_automic(struct hda_codec *codec)
 		{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
 		{}
 	};
-	static struct hda_verb ext_mic_absent[] = {
+	static const struct hda_verb ext_mic_absent[] = {
 		{0x14, AC_VERB_SET_CONNECT_SEL, 2},
 		{0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 		{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
@@ -2537,7 +2597,7 @@ static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec)
 }
 
 static void conexant_check_dig_outs(struct hda_codec *codec,
-				    hda_nid_t *dig_pins,
+				    const hda_nid_t *dig_pins,
 				    int num_pins)
 {
 	struct conexant_spec *spec = codec->spec;
@@ -2557,7 +2617,7 @@ static void conexant_check_dig_outs(struct hda_codec *codec,
 	}
 }
 
-static struct hda_input_mux cxt5066_capture_source = {
+static const struct hda_input_mux cxt5066_capture_source = {
 	.num_items = 4,
 	.items = {
 		{ "Mic B", 0 },
@@ -2567,7 +2627,7 @@ static struct hda_input_mux cxt5066_capture_source = {
 	},
 };
 
-static struct hda_bind_ctls cxt5066_bind_capture_vol_others = {
+static const struct hda_bind_ctls cxt5066_bind_capture_vol_others = {
 	.ops = &snd_hda_bind_vol,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT),
@@ -2576,7 +2636,7 @@ static struct hda_bind_ctls cxt5066_bind_capture_vol_others = {
 	},
 };
 
-static struct hda_bind_ctls cxt5066_bind_capture_sw_others = {
+static const struct hda_bind_ctls cxt5066_bind_capture_sw_others = {
 	.ops = &snd_hda_bind_sw,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT),
@@ -2585,12 +2645,12 @@ static struct hda_bind_ctls cxt5066_bind_capture_sw_others = {
 	},
 };
 
-static struct snd_kcontrol_new cxt5066_mixer_master[] = {
+static const struct snd_kcontrol_new cxt5066_mixer_master[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT),
 	{}
 };
 
-static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = {
+static const struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Master Playback Volume",
@@ -2609,7 +2669,7 @@ static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = {
 	{}
 };
 
-static struct snd_kcontrol_new cxt5066_mixer_olpc_dc[] = {
+static const struct snd_kcontrol_new cxt5066_mixer_olpc_dc[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "DC Mode Enable Switch",
@@ -2627,7 +2687,7 @@ static struct snd_kcontrol_new cxt5066_mixer_olpc_dc[] = {
 	{}
 };
 
-static struct snd_kcontrol_new cxt5066_mixers[] = {
+static const struct snd_kcontrol_new cxt5066_mixers[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Master Playback Switch",
@@ -2650,7 +2710,7 @@ static struct snd_kcontrol_new cxt5066_mixers[] = {
 	{}
 };
 
-static struct snd_kcontrol_new cxt5066_vostro_mixers[] = {
+static const struct snd_kcontrol_new cxt5066_vostro_mixers[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Internal Mic Boost Capture Enum",
@@ -2662,7 +2722,7 @@ static struct snd_kcontrol_new cxt5066_vostro_mixers[] = {
 	{}
 };
 
-static struct hda_verb cxt5066_init_verbs[] = {
+static const struct hda_verb cxt5066_init_verbs[] = {
 	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */
 	{0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */
@@ -2717,7 +2777,7 @@ static struct hda_verb cxt5066_init_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb cxt5066_init_verbs_olpc[] = {
+static const struct hda_verb cxt5066_init_verbs_olpc[] = {
 	/* Port A: headphones */
 	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
@@ -2778,7 +2838,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = {
 	{ } /* end */
 };
 
-static struct hda_verb cxt5066_init_verbs_vostro[] = {
+static const struct hda_verb cxt5066_init_verbs_vostro[] = {
 	/* Port A: headphones */
 	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
 	{0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
@@ -2839,7 +2899,7 @@ static struct hda_verb cxt5066_init_verbs_vostro[] = {
 	{ } /* end */
 };
 
-static struct hda_verb cxt5066_init_verbs_ideapad[] = {
+static const struct hda_verb cxt5066_init_verbs_ideapad[] = {
 	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */
 	{0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */
@@ -2889,7 +2949,7 @@ static struct hda_verb cxt5066_init_verbs_ideapad[] = {
 	{ } /* end */
 };
 
-static struct hda_verb cxt5066_init_verbs_thinkpad[] = {
+static const struct hda_verb cxt5066_init_verbs_thinkpad[] = {
 	{0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */
 	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */
 
@@ -2947,13 +3007,13 @@ static struct hda_verb cxt5066_init_verbs_thinkpad[] = {
 	{ } /* end */
 };
 
-static struct hda_verb cxt5066_init_verbs_portd_lo[] = {
+static const struct hda_verb cxt5066_init_verbs_portd_lo[] = {
 	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{ } /* end */
 };
 
 
-static struct hda_verb cxt5066_init_verbs_hp_laptop[] = {
+static const struct hda_verb cxt5066_init_verbs_hp_laptop[] = {
 	{0x14, AC_VERB_SET_CONNECT_SEL, 0x0},
 	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
 	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
@@ -2997,6 +3057,7 @@ enum {
 	CXT5066_THINKPAD,	/* Lenovo ThinkPad T410s, others? */
 	CXT5066_ASUS,		/* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */
 	CXT5066_HP_LAPTOP,      /* HP Laptop */
+	CXT5066_AUTO,		/* BIOS auto-parser */
 	CXT5066_MODELS
 };
 
@@ -3009,9 +3070,10 @@ static const char * const cxt5066_models[CXT5066_MODELS] = {
 	[CXT5066_THINKPAD]	= "thinkpad",
 	[CXT5066_ASUS]		= "asus",
 	[CXT5066_HP_LAPTOP]	= "hp-laptop",
+	[CXT5066_AUTO]		= "auto",
 };
 
-static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
+static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD),
@@ -3036,7 +3098,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
  	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
+	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
+	SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO),
 	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
 	{}
 };
@@ -3046,6 +3110,15 @@ static int patch_cxt5066(struct hda_codec *codec)
 	struct conexant_spec *spec;
 	int board_config;
 
+	board_config = snd_hda_check_board_config(codec, CXT5066_MODELS,
+						  cxt5066_models, cxt5066_cfg_tbl);
+#if 0 /* use the old method just for safety */
+	if (board_config < 0)
+		board_config = CXT5066_AUTO;
+#endif
+	if (board_config == CXT5066_AUTO)
+		return patch_conexant_auto(codec);
+
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (!spec)
 		return -ENOMEM;
@@ -3076,8 +3149,6 @@ static int patch_cxt5066(struct hda_codec *codec)
 
 	set_beep_amp(spec, 0x13, 0, HDA_OUTPUT);
 
-	board_config = snd_hda_check_board_config(codec, CXT5066_MODELS,
-						  cxt5066_models, cxt5066_cfg_tbl);
 	switch (board_config) {
 	default:
 	case CXT5066_LAPTOP:
@@ -3195,7 +3266,45 @@ static int patch_cxt5066(struct hda_codec *codec)
  * Automatic parser for CX20641 & co
  */
 
-static hda_nid_t cx_auto_adc_nids[] = { 0x14 };
+static int cx_auto_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+				       struct hda_codec *codec,
+				       unsigned int stream_tag,
+				       unsigned int format,
+				       struct snd_pcm_substream *substream)
+{
+	struct conexant_spec *spec = codec->spec;
+	hda_nid_t adc = spec->imux_info[spec->cur_mux[0]].adc;
+	if (spec->adc_switching) {
+		spec->cur_adc = adc;
+		spec->cur_adc_stream_tag = stream_tag;
+		spec->cur_adc_format = format;
+	}
+	snd_hda_codec_setup_stream(codec, adc, stream_tag, 0, format);
+	return 0;
+}
+
+static int cx_auto_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				       struct hda_codec *codec,
+				       struct snd_pcm_substream *substream)
+{
+	struct conexant_spec *spec = codec->spec;
+	snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+	spec->cur_adc = 0;
+	return 0;
+}
+
+static const struct hda_pcm_stream cx_auto_pcm_analog_capture = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.nid = 0, /* fill later */
+	.ops = {
+		.prepare = cx_auto_capture_pcm_prepare,
+		.cleanup = cx_auto_capture_pcm_cleanup
+	},
+};
+
+static const hda_nid_t cx_auto_adc_nids[] = { 0x14 };
 
 /* get the connection index of @nid in the widget @mux */
 static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
@@ -3320,61 +3429,349 @@ static void cx_auto_parse_output(struct hda_codec *codec)
 	spec->multiout.dac_nids = spec->private_dac_nids;
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (cfg->hp_outs > 0)
-		spec->auto_mute = 1;
+	for (i = 0; i < cfg->hp_outs; i++) {
+		if (is_jack_detectable(codec, cfg->hp_pins[i])) {
+			spec->auto_mute = 1;
+			break;
+		}
+	}
+	if (spec->auto_mute &&
+	    cfg->line_out_pins[0] &&
+	    cfg->line_out_type != AUTO_PIN_SPEAKER_OUT &&
+	    cfg->line_out_pins[0] != cfg->hp_pins[0] &&
+	    cfg->line_out_pins[0] != cfg->speaker_pins[0]) {
+		for (i = 0; i < cfg->line_outs; i++) {
+			if (is_jack_detectable(codec, cfg->line_out_pins[i])) {
+				spec->detect_line = 1;
+				break;
+			}
+		}
+		spec->automute_lines = spec->detect_line;
+	}
+
 	spec->vmaster_nid = spec->private_dac_nids[0];
 }
 
+static void cx_auto_turn_eapd(struct hda_codec *codec, int num_pins,
+			      hda_nid_t *pins, bool on);
+
+static void do_automute(struct hda_codec *codec, int num_pins,
+			hda_nid_t *pins, bool on)
+{
+	int i;
+	for (i = 0; i < num_pins; i++)
+		snd_hda_codec_write(codec, pins[i], 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL,
+				    on ? PIN_OUT : 0);
+	cx_auto_turn_eapd(codec, num_pins, pins, on);
+}
+
+static int detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
+{
+	int i, present = 0;
+
+	for (i = 0; i < num_pins; i++) {
+		hda_nid_t nid = pins[i];
+		if (!nid || !is_jack_detectable(codec, nid))
+			break;
+		snd_hda_input_jack_report(codec, nid);
+		present |= snd_hda_jack_detect(codec, nid);
+	}
+	return present;
+}
+
 /* auto-mute/unmute speaker and line outs according to headphone jack */
+static void cx_auto_update_speakers(struct hda_codec *codec)
+{
+	struct conexant_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int on = 1;
+
+	/* turn on HP EAPD when HP jacks are present */
+	if (spec->auto_mute)
+		on = spec->hp_present;
+	cx_auto_turn_eapd(codec, cfg->hp_outs, cfg->hp_pins, on);
+	/* mute speakers in auto-mode if HP or LO jacks are plugged */
+	if (spec->auto_mute)
+		on = !(spec->hp_present ||
+		       (spec->detect_line && spec->line_present));
+	do_automute(codec, cfg->speaker_outs, cfg->speaker_pins, on);
+
+	/* toggle line-out mutes if needed, too */
+	/* if LO is a copy of either HP or Speaker, don't need to handle it */
+	if (cfg->line_out_pins[0] == cfg->hp_pins[0] ||
+	    cfg->line_out_pins[0] == cfg->speaker_pins[0])
+		return;
+	if (spec->auto_mute) {
+		/* mute LO in auto-mode when HP jack is present */
+		if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT ||
+		    spec->automute_lines)
+			on = !spec->hp_present;
+		else
+			on = 1;
+	}
+	do_automute(codec, cfg->line_outs, cfg->line_out_pins, on);
+}
+
 static void cx_auto_hp_automute(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
-	int i, present;
 
 	if (!spec->auto_mute)
 		return;
-	present = 0;
-	for (i = 0; i < cfg->hp_outs; i++) {
-		if (snd_hda_jack_detect(codec, cfg->hp_pins[i])) {
-			present = 1;
-			break;
-		}
+	spec->hp_present = detect_jacks(codec, cfg->hp_outs, cfg->hp_pins);
+	cx_auto_update_speakers(codec);
+}
+
+static void cx_auto_line_automute(struct hda_codec *codec)
+{
+	struct conexant_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+
+	if (!spec->auto_mute || !spec->detect_line)
+		return;
+	spec->line_present = detect_jacks(codec, cfg->line_outs,
+					  cfg->line_out_pins);
+	cx_auto_update_speakers(codec);
+}
+
+static int cx_automute_mode_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct conexant_spec *spec = codec->spec;
+	static const char * const texts2[] = {
+		"Disabled", "Enabled"
+	};
+	static const char * const texts3[] = {
+		"Disabled", "Speaker Only", "Line-Out+Speaker"
+	};
+	const char * const *texts;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	if (spec->automute_hp_lo) {
+		uinfo->value.enumerated.items = 3;
+		texts = texts3;
+	} else {
+		uinfo->value.enumerated.items = 2;
+		texts = texts2;
 	}
-	for (i = 0; i < cfg->line_outs; i++) {
-		snd_hda_codec_write(codec, cfg->line_out_pins[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL,
-				    present ? 0 : PIN_OUT);
+	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+	strcpy(uinfo->value.enumerated.name,
+	       texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+
+static int cx_automute_mode_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct conexant_spec *spec = codec->spec;
+	unsigned int val;
+	if (!spec->auto_mute)
+		val = 0;
+	else if (!spec->automute_lines)
+		val = 1;
+	else
+		val = 2;
+	ucontrol->value.enumerated.item[0] = val;
+	return 0;
+}
+
+static int cx_automute_mode_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct conexant_spec *spec = codec->spec;
+
+	switch (ucontrol->value.enumerated.item[0]) {
+	case 0:
+		if (!spec->auto_mute)
+			return 0;
+		spec->auto_mute = 0;
+		break;
+	case 1:
+		if (spec->auto_mute && !spec->automute_lines)
+			return 0;
+		spec->auto_mute = 1;
+		spec->automute_lines = 0;
+		break;
+	case 2:
+		if (!spec->automute_hp_lo)
+			return -EINVAL;
+		if (spec->auto_mute && spec->automute_lines)
+			return 0;
+		spec->auto_mute = 1;
+		spec->automute_lines = 1;
+		break;
+	default:
+		return -EINVAL;
 	}
-	for (i = 0; !present && i < cfg->line_outs; i++)
-		if (snd_hda_jack_detect(codec, cfg->line_out_pins[i]))
-			present = 1;
-	for (i = 0; i < cfg->speaker_outs; i++) {
-		snd_hda_codec_write(codec, cfg->speaker_pins[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL,
-				    present ? 0 : PIN_OUT);
+	cx_auto_update_speakers(codec);
+	return 1;
+}
+
+static const struct snd_kcontrol_new cx_automute_mode_enum[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Auto-Mute Mode",
+		.info = cx_automute_mode_info,
+		.get = cx_automute_mode_get,
+		.put = cx_automute_mode_put,
+	},
+	{ }
+};
+
+static int cx_auto_mux_enum_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct conexant_spec *spec = codec->spec;
+
+	return snd_hda_input_mux_info(&spec->private_imux, uinfo);
+}
+
+static int cx_auto_mux_enum_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct conexant_spec *spec = codec->spec;
+
+	ucontrol->value.enumerated.item[0] = spec->cur_mux[0];
+	return 0;
+}
+
+/* look for the route the given pin from mux and return the index;
+ * if do_select is set, actually select the route.
+ */
+static int __select_input_connection(struct hda_codec *codec, hda_nid_t mux,
+				     hda_nid_t pin, hda_nid_t *srcp,
+				     bool do_select, int depth)
+{
+	hda_nid_t conn[HDA_MAX_NUM_INPUTS];
+	int i, nums;
+
+	switch (get_wcaps_type(get_wcaps(codec, mux))) {
+	case AC_WID_AUD_IN:
+	case AC_WID_AUD_SEL:
+	case AC_WID_AUD_MIX:
+		break;
+	default:
+		return -1;
+	}
+
+	nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
+	for (i = 0; i < nums; i++)
+		if (conn[i] == pin) {
+			if (do_select)
+				snd_hda_codec_write(codec, mux, 0,
+						    AC_VERB_SET_CONNECT_SEL, i);
+			if (srcp)
+				*srcp = mux;
+			return i;
+		}
+	depth++;
+	if (depth == 2)
+		return -1;
+	for (i = 0; i < nums; i++) {
+		int ret  = __select_input_connection(codec, conn[i], pin, srcp,
+						     do_select, depth);
+		if (ret >= 0) {
+			if (do_select)
+				snd_hda_codec_write(codec, mux, 0,
+						    AC_VERB_SET_CONNECT_SEL, i);
+			return i;
+		}
 	}
+	return -1;
+}
+
+static void select_input_connection(struct hda_codec *codec, hda_nid_t mux,
+				   hda_nid_t pin)
+{
+	__select_input_connection(codec, mux, pin, NULL, true, 0);
+}
+
+static int get_input_connection(struct hda_codec *codec, hda_nid_t mux,
+				hda_nid_t pin)
+{
+	return __select_input_connection(codec, mux, pin, NULL, false, 0);
+}
+
+static int cx_auto_mux_enum_update(struct hda_codec *codec,
+				   const struct hda_input_mux *imux,
+				   unsigned int idx)
+{
+	struct conexant_spec *spec = codec->spec;
+	hda_nid_t adc;
+	int changed = 1;
+
+	if (!imux->num_items)
+		return 0;
+	if (idx >= imux->num_items)
+		idx = imux->num_items - 1;
+	if (spec->cur_mux[0] == idx)
+		changed = 0;
+	adc = spec->imux_info[idx].adc;
+	select_input_connection(codec, spec->imux_info[idx].adc,
+				spec->imux_info[idx].pin);
+	if (spec->cur_adc && spec->cur_adc != adc) {
+		/* stream is running, let's swap the current ADC */
+		__snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1);
+		spec->cur_adc = adc;
+		snd_hda_codec_setup_stream(codec, adc,
+					   spec->cur_adc_stream_tag, 0,
+					   spec->cur_adc_format);
+	}
+	spec->cur_mux[0] = idx;
+	return changed;
+}
+
+static int cx_auto_mux_enum_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct conexant_spec *spec = codec->spec;
+
+	return cx_auto_mux_enum_update(codec, &spec->private_imux,
+				       ucontrol->value.enumerated.item[0]);
+}
+
+static const struct snd_kcontrol_new cx_auto_capture_mixers[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Capture Source",
+		.info = cx_auto_mux_enum_info,
+		.get = cx_auto_mux_enum_get,
+		.put = cx_auto_mux_enum_put
+	},
+	{}
+};
+
+static bool select_automic(struct hda_codec *codec, int idx, bool detect)
+{
+	struct conexant_spec *spec = codec->spec;
+	if (idx < 0)
+		return false;
+	if (detect && !snd_hda_jack_detect(codec, spec->imux_info[idx].pin))
+		return false;
+	cx_auto_mux_enum_update(codec, &spec->private_imux, idx);
+	return true;
 }
 
 /* automatic switch internal and external mic */
 static void cx_auto_automic(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
-	struct auto_pin_cfg *cfg = &spec->autocfg;
-	struct hda_input_mux *imux = &spec->private_imux;
-	int ext_idx = spec->auto_mic_ext;
 
 	if (!spec->auto_mic)
 		return;
-	if (snd_hda_jack_detect(codec, cfg->inputs[ext_idx].pin)) {
-		snd_hda_codec_write(codec, spec->adc_nids[0], 0,
-				    AC_VERB_SET_CONNECT_SEL,
-				    imux->items[ext_idx].index);
-	} else {
-		snd_hda_codec_write(codec, spec->adc_nids[0], 0,
-				    AC_VERB_SET_CONNECT_SEL,
-				    imux->items[!ext_idx].index);
-	}
+	if (!select_automic(codec, spec->auto_mic_ext, true))
+		if (!select_automic(codec, spec->auto_mic_dock, true))
+			select_automic(codec, spec->auto_mic_int, false);
 }
 
 static void cx_auto_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -3383,7 +3780,9 @@ static void cx_auto_unsol_event(struct hda_codec *codec, unsigned int res)
 	switch (res >> 26) {
 	case CONEXANT_HP_EVENT:
 		cx_auto_hp_automute(codec);
-		snd_hda_input_jack_report(codec, nid);
+		break;
+	case CONEXANT_LINE_EVENT:
+		cx_auto_line_automute(codec);
 		break;
 	case CONEXANT_MIC_EVENT:
 		cx_auto_automic(codec);
@@ -3392,43 +3791,45 @@ static void cx_auto_unsol_event(struct hda_codec *codec, unsigned int res)
 	}
 }
 
-/* return true if it's an internal-mic pin */
-static int is_int_mic(struct hda_codec *codec, hda_nid_t pin)
-{
-	unsigned int def_conf = snd_hda_codec_get_pincfg(codec, pin);
-	return get_defcfg_device(def_conf) == AC_JACK_MIC_IN &&
-		snd_hda_get_input_pin_attr(def_conf) == INPUT_PIN_ATTR_INT;
-}
-
-/* return true if it's an external-mic pin */
-static int is_ext_mic(struct hda_codec *codec, hda_nid_t pin)
-{
-	unsigned int def_conf = snd_hda_codec_get_pincfg(codec, pin);
-	return get_defcfg_device(def_conf) == AC_JACK_MIC_IN &&
-		snd_hda_get_input_pin_attr(def_conf) >= INPUT_PIN_ATTR_NORMAL &&
-		(snd_hda_query_pin_caps(codec, pin) & AC_PINCAP_PRES_DETECT);
-}
-
 /* check whether the pin config is suitable for auto-mic switching;
- * auto-mic is enabled only when one int-mic and one-ext mic exist
+ * auto-mic is enabled only when one int-mic and one ext- and/or
+ * one dock-mic exist
  */
 static void cx_auto_check_auto_mic(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
-	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int pset[INPUT_PIN_ATTR_NORMAL + 1];
+	int i;
 
-	if (is_ext_mic(codec, cfg->inputs[0].pin) &&
-	    is_int_mic(codec, cfg->inputs[1].pin)) {
-		spec->auto_mic = 1;
-		spec->auto_mic_ext = 1;
-		return;
-	}
-	if (is_int_mic(codec, cfg->inputs[1].pin) &&
-	    is_ext_mic(codec, cfg->inputs[0].pin)) {
-		spec->auto_mic = 1;
-		spec->auto_mic_ext = 0;
-		return;
+	for (i = 0; i < ARRAY_SIZE(pset); i++)
+		pset[i] = -1;
+	for (i = 0; i < spec->private_imux.num_items; i++) {
+		hda_nid_t pin = spec->imux_info[i].pin;
+		unsigned int def_conf = snd_hda_codec_get_pincfg(codec, pin);
+		int type, attr;
+		attr = snd_hda_get_input_pin_attr(def_conf);
+		if (attr == INPUT_PIN_ATTR_UNUSED)
+			return; /* invalid entry */
+		if (attr > INPUT_PIN_ATTR_NORMAL)
+			attr = INPUT_PIN_ATTR_NORMAL;
+		if (attr != INPUT_PIN_ATTR_INT &&
+		    !is_jack_detectable(codec, pin))
+			return; /* non-detectable pin */
+		type = get_defcfg_device(def_conf);
+		if (type != AC_JACK_MIC_IN &&
+		    (attr != INPUT_PIN_ATTR_DOCK || type != AC_JACK_LINE_IN))
+			return; /* no valid input type */
+		if (pset[attr] >= 0)
+			return; /* already occupied */
+		pset[attr] = i;
 	}
+	if (pset[INPUT_PIN_ATTR_INT] < 0 ||
+	    (pset[INPUT_PIN_ATTR_NORMAL] < 0 && pset[INPUT_PIN_ATTR_DOCK]))
+		return; /* no input to switch*/
+	spec->auto_mic = 1;
+	spec->auto_mic_ext = pset[INPUT_PIN_ATTR_NORMAL];
+	spec->auto_mic_dock = pset[INPUT_PIN_ATTR_DOCK];
+	spec->auto_mic_int = pset[INPUT_PIN_ATTR_INT];
 }
 
 static void cx_auto_parse_input(struct hda_codec *codec)
@@ -3436,22 +3837,37 @@ static void cx_auto_parse_input(struct hda_codec *codec)
 	struct conexant_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 	struct hda_input_mux *imux;
-	int i;
+	int i, j;
 
 	imux = &spec->private_imux;
 	for (i = 0; i < cfg->num_inputs; i++) {
-		int idx = get_connection_index(codec, spec->adc_nids[0],
-					       cfg->inputs[i].pin);
-		if (idx >= 0) {
-			const char *label;
-			label = hda_get_autocfg_input_label(codec, cfg, i);
-			snd_hda_add_imux_item(imux, label, idx, NULL);
+		for (j = 0; j < spec->num_adc_nids; j++) {
+			hda_nid_t adc = spec->adc_nids[j];
+			int idx = get_input_connection(codec, adc,
+						       cfg->inputs[i].pin);
+			if (idx >= 0) {
+				const char *label;
+				label = hda_get_autocfg_input_label(codec, cfg, i);
+				spec->imux_info[imux->num_items].index = i;
+				spec->imux_info[imux->num_items].boost = 0;
+				spec->imux_info[imux->num_items].adc = adc;
+				spec->imux_info[imux->num_items].pin =
+					cfg->inputs[i].pin;
+				snd_hda_add_imux_item(imux, label, idx, NULL);
+				break;
+			}
 		}
 	}
-	if (imux->num_items == 2 && cfg->num_inputs == 2)
+	if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items)
 		cx_auto_check_auto_mic(codec);
-	if (imux->num_items > 1 && !spec->auto_mic)
-		spec->input_mux = imux;
+	if (imux->num_items > 1 && !spec->auto_mic) {
+		for (i = 1; i < imux->num_items; i++) {
+			if (spec->imux_info[i].adc != spec->imux_info[0].adc) {
+				spec->adc_switching = 1;
+				break;
+			}
+		}
+	}
 }
 
 /* get digital-input audio widget corresponding to the given pin */
@@ -3517,14 +3933,15 @@ static int cx_auto_parse_auto_config(struct hda_codec *codec)
 	return 0;
 }
 
-static void cx_auto_turn_on_eapd(struct hda_codec *codec, int num_pins,
-				 hda_nid_t *pins)
+static void cx_auto_turn_eapd(struct hda_codec *codec, int num_pins,
+			      hda_nid_t *pins, bool on)
 {
 	int i;
 	for (i = 0; i < num_pins; i++) {
 		if (snd_hda_query_pin_caps(codec, pins[i]) & AC_PINCAP_EAPD)
 			snd_hda_codec_write(codec, pins[i], 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 0x02);
+					    AC_VERB_SET_EAPD_BTLENABLE,
+					    on ? 0x02 : 0);
 	}
 }
 
@@ -3537,6 +3954,34 @@ static void select_connection(struct hda_codec *codec, hda_nid_t pin,
 				    AC_VERB_SET_CONNECT_SEL, idx);
 }
 
+static void mute_outputs(struct hda_codec *codec, int num_nids,
+			 const hda_nid_t *nids)
+{
+	int i, val;
+
+	for (i = 0; i < num_nids; i++) {
+		hda_nid_t nid = nids[i];
+		if (!(get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
+			continue;
+		if (query_amp_caps(codec, nid, HDA_OUTPUT) & AC_AMPCAP_MUTE)
+			val = AMP_OUT_MUTE;
+		else
+			val = AMP_OUT_ZERO;
+		snd_hda_codec_write(codec, nid, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, val);
+	}
+}
+
+static void enable_unsol_pins(struct hda_codec *codec, int num_pins,
+			      hda_nid_t *pins, unsigned int tag)
+{
+	int i;
+	for (i = 0; i < num_pins; i++)
+		snd_hda_codec_write(codec, pins[i], 0,
+				    AC_VERB_SET_UNSOLICITED_ENABLE,
+				    AC_USRSP_EN | tag);
+}
+
 static void cx_auto_init_output(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
@@ -3544,51 +3989,53 @@ static void cx_auto_init_output(struct hda_codec *codec)
 	hda_nid_t nid;
 	int i;
 
-	for (i = 0; i < spec->multiout.num_dacs; i++)
-		snd_hda_codec_write(codec, spec->multiout.dac_nids[i], 0,
-				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
-
+	mute_outputs(codec, spec->multiout.num_dacs, spec->multiout.dac_nids);
 	for (i = 0; i < cfg->hp_outs; i++)
 		snd_hda_codec_write(codec, cfg->hp_pins[i], 0,
 				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
-	if (spec->auto_mute) {
-		for (i = 0; i < cfg->hp_outs; i++) {
-			snd_hda_codec_write(codec, cfg->hp_pins[i], 0,
-				    AC_VERB_SET_UNSOLICITED_ENABLE,
-				    AC_USRSP_EN | CONEXANT_HP_EVENT);
-		}
-		cx_auto_hp_automute(codec);
-	} else {
-		for (i = 0; i < cfg->line_outs; i++)
-			snd_hda_codec_write(codec, cfg->line_out_pins[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
-		for (i = 0; i < cfg->speaker_outs; i++)
-			snd_hda_codec_write(codec, cfg->speaker_pins[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
-	}
-
+	mute_outputs(codec, cfg->hp_outs, cfg->hp_pins);
+	mute_outputs(codec, cfg->line_outs, cfg->line_out_pins);
+	mute_outputs(codec, cfg->speaker_outs, cfg->speaker_pins);
 	for (i = 0; i < spec->dac_info_filled; i++) {
 		nid = spec->dac_info[i].dac;
 		if (!nid)
 			nid = spec->multiout.dac_nids[0];
 		select_connection(codec, spec->dac_info[i].pin, nid);
 	}
-
-	/* turn on EAPD */
-	cx_auto_turn_on_eapd(codec, cfg->line_outs, cfg->line_out_pins);
-	cx_auto_turn_on_eapd(codec, cfg->hp_outs, cfg->hp_pins);
-	cx_auto_turn_on_eapd(codec, cfg->speaker_outs, cfg->speaker_pins);
+	if (spec->auto_mute) {
+		enable_unsol_pins(codec, cfg->hp_outs, cfg->hp_pins,
+				  CONEXANT_HP_EVENT);
+		spec->hp_present = detect_jacks(codec, cfg->hp_outs,
+						cfg->hp_pins);
+		if (spec->detect_line) {
+			enable_unsol_pins(codec, cfg->line_outs,
+					  cfg->line_out_pins,
+					  CONEXANT_LINE_EVENT);
+			spec->line_present =
+				detect_jacks(codec, cfg->line_outs,
+					     cfg->line_out_pins);
+		}
+	}
+	cx_auto_update_speakers(codec);
 }
 
 static void cx_auto_init_input(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
-	int i;
+	int i, val;
 
-	for (i = 0; i < spec->num_adc_nids; i++)
-		snd_hda_codec_write(codec, spec->adc_nids[i], 0,
-				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0));
+	for (i = 0; i < spec->num_adc_nids; i++) {
+		hda_nid_t nid = spec->adc_nids[i];
+		if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP))
+			continue;
+		if (query_amp_caps(codec, nid, HDA_INPUT) & AC_AMPCAP_MUTE)
+			val = AMP_IN_MUTE(0);
+		else
+			val = AMP_IN_UNMUTE(0);
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+				    val);
+	}
 
 	for (i = 0; i < cfg->num_inputs; i++) {
 		unsigned int type;
@@ -3601,17 +4048,22 @@ static void cx_auto_init_input(struct hda_codec *codec)
 	}
 
 	if (spec->auto_mic) {
-		int ext_idx = spec->auto_mic_ext;
-		snd_hda_codec_write(codec, cfg->inputs[ext_idx].pin, 0,
-				    AC_VERB_SET_UNSOLICITED_ENABLE,
-				    AC_USRSP_EN | CONEXANT_MIC_EVENT);
+		if (spec->auto_mic_ext >= 0) {
+			snd_hda_codec_write(codec,
+				cfg->inputs[spec->auto_mic_ext].pin, 0,
+				AC_VERB_SET_UNSOLICITED_ENABLE,
+				AC_USRSP_EN | CONEXANT_MIC_EVENT);
+		}
+		if (spec->auto_mic_dock >= 0) {
+			snd_hda_codec_write(codec,
+				cfg->inputs[spec->auto_mic_dock].pin, 0,
+				AC_VERB_SET_UNSOLICITED_ENABLE,
+				AC_USRSP_EN | CONEXANT_MIC_EVENT);
+		}
 		cx_auto_automic(codec);
 	} else {
-		for (i = 0; i < spec->num_adc_nids; i++) {
-			snd_hda_codec_write(codec, spec->adc_nids[i], 0,
-					    AC_VERB_SET_CONNECT_SEL,
-					    spec->private_imux.items[0].index);
-		}
+		select_input_connection(codec, spec->imux_info[0].adc,
+					spec->imux_info[0].pin);
 	}
 }
 
@@ -3646,7 +4098,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
 		HDA_CODEC_VOLUME(name, 0, 0, 0),
 		HDA_CODEC_MUTE(name, 0, 0, 0),
 	};
-	static char *sfx[2] = { "Volume", "Switch" };
+	static const char * const sfx[2] = { "Volume", "Switch" };
 	int i, err;
 
 	for (i = 0; i < 2; i++) {
@@ -3674,6 +4126,19 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
 #define cx_auto_add_pb_volume(codec, nid, str, idx)			\
 	cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT)
 
+static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac,
+			     hda_nid_t pin, const char *name, int idx)
+{
+	unsigned int caps;
+	caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+	if (caps & AC_AMPCAP_NUM_STEPS)
+		return cx_auto_add_pb_volume(codec, dac, name, idx);
+	caps = query_amp_caps(codec, pin, HDA_OUTPUT);
+	if (caps & AC_AMPCAP_NUM_STEPS)
+		return cx_auto_add_pb_volume(codec, pin, name, idx);
+	return 0;
+}
+
 static int cx_auto_build_output_controls(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
@@ -3682,8 +4147,10 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 	static const char * const texts[3] = { "Front", "Surround", "CLFE" };
 
 	if (spec->dac_info_filled == 1)
-		return cx_auto_add_pb_volume(codec, spec->dac_info[0].dac,
-					     "Master", 0);
+		return try_add_pb_volume(codec, spec->dac_info[0].dac,
+					 spec->dac_info[0].pin,
+					 "Master", 0);
+
 	for (i = 0; i < spec->dac_info_filled; i++) {
 		const char *label;
 		int idx, type;
@@ -3707,74 +4174,123 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 			idx = num_spk++;
 			break;
 		}
-		err = cx_auto_add_pb_volume(codec, spec->dac_info[i].dac,
-					    label, idx);
+		err = try_add_pb_volume(codec, spec->dac_info[i].dac,
+					spec->dac_info[i].pin,
+					label, idx);
 		if (err < 0)
 			return err;
 	}
+
+	if (spec->auto_mute) {
+		err = snd_hda_add_new_ctls(codec, cx_automute_mode_enum);
+		if (err < 0)
+			return err;
+	}
+	
+	return 0;
+}
+
+static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid,
+				      const char *label, const char *pfx,
+				      int cidx)
+{
+	struct conexant_spec *spec = codec->spec;
+	int i;
+
+	for (i = 0; i < spec->num_adc_nids; i++) {
+		hda_nid_t adc_nid = spec->adc_nids[i];
+		int idx = get_input_connection(codec, adc_nid, nid);
+		if (idx < 0)
+			continue;
+		return cx_auto_add_volume_idx(codec, label, pfx,
+					      cidx, adc_nid, HDA_INPUT, idx);
+	}
+	return 0;
+}
+
+static int cx_auto_add_boost_volume(struct hda_codec *codec, int idx,
+				    const char *label, int cidx)
+{
+	struct conexant_spec *spec = codec->spec;
+	hda_nid_t mux, nid;
+	int i, con;
+
+	nid = spec->imux_info[idx].pin;
+	if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)
+		return cx_auto_add_volume(codec, label, " Boost", cidx,
+					  nid, HDA_INPUT);
+	con = __select_input_connection(codec, spec->imux_info[idx].adc, nid,
+					&mux, false, 0);
+	if (con < 0)
+		return 0;
+	for (i = 0; i < idx; i++) {
+		if (spec->imux_info[i].boost == mux)
+			return 0; /* already present */
+	}
+
+	if (get_wcaps(codec, mux) & AC_WCAP_OUT_AMP) {
+		spec->imux_info[idx].boost = mux;
+		return cx_auto_add_volume(codec, label, " Boost", 0,
+					  mux, HDA_OUTPUT);
+	}
 	return 0;
 }
 
 static int cx_auto_build_input_controls(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
-	struct auto_pin_cfg *cfg = &spec->autocfg;
-	static const char *prev_label;
-	int i, err, cidx, conn_len;
-	hda_nid_t conn[HDA_MAX_CONNECTIONS];
-
-	int multi_adc_volume = 0; /* If the ADC nid has several input volumes */
-	int adc_nid = spec->adc_nids[0];
-
-	conn_len = snd_hda_get_connections(codec, adc_nid, conn,
-					   HDA_MAX_CONNECTIONS);
-	if (conn_len < 0)
-		return conn_len;
-
-	multi_adc_volume = cfg->num_inputs > 1 && conn_len > 1;
-	if (!multi_adc_volume) {
-		err = cx_auto_add_volume(codec, "Capture", "", 0, adc_nid,
-					 HDA_INPUT);
-		if (err < 0)
-			return err;
+	struct hda_input_mux *imux = &spec->private_imux;
+	const char *prev_label;
+	int input_conn[HDA_MAX_NUM_INPUTS];
+	int i, err, cidx;
+	int multi_connection;
+
+	multi_connection = 0;
+	for (i = 0; i < imux->num_items; i++) {
+		cidx = get_input_connection(codec, spec->imux_info[i].adc,
+					    spec->imux_info[i].pin);
+		input_conn[i] = (spec->imux_info[i].adc << 8) | cidx;
+		if (i > 0 && input_conn[i] != input_conn[0])
+			multi_connection = 1;
 	}
 
 	prev_label = NULL;
 	cidx = 0;
-	for (i = 0; i < cfg->num_inputs; i++) {
-		hda_nid_t nid = cfg->inputs[i].pin;
+	for (i = 0; i < imux->num_items; i++) {
+		hda_nid_t nid = spec->imux_info[i].pin;
 		const char *label;
-		int j;
-		int pin_amp = get_wcaps(codec, nid) & AC_WCAP_IN_AMP;
-		if (!pin_amp && !multi_adc_volume)
-			continue;
 
-		label = hda_get_autocfg_input_label(codec, cfg, i);
+		label = hda_get_autocfg_input_label(codec, &spec->autocfg,
+						    spec->imux_info[i].index);
 		if (label == prev_label)
 			cidx++;
 		else
 			cidx = 0;
 		prev_label = label;
 
-		if (pin_amp) {
-			err = cx_auto_add_volume(codec, label, " Boost", cidx,
-						 nid, HDA_INPUT);
-			if (err < 0)
-				return err;
-		}
+		err = cx_auto_add_boost_volume(codec, i, label, cidx);
+		if (err < 0)
+			return err;
 
-		if (!multi_adc_volume)
-			continue;
-		for (j = 0; j < conn_len; j++) {
-			if (conn[j] == nid) {
-				err = cx_auto_add_volume_idx(codec, label,
-				    " Capture", cidx, adc_nid, HDA_INPUT, j);
-				if (err < 0)
-					return err;
-				break;
-			}
+		if (!multi_connection) {
+			if (i > 0)
+				continue;
+			err = cx_auto_add_capture_volume(codec, nid,
+							 "Capture", "", cidx);
+		} else {
+			err = cx_auto_add_capture_volume(codec, nid,
+							 label, " Capture", cidx);
 		}
+		if (err < 0)
+			return err;
+	}
+
+	if (spec->private_imux.num_items > 1 && !spec->auto_mic) {
+		err = snd_hda_add_new_ctls(codec, cx_auto_capture_mixers);
+		if (err < 0)
+			return err;
 	}
+
 	return 0;
 }
 
@@ -3791,7 +4307,29 @@ static int cx_auto_build_controls(struct hda_codec *codec)
 	return conexant_build_controls(codec);
 }
 
-static struct hda_codec_ops cx_auto_patch_ops = {
+static int cx_auto_search_adcs(struct hda_codec *codec)
+{
+	struct conexant_spec *spec = codec->spec;
+	hda_nid_t nid, end_nid;
+
+	end_nid = codec->start_nid + codec->num_nodes;
+	for (nid = codec->start_nid; nid < end_nid; nid++) {
+		unsigned int caps = get_wcaps(codec, nid);
+		if (get_wcaps_type(caps) != AC_WID_AUD_IN)
+			continue;
+		if (caps & AC_WCAP_DIGITAL)
+			continue;
+		if (snd_BUG_ON(spec->num_adc_nids >=
+			       ARRAY_SIZE(spec->private_adc_nids)))
+			break;
+		spec->private_adc_nids[spec->num_adc_nids++] = nid;
+	}
+	spec->adc_nids = spec->private_adc_nids;
+	return 0;
+}
+
+
+static const struct hda_codec_ops cx_auto_patch_ops = {
 	.build_controls = cx_auto_build_controls,
 	.build_pcms = conexant_build_pcms,
 	.init = cx_auto_init,
@@ -3808,19 +4346,24 @@ static int patch_conexant_auto(struct hda_codec *codec)
 	struct conexant_spec *spec;
 	int err;
 
+	printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+	       codec->chip_name);
+
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
-	spec->adc_nids = cx_auto_adc_nids;
-	spec->num_adc_nids = ARRAY_SIZE(cx_auto_adc_nids);
-	spec->capsrc_nids = spec->adc_nids;
+	codec->pin_amp_workaround = 1;
+	err = cx_auto_search_adcs(codec);
+	if (err < 0)
+		return err;
 	err = cx_auto_parse_auto_config(codec);
 	if (err < 0) {
 		kfree(codec->spec);
 		codec->spec = NULL;
 		return err;
 	}
+	spec->capture_stream = &cx_auto_pcm_analog_capture;
 	codec->patch_ops = cx_auto_patch_ops;
 	if (spec->beep_amp)
 		snd_hda_attach_beep_device(codec, spec->beep_amp);
@@ -3830,7 +4373,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
 /*
  */
 
-static struct hda_codec_preset snd_hda_preset_conexant[] = {
+static const struct hda_codec_preset snd_hda_preset_conexant[] = {
 	{ .id = 0x14f15045, .name = "CX20549 (Venice)",
 	  .patch = patch_cxt5045 },
 	{ .id = 0x14f15047, .name = "CX20551 (Waikiki)",
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 715615a88a8..bd0ae697f9c 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -33,6 +33,7 @@
 #include <linux/slab.h>
 #include <linux/moduleparam.h>
 #include <sound/core.h>
+#include <sound/jack.h>
 #include "hda_codec.h"
 #include "hda_local.h"
 
@@ -47,8 +48,8 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info");
  *
  * The HDA correspondence of pipes/ports are converter/pin nodes.
  */
-#define MAX_HDMI_CVTS	3
-#define MAX_HDMI_PINS	3
+#define MAX_HDMI_CVTS	4
+#define MAX_HDMI_PINS	4
 
 struct hdmi_spec {
 	int num_cvts;
@@ -76,11 +77,7 @@ struct hdmi_spec {
 	 * ati/nvhdmi specific
 	 */
 	struct hda_multi_out multiout;
-	struct hda_pcm_stream *pcm_playback;
-
-	/* misc flags */
-	/* PD bit indicates only the update, not the current state */
-	unsigned int old_pin_detect:1;
+	const struct hda_pcm_stream *pcm_playback;
 };
 
 
@@ -299,13 +296,6 @@ static int hda_node_index(hda_nid_t *nids, hda_nid_t nid)
 	return -EINVAL;
 }
 
-static void hdmi_get_show_eld(struct hda_codec *codec, hda_nid_t pin_nid,
-			      struct hdmi_eld *eld)
-{
-	if (!snd_hdmi_get_eld(eld, codec, pin_nid))
-		snd_hdmi_show_eld(eld);
-}
-
 #ifdef BE_PARANOID
 static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid,
 				int *packet_index, int *byte_index)
@@ -693,33 +683,20 @@ static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid,
 static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 {
 	struct hdmi_spec *spec = codec->spec;
-	int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
-	int pind = !!(res & AC_UNSOL_RES_PD);
+	int pin_nid = res >> AC_UNSOL_RES_TAG_SHIFT;
+	int pd = !!(res & AC_UNSOL_RES_PD);
 	int eldv = !!(res & AC_UNSOL_RES_ELDV);
 	int index;
 
 	printk(KERN_INFO
 		"HDMI hot plug event: Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
-		tag, pind, eldv);
+		pin_nid, pd, eldv);
 
-	index = hda_node_index(spec->pin, tag);
+	index = hda_node_index(spec->pin, pin_nid);
 	if (index < 0)
 		return;
 
-	if (spec->old_pin_detect) {
-		if (pind)
-			hdmi_present_sense(codec, tag, &spec->sink_eld[index]);
-		pind = spec->sink_eld[index].monitor_present;
-	}
-
-	spec->sink_eld[index].monitor_present = pind;
-	spec->sink_eld[index].eld_valid = eldv;
-
-	if (pind && eldv) {
-		hdmi_get_show_eld(codec, spec->pin[index],
-				  &spec->sink_eld[index]);
-		/* TODO: do real things about ELD */
-	}
+	hdmi_present_sense(codec, pin_nid, &spec->sink_eld[index]);
 }
 
 static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res)
@@ -900,18 +877,39 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, hda_nid_t pin_nid)
 static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid,
 			       struct hdmi_eld *eld)
 {
+	/*
+	 * Always execute a GetPinSense verb here, even when called from
+	 * hdmi_intrinsic_event; for some NVIDIA HW, the unsolicited
+	 * response's PD bit is not the real PD value, but indicates that
+	 * the real PD value changed. An older version of the HD-audio
+	 * specification worked this way. Hence, we just ignore the data in
+	 * the unsolicited response to avoid custom WARs.
+	 */
 	int present = snd_hda_pin_sense(codec, pin_nid);
 
+	memset(eld, 0, sizeof(*eld));
+
 	eld->monitor_present	= !!(present & AC_PINSENSE_PRESENCE);
-	eld->eld_valid		= !!(present & AC_PINSENSE_ELDV);
+	if (eld->monitor_present)
+		eld->eld_valid	= !!(present & AC_PINSENSE_ELDV);
+	else
+		eld->eld_valid	= 0;
+
+	printk(KERN_INFO
+		"HDMI status: Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
+		pin_nid, eld->monitor_present, eld->eld_valid);
+
+	if (eld->eld_valid)
+		if (!snd_hdmi_get_eld(eld, codec, pin_nid))
+			snd_hdmi_show_eld(eld);
 
-	if (present & AC_PINSENSE_ELDV)
-		hdmi_get_show_eld(codec, pin_nid, eld);
+	snd_hda_input_jack_report(codec, pin_nid);
 }
 
 static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
 {
 	struct hdmi_spec *spec = codec->spec;
+	int err;
 
 	if (spec->num_pins >= MAX_HDMI_PINS) {
 		snd_printk(KERN_WARNING
@@ -919,6 +917,11 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
 		return -E2BIG;
 	}
 
+	err = snd_hda_input_jack_add(codec, pin_nid,
+				     SND_JACK_VIDEOOUT, NULL);
+	if (err < 0)
+		return err;
+
 	hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]);
 
 	spec->pin[spec->num_pins] = pin_nid;
@@ -1024,6 +1027,7 @@ static char *generic_hdmi_pcm_names[MAX_HDMI_CVTS] = {
 	"HDMI 0",
 	"HDMI 1",
 	"HDMI 2",
+	"HDMI 3",
 };
 
 /*
@@ -1044,7 +1048,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 	return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format);
 }
 
-static struct hda_pcm_stream generic_hdmi_pcm_playback = {
+static const struct hda_pcm_stream generic_hdmi_pcm_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.ops = {
@@ -1120,11 +1124,12 @@ static void generic_hdmi_free(struct hda_codec *codec)
 
 	for (i = 0; i < spec->num_pins; i++)
 		snd_hda_eld_proc_free(codec, &spec->sink_eld[i]);
+	snd_hda_input_jack_free(codec);
 
 	kfree(spec);
 }
 
-static struct hda_codec_ops generic_hdmi_patch_ops = {
+static const struct hda_codec_ops generic_hdmi_patch_ops = {
 	.init			= generic_hdmi_init,
 	.free			= generic_hdmi_free,
 	.build_pcms		= generic_hdmi_build_pcms,
@@ -1169,12 +1174,12 @@ static int patch_generic_hdmi(struct hda_codec *codec)
 #define nvhdmi_master_con_nid_7x	0x04
 #define nvhdmi_master_pin_nid_7x	0x05
 
-static hda_nid_t nvhdmi_con_nids_7x[4] = {
+static const hda_nid_t nvhdmi_con_nids_7x[4] = {
 	/*front, rear, clfe, rear_surr */
 	0x6, 0x8, 0xa, 0xc,
 };
 
-static struct hda_verb nvhdmi_basic_init_7x[] = {
+static const struct hda_verb nvhdmi_basic_init_7x[] = {
 	/* set audio protect on */
 	{ 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1},
 	/* enable digital output on pin widget */
@@ -1435,7 +1440,7 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo,
 	return 0;
 }
 
-static struct hda_pcm_stream nvhdmi_pcm_playback_8ch_7x = {
+static const struct hda_pcm_stream nvhdmi_pcm_playback_8ch_7x = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 8,
@@ -1450,7 +1455,7 @@ static struct hda_pcm_stream nvhdmi_pcm_playback_8ch_7x = {
 	},
 };
 
-static struct hda_pcm_stream nvhdmi_pcm_playback_2ch = {
+static const struct hda_pcm_stream nvhdmi_pcm_playback_2ch = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -1465,32 +1470,20 @@ static struct hda_pcm_stream nvhdmi_pcm_playback_2ch = {
 	},
 };
 
-static struct hda_codec_ops nvhdmi_patch_ops_8ch_7x = {
+static const struct hda_codec_ops nvhdmi_patch_ops_8ch_7x = {
 	.build_controls = generic_hdmi_build_controls,
 	.build_pcms = generic_hdmi_build_pcms,
 	.init = nvhdmi_7x_init,
 	.free = generic_hdmi_free,
 };
 
-static struct hda_codec_ops nvhdmi_patch_ops_2ch = {
+static const struct hda_codec_ops nvhdmi_patch_ops_2ch = {
 	.build_controls = generic_hdmi_build_controls,
 	.build_pcms = generic_hdmi_build_pcms,
 	.init = nvhdmi_7x_init,
 	.free = generic_hdmi_free,
 };
 
-static int patch_nvhdmi_8ch_89(struct hda_codec *codec)
-{
-	struct hdmi_spec *spec;
-	int err = patch_generic_hdmi(codec);
-
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-	spec->old_pin_detect = 1;
-	return 0;
-}
-
 static int patch_nvhdmi_2ch(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec;
@@ -1504,7 +1497,6 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec)
 	spec->multiout.num_dacs = 0;  /* no analog */
 	spec->multiout.max_channels = 2;
 	spec->multiout.dig_out_nid = nvhdmi_master_con_nid_7x;
-	spec->old_pin_detect = 1;
 	spec->num_cvts = 1;
 	spec->cvt[0] = nvhdmi_master_con_nid_7x;
 	spec->pcm_playback = &nvhdmi_pcm_playback_2ch;
@@ -1568,7 +1560,7 @@ static int atihdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 	return 0;
 }
 
-static struct hda_pcm_stream atihdmi_pcm_digital_playback = {
+static const struct hda_pcm_stream atihdmi_pcm_digital_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -1580,7 +1572,7 @@ static struct hda_pcm_stream atihdmi_pcm_digital_playback = {
 	},
 };
 
-static struct hda_verb atihdmi_basic_init[] = {
+static const struct hda_verb atihdmi_basic_init[] = {
 	/* enable digital output on pin widget */
 	{ 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{} /* terminator */
@@ -1599,7 +1591,7 @@ static int atihdmi_init(struct hda_codec *codec)
 	return 0;
 }
 
-static struct hda_codec_ops atihdmi_patch_ops = {
+static const struct hda_codec_ops atihdmi_patch_ops = {
 	.build_controls = generic_hdmi_build_controls,
 	.build_pcms = generic_hdmi_build_pcms,
 	.init = atihdmi_init,
@@ -1634,7 +1626,7 @@ static int patch_atihdmi(struct hda_codec *codec)
 /*
  * patch entries
  */
-static struct hda_codec_preset snd_hda_preset_hdmi[] = {
+static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
 { .id = 0x1002793c, .name = "RS600 HDMI",	.patch = patch_atihdmi },
 { .id = 0x10027919, .name = "RS600 HDMI",	.patch = patch_atihdmi },
 { .id = 0x1002791a, .name = "RS690/780 HDMI",	.patch = patch_atihdmi },
@@ -1647,28 +1639,28 @@ static struct hda_codec_preset snd_hda_preset_hdmi[] = {
 { .id = 0x10de0005, .name = "MCP77/78 HDMI",	.patch = patch_nvhdmi_8ch_7x },
 { .id = 0x10de0006, .name = "MCP77/78 HDMI",	.patch = patch_nvhdmi_8ch_7x },
 { .id = 0x10de0007, .name = "MCP79/7A HDMI",	.patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de000a, .name = "GPU 0a HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de000b, .name = "GPU 0b HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de000c, .name = "MCP89 HDMI",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de000d, .name = "GPU 0d HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de0010, .name = "GPU 10 HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de0011, .name = "GPU 11 HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de0012, .name = "GPU 12 HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de0013, .name = "GPU 13 HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de0014, .name = "GPU 14 HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de0015, .name = "GPU 15 HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
+{ .id = 0x10de000a, .name = "GPU 0a HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de000b, .name = "GPU 0b HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de000c, .name = "MCP89 HDMI",	.patch = patch_generic_hdmi },
+{ .id = 0x10de000d, .name = "GPU 0d HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0010, .name = "GPU 10 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0011, .name = "GPU 11 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0012, .name = "GPU 12 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0013, .name = "GPU 13 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0014, .name = "GPU 14 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0015, .name = "GPU 15 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP",	.patch = patch_generic_hdmi },
 /* 17 is known to be absent */
-{ .id = 0x10de0018, .name = "GPU 18 HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de0019, .name = "GPU 19 HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de001a, .name = "GPU 1a HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de001b, .name = "GPU 1b HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de001c, .name = "GPU 1c HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de0040, .name = "GPU 40 HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de0041, .name = "GPU 41 HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de0042, .name = "GPU 42 HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de0043, .name = "GPU 43 HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
-{ .id = 0x10de0044, .name = "GPU 44 HDMI/DP",	.patch = patch_nvhdmi_8ch_89 },
+{ .id = 0x10de0018, .name = "GPU 18 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0019, .name = "GPU 19 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de001a, .name = "GPU 1a HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de001b, .name = "GPU 1b HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de001c, .name = "GPU 1c HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0040, .name = "GPU 40 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0041, .name = "GPU 41 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0042, .name = "GPU 42 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0043, .name = "GPU 43 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0044, .name = "GPU 44 HDMI/DP",	.patch = patch_generic_hdmi },
 { .id = 0x10de0067, .name = "MCP67 HDMI",	.patch = patch_nvhdmi_2ch },
 { .id = 0x10de8001, .name = "MCP73 HDMI",	.patch = patch_nvhdmi_2ch },
 { .id = 0x80860054, .name = "IbexPeak HDMI",	.patch = patch_generic_hdmi },
@@ -1677,6 +1669,7 @@ static struct hda_codec_preset snd_hda_preset_hdmi[] = {
 { .id = 0x80862803, .name = "Eaglelake HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862804, .name = "IbexPeak HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862805, .name = "CougarPoint HDMI",	.patch = patch_generic_hdmi },
+{ .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi },
 { .id = 0x808629fb, .name = "Crestline HDMI",	.patch = patch_generic_hdmi },
 {} /* terminator */
 };
@@ -1722,6 +1715,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862802");
 MODULE_ALIAS("snd-hda-codec-id:80862803");
 MODULE_ALIAS("snd-hda-codec-id:80862804");
 MODULE_ALIAS("snd-hda-codec-id:80862805");
+MODULE_ALIAS("snd-hda-codec-id:80862806");
 MODULE_ALIAS("snd-hda-codec-id:808629fb");
 
 MODULE_LICENSE("GPL");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 52928d9a72d..7a4e10002f5 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -299,11 +299,23 @@ struct alc_customize_define {
 
 struct alc_fixup;
 
+struct alc_multi_io {
+	hda_nid_t pin;		/* multi-io widget pin NID */
+	hda_nid_t dac;		/* DAC to be connected */
+	unsigned int ctl_in;	/* cached input-pin control value */
+};
+
+enum {
+	ALC_AUTOMUTE_PIN,	/* change the pin control */
+	ALC_AUTOMUTE_AMP,	/* mute/unmute the pin AMP */
+	ALC_AUTOMUTE_MIXER,	/* mute/unmute mixer widget AMP */
+};
+
 struct alc_spec {
 	/* codec parameterization */
-	struct snd_kcontrol_new *mixers[5];	/* mixer arrays */
+	const struct snd_kcontrol_new *mixers[5];	/* mixer arrays */
 	unsigned int num_mixers;
-	struct snd_kcontrol_new *cap_mixer;	/* capture mixer */
+	const struct snd_kcontrol_new *cap_mixer;	/* capture mixer */
 	unsigned int beep_amp;	/* beep amp value, set via set_beep_amp() */
 
 	const struct hda_verb *init_verbs[10];	/* initialization verbs
@@ -313,14 +325,14 @@ struct alc_spec {
 	unsigned int num_init_verbs;
 
 	char stream_name_analog[32];	/* analog PCM stream */
-	struct hda_pcm_stream *stream_analog_playback;
-	struct hda_pcm_stream *stream_analog_capture;
-	struct hda_pcm_stream *stream_analog_alt_playback;
-	struct hda_pcm_stream *stream_analog_alt_capture;
+	const struct hda_pcm_stream *stream_analog_playback;
+	const struct hda_pcm_stream *stream_analog_capture;
+	const struct hda_pcm_stream *stream_analog_alt_playback;
+	const struct hda_pcm_stream *stream_analog_alt_capture;
 
 	char stream_name_digital[32];	/* digital PCM stream */
-	struct hda_pcm_stream *stream_digital_playback;
-	struct hda_pcm_stream *stream_digital_capture;
+	const struct hda_pcm_stream *stream_digital_playback;
+	const struct hda_pcm_stream *stream_digital_capture;
 
 	/* playback */
 	struct hda_multi_out multiout;	/* playback set-up
@@ -333,8 +345,8 @@ struct alc_spec {
 
 	/* capture */
 	unsigned int num_adc_nids;
-	hda_nid_t *adc_nids;
-	hda_nid_t *capsrc_nids;
+	const hda_nid_t *adc_nids;
+	const hda_nid_t *capsrc_nids;
 	hda_nid_t dig_in_nid;		/* digital-in NID; optional */
 
 	/* capture setup for dynamic dual-adc switch */
@@ -348,6 +360,7 @@ struct alc_spec {
 	const struct hda_input_mux *input_mux;
 	unsigned int cur_mux[3];
 	struct alc_mic_route ext_mic;
+	struct alc_mic_route dock_mic;
 	struct alc_mic_route int_mic;
 
 	/* channel model */
@@ -375,17 +388,27 @@ struct alc_spec {
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	void (*power_hook)(struct hda_codec *codec);
 #endif
+	void (*shutup)(struct hda_codec *codec);
 
 	/* for pin sensing */
-	unsigned int sense_updated: 1;
 	unsigned int jack_present: 1;
-	unsigned int master_sw: 1;
+	unsigned int line_jack_present:1;
+	unsigned int master_mute:1;
 	unsigned int auto_mic:1;
+	unsigned int automute:1;	/* HP automute enabled */
+	unsigned int detect_line:1;	/* Line-out detection enabled */
+	unsigned int automute_lines:1;	/* automute line-out as well */
+	unsigned int automute_hp_lo:1;	/* both HP and LO available */
 
 	/* other flags */
 	unsigned int no_analog :1; /* digital I/O only */
 	unsigned int dual_adc_switch:1; /* switch ADCs (for ALC275) */
 	unsigned int single_input_src:1;
+
+	/* auto-mute control */
+	int automute_mode;
+	hda_nid_t automute_mixer_nid[AUTO_CFG_MAX_OUTS];
+
 	int init_amp;
 	int codec_variant;	/* flag for other variants */
 
@@ -403,25 +426,29 @@ struct alc_spec {
 	int fixup_id;
 	const struct alc_fixup *fixup_list;
 	const char *fixup_name;
+
+	/* multi-io */
+	int multi_ios;
+	struct alc_multi_io multi_io[4];
 };
 
 /*
  * configuration template - to be copied to the spec instance
  */
 struct alc_config_preset {
-	struct snd_kcontrol_new *mixers[5]; /* should be identical size
+	const struct snd_kcontrol_new *mixers[5]; /* should be identical size
 					     * with spec
 					     */
-	struct snd_kcontrol_new *cap_mixer; /* capture mixer */
+	const struct snd_kcontrol_new *cap_mixer; /* capture mixer */
 	const struct hda_verb *init_verbs[5];
 	unsigned int num_dacs;
-	hda_nid_t *dac_nids;
+	const hda_nid_t *dac_nids;
 	hda_nid_t dig_out_nid;		/* optional */
 	hda_nid_t hp_nid;		/* optional */
-	hda_nid_t *slave_dig_outs;
+	const hda_nid_t *slave_dig_outs;
 	unsigned int num_adc_nids;
-	hda_nid_t *adc_nids;
-	hda_nid_t *capsrc_nids;
+	const hda_nid_t *adc_nids;
+	const hda_nid_t *capsrc_nids;
 	hda_nid_t dig_in_nid;
 	unsigned int num_channel_mode;
 	const struct hda_channel_mode *channel_mode;
@@ -433,7 +460,7 @@ struct alc_config_preset {
 	void (*setup)(struct hda_codec *);
 	void (*init_hook)(struct hda_codec *);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-	struct hda_amp_list *loopbacks;
+	const struct hda_amp_list *loopbacks;
 	void (*power_hook)(struct hda_codec *codec);
 #endif
 };
@@ -560,11 +587,11 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
  * NIDs 0x0f and 0x10 have been observed to have this behaviour as of
  * March 2006.
  */
-static char *alc_pin_mode_names[] = {
+static const char * const alc_pin_mode_names[] = {
 	"Mic 50pc bias", "Mic 80pc bias",
 	"Line in", "Line out", "Headphone out",
 };
-static unsigned char alc_pin_mode_values[] = {
+static const unsigned char alc_pin_mode_values[] = {
 	PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP,
 };
 /* The control can present all 5 options, or it can limit the options based
@@ -583,7 +610,7 @@ static unsigned char alc_pin_mode_values[] = {
 /* Info about the pin modes supported by the different pin direction modes.
  * For each direction the minimum and maximum values are given.
  */
-static signed char alc_pin_mode_dir_info[5][2] = {
+static const signed char alc_pin_mode_dir_info[5][2] = {
 	{ 0, 2 },    /* ALC_PIN_DIR_IN */
 	{ 3, 4 },    /* ALC_PIN_DIR_OUT */
 	{ 0, 4 },    /* ALC_PIN_DIR_INOUT */
@@ -900,7 +927,7 @@ static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec)
 
 /*
  */
-static void add_mixer(struct alc_spec *spec, struct snd_kcontrol_new *mix)
+static void add_mixer(struct alc_spec *spec, const struct snd_kcontrol_new *mix)
 {
 	if (snd_BUG_ON(spec->num_mixers >= ARRAY_SIZE(spec->mixers)))
 		return;
@@ -971,21 +998,21 @@ static void setup_preset(struct hda_codec *codec,
 }
 
 /* Enable GPIO mask and set output */
-static struct hda_verb alc_gpio1_init_verbs[] = {
+static const struct hda_verb alc_gpio1_init_verbs[] = {
 	{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
 	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
 	{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
 	{ }
 };
 
-static struct hda_verb alc_gpio2_init_verbs[] = {
+static const struct hda_verb alc_gpio2_init_verbs[] = {
 	{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
 	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
 	{0x01, AC_VERB_SET_GPIO_DATA, 0x02},
 	{ }
 };
 
-static struct hda_verb alc_gpio3_init_verbs[] = {
+static const struct hda_verb alc_gpio3_init_verbs[] = {
 	{0x01, AC_VERB_SET_GPIO_MASK, 0x03},
 	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03},
 	{0x01, AC_VERB_SET_GPIO_DATA, 0x03},
@@ -1031,6 +1058,7 @@ static int alc_init_jacks(struct hda_codec *codec)
 	int err;
 	unsigned int hp_nid = spec->autocfg.hp_pins[0];
 	unsigned int mic_nid = spec->ext_mic.pin;
+	unsigned int dock_nid = spec->dock_mic.pin;
 
 	if (hp_nid) {
 		err = snd_hda_input_jack_add(codec, hp_nid,
@@ -1047,46 +1075,116 @@ static int alc_init_jacks(struct hda_codec *codec)
 			return err;
 		snd_hda_input_jack_report(codec, mic_nid);
 	}
+	if (dock_nid) {
+		err = snd_hda_input_jack_add(codec, dock_nid,
+					     SND_JACK_MICROPHONE, NULL);
+		if (err < 0)
+			return err;
+		snd_hda_input_jack_report(codec, dock_nid);
+	}
 #endif /* CONFIG_SND_HDA_INPUT_JACK */
 	return 0;
 }
 
-static void alc_automute_speaker(struct hda_codec *codec, int pinctl)
+static int detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
 {
-	struct alc_spec *spec = codec->spec;
-	unsigned int mute;
-	hda_nid_t nid;
-	int i;
+	int i, present = 0;
 
-	spec->jack_present = 0;
-	for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) {
-		nid = spec->autocfg.hp_pins[i];
+	for (i = 0; i < num_pins; i++) {
+		hda_nid_t nid = pins[i];
 		if (!nid)
 			break;
 		snd_hda_input_jack_report(codec, nid);
-		spec->jack_present |= snd_hda_jack_detect(codec, nid);
+		present |= snd_hda_jack_detect(codec, nid);
 	}
+	return present;
+}
 
-	mute = spec->jack_present ? HDA_AMP_MUTE : 0;
-	/* Toggle internal speakers muting */
-	for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) {
-		nid = spec->autocfg.speaker_pins[i];
+static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
+			bool mute, bool hp_out)
+{
+	struct alc_spec *spec = codec->spec;
+	unsigned int mute_bits = mute ? HDA_AMP_MUTE : 0;
+	unsigned int pin_bits = mute ? 0 : (hp_out ? PIN_HP : PIN_OUT);
+	int i;
+
+	for (i = 0; i < num_pins; i++) {
+		hda_nid_t nid = pins[i];
 		if (!nid)
 			break;
-		if (pinctl) {
+		switch (spec->automute_mode) {
+		case ALC_AUTOMUTE_PIN:
 			snd_hda_codec_write(codec, nid, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL,
-				    spec->jack_present ? 0 : PIN_OUT);
-		} else {
+					    AC_VERB_SET_PIN_WIDGET_CONTROL,
+					    pin_bits);
+			break;
+		case ALC_AUTOMUTE_AMP:
 			snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
-					 HDA_AMP_MUTE, mute);
+						 HDA_AMP_MUTE, mute_bits);
+			break;
+		case ALC_AUTOMUTE_MIXER:
+			nid = spec->automute_mixer_nid[i];
+			if (!nid)
+				break;
+			snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
+						 HDA_AMP_MUTE, mute_bits);
+			snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 1,
+						 HDA_AMP_MUTE, mute_bits);
+			break;
 		}
 	}
 }
 
-static void alc_automute_pin(struct hda_codec *codec)
+/* Toggle internal speakers muting */
+static void update_speakers(struct hda_codec *codec)
 {
-	alc_automute_speaker(codec, 1);
+	struct alc_spec *spec = codec->spec;
+	int on;
+
+	if (!spec->automute)
+		on = 0;
+	else
+		on = spec->jack_present | spec->line_jack_present;
+	on |= spec->master_mute;
+	do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins),
+		    spec->autocfg.speaker_pins, on, false);
+
+	/* toggle line-out mutes if needed, too */
+	/* if LO is a copy of either HP or Speaker, don't need to handle it */
+	if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] ||
+	    spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0])
+		return;
+	if (!spec->automute_lines || !spec->automute)
+		on = 0;
+	else
+		on = spec->jack_present;
+	on |= spec->master_mute;
+	do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
+		    spec->autocfg.line_out_pins, on, false);
+}
+
+static void alc_hp_automute(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (!spec->automute)
+		return;
+	spec->jack_present =
+		detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
+			     spec->autocfg.hp_pins);
+	update_speakers(codec);
+}
+
+static void alc_line_automute(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (!spec->automute || !spec->detect_line)
+		return;
+	spec->line_jack_present =
+		detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
+			     spec->autocfg.line_out_pins);
+	update_speakers(codec);
 }
 
 static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
@@ -1128,7 +1226,7 @@ static void alc_dual_mic_adc_auto_switch(struct hda_codec *codec)
 static void alc_mic_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	struct alc_mic_route *dead, *alive;
+	struct alc_mic_route *dead1, *dead2, *alive;
 	unsigned int present, type;
 	hda_nid_t cap_nid;
 
@@ -1146,13 +1244,24 @@ static void alc_mic_automute(struct hda_codec *codec)
 
 	cap_nid = spec->capsrc_nids ? spec->capsrc_nids[0] : spec->adc_nids[0];
 
+	alive = &spec->int_mic;
+	dead1 = &spec->ext_mic;
+	dead2 = &spec->dock_mic;
+
 	present = snd_hda_jack_detect(codec, spec->ext_mic.pin);
 	if (present) {
 		alive = &spec->ext_mic;
-		dead = &spec->int_mic;
-	} else {
-		alive = &spec->int_mic;
-		dead = &spec->ext_mic;
+		dead1 = &spec->int_mic;
+		dead2 = &spec->dock_mic;
+	}
+	if (!present && spec->dock_mic.pin > 0) {
+		present = snd_hda_jack_detect(codec, spec->dock_mic.pin);
+		if (present) {
+			alive = &spec->dock_mic;
+			dead1 = &spec->int_mic;
+			dead2 = &spec->ext_mic;
+		}
+		snd_hda_input_jack_report(codec, spec->dock_mic.pin);
 	}
 
 	type = get_wcaps_type(get_wcaps(codec, cap_nid));
@@ -1161,9 +1270,14 @@ static void alc_mic_automute(struct hda_codec *codec)
 		snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT,
 					 alive->mux_idx,
 					 HDA_AMP_MUTE, 0);
-		snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT,
-					 dead->mux_idx,
-					 HDA_AMP_MUTE, HDA_AMP_MUTE);
+		if (dead1->pin > 0)
+			snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT,
+						 dead1->mux_idx,
+						 HDA_AMP_MUTE, HDA_AMP_MUTE);
+		if (dead2->pin > 0)
+			snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT,
+						 dead2->mux_idx,
+						 HDA_AMP_MUTE, HDA_AMP_MUTE);
 	} else {
 		/* MUX style (e.g. ALC880) */
 		snd_hda_codec_write_cache(codec, cap_nid, 0,
@@ -1184,7 +1298,10 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
 		res >>= 26;
 	switch (res) {
 	case ALC880_HP_EVENT:
-		alc_automute_pin(codec);
+		alc_hp_automute(codec);
+		break;
+	case ALC880_FRONT_EVENT:
+		alc_line_automute(codec);
 		break;
 	case ALC880_MIC_EVENT:
 		alc_mic_automute(codec);
@@ -1194,7 +1311,8 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
 
 static void alc_inithook(struct hda_codec *codec)
 {
-	alc_automute_pin(codec);
+	alc_hp_automute(codec);
+	alc_line_automute(codec);
 	alc_mic_automute(codec);
 }
 
@@ -1236,6 +1354,43 @@ static void set_eapd(struct hda_codec *codec, hda_nid_t nid, int on)
 				    on ? 2 : 0);
 }
 
+/* turn on/off EAPD controls of the codec */
+static void alc_auto_setup_eapd(struct hda_codec *codec, bool on)
+{
+	/* We currently only handle front, HP */
+	switch (codec->vendor_id) {
+	case 0x10ec0260:
+		set_eapd(codec, 0x0f, on);
+		set_eapd(codec, 0x10, on);
+		break;
+	case 0x10ec0262:
+	case 0x10ec0267:
+	case 0x10ec0268:
+	case 0x10ec0269:
+	case 0x10ec0270:
+	case 0x10ec0272:
+	case 0x10ec0660:
+	case 0x10ec0662:
+	case 0x10ec0663:
+	case 0x10ec0665:
+	case 0x10ec0862:
+	case 0x10ec0889:
+	case 0x10ec0892:
+		set_eapd(codec, 0x14, on);
+		set_eapd(codec, 0x15, on);
+		break;
+	}
+}
+
+/* generic shutup callback;
+ * just turning off EPAD and a little pause for avoiding pop-noise
+ */
+static void alc_eapd_shutup(struct hda_codec *codec)
+{
+	alc_auto_setup_eapd(codec, false);
+	msleep(200);
+}
+
 static void alc_auto_init_amp(struct hda_codec *codec, int type)
 {
 	unsigned int tmp;
@@ -1251,27 +1406,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
 		snd_hda_sequence_write(codec, alc_gpio3_init_verbs);
 		break;
 	case ALC_INIT_DEFAULT:
-		switch (codec->vendor_id) {
-		case 0x10ec0260:
-			set_eapd(codec, 0x0f, 1);
-			set_eapd(codec, 0x10, 1);
-			break;
-		case 0x10ec0262:
-		case 0x10ec0267:
-		case 0x10ec0268:
-		case 0x10ec0269:
-		case 0x10ec0270:
-		case 0x10ec0272:
-		case 0x10ec0660:
-		case 0x10ec0662:
-		case 0x10ec0663:
-		case 0x10ec0665:
-		case 0x10ec0862:
-		case 0x10ec0889:
-			set_eapd(codec, 0x14, 1);
-			set_eapd(codec, 0x15, 1);
-			break;
-		}
+		alc_auto_setup_eapd(codec, true);
 		switch (codec->vendor_id) {
 		case 0x10ec0260:
 			snd_hda_codec_write(codec, 0x1a, 0,
@@ -1315,20 +1450,128 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
 	}
 }
 
+static int alc_automute_mode_info(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_info *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+	static const char * const texts2[] = {
+		"Disabled", "Enabled"
+	};
+	static const char * const texts3[] = {
+		"Disabled", "Speaker Only", "Line-Out+Speaker"
+	};
+	const char * const *texts;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	if (spec->automute_hp_lo) {
+		uinfo->value.enumerated.items = 3;
+		texts = texts3;
+	} else {
+		uinfo->value.enumerated.items = 2;
+		texts = texts2;
+	}
+	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+	strcpy(uinfo->value.enumerated.name,
+	       texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+
+static int alc_automute_mode_get(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+	unsigned int val;
+	if (!spec->automute)
+		val = 0;
+	else if (!spec->automute_lines)
+		val = 1;
+	else
+		val = 2;
+	ucontrol->value.enumerated.item[0] = val;
+	return 0;
+}
+
+static int alc_automute_mode_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+
+	switch (ucontrol->value.enumerated.item[0]) {
+	case 0:
+		if (!spec->automute)
+			return 0;
+		spec->automute = 0;
+		break;
+	case 1:
+		if (spec->automute && !spec->automute_lines)
+			return 0;
+		spec->automute = 1;
+		spec->automute_lines = 0;
+		break;
+	case 2:
+		if (!spec->automute_hp_lo)
+			return -EINVAL;
+		if (spec->automute && spec->automute_lines)
+			return 0;
+		spec->automute = 1;
+		spec->automute_lines = 1;
+		break;
+	default:
+		return -EINVAL;
+	}
+	update_speakers(codec);
+	return 1;
+}
+
+static const struct snd_kcontrol_new alc_automute_mode_enum = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Auto-Mute Mode",
+	.info = alc_automute_mode_info,
+	.get = alc_automute_mode_get,
+	.put = alc_automute_mode_put,
+};
+
+static struct snd_kcontrol_new *alc_kcontrol_new(struct alc_spec *spec);
+
+static int alc_add_automute_mode_enum(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	struct snd_kcontrol_new *knew;
+
+	knew = alc_kcontrol_new(spec);
+	if (!knew)
+		return -ENOMEM;
+	*knew = alc_automute_mode_enum;
+	knew->name = kstrdup("Auto-Mute Mode", GFP_KERNEL);
+	if (!knew->name)
+		return -ENOMEM;
+	return 0;
+}
+
 static void alc_init_auto_hp(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int present = 0;
 	int i;
 
-	if (!cfg->hp_pins[0]) {
-		if (cfg->line_out_type != AUTO_PIN_HP_OUT)
-			return;
-	}
+	if (cfg->hp_pins[0])
+		present++;
+	if (cfg->line_out_pins[0])
+		present++;
+	if (cfg->speaker_pins[0])
+		present++;
+	if (present < 2) /* need two different output types */
+		return;
+	if (present == 3)
+		spec->automute_hp_lo = 1; /* both HP and LO automute */
 
 	if (!cfg->speaker_pins[0]) {
-		if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT)
-			return;
 		memcpy(cfg->speaker_pins, cfg->line_out_pins,
 		       sizeof(cfg->speaker_pins));
 		cfg->speaker_outs = cfg->line_outs;
@@ -1341,28 +1584,49 @@ static void alc_init_auto_hp(struct hda_codec *codec)
 	}
 
 	for (i = 0; i < cfg->hp_outs; i++) {
+		hda_nid_t nid = cfg->hp_pins[i];
+		if (!is_jack_detectable(codec, nid))
+			continue;
 		snd_printdd("realtek: Enable HP auto-muting on NID 0x%x\n",
-			    cfg->hp_pins[i]);
-		snd_hda_codec_write_cache(codec, cfg->hp_pins[i], 0,
+			    nid);
+		snd_hda_codec_write_cache(codec, nid, 0,
 				  AC_VERB_SET_UNSOLICITED_ENABLE,
 				  AC_USRSP_EN | ALC880_HP_EVENT);
+		spec->automute = 1;
+		spec->automute_mode = ALC_AUTOMUTE_PIN;
+	}
+	if (spec->automute && cfg->line_out_pins[0] &&
+	    cfg->line_out_pins[0] != cfg->hp_pins[0] &&
+	    cfg->line_out_pins[0] != cfg->speaker_pins[0]) {
+		for (i = 0; i < cfg->line_outs; i++) {
+			hda_nid_t nid = cfg->line_out_pins[i];
+			if (!is_jack_detectable(codec, nid))
+				continue;
+			snd_printdd("realtek: Enable Line-Out auto-muting "
+				    "on NID 0x%x\n", nid);
+			snd_hda_codec_write_cache(codec, nid, 0,
+					AC_VERB_SET_UNSOLICITED_ENABLE,
+					AC_USRSP_EN | ALC880_FRONT_EVENT);
+			spec->detect_line = 1;
+		}
+		spec->automute_lines = spec->detect_line;
+	}
+
+	if (spec->automute) {
+		/* create a control for automute mode */
+		alc_add_automute_mode_enum(codec);
+		spec->unsol_event = alc_sku_unsol_event;
 	}
-	spec->unsol_event = alc_sku_unsol_event;
 }
 
 static void alc_init_auto_mic(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
-	hda_nid_t fixed, ext;
+	hda_nid_t fixed, ext, dock;
 	int i;
 
-	/* there must be only two mic inputs exclusively */
-	for (i = 0; i < cfg->num_inputs; i++)
-		if (cfg->inputs[i].type >= AUTO_PIN_LINE_IN)
-			return;
-
-	fixed = ext = 0;
+	fixed = ext = dock = 0;
 	for (i = 0; i < cfg->num_inputs; i++) {
 		hda_nid_t nid = cfg->inputs[i].pin;
 		unsigned int defcfg;
@@ -1371,26 +1635,45 @@ static void alc_init_auto_mic(struct hda_codec *codec)
 		case INPUT_PIN_ATTR_INT:
 			if (fixed)
 				return; /* already occupied */
+			if (cfg->inputs[i].type != AUTO_PIN_MIC)
+				return; /* invalid type */
 			fixed = nid;
 			break;
 		case INPUT_PIN_ATTR_UNUSED:
 			return; /* invalid entry */
+		case INPUT_PIN_ATTR_DOCK:
+			if (dock)
+				return; /* already occupied */
+			if (cfg->inputs[i].type > AUTO_PIN_LINE_IN)
+				return; /* invalid type */
+			dock = nid;
+			break;
 		default:
 			if (ext)
 				return; /* already occupied */
+			if (cfg->inputs[i].type != AUTO_PIN_MIC)
+				return; /* invalid type */
 			ext = nid;
 			break;
 		}
 	}
+	if (!ext && dock) {
+		ext = dock;
+		dock = 0;
+	}
 	if (!ext || !fixed)
 		return;
-	if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP))
+	if (!is_jack_detectable(codec, ext))
 		return; /* no unsol support */
-	snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n",
-		    ext, fixed);
+	if (dock && !is_jack_detectable(codec, dock))
+		return; /* no unsol support */
+	snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x/0x%x\n",
+		    ext, fixed, dock);
 	spec->ext_mic.pin = ext;
+	spec->dock_mic.pin = dock;
 	spec->int_mic.pin = fixed;
 	spec->ext_mic.mux_idx = MUX_IDX_UNDEF; /* set later */
+	spec->dock_mic.mux_idx = MUX_IDX_UNDEF; /* set later */
 	spec->int_mic.mux_idx = MUX_IDX_UNDEF; /* set later */
 	spec->auto_mic = 1;
 	snd_hda_codec_write_cache(codec, spec->ext_mic.pin, 0,
@@ -1583,9 +1866,6 @@ do_sku:
 				return 1;
 		spec->autocfg.hp_pins[0] = nid;
 	}
-
-	alc_init_auto_hp(codec);
-	alc_init_auto_mic(codec);
 	return 1;
 }
 
@@ -1598,9 +1878,10 @@ static void alc_ssid_check(struct hda_codec *codec,
 		snd_printd("realtek: "
 			   "Enable default setup for auto mode as fallback\n");
 		spec->init_amp = ALC_INIT_DEFAULT;
-		alc_init_auto_hp(codec);
-		alc_init_auto_mic(codec);
 	}
+
+	alc_init_auto_hp(codec);
+	alc_init_auto_mic(codec);
 }
 
 /*
@@ -1704,11 +1985,11 @@ static void alc_apply_fixup(struct hda_codec *codec, int action)
 				   codec->chip_name, fix->type);
 			break;
 		}
-		if (!fix[id].chained)
+		if (!fix->chained)
 			break;
 		if (++depth > 10)
 			break;
-		id = fix[id].chain_id;
+		id = fix->chain_id;
 	}
 }
 
@@ -1842,7 +2123,7 @@ static void alc_auto_parse_digital(struct hda_codec *codec)
 /*
  * 2ch mode
  */
-static struct hda_verb alc888_4ST_ch2_intel_init[] = {
+static const struct hda_verb alc888_4ST_ch2_intel_init[] = {
 /* Mic-in jack as mic in */
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
@@ -1857,7 +2138,7 @@ static struct hda_verb alc888_4ST_ch2_intel_init[] = {
 /*
  * 4ch mode
  */
-static struct hda_verb alc888_4ST_ch4_intel_init[] = {
+static const struct hda_verb alc888_4ST_ch4_intel_init[] = {
 /* Mic-in jack as mic in */
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
@@ -1872,7 +2153,7 @@ static struct hda_verb alc888_4ST_ch4_intel_init[] = {
 /*
  * 6ch mode
  */
-static struct hda_verb alc888_4ST_ch6_intel_init[] = {
+static const struct hda_verb alc888_4ST_ch6_intel_init[] = {
 /* Mic-in jack as CLFE */
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
@@ -1887,7 +2168,7 @@ static struct hda_verb alc888_4ST_ch6_intel_init[] = {
 /*
  * 8ch mode
  */
-static struct hda_verb alc888_4ST_ch8_intel_init[] = {
+static const struct hda_verb alc888_4ST_ch8_intel_init[] = {
 /* Mic-in jack as CLFE */
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
@@ -1899,7 +2180,7 @@ static struct hda_verb alc888_4ST_ch8_intel_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc888_4ST_8ch_intel_modes[4] = {
+static const struct hda_channel_mode alc888_4ST_8ch_intel_modes[4] = {
 	{ 2, alc888_4ST_ch2_intel_init },
 	{ 4, alc888_4ST_ch4_intel_init },
 	{ 6, alc888_4ST_ch6_intel_init },
@@ -1910,7 +2191,7 @@ static struct hda_channel_mode alc888_4ST_8ch_intel_modes[4] = {
  * ALC888 Fujitsu Siemens Amillo xa3530
  */
 
-static struct hda_verb alc888_fujitsu_xa3530_verbs[] = {
+static const struct hda_verb alc888_fujitsu_xa3530_verbs[] = {
 /* Front Mic: set to PIN_IN (empty by default) */
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 /* Connect Internal HP to Front */
@@ -1943,22 +2224,6 @@ static struct hda_verb alc888_fujitsu_xa3530_verbs[] = {
 	{}
 };
 
-static void alc_automute_amp(struct hda_codec *codec)
-{
-	alc_automute_speaker(codec, 0);
-}
-
-static void alc_automute_amp_unsol_event(struct hda_codec *codec,
-					 unsigned int res)
-{
-	if (codec->vendor_id == 0x10ec0880)
-		res >>= 28;
-	else
-		res >>= 26;
-	if (res == ALC880_HP_EVENT)
-		alc_automute_amp(codec);
-}
-
 static void alc889_automute_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -1969,12 +2234,14 @@ static void alc889_automute_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[2] = 0x17;
 	spec->autocfg.speaker_pins[3] = 0x19;
 	spec->autocfg.speaker_pins[4] = 0x1a;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc889_intel_init_hook(struct hda_codec *codec)
 {
 	alc889_coef_init(codec);
-	alc_automute_amp(codec);
+	alc_hp_automute(codec);
 }
 
 static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec)
@@ -1985,13 +2252,15 @@ static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec)
 	spec->autocfg.hp_pins[1] = 0x1b; /* hp */
 	spec->autocfg.speaker_pins[0] = 0x14; /* speaker */
 	spec->autocfg.speaker_pins[1] = 0x15; /* bass */
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 /*
  * ALC888 Acer Aspire 4930G model
  */
 
-static struct hda_verb alc888_acer_aspire_4930g_verbs[] = {
+static const struct hda_verb alc888_acer_aspire_4930g_verbs[] = {
 /* Front Mic: set to PIN_IN (empty by default) */
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 /* Unselect Front Mic by default in input mixer 3 */
@@ -2014,7 +2283,7 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = {
  * ALC888 Acer Aspire 6530G model
  */
 
-static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
+static const struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
 /* Route to built-in subwoofer as well as speakers */
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
@@ -2044,7 +2313,7 @@ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
  *ALC888 Acer Aspire 7730G model
  */
 
-static struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
+static const struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
 /* Bias voltage on for external mic port */
 	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
 /* Front Mic: set to PIN_IN (empty by default) */
@@ -2074,7 +2343,7 @@ static struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
  * ALC889 Acer Aspire 8930G model
  */
 
-static struct hda_verb alc889_acer_aspire_8930g_verbs[] = {
+static const struct hda_verb alc889_acer_aspire_8930g_verbs[] = {
 /* Front Mic: set to PIN_IN (empty by default) */
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 /* Unselect Front Mic by default in input mixer 3 */
@@ -2120,7 +2389,7 @@ static struct hda_verb alc889_acer_aspire_8930g_verbs[] = {
 	{ }
 };
 
-static struct hda_input_mux alc888_2_capture_sources[2] = {
+static const struct hda_input_mux alc888_2_capture_sources[2] = {
 	/* Front mic only available on one ADC */
 	{
 		.num_items = 4,
@@ -2141,7 +2410,7 @@ static struct hda_input_mux alc888_2_capture_sources[2] = {
 	}
 };
 
-static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = {
+static const struct hda_input_mux alc888_acer_aspire_6530_sources[2] = {
 	/* Interal mic only available on one ADC */
 	{
 		.num_items = 5,
@@ -2164,7 +2433,7 @@ static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = {
 	}
 };
 
-static struct hda_input_mux alc889_capture_sources[3] = {
+static const struct hda_input_mux alc889_capture_sources[3] = {
 	/* Digital mic only available on first "ADC" */
 	{
 		.num_items = 5,
@@ -2196,7 +2465,7 @@ static struct hda_input_mux alc889_capture_sources[3] = {
 	}
 };
 
-static struct snd_kcontrol_new alc888_base_mixer[] = {
+static const struct snd_kcontrol_new alc888_base_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -2218,7 +2487,7 @@ static struct snd_kcontrol_new alc888_base_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = {
+static const struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -2240,7 +2509,7 @@ static struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = {
+static const struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -2267,6 +2536,8 @@ static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x16;
 	spec->autocfg.speaker_pins[2] = 0x17;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
@@ -2277,6 +2548,8 @@ static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x16;
 	spec->autocfg.speaker_pins[2] = 0x17;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec)
@@ -2287,6 +2560,8 @@ static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x16;
 	spec->autocfg.speaker_pins[2] = 0x17;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
@@ -2297,6 +2572,8 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x16;
 	spec->autocfg.speaker_pins[2] = 0x1b;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 /*
@@ -2307,12 +2584,12 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
  *                 F-Mic = 0x1b, HP = 0x19
  */
 
-static hda_nid_t alc880_dac_nids[4] = {
+static const hda_nid_t alc880_dac_nids[4] = {
 	/* front, rear, clfe, rear_surr */
 	0x02, 0x05, 0x04, 0x03
 };
 
-static hda_nid_t alc880_adc_nids[3] = {
+static const hda_nid_t alc880_adc_nids[3] = {
 	/* ADC0-2 */
 	0x07, 0x08, 0x09,
 };
@@ -2321,7 +2598,7 @@ static hda_nid_t alc880_adc_nids[3] = {
  * but it shows zero connection in the real implementation on some devices.
  * Note: this is a 915GAV bug, fixed on 915GLV
  */
-static hda_nid_t alc880_adc_nids_alt[2] = {
+static const hda_nid_t alc880_adc_nids_alt[2] = {
 	/* ADC1-2 */
 	0x08, 0x09,
 };
@@ -2329,7 +2606,7 @@ static hda_nid_t alc880_adc_nids_alt[2] = {
 #define ALC880_DIGOUT_NID	0x06
 #define ALC880_DIGIN_NID	0x0a
 
-static struct hda_input_mux alc880_capture_source = {
+static const struct hda_input_mux alc880_capture_source = {
 	.num_items = 4,
 	.items = {
 		{ "Mic", 0x0 },
@@ -2341,7 +2618,7 @@ static struct hda_input_mux alc880_capture_source = {
 
 /* channel source setting (2/6 channel selection for 3-stack) */
 /* 2ch mode */
-static struct hda_verb alc880_threestack_ch2_init[] = {
+static const struct hda_verb alc880_threestack_ch2_init[] = {
 	/* set line-in to input, mute it */
 	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
 	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
@@ -2352,7 +2629,7 @@ static struct hda_verb alc880_threestack_ch2_init[] = {
 };
 
 /* 6ch mode */
-static struct hda_verb alc880_threestack_ch6_init[] = {
+static const struct hda_verb alc880_threestack_ch6_init[] = {
 	/* set line-in to output, unmute it */
 	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
@@ -2362,12 +2639,12 @@ static struct hda_verb alc880_threestack_ch6_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc880_threestack_modes[2] = {
+static const struct hda_channel_mode alc880_threestack_modes[2] = {
 	{ 2, alc880_threestack_ch2_init },
 	{ 6, alc880_threestack_ch6_init },
 };
 
-static struct snd_kcontrol_new alc880_three_stack_mixer[] = {
+static const struct snd_kcontrol_new alc880_three_stack_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
@@ -2512,14 +2789,14 @@ static int alc_cap_sw_put(struct snd_kcontrol *kcontrol,
 	}
 
 #define DEFINE_CAPMIX(num) \
-static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \
+static const struct snd_kcontrol_new alc_capture_mixer ## num[] = { \
 	_DEFINE_CAPMIX(num),				      \
 	_DEFINE_CAPSRC(num),				      \
 	{ } /* end */					      \
 }
 
 #define DEFINE_CAPMIX_NOSRC(num) \
-static struct snd_kcontrol_new alc_capture_mixer_nosrc ## num[] = { \
+static const struct snd_kcontrol_new alc_capture_mixer_nosrc ## num[] = { \
 	_DEFINE_CAPMIX(num),					    \
 	{ } /* end */						    \
 }
@@ -2542,7 +2819,7 @@ DEFINE_CAPMIX_NOSRC(3);
  */
 
 /* additional mixers to alc880_three_stack_mixer */
-static struct snd_kcontrol_new alc880_five_stack_mixer[] = {
+static const struct snd_kcontrol_new alc880_five_stack_mixer[] = {
 	HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT),
 	{ } /* end */
@@ -2550,7 +2827,7 @@ static struct snd_kcontrol_new alc880_five_stack_mixer[] = {
 
 /* channel source setting (6/8 channel selection for 5-stack) */
 /* 6ch mode */
-static struct hda_verb alc880_fivestack_ch6_init[] = {
+static const struct hda_verb alc880_fivestack_ch6_init[] = {
 	/* set line-in to input, mute it */
 	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
 	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
@@ -2558,14 +2835,14 @@ static struct hda_verb alc880_fivestack_ch6_init[] = {
 };
 
 /* 8ch mode */
-static struct hda_verb alc880_fivestack_ch8_init[] = {
+static const struct hda_verb alc880_fivestack_ch8_init[] = {
 	/* set line-in to output, unmute it */
 	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc880_fivestack_modes[2] = {
+static const struct hda_channel_mode alc880_fivestack_modes[2] = {
 	{ 6, alc880_fivestack_ch6_init },
 	{ 8, alc880_fivestack_ch8_init },
 };
@@ -2580,12 +2857,12 @@ static struct hda_channel_mode alc880_fivestack_modes[2] = {
  *   Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b
  */
 
-static hda_nid_t alc880_6st_dac_nids[4] = {
+static const hda_nid_t alc880_6st_dac_nids[4] = {
 	/* front, rear, clfe, rear_surr */
 	0x02, 0x03, 0x04, 0x05
 };
 
-static struct hda_input_mux alc880_6stack_capture_source = {
+static const struct hda_input_mux alc880_6stack_capture_source = {
 	.num_items = 4,
 	.items = {
 		{ "Mic", 0x0 },
@@ -2596,11 +2873,11 @@ static struct hda_input_mux alc880_6stack_capture_source = {
 };
 
 /* fixed 8-channels */
-static struct hda_channel_mode alc880_sixstack_modes[1] = {
+static const struct hda_channel_mode alc880_sixstack_modes[1] = {
 	{ 8, NULL },
 };
 
-static struct snd_kcontrol_new alc880_six_stack_mixer[] = {
+static const struct snd_kcontrol_new alc880_six_stack_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -2655,18 +2932,18 @@ static struct snd_kcontrol_new alc880_six_stack_mixer[] = {
  * haven't setup any initialization verbs for these yet...
  */
 
-static hda_nid_t alc880_w810_dac_nids[3] = {
+static const hda_nid_t alc880_w810_dac_nids[3] = {
 	/* front, rear/surround, clfe */
 	0x02, 0x03, 0x04
 };
 
 /* fixed 6 channels */
-static struct hda_channel_mode alc880_w810_modes[1] = {
+static const struct hda_channel_mode alc880_w810_modes[1] = {
 	{ 6, NULL }
 };
 
 /* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */
-static struct snd_kcontrol_new alc880_w810_base_mixer[] = {
+static const struct snd_kcontrol_new alc880_w810_base_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -2688,17 +2965,17 @@ static struct snd_kcontrol_new alc880_w810_base_mixer[] = {
  *                 Line = 0x1a
  */
 
-static hda_nid_t alc880_z71v_dac_nids[1] = {
+static const hda_nid_t alc880_z71v_dac_nids[1] = {
 	0x02
 };
 #define ALC880_Z71V_HP_DAC	0x03
 
 /* fixed 2 channels */
-static struct hda_channel_mode alc880_2_jack_modes[1] = {
+static const struct hda_channel_mode alc880_2_jack_modes[1] = {
 	{ 2, NULL }
 };
 
-static struct snd_kcontrol_new alc880_z71v_mixer[] = {
+static const struct snd_kcontrol_new alc880_z71v_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -2718,12 +2995,12 @@ static struct snd_kcontrol_new alc880_z71v_mixer[] = {
  * Pin assignment: HP = 0x14, Front = 0x15, Mic = 0x18
  */
 
-static hda_nid_t alc880_f1734_dac_nids[1] = {
+static const hda_nid_t alc880_f1734_dac_nids[1] = {
 	0x03
 };
 #define ALC880_F1734_HP_DAC	0x02
 
-static struct snd_kcontrol_new alc880_f1734_mixer[] = {
+static const struct snd_kcontrol_new alc880_f1734_mixer[] = {
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -2735,7 +3012,7 @@ static struct snd_kcontrol_new alc880_f1734_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_input_mux alc880_f1734_capture_source = {
+static const struct hda_input_mux alc880_f1734_capture_source = {
 	.num_items = 2,
 	.items = {
 		{ "Mic", 0x1 },
@@ -2755,7 +3032,7 @@ static struct hda_input_mux alc880_f1734_capture_source = {
 #define alc880_asus_dac_nids	alc880_w810_dac_nids	/* identical with w810 */
 #define alc880_asus_modes	alc880_threestack_modes	/* 2/6 channel mode */
 
-static struct snd_kcontrol_new alc880_asus_mixer[] = {
+static const struct snd_kcontrol_new alc880_asus_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -2789,14 +3066,14 @@ static struct snd_kcontrol_new alc880_asus_mixer[] = {
  */
 
 /* additional mixers to alc880_asus_mixer */
-static struct snd_kcontrol_new alc880_asus_w1v_mixer[] = {
+static const struct snd_kcontrol_new alc880_asus_w1v_mixer[] = {
 	HDA_CODEC_VOLUME("Line2 Playback Volume", 0x0b, 0x03, HDA_INPUT),
 	HDA_CODEC_MUTE("Line2 Playback Switch", 0x0b, 0x03, HDA_INPUT),
 	{ } /* end */
 };
 
 /* TCL S700 */
-static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = {
+static const struct snd_kcontrol_new alc880_tcl_s700_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
@@ -2810,7 +3087,7 @@ static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = {
 };
 
 /* Uniwill */
-static struct snd_kcontrol_new alc880_uniwill_mixer[] = {
+static const struct snd_kcontrol_new alc880_uniwill_mixer[] = {
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -2837,7 +3114,7 @@ static struct snd_kcontrol_new alc880_uniwill_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc880_fujitsu_mixer[] = {
+static const struct snd_kcontrol_new alc880_fujitsu_mixer[] = {
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -2851,7 +3128,7 @@ static struct snd_kcontrol_new alc880_fujitsu_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = {
+static const struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = {
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -2878,7 +3155,6 @@ static const char * const alc_slave_vols[] = {
 	"Speaker Playback Volume",
 	"Mono Playback Volume",
 	"Line-Out Playback Volume",
-	"PCM Playback Volume",
 	NULL,
 };
 
@@ -2893,7 +3169,6 @@ static const char * const alc_slave_sws[] = {
 	"Mono Playback Switch",
 	"IEC958 Playback Switch",
 	"Line-Out Playback Switch",
-	"PCM Playback Switch",
 	NULL,
 };
 
@@ -2914,7 +3189,7 @@ static void alc_free_kctls(struct hda_codec *codec);
 
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
 /* additional beep mixers; the actual parameters are overwritten at build */
-static struct snd_kcontrol_new alc_beep_mixer[] = {
+static const struct snd_kcontrol_new alc_beep_mixer[] = {
 	HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT),
 	HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_INPUT),
 	{ } /* end */
@@ -2925,7 +3200,7 @@ static int alc_build_controls(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	struct snd_kcontrol *kctl = NULL;
-	struct snd_kcontrol_new *knew;
+	const struct snd_kcontrol_new *knew;
 	int i, j, err;
 	unsigned int u;
 	hda_nid_t nid;
@@ -2962,7 +3237,7 @@ static int alc_build_controls(struct hda_codec *codec)
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
 	/* create beep controls if needed */
 	if (spec->beep_amp) {
-		struct snd_kcontrol_new *knew;
+		const struct snd_kcontrol_new *knew;
 		for (knew = alc_beep_mixer; knew->name; knew++) {
 			struct snd_kcontrol *kctl;
 			kctl = snd_ctl_new1(knew, codec);
@@ -3001,7 +3276,7 @@ static int alc_build_controls(struct hda_codec *codec)
 		if (!kctl)
 			kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
 		for (i = 0; kctl && i < kctl->count; i++) {
-			hda_nid_t *nids = spec->capsrc_nids;
+			const hda_nid_t *nids = spec->capsrc_nids;
 			if (!nids)
 				nids = spec->adc_nids;
 			err = snd_hda_add_nid(codec, kctl, i, nids[i]);
@@ -3079,7 +3354,7 @@ static int alc_build_controls(struct hda_codec *codec)
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb alc880_volume_init_verbs[] = {
+static const struct hda_verb alc880_volume_init_verbs[] = {
 	/*
 	 * Unmute ADC0-2 and set the default input to mic-in
 	 */
@@ -3130,7 +3405,7 @@ static struct hda_verb alc880_volume_init_verbs[] = {
  * 3-stack pin configuration:
  * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
  */
-static struct hda_verb alc880_pin_3stack_init_verbs[] = {
+static const struct hda_verb alc880_pin_3stack_init_verbs[] = {
 	/*
 	 * preset connection lists of input pins
 	 * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
@@ -3168,7 +3443,7 @@ static struct hda_verb alc880_pin_3stack_init_verbs[] = {
  * front = 0x14, surround = 0x17, clfe = 0x16, mic = 0x18, HP = 0x19,
  * line-in/side = 0x1a, f-mic = 0x1b
  */
-static struct hda_verb alc880_pin_5stack_init_verbs[] = {
+static const struct hda_verb alc880_pin_5stack_init_verbs[] = {
 	/*
 	 * preset connection lists of input pins
 	 * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
@@ -3212,7 +3487,7 @@ static struct hda_verb alc880_pin_5stack_init_verbs[] = {
  * W810 pin configuration:
  * front = 0x14, surround = 0x15, clfe = 0x16, HP = 0x1b
  */
-static struct hda_verb alc880_pin_w810_init_verbs[] = {
+static const struct hda_verb alc880_pin_w810_init_verbs[] = {
 	/* hphone/speaker input selector: front DAC */
 	{0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
 
@@ -3233,7 +3508,7 @@ static struct hda_verb alc880_pin_w810_init_verbs[] = {
  * Z71V pin configuration:
  * Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?)
  */
-static struct hda_verb alc880_pin_z71v_init_verbs[] = {
+static const struct hda_verb alc880_pin_z71v_init_verbs[] = {
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -3252,7 +3527,7 @@ static struct hda_verb alc880_pin_z71v_init_verbs[] = {
  * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18,
  * f-mic = 0x19, line = 0x1a, HP = 0x1b
  */
-static struct hda_verb alc880_pin_6stack_init_verbs[] = {
+static const struct hda_verb alc880_pin_6stack_init_verbs[] = {
 	{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
 
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
@@ -3282,7 +3557,7 @@ static struct hda_verb alc880_pin_6stack_init_verbs[] = {
  * HP = 0x14, InternalSpeaker = 0x15, mic = 0x18, internal mic = 0x19,
  * line = 0x1a
  */
-static struct hda_verb alc880_uniwill_init_verbs[] = {
+static const struct hda_verb alc880_uniwill_init_verbs[] = {
 	{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
 
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -3320,7 +3595,7 @@ static struct hda_verb alc880_uniwill_init_verbs[] = {
 * Uniwill P53
 * HP = 0x14, InternalSpeaker = 0x15, mic = 0x19,
  */
-static struct hda_verb alc880_uniwill_p53_init_verbs[] = {
+static const struct hda_verb alc880_uniwill_p53_init_verbs[] = {
 	{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
 
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -3349,7 +3624,7 @@ static struct hda_verb alc880_uniwill_p53_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc880_beep_init_verbs[] = {
+static const struct hda_verb alc880_beep_init_verbs[] = {
 	{ 0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) },
 	{ }
 };
@@ -3372,11 +3647,13 @@ static void alc880_uniwill_setup(struct hda_codec *codec)
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x16;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc880_uniwill_init_hook(struct hda_codec *codec)
 {
-	alc_automute_amp(codec);
+	alc_hp_automute(codec);
 	alc88x_simple_mic_automute(codec);
 }
 
@@ -3391,7 +3668,7 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec,
 		alc88x_simple_mic_automute(codec);
 		break;
 	default:
-		alc_automute_amp_unsol_event(codec, res);
+		alc_sku_unsol_event(codec, res);
 		break;
 	}
 }
@@ -3402,6 +3679,8 @@ static void alc880_uniwill_p53_setup(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
@@ -3426,14 +3705,14 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
 	if ((res >> 28) == ALC880_DCVOL_EVENT)
 		alc880_uniwill_p53_dcvol_automute(codec);
 	else
-		alc_automute_amp_unsol_event(codec, res);
+		alc_sku_unsol_event(codec, res);
 }
 
 /*
  * F1734 pin configuration:
  * HP = 0x14, speaker-out = 0x15, mic = 0x18
  */
-static struct hda_verb alc880_pin_f1734_init_verbs[] = {
+static const struct hda_verb alc880_pin_f1734_init_verbs[] = {
 	{0x07, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
 	{0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -3465,7 +3744,7 @@ static struct hda_verb alc880_pin_f1734_init_verbs[] = {
  * ASUS pin configuration:
  * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a
  */
-static struct hda_verb alc880_pin_asus_init_verbs[] = {
+static const struct hda_verb alc880_pin_asus_init_verbs[] = {
 	{0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
 	{0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
@@ -3499,7 +3778,7 @@ static struct hda_verb alc880_pin_asus_init_verbs[] = {
 #define alc880_gpio3_init_verbs	alc_gpio3_init_verbs
 
 /* Clevo m520g init */
-static struct hda_verb alc880_pin_clevo_init_verbs[] = {
+static const struct hda_verb alc880_pin_clevo_init_verbs[] = {
 	/* headphone output */
 	{0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
 	/* line-out */
@@ -3527,7 +3806,7 @@ static struct hda_verb alc880_pin_clevo_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc880_pin_tcl_S700_init_verbs[] = {
+static const struct hda_verb alc880_pin_tcl_S700_init_verbs[] = {
 	/* change to EAPD mode */
 	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
 	{0x20, AC_VERB_SET_PROC_COEF,  0x3060},
@@ -3565,12 +3844,12 @@ static struct hda_verb alc880_pin_tcl_S700_init_verbs[] = {
  */
 
 /* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */
-static hda_nid_t alc880_lg_dac_nids[3] = {
+static const hda_nid_t alc880_lg_dac_nids[3] = {
 	0x05, 0x02, 0x03
 };
 
 /* seems analog CD is not working */
-static struct hda_input_mux alc880_lg_capture_source = {
+static const struct hda_input_mux alc880_lg_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic", 0x1 },
@@ -3580,34 +3859,34 @@ static struct hda_input_mux alc880_lg_capture_source = {
 };
 
 /* 2,4,6 channel modes */
-static struct hda_verb alc880_lg_ch2_init[] = {
+static const struct hda_verb alc880_lg_ch2_init[] = {
 	/* set line-in and mic-in to input */
 	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
 	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
 	{ }
 };
 
-static struct hda_verb alc880_lg_ch4_init[] = {
+static const struct hda_verb alc880_lg_ch4_init[] = {
 	/* set line-in to out and mic-in to input */
 	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
 	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
 	{ }
 };
 
-static struct hda_verb alc880_lg_ch6_init[] = {
+static const struct hda_verb alc880_lg_ch6_init[] = {
 	/* set line-in and mic-in to output */
 	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
 	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
 	{ }
 };
 
-static struct hda_channel_mode alc880_lg_ch_modes[3] = {
+static const struct hda_channel_mode alc880_lg_ch_modes[3] = {
 	{ 2, alc880_lg_ch2_init },
 	{ 4, alc880_lg_ch4_init },
 	{ 6, alc880_lg_ch6_init },
 };
 
-static struct snd_kcontrol_new alc880_lg_mixer[] = {
+static const struct snd_kcontrol_new alc880_lg_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0f, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -3632,7 +3911,7 @@ static struct snd_kcontrol_new alc880_lg_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc880_lg_init_verbs[] = {
+static const struct hda_verb alc880_lg_init_verbs[] = {
 	/* set capture source to mic-in */
 	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
@@ -3670,6 +3949,8 @@ static void alc880_lg_setup(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x17;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 /*
@@ -3684,7 +3965,7 @@ static void alc880_lg_setup(struct hda_codec *codec)
  *   SPDIF-Out: 0x1e
  */
 
-static struct hda_input_mux alc880_lg_lw_capture_source = {
+static const struct hda_input_mux alc880_lg_lw_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic", 0x0 },
@@ -3695,7 +3976,7 @@ static struct hda_input_mux alc880_lg_lw_capture_source = {
 
 #define alc880_lg_lw_modes alc880_threestack_modes
 
-static struct snd_kcontrol_new alc880_lg_lw_mixer[] = {
+static const struct snd_kcontrol_new alc880_lg_lw_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
@@ -3720,7 +4001,7 @@ static struct snd_kcontrol_new alc880_lg_lw_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc880_lg_lw_init_verbs[] = {
+static const struct hda_verb alc880_lg_lw_init_verbs[] = {
 	{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
 	{0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
 	{0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
@@ -3754,9 +4035,11 @@ static void alc880_lg_lw_setup(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
-static struct snd_kcontrol_new alc880_medion_rim_mixer[] = {
+static const struct snd_kcontrol_new alc880_medion_rim_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
@@ -3766,7 +4049,7 @@ static struct snd_kcontrol_new alc880_medion_rim_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_input_mux alc880_medion_rim_capture_source = {
+static const struct hda_input_mux alc880_medion_rim_capture_source = {
 	.num_items = 2,
 	.items = {
 		{ "Mic", 0x0 },
@@ -3774,7 +4057,7 @@ static struct hda_input_mux alc880_medion_rim_capture_source = {
 	},
 };
 
-static struct hda_verb alc880_medion_rim_init_verbs[] = {
+static const struct hda_verb alc880_medion_rim_init_verbs[] = {
 	{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
 
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -3801,7 +4084,7 @@ static struct hda_verb alc880_medion_rim_init_verbs[] = {
 static void alc880_medion_rim_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	alc_automute_amp(codec);
+	alc_hp_automute(codec);
 	/* toggle EAPD */
 	if (spec->jack_present)
 		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0);
@@ -3825,10 +4108,12 @@ static void alc880_medion_rim_setup(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x1b;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list alc880_loopbacks[] = {
+static const struct hda_amp_list alc880_loopbacks[] = {
 	{ 0x0b, HDA_INPUT, 0 },
 	{ 0x0b, HDA_INPUT, 1 },
 	{ 0x0b, HDA_INPUT, 2 },
@@ -3837,7 +4122,7 @@ static struct hda_amp_list alc880_loopbacks[] = {
 	{ } /* end */
 };
 
-static struct hda_amp_list alc880_lg_loopbacks[] = {
+static const struct hda_amp_list alc880_lg_loopbacks[] = {
 	{ 0x0b, HDA_INPUT, 1 },
 	{ 0x0b, HDA_INPUT, 6 },
 	{ 0x0b, HDA_INPUT, 7 },
@@ -4009,7 +4294,7 @@ static int dualmic_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 	return 0;
 }
 
-static struct hda_pcm_stream dualmic_pcm_analog_capture = {
+static const struct hda_pcm_stream dualmic_pcm_analog_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -4022,7 +4307,7 @@ static struct hda_pcm_stream dualmic_pcm_analog_capture = {
 
 /*
  */
-static struct hda_pcm_stream alc880_pcm_analog_playback = {
+static const struct hda_pcm_stream alc880_pcm_analog_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 8,
@@ -4034,21 +4319,21 @@ static struct hda_pcm_stream alc880_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream alc880_pcm_analog_capture = {
+static const struct hda_pcm_stream alc880_pcm_analog_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
 	/* NID is set in alc_build_pcms */
 };
 
-static struct hda_pcm_stream alc880_pcm_analog_alt_playback = {
+static const struct hda_pcm_stream alc880_pcm_analog_alt_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
 	/* NID is set in alc_build_pcms */
 };
 
-static struct hda_pcm_stream alc880_pcm_analog_alt_capture = {
+static const struct hda_pcm_stream alc880_pcm_analog_alt_capture = {
 	.substreams = 2, /* can be overridden */
 	.channels_min = 2,
 	.channels_max = 2,
@@ -4059,7 +4344,7 @@ static struct hda_pcm_stream alc880_pcm_analog_alt_capture = {
 	},
 };
 
-static struct hda_pcm_stream alc880_pcm_digital_playback = {
+static const struct hda_pcm_stream alc880_pcm_digital_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -4072,7 +4357,7 @@ static struct hda_pcm_stream alc880_pcm_digital_playback = {
 	},
 };
 
-static struct hda_pcm_stream alc880_pcm_digital_capture = {
+static const struct hda_pcm_stream alc880_pcm_digital_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -4080,7 +4365,7 @@ static struct hda_pcm_stream alc880_pcm_digital_capture = {
 };
 
 /* Used by alc_build_pcms to flag that a PCM has no playback stream */
-static struct hda_pcm_stream alc_pcm_null_stream = {
+static const struct hda_pcm_stream alc_pcm_null_stream = {
 	.substreams = 0,
 	.channels_min = 0,
 	.channels_max = 0,
@@ -4174,7 +4459,7 @@ static int alc_build_pcms(struct hda_codec *codec)
 				alc_pcm_null_stream;
 			info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0;
 		}
-		if (spec->num_adc_nids > 1) {
+		if (spec->num_adc_nids > 1 && spec->stream_analog_alt_capture) {
 			info->stream[SNDRV_PCM_STREAM_CAPTURE] =
 				*spec->stream_analog_alt_capture;
 			info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
@@ -4193,6 +4478,10 @@ static int alc_build_pcms(struct hda_codec *codec)
 
 static inline void alc_shutup(struct hda_codec *codec)
 {
+	struct alc_spec *spec = codec->spec;
+
+	if (spec && spec->shutup)
+		spec->shutup(codec);
 	snd_hda_shutup_pins(codec);
 }
 
@@ -4226,28 +4515,7 @@ static void alc_free(struct hda_codec *codec)
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 static void alc_power_eapd(struct hda_codec *codec)
 {
-	/* We currently only handle front, HP */
-	switch (codec->vendor_id) {
-	case 0x10ec0260:
-		set_eapd(codec, 0x0f, 0);
-		set_eapd(codec, 0x10, 0);
-		break;
-	case 0x10ec0262:
-	case 0x10ec0267:
-	case 0x10ec0268:
-	case 0x10ec0269:
-	case 0x10ec0270:
-	case 0x10ec0272:
-	case 0x10ec0660:
-	case 0x10ec0662:
-	case 0x10ec0663:
-	case 0x10ec0665:
-	case 0x10ec0862:
-	case 0x10ec0889:
-		set_eapd(codec, 0x14, 0);
-		set_eapd(codec, 0x15, 0);
-		break;
-	}
+	alc_auto_setup_eapd(codec, false);
 }
 
 static int alc_suspend(struct hda_codec *codec, pm_message_t state)
@@ -4263,6 +4531,7 @@ static int alc_suspend(struct hda_codec *codec, pm_message_t state)
 #ifdef SND_HDA_NEEDS_RESUME
 static int alc_resume(struct hda_codec *codec)
 {
+	msleep(150); /* to avoid pop noise */
 	codec->patch_ops.init(codec);
 	snd_hda_codec_resume_amp(codec);
 	snd_hda_codec_resume_cache(codec);
@@ -4273,7 +4542,7 @@ static int alc_resume(struct hda_codec *codec)
 
 /*
  */
-static struct hda_codec_ops alc_patch_ops = {
+static const struct hda_codec_ops alc_patch_ops = {
 	.build_controls = alc_build_controls,
 	.build_pcms = alc_build_pcms,
 	.init = alc_init,
@@ -4308,11 +4577,11 @@ static int alc_codec_rename(struct hda_codec *codec, const char *name)
  * enum controls.
  */
 #ifdef CONFIG_SND_DEBUG
-static hda_nid_t alc880_test_dac_nids[4] = {
+static const hda_nid_t alc880_test_dac_nids[4] = {
 	0x02, 0x03, 0x04, 0x05
 };
 
-static struct hda_input_mux alc880_test_capture_source = {
+static const struct hda_input_mux alc880_test_capture_source = {
 	.num_items = 7,
 	.items = {
 		{ "In-1", 0x0 },
@@ -4325,7 +4594,7 @@ static struct hda_input_mux alc880_test_capture_source = {
 	},
 };
 
-static struct hda_channel_mode alc880_test_modes[4] = {
+static const struct hda_channel_mode alc880_test_modes[4] = {
 	{ 2, NULL },
 	{ 4, NULL },
 	{ 6, NULL },
@@ -4335,7 +4604,7 @@ static struct hda_channel_mode alc880_test_modes[4] = {
 static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {
+	static const char * const texts[] = {
 		"N/A", "Line Out", "HP Out",
 		"In Hi-Z", "In 50%", "In Grd", "In 80%", "In 100%"
 	};
@@ -4380,7 +4649,7 @@ static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
-	static unsigned int ctls[] = {
+	static const unsigned int ctls[] = {
 		0, AC_PINCTL_OUT_EN, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN,
 		AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ,
 		AC_PINCTL_IN_EN | AC_PINCTL_VREF_50,
@@ -4410,7 +4679,7 @@ static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol,
 static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {
+	static const char * const texts[] = {
 		"Front", "Surround", "CLFE", "Side"
 	};
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
@@ -4471,7 +4740,7 @@ static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol,
 			.private_value = nid	       \
 			}
 
-static struct snd_kcontrol_new alc880_test_mixer[] = {
+static const struct snd_kcontrol_new alc880_test_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("CLFE Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
@@ -4512,7 +4781,7 @@ static struct snd_kcontrol_new alc880_test_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc880_test_init_verbs[] = {
+static const struct hda_verb alc880_test_init_verbs[] = {
 	/* Unmute inputs of 0x0c - 0x0f */
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
@@ -4596,7 +4865,7 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = {
 	[ALC880_AUTO]		= "auto",
 };
 
-static struct snd_pci_quirk alc880_cfg_tbl[] = {
+static const struct snd_pci_quirk alc880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_W810),
 	SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_5ST_DIG),
 	SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_6ST),
@@ -4676,7 +4945,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = {
 /*
  * ALC880 codec presets
  */
-static struct alc_config_preset alc880_presets[] = {
+static const struct alc_config_preset alc880_presets[] = {
 	[ALC880_3ST] = {
 		.mixers = { alc880_three_stack_mixer },
 		.init_verbs = { alc880_volume_init_verbs,
@@ -4794,7 +5063,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.input_mux = &alc880_f1734_capture_source,
 		.unsol_event = alc880_uniwill_p53_unsol_event,
 		.setup = alc880_uniwill_p53_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC880_ASUS] = {
 		.mixers = { alc880_asus_mixer },
@@ -4885,7 +5154,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.input_mux = &alc880_capture_source,
 		.unsol_event = alc880_uniwill_p53_unsol_event,
 		.setup = alc880_uniwill_p53_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC880_FUJITSU] = {
 		.mixers = { alc880_fujitsu_mixer },
@@ -4900,7 +5169,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.input_mux = &alc880_capture_source,
 		.unsol_event = alc880_uniwill_p53_unsol_event,
 		.setup = alc880_uniwill_p53_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC880_CLEVO] = {
 		.mixers = { alc880_three_stack_mixer },
@@ -4925,9 +5194,9 @@ static struct alc_config_preset alc880_presets[] = {
 		.channel_mode = alc880_lg_ch_modes,
 		.need_dac_fix = 1,
 		.input_mux = &alc880_lg_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc880_lg_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 		.loopbacks = alc880_lg_loopbacks,
 #endif
@@ -4942,9 +5211,9 @@ static struct alc_config_preset alc880_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc880_lg_lw_modes),
 		.channel_mode = alc880_lg_lw_modes,
 		.input_mux = &alc880_lg_lw_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc880_lg_lw_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC880_MEDION_RIM] = {
 		.mixers = { alc880_medion_rim_mixer },
@@ -4984,20 +5253,25 @@ enum {
 	ALC_CTL_WIDGET_MUTE,
 	ALC_CTL_BIND_MUTE,
 };
-static struct snd_kcontrol_new alc880_control_templates[] = {
+static const struct snd_kcontrol_new alc880_control_templates[] = {
 	HDA_CODEC_VOLUME(NULL, 0, 0, 0),
 	HDA_CODEC_MUTE(NULL, 0, 0, 0),
 	HDA_BIND_MUTE(NULL, 0, 0, 0),
 };
 
+static struct snd_kcontrol_new *alc_kcontrol_new(struct alc_spec *spec)
+{
+	snd_array_init(&spec->kctls, sizeof(struct snd_kcontrol_new), 32);
+	return snd_array_new(&spec->kctls);
+}
+
 /* add dynamic controls */
 static int add_control(struct alc_spec *spec, int type, const char *name,
 		       int cidx, unsigned long val)
 {
 	struct snd_kcontrol_new *knew;
 
-	snd_array_init(&spec->kctls, sizeof(*knew), 32);
-	knew = snd_array_new(&spec->kctls);
+	knew = alc_kcontrol_new(spec);
 	if (!knew)
 		return -ENOMEM;
 	*knew = alc880_control_templates[type];
@@ -5055,7 +5329,7 @@ static int alc880_auto_fill_dac_nids(struct alc_spec *spec,
 		nid = cfg->line_out_pins[i];
 		if (alc880_is_fixed_pin(nid)) {
 			int idx = alc880_fixed_pin_idx(nid);
-			spec->multiout.dac_nids[i] = alc880_idx_to_dac(idx);
+			spec->private_dac_nids[i] = alc880_idx_to_dac(idx);
 			assigned[idx] = 1;
 		}
 	}
@@ -5067,7 +5341,7 @@ static int alc880_auto_fill_dac_nids(struct alc_spec *spec,
 		/* search for an empty channel */
 		for (j = 0; j < cfg->line_outs; j++) {
 			if (!assigned[j]) {
-				spec->multiout.dac_nids[i] =
+				spec->private_dac_nids[i] =
 					alc880_idx_to_dac(j);
 				assigned[j] = 1;
 				break;
@@ -5078,10 +5352,13 @@ static int alc880_auto_fill_dac_nids(struct alc_spec *spec,
 	return 0;
 }
 
-static const char *alc_get_line_out_pfx(const struct auto_pin_cfg *cfg,
+static const char *alc_get_line_out_pfx(struct alc_spec *spec,
 					bool can_be_master)
 {
-	if (!cfg->hp_outs && !cfg->speaker_outs && can_be_master)
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+
+	if (cfg->line_outs == 1 && !spec->multi_ios &&
+	    !cfg->hp_outs && !cfg->speaker_outs && can_be_master)
 		return "Master";
 
 	switch (cfg->line_out_type) {
@@ -5092,7 +5369,7 @@ static const char *alc_get_line_out_pfx(const struct auto_pin_cfg *cfg,
 	case AUTO_PIN_HP_OUT:
 		return "Headphone";
 	default:
-		if (cfg->line_outs == 1)
+		if (cfg->line_outs == 1 && !spec->multi_ios)
 			return "PCM";
 		break;
 	}
@@ -5106,11 +5383,15 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec,
 	static const char * const chname[4] = {
 		"Front", "Surround", NULL /*CLFE*/, "Side"
 	};
-	const char *pfx = alc_get_line_out_pfx(cfg, false);
+	const char *pfx = alc_get_line_out_pfx(spec, false);
 	hda_nid_t nid;
-	int i, err;
+	int i, err, noutputs;
 
-	for (i = 0; i < cfg->line_outs; i++) {
+	noutputs = cfg->line_outs;
+	if (spec->multi_ios > 0)
+		noutputs += spec->multi_ios;
+
+	for (i = 0; i < noutputs; i++) {
 		if (!spec->multiout.dac_nids[i])
 			continue;
 		nid = alc880_idx_to_mixer(alc880_dac_to_idx(spec->multiout.dac_nids[i]));
@@ -5376,6 +5657,8 @@ static void alc880_auto_init_input_src(struct hda_codec *codec)
 	}
 }
 
+static int alc_auto_add_multi_channel_mode(struct hda_codec *codec);
+
 /* parse the BIOS configuration and set up the alc_spec */
 /* return 1 if successful, 0 if the proper config is not found,
  * or a negative error code
@@ -5384,7 +5667,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	int err;
-	static hda_nid_t alc880_ignore[] = { 0x1d, 0 };
+	static const hda_nid_t alc880_ignore[] = { 0x1d, 0 };
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
 					   alc880_ignore);
@@ -5396,6 +5679,9 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 	err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
+	err = alc_auto_add_multi_channel_mode(codec);
+	if (err < 0)
+		return err;
 	err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
@@ -5467,6 +5753,12 @@ static void fixup_automic_adc(struct hda_codec *codec)
 			spec->capsrc_nids += i;
 		spec->adc_nids += i;
 		spec->num_adc_nids = 1;
+		/* optional dock-mic */
+		eidx = get_connection_index(codec, cap, spec->dock_mic.pin);
+		if (eidx < 0)
+			spec->dock_mic.pin = 0;
+		else
+			spec->dock_mic.mux_idx = eidx;
 		return;
 	}
 	snd_printd(KERN_INFO "hda_codec: %s: "
@@ -5494,6 +5786,8 @@ static int init_capsrc_for_pin(struct hda_codec *codec, hda_nid_t pin)
 	struct alc_spec *spec = codec->spec;
 	int i;
 
+	if (!pin)
+		return 0;
 	for (i = 0; i < spec->num_adc_nids; i++) {
 		hda_nid_t cap = spec->capsrc_nids ?
 			spec->capsrc_nids[i] : spec->adc_nids[i];
@@ -5534,6 +5828,7 @@ static void fixup_dual_adc_switch(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	init_capsrc_for_pin(codec, spec->ext_mic.pin);
+	init_capsrc_for_pin(codec, spec->dock_mic.pin);
 	init_capsrc_for_pin(codec, spec->int_mic.pin);
 }
 
@@ -5550,7 +5845,7 @@ static void alc_init_special_input_src(struct hda_codec *codec)
 static void set_capture_mixer(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	static struct snd_kcontrol_new *caps[2][3] = {
+	static const struct snd_kcontrol_new *caps[2][3] = {
 		{ alc_capture_mixer_nosrc1,
 		  alc_capture_mixer_nosrc2,
 		  alc_capture_mixer_nosrc3 },
@@ -5576,7 +5871,7 @@ static void set_capture_mixer(struct hda_codec *codec)
 }
 
 /* fill adc_nids (and capsrc_nids) containing all active input pins */
-static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids,
+static void fillup_priv_adc_nids(struct hda_codec *codec, const hda_nid_t *nids,
 				 int num_nids)
 {
 	struct alc_spec *spec = codec->spec;
@@ -5642,9 +5937,11 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids,
 #define set_beep_amp(spec, nid, idx, dir) \
 	((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir))
 
-static struct snd_pci_quirk beep_white_list[] = {
+static const struct snd_pci_quirk beep_white_list[] = {
 	SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1),
 	SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1),
+	SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1),
+	SND_PCI_QUIRK(0x1043, 0x834a, "EeePC", 1),
 	SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1),
 	{}
 };
@@ -5752,17 +6049,17 @@ static int patch_alc880(struct hda_codec *codec)
  * ALC260 support
  */
 
-static hda_nid_t alc260_dac_nids[1] = {
+static const hda_nid_t alc260_dac_nids[1] = {
 	/* front */
 	0x02,
 };
 
-static hda_nid_t alc260_adc_nids[1] = {
+static const hda_nid_t alc260_adc_nids[1] = {
 	/* ADC0 */
 	0x04,
 };
 
-static hda_nid_t alc260_adc_nids_alt[1] = {
+static const hda_nid_t alc260_adc_nids_alt[1] = {
 	/* ADC1 */
 	0x05,
 };
@@ -5770,7 +6067,7 @@ static hda_nid_t alc260_adc_nids_alt[1] = {
 /* NIDs used when simultaneous access to both ADCs makes sense.  Note that
  * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC.
  */
-static hda_nid_t alc260_dual_adc_nids[2] = {
+static const hda_nid_t alc260_dual_adc_nids[2] = {
 	/* ADC0, ADC1 */
 	0x04, 0x05
 };
@@ -5778,7 +6075,7 @@ static hda_nid_t alc260_dual_adc_nids[2] = {
 #define ALC260_DIGOUT_NID	0x03
 #define ALC260_DIGIN_NID	0x06
 
-static struct hda_input_mux alc260_capture_source = {
+static const struct hda_input_mux alc260_capture_source = {
 	.num_items = 4,
 	.items = {
 		{ "Mic", 0x0 },
@@ -5794,7 +6091,7 @@ static struct hda_input_mux alc260_capture_source = {
  * recording the mixer output on the second ADC (ADC0 doesn't have a
  * connection to the mixer output).
  */
-static struct hda_input_mux alc260_fujitsu_capture_sources[2] = {
+static const struct hda_input_mux alc260_fujitsu_capture_sources[2] = {
 	{
 		.num_items = 3,
 		.items = {
@@ -5818,7 +6115,7 @@ static struct hda_input_mux alc260_fujitsu_capture_sources[2] = {
 /* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to
  * the Fujitsu S702x, but jacks are marked differently.
  */
-static struct hda_input_mux alc260_acer_capture_sources[2] = {
+static const struct hda_input_mux alc260_acer_capture_sources[2] = {
 	{
 		.num_items = 4,
 		.items = {
@@ -5841,7 +6138,7 @@ static struct hda_input_mux alc260_acer_capture_sources[2] = {
 };
 
 /* Maxdata Favorit 100XS */
-static struct hda_input_mux alc260_favorit100_capture_sources[2] = {
+static const struct hda_input_mux alc260_favorit100_capture_sources[2] = {
 	{
 		.num_items = 2,
 		.items = {
@@ -5865,7 +6162,7 @@ static struct hda_input_mux alc260_favorit100_capture_sources[2] = {
  * element which allows changing the channel mode, so the verb list is
  * never used.
  */
-static struct hda_channel_mode alc260_modes[1] = {
+static const struct hda_channel_mode alc260_modes[1] = {
 	{ 2, NULL },
 };
 
@@ -5879,7 +6176,7 @@ static struct hda_channel_mode alc260_modes[1] = {
  * acer: acer + capture
  */
 
-static struct snd_kcontrol_new alc260_base_output_mixer[] = {
+static const struct snd_kcontrol_new alc260_base_output_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
@@ -5889,7 +6186,7 @@ static struct snd_kcontrol_new alc260_base_output_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc260_input_mixer[] = {
+static const struct snd_kcontrol_new alc260_input_mixer[] = {
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
@@ -5902,21 +6199,14 @@ static struct snd_kcontrol_new alc260_input_mixer[] = {
 };
 
 /* update HP, line and mono out pins according to the master switch */
-static void alc260_hp_master_update(struct hda_codec *codec,
-				    hda_nid_t hp, hda_nid_t line,
-				    hda_nid_t mono)
+static void alc260_hp_master_update(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	unsigned int val = spec->master_sw ? PIN_HP : 0;
-	/* change HP and line-out pins */
-	snd_hda_codec_write(codec, hp, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    val);
-	snd_hda_codec_write(codec, line, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    val);
-	/* mono (speaker) depending on the HP jack sense */
-	val = (val && !spec->jack_present) ? PIN_OUT : 0;
-	snd_hda_codec_write(codec, mono, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    val);
+
+	/* change HP pins */
+	do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
+		    spec->autocfg.hp_pins, spec->master_mute, true);
+	update_speakers(codec);
 }
 
 static int alc260_hp_master_sw_get(struct snd_kcontrol *kcontrol,
@@ -5924,7 +6214,7 @@ static int alc260_hp_master_sw_get(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct alc_spec *spec = codec->spec;
-	*ucontrol->value.integer.value = spec->master_sw;
+	*ucontrol->value.integer.value = !spec->master_mute;
 	return 0;
 }
 
@@ -5933,20 +6223,16 @@ static int alc260_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct alc_spec *spec = codec->spec;
-	int val = !!*ucontrol->value.integer.value;
-	hda_nid_t hp, line, mono;
+	int val = !*ucontrol->value.integer.value;
 
-	if (val == spec->master_sw)
+	if (val == spec->master_mute)
 		return 0;
-	spec->master_sw = val;
-	hp = (kcontrol->private_value >> 16) & 0xff;
-	line = (kcontrol->private_value >> 8) & 0xff;
-	mono = kcontrol->private_value & 0xff;
-	alc260_hp_master_update(codec, hp, line, mono);
+	spec->master_mute = val;
+	alc260_hp_master_update(codec);
 	return 1;
 }
 
-static struct snd_kcontrol_new alc260_hp_output_mixer[] = {
+static const struct snd_kcontrol_new alc260_hp_output_mixer[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Master Playback Switch",
@@ -5954,7 +6240,6 @@ static struct snd_kcontrol_new alc260_hp_output_mixer[] = {
 		.info = snd_ctl_boolean_mono_info,
 		.get = alc260_hp_master_sw_get,
 		.put = alc260_hp_master_sw_put,
-		.private_value = (0x0f << 16) | (0x10 << 8) | 0x11
 	},
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
@@ -5966,26 +6251,23 @@ static struct snd_kcontrol_new alc260_hp_output_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc260_hp_unsol_verbs[] = {
+static const struct hda_verb alc260_hp_unsol_verbs[] = {
 	{0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
 	{},
 };
 
-static void alc260_hp_automute(struct hda_codec *codec)
+static void alc260_hp_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	spec->jack_present = snd_hda_jack_detect(codec, 0x10);
-	alc260_hp_master_update(codec, 0x0f, 0x10, 0x11);
-}
-
-static void alc260_hp_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	if ((res >> 26) == ALC880_HP_EVENT)
-		alc260_hp_automute(codec);
+	spec->autocfg.hp_pins[0] = 0x0f;
+	spec->autocfg.speaker_pins[0] = 0x10;
+	spec->autocfg.speaker_pins[1] = 0x11;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
 }
 
-static struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
+static const struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Master Playback Switch",
@@ -5993,7 +6275,6 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
 		.info = snd_ctl_boolean_mono_info,
 		.get = alc260_hp_master_sw_get,
 		.put = alc260_hp_master_sw_put,
-		.private_value = (0x15 << 16) | (0x10 << 8) | 0x11
 	},
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT),
@@ -6006,7 +6287,18 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_bind_ctls alc260_dc7600_bind_master_vol = {
+static void alc260_hp_3013_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x10;
+	spec->autocfg.speaker_pins[1] = 0x11;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
+static const struct hda_bind_ctls alc260_dc7600_bind_master_vol = {
 	.ops = &snd_hda_bind_vol,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_OUTPUT),
@@ -6016,7 +6308,7 @@ static struct hda_bind_ctls alc260_dc7600_bind_master_vol = {
 	},
 };
 
-static struct hda_bind_ctls alc260_dc7600_bind_switch = {
+static const struct hda_bind_ctls alc260_dc7600_bind_switch = {
 	.ops = &snd_hda_bind_sw,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT),
@@ -6025,7 +6317,7 @@ static struct hda_bind_ctls alc260_dc7600_bind_switch = {
 	},
 };
 
-static struct snd_kcontrol_new alc260_hp_dc7600_mixer[] = {
+static const struct snd_kcontrol_new alc260_hp_dc7600_mixer[] = {
 	HDA_BIND_VOL("Master Playback Volume", &alc260_dc7600_bind_master_vol),
 	HDA_BIND_SW("LineOut Playback Switch", &alc260_dc7600_bind_switch),
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x0f, 0x0, HDA_OUTPUT),
@@ -6033,49 +6325,27 @@ static struct snd_kcontrol_new alc260_hp_dc7600_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc260_hp_3013_unsol_verbs[] = {
+static const struct hda_verb alc260_hp_3013_unsol_verbs[] = {
 	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
 	{},
 };
 
-static void alc260_hp_3013_automute(struct hda_codec *codec)
+static void alc260_hp_3012_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	spec->jack_present = snd_hda_jack_detect(codec, 0x15);
-	alc260_hp_master_update(codec, 0x15, 0x10, 0x11);
-}
-
-static void alc260_hp_3013_unsol_event(struct hda_codec *codec,
-				       unsigned int res)
-{
-	if ((res >> 26) == ALC880_HP_EVENT)
-		alc260_hp_3013_automute(codec);
-}
-
-static void alc260_hp_3012_automute(struct hda_codec *codec)
-{
-	unsigned int bits = snd_hda_jack_detect(codec, 0x10) ? 0 : PIN_OUT;
-
-	snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    bits);
-	snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    bits);
-	snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    bits);
-}
-
-static void alc260_hp_3012_unsol_event(struct hda_codec *codec,
-				       unsigned int res)
-{
-	if ((res >> 26) == ALC880_HP_EVENT)
-		alc260_hp_3012_automute(codec);
+	spec->autocfg.hp_pins[0] = 0x10;
+	spec->autocfg.speaker_pins[0] = 0x0f;
+	spec->autocfg.speaker_pins[1] = 0x11;
+	spec->autocfg.speaker_pins[2] = 0x15;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
 }
 
 /* Fujitsu S702x series laptops.  ALC260 pin usage: Mic/Line jack = 0x12,
  * HP jack = 0x14, CD audio =  0x16, internal speaker = 0x10.
  */
-static struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
+static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT),
 	ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
@@ -6112,7 +6382,7 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
  * controls for such models.  On models without a "mono speaker" the control
  * won't do anything.
  */
-static struct snd_kcontrol_new alc260_acer_mixer[] = {
+static const struct snd_kcontrol_new alc260_acer_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
 	ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
@@ -6133,7 +6403,7 @@ static struct snd_kcontrol_new alc260_acer_mixer[] = {
 
 /* Maxdata Favorit 100XS: one output and one input (0x12) jack
  */
-static struct snd_kcontrol_new alc260_favorit100_mixer[] = {
+static const struct snd_kcontrol_new alc260_favorit100_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
 	ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
@@ -6146,7 +6416,7 @@ static struct snd_kcontrol_new alc260_favorit100_mixer[] = {
 /* Packard bell V7900  ALC260 pin usage: HP = 0x0f, Mic jack = 0x12,
  * Line In jack = 0x14, CD audio =  0x16, pc beep = 0x17.
  */
-static struct snd_kcontrol_new alc260_will_mixer[] = {
+static const struct snd_kcontrol_new alc260_will_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
@@ -6163,7 +6433,7 @@ static struct snd_kcontrol_new alc260_will_mixer[] = {
 /* Replacer 672V ALC260 pin usage: Mic jack = 0x12,
  * Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f.
  */
-static struct snd_kcontrol_new alc260_replacer_672v_mixer[] = {
+static const struct snd_kcontrol_new alc260_replacer_672v_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
@@ -6180,7 +6450,7 @@ static struct snd_kcontrol_new alc260_replacer_672v_mixer[] = {
 /*
  * initialization verbs
  */
-static struct hda_verb alc260_init_verbs[] = {
+static const struct hda_verb alc260_init_verbs[] = {
 	/* Line In pin widget for input */
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	/* CD pin widget for input */
@@ -6244,7 +6514,7 @@ static struct hda_verb alc260_init_verbs[] = {
 };
 
 #if 0 /* should be identical with alc260_init_verbs? */
-static struct hda_verb alc260_hp_init_verbs[] = {
+static const struct hda_verb alc260_hp_init_verbs[] = {
 	/* Headphone and output */
 	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
 	/* mono output */
@@ -6294,7 +6564,7 @@ static struct hda_verb alc260_hp_init_verbs[] = {
 };
 #endif
 
-static struct hda_verb alc260_hp_3013_init_verbs[] = {
+static const struct hda_verb alc260_hp_3013_init_verbs[] = {
 	/* Line out and output */
 	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
 	/* mono output */
@@ -6347,7 +6617,7 @@ static struct hda_verb alc260_hp_3013_init_verbs[] = {
  * laptops.  ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD
  * audio = 0x16, internal speaker = 0x10.
  */
-static struct hda_verb alc260_fujitsu_init_verbs[] = {
+static const struct hda_verb alc260_fujitsu_init_verbs[] = {
 	/* Disable all GPIOs */
 	{0x01, AC_VERB_SET_GPIO_MASK, 0},
 	/* Internal speaker is connected to headphone pin */
@@ -6429,7 +6699,7 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = {
 /* Initialisation sequence for ALC260 as configured in Acer TravelMate and
  * similar laptops (adapted from Fujitsu init verbs).
  */
-static struct hda_verb alc260_acer_init_verbs[] = {
+static const struct hda_verb alc260_acer_init_verbs[] = {
 	/* On TravelMate laptops, GPIO 0 enables the internal speaker and
 	 * the headphone jack.  Turn this on and rely on the standard mute
 	 * methods whenever the user wants to turn these outputs off.
@@ -6517,7 +6787,7 @@ static struct hda_verb alc260_acer_init_verbs[] = {
 /* Initialisation sequence for Maxdata Favorit 100XS
  * (adapted from Acer init verbs).
  */
-static struct hda_verb alc260_favorit100_init_verbs[] = {
+static const struct hda_verb alc260_favorit100_init_verbs[] = {
 	/* GPIO 0 enables the output jack.
 	 * Turn this on and rely on the standard mute
 	 * methods whenever the user wants to turn these outputs off.
@@ -6597,7 +6867,7 @@ static struct hda_verb alc260_favorit100_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc260_will_verbs[] = {
+static const struct hda_verb alc260_will_verbs[] = {
 	{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -6607,7 +6877,7 @@ static struct hda_verb alc260_will_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc260_replacer_672v_verbs[] = {
+static const struct hda_verb alc260_replacer_672v_verbs[] = {
 	{0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
 	{0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
 	{0x1a, AC_VERB_SET_PROC_COEF, 0x3050},
@@ -6649,7 +6919,7 @@ static void alc260_replacer_672v_unsol_event(struct hda_codec *codec,
                 alc260_replacer_672v_automute(codec);
 }
 
-static struct hda_verb alc260_hp_dc7600_verbs[] = {
+static const struct hda_verb alc260_hp_dc7600_verbs[] = {
 	{0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
@@ -6667,17 +6937,17 @@ static struct hda_verb alc260_hp_dc7600_verbs[] = {
  * configuration.
  */
 #ifdef CONFIG_SND_DEBUG
-static hda_nid_t alc260_test_dac_nids[1] = {
+static const hda_nid_t alc260_test_dac_nids[1] = {
 	0x02,
 };
-static hda_nid_t alc260_test_adc_nids[2] = {
+static const hda_nid_t alc260_test_adc_nids[2] = {
 	0x04, 0x05,
 };
 /* For testing the ALC260, each input MUX needs its own definition since
  * the signal assignments are different.  This assumes that the first ADC
  * is NID 0x04.
  */
-static struct hda_input_mux alc260_test_capture_sources[2] = {
+static const struct hda_input_mux alc260_test_capture_sources[2] = {
 	{
 		.num_items = 7,
 		.items = {
@@ -6704,7 +6974,7 @@ static struct hda_input_mux alc260_test_capture_sources[2] = {
 		},
         },
 };
-static struct snd_kcontrol_new alc260_test_mixer[] = {
+static const struct snd_kcontrol_new alc260_test_mixer[] = {
 	/* Output driver widgets */
 	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
@@ -6768,7 +7038,7 @@ static struct snd_kcontrol_new alc260_test_mixer[] = {
 
 	{ } /* end */
 };
-static struct hda_verb alc260_test_init_verbs[] = {
+static const struct hda_verb alc260_test_init_verbs[] = {
 	/* Enable all GPIOs as outputs with an initial value of 0 */
 	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f},
 	{0x01, AC_VERB_SET_GPIO_DATA, 0x00},
@@ -6906,7 +7176,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
 
 	spec->multiout.num_dacs = 1;
 	spec->multiout.dac_nids = spec->private_dac_nids;
-	spec->multiout.dac_nids[0] = 0x02;
+	spec->private_dac_nids[0] = 0x02;
 
 	nid = cfg->line_out_pins[0];
 	if (nid) {
@@ -7004,7 +7274,7 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec)
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb alc260_volume_init_verbs[] = {
+static const struct hda_verb alc260_volume_init_verbs[] = {
 	/*
 	 * Unmute ADC0-1 and set the default input to mic-in
 	 */
@@ -7049,7 +7319,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	int err;
-	static hda_nid_t alc260_ignore[] = { 0x17, 0 };
+	static const hda_nid_t alc260_ignore[] = { 0x17, 0 };
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
 					   alc260_ignore);
@@ -7094,7 +7364,7 @@ static void alc260_auto_init(struct hda_codec *codec)
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list alc260_loopbacks[] = {
+static const struct hda_amp_list alc260_loopbacks[] = {
 	{ 0x07, HDA_INPUT, 0 },
 	{ 0x07, HDA_INPUT, 1 },
 	{ 0x07, HDA_INPUT, 2 },
@@ -7121,7 +7391,7 @@ static const struct alc_fixup alc260_fixups[] = {
 	},
 };
 
-static struct snd_pci_quirk alc260_fixup_tbl[] = {
+static const struct snd_pci_quirk alc260_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", PINFIX_HP_DC5750),
 	{}
 };
@@ -7145,7 +7415,7 @@ static const char * const alc260_models[ALC260_MODEL_LAST] = {
 	[ALC260_AUTO]		= "auto",
 };
 
-static struct snd_pci_quirk alc260_cfg_tbl[] = {
+static const struct snd_pci_quirk alc260_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER),
 	SND_PCI_QUIRK(0x1025, 0x007f, "Acer", ALC260_WILL),
 	SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
@@ -7169,7 +7439,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = {
 	{}
 };
 
-static struct alc_config_preset alc260_presets[] = {
+static const struct alc_config_preset alc260_presets[] = {
 	[ALC260_BASIC] = {
 		.mixers = { alc260_base_output_mixer,
 			    alc260_input_mixer },
@@ -7194,8 +7464,9 @@ static struct alc_config_preset alc260_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc260_modes),
 		.channel_mode = alc260_modes,
 		.input_mux = &alc260_capture_source,
-		.unsol_event = alc260_hp_unsol_event,
-		.init_hook = alc260_hp_automute,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc260_hp_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC260_HP_DC7600] = {
 		.mixers = { alc260_hp_dc7600_mixer,
@@ -7209,8 +7480,9 @@ static struct alc_config_preset alc260_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc260_modes),
 		.channel_mode = alc260_modes,
 		.input_mux = &alc260_capture_source,
-		.unsol_event = alc260_hp_3012_unsol_event,
-		.init_hook = alc260_hp_3012_automute,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc260_hp_3012_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC260_HP_3013] = {
 		.mixers = { alc260_hp_3013_mixer,
@@ -7224,8 +7496,9 @@ static struct alc_config_preset alc260_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc260_modes),
 		.channel_mode = alc260_modes,
 		.input_mux = &alc260_capture_source,
-		.unsol_event = alc260_hp_3013_unsol_event,
-		.init_hook = alc260_hp_3013_automute,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc260_hp_3013_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC260_FUJITSU_S702X] = {
 		.mixers = { alc260_fujitsu_mixer },
@@ -7383,6 +7656,7 @@ static int patch_alc260(struct hda_codec *codec)
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC260_AUTO)
 		spec->init_hook = alc260_auto_init;
+	spec->shutup = alc_eapd_shutup;
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc260_loopbacks;
@@ -7410,12 +7684,12 @@ static int patch_alc260(struct hda_codec *codec)
 #define ALC1200_DIGOUT_NID	0x10
 
 
-static struct hda_channel_mode alc882_ch_modes[1] = {
+static const struct hda_channel_mode alc882_ch_modes[1] = {
 	{ 8, NULL }
 };
 
 /* DACs */
-static hda_nid_t alc882_dac_nids[4] = {
+static const hda_nid_t alc882_dac_nids[4] = {
 	/* front, rear, clfe, rear_surr */
 	0x02, 0x03, 0x04, 0x05
 };
@@ -7425,20 +7699,20 @@ static hda_nid_t alc882_dac_nids[4] = {
 #define alc882_adc_nids		alc880_adc_nids
 #define alc882_adc_nids_alt	alc880_adc_nids_alt
 #define alc883_adc_nids		alc882_adc_nids_alt
-static hda_nid_t alc883_adc_nids_alt[1] = { 0x08 };
-static hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 };
+static const hda_nid_t alc883_adc_nids_alt[1] = { 0x08 };
+static const hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 };
 #define alc889_adc_nids		alc880_adc_nids
 
-static hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 };
-static hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 };
+static const hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 };
+static const hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 };
 #define alc883_capsrc_nids	alc882_capsrc_nids_alt
-static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 };
+static const hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 };
 #define alc889_capsrc_nids	alc882_capsrc_nids
 
 /* input MUX */
 /* FIXME: should be a matrix-type input source selection */
 
-static struct hda_input_mux alc882_capture_source = {
+static const struct hda_input_mux alc882_capture_source = {
 	.num_items = 4,
 	.items = {
 		{ "Mic", 0x0 },
@@ -7450,7 +7724,7 @@ static struct hda_input_mux alc882_capture_source = {
 
 #define alc883_capture_source	alc882_capture_source
 
-static struct hda_input_mux alc889_capture_source = {
+static const struct hda_input_mux alc889_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Front Mic", 0x0 },
@@ -7459,7 +7733,7 @@ static struct hda_input_mux alc889_capture_source = {
 	},
 };
 
-static struct hda_input_mux mb5_capture_source = {
+static const struct hda_input_mux mb5_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic", 0x1 },
@@ -7468,7 +7742,7 @@ static struct hda_input_mux mb5_capture_source = {
 	},
 };
 
-static struct hda_input_mux macmini3_capture_source = {
+static const struct hda_input_mux macmini3_capture_source = {
 	.num_items = 2,
 	.items = {
 		{ "Line", 0x2 },
@@ -7476,7 +7750,7 @@ static struct hda_input_mux macmini3_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc883_3stack_6ch_intel = {
+static const struct hda_input_mux alc883_3stack_6ch_intel = {
 	.num_items = 4,
 	.items = {
 		{ "Mic", 0x1 },
@@ -7486,7 +7760,7 @@ static struct hda_input_mux alc883_3stack_6ch_intel = {
 	},
 };
 
-static struct hda_input_mux alc883_lenovo_101e_capture_source = {
+static const struct hda_input_mux alc883_lenovo_101e_capture_source = {
 	.num_items = 2,
 	.items = {
 		{ "Mic", 0x1 },
@@ -7494,7 +7768,7 @@ static struct hda_input_mux alc883_lenovo_101e_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc883_lenovo_nb0763_capture_source = {
+static const struct hda_input_mux alc883_lenovo_nb0763_capture_source = {
 	.num_items = 4,
 	.items = {
 		{ "Mic", 0x0 },
@@ -7504,7 +7778,7 @@ static struct hda_input_mux alc883_lenovo_nb0763_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = {
+static const struct hda_input_mux alc883_fujitsu_pi2515_capture_source = {
 	.num_items = 2,
 	.items = {
 		{ "Mic", 0x0 },
@@ -7512,7 +7786,7 @@ static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc883_lenovo_sky_capture_source = {
+static const struct hda_input_mux alc883_lenovo_sky_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic", 0x0 },
@@ -7521,7 +7795,7 @@ static struct hda_input_mux alc883_lenovo_sky_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc883_asus_eee1601_capture_source = {
+static const struct hda_input_mux alc883_asus_eee1601_capture_source = {
 	.num_items = 2,
 	.items = {
 		{ "Mic", 0x0 },
@@ -7529,7 +7803,7 @@ static struct hda_input_mux alc883_asus_eee1601_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc889A_mb31_capture_source = {
+static const struct hda_input_mux alc889A_mb31_capture_source = {
 	.num_items = 2,
 	.items = {
 		{ "Mic", 0x0 },
@@ -7540,7 +7814,7 @@ static struct hda_input_mux alc889A_mb31_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc889A_imac91_capture_source = {
+static const struct hda_input_mux alc889A_imac91_capture_source = {
 	.num_items = 2,
 	.items = {
 		{ "Mic", 0x01 },
@@ -7551,14 +7825,14 @@ static struct hda_input_mux alc889A_imac91_capture_source = {
 /*
  * 2ch mode
  */
-static struct hda_channel_mode alc883_3ST_2ch_modes[1] = {
+static const struct hda_channel_mode alc883_3ST_2ch_modes[1] = {
 	{ 2, NULL }
 };
 
 /*
  * 2ch mode
  */
-static struct hda_verb alc882_3ST_ch2_init[] = {
+static const struct hda_verb alc882_3ST_ch2_init[] = {
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
 	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
@@ -7569,7 +7843,7 @@ static struct hda_verb alc882_3ST_ch2_init[] = {
 /*
  * 4ch mode
  */
-static struct hda_verb alc882_3ST_ch4_init[] = {
+static const struct hda_verb alc882_3ST_ch4_init[] = {
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
 	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
@@ -7581,7 +7855,7 @@ static struct hda_verb alc882_3ST_ch4_init[] = {
 /*
  * 6ch mode
  */
-static struct hda_verb alc882_3ST_ch6_init[] = {
+static const struct hda_verb alc882_3ST_ch6_init[] = {
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
 	{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
@@ -7591,7 +7865,7 @@ static struct hda_verb alc882_3ST_ch6_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc882_3ST_6ch_modes[3] = {
+static const struct hda_channel_mode alc882_3ST_6ch_modes[3] = {
 	{ 2, alc882_3ST_ch2_init },
 	{ 4, alc882_3ST_ch4_init },
 	{ 6, alc882_3ST_ch6_init },
@@ -7602,7 +7876,7 @@ static struct hda_channel_mode alc882_3ST_6ch_modes[3] = {
 /*
  * 2ch mode
  */
-static struct hda_verb alc883_3ST_ch2_clevo_init[] = {
+static const struct hda_verb alc883_3ST_ch2_clevo_init[] = {
 	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
@@ -7614,7 +7888,7 @@ static struct hda_verb alc883_3ST_ch2_clevo_init[] = {
 /*
  * 4ch mode
  */
-static struct hda_verb alc883_3ST_ch4_clevo_init[] = {
+static const struct hda_verb alc883_3ST_ch4_clevo_init[] = {
 	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
@@ -7627,7 +7901,7 @@ static struct hda_verb alc883_3ST_ch4_clevo_init[] = {
 /*
  * 6ch mode
  */
-static struct hda_verb alc883_3ST_ch6_clevo_init[] = {
+static const struct hda_verb alc883_3ST_ch6_clevo_init[] = {
 	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
@@ -7638,7 +7912,7 @@ static struct hda_verb alc883_3ST_ch6_clevo_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc883_3ST_6ch_clevo_modes[3] = {
+static const struct hda_channel_mode alc883_3ST_6ch_clevo_modes[3] = {
 	{ 2, alc883_3ST_ch2_clevo_init },
 	{ 4, alc883_3ST_ch4_clevo_init },
 	{ 6, alc883_3ST_ch6_clevo_init },
@@ -7648,7 +7922,7 @@ static struct hda_channel_mode alc883_3ST_6ch_clevo_modes[3] = {
 /*
  * 6ch mode
  */
-static struct hda_verb alc882_sixstack_ch6_init[] = {
+static const struct hda_verb alc882_sixstack_ch6_init[] = {
 	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
 	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
@@ -7659,7 +7933,7 @@ static struct hda_verb alc882_sixstack_ch6_init[] = {
 /*
  * 8ch mode
  */
-static struct hda_verb alc882_sixstack_ch8_init[] = {
+static const struct hda_verb alc882_sixstack_ch8_init[] = {
 	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
@@ -7667,7 +7941,7 @@ static struct hda_verb alc882_sixstack_ch8_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc882_sixstack_modes[2] = {
+static const struct hda_channel_mode alc882_sixstack_modes[2] = {
 	{ 6, alc882_sixstack_ch6_init },
 	{ 8, alc882_sixstack_ch8_init },
 };
@@ -7675,7 +7949,7 @@ static struct hda_channel_mode alc882_sixstack_modes[2] = {
 
 /* Macbook Air 2,1 */
 
-static struct hda_channel_mode alc885_mba21_ch_modes[1] = {
+static const struct hda_channel_mode alc885_mba21_ch_modes[1] = {
       { 2, NULL },
 };
 
@@ -7686,7 +7960,7 @@ static struct hda_channel_mode alc885_mba21_ch_modes[1] = {
 /*
  * 2ch mode
  */
-static struct hda_verb alc885_mbp_ch2_init[] = {
+static const struct hda_verb alc885_mbp_ch2_init[] = {
 	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
 	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
@@ -7696,7 +7970,7 @@ static struct hda_verb alc885_mbp_ch2_init[] = {
 /*
  * 4ch mode
  */
-static struct hda_verb alc885_mbp_ch4_init[] = {
+static const struct hda_verb alc885_mbp_ch4_init[] = {
 	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
@@ -7705,7 +7979,7 @@ static struct hda_verb alc885_mbp_ch4_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc885_mbp_4ch_modes[2] = {
+static const struct hda_channel_mode alc885_mbp_4ch_modes[2] = {
 	{ 2, alc885_mbp_ch2_init },
 	{ 4, alc885_mbp_ch4_init },
 };
@@ -7715,7 +7989,7 @@ static struct hda_channel_mode alc885_mbp_4ch_modes[2] = {
  * Speakers/Woofer/HP = Front
  * LineIn = Input
  */
-static struct hda_verb alc885_mb5_ch2_init[] = {
+static const struct hda_verb alc885_mb5_ch2_init[] = {
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{ } /* end */
@@ -7727,14 +8001,14 @@ static struct hda_verb alc885_mb5_ch2_init[] = {
  * Woofer = LFE
  * LineIn = Surround
  */
-static struct hda_verb alc885_mb5_ch6_init[] = {
+static const struct hda_verb alc885_mb5_ch6_init[] = {
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc885_mb5_6ch_modes[2] = {
+static const struct hda_channel_mode alc885_mb5_6ch_modes[2] = {
 	{ 2, alc885_mb5_ch2_init },
 	{ 6, alc885_mb5_ch6_init },
 };
@@ -7744,7 +8018,7 @@ static struct hda_channel_mode alc885_mb5_6ch_modes[2] = {
 /*
  * 2ch mode
  */
-static struct hda_verb alc883_4ST_ch2_init[] = {
+static const struct hda_verb alc883_4ST_ch2_init[] = {
 	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
@@ -7757,7 +8031,7 @@ static struct hda_verb alc883_4ST_ch2_init[] = {
 /*
  * 4ch mode
  */
-static struct hda_verb alc883_4ST_ch4_init[] = {
+static const struct hda_verb alc883_4ST_ch4_init[] = {
 	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
@@ -7771,7 +8045,7 @@ static struct hda_verb alc883_4ST_ch4_init[] = {
 /*
  * 6ch mode
  */
-static struct hda_verb alc883_4ST_ch6_init[] = {
+static const struct hda_verb alc883_4ST_ch6_init[] = {
 	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
@@ -7786,7 +8060,7 @@ static struct hda_verb alc883_4ST_ch6_init[] = {
 /*
  * 8ch mode
  */
-static struct hda_verb alc883_4ST_ch8_init[] = {
+static const struct hda_verb alc883_4ST_ch8_init[] = {
 	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
 	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
@@ -7799,7 +8073,7 @@ static struct hda_verb alc883_4ST_ch8_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc883_4ST_8ch_modes[4] = {
+static const struct hda_channel_mode alc883_4ST_8ch_modes[4] = {
 	{ 2, alc883_4ST_ch2_init },
 	{ 4, alc883_4ST_ch4_init },
 	{ 6, alc883_4ST_ch6_init },
@@ -7810,7 +8084,7 @@ static struct hda_channel_mode alc883_4ST_8ch_modes[4] = {
 /*
  * 2ch mode
  */
-static struct hda_verb alc883_3ST_ch2_intel_init[] = {
+static const struct hda_verb alc883_3ST_ch2_intel_init[] = {
 	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
 	{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
 	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
@@ -7821,7 +8095,7 @@ static struct hda_verb alc883_3ST_ch2_intel_init[] = {
 /*
  * 4ch mode
  */
-static struct hda_verb alc883_3ST_ch4_intel_init[] = {
+static const struct hda_verb alc883_3ST_ch4_intel_init[] = {
 	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
 	{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
 	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
@@ -7833,7 +8107,7 @@ static struct hda_verb alc883_3ST_ch4_intel_init[] = {
 /*
  * 6ch mode
  */
-static struct hda_verb alc883_3ST_ch6_intel_init[] = {
+static const struct hda_verb alc883_3ST_ch6_intel_init[] = {
 	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
 	{ 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 },
@@ -7843,7 +8117,7 @@ static struct hda_verb alc883_3ST_ch6_intel_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = {
+static const struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = {
 	{ 2, alc883_3ST_ch2_intel_init },
 	{ 4, alc883_3ST_ch4_intel_init },
 	{ 6, alc883_3ST_ch6_intel_init },
@@ -7852,7 +8126,7 @@ static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = {
 /*
  * 2ch mode
  */
-static struct hda_verb alc889_ch2_intel_init[] = {
+static const struct hda_verb alc889_ch2_intel_init[] = {
 	{ 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
 	{ 0x19, AC_VERB_SET_CONNECT_SEL, 0x00 },
 	{ 0x16, AC_VERB_SET_CONNECT_SEL, 0x00 },
@@ -7865,7 +8139,7 @@ static struct hda_verb alc889_ch2_intel_init[] = {
 /*
  * 6ch mode
  */
-static struct hda_verb alc889_ch6_intel_init[] = {
+static const struct hda_verb alc889_ch6_intel_init[] = {
 	{ 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
 	{ 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 },
 	{ 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 },
@@ -7878,7 +8152,7 @@ static struct hda_verb alc889_ch6_intel_init[] = {
 /*
  * 8ch mode
  */
-static struct hda_verb alc889_ch8_intel_init[] = {
+static const struct hda_verb alc889_ch8_intel_init[] = {
 	{ 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
 	{ 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 },
 	{ 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 },
@@ -7889,7 +8163,7 @@ static struct hda_verb alc889_ch8_intel_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc889_8ch_intel_modes[3] = {
+static const struct hda_channel_mode alc889_8ch_intel_modes[3] = {
 	{ 2, alc889_ch2_intel_init },
 	{ 6, alc889_ch6_intel_init },
 	{ 8, alc889_ch8_intel_init },
@@ -7898,7 +8172,7 @@ static struct hda_channel_mode alc889_8ch_intel_modes[3] = {
 /*
  * 6ch mode
  */
-static struct hda_verb alc883_sixstack_ch6_init[] = {
+static const struct hda_verb alc883_sixstack_ch6_init[] = {
 	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
 	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
@@ -7909,7 +8183,7 @@ static struct hda_verb alc883_sixstack_ch6_init[] = {
 /*
  * 8ch mode
  */
-static struct hda_verb alc883_sixstack_ch8_init[] = {
+static const struct hda_verb alc883_sixstack_ch8_init[] = {
 	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
@@ -7917,7 +8191,7 @@ static struct hda_verb alc883_sixstack_ch8_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc883_sixstack_modes[2] = {
+static const struct hda_channel_mode alc883_sixstack_modes[2] = {
 	{ 6, alc883_sixstack_ch6_init },
 	{ 8, alc883_sixstack_ch8_init },
 };
@@ -7926,7 +8200,7 @@ static struct hda_channel_mode alc883_sixstack_modes[2] = {
 /* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
  *                 Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
  */
-static struct snd_kcontrol_new alc882_base_mixer[] = {
+static const struct snd_kcontrol_new alc882_base_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -7953,14 +8227,14 @@ static struct snd_kcontrol_new alc882_base_mixer[] = {
 
 /* Macbook Air 2,1 same control for HP and internal Speaker */
 
-static struct snd_kcontrol_new alc885_mba21_mixer[] = {
+static const struct snd_kcontrol_new alc885_mba21_mixer[] = {
       HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
       HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_OUTPUT),
      { }
 };
 
 
-static struct snd_kcontrol_new alc885_mbp3_mixer[] = {
+static const struct snd_kcontrol_new alc885_mbp3_mixer[] = {
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
 	HDA_BIND_MUTE   ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
@@ -7975,7 +8249,7 @@ static struct snd_kcontrol_new alc885_mbp3_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc885_mb5_mixer[] = {
+static const struct snd_kcontrol_new alc885_mb5_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
 	HDA_BIND_MUTE   ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
@@ -7993,7 +8267,7 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc885_macmini3_mixer[] = {
+static const struct snd_kcontrol_new alc885_macmini3_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
 	HDA_BIND_MUTE   ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
@@ -8008,14 +8282,14 @@ static struct snd_kcontrol_new alc885_macmini3_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc885_imac91_mixer[] = {
+static const struct snd_kcontrol_new alc885_imac91_mixer[] = {
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
 	HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT),
 	{ } /* end */
 };
 
 
-static struct snd_kcontrol_new alc882_w2jc_mixer[] = {
+static const struct snd_kcontrol_new alc882_w2jc_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
@@ -8028,7 +8302,7 @@ static struct snd_kcontrol_new alc882_w2jc_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc882_targa_mixer[] = {
+static const struct snd_kcontrol_new alc882_targa_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
@@ -8048,7 +8322,7 @@ static struct snd_kcontrol_new alc882_targa_mixer[] = {
 /* Pin assignment: Front=0x14, HP = 0x15, Front = 0x16, ???
  *                 Front Mic=0x18, Line In = 0x1a, Line In = 0x1b, CD = 0x1c
  */
-static struct snd_kcontrol_new alc882_asus_a7j_mixer[] = {
+static const struct snd_kcontrol_new alc882_asus_a7j_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
@@ -8065,7 +8339,7 @@ static struct snd_kcontrol_new alc882_asus_a7j_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc882_asus_a7m_mixer[] = {
+static const struct snd_kcontrol_new alc882_asus_a7m_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
@@ -8079,7 +8353,7 @@ static struct snd_kcontrol_new alc882_asus_a7m_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc882_chmode_mixer[] = {
+static const struct snd_kcontrol_new alc882_chmode_mixer[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Channel Mode",
@@ -8090,7 +8364,7 @@ static struct snd_kcontrol_new alc882_chmode_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc882_base_init_verbs[] = {
+static const struct hda_verb alc882_base_init_verbs[] = {
 	/* Front mixer: unmute input/output amp left and right (volume = 0) */
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
@@ -8152,7 +8426,7 @@ static struct hda_verb alc882_base_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc882_adc1_init_verbs[] = {
+static const struct hda_verb alc882_adc1_init_verbs[] = {
 	/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
 	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
@@ -8164,26 +8438,26 @@ static struct hda_verb alc882_adc1_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc882_eapd_verbs[] = {
+static const struct hda_verb alc882_eapd_verbs[] = {
 	/* change to EAPD mode */
 	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
 	{0x20, AC_VERB_SET_PROC_COEF, 0x3060},
 	{ }
 };
 
-static struct hda_verb alc889_eapd_verbs[] = {
+static const struct hda_verb alc889_eapd_verbs[] = {
 	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
 	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
 	{ }
 };
 
-static struct hda_verb alc_hp15_unsol_verbs[] = {
+static const struct hda_verb alc_hp15_unsol_verbs[] = {
 	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{}
 };
 
-static struct hda_verb alc885_init_verbs[] = {
+static const struct hda_verb alc885_init_verbs[] = {
 	/* Front mixer: unmute input/output amp left and right (volume = 0) */
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
@@ -8242,7 +8516,7 @@ static struct hda_verb alc885_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc885_init_input_verbs[] = {
+static const struct hda_verb alc885_init_input_verbs[] = {
 	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
 	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
@@ -8251,7 +8525,7 @@ static struct hda_verb alc885_init_input_verbs[] = {
 
 
 /* Unmute Selector 24h and set the default input to front mic */
-static struct hda_verb alc889_init_input_verbs[] = {
+static const struct hda_verb alc889_init_input_verbs[] = {
 	{0x24, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{ }
@@ -8261,7 +8535,7 @@ static struct hda_verb alc889_init_input_verbs[] = {
 #define alc883_init_verbs	alc882_base_init_verbs
 
 /* Mac Pro test */
-static struct snd_kcontrol_new alc882_macpro_mixer[] = {
+static const struct snd_kcontrol_new alc882_macpro_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x18, 0x0, HDA_OUTPUT),
@@ -8274,7 +8548,7 @@ static struct snd_kcontrol_new alc882_macpro_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc882_macpro_init_verbs[] = {
+static const struct hda_verb alc882_macpro_init_verbs[] = {
 	/* Front mixer: unmute input/output amp left and right (volume = 0) */
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
@@ -8326,7 +8600,7 @@ static struct hda_verb alc882_macpro_init_verbs[] = {
 };
 
 /* Macbook 5,1 */
-static struct hda_verb alc885_mb5_init_verbs[] = {
+static const struct hda_verb alc885_mb5_init_verbs[] = {
 	/* DACs */
 	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -8375,7 +8649,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = {
 };
 
 /* Macmini 3,1 */
-static struct hda_verb alc885_macmini3_init_verbs[] = {
+static const struct hda_verb alc885_macmini3_init_verbs[] = {
 	/* DACs */
 	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -8422,7 +8696,7 @@ static struct hda_verb alc885_macmini3_init_verbs[] = {
 };
 
 
-static struct hda_verb alc885_mba21_init_verbs[] = {
+static const struct hda_verb alc885_mba21_init_verbs[] = {
 	/*Internal and HP Speaker Mixer*/
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -8445,7 +8719,7 @@ static struct hda_verb alc885_mba21_init_verbs[] = {
 
 
 /* Macbook Pro rev3 */
-static struct hda_verb alc885_mbp3_init_verbs[] = {
+static const struct hda_verb alc885_mbp3_init_verbs[] = {
 	/* Front mixer: unmute input/output amp left and right (volume = 0) */
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
@@ -8509,7 +8783,7 @@ static struct hda_verb alc885_mbp3_init_verbs[] = {
 };
 
 /* iMac 9,1 */
-static struct hda_verb alc885_imac91_init_verbs[] = {
+static const struct hda_verb alc885_imac91_init_verbs[] = {
 	/* Internal Speaker Pin (0x0c) */
 	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
 	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -8564,14 +8838,14 @@ static struct hda_verb alc885_imac91_init_verbs[] = {
 };
 
 /* iMac 24 mixer. */
-static struct snd_kcontrol_new alc885_imac24_mixer[] = {
+static const struct snd_kcontrol_new alc885_imac24_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x00, HDA_INPUT),
 	{ } /* end */
 };
 
 /* iMac 24 init verbs. */
-static struct hda_verb alc885_imac24_init_verbs[] = {
+static const struct hda_verb alc885_imac24_init_verbs[] = {
 	/* Internal speakers: output 0 (0x0c) */
 	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -8599,6 +8873,8 @@ static void alc885_imac24_setup(struct hda_codec *codec)
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x18;
 	spec->autocfg.speaker_pins[1] = 0x1a;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 #define alc885_mb5_setup	alc885_imac24_setup
@@ -8611,6 +8887,8 @@ static void alc885_mba21_setup(struct hda_codec *codec)
 
        spec->autocfg.hp_pins[0] = 0x14;
        spec->autocfg.speaker_pins[0] = 0x18;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 
@@ -8621,6 +8899,8 @@ static void alc885_mbp3_setup(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc885_imac91_setup(struct hda_codec *codec)
@@ -8630,9 +8910,11 @@ static void alc885_imac91_setup(struct hda_codec *codec)
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x18;
 	spec->autocfg.speaker_pins[1] = 0x1a;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
-static struct hda_verb alc882_targa_verbs[] = {
+static const struct hda_verb alc882_targa_verbs[] = {
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 
@@ -8651,7 +8933,7 @@ static struct hda_verb alc882_targa_verbs[] = {
 static void alc882_targa_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	alc_automute_amp(codec);
+	alc_hp_automute(codec);
 	snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
 				  spec->jack_present ? 1 : 3);
 }
@@ -8662,6 +8944,8 @@ static void alc882_targa_setup(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x1b;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -8670,7 +8954,7 @@ static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res)
 		alc882_targa_automute(codec);
 }
 
-static struct hda_verb alc882_asus_a7j_verbs[] = {
+static const struct hda_verb alc882_asus_a7j_verbs[] = {
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 
@@ -8688,7 +8972,7 @@ static struct hda_verb alc882_asus_a7j_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc882_asus_a7m_verbs[] = {
+static const struct hda_verb alc882_asus_a7m_verbs[] = {
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 
@@ -8749,13 +9033,13 @@ static void alc885_macpro_init_hook(struct hda_codec *codec)
 static void alc885_imac24_init_hook(struct hda_codec *codec)
 {
 	alc885_macpro_init_hook(codec);
-	alc_automute_amp(codec);
+	alc_hp_automute(codec);
 }
 
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb alc883_auto_init_verbs[] = {
+static const struct hda_verb alc883_auto_init_verbs[] = {
 	/*
 	 * Unmute ADC0-2 and set the default input to mic-in
 	 */
@@ -8795,7 +9079,7 @@ static struct hda_verb alc883_auto_init_verbs[] = {
 };
 
 /* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */
-static struct hda_verb alc889A_mb31_ch2_init[] = {
+static const struct hda_verb alc889A_mb31_ch2_init[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},             /* HP as front */
 	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
 	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},    /* Line as input */
@@ -8804,7 +9088,7 @@ static struct hda_verb alc889A_mb31_ch2_init[] = {
 };
 
 /* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */
-static struct hda_verb alc889A_mb31_ch4_init[] = {
+static const struct hda_verb alc889A_mb31_ch4_init[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},             /* HP as front */
 	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
 	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},   /* Line as output */
@@ -8813,7 +9097,7 @@ static struct hda_verb alc889A_mb31_ch4_init[] = {
 };
 
 /* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */
-static struct hda_verb alc889A_mb31_ch5_init[] = {
+static const struct hda_verb alc889A_mb31_ch5_init[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},             /* HP as rear */
 	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
 	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},    /* Line as input */
@@ -8822,7 +9106,7 @@ static struct hda_verb alc889A_mb31_ch5_init[] = {
 };
 
 /* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */
-static struct hda_verb alc889A_mb31_ch6_init[] = {
+static const struct hda_verb alc889A_mb31_ch6_init[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},             /* HP as front */
 	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},   /* Subwoofer off */
 	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},   /* Line as output */
@@ -8830,14 +9114,14 @@ static struct hda_verb alc889A_mb31_ch6_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc889A_mb31_6ch_modes[4] = {
+static const struct hda_channel_mode alc889A_mb31_6ch_modes[4] = {
 	{ 2, alc889A_mb31_ch2_init },
 	{ 4, alc889A_mb31_ch4_init },
 	{ 5, alc889A_mb31_ch5_init },
 	{ 6, alc889A_mb31_ch6_init },
 };
 
-static struct hda_verb alc883_medion_eapd_verbs[] = {
+static const struct hda_verb alc883_medion_eapd_verbs[] = {
         /* eanable EAPD on medion laptop */
 	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
 	{0x20, AC_VERB_SET_PROC_COEF, 0x3070},
@@ -8846,7 +9130,7 @@ static struct hda_verb alc883_medion_eapd_verbs[] = {
 
 #define alc883_base_mixer	alc882_base_mixer
 
-static struct snd_kcontrol_new alc883_mitac_mixer[] = {
+static const struct snd_kcontrol_new alc883_mitac_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
@@ -8863,7 +9147,7 @@ static struct snd_kcontrol_new alc883_mitac_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_clevo_m720_mixer[] = {
+static const struct snd_kcontrol_new alc883_clevo_m720_mixer[] = {
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -8877,7 +9161,7 @@ static struct snd_kcontrol_new alc883_clevo_m720_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = {
+static const struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = {
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -8891,7 +9175,7 @@ static struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = {
+static const struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
@@ -8908,7 +9192,7 @@ static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
+static const struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -8931,7 +9215,7 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = {
+static const struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -8955,7 +9239,7 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc885_8ch_intel_mixer[] = {
+static const struct snd_kcontrol_new alc885_8ch_intel_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -8979,7 +9263,7 @@ static struct snd_kcontrol_new alc885_8ch_intel_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_fivestack_mixer[] = {
+static const struct snd_kcontrol_new alc883_fivestack_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -9002,7 +9286,7 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_targa_mixer[] = {
+static const struct snd_kcontrol_new alc883_targa_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
@@ -9023,7 +9307,7 @@ static struct snd_kcontrol_new alc883_targa_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = {
+static const struct snd_kcontrol_new alc883_targa_2ch_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
@@ -9039,7 +9323,7 @@ static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_targa_8ch_mixer[] = {
+static const struct snd_kcontrol_new alc883_targa_8ch_mixer[] = {
 	HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
@@ -9048,7 +9332,7 @@ static struct snd_kcontrol_new alc883_targa_8ch_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = {
+static const struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -9060,7 +9344,7 @@ static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = {
+static const struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = {
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
@@ -9073,7 +9357,7 @@ static struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = {
+static const struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT),
@@ -9083,7 +9367,7 @@ static struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc883_medion_wim2160_verbs[] = {
+static const struct hda_verb alc883_medion_wim2160_verbs[] = {
 	/* Unmute front mixer */
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
@@ -9107,9 +9391,11 @@ static void alc883_medion_wim2160_setup(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x1a;
 	spec->autocfg.speaker_pins[0] = 0x15;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
-static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
+static const struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
@@ -9121,7 +9407,7 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = {
+static const struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
@@ -9134,7 +9420,7 @@ static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = {
+static const struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
@@ -9159,7 +9445,7 @@ static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc889A_mb31_mixer[] = {
+static const struct snd_kcontrol_new alc889A_mb31_mixer[] = {
 	/* Output mixers */
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
@@ -9185,7 +9471,7 @@ static struct snd_kcontrol_new alc889A_mb31_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_vaiott_mixer[] = {
+static const struct snd_kcontrol_new alc883_vaiott_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
@@ -9195,7 +9481,7 @@ static struct snd_kcontrol_new alc883_vaiott_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_bind_ctls alc883_bind_cap_vol = {
+static const struct hda_bind_ctls alc883_bind_cap_vol = {
 	.ops = &snd_hda_bind_vol,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
@@ -9204,7 +9490,7 @@ static struct hda_bind_ctls alc883_bind_cap_vol = {
 	},
 };
 
-static struct hda_bind_ctls alc883_bind_cap_switch = {
+static const struct hda_bind_ctls alc883_bind_cap_switch = {
 	.ops = &snd_hda_bind_sw,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
@@ -9213,7 +9499,7 @@ static struct hda_bind_ctls alc883_bind_cap_switch = {
 	},
 };
 
-static struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = {
+static const struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
@@ -9225,7 +9511,7 @@ static struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = {
+static const struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = {
 	HDA_BIND_VOL("Capture Volume", &alc883_bind_cap_vol),
 	HDA_BIND_SW("Capture Switch", &alc883_bind_cap_switch),
 	{
@@ -9240,7 +9526,7 @@ static struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_chmode_mixer[] = {
+static const struct snd_kcontrol_new alc883_chmode_mixer[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Channel Mode",
@@ -9259,9 +9545,11 @@ static void alc883_mitac_setup(struct hda_codec *codec)
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x17;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
-static struct hda_verb alc883_mitac_verbs[] = {
+static const struct hda_verb alc883_mitac_verbs[] = {
 	/* HP */
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -9276,7 +9564,7 @@ static struct hda_verb alc883_mitac_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc883_clevo_m540r_verbs[] = {
+static const struct hda_verb alc883_clevo_m540r_verbs[] = {
 	/* HP */
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -9292,7 +9580,7 @@ static struct hda_verb alc883_clevo_m540r_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc883_clevo_m720_verbs[] = {
+static const struct hda_verb alc883_clevo_m720_verbs[] = {
 	/* HP */
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -9307,7 +9595,7 @@ static struct hda_verb alc883_clevo_m720_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = {
+static const struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = {
 	/* HP */
 	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -9321,7 +9609,7 @@ static struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc883_targa_verbs[] = {
+static const struct hda_verb alc883_targa_verbs[] = {
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 
@@ -9350,14 +9638,14 @@ static struct hda_verb alc883_targa_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc883_lenovo_101e_verbs[] = {
+static const struct hda_verb alc883_lenovo_101e_verbs[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_FRONT_EVENT|AC_USRSP_EN},
         {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT|AC_USRSP_EN},
 	{ } /* end */
 };
 
-static struct hda_verb alc883_lenovo_nb0763_verbs[] = {
+static const struct hda_verb alc883_lenovo_nb0763_verbs[] = {
         {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
         {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
@@ -9365,7 +9653,7 @@ static struct hda_verb alc883_lenovo_nb0763_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc888_lenovo_ms7195_verbs[] = {
+static const struct hda_verb alc888_lenovo_ms7195_verbs[] = {
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -9374,7 +9662,7 @@ static struct hda_verb alc888_lenovo_ms7195_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc883_haier_w66_verbs[] = {
+static const struct hda_verb alc883_haier_w66_verbs[] = {
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 
@@ -9387,7 +9675,7 @@ static struct hda_verb alc883_haier_w66_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc888_lenovo_sky_verbs[] = {
+static const struct hda_verb alc888_lenovo_sky_verbs[] = {
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -9399,12 +9687,12 @@ static struct hda_verb alc888_lenovo_sky_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc888_6st_dell_verbs[] = {
+static const struct hda_verb alc888_6st_dell_verbs[] = {
 	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
 	{ }
 };
 
-static struct hda_verb alc883_vaiott_verbs[] = {
+static const struct hda_verb alc883_vaiott_verbs[] = {
 	/* HP */
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -9423,9 +9711,11 @@ static void alc888_3st_hp_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x16;
 	spec->autocfg.speaker_pins[2] = 0x18;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
-static struct hda_verb alc888_3st_hp_verbs[] = {
+static const struct hda_verb alc888_3st_hp_verbs[] = {
 	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},	/* Front: output 0 (0x0c) */
 	{0x16, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Rear : output 1 (0x0d) */
 	{0x18, AC_VERB_SET_CONNECT_SEL, 0x02},	/* CLFE : output 2 (0x0e) */
@@ -9436,7 +9726,7 @@ static struct hda_verb alc888_3st_hp_verbs[] = {
 /*
  * 2ch mode
  */
-static struct hda_verb alc888_3st_hp_2ch_init[] = {
+static const struct hda_verb alc888_3st_hp_2ch_init[] = {
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
 	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
@@ -9447,7 +9737,7 @@ static struct hda_verb alc888_3st_hp_2ch_init[] = {
 /*
  * 4ch mode
  */
-static struct hda_verb alc888_3st_hp_4ch_init[] = {
+static const struct hda_verb alc888_3st_hp_4ch_init[] = {
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
 	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
@@ -9459,7 +9749,7 @@ static struct hda_verb alc888_3st_hp_4ch_init[] = {
 /*
  * 6ch mode
  */
-static struct hda_verb alc888_3st_hp_6ch_init[] = {
+static const struct hda_verb alc888_3st_hp_6ch_init[] = {
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
 	{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
@@ -9469,39 +9759,21 @@ static struct hda_verb alc888_3st_hp_6ch_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc888_3st_hp_modes[3] = {
+static const struct hda_channel_mode alc888_3st_hp_modes[3] = {
 	{ 2, alc888_3st_hp_2ch_init },
 	{ 4, alc888_3st_hp_4ch_init },
 	{ 6, alc888_3st_hp_6ch_init },
 };
 
-/* toggle front-jack and RCA according to the hp-jack state */
-static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec)
+static void alc888_lenovo_ms7195_setup(struct hda_codec *codec)
 {
- 	unsigned int present = snd_hda_jack_detect(codec, 0x1b);
-
-	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-	snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-/* toggle RCA according to the front-jack state */
-static void alc888_lenovo_ms7195_rca_automute(struct hda_codec *codec)
-{
- 	unsigned int present = snd_hda_jack_detect(codec, 0x14);
-
-	snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
+	struct alc_spec *spec = codec->spec;
 
-static void alc883_lenovo_ms7195_unsol_event(struct hda_codec *codec,
-					     unsigned int res)
-{
-	if ((res >> 26) == ALC880_HP_EVENT)
-		alc888_lenovo_ms7195_front_automute(codec);
-	if ((res >> 26) == ALC880_FRONT_EVENT)
-		alc888_lenovo_ms7195_rca_automute(codec);
+	spec->autocfg.hp_pins[0] = 0x1b;
+	spec->autocfg.line_out_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[0] = 0x15;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 /* toggle speaker-output according to the hp-jack state */
@@ -9511,6 +9783,8 @@ static void alc883_lenovo_nb0763_setup(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 /* toggle speaker-output according to the hp-jack state */
@@ -9523,11 +9797,13 @@ static void alc883_clevo_m720_setup(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc883_clevo_m720_init_hook(struct hda_codec *codec)
 {
-	alc_automute_amp(codec);
+	alc_hp_automute(codec);
 	alc88x_simple_mic_automute(codec);
 }
 
@@ -9539,7 +9815,7 @@ static void alc883_clevo_m720_unsol_event(struct hda_codec *codec,
 		alc88x_simple_mic_automute(codec);
 		break;
 	default:
-		alc_automute_amp_unsol_event(codec, res);
+		alc_sku_unsol_event(codec, res);
 		break;
 	}
 }
@@ -9551,6 +9827,8 @@ static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc883_haier_w66_setup(struct hda_codec *codec)
@@ -9559,33 +9837,21 @@ static void alc883_haier_w66_setup(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
-static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
-{
-	int bits = snd_hda_jack_detect(codec, 0x14) ? HDA_AMP_MUTE : 0;
-
-	snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
-}
-
-static void alc883_lenovo_101e_all_automute(struct hda_codec *codec)
+static void alc883_lenovo_101e_setup(struct hda_codec *codec)
 {
-	int bits = snd_hda_jack_detect(codec, 0x1b) ? HDA_AMP_MUTE : 0;
-
-	snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
-	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
-}
+	struct alc_spec *spec = codec->spec;
 
-static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
-{
-	if ((res >> 26) == ALC880_HP_EVENT)
-		alc883_lenovo_101e_all_automute(codec);
-	if ((res >> 26) == ALC880_FRONT_EVENT)
-		alc883_lenovo_101e_ispeaker_automute(codec);
+	spec->autocfg.hp_pins[0] = 0x1b;
+	spec->autocfg.line_out_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[0] = 0x15;
+	spec->automute = 1;
+	spec->detect_line = 1;
+	spec->automute_lines = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 /* toggle speaker-output according to the hp-jack state */
@@ -9596,9 +9862,11 @@ static void alc883_acer_aspire_setup(struct hda_codec *codec)
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[1] = 0x16;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
-static struct hda_verb alc883_acer_eapd_verbs[] = {
+static const struct hda_verb alc883_acer_eapd_verbs[] = {
 	/* HP Pin: output 0 (0x0c) */
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -9625,6 +9893,8 @@ static void alc888_6st_dell_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[1] = 0x15;
 	spec->autocfg.speaker_pins[2] = 0x16;
 	spec->autocfg.speaker_pins[3] = 0x17;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc888_lenovo_sky_setup(struct hda_codec *codec)
@@ -9637,6 +9907,8 @@ static void alc888_lenovo_sky_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[2] = 0x16;
 	spec->autocfg.speaker_pins[3] = 0x17;
 	spec->autocfg.speaker_pins[4] = 0x1a;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc883_vaiott_setup(struct hda_codec *codec)
@@ -9646,9 +9918,11 @@ static void alc883_vaiott_setup(struct hda_codec *codec)
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x17;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
-static struct hda_verb alc888_asus_m90v_verbs[] = {
+static const struct hda_verb alc888_asus_m90v_verbs[] = {
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
@@ -9671,9 +9945,11 @@ static void alc883_mode2_setup(struct hda_codec *codec)
 	spec->ext_mic.mux_idx = 0;
 	spec->int_mic.mux_idx = 1;
 	spec->auto_mic = 1;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
-static struct hda_verb alc888_asus_eee1601_verbs[] = {
+static const struct hda_verb alc888_asus_eee1601_verbs[] = {
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
@@ -9692,10 +9968,10 @@ static void alc883_eee1601_inithook(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x1b;
-	alc_automute_pin(codec);
+	alc_hp_automute(codec);
 }
 
-static struct hda_verb alc889A_mb31_verbs[] = {
+static const struct hda_verb alc889A_mb31_verbs[] = {
 	/* Init rear pin (used as headphone output) */
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},    /* Apple Headphones */
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},           /* Connect to front */
@@ -9741,11 +10017,11 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
 #define alc882_pcm_digital_playback	alc880_pcm_digital_playback
 #define alc882_pcm_digital_capture	alc880_pcm_digital_capture
 
-static hda_nid_t alc883_slave_dig_outs[] = {
+static const hda_nid_t alc883_slave_dig_outs[] = {
 	ALC1200_DIGOUT_NID, 0,
 };
 
-static hda_nid_t alc1200_slave_dig_outs[] = {
+static const hda_nid_t alc1200_slave_dig_outs[] = {
 	ALC883_DIGOUT_NID, 0,
 };
 
@@ -9804,7 +10080,7 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = {
 	[ALC882_AUTO]		= "auto",
 };
 
-static struct snd_pci_quirk alc882_cfg_tbl[] = {
+static const struct snd_pci_quirk alc882_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG),
 
 	SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
@@ -9863,6 +10139,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
 	SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL),
 	SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch),
+	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
 
 	SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
 	SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
@@ -9930,7 +10207,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 };
 
 /* codec SSID table for Intel Mac */
-static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
+static const struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC885_MBP3),
 	SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC885_MBP3),
 	SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC885_MBP3),
@@ -9957,7 +10234,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
 	{} /* terminator */
 };
 
-static struct alc_config_preset alc882_presets[] = {
+static const struct alc_config_preset alc882_presets[] = {
 	[ALC882_3ST_DIG] = {
 		.mixers = { alc882_base_mixer },
 		.init_verbs = { alc882_base_init_verbs,
@@ -10013,9 +10290,9 @@ static struct alc_config_preset alc882_presets[] = {
 			.channel_mode = alc885_mba21_ch_modes,
 			.num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
 			.input_mux = &alc882_capture_source,
-			.unsol_event = alc_automute_amp_unsol_event,
+			.unsol_event = alc_sku_unsol_event,
 			.setup = alc885_mba21_setup,
-			.init_hook = alc_automute_amp,
+			.init_hook = alc_hp_automute,
        },
 	[ALC885_MBP3] = {
 		.mixers = { alc885_mbp3_mixer, alc882_chmode_mixer },
@@ -10029,9 +10306,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.input_mux = &alc882_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc885_mbp3_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC885_MB5] = {
 		.mixers = { alc885_mb5_mixer, alc882_chmode_mixer },
@@ -10044,9 +10321,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.input_mux = &mb5_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc885_mb5_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC885_MACMINI3] = {
 		.mixers = { alc885_macmini3_mixer, alc882_chmode_mixer },
@@ -10059,9 +10336,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.input_mux = &macmini3_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc885_macmini3_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC885_MACPRO] = {
 		.mixers = { alc882_macpro_mixer },
@@ -10085,7 +10362,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
 		.channel_mode = alc882_ch_modes,
 		.input_mux = &alc882_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc885_imac24_setup,
 		.init_hook = alc885_imac24_init_hook,
 	},
@@ -10100,9 +10377,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.input_mux = &alc889A_imac91_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc885_imac91_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC882_TARGA] = {
 		.mixers = { alc882_targa_mixer, alc882_chmode_mixer },
@@ -10118,7 +10395,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc882_3ST_6ch_modes,
 		.need_dac_fix = 1,
 		.input_mux = &alc882_capture_source,
-		.unsol_event = alc882_targa_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc882_targa_setup,
 		.init_hook = alc882_targa_automute,
 	},
@@ -10212,8 +10489,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.capsrc_nids = alc889_capsrc_nids,
 		.input_mux = &alc889_capture_source,
 		.setup = alc889_automute_setup,
-		.init_hook = alc_automute_amp,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.init_hook = alc_hp_automute,
+		.unsol_event = alc_sku_unsol_event,
 		.need_dac_fix = 1,
 	},
 	[ALC889_INTEL] = {
@@ -10233,7 +10510,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.input_mux = &alc889_capture_source,
 		.setup = alc889_automute_setup,
 		.init_hook = alc889_intel_init_hook,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.need_dac_fix = 1,
 	},
 	[ALC883_6ST_DIG] = {
@@ -10322,9 +10599,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc883_acer_aspire_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC888_ACER_ASPIRE_4930G] = {
 		.mixers = { alc888_acer_aspire_4930g_mixer,
@@ -10344,9 +10621,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_mux_defs =
 			ARRAY_SIZE(alc888_2_capture_sources),
 		.input_mux = alc888_2_capture_sources,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc888_acer_aspire_4930g_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC888_ACER_ASPIRE_6530G] = {
 		.mixers = { alc888_acer_aspire_6530_mixer },
@@ -10363,9 +10640,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_mux_defs =
 			ARRAY_SIZE(alc888_2_capture_sources),
 		.input_mux = alc888_acer_aspire_6530_sources,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc888_acer_aspire_6530g_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC888_ACER_ASPIRE_8930G] = {
 		.mixers = { alc889_acer_aspire_8930g_mixer,
@@ -10386,9 +10663,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_mux_defs =
 			ARRAY_SIZE(alc889_capture_sources),
 		.input_mux = alc889_capture_sources,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc889_acer_aspire_8930g_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 		.power_hook = alc_power_eapd,
 #endif
@@ -10409,9 +10686,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.need_dac_fix = 1,
 		.const_channel_count = 6,
 		.input_mux = &alc883_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc888_acer_aspire_7730g_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC883_MEDION] = {
 		.mixers = { alc883_fivestack_mixer,
@@ -10438,9 +10715,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc883_medion_wim2160_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC883_LAPTOP_EAPD] = {
 		.mixers = { alc883_base_mixer },
@@ -10490,8 +10767,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_lenovo_101e_capture_source,
-		.unsol_event = alc883_lenovo_101e_unsol_event,
-		.init_hook = alc883_lenovo_101e_all_automute,
+		.setup = alc883_lenovo_101e_setup,
+		.unsol_event = alc_sku_unsol_event,
+		.init_hook = alc_inithook,
 	},
 	[ALC883_LENOVO_NB0763] = {
 		.mixers = { alc883_lenovo_nb0763_mixer },
@@ -10502,9 +10780,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc883_3ST_2ch_modes,
 		.need_dac_fix = 1,
 		.input_mux = &alc883_lenovo_nb0763_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc883_lenovo_nb0763_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC888_LENOVO_MS7195_DIG] = {
 		.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
@@ -10516,8 +10794,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc883_3ST_6ch_modes,
 		.need_dac_fix = 1,
 		.input_mux = &alc883_capture_source,
-		.unsol_event = alc883_lenovo_ms7195_unsol_event,
-		.init_hook = alc888_lenovo_ms7195_front_automute,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc888_lenovo_ms7195_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC883_HAIER_W66] = {
 		.mixers = { alc883_targa_2ch_mixer},
@@ -10528,9 +10807,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc883_haier_w66_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC888_3ST_HP] = {
 		.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
@@ -10541,9 +10820,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc888_3st_hp_modes,
 		.need_dac_fix = 1,
 		.input_mux = &alc883_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc888_3st_hp_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC888_6ST_DELL] = {
 		.mixers = { alc883_base_mixer, alc883_chmode_mixer },
@@ -10555,9 +10834,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
 		.channel_mode = alc883_sixstack_modes,
 		.input_mux = &alc883_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc888_6st_dell_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC883_MITAC] = {
 		.mixers = { alc883_mitac_mixer },
@@ -10567,9 +10846,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc883_mitac_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC883_FUJITSU_PI2515] = {
 		.mixers = { alc883_2ch_fujitsu_pi2515_mixer },
@@ -10581,9 +10860,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_fujitsu_pi2515_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc883_2ch_fujitsu_pi2515_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC888_FUJITSU_XA3530] = {
 		.mixers = { alc888_base_mixer, alc883_chmode_mixer },
@@ -10600,9 +10879,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_mux_defs =
 			ARRAY_SIZE(alc888_2_capture_sources),
 		.input_mux = alc888_2_capture_sources,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc888_fujitsu_xa3530_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC888_LENOVO_SKY] = {
 		.mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer },
@@ -10614,9 +10893,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc883_sixstack_modes,
 		.need_dac_fix = 1,
 		.input_mux = &alc883_lenovo_sky_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc888_lenovo_sky_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC888_ASUS_M90V] = {
 		.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
@@ -10684,9 +10963,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc883_vaiott_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 };
 
@@ -10699,7 +10978,6 @@ enum {
 	PINFIX_LENOVO_Y530,
 	PINFIX_PB_M5210,
 	PINFIX_ACER_ASPIRE_7736,
-	PINFIX_GIGABYTE_880GM,
 };
 
 static const struct alc_fixup alc882_fixups[] = {
@@ -10731,21 +11009,13 @@ static const struct alc_fixup alc882_fixups[] = {
 		.type = ALC_FIXUP_SKU,
 		.v.sku = ALC_FIXUP_SKU_IGNORE,
 	},
-	[PINFIX_GIGABYTE_880GM] = {
-		.type = ALC_FIXUP_PINS,
-		.v.pins = (const struct alc_pincfg[]) {
-			{ 0x14, 0x1114410 }, /* set as speaker */
-			{ }
-		}
-	},
 };
 
-static struct snd_pci_quirk alc882_fixup_tbl[] = {
+static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530),
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
 	SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736),
-	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", PINFIX_GIGABYTE_880GM),
 	{}
 };
 
@@ -10849,6 +11119,11 @@ static void alc882_auto_init_input_src(struct hda_codec *codec)
 		const struct hda_input_mux *imux;
 		int conns, mute, idx, item;
 
+		/* mute ADC */
+		snd_hda_codec_write(codec, spec->adc_nids[c], 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE,
+				    AMP_IN_MUTE(0));
+
 		conns = snd_hda_get_connections(codec, nid, conn_list,
 						ARRAY_SIZE(conn_list));
 		if (conns < 0)
@@ -10928,7 +11203,7 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec)
 static int alc882_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	static hda_nid_t alc882_ignore[] = { 0x1d, 0 };
+	static const hda_nid_t alc882_ignore[] = { 0x1d, 0 };
 	int err;
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
@@ -10941,6 +11216,9 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
 	err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
+	err = alc_auto_add_multi_channel_mode(codec);
+	if (err < 0)
+		return err;
 	err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
@@ -11142,14 +11420,14 @@ static int patch_alc882(struct hda_codec *codec)
 #define alc262_modes		alc260_modes
 #define alc262_capture_source	alc882_capture_source
 
-static hda_nid_t alc262_dmic_adc_nids[1] = {
+static const hda_nid_t alc262_dmic_adc_nids[1] = {
 	/* ADC0 */
 	0x09
 };
 
-static hda_nid_t alc262_dmic_capsrc_nids[1] = { 0x22 };
+static const hda_nid_t alc262_dmic_capsrc_nids[1] = { 0x22 };
 
-static struct snd_kcontrol_new alc262_base_mixer[] = {
+static const struct snd_kcontrol_new alc262_base_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
@@ -11170,71 +11448,30 @@ static struct snd_kcontrol_new alc262_base_mixer[] = {
 };
 
 /* update HP, line and mono-out pins according to the master switch */
-static void alc262_hp_master_update(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	int val = spec->master_sw;
-
-	/* HP & line-out */
-	snd_hda_codec_write_cache(codec, 0x1b, 0,
-				  AC_VERB_SET_PIN_WIDGET_CONTROL,
-				  val ? PIN_HP : 0);
-	snd_hda_codec_write_cache(codec, 0x15, 0,
-				  AC_VERB_SET_PIN_WIDGET_CONTROL,
-				  val ? PIN_HP : 0);
-	/* mono (speaker) depending on the HP jack sense */
-	val = val && !spec->jack_present;
-	snd_hda_codec_write_cache(codec, 0x16, 0,
-				  AC_VERB_SET_PIN_WIDGET_CONTROL,
-				  val ? PIN_OUT : 0);
-}
+#define alc262_hp_master_update		alc260_hp_master_update
 
-static void alc262_hp_bpc_automute(struct hda_codec *codec)
+static void alc262_hp_bpc_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	spec->jack_present = snd_hda_jack_detect(codec, 0x1b);
-	alc262_hp_master_update(codec);
-}
-
-static void alc262_hp_bpc_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	if ((res >> 26) != ALC880_HP_EVENT)
-		return;
-	alc262_hp_bpc_automute(codec);
+	spec->autocfg.hp_pins[0] = 0x1b;
+	spec->autocfg.speaker_pins[0] = 0x16;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
 }
 
-static void alc262_hp_wildwest_automute(struct hda_codec *codec)
+static void alc262_hp_wildwest_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	spec->jack_present = snd_hda_jack_detect(codec, 0x15);
-	alc262_hp_master_update(codec);
-}
-
-static void alc262_hp_wildwest_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
-{
-	if ((res >> 26) != ALC880_HP_EVENT)
-		return;
-	alc262_hp_wildwest_automute(codec);
+	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x16;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
 }
 
 #define alc262_hp_master_sw_get		alc260_hp_master_sw_get
-
-static int alc262_hp_master_sw_put(struct snd_kcontrol *kcontrol,
-				   struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct alc_spec *spec = codec->spec;
-	int val = !!*ucontrol->value.integer.value;
-
-	if (val == spec->master_sw)
-		return 0;
-	spec->master_sw = val;
-	alc262_hp_master_update(codec);
-	return 1;
-}
+#define alc262_hp_master_sw_put		alc260_hp_master_sw_put
 
 #define ALC262_HP_MASTER_SWITCH					\
 	{							\
@@ -11251,7 +11488,7 @@ static int alc262_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 	}
 
 
-static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
+static const struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
 	ALC262_HP_MASTER_SWITCH,
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
@@ -11275,7 +11512,7 @@ static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = {
+static const struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = {
 	ALC262_HP_MASTER_SWITCH,
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
@@ -11295,7 +11532,7 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = {
+static const struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = {
 	HDA_CODEC_VOLUME("Rear Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Rear Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Rear Mic Boost Volume", 0x18, 0, HDA_INPUT),
@@ -11309,9 +11546,11 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
 }
 
-static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
+static const struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -11322,7 +11561,7 @@ static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc262_hp_t5735_verbs[] = {
+static const struct hda_verb alc262_hp_t5735_verbs[] = {
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 
@@ -11330,7 +11569,7 @@ static struct hda_verb alc262_hp_t5735_verbs[] = {
 	{ }
 };
 
-static struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = {
+static const struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = {
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
@@ -11340,7 +11579,7 @@ static struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc262_hp_rp5700_verbs[] = {
+static const struct hda_verb alc262_hp_rp5700_verbs[] = {
 	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
@@ -11354,7 +11593,7 @@ static struct hda_verb alc262_hp_rp5700_verbs[] = {
 	{}
 };
 
-static struct hda_input_mux alc262_hp_rp5700_capture_source = {
+static const struct hda_input_mux alc262_hp_rp5700_capture_source = {
 	.num_items = 1,
 	.items = {
 		{ "Line", 0x1 },
@@ -11362,44 +11601,9 @@ static struct hda_input_mux alc262_hp_rp5700_capture_source = {
 };
 
 /* bind hp and internal speaker mute (with plug check) as master switch */
-static void alc262_hippo_master_update(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	hda_nid_t hp_nid = spec->autocfg.hp_pins[0];
-	hda_nid_t line_nid = spec->autocfg.line_out_pins[0];
-	hda_nid_t speaker_nid = spec->autocfg.speaker_pins[0];
-	unsigned int mute;
-
-	/* HP */
-	mute = spec->master_sw ? 0 : HDA_AMP_MUTE;
-	snd_hda_codec_amp_stereo(codec, hp_nid, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, mute);
-	/* mute internal speaker per jack sense */
-	if (spec->jack_present)
-		mute = HDA_AMP_MUTE;
-	if (line_nid)
-		snd_hda_codec_amp_stereo(codec, line_nid, HDA_OUTPUT, 0,
-					 HDA_AMP_MUTE, mute);
-	if (speaker_nid && speaker_nid != line_nid)
-		snd_hda_codec_amp_stereo(codec, speaker_nid, HDA_OUTPUT, 0,
-					 HDA_AMP_MUTE, mute);
-}
-
+#define alc262_hippo_master_update	alc262_hp_master_update
 #define alc262_hippo_master_sw_get	alc262_hp_master_sw_get
-
-static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol,
-				      struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct alc_spec *spec = codec->spec;
-	int val = !!*ucontrol->value.integer.value;
-
-	if (val == spec->master_sw)
-		return 0;
-	spec->master_sw = val;
-	alc262_hippo_master_update(codec);
-	return 1;
-}
+#define alc262_hippo_master_sw_put	alc262_hp_master_sw_put
 
 #define ALC262_HIPPO_MASTER_SWITCH				\
 	{							\
@@ -11416,7 +11620,7 @@ static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol,
 			     (SUBDEV_SPEAKER(0) << 16), \
 	}
 
-static struct snd_kcontrol_new alc262_hippo_mixer[] = {
+static const struct snd_kcontrol_new alc262_hippo_mixer[] = {
 	ALC262_HIPPO_MASTER_SWITCH,
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
@@ -11433,7 +11637,7 @@ static struct snd_kcontrol_new alc262_hippo_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc262_hippo1_mixer[] = {
+static const struct snd_kcontrol_new alc262_hippo1_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	ALC262_HIPPO_MASTER_SWITCH,
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
@@ -11450,28 +11654,14 @@ static struct snd_kcontrol_new alc262_hippo1_mixer[] = {
 };
 
 /* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc262_hippo_automute(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	hda_nid_t hp_nid = spec->autocfg.hp_pins[0];
-
-	spec->jack_present = snd_hda_jack_detect(codec, hp_nid);
-	alc262_hippo_master_update(codec);
-}
-
-static void alc262_hippo_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	if ((res >> 26) != ALC880_HP_EVENT)
-		return;
-	alc262_hippo_automute(codec);
-}
-
 static void alc262_hippo_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc262_hippo1_setup(struct hda_codec *codec)
@@ -11480,10 +11670,12 @@ static void alc262_hippo1_setup(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 
-static struct snd_kcontrol_new alc262_sony_mixer[] = {
+static const struct snd_kcontrol_new alc262_sony_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	ALC262_HIPPO_MASTER_SWITCH,
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
@@ -11493,7 +11685,7 @@ static struct snd_kcontrol_new alc262_sony_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc262_benq_t31_mixer[] = {
+static const struct snd_kcontrol_new alc262_benq_t31_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	ALC262_HIPPO_MASTER_SWITCH,
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
@@ -11504,7 +11696,7 @@ static struct snd_kcontrol_new alc262_benq_t31_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc262_tyan_mixer[] = {
+static const struct snd_kcontrol_new alc262_tyan_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Aux Playback Volume", 0x0b, 0x06, HDA_INPUT),
@@ -11520,7 +11712,7 @@ static struct snd_kcontrol_new alc262_tyan_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc262_tyan_verbs[] = {
+static const struct hda_verb alc262_tyan_verbs[] = {
 	/* Headphone automute */
 	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -11542,6 +11734,8 @@ static void alc262_tyan_setup(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x15;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 
@@ -11551,7 +11745,7 @@ static void alc262_tyan_setup(struct hda_codec *codec)
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb alc262_init_verbs[] = {
+static const struct hda_verb alc262_init_verbs[] = {
 	/*
 	 * Unmute ADC0-2 and set the default input to mic-in
 	 */
@@ -11627,13 +11821,13 @@ static struct hda_verb alc262_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc262_eapd_verbs[] = {
+static const struct hda_verb alc262_eapd_verbs[] = {
 	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
 	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
 	{ }
 };
 
-static struct hda_verb alc262_hippo1_unsol_verbs[] = {
+static const struct hda_verb alc262_hippo1_unsol_verbs[] = {
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
 	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
@@ -11643,7 +11837,7 @@ static struct hda_verb alc262_hippo1_unsol_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc262_sony_unsol_verbs[] = {
+static const struct hda_verb alc262_sony_unsol_verbs[] = {
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},	// Front Mic
@@ -11653,7 +11847,7 @@ static struct hda_verb alc262_sony_unsol_verbs[] = {
 	{}
 };
 
-static struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = {
+static const struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = {
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
@@ -11662,7 +11856,7 @@ static struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc262_toshiba_s06_verbs[] = {
+static const struct hda_verb alc262_toshiba_s06_verbs[] = {
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -11685,6 +11879,8 @@ static void alc262_toshiba_s06_setup(struct hda_codec *codec)
 	spec->int_mic.pin = 0x12;
 	spec->int_mic.mux_idx = 9;
 	spec->auto_mic = 1;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
 }
 
 /*
@@ -11694,7 +11890,7 @@ static void alc262_toshiba_s06_setup(struct hda_codec *codec)
  *  0x18 = external mic
  */
 
-static struct snd_kcontrol_new alc262_nec_mixer[] = {
+static const struct snd_kcontrol_new alc262_nec_mixer[] = {
 	HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 0, 0x0, HDA_OUTPUT),
 
@@ -11707,7 +11903,7 @@ static struct snd_kcontrol_new alc262_nec_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc262_nec_verbs[] = {
+static const struct hda_verb alc262_nec_verbs[] = {
 	/* Unmute Speaker */
 	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 
@@ -11730,7 +11926,7 @@ static struct hda_verb alc262_nec_verbs[] = {
 
 #define ALC_HP_EVENT	0x37
 
-static struct hda_verb alc262_fujitsu_unsol_verbs[] = {
+static const struct hda_verb alc262_fujitsu_unsol_verbs[] = {
 	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
@@ -11738,20 +11934,20 @@ static struct hda_verb alc262_fujitsu_unsol_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc262_lenovo_3000_unsol_verbs[] = {
+static const struct hda_verb alc262_lenovo_3000_unsol_verbs[] = {
 	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{}
 };
 
-static struct hda_verb alc262_lenovo_3000_init_verbs[] = {
+static const struct hda_verb alc262_lenovo_3000_init_verbs[] = {
 	/* Front Mic pin: input vref at 50% */
 	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{}
 };
 
-static struct hda_input_mux alc262_fujitsu_capture_source = {
+static const struct hda_input_mux alc262_fujitsu_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic", 0x0 },
@@ -11760,7 +11956,7 @@ static struct hda_input_mux alc262_fujitsu_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc262_HP_capture_source = {
+static const struct hda_input_mux alc262_HP_capture_source = {
 	.num_items = 5,
 	.items = {
 		{ "Mic", 0x0 },
@@ -11771,7 +11967,7 @@ static struct hda_input_mux alc262_HP_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc262_HP_D7000_capture_source = {
+static const struct hda_input_mux alc262_HP_D7000_capture_source = {
 	.num_items = 4,
 	.items = {
 		{ "Mic", 0x0 },
@@ -11781,44 +11977,19 @@ static struct hda_input_mux alc262_HP_D7000_capture_source = {
 	},
 };
 
-/* mute/unmute internal speaker according to the hp jacks and mute state */
-static void alc262_fujitsu_automute(struct hda_codec *codec, int force)
+static void alc262_fujitsu_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	unsigned int mute;
 
-	if (force || !spec->sense_updated) {
-		spec->jack_present = snd_hda_jack_detect(codec, 0x14) ||
-				     snd_hda_jack_detect(codec, 0x1b);
-		spec->sense_updated = 1;
-	}
-	/* unmute internal speaker only if both HPs are unplugged and
-	 * master switch is on
-	 */
-	if (spec->jack_present)
-		mute = HDA_AMP_MUTE;
-	else
-		mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0);
-	snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, mute);
-}
-
-/* unsolicited event for HP jack sensing */
-static void alc262_fujitsu_unsol_event(struct hda_codec *codec,
-				       unsigned int res)
-{
-	if ((res >> 26) != ALC_HP_EVENT)
-		return;
-	alc262_fujitsu_automute(codec, 1);
-}
-
-static void alc262_fujitsu_init_hook(struct hda_codec *codec)
-{
-	alc262_fujitsu_automute(codec, 1);
+	spec->autocfg.hp_pins[0] = 0x14;
+	spec->autocfg.hp_pins[1] = 0x1b;
+	spec->autocfg.speaker_pins[0] = 0x15;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 /* bind volumes of both NID 0x0c and 0x0d */
-static struct hda_bind_ctls alc262_fujitsu_bind_master_vol = {
+static const struct hda_bind_ctls alc262_fujitsu_bind_master_vol = {
 	.ops = &snd_hda_bind_vol,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT),
@@ -11827,78 +11998,15 @@ static struct hda_bind_ctls alc262_fujitsu_bind_master_vol = {
 	},
 };
 
-/* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc262_lenovo_3000_automute(struct hda_codec *codec, int force)
-{
-	struct alc_spec *spec = codec->spec;
-	unsigned int mute;
-
-	if (force || !spec->sense_updated) {
-		spec->jack_present = snd_hda_jack_detect(codec, 0x1b);
-		spec->sense_updated = 1;
-	}
-	if (spec->jack_present) {
-		/* mute internal speaker */
-		snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
-					 HDA_AMP_MUTE, HDA_AMP_MUTE);
-		snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
-					 HDA_AMP_MUTE, HDA_AMP_MUTE);
-	} else {
-		/* unmute internal speaker if necessary */
-		mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0);
-		snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
-					 HDA_AMP_MUTE, mute);
-		snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
-					 HDA_AMP_MUTE, mute);
-	}
-}
-
-/* unsolicited event for HP jack sensing */
-static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec,
-				       unsigned int res)
-{
-	if ((res >> 26) != ALC_HP_EVENT)
-		return;
-	alc262_lenovo_3000_automute(codec, 1);
-}
-
-static int amp_stereo_mute_update(struct hda_codec *codec, hda_nid_t nid,
-				  int dir, int idx, long *valp)
-{
-	int i, change = 0;
-
-	for (i = 0; i < 2; i++, valp++)
-		change |= snd_hda_codec_amp_update(codec, nid, i, dir, idx,
-						   HDA_AMP_MUTE,
-						   *valp ? 0 : HDA_AMP_MUTE);
-	return change;
-}
-
-/* bind hp and internal speaker mute (with plug check) */
-static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
-					 struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	long *valp = ucontrol->value.integer.value;
-	int change;
-
-	change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
-	change |= amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
-	if (change)
-		alc262_fujitsu_automute(codec, 0);
-	return change;
-}
-
-static struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
+static const struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
 	HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = alc262_fujitsu_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+		.subdevice = HDA_SUBDEV_NID_FLAG | 0x14,
+		.info = snd_ctl_boolean_mono_info,
+		.get = alc262_hp_master_sw_get,
+		.put = alc262_hp_master_sw_put,
 	},
 	{
 		.iface = NID_MAPPING,
@@ -11916,30 +12024,26 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
 	{ } /* end */
 };
 
-/* bind hp and internal speaker mute (with plug check) */
-static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol,
-					 struct snd_ctl_elem_value *ucontrol)
+static void alc262_lenovo_3000_setup(struct hda_codec *codec)
 {
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	long *valp = ucontrol->value.integer.value;
-	int change;
+	struct alc_spec *spec = codec->spec;
 
-	change = amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
-	if (change)
-		alc262_lenovo_3000_automute(codec, 0);
-	return change;
+	spec->autocfg.hp_pins[0] = 0x1b;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[1] = 0x16;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
-static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = {
+static const struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = {
 	HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = alc262_lenovo_3000_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+		.subdevice = HDA_SUBDEV_NID_FLAG | 0x1b,
+		.info = snd_ctl_boolean_mono_info,
+		.get = alc262_hp_master_sw_get,
+		.put = alc262_hp_master_sw_put,
 	},
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
@@ -11952,7 +12056,7 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = {
+static const struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = {
 	HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
 	ALC262_HIPPO_MASTER_SWITCH,
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
@@ -11965,13 +12069,13 @@ static struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = {
 };
 
 /* additional init verbs for Benq laptops */
-static struct hda_verb alc262_EAPD_verbs[] = {
+static const struct hda_verb alc262_EAPD_verbs[] = {
 	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
 	{0x20, AC_VERB_SET_PROC_COEF,  0x3070},
 	{}
 };
 
-static struct hda_verb alc262_benq_t31_EAPD_verbs[] = {
+static const struct hda_verb alc262_benq_t31_EAPD_verbs[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
 
@@ -11981,7 +12085,7 @@ static struct hda_verb alc262_benq_t31_EAPD_verbs[] = {
 };
 
 /* Samsung Q1 Ultra Vista model setup */
-static struct snd_kcontrol_new alc262_ultra_mixer[] = {
+static const struct snd_kcontrol_new alc262_ultra_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
@@ -11991,7 +12095,7 @@ static struct snd_kcontrol_new alc262_ultra_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc262_ultra_verbs[] = {
+static const struct hda_verb alc262_ultra_verbs[] = {
 	/* output mixer */
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
@@ -12054,7 +12158,7 @@ static void alc262_ultra_unsol_event(struct hda_codec *codec,
 	alc262_ultra_automute(codec);
 }
 
-static struct hda_input_mux alc262_ultra_capture_source = {
+static const struct hda_input_mux alc262_ultra_capture_source = {
 	.num_items = 2,
 	.items = {
 		{ "Mic", 0x1 },
@@ -12080,7 +12184,7 @@ static int alc262_ultra_mux_enum_put(struct snd_kcontrol *kcontrol,
 	return ret;
 }
 
-static struct snd_kcontrol_new alc262_ultra_capture_mixer[] = {
+static const struct snd_kcontrol_new alc262_ultra_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
 	{
@@ -12155,9 +12259,9 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
 
 	spec->multiout.num_dacs = 1;	/* only use one dac */
 	spec->multiout.dac_nids = spec->private_dac_nids;
-	spec->multiout.dac_nids[0] = 2;
+	spec->private_dac_nids[0] = 2;
 
-	pfx = alc_get_line_out_pfx(cfg, true);
+	pfx = alc_get_line_out_pfx(spec, true);
 	if (!pfx)
 		pfx = "Front";
 	for (i = 0; i < 2; i++) {
@@ -12211,7 +12315,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb alc262_volume_init_verbs[] = {
+static const struct hda_verb alc262_volume_init_verbs[] = {
 	/*
 	 * Unmute ADC0-2 and set the default input to mic-in
 	 */
@@ -12272,7 +12376,7 @@ static struct hda_verb alc262_volume_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc262_HP_BPC_init_verbs[] = {
+static const struct hda_verb alc262_HP_BPC_init_verbs[] = {
 	/*
 	 * Unmute ADC0-2 and set the default input to mic-in
 	 */
@@ -12376,7 +12480,7 @@ static struct hda_verb alc262_HP_BPC_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = {
+static const struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = {
 	/*
 	 * Unmute ADC0-2 and set the default input to mic-in
 	 */
@@ -12472,7 +12576,7 @@ static struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
+static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
 
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },	/* Front Speaker */
 	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
@@ -12508,7 +12612,7 @@ static const struct alc_fixup alc262_fixups[] = {
 	},
 };
 
-static struct snd_pci_quirk alc262_fixup_tbl[] = {
+static const struct snd_pci_quirk alc262_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", PINFIX_FSC_H270),
 	{}
 };
@@ -12531,7 +12635,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	int err;
-	static hda_nid_t alc262_ignore[] = { 0x1d, 0 };
+	static const hda_nid_t alc262_ignore[] = { 0x1d, 0 };
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
 					   alc262_ignore);
@@ -12616,7 +12720,7 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = {
 	[ALC262_AUTO]		= "auto",
 };
 
-static struct snd_pci_quirk alc262_cfg_tbl[] = {
+static const struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO),
 	SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1200, "HP xw series",
@@ -12668,7 +12772,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	{}
 };
 
-static struct alc_config_preset alc262_presets[] = {
+static const struct alc_config_preset alc262_presets[] = {
 	[ALC262_BASIC] = {
 		.mixers = { alc262_base_mixer },
 		.init_verbs = { alc262_init_verbs },
@@ -12689,9 +12793,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
-		.unsol_event = alc262_hippo_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc262_hippo_setup,
-		.init_hook = alc262_hippo_automute,
+		.init_hook = alc_inithook,
 	},
 	[ALC262_HIPPO_1] = {
 		.mixers = { alc262_hippo1_mixer },
@@ -12703,9 +12807,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
-		.unsol_event = alc262_hippo_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc262_hippo1_setup,
-		.init_hook = alc262_hippo_automute,
+		.init_hook = alc_inithook,
 	},
 	[ALC262_FUJITSU] = {
 		.mixers = { alc262_fujitsu_mixer },
@@ -12718,8 +12822,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_fujitsu_capture_source,
-		.unsol_event = alc262_fujitsu_unsol_event,
-		.init_hook = alc262_fujitsu_init_hook,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_fujitsu_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC262_HP_BPC] = {
 		.mixers = { alc262_HP_BPC_mixer },
@@ -12730,8 +12835,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_HP_capture_source,
-		.unsol_event = alc262_hp_bpc_unsol_event,
-		.init_hook = alc262_hp_bpc_automute,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_hp_bpc_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC262_HP_BPC_D7000_WF] = {
 		.mixers = { alc262_HP_BPC_WildWest_mixer },
@@ -12742,8 +12848,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_HP_D7000_capture_source,
-		.unsol_event = alc262_hp_wildwest_unsol_event,
-		.init_hook = alc262_hp_wildwest_automute,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_hp_wildwest_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC262_HP_BPC_D7000_WL] = {
 		.mixers = { alc262_HP_BPC_WildWest_mixer,
@@ -12755,8 +12862,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_HP_D7000_capture_source,
-		.unsol_event = alc262_hp_wildwest_unsol_event,
-		.init_hook = alc262_hp_wildwest_automute,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_hp_wildwest_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC262_HP_TC_T5735] = {
 		.mixers = { alc262_hp_t5735_mixer },
@@ -12799,9 +12907,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
-		.unsol_event = alc262_hippo_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc262_hippo_setup,
-		.init_hook = alc262_hippo_automute,
+		.init_hook = alc_inithook,
 	},
 	[ALC262_BENQ_T31] = {
 		.mixers = { alc262_benq_t31_mixer },
@@ -12813,9 +12921,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
-		.unsol_event = alc262_hippo_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc262_hippo_setup,
-		.init_hook = alc262_hippo_automute,
+		.init_hook = alc_inithook,
 	},
 	[ALC262_ULTRA] = {
 		.mixers = { alc262_ultra_mixer },
@@ -12844,7 +12952,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_fujitsu_capture_source,
-		.unsol_event = alc262_lenovo_3000_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_lenovo_3000_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC262_NEC] = {
 		.mixers = { alc262_nec_mixer },
@@ -12881,9 +12991,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
-		.unsol_event = alc262_hippo_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc262_hippo_setup,
-		.init_hook = alc262_hippo_automute,
+		.init_hook = alc_inithook,
 	},
 	[ALC262_TYAN] = {
 		.mixers = { alc262_tyan_mixer },
@@ -12895,9 +13005,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc262_tyan_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 };
 
@@ -13018,6 +13128,7 @@ static int patch_alc262(struct hda_codec *codec)
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC262_AUTO)
 		spec->init_hook = alc262_auto_init;
+	spec->shutup = alc_eapd_shutup;
 
 	alc_init_jacks(codec);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -13034,24 +13145,24 @@ static int patch_alc262(struct hda_codec *codec)
 #define ALC268_DIGOUT_NID	ALC880_DIGOUT_NID
 #define alc268_modes		alc260_modes
 
-static hda_nid_t alc268_dac_nids[2] = {
+static const hda_nid_t alc268_dac_nids[2] = {
 	/* front, hp */
 	0x02, 0x03
 };
 
-static hda_nid_t alc268_adc_nids[2] = {
+static const hda_nid_t alc268_adc_nids[2] = {
 	/* ADC0-1 */
 	0x08, 0x07
 };
 
-static hda_nid_t alc268_adc_nids_alt[1] = {
+static const hda_nid_t alc268_adc_nids_alt[1] = {
 	/* ADC0 */
 	0x08
 };
 
-static hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 };
+static const hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 };
 
-static struct snd_kcontrol_new alc268_base_mixer[] = {
+static const struct snd_kcontrol_new alc268_base_mixer[] = {
 	/* output mixer control */
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
@@ -13063,7 +13174,7 @@ static struct snd_kcontrol_new alc268_base_mixer[] = {
 	{ }
 };
 
-static struct snd_kcontrol_new alc268_toshiba_mixer[] = {
+static const struct snd_kcontrol_new alc268_toshiba_mixer[] = {
 	/* output mixer control */
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
@@ -13075,7 +13186,7 @@ static struct snd_kcontrol_new alc268_toshiba_mixer[] = {
 };
 
 /* bind Beep switches of both NID 0x0f and 0x10 */
-static struct hda_bind_ctls alc268_bind_beep_sw = {
+static const struct hda_bind_ctls alc268_bind_beep_sw = {
 	.ops = &snd_hda_bind_sw,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x0f, 3, 1, HDA_INPUT),
@@ -13084,27 +13195,27 @@ static struct hda_bind_ctls alc268_bind_beep_sw = {
 	},
 };
 
-static struct snd_kcontrol_new alc268_beep_mixer[] = {
+static const struct snd_kcontrol_new alc268_beep_mixer[] = {
 	HDA_CODEC_VOLUME("Beep Playback Volume", 0x1d, 0x0, HDA_INPUT),
 	HDA_BIND_SW("Beep Playback Switch", &alc268_bind_beep_sw),
 	{ }
 };
 
-static struct hda_verb alc268_eapd_verbs[] = {
+static const struct hda_verb alc268_eapd_verbs[] = {
 	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
 	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
 	{ }
 };
 
 /* Toshiba specific */
-static struct hda_verb alc268_toshiba_verbs[] = {
+static const struct hda_verb alc268_toshiba_verbs[] = {
 	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
 	{ } /* end */
 };
 
 /* Acer specific */
 /* bind volumes of both NID 0x02 and 0x03 */
-static struct hda_bind_ctls alc268_acer_bind_master_vol = {
+static const struct hda_bind_ctls alc268_acer_bind_master_vol = {
 	.ops = &snd_hda_bind_vol,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
@@ -13113,66 +13224,44 @@ static struct hda_bind_ctls alc268_acer_bind_master_vol = {
 	},
 };
 
-/* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc268_acer_automute(struct hda_codec *codec, int force)
+static void alc268_acer_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	unsigned int mute;
 
-	if (force || !spec->sense_updated) {
-		spec->jack_present = snd_hda_jack_detect(codec, 0x14);
-		spec->sense_updated = 1;
-	}
-	if (spec->jack_present)
-		mute = HDA_AMP_MUTE; /* mute internal speaker */
-	else /* unmute internal speaker if necessary */
-		mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0);
-	snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, mute);
+	spec->autocfg.hp_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[0] = 0x15;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
+#define alc268_acer_master_sw_get	alc262_hp_master_sw_get
+#define alc268_acer_master_sw_put	alc262_hp_master_sw_put
 
-/* bind hp and internal speaker mute (with plug check) */
-static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol,
-				     struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	long *valp = ucontrol->value.integer.value;
-	int change;
-
-	change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
-	if (change)
-		alc268_acer_automute(codec, 0);
-	return change;
-}
-
-static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = {
+static const struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = {
 	/* output mixer control */
 	HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
+		.subdevice = HDA_SUBDEV_NID_FLAG | 0x15,
+		.info = snd_ctl_boolean_mono_info,
+		.get = alc268_acer_master_sw_get,
 		.put = alc268_acer_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
 	},
 	HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x18, 0, HDA_INPUT),
 	{ }
 };
 
-static struct snd_kcontrol_new alc268_acer_mixer[] = {
+static const struct snd_kcontrol_new alc268_acer_mixer[] = {
 	/* output mixer control */
 	HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
+		.subdevice = HDA_SUBDEV_NID_FLAG | 0x14,
+		.info = snd_ctl_boolean_mono_info,
+		.get = alc268_acer_master_sw_get,
 		.put = alc268_acer_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
 	},
 	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
@@ -13180,24 +13269,23 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = {
 	{ }
 };
 
-static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = {
+static const struct snd_kcontrol_new alc268_acer_dmic_mixer[] = {
 	/* output mixer control */
 	HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
+		.subdevice = HDA_SUBDEV_NID_FLAG | 0x14,
+		.info = snd_ctl_boolean_mono_info,
+		.get = alc268_acer_master_sw_get,
 		.put = alc268_acer_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
 	},
 	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
 	{ }
 };
 
-static struct hda_verb alc268_acer_aspire_one_verbs[] = {
+static const struct hda_verb alc268_acer_aspire_one_verbs[] = {
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
@@ -13207,7 +13295,7 @@ static struct hda_verb alc268_acer_aspire_one_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc268_acer_verbs[] = {
+static const struct hda_verb alc268_acer_verbs[] = {
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* internal dmic? */
 	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -13219,53 +13307,16 @@ static struct hda_verb alc268_acer_verbs[] = {
 };
 
 /* unsolicited event for HP jack sensing */
-#define alc268_toshiba_unsol_event	alc262_hippo_unsol_event
 #define alc268_toshiba_setup		alc262_hippo_setup
-#define alc268_toshiba_automute		alc262_hippo_automute
-
-static void alc268_acer_unsol_event(struct hda_codec *codec,
-				       unsigned int res)
-{
-	if ((res >> 26) != ALC880_HP_EVENT)
-		return;
-	alc268_acer_automute(codec, 1);
-}
-
-static void alc268_acer_init_hook(struct hda_codec *codec)
-{
-	alc268_acer_automute(codec, 1);
-}
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc268_aspire_one_speaker_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-	unsigned char bits;
-
-	present = snd_hda_jack_detect(codec, 0x15);
-	bits = present ? HDA_AMP_MUTE : 0;
-	snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0,
-				 HDA_AMP_MUTE, bits);
-	snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 1,
-				 HDA_AMP_MUTE, bits);
-}
-
-static void alc268_acer_lc_unsol_event(struct hda_codec *codec,
-				    unsigned int res)
-{
-	switch (res >> 26) {
-	case ALC880_HP_EVENT:
-		alc268_aspire_one_speaker_automute(codec);
-		break;
-	case ALC880_MIC_EVENT:
-		alc_mic_automute(codec);
-		break;
-	}
-}
 
 static void alc268_acer_lc_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
+	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute_mixer_nid[0] = 0x0f;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_MIXER;
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
 	spec->int_mic.pin = 0x12;
@@ -13273,13 +13324,7 @@ static void alc268_acer_lc_setup(struct hda_codec *codec)
 	spec->auto_mic = 1;
 }
 
-static void alc268_acer_lc_init_hook(struct hda_codec *codec)
-{
-	alc268_aspire_one_speaker_automute(codec);
-	alc_mic_automute(codec);
-}
-
-static struct snd_kcontrol_new alc268_dell_mixer[] = {
+static const struct snd_kcontrol_new alc268_dell_mixer[] = {
 	/* output mixer control */
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
@@ -13290,7 +13335,7 @@ static struct snd_kcontrol_new alc268_dell_mixer[] = {
 	{ }
 };
 
-static struct hda_verb alc268_dell_verbs[] = {
+static const struct hda_verb alc268_dell_verbs[] = {
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
@@ -13310,9 +13355,11 @@ static void alc268_dell_setup(struct hda_codec *codec)
 	spec->int_mic.pin = 0x19;
 	spec->int_mic.mux_idx = 1;
 	spec->auto_mic = 1;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
 }
 
-static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = {
+static const struct snd_kcontrol_new alc267_quanta_il1_mixer[] = {
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
@@ -13324,7 +13371,7 @@ static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = {
 	{ }
 };
 
-static struct hda_verb alc267_quanta_il1_verbs[] = {
+static const struct hda_verb alc267_quanta_il1_verbs[] = {
 	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
 	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
 	{ }
@@ -13340,12 +13387,14 @@ static void alc267_quanta_il1_setup(struct hda_codec *codec)
 	spec->int_mic.pin = 0x19;
 	spec->int_mic.mux_idx = 1;
 	spec->auto_mic = 1;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
 }
 
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb alc268_base_init_verbs[] = {
+static const struct hda_verb alc268_base_init_verbs[] = {
 	/* Unmute DAC0-1 and set vol = 0 */
 	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
@@ -13393,7 +13442,7 @@ static struct hda_verb alc268_base_init_verbs[] = {
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb alc268_volume_init_verbs[] = {
+static const struct hda_verb alc268_volume_init_verbs[] = {
 	/* set output DAC */
 	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
@@ -13419,20 +13468,20 @@ static struct hda_verb alc268_volume_init_verbs[] = {
 	{ }
 };
 
-static struct snd_kcontrol_new alc268_capture_nosrc_mixer[] = {
+static const struct snd_kcontrol_new alc268_capture_nosrc_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc268_capture_alt_mixer[] = {
+static const struct snd_kcontrol_new alc268_capture_alt_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
 	_DEFINE_CAPSRC(1),
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc268_capture_mixer[] = {
+static const struct snd_kcontrol_new alc268_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT),
@@ -13441,7 +13490,7 @@ static struct snd_kcontrol_new alc268_capture_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_input_mux alc268_capture_source = {
+static const struct hda_input_mux alc268_capture_source = {
 	.num_items = 4,
 	.items = {
 		{ "Mic", 0x0 },
@@ -13451,7 +13500,7 @@ static struct hda_input_mux alc268_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc268_acer_capture_source = {
+static const struct hda_input_mux alc268_acer_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic", 0x0 },
@@ -13460,7 +13509,7 @@ static struct hda_input_mux alc268_acer_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc268_acer_dmic_capture_source = {
+static const struct hda_input_mux alc268_acer_dmic_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic", 0x0 },
@@ -13470,7 +13519,7 @@ static struct hda_input_mux alc268_acer_dmic_capture_source = {
 };
 
 #ifdef CONFIG_SND_DEBUG
-static struct snd_kcontrol_new alc268_test_mixer[] = {
+static const struct snd_kcontrol_new alc268_test_mixer[] = {
 	/* Volume widgets */
 	HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x03, 0x0, HDA_OUTPUT),
@@ -13549,7 +13598,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
 						      HDA_OUTPUT));
 		if (err < 0)
 			return err;
-		spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
+		spec->private_dac_nids[spec->multiout.num_dacs++] = dac;
 	}
 
 	if (nid != 0x16)
@@ -13722,7 +13771,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	int err;
-	static hda_nid_t alc268_ignore[] = { 0 };
+	static const hda_nid_t alc268_ignore[] = { 0 };
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
 					   alc268_ignore);
@@ -13802,7 +13851,7 @@ static const char * const alc268_models[ALC268_MODEL_LAST] = {
 	[ALC268_AUTO]		= "auto",
 };
 
-static struct snd_pci_quirk alc268_cfg_tbl[] = {
+static const struct snd_pci_quirk alc268_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x011e, "Acer Aspire 5720z", ALC268_ACER),
 	SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER),
 	SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER),
@@ -13827,7 +13876,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
 };
 
 /* Toshiba laptops have no unique PCI SSID but only codec SSID */
-static struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
+static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1179, 0xff0a, "TOSHIBA X-200", ALC268_AUTO),
 	SND_PCI_QUIRK(0x1179, 0xff0e, "TOSHIBA X-200 HDMI", ALC268_AUTO),
 	SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05",
@@ -13835,7 +13884,7 @@ static struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
 	{}
 };
 
-static struct alc_config_preset alc268_presets[] = {
+static const struct alc_config_preset alc268_presets[] = {
 	[ALC267_QUANTA_IL1] = {
 		.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
 			    alc268_capture_nosrc_mixer },
@@ -13881,9 +13930,9 @@ static struct alc_config_preset alc268_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc268_modes),
 		.channel_mode = alc268_modes,
 		.input_mux = &alc268_capture_source,
-		.unsol_event = alc268_toshiba_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc268_toshiba_setup,
-		.init_hook = alc268_toshiba_automute,
+		.init_hook = alc_inithook,
 	},
 	[ALC268_ACER] = {
 		.mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
@@ -13899,8 +13948,9 @@ static struct alc_config_preset alc268_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc268_modes),
 		.channel_mode = alc268_modes,
 		.input_mux = &alc268_acer_capture_source,
-		.unsol_event = alc268_acer_unsol_event,
-		.init_hook = alc268_acer_init_hook,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc268_acer_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC268_ACER_DMIC] = {
 		.mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer,
@@ -13916,8 +13966,9 @@ static struct alc_config_preset alc268_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc268_modes),
 		.channel_mode = alc268_modes,
 		.input_mux = &alc268_acer_dmic_capture_source,
-		.unsol_event = alc268_acer_unsol_event,
-		.init_hook = alc268_acer_init_hook,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc268_acer_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC268_ACER_ASPIRE_ONE] = {
 		.mixers = { alc268_acer_aspire_one_mixer,
@@ -13933,9 +13984,9 @@ static struct alc_config_preset alc268_presets[] = {
 		.hp_nid = 0x03,
 		.num_channel_mode = ARRAY_SIZE(alc268_modes),
 		.channel_mode = alc268_modes,
-		.unsol_event = alc268_acer_lc_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc268_acer_lc_setup,
-		.init_hook = alc268_acer_lc_init_hook,
+		.init_hook = alc_inithook,
 	},
 	[ALC268_DELL] = {
 		.mixers = { alc268_dell_mixer, alc268_beep_mixer,
@@ -13969,8 +14020,9 @@ static struct alc_config_preset alc268_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc268_modes),
 		.channel_mode = alc268_modes,
 		.input_mux = &alc268_capture_source,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc268_toshiba_setup,
-		.init_hook = alc268_toshiba_automute,
+		.init_hook = alc_inithook,
 	},
 #ifdef CONFIG_SND_DEBUG
 	[ALC268_TEST] = {
@@ -14092,6 +14144,7 @@ static int patch_alc268(struct hda_codec *codec)
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC268_AUTO)
 		spec->init_hook = alc268_auto_init;
+	spec->shutup = alc_eapd_shutup;
 
 	alc_init_jacks(codec);
 
@@ -14105,32 +14158,32 @@ static int patch_alc268(struct hda_codec *codec)
 
 #define alc269_dac_nids		alc260_dac_nids
 
-static hda_nid_t alc269_adc_nids[1] = {
+static const hda_nid_t alc269_adc_nids[1] = {
 	/* ADC1 */
 	0x08,
 };
 
-static hda_nid_t alc269_capsrc_nids[1] = {
+static const hda_nid_t alc269_capsrc_nids[1] = {
 	0x23,
 };
 
-static hda_nid_t alc269vb_adc_nids[1] = {
+static const hda_nid_t alc269vb_adc_nids[1] = {
 	/* ADC1 */
 	0x09,
 };
 
-static hda_nid_t alc269vb_capsrc_nids[1] = {
+static const hda_nid_t alc269vb_capsrc_nids[1] = {
 	0x22,
 };
 
-static hda_nid_t alc269_adc_candidates[] = {
+static const hda_nid_t alc269_adc_candidates[] = {
 	0x08, 0x09, 0x07, 0x11,
 };
 
 #define alc269_modes		alc260_modes
 #define alc269_capture_source	alc880_lg_lw_capture_source
 
-static struct snd_kcontrol_new alc269_base_mixer[] = {
+static const struct snd_kcontrol_new alc269_base_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
@@ -14146,7 +14199,7 @@ static struct snd_kcontrol_new alc269_base_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = {
+static const struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = {
 	/* output mixer control */
 	HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
 	{
@@ -14167,7 +14220,7 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = {
 	{ }
 };
 
-static struct snd_kcontrol_new alc269_lifebook_mixer[] = {
+static const struct snd_kcontrol_new alc269_lifebook_mixer[] = {
 	/* output mixer control */
 	HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
 	{
@@ -14191,7 +14244,7 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = {
 	{ }
 };
 
-static struct snd_kcontrol_new alc269_laptop_mixer[] = {
+static const struct snd_kcontrol_new alc269_laptop_mixer[] = {
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
@@ -14199,7 +14252,7 @@ static struct snd_kcontrol_new alc269_laptop_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc269vb_laptop_mixer[] = {
+static const struct snd_kcontrol_new alc269vb_laptop_mixer[] = {
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
@@ -14207,14 +14260,14 @@ static struct snd_kcontrol_new alc269vb_laptop_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc269_asus_mixer[] = {
+static const struct snd_kcontrol_new alc269_asus_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x0, HDA_INPUT),
 	{ } /* end */
 };
 
 /* capture mixer elements */
-static struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = {
+static const struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
@@ -14222,14 +14275,14 @@ static struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc269_laptop_digital_capture_mixer[] = {
+static const struct snd_kcontrol_new alc269_laptop_digital_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc269vb_laptop_analog_capture_mixer[] = {
+static const struct snd_kcontrol_new alc269vb_laptop_analog_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
@@ -14237,7 +14290,7 @@ static struct snd_kcontrol_new alc269vb_laptop_analog_capture_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc269vb_laptop_digital_capture_mixer[] = {
+static const struct snd_kcontrol_new alc269vb_laptop_digital_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
@@ -14247,7 +14300,7 @@ static struct snd_kcontrol_new alc269vb_laptop_digital_capture_mixer[] = {
 /* FSC amilo */
 #define alc269_fujitsu_mixer	alc269_laptop_mixer
 
-static struct hda_verb alc269_quanta_fl1_verbs[] = {
+static const struct hda_verb alc269_quanta_fl1_verbs[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -14257,7 +14310,7 @@ static struct hda_verb alc269_quanta_fl1_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc269_lifebook_verbs[] = {
+static const struct hda_verb alc269_lifebook_verbs[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
@@ -14274,15 +14327,7 @@ static struct hda_verb alc269_lifebook_verbs[] = {
 /* toggle speaker-output according to the hp-jack state */
 static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
 {
-	unsigned int present;
-	unsigned char bits;
-
-	present = snd_hda_jack_detect(codec, 0x15);
-	bits = present ? HDA_AMP_MUTE : 0;
-	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				 HDA_AMP_MUTE, bits);
-	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				 HDA_AMP_MUTE, bits);
+	alc_hp_automute(codec);
 
 	snd_hda_codec_write(codec, 0x20, 0,
 			AC_VERB_SET_COEF_INDEX, 0x0c);
@@ -14295,34 +14340,8 @@ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
 			AC_VERB_SET_PROC_COEF, 0x480);
 }
 
-/* toggle speaker-output according to the hp-jacks state */
-static void alc269_lifebook_speaker_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-	unsigned char bits;
-
-	/* Check laptop headphone socket */
-	present = snd_hda_jack_detect(codec, 0x15);
-
-	/* Check port replicator headphone socket */
-	present |= snd_hda_jack_detect(codec, 0x1a);
-
-	bits = present ? HDA_AMP_MUTE : 0;
-	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				 HDA_AMP_MUTE, bits);
-	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				 HDA_AMP_MUTE, bits);
-
-	snd_hda_codec_write(codec, 0x20, 0,
-			AC_VERB_SET_COEF_INDEX, 0x0c);
-	snd_hda_codec_write(codec, 0x20, 0,
-			AC_VERB_SET_PROC_COEF, 0x680);
-
-	snd_hda_codec_write(codec, 0x20, 0,
-			AC_VERB_SET_COEF_INDEX, 0x0c);
-	snd_hda_codec_write(codec, 0x20, 0,
-			AC_VERB_SET_PROC_COEF, 0x480);
-}
+#define alc269_lifebook_speaker_automute \
+	alc269_quanta_fl1_speaker_automute
 
 static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec)
 {
@@ -14371,6 +14390,9 @@ static void alc269_quanta_fl1_setup(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute_mixer_nid[0] = 0x0c;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_MIXER;
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
 	spec->int_mic.pin = 0x19;
@@ -14384,13 +14406,24 @@ static void alc269_quanta_fl1_init_hook(struct hda_codec *codec)
 	alc_mic_automute(codec);
 }
 
+static void alc269_lifebook_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.hp_pins[1] = 0x1a;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute_mixer_nid[0] = 0x0c;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_MIXER;
+}
+
 static void alc269_lifebook_init_hook(struct hda_codec *codec)
 {
 	alc269_lifebook_speaker_automute(codec);
 	alc269_lifebook_mic_autoswitch(codec);
 }
 
-static struct hda_verb alc269_laptop_dmic_init_verbs[] = {
+static const struct hda_verb alc269_laptop_dmic_init_verbs[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{0x23, AC_VERB_SET_CONNECT_SEL, 0x05},
 	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
@@ -14401,7 +14434,7 @@ static struct hda_verb alc269_laptop_dmic_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc269_laptop_amic_init_verbs[] = {
+static const struct hda_verb alc269_laptop_amic_init_verbs[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{0x23, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
@@ -14411,7 +14444,7 @@ static struct hda_verb alc269_laptop_amic_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc269vb_laptop_dmic_init_verbs[] = {
+static const struct hda_verb alc269vb_laptop_dmic_init_verbs[] = {
 	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{0x22, AC_VERB_SET_CONNECT_SEL, 0x06},
 	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
@@ -14422,7 +14455,7 @@ static struct hda_verb alc269vb_laptop_dmic_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc269vb_laptop_amic_init_verbs[] = {
+static const struct hda_verb alc269vb_laptop_amic_init_verbs[] = {
 	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{0x22, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
@@ -14433,7 +14466,7 @@ static struct hda_verb alc269vb_laptop_amic_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc271_acer_dmic_verbs[] = {
+static const struct hda_verb alc271_acer_dmic_verbs[] = {
 	{0x20, AC_VERB_SET_COEF_INDEX, 0x0d},
 	{0x20, AC_VERB_SET_PROC_COEF, 0x4000},
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
@@ -14447,42 +14480,14 @@ static struct hda_verb alc271_acer_dmic_verbs[] = {
 	{ }
 };
 
-/* toggle speaker-output according to the hp-jack state */
-static void alc269_speaker_automute(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	unsigned int nid = spec->autocfg.hp_pins[0];
-	unsigned int present;
-	unsigned char bits;
-
-	present = snd_hda_jack_detect(codec, nid);
-	bits = present ? HDA_AMP_MUTE : 0;
-	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				 HDA_AMP_MUTE, bits);
-	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				 HDA_AMP_MUTE, bits);
-	snd_hda_input_jack_report(codec, nid);
-}
-
-/* unsolicited event for HP jack sensing */
-static void alc269_laptop_unsol_event(struct hda_codec *codec,
-				     unsigned int res)
-{
-	switch (res >> 26) {
-	case ALC880_HP_EVENT:
-		alc269_speaker_automute(codec);
-		break;
-	case ALC880_MIC_EVENT:
-		alc_mic_automute(codec);
-		break;
-	}
-}
-
 static void alc269_laptop_amic_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute_mixer_nid[0] = 0x0c;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_MIXER;
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
 	spec->int_mic.pin = 0x19;
@@ -14495,6 +14500,9 @@ static void alc269_laptop_dmic_setup(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute_mixer_nid[0] = 0x0c;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_MIXER;
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
 	spec->int_mic.pin = 0x12;
@@ -14507,6 +14515,9 @@ static void alc269vb_laptop_amic_setup(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	spec->autocfg.hp_pins[0] = 0x21;
 	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute_mixer_nid[0] = 0x0c;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_MIXER;
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
 	spec->int_mic.pin = 0x19;
@@ -14519,6 +14530,9 @@ static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	spec->autocfg.hp_pins[0] = 0x21;
 	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute_mixer_nid[0] = 0x0c;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_MIXER;
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
 	spec->int_mic.pin = 0x12;
@@ -14526,16 +14540,10 @@ static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
 	spec->auto_mic = 1;
 }
 
-static void alc269_laptop_inithook(struct hda_codec *codec)
-{
-	alc269_speaker_automute(codec);
-	alc_mic_automute(codec);
-}
-
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb alc269_init_verbs[] = {
+static const struct hda_verb alc269_init_verbs[] = {
 	/*
 	 * Unmute ADC0 and set the default input to mic-in
 	 */
@@ -14578,7 +14586,7 @@ static struct hda_verb alc269_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc269vb_init_verbs[] = {
+static const struct hda_verb alc269vb_init_verbs[] = {
 	/*
 	 * Unmute ADC0 and set the default input to mic-in
 	 */
@@ -14636,7 +14644,7 @@ static struct hda_verb alc269vb_init_verbs[] = {
 #define alc269_pcm_digital_playback	alc880_pcm_digital_playback
 #define alc269_pcm_digital_capture	alc880_pcm_digital_capture
 
-static struct hda_pcm_stream alc269_44k_pcm_analog_playback = {
+static const struct hda_pcm_stream alc269_44k_pcm_analog_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 8,
@@ -14649,7 +14657,7 @@ static struct hda_pcm_stream alc269_44k_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream alc269_44k_pcm_analog_capture = {
+static const struct hda_pcm_stream alc269_44k_pcm_analog_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -14733,7 +14741,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	int err;
-	static hda_nid_t alc269_ignore[] = { 0x1d, 0 };
+	static const hda_nid_t alc269_ignore[] = { 0x1d, 0 };
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
 					   alc269_ignore);
@@ -14803,7 +14811,6 @@ static void alc269_auto_init(struct hda_codec *codec)
 		alc_inithook(codec);
 }
 
-#ifdef SND_HDA_NEEDS_RESUME
 static void alc269_toggle_power_output(struct hda_codec *codec, int power_up)
 {
 	int val = alc_read_coef_idx(codec, 0x04);
@@ -14814,25 +14821,17 @@ static void alc269_toggle_power_output(struct hda_codec *codec, int power_up)
 	alc_write_coef_idx(codec, 0x04, val);
 }
 
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static int alc269_suspend(struct hda_codec *codec, pm_message_t state)
+static void alc269_shutup(struct hda_codec *codec)
 {
-	struct alc_spec *spec = codec->spec;
-
 	if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017)
 		alc269_toggle_power_output(codec, 0);
 	if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
 		alc269_toggle_power_output(codec, 0);
 		msleep(150);
 	}
-
-	alc_shutup(codec);
-	if (spec && spec->power_hook)
-		spec->power_hook(codec);
-	return 0;
 }
-#endif /* CONFIG_SND_HDA_POWER_SAVE */
 
+#ifdef SND_HDA_NEEDS_RESUME
 static int alc269_resume(struct hda_codec *codec)
 {
 	if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
@@ -14868,6 +14867,23 @@ static void alc269_fixup_hweq(struct hda_codec *codec,
 	alc_write_coef_idx(codec, 0x1e, coef | 0x80);
 }
 
+static void alc271_fixup_dmic(struct hda_codec *codec,
+			      const struct alc_fixup *fix, int action)
+{
+	static const struct hda_verb verbs[] = {
+		{0x20, AC_VERB_SET_COEF_INDEX, 0x0d},
+		{0x20, AC_VERB_SET_PROC_COEF, 0x4000},
+		{}
+	};
+	unsigned int cfg;
+
+	if (strcmp(codec->chip_name, "ALC271X"))
+		return;
+	cfg = snd_hda_codec_get_pincfg(codec, 0x12);
+	if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED)
+		snd_hda_sequence_write(codec, verbs);
+}
+
 enum {
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -14876,6 +14892,7 @@ enum {
 	ALC269_FIXUP_ASUS_G73JW,
 	ALC269_FIXUP_LENOVO_EAPD,
 	ALC275_FIXUP_SONY_HWEQ,
+	ALC271_FIXUP_DMIC,
 };
 
 static const struct alc_fixup alc269_fixups[] = {
@@ -14929,15 +14946,20 @@ static const struct alc_fixup alc269_fixups[] = {
 		.v.func = alc269_fixup_hweq,
 		.chained = true,
 		.chain_id = ALC275_FIXUP_SONY_VAIO_GPIO2
-	}
+	},
+	[ALC271_FIXUP_DMIC] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc271_fixup_dmic,
+	},
 };
 
-static struct snd_pci_quirk alc269_fixup_tbl[] = {
+static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
 	SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
+	SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
@@ -14962,7 +14984,7 @@ static const char * const alc269_models[ALC269_MODEL_LAST] = {
 	[ALC269_AUTO]			= "auto",
 };
 
-static struct snd_pci_quirk alc269_cfg_tbl[] = {
+static const struct snd_pci_quirk alc269_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1),
 	SND_PCI_QUIRK(0x1025, 0x047c, "ACER ZGA", ALC271_ACER),
 	SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
@@ -15020,7 +15042,7 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = {
 	{}
 };
 
-static struct alc_config_preset alc269_presets[] = {
+static const struct alc_config_preset alc269_presets[] = {
 	[ALC269_BASIC] = {
 		.mixers = { alc269_base_mixer },
 		.init_verbs = { alc269_init_verbs },
@@ -15054,9 +15076,9 @@ static struct alc_config_preset alc269_presets[] = {
 		.hp_nid = 0x03,
 		.num_channel_mode = ARRAY_SIZE(alc269_modes),
 		.channel_mode = alc269_modes,
-		.unsol_event = alc269_laptop_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc269_laptop_amic_setup,
-		.init_hook = alc269_laptop_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC269_DMIC] = {
 		.mixers = { alc269_laptop_mixer },
@@ -15068,9 +15090,9 @@ static struct alc_config_preset alc269_presets[] = {
 		.hp_nid = 0x03,
 		.num_channel_mode = ARRAY_SIZE(alc269_modes),
 		.channel_mode = alc269_modes,
-		.unsol_event = alc269_laptop_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc269_laptop_dmic_setup,
-		.init_hook = alc269_laptop_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC269VB_AMIC] = {
 		.mixers = { alc269vb_laptop_mixer },
@@ -15082,9 +15104,9 @@ static struct alc_config_preset alc269_presets[] = {
 		.hp_nid = 0x03,
 		.num_channel_mode = ARRAY_SIZE(alc269_modes),
 		.channel_mode = alc269_modes,
-		.unsol_event = alc269_laptop_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc269vb_laptop_amic_setup,
-		.init_hook = alc269_laptop_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC269VB_DMIC] = {
 		.mixers = { alc269vb_laptop_mixer },
@@ -15096,9 +15118,9 @@ static struct alc_config_preset alc269_presets[] = {
 		.hp_nid = 0x03,
 		.num_channel_mode = ARRAY_SIZE(alc269_modes),
 		.channel_mode = alc269_modes,
-		.unsol_event = alc269_laptop_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc269vb_laptop_dmic_setup,
-		.init_hook = alc269_laptop_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC269_FUJITSU] = {
 		.mixers = { alc269_fujitsu_mixer },
@@ -15110,9 +15132,9 @@ static struct alc_config_preset alc269_presets[] = {
 		.hp_nid = 0x03,
 		.num_channel_mode = ARRAY_SIZE(alc269_modes),
 		.channel_mode = alc269_modes,
-		.unsol_event = alc269_laptop_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc269_laptop_dmic_setup,
-		.init_hook = alc269_laptop_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC269_LIFEBOOK] = {
 		.mixers = { alc269_lifebook_mixer },
@@ -15124,6 +15146,7 @@ static struct alc_config_preset alc269_presets[] = {
 		.channel_mode = alc269_modes,
 		.input_mux = &alc269_capture_source,
 		.unsol_event = alc269_lifebook_unsol_event,
+		.setup = alc269_lifebook_setup,
 		.init_hook = alc269_lifebook_init_hook,
 	},
 	[ALC271_ACER] = {
@@ -15169,14 +15192,21 @@ static int alc269_fill_coef(struct hda_codec *codec)
 		val = alc_read_coef_idx(codec, 0xd);
 		if ((val & 0x0c00) >> 10 != 0x1) {
 			/* Capless ramp up clock control */
-			alc_write_coef_idx(codec, 0xd, val | 1<<10);
+			alc_write_coef_idx(codec, 0xd, val | (1<<10));
 		}
 		val = alc_read_coef_idx(codec, 0x17);
 		if ((val & 0x01c0) >> 6 != 0x4) {
 			/* Class D power on reset */
-			alc_write_coef_idx(codec, 0x17, val | 1<<7);
+			alc_write_coef_idx(codec, 0x17, val | (1<<7));
 		}
 	}
+
+	val = alc_read_coef_idx(codec, 0xd); /* Class D */
+	alc_write_coef_idx(codec, 0xd, val | (1<<14));
+
+	val = alc_read_coef_idx(codec, 0x4); /* HP */
+	alc_write_coef_idx(codec, 0x4, val | (1<<11));
+
 	return 0;
 }
 
@@ -15297,14 +15327,12 @@ static int patch_alc269(struct hda_codec *codec)
 	spec->vmaster_nid = 0x02;
 
 	codec->patch_ops = alc_patch_ops;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-	codec->patch_ops.suspend = alc269_suspend;
-#endif
 #ifdef SND_HDA_NEEDS_RESUME
 	codec->patch_ops.resume = alc269_resume;
 #endif
 	if (board_config == ALC269_AUTO)
 		spec->init_hook = alc269_auto_init;
+	spec->shutup = alc269_shutup;
 
 	alc_init_jacks(codec);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -15325,7 +15353,7 @@ static int patch_alc269(struct hda_codec *codec)
  * set the path ways for 2 channel output
  * need to set the codec line out and mic 1 pin widgets to inputs
  */
-static struct hda_verb alc861_threestack_ch2_init[] = {
+static const struct hda_verb alc861_threestack_ch2_init[] = {
 	/* set pin widget 1Ah (line in) for input */
 	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
 	/* set pin widget 18h (mic1/2) for input, for mic also enable
@@ -15344,7 +15372,7 @@ static struct hda_verb alc861_threestack_ch2_init[] = {
  * 6ch mode
  * need to set the codec line out and mic 1 pin widgets to outputs
  */
-static struct hda_verb alc861_threestack_ch6_init[] = {
+static const struct hda_verb alc861_threestack_ch6_init[] = {
 	/* set pin widget 1Ah (line in) for output (Back Surround)*/
 	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
 	/* set pin widget 18h (mic1) for output (CLFE)*/
@@ -15361,30 +15389,30 @@ static struct hda_verb alc861_threestack_ch6_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc861_threestack_modes[2] = {
+static const struct hda_channel_mode alc861_threestack_modes[2] = {
 	{ 2, alc861_threestack_ch2_init },
 	{ 6, alc861_threestack_ch6_init },
 };
 /* Set mic1 as input and unmute the mixer */
-static struct hda_verb alc861_uniwill_m31_ch2_init[] = {
+static const struct hda_verb alc861_uniwill_m31_ch2_init[] = {
 	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
 	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
 	{ } /* end */
 };
 /* Set mic1 as output and mute mixer */
-static struct hda_verb alc861_uniwill_m31_ch4_init[] = {
+static const struct hda_verb alc861_uniwill_m31_ch4_init[] = {
 	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
 	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc861_uniwill_m31_modes[2] = {
+static const struct hda_channel_mode alc861_uniwill_m31_modes[2] = {
 	{ 2, alc861_uniwill_m31_ch2_init },
 	{ 4, alc861_uniwill_m31_ch4_init },
 };
 
 /* Set mic1 and line-in as input and unmute the mixer */
-static struct hda_verb alc861_asus_ch2_init[] = {
+static const struct hda_verb alc861_asus_ch2_init[] = {
 	/* set pin widget 1Ah (line in) for input */
 	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
 	/* set pin widget 18h (mic1/2) for input, for mic also enable
@@ -15400,7 +15428,7 @@ static struct hda_verb alc861_asus_ch2_init[] = {
 	{ } /* end */
 };
 /* Set mic1 nad line-in as output and mute mixer */
-static struct hda_verb alc861_asus_ch6_init[] = {
+static const struct hda_verb alc861_asus_ch6_init[] = {
 	/* set pin widget 1Ah (line in) for output (Back Surround)*/
 	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
 	/* { 0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
@@ -15418,14 +15446,14 @@ static struct hda_verb alc861_asus_ch6_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc861_asus_modes[2] = {
+static const struct hda_channel_mode alc861_asus_modes[2] = {
 	{ 2, alc861_asus_ch2_init },
 	{ 6, alc861_asus_ch6_init },
 };
 
 /* patch-ALC861 */
 
-static struct snd_kcontrol_new alc861_base_mixer[] = {
+static const struct snd_kcontrol_new alc861_base_mixer[] = {
         /* output mixer control */
 	HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
@@ -15448,7 +15476,7 @@ static struct snd_kcontrol_new alc861_base_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc861_3ST_mixer[] = {
+static const struct snd_kcontrol_new alc861_3ST_mixer[] = {
         /* output mixer control */
 	HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
@@ -15479,7 +15507,7 @@ static struct snd_kcontrol_new alc861_3ST_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc861_toshiba_mixer[] = {
+static const struct snd_kcontrol_new alc861_toshiba_mixer[] = {
         /* output mixer control */
 	HDA_CODEC_MUTE("Master Playback Switch", 0x03, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
@@ -15488,7 +15516,7 @@ static struct snd_kcontrol_new alc861_toshiba_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = {
+static const struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = {
         /* output mixer control */
 	HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
@@ -15519,7 +15547,7 @@ static struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc861_asus_mixer[] = {
+static const struct snd_kcontrol_new alc861_asus_mixer[] = {
         /* output mixer control */
 	HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
@@ -15551,7 +15579,7 @@ static struct snd_kcontrol_new alc861_asus_mixer[] = {
 };
 
 /* additional mixer */
-static struct snd_kcontrol_new alc861_asus_laptop_mixer[] = {
+static const struct snd_kcontrol_new alc861_asus_laptop_mixer[] = {
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
 	{ }
@@ -15560,7 +15588,7 @@ static struct snd_kcontrol_new alc861_asus_laptop_mixer[] = {
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb alc861_base_init_verbs[] = {
+static const struct hda_verb alc861_base_init_verbs[] = {
 	/*
 	 * Unmute ADC0 and set the default input to mic-in
 	 */
@@ -15626,7 +15654,7 @@ static struct hda_verb alc861_base_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc861_threestack_init_verbs[] = {
+static const struct hda_verb alc861_threestack_init_verbs[] = {
 	/*
 	 * Unmute ADC0 and set the default input to mic-in
 	 */
@@ -15687,7 +15715,7 @@ static struct hda_verb alc861_threestack_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc861_uniwill_m31_init_verbs[] = {
+static const struct hda_verb alc861_uniwill_m31_init_verbs[] = {
 	/*
 	 * Unmute ADC0 and set the default input to mic-in
 	 */
@@ -15749,7 +15777,7 @@ static struct hda_verb alc861_uniwill_m31_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc861_asus_init_verbs[] = {
+static const struct hda_verb alc861_asus_init_verbs[] = {
 	/*
 	 * Unmute ADC0 and set the default input to mic-in
 	 */
@@ -15815,7 +15843,7 @@ static struct hda_verb alc861_asus_init_verbs[] = {
 };
 
 /* additional init verbs for ASUS laptops */
-static struct hda_verb alc861_asus_laptop_init_verbs[] = {
+static const struct hda_verb alc861_asus_laptop_init_verbs[] = {
 	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x45 }, /* HP-out */
 	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2) }, /* mute line-in */
 	{ }
@@ -15824,7 +15852,7 @@ static struct hda_verb alc861_asus_laptop_init_verbs[] = {
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb alc861_auto_init_verbs[] = {
+static const struct hda_verb alc861_auto_init_verbs[] = {
 	/*
 	 * Unmute ADC0 and set the default input to mic-in
 	 */
@@ -15873,7 +15901,7 @@ static struct hda_verb alc861_auto_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc861_toshiba_init_verbs[] = {
+static const struct hda_verb alc861_toshiba_init_verbs[] = {
 	{0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
 
 	{ }
@@ -15906,26 +15934,26 @@ static void alc861_toshiba_unsol_event(struct hda_codec *codec,
 
 #define ALC861_DIGOUT_NID	0x07
 
-static struct hda_channel_mode alc861_8ch_modes[1] = {
+static const struct hda_channel_mode alc861_8ch_modes[1] = {
 	{ 8, NULL }
 };
 
-static hda_nid_t alc861_dac_nids[4] = {
+static const hda_nid_t alc861_dac_nids[4] = {
 	/* front, surround, clfe, side */
 	0x03, 0x06, 0x05, 0x04
 };
 
-static hda_nid_t alc660_dac_nids[3] = {
+static const hda_nid_t alc660_dac_nids[3] = {
 	/* front, clfe, surround */
 	0x03, 0x05, 0x06
 };
 
-static hda_nid_t alc861_adc_nids[1] = {
+static const hda_nid_t alc861_adc_nids[1] = {
 	/* ADC0-2 */
 	0x08,
 };
 
-static struct hda_input_mux alc861_capture_source = {
+static const struct hda_input_mux alc861_capture_source = {
 	.num_items = 5,
 	.items = {
 		{ "Mic", 0x0 },
@@ -15975,7 +16003,7 @@ static int alc861_auto_fill_dac_nids(struct hda_codec *codec,
 		dac = alc861_look_for_dac(codec, nid);
 		if (!dac)
 			continue;
-		spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
+		spec->private_dac_nids[spec->multiout.num_dacs++] = dac;
 	}
 	return 0;
 }
@@ -15998,11 +16026,15 @@ static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec,
 	static const char * const chname[4] = {
 		"Front", "Surround", NULL /*CLFE*/, "Side"
 	};
-	const char *pfx = alc_get_line_out_pfx(cfg, true);
+	const char *pfx = alc_get_line_out_pfx(spec, true);
 	hda_nid_t nid;
-	int i, err;
+	int i, err, noutputs;
 
-	for (i = 0; i < cfg->line_outs; i++) {
+	noutputs = cfg->line_outs;
+	if (spec->multi_ios > 0)
+		noutputs += spec->multi_ios;
+
+	for (i = 0; i < noutputs; i++) {
 		nid = spec->multiout.dac_nids[i];
 		if (!nid)
 			continue;
@@ -16135,7 +16167,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	int err;
-	static hda_nid_t alc861_ignore[] = { 0x1d, 0 };
+	static const hda_nid_t alc861_ignore[] = { 0x1d, 0 };
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
 					   alc861_ignore);
@@ -16147,6 +16179,9 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 	err = alc861_auto_fill_dac_nids(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
+	err = alc_auto_add_multi_channel_mode(codec);
+	if (err < 0)
+		return err;
 	err = alc861_auto_create_multi_out_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
@@ -16191,7 +16226,7 @@ static void alc861_auto_init(struct hda_codec *codec)
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list alc861_loopbacks[] = {
+static const struct hda_amp_list alc861_loopbacks[] = {
 	{ 0x15, HDA_INPUT, 0 },
 	{ 0x15, HDA_INPUT, 1 },
 	{ 0x15, HDA_INPUT, 2 },
@@ -16216,7 +16251,7 @@ static const char * const alc861_models[ALC861_MODEL_LAST] = {
 	[ALC861_AUTO]		= "auto",
 };
 
-static struct snd_pci_quirk alc861_cfg_tbl[] = {
+static const struct snd_pci_quirk alc861_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC861_3ST),
 	SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP),
 	SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP),
@@ -16240,7 +16275,7 @@ static struct snd_pci_quirk alc861_cfg_tbl[] = {
 	{}
 };
 
-static struct alc_config_preset alc861_presets[] = {
+static const struct alc_config_preset alc861_presets[] = {
 	[ALC861_3ST] = {
 		.mixers = { alc861_3ST_mixer },
 		.init_verbs = { alc861_threestack_init_verbs },
@@ -16363,7 +16398,7 @@ static const struct alc_fixup alc861_fixups[] = {
 	},
 };
 
-static struct snd_pci_quirk alc861_fixup_tbl[] = {
+static const struct snd_pci_quirk alc861_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505),
 	{}
 };
@@ -16456,7 +16491,7 @@ static int patch_alc861(struct hda_codec *codec)
  */
 #define ALC861VD_DIGOUT_NID	0x06
 
-static hda_nid_t alc861vd_dac_nids[4] = {
+static const hda_nid_t alc861vd_dac_nids[4] = {
 	/* front, surr, clfe, side surr */
 	0x02, 0x03, 0x04, 0x05
 };
@@ -16468,21 +16503,21 @@ static hda_nid_t alc861vd_dac_nids[4] = {
  * - and it is the same as in 861vd.
  * adc_nids in ALC660vd are (is) the same as in 861vd
  */
-static hda_nid_t alc660vd_dac_nids[3] = {
+static const hda_nid_t alc660vd_dac_nids[3] = {
 	/* front, rear, clfe, rear_surr */
 	0x02, 0x04, 0x03
 };
 
-static hda_nid_t alc861vd_adc_nids[1] = {
+static const hda_nid_t alc861vd_adc_nids[1] = {
 	/* ADC0 */
 	0x09,
 };
 
-static hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 };
+static const hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 };
 
 /* input MUX */
 /* FIXME: should be a matrix-type input source selection */
-static struct hda_input_mux alc861vd_capture_source = {
+static const struct hda_input_mux alc861vd_capture_source = {
 	.num_items = 4,
 	.items = {
 		{ "Mic", 0x0 },
@@ -16492,7 +16527,7 @@ static struct hda_input_mux alc861vd_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc861vd_dallas_capture_source = {
+static const struct hda_input_mux alc861vd_dallas_capture_source = {
 	.num_items = 2,
 	.items = {
 		{ "Mic", 0x0 },
@@ -16500,7 +16535,7 @@ static struct hda_input_mux alc861vd_dallas_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc861vd_hp_capture_source = {
+static const struct hda_input_mux alc861vd_hp_capture_source = {
 	.num_items = 2,
 	.items = {
 		{ "Front Mic", 0x0 },
@@ -16511,14 +16546,14 @@ static struct hda_input_mux alc861vd_hp_capture_source = {
 /*
  * 2ch mode
  */
-static struct hda_channel_mode alc861vd_3stack_2ch_modes[1] = {
+static const struct hda_channel_mode alc861vd_3stack_2ch_modes[1] = {
 	{ 2, NULL }
 };
 
 /*
  * 6ch mode
  */
-static struct hda_verb alc861vd_6stack_ch6_init[] = {
+static const struct hda_verb alc861vd_6stack_ch6_init[] = {
 	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
 	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
@@ -16529,7 +16564,7 @@ static struct hda_verb alc861vd_6stack_ch6_init[] = {
 /*
  * 8ch mode
  */
-static struct hda_verb alc861vd_6stack_ch8_init[] = {
+static const struct hda_verb alc861vd_6stack_ch8_init[] = {
 	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
@@ -16537,12 +16572,12 @@ static struct hda_verb alc861vd_6stack_ch8_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc861vd_6stack_modes[2] = {
+static const struct hda_channel_mode alc861vd_6stack_modes[2] = {
 	{ 6, alc861vd_6stack_ch6_init },
 	{ 8, alc861vd_6stack_ch8_init },
 };
 
-static struct snd_kcontrol_new alc861vd_chmode_mixer[] = {
+static const struct snd_kcontrol_new alc861vd_chmode_mixer[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Channel Mode",
@@ -16556,7 +16591,7 @@ static struct snd_kcontrol_new alc861vd_chmode_mixer[] = {
 /* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
  *                 Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
  */
-static struct snd_kcontrol_new alc861vd_6st_mixer[] = {
+static const struct snd_kcontrol_new alc861vd_6st_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 
@@ -16592,7 +16627,7 @@ static struct snd_kcontrol_new alc861vd_6st_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc861vd_3st_mixer[] = {
+static const struct snd_kcontrol_new alc861vd_3st_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 
@@ -16615,7 +16650,7 @@ static struct snd_kcontrol_new alc861vd_3st_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc861vd_lenovo_mixer[] = {
+static const struct snd_kcontrol_new alc861vd_lenovo_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	/*HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),*/
 	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
@@ -16639,7 +16674,7 @@ static struct snd_kcontrol_new alc861vd_lenovo_mixer[] = {
 /* Pin assignment: Speaker=0x14, HP = 0x15,
  *                 Mic=0x18, Internal Mic = 0x19, CD = 0x1c, PC Beep = 0x1d
  */
-static struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
+static const struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
@@ -16656,7 +16691,7 @@ static struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
 /* Pin assignment: Speaker=0x14, Line-out = 0x15,
  *                 Front Mic=0x18, ATAPI Mic = 0x19,
  */
-static struct snd_kcontrol_new alc861vd_hp_mixer[] = {
+static const struct snd_kcontrol_new alc861vd_hp_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
@@ -16672,7 +16707,7 @@ static struct snd_kcontrol_new alc861vd_hp_mixer[] = {
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb alc861vd_volume_init_verbs[] = {
+static const struct hda_verb alc861vd_volume_init_verbs[] = {
 	/*
 	 * Unmute ADC0 and set the default input to mic-in
 	 */
@@ -16722,7 +16757,7 @@ static struct hda_verb alc861vd_volume_init_verbs[] = {
  * 3-stack pin configuration:
  * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
  */
-static struct hda_verb alc861vd_3stack_init_verbs[] = {
+static const struct hda_verb alc861vd_3stack_init_verbs[] = {
 	/*
 	 * Set pin mode and muting
 	 */
@@ -16753,7 +16788,7 @@ static struct hda_verb alc861vd_3stack_init_verbs[] = {
 /*
  * 6-stack pin configuration:
  */
-static struct hda_verb alc861vd_6stack_init_verbs[] = {
+static const struct hda_verb alc861vd_6stack_init_verbs[] = {
 	/*
 	 * Set pin mode and muting
 	 */
@@ -16794,18 +16829,18 @@ static struct hda_verb alc861vd_6stack_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc861vd_eapd_verbs[] = {
+static const struct hda_verb alc861vd_eapd_verbs[] = {
 	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
 	{ }
 };
 
-static struct hda_verb alc660vd_eapd_verbs[] = {
+static const struct hda_verb alc660vd_eapd_verbs[] = {
 	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
 	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
 	{ }
 };
 
-static struct hda_verb alc861vd_lenovo_unsol_verbs[] = {
+static const struct hda_verb alc861vd_lenovo_unsol_verbs[] = {
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
@@ -16819,11 +16854,13 @@ static void alc861vd_lenovo_setup(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc861vd_lenovo_init_hook(struct hda_codec *codec)
 {
-	alc_automute_amp(codec);
+	alc_hp_automute(codec);
 	alc88x_simple_mic_automute(codec);
 }
 
@@ -16835,12 +16872,12 @@ static void alc861vd_lenovo_unsol_event(struct hda_codec *codec,
 		alc88x_simple_mic_automute(codec);
 		break;
 	default:
-		alc_automute_amp_unsol_event(codec, res);
+		alc_sku_unsol_event(codec, res);
 		break;
 	}
 }
 
-static struct hda_verb alc861vd_dallas_verbs[] = {
+static const struct hda_verb alc861vd_dallas_verbs[] = {
 	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
@@ -16892,6 +16929,8 @@ static void alc861vd_dallas_setup(struct hda_codec *codec)
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -16920,7 +16959,7 @@ static const char * const alc861vd_models[ALC861VD_MODEL_LAST] = {
 	[ALC861VD_AUTO]		= "auto",
 };
 
-static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
+static const struct snd_pci_quirk alc861vd_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),
 	SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
 	SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
@@ -16939,7 +16978,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
 	{}
 };
 
-static struct alc_config_preset alc861vd_presets[] = {
+static const struct alc_config_preset alc861vd_presets[] = {
 	[ALC660VD_3ST] = {
 		.mixers = { alc861vd_3st_mixer },
 		.init_verbs = { alc861vd_volume_init_verbs,
@@ -17016,9 +17055,9 @@ static struct alc_config_preset alc861vd_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
 		.channel_mode = alc861vd_3stack_2ch_modes,
 		.input_mux = &alc861vd_dallas_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc861vd_dallas_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC861VD_HP] = {
 		.mixers = { alc861vd_hp_mixer },
@@ -17029,9 +17068,9 @@ static struct alc_config_preset alc861vd_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
 		.channel_mode = alc861vd_3stack_2ch_modes,
 		.input_mux = &alc861vd_hp_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc861vd_dallas_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_hp_automute,
 	},
 	[ALC660VD_ASUS_V1S] = {
 		.mixers = { alc861vd_lenovo_mixer },
@@ -17130,11 +17169,15 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
 	static const char * const chname[4] = {
 		"Front", "Surround", "CLFE", "Side"
 	};
-	const char *pfx = alc_get_line_out_pfx(cfg, true);
+	const char *pfx = alc_get_line_out_pfx(spec, true);
 	hda_nid_t nid_v, nid_s;
-	int i, err;
+	int i, err, noutputs;
 
-	for (i = 0; i < cfg->line_outs; i++) {
+	noutputs = cfg->line_outs;
+	if (spec->multi_ios > 0)
+		noutputs += spec->multi_ios;
+
+	for (i = 0; i < noutputs; i++) {
 		if (!spec->multiout.dac_nids[i])
 			continue;
 		nid_v = alc861vd_idx_to_mixer_vol(
@@ -17247,7 +17290,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	int err;
-	static hda_nid_t alc861vd_ignore[] = { 0x1d, 0 };
+	static const hda_nid_t alc861vd_ignore[] = { 0x1d, 0 };
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
 					   alc861vd_ignore);
@@ -17259,6 +17302,9 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
 	err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
+	err = alc_auto_add_multi_channel_mode(codec);
+	if (err < 0)
+		return err;
 	err = alc861vd_auto_create_multi_out_ctls(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
@@ -17327,7 +17373,7 @@ static const struct alc_fixup alc861vd_fixups[] = {
 	},
 };
 
-static struct snd_pci_quirk alc861vd_fixup_tbl[] = {
+static const struct snd_pci_quirk alc861vd_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1339, "ASUS A7-K", ALC660VD_FIX_ASUS_GPIO1),
 	{}
 };
@@ -17410,6 +17456,7 @@ static int patch_alc861vd(struct hda_codec *codec)
 
 	if (board_config == ALC861VD_AUTO)
 		spec->init_hook = alc861vd_auto_init;
+	spec->shutup = alc_eapd_shutup;
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc861vd_loopbacks;
@@ -17432,32 +17479,32 @@ static int patch_alc861vd(struct hda_codec *codec)
 #define ALC662_DIGOUT_NID	0x06
 #define ALC662_DIGIN_NID	0x0a
 
-static hda_nid_t alc662_dac_nids[4] = {
-	/* front, rear, clfe, rear_surr */
+static const hda_nid_t alc662_dac_nids[3] = {
+	/* front, rear, clfe */
 	0x02, 0x03, 0x04
 };
 
-static hda_nid_t alc272_dac_nids[2] = {
+static const hda_nid_t alc272_dac_nids[2] = {
 	0x02, 0x03
 };
 
-static hda_nid_t alc662_adc_nids[2] = {
+static const hda_nid_t alc662_adc_nids[2] = {
 	/* ADC1-2 */
 	0x09, 0x08
 };
 
-static hda_nid_t alc272_adc_nids[1] = {
+static const hda_nid_t alc272_adc_nids[1] = {
 	/* ADC1-2 */
 	0x08,
 };
 
-static hda_nid_t alc662_capsrc_nids[2] = { 0x22, 0x23 };
-static hda_nid_t alc272_capsrc_nids[1] = { 0x23 };
+static const hda_nid_t alc662_capsrc_nids[2] = { 0x22, 0x23 };
+static const hda_nid_t alc272_capsrc_nids[1] = { 0x23 };
 
 
 /* input MUX */
 /* FIXME: should be a matrix-type input source selection */
-static struct hda_input_mux alc662_capture_source = {
+static const struct hda_input_mux alc662_capture_source = {
 	.num_items = 4,
 	.items = {
 		{ "Mic", 0x0 },
@@ -17467,7 +17514,7 @@ static struct hda_input_mux alc662_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc662_lenovo_101e_capture_source = {
+static const struct hda_input_mux alc662_lenovo_101e_capture_source = {
 	.num_items = 2,
 	.items = {
 		{ "Mic", 0x1 },
@@ -17475,7 +17522,7 @@ static struct hda_input_mux alc662_lenovo_101e_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc663_capture_source = {
+static const struct hda_input_mux alc663_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic", 0x0 },
@@ -17485,7 +17532,7 @@ static struct hda_input_mux alc663_capture_source = {
 };
 
 #if 0 /* set to 1 for testing other input sources below */
-static struct hda_input_mux alc272_nc10_capture_source = {
+static const struct hda_input_mux alc272_nc10_capture_source = {
 	.num_items = 16,
 	.items = {
 		{ "Autoselect Mic", 0x0 },
@@ -17511,14 +17558,14 @@ static struct hda_input_mux alc272_nc10_capture_source = {
 /*
  * 2ch mode
  */
-static struct hda_channel_mode alc662_3ST_2ch_modes[1] = {
+static const struct hda_channel_mode alc662_3ST_2ch_modes[1] = {
 	{ 2, NULL }
 };
 
 /*
  * 2ch mode
  */
-static struct hda_verb alc662_3ST_ch2_init[] = {
+static const struct hda_verb alc662_3ST_ch2_init[] = {
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
 	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
@@ -17529,7 +17576,7 @@ static struct hda_verb alc662_3ST_ch2_init[] = {
 /*
  * 6ch mode
  */
-static struct hda_verb alc662_3ST_ch6_init[] = {
+static const struct hda_verb alc662_3ST_ch6_init[] = {
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
 	{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
@@ -17539,7 +17586,7 @@ static struct hda_verb alc662_3ST_ch6_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc662_3ST_6ch_modes[2] = {
+static const struct hda_channel_mode alc662_3ST_6ch_modes[2] = {
 	{ 2, alc662_3ST_ch2_init },
 	{ 6, alc662_3ST_ch6_init },
 };
@@ -17547,7 +17594,7 @@ static struct hda_channel_mode alc662_3ST_6ch_modes[2] = {
 /*
  * 2ch mode
  */
-static struct hda_verb alc662_sixstack_ch6_init[] = {
+static const struct hda_verb alc662_sixstack_ch6_init[] = {
 	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
 	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
 	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
@@ -17557,14 +17604,14 @@ static struct hda_verb alc662_sixstack_ch6_init[] = {
 /*
  * 6ch mode
  */
-static struct hda_verb alc662_sixstack_ch8_init[] = {
+static const struct hda_verb alc662_sixstack_ch8_init[] = {
 	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc662_5stack_modes[2] = {
+static const struct hda_channel_mode alc662_5stack_modes[2] = {
 	{ 2, alc662_sixstack_ch6_init },
 	{ 6, alc662_sixstack_ch8_init },
 };
@@ -17573,7 +17620,7 @@ static struct hda_channel_mode alc662_5stack_modes[2] = {
  *                 Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
  */
 
-static struct snd_kcontrol_new alc662_base_mixer[] = {
+static const struct snd_kcontrol_new alc662_base_mixer[] = {
 	/* output mixer control */
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
@@ -17597,7 +17644,7 @@ static struct snd_kcontrol_new alc662_base_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = {
+static const struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
@@ -17612,7 +17659,7 @@ static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = {
+static const struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
@@ -17633,7 +17680,7 @@ static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = {
+static const struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x03, 0x0, HDA_OUTPUT),
@@ -17646,7 +17693,7 @@ static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
+static const struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	ALC262_HIPPO_MASTER_SWITCH,
 
@@ -17660,7 +17707,7 @@ static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = {
+static const struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = {
 	ALC262_HIPPO_MASTER_SWITCH,
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
@@ -17674,7 +17721,7 @@ static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_bind_ctls alc663_asus_bind_master_vol = {
+static const struct hda_bind_ctls alc663_asus_bind_master_vol = {
 	.ops = &snd_hda_bind_vol,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
@@ -17683,7 +17730,7 @@ static struct hda_bind_ctls alc663_asus_bind_master_vol = {
 	},
 };
 
-static struct hda_bind_ctls alc663_asus_one_bind_switch = {
+static const struct hda_bind_ctls alc663_asus_one_bind_switch = {
 	.ops = &snd_hda_bind_sw,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
@@ -17692,7 +17739,7 @@ static struct hda_bind_ctls alc663_asus_one_bind_switch = {
 	},
 };
 
-static struct snd_kcontrol_new alc663_m51va_mixer[] = {
+static const struct snd_kcontrol_new alc663_m51va_mixer[] = {
 	HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
 	HDA_BIND_SW("Master Playback Switch", &alc663_asus_one_bind_switch),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
@@ -17700,7 +17747,7 @@ static struct snd_kcontrol_new alc663_m51va_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_bind_ctls alc663_asus_tree_bind_switch = {
+static const struct hda_bind_ctls alc663_asus_tree_bind_switch = {
 	.ops = &snd_hda_bind_sw,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
@@ -17710,7 +17757,7 @@ static struct hda_bind_ctls alc663_asus_tree_bind_switch = {
 	},
 };
 
-static struct snd_kcontrol_new alc663_two_hp_m1_mixer[] = {
+static const struct snd_kcontrol_new alc663_two_hp_m1_mixer[] = {
 	HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
 	HDA_BIND_SW("Master Playback Switch", &alc663_asus_tree_bind_switch),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
@@ -17721,7 +17768,7 @@ static struct snd_kcontrol_new alc663_two_hp_m1_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_bind_ctls alc663_asus_four_bind_switch = {
+static const struct hda_bind_ctls alc663_asus_four_bind_switch = {
 	.ops = &snd_hda_bind_sw,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
@@ -17731,7 +17778,7 @@ static struct hda_bind_ctls alc663_asus_four_bind_switch = {
 	},
 };
 
-static struct snd_kcontrol_new alc663_two_hp_m2_mixer[] = {
+static const struct snd_kcontrol_new alc663_two_hp_m2_mixer[] = {
 	HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
 	HDA_BIND_SW("Master Playback Switch", &alc663_asus_four_bind_switch),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
@@ -17741,7 +17788,7 @@ static struct snd_kcontrol_new alc663_two_hp_m2_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc662_1bjd_mixer[] = {
+static const struct snd_kcontrol_new alc662_1bjd_mixer[] = {
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
@@ -17752,7 +17799,7 @@ static struct snd_kcontrol_new alc662_1bjd_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_bind_ctls alc663_asus_two_bind_master_vol = {
+static const struct hda_bind_ctls alc663_asus_two_bind_master_vol = {
 	.ops = &snd_hda_bind_vol,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
@@ -17761,7 +17808,7 @@ static struct hda_bind_ctls alc663_asus_two_bind_master_vol = {
 	},
 };
 
-static struct hda_bind_ctls alc663_asus_two_bind_switch = {
+static const struct hda_bind_ctls alc663_asus_two_bind_switch = {
 	.ops = &snd_hda_bind_sw,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
@@ -17770,7 +17817,7 @@ static struct hda_bind_ctls alc663_asus_two_bind_switch = {
 	},
 };
 
-static struct snd_kcontrol_new alc663_asus_21jd_clfe_mixer[] = {
+static const struct snd_kcontrol_new alc663_asus_21jd_clfe_mixer[] = {
 	HDA_BIND_VOL("Master Playback Volume",
 				&alc663_asus_two_bind_master_vol),
 	HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch),
@@ -17781,7 +17828,7 @@ static struct snd_kcontrol_new alc663_asus_21jd_clfe_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc663_asus_15jd_clfe_mixer[] = {
+static const struct snd_kcontrol_new alc663_asus_15jd_clfe_mixer[] = {
 	HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
 	HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
@@ -17791,7 +17838,7 @@ static struct snd_kcontrol_new alc663_asus_15jd_clfe_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc663_g71v_mixer[] = {
+static const struct snd_kcontrol_new alc663_g71v_mixer[] = {
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x03, 0x0, HDA_OUTPUT),
@@ -17805,7 +17852,7 @@ static struct snd_kcontrol_new alc663_g71v_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc663_g50v_mixer[] = {
+static const struct snd_kcontrol_new alc663_g50v_mixer[] = {
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
@@ -17819,7 +17866,7 @@ static struct snd_kcontrol_new alc663_g50v_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_bind_ctls alc663_asus_mode7_8_all_bind_switch = {
+static const struct hda_bind_ctls alc663_asus_mode7_8_all_bind_switch = {
 	.ops = &snd_hda_bind_sw,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
@@ -17831,7 +17878,7 @@ static struct hda_bind_ctls alc663_asus_mode7_8_all_bind_switch = {
 	},
 };
 
-static struct hda_bind_ctls alc663_asus_mode7_8_sp_bind_switch = {
+static const struct hda_bind_ctls alc663_asus_mode7_8_sp_bind_switch = {
 	.ops = &snd_hda_bind_sw,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
@@ -17840,7 +17887,7 @@ static struct hda_bind_ctls alc663_asus_mode7_8_sp_bind_switch = {
 	},
 };
 
-static struct snd_kcontrol_new alc663_mode7_mixer[] = {
+static const struct snd_kcontrol_new alc663_mode7_mixer[] = {
 	HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
 	HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
 	HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
@@ -17853,7 +17900,7 @@ static struct snd_kcontrol_new alc663_mode7_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc663_mode8_mixer[] = {
+static const struct snd_kcontrol_new alc663_mode8_mixer[] = {
 	HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
 	HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
 	HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
@@ -17865,7 +17912,7 @@ static struct snd_kcontrol_new alc663_mode8_mixer[] = {
 };
 
 
-static struct snd_kcontrol_new alc662_chmode_mixer[] = {
+static const struct snd_kcontrol_new alc662_chmode_mixer[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Channel Mode",
@@ -17876,7 +17923,7 @@ static struct snd_kcontrol_new alc662_chmode_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc662_init_verbs[] = {
+static const struct hda_verb alc662_init_verbs[] = {
 	/* ADC: mute amp left and right */
 	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -17922,55 +17969,36 @@ static struct hda_verb alc662_init_verbs[] = {
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 
-	/* always trun on EAPD */
-	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
-	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
-
-	{ }
-};
-
-static struct hda_verb alc663_init_verbs[] = {
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 	{ }
 };
 
-static struct hda_verb alc272_init_verbs[] = {
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+static const struct hda_verb alc662_eapd_init_verbs[] = {
+	/* always trun on EAPD */
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
 	{ }
 };
 
-static struct hda_verb alc662_sue_init_verbs[] = {
+static const struct hda_verb alc662_sue_init_verbs[] = {
 	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_FRONT_EVENT},
 	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT},
 	{}
 };
 
-static struct hda_verb alc662_eeepc_sue_init_verbs[] = {
+static const struct hda_verb alc662_eeepc_sue_init_verbs[] = {
 	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
 	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
 	{}
 };
 
 /* Set Unsolicited Event*/
-static struct hda_verb alc662_eeepc_ep20_sue_init_verbs[] = {
+static const struct hda_verb alc662_eeepc_ep20_sue_init_verbs[] = {
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
 	{}
 };
 
-static struct hda_verb alc663_m51va_init_verbs[] = {
+static const struct hda_verb alc663_m51va_init_verbs[] = {
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -17983,7 +18011,7 @@ static struct hda_verb alc663_m51va_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc663_21jd_amic_init_verbs[] = {
+static const struct hda_verb alc663_21jd_amic_init_verbs[] = {
 	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
@@ -17994,7 +18022,7 @@ static struct hda_verb alc663_21jd_amic_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc662_1bjd_amic_init_verbs[] = {
+static const struct hda_verb alc662_1bjd_amic_init_verbs[] = {
 	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -18006,7 +18034,7 @@ static struct hda_verb alc662_1bjd_amic_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc663_15jd_amic_init_verbs[] = {
+static const struct hda_verb alc663_15jd_amic_init_verbs[] = {
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
@@ -18017,7 +18045,7 @@ static struct hda_verb alc663_15jd_amic_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc663_two_hp_amic_m1_init_verbs[] = {
+static const struct hda_verb alc663_two_hp_amic_m1_init_verbs[] = {
 	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -18033,7 +18061,7 @@ static struct hda_verb alc663_two_hp_amic_m1_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc663_two_hp_amic_m2_init_verbs[] = {
+static const struct hda_verb alc663_two_hp_amic_m2_init_verbs[] = {
 	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -18049,7 +18077,7 @@ static struct hda_verb alc663_two_hp_amic_m2_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc663_g71v_init_verbs[] = {
+static const struct hda_verb alc663_g71v_init_verbs[] = {
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	/* {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */
 	/* {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, */ /* Headphone */
@@ -18064,7 +18092,7 @@ static struct hda_verb alc663_g71v_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc663_g50v_init_verbs[] = {
+static const struct hda_verb alc663_g50v_init_verbs[] = {
 	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x21, AC_VERB_SET_CONNECT_SEL, 0x00},	/* Headphone */
@@ -18074,7 +18102,7 @@ static struct hda_verb alc663_g50v_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc662_ecs_init_verbs[] = {
+static const struct hda_verb alc662_ecs_init_verbs[] = {
 	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f},
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
@@ -18082,7 +18110,7 @@ static struct hda_verb alc662_ecs_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc272_dell_zm1_init_verbs[] = {
+static const struct hda_verb alc272_dell_zm1_init_verbs[] = {
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
@@ -18097,7 +18125,7 @@ static struct hda_verb alc272_dell_zm1_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc272_dell_init_verbs[] = {
+static const struct hda_verb alc272_dell_init_verbs[] = {
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
@@ -18112,7 +18140,7 @@ static struct hda_verb alc272_dell_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc663_mode7_init_verbs[] = {
+static const struct hda_verb alc663_mode7_init_verbs[] = {
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
@@ -18131,7 +18159,7 @@ static struct hda_verb alc663_mode7_init_verbs[] = {
 	{}
 };
 
-static struct hda_verb alc663_mode8_init_verbs[] = {
+static const struct hda_verb alc663_mode8_init_verbs[] = {
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -18151,61 +18179,29 @@ static struct hda_verb alc663_mode8_init_verbs[] = {
 	{}
 };
 
-static struct snd_kcontrol_new alc662_auto_capture_mixer[] = {
+static const struct snd_kcontrol_new alc662_auto_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc272_auto_capture_mixer[] = {
+static const struct snd_kcontrol_new alc272_auto_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
 	{ } /* end */
 };
 
-static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
+static void alc662_lenovo_101e_setup(struct hda_codec *codec)
 {
-	unsigned int present;
-	unsigned char bits;
-
-	present = snd_hda_jack_detect(codec, 0x14);
-	bits = present ? HDA_AMP_MUTE : 0;
-
-	snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
-}
-
-static void alc662_lenovo_101e_all_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-	unsigned char bits;
-
- 	present = snd_hda_jack_detect(codec, 0x1b);
-	bits = present ? HDA_AMP_MUTE : 0;
-
-	snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
-	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
-}
-
-static void alc662_lenovo_101e_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
-{
-	if ((res >> 26) == ALC880_HP_EVENT)
-		alc662_lenovo_101e_all_automute(codec);
-	if ((res >> 26) == ALC880_FRONT_EVENT)
-		alc662_lenovo_101e_ispeaker_automute(codec);
-}
+	struct alc_spec *spec = codec->spec;
 
-/* unsolicited event for HP jack sensing */
-static void alc662_eeepc_unsol_event(struct hda_codec *codec,
-				     unsigned int res)
-{
-	if ((res >> 26) == ALC880_MIC_EVENT)
-		alc_mic_automute(codec);
-	else
-		alc262_hippo_unsol_event(codec, res);
+	spec->autocfg.hp_pins[0] = 0x1b;
+	spec->autocfg.line_out_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[0] = 0x15;
+	spec->automute = 1;
+	spec->detect_line = 1;
+	spec->automute_lines = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc662_eeepc_setup(struct hda_codec *codec)
@@ -18220,180 +18216,24 @@ static void alc662_eeepc_setup(struct hda_codec *codec)
 	spec->auto_mic = 1;
 }
 
-static void alc662_eeepc_inithook(struct hda_codec *codec)
-{
-	alc262_hippo_automute(codec);
-	alc_mic_automute(codec);
-}
-
 static void alc662_eeepc_ep20_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x1b;
-}
-
-#define alc662_eeepc_ep20_inithook	alc262_hippo_master_update
-
-static void alc663_m51va_speaker_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-	unsigned char bits;
-
-	present = snd_hda_jack_detect(codec, 0x21);
-	bits = present ? HDA_AMP_MUTE : 0;
-	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				 HDA_AMP_MUTE, bits);
-	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				 HDA_AMP_MUTE, bits);
-}
-
-static void alc663_21jd_two_speaker_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-	unsigned char bits;
-
-	present = snd_hda_jack_detect(codec, 0x21);
-	bits = present ? HDA_AMP_MUTE : 0;
-	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				 HDA_AMP_MUTE, bits);
-	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				 HDA_AMP_MUTE, bits);
-	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
-				 HDA_AMP_MUTE, bits);
-	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
-				 HDA_AMP_MUTE, bits);
-}
-
-static void alc663_15jd_two_speaker_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-	unsigned char bits;
-
-	present = snd_hda_jack_detect(codec, 0x15);
-	bits = present ? HDA_AMP_MUTE : 0;
-	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				 HDA_AMP_MUTE, bits);
-	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				 HDA_AMP_MUTE, bits);
-	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
-				 HDA_AMP_MUTE, bits);
-	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
-				 HDA_AMP_MUTE, bits);
-}
-
-static void alc662_f5z_speaker_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-	unsigned char bits;
-
-	present = snd_hda_jack_detect(codec, 0x1b);
-	bits = present ? 0 : PIN_OUT;
-	snd_hda_codec_write(codec, 0x14, 0,
-			 AC_VERB_SET_PIN_WIDGET_CONTROL, bits);
-}
-
-static void alc663_two_hp_m1_speaker_automute(struct hda_codec *codec)
-{
-	unsigned int present1, present2;
-
-	present1 = snd_hda_jack_detect(codec, 0x21);
-	present2 = snd_hda_jack_detect(codec, 0x15);
-
-	if (present1 || present2) {
-		snd_hda_codec_write_cache(codec, 0x14, 0,
-			AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
-	} else {
-		snd_hda_codec_write_cache(codec, 0x14, 0,
-			AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
-	}
-}
-
-static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec)
-{
-	unsigned int present1, present2;
-
-	present1 = snd_hda_jack_detect(codec, 0x1b);
-	present2 = snd_hda_jack_detect(codec, 0x15);
-
-	if (present1 || present2) {
-		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-					 HDA_AMP_MUTE, HDA_AMP_MUTE);
-		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-					 HDA_AMP_MUTE, HDA_AMP_MUTE);
-	} else {
-		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-					 HDA_AMP_MUTE, 0);
-		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-					 HDA_AMP_MUTE, 0);
-	}
-}
-
-static void alc663_two_hp_m7_speaker_automute(struct hda_codec *codec)
-{
-	unsigned int present1, present2;
-
-	present1 = snd_hda_codec_read(codec, 0x1b, 0,
-			AC_VERB_GET_PIN_SENSE, 0)
-			& AC_PINSENSE_PRESENCE;
-	present2 = snd_hda_codec_read(codec, 0x21, 0,
-			AC_VERB_GET_PIN_SENSE, 0)
-			& AC_PINSENSE_PRESENCE;
-
-	if (present1 || present2) {
-		snd_hda_codec_write_cache(codec, 0x14, 0,
-			AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_codec_write_cache(codec, 0x17, 0,
-			AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
-	} else {
-		snd_hda_codec_write_cache(codec, 0x14, 0,
-			AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
-		snd_hda_codec_write_cache(codec, 0x17, 0,
-			AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
-	}
-}
-
-static void alc663_two_hp_m8_speaker_automute(struct hda_codec *codec)
-{
-	unsigned int present1, present2;
-
-	present1 = snd_hda_codec_read(codec, 0x21, 0,
-			AC_VERB_GET_PIN_SENSE, 0)
-			& AC_PINSENSE_PRESENCE;
-	present2 = snd_hda_codec_read(codec, 0x15, 0,
-			AC_VERB_GET_PIN_SENSE, 0)
-			& AC_PINSENSE_PRESENCE;
-
-	if (present1 || present2) {
-		snd_hda_codec_write_cache(codec, 0x14, 0,
-			AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_codec_write_cache(codec, 0x17, 0,
-			AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
-	} else {
-		snd_hda_codec_write_cache(codec, 0x14, 0,
-			AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
-		snd_hda_codec_write_cache(codec, 0x17, 0,
-			AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
-	}
-}
-
-static void alc663_m51va_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
-{
-	switch (res >> 26) {
-	case ALC880_HP_EVENT:
-		alc663_m51va_speaker_automute(codec);
-		break;
-	case ALC880_MIC_EVENT:
-		alc_mic_automute(codec);
-		break;
-	}
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc663_m51va_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
+	spec->autocfg.hp_pins[0] = 0x21;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute_mixer_nid[0] = 0x0c;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_MIXER;
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
 	spec->int_mic.pin = 0x12;
@@ -18401,18 +18241,15 @@ static void alc663_m51va_setup(struct hda_codec *codec)
 	spec->auto_mic = 1;
 }
 
-static void alc663_m51va_inithook(struct hda_codec *codec)
-{
-	alc663_m51va_speaker_automute(codec);
-	alc_mic_automute(codec);
-}
-
 /* ***************** Mode1 ******************************/
-#define alc663_mode1_unsol_event	alc663_m51va_unsol_event
-
 static void alc663_mode1_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
+	spec->autocfg.hp_pins[0] = 0x21;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute_mixer_nid[0] = 0x0c;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_MIXER;
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
 	spec->int_mic.pin = 0x19;
@@ -18420,229 +18257,144 @@ static void alc663_mode1_setup(struct hda_codec *codec)
 	spec->auto_mic = 1;
 }
 
-#define alc663_mode1_inithook		alc663_m51va_inithook
-
 /* ***************** Mode2 ******************************/
-static void alc662_mode2_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
+static void alc662_mode2_setup(struct hda_codec *codec)
 {
-	switch (res >> 26) {
-	case ALC880_HP_EVENT:
-		alc662_f5z_speaker_automute(codec);
-		break;
-	case ALC880_MIC_EVENT:
-		alc_mic_automute(codec);
-		break;
-	}
+	struct alc_spec *spec = codec->spec;
+	spec->autocfg.hp_pins[0] = 0x1b;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x19;
+	spec->int_mic.mux_idx = 1;
+	spec->auto_mic = 1;
 }
 
-#define alc662_mode2_setup	alc663_mode1_setup
-
-static void alc662_mode2_inithook(struct hda_codec *codec)
-{
-	alc662_f5z_speaker_automute(codec);
-	alc_mic_automute(codec);
-}
 /* ***************** Mode3 ******************************/
-static void alc663_mode3_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
+static void alc663_mode3_setup(struct hda_codec *codec)
 {
-	switch (res >> 26) {
-	case ALC880_HP_EVENT:
-		alc663_two_hp_m1_speaker_automute(codec);
-		break;
-	case ALC880_MIC_EVENT:
-		alc_mic_automute(codec);
-		break;
-	}
+	struct alc_spec *spec = codec->spec;
+	spec->autocfg.hp_pins[0] = 0x21;
+	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x19;
+	spec->int_mic.mux_idx = 1;
+	spec->auto_mic = 1;
 }
 
-#define alc663_mode3_setup	alc663_mode1_setup
-
-static void alc663_mode3_inithook(struct hda_codec *codec)
-{
-	alc663_two_hp_m1_speaker_automute(codec);
-	alc_mic_automute(codec);
-}
 /* ***************** Mode4 ******************************/
-static void alc663_mode4_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
+static void alc663_mode4_setup(struct hda_codec *codec)
 {
-	switch (res >> 26) {
-	case ALC880_HP_EVENT:
-		alc663_21jd_two_speaker_automute(codec);
-		break;
-	case ALC880_MIC_EVENT:
-		alc_mic_automute(codec);
-		break;
-	}
+	struct alc_spec *spec = codec->spec;
+	spec->autocfg.hp_pins[0] = 0x21;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[1] = 0x16;
+	spec->automute_mixer_nid[0] = 0x0c;
+	spec->automute_mixer_nid[1] = 0x0e;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_MIXER;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x19;
+	spec->int_mic.mux_idx = 1;
+	spec->auto_mic = 1;
 }
 
-#define alc663_mode4_setup	alc663_mode1_setup
-
-static void alc663_mode4_inithook(struct hda_codec *codec)
-{
-	alc663_21jd_two_speaker_automute(codec);
-	alc_mic_automute(codec);
-}
 /* ***************** Mode5 ******************************/
-static void alc663_mode5_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
+static void alc663_mode5_setup(struct hda_codec *codec)
 {
-	switch (res >> 26) {
-	case ALC880_HP_EVENT:
-		alc663_15jd_two_speaker_automute(codec);
-		break;
-	case ALC880_MIC_EVENT:
-		alc_mic_automute(codec);
-		break;
-	}
+	struct alc_spec *spec = codec->spec;
+	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[1] = 0x16;
+	spec->automute_mixer_nid[0] = 0x0c;
+	spec->automute_mixer_nid[1] = 0x0e;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_MIXER;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x19;
+	spec->int_mic.mux_idx = 1;
+	spec->auto_mic = 1;
 }
 
-#define alc663_mode5_setup	alc663_mode1_setup
-
-static void alc663_mode5_inithook(struct hda_codec *codec)
-{
-	alc663_15jd_two_speaker_automute(codec);
-	alc_mic_automute(codec);
-}
 /* ***************** Mode6 ******************************/
-static void alc663_mode6_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
-{
-	switch (res >> 26) {
-	case ALC880_HP_EVENT:
-		alc663_two_hp_m2_speaker_automute(codec);
-		break;
-	case ALC880_MIC_EVENT:
-		alc_mic_automute(codec);
-		break;
-	}
-}
-
-#define alc663_mode6_setup	alc663_mode1_setup
-
-static void alc663_mode6_inithook(struct hda_codec *codec)
+static void alc663_mode6_setup(struct hda_codec *codec)
 {
-	alc663_two_hp_m2_speaker_automute(codec);
-	alc_mic_automute(codec);
+	struct alc_spec *spec = codec->spec;
+	spec->autocfg.hp_pins[0] = 0x1b;
+	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute_mixer_nid[0] = 0x0c;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_MIXER;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x19;
+	spec->int_mic.mux_idx = 1;
+	spec->auto_mic = 1;
 }
 
 /* ***************** Mode7 ******************************/
-static void alc663_mode7_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
+static void alc663_mode7_setup(struct hda_codec *codec)
 {
-	switch (res >> 26) {
-	case ALC880_HP_EVENT:
-		alc663_two_hp_m7_speaker_automute(codec);
-		break;
-	case ALC880_MIC_EVENT:
-		alc_mic_automute(codec);
-		break;
-	}
-}
-
-#define alc663_mode7_setup	alc663_mode1_setup
-
-static void alc663_mode7_inithook(struct hda_codec *codec)
-{
-	alc663_two_hp_m7_speaker_automute(codec);
-	alc_mic_automute(codec);
+	struct alc_spec *spec = codec->spec;
+	spec->autocfg.hp_pins[0] = 0x1b;
+	spec->autocfg.hp_pins[0] = 0x21;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[0] = 0x17;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x19;
+	spec->int_mic.mux_idx = 1;
+	spec->auto_mic = 1;
 }
 
 /* ***************** Mode8 ******************************/
-static void alc663_mode8_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
-{
-	switch (res >> 26) {
-	case ALC880_HP_EVENT:
-		alc663_two_hp_m8_speaker_automute(codec);
-		break;
-	case ALC880_MIC_EVENT:
-		alc_mic_automute(codec);
-		break;
-	}
-}
-
-#define alc663_mode8_setup	alc663_m51va_setup
-
-static void alc663_mode8_inithook(struct hda_codec *codec)
-{
-	alc663_two_hp_m8_speaker_automute(codec);
-	alc_mic_automute(codec);
-}
-
-static void alc663_g71v_hp_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-	unsigned char bits;
-
-	present = snd_hda_jack_detect(codec, 0x21);
-	bits = present ? HDA_AMP_MUTE : 0;
-	snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
-	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
-}
-
-static void alc663_g71v_front_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-	unsigned char bits;
-
-	present = snd_hda_jack_detect(codec, 0x15);
-	bits = present ? HDA_AMP_MUTE : 0;
-	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
-}
-
-static void alc663_g71v_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
-{
-	switch (res >> 26) {
-	case ALC880_HP_EVENT:
-		alc663_g71v_hp_automute(codec);
-		break;
-	case ALC880_FRONT_EVENT:
-		alc663_g71v_front_automute(codec);
-		break;
-	case ALC880_MIC_EVENT:
-		alc_mic_automute(codec);
-		break;
-	}
-}
-
-#define alc663_g71v_setup	alc663_m51va_setup
-
-static void alc663_g71v_inithook(struct hda_codec *codec)
+static void alc663_mode8_setup(struct hda_codec *codec)
 {
-	alc663_g71v_front_automute(codec);
-	alc663_g71v_hp_automute(codec);
-	alc_mic_automute(codec);
+	struct alc_spec *spec = codec->spec;
+	spec->autocfg.hp_pins[0] = 0x21;
+	spec->autocfg.hp_pins[1] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[0] = 0x17;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x12;
+	spec->int_mic.mux_idx = 9;
+	spec->auto_mic = 1;
 }
 
-static void alc663_g50v_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
+static void alc663_g71v_setup(struct hda_codec *codec)
 {
-	switch (res >> 26) {
-	case ALC880_HP_EVENT:
-		alc663_m51va_speaker_automute(codec);
-		break;
-	case ALC880_MIC_EVENT:
-		alc_mic_automute(codec);
-		break;
-	}
+	struct alc_spec *spec = codec->spec;
+	spec->autocfg.hp_pins[0] = 0x21;
+	spec->autocfg.line_out_pins[0] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	spec->detect_line = 1;
+	spec->automute_lines = 1;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x12;
+	spec->int_mic.mux_idx = 9;
+	spec->auto_mic = 1;
 }
 
 #define alc663_g50v_setup	alc663_m51va_setup
 
-static void alc663_g50v_inithook(struct hda_codec *codec)
-{
-	alc663_m51va_speaker_automute(codec);
-	alc_mic_automute(codec);
-}
-
-static struct snd_kcontrol_new alc662_ecs_mixer[] = {
+static const struct snd_kcontrol_new alc662_ecs_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	ALC262_HIPPO_MASTER_SWITCH,
 
@@ -18656,7 +18408,7 @@ static struct snd_kcontrol_new alc662_ecs_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc272_nc10_mixer[] = {
+static const struct snd_kcontrol_new alc272_nc10_mixer[] = {
 	/* Master Playback automatically created from Speaker and Headphone */
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
@@ -18691,7 +18443,7 @@ static const char * const alc662_models[ALC662_MODEL_LAST] = {
 	[ALC662_3ST_2ch_DIG]	= "3stack-dig",
 	[ALC662_3ST_6ch_DIG]	= "3stack-6ch-dig",
 	[ALC662_3ST_6ch]	= "3stack-6ch",
-	[ALC662_5ST_DIG]	= "6stack-dig",
+	[ALC662_5ST_DIG]	= "5stack-dig",
 	[ALC662_LENOVO_101E]	= "lenovo-101e",
 	[ALC662_ASUS_EEEPC_P701] = "eeepc-p701",
 	[ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20",
@@ -18714,7 +18466,7 @@ static const char * const alc662_models[ALC662_MODEL_LAST] = {
 	[ALC662_AUTO]		= "auto",
 };
 
-static struct snd_pci_quirk alc662_cfg_tbl[] = {
+static const struct snd_pci_quirk alc662_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS),
 	SND_PCI_QUIRK(0x1028, 0x02d6, "DELL", ALC272_DELL),
 	SND_PCI_QUIRK(0x1028, 0x02f4, "DELL ZM1", ALC272_DELL_ZM1),
@@ -18782,6 +18534,8 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
 		      ALC662_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
 	SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
+	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
+		      ALC662_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13),
 	SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA),
@@ -18794,10 +18548,10 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
 	{}
 };
 
-static struct alc_config_preset alc662_presets[] = {
+static const struct alc_config_preset alc662_presets[] = {
 	[ALC662_3ST_2ch_DIG] = {
 		.mixers = { alc662_3ST_2ch_mixer },
-		.init_verbs = { alc662_init_verbs },
+		.init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
 		.dig_out_nid = ALC662_DIGOUT_NID,
@@ -18808,7 +18562,7 @@ static struct alc_config_preset alc662_presets[] = {
 	},
 	[ALC662_3ST_6ch_DIG] = {
 		.mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
-		.init_verbs = { alc662_init_verbs },
+		.init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
 		.dig_out_nid = ALC662_DIGOUT_NID,
@@ -18820,7 +18574,7 @@ static struct alc_config_preset alc662_presets[] = {
 	},
 	[ALC662_3ST_6ch] = {
 		.mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
-		.init_verbs = { alc662_init_verbs },
+		.init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
@@ -18830,7 +18584,7 @@ static struct alc_config_preset alc662_presets[] = {
 	},
 	[ALC662_5ST_DIG] = {
 		.mixers = { alc662_base_mixer, alc662_chmode_mixer },
-		.init_verbs = { alc662_init_verbs },
+		.init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
 		.dig_out_nid = ALC662_DIGOUT_NID,
@@ -18841,104 +18595,120 @@ static struct alc_config_preset alc662_presets[] = {
 	},
 	[ALC662_LENOVO_101E] = {
 		.mixers = { alc662_lenovo_101e_mixer },
-		.init_verbs = { alc662_init_verbs, alc662_sue_init_verbs },
+		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
+				alc662_sue_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
 		.input_mux = &alc662_lenovo_101e_capture_source,
-		.unsol_event = alc662_lenovo_101e_unsol_event,
-		.init_hook = alc662_lenovo_101e_all_automute,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc662_lenovo_101e_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC662_ASUS_EEEPC_P701] = {
 		.mixers = { alc662_eeepc_p701_mixer },
 		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
 				alc662_eeepc_sue_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc662_eeepc_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc662_eeepc_setup,
-		.init_hook = alc662_eeepc_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC662_ASUS_EEEPC_EP20] = {
 		.mixers = { alc662_eeepc_ep20_mixer,
 			    alc662_chmode_mixer },
 		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
 				alc662_eeepc_ep20_sue_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
 		.channel_mode = alc662_3ST_6ch_modes,
 		.input_mux = &alc662_lenovo_101e_capture_source,
-		.unsol_event = alc662_eeepc_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc662_eeepc_ep20_setup,
-		.init_hook = alc662_eeepc_ep20_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC662_ECS] = {
 		.mixers = { alc662_ecs_mixer },
 		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
 				alc662_ecs_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc662_eeepc_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc662_eeepc_setup,
-		.init_hook = alc662_eeepc_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC663_ASUS_M51VA] = {
 		.mixers = { alc663_m51va_mixer },
-		.init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs },
+		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
+				alc663_m51va_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
 		.dig_out_nid = ALC662_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc663_m51va_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc663_m51va_setup,
-		.init_hook = alc663_m51va_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC663_ASUS_G71V] = {
 		.mixers = { alc663_g71v_mixer },
-		.init_verbs = { alc662_init_verbs, alc663_g71v_init_verbs },
+		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
+				alc663_g71v_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
 		.dig_out_nid = ALC662_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc663_g71v_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc663_g71v_setup,
-		.init_hook = alc663_g71v_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC663_ASUS_H13] = {
 		.mixers = { alc663_m51va_mixer },
-		.init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs },
+		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
+				alc663_m51va_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc663_m51va_unsol_event,
-		.init_hook = alc663_m51va_inithook,
+		.setup = alc663_m51va_setup,
+		.unsol_event = alc_sku_unsol_event,
+		.init_hook = alc_inithook,
 	},
 	[ALC663_ASUS_G50V] = {
 		.mixers = { alc663_g50v_mixer },
-		.init_verbs = { alc662_init_verbs, alc663_g50v_init_verbs },
+		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
+				alc663_g50v_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
 		.dig_out_nid = ALC662_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
 		.channel_mode = alc662_3ST_6ch_modes,
 		.input_mux = &alc663_capture_source,
-		.unsol_event = alc663_g50v_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc663_g50v_setup,
-		.init_hook = alc663_g50v_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC663_ASUS_MODE1] = {
 		.mixers = { alc663_m51va_mixer },
 		.cap_mixer = alc662_auto_capture_mixer,
 		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
 				alc663_21jd_amic_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.hp_nid = 0x03,
@@ -18946,28 +18716,30 @@ static struct alc_config_preset alc662_presets[] = {
 		.dig_out_nid = ALC662_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc663_mode1_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc663_mode1_setup,
-		.init_hook = alc663_mode1_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC662_ASUS_MODE2] = {
 		.mixers = { alc662_1bjd_mixer },
 		.cap_mixer = alc662_auto_capture_mixer,
 		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
 				alc662_1bjd_amic_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.dac_nids = alc662_dac_nids,
 		.dig_out_nid = ALC662_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc662_mode2_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc662_mode2_setup,
-		.init_hook = alc662_mode2_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC663_ASUS_MODE3] = {
 		.mixers = { alc663_two_hp_m1_mixer },
 		.cap_mixer = alc662_auto_capture_mixer,
 		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
 				alc663_two_hp_amic_m1_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.hp_nid = 0x03,
@@ -18975,14 +18747,15 @@ static struct alc_config_preset alc662_presets[] = {
 		.dig_out_nid = ALC662_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc663_mode3_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc663_mode3_setup,
-		.init_hook = alc663_mode3_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC663_ASUS_MODE4] = {
 		.mixers = { alc663_asus_21jd_clfe_mixer },
 		.cap_mixer = alc662_auto_capture_mixer,
 		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
 				alc663_21jd_amic_init_verbs},
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.hp_nid = 0x03,
@@ -18990,14 +18763,15 @@ static struct alc_config_preset alc662_presets[] = {
 		.dig_out_nid = ALC662_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc663_mode4_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc663_mode4_setup,
-		.init_hook = alc663_mode4_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC663_ASUS_MODE5] = {
 		.mixers = { alc663_asus_15jd_clfe_mixer },
 		.cap_mixer = alc662_auto_capture_mixer,
 		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
 				alc663_15jd_amic_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.hp_nid = 0x03,
@@ -19005,14 +18779,15 @@ static struct alc_config_preset alc662_presets[] = {
 		.dig_out_nid = ALC662_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc663_mode5_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc663_mode5_setup,
-		.init_hook = alc663_mode5_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC663_ASUS_MODE6] = {
 		.mixers = { alc663_two_hp_m2_mixer },
 		.cap_mixer = alc662_auto_capture_mixer,
 		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
 				alc663_two_hp_amic_m2_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.hp_nid = 0x03,
@@ -19020,14 +18795,15 @@ static struct alc_config_preset alc662_presets[] = {
 		.dig_out_nid = ALC662_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc663_mode6_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc663_mode6_setup,
-		.init_hook = alc663_mode6_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC663_ASUS_MODE7] = {
 		.mixers = { alc663_mode7_mixer },
 		.cap_mixer = alc662_auto_capture_mixer,
 		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
 				alc663_mode7_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.hp_nid = 0x03,
@@ -19035,14 +18811,15 @@ static struct alc_config_preset alc662_presets[] = {
 		.dig_out_nid = ALC662_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc663_mode7_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc663_mode7_setup,
-		.init_hook = alc663_mode7_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC663_ASUS_MODE8] = {
 		.mixers = { alc663_mode8_mixer },
 		.cap_mixer = alc662_auto_capture_mixer,
 		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
 				alc663_mode8_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc662_dac_nids),
 		.hp_nid = 0x03,
@@ -19050,52 +18827,57 @@ static struct alc_config_preset alc662_presets[] = {
 		.dig_out_nid = ALC662_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc663_mode8_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc663_mode8_setup,
-		.init_hook = alc663_mode8_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC272_DELL] = {
 		.mixers = { alc663_m51va_mixer },
 		.cap_mixer = alc272_auto_capture_mixer,
-		.init_verbs = { alc662_init_verbs, alc272_dell_init_verbs },
+		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
+				alc272_dell_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc272_dac_nids),
-		.dac_nids = alc662_dac_nids,
+		.dac_nids = alc272_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.adc_nids = alc272_adc_nids,
 		.num_adc_nids = ARRAY_SIZE(alc272_adc_nids),
 		.capsrc_nids = alc272_capsrc_nids,
 		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc663_m51va_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc663_m51va_setup,
-		.init_hook = alc663_m51va_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC272_DELL_ZM1] = {
 		.mixers = { alc663_m51va_mixer },
 		.cap_mixer = alc662_auto_capture_mixer,
-		.init_verbs = { alc662_init_verbs, alc272_dell_zm1_init_verbs },
+		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
+				alc272_dell_zm1_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc272_dac_nids),
-		.dac_nids = alc662_dac_nids,
+		.dac_nids = alc272_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.adc_nids = alc662_adc_nids,
 		.num_adc_nids = 1,
 		.capsrc_nids = alc662_capsrc_nids,
 		.channel_mode = alc662_3ST_2ch_modes,
-		.unsol_event = alc663_m51va_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc663_m51va_setup,
-		.init_hook = alc663_m51va_inithook,
+		.init_hook = alc_inithook,
 	},
 	[ALC272_SAMSUNG_NC10] = {
 		.mixers = { alc272_nc10_mixer },
 		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
 				alc663_21jd_amic_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc272_dac_nids),
 		.dac_nids = alc272_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
 		/*.input_mux = &alc272_nc10_capture_source,*/
-		.unsol_event = alc663_mode4_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc663_mode4_setup,
-		.init_hook = alc663_mode4_inithook,
+		.init_hook = alc_inithook,
 	},
 };
 
@@ -19105,45 +18887,79 @@ static struct alc_config_preset alc662_presets[] = {
  */
 
 /* convert from MIX nid to DAC */
-static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid)
-{
-	if (nid == 0x0f)
-		return 0x02;
-	else if (nid >= 0x0c && nid <= 0x0e)
-		return nid - 0x0c + 0x02;
-	else if (nid == 0x26) /* ALC887-VD has this DAC too */
-		return 0x25;
-	else
-		return 0;
+static hda_nid_t alc_auto_mix_to_dac(struct hda_codec *codec, hda_nid_t nid)
+{
+	hda_nid_t list[5];
+	int i, num;
+
+	num = snd_hda_get_connections(codec, nid, list, ARRAY_SIZE(list));
+	for (i = 0; i < num; i++) {
+		if (get_wcaps_type(get_wcaps(codec, list[i])) == AC_WID_AUD_OUT)
+			return list[i];
+	}
+	return 0;
+}
+
+/* go down to the selector widget before the mixer */
+static hda_nid_t alc_go_down_to_selector(struct hda_codec *codec, hda_nid_t pin)
+{
+	hda_nid_t srcs[5];
+	int num = snd_hda_get_connections(codec, pin, srcs,
+					  ARRAY_SIZE(srcs));
+	if (num != 1 ||
+	    get_wcaps_type(get_wcaps(codec, srcs[0])) != AC_WID_AUD_SEL)
+		return pin;
+	return srcs[0];
 }
 
 /* get MIX nid connected to the given pin targeted to DAC */
-static hda_nid_t alc662_dac_to_mix(struct hda_codec *codec, hda_nid_t pin,
+static hda_nid_t alc_auto_dac_to_mix(struct hda_codec *codec, hda_nid_t pin,
 				   hda_nid_t dac)
 {
 	hda_nid_t mix[5];
 	int i, num;
 
+	pin = alc_go_down_to_selector(codec, pin);
 	num = snd_hda_get_connections(codec, pin, mix, ARRAY_SIZE(mix));
 	for (i = 0; i < num; i++) {
-		if (alc662_mix_to_dac(mix[i]) == dac)
+		if (alc_auto_mix_to_dac(codec, mix[i]) == dac)
 			return mix[i];
 	}
 	return 0;
 }
 
+/* select the connection from pin to DAC if needed */
+static int alc_auto_select_dac(struct hda_codec *codec, hda_nid_t pin,
+			       hda_nid_t dac)
+{
+	hda_nid_t mix[5];
+	int i, num;
+
+	pin = alc_go_down_to_selector(codec, pin);
+	num = snd_hda_get_connections(codec, pin, mix, ARRAY_SIZE(mix));
+	if (num < 2)
+		return 0;
+	for (i = 0; i < num; i++) {
+		if (alc_auto_mix_to_dac(codec, mix[i]) == dac) {
+			snd_hda_codec_update_cache(codec, pin, 0,
+						   AC_VERB_SET_CONNECT_SEL, i);
+			return 0;
+		}
+	}
+	return 0;
+}
+
 /* look for an empty DAC slot */
-static hda_nid_t alc662_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
+static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
 {
 	struct alc_spec *spec = codec->spec;
 	hda_nid_t srcs[5];
 	int i, j, num;
 
+	pin = alc_go_down_to_selector(codec, pin);
 	num = snd_hda_get_connections(codec, pin, srcs, ARRAY_SIZE(srcs));
-	if (num < 0)
-		return 0;
 	for (i = 0; i < num; i++) {
-		hda_nid_t nid = alc662_mix_to_dac(srcs[i]);
+		hda_nid_t nid = alc_auto_mix_to_dac(codec, srcs[i]);
 		if (!nid)
 			continue;
 		for (j = 0; j < spec->multiout.num_dacs; j++)
@@ -19165,10 +18981,10 @@ static int alc662_auto_fill_dac_nids(struct hda_codec *codec,
 
 	spec->multiout.dac_nids = spec->private_dac_nids;
 	for (i = 0; i < cfg->line_outs; i++) {
-		dac = alc662_look_for_dac(codec, cfg->line_out_pins[i]);
+		dac = alc_auto_look_for_dac(codec, cfg->line_out_pins[i]);
 		if (!dac)
 			continue;
-		spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
+		spec->private_dac_nids[spec->multiout.num_dacs++] = dac;
 	}
 	return 0;
 }
@@ -19204,15 +19020,23 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec,
 	static const char * const chname[4] = {
 		"Front", "Surround", NULL /*CLFE*/, "Side"
 	};
-	const char *pfx = alc_get_line_out_pfx(cfg, true);
-	hda_nid_t nid, mix;
-	int i, err;
+	const char *pfx = alc_get_line_out_pfx(spec, true);
+	hda_nid_t nid, mix, pin;
+	int i, err, noutputs;
 
-	for (i = 0; i < cfg->line_outs; i++) {
+	noutputs = cfg->line_outs;
+	if (spec->multi_ios > 0)
+		noutputs += spec->multi_ios;
+
+	for (i = 0; i < noutputs; i++) {
 		nid = spec->multiout.dac_nids[i];
 		if (!nid)
 			continue;
-		mix = alc662_dac_to_mix(codec, cfg->line_out_pins[i], nid);
+		if (i >= cfg->line_outs)
+			pin = spec->multi_io[i - 1].pin;
+		else
+			pin = cfg->line_out_pins[i];
+		mix = alc_auto_dac_to_mix(codec, pin, nid);
 		if (!mix)
 			continue;
 		if (!pfx && i == 2) {
@@ -19258,7 +19082,7 @@ static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
 
 	if (!pin)
 		return 0;
-	nid = alc662_look_for_dac(codec, pin);
+	nid = alc_auto_look_for_dac(codec, pin);
 	if (!nid) {
 		/* the corresponding DAC is already occupied */
 		if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP))
@@ -19268,7 +19092,7 @@ static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
 				   HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
 	}
 
-	mix = alc662_dac_to_mix(codec, pin, nid);
+	mix = alc_auto_dac_to_mix(codec, pin, nid);
 	if (!mix)
 		return 0;
 	err = alc662_add_vol_ctl(spec, pfx, nid, 3);
@@ -19292,14 +19116,21 @@ static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
 	hda_nid_t srcs[HDA_MAX_CONNECTIONS];
 
 	alc_set_pin_output(codec, nid, pin_type);
-	/* need the manual connection? */
 	num = snd_hda_get_connections(codec, nid, srcs, ARRAY_SIZE(srcs));
-	if (num <= 1)
-		return;
 	for (i = 0; i < num; i++) {
-		if (alc662_mix_to_dac(srcs[i]) != dac)
+		if (alc_auto_mix_to_dac(codec, srcs[i]) != dac)
 			continue;
-		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, i);
+		/* need the manual connection? */
+		if (num > 1)
+			snd_hda_codec_write(codec, nid, 0,
+					    AC_VERB_SET_CONNECT_SEL, i);
+		/* unmute mixer widget inputs */
+		snd_hda_codec_write(codec, srcs[i], 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE,
+				    AMP_IN_UNMUTE(0));
+		snd_hda_codec_write(codec, srcs[i], 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE,
+				    AMP_IN_UNMUTE(1));
 		return;
 	}
 }
@@ -19356,11 +19187,164 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec)
 
 #define alc662_auto_init_input_src	alc882_auto_init_input_src
 
+/*
+ * multi-io helper
+ */
+static int alc_auto_fill_multi_ios(struct hda_codec *codec,
+				   unsigned int location)
+{
+	struct alc_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int type, i, num_pins = 0;
+
+	for (type = AUTO_PIN_LINE_IN; type >= AUTO_PIN_MIC; type--) {
+		for (i = 0; i < cfg->num_inputs; i++) {
+			hda_nid_t nid = cfg->inputs[i].pin;
+			hda_nid_t dac;
+			unsigned int defcfg, caps;
+			if (cfg->inputs[i].type != type)
+				continue;
+			defcfg = snd_hda_codec_get_pincfg(codec, nid);
+			if (get_defcfg_connect(defcfg) != AC_JACK_PORT_COMPLEX)
+				continue;
+			if (location && get_defcfg_location(defcfg) != location)
+				continue;
+			caps = snd_hda_query_pin_caps(codec, nid);
+			if (!(caps & AC_PINCAP_OUT))
+				continue;
+			dac = alc_auto_look_for_dac(codec, nid);
+			if (!dac)
+				continue;
+			spec->multi_io[num_pins].pin = nid;
+			spec->multi_io[num_pins].dac = dac;
+			num_pins++;
+			spec->private_dac_nids[spec->multiout.num_dacs++] = dac;
+		}
+	}
+	spec->multiout.num_dacs = 1;
+	if (num_pins < 2)
+		return 0;
+	return num_pins;
+}
+
+static int alc_auto_ch_mode_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = spec->multi_ios + 1;
+	if (uinfo->value.enumerated.item > spec->multi_ios)
+		uinfo->value.enumerated.item = spec->multi_ios;
+	sprintf(uinfo->value.enumerated.name, "%dch",
+		(uinfo->value.enumerated.item + 1) * 2);
+	return 0;
+}
+
+static int alc_auto_ch_mode_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+	ucontrol->value.enumerated.item[0] = (spec->ext_channel_count - 1) / 2;
+	return 0;
+}
+
+static int alc_set_multi_io(struct hda_codec *codec, int idx, bool output)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t nid = spec->multi_io[idx].pin;
+
+	if (!spec->multi_io[idx].ctl_in)
+		spec->multi_io[idx].ctl_in =
+			snd_hda_codec_read(codec, nid, 0,
+					   AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+	if (output) {
+		snd_hda_codec_update_cache(codec, nid, 0,
+					   AC_VERB_SET_PIN_WIDGET_CONTROL,
+					   PIN_OUT);
+		if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
+			snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+						 HDA_AMP_MUTE, 0);
+		alc_auto_select_dac(codec, nid, spec->multi_io[idx].dac);
+	} else {
+		if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
+			snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+						 HDA_AMP_MUTE, HDA_AMP_MUTE);
+		snd_hda_codec_update_cache(codec, nid, 0,
+					   AC_VERB_SET_PIN_WIDGET_CONTROL,
+					   spec->multi_io[idx].ctl_in);
+	}
+	return 0;
+}
+
+static int alc_auto_ch_mode_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+	int i, ch;
+
+	ch = ucontrol->value.enumerated.item[0];
+	if (ch < 0 || ch > spec->multi_ios)
+		return -EINVAL;
+	if (ch == (spec->ext_channel_count - 1) / 2)
+		return 0;
+	spec->ext_channel_count = (ch + 1) * 2;
+	for (i = 0; i < spec->multi_ios; i++)
+		alc_set_multi_io(codec, i, i < ch);
+	spec->multiout.max_channels = spec->ext_channel_count;
+	return 1;
+}
+
+static const struct snd_kcontrol_new alc_auto_channel_mode_enum = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Channel Mode",
+	.info = alc_auto_ch_mode_info,
+	.get = alc_auto_ch_mode_get,
+	.put = alc_auto_ch_mode_put,
+};
+
+static int alc_auto_add_multi_channel_mode(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	unsigned int location, defcfg;
+	int num_pins;
+
+	if (cfg->line_outs != 1 ||
+	    cfg->line_out_type != AUTO_PIN_LINE_OUT)
+		return 0;
+
+	defcfg = snd_hda_codec_get_pincfg(codec, cfg->line_out_pins[0]);
+	location = get_defcfg_location(defcfg);
+
+	num_pins = alc_auto_fill_multi_ios(codec, location);
+	if (num_pins > 0) {
+		struct snd_kcontrol_new *knew;
+
+		knew = alc_kcontrol_new(spec);
+		if (!knew)
+			return -ENOMEM;
+		*knew = alc_auto_channel_mode_enum;
+		knew->name = kstrdup("Channel Mode", GFP_KERNEL);
+		if (!knew->name)
+			return -ENOMEM;
+
+		spec->multi_ios = num_pins;
+		spec->ext_channel_count = 2;
+		spec->multiout.num_dacs = num_pins + 1;
+	}
+	return 0;
+}
+
 static int alc662_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	int err;
-	static hda_nid_t alc662_ignore[] = { 0x1d, 0 };
+	static const hda_nid_t alc662_ignore[] = { 0x1d, 0 };
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
 					   alc662_ignore);
@@ -19372,6 +19356,9 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 	err = alc662_auto_fill_dac_nids(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
+	err = alc_auto_add_multi_channel_mode(codec);
+	if (err < 0)
+		return err;
 	err = alc662_auto_create_multi_out_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
@@ -19402,14 +19389,6 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux[0];
 
-	add_verb(spec, alc662_init_verbs);
-	if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 ||
-	    codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670)
-		add_verb(spec, alc663_init_verbs);
-
-	if (codec->vendor_id == 0x10ec0272)
-		add_verb(spec, alc272_init_verbs);
-
 	err = alc_auto_add_mic_boost(codec);
 	if (err < 0)
 		return err;
@@ -19455,7 +19434,7 @@ enum {
 	ALC662_FIXUP_IDEAPAD,
 	ALC272_FIXUP_MARIO,
 	ALC662_FIXUP_CZC_P10T,
-	ALC662_FIXUP_GIGABYTE,
+	ALC662_FIXUP_SKU_IGNORE,
 };
 
 static const struct alc_fixup alc662_fixups[] = {
@@ -19484,20 +19463,17 @@ static const struct alc_fixup alc662_fixups[] = {
 			{}
 		}
 	},
-	[ALC662_FIXUP_GIGABYTE] = {
-		.type = ALC_FIXUP_PINS,
-		.v.pins = (const struct alc_pincfg[]) {
-			{ 0x14, 0x1114410 }, /* set as speaker */
-			{ }
-		}
+	[ALC662_FIXUP_SKU_IGNORE] = {
+		.type = ALC_FIXUP_SKU,
+		.v.sku = ALC_FIXUP_SKU_IGNORE,
 	},
 };
 
-static struct snd_pci_quirk alc662_fixup_tbl[] = {
+static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
+	SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
-	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", ALC662_FIXUP_GIGABYTE),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
@@ -19611,6 +19587,7 @@ static int patch_alc662(struct hda_codec *codec)
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC662_AUTO)
 		spec->init_hook = alc662_auto_init;
+	spec->shutup = alc_eapd_shutup;
 
 	alc_init_jacks(codec);
 
@@ -19639,6 +19616,15 @@ static int patch_alc888(struct hda_codec *codec)
 	return patch_alc882(codec);
 }
 
+static int patch_alc899(struct hda_codec *codec)
+{
+	if ((alc_read_coef_idx(codec, 0) & 0x2000) != 0x2000) {
+		kfree(codec->chip_name);
+		codec->chip_name = kstrdup("ALC898", GFP_KERNEL);
+	}
+	return patch_alc882(codec);
+}
+
 /*
  * ALC680 support
  */
@@ -19646,12 +19632,12 @@ static int patch_alc888(struct hda_codec *codec)
 #define ALC680_DIGOUT_NID	ALC880_DIGOUT_NID
 #define alc680_modes		alc260_modes
 
-static hda_nid_t alc680_dac_nids[3] = {
+static const hda_nid_t alc680_dac_nids[3] = {
 	/* Lout1, Lout2, hp */
 	0x02, 0x03, 0x04
 };
 
-static hda_nid_t alc680_adc_nids[3] = {
+static const hda_nid_t alc680_adc_nids[3] = {
 	/* ADC0-2 */
 	/* DMIC, MIC, Line-in*/
 	0x07, 0x08, 0x09
@@ -19671,8 +19657,7 @@ static void alc680_rec_autoswitch(struct hda_codec *codec)
 
 	for (i = 0; i < cfg->num_inputs; i++) {
 		nid = cfg->inputs[i].pin;
-		if (!(snd_hda_query_pin_caps(codec, nid) &
-		      AC_PINCAP_PRES_DETECT))
+		if (!is_jack_detectable(codec, nid))
 			continue;
 		if (snd_hda_jack_detect(codec, nid)) {
 			if (cfg->inputs[i].type < type_found) {
@@ -19719,7 +19704,7 @@ static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 	return 0;
 }
 
-static struct hda_pcm_stream alc680_pcm_analog_auto_capture = {
+static const struct hda_pcm_stream alc680_pcm_analog_auto_capture = {
 	.substreams = 1, /* can be overridden */
 	.channels_min = 2,
 	.channels_max = 2,
@@ -19730,7 +19715,7 @@ static struct hda_pcm_stream alc680_pcm_analog_auto_capture = {
 	},
 };
 
-static struct snd_kcontrol_new alc680_base_mixer[] = {
+static const struct snd_kcontrol_new alc680_base_mixer[] = {
 	/* output mixer control */
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
@@ -19742,7 +19727,7 @@ static struct snd_kcontrol_new alc680_base_mixer[] = {
 	{ }
 };
 
-static struct hda_bind_ctls alc680_bind_cap_vol = {
+static const struct hda_bind_ctls alc680_bind_cap_vol = {
 	.ops = &snd_hda_bind_vol,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
@@ -19752,7 +19737,7 @@ static struct hda_bind_ctls alc680_bind_cap_vol = {
 	},
 };
 
-static struct hda_bind_ctls alc680_bind_cap_switch = {
+static const struct hda_bind_ctls alc680_bind_cap_switch = {
 	.ops = &snd_hda_bind_sw,
 	.values = {
 		HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
@@ -19762,7 +19747,7 @@ static struct hda_bind_ctls alc680_bind_cap_switch = {
 	},
 };
 
-static struct snd_kcontrol_new alc680_master_capture_mixer[] = {
+static const struct snd_kcontrol_new alc680_master_capture_mixer[] = {
 	HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol),
 	HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch),
 	{ } /* end */
@@ -19771,7 +19756,7 @@ static struct snd_kcontrol_new alc680_master_capture_mixer[] = {
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb alc680_init_verbs[] = {
+static const struct hda_verb alc680_init_verbs[] = {
 	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -19809,20 +19794,22 @@ static void alc680_base_setup(struct hda_codec *codec)
 	spec->autocfg.inputs[0].type = AUTO_PIN_MIC;
 	spec->autocfg.inputs[1].pin = 0x19;
 	spec->autocfg.inputs[1].type = AUTO_PIN_LINE_IN;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 }
 
 static void alc680_unsol_event(struct hda_codec *codec,
 					   unsigned int res)
 {
 	if ((res >> 26) == ALC880_HP_EVENT)
-		alc_automute_amp(codec);
+		alc_hp_automute(codec);
 	if ((res >> 26) == ALC880_MIC_EVENT)
 		alc680_rec_autoswitch(codec);
 }
 
 static void alc680_inithook(struct hda_codec *codec)
 {
-	alc_automute_amp(codec);
+	alc_hp_automute(codec);
 	alc680_rec_autoswitch(codec);
 }
 
@@ -19859,7 +19846,7 @@ static int alc680_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
 
 		if (err < 0)
 			return err;
-		spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
+		spec->private_dac_nids[spec->multiout.num_dacs++] = dac;
 	}
 
 	return 0;
@@ -19945,7 +19932,7 @@ static int alc680_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	int err;
-	static hda_nid_t alc680_ignore[] = { 0 };
+	static const hda_nid_t alc680_ignore[] = { 0 };
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
 					   alc680_ignore);
@@ -20003,12 +19990,12 @@ static const char * const alc680_models[ALC680_MODEL_LAST] = {
 	[ALC680_AUTO]		= "auto",
 };
 
-static struct snd_pci_quirk alc680_cfg_tbl[] = {
+static const struct snd_pci_quirk alc680_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x12f3, "ASUS NX90", ALC680_BASE),
 	{}
 };
 
-static struct alc_config_preset alc680_presets[] = {
+static const struct alc_config_preset alc680_presets[] = {
 	[ALC680_BASE] = {
 		.mixers = { alc680_base_mixer },
 		.cap_mixer =  alc680_master_capture_mixer,
@@ -20089,7 +20076,8 @@ static int patch_alc680(struct hda_codec *codec)
 /*
  * patch entries
  */
-static struct hda_codec_preset snd_hda_preset_realtek[] = {
+static const struct hda_codec_preset snd_hda_preset_realtek[] = {
+	{ .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 },
 	{ .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
 	{ .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
 	{ .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 },
@@ -20098,6 +20086,7 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0270, .name = "ALC270", .patch = patch_alc269 },
 	{ .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 },
 	{ .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 },
+	{ .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 },
 	{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
 	  .patch = patch_alc861 },
 	{ .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
@@ -20125,6 +20114,7 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc888 },
 	{ .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 },
 	{ .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 },
+	{ .id = 0x10ec0899, .name = "ALC899", .patch = patch_alc899 },
 	{} /* terminator */
 };
 
diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c
index f419ee8d75f..2f55f32876f 100644
--- a/sound/pci/hda/patch_si3054.c
+++ b/sound/pci/hda/patch_si3054.c
@@ -130,7 +130,7 @@ static int si3054_switch_put(struct snd_kcontrol *kcontrol,
 }
 		
 
-static struct snd_kcontrol_new si3054_modem_mixer[] = {
+static const struct snd_kcontrol_new si3054_modem_mixer[] = {
 	SI3054_KCONTROL("Off-hook Switch", SI3054_GPIO_CONTROL, SI3054_GPIO_OH),
 	SI3054_KCONTROL("Caller ID Switch", SI3054_GPIO_CONTROL, SI3054_GPIO_CID),
 	{}
@@ -181,7 +181,7 @@ static int si3054_pcm_open(struct hda_pcm_stream *hinfo,
 }
 
 
-static struct hda_pcm_stream si3054_pcm = {
+static const struct hda_pcm_stream si3054_pcm = {
 	.substreams = 1,
 	.channels_min = 1,
 	.channels_max = 1,
@@ -200,12 +200,13 @@ static int si3054_build_pcms(struct hda_codec *codec)
 {
 	struct si3054_spec *spec = codec->spec;
 	struct hda_pcm *info = &spec->pcm;
-	si3054_pcm.nid = codec->mfg;
 	codec->num_pcms = 1;
 	codec->pcm_info = info;
 	info->name = "Si3054 Modem";
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = si3054_pcm;
 	info->stream[SNDRV_PCM_STREAM_CAPTURE]  = si3054_pcm;
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = codec->mfg;
+	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = codec->mfg;
 	info->pcm_type = HDA_PCM_TYPE_MODEM;
 	return 0;
 }
@@ -263,7 +264,7 @@ static void si3054_free(struct hda_codec *codec)
 /*
  */
 
-static struct hda_codec_ops si3054_patch_ops = {
+static const struct hda_codec_ops si3054_patch_ops = {
 	.build_controls = si3054_build_controls,
 	.build_pcms = si3054_build_pcms,
 	.init = si3054_init,
@@ -283,7 +284,7 @@ static int patch_si3054(struct hda_codec *codec)
 /*
  * patch entries
  */
-static struct hda_codec_preset snd_hda_preset_si3054[] = {
+static const struct hda_codec_preset snd_hda_preset_si3054[] = {
  	{ .id = 0x163c3055, .name = "Si3054", .patch = patch_si3054 },
  	{ .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 },
  	{ .id = 0x11c13026, .name = "Si3054", .patch = patch_si3054 },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 94d19c03a7f..7f81cc2274f 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -217,15 +217,15 @@ struct sigmatel_spec {
 	unsigned int stream_delay;
 
 	/* analog loopback */
-	struct snd_kcontrol_new *aloopback_ctl;
+	const struct snd_kcontrol_new *aloopback_ctl;
 	unsigned char aloopback_mask;
 	unsigned char aloopback_shift;
 
 	/* power management */
 	unsigned int num_pwrs;
-	unsigned int *pwr_mapping;
-	hda_nid_t *pwr_nids;
-	hda_nid_t *dac_list;
+	const unsigned int *pwr_mapping;
+	const hda_nid_t *pwr_nids;
+	const hda_nid_t *dac_list;
 
 	/* events */
 	struct snd_array events;
@@ -241,20 +241,20 @@ struct sigmatel_spec {
 	int volume_offset;
 
 	/* capture */
-	hda_nid_t *adc_nids;
+	const hda_nid_t *adc_nids;
 	unsigned int num_adcs;
-	hda_nid_t *mux_nids;
+	const hda_nid_t *mux_nids;
 	unsigned int num_muxes;
-	hda_nid_t *dmic_nids;
+	const hda_nid_t *dmic_nids;
 	unsigned int num_dmics;
-	hda_nid_t *dmux_nids;
+	const hda_nid_t *dmux_nids;
 	unsigned int num_dmuxes;
-	hda_nid_t *smux_nids;
+	const hda_nid_t *smux_nids;
 	unsigned int num_smuxes;
 	unsigned int num_analog_muxes;
 
-	unsigned long *capvols; /* amp-volume attr: HDA_COMPOSE_AMP_VAL() */
-	unsigned long *capsws; /* amp-mute attr: HDA_COMPOSE_AMP_VAL() */
+	const unsigned long *capvols; /* amp-volume attr: HDA_COMPOSE_AMP_VAL() */
+	const unsigned long *capsws; /* amp-mute attr: HDA_COMPOSE_AMP_VAL() */
 	unsigned int num_caps; /* number of capture volume/switch elements */
 
 	struct sigmatel_mic_route ext_mic;
@@ -269,12 +269,12 @@ struct sigmatel_spec {
 	hda_nid_t digbeep_nid;
 
 	/* pin widgets */
-	hda_nid_t *pin_nids;
+	const hda_nid_t *pin_nids;
 	unsigned int num_pins;
 
 	/* codec specific stuff */
-	struct hda_verb *init;
-	struct snd_kcontrol_new *mixer;
+	const struct hda_verb *init;
+	const struct snd_kcontrol_new *mixer;
 
 	/* capture source */
 	struct hda_input_mux *dinput_mux;
@@ -317,52 +317,52 @@ struct sigmatel_spec {
 	hda_nid_t auto_dmic_nids[MAX_DMICS_NUM];
 };
 
-static hda_nid_t stac9200_adc_nids[1] = {
+static const hda_nid_t stac9200_adc_nids[1] = {
         0x03,
 };
 
-static hda_nid_t stac9200_mux_nids[1] = {
+static const hda_nid_t stac9200_mux_nids[1] = {
         0x0c,
 };
 
-static hda_nid_t stac9200_dac_nids[1] = {
+static const hda_nid_t stac9200_dac_nids[1] = {
         0x02,
 };
 
-static hda_nid_t stac92hd73xx_pwr_nids[8] = {
+static const hda_nid_t stac92hd73xx_pwr_nids[8] = {
 	0x0a, 0x0b, 0x0c, 0xd, 0x0e,
 	0x0f, 0x10, 0x11
 };
 
-static hda_nid_t stac92hd73xx_slave_dig_outs[2] = {
+static const hda_nid_t stac92hd73xx_slave_dig_outs[2] = {
 	0x26, 0,
 };
 
-static hda_nid_t stac92hd73xx_adc_nids[2] = {
+static const hda_nid_t stac92hd73xx_adc_nids[2] = {
 	0x1a, 0x1b
 };
 
 #define STAC92HD73XX_NUM_DMICS	2
-static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = {
+static const hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = {
 	0x13, 0x14, 0
 };
 
 #define STAC92HD73_DAC_COUNT 5
 
-static hda_nid_t stac92hd73xx_mux_nids[2] = {
+static const hda_nid_t stac92hd73xx_mux_nids[2] = {
 	0x20, 0x21,
 };
 
-static hda_nid_t stac92hd73xx_dmux_nids[2] = {
+static const hda_nid_t stac92hd73xx_dmux_nids[2] = {
 	0x20, 0x21,
 };
 
-static hda_nid_t stac92hd73xx_smux_nids[2] = {
+static const hda_nid_t stac92hd73xx_smux_nids[2] = {
 	0x22, 0x23,
 };
 
 #define STAC92HD73XX_NUM_CAPS	2
-static unsigned long stac92hd73xx_capvols[] = {
+static const unsigned long stac92hd73xx_capvols[] = {
 	HDA_COMPOSE_AMP_VAL(0x20, 3, 0, HDA_OUTPUT),
 	HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
 };
@@ -370,137 +370,141 @@ static unsigned long stac92hd73xx_capvols[] = {
 
 #define STAC92HD83_DAC_COUNT 3
 
-static hda_nid_t stac92hd83xxx_pwr_nids[4] = {
+static const hda_nid_t stac92hd83xxx_pwr_nids[4] = {
 	0xa, 0xb, 0xd, 0xe,
 };
 
-static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = {
+static const hda_nid_t stac92hd83xxx_slave_dig_outs[2] = {
 	0x1e, 0,
 };
 
-static unsigned int stac92hd83xxx_pwr_mapping[4] = {
+static const unsigned int stac92hd83xxx_pwr_mapping[4] = {
 	0x03, 0x0c, 0x20, 0x40,
 };
 
-static hda_nid_t stac92hd83xxx_dmic_nids[] = {
+static const hda_nid_t stac92hd83xxx_dmic_nids[] = {
 		0x11, 0x20,
 };
 
-static hda_nid_t stac92hd71bxx_pwr_nids[3] = {
+static const hda_nid_t stac92hd71bxx_pwr_nids[3] = {
 	0x0a, 0x0d, 0x0f
 };
 
-static hda_nid_t stac92hd71bxx_adc_nids[2] = {
+static const hda_nid_t stac92hd71bxx_adc_nids[2] = {
 	0x12, 0x13,
 };
 
-static hda_nid_t stac92hd71bxx_mux_nids[2] = {
+static const hda_nid_t stac92hd71bxx_mux_nids[2] = {
 	0x1a, 0x1b
 };
 
-static hda_nid_t stac92hd71bxx_dmux_nids[2] = {
+static const hda_nid_t stac92hd71bxx_dmux_nids[2] = {
 	0x1c, 0x1d,
 };
 
-static hda_nid_t stac92hd71bxx_smux_nids[2] = {
+static const hda_nid_t stac92hd71bxx_smux_nids[2] = {
 	0x24, 0x25,
 };
 
 #define STAC92HD71BXX_NUM_DMICS	2
-static hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = {
+static const hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = {
 	0x18, 0x19, 0
 };
 
-static hda_nid_t stac92hd71bxx_slave_dig_outs[2] = {
+static const hda_nid_t stac92hd71bxx_dmic_5port_nids[STAC92HD71BXX_NUM_DMICS] = {
+	0x18, 0
+};
+
+static const hda_nid_t stac92hd71bxx_slave_dig_outs[2] = {
 	0x22, 0
 };
 
 #define STAC92HD71BXX_NUM_CAPS		2
-static unsigned long stac92hd71bxx_capvols[] = {
+static const unsigned long stac92hd71bxx_capvols[] = {
 	HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_OUTPUT),
 	HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT),
 };
 #define stac92hd71bxx_capsws	stac92hd71bxx_capvols
 
-static hda_nid_t stac925x_adc_nids[1] = {
+static const hda_nid_t stac925x_adc_nids[1] = {
         0x03,
 };
 
-static hda_nid_t stac925x_mux_nids[1] = {
+static const hda_nid_t stac925x_mux_nids[1] = {
         0x0f,
 };
 
-static hda_nid_t stac925x_dac_nids[1] = {
+static const hda_nid_t stac925x_dac_nids[1] = {
         0x02,
 };
 
 #define STAC925X_NUM_DMICS	1
-static hda_nid_t stac925x_dmic_nids[STAC925X_NUM_DMICS + 1] = {
+static const hda_nid_t stac925x_dmic_nids[STAC925X_NUM_DMICS + 1] = {
 	0x15, 0
 };
 
-static hda_nid_t stac925x_dmux_nids[1] = {
+static const hda_nid_t stac925x_dmux_nids[1] = {
 	0x14,
 };
 
-static unsigned long stac925x_capvols[] = {
+static const unsigned long stac925x_capvols[] = {
 	HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT),
 };
-static unsigned long stac925x_capsws[] = {
+static const unsigned long stac925x_capsws[] = {
 	HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
 };
 
-static hda_nid_t stac922x_adc_nids[2] = {
+static const hda_nid_t stac922x_adc_nids[2] = {
         0x06, 0x07,
 };
 
-static hda_nid_t stac922x_mux_nids[2] = {
+static const hda_nid_t stac922x_mux_nids[2] = {
         0x12, 0x13,
 };
 
 #define STAC922X_NUM_CAPS	2
-static unsigned long stac922x_capvols[] = {
+static const unsigned long stac922x_capvols[] = {
 	HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_INPUT),
 	HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT),
 };
 #define stac922x_capsws		stac922x_capvols
 
-static hda_nid_t stac927x_slave_dig_outs[2] = {
+static const hda_nid_t stac927x_slave_dig_outs[2] = {
 	0x1f, 0,
 };
 
-static hda_nid_t stac927x_adc_nids[3] = {
+static const hda_nid_t stac927x_adc_nids[3] = {
         0x07, 0x08, 0x09
 };
 
-static hda_nid_t stac927x_mux_nids[3] = {
+static const hda_nid_t stac927x_mux_nids[3] = {
         0x15, 0x16, 0x17
 };
 
-static hda_nid_t stac927x_smux_nids[1] = {
+static const hda_nid_t stac927x_smux_nids[1] = {
 	0x21,
 };
 
-static hda_nid_t stac927x_dac_nids[6] = {
+static const hda_nid_t stac927x_dac_nids[6] = {
 	0x02, 0x03, 0x04, 0x05, 0x06, 0
 };
 
-static hda_nid_t stac927x_dmux_nids[1] = {
+static const hda_nid_t stac927x_dmux_nids[1] = {
 	0x1b,
 };
 
 #define STAC927X_NUM_DMICS 2
-static hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = {
+static const hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = {
 	0x13, 0x14, 0
 };
 
 #define STAC927X_NUM_CAPS	3
-static unsigned long stac927x_capvols[] = {
+static const unsigned long stac927x_capvols[] = {
 	HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT),
 	HDA_COMPOSE_AMP_VAL(0x19, 3, 0, HDA_INPUT),
 	HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_INPUT),
 };
-static unsigned long stac927x_capsws[] = {
+static const unsigned long stac927x_capsws[] = {
 	HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
 	HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_OUTPUT),
 	HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT),
@@ -511,77 +515,77 @@ static const char * const stac927x_spdif_labels[5] = {
 	"Analog Mux 2", "Analog Mux 3"
 };
 
-static hda_nid_t stac9205_adc_nids[2] = {
+static const hda_nid_t stac9205_adc_nids[2] = {
         0x12, 0x13
 };
 
-static hda_nid_t stac9205_mux_nids[2] = {
+static const hda_nid_t stac9205_mux_nids[2] = {
         0x19, 0x1a
 };
 
-static hda_nid_t stac9205_dmux_nids[1] = {
+static const hda_nid_t stac9205_dmux_nids[1] = {
 	0x1d,
 };
 
-static hda_nid_t stac9205_smux_nids[1] = {
+static const hda_nid_t stac9205_smux_nids[1] = {
 	0x21,
 };
 
 #define STAC9205_NUM_DMICS	2
-static hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = {
+static const hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = {
         0x17, 0x18, 0
 };
 
 #define STAC9205_NUM_CAPS	2
-static unsigned long stac9205_capvols[] = {
+static const unsigned long stac9205_capvols[] = {
 	HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_INPUT),
 	HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_INPUT),
 };
-static unsigned long stac9205_capsws[] = {
+static const unsigned long stac9205_capsws[] = {
 	HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT),
 	HDA_COMPOSE_AMP_VAL(0x1e, 3, 0, HDA_OUTPUT),
 };
 
-static hda_nid_t stac9200_pin_nids[8] = {
+static const hda_nid_t stac9200_pin_nids[8] = {
 	0x08, 0x09, 0x0d, 0x0e, 
 	0x0f, 0x10, 0x11, 0x12,
 };
 
-static hda_nid_t stac925x_pin_nids[8] = {
+static const hda_nid_t stac925x_pin_nids[8] = {
 	0x07, 0x08, 0x0a, 0x0b, 
 	0x0c, 0x0d, 0x10, 0x11,
 };
 
-static hda_nid_t stac922x_pin_nids[10] = {
+static const hda_nid_t stac922x_pin_nids[10] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
 	0x0f, 0x10, 0x11, 0x15, 0x1b,
 };
 
-static hda_nid_t stac92hd73xx_pin_nids[13] = {
+static const hda_nid_t stac92hd73xx_pin_nids[13] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
 	0x0f, 0x10, 0x11, 0x12, 0x13,
 	0x14, 0x22, 0x23
 };
 
 #define STAC92HD71BXX_NUM_PINS 13
-static hda_nid_t stac92hd71bxx_pin_nids_4port[STAC92HD71BXX_NUM_PINS] = {
+static const hda_nid_t stac92hd71bxx_pin_nids_4port[STAC92HD71BXX_NUM_PINS] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x00,
 	0x00, 0x14, 0x18, 0x19, 0x1e,
 	0x1f, 0x20, 0x27
 };
-static hda_nid_t stac92hd71bxx_pin_nids_6port[STAC92HD71BXX_NUM_PINS] = {
+static const hda_nid_t stac92hd71bxx_pin_nids_6port[STAC92HD71BXX_NUM_PINS] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
 	0x0f, 0x14, 0x18, 0x19, 0x1e,
 	0x1f, 0x20, 0x27
 };
 
-static hda_nid_t stac927x_pin_nids[14] = {
+static const hda_nid_t stac927x_pin_nids[14] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
 	0x0f, 0x10, 0x11, 0x12, 0x13,
 	0x14, 0x21, 0x22, 0x23,
 };
 
-static hda_nid_t stac9205_pin_nids[12] = {
+static const hda_nid_t stac9205_pin_nids[12] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
 	0x0f, 0x14, 0x16, 0x17, 0x18,
 	0x21, 0x22,
@@ -841,45 +845,45 @@ static int stac92xx_aloopback_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static struct hda_verb stac9200_core_init[] = {
+static const struct hda_verb stac9200_core_init[] = {
 	/* set dac0mux for dac converter */
 	{ 0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{}
 };
 
-static struct hda_verb stac9200_eapd_init[] = {
+static const struct hda_verb stac9200_eapd_init[] = {
 	/* set dac0mux for dac converter */
 	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x08, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
 	{}
 };
 
-static struct hda_verb dell_eq_core_init[] = {
+static const struct hda_verb dell_eq_core_init[] = {
 	/* set master volume to max value without distortion
 	 * and direct control */
 	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec},
 	{}
 };
 
-static struct hda_verb stac92hd73xx_core_init[] = {
+static const struct hda_verb stac92hd73xx_core_init[] = {
 	/* set master volume and direct control */
 	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	{}
 };
 
-static struct hda_verb stac92hd83xxx_core_init[] = {
+static const struct hda_verb stac92hd83xxx_core_init[] = {
 	/* power state controls amps */
 	{ 0x01, AC_VERB_SET_EAPD, 1 << 2},
 	{}
 };
 
-static struct hda_verb stac92hd71bxx_core_init[] = {
+static const struct hda_verb stac92hd71bxx_core_init[] = {
 	/* set master volume and direct control */
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	{}
 };
 
-static struct hda_verb stac92hd71bxx_unmute_core_init[] = {
+static const struct hda_verb stac92hd71bxx_unmute_core_init[] = {
 	/* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */
 	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -887,7 +891,7 @@ static struct hda_verb stac92hd71bxx_unmute_core_init[] = {
 	{}
 };
 
-static struct hda_verb stac925x_core_init[] = {
+static const struct hda_verb stac925x_core_init[] = {
 	/* set dac0mux for dac converter */
 	{ 0x06, AC_VERB_SET_CONNECT_SEL, 0x00},
 	/* mute the master volume */
@@ -895,13 +899,13 @@ static struct hda_verb stac925x_core_init[] = {
 	{}
 };
 
-static struct hda_verb stac922x_core_init[] = {
+static const struct hda_verb stac922x_core_init[] = {
 	/* set master volume and direct control */	
 	{ 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	{}
 };
 
-static struct hda_verb d965_core_init[] = {
+static const struct hda_verb d965_core_init[] = {
 	/* set master volume and direct control */	
 	{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	/* unmute node 0x1b */
@@ -911,7 +915,7 @@ static struct hda_verb d965_core_init[] = {
 	{}
 };
 
-static struct hda_verb dell_3st_core_init[] = {
+static const struct hda_verb dell_3st_core_init[] = {
 	/* don't set delta bit */
 	{0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f},
 	/* unmute node 0x1b */
@@ -921,7 +925,7 @@ static struct hda_verb dell_3st_core_init[] = {
 	{}
 };
 
-static struct hda_verb stac927x_core_init[] = {
+static const struct hda_verb stac927x_core_init[] = {
 	/* set master volume and direct control */	
 	{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	/* enable analog pc beep path */
@@ -929,7 +933,7 @@ static struct hda_verb stac927x_core_init[] = {
 	{}
 };
 
-static struct hda_verb stac927x_volknob_core_init[] = {
+static const struct hda_verb stac927x_volknob_core_init[] = {
 	/* don't set delta bit */
 	{0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f},
 	/* enable analog pc beep path */
@@ -937,7 +941,7 @@ static struct hda_verb stac927x_volknob_core_init[] = {
 	{}
 };
 
-static struct hda_verb stac9205_core_init[] = {
+static const struct hda_verb stac9205_core_init[] = {
 	/* set master volume and direct control */	
 	{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	/* enable analog pc beep path */
@@ -977,7 +981,7 @@ static struct hda_verb stac9205_core_init[] = {
 		.private_value = nid, \
 	}
 
-static struct snd_kcontrol_new stac9200_mixer[] = {
+static const struct snd_kcontrol_new stac9200_mixer[] = {
 	HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xb, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT),
@@ -985,38 +989,38 @@ static struct snd_kcontrol_new stac9200_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new stac92hd73xx_6ch_loopback[] = {
+static const struct snd_kcontrol_new stac92hd73xx_6ch_loopback[] = {
 	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3),
 	{}
 };
 
-static struct snd_kcontrol_new stac92hd73xx_8ch_loopback[] = {
+static const struct snd_kcontrol_new stac92hd73xx_8ch_loopback[] = {
 	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4),
 	{}
 };
 
-static struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = {
+static const struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = {
 	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5),
 	{}
 };
 
 
-static struct snd_kcontrol_new stac92hd71bxx_loopback[] = {
+static const struct snd_kcontrol_new stac92hd71bxx_loopback[] = {
 	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2)
 };
 
-static struct snd_kcontrol_new stac925x_mixer[] = {
+static const struct snd_kcontrol_new stac925x_mixer[] = {
 	HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xe, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT),
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new stac9205_loopback[] = {
+static const struct snd_kcontrol_new stac9205_loopback[] = {
 	STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1),
 	{}
 };
 
-static struct snd_kcontrol_new stac927x_loopback[] = {
+static const struct snd_kcontrol_new stac927x_loopback[] = {
 	STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1),
 	{}
 };
@@ -1182,16 +1186,16 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 	return 0;	
 }
 
-static unsigned int ref9200_pin_configs[8] = {
+static const unsigned int ref9200_pin_configs[8] = {
 	0x01c47010, 0x01447010, 0x0221401f, 0x01114010,
 	0x02a19020, 0x01a19021, 0x90100140, 0x01813122,
 };
 
-static unsigned int gateway9200_m4_pin_configs[8] = {
+static const unsigned int gateway9200_m4_pin_configs[8] = {
 	0x400000fe, 0x404500f4, 0x400100f0, 0x90110010,
 	0x400100f1, 0x02a1902e, 0x500000f2, 0x500000f3,
 };
-static unsigned int gateway9200_m4_2_pin_configs[8] = {
+static const unsigned int gateway9200_m4_2_pin_configs[8] = {
 	0x400000fe, 0x404500f4, 0x400100f0, 0x90110010,
 	0x400100f1, 0x02a1902e, 0x500000f2, 0x500000f3,
 };
@@ -1202,7 +1206,7 @@ static unsigned int gateway9200_m4_2_pin_configs[8] = {
     102801DE
     102801E8
 */
-static unsigned int dell9200_d21_pin_configs[8] = {
+static const unsigned int dell9200_d21_pin_configs[8] = {
 	0x400001f0, 0x400001f1, 0x02214030, 0x01014010, 
 	0x02a19020, 0x01a19021, 0x90100140, 0x01813122,
 };
@@ -1212,7 +1216,7 @@ static unsigned int dell9200_d21_pin_configs[8] = {
     102801C0
     102801C1
 */
-static unsigned int dell9200_d22_pin_configs[8] = {
+static const unsigned int dell9200_d22_pin_configs[8] = {
 	0x400001f0, 0x400001f1, 0x0221401f, 0x01014010, 
 	0x01813020, 0x02a19021, 0x90100140, 0x400001f2,
 };
@@ -1226,7 +1230,7 @@ static unsigned int dell9200_d22_pin_configs[8] = {
     102801DA
     102801E3
 */
-static unsigned int dell9200_d23_pin_configs[8] = {
+static const unsigned int dell9200_d23_pin_configs[8] = {
 	0x400001f0, 0x400001f1, 0x0221401f, 0x01014010, 
 	0x01813020, 0x01a19021, 0x90100140, 0x400001f2, 
 };
@@ -1237,7 +1241,7 @@ static unsigned int dell9200_d23_pin_configs[8] = {
     102801B5 (Dell Inspiron 630m)
     102801D8 (Dell Inspiron 640m)
 */
-static unsigned int dell9200_m21_pin_configs[8] = {
+static const unsigned int dell9200_m21_pin_configs[8] = {
 	0x40c003fa, 0x03441340, 0x0321121f, 0x90170310,
 	0x408003fb, 0x03a11020, 0x401003fc, 0x403003fd,
 };
@@ -1250,7 +1254,7 @@ static unsigned int dell9200_m21_pin_configs[8] = {
     102801D4 
     102801D6 
 */
-static unsigned int dell9200_m22_pin_configs[8] = {
+static const unsigned int dell9200_m22_pin_configs[8] = {
 	0x40c003fa, 0x0144131f, 0x0321121f, 0x90170310, 
 	0x90a70321, 0x03a11020, 0x401003fb, 0x40f000fc,
 };
@@ -1260,7 +1264,7 @@ static unsigned int dell9200_m22_pin_configs[8] = {
     102801CE (Dell XPS M1710)
     102801CF (Dell Precision M90)
 */
-static unsigned int dell9200_m23_pin_configs[8] = {
+static const unsigned int dell9200_m23_pin_configs[8] = {
 	0x40c003fa, 0x01441340, 0x0421421f, 0x90170310,
 	0x408003fb, 0x04a1102e, 0x90170311, 0x403003fc,
 };
@@ -1272,7 +1276,7 @@ static unsigned int dell9200_m23_pin_configs[8] = {
     102801CB (Dell Latitude 120L)
     102801D3
 */
-static unsigned int dell9200_m24_pin_configs[8] = {
+static const unsigned int dell9200_m24_pin_configs[8] = {
 	0x40c003fa, 0x404003fb, 0x0321121f, 0x90170310, 
 	0x408003fc, 0x03a11020, 0x401003fd, 0x403003fe, 
 };
@@ -1283,7 +1287,7 @@ static unsigned int dell9200_m24_pin_configs[8] = {
     102801EE
     102801EF
 */
-static unsigned int dell9200_m25_pin_configs[8] = {
+static const unsigned int dell9200_m25_pin_configs[8] = {
 	0x40c003fa, 0x01441340, 0x0421121f, 0x90170310, 
 	0x408003fb, 0x04a11020, 0x401003fc, 0x403003fd,
 };
@@ -1293,7 +1297,7 @@ static unsigned int dell9200_m25_pin_configs[8] = {
     102801F5 (Dell Inspiron 1501)
     102801F6
 */
-static unsigned int dell9200_m26_pin_configs[8] = {
+static const unsigned int dell9200_m26_pin_configs[8] = {
 	0x40c003fa, 0x404003fb, 0x0421121f, 0x90170310, 
 	0x408003fc, 0x04a11020, 0x401003fd, 0x403003fe,
 };
@@ -1302,18 +1306,18 @@ static unsigned int dell9200_m26_pin_configs[8] = {
     STAC 9200-32
     102801CD (Dell Inspiron E1705/9400)
 */
-static unsigned int dell9200_m27_pin_configs[8] = {
+static const unsigned int dell9200_m27_pin_configs[8] = {
 	0x40c003fa, 0x01441340, 0x0421121f, 0x90170310,
 	0x90170310, 0x04a11020, 0x90170310, 0x40f003fc,
 };
 
-static unsigned int oqo9200_pin_configs[8] = {
+static const unsigned int oqo9200_pin_configs[8] = {
 	0x40c000f0, 0x404000f1, 0x0221121f, 0x02211210,
 	0x90170111, 0x90a70120, 0x400000f2, 0x400000f3,
 };
 
 
-static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = {
+static const unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = {
 	[STAC_REF] = ref9200_pin_configs,
 	[STAC_9200_OQO] = oqo9200_pin_configs,
 	[STAC_9200_DELL_D21] = dell9200_d21_pin_configs,
@@ -1350,7 +1354,7 @@ static const char * const stac9200_models[STAC_9200_MODELS] = {
 	[STAC_9200_PANASONIC] = "panasonic",
 };
 
-static struct snd_pci_quirk stac9200_cfg_tbl[] = {
+static const struct snd_pci_quirk stac9200_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_REF),
@@ -1426,47 +1430,47 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = {
 	{} /* terminator */
 };
 
-static unsigned int ref925x_pin_configs[8] = {
+static const unsigned int ref925x_pin_configs[8] = {
 	0x40c003f0, 0x424503f2, 0x01813022, 0x02a19021,
 	0x90a70320, 0x02214210, 0x01019020, 0x9033032e,
 };
 
-static unsigned int stac925xM1_pin_configs[8] = {
+static const unsigned int stac925xM1_pin_configs[8] = {
 	0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020,
 	0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e,
 };
 
-static unsigned int stac925xM1_2_pin_configs[8] = {
+static const unsigned int stac925xM1_2_pin_configs[8] = {
 	0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020,
 	0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e,
 };
 
-static unsigned int stac925xM2_pin_configs[8] = {
+static const unsigned int stac925xM2_pin_configs[8] = {
 	0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020,
 	0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e,
 };
 
-static unsigned int stac925xM2_2_pin_configs[8] = {
+static const unsigned int stac925xM2_2_pin_configs[8] = {
 	0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020,
 	0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e,
 };
 
-static unsigned int stac925xM3_pin_configs[8] = {
+static const unsigned int stac925xM3_pin_configs[8] = {
 	0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020,
 	0x40a000f0, 0x90100210, 0x400003f1, 0x503303f3,
 };
 
-static unsigned int stac925xM5_pin_configs[8] = {
+static const unsigned int stac925xM5_pin_configs[8] = {
 	0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020,
 	0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e,
 };
 
-static unsigned int stac925xM6_pin_configs[8] = {
+static const unsigned int stac925xM6_pin_configs[8] = {
 	0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020,
 	0x40a000f0, 0x90100210, 0x400003f1, 0x90330320,
 };
 
-static unsigned int *stac925x_brd_tbl[STAC_925x_MODELS] = {
+static const unsigned int *stac925x_brd_tbl[STAC_925x_MODELS] = {
 	[STAC_REF] = ref925x_pin_configs,
 	[STAC_M1] = stac925xM1_pin_configs,
 	[STAC_M1_2] = stac925xM1_2_pin_configs,
@@ -1489,7 +1493,7 @@ static const char * const stac925x_models[STAC_925x_MODELS] = {
 	[STAC_M6] = "m6",
 };
 
-static struct snd_pci_quirk stac925x_codec_id_cfg_tbl[] = {
+static const struct snd_pci_quirk stac925x_codec_id_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x107b, 0x0316, "Gateway M255", STAC_M2),
 	SND_PCI_QUIRK(0x107b, 0x0366, "Gateway MP6954", STAC_M5),
 	SND_PCI_QUIRK(0x107b, 0x0461, "Gateway NX560XL", STAC_M1),
@@ -1503,7 +1507,7 @@ static struct snd_pci_quirk stac925x_codec_id_cfg_tbl[] = {
 	{} /* terminator */
 };
 
-static struct snd_pci_quirk stac925x_cfg_tbl[] = {
+static const struct snd_pci_quirk stac925x_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF),
@@ -1515,33 +1519,33 @@ static struct snd_pci_quirk stac925x_cfg_tbl[] = {
 	{} /* terminator */
 };
 
-static unsigned int ref92hd73xx_pin_configs[13] = {
+static const unsigned int ref92hd73xx_pin_configs[13] = {
 	0x02214030, 0x02a19040, 0x01a19020, 0x02214030,
 	0x0181302e, 0x01014010, 0x01014020, 0x01014030,
 	0x02319040, 0x90a000f0, 0x90a000f0, 0x01452050,
 	0x01452050,
 };
 
-static unsigned int dell_m6_pin_configs[13] = {
+static const unsigned int dell_m6_pin_configs[13] = {
 	0x0321101f, 0x4f00000f, 0x4f0000f0, 0x90170110,
 	0x03a11020, 0x0321101f, 0x4f0000f0, 0x4f0000f0,
 	0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0,
 	0x4f0000f0,
 };
 
-static unsigned int alienware_m17x_pin_configs[13] = {
+static const unsigned int alienware_m17x_pin_configs[13] = {
 	0x0321101f, 0x0321101f, 0x03a11020, 0x03014020,
 	0x90170110, 0x4f0000f0, 0x4f0000f0, 0x4f0000f0,
 	0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0,
 	0x904601b0,
 };
 
-static unsigned int intel_dg45id_pin_configs[13] = {
+static const unsigned int intel_dg45id_pin_configs[13] = {
 	0x02214230, 0x02A19240, 0x01013214, 0x01014210,
 	0x01A19250, 0x01011212, 0x01016211
 };
 
-static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = {
+static const unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = {
 	[STAC_92HD73XX_REF]	= ref92hd73xx_pin_configs,
 	[STAC_DELL_M6_AMIC]	= dell_m6_pin_configs,
 	[STAC_DELL_M6_DMIC]	= dell_m6_pin_configs,
@@ -1563,7 +1567,7 @@ static const char * const stac92hd73xx_models[STAC_92HD73XX_MODELS] = {
 	[STAC_ALIENWARE_M17X] = "alienware",
 };
 
-static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
+static const struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 				"DFI LanParty", STAC_92HD73XX_REF),
@@ -1600,11 +1604,11 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02fe,
 				"Dell Studio XPS 1645", STAC_DELL_M6_BOTH),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0413,
-				"Dell Studio 1558", STAC_DELL_M6_BOTH),
+				"Dell Studio 1558", STAC_DELL_M6_DMIC),
 	{} /* terminator */
 };
 
-static struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = {
+static const struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a1,
 		      "Alienware M17x", STAC_ALIENWARE_M17X),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a,
@@ -1612,25 +1616,25 @@ static struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = {
 	{} /* terminator */
 };
 
-static unsigned int ref92hd83xxx_pin_configs[10] = {
+static const unsigned int ref92hd83xxx_pin_configs[10] = {
 	0x02214030, 0x02211010, 0x02a19020, 0x02170130,
 	0x01014050, 0x01819040, 0x01014020, 0x90a3014e,
 	0x01451160, 0x98560170,
 };
 
-static unsigned int dell_s14_pin_configs[10] = {
+static const unsigned int dell_s14_pin_configs[10] = {
 	0x0221403f, 0x0221101f, 0x02a19020, 0x90170110,
 	0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a60160,
 	0x40f000f0, 0x40f000f0,
 };
 
-static unsigned int hp_dv7_4000_pin_configs[10] = {
+static const unsigned int hp_dv7_4000_pin_configs[10] = {
 	0x03a12050, 0x0321201f, 0x40f000f0, 0x90170110,
 	0x40f000f0, 0x40f000f0, 0x90170110, 0xd5a30140,
 	0x40f000f0, 0x40f000f0,
 };
 
-static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
+static const unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
 	[STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs,
 	[STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs,
 	[STAC_DELL_S14] = dell_s14_pin_configs,
@@ -1646,7 +1650,7 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
 	[STAC_HP_DV7_4000] = "hp-dv7-4000",
 };
 
-static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
+static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_92HD83XXX_REF),
@@ -1659,35 +1663,35 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
 	{} /* terminator */
 };
 
-static unsigned int ref92hd71bxx_pin_configs[STAC92HD71BXX_NUM_PINS] = {
+static const unsigned int ref92hd71bxx_pin_configs[STAC92HD71BXX_NUM_PINS] = {
 	0x02214030, 0x02a19040, 0x01a19020, 0x01014010,
 	0x0181302e, 0x01014010, 0x01019020, 0x90a000f0,
 	0x90a000f0, 0x01452050, 0x01452050, 0x00000000,
 	0x00000000
 };
 
-static unsigned int dell_m4_1_pin_configs[STAC92HD71BXX_NUM_PINS] = {
+static const unsigned int dell_m4_1_pin_configs[STAC92HD71BXX_NUM_PINS] = {
 	0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110,
 	0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0,
 	0x40f000f0, 0x4f0000f0, 0x4f0000f0, 0x00000000,
 	0x00000000
 };
 
-static unsigned int dell_m4_2_pin_configs[STAC92HD71BXX_NUM_PINS] = {
+static const unsigned int dell_m4_2_pin_configs[STAC92HD71BXX_NUM_PINS] = {
 	0x0421101f, 0x04a11221, 0x90a70330, 0x90170110,
 	0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0,
 	0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000,
 	0x00000000
 };
 
-static unsigned int dell_m4_3_pin_configs[STAC92HD71BXX_NUM_PINS] = {
+static const unsigned int dell_m4_3_pin_configs[STAC92HD71BXX_NUM_PINS] = {
 	0x0421101f, 0x04a11221, 0x90a70330, 0x90170110,
 	0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a000f0,
 	0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000,
 	0x00000000
 };
 
-static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = {
+static const unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = {
 	[STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs,
 	[STAC_DELL_M4_1]	= dell_m4_1_pin_configs,
 	[STAC_DELL_M4_2]	= dell_m4_2_pin_configs,
@@ -1712,7 +1716,7 @@ static const char * const stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = {
 	[STAC_HP_DV4_1222NR] = "hp-dv4-1222nr",
 };
 
-static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
+static const struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_92HD71BXX_REF),
@@ -1769,7 +1773,7 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 	{} /* terminator */
 };
 
-static unsigned int ref922x_pin_configs[10] = {
+static const unsigned int ref922x_pin_configs[10] = {
 	0x01014010, 0x01016011, 0x01012012, 0x0221401f,
 	0x01813122, 0x01011014, 0x01441030, 0x01c41030,
 	0x40000100, 0x40000100,
@@ -1783,7 +1787,7 @@ static unsigned int ref922x_pin_configs[10] = {
     102801D1
     102801D2
 */
-static unsigned int dell_922x_d81_pin_configs[10] = {
+static const unsigned int dell_922x_d81_pin_configs[10] = {
 	0x02214030, 0x01a19021, 0x01111012, 0x01114010,
 	0x02a19020, 0x01117011, 0x400001f0, 0x400001f1,
 	0x01813122, 0x400001f2,
@@ -1794,7 +1798,7 @@ static unsigned int dell_922x_d81_pin_configs[10] = {
     102801AC
     102801D0
 */
-static unsigned int dell_922x_d82_pin_configs[10] = {
+static const unsigned int dell_922x_d82_pin_configs[10] = {
 	0x02214030, 0x01a19021, 0x01111012, 0x01114010,
 	0x02a19020, 0x01117011, 0x01451140, 0x400001f0,
 	0x01813122, 0x400001f1,
@@ -1804,7 +1808,7 @@ static unsigned int dell_922x_d82_pin_configs[10] = {
     STAC 922X pin configs for
     102801BF
 */
-static unsigned int dell_922x_m81_pin_configs[10] = {
+static const unsigned int dell_922x_m81_pin_configs[10] = {
 	0x0321101f, 0x01112024, 0x01111222, 0x91174220,
 	0x03a11050, 0x01116221, 0x90a70330, 0x01452340, 
 	0x40C003f1, 0x405003f0,
@@ -1814,61 +1818,61 @@ static unsigned int dell_922x_m81_pin_configs[10] = {
     STAC 9221 A1 pin configs for
     102801D7 (Dell XPS M1210)
 */
-static unsigned int dell_922x_m82_pin_configs[10] = {
+static const unsigned int dell_922x_m82_pin_configs[10] = {
 	0x02211211, 0x408103ff, 0x02a1123e, 0x90100310, 
 	0x408003f1, 0x0221121f, 0x03451340, 0x40c003f2, 
 	0x508003f3, 0x405003f4, 
 };
 
-static unsigned int d945gtp3_pin_configs[10] = {
+static const unsigned int d945gtp3_pin_configs[10] = {
 	0x0221401f, 0x01a19022, 0x01813021, 0x01014010,
 	0x40000100, 0x40000100, 0x40000100, 0x40000100,
 	0x02a19120, 0x40000100,
 };
 
-static unsigned int d945gtp5_pin_configs[10] = {
+static const unsigned int d945gtp5_pin_configs[10] = {
 	0x0221401f, 0x01011012, 0x01813024, 0x01014010,
 	0x01a19021, 0x01016011, 0x01452130, 0x40000100,
 	0x02a19320, 0x40000100,
 };
 
-static unsigned int intel_mac_v1_pin_configs[10] = {
+static const unsigned int intel_mac_v1_pin_configs[10] = {
 	0x0121e21f, 0x400000ff, 0x9017e110, 0x400000fd,
 	0x400000fe, 0x0181e020, 0x1145e030, 0x11c5e240,
 	0x400000fc, 0x400000fb,
 };
 
-static unsigned int intel_mac_v2_pin_configs[10] = {
+static const unsigned int intel_mac_v2_pin_configs[10] = {
 	0x0121e21f, 0x90a7012e, 0x9017e110, 0x400000fd,
 	0x400000fe, 0x0181e020, 0x1145e230, 0x500000fa,
 	0x400000fc, 0x400000fb,
 };
 
-static unsigned int intel_mac_v3_pin_configs[10] = {
+static const unsigned int intel_mac_v3_pin_configs[10] = {
 	0x0121e21f, 0x90a7012e, 0x9017e110, 0x400000fd,
 	0x400000fe, 0x0181e020, 0x1145e230, 0x11c5e240,
 	0x400000fc, 0x400000fb,
 };
 
-static unsigned int intel_mac_v4_pin_configs[10] = {
+static const unsigned int intel_mac_v4_pin_configs[10] = {
 	0x0321e21f, 0x03a1e02e, 0x9017e110, 0x9017e11f,
 	0x400000fe, 0x0381e020, 0x1345e230, 0x13c5e240,
 	0x400000fc, 0x400000fb,
 };
 
-static unsigned int intel_mac_v5_pin_configs[10] = {
+static const unsigned int intel_mac_v5_pin_configs[10] = {
 	0x0321e21f, 0x03a1e02e, 0x9017e110, 0x9017e11f,
 	0x400000fe, 0x0381e020, 0x1345e230, 0x13c5e240,
 	0x400000fc, 0x400000fb,
 };
 
-static unsigned int ecs202_pin_configs[10] = {
+static const unsigned int ecs202_pin_configs[10] = {
 	0x0221401f, 0x02a19020, 0x01a19020, 0x01114010,
 	0x408000f0, 0x01813022, 0x074510a0, 0x40c400f1,
 	0x9037012e, 0x40e000f2,
 };
 
-static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = {
+static const unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = {
 	[STAC_D945_REF] = ref922x_pin_configs,
 	[STAC_D945GTP3] = d945gtp3_pin_configs,
 	[STAC_D945GTP5] = d945gtp5_pin_configs,
@@ -1917,7 +1921,7 @@ static const char * const stac922x_models[STAC_922X_MODELS] = {
 	[STAC_922X_DELL_M82] = "dell-m82",
 };
 
-static struct snd_pci_quirk stac922x_cfg_tbl[] = {
+static const struct snd_pci_quirk stac922x_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_D945_REF),
@@ -2008,42 +2012,42 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = {
 	{} /* terminator */
 };
 
-static unsigned int ref927x_pin_configs[14] = {
+static const unsigned int ref927x_pin_configs[14] = {
 	0x02214020, 0x02a19080, 0x0181304e, 0x01014010,
 	0x01a19040, 0x01011012, 0x01016011, 0x0101201f, 
 	0x183301f0, 0x18a001f0, 0x18a001f0, 0x01442070,
 	0x01c42190, 0x40000100,
 };
 
-static unsigned int d965_3st_pin_configs[14] = {
+static const unsigned int d965_3st_pin_configs[14] = {
 	0x0221401f, 0x02a19120, 0x40000100, 0x01014011,
 	0x01a19021, 0x01813024, 0x40000100, 0x40000100,
 	0x40000100, 0x40000100, 0x40000100, 0x40000100,
 	0x40000100, 0x40000100
 };
 
-static unsigned int d965_5st_pin_configs[14] = {
+static const unsigned int d965_5st_pin_configs[14] = {
 	0x02214020, 0x02a19080, 0x0181304e, 0x01014010,
 	0x01a19040, 0x01011012, 0x01016011, 0x40000100,
 	0x40000100, 0x40000100, 0x40000100, 0x01442070,
 	0x40000100, 0x40000100
 };
 
-static unsigned int d965_5st_no_fp_pin_configs[14] = {
+static const unsigned int d965_5st_no_fp_pin_configs[14] = {
 	0x40000100, 0x40000100, 0x0181304e, 0x01014010,
 	0x01a19040, 0x01011012, 0x01016011, 0x40000100,
 	0x40000100, 0x40000100, 0x40000100, 0x01442070,
 	0x40000100, 0x40000100
 };
 
-static unsigned int dell_3st_pin_configs[14] = {
+static const unsigned int dell_3st_pin_configs[14] = {
 	0x02211230, 0x02a11220, 0x01a19040, 0x01114210,
 	0x01111212, 0x01116211, 0x01813050, 0x01112214,
 	0x403003fa, 0x90a60040, 0x90a60040, 0x404003fb,
 	0x40c003fc, 0x40000100
 };
 
-static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
+static const unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
 	[STAC_D965_REF_NO_JD] = ref927x_pin_configs,
 	[STAC_D965_REF]  = ref927x_pin_configs,
 	[STAC_D965_3ST]  = d965_3st_pin_configs,
@@ -2066,7 +2070,7 @@ static const char * const stac927x_models[STAC_927X_MODELS] = {
 	[STAC_927X_VOLKNOB]	= "volknob",
 };
 
-static struct snd_pci_quirk stac927x_cfg_tbl[] = {
+static const struct snd_pci_quirk stac927x_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_D965_REF),
@@ -2104,7 +2108,7 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
 	{} /* terminator */
 };
 
-static unsigned int ref9205_pin_configs[12] = {
+static const unsigned int ref9205_pin_configs[12] = {
 	0x40000100, 0x40000100, 0x01016011, 0x01014010,
 	0x01813122, 0x01a19021, 0x01019020, 0x40000100,
 	0x90a000f0, 0x90a000f0, 0x01441030, 0x01c41030
@@ -2121,7 +2125,7 @@ static unsigned int ref9205_pin_configs[12] = {
     10280228 (Dell Vostro 1500)
     10280229 (Dell Vostro 1700)
 */
-static unsigned int dell_9205_m42_pin_configs[12] = {
+static const unsigned int dell_9205_m42_pin_configs[12] = {
 	0x0321101F, 0x03A11020, 0x400003FA, 0x90170310,
 	0x400003FB, 0x400003FC, 0x400003FD, 0x40F000F9,
 	0x90A60330, 0x400003FF, 0x0144131F, 0x40C003FE,
@@ -2137,19 +2141,19 @@ static unsigned int dell_9205_m42_pin_configs[12] = {
     10280200
     10280201
 */
-static unsigned int dell_9205_m43_pin_configs[12] = {
+static const unsigned int dell_9205_m43_pin_configs[12] = {
 	0x0321101f, 0x03a11020, 0x90a70330, 0x90170310,
 	0x400000fe, 0x400000ff, 0x400000fd, 0x40f000f9,
 	0x400000fa, 0x400000fc, 0x0144131f, 0x40c003f8,
 };
 
-static unsigned int dell_9205_m44_pin_configs[12] = {
+static const unsigned int dell_9205_m44_pin_configs[12] = {
 	0x0421101f, 0x04a11020, 0x400003fa, 0x90170310,
 	0x400003fb, 0x400003fc, 0x400003fd, 0x400003f9,
 	0x90a60330, 0x400003ff, 0x01441340, 0x40c003fe,
 };
 
-static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = {
+static const unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = {
 	[STAC_9205_REF] = ref9205_pin_configs,
 	[STAC_9205_DELL_M42] = dell_9205_m42_pin_configs,
 	[STAC_9205_DELL_M43] = dell_9205_m43_pin_configs,
@@ -2166,7 +2170,7 @@ static const char * const stac9205_models[STAC_9205_MODELS] = {
 	[STAC_9205_EAPD] = "eapd",
 };
 
-static struct snd_pci_quirk stac9205_cfg_tbl[] = {
+static const struct snd_pci_quirk stac9205_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_9205_REF),
@@ -2214,7 +2218,7 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
 };
 
 static void stac92xx_set_config_regs(struct hda_codec *codec,
-				     unsigned int *pincfgs)
+				     const unsigned int *pincfgs)
 {
 	int i;
 	struct sigmatel_spec *spec = codec->spec;
@@ -2334,7 +2338,7 @@ static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 	return 0;
 }
 
-static struct hda_pcm_stream stac92xx_pcm_digital_playback = {
+static const struct hda_pcm_stream stac92xx_pcm_digital_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -2347,14 +2351,14 @@ static struct hda_pcm_stream stac92xx_pcm_digital_playback = {
 	},
 };
 
-static struct hda_pcm_stream stac92xx_pcm_digital_capture = {
+static const struct hda_pcm_stream stac92xx_pcm_digital_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
 	/* NID is set in stac92xx_build_pcms */
 };
 
-static struct hda_pcm_stream stac92xx_pcm_analog_playback = {
+static const struct hda_pcm_stream stac92xx_pcm_analog_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 8,
@@ -2366,7 +2370,7 @@ static struct hda_pcm_stream stac92xx_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream stac92xx_pcm_analog_alt_playback = {
+static const struct hda_pcm_stream stac92xx_pcm_analog_alt_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -2378,7 +2382,7 @@ static struct hda_pcm_stream stac92xx_pcm_analog_alt_playback = {
 	},
 };
 
-static struct hda_pcm_stream stac92xx_pcm_analog_capture = {
+static const struct hda_pcm_stream stac92xx_pcm_analog_capture = {
 	.channels_min = 2,
 	.channels_max = 2,
 	/* NID + .substreams is set in stac92xx_build_pcms */
@@ -2487,7 +2491,7 @@ static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_info *uinfo)
 {
 	int i;
-	static char *texts[] = {
+	static const char * const texts[] = {
 		"Mic In", "Line In", "Line Out"
 	};
 
@@ -2556,7 +2560,7 @@ static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol,
 static int stac92xx_io_switch_info(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[2];
+	char *texts[2];
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct sigmatel_spec *spec = codec->spec;
 
@@ -2687,7 +2691,7 @@ enum {
 	STAC_CTL_WIDGET_DC_BIAS
 };
 
-static struct snd_kcontrol_new stac92xx_control_templates[] = {
+static const struct snd_kcontrol_new stac92xx_control_templates[] = {
 	HDA_CODEC_VOLUME(NULL, 0, 0, 0),
 	HDA_CODEC_MUTE(NULL, 0, 0, 0),
 	HDA_CODEC_MUTE_BEEP(NULL, 0, 0, 0),
@@ -2701,7 +2705,7 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = {
 /* add dynamic controls */
 static struct snd_kcontrol_new *
 stac_control_new(struct sigmatel_spec *spec,
-		 struct snd_kcontrol_new *ktemp,
+		 const struct snd_kcontrol_new *ktemp,
 		 const char *name,
 		 unsigned int subdev)
 {
@@ -2724,7 +2728,7 @@ stac_control_new(struct sigmatel_spec *spec,
 }
 
 static int stac92xx_add_control_temp(struct sigmatel_spec *spec,
-				     struct snd_kcontrol_new *ktemp,
+				     const struct snd_kcontrol_new *ktemp,
 				     int idx, const char *name,
 				     unsigned long val)
 {
@@ -2754,7 +2758,7 @@ static inline int stac92xx_add_control(struct sigmatel_spec *spec, int type,
 	return stac92xx_add_control_idx(spec, type, 0, name, val);
 }
 
-static struct snd_kcontrol_new stac_input_src_temp = {
+static const struct snd_kcontrol_new stac_input_src_temp = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Input Source",
 	.info = stac92xx_mux_enum_info,
@@ -3072,7 +3076,8 @@ static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid)
 		printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid);
 		return 1;
 	} else {
-		spec->multiout.dac_nids[spec->multiout.num_dacs] = nid;
+		snd_BUG_ON(spec->multiout.dac_nids != spec->dac_nids);
+		spec->dac_nids[spec->multiout.num_dacs] = nid;
 		spec->multiout.num_dacs++;
 	}
 	return 0;
@@ -3109,8 +3114,7 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs,
 
 	for (i = 0; i < num_outs && i < ARRAY_SIZE(chname); i++) {
 		if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) {
-			wid_caps = get_wcaps(codec, pins[i]);
-			if (wid_caps & AC_WCAP_UNSOL_CAP)
+			if (is_jack_detectable(codec, pins[i]))
 				spec->hp_detect = 1;
 		}
 		nid = dac_nids[i];
@@ -3309,7 +3313,7 @@ static int stac92xx_dig_beep_switch_put(struct snd_kcontrol *kcontrol,
 	return snd_hda_enable_beep_device(codec, ucontrol->value.integer.value[0]);
 }
 
-static struct snd_kcontrol_new stac92xx_dig_beep_ctrl = {
+static const struct snd_kcontrol_new stac92xx_dig_beep_ctrl = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.info = stac92xx_dig_beep_switch_info,
 	.get = stac92xx_dig_beep_switch_get,
@@ -3516,14 +3520,18 @@ static int check_mic_pin(struct hda_codec *codec, hda_nid_t nid,
 			 hda_nid_t *fixed, hda_nid_t *ext, hda_nid_t *dock)
 {
 	unsigned int cfg;
+	unsigned int type;
 
 	if (!nid)
 		return 0;
 	cfg = snd_hda_codec_get_pincfg(codec, nid);
+	type = get_defcfg_device(cfg);
 	switch (snd_hda_get_input_pin_attr(cfg)) {
 	case INPUT_PIN_ATTR_INT:
 		if (*fixed)
 			return 1; /* already occupied */
+		if (type != AC_JACK_MIC_IN)
+			return 1; /* invalid type */
 		*fixed = nid;
 		break;
 	case INPUT_PIN_ATTR_UNUSED:
@@ -3531,11 +3539,15 @@ static int check_mic_pin(struct hda_codec *codec, hda_nid_t nid,
 	case INPUT_PIN_ATTR_DOCK:
 		if (*dock)
 			return 1; /* already occupied */
+		if (type != AC_JACK_MIC_IN && type != AC_JACK_LINE_IN)
+			return 1; /* invalid type */
 		*dock = nid;
 		break;
 	default:
 		if (*ext)
 			return 1; /* already occupied */
+		if (type != AC_JACK_MIC_IN)
+			return 1; /* invalid type */
 		*ext = nid;
 		break;
 	}
@@ -3591,10 +3603,6 @@ static int stac_check_auto_mic(struct hda_codec *codec)
 	hda_nid_t fixed, ext, dock;
 	int i;
 
-	for (i = 0; i < cfg->num_inputs; i++) {
-		if (cfg->inputs[i].type >= AUTO_PIN_LINE_IN)
-			return 0; /* must be exclusively mics */
-	}
 	fixed = ext = dock = 0;
 	for (i = 0; i < cfg->num_inputs; i++)
 		if (check_mic_pin(codec, cfg->inputs[i].pin,
@@ -3606,7 +3614,7 @@ static int stac_check_auto_mic(struct hda_codec *codec)
 			return 0;
 	if (!fixed || (!ext && !dock))
 		return 0; /* no input to switch */
-	if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP))
+	if (!is_jack_detectable(codec, ext))
 		return 0; /* no unsol support */
 	if (set_mic_route(codec, &spec->ext_mic, ext) ||
 	    set_mic_route(codec, &spec->int_mic, fixed) ||
@@ -3921,13 +3929,11 @@ static int stac9200_auto_create_hp_ctls(struct hda_codec *codec,
 {
 	struct sigmatel_spec *spec = codec->spec;
 	hda_nid_t pin = cfg->hp_pins[0];
-	unsigned int wid_caps;
 
 	if (! pin)
 		return 0;
 
-	wid_caps = get_wcaps(codec, pin);
-	if (wid_caps & AC_WCAP_UNSOL_CAP)
+	if (is_jack_detectable(codec, pin))
 		spec->hp_detect = 1;
 
 	return 0;
@@ -4138,7 +4144,7 @@ static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid,
 	struct sigmatel_event *event;
 	int tag;
 
-	if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP))
+	if (!is_jack_detectable(codec, nid))
 		return 0;
 	event = stac_get_event(codec, nid);
 	if (event) {
@@ -4171,7 +4177,7 @@ static void stac92xx_power_down(struct hda_codec *codec)
 	struct sigmatel_spec *spec = codec->spec;
 
 	/* power down inactive DACs */
-	hda_nid_t *dac;
+	const hda_nid_t *dac;
 	for (dac = spec->dac_list; *dac; dac++)
 		if (!check_all_dac_nids(spec, *dac))
 			snd_hda_codec_write(codec, *dac, 0,
@@ -4644,7 +4650,7 @@ static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx)
 }
 
 static int stac92xx_connected_ports(struct hda_codec *codec,
-					 hda_nid_t *nids, int num_nids)
+				    const hda_nid_t *nids, int num_nids)
 {
 	struct sigmatel_spec *spec = codec->spec;
 	int idx, num;
@@ -4968,7 +4974,7 @@ static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
 }
 #endif
 
-static struct hda_codec_ops stac92xx_patch_ops = {
+static const struct hda_codec_ops stac92xx_patch_ops = {
 	.build_controls = stac92xx_build_controls,
 	.build_pcms = stac92xx_build_pcms,
 	.init = stac92xx_init,
@@ -5588,7 +5594,7 @@ static int stac_hp_bass_gpio_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static struct snd_kcontrol_new stac_hp_bass_sw_ctrl = {
+static const struct snd_kcontrol_new stac_hp_bass_sw_ctrl = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.info = stac_hp_bass_gpio_info,
 	.get = stac_hp_bass_gpio_get,
@@ -5612,7 +5618,7 @@ static int stac_add_hp_bass_switch(struct hda_codec *codec)
 static int patch_stac92hd71bxx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
-	struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init;
+	const struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init;
 	unsigned int pin_cfg;
 	int err = 0;
 
@@ -5705,9 +5711,9 @@ again:
 		unmute_init++;
 		snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0);
 		snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3);
-		stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS - 1] = 0;
+		spec->dmic_nids = stac92hd71bxx_dmic_5port_nids;
 		spec->num_dmics = stac92xx_connected_ports(codec,
-					stac92hd71bxx_dmic_nids,
+					stac92hd71bxx_dmic_5port_nids,
 					STAC92HD71BXX_NUM_DMICS - 1);
 		break;
 	case 0x111d7603: /* 6 Port with Analog Mixer */
@@ -5729,15 +5735,6 @@ again:
 	if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP)
 		snd_hda_sequence_write_cache(codec, unmute_init);
 
-	/* Some HP machines seem to have unstable codec communications
-	 * especially with ATI fglrx driver.  For recovering from the
-	 * CORB/RIRB stall, allow the BUS reset and keep always sync
-	 */
-	if (spec->board_config == STAC_HP_DV5) {
-		codec->bus->sync_write = 1;
-		codec->bus->allow_bus_reset = 1;
-	}
-
 	spec->aloopback_ctl = stac92hd71bxx_loopback;
 	spec->aloopback_mask = 0x50;
 	spec->aloopback_shift = 0;
@@ -6223,31 +6220,31 @@ static int patch_stac9205(struct hda_codec *codec)
  * STAC9872 hack
  */
 
-static struct hda_verb stac9872_core_init[] = {
+static const struct hda_verb stac9872_core_init[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */
 	{}
 };
 
-static hda_nid_t stac9872_pin_nids[] = {
+static const hda_nid_t stac9872_pin_nids[] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f,
 	0x11, 0x13, 0x14,
 };
 
-static hda_nid_t stac9872_adc_nids[] = {
+static const hda_nid_t stac9872_adc_nids[] = {
 	0x8 /*,0x6*/
 };
 
-static hda_nid_t stac9872_mux_nids[] = {
+static const hda_nid_t stac9872_mux_nids[] = {
 	0x15
 };
 
-static unsigned long stac9872_capvols[] = {
+static const unsigned long stac9872_capvols[] = {
 	HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
 };
 #define stac9872_capsws		stac9872_capvols
 
-static unsigned int stac9872_vaio_pin_configs[9] = {
+static const unsigned int stac9872_vaio_pin_configs[9] = {
 	0x03211020, 0x411111f0, 0x411111f0, 0x03a15030,
 	0x411111f0, 0x90170110, 0x411111f0, 0x411111f0,
 	0x90a7013e
@@ -6258,11 +6255,11 @@ static const char * const stac9872_models[STAC_9872_MODELS] = {
 	[STAC_9872_VAIO] = "vaio",
 };
 
-static unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = {
+static const unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = {
 	[STAC_9872_VAIO] = stac9872_vaio_pin_configs,
 };
 
-static struct snd_pci_quirk stac9872_cfg_tbl[] = {
+static const struct snd_pci_quirk stac9872_cfg_tbl[] = {
 	SND_PCI_QUIRK_MASK(0x104d, 0xfff0, 0x81e0,
 			   "Sony VAIO F/S", STAC_9872_VAIO),
 	{} /* terminator */
@@ -6316,7 +6313,7 @@ static int patch_stac9872(struct hda_codec *codec)
 /*
  * patch entries
  */
-static struct hda_codec_preset snd_hda_preset_sigmatel[] = {
+static const struct hda_codec_preset snd_hda_preset_sigmatel[] = {
  	{ .id = 0x83847690, .name = "STAC9200", .patch = patch_stac9200 },
  	{ .id = 0x83847882, .name = "STAC9220 A1", .patch = patch_stac922x },
  	{ .id = 0x83847680, .name = "STAC9221 A1", .patch = patch_stac922x },
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 1371b57c11e..605c99e1e52 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -98,24 +98,30 @@ enum VIA_HDA_CODEC {
 	VT1716S,
 	VT2002P,
 	VT1812,
+	VT1802,
 	CODEC_TYPES,
 };
 
+#define VT2002P_COMPATIBLE(spec) \
+	((spec)->codec_type == VT2002P ||\
+	 (spec)->codec_type == VT1812 ||\
+	 (spec)->codec_type == VT1802)
+
 struct via_spec {
 	/* codec parameterization */
-	struct snd_kcontrol_new *mixers[6];
+	const struct snd_kcontrol_new *mixers[6];
 	unsigned int num_mixers;
 
-	struct hda_verb *init_verbs[5];
+	const struct hda_verb *init_verbs[5];
 	unsigned int num_iverbs;
 
 	char *stream_name_analog;
-	struct hda_pcm_stream *stream_analog_playback;
-	struct hda_pcm_stream *stream_analog_capture;
+	const struct hda_pcm_stream *stream_analog_playback;
+	const struct hda_pcm_stream *stream_analog_capture;
 
 	char *stream_name_digital;
-	struct hda_pcm_stream *stream_digital_playback;
-	struct hda_pcm_stream *stream_digital_capture;
+	const struct hda_pcm_stream *stream_digital_playback;
+	const struct hda_pcm_stream *stream_digital_capture;
 
 	/* playback */
 	struct hda_multi_out multiout;
@@ -123,7 +129,7 @@ struct via_spec {
 
 	/* capture */
 	unsigned int num_adc_nids;
-	hda_nid_t *adc_nids;
+	const hda_nid_t *adc_nids;
 	hda_nid_t mux_nids[3];
 	hda_nid_t dig_in_nid;
 	hda_nid_t dig_in_pin;
@@ -154,6 +160,9 @@ struct via_spec {
 	struct delayed_work vt1708_hp_work;
 	int vt1708_jack_detectect;
 	int vt1708_hp_present;
+
+	void (*set_widgets_power_state)(struct hda_codec *codec);
+
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	struct hda_loopback_check loopback;
 #endif
@@ -218,17 +227,19 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec)
 		codec_type = VT1812;
 	else if (dev_id == 0x0440)
 		codec_type = VT1708S;
+	else if ((dev_id & 0xfff) == 0x446)
+		codec_type = VT1802;
 	else
 		codec_type = UNKNOWN;
 	return codec_type;
 };
 
+#define VIA_JACK_EVENT		0x20
 #define VIA_HP_EVENT		0x01
 #define VIA_GPIO_EVENT		0x02
-#define VIA_JACK_EVENT		0x04
-#define VIA_MONO_EVENT		0x08
-#define VIA_SPEAKER_EVENT	0x10
-#define VIA_BIND_HP_EVENT	0x20
+#define VIA_MONO_EVENT		0x03
+#define VIA_SPEAKER_EVENT	0x04
+#define VIA_BIND_HP_EVENT	0x05
 
 enum {
 	VIA_CTL_WIDGET_VOL,
@@ -245,7 +256,6 @@ enum {
 };
 
 static void analog_low_current_mode(struct hda_codec *codec, int stream_idle);
-static void set_jack_power_state(struct hda_codec *codec);
 static int is_aa_path_mute(struct hda_codec *codec);
 
 static void vt1708_start_hp_work(struct via_spec *spec)
@@ -271,6 +281,12 @@ static void vt1708_stop_hp_work(struct via_spec *spec)
 	cancel_delayed_work_sync(&spec->vt1708_hp_work);
 }
 
+static void set_widgets_power_state(struct hda_codec *codec)
+{
+	struct via_spec *spec = codec->spec;
+	if (spec->set_widgets_power_state)
+		spec->set_widgets_power_state(codec);
+}
 
 static int analog_input_switch_put(struct snd_kcontrol *kcontrol,
 				   struct snd_ctl_elem_value *ucontrol)
@@ -278,7 +294,7 @@ static int analog_input_switch_put(struct snd_kcontrol *kcontrol,
 	int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 
-	set_jack_power_state(codec);
+	set_widgets_power_state(codec);
 	analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1);
 	if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) {
 		if (is_aa_path_mute(codec))
@@ -394,54 +410,54 @@ static int bind_pin_switch_put(struct snd_kcontrol *kcontrol,
 			.put = bind_pin_switch_put,			\
 			.private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) }
 
-static struct snd_kcontrol_new via_control_templates[] = {
+static const struct snd_kcontrol_new via_control_templates[] = {
 	HDA_CODEC_VOLUME(NULL, 0, 0, 0),
 	HDA_CODEC_MUTE(NULL, 0, 0, 0),
 	ANALOG_INPUT_MUTE,
 	BIND_PIN_MUTE,
 };
 
-static hda_nid_t vt1708_adc_nids[2] = {
+static const hda_nid_t vt1708_adc_nids[2] = {
 	/* ADC1-2 */
 	0x15, 0x27
 };
 
-static hda_nid_t vt1709_adc_nids[3] = {
+static const hda_nid_t vt1709_adc_nids[3] = {
 	/* ADC1-2 */
 	0x14, 0x15, 0x16
 };
 
-static hda_nid_t vt1708B_adc_nids[2] = {
+static const hda_nid_t vt1708B_adc_nids[2] = {
 	/* ADC1-2 */
 	0x13, 0x14
 };
 
-static hda_nid_t vt1708S_adc_nids[2] = {
+static const hda_nid_t vt1708S_adc_nids[2] = {
 	/* ADC1-2 */
 	0x13, 0x14
 };
 
-static hda_nid_t vt1702_adc_nids[3] = {
+static const hda_nid_t vt1702_adc_nids[3] = {
 	/* ADC1-2 */
 	0x12, 0x20, 0x1F
 };
 
-static hda_nid_t vt1718S_adc_nids[2] = {
+static const hda_nid_t vt1718S_adc_nids[2] = {
 	/* ADC1-2 */
 	0x10, 0x11
 };
 
-static hda_nid_t vt1716S_adc_nids[2] = {
+static const hda_nid_t vt1716S_adc_nids[2] = {
 	/* ADC1-2 */
 	0x13, 0x14
 };
 
-static hda_nid_t vt2002P_adc_nids[2] = {
+static const hda_nid_t vt2002P_adc_nids[2] = {
 	/* ADC1-2 */
 	0x10, 0x11
 };
 
-static hda_nid_t vt1812_adc_nids[2] = {
+static const hda_nid_t vt1812_adc_nids[2] = {
 	/* ADC1-2 */
 	0x10, 0x11
 };
@@ -471,7 +487,7 @@ static int __via_add_control(struct via_spec *spec, int type, const char *name,
 	__via_add_control(spec, type, name, 0, val)
 
 static struct snd_kcontrol_new *via_clone_control(struct via_spec *spec,
-						struct snd_kcontrol_new *tmpl)
+				const struct snd_kcontrol_new *tmpl)
 {
 	struct snd_kcontrol_new *knew;
 
@@ -602,482 +618,6 @@ static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid,
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm);
 }
 
-static void set_jack_power_state(struct hda_codec *codec)
-{
-	struct via_spec *spec = codec->spec;
-	int imux_is_smixer;
-	unsigned int parm;
-
-	if (spec->codec_type == VT1702) {
-		imux_is_smixer = snd_hda_codec_read(
-			codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3;
-		/* inputs */
-		/* PW 1/2/5 (14h/15h/18h) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x14, &parm);
-		set_pin_power_state(codec, 0x15, &parm);
-		set_pin_power_state(codec, 0x18, &parm);
-		if (imux_is_smixer)
-			parm = AC_PWRST_D0; /* SW0 = stereo mixer (idx 3) */
-		/* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */
-		snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-
-		/* outputs */
-		/* PW 3/4 (16h/17h) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x16, &parm);
-		set_pin_power_state(codec, 0x17, &parm);
-		/* MW0 (1ah), AOW 0/1 (10h/1dh) */
-		snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE,
-				    imux_is_smixer ? AC_PWRST_D0 : parm);
-		snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-	} else if (spec->codec_type == VT1708B_8CH
-		   || spec->codec_type == VT1708B_4CH
-		   || spec->codec_type == VT1708S) {
-		/* SW0 (17h) = stereo mixer */
-		int is_8ch = spec->codec_type != VT1708B_4CH;
-		imux_is_smixer = snd_hda_codec_read(
-			codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00)
-			== ((spec->codec_type == VT1708S)  ? 5 : 0);
-		/* inputs */
-		/* PW 1/2/5 (1ah/1bh/1eh) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x1a, &parm);
-		set_pin_power_state(codec, 0x1b, &parm);
-		set_pin_power_state(codec, 0x1e, &parm);
-		if (imux_is_smixer)
-			parm = AC_PWRST_D0;
-		/* SW0 (17h), AIW 0/1 (13h/14h) */
-		snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-
-		/* outputs */
-		/* PW0 (19h), SW1 (18h), AOW1 (11h) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x19, &parm);
-		if (spec->smart51_enabled)
-			parm = AC_PWRST_D0;
-		snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-
-		/* PW6 (22h), SW2 (26h), AOW2 (24h) */
-		if (is_8ch) {
-			parm = AC_PWRST_D3;
-			set_pin_power_state(codec, 0x22, &parm);
-			if (spec->smart51_enabled)
-				parm = AC_PWRST_D0;
-			snd_hda_codec_write(codec, 0x26, 0,
-					    AC_VERB_SET_POWER_STATE, parm);
-			snd_hda_codec_write(codec, 0x24, 0,
-					    AC_VERB_SET_POWER_STATE, parm);
-		}
-
-		/* PW 3/4/7 (1ch/1dh/23h) */
-		parm = AC_PWRST_D3;
-		/* force to D0 for internal Speaker */
-		set_pin_power_state(codec, 0x1c, &parm);
-		set_pin_power_state(codec, 0x1d, &parm);
-		if (is_8ch)
-			set_pin_power_state(codec, 0x23, &parm);
-		/* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */
-		snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE,
-				    imux_is_smixer ? AC_PWRST_D0 : parm);
-		snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		if (is_8ch) {
-			snd_hda_codec_write(codec, 0x25, 0,
-					    AC_VERB_SET_POWER_STATE, parm);
-			snd_hda_codec_write(codec, 0x27, 0,
-					    AC_VERB_SET_POWER_STATE, parm);
-		}
-	}  else if (spec->codec_type == VT1718S) {
-		/* MUX6 (1eh) = stereo mixer */
-		imux_is_smixer = snd_hda_codec_read(
-			codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5;
-		/* inputs */
-		/* PW 5/6/7 (29h/2ah/2bh) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x29, &parm);
-		set_pin_power_state(codec, 0x2a, &parm);
-		set_pin_power_state(codec, 0x2b, &parm);
-		if (imux_is_smixer)
-			parm = AC_PWRST_D0;
-		/* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */
-		snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-
-		/* outputs */
-		/* PW3 (27h), MW2 (1ah), AOW3 (bh) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x27, &parm);
-		snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-
-		/* PW2 (26h), AOW2 (ah) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x26, &parm);
-		snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-
-		/* PW0/1 (24h/25h) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x24, &parm);
-		set_pin_power_state(codec, 0x25, &parm);
-		if (!spec->hp_independent_mode) /* check for redirected HP */
-			set_pin_power_state(codec, 0x28, &parm);
-		snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		/* MW9 (21h), Mw2 (1ah), AOW0 (8h) */
-		snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE,
-				    imux_is_smixer ? AC_PWRST_D0 : parm);
-		if (spec->hp_independent_mode) {
-			/* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */
-			parm = AC_PWRST_D3;
-			set_pin_power_state(codec, 0x28, &parm);
-			snd_hda_codec_write(codec, 0x1b, 0,
-					    AC_VERB_SET_POWER_STATE, parm);
-			snd_hda_codec_write(codec, 0x34, 0,
-					    AC_VERB_SET_POWER_STATE, parm);
-			snd_hda_codec_write(codec, 0xc, 0,
-					    AC_VERB_SET_POWER_STATE, parm);
-		}
-	} else if (spec->codec_type == VT1716S) {
-		unsigned int mono_out, present;
-		/* SW0 (17h) = stereo mixer */
-		imux_is_smixer = snd_hda_codec_read(
-			codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) ==  5;
-		/* inputs */
-		/* PW 1/2/5 (1ah/1bh/1eh) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x1a, &parm);
-		set_pin_power_state(codec, 0x1b, &parm);
-		set_pin_power_state(codec, 0x1e, &parm);
-		if (imux_is_smixer)
-			parm = AC_PWRST_D0;
-		/* SW0 (17h), AIW0(13h) */
-		snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x1e, &parm);
-		/* PW11 (22h) */
-		if (spec->dmic_enabled)
-			set_pin_power_state(codec, 0x22, &parm);
-		else
-			snd_hda_codec_write(
-				codec, 0x22, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-
-		/* SW2(26h), AIW1(14h) */
-		snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-
-		/* outputs */
-		/* PW0 (19h), SW1 (18h), AOW1 (11h) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x19, &parm);
-		/* Smart 5.1 PW2(1bh) */
-		if (spec->smart51_enabled)
-			set_pin_power_state(codec, 0x1b, &parm);
-		snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-
-		/* PW7 (23h), SW3 (27h), AOW3 (25h) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x23, &parm);
-		/* Smart 5.1 PW1(1ah) */
-		if (spec->smart51_enabled)
-			set_pin_power_state(codec, 0x1a, &parm);
-		snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-
-		/* Smart 5.1 PW5(1eh) */
-		if (spec->smart51_enabled)
-			set_pin_power_state(codec, 0x1e, &parm);
-		snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-
-		/* Mono out */
-		/* SW4(28h)->MW1(29h)-> PW12 (2ah)*/
-		present = snd_hda_jack_detect(codec, 0x1c);
-		if (present)
-			mono_out = 0;
-		else {
-			present = snd_hda_jack_detect(codec, 0x1d);
-			if (!spec->hp_independent_mode && present)
-				mono_out = 0;
-			else
-				mono_out = 1;
-		}
-		parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3;
-		snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-		snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE,
-				    parm);
-
-		/* PW 3/4 (1ch/1dh) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x1c, &parm);
-		set_pin_power_state(codec, 0x1d, &parm);
-		/* HP Independent Mode, power on AOW3 */
-		if (spec->hp_independent_mode)
-			snd_hda_codec_write(codec, 0x25, 0,
-					    AC_VERB_SET_POWER_STATE, parm);
-
-		/* force to D0 for internal Speaker */
-		/* MW0 (16h), AOW0 (10h) */
-		snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE,
-				    imux_is_smixer ? AC_PWRST_D0 : parm);
-		snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE,
-				    mono_out ? AC_PWRST_D0 : parm);
-	} else if (spec->codec_type == VT2002P) {
-		unsigned int present;
-		/* MUX9 (1eh) = stereo mixer */
-		imux_is_smixer = snd_hda_codec_read(
-			codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3;
-		/* inputs */
-		/* PW 5/6/7 (29h/2ah/2bh) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x29, &parm);
-		set_pin_power_state(codec, 0x2a, &parm);
-		set_pin_power_state(codec, 0x2b, &parm);
-		if (imux_is_smixer)
-			parm = AC_PWRST_D0;
-		/* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */
-		snd_hda_codec_write(codec, 0x1e, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-		snd_hda_codec_write(codec, 0x1f, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-		snd_hda_codec_write(codec, 0x10, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-		snd_hda_codec_write(codec, 0x11, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-
-		/* outputs */
-		/* AOW0 (8h)*/
-		snd_hda_codec_write(codec, 0x8, 0,
-				    AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
-
-		/* PW4 (26h), MW4 (1ch), MUX4(37h) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x26, &parm);
-		snd_hda_codec_write(codec, 0x1c, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-		snd_hda_codec_write(codec, 0x37,
-				    0, AC_VERB_SET_POWER_STATE, parm);
-
-		/* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x25, &parm);
-		snd_hda_codec_write(codec, 0x19, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-		snd_hda_codec_write(codec, 0x35, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-		if (spec->hp_independent_mode)	{
-			snd_hda_codec_write(codec, 0x9, 0,
-					    AC_VERB_SET_POWER_STATE, parm);
-		}
-
-		/* Class-D */
-		/* PW0 (24h), MW0(18h), MUX0(34h) */
-		present = snd_hda_jack_detect(codec, 0x25);
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x24, &parm);
-		if (present) {
-			snd_hda_codec_write(
-				codec, 0x18, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-			snd_hda_codec_write(
-				codec, 0x34, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-		} else {
-			snd_hda_codec_write(
-				codec, 0x18, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
-			snd_hda_codec_write(
-				codec, 0x34, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
-		}
-
-		/* Mono Out */
-		/* PW15 (31h), MW8(17h), MUX8(3bh) */
-		present = snd_hda_jack_detect(codec, 0x26);
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x31, &parm);
-		if (present) {
-			snd_hda_codec_write(
-				codec, 0x17, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-			snd_hda_codec_write(
-				codec, 0x3b, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-		} else {
-			snd_hda_codec_write(
-				codec, 0x17, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
-			snd_hda_codec_write(
-				codec, 0x3b, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
-		}
-
-		/* MW9 (21h) */
-		if (imux_is_smixer || !is_aa_path_mute(codec))
-			snd_hda_codec_write(
-				codec, 0x21, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
-		else
-			snd_hda_codec_write(
-				codec, 0x21, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-	} else if (spec->codec_type == VT1812) {
-		unsigned int present;
-		/* MUX10 (1eh) = stereo mixer */
-		imux_is_smixer = snd_hda_codec_read(
-			codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5;
-		/* inputs */
-		/* PW 5/6/7 (29h/2ah/2bh) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x29, &parm);
-		set_pin_power_state(codec, 0x2a, &parm);
-		set_pin_power_state(codec, 0x2b, &parm);
-		if (imux_is_smixer)
-			parm = AC_PWRST_D0;
-		/* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */
-		snd_hda_codec_write(codec, 0x1e, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-		snd_hda_codec_write(codec, 0x1f, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-		snd_hda_codec_write(codec, 0x10, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-		snd_hda_codec_write(codec, 0x11, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-
-		/* outputs */
-		/* AOW0 (8h)*/
-		snd_hda_codec_write(codec, 0x8, 0,
-				    AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
-
-		/* PW4 (28h), MW4 (18h), MUX4(38h) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x28, &parm);
-		snd_hda_codec_write(codec, 0x18, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-		snd_hda_codec_write(codec, 0x38, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-
-		/* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x25, &parm);
-		snd_hda_codec_write(codec, 0x15, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-		snd_hda_codec_write(codec, 0x35, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-		if (spec->hp_independent_mode)	{
-			snd_hda_codec_write(codec, 0x9, 0,
-					    AC_VERB_SET_POWER_STATE, parm);
-		}
-
-		/* Internal Speaker */
-		/* PW0 (24h), MW0(14h), MUX0(34h) */
-		present = snd_hda_jack_detect(codec, 0x25);
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x24, &parm);
-		if (present) {
-			snd_hda_codec_write(codec, 0x14, 0,
-					    AC_VERB_SET_POWER_STATE,
-					    AC_PWRST_D3);
-			snd_hda_codec_write(codec, 0x34, 0,
-					    AC_VERB_SET_POWER_STATE,
-					    AC_PWRST_D3);
-		} else {
-			snd_hda_codec_write(codec, 0x14, 0,
-					    AC_VERB_SET_POWER_STATE,
-					    AC_PWRST_D0);
-			snd_hda_codec_write(codec, 0x34, 0,
-					    AC_VERB_SET_POWER_STATE,
-					    AC_PWRST_D0);
-		}
-		/* Mono Out */
-		/* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */
-		present = snd_hda_jack_detect(codec, 0x28);
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x31, &parm);
-		if (present) {
-			snd_hda_codec_write(codec, 0x1c, 0,
-					    AC_VERB_SET_POWER_STATE,
-					    AC_PWRST_D3);
-			snd_hda_codec_write(codec, 0x3c, 0,
-					    AC_VERB_SET_POWER_STATE,
-					    AC_PWRST_D3);
-			snd_hda_codec_write(codec, 0x3e, 0,
-					    AC_VERB_SET_POWER_STATE,
-					    AC_PWRST_D3);
-		} else {
-			snd_hda_codec_write(codec, 0x1c, 0,
-					    AC_VERB_SET_POWER_STATE,
-					    AC_PWRST_D0);
-			snd_hda_codec_write(codec, 0x3c, 0,
-					    AC_VERB_SET_POWER_STATE,
-					    AC_PWRST_D0);
-			snd_hda_codec_write(codec, 0x3e, 0,
-					    AC_VERB_SET_POWER_STATE,
-					    AC_PWRST_D0);
-		}
-
-		/* PW15 (33h), MW15 (1dh), MUX15(3dh) */
-		parm = AC_PWRST_D3;
-		set_pin_power_state(codec, 0x33, &parm);
-		snd_hda_codec_write(codec, 0x1d, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-		snd_hda_codec_write(codec, 0x3d, 0,
-				    AC_VERB_SET_POWER_STATE, parm);
-
-		/* MW9 (21h) */
-		if (imux_is_smixer || !is_aa_path_mute(codec))
-			snd_hda_codec_write(
-				codec, 0x21, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
-		else
-			snd_hda_codec_write(
-				codec, 0x21, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-	}
-}
-
 /*
  * input MUX handling
  */
@@ -1120,7 +660,7 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol,
 				     spec->mux_nids[adc_idx],
 				     &spec->cur_mux[adc_idx]);
 	/* update jack power state */
-	set_jack_power_state(codec);
+	set_widgets_power_state(codec);
 
 	return ret;
 }
@@ -1168,6 +708,9 @@ static hda_nid_t side_mute_channel(struct via_spec *spec)
 	case VT1709_10CH:	return 0x29;
 	case VT1708B_8CH:	/* fall thru */
 	case VT1708S:		return 0x27;
+	case VT2002P:		return 0x19;
+	case VT1802:		return 0x15;
+	case VT1812:		return 0x15;
 	default:		return 0;
 	}
 }
@@ -1176,13 +719,22 @@ static int update_side_mute_status(struct hda_codec *codec)
 {
 	/* mute side channel */
 	struct via_spec *spec = codec->spec;
-	unsigned int parm = spec->hp_independent_mode
-		? AMP_OUT_MUTE : AMP_OUT_UNMUTE;
+	unsigned int parm;
 	hda_nid_t sw3 = side_mute_channel(spec);
 
-	if (sw3)
-		snd_hda_codec_write(codec, sw3, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-				    parm);
+	if (sw3) {
+		if (VT2002P_COMPATIBLE(spec))
+			parm = spec->hp_independent_mode ?
+			       AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1);
+		else
+			parm = spec->hp_independent_mode ?
+			       AMP_OUT_MUTE : AMP_OUT_UNMUTE;
+		snd_hda_codec_write(codec, sw3, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, parm);
+		if (spec->codec_type == VT1812)
+			snd_hda_codec_write(codec, 0x1d, 0,
+					    AC_VERB_SET_AMP_GAIN_MUTE, parm);
+	}
 	return 0;
 }
 
@@ -1217,19 +769,18 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol,
 	    || spec->codec_type == VT1702
 	    || spec->codec_type == VT1718S
 	    || spec->codec_type == VT1716S
-	    || spec->codec_type == VT2002P
-	    || spec->codec_type == VT1812) {
+	    || VT2002P_COMPATIBLE(spec)) {
 		activate_ctl(codec, "Headphone Playback Volume",
 			     spec->hp_independent_mode);
 		activate_ctl(codec, "Headphone Playback Switch",
 			     spec->hp_independent_mode);
 	}
 	/* update jack power state */
-	set_jack_power_state(codec);
+	set_widgets_power_state(codec);
 	return 0;
 }
 
-static struct snd_kcontrol_new via_hp_mixer[2] = {
+static const struct snd_kcontrol_new via_hp_mixer[2] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Independent HP",
@@ -1256,6 +807,7 @@ static int via_hp_build(struct hda_codec *codec)
 		nid = 0x34;
 		break;
 	case VT2002P:
+	case VT1802:
 		nid = 0x35;
 		break;
 	case VT1812:
@@ -1292,14 +844,18 @@ static void notify_aa_path_ctls(struct hda_codec *codec)
 {
 	int i;
 	struct snd_ctl_elem_id id;
-	const char *labels[] = {"Mic", "Front Mic", "Line"};
+	const char *labels[] = {"Mic", "Front Mic", "Line", "Rear Mic"};
+	struct snd_kcontrol *ctl;
 
 	memset(&id, 0, sizeof(id));
 	id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
 	for (i = 0; i < ARRAY_SIZE(labels); i++) {
 		sprintf(id.name, "%s Playback Volume", labels[i]);
-		snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE,
-			       &id);
+		ctl = snd_hda_find_mixer_ctl(codec, id.name);
+		if (ctl)
+			snd_ctl_notify(codec->bus->card,
+					SNDRV_CTL_EVENT_MASK_VALUE,
+					&ctl->id);
 	}
 }
 
@@ -1443,11 +999,11 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol,
 		}
 	}
 	spec->smart51_enabled = *ucontrol->value.integer.value;
-	set_jack_power_state(codec);
+	set_widgets_power_state(codec);
 	return 1;
 }
 
-static struct snd_kcontrol_new via_smart51_mixer[2] = {
+static const struct snd_kcontrol_new via_smart51_mixer[2] = {
 	{
 	 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	 .name = "Smart 5.1",
@@ -1469,6 +1025,11 @@ static int via_smart51_build(struct via_spec *spec)
 	hda_nid_t nid;
 	int i;
 
+	if (!cfg)
+		return 0;
+	if (cfg->line_outs > 2)
+		return 0;
+
 	knew = via_clone_control(spec, &via_smart51_mixer[0]);
 	if (knew == NULL)
 		return -ENOMEM;
@@ -1488,7 +1049,7 @@ static int via_smart51_build(struct via_spec *spec)
 }
 
 /* capture mixer elements */
-static struct snd_kcontrol_new vt1708_capture_mixer[] = {
+static const struct snd_kcontrol_new vt1708_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x27, 0x0, HDA_INPUT),
@@ -1539,6 +1100,7 @@ static int is_aa_path_mute(struct hda_codec *codec)
 		break;
 	case VT2002P:
 	case VT1812:
+	case VT1802:
 		nid_mixer = 0x21;
 		start_idx = 0;
 		end_idx = 2;
@@ -1603,6 +1165,7 @@ static void analog_low_current_mode(struct hda_codec *codec, int stream_idle)
 		break;
 	case VT2002P:
 	case VT1812:
+	case VT1802:
 		verb = 0xf93;
 		parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */
 		break;
@@ -1616,7 +1179,7 @@ static void analog_low_current_mode(struct hda_codec *codec, int stream_idle)
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb vt1708_volume_init_verbs[] = {
+static const struct hda_verb vt1708_volume_init_verbs[] = {
 	/*
 	 * Unmute ADC0-1 and set the default input to mic-in
 	 */
@@ -1646,6 +1209,8 @@ static struct hda_verb vt1708_volume_init_verbs[] = {
 	{0x20, AC_VERB_SET_CONNECT_SEL, 0},
 	/* PW9 Output enable */
 	{0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	/* power down jack detect function */
+	{0x1, 0xf81, 0x1},
 	{ }
 };
 
@@ -1668,7 +1233,7 @@ static void playback_multi_pcm_prep_0(struct hda_codec *codec,
 {
 	struct via_spec *spec = codec->spec;
 	struct hda_multi_out *mout = &spec->multiout;
-	hda_nid_t *nids = mout->dac_nids;
+	const hda_nid_t *nids = mout->dac_nids;
 	int chs = substream->runtime->channels;
 	int i;
 
@@ -1737,7 +1302,7 @@ static int via_playback_multi_pcm_prepare(struct hda_pcm_stream *hinfo,
 {
 	struct via_spec *spec = codec->spec;
 	struct hda_multi_out *mout = &spec->multiout;
-	hda_nid_t *nids = mout->dac_nids;
+	const hda_nid_t *nids = mout->dac_nids;
 
 	if (substream->number == 0)
 		playback_multi_pcm_prep_0(codec, stream_tag, format,
@@ -1758,7 +1323,7 @@ static int via_playback_multi_pcm_cleanup(struct hda_pcm_stream *hinfo,
 {
 	struct via_spec *spec = codec->spec;
 	struct hda_multi_out *mout = &spec->multiout;
-	hda_nid_t *nids = mout->dac_nids;
+	const hda_nid_t *nids = mout->dac_nids;
 	int i;
 
 	if (substream->number == 0) {
@@ -1856,7 +1421,7 @@ static int via_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 	return 0;
 }
 
-static struct hda_pcm_stream vt1708_pcm_analog_playback = {
+static const struct hda_pcm_stream vt1708_pcm_analog_playback = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 8,
@@ -1868,7 +1433,7 @@ static struct hda_pcm_stream vt1708_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = {
+static const struct hda_pcm_stream vt1708_pcm_analog_s16_playback = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 8,
@@ -1885,7 +1450,7 @@ static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt1708_pcm_analog_capture = {
+static const struct hda_pcm_stream vt1708_pcm_analog_capture = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -1896,7 +1461,7 @@ static struct hda_pcm_stream vt1708_pcm_analog_capture = {
 	},
 };
 
-static struct hda_pcm_stream vt1708_pcm_digital_playback = {
+static const struct hda_pcm_stream vt1708_pcm_digital_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -1909,7 +1474,7 @@ static struct hda_pcm_stream vt1708_pcm_digital_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt1708_pcm_digital_capture = {
+static const struct hda_pcm_stream vt1708_pcm_digital_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -1919,7 +1484,7 @@ static int via_build_controls(struct hda_codec *codec)
 {
 	struct via_spec *spec = codec->spec;
 	struct snd_kcontrol *kctl;
-	struct snd_kcontrol_new *knew;
+	const struct snd_kcontrol_new *knew;
 	int err, i;
 
 	for (i = 0; i < spec->num_mixers; i++) {
@@ -1967,7 +1532,7 @@ static int via_build_controls(struct hda_codec *codec)
 	}
 
 	/* init power states */
-	set_jack_power_state(codec);
+	set_widgets_power_state(codec);
 	analog_low_current_mode(codec, 1);
 
 	via_free_kctls(codec); /* no longer needed */
@@ -2131,7 +1696,7 @@ static void via_speaker_automute(struct hda_codec *codec)
 	unsigned int hp_present;
 	struct via_spec *spec = codec->spec;
 
-	if (spec->codec_type != VT2002P && spec->codec_type != VT1812)
+	if (!VT2002P_COMPATIBLE(spec))
 		return;
 
 	hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]);
@@ -2190,17 +1755,21 @@ static void via_unsol_event(struct hda_codec *codec,
 				  unsigned int res)
 {
 	res >>= 26;
-	if (res & VIA_HP_EVENT)
+
+	if (res & VIA_JACK_EVENT)
+		set_widgets_power_state(codec);
+
+	res &= ~VIA_JACK_EVENT;
+
+	if (res == VIA_HP_EVENT)
 		via_hp_automute(codec);
-	if (res & VIA_GPIO_EVENT)
+	else if (res == VIA_GPIO_EVENT)
 		via_gpio_control(codec);
-	if (res & VIA_JACK_EVENT)
-		set_jack_power_state(codec);
-	if (res & VIA_MONO_EVENT)
+	else if (res == VIA_MONO_EVENT)
 		via_mono_automute(codec);
-	if (res & VIA_SPEAKER_EVENT)
+	else if (res == VIA_SPEAKER_EVENT)
 		via_speaker_automute(codec);
-	if (res & VIA_BIND_HP_EVENT)
+	else if (res == VIA_BIND_HP_EVENT)
 		via_hp_bind_automute(codec);
 }
 
@@ -2250,7 +1819,7 @@ static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid)
 
 /*
  */
-static struct hda_codec_ops via_patch_ops = {
+static const struct hda_codec_ops via_patch_ops = {
 	.build_controls = via_build_controls,
 	.build_pcms = via_build_pcms,
 	.init = via_init,
@@ -2280,16 +1849,16 @@ static int vt1708_auto_fill_dac_nids(struct via_spec *spec,
 			/* config dac list */
 			switch (i) {
 			case AUTO_SEQ_FRONT:
-				spec->multiout.dac_nids[i] = 0x10;
+				spec->private_dac_nids[i] = 0x10;
 				break;
 			case AUTO_SEQ_CENLFE:
-				spec->multiout.dac_nids[i] = 0x12;
+				spec->private_dac_nids[i] = 0x12;
 				break;
 			case AUTO_SEQ_SURROUND:
-				spec->multiout.dac_nids[i] = 0x11;
+				spec->private_dac_nids[i] = 0x11;
 				break;
 			case AUTO_SEQ_SIDE:
-				spec->multiout.dac_nids[i] = 0x13;
+				spec->private_dac_nids[i] = 0x13;
 				break;
 			}
 		}
@@ -2433,7 +2002,8 @@ static int vt1708_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
 static int vt_auto_create_analog_input_ctls(struct hda_codec *codec,
 					    const struct auto_pin_cfg *cfg,
 					    hda_nid_t cap_nid,
-					    hda_nid_t pin_idxs[], int num_idxs)
+					    const hda_nid_t pin_idxs[],
+					    int num_idxs)
 {
 	struct via_spec *spec = codec->spec;
 	struct hda_input_mux *imux = &spec->private_imux[0];
@@ -2479,13 +2049,13 @@ static int vt_auto_create_analog_input_ctls(struct hda_codec *codec,
 static int vt1708_auto_create_analog_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	static hda_nid_t pin_idxs[] = { 0xff, 0x24, 0x1d, 0x1e, 0x21 };
+	static const hda_nid_t pin_idxs[] = { 0xff, 0x24, 0x1d, 0x1e, 0x21 };
 	return vt_auto_create_analog_input_ctls(codec, cfg, 0x17, pin_idxs,
 						ARRAY_SIZE(pin_idxs));
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list vt1708_loopbacks[] = {
+static const struct hda_amp_list vt1708_loopbacks[] = {
 	{ 0x17, HDA_INPUT, 1 },
 	{ 0x17, HDA_INPUT, 2 },
 	{ 0x17, HDA_INPUT, 3 },
@@ -2544,7 +2114,7 @@ static int vt1708_jack_detectect_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new vt1708_jack_detectect[] = {
+static const struct snd_kcontrol_new vt1708_jack_detectect[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Jack Detect",
@@ -2619,7 +2189,8 @@ static int via_auto_init(struct hda_codec *codec)
 	via_auto_init_multi_out(codec);
 	via_auto_init_hp_out(codec);
 	via_auto_init_analog_input(codec);
-	if (spec->codec_type == VT2002P || spec->codec_type == VT1812) {
+
+	if (VT2002P_COMPATIBLE(spec)) {
 		via_hp_bind_automute(codec);
 	} else {
 		via_hp_automute(codec);
@@ -2723,7 +2294,7 @@ static int patch_vt1708(struct hda_codec *codec)
 }
 
 /* capture mixer elements */
-static struct snd_kcontrol_new vt1709_capture_mixer[] = {
+static const struct snd_kcontrol_new vt1709_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x14, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x15, 0x0, HDA_INPUT),
@@ -2745,7 +2316,7 @@ static struct snd_kcontrol_new vt1709_capture_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb vt1709_uniwill_init_verbs[] = {
+static const struct hda_verb vt1709_uniwill_init_verbs[] = {
 	{0x20, AC_VERB_SET_UNSOLICITED_ENABLE,
 	 AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
 	{ }
@@ -2754,7 +2325,7 @@ static struct hda_verb vt1709_uniwill_init_verbs[] = {
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb vt1709_10ch_volume_init_verbs[] = {
+static const struct hda_verb vt1709_10ch_volume_init_verbs[] = {
 	/*
 	 * Unmute ADC0-2 and set the default input to mic-in
 	 */
@@ -2794,7 +2365,7 @@ static struct hda_verb vt1709_10ch_volume_init_verbs[] = {
 	{ }
 };
 
-static struct hda_pcm_stream vt1709_10ch_pcm_analog_playback = {
+static const struct hda_pcm_stream vt1709_10ch_pcm_analog_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 10,
@@ -2806,7 +2377,7 @@ static struct hda_pcm_stream vt1709_10ch_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt1709_6ch_pcm_analog_playback = {
+static const struct hda_pcm_stream vt1709_6ch_pcm_analog_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 6,
@@ -2818,7 +2389,7 @@ static struct hda_pcm_stream vt1709_6ch_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt1709_pcm_analog_capture = {
+static const struct hda_pcm_stream vt1709_pcm_analog_capture = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -2829,7 +2400,7 @@ static struct hda_pcm_stream vt1709_pcm_analog_capture = {
 	},
 };
 
-static struct hda_pcm_stream vt1709_pcm_digital_playback = {
+static const struct hda_pcm_stream vt1709_pcm_digital_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -2840,7 +2411,7 @@ static struct hda_pcm_stream vt1709_pcm_digital_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt1709_pcm_digital_capture = {
+static const struct hda_pcm_stream vt1709_pcm_digital_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -2867,26 +2438,26 @@ static int vt1709_auto_fill_dac_nids(struct via_spec *spec,
 				switch (i) {
 				case AUTO_SEQ_FRONT:
 					/* AOW0 */
-					spec->multiout.dac_nids[i] = 0x10;
+					spec->private_dac_nids[i] = 0x10;
 					break;
 				case AUTO_SEQ_CENLFE:
 					/* AOW2 */
-					spec->multiout.dac_nids[i] = 0x12;
+					spec->private_dac_nids[i] = 0x12;
 					break;
 				case AUTO_SEQ_SURROUND:
 					/* AOW3 */
-					spec->multiout.dac_nids[i] = 0x11;
+					spec->private_dac_nids[i] = 0x11;
 					break;
 				case AUTO_SEQ_SIDE:
 					/* AOW1 */
-					spec->multiout.dac_nids[i] = 0x27;
+					spec->private_dac_nids[i] = 0x27;
 					break;
 				default:
 					break;
 				}
 			}
 		}
-		spec->multiout.dac_nids[cfg->line_outs] = 0x28; /* AOW4 */
+		spec->private_dac_nids[cfg->line_outs] = 0x28; /* AOW4 */
 
 	} else if (cfg->line_outs == 3) { /* 6 channels */
 		for (i = 0; i < cfg->line_outs; i++) {
@@ -2896,15 +2467,15 @@ static int vt1709_auto_fill_dac_nids(struct via_spec *spec,
 				switch (i) {
 				case AUTO_SEQ_FRONT:
 					/* AOW0 */
-					spec->multiout.dac_nids[i] = 0x10;
+					spec->private_dac_nids[i] = 0x10;
 					break;
 				case AUTO_SEQ_CENLFE:
 					/* AOW2 */
-					spec->multiout.dac_nids[i] = 0x12;
+					spec->private_dac_nids[i] = 0x12;
 					break;
 				case AUTO_SEQ_SURROUND:
 					/* AOW1 */
-					spec->multiout.dac_nids[i] = 0x11;
+					spec->private_dac_nids[i] = 0x11;
 					break;
 				default:
 					break;
@@ -3052,7 +2623,7 @@ static int vt1709_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
 static int vt1709_auto_create_analog_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	static hda_nid_t pin_idxs[] = { 0xff, 0x23, 0x1d, 0x1e, 0x21 };
+	static const hda_nid_t pin_idxs[] = { 0xff, 0x23, 0x1d, 0x1e, 0x21 };
 	return vt_auto_create_analog_input_ctls(codec, cfg, 0x18, pin_idxs,
 						ARRAY_SIZE(pin_idxs));
 }
@@ -3102,7 +2673,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec)
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list vt1709_loopbacks[] = {
+static const struct hda_amp_list vt1709_loopbacks[] = {
 	{ 0x18, HDA_INPUT, 1 },
 	{ 0x18, HDA_INPUT, 2 },
 	{ 0x18, HDA_INPUT, 3 },
@@ -3163,7 +2734,7 @@ static int patch_vt1709_10ch(struct hda_codec *codec)
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb vt1709_6ch_volume_init_verbs[] = {
+static const struct hda_verb vt1709_6ch_volume_init_verbs[] = {
 	/*
 	 * Unmute ADC0-2 and set the default input to mic-in
 	 */
@@ -3253,7 +2824,7 @@ static int patch_vt1709_6ch(struct hda_codec *codec)
 }
 
 /* capture mixer elements */
-static struct snd_kcontrol_new vt1708B_capture_mixer[] = {
+static const struct snd_kcontrol_new vt1708B_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT),
@@ -3275,7 +2846,7 @@ static struct snd_kcontrol_new vt1708B_capture_mixer[] = {
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb vt1708B_8ch_volume_init_verbs[] = {
+static const struct hda_verb vt1708B_8ch_volume_init_verbs[] = {
 	/*
 	 * Unmute ADC0-1 and set the default input to mic-in
 	 */
@@ -3310,7 +2881,7 @@ static struct hda_verb vt1708B_8ch_volume_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb vt1708B_4ch_volume_init_verbs[] = {
+static const struct hda_verb vt1708B_4ch_volume_init_verbs[] = {
 	/*
 	 * Unmute ADC0-1 and set the default input to mic-in
 	 */
@@ -3345,7 +2916,7 @@ static struct hda_verb vt1708B_4ch_volume_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb vt1708B_uniwill_init_verbs[] = {
+static const struct hda_verb vt1708B_uniwill_init_verbs[] = {
 	{0x1d, AC_VERB_SET_UNSOLICITED_ENABLE,
 	 AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
 	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
@@ -3369,7 +2940,7 @@ static int via_pcm_open_close(struct hda_pcm_stream *hinfo,
 	return 0;
 }
 
-static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = {
+static const struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 8,
@@ -3382,7 +2953,7 @@ static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt1708B_4ch_pcm_analog_playback = {
+static const struct hda_pcm_stream vt1708B_4ch_pcm_analog_playback = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 4,
@@ -3394,7 +2965,7 @@ static struct hda_pcm_stream vt1708B_4ch_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt1708B_pcm_analog_capture = {
+static const struct hda_pcm_stream vt1708B_pcm_analog_capture = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -3407,7 +2978,7 @@ static struct hda_pcm_stream vt1708B_pcm_analog_capture = {
 	},
 };
 
-static struct hda_pcm_stream vt1708B_pcm_digital_playback = {
+static const struct hda_pcm_stream vt1708B_pcm_digital_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -3420,7 +2991,7 @@ static struct hda_pcm_stream vt1708B_pcm_digital_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt1708B_pcm_digital_capture = {
+static const struct hda_pcm_stream vt1708B_pcm_digital_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -3443,16 +3014,16 @@ static int vt1708B_auto_fill_dac_nids(struct via_spec *spec,
 			/* config dac list */
 			switch (i) {
 			case AUTO_SEQ_FRONT:
-				spec->multiout.dac_nids[i] = 0x10;
+				spec->private_dac_nids[i] = 0x10;
 				break;
 			case AUTO_SEQ_CENLFE:
-				spec->multiout.dac_nids[i] = 0x24;
+				spec->private_dac_nids[i] = 0x24;
 				break;
 			case AUTO_SEQ_SURROUND:
-				spec->multiout.dac_nids[i] = 0x11;
+				spec->private_dac_nids[i] = 0x11;
 				break;
 			case AUTO_SEQ_SIDE:
-				spec->multiout.dac_nids[i] = 0x25;
+				spec->private_dac_nids[i] = 0x25;
 				break;
 			}
 		}
@@ -3584,7 +3155,7 @@ static int vt1708B_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
 static int vt1708B_auto_create_analog_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	static hda_nid_t pin_idxs[] = { 0xff, 0x1f, 0x1a, 0x1b, 0x1e };
+	static const hda_nid_t pin_idxs[] = { 0xff, 0x1f, 0x1a, 0x1b, 0x1e };
 	return vt_auto_create_analog_input_ctls(codec, cfg, 0x16, pin_idxs,
 						ARRAY_SIZE(pin_idxs));
 }
@@ -3634,7 +3205,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec)
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list vt1708B_loopbacks[] = {
+static const struct hda_amp_list vt1708B_loopbacks[] = {
 	{ 0x16, HDA_INPUT, 1 },
 	{ 0x16, HDA_INPUT, 2 },
 	{ 0x16, HDA_INPUT, 3 },
@@ -3642,6 +3213,87 @@ static struct hda_amp_list vt1708B_loopbacks[] = {
 	{ } /* end */
 };
 #endif
+
+static void set_widgets_power_state_vt1708B(struct hda_codec *codec)
+{
+	struct via_spec *spec = codec->spec;
+	int imux_is_smixer;
+	unsigned int parm;
+	int is_8ch = 0;
+	if ((spec->codec_type != VT1708B_4CH) &&
+	    (codec->vendor_id != 0x11064397))
+		is_8ch = 1;
+
+	/* SW0 (17h) = stereo mixer */
+	imux_is_smixer =
+	(snd_hda_codec_read(codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00)
+	 == ((spec->codec_type == VT1708S) ? 5 : 0));
+	/* inputs */
+	/* PW 1/2/5 (1ah/1bh/1eh) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x1a, &parm);
+	set_pin_power_state(codec, 0x1b, &parm);
+	set_pin_power_state(codec, 0x1e, &parm);
+	if (imux_is_smixer)
+		parm = AC_PWRST_D0;
+	/* SW0 (17h), AIW 0/1 (13h/14h) */
+	snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	/* outputs */
+	/* PW0 (19h), SW1 (18h), AOW1 (11h) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x19, &parm);
+	if (spec->smart51_enabled)
+		set_pin_power_state(codec, 0x1b, &parm);
+	snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	/* PW6 (22h), SW2 (26h), AOW2 (24h) */
+	if (is_8ch) {
+		parm = AC_PWRST_D3;
+		set_pin_power_state(codec, 0x22, &parm);
+		if (spec->smart51_enabled)
+			set_pin_power_state(codec, 0x1a, &parm);
+		snd_hda_codec_write(codec, 0x26, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+		snd_hda_codec_write(codec, 0x24, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+	} else if (codec->vendor_id == 0x11064397) {
+		/* PW7(23h), SW2(27h), AOW2(25h) */
+		parm = AC_PWRST_D3;
+		set_pin_power_state(codec, 0x23, &parm);
+		if (spec->smart51_enabled)
+			set_pin_power_state(codec, 0x1a, &parm);
+		snd_hda_codec_write(codec, 0x27, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+		snd_hda_codec_write(codec, 0x25, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+	}
+
+	/* PW 3/4/7 (1ch/1dh/23h) */
+	parm = AC_PWRST_D3;
+	/* force to D0 for internal Speaker */
+	set_pin_power_state(codec, 0x1c, &parm);
+	set_pin_power_state(codec, 0x1d, &parm);
+	if (is_8ch)
+		set_pin_power_state(codec, 0x23, &parm);
+
+	/* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */
+	snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE,
+			    imux_is_smixer ? AC_PWRST_D0 : parm);
+	snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
+	if (is_8ch) {
+		snd_hda_codec_write(codec, 0x25, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+		snd_hda_codec_write(codec, 0x27, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+	} else if (codec->vendor_id == 0x11064397 && spec->hp_independent_mode)
+		snd_hda_codec_write(codec, 0x25, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+}
+
 static int patch_vt1708S(struct hda_codec *codec);
 static int patch_vt1708B_8ch(struct hda_codec *codec)
 {
@@ -3692,6 +3344,8 @@ static int patch_vt1708B_8ch(struct hda_codec *codec)
 	spec->loopback.amplist = vt1708B_loopbacks;
 #endif
 
+	spec->set_widgets_power_state =  set_widgets_power_state_vt1708B;
+
 	return 0;
 }
 
@@ -3742,13 +3396,15 @@ static int patch_vt1708B_4ch(struct hda_codec *codec)
 	spec->loopback.amplist = vt1708B_loopbacks;
 #endif
 
+	spec->set_widgets_power_state =  set_widgets_power_state_vt1708B;
+
 	return 0;
 }
 
 /* Patch for VT1708S */
 
 /* capture mixer elements */
-static struct snd_kcontrol_new vt1708S_capture_mixer[] = {
+static const struct snd_kcontrol_new vt1708S_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT),
@@ -3771,7 +3427,7 @@ static struct snd_kcontrol_new vt1708S_capture_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb vt1708S_volume_init_verbs[] = {
+static const struct hda_verb vt1708S_volume_init_verbs[] = {
 	/* Unmute ADC0-1 and set the default input to mic-in */
 	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -3797,7 +3453,7 @@ static struct hda_verb vt1708S_volume_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb vt1708S_uniwill_init_verbs[] = {
+static const struct hda_verb vt1708S_uniwill_init_verbs[] = {
 	{0x1d, AC_VERB_SET_UNSOLICITED_ENABLE,
 	 AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
 	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
@@ -3810,7 +3466,19 @@ static struct hda_verb vt1708S_uniwill_init_verbs[] = {
 	{ }
 };
 
-static struct hda_pcm_stream vt1708S_pcm_analog_playback = {
+static const struct hda_verb vt1705_uniwill_init_verbs[] = {
+	{0x1d, AC_VERB_SET_UNSOLICITED_ENABLE,
+	 AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
+	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+	{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+	{0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+	{0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+	{0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+	{ }
+};
+
+static const struct hda_pcm_stream vt1708S_pcm_analog_playback = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 8,
@@ -3823,7 +3491,20 @@ static struct hda_pcm_stream vt1708S_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt1708S_pcm_analog_capture = {
+static const struct hda_pcm_stream vt1705_pcm_analog_playback = {
+	.substreams = 2,
+	.channels_min = 2,
+	.channels_max = 6,
+	.nid = 0x10, /* NID to query formats and rates */
+	.ops = {
+		.open = via_playback_pcm_open,
+		.prepare = via_playback_multi_pcm_prepare,
+		.cleanup = via_playback_multi_pcm_cleanup,
+		.close = via_pcm_open_close
+	},
+};
+
+static const struct hda_pcm_stream vt1708S_pcm_analog_capture = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -3836,7 +3517,7 @@ static struct hda_pcm_stream vt1708S_pcm_analog_capture = {
 	},
 };
 
-static struct hda_pcm_stream vt1708S_pcm_digital_playback = {
+static const struct hda_pcm_stream vt1708S_pcm_digital_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -3866,16 +3547,19 @@ static int vt1708S_auto_fill_dac_nids(struct via_spec *spec,
 			/* config dac list */
 			switch (i) {
 			case AUTO_SEQ_FRONT:
-				spec->multiout.dac_nids[i] = 0x10;
+				spec->private_dac_nids[i] = 0x10;
 				break;
 			case AUTO_SEQ_CENLFE:
-				spec->multiout.dac_nids[i] = 0x24;
+				if (spec->codec->vendor_id == 0x11064397)
+					spec->private_dac_nids[i] = 0x25;
+				else
+					spec->private_dac_nids[i] = 0x24;
 				break;
 			case AUTO_SEQ_SURROUND:
-				spec->multiout.dac_nids[i] = 0x11;
+				spec->private_dac_nids[i] = 0x11;
 				break;
 			case AUTO_SEQ_SIDE:
-				spec->multiout.dac_nids[i] = 0x25;
+				spec->private_dac_nids[i] = 0x25;
 				break;
 			}
 		}
@@ -3884,23 +3568,29 @@ static int vt1708S_auto_fill_dac_nids(struct via_spec *spec,
 	/* for Smart 5.1, line/mic inputs double as output pins */
 	if (cfg->line_outs == 1) {
 		spec->multiout.num_dacs = 3;
-		spec->multiout.dac_nids[AUTO_SEQ_SURROUND] = 0x11;
-		spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x24;
+		spec->private_dac_nids[AUTO_SEQ_SURROUND] = 0x11;
+		if (spec->codec->vendor_id == 0x11064397)
+			spec->private_dac_nids[AUTO_SEQ_CENLFE] = 0x25;
+		else
+			spec->private_dac_nids[AUTO_SEQ_CENLFE] = 0x24;
 	}
 
 	return 0;
 }
 
 /* add playback controls from the parsed DAC table */
-static int vt1708S_auto_create_multi_out_ctls(struct via_spec *spec,
+static int vt1708S_auto_create_multi_out_ctls(struct hda_codec *codec,
 					     const struct auto_pin_cfg *cfg)
 {
+	struct via_spec *spec = codec->spec;
 	char name[32];
 	static const char * const chname[4] = {
 		"Front", "Surround", "C/LFE", "Side"
 	};
-	hda_nid_t nid_vols[] = {0x10, 0x11, 0x24, 0x25};
-	hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x26, 0x27};
+	hda_nid_t nid_vols[2][4] = { {0x10, 0x11, 0x24, 0x25},
+				     {0x10, 0x11, 0x25, 0} };
+	hda_nid_t nid_mutes[2][4] = { {0x1C, 0x18, 0x26, 0x27},
+				      {0x1C, 0x18, 0x27, 0} };
 	hda_nid_t nid, nid_vol, nid_mute;
 	int i, err;
 
@@ -3911,8 +3601,15 @@ static int vt1708S_auto_create_multi_out_ctls(struct via_spec *spec,
 		if (!nid && i > AUTO_SEQ_CENLFE)
 			continue;
 
-		nid_vol = nid_vols[i];
-		nid_mute = nid_mutes[i];
+		if (codec->vendor_id == 0x11064397) {
+			nid_vol = nid_vols[1][i];
+			nid_mute = nid_mutes[1][i];
+		} else {
+			nid_vol = nid_vols[0][i];
+			nid_mute = nid_mutes[0][i];
+		}
+		if (!nid_vol && !nid_mute)
+			continue;
 
 		if (i == AUTO_SEQ_CENLFE) {
 			/* Center/LFE */
@@ -4022,7 +3719,7 @@ static int vt1708S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
 static int vt1708S_auto_create_analog_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	static hda_nid_t pin_idxs[] = { 0x1f, 0x1a, 0x1b, 0x1e, 0, 0xff };
+	static const hda_nid_t pin_idxs[] = { 0x1f, 0x1a, 0x1b, 0x1e, 0, 0xff };
 	return vt_auto_create_analog_input_ctls(codec, cfg, 0x16, pin_idxs,
 						ARRAY_SIZE(pin_idxs));
 }
@@ -4066,7 +3763,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec)
 	if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
 		return 0; /* can't find valid BIOS pin config */
 
-	err = vt1708S_auto_create_multi_out_ctls(spec, &spec->autocfg);
+	err = vt1708S_auto_create_multi_out_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
 	err = vt1708S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
@@ -4093,7 +3790,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec)
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list vt1708S_loopbacks[] = {
+static const struct hda_amp_list vt1708S_loopbacks[] = {
 	{ 0x16, HDA_INPUT, 1 },
 	{ 0x16, HDA_INPUT, 2 },
 	{ 0x16, HDA_INPUT, 3 },
@@ -4133,17 +3830,29 @@ static int patch_vt1708S(struct hda_codec *codec)
 	}
 
 	spec->init_verbs[spec->num_iverbs++] = vt1708S_volume_init_verbs;
-	spec->init_verbs[spec->num_iverbs++] = vt1708S_uniwill_init_verbs;
+	if (codec->vendor_id == 0x11064397)
+		spec->init_verbs[spec->num_iverbs++] =
+			vt1705_uniwill_init_verbs;
+	else
+		spec->init_verbs[spec->num_iverbs++] =
+			vt1708S_uniwill_init_verbs;
 
 	if (codec->vendor_id == 0x11060440)
 		spec->stream_name_analog = "VT1818S Analog";
+	else if (codec->vendor_id == 0x11064397)
+		spec->stream_name_analog = "VT1705 Analog";
 	else
 		spec->stream_name_analog = "VT1708S Analog";
-	spec->stream_analog_playback = &vt1708S_pcm_analog_playback;
+	if (codec->vendor_id == 0x11064397)
+		spec->stream_analog_playback = &vt1705_pcm_analog_playback;
+	else
+		spec->stream_analog_playback = &vt1708S_pcm_analog_playback;
 	spec->stream_analog_capture = &vt1708S_pcm_analog_capture;
 
 	if (codec->vendor_id == 0x11060440)
 		spec->stream_name_digital = "VT1818S Digital";
+	else if (codec->vendor_id == 0x11064397)
+		spec->stream_name_digital = "VT1705 Digital";
 	else
 		spec->stream_name_digital = "VT1708S Digital";
 	spec->stream_digital_playback = &vt1708S_pcm_digital_playback;
@@ -4181,13 +3890,22 @@ static int patch_vt1708S(struct hda_codec *codec)
 		spec->stream_name_analog = "VT1818S Analog";
 		spec->stream_name_digital = "VT1818S Digital";
 	}
+	/* correct names for VT1705 */
+	if (codec->vendor_id == 0x11064397)	{
+		kfree(codec->chip_name);
+		codec->chip_name = kstrdup("VT1705", GFP_KERNEL);
+		snprintf(codec->bus->card->mixername,
+			 sizeof(codec->bus->card->mixername),
+			 "%s %s", codec->vendor_name, codec->chip_name);
+	}
+	spec->set_widgets_power_state =  set_widgets_power_state_vt1708B;
 	return 0;
 }
 
 /* Patch for VT1702 */
 
 /* capture mixer elements */
-static struct snd_kcontrol_new vt1702_capture_mixer[] = {
+static const struct snd_kcontrol_new vt1702_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x20, 0x0, HDA_INPUT),
@@ -4211,7 +3929,7 @@ static struct snd_kcontrol_new vt1702_capture_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb vt1702_volume_init_verbs[] = {
+static const struct hda_verb vt1702_volume_init_verbs[] = {
 	/*
 	 * Unmute ADC0-1 and set the default input to mic-in
 	 */
@@ -4242,7 +3960,7 @@ static struct hda_verb vt1702_volume_init_verbs[] = {
 	{ }
 };
 
-static struct hda_verb vt1702_uniwill_init_verbs[] = {
+static const struct hda_verb vt1702_uniwill_init_verbs[] = {
 	{0x17, AC_VERB_SET_UNSOLICITED_ENABLE,
 	 AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
 	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
@@ -4252,7 +3970,7 @@ static struct hda_verb vt1702_uniwill_init_verbs[] = {
 	{ }
 };
 
-static struct hda_pcm_stream vt1702_pcm_analog_playback = {
+static const struct hda_pcm_stream vt1702_pcm_analog_playback = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -4265,7 +3983,7 @@ static struct hda_pcm_stream vt1702_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt1702_pcm_analog_capture = {
+static const struct hda_pcm_stream vt1702_pcm_analog_capture = {
 	.substreams = 3,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -4278,7 +3996,7 @@ static struct hda_pcm_stream vt1702_pcm_analog_capture = {
 	},
 };
 
-static struct hda_pcm_stream vt1702_pcm_digital_playback = {
+static const struct hda_pcm_stream vt1702_pcm_digital_playback = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -4300,7 +4018,7 @@ static int vt1702_auto_fill_dac_nids(struct via_spec *spec,
 
 	if (cfg->line_out_pins[0]) {
 		/* config dac list */
-		spec->multiout.dac_nids[0] = 0x10;
+		spec->private_dac_nids[0] = 0x10;
 	}
 
 	return 0;
@@ -4378,7 +4096,7 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
 static int vt1702_auto_create_analog_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	static hda_nid_t pin_idxs[] = { 0x14, 0x15, 0x18, 0xff };
+	static const hda_nid_t pin_idxs[] = { 0x14, 0x15, 0x18, 0xff };
 	return vt_auto_create_analog_input_ctls(codec, cfg, 0x1a, pin_idxs,
 						ARRAY_SIZE(pin_idxs));
 }
@@ -4429,7 +4147,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec)
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list vt1702_loopbacks[] = {
+static const struct hda_amp_list vt1702_loopbacks[] = {
 	{ 0x1A, HDA_INPUT, 1 },
 	{ 0x1A, HDA_INPUT, 2 },
 	{ 0x1A, HDA_INPUT, 3 },
@@ -4438,6 +4156,37 @@ static struct hda_amp_list vt1702_loopbacks[] = {
 };
 #endif
 
+static void set_widgets_power_state_vt1702(struct hda_codec *codec)
+{
+	int imux_is_smixer =
+	snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3;
+	unsigned int parm;
+	/* inputs */
+	/* PW 1/2/5 (14h/15h/18h) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x14, &parm);
+	set_pin_power_state(codec, 0x15, &parm);
+	set_pin_power_state(codec, 0x18, &parm);
+	if (imux_is_smixer)
+		parm = AC_PWRST_D0; /* SW0 (13h) = stereo mixer (idx 3) */
+	/* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */
+	snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	/* outputs */
+	/* PW 3/4 (16h/17h) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x17, &parm);
+	set_pin_power_state(codec, 0x16, &parm);
+	/* MW0 (1ah), AOW 0/1 (10h/1dh) */
+	snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE,
+			    imux_is_smixer ? AC_PWRST_D0 : parm);
+	snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm);
+}
+
 static int patch_vt1702(struct hda_codec *codec)
 {
 	struct via_spec *spec;
@@ -4484,13 +4233,14 @@ static int patch_vt1702(struct hda_codec *codec)
 	spec->loopback.amplist = vt1702_loopbacks;
 #endif
 
+	spec->set_widgets_power_state =  set_widgets_power_state_vt1702;
 	return 0;
 }
 
 /* Patch for VT1718S */
 
 /* capture mixer elements */
-static struct snd_kcontrol_new vt1718S_capture_mixer[] = {
+static const struct snd_kcontrol_new vt1718S_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT),
@@ -4512,14 +4262,15 @@ static struct snd_kcontrol_new vt1718S_capture_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb vt1718S_volume_init_verbs[] = {
+static const struct hda_verb vt1718S_volume_init_verbs[] = {
 	/*
 	 * Unmute ADC0-1 and set the default input to mic-in
 	 */
 	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 
-
+	/* Enable MW0 adjust Gain 5 */
+	{0x1, 0xfb2, 0x10},
 	/* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
 	 * mixer widget
 	 */
@@ -4528,7 +4279,7 @@ static struct hda_verb vt1718S_volume_init_verbs[] = {
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
 
 	/* Setup default input of Front HP to MW9 */
 	{0x28, AC_VERB_SET_CONNECT_SEL, 0x1},
@@ -4559,7 +4310,7 @@ static struct hda_verb vt1718S_volume_init_verbs[] = {
 };
 
 
-static struct hda_verb vt1718S_uniwill_init_verbs[] = {
+static const struct hda_verb vt1718S_uniwill_init_verbs[] = {
 	{0x28, AC_VERB_SET_UNSOLICITED_ENABLE,
 	 AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
 	{0x24, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
@@ -4572,7 +4323,7 @@ static struct hda_verb vt1718S_uniwill_init_verbs[] = {
 	{ }
 };
 
-static struct hda_pcm_stream vt1718S_pcm_analog_playback = {
+static const struct hda_pcm_stream vt1718S_pcm_analog_playback = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 10,
@@ -4585,7 +4336,7 @@ static struct hda_pcm_stream vt1718S_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt1718S_pcm_analog_capture = {
+static const struct hda_pcm_stream vt1718S_pcm_analog_capture = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -4598,7 +4349,7 @@ static struct hda_pcm_stream vt1718S_pcm_analog_capture = {
 	},
 };
 
-static struct hda_pcm_stream vt1718S_pcm_digital_playback = {
+static const struct hda_pcm_stream vt1718S_pcm_digital_playback = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -4611,7 +4362,7 @@ static struct hda_pcm_stream vt1718S_pcm_digital_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt1718S_pcm_digital_capture = {
+static const struct hda_pcm_stream vt1718S_pcm_digital_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -4634,16 +4385,16 @@ static int vt1718S_auto_fill_dac_nids(struct via_spec *spec,
 			/* config dac list */
 			switch (i) {
 			case AUTO_SEQ_FRONT:
-				spec->multiout.dac_nids[i] = 0x8;
+				spec->private_dac_nids[i] = 0x8;
 				break;
 			case AUTO_SEQ_CENLFE:
-				spec->multiout.dac_nids[i] = 0xa;
+				spec->private_dac_nids[i] = 0xa;
 				break;
 			case AUTO_SEQ_SURROUND:
-				spec->multiout.dac_nids[i] = 0x9;
+				spec->private_dac_nids[i] = 0x9;
 				break;
 			case AUTO_SEQ_SIDE:
-				spec->multiout.dac_nids[i] = 0xb;
+				spec->private_dac_nids[i] = 0xb;
 				break;
 			}
 		}
@@ -4765,7 +4516,7 @@ static int vt1718S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
 static int vt1718S_auto_create_analog_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	static hda_nid_t pin_idxs[] = { 0x2c, 0x2b, 0x2a, 0x29, 0, 0xff };
+	static const hda_nid_t pin_idxs[] = { 0x2c, 0x2b, 0x2a, 0x29, 0, 0xff };
 	return vt_auto_create_analog_input_ctls(codec, cfg, 0x21, pin_idxs,
 						ARRAY_SIZE(pin_idxs));
 }
@@ -4816,7 +4567,7 @@ static int vt1718S_parse_auto_config(struct hda_codec *codec)
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list vt1718S_loopbacks[] = {
+static const struct hda_amp_list vt1718S_loopbacks[] = {
 	{ 0x21, HDA_INPUT, 1 },
 	{ 0x21, HDA_INPUT, 2 },
 	{ 0x21, HDA_INPUT, 3 },
@@ -4825,6 +4576,72 @@ static struct hda_amp_list vt1718S_loopbacks[] = {
 };
 #endif
 
+static void set_widgets_power_state_vt1718S(struct hda_codec *codec)
+{
+	struct via_spec *spec = codec->spec;
+	int imux_is_smixer;
+	unsigned int parm;
+	/* MUX6 (1eh) = stereo mixer */
+	imux_is_smixer =
+	snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5;
+	/* inputs */
+	/* PW 5/6/7 (29h/2ah/2bh) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x29, &parm);
+	set_pin_power_state(codec, 0x2a, &parm);
+	set_pin_power_state(codec, 0x2b, &parm);
+	if (imux_is_smixer)
+		parm = AC_PWRST_D0;
+	/* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */
+	snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	/* outputs */
+	/* PW3 (27h), MW2 (1ah), AOW3 (bh) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x27, &parm);
+	snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	/* PW2 (26h), AOW2 (ah) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x26, &parm);
+	if (spec->smart51_enabled)
+		set_pin_power_state(codec, 0x2b, &parm);
+	snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	/* PW0 (24h), AOW0 (8h) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x24, &parm);
+	if (!spec->hp_independent_mode) /* check for redirected HP */
+		set_pin_power_state(codec, 0x28, &parm);
+	snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm);
+	/* MW9 (21h), Mw2 (1ah), AOW0 (8h) */
+	snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE,
+			    imux_is_smixer ? AC_PWRST_D0 : parm);
+
+	/* PW1 (25h), AOW1 (9h) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x25, &parm);
+	if (spec->smart51_enabled)
+		set_pin_power_state(codec, 0x2a, &parm);
+	snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	if (spec->hp_independent_mode) {
+		/* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */
+		parm = AC_PWRST_D3;
+		set_pin_power_state(codec, 0x28, &parm);
+		snd_hda_codec_write(codec, 0x1b, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+		snd_hda_codec_write(codec, 0x34, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+		snd_hda_codec_write(codec, 0xc, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+	}
+}
+
 static int patch_vt1718S(struct hda_codec *codec)
 {
 	struct via_spec *spec;
@@ -4886,6 +4703,8 @@ static int patch_vt1718S(struct hda_codec *codec)
 	spec->loopback.amplist = vt1718S_loopbacks;
 #endif
 
+	spec->set_widgets_power_state =  set_widgets_power_state_vt1718S;
+
 	return 0;
 }
 
@@ -4925,13 +4744,12 @@ static int vt1716s_dmic_put(struct snd_kcontrol *kcontrol,
 	snd_hda_codec_write(codec, 0x26, 0,
 					       AC_VERB_SET_CONNECT_SEL, index);
 	spec->dmic_enabled = index;
-	set_jack_power_state(codec);
-
+	set_widgets_power_state(codec);
 	return 1;
 }
 
 /* capture mixer elements */
-static struct snd_kcontrol_new vt1716S_capture_mixer[] = {
+static const struct snd_kcontrol_new vt1716S_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT),
@@ -4950,7 +4768,7 @@ static struct snd_kcontrol_new vt1716S_capture_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new vt1716s_dmic_mixer[] = {
+static const struct snd_kcontrol_new vt1716s_dmic_mixer[] = {
 	HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x22, 0x0, HDA_INPUT),
 	{
 	 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -4966,12 +4784,12 @@ static struct snd_kcontrol_new vt1716s_dmic_mixer[] = {
 
 
 /* mono-out mixer elements */
-static struct snd_kcontrol_new vt1716S_mono_out_mixer[] = {
+static const struct snd_kcontrol_new vt1716S_mono_out_mixer[] = {
 	HDA_CODEC_MUTE("Mono Playback Switch", 0x2a, 0x0, HDA_OUTPUT),
 	{ } /* end */
 };
 
-static struct hda_verb vt1716S_volume_init_verbs[] = {
+static const struct hda_verb vt1716S_volume_init_verbs[] = {
 	/*
 	 * Unmute ADC0-1 and set the default input to mic-in
 	 */
@@ -5020,7 +4838,7 @@ static struct hda_verb vt1716S_volume_init_verbs[] = {
 };
 
 
-static struct hda_verb vt1716S_uniwill_init_verbs[] = {
+static const struct hda_verb vt1716S_uniwill_init_verbs[] = {
 	{0x1d, AC_VERB_SET_UNSOLICITED_ENABLE,
 	 AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
 	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
@@ -5033,7 +4851,7 @@ static struct hda_verb vt1716S_uniwill_init_verbs[] = {
 	{ }
 };
 
-static struct hda_pcm_stream vt1716S_pcm_analog_playback = {
+static const struct hda_pcm_stream vt1716S_pcm_analog_playback = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 6,
@@ -5046,7 +4864,7 @@ static struct hda_pcm_stream vt1716S_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt1716S_pcm_analog_capture = {
+static const struct hda_pcm_stream vt1716S_pcm_analog_capture = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -5059,7 +4877,7 @@ static struct hda_pcm_stream vt1716S_pcm_analog_capture = {
 	},
 };
 
-static struct hda_pcm_stream vt1716S_pcm_digital_playback = {
+static const struct hda_pcm_stream vt1716S_pcm_digital_playback = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -5088,13 +4906,13 @@ static int vt1716S_auto_fill_dac_nids(struct via_spec *spec,
 			/* config dac list */
 			switch (i) {
 			case AUTO_SEQ_FRONT:
-				spec->multiout.dac_nids[i] = 0x10;
+				spec->private_dac_nids[i] = 0x10;
 				break;
 			case AUTO_SEQ_CENLFE:
-				spec->multiout.dac_nids[i] = 0x25;
+				spec->private_dac_nids[i] = 0x25;
 				break;
 			case AUTO_SEQ_SURROUND:
-				spec->multiout.dac_nids[i] = 0x11;
+				spec->private_dac_nids[i] = 0x11;
 				break;
 			}
 		}
@@ -5229,7 +5047,7 @@ static int vt1716S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
 static int vt1716S_auto_create_analog_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	static hda_nid_t pin_idxs[] = { 0x1f, 0x1a, 0x1b, 0x1e, 0, 0xff };
+	static const hda_nid_t pin_idxs[] = { 0x1f, 0x1a, 0x1b, 0x1e, 0, 0xff };
 	return vt_auto_create_analog_input_ctls(codec, cfg, 0x16, pin_idxs,
 						ARRAY_SIZE(pin_idxs));
 }
@@ -5276,7 +5094,7 @@ static int vt1716S_parse_auto_config(struct hda_codec *codec)
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list vt1716S_loopbacks[] = {
+static const struct hda_amp_list vt1716S_loopbacks[] = {
 	{ 0x16, HDA_INPUT, 1 },
 	{ 0x16, HDA_INPUT, 2 },
 	{ 0x16, HDA_INPUT, 3 },
@@ -5285,6 +5103,99 @@ static struct hda_amp_list vt1716S_loopbacks[] = {
 };
 #endif
 
+static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
+{
+	struct via_spec *spec = codec->spec;
+	int imux_is_smixer;
+	unsigned int parm;
+	unsigned int mono_out, present;
+	/* SW0 (17h) = stereo mixer */
+	imux_is_smixer =
+	(snd_hda_codec_read(codec, 0x17, 0,
+			    AC_VERB_GET_CONNECT_SEL, 0x00) ==  5);
+	/* inputs */
+	/* PW 1/2/5 (1ah/1bh/1eh) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x1a, &parm);
+	set_pin_power_state(codec, 0x1b, &parm);
+	set_pin_power_state(codec, 0x1e, &parm);
+	if (imux_is_smixer)
+		parm = AC_PWRST_D0;
+	/* SW0 (17h), AIW0(13h) */
+	snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x1e, &parm);
+	/* PW11 (22h) */
+	if (spec->dmic_enabled)
+		set_pin_power_state(codec, 0x22, &parm);
+	else
+		snd_hda_codec_write(codec, 0x22, 0,
+				    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+
+	/* SW2(26h), AIW1(14h) */
+	snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	/* outputs */
+	/* PW0 (19h), SW1 (18h), AOW1 (11h) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x19, &parm);
+	/* Smart 5.1 PW2(1bh) */
+	if (spec->smart51_enabled)
+		set_pin_power_state(codec, 0x1b, &parm);
+	snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	/* PW7 (23h), SW3 (27h), AOW3 (25h) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x23, &parm);
+	/* Smart 5.1 PW1(1ah) */
+	if (spec->smart51_enabled)
+		set_pin_power_state(codec, 0x1a, &parm);
+	snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	/* Smart 5.1 PW5(1eh) */
+	if (spec->smart51_enabled)
+		set_pin_power_state(codec, 0x1e, &parm);
+	snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	/* Mono out */
+	/* SW4(28h)->MW1(29h)-> PW12 (2ah)*/
+	present = snd_hda_jack_detect(codec, 0x1c);
+
+	if (present)
+		mono_out = 0;
+	else {
+		present = snd_hda_jack_detect(codec, 0x1d);
+		if (!spec->hp_independent_mode && present)
+			mono_out = 0;
+		else
+			mono_out = 1;
+	}
+	parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3;
+	snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	/* PW 3/4 (1ch/1dh) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x1c, &parm);
+	set_pin_power_state(codec, 0x1d, &parm);
+	/* HP Independent Mode, power on AOW3 */
+	if (spec->hp_independent_mode)
+		snd_hda_codec_write(codec, 0x25, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+
+	/* force to D0 for internal Speaker */
+	/* MW0 (16h), AOW0 (10h) */
+	snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE,
+			    imux_is_smixer ? AC_PWRST_D0 : parm);
+	snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE,
+			    mono_out ? AC_PWRST_D0 : parm);
+}
+
 static int patch_vt1716S(struct hda_codec *codec)
 {
 	struct via_spec *spec;
@@ -5339,13 +5250,14 @@ static int patch_vt1716S(struct hda_codec *codec)
 	spec->loopback.amplist = vt1716S_loopbacks;
 #endif
 
+	spec->set_widgets_power_state = set_widgets_power_state_vt1716S;
 	return 0;
 }
 
 /* for vt2002P */
 
 /* capture mixer elements */
-static struct snd_kcontrol_new vt2002P_capture_mixer[] = {
+static const struct snd_kcontrol_new vt2002P_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT),
@@ -5368,7 +5280,11 @@ static struct snd_kcontrol_new vt2002P_capture_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb vt2002P_volume_init_verbs[] = {
+static const struct hda_verb vt2002P_volume_init_verbs[] = {
+	/* Class-D speaker related verbs */
+	{0x1, 0xfe0, 0x4},
+	{0x1, 0xfe9, 0x80},
+	{0x1, 0xfe2, 0x22},
 	/*
 	 * Unmute ADC0-1 and set the default input to mic-in
 	 */
@@ -5419,9 +5335,60 @@ static struct hda_verb vt2002P_volume_init_verbs[] = {
 	{0x1, 0xfb8, 0x88},
 	{ }
 };
+static const struct hda_verb vt1802_volume_init_verbs[] = {
+	/*
+	 * Unmute ADC0-1 and set the default input to mic-in
+	 */
+	{0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+
+	/* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+	 * mixer widget
+	 */
+	/* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+	/* MUX Indices: Mic = 0 */
+	{0x1e, AC_VERB_SET_CONNECT_SEL, 0},
+	{0x1f, AC_VERB_SET_CONNECT_SEL, 0},
+
+	/* PW9 Output enable */
+	{0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
 
+	/* Enable Boost Volume backdoor */
+	{0x1, 0xfb9, 0x24},
+
+	/* MW0/1/4/8: un-mute index 0 (MUXx), un-mute index 1 (MW9) */
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 
-static struct hda_verb vt2002P_uniwill_init_verbs[] = {
+	/* set MUX0/1/4/8 = 0 (AOW0) */
+	{0x34, AC_VERB_SET_CONNECT_SEL, 0},
+	{0x35, AC_VERB_SET_CONNECT_SEL, 0},
+	{0x38, AC_VERB_SET_CONNECT_SEL, 0},
+	{0x3c, AC_VERB_SET_CONNECT_SEL, 0},
+
+	/* set PW0 index=0 (MW0) */
+	{0x24, AC_VERB_SET_CONNECT_SEL, 0},
+
+	/* Enable AOW0 to MW9 */
+	{0x1, 0xfb8, 0x88},
+	{ }
+};
+
+
+static const struct hda_verb vt2002P_uniwill_init_verbs[] = {
 	{0x25, AC_VERB_SET_UNSOLICITED_ENABLE,
 	 AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT},
 	{0x26, AC_VERB_SET_UNSOLICITED_ENABLE,
@@ -5431,8 +5398,18 @@ static struct hda_verb vt2002P_uniwill_init_verbs[] = {
 	{0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
 	{ }
 };
+static const struct hda_verb vt1802_uniwill_init_verbs[] = {
+	{0x25, AC_VERB_SET_UNSOLICITED_ENABLE,
+	 AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT},
+	{0x28, AC_VERB_SET_UNSOLICITED_ENABLE,
+	 AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT},
+	{0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+	{0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+	{0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+	{ }
+};
 
-static struct hda_pcm_stream vt2002P_pcm_analog_playback = {
+static const struct hda_pcm_stream vt2002P_pcm_analog_playback = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -5445,7 +5422,7 @@ static struct hda_pcm_stream vt2002P_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt2002P_pcm_analog_capture = {
+static const struct hda_pcm_stream vt2002P_pcm_analog_capture = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -5458,7 +5435,7 @@ static struct hda_pcm_stream vt2002P_pcm_analog_capture = {
 	},
 };
 
-static struct hda_pcm_stream vt2002P_pcm_digital_playback = {
+static const struct hda_pcm_stream vt2002P_pcm_digital_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -5478,7 +5455,7 @@ static int vt2002P_auto_fill_dac_nids(struct via_spec *spec,
 	spec->multiout.num_dacs = 1;
 	spec->multiout.dac_nids = spec->private_dac_nids;
 	if (cfg->line_out_pins[0])
-		spec->multiout.dac_nids[0] = 0x8;
+		spec->private_dac_nids[0] = 0x8;
 	return 0;
 }
 
@@ -5487,10 +5464,15 @@ static int vt2002P_auto_create_multi_out_ctls(struct via_spec *spec,
 					     const struct auto_pin_cfg *cfg)
 {
 	int err;
+	hda_nid_t sw_nid;
 
 	if (!cfg->line_out_pins[0])
 		return -1;
 
+	if (spec->codec_type == VT1802)
+		sw_nid = 0x28;
+	else
+		sw_nid = 0x26;
 
 	/* Line-Out: PortE */
 	err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
@@ -5500,7 +5482,7 @@ static int vt2002P_auto_create_multi_out_ctls(struct via_spec *spec,
 		return err;
 	err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE,
 			      "Master Front Playback Switch",
-			      HDA_COMPOSE_AMP_VAL(0x26, 3, 0, HDA_OUTPUT));
+			      HDA_COMPOSE_AMP_VAL(sw_nid, 3, 0, HDA_OUTPUT));
 	if (err < 0)
 		return err;
 
@@ -5540,7 +5522,7 @@ static int vt2002P_auto_create_analog_input_ctls(struct hda_codec *codec,
 {
 	struct via_spec *spec = codec->spec;
 	struct hda_input_mux *imux = &spec->private_imux[0];
-	static hda_nid_t pin_idxs[] = { 0x2b, 0x2a, 0x29, 0xff };
+	static const hda_nid_t pin_idxs[] = { 0x2b, 0x2a, 0x29, 0xff };
 	int err;
 
 	err = vt_auto_create_analog_input_ctls(codec, cfg, 0x21, pin_idxs,
@@ -5601,7 +5583,7 @@ static int vt2002P_parse_auto_config(struct hda_codec *codec)
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list vt2002P_loopbacks[] = {
+static const struct hda_amp_list vt2002P_loopbacks[] = {
 	{ 0x21, HDA_INPUT, 0 },
 	{ 0x21, HDA_INPUT, 1 },
 	{ 0x21, HDA_INPUT, 2 },
@@ -5609,6 +5591,116 @@ static struct hda_amp_list vt2002P_loopbacks[] = {
 };
 #endif
 
+static void set_widgets_power_state_vt2002P(struct hda_codec *codec)
+{
+	struct via_spec *spec = codec->spec;
+	int imux_is_smixer;
+	unsigned int parm;
+	unsigned int present;
+	/* MUX9 (1eh) = stereo mixer */
+	imux_is_smixer =
+	snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3;
+	/* inputs */
+	/* PW 5/6/7 (29h/2ah/2bh) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x29, &parm);
+	set_pin_power_state(codec, 0x2a, &parm);
+	set_pin_power_state(codec, 0x2b, &parm);
+	parm = AC_PWRST_D0;
+	/* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */
+	snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	/* outputs */
+	/* AOW0 (8h)*/
+	snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	if (spec->codec_type == VT1802) {
+		/* PW4 (28h), MW4 (18h), MUX4(38h) */
+		parm = AC_PWRST_D3;
+		set_pin_power_state(codec, 0x28, &parm);
+		snd_hda_codec_write(codec, 0x18, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+		snd_hda_codec_write(codec, 0x38, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+	} else {
+		/* PW4 (26h), MW4 (1ch), MUX4(37h) */
+		parm = AC_PWRST_D3;
+		set_pin_power_state(codec, 0x26, &parm);
+		snd_hda_codec_write(codec, 0x1c, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+		snd_hda_codec_write(codec, 0x37, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+	}
+
+	if (spec->codec_type == VT1802) {
+		/* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */
+		parm = AC_PWRST_D3;
+		set_pin_power_state(codec, 0x25, &parm);
+		snd_hda_codec_write(codec, 0x15, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+		snd_hda_codec_write(codec, 0x35, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+	} else {
+		/* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */
+		parm = AC_PWRST_D3;
+		set_pin_power_state(codec, 0x25, &parm);
+		snd_hda_codec_write(codec, 0x19, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+		snd_hda_codec_write(codec, 0x35, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+	}
+
+	if (spec->hp_independent_mode)
+		snd_hda_codec_write(codec, 0x9, 0,
+				    AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+
+	/* Class-D */
+	/* PW0 (24h), MW0(18h/14h), MUX0(34h) */
+	present = snd_hda_jack_detect(codec, 0x25);
+
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x24, &parm);
+	parm = present ? AC_PWRST_D3 : AC_PWRST_D0;
+	if (spec->codec_type == VT1802)
+		snd_hda_codec_write(codec, 0x14, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+	else
+		snd_hda_codec_write(codec, 0x18, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	/* Mono Out */
+	present = snd_hda_jack_detect(codec, 0x26);
+
+	parm = present ? AC_PWRST_D3 : AC_PWRST_D0;
+	if (spec->codec_type == VT1802) {
+		/* PW15 (33h), MW8(1ch), MUX8(3ch) */
+		snd_hda_codec_write(codec, 0x33, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+		snd_hda_codec_write(codec, 0x1c, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+		snd_hda_codec_write(codec, 0x3c, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+	} else {
+		/* PW15 (31h), MW8(17h), MUX8(3bh) */
+		snd_hda_codec_write(codec, 0x31, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+		snd_hda_codec_write(codec, 0x17, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+		snd_hda_codec_write(codec, 0x3b, 0,
+				    AC_VERB_SET_POWER_STATE, parm);
+	}
+	/* MW9 (21h) */
+	if (imux_is_smixer || !is_aa_path_mute(codec))
+		snd_hda_codec_write(codec, 0x21, 0,
+				    AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+	else
+		snd_hda_codec_write(codec, 0x21, 0,
+				    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+}
 
 /* patch for vt2002P */
 static int patch_vt2002P(struct hda_codec *codec)
@@ -5631,14 +5723,31 @@ static int patch_vt2002P(struct hda_codec *codec)
 		       "from BIOS.  Using genenic mode...\n");
 	}
 
-	spec->init_verbs[spec->num_iverbs++]  = vt2002P_volume_init_verbs;
-	spec->init_verbs[spec->num_iverbs++] = vt2002P_uniwill_init_verbs;
+	if (spec->codec_type == VT1802)
+		spec->init_verbs[spec->num_iverbs++]  =
+			vt1802_volume_init_verbs;
+	else
+		spec->init_verbs[spec->num_iverbs++]  =
+			vt2002P_volume_init_verbs;
+
+	if (spec->codec_type == VT1802)
+		spec->init_verbs[spec->num_iverbs++] =
+			vt1802_uniwill_init_verbs;
+	else
+		spec->init_verbs[spec->num_iverbs++] =
+			vt2002P_uniwill_init_verbs;
 
-	spec->stream_name_analog = "VT2002P Analog";
+	if (spec->codec_type == VT1802)
+		spec->stream_name_analog = "VT1802 Analog";
+	else
+		spec->stream_name_analog = "VT2002P Analog";
 	spec->stream_analog_playback = &vt2002P_pcm_analog_playback;
 	spec->stream_analog_capture = &vt2002P_pcm_analog_capture;
 
-	spec->stream_name_digital = "VT2002P Digital";
+	if (spec->codec_type == VT1802)
+		spec->stream_name_digital = "VT1802 Digital";
+	else
+		spec->stream_name_digital = "VT2002P Digital";
 	spec->stream_digital_playback = &vt2002P_pcm_digital_playback;
 
 	if (!spec->adc_nids && spec->input_mux) {
@@ -5660,13 +5769,14 @@ static int patch_vt2002P(struct hda_codec *codec)
 	spec->loopback.amplist = vt2002P_loopbacks;
 #endif
 
+	spec->set_widgets_power_state =  set_widgets_power_state_vt2002P;
 	return 0;
 }
 
 /* for vt1812 */
 
 /* capture mixer elements */
-static struct snd_kcontrol_new vt1812_capture_mixer[] = {
+static const struct snd_kcontrol_new vt1812_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT),
@@ -5688,7 +5798,7 @@ static struct snd_kcontrol_new vt1812_capture_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb vt1812_volume_init_verbs[] = {
+static const struct hda_verb vt1812_volume_init_verbs[] = {
 	/*
 	 * Unmute ADC0-1 and set the default input to mic-in
 	 */
@@ -5741,7 +5851,7 @@ static struct hda_verb vt1812_volume_init_verbs[] = {
 };
 
 
-static struct hda_verb vt1812_uniwill_init_verbs[] = {
+static const struct hda_verb vt1812_uniwill_init_verbs[] = {
 	{0x33, AC_VERB_SET_UNSOLICITED_ENABLE,
 	 AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT},
 	{0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT },
@@ -5753,7 +5863,7 @@ static struct hda_verb vt1812_uniwill_init_verbs[] = {
 	{ }
 };
 
-static struct hda_pcm_stream vt1812_pcm_analog_playback = {
+static const struct hda_pcm_stream vt1812_pcm_analog_playback = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -5766,7 +5876,7 @@ static struct hda_pcm_stream vt1812_pcm_analog_playback = {
 	},
 };
 
-static struct hda_pcm_stream vt1812_pcm_analog_capture = {
+static const struct hda_pcm_stream vt1812_pcm_analog_capture = {
 	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -5779,7 +5889,7 @@ static struct hda_pcm_stream vt1812_pcm_analog_capture = {
 	},
 };
 
-static struct hda_pcm_stream vt1812_pcm_digital_playback = {
+static const struct hda_pcm_stream vt1812_pcm_digital_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
@@ -5798,7 +5908,7 @@ static int vt1812_auto_fill_dac_nids(struct via_spec *spec,
 	spec->multiout.num_dacs = 1;
 	spec->multiout.dac_nids = spec->private_dac_nids;
 	if (cfg->line_out_pins[0])
-		spec->multiout.dac_nids[0] = 0x8;
+		spec->private_dac_nids[0] = 0x8;
 	return 0;
 }
 
@@ -5861,7 +5971,7 @@ static int vt1812_auto_create_analog_input_ctls(struct hda_codec *codec,
 {
 	struct via_spec *spec = codec->spec;
 	struct hda_input_mux *imux = &spec->private_imux[0];
-	static hda_nid_t pin_idxs[] = { 0x2b, 0x2a, 0x29, 0, 0, 0xff };
+	static const hda_nid_t pin_idxs[] = { 0x2b, 0x2a, 0x29, 0, 0, 0xff };
 	int err;
 
 	err = vt_auto_create_analog_input_ctls(codec, cfg, 0x21, pin_idxs,
@@ -5923,7 +6033,7 @@ static int vt1812_parse_auto_config(struct hda_codec *codec)
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static struct hda_amp_list vt1812_loopbacks[] = {
+static const struct hda_amp_list vt1812_loopbacks[] = {
 	{ 0x21, HDA_INPUT, 0 },
 	{ 0x21, HDA_INPUT, 1 },
 	{ 0x21, HDA_INPUT, 2 },
@@ -5931,6 +6041,97 @@ static struct hda_amp_list vt1812_loopbacks[] = {
 };
 #endif
 
+static void set_widgets_power_state_vt1812(struct hda_codec *codec)
+{
+	struct via_spec *spec = codec->spec;
+	int imux_is_smixer =
+	snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3;
+	unsigned int parm;
+	unsigned int present;
+	/* MUX10 (1eh) = stereo mixer */
+	imux_is_smixer =
+	snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5;
+	/* inputs */
+	/* PW 5/6/7 (29h/2ah/2bh) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x29, &parm);
+	set_pin_power_state(codec, 0x2a, &parm);
+	set_pin_power_state(codec, 0x2b, &parm);
+	parm = AC_PWRST_D0;
+	/* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */
+	snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	/* outputs */
+	/* AOW0 (8h)*/
+	snd_hda_codec_write(codec, 0x8, 0,
+			    AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+
+	/* PW4 (28h), MW4 (18h), MUX4(38h) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x28, &parm);
+	snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x38, 0, AC_VERB_SET_POWER_STATE, parm);
+
+	/* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x25, &parm);
+	snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x35, 0, AC_VERB_SET_POWER_STATE, parm);
+	if (spec->hp_independent_mode)
+		snd_hda_codec_write(codec, 0x9, 0,
+				    AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+
+	/* Internal Speaker */
+	/* PW0 (24h), MW0(14h), MUX0(34h) */
+	present = snd_hda_jack_detect(codec, 0x25);
+
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x24, &parm);
+	if (present) {
+		snd_hda_codec_write(codec, 0x14, 0,
+				    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+		snd_hda_codec_write(codec, 0x34, 0,
+				    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+	} else {
+		snd_hda_codec_write(codec, 0x14, 0,
+				    AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+		snd_hda_codec_write(codec, 0x34, 0,
+				    AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+	}
+
+
+	/* Mono Out */
+	/* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */
+	present = snd_hda_jack_detect(codec, 0x28);
+
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x31, &parm);
+	if (present) {
+		snd_hda_codec_write(codec, 0x1c, 0,
+				    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+		snd_hda_codec_write(codec, 0x3c, 0,
+				    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+		snd_hda_codec_write(codec, 0x3e, 0,
+				    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+	} else {
+		snd_hda_codec_write(codec, 0x1c, 0,
+				    AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+		snd_hda_codec_write(codec, 0x3c, 0,
+				    AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+		snd_hda_codec_write(codec, 0x3e, 0,
+				    AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+	}
+
+	/* PW15 (33h), MW15 (1dh), MUX15(3dh) */
+	parm = AC_PWRST_D3;
+	set_pin_power_state(codec, 0x33, &parm);
+	snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm);
+	snd_hda_codec_write(codec, 0x3d, 0, AC_VERB_SET_POWER_STATE, parm);
+
+}
 
 /* patch for vt1812 */
 static int patch_vt1812(struct hda_codec *codec)
@@ -5984,13 +6185,14 @@ static int patch_vt1812(struct hda_codec *codec)
 	spec->loopback.amplist = vt1812_loopbacks;
 #endif
 
+	spec->set_widgets_power_state =  set_widgets_power_state_vt1812;
 	return 0;
 }
 
 /*
  * patch entries
  */
-static struct hda_codec_preset snd_hda_preset_via[] = {
+static const struct hda_codec_preset snd_hda_preset_via[] = {
 	{ .id = 0x11061708, .name = "VT1708", .patch = patch_vt1708},
 	{ .id = 0x11061709, .name = "VT1708", .patch = patch_vt1708},
 	{ .id = 0x1106170a, .name = "VT1708", .patch = patch_vt1708},
@@ -6035,7 +6237,7 @@ static struct hda_codec_preset snd_hda_preset_via[] = {
 	  .patch = patch_vt1708S},
 	{ .id = 0x11063397, .name = "VT1708S",
 	  .patch = patch_vt1708S},
-	{ .id = 0x11064397, .name = "VT1708S",
+	{ .id = 0x11064397, .name = "VT1705",
 	  .patch = patch_vt1708S},
 	{ .id = 0x11065397, .name = "VT1708S",
 	  .patch = patch_vt1708S},
@@ -6076,6 +6278,10 @@ static struct hda_codec_preset snd_hda_preset_via[] = {
 	{ .id = 0x11060448, .name = "VT1812", .patch = patch_vt1812},
 	{ .id = 0x11060440, .name = "VT1818S",
 	  .patch = patch_vt1708S},
+	{ .id = 0x11060446, .name = "VT1802",
+		.patch = patch_vt2002P},
+	{ .id = 0x11068446, .name = "VT1802",
+		.patch = patch_vt2002P},
 	{} /* terminator */
 };
 
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 27709f0cd2a..f3353b49c78 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -235,8 +235,8 @@ static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0m_ids) = {
 	{ PCI_VDEVICE(NVIDIA, 0x0069), DEVICE_NFORCE }, /* NFORCE2 */
 	{ PCI_VDEVICE(NVIDIA, 0x0089), DEVICE_NFORCE }, /* NFORCE2s */
 	{ PCI_VDEVICE(NVIDIA, 0x00d9), DEVICE_NFORCE }, /* NFORCE3 */
+	{ PCI_VDEVICE(AMD, 0x746e), DEVICE_INTEL },	/* AMD8111 */
 #if 0
-	{ PCI_VDEVICE(AMD, 0x746d), DEVICE_INTEL },	/* AMD8111 */
 	{ PCI_VDEVICE(AL, 0x5455), DEVICE_ALI },   /* Ali5455 */
 #endif
 	{ 0, }
@@ -1261,9 +1261,9 @@ static struct shortname_table {
 	{ PCI_DEVICE_ID_NVIDIA_MCP2_MODEM, "NVidia nForce2" },
 	{ PCI_DEVICE_ID_NVIDIA_MCP2S_MODEM, "NVidia nForce2s" },
 	{ PCI_DEVICE_ID_NVIDIA_MCP3_MODEM, "NVidia nForce3" },
+	{ 0x746e, "AMD AMD8111" },
 #if 0
 	{ 0x5455, "ALi M5455" },
-	{ 0x746d, "AMD AMD8111" },
 #endif
 	{ 0 },
 };
diff --git a/sound/pci/lola/Makefile b/sound/pci/lola/Makefile
new file mode 100644
index 00000000000..8178a2a59d0
--- /dev/null
+++ b/sound/pci/lola/Makefile
@@ -0,0 +1,4 @@
+snd-lola-y := lola.o lola_pcm.o lola_clock.o lola_mixer.o
+snd-lola-$(CONFIG_SND_DEBUG) += lola_proc.o
+
+obj-$(CONFIG_SND_LOLA) += snd-lola.o
diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c
new file mode 100644
index 00000000000..34b24286d27
--- /dev/null
+++ b/sound/pci/lola/lola.c
@@ -0,0 +1,791 @@
+/*
+ *  Support for Digigram Lola PCI-e boards
+ *
+ *  Copyright (c) 2011 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.
+ */
+
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/moduleparam.h>
+#include <linux/dma-mapping.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include "lola.h"
+
+/* Standard options */
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Digigram Lola driver.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Digigram Lola driver.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Digigram Lola driver.");
+
+/* Lola-specific options */
+
+/* for instance use always max granularity which is compatible
+ * with all sample rates
+ */
+static int granularity[SNDRV_CARDS] = {
+	[0 ... (SNDRV_CARDS - 1)] = LOLA_GRANULARITY_MAX
+};
+
+/* below a sample_rate of 16kHz the analogue audio quality is NOT excellent */
+static int sample_rate_min[SNDRV_CARDS] = {
+	[0 ... (SNDRV_CARDS - 1) ] = 16000
+};
+
+module_param_array(granularity, int, NULL, 0444);
+MODULE_PARM_DESC(granularity, "Granularity value");
+module_param_array(sample_rate_min, int, NULL, 0444);
+MODULE_PARM_DESC(sample_rate_min, "Minimal sample rate");
+
+/*
+ */
+
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Digigram, Lola}}");
+MODULE_DESCRIPTION("Digigram Lola driver");
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+static int debug;
+module_param(debug, int, 0644);
+#define verbose_debug(fmt, args...)			\
+	do { if (debug > 1) printk(KERN_DEBUG SFX fmt, ##args); } while (0)
+#else
+#define verbose_debug(fmt, args...)
+#endif
+
+/*
+ * pseudo-codec read/write via CORB/RIRB
+ */
+
+static int corb_send_verb(struct lola *chip, unsigned int nid,
+			  unsigned int verb, unsigned int data,
+			  unsigned int extdata)
+{
+	unsigned long flags;
+	int ret = -EIO;
+
+	chip->last_cmd_nid = nid;
+	chip->last_verb = verb;
+	chip->last_data = data;
+	chip->last_extdata = extdata;
+	data |= (nid << 20) | (verb << 8);
+
+	spin_lock_irqsave(&chip->reg_lock, flags);
+	if (chip->rirb.cmds < LOLA_CORB_ENTRIES - 1) {
+		unsigned int wp = chip->corb.wp + 1;
+		wp %= LOLA_CORB_ENTRIES;
+		chip->corb.wp = wp;
+		chip->corb.buf[wp * 2] = cpu_to_le32(data);
+		chip->corb.buf[wp * 2 + 1] = cpu_to_le32(extdata);
+		lola_writew(chip, BAR0, CORBWP, wp);
+		chip->rirb.cmds++;
+		smp_wmb();
+		ret = 0;
+	}
+	spin_unlock_irqrestore(&chip->reg_lock, flags);
+	return ret;
+}
+
+static void lola_queue_unsol_event(struct lola *chip, unsigned int res,
+				   unsigned int res_ex)
+{
+	lola_update_ext_clock_freq(chip, res);
+}
+
+/* retrieve RIRB entry - called from interrupt handler */
+static void lola_update_rirb(struct lola *chip)
+{
+	unsigned int rp, wp;
+	u32 res, res_ex;
+
+	wp = lola_readw(chip, BAR0, RIRBWP);
+	if (wp == chip->rirb.wp)
+		return;
+	chip->rirb.wp = wp;
+
+	while (chip->rirb.rp != wp) {
+		chip->rirb.rp++;
+		chip->rirb.rp %= LOLA_CORB_ENTRIES;
+
+		rp = chip->rirb.rp << 1; /* an RIRB entry is 8-bytes */
+		res_ex = le32_to_cpu(chip->rirb.buf[rp + 1]);
+		res = le32_to_cpu(chip->rirb.buf[rp]);
+		if (res_ex & LOLA_RIRB_EX_UNSOL_EV)
+			lola_queue_unsol_event(chip, res, res_ex);
+		else if (chip->rirb.cmds) {
+			chip->res = res;
+			chip->res_ex = res_ex;
+			smp_wmb();
+			chip->rirb.cmds--;
+		}
+	}
+}
+
+static int rirb_get_response(struct lola *chip, unsigned int *val,
+			     unsigned int *extval)
+{
+	unsigned long timeout;
+
+ again:
+	timeout = jiffies + msecs_to_jiffies(1000);
+	for (;;) {
+		if (chip->polling_mode) {
+			spin_lock_irq(&chip->reg_lock);
+			lola_update_rirb(chip);
+			spin_unlock_irq(&chip->reg_lock);
+		}
+		if (!chip->rirb.cmds) {
+			*val = chip->res;
+			if (extval)
+				*extval = chip->res_ex;
+			verbose_debug("get_response: %x, %x\n",
+				      chip->res, chip->res_ex);
+			if (chip->res_ex & LOLA_RIRB_EX_ERROR) {
+				printk(KERN_WARNING SFX "RIRB ERROR: "
+				       "NID=%x, verb=%x, data=%x, ext=%x\n",
+				       chip->last_cmd_nid,
+				       chip->last_verb, chip->last_data,
+				       chip->last_extdata);
+				return -EIO;
+			}
+			return 0;
+		}
+		if (time_after(jiffies, timeout))
+			break;
+		udelay(20);
+		cond_resched();
+	}
+	printk(KERN_WARNING SFX "RIRB response error\n");
+	if (!chip->polling_mode) {
+		printk(KERN_WARNING SFX "switching to polling mode\n");
+		chip->polling_mode = 1;
+		goto again;
+	}
+	return -EIO;
+}
+
+/* aynchronous write of a codec verb with data */
+int lola_codec_write(struct lola *chip, unsigned int nid, unsigned int verb,
+		     unsigned int data, unsigned int extdata)
+{
+	verbose_debug("codec_write NID=%x, verb=%x, data=%x, ext=%x\n",
+		      nid, verb, data, extdata);
+	return corb_send_verb(chip, nid, verb, data, extdata);
+}
+
+/* write a codec verb with data and read the returned status */
+int lola_codec_read(struct lola *chip, unsigned int nid, unsigned int verb,
+		    unsigned int data, unsigned int extdata,
+		    unsigned int *val, unsigned int *extval)
+{
+	int err;
+
+	verbose_debug("codec_read NID=%x, verb=%x, data=%x, ext=%x\n",
+		      nid, verb, data, extdata);
+	err = corb_send_verb(chip, nid, verb, data, extdata);
+	if (err < 0)
+		return err;
+	err = rirb_get_response(chip, val, extval);
+	return err;
+}
+
+/* flush all pending codec writes */
+int lola_codec_flush(struct lola *chip)
+{
+	unsigned int tmp;
+	return rirb_get_response(chip, &tmp, NULL);
+}
+
+/*
+ * interrupt handler
+ */
+static irqreturn_t lola_interrupt(int irq, void *dev_id)
+{
+	struct lola *chip = dev_id;
+	unsigned int notify_ins, notify_outs, error_ins, error_outs;
+	int handled = 0;
+	int i;
+
+	notify_ins = notify_outs = error_ins = error_outs = 0;
+	spin_lock(&chip->reg_lock);
+	for (;;) {
+		unsigned int status, in_sts, out_sts;
+		unsigned int reg;
+
+		status = lola_readl(chip, BAR1, DINTSTS);
+		if (!status || status == -1)
+			break;
+
+		in_sts = lola_readl(chip, BAR1, DIINTSTS);
+		out_sts = lola_readl(chip, BAR1, DOINTSTS);
+
+		/* clear Input Interrupts */
+		for (i = 0; in_sts && i < chip->pcm[CAPT].num_streams; i++) {
+			if (!(in_sts & (1 << i)))
+				continue;
+			in_sts &= ~(1 << i);
+			reg = lola_dsd_read(chip, i, STS);
+			if (reg & LOLA_DSD_STS_DESE) /* error */
+				error_ins |= (1 << i);
+			if (reg & LOLA_DSD_STS_BCIS) /* notify */
+				notify_ins |= (1 << i);
+			/* clear */
+			lola_dsd_write(chip, i, STS, reg);
+		}
+
+		/* clear Output Interrupts */
+		for (i = 0; out_sts && i < chip->pcm[PLAY].num_streams; i++) {
+			if (!(out_sts & (1 << i)))
+				continue;
+			out_sts &= ~(1 << i);
+			reg = lola_dsd_read(chip, i + MAX_STREAM_IN_COUNT, STS);
+			if (reg & LOLA_DSD_STS_DESE) /* error */
+				error_outs |= (1 << i);
+			if (reg & LOLA_DSD_STS_BCIS) /* notify */
+				notify_outs |= (1 << i);
+			lola_dsd_write(chip, i + MAX_STREAM_IN_COUNT, STS, reg);
+		}
+
+		if (status & LOLA_DINT_CTRL) {
+			unsigned char rbsts; /* ring status is byte access */
+			rbsts = lola_readb(chip, BAR0, RIRBSTS);
+			rbsts &= LOLA_RIRB_INT_MASK;
+			if (rbsts)
+				lola_writeb(chip, BAR0, RIRBSTS, rbsts);
+			rbsts = lola_readb(chip, BAR0, CORBSTS);
+			rbsts &= LOLA_CORB_INT_MASK;
+			if (rbsts)
+				lola_writeb(chip, BAR0, CORBSTS, rbsts);
+
+			lola_update_rirb(chip);
+		}
+
+		if (status & (LOLA_DINT_FIFOERR | LOLA_DINT_MUERR)) {
+			/* clear global fifo error interrupt */
+			lola_writel(chip, BAR1, DINTSTS,
+				    (status & (LOLA_DINT_FIFOERR | LOLA_DINT_MUERR)));
+		}
+		handled = 1;
+	}
+	spin_unlock(&chip->reg_lock);
+
+	lola_pcm_update(chip, &chip->pcm[CAPT], notify_ins);
+	lola_pcm_update(chip, &chip->pcm[PLAY], notify_outs);
+
+	return IRQ_RETVAL(handled);
+}
+
+
+/*
+ * controller
+ */
+static int reset_controller(struct lola *chip)
+{
+	unsigned int gctl = lola_readl(chip, BAR0, GCTL);
+	unsigned long end_time;
+
+	if (gctl) {
+		/* to be sure */
+		lola_writel(chip, BAR1, BOARD_MODE, 0);
+		return 0;
+	}
+
+	chip->cold_reset = 1;
+	lola_writel(chip, BAR0, GCTL, LOLA_GCTL_RESET);
+	end_time = jiffies + msecs_to_jiffies(200);
+	do {
+		msleep(1);
+		gctl = lola_readl(chip, BAR0, GCTL);
+		if (gctl)
+			break;
+	} while (time_before(jiffies, end_time));
+	if (!gctl) {
+		printk(KERN_ERR SFX "cannot reset controller\n");
+		return -EIO;
+	}
+	return 0;
+}
+
+static void lola_irq_enable(struct lola *chip)
+{
+	unsigned int val;
+
+	/* enalbe all I/O streams */
+	val = (1 << chip->pcm[PLAY].num_streams) - 1;
+	lola_writel(chip, BAR1, DOINTCTL, val);
+	val = (1 << chip->pcm[CAPT].num_streams) - 1;
+	lola_writel(chip, BAR1, DIINTCTL, val);
+
+	/* enable global irqs */
+	val = LOLA_DINT_GLOBAL | LOLA_DINT_CTRL | LOLA_DINT_FIFOERR |
+		LOLA_DINT_MUERR;
+	lola_writel(chip, BAR1, DINTCTL, val);
+}
+
+static void lola_irq_disable(struct lola *chip)
+{
+	lola_writel(chip, BAR1, DINTCTL, 0);
+	lola_writel(chip, BAR1, DIINTCTL, 0);
+	lola_writel(chip, BAR1, DOINTCTL, 0);
+}
+
+static int setup_corb_rirb(struct lola *chip)
+{
+	int err;
+	unsigned char tmp;
+	unsigned long end_time;
+
+	err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+				  snd_dma_pci_data(chip->pci),
+				  PAGE_SIZE, &chip->rb);
+	if (err < 0)
+		return err;
+
+	chip->corb.addr = chip->rb.addr;
+	chip->corb.buf = (u32 *)chip->rb.area;
+	chip->rirb.addr = chip->rb.addr + 2048;
+	chip->rirb.buf = (u32 *)(chip->rb.area + 2048);
+
+	/* disable ringbuffer DMAs */
+	lola_writeb(chip, BAR0, RIRBCTL, 0);
+	lola_writeb(chip, BAR0, CORBCTL, 0);
+
+	end_time = jiffies + msecs_to_jiffies(200);
+	do {
+		if (!lola_readb(chip, BAR0, RIRBCTL) &&
+		    !lola_readb(chip, BAR0, CORBCTL))
+			break;
+		msleep(1);
+	} while (time_before(jiffies, end_time));
+
+	/* CORB set up */
+	lola_writel(chip, BAR0, CORBLBASE, (u32)chip->corb.addr);
+	lola_writel(chip, BAR0, CORBUBASE, upper_32_bits(chip->corb.addr));
+	/* set the corb size to 256 entries */
+	lola_writeb(chip, BAR0, CORBSIZE, 0x02);
+	/* set the corb write pointer to 0 */
+	lola_writew(chip, BAR0, CORBWP, 0);
+	/* reset the corb hw read pointer */
+	lola_writew(chip, BAR0, CORBRP, LOLA_RBRWP_CLR);
+	/* enable corb dma */
+	lola_writeb(chip, BAR0, CORBCTL, LOLA_RBCTL_DMA_EN);
+	/* clear flags if set */
+	tmp = lola_readb(chip, BAR0, CORBSTS) & LOLA_CORB_INT_MASK;
+	if (tmp)
+		lola_writeb(chip, BAR0, CORBSTS, tmp);
+	chip->corb.wp = 0;
+
+	/* RIRB set up */
+	lola_writel(chip, BAR0, RIRBLBASE, (u32)chip->rirb.addr);
+	lola_writel(chip, BAR0, RIRBUBASE, upper_32_bits(chip->rirb.addr));
+	/* set the rirb size to 256 entries */
+	lola_writeb(chip, BAR0, RIRBSIZE, 0x02);
+	/* reset the rirb hw write pointer */
+	lola_writew(chip, BAR0, RIRBWP, LOLA_RBRWP_CLR);
+	/* set N=1, get RIRB response interrupt for new entry */
+	lola_writew(chip, BAR0, RINTCNT, 1);
+	/* enable rirb dma and response irq */
+	lola_writeb(chip, BAR0, RIRBCTL, LOLA_RBCTL_DMA_EN | LOLA_RBCTL_IRQ_EN);
+	/* clear flags if set */
+	tmp =  lola_readb(chip, BAR0, RIRBSTS) & LOLA_RIRB_INT_MASK;
+	if (tmp)
+		lola_writeb(chip, BAR0, RIRBSTS, tmp);
+	chip->rirb.rp = chip->rirb.cmds = 0;
+
+	return 0;
+}
+
+static void stop_corb_rirb(struct lola *chip)
+{
+	/* disable ringbuffer DMAs */
+	lola_writeb(chip, BAR0, RIRBCTL, 0);
+	lola_writeb(chip, BAR0, CORBCTL, 0);
+}
+
+static void lola_reset_setups(struct lola *chip)
+{
+	/* update the granularity */
+	lola_set_granularity(chip, chip->granularity, true);
+	/* update the sample clock */
+	lola_set_clock_index(chip, chip->clock.cur_index);
+	/* enable unsolicited events of the clock widget */
+	lola_enable_clock_events(chip);
+	/* update the analog gains */
+	lola_setup_all_analog_gains(chip, CAPT, false); /* input, update */
+	/* update SRC configuration if applicable */
+	lola_set_src_config(chip, chip->input_src_mask, false);
+	/* update the analog outputs */
+	lola_setup_all_analog_gains(chip, PLAY, false); /* output, update */
+}
+
+static int lola_parse_tree(struct lola *chip)
+{
+	unsigned int val;
+	int nid, err;
+
+	err = lola_read_param(chip, 0, LOLA_PAR_VENDOR_ID, &val);
+	if (err < 0) {
+		printk(KERN_ERR SFX "Can't read VENDOR_ID\n");
+		return err;
+	}
+	val >>= 16;
+	if (val != 0x1369) {
+		printk(KERN_ERR SFX "Unknown codec vendor 0x%x\n", val);
+		return -EINVAL;
+	}
+
+	err = lola_read_param(chip, 1, LOLA_PAR_FUNCTION_TYPE, &val);
+	if (err < 0) {
+		printk(KERN_ERR SFX "Can't read FUNCTION_TYPE for 0x%x\n", nid);
+		return err;
+	}
+	if (val != 1) {
+		printk(KERN_ERR SFX "Unknown function type %d\n", val);
+		return -EINVAL;
+	}
+
+	err = lola_read_param(chip, 1, LOLA_PAR_SPECIFIC_CAPS, &val);
+	if (err < 0) {
+		printk(KERN_ERR SFX "Can't read SPECCAPS\n");
+		return err;
+	}
+	chip->lola_caps = val;
+	chip->pin[CAPT].num_pins = LOLA_AFG_INPUT_PIN_COUNT(chip->lola_caps);
+	chip->pin[PLAY].num_pins = LOLA_AFG_OUTPUT_PIN_COUNT(chip->lola_caps);
+	snd_printdd(SFX "speccaps=0x%x, pins in=%d, out=%d\n",
+		    chip->lola_caps,
+		    chip->pin[CAPT].num_pins, chip->pin[PLAY].num_pins);
+
+	if (chip->pin[CAPT].num_pins > MAX_AUDIO_INOUT_COUNT ||
+	    chip->pin[PLAY].num_pins > MAX_AUDIO_INOUT_COUNT) {
+		printk(KERN_ERR SFX "Invalid Lola-spec caps 0x%x\n", val);
+		return -EINVAL;
+	}
+
+	nid = 0x02;
+	err = lola_init_pcm(chip, CAPT, &nid);
+	if (err < 0)
+		return err;
+	err = lola_init_pcm(chip, PLAY, &nid);
+	if (err < 0)
+		return err;
+
+	err = lola_init_pins(chip, CAPT, &nid);
+	if (err < 0)
+		return err;
+	err = lola_init_pins(chip, PLAY, &nid);
+	if (err < 0)
+		return err;
+
+	if (LOLA_AFG_CLOCK_WIDGET_PRESENT(chip->lola_caps)) {
+		err = lola_init_clock_widget(chip, nid);
+		if (err < 0)
+			return err;
+		nid++;
+	}
+	if (LOLA_AFG_MIXER_WIDGET_PRESENT(chip->lola_caps)) {
+		err = lola_init_mixer_widget(chip, nid);
+		if (err < 0)
+			return err;
+		nid++;
+	}
+
+	/* enable unsolicited events of the clock widget */
+	err = lola_enable_clock_events(chip);
+	if (err < 0)
+		return err;
+
+	/* if last ResetController was not a ColdReset, we don't know
+	 * the state of the card; initialize here again
+	 */
+	if (!chip->cold_reset) {
+		lola_reset_setups(chip);
+		chip->cold_reset = 1;
+	} else {
+		/* set the granularity if it is not the default */
+		if (chip->granularity != LOLA_GRANULARITY_MIN)
+			lola_set_granularity(chip, chip->granularity, true);
+	}
+
+	return 0;
+}
+
+static void lola_stop_hw(struct lola *chip)
+{
+	stop_corb_rirb(chip);
+	lola_irq_disable(chip);
+}
+
+static void lola_free(struct lola *chip)
+{
+	if (chip->initialized)
+		lola_stop_hw(chip);
+	lola_free_pcm(chip);
+	lola_free_mixer(chip);
+	if (chip->irq >= 0)
+		free_irq(chip->irq, (void *)chip);
+	if (chip->bar[0].remap_addr)
+		iounmap(chip->bar[0].remap_addr);
+	if (chip->bar[1].remap_addr)
+		iounmap(chip->bar[1].remap_addr);
+	if (chip->rb.area)
+		snd_dma_free_pages(&chip->rb);
+	pci_release_regions(chip->pci);
+	pci_disable_device(chip->pci);
+	kfree(chip);
+}
+
+static int lola_dev_free(struct snd_device *device)
+{
+	lola_free(device->device_data);
+	return 0;
+}
+
+static int __devinit lola_create(struct snd_card *card, struct pci_dev *pci,
+				 int dev, struct lola **rchip)
+{
+	struct lola *chip;
+	int err;
+	unsigned int dever;
+	static struct snd_device_ops ops = {
+		.dev_free = lola_dev_free,
+	};
+
+	*rchip = NULL;
+
+	err = pci_enable_device(pci);
+	if (err < 0)
+		return err;
+
+	chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+	if (!chip) {
+		snd_printk(KERN_ERR SFX "cannot allocate chip\n");
+		pci_disable_device(pci);
+		return -ENOMEM;
+	}
+
+	spin_lock_init(&chip->reg_lock);
+	mutex_init(&chip->open_mutex);
+	chip->card = card;
+	chip->pci = pci;
+	chip->irq = -1;
+
+	chip->granularity = granularity[dev];
+	switch (chip->granularity) {
+	case 8:
+		chip->sample_rate_max = 48000;
+		break;
+	case 16:
+		chip->sample_rate_max = 96000;
+		break;
+	case 32:
+		chip->sample_rate_max = 192000;
+		break;
+	default:
+		snd_printk(KERN_WARNING SFX
+			   "Invalid granularity %d, reset to %d\n",
+			   chip->granularity, LOLA_GRANULARITY_MAX);
+		chip->granularity = LOLA_GRANULARITY_MAX;
+		chip->sample_rate_max = 192000;
+		break;
+	}
+	chip->sample_rate_min = sample_rate_min[dev];
+	if (chip->sample_rate_min > chip->sample_rate_max) {
+		snd_printk(KERN_WARNING SFX
+			   "Invalid sample_rate_min %d, reset to 16000\n",
+			   chip->sample_rate_min);
+		chip->sample_rate_min = 16000;
+	}
+
+	err = pci_request_regions(pci, DRVNAME);
+	if (err < 0) {
+		kfree(chip);
+		pci_disable_device(pci);
+		return err;
+	}
+
+	chip->bar[0].addr = pci_resource_start(pci, 0);
+	chip->bar[0].remap_addr = pci_ioremap_bar(pci, 0);
+	chip->bar[1].addr = pci_resource_start(pci, 2);
+	chip->bar[1].remap_addr = pci_ioremap_bar(pci, 2);
+	if (!chip->bar[0].remap_addr || !chip->bar[1].remap_addr) {
+		snd_printk(KERN_ERR SFX "ioremap error\n");
+		err = -ENXIO;
+		goto errout;
+	}
+
+	pci_set_master(pci);
+
+	err = reset_controller(chip);
+	if (err < 0)
+		goto errout;
+
+	if (request_irq(pci->irq, lola_interrupt, IRQF_SHARED,
+			DRVNAME, chip)) {
+		printk(KERN_ERR SFX "unable to grab IRQ %d\n", pci->irq);
+		err = -EBUSY;
+		goto errout;
+	}
+	chip->irq = pci->irq;
+	synchronize_irq(chip->irq);
+
+	dever = lola_readl(chip, BAR1, DEVER);
+	chip->pcm[CAPT].num_streams = (dever >> 0) & 0x3ff;
+	chip->pcm[PLAY].num_streams = (dever >> 10) & 0x3ff;
+	chip->version = (dever >> 24) & 0xff;
+	snd_printdd(SFX "streams in=%d, out=%d, version=0x%x\n",
+		    chip->pcm[CAPT].num_streams, chip->pcm[PLAY].num_streams,
+		    chip->version);
+
+	/* Test LOLA_BAR1_DEVER */
+	if (chip->pcm[CAPT].num_streams > MAX_STREAM_IN_COUNT ||
+	    chip->pcm[PLAY].num_streams > MAX_STREAM_OUT_COUNT ||
+	    (!chip->pcm[CAPT].num_streams &&
+	     !chip->pcm[PLAY].num_streams)) {
+		printk(KERN_ERR SFX "invalid DEVER = %x\n", dever);
+		err = -EINVAL;
+		goto errout;
+	}
+
+	err = setup_corb_rirb(chip);
+	if (err < 0)
+		goto errout;
+
+	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+	if (err < 0) {
+		snd_printk(KERN_ERR SFX "Error creating device [card]!\n");
+		goto errout;
+	}
+
+	strcpy(card->driver, "Lola");
+	strlcpy(card->shortname, "Digigram Lola", sizeof(card->shortname));
+	snprintf(card->longname, sizeof(card->longname),
+		 "%s at 0x%lx irq %i",
+		 card->shortname, chip->bar[0].addr, chip->irq);
+	strcpy(card->mixername, card->shortname);
+
+	lola_irq_enable(chip);
+
+	chip->initialized = 1;
+	*rchip = chip;
+	return 0;
+
+ errout:
+	lola_free(chip);
+	return err;
+}
+
+static int __devinit lola_probe(struct pci_dev *pci,
+				const struct pci_device_id *pci_id)
+{
+	static int dev;
+	struct snd_card *card;
+	struct lola *chip;
+	int err;
+
+	if (dev >= SNDRV_CARDS)
+		return -ENODEV;
+	if (!enable[dev]) {
+		dev++;
+		return -ENOENT;
+	}
+
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0) {
+		snd_printk(KERN_ERR SFX "Error creating card!\n");
+		return err;
+	}
+
+	snd_card_set_dev(card, &pci->dev);
+
+	err = lola_create(card, pci, dev, &chip);
+	if (err < 0)
+		goto out_free;
+	card->private_data = chip;
+
+	err = lola_parse_tree(chip);
+	if (err < 0)
+		goto out_free;
+
+	err = lola_create_pcm(chip);
+	if (err < 0)
+		goto out_free;
+
+	err = lola_create_mixer(chip);
+	if (err < 0)
+		goto out_free;
+
+	lola_proc_debug_new(chip);
+
+	err = snd_card_register(card);
+	if (err < 0)
+		goto out_free;
+
+	pci_set_drvdata(pci, card);
+	dev++;
+	return err;
+out_free:
+	snd_card_free(card);
+	return err;
+}
+
+static void __devexit lola_remove(struct pci_dev *pci)
+{
+	snd_card_free(pci_get_drvdata(pci));
+	pci_set_drvdata(pci, NULL);
+}
+
+/* PCI IDs */
+static DEFINE_PCI_DEVICE_TABLE(lola_ids) = {
+	{ PCI_VDEVICE(DIGIGRAM, 0x0001) },
+	{ 0, }
+};
+MODULE_DEVICE_TABLE(pci, lola_ids);
+
+/* pci_driver definition */
+static struct pci_driver driver = {
+	.name = DRVNAME,
+	.id_table = lola_ids,
+	.probe = lola_probe,
+	.remove = __devexit_p(lola_remove),
+};
+
+static int __init alsa_card_lola_init(void)
+{
+	return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_lola_exit(void)
+{
+	pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_lola_init)
+module_exit(alsa_card_lola_exit)
diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h
new file mode 100644
index 00000000000..d5708e29b16
--- /dev/null
+++ b/sound/pci/lola/lola.h
@@ -0,0 +1,527 @@
+/*
+ *  Support for Digigram Lola PCI-e boards
+ *
+ *  Copyright (c) 2011 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 _LOLA_H
+#define _LOLA_H
+
+#define DRVNAME	"snd-lola"
+#define SFX	DRVNAME ": "
+
+/*
+ * Lola HD Audio Registers BAR0
+ */
+#define LOLA_BAR0_GCAP		0x00
+#define LOLA_BAR0_VMIN		0x02
+#define LOLA_BAR0_VMAJ		0x03
+#define LOLA_BAR0_OUTPAY	0x04
+#define LOLA_BAR0_INPAY		0x06
+#define LOLA_BAR0_GCTL		0x08
+#define LOLA_BAR0_WAKEEN	0x0c
+#define LOLA_BAR0_STATESTS	0x0e
+#define LOLA_BAR0_GSTS		0x10
+#define LOLA_BAR0_OUTSTRMPAY	0x18
+#define LOLA_BAR0_INSTRMPAY	0x1a
+#define LOLA_BAR0_INTCTL	0x20
+#define LOLA_BAR0_INTSTS	0x24
+#define LOLA_BAR0_WALCLK	0x30
+#define LOLA_BAR0_SSYNC		0x38
+
+#define LOLA_BAR0_CORBLBASE	0x40
+#define LOLA_BAR0_CORBUBASE	0x44
+#define LOLA_BAR0_CORBWP	0x48	/* no ULONG access */
+#define LOLA_BAR0_CORBRP	0x4a	/* no ULONG access */
+#define LOLA_BAR0_CORBCTL	0x4c	/* no ULONG access */
+#define LOLA_BAR0_CORBSTS	0x4d	/* UCHAR access only */
+#define LOLA_BAR0_CORBSIZE	0x4e	/* no ULONG access */
+
+#define LOLA_BAR0_RIRBLBASE	0x50
+#define LOLA_BAR0_RIRBUBASE	0x54
+#define LOLA_BAR0_RIRBWP	0x58
+#define LOLA_BAR0_RINTCNT	0x5a	/* no ULONG access */
+#define LOLA_BAR0_RIRBCTL	0x5c
+#define LOLA_BAR0_RIRBSTS	0x5d	/* UCHAR access only */
+#define LOLA_BAR0_RIRBSIZE	0x5e	/* no ULONG access */
+
+#define LOLA_BAR0_ICW		0x60
+#define LOLA_BAR0_IRR		0x64
+#define LOLA_BAR0_ICS		0x68
+#define LOLA_BAR0_DPLBASE	0x70
+#define LOLA_BAR0_DPUBASE	0x74
+
+/* stream register offsets from stream base 0x80 */
+#define LOLA_BAR0_SD0_OFFSET	0x80
+#define LOLA_REG0_SD_CTL	0x00
+#define LOLA_REG0_SD_STS	0x03
+#define LOLA_REG0_SD_LPIB	0x04
+#define LOLA_REG0_SD_CBL	0x08
+#define LOLA_REG0_SD_LVI	0x0c
+#define LOLA_REG0_SD_FIFOW	0x0e
+#define LOLA_REG0_SD_FIFOSIZE	0x10
+#define LOLA_REG0_SD_FORMAT	0x12
+#define LOLA_REG0_SD_BDLPL	0x18
+#define LOLA_REG0_SD_BDLPU	0x1c
+
+/*
+ * Lola Digigram Registers BAR1
+ */
+#define LOLA_BAR1_FPGAVER	0x00
+#define LOLA_BAR1_DEVER		0x04
+#define LOLA_BAR1_UCBMV		0x08
+#define LOLA_BAR1_JTAG		0x0c
+#define LOLA_BAR1_UARTRX	0x10
+#define LOLA_BAR1_UARTTX	0x14
+#define LOLA_BAR1_UARTCR	0x18
+#define LOLA_BAR1_NVRAMVER	0x1c
+#define LOLA_BAR1_CTRLSPI	0x20
+#define LOLA_BAR1_DSPI		0x24
+#define LOLA_BAR1_AISPI		0x28
+#define LOLA_BAR1_GRAN		0x2c
+
+#define LOLA_BAR1_DINTCTL	0x80
+#define LOLA_BAR1_DIINTCTL	0x84
+#define LOLA_BAR1_DOINTCTL	0x88
+#define LOLA_BAR1_LRC		0x90
+#define LOLA_BAR1_DINTSTS	0x94
+#define LOLA_BAR1_DIINTSTS	0x98
+#define LOLA_BAR1_DOINTSTS	0x9c
+
+#define LOLA_BAR1_DSD0_OFFSET	0xa0
+#define LOLA_BAR1_DSD_SIZE	0x18
+
+#define LOLA_BAR1_DSDnSTS       0x00
+#define LOLA_BAR1_DSDnLPIB      0x04
+#define LOLA_BAR1_DSDnCTL       0x08
+#define LOLA_BAR1_DSDnLVI       0x0c
+#define LOLA_BAR1_DSDnBDPL      0x10
+#define LOLA_BAR1_DSDnBDPU      0x14
+
+#define LOLA_BAR1_SSYNC		0x03e8
+
+#define LOLA_BAR1_BOARD_CTRL	0x0f00
+#define LOLA_BAR1_BOARD_MODE	0x0f02
+
+#define LOLA_BAR1_SOURCE_GAIN_ENABLE		0x1000
+#define LOLA_BAR1_DEST00_MIX_GAIN_ENABLE	0x1004
+#define LOLA_BAR1_DEST31_MIX_GAIN_ENABLE	0x1080
+#define LOLA_BAR1_SOURCE00_01_GAIN		0x1084
+#define LOLA_BAR1_SOURCE30_31_GAIN		0x10c0
+#define LOLA_BAR1_SOURCE_GAIN(src) \
+	(LOLA_BAR1_SOURCE00_01_GAIN + (src) * 2)
+#define LOLA_BAR1_DEST00_MIX00_01_GAIN		0x10c4
+#define LOLA_BAR1_DEST00_MIX30_31_GAIN		0x1100
+#define LOLA_BAR1_DEST01_MIX00_01_GAIN		0x1104
+#define LOLA_BAR1_DEST01_MIX30_31_GAIN		0x1140
+#define LOLA_BAR1_DEST31_MIX00_01_GAIN		0x1884
+#define LOLA_BAR1_DEST31_MIX30_31_GAIN		0x18c0
+#define LOLA_BAR1_MIX_GAIN(dest, mix) \
+	(LOLA_BAR1_DEST00_MIX00_01_GAIN + (dest) * 0x40 + (mix) * 2)
+#define LOLA_BAR1_ANALOG_CLIP_IN		0x18c4
+#define LOLA_BAR1_PEAKMETERS_SOURCE00_01	0x18c8
+#define LOLA_BAR1_PEAKMETERS_SOURCE30_31	0x1904
+#define LOLA_BAR1_PEAKMETERS_SOURCE(src) \
+	(LOLA_BAR1_PEAKMETERS_SOURCE00_01 + (src) * 2)
+#define LOLA_BAR1_PEAKMETERS_DEST00_01		0x1908
+#define LOLA_BAR1_PEAKMETERS_DEST30_31		0x1944
+#define LOLA_BAR1_PEAKMETERS_DEST(dest) \
+	(LOLA_BAR1_PEAKMETERS_DEST00_01 + (dest) * 2)
+#define LOLA_BAR1_PEAKMETERS_AGC00_01		0x1948
+#define LOLA_BAR1_PEAKMETERS_AGC14_15		0x1964
+#define LOLA_BAR1_PEAKMETERS_AGC(x) \
+	(LOLA_BAR1_PEAKMETERS_AGC00_01 + (x) * 2)
+
+/* GCTL reset bit */
+#define LOLA_GCTL_RESET		(1 << 0)
+/* GCTL unsolicited response enable bit */
+#define LOLA_GCTL_UREN		(1 << 8)
+
+/* CORB/RIRB control, read/write pointer */
+#define LOLA_RBCTL_DMA_EN	0x02	/* enable DMA */
+#define LOLA_RBCTL_IRQ_EN	0x01	/* enable IRQ */
+#define LOLA_RBRWP_CLR		0x8000	/* read/write pointer clear */
+
+#define LOLA_RIRB_EX_UNSOL_EV	0x40000000
+#define LOLA_RIRB_EX_ERROR	0x80000000
+
+/* CORB int mask: CMEI[0] */
+#define LOLA_CORB_INT_CMEI	0x01
+#define LOLA_CORB_INT_MASK	LOLA_CORB_INT_CMEI
+
+/* RIRB int mask: overrun[2], response[0] */
+#define LOLA_RIRB_INT_RESPONSE	0x01
+#define LOLA_RIRB_INT_OVERRUN	0x04
+#define LOLA_RIRB_INT_MASK	(LOLA_RIRB_INT_RESPONSE | LOLA_RIRB_INT_OVERRUN)
+
+/* DINTCTL and DINTSTS */
+#define LOLA_DINT_GLOBAL	0x80000000 /* global interrupt enable bit */
+#define LOLA_DINT_CTRL		0x40000000 /* controller interrupt enable bit */
+#define LOLA_DINT_FIFOERR	0x20000000 /* global fifo error enable bit */
+#define LOLA_DINT_MUERR		0x10000000 /* global microcontroller underrun error */
+
+/* DSDnCTL bits */
+#define LOLA_DSD_CTL_SRST	0x01	/* stream reset bit */
+#define LOLA_DSD_CTL_SRUN	0x02	/* stream DMA start bit */
+#define LOLA_DSD_CTL_IOCE	0x04	/* interrupt on completion enable */
+#define LOLA_DSD_CTL_DEIE	0x10	/* descriptor error interrupt enable */
+#define LOLA_DSD_CTL_VLRCV	0x20	/* valid LRCountValue information in bits 8..31 */
+#define LOLA_LRC_MASK		0xffffff00
+
+/* DSDnSTS */
+#define LOLA_DSD_STS_BCIS	0x04	/* buffer completion interrupt status */
+#define LOLA_DSD_STS_DESE	0x10	/* descriptor error interrupt */
+#define LOLA_DSD_STS_FIFORDY	0x20	/* fifo ready */
+
+#define LOLA_CORB_ENTRIES	256
+
+#define MAX_STREAM_IN_COUNT	16
+#define MAX_STREAM_OUT_COUNT	16
+#define MAX_STREAM_COUNT	16
+#define MAX_PINS		MAX_STREAM_COUNT
+#define MAX_STREAM_BUFFER_COUNT	16
+#define MAX_AUDIO_INOUT_COUNT	16
+
+#define LOLA_CLOCK_TYPE_INTERNAL    0
+#define LOLA_CLOCK_TYPE_AES         1
+#define LOLA_CLOCK_TYPE_AES_SYNC    2
+#define LOLA_CLOCK_TYPE_WORDCLOCK   3
+#define LOLA_CLOCK_TYPE_ETHERSOUND  4
+#define LOLA_CLOCK_TYPE_VIDEO       5
+
+#define LOLA_CLOCK_FORMAT_NONE      0
+#define LOLA_CLOCK_FORMAT_NTSC      1
+#define LOLA_CLOCK_FORMAT_PAL       2
+
+#define MAX_SAMPLE_CLOCK_COUNT  48
+
+/* parameters used with mixer widget's mixer capabilities */
+#define LOLA_PEAK_METER_CAN_AGC_MASK		1
+#define LOLA_PEAK_METER_CAN_ANALOG_CLIP_MASK	2
+
+struct lola_bar {
+	unsigned long addr;
+	void __iomem *remap_addr;
+};
+
+/* CORB/RIRB */
+struct lola_rb {
+	u32 *buf;		/* CORB/RIRB buffer, 8 byte per each entry */
+	dma_addr_t addr;	/* physical address of CORB/RIRB buffer */
+	unsigned short rp, wp;	/* read/write pointers */
+	int cmds;		/* number of pending requests */
+};
+
+/* Pin widget setup */
+struct lola_pin {
+	unsigned int nid;
+	bool is_analog;
+	unsigned int amp_mute;
+	unsigned int amp_step_size;
+	unsigned int amp_num_steps;
+	unsigned int amp_offset;
+	unsigned int max_level;
+	unsigned int config_default_reg;
+	unsigned int fixed_gain_list_len;
+	unsigned int cur_gain_step;
+};
+
+struct lola_pin_array {
+	unsigned int num_pins;
+	unsigned int num_analog_pins;
+	struct lola_pin pins[MAX_PINS];
+};
+
+/* Clock widget setup */
+struct lola_sample_clock {
+	unsigned int type;
+	unsigned int format;
+	unsigned int freq;
+};
+
+struct lola_clock_widget {
+	unsigned int nid;
+	unsigned int items;
+	unsigned int cur_index;
+	unsigned int cur_freq;
+	bool cur_valid;
+	struct lola_sample_clock sample_clock[MAX_SAMPLE_CLOCK_COUNT];
+	unsigned int idx_lookup[MAX_SAMPLE_CLOCK_COUNT];
+};
+
+#define LOLA_MIXER_DIM      32
+struct lola_mixer_array {
+	u32 src_gain_enable;
+	u32 dest_mix_gain_enable[LOLA_MIXER_DIM];
+	u16 src_gain[LOLA_MIXER_DIM];
+	u16 dest_mix_gain[LOLA_MIXER_DIM][LOLA_MIXER_DIM];
+};
+
+/* Mixer widget setup */
+struct lola_mixer_widget {
+	unsigned int nid;
+	unsigned int caps;
+	struct lola_mixer_array __user *array;
+	struct lola_mixer_array *array_saved;
+	unsigned int src_stream_outs;
+	unsigned int src_phys_ins;
+	unsigned int dest_stream_ins;
+	unsigned int dest_phys_outs;
+	unsigned int src_stream_out_ofs;
+	unsigned int dest_phys_out_ofs;
+	unsigned int src_mask;
+	unsigned int dest_mask;
+};
+
+/* Audio stream */
+struct lola_stream {
+	unsigned int nid;	/* audio widget NID */
+	unsigned int index;	/* array index */
+	unsigned int dsd;	/* DSD index */
+	bool can_float;
+	struct snd_pcm_substream *substream; /* assigned PCM substream */
+	struct lola_stream *master;	/* master stream (for multi-channel) */
+
+	/* buffer setup */
+	unsigned int bufsize;
+	unsigned int period_bytes;
+	unsigned int frags;
+
+	/* format + channel setup */
+	unsigned int format_verb;
+
+	/* flags */
+	unsigned int opened:1;
+	unsigned int prepared:1;
+	unsigned int paused:1;
+	unsigned int running:1;
+};
+
+#define PLAY	SNDRV_PCM_STREAM_PLAYBACK
+#define CAPT	SNDRV_PCM_STREAM_CAPTURE
+
+struct lola_pcm {
+	unsigned int num_streams;
+	struct snd_dma_buffer bdl; /* BDL buffer */
+	struct lola_stream streams[MAX_STREAM_COUNT];
+};
+
+/* card instance */
+struct lola {
+	struct snd_card *card;
+	struct pci_dev *pci;
+
+	/* pci resources */
+	struct lola_bar bar[2];
+	int irq;
+
+	/* locks */
+	spinlock_t reg_lock;
+	struct mutex open_mutex;
+
+	/* CORB/RIRB */
+	struct lola_rb corb;
+	struct lola_rb rirb;
+	unsigned int res, res_ex;	/* last read values */
+	/* last command (for debugging) */
+	unsigned int last_cmd_nid, last_verb, last_data, last_extdata;
+
+	/* CORB/RIRB buffers */
+	struct snd_dma_buffer rb;
+
+	/* unsolicited events */
+	unsigned int last_unsol_res;
+
+	/* streams */
+	struct lola_pcm pcm[2];
+
+	/* input src */
+	unsigned int input_src_caps_mask;
+	unsigned int input_src_mask;
+
+	/* pins */
+	struct lola_pin_array pin[2];
+
+	/* clock */
+	struct lola_clock_widget clock;
+	int ref_count_rate;
+	unsigned int sample_rate;
+
+	/* mixer */
+	struct lola_mixer_widget mixer;
+
+	/* hw info */
+	unsigned int version;
+	unsigned int lola_caps;
+
+	/* parameters */
+	unsigned int granularity;
+	unsigned int sample_rate_min;
+	unsigned int sample_rate_max;
+
+	/* flags */
+	unsigned int initialized:1;
+	unsigned int cold_reset:1;
+	unsigned int polling_mode:1;
+
+	/* for debugging */
+	unsigned int debug_res;
+	unsigned int debug_res_ex;
+};
+
+#define BAR0	0
+#define BAR1	1
+
+/* Helper macros */
+#define lola_readl(chip, idx, name) \
+	readl((chip)->bar[idx].remap_addr + LOLA_##idx##_##name)
+#define lola_readw(chip, idx, name) \
+	readw((chip)->bar[idx].remap_addr + LOLA_##idx##_##name)
+#define lola_readb(chip, idx, name) \
+	readb((chip)->bar[idx].remap_addr + LOLA_##idx##_##name)
+#define lola_writel(chip, idx, name, val) \
+	writel((val), (chip)->bar[idx].remap_addr + LOLA_##idx##_##name)
+#define lola_writew(chip, idx, name, val) \
+	writew((val), (chip)->bar[idx].remap_addr + LOLA_##idx##_##name)
+#define lola_writeb(chip, idx, name, val) \
+	writeb((val), (chip)->bar[idx].remap_addr + LOLA_##idx##_##name)
+
+#define lola_dsd_read(chip, dsd, name) \
+	readl((chip)->bar[BAR1].remap_addr + LOLA_BAR1_DSD0_OFFSET + \
+	      (LOLA_BAR1_DSD_SIZE * (dsd)) + LOLA_BAR1_DSDn##name)
+#define lola_dsd_write(chip, dsd, name, val) \
+	writel((val), (chip)->bar[BAR1].remap_addr + LOLA_BAR1_DSD0_OFFSET + \
+	       (LOLA_BAR1_DSD_SIZE * (dsd)) + LOLA_BAR1_DSDn##name)
+
+/* GET verbs HDAudio */
+#define LOLA_VERB_GET_STREAM_FORMAT		0xa00
+#define LOLA_VERB_GET_AMP_GAIN_MUTE		0xb00
+#define LOLA_VERB_PARAMETERS			0xf00
+#define LOLA_VERB_GET_POWER_STATE		0xf05
+#define LOLA_VERB_GET_CONV			0xf06
+#define LOLA_VERB_GET_UNSOLICITED_RESPONSE	0xf08
+#define LOLA_VERB_GET_DIGI_CONVERT_1		0xf0d
+#define LOLA_VERB_GET_CONFIG_DEFAULT		0xf1c
+#define LOLA_VERB_GET_SUBSYSTEM_ID		0xf20
+/* GET verbs Digigram */
+#define LOLA_VERB_GET_FIXED_GAIN		0xfc0
+#define LOLA_VERB_GET_GAIN_SELECT		0xfc1
+#define LOLA_VERB_GET_MAX_LEVEL			0xfc2
+#define LOLA_VERB_GET_CLOCK_LIST		0xfc3
+#define LOLA_VERB_GET_CLOCK_SELECT		0xfc4
+#define LOLA_VERB_GET_CLOCK_STATUS		0xfc5
+
+/* SET verbs HDAudio */
+#define LOLA_VERB_SET_STREAM_FORMAT		0x200
+#define LOLA_VERB_SET_AMP_GAIN_MUTE		0x300
+#define LOLA_VERB_SET_POWER_STATE		0x705
+#define LOLA_VERB_SET_CHANNEL_STREAMID		0x706
+#define LOLA_VERB_SET_UNSOLICITED_ENABLE	0x708
+#define LOLA_VERB_SET_DIGI_CONVERT_1		0x70d
+/* SET verbs Digigram */
+#define LOLA_VERB_SET_GAIN_SELECT		0xf81
+#define LOLA_VERB_SET_CLOCK_SELECT		0xf84
+#define LOLA_VERB_SET_GRANULARITY_STEPS		0xf86
+#define LOLA_VERB_SET_SOURCE_GAIN		0xf87
+#define LOLA_VERB_SET_MIX_GAIN			0xf88
+#define LOLA_VERB_SET_DESTINATION_GAIN		0xf89
+#define LOLA_VERB_SET_SRC			0xf8a
+
+/* Parameter IDs used with LOLA_VERB_PARAMETERS */
+#define LOLA_PAR_VENDOR_ID			0x00
+#define LOLA_PAR_FUNCTION_TYPE			0x05
+#define LOLA_PAR_AUDIO_WIDGET_CAP		0x09
+#define LOLA_PAR_PCM				0x0a
+#define LOLA_PAR_STREAM_FORMATS			0x0b
+#define LOLA_PAR_PIN_CAP			0x0c
+#define LOLA_PAR_AMP_IN_CAP			0x0d
+#define LOLA_PAR_CONNLIST_LEN			0x0e
+#define LOLA_PAR_POWER_STATE			0x0f
+#define LOLA_PAR_GPIO_CAP			0x11
+#define LOLA_PAR_AMP_OUT_CAP			0x12
+#define LOLA_PAR_SPECIFIC_CAPS			0x80
+#define LOLA_PAR_FIXED_GAIN_LIST		0x81
+
+/* extract results of LOLA_PAR_SPECIFIC_CAPS */
+#define LOLA_AFG_MIXER_WIDGET_PRESENT(res)	((res & (1 << 21)) != 0)
+#define LOLA_AFG_CLOCK_WIDGET_PRESENT(res)	((res & (1 << 20)) != 0)
+#define LOLA_AFG_INPUT_PIN_COUNT(res)		((res >> 10) & 0x2ff)
+#define LOLA_AFG_OUTPUT_PIN_COUNT(res)		((res) & 0x2ff)
+
+/* extract results of LOLA_PAR_AMP_IN_CAP / LOLA_PAR_AMP_OUT_CAP */
+#define LOLA_AMP_MUTE_CAPABLE(res)		((res & (1 << 31)) != 0)
+#define LOLA_AMP_STEP_SIZE(res)			((res >> 24) & 0x7f)
+#define LOLA_AMP_NUM_STEPS(res)			((res >> 12) & 0x3ff)
+#define LOLA_AMP_OFFSET(res)			((res) & 0x3ff)
+
+#define LOLA_GRANULARITY_MIN		8
+#define LOLA_GRANULARITY_MAX		32
+#define LOLA_GRANULARITY_STEP		8
+
+/* parameters used with unsolicited command/response */
+#define LOLA_UNSOLICITED_TAG_MASK	0x3f
+#define LOLA_UNSOLICITED_TAG		0x1a
+#define LOLA_UNSOLICITED_ENABLE		0x80
+#define LOLA_UNSOL_RESP_TAG_OFFSET	26
+
+/* count values in the Vendor Specific Mixer Widget's Audio Widget Capabilities */
+#define LOLA_MIXER_SRC_INPUT_PLAY_SEPARATION(res)   ((res >> 2) & 0x1f)
+#define LOLA_MIXER_DEST_REC_OUTPUT_SEPATATION(res)  ((res >> 7) & 0x1f)
+
+int lola_codec_write(struct lola *chip, unsigned int nid, unsigned int verb,
+		     unsigned int data, unsigned int extdata);
+int lola_codec_read(struct lola *chip, unsigned int nid, unsigned int verb,
+		    unsigned int data, unsigned int extdata,
+		    unsigned int *val, unsigned int *extval);
+int lola_codec_flush(struct lola *chip);
+#define lola_read_param(chip, nid, param, val) \
+	lola_codec_read(chip, nid, LOLA_VERB_PARAMETERS, param, 0, val, NULL)
+
+/* PCM */
+int lola_create_pcm(struct lola *chip);
+void lola_free_pcm(struct lola *chip);
+int lola_init_pcm(struct lola *chip, int dir, int *nidp);
+void lola_pcm_update(struct lola *chip, struct lola_pcm *pcm, unsigned int bits);
+
+/* clock */
+int lola_init_clock_widget(struct lola *chip, int nid);
+int lola_set_granularity(struct lola *chip, unsigned int val, bool force);
+int lola_enable_clock_events(struct lola *chip);
+int lola_set_clock_index(struct lola *chip, unsigned int idx);
+int lola_set_clock(struct lola *chip, int idx);
+int lola_set_sample_rate(struct lola *chip, int rate);
+bool lola_update_ext_clock_freq(struct lola *chip, unsigned int val);
+unsigned int lola_sample_rate_convert(unsigned int coded);
+
+/* mixer */
+int lola_init_pins(struct lola *chip, int dir, int *nidp);
+int lola_init_mixer_widget(struct lola *chip, int nid);
+void lola_free_mixer(struct lola *chip);
+int lola_create_mixer(struct lola *chip);
+int lola_setup_all_analog_gains(struct lola *chip, int dir, bool mute);
+void lola_save_mixer(struct lola *chip);
+void lola_restore_mixer(struct lola *chip);
+int lola_set_src_config(struct lola *chip, unsigned int src_mask, bool update);
+
+/* proc */
+#ifdef CONFIG_SND_DEBUG
+void lola_proc_debug_new(struct lola *chip);
+#else
+#define lola_proc_debug_new(chip)
+#endif
+
+#endif /* _LOLA_H */
diff --git a/sound/pci/lola/lola_clock.c b/sound/pci/lola/lola_clock.c
new file mode 100644
index 00000000000..72f8ef0ac86
--- /dev/null
+++ b/sound/pci/lola/lola_clock.c
@@ -0,0 +1,323 @@
+/*
+ *  Support for Digigram Lola PCI-e boards
+ *
+ *  Copyright (c) 2011 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.
+ */
+
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include "lola.h"
+
+unsigned int lola_sample_rate_convert(unsigned int coded)
+{
+	unsigned int freq;
+
+	/* base frequency */
+	switch (coded & 0x3) {
+	case 0:     freq = 48000; break;
+	case 1:     freq = 44100; break;
+	case 2:     freq = 32000; break;
+	default:    return 0;   /* error */
+	}
+
+	/* multiplier / devisor */
+	switch (coded & 0x1c) {
+	case (0 << 2):    break;
+	case (4 << 2):    break;
+	case (1 << 2):    freq *= 2; break;
+	case (2 << 2):    freq *= 4; break;
+	case (5 << 2):    freq /= 2; break;
+	case (6 << 2):    freq /= 4; break;
+	default:        return 0;   /* error */
+	}
+
+	/* ajustement */
+	switch (coded & 0x60) {
+	case (0 << 5):    break;
+	case (1 << 5):    freq = (freq * 999) / 1000; break;
+	case (2 << 5):    freq = (freq * 1001) / 1000; break;
+	default:        return 0;   /* error */
+	}
+	return freq;
+}
+
+/*
+ * Granualrity
+ */
+
+#define LOLA_MAXFREQ_AT_GRANULARITY_MIN         48000
+#define LOLA_MAXFREQ_AT_GRANULARITY_BELOW_MAX   96000
+
+static bool check_gran_clock_compatibility(struct lola *chip,
+					   unsigned int val,
+					   unsigned int freq)
+{
+	if (!chip->granularity)
+		return true;
+
+	if (val < LOLA_GRANULARITY_MIN || val > LOLA_GRANULARITY_MAX ||
+	    (val % LOLA_GRANULARITY_STEP) != 0)
+		return false;
+
+	if (val == LOLA_GRANULARITY_MIN) {
+		if (freq > LOLA_MAXFREQ_AT_GRANULARITY_MIN)
+			return false;
+	} else if (val < LOLA_GRANULARITY_MAX) {
+		if (freq > LOLA_MAXFREQ_AT_GRANULARITY_BELOW_MAX)
+			return false;
+	}
+	return true;
+}
+
+int lola_set_granularity(struct lola *chip, unsigned int val, bool force)
+{
+	int err;
+
+	if (!force) {
+		if (val == chip->granularity)
+			return 0;
+#if 0
+		/* change Gran only if there are no streams allocated ! */
+		if (chip->audio_in_alloc_mask || chip->audio_out_alloc_mask)
+			return -EBUSY;
+#endif
+		if (!check_gran_clock_compatibility(chip, val,
+						    chip->clock.cur_freq))
+			return -EINVAL;
+	}
+
+	chip->granularity = val;
+	val /= LOLA_GRANULARITY_STEP;
+
+	/* audio function group */
+	err = lola_codec_write(chip, 1, LOLA_VERB_SET_GRANULARITY_STEPS,
+			       val, 0);
+	if (err < 0)
+		return err;
+	/* this can be a very slow function !!! */
+	usleep_range(400 * val, 20000);
+	return lola_codec_flush(chip);
+}
+
+/*
+ * Clock widget handling
+ */
+
+int __devinit lola_init_clock_widget(struct lola *chip, int nid)
+{
+	unsigned int val;
+	int i, j, nitems, nb_verbs, idx, idx_list;
+	int err;
+
+	err = lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val);
+	if (err < 0) {
+		printk(KERN_ERR SFX "Can't read wcaps for 0x%x\n", nid);
+		return err;
+	}
+
+	if ((val & 0xfff00000) != 0x01f00000) { /* test SubType and Type */
+		snd_printdd("No valid clock widget\n");
+		return 0;
+	}
+
+	chip->clock.nid = nid;
+	chip->clock.items = val & 0xff;
+	snd_printdd("clock_list nid=%x, entries=%d\n", nid,
+		    chip->clock.items);
+	if (chip->clock.items > MAX_SAMPLE_CLOCK_COUNT) {
+		printk(KERN_ERR SFX "CLOCK_LIST too big: %d\n",
+		       chip->clock.items);
+		return -EINVAL;
+	}
+
+	nitems = chip->clock.items;
+	nb_verbs = (nitems + 3) / 4;
+	idx = 0;
+	idx_list = 0;
+	for (i = 0; i < nb_verbs; i++) {
+		unsigned int res_ex;
+		unsigned short items[4];
+
+		err = lola_codec_read(chip, nid, LOLA_VERB_GET_CLOCK_LIST,
+				      idx, 0, &val, &res_ex);
+		if (err < 0) {
+			printk(KERN_ERR SFX "Can't read CLOCK_LIST\n");
+			return -EINVAL;
+		}
+
+		items[0] = val & 0xfff;
+		items[1] = (val >> 16) & 0xfff;
+		items[2] = res_ex & 0xfff;
+		items[3] = (res_ex >> 16) & 0xfff;
+
+		for (j = 0; j < 4; j++) {
+			unsigned char type = items[j] >> 8;
+			unsigned int freq = items[j] & 0xff;
+			int format = LOLA_CLOCK_FORMAT_NONE;
+			bool add_clock = true;
+			if (type == LOLA_CLOCK_TYPE_INTERNAL) {
+				freq = lola_sample_rate_convert(freq);
+				if (freq < chip->sample_rate_min)
+					add_clock = false;
+				else if (freq == 48000) {
+					chip->clock.cur_index = idx_list;
+					chip->clock.cur_freq = 48000;
+					chip->clock.cur_valid = true;
+				}
+			} else if (type == LOLA_CLOCK_TYPE_VIDEO) {
+				freq = lola_sample_rate_convert(freq);
+				if (freq < chip->sample_rate_min)
+					add_clock = false;
+				/* video clock has a format (0:NTSC, 1:PAL)*/
+				if (items[j] & 0x80)
+					format = LOLA_CLOCK_FORMAT_NTSC;
+				else
+					format = LOLA_CLOCK_FORMAT_PAL;
+			}
+			if (add_clock) {
+				struct lola_sample_clock *sc;
+				sc = &chip->clock.sample_clock[idx_list];
+				sc->type = type;
+				sc->format = format;
+				sc->freq = freq;
+				/* keep the index used with the board */
+				chip->clock.idx_lookup[idx_list] = idx;
+				idx_list++;
+			} else {
+				chip->clock.items--;
+			}
+			if (++idx >= nitems)
+				break;
+		}
+	}
+	return 0;
+}
+
+/* enable unsolicited events of the clock widget */
+int lola_enable_clock_events(struct lola *chip)
+{
+	unsigned int res;
+	int err;
+
+	err = lola_codec_read(chip, chip->clock.nid,
+			      LOLA_VERB_SET_UNSOLICITED_ENABLE,
+			      LOLA_UNSOLICITED_ENABLE | LOLA_UNSOLICITED_TAG,
+			      0, &res, NULL);
+	if (err < 0)
+		return err;
+	if (res) {
+		printk(KERN_WARNING SFX "error in enable_clock_events %d\n",
+		       res);
+		return -EINVAL;
+	}
+	return 0;
+}
+
+int lola_set_clock_index(struct lola *chip, unsigned int idx)
+{
+	unsigned int res;
+	int err;
+
+	err = lola_codec_read(chip, chip->clock.nid,
+			      LOLA_VERB_SET_CLOCK_SELECT,
+			      chip->clock.idx_lookup[idx],
+			      0, &res, NULL);
+	if (err < 0)
+		return err;
+	if (res) {
+		printk(KERN_WARNING SFX "error in set_clock %d\n", res);
+		return -EINVAL;
+	}
+	return 0;
+}
+
+bool lola_update_ext_clock_freq(struct lola *chip, unsigned int val)
+{
+	unsigned int tag;
+
+	/* the current EXTERNAL clock information gets updated by interrupt
+	 * with an unsolicited response
+	 */
+	if (!val)
+		return false;
+	tag = (val >> LOLA_UNSOL_RESP_TAG_OFFSET) & LOLA_UNSOLICITED_TAG_MASK;
+	if (tag != LOLA_UNSOLICITED_TAG)
+		return false;
+
+	/* only for current = external clocks */
+	if (chip->clock.sample_clock[chip->clock.cur_index].type !=
+	    LOLA_CLOCK_TYPE_INTERNAL) {
+		chip->clock.cur_freq = lola_sample_rate_convert(val & 0x7f);
+		chip->clock.cur_valid = (val & 0x100) != 0;
+	}
+	return true;
+}
+
+int lola_set_clock(struct lola *chip, int idx)
+{
+	int freq = 0;
+	bool valid = false;
+
+	if (idx == chip->clock.cur_index) {
+		/* current clock is allowed */
+		freq = chip->clock.cur_freq;
+		valid = chip->clock.cur_valid;
+	} else if (chip->clock.sample_clock[idx].type ==
+		   LOLA_CLOCK_TYPE_INTERNAL) {
+		/* internal clocks allowed */
+		freq = chip->clock.sample_clock[idx].freq;
+		valid = true;
+	}
+
+	if (!freq || !valid)
+		return -EINVAL;
+
+	if (!check_gran_clock_compatibility(chip, chip->granularity, freq))
+		return -EINVAL;
+
+	if (idx != chip->clock.cur_index) {
+		int err = lola_set_clock_index(chip, idx);
+		if (err < 0)
+			return err;
+		/* update new settings */
+		chip->clock.cur_index = idx;
+		chip->clock.cur_freq = freq;
+		chip->clock.cur_valid = true;
+	}
+	return 0;
+}
+
+int lola_set_sample_rate(struct lola *chip, int rate)
+{
+	int i;
+
+	if (chip->clock.cur_freq == rate && chip->clock.cur_valid)
+		return 0;
+	/* search for new dwClockIndex */
+	for (i = 0; i < chip->clock.items; i++) {
+		if (chip->clock.sample_clock[i].type == LOLA_CLOCK_TYPE_INTERNAL &&
+		    chip->clock.sample_clock[i].freq == rate)
+			break;
+	}
+	if (i >= chip->clock.items)
+		return -EINVAL;
+	return lola_set_clock(chip, i);
+}
+
diff --git a/sound/pci/lola/lola_mixer.c b/sound/pci/lola/lola_mixer.c
new file mode 100644
index 00000000000..5d518f1a712
--- /dev/null
+++ b/sound/pci/lola/lola_mixer.c
@@ -0,0 +1,839 @@
+/*
+ *  Support for Digigram Lola PCI-e boards
+ *
+ *  Copyright (c) 2011 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.
+ */
+
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/vmalloc.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/tlv.h>
+#include "lola.h"
+
+static int __devinit lola_init_pin(struct lola *chip, struct lola_pin *pin,
+				   int dir, int nid)
+{
+	unsigned int val;
+	int err;
+
+	pin->nid = nid;
+	err = lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val);
+	if (err < 0) {
+		printk(KERN_ERR SFX "Can't read wcaps for 0x%x\n", nid);
+		return err;
+	}
+	val &= 0x00f00fff; /* test TYPE and bits 0..11 */
+	if (val == 0x00400200)    /* Type = 4, Digital = 1 */
+		pin->is_analog = false;
+	else if (val == 0x0040000a && dir == CAPT) /* Dig=0, InAmp/ovrd */
+		pin->is_analog = true;
+	else if (val == 0x0040000c && dir == PLAY) /* Dig=0, OutAmp/ovrd */
+		pin->is_analog = true;
+	else {
+		printk(KERN_ERR SFX "Invalid wcaps 0x%x for 0x%x\n", val, nid);
+		return -EINVAL;
+	}
+
+	/* analog parameters only following, so continue in case of Digital pin
+	 */
+	if (!pin->is_analog)
+		return 0;
+
+	if (dir == PLAY)
+		err = lola_read_param(chip, nid, LOLA_PAR_AMP_OUT_CAP, &val);
+	else
+		err = lola_read_param(chip, nid, LOLA_PAR_AMP_IN_CAP, &val);
+	if (err < 0) {
+		printk(KERN_ERR SFX "Can't read AMP-caps for 0x%x\n", nid);
+		return err;
+	}
+
+	pin->amp_mute = LOLA_AMP_MUTE_CAPABLE(val);
+	pin->amp_step_size = LOLA_AMP_STEP_SIZE(val);
+	pin->amp_num_steps = LOLA_AMP_NUM_STEPS(val);
+	if (pin->amp_num_steps) {
+		/* zero as mute state */
+		pin->amp_num_steps++;
+		pin->amp_step_size++;
+	}
+	pin->amp_offset = LOLA_AMP_OFFSET(val);
+
+	err = lola_codec_read(chip, nid, LOLA_VERB_GET_MAX_LEVEL, 0, 0, &val,
+			      NULL);
+	if (err < 0) {
+		printk(KERN_ERR SFX "Can't get MAX_LEVEL 0x%x\n", nid);
+		return err;
+	}
+	pin->max_level = val & 0x3ff;   /* 10 bits */
+
+	pin->config_default_reg = 0;
+	pin->fixed_gain_list_len = 0;
+	pin->cur_gain_step = 0;
+
+	return 0;
+}
+
+int __devinit lola_init_pins(struct lola *chip, int dir, int *nidp)
+{
+	int i, err, nid;
+	nid = *nidp;
+	for (i = 0; i < chip->pin[dir].num_pins; i++, nid++) {
+		err = lola_init_pin(chip, &chip->pin[dir].pins[i], dir, nid);
+		if (err < 0)
+			return err;
+		if (chip->pin[dir].pins[i].is_analog)
+			chip->pin[dir].num_analog_pins++;
+	}
+	*nidp = nid;
+	return 0;
+}
+
+void lola_free_mixer(struct lola *chip)
+{
+	if (chip->mixer.array_saved)
+		vfree(chip->mixer.array_saved);
+}
+
+int __devinit lola_init_mixer_widget(struct lola *chip, int nid)
+{
+	unsigned int val;
+	int err;
+
+	err = lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val);
+	if (err < 0) {
+		printk(KERN_ERR SFX "Can't read wcaps for 0x%x\n", nid);
+		return err;
+	}
+
+	if ((val & 0xfff00000) != 0x02f00000) { /* test SubType and Type */
+		snd_printdd("No valid mixer widget\n");
+		return 0;
+	}
+
+	chip->mixer.nid = nid;
+	chip->mixer.caps = val;
+	chip->mixer.array = (struct lola_mixer_array __iomem *)
+		(chip->bar[BAR1].remap_addr + LOLA_BAR1_SOURCE_GAIN_ENABLE);
+
+	/* reserve memory to copy mixer data for sleep mode transitions */
+	chip->mixer.array_saved = vmalloc(sizeof(struct lola_mixer_array));
+
+	/* mixer matrix sources are physical input data and play streams */
+	chip->mixer.src_stream_outs = chip->pcm[PLAY].num_streams;
+	chip->mixer.src_phys_ins = chip->pin[CAPT].num_pins;
+
+	/* mixer matrix destinations are record streams and physical output */
+	chip->mixer.dest_stream_ins = chip->pcm[CAPT].num_streams;
+	chip->mixer.dest_phys_outs = chip->pin[PLAY].num_pins;
+
+	/* mixer matrix can have unused areas between PhysIn and
+	 * Play or Record and PhysOut zones
+	 */
+	chip->mixer.src_stream_out_ofs = chip->mixer.src_phys_ins +
+		LOLA_MIXER_SRC_INPUT_PLAY_SEPARATION(val);
+	chip->mixer.dest_phys_out_ofs = chip->mixer.dest_stream_ins +
+		LOLA_MIXER_DEST_REC_OUTPUT_SEPATATION(val);
+
+	/* example : MixerMatrix of LoLa881
+	 * 0-------8------16-------8------16
+	 * |       |       |       |       |
+	 * | INPUT |       | INPUT |       |
+	 * | ->    |unused | ->    |unused |
+	 * | RECORD|       | OUTPUT|       |
+	 * |       |       |       |       |
+	 * 8--------------------------------
+	 * |       |       |       |       |
+	 * |       |       |       |       |
+	 * |unused |unused |unused |unused |
+	 * |       |       |       |       |
+	 * |       |       |       |       |
+	 * 16-------------------------------
+	 * |       |       |       |       |
+	 * | PLAY  |       | PLAY  |       |
+	 * |  ->   |unused | ->    |unused |
+	 * | RECORD|       | OUTPUT|       |
+	 * |       |       |       |       |
+	 * 8--------------------------------
+	 * |       |       |       |       |
+	 * |       |       |       |       |
+	 * |unused |unused |unused |unused |
+	 * |       |       |       |       |
+	 * |       |       |       |       |
+	 * 16-------------------------------
+	 */
+	if (chip->mixer.src_stream_out_ofs > MAX_AUDIO_INOUT_COUNT ||
+	    chip->mixer.dest_phys_out_ofs > MAX_STREAM_IN_COUNT) {
+		printk(KERN_ERR SFX "Invalid mixer widget size\n");
+		return -EINVAL;
+	}
+
+	chip->mixer.src_mask = ((1U << chip->mixer.src_phys_ins) - 1) |
+		(((1U << chip->mixer.src_stream_outs) - 1)
+		 << chip->mixer.src_stream_out_ofs);
+	chip->mixer.dest_mask = ((1U << chip->mixer.dest_stream_ins) - 1) |
+		(((1U << chip->mixer.dest_phys_outs) - 1)
+		 << chip->mixer.dest_phys_out_ofs);
+
+	return 0;
+}
+
+static int lola_mixer_set_src_gain(struct lola *chip, unsigned int id,
+				   unsigned short gain, bool on)
+{
+	unsigned int oldval, val;
+
+	if (!(chip->mixer.src_mask & (1 << id)))
+		return -EINVAL;
+	writew(gain, &chip->mixer.array->src_gain[id]);
+	oldval = val = readl(&chip->mixer.array->src_gain_enable);
+	if (on)
+		val |= (1 << id);
+	else
+		val &= ~(1 << id);
+	writel(val, &chip->mixer.array->src_gain_enable);
+	lola_codec_flush(chip);
+	/* inform micro-controller about the new source gain */
+	return lola_codec_write(chip, chip->mixer.nid,
+				LOLA_VERB_SET_SOURCE_GAIN, id, 0);
+}
+
+#if 0 /* not used */
+static int lola_mixer_set_src_gains(struct lola *chip, unsigned int mask,
+				    unsigned short *gains)
+{
+	int i;
+
+	if ((chip->mixer.src_mask & mask) != mask)
+		return -EINVAL;
+	for (i = 0; i < LOLA_MIXER_DIM; i++) {
+		if (mask & (1 << i)) {
+			writew(*gains, &chip->mixer.array->src_gain[i]);
+			gains++;
+		}
+	}
+	writel(mask, &chip->mixer.array->src_gain_enable);
+	lola_codec_flush(chip);
+	if (chip->mixer.caps & LOLA_PEAK_METER_CAN_AGC_MASK) {
+		/* update for all srcs at once */
+		return lola_codec_write(chip, chip->mixer.nid,
+					LOLA_VERB_SET_SOURCE_GAIN, 0x80, 0);
+	}
+	/* update manually */
+	for (i = 0; i < LOLA_MIXER_DIM; i++) {
+		if (mask & (1 << i)) {
+			lola_codec_write(chip, chip->mixer.nid,
+					 LOLA_VERB_SET_SOURCE_GAIN, i, 0);
+		}
+	}
+	return 0;
+}
+#endif /* not used */
+
+static int lola_mixer_set_mapping_gain(struct lola *chip,
+				       unsigned int src, unsigned int dest,
+				       unsigned short gain, bool on)
+{
+	unsigned int val;
+
+	if (!(chip->mixer.src_mask & (1 << src)) ||
+	    !(chip->mixer.dest_mask & (1 << dest)))
+		return -EINVAL;
+	if (on)
+		writew(gain, &chip->mixer.array->dest_mix_gain[dest][src]);
+	val = readl(&chip->mixer.array->dest_mix_gain_enable[dest]);
+	if (on)
+		val |= (1 << src);
+	else
+		val &= ~(1 << src);
+	writel(val, &chip->mixer.array->dest_mix_gain_enable[dest]);
+	lola_codec_flush(chip);
+	return lola_codec_write(chip, chip->mixer.nid, LOLA_VERB_SET_MIX_GAIN,
+				src, dest);
+}
+
+static int lola_mixer_set_dest_gains(struct lola *chip, unsigned int id,
+				     unsigned int mask, unsigned short *gains)
+{
+	int i;
+
+	if (!(chip->mixer.dest_mask & (1 << id)) ||
+	    (chip->mixer.src_mask & mask) != mask)
+		return -EINVAL;
+	for (i = 0; i < LOLA_MIXER_DIM; i++) {
+		if (mask & (1 << i)) {
+			writew(*gains, &chip->mixer.array->dest_mix_gain[id][i]);
+			gains++;
+		}
+	}
+	writel(mask, &chip->mixer.array->dest_mix_gain_enable[id]);
+	lola_codec_flush(chip);
+	/* update for all dests at once */
+	return lola_codec_write(chip, chip->mixer.nid,
+				LOLA_VERB_SET_DESTINATION_GAIN, id, 0);
+}
+
+/*
+ */
+
+static int set_analog_volume(struct lola *chip, int dir,
+			     unsigned int idx, unsigned int val,
+			     bool external_call);
+
+int lola_setup_all_analog_gains(struct lola *chip, int dir, bool mute)
+{
+	struct lola_pin *pin;
+	int idx, max_idx;
+
+	pin = chip->pin[dir].pins;
+	max_idx = chip->pin[dir].num_pins;
+	for (idx = 0; idx < max_idx; idx++) {
+		if (pin[idx].is_analog) {
+			unsigned int val = mute ? 0 : pin[idx].cur_gain_step;
+			/* set volume and do not save the value */
+			set_analog_volume(chip, dir, idx, val, false);
+		}
+	}
+	return lola_codec_flush(chip);
+}
+
+void lola_save_mixer(struct lola *chip)
+{
+	/* mute analog output */
+	if (chip->mixer.array_saved) {
+		/* store contents of mixer array */
+		memcpy_fromio(chip->mixer.array_saved, chip->mixer.array,
+			      sizeof(*chip->mixer.array));
+	}
+	lola_setup_all_analog_gains(chip, PLAY, true); /* output mute */
+}
+
+void lola_restore_mixer(struct lola *chip)
+{
+	int i;
+
+	/*lola_reset_setups(chip);*/
+	if (chip->mixer.array_saved) {
+		/* restore contents of mixer array */
+		memcpy_toio(chip->mixer.array, chip->mixer.array_saved,
+			    sizeof(*chip->mixer.array));
+		/* inform micro-controller about all restored values
+		 * and ignore return values
+		 */
+		for (i = 0; i < chip->mixer.src_phys_ins; i++)
+			lola_codec_write(chip, chip->mixer.nid,
+					 LOLA_VERB_SET_SOURCE_GAIN,
+					 i, 0);
+		for (i = 0; i < chip->mixer.src_stream_outs; i++)
+			lola_codec_write(chip, chip->mixer.nid,
+					 LOLA_VERB_SET_SOURCE_GAIN,
+					 chip->mixer.src_stream_out_ofs + i, 0);
+		for (i = 0; i < chip->mixer.dest_stream_ins; i++)
+			lola_codec_write(chip, chip->mixer.nid,
+					 LOLA_VERB_SET_DESTINATION_GAIN,
+					 i, 0);
+		for (i = 0; i < chip->mixer.dest_phys_outs; i++)
+			lola_codec_write(chip, chip->mixer.nid,
+					 LOLA_VERB_SET_DESTINATION_GAIN,
+					 chip->mixer.dest_phys_out_ofs + i, 0);
+		lola_codec_flush(chip);
+	}
+}
+
+/*
+ */
+
+static int set_analog_volume(struct lola *chip, int dir,
+			     unsigned int idx, unsigned int val,
+			     bool external_call)
+{
+	struct lola_pin *pin;
+	int err;
+
+	if (idx >= chip->pin[dir].num_pins)
+		return -EINVAL;
+	pin = &chip->pin[dir].pins[idx];
+	if (!pin->is_analog || pin->amp_num_steps <= val)
+		return -EINVAL;
+	if (external_call && pin->cur_gain_step == val)
+		return 0;
+	if (external_call)
+		lola_codec_flush(chip);
+	err = lola_codec_write(chip, pin->nid,
+			       LOLA_VERB_SET_AMP_GAIN_MUTE, val, 0);
+	if (err < 0)
+		return err;
+	if (external_call)
+		pin->cur_gain_step = val;
+	return 0;
+}
+
+int lola_set_src_config(struct lola *chip, unsigned int src_mask, bool update)
+{
+	int ret = 0;
+	int success = 0;
+	int n, err;
+
+	/* SRC can be activated and the dwInputSRCMask is valid? */
+	if ((chip->input_src_caps_mask & src_mask) != src_mask)
+		return -EINVAL;
+	/* handle all even Inputs - SRC is a stereo setting !!! */
+	for (n = 0; n < chip->pin[CAPT].num_pins; n += 2) {
+		unsigned int mask = 3U << n; /* handle the stereo case */
+		unsigned int new_src, src_state;
+		if (!(chip->input_src_caps_mask & mask))
+			continue;
+		/* if one IO needs SRC, both stereo IO will get SRC */
+		new_src = (src_mask & mask) != 0;
+		if (update) {
+			src_state = (chip->input_src_mask & mask) != 0;
+			if (src_state == new_src)
+				continue;   /* nothing to change for this IO */
+		}
+		err = lola_codec_write(chip, chip->pcm[CAPT].streams[n].nid,
+				       LOLA_VERB_SET_SRC, new_src, 0);
+		if (!err)
+			success++;
+		else
+			ret = err;
+	}
+	if (success)
+		ret = lola_codec_flush(chip);
+	if (!ret)
+		chip->input_src_mask = src_mask;
+	return ret;
+}
+
+/*
+ */
+static int init_mixer_values(struct lola *chip)
+{
+	int i;
+
+	/* all src on */
+	lola_set_src_config(chip, (1 << chip->pin[CAPT].num_pins) - 1, false);
+
+	/* clear all matrix */
+	memset_io(chip->mixer.array, 0, sizeof(*chip->mixer.array));
+	/* set src gain to 0dB */
+	for (i = 0; i < chip->mixer.src_phys_ins; i++)
+		lola_mixer_set_src_gain(chip, i, 336, true); /* 0dB */
+	for (i = 0; i < chip->mixer.src_stream_outs; i++)
+		lola_mixer_set_src_gain(chip,
+					i + chip->mixer.src_stream_out_ofs,
+					336, true); /* 0dB */
+	/* set 1:1 dest gain */
+	for (i = 0; i < chip->mixer.dest_stream_ins; i++) {
+		int src = i % chip->mixer.src_phys_ins;
+		lola_mixer_set_mapping_gain(chip, src, i, 336, true);
+	}
+	for (i = 0; i < chip->mixer.src_stream_outs; i++) {
+		int src = chip->mixer.src_stream_out_ofs + i;
+		int dst = chip->mixer.dest_phys_out_ofs +
+			i % chip->mixer.dest_phys_outs;
+		lola_mixer_set_mapping_gain(chip, src, dst, 336, true);
+	}
+	return 0;
+}
+
+/*
+ * analog mixer control element
+ */
+static int lola_analog_vol_info(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_info *uinfo)
+{
+	struct lola *chip = snd_kcontrol_chip(kcontrol);
+	int dir = kcontrol->private_value;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = chip->pin[dir].num_pins;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = chip->pin[dir].pins[0].amp_num_steps;
+	return 0;
+}
+
+static int lola_analog_vol_get(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct lola *chip = snd_kcontrol_chip(kcontrol);
+	int dir = kcontrol->private_value;
+	int i;
+
+	for (i = 0; i < chip->pin[dir].num_pins; i++)
+		ucontrol->value.integer.value[i] =
+			chip->pin[dir].pins[i].cur_gain_step;
+	return 0;
+}
+
+static int lola_analog_vol_put(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct lola *chip = snd_kcontrol_chip(kcontrol);
+	int dir = kcontrol->private_value;
+	int i, err;
+
+	for (i = 0; i < chip->pin[dir].num_pins; i++) {
+		err = set_analog_volume(chip, dir, i,
+					ucontrol->value.integer.value[i],
+					true);
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
+
+static int lola_analog_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+			       unsigned int size, unsigned int __user *tlv)
+{
+	struct lola *chip = snd_kcontrol_chip(kcontrol);
+	int dir = kcontrol->private_value;
+	unsigned int val1, val2;
+	struct lola_pin *pin;
+
+	if (size < 4 * sizeof(unsigned int))
+		return -ENOMEM;
+	pin = &chip->pin[dir].pins[0];
+
+	val2 = pin->amp_step_size * 25;
+	val1 = -1 * (int)pin->amp_offset * (int)val2;
+#ifdef TLV_DB_SCALE_MUTE
+	val2 |= TLV_DB_SCALE_MUTE;
+#endif
+	if (put_user(SNDRV_CTL_TLVT_DB_SCALE, tlv))
+		return -EFAULT;
+	if (put_user(2 * sizeof(unsigned int), tlv + 1))
+		return -EFAULT;
+	if (put_user(val1, tlv + 2))
+		return -EFAULT;
+	if (put_user(val2, tlv + 3))
+		return -EFAULT;
+	return 0;
+}
+
+static struct snd_kcontrol_new lola_analog_mixer __devinitdata = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+		   SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+		   SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK),
+	.info = lola_analog_vol_info,
+	.get = lola_analog_vol_get,
+	.put = lola_analog_vol_put,
+	.tlv.c = lola_analog_vol_tlv,
+};
+
+static int __devinit create_analog_mixer(struct lola *chip, int dir, char *name)
+{
+	if (!chip->pin[dir].num_pins)
+		return 0;
+	/* no analog volumes on digital only adapters */
+	if (chip->pin[dir].num_pins != chip->pin[dir].num_analog_pins)
+		return 0;
+	lola_analog_mixer.name = name;
+	lola_analog_mixer.private_value = dir;
+	return snd_ctl_add(chip->card,
+			   snd_ctl_new1(&lola_analog_mixer, chip));
+}
+
+/*
+ * Hardware sample rate converter on digital input
+ */
+static int lola_input_src_info(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_info *uinfo)
+{
+	struct lola *chip = snd_kcontrol_chip(kcontrol);
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	uinfo->count = chip->pin[CAPT].num_pins;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 1;
+	return 0;
+}
+
+static int lola_input_src_get(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct lola *chip = snd_kcontrol_chip(kcontrol);
+	int i;
+
+	for (i = 0; i < chip->pin[CAPT].num_pins; i++)
+		ucontrol->value.integer.value[i] =
+			!!(chip->input_src_mask & (1 << i));
+	return 0;
+}
+
+static int lola_input_src_put(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct lola *chip = snd_kcontrol_chip(kcontrol);
+	int i;
+	unsigned int mask;
+
+	mask = 0;
+	for (i = 0; i < chip->pin[CAPT].num_pins; i++)
+		if (ucontrol->value.integer.value[i])
+			mask |= 1 << i;
+	return lola_set_src_config(chip, mask, true);
+}
+
+static struct snd_kcontrol_new lola_input_src_mixer __devinitdata = {
+	.name = "Digital SRC Capture Switch",
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.info = lola_input_src_info,
+	.get = lola_input_src_get,
+	.put = lola_input_src_put,
+};
+
+/*
+ * Lola16161 or Lola881 can have Hardware sample rate converters
+ * on its digital input pins
+ */
+static int __devinit create_input_src_mixer(struct lola *chip)
+{
+	if (!chip->input_src_caps_mask)
+		return 0;
+
+	return snd_ctl_add(chip->card,
+			   snd_ctl_new1(&lola_input_src_mixer, chip));
+}
+
+/*
+ * src gain mixer
+ */
+static int lola_src_gain_info(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_info *uinfo)
+{
+	unsigned int count = (kcontrol->private_value >> 8) & 0xff;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = count;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 409;
+	return 0;
+}
+
+static int lola_src_gain_get(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct lola *chip = snd_kcontrol_chip(kcontrol);
+	unsigned int ofs = kcontrol->private_value & 0xff;
+	unsigned int count = (kcontrol->private_value >> 8) & 0xff;
+	unsigned int mask, i;
+
+	mask = readl(&chip->mixer.array->src_gain_enable);
+	for (i = 0; i < count; i++) {
+		unsigned int idx = ofs + i;
+		unsigned short val;
+		if (!(chip->mixer.src_mask & (1 << idx)))
+			return -EINVAL;
+		if (mask & (1 << idx))
+			val = readw(&chip->mixer.array->src_gain[idx]) + 1;
+		else
+			val = 0;
+		ucontrol->value.integer.value[i] = val;
+	}
+	return 0;
+}
+
+static int lola_src_gain_put(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct lola *chip = snd_kcontrol_chip(kcontrol);
+	unsigned int ofs = kcontrol->private_value & 0xff;
+	unsigned int count = (kcontrol->private_value >> 8) & 0xff;
+	int i, err;
+
+	for (i = 0; i < count; i++) {
+		unsigned int idx = ofs + i;
+		unsigned short val = ucontrol->value.integer.value[i];
+		if (val)
+			val--;
+		err = lola_mixer_set_src_gain(chip, idx, val, !!val);
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
+
+/* raw value: 0 = -84dB, 336 = 0dB, 408=18dB, incremented 1 for mute */
+static const DECLARE_TLV_DB_SCALE(lola_src_gain_tlv, -8425, 25, 1);
+
+static struct snd_kcontrol_new lola_src_gain_mixer __devinitdata = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+		   SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+	.info = lola_src_gain_info,
+	.get = lola_src_gain_get,
+	.put = lola_src_gain_put,
+	.tlv.p = lola_src_gain_tlv,
+};
+
+static int __devinit create_src_gain_mixer(struct lola *chip,
+					   int num, int ofs, char *name)
+{
+	lola_src_gain_mixer.name = name;
+	lola_src_gain_mixer.private_value = ofs + (num << 8);
+	return snd_ctl_add(chip->card,
+			   snd_ctl_new1(&lola_src_gain_mixer, chip));
+}
+
+/*
+ * destination gain (matrix-like) mixer
+ */
+static int lola_dest_gain_info(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_info *uinfo)
+{
+	unsigned int src_num = (kcontrol->private_value >> 8) & 0xff;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = src_num;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 433;
+	return 0;
+}
+
+static int lola_dest_gain_get(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct lola *chip = snd_kcontrol_chip(kcontrol);
+	unsigned int src_ofs = kcontrol->private_value & 0xff;
+	unsigned int src_num = (kcontrol->private_value >> 8) & 0xff;
+	unsigned int dst_ofs = (kcontrol->private_value >> 16) & 0xff;
+	unsigned int dst, mask, i;
+
+	dst = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + dst_ofs;
+	mask = readl(&chip->mixer.array->dest_mix_gain_enable[dst]);
+	for (i = 0; i < src_num; i++) {
+		unsigned int src = src_ofs + i;
+		unsigned short val;
+		if (!(chip->mixer.src_mask & (1 << src)))
+			return -EINVAL;
+		if (mask & (1 << dst))
+			val = readw(&chip->mixer.array->dest_mix_gain[dst][src]) + 1;
+		else
+			val = 0;
+		ucontrol->value.integer.value[i] = val;
+	}
+	return 0;
+}
+
+static int lola_dest_gain_put(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct lola *chip = snd_kcontrol_chip(kcontrol);
+	unsigned int src_ofs = kcontrol->private_value & 0xff;
+	unsigned int src_num = (kcontrol->private_value >> 8) & 0xff;
+	unsigned int dst_ofs = (kcontrol->private_value >> 16) & 0xff;
+	unsigned int dst, mask;
+	unsigned short gains[MAX_STREAM_COUNT];
+	int i, num;
+
+	mask = 0;
+	num = 0;
+	for (i = 0; i < src_num; i++) {
+		unsigned short val = ucontrol->value.integer.value[i];
+		if (val) {
+			gains[num++] = val - 1;
+			mask |= 1 << i;
+		}
+	}
+	mask <<= src_ofs;
+	dst = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + dst_ofs;
+	return lola_mixer_set_dest_gains(chip, dst, mask, gains);
+}
+
+static const DECLARE_TLV_DB_SCALE(lola_dest_gain_tlv, -8425, 25, 1);
+
+static struct snd_kcontrol_new lola_dest_gain_mixer __devinitdata = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+		   SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+	.info = lola_dest_gain_info,
+	.get = lola_dest_gain_get,
+	.put = lola_dest_gain_put,
+	.tlv.p = lola_dest_gain_tlv,
+};
+
+static int __devinit create_dest_gain_mixer(struct lola *chip,
+					    int src_num, int src_ofs,
+					    int num, int ofs, char *name)
+{
+	lola_dest_gain_mixer.count = num;
+	lola_dest_gain_mixer.name = name;
+	lola_dest_gain_mixer.private_value =
+		src_ofs + (src_num << 8) + (ofs << 16) + (num << 24);
+	return snd_ctl_add(chip->card,
+			  snd_ctl_new1(&lola_dest_gain_mixer, chip));
+}
+
+/*
+ */
+int __devinit lola_create_mixer(struct lola *chip)
+{
+	int err;
+
+	err = create_analog_mixer(chip, PLAY, "Analog Playback Volume");
+	if (err < 0)
+		return err;
+	err = create_analog_mixer(chip, CAPT, "Analog Capture Volume");
+	if (err < 0)
+		return err;
+	err = create_input_src_mixer(chip);
+	if (err < 0)
+		return err;
+	err = create_src_gain_mixer(chip, chip->mixer.src_phys_ins, 0,
+				    "Line Source Gain Volume");
+	if (err < 0)
+		return err;
+	err = create_src_gain_mixer(chip, chip->mixer.src_stream_outs,
+				    chip->mixer.src_stream_out_ofs,
+				    "Stream Source Gain Volume");
+	if (err < 0)
+		return err;
+	err = create_dest_gain_mixer(chip,
+				     chip->mixer.src_phys_ins, 0,
+				     chip->mixer.dest_stream_ins, 0,
+				     "Line Capture Volume");
+	if (err < 0)
+		return err;
+	err = create_dest_gain_mixer(chip,
+				     chip->mixer.src_stream_outs,
+				     chip->mixer.src_stream_out_ofs,
+				     chip->mixer.dest_stream_ins, 0,
+				     "Stream-Loopback Capture Volume");
+	if (err < 0)
+		return err;
+	err = create_dest_gain_mixer(chip,
+				     chip->mixer.src_phys_ins, 0,
+				     chip->mixer.dest_phys_outs,
+				     chip->mixer.dest_phys_out_ofs,
+				     "Line-Loopback Playback Volume");
+	if (err < 0)
+		return err;
+	err = create_dest_gain_mixer(chip,
+				     chip->mixer.src_stream_outs,
+				     chip->mixer.src_stream_out_ofs,
+				     chip->mixer.dest_phys_outs,
+				     chip->mixer.dest_phys_out_ofs,
+				     "Stream Playback Volume");
+	if (err < 0)
+		return err;
+
+	return init_mixer_values(chip);
+}
diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c
new file mode 100644
index 00000000000..c44db68eecb
--- /dev/null
+++ b/sound/pci/lola/lola_pcm.c
@@ -0,0 +1,706 @@
+/*
+ *  Support for Digigram Lola PCI-e boards
+ *
+ *  Copyright (c) 2011 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.
+ */
+
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/dma-mapping.h>
+#include <linux/pci.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include "lola.h"
+
+#define LOLA_MAX_BDL_ENTRIES	8
+#define LOLA_MAX_BUF_SIZE	(1024*1024*1024)
+#define LOLA_BDL_ENTRY_SIZE	(16 * 16)
+
+static struct lola_pcm *lola_get_pcm(struct snd_pcm_substream *substream)
+{
+	struct lola *chip = snd_pcm_substream_chip(substream);
+	return &chip->pcm[substream->stream];
+}
+
+static struct lola_stream *lola_get_stream(struct snd_pcm_substream *substream)
+{
+	struct lola_pcm *pcm = lola_get_pcm(substream);
+	unsigned int idx = substream->number;
+	return &pcm->streams[idx];
+}
+
+static unsigned int lola_get_lrc(struct lola *chip)
+{
+	return lola_readl(chip, BAR1, LRC);
+}
+
+static unsigned int lola_get_tstamp(struct lola *chip, bool quick_no_sync)
+{
+	unsigned int tstamp = lola_get_lrc(chip) >> 8;
+	if (chip->granularity) {
+		unsigned int wait_banks = quick_no_sync ? 0 : 8;
+		tstamp += (wait_banks + 1) * chip->granularity - 1;
+		tstamp -= tstamp % chip->granularity;
+	}
+	return tstamp << 8;
+}
+
+/* clear any pending interrupt status */
+static void lola_stream_clear_pending_irq(struct lola *chip,
+					  struct lola_stream *str)
+{
+	unsigned int val = lola_dsd_read(chip, str->dsd, STS);
+	val &= LOLA_DSD_STS_DESE | LOLA_DSD_STS_BCIS;
+	if (val)
+		lola_dsd_write(chip, str->dsd, STS, val);
+}
+
+static void lola_stream_start(struct lola *chip, struct lola_stream *str,
+			      unsigned int tstamp)
+{
+	lola_stream_clear_pending_irq(chip, str);
+	lola_dsd_write(chip, str->dsd, CTL,
+		       LOLA_DSD_CTL_SRUN |
+		       LOLA_DSD_CTL_IOCE |
+		       LOLA_DSD_CTL_DEIE |
+		       LOLA_DSD_CTL_VLRCV |
+		       tstamp);
+}
+
+static void lola_stream_stop(struct lola *chip, struct lola_stream *str,
+			     unsigned int tstamp)
+{
+	lola_dsd_write(chip, str->dsd, CTL,
+		       LOLA_DSD_CTL_IOCE |
+		       LOLA_DSD_CTL_DEIE |
+		       LOLA_DSD_CTL_VLRCV |
+		       tstamp);
+	lola_stream_clear_pending_irq(chip, str);
+}
+
+static void wait_for_srst_clear(struct lola *chip, struct lola_stream *str)
+{
+	unsigned long end_time = jiffies + msecs_to_jiffies(200);
+	while (time_before(jiffies, end_time)) {
+		unsigned int val;
+		val = lola_dsd_read(chip, str->dsd, CTL);
+		if (!(val & LOLA_DSD_CTL_SRST))
+			return;
+		msleep(1);
+	}
+	printk(KERN_WARNING SFX "SRST not clear (stream %d)\n", str->dsd);
+}
+
+static int lola_stream_wait_for_fifo(struct lola *chip,
+				     struct lola_stream *str,
+				     bool ready)
+{
+	unsigned int val = ready ? LOLA_DSD_STS_FIFORDY : 0;
+	unsigned long end_time = jiffies + msecs_to_jiffies(200);
+	while (time_before(jiffies, end_time)) {
+		unsigned int reg = lola_dsd_read(chip, str->dsd, STS);
+		if ((reg & LOLA_DSD_STS_FIFORDY) == val)
+			return 0;
+		msleep(1);
+	}
+	printk(KERN_WARNING SFX "FIFO not ready (stream %d)\n", str->dsd);
+	return -EIO;
+}
+
+/* sync for FIFO ready/empty for all linked streams;
+ * clear paused flag when FIFO gets ready again
+ */
+static int lola_sync_wait_for_fifo(struct lola *chip,
+				   struct snd_pcm_substream *substream,
+				   bool ready)
+{
+	unsigned int val = ready ? LOLA_DSD_STS_FIFORDY : 0;
+	unsigned long end_time = jiffies + msecs_to_jiffies(200);
+	struct snd_pcm_substream *s;
+	int pending = 0;
+
+	while (time_before(jiffies, end_time)) {
+		pending = 0;
+		snd_pcm_group_for_each_entry(s, substream) {
+			struct lola_stream *str;
+			if (s->pcm->card != substream->pcm->card)
+				continue;
+			str = lola_get_stream(s);
+			if (str->prepared && str->paused) {
+				unsigned int reg;
+				reg = lola_dsd_read(chip, str->dsd, STS);
+				if ((reg & LOLA_DSD_STS_FIFORDY) != val) {
+					pending = str->dsd + 1;
+					break;
+				}
+				if (ready)
+					str->paused = 0;
+			}
+		}
+		if (!pending)
+			return 0;
+		msleep(1);
+	}
+	printk(KERN_WARNING SFX "FIFO not ready (pending %d)\n", pending - 1);
+	return -EIO;
+}
+
+/* finish pause - prepare for a new resume */
+static void lola_sync_pause(struct lola *chip,
+			    struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_substream *s;
+
+	lola_sync_wait_for_fifo(chip, substream, false);
+	snd_pcm_group_for_each_entry(s, substream) {
+		struct lola_stream *str;
+		if (s->pcm->card != substream->pcm->card)
+			continue;
+		str = lola_get_stream(s);
+		if (str->paused && str->prepared)
+			lola_dsd_write(chip, str->dsd, CTL, LOLA_DSD_CTL_SRUN |
+				       LOLA_DSD_CTL_IOCE | LOLA_DSD_CTL_DEIE);
+	}
+	lola_sync_wait_for_fifo(chip, substream, true);
+}
+
+static void lola_stream_reset(struct lola *chip, struct lola_stream *str)
+{
+	if (str->prepared) {
+		if (str->paused)
+			lola_sync_pause(chip, str->substream);
+		str->prepared = 0;
+		lola_dsd_write(chip, str->dsd, CTL,
+			       LOLA_DSD_CTL_IOCE | LOLA_DSD_CTL_DEIE);
+		lola_stream_wait_for_fifo(chip, str, false);
+		lola_stream_clear_pending_irq(chip, str);
+		lola_dsd_write(chip, str->dsd, CTL, LOLA_DSD_CTL_SRST);
+		lola_dsd_write(chip, str->dsd, LVI, 0);
+		lola_dsd_write(chip, str->dsd, BDPU, 0);
+		lola_dsd_write(chip, str->dsd, BDPL, 0);
+		wait_for_srst_clear(chip, str);
+	}
+}
+
+static struct snd_pcm_hardware lola_pcm_hw = {
+	.info =			(SNDRV_PCM_INFO_MMAP |
+				 SNDRV_PCM_INFO_INTERLEAVED |
+				 SNDRV_PCM_INFO_BLOCK_TRANSFER |
+				 SNDRV_PCM_INFO_MMAP_VALID |
+				 SNDRV_PCM_INFO_PAUSE),
+	.formats =		(SNDRV_PCM_FMTBIT_S16_LE |
+				 SNDRV_PCM_FMTBIT_S24_LE |
+				 SNDRV_PCM_FMTBIT_S32_LE |
+				 SNDRV_PCM_FMTBIT_FLOAT_LE),
+	.rates =		SNDRV_PCM_RATE_8000_192000,
+	.rate_min =		8000,
+	.rate_max =		192000,
+	.channels_min =		1,
+	.channels_max =		2,
+	.buffer_bytes_max =	LOLA_MAX_BUF_SIZE,
+	.period_bytes_min =	128,
+	.period_bytes_max =	LOLA_MAX_BUF_SIZE / 2,
+	.periods_min =		2,
+	.periods_max =		LOLA_MAX_BDL_ENTRIES,
+	.fifo_size =		0,
+};
+
+static int lola_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct lola *chip = snd_pcm_substream_chip(substream);
+	struct lola_pcm *pcm = lola_get_pcm(substream);
+	struct lola_stream *str = lola_get_stream(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	mutex_lock(&chip->open_mutex);
+	if (str->opened) {
+		mutex_unlock(&chip->open_mutex);
+		return -EBUSY;
+	}
+	str->substream = substream;
+	str->master = NULL;
+	str->opened = 1;
+	runtime->hw = lola_pcm_hw;
+	runtime->hw.channels_max = pcm->num_streams - str->index;
+	if (chip->sample_rate) {
+		/* sample rate is locked */
+		runtime->hw.rate_min = chip->sample_rate;
+		runtime->hw.rate_max = chip->sample_rate;
+	} else {
+		runtime->hw.rate_min = chip->sample_rate_min;
+		runtime->hw.rate_max = chip->sample_rate_max;
+	}
+	chip->ref_count_rate++;
+	snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+	/* period size = multiple of chip->granularity (8, 16 or 32 frames)*/
+	snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+				   chip->granularity);
+	snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+				   chip->granularity);
+	mutex_unlock(&chip->open_mutex);
+	return 0;
+}
+
+static void lola_cleanup_slave_streams(struct lola_pcm *pcm,
+				       struct lola_stream *str)
+{
+	int i;
+	for (i = str->index + 1; i < pcm->num_streams; i++) {
+		struct lola_stream *s = &pcm->streams[i];
+		if (s->master != str)
+			break;
+		s->master = NULL;
+		s->opened = 0;
+	}
+}
+
+static int lola_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct lola *chip = snd_pcm_substream_chip(substream);
+	struct lola_stream *str = lola_get_stream(substream);
+
+	mutex_lock(&chip->open_mutex);
+	if (str->substream == substream) {
+		str->substream = NULL;
+		str->opened = 0;
+	}
+	if (--chip->ref_count_rate == 0) {
+		/* release sample rate */
+		chip->sample_rate = 0;
+	}
+	mutex_unlock(&chip->open_mutex);
+	return 0;
+}
+
+static int lola_pcm_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *hw_params)
+{
+	struct lola_stream *str = lola_get_stream(substream);
+
+	str->bufsize = 0;
+	str->period_bytes = 0;
+	str->format_verb = 0;
+	return snd_pcm_lib_malloc_pages(substream,
+					params_buffer_bytes(hw_params));
+}
+
+static int lola_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	struct lola *chip = snd_pcm_substream_chip(substream);
+	struct lola_pcm *pcm = lola_get_pcm(substream);
+	struct lola_stream *str = lola_get_stream(substream);
+
+	mutex_lock(&chip->open_mutex);
+	lola_stream_reset(chip, str);
+	lola_cleanup_slave_streams(pcm, str);
+	mutex_unlock(&chip->open_mutex);
+	return snd_pcm_lib_free_pages(substream);
+}
+
+/*
+ * set up a BDL entry
+ */
+static int setup_bdle(struct snd_pcm_substream *substream,
+		      struct lola_stream *str, u32 **bdlp,
+		      int ofs, int size)
+{
+	u32 *bdl = *bdlp;
+
+	while (size > 0) {
+		dma_addr_t addr;
+		int chunk;
+
+		if (str->frags >= LOLA_MAX_BDL_ENTRIES)
+			return -EINVAL;
+
+		addr = snd_pcm_sgbuf_get_addr(substream, ofs);
+		/* program the address field of the BDL entry */
+		bdl[0] = cpu_to_le32((u32)addr);
+		bdl[1] = cpu_to_le32(upper_32_bits(addr));
+		/* program the size field of the BDL entry */
+		chunk = snd_pcm_sgbuf_get_chunk_size(substream, ofs, size);
+		bdl[2] = cpu_to_le32(chunk);
+		/* program the IOC to enable interrupt
+		 * only when the whole fragment is processed
+		 */
+		size -= chunk;
+		bdl[3] = size ? 0 : cpu_to_le32(0x01);
+		bdl += 4;
+		str->frags++;
+		ofs += chunk;
+	}
+	*bdlp = bdl;
+	return ofs;
+}
+
+/*
+ * set up BDL entries
+ */
+static int lola_setup_periods(struct lola *chip, struct lola_pcm *pcm,
+			      struct snd_pcm_substream *substream,
+			      struct lola_stream *str)
+{
+	u32 *bdl;
+	int i, ofs, periods, period_bytes;
+
+	period_bytes = str->period_bytes;
+	periods = str->bufsize / period_bytes;
+
+	/* program the initial BDL entries */
+	bdl = (u32 *)(pcm->bdl.area + LOLA_BDL_ENTRY_SIZE * str->index);
+	ofs = 0;
+	str->frags = 0;
+	for (i = 0; i < periods; i++) {
+		ofs = setup_bdle(substream, str, &bdl, ofs, period_bytes);
+		if (ofs < 0)
+			goto error;
+	}
+	return 0;
+
+ error:
+	snd_printk(KERN_ERR SFX "Too many BDL entries: buffer=%d, period=%d\n",
+		   str->bufsize, period_bytes);
+	return -EINVAL;
+}
+
+static unsigned int lola_get_format_verb(struct snd_pcm_substream *substream)
+{
+	unsigned int verb;
+
+	switch (substream->runtime->format) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		verb = 0x00000000;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		verb = 0x00000200;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		verb = 0x00000300;
+		break;
+	case SNDRV_PCM_FORMAT_FLOAT_LE:
+		verb = 0x00001300;
+		break;
+	default:
+		return 0;
+	}
+	verb |= substream->runtime->channels;
+	return verb;
+}
+
+static int lola_set_stream_config(struct lola *chip,
+				  struct lola_stream *str,
+				  int channels)
+{
+	int i, err;
+	unsigned int verb, val;
+
+	/* set format info for all channels
+	 * (with only one command for the first channel)
+	 */
+	err = lola_codec_read(chip, str->nid, LOLA_VERB_SET_STREAM_FORMAT,
+			      str->format_verb, 0, &val, NULL);
+	if (err < 0) {
+		printk(KERN_ERR SFX "Cannot set stream format 0x%x\n",
+		       str->format_verb);
+		return err;
+	}
+
+	/* update stream - channel config */
+	for (i = 0; i < channels; i++) {
+		verb = (str->index << 6) | i;
+		err = lola_codec_read(chip, str[i].nid,
+				      LOLA_VERB_SET_CHANNEL_STREAMID, 0, verb,
+				      &val, NULL);
+		if (err < 0) {
+			printk(KERN_ERR SFX "Cannot set stream channel %d\n", i);
+			return err;
+		}
+	}
+	return 0;
+}
+
+/*
+ * set up the SD for streaming
+ */
+static int lola_setup_controller(struct lola *chip, struct lola_pcm *pcm,
+				 struct lola_stream *str)
+{
+	dma_addr_t bdl;
+
+	if (str->prepared)
+		return -EINVAL;
+
+	/* set up BDL */
+	bdl = pcm->bdl.addr + LOLA_BDL_ENTRY_SIZE * str->index;
+	lola_dsd_write(chip, str->dsd, BDPL, (u32)bdl);
+	lola_dsd_write(chip, str->dsd, BDPU, upper_32_bits(bdl));
+	/* program the stream LVI (last valid index) of the BDL */
+	lola_dsd_write(chip, str->dsd, LVI, str->frags - 1);
+	lola_stream_clear_pending_irq(chip, str);
+
+ 	lola_dsd_write(chip, str->dsd, CTL,
+		       LOLA_DSD_CTL_IOCE | LOLA_DSD_CTL_DEIE | LOLA_DSD_CTL_SRUN);
+
+	str->prepared = 1;
+
+	return lola_stream_wait_for_fifo(chip, str, true);
+}
+
+static int lola_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct lola *chip = snd_pcm_substream_chip(substream);
+	struct lola_pcm *pcm = lola_get_pcm(substream);
+	struct lola_stream *str = lola_get_stream(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned int bufsize, period_bytes, format_verb;
+	int i, err;
+
+	mutex_lock(&chip->open_mutex);
+	lola_stream_reset(chip, str);
+	lola_cleanup_slave_streams(pcm, str);
+	if (str->index + runtime->channels > pcm->num_streams) {
+		mutex_unlock(&chip->open_mutex);
+		return -EINVAL;
+	}
+	for (i = 1; i < runtime->channels; i++) {
+		str[i].master = str;
+		str[i].opened = 1;
+	}
+	mutex_unlock(&chip->open_mutex);
+
+	bufsize = snd_pcm_lib_buffer_bytes(substream);
+	period_bytes = snd_pcm_lib_period_bytes(substream);
+	format_verb = lola_get_format_verb(substream);
+
+	str->bufsize = bufsize;
+	str->period_bytes = period_bytes;
+	str->format_verb = format_verb;
+
+	err = lola_setup_periods(chip, pcm, substream, str);
+	if (err < 0)
+		return err;
+
+	err = lola_set_sample_rate(chip, runtime->rate);
+	if (err < 0)
+		return err;
+	chip->sample_rate = runtime->rate;	/* sample rate gets locked */
+
+	err = lola_set_stream_config(chip, str, runtime->channels);
+	if (err < 0)
+		return err;
+
+	err = lola_setup_controller(chip, pcm, str);
+	if (err < 0) {
+		lola_stream_reset(chip, str);
+		return err;
+	}
+
+	return 0;
+}
+
+static int lola_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct lola *chip = snd_pcm_substream_chip(substream);
+	struct lola_stream *str;
+	struct snd_pcm_substream *s;
+	unsigned int start;
+	unsigned int tstamp;
+	bool sync_streams;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		start = 1;
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_STOP:
+		start = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/*
+	 * sample correct synchronization is only needed starting several
+	 * streams. On stop or if only one stream do as quick as possible
+	 */
+	sync_streams = (start && snd_pcm_stream_linked(substream));
+	tstamp = lola_get_tstamp(chip, !sync_streams);
+	spin_lock(&chip->reg_lock);
+	snd_pcm_group_for_each_entry(s, substream) {
+		if (s->pcm->card != substream->pcm->card)
+			continue;
+		str = lola_get_stream(s);
+		if (start)
+			lola_stream_start(chip, str, tstamp);
+		else
+			lola_stream_stop(chip, str, tstamp);
+		str->running = start;
+		str->paused = !start;
+		snd_pcm_trigger_done(s, substream);
+	}
+	spin_unlock(&chip->reg_lock);
+	return 0;
+}
+
+static snd_pcm_uframes_t lola_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct lola *chip = snd_pcm_substream_chip(substream);
+	struct lola_stream *str = lola_get_stream(substream);
+	unsigned int pos = lola_dsd_read(chip, str->dsd, LPIB);
+
+	if (pos >= str->bufsize)
+		pos = 0;
+	return bytes_to_frames(substream->runtime, pos);
+}
+
+void lola_pcm_update(struct lola *chip, struct lola_pcm *pcm, unsigned int bits)
+{
+	int i;
+
+	for (i = 0; bits && i < pcm->num_streams; i++) {
+		if (bits & (1 << i)) {
+			struct lola_stream *str = &pcm->streams[i];
+			if (str->substream && str->running)
+				snd_pcm_period_elapsed(str->substream);
+			bits &= ~(1 << i);
+		}
+	}
+}
+
+static struct snd_pcm_ops lola_pcm_ops = {
+	.open = lola_pcm_open,
+	.close = lola_pcm_close,
+	.ioctl = snd_pcm_lib_ioctl,
+	.hw_params = lola_pcm_hw_params,
+	.hw_free = lola_pcm_hw_free,
+	.prepare = lola_pcm_prepare,
+	.trigger = lola_pcm_trigger,
+	.pointer = lola_pcm_pointer,
+	.page = snd_pcm_sgbuf_ops_page,
+};
+
+int __devinit lola_create_pcm(struct lola *chip)
+{
+	struct snd_pcm *pcm;
+	int i, err;
+
+	for (i = 0; i < 2; i++) {
+		err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+					  snd_dma_pci_data(chip->pci),
+					  PAGE_SIZE, &chip->pcm[i].bdl);
+		if (err < 0)
+			return err;
+	}
+
+	err = snd_pcm_new(chip->card, "Digigram Lola", 0,
+			  chip->pcm[SNDRV_PCM_STREAM_PLAYBACK].num_streams,
+			  chip->pcm[SNDRV_PCM_STREAM_CAPTURE].num_streams,
+			  &pcm);
+	if (err < 0)
+		return err;
+	strlcpy(pcm->name, "Digigram Lola", sizeof(pcm->name));
+	pcm->private_data = chip;
+	for (i = 0; i < 2; i++) {
+		if (chip->pcm[i].num_streams)
+			snd_pcm_set_ops(pcm, i, &lola_pcm_ops);
+	}
+	/* buffer pre-allocation */
+	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
+					      snd_dma_pci_data(chip->pci),
+					      1024 * 64, 32 * 1024 * 1024);
+	return 0;
+}
+
+void lola_free_pcm(struct lola *chip)
+{
+	snd_dma_free_pages(&chip->pcm[0].bdl);
+	snd_dma_free_pages(&chip->pcm[1].bdl);
+}
+
+/*
+ */
+
+static int lola_init_stream(struct lola *chip, struct lola_stream *str,
+			    int idx, int nid, int dir)
+{
+	unsigned int val;
+	int err;
+
+	str->nid = nid;
+	str->index = idx;
+	str->dsd = idx;
+	if (dir == PLAY)
+		str->dsd += MAX_STREAM_IN_COUNT;
+	err = lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val);
+	if (err < 0) {
+		printk(KERN_ERR SFX "Can't read wcaps for 0x%x\n", nid);
+		return err;
+	}
+	if (dir == PLAY) {
+		/* test TYPE and bits 0..11 (no test bit9 : Digital = 0/1) */
+		if ((val & 0x00f00dff) != 0x00000010) {
+			printk(KERN_ERR SFX "Invalid wcaps 0x%x for 0x%x\n",
+			       val, nid);
+			return -EINVAL;
+		}
+	} else {
+		/* test TYPE and bits 0..11 (no test bit9 : Digital = 0/1)
+		 * (bug : ignore bit8: Conn list = 0/1)
+		 */
+		if ((val & 0x00f00cff) != 0x00100010) {
+			printk(KERN_ERR SFX "Invalid wcaps 0x%x for 0x%x\n",
+			       val, nid);
+			return -EINVAL;
+		}
+		/* test bit9:DIGITAL and bit12:SRC_PRESENT*/
+		if ((val & 0x00001200) == 0x00001200)
+			chip->input_src_caps_mask |= (1 << idx);
+	}
+
+	err = lola_read_param(chip, nid, LOLA_PAR_STREAM_FORMATS, &val);
+	if (err < 0) {
+		printk(KERN_ERR SFX "Can't read FORMATS 0x%x\n", nid);
+		return err;
+	}
+	val &= 3;
+	if (val == 3)
+		str->can_float = true;
+	if (!(val & 1)) {
+		printk(KERN_ERR SFX "Invalid formats 0x%x for 0x%x", val, nid);
+		return -EINVAL;
+	}
+	return 0;
+}
+
+int __devinit lola_init_pcm(struct lola *chip, int dir, int *nidp)
+{
+	struct lola_pcm *pcm = &chip->pcm[dir];
+	int i, nid, err;
+
+	nid = *nidp;
+	for (i = 0; i < pcm->num_streams; i++, nid++) {
+		err = lola_init_stream(chip, &pcm->streams[i], i, nid, dir);
+		if (err < 0)
+			return err;
+	}
+	*nidp = nid;
+	return 0;
+}
diff --git a/sound/pci/lola/lola_proc.c b/sound/pci/lola/lola_proc.c
new file mode 100644
index 00000000000..9d7daf897c9
--- /dev/null
+++ b/sound/pci/lola/lola_proc.c
@@ -0,0 +1,222 @@
+/*
+ *  Support for Digigram Lola PCI-e boards
+ *
+ *  Copyright (c) 2011 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.
+ */
+
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include "lola.h"
+
+static void print_audio_widget(struct snd_info_buffer *buffer,
+			       struct lola *chip, int nid, const char *name)
+{
+	unsigned int val;
+
+	lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val);
+	snd_iprintf(buffer, "Node 0x%02x %s wcaps 0x%x\n", nid, name, val);
+	lola_read_param(chip, nid, LOLA_PAR_STREAM_FORMATS, &val);
+	snd_iprintf(buffer, "  Formats: 0x%x\n", val);
+}
+
+static void print_pin_widget(struct snd_info_buffer *buffer,
+			     struct lola *chip, int nid, unsigned int ampcap,
+			     const char *name)
+{
+	unsigned int val;
+
+	lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val);
+	snd_iprintf(buffer, "Node 0x%02x %s wcaps 0x%x\n", nid, name, val);
+	if (val == 0x00400200)
+		return;
+	lola_read_param(chip, nid, ampcap, &val);
+	snd_iprintf(buffer, "  Amp-Caps: 0x%x\n", val);
+	snd_iprintf(buffer, "    mute=%d, step-size=%d, steps=%d, ofs=%d\n",
+		    LOLA_AMP_MUTE_CAPABLE(val),
+		    LOLA_AMP_STEP_SIZE(val),
+		    LOLA_AMP_NUM_STEPS(val),
+		    LOLA_AMP_OFFSET(val));
+	lola_codec_read(chip, nid, LOLA_VERB_GET_MAX_LEVEL, 0, 0, &val, NULL);
+	snd_iprintf(buffer, "  Max-level: 0x%x\n", val);
+}
+
+static void print_clock_widget(struct snd_info_buffer *buffer,
+			       struct lola *chip, int nid)
+{
+	int i, j, num_clocks;
+	unsigned int val;
+
+	lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val);
+	snd_iprintf(buffer, "Node 0x%02x [Clock] wcaps 0x%x\n", nid, val);
+	num_clocks = val & 0xff;
+	for (i = 0; i < num_clocks; i += 4) {
+		unsigned int res_ex;
+		unsigned short items[4];
+		const char *name;
+
+		lola_codec_read(chip, nid, LOLA_VERB_GET_CLOCK_LIST,
+				i, 0, &val, &res_ex);
+		items[0] = val & 0xfff;
+		items[1] = (val >> 16) & 0xfff;
+		items[2] = res_ex & 0xfff;
+		items[3] = (res_ex >> 16) & 0xfff;
+		for (j = 0; j < 4; j++) {
+			unsigned char type = items[j] >> 8;
+			unsigned int freq = items[j] & 0xff;
+			if (i + j >= num_clocks)
+				break;
+			if (type == LOLA_CLOCK_TYPE_INTERNAL) {
+				name = "Internal";
+				freq = lola_sample_rate_convert(freq);
+			} else if (type == LOLA_CLOCK_TYPE_VIDEO) {
+				name = "Video";
+				freq = lola_sample_rate_convert(freq);
+			} else {
+				name = "Other";
+			}
+			snd_iprintf(buffer, "  Clock %d: Type %d:%s, freq=%d\n",
+				    i + j, type, name, freq);
+		}
+	}
+}
+
+static void print_mixer_widget(struct snd_info_buffer *buffer,
+			       struct lola *chip, int nid)
+{
+	unsigned int val;
+
+	lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val);
+	snd_iprintf(buffer, "Node 0x%02x [Mixer] wcaps 0x%x\n", nid, val);
+}
+
+static void lola_proc_codec_read(struct snd_info_entry *entry,
+				 struct snd_info_buffer *buffer)
+{
+	struct lola *chip = entry->private_data;
+	unsigned int val;
+	int i, nid;
+
+	lola_read_param(chip, 0, LOLA_PAR_VENDOR_ID, &val);
+	snd_iprintf(buffer, "Vendor: 0x%08x\n", val);
+	lola_read_param(chip, 1, LOLA_PAR_FUNCTION_TYPE, &val);
+	snd_iprintf(buffer, "Function Type: %d\n", val);
+	lola_read_param(chip, 1, LOLA_PAR_SPECIFIC_CAPS, &val);
+	snd_iprintf(buffer, "Specific-Caps: 0x%08x\n", val);
+	snd_iprintf(buffer, "  Pins-In %d, Pins-Out %d\n",
+		    chip->pin[CAPT].num_pins, chip->pin[PLAY].num_pins);
+	nid = 2;
+	for (i = 0; i < chip->pcm[CAPT].num_streams; i++, nid++)
+		print_audio_widget(buffer, chip, nid, "[Audio-In]");
+	for (i = 0; i < chip->pcm[PLAY].num_streams; i++, nid++)
+		print_audio_widget(buffer, chip, nid, "[Audio-Out]");
+	for (i = 0; i < chip->pin[CAPT].num_pins; i++, nid++)
+		print_pin_widget(buffer, chip, nid, LOLA_PAR_AMP_IN_CAP,
+				 "[Pin-In]");
+	for (i = 0; i < chip->pin[PLAY].num_pins; i++, nid++)
+		print_pin_widget(buffer, chip, nid, LOLA_PAR_AMP_OUT_CAP,
+				 "[Pin-Out]");
+	if (LOLA_AFG_CLOCK_WIDGET_PRESENT(chip->lola_caps)) {
+		print_clock_widget(buffer, chip, nid);
+		nid++;
+	}
+	if (LOLA_AFG_MIXER_WIDGET_PRESENT(chip->lola_caps)) {
+		print_mixer_widget(buffer, chip, nid);
+		nid++;
+	}
+}
+
+/* direct codec access for debugging */
+static void lola_proc_codec_rw_write(struct snd_info_entry *entry,
+				     struct snd_info_buffer *buffer)
+{
+	struct lola *chip = entry->private_data;
+	char line[64];
+	unsigned int id, verb, data, extdata;
+	while (!snd_info_get_line(buffer, line, sizeof(line))) {
+		if (sscanf(line, "%i %i %i %i", &id, &verb, &data, &extdata) != 4)
+			continue;
+		lola_codec_read(chip, id, verb, data, extdata,
+				&chip->debug_res,
+				&chip->debug_res_ex);
+	}
+}
+
+static void lola_proc_codec_rw_read(struct snd_info_entry *entry,
+				    struct snd_info_buffer *buffer)
+{
+	struct lola *chip = entry->private_data;
+	snd_iprintf(buffer, "0x%x 0x%x\n", chip->debug_res, chip->debug_res_ex);
+}
+
+/*
+ * dump some registers
+ */
+static void lola_proc_regs_read(struct snd_info_entry *entry,
+				struct snd_info_buffer *buffer)
+{
+	struct lola *chip = entry->private_data;
+	int i;
+
+	for (i = 0; i < 0x40; i += 4) {
+		snd_iprintf(buffer, "BAR0 %02x: %08x\n", i,
+			    readl(chip->bar[BAR0].remap_addr + i));
+	}
+	snd_iprintf(buffer, "\n");
+	for (i = 0; i < 0x30; i += 4) {
+		snd_iprintf(buffer, "BAR1 %02x: %08x\n", i,
+			    readl(chip->bar[BAR1].remap_addr + i));
+	}
+	snd_iprintf(buffer, "\n");
+	for (i = 0x80; i < 0xa0; i += 4) {
+		snd_iprintf(buffer, "BAR1 %02x: %08x\n", i,
+			    readl(chip->bar[BAR1].remap_addr + i));
+	}
+	snd_iprintf(buffer, "\n");
+	for (i = 0; i < 32; i++) {
+		snd_iprintf(buffer, "DSD %02x STS  %08x\n", i,
+			    lola_dsd_read(chip, i, STS));
+		snd_iprintf(buffer, "DSD %02x LPIB %08x\n", i,
+			    lola_dsd_read(chip, i, LPIB));
+		snd_iprintf(buffer, "DSD %02x CTL  %08x\n", i,
+			    lola_dsd_read(chip, i, CTL));
+		snd_iprintf(buffer, "DSD %02x LVIL %08x\n", i,
+			    lola_dsd_read(chip, i, LVI));
+		snd_iprintf(buffer, "DSD %02x BDPL %08x\n", i,
+			    lola_dsd_read(chip, i, BDPL));
+		snd_iprintf(buffer, "DSD %02x BDPU %08x\n", i,
+			    lola_dsd_read(chip, i, BDPU));
+	}
+}
+
+void __devinit lola_proc_debug_new(struct lola *chip)
+{
+	struct snd_info_entry *entry;
+
+	if (!snd_card_proc_new(chip->card, "codec", &entry))
+		snd_info_set_text_ops(entry, chip, lola_proc_codec_read);
+	if (!snd_card_proc_new(chip->card, "codec_rw", &entry)) {
+		snd_info_set_text_ops(entry, chip, lola_proc_codec_rw_read);
+		entry->mode |= S_IWUSR;
+		entry->c.text.write = lola_proc_codec_rw_write;
+	}
+	if (!snd_card_proc_new(chip->card, "regs", &entry))
+		snd_info_set_text_ops(entry, chip, lola_proc_regs_read);
+}
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index 8cc4733698a..ce33be0e4e9 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -278,7 +278,7 @@ static int pdacf_resume(struct pcmcia_device *link)
 /*
  * Module entry points
  */
-static struct pcmcia_device_id snd_pdacf_ids[] = {
+static const struct pcmcia_device_id snd_pdacf_ids[] = {
 	/* this is too general PCMCIA_DEVICE_MANF_CARD(0x015d, 0x4c45), */
 	PCMCIA_DEVICE_PROD_ID12("Core Sound","PDAudio-CF",0x396d19d2,0x71717b49),
 	PCMCIA_DEVICE_NULL
diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c
index 80000d631f8..d9ef21d8fa7 100644
--- a/sound/pcmcia/vx/vxpocket.c
+++ b/sound/pcmcia/vx/vxpocket.c
@@ -350,7 +350,7 @@ static void vxpocket_detach(struct pcmcia_device *link)
  * Module entry points
  */
 
-static struct pcmcia_device_id vxp_ids[] = {
+static const struct pcmcia_device_id vxp_ids[] = {
 	PCMCIA_DEVICE_MANF_CARD(0x01f1, 0x0100),
 	PCMCIA_DEVICE_NULL
 };
diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c
index 961d9829769..9cea84c3e0c 100644
--- a/sound/ppc/tumbler.c
+++ b/sound/ppc/tumbler.c
@@ -1000,7 +1000,7 @@ static void device_change_handler(struct work_struct *work)
 				   chip->lineout_sw_ctl);
 		if (mix->anded_reset)
 			msleep(10);
-		check_mute(chip, &mix->amp_mute, 1, mix->auto_mute_notify,
+		check_mute(chip, &mix->amp_mute, !IS_G4DA, mix->auto_mute_notify,
 			   chip->speaker_sw_ctl);
 	} else {
 		/* unmute speaker, mute others */
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index a7a7bbc0762..f53dc09c48f 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -846,9 +846,10 @@ int atmel_ssc_set_audio(int ssc_id)
 	if (IS_ERR(ssc))
 		pr_warn("Unable to parent ASoC SSC DAI on SSC: %ld\n",
 			PTR_ERR(ssc));
-	else
+	else {
 		ssc_pdev->dev.parent = &(ssc->pdev->dev);
-	ssc_free(ssc);
+		ssc_free(ssc);
+	}
 
 	ret = platform_device_add(ssc_pdev);
 	if (ret < 0)
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index b8066ef10bb..46dbfd067f7 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -153,8 +153,7 @@ static int cq93vc_resume(struct snd_soc_codec *codec)
 
 static int cq93vc_probe(struct snd_soc_codec *codec)
 {
-	struct davinci_vc *davinci_vc =
-			mfd_get_data(to_platform_device(codec->dev));
+	struct davinci_vc *davinci_vc = codec->dev->platform_data;
 
 	davinci_vc->cq93vc.codec = codec;
 	codec->control_data = davinci_vc;
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index d68ea532cc7..f8c663dcff0 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -262,14 +262,14 @@ static int v253_hangup(struct tty_struct *tty)
 }
 
 /* Line discipline .receive_buf() */
-static void v253_receive(struct tty_struct *tty,
-				const unsigned char *cp, char *fp, int count)
+static unsigned int v253_receive(struct tty_struct *tty,
+				 const unsigned char *cp, char *fp, int count)
 {
 	struct snd_soc_codec *codec = tty->disc_data;
 	struct cx20442_priv *cx20442;
 
 	if (!codec)
-		return;
+		return count;
 
 	cx20442 = snd_soc_codec_get_drvdata(codec);
 
@@ -281,6 +281,8 @@ static void v253_receive(struct tty_struct *tty,
 		codec->hw_write = (hw_write_t)tty->ops->write;
 		codec->card->pop_time = 1;
 	}
+
+	return count;
 }
 
 /* Line discipline .write_wakeup() */
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 575238d68e5..bec788b1261 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -26,7 +26,6 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
-#include <linux/mfd/core.h>
 #include <linux/i2c/twl.h>
 #include <linux/slab.h>
 #include <sound/core.h>
@@ -733,8 +732,7 @@ static int aif_event(struct snd_soc_dapm_widget *w,
 
 static void headset_ramp(struct snd_soc_codec *codec, int ramp)
 {
-	struct twl4030_codec_audio_data *pdata =
-			mfd_get_data(to_platform_device(codec->dev));
+	struct twl4030_codec_audio_data *pdata = codec->dev->platform_data;
 	unsigned char hs_gain, hs_pop;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	/* Base values for ramp delay calculation: 2^19 - 2^26 */
@@ -2299,7 +2297,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = {
 
 static int __devinit twl4030_codec_probe(struct platform_device *pdev)
 {
-	struct twl4030_codec_audio_data *pdata = mfd_get_data(pdev);
+	struct twl4030_codec_audio_data *pdata = pdev->dev.platform_data;
 
 	if (!pdata) {
 		dev_err(&pdev->dev, "platform_data is missing\n");
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index c8a874d0d4c..5836201834d 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -441,8 +441,7 @@ EXPORT_SYMBOL_GPL(wl1273_get_format);
 
 static int wl1273_probe(struct snd_soc_codec *codec)
 {
-	struct wl1273_core **core =
-			mfd_get_data(to_platform_device(codec->dev));
+	struct wl1273_core **core = codec->dev->platform_data;
 	struct wl1273_priv *wl1273;
 	int r;
 
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 736b785e375..fbee556cbf3 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -1378,7 +1378,7 @@ static void wm8400_probe_deferred(struct work_struct *work)
 
 static int wm8400_codec_probe(struct snd_soc_codec *codec)
 {
-	struct wm8400 *wm8400 = mfd_get_data(to_platform_device(codec->dev));
+	struct wm8400 *wm8400 = dev_get_platdata(codec->dev);
 	struct wm8400_priv *priv;
 	int ret;
 	u16 reg;
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
index 13e05a302a9..9259f1f3489 100644
--- a/sound/soc/davinci/davinci-vcif.c
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -205,7 +205,7 @@ static struct snd_soc_dai_driver davinci_vcif_dai = {
 
 static int davinci_vcif_probe(struct platform_device *pdev)
 {
-	struct davinci_vc *davinci_vc = mfd_get_data(pdev);
+	struct davinci_vc *davinci_vc = pdev->dev.platform_data;
 	struct davinci_vcif_dev *davinci_vcif_dev;
 	int ret;
 
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index b5922984eac..99054cf1f68 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -65,14 +65,6 @@ config SND_OMAP_SOC_OVERO
 	  Say Y if you want to add support for SoC audio on the
 	  Gumstix Overo or CompuLab CM-T35
 
-config SND_OMAP_SOC_OMAP2EVM
-	tristate "SoC Audio support for OMAP2EVM board"
-	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP2EVM
-	select SND_OMAP_SOC_MCBSP
-	select SND_SOC_TWL4030
-	help
-	  Say Y if you want to add support for SoC audio on the omap2evm board.
-
 config SND_OMAP_SOC_OMAP3EVM
 	tristate "SoC Audio support for OMAP3EVM board"
 	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index ba9fc650db2..6c2c87eed5b 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -13,7 +13,6 @@ snd-soc-rx51-objs := rx51.o
 snd-soc-ams-delta-objs := ams-delta.o
 snd-soc-osk5912-objs := osk5912.o
 snd-soc-overo-objs := overo.o
-snd-soc-omap2evm-objs := omap2evm.o
 snd-soc-omap3evm-objs := omap3evm.o
 snd-soc-am3517evm-objs := am3517evm.o
 snd-soc-sdp3430-objs := sdp3430.o
diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c
deleted file mode 100644
index 29b60d6796e..00000000000
--- a/sound/soc/omap/omap2evm.c
+++ /dev/null
@@ -1,139 +0,0 @@
-/*
- * omap2evm.c  --  SoC audio machine driver for omap2evm board
- *
- * Author: Arun KS <arunks@mistralsolutions.com>
- *
- * Based on sound/soc/omap/overo.c by Steve Sakoman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <mach/hardware.h>
-#include <mach/gpio.h>
-#include <plat/mcbsp.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-
-static int omap2evm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	int ret;
-
-	/* Set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai,
-				  SND_SOC_DAIFMT_I2S |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0) {
-		printk(KERN_ERR "can't set codec DAI configuration\n");
-		return ret;
-	}
-
-	/* Set cpu DAI configuration */
-	ret = snd_soc_dai_set_fmt(cpu_dai,
-				  SND_SOC_DAIFMT_I2S |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0) {
-		printk(KERN_ERR "can't set cpu DAI configuration\n");
-		return ret;
-	}
-
-	/* Set the codec system clock for DAC and ADC */
-	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
-					    SND_SOC_CLOCK_IN);
-	if (ret < 0) {
-		printk(KERN_ERR "can't set codec system clock\n");
-		return ret;
-	}
-
-	return 0;
-}
-
-static struct snd_soc_ops omap2evm_ops = {
-	.hw_params = omap2evm_hw_params,
-};
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link omap2evm_dai = {
-	.name = "TWL4030",
-	.stream_name = "TWL4030",
-	.cpu_dai_name = "omap-mcbsp-dai.1",
-	.codec_dai_name = "twl4030-hifi",
-	.platform_name = "omap-pcm-audio",
-	.codec_name = "twl4030-codec",
-	.ops = &omap2evm_ops,
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_omap2evm = {
-	.name = "omap2evm",
-	.dai_link = &omap2evm_dai,
-	.num_links = 1,
-};
-
-static struct platform_device *omap2evm_snd_device;
-
-static int __init omap2evm_soc_init(void)
-{
-	int ret;
-
-	if (!machine_is_omap2evm())
-		return -ENODEV;
-	printk(KERN_INFO "omap2evm SoC init\n");
-
-	omap2evm_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!omap2evm_snd_device) {
-		printk(KERN_ERR "Platform device allocation failed\n");
-		return -ENOMEM;
-	}
-
-	platform_set_drvdata(omap2evm_snd_device, &snd_soc_omap2evm);
-
-	ret = platform_device_add(omap2evm_snd_device);
-	if (ret)
-		goto err1;
-
-	return 0;
-
-err1:
-	printk(KERN_ERR "Unable to add platform device\n");
-	platform_device_put(omap2evm_snd_device);
-
-	return ret;
-}
-module_init(omap2evm_soc_init);
-
-static void __exit omap2evm_soc_exit(void)
-{
-	platform_device_unregister(omap2evm_snd_device);
-}
-module_exit(omap2evm_soc_exit);
-
-MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
-MODULE_DESCRIPTION("ALSA SoC omap2evm");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 459566bfcd3..d155cbb58e1 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -1,6 +1,6 @@
 config SND_SOC_SAMSUNG
 	tristate "ASoC support for Samsung"
-	depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_S5P6442 || ARCH_EXYNOS4
+	depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS4
 	select S3C64XX_DMA if ARCH_S3C64XX
 	select S3C2410_DMA if ARCH_S3C2410
 	help
@@ -55,7 +55,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750
 
 config SND_SOC_SAMSUNG_SMDK_WM8580
 	tristate "SoC I2S Audio support for WM8580 on SMDK"
-	depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDK6440 || MACH_SMDK6450 || MACH_SMDK6442 || MACH_SMDKV210 || MACH_SMDKC110)
+	depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDK6440 || MACH_SMDK6450 || MACH_SMDKV210 || MACH_SMDKC110)
 	select SND_SOC_WM8580
 	select SND_SAMSUNG_I2S
 	help
diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c
index 8aacf23d6f3..3d26f6607aa 100644
--- a/sound/soc/samsung/smdk_wm8580.c
+++ b/sound/soc/samsung/smdk_wm8580.c
@@ -249,7 +249,7 @@ static int __init smdk_audio_init(void)
 	int ret;
 	char *str;
 
-	if (machine_is_smdkc100() || machine_is_smdk6442()
+	if (machine_is_smdkc100()
 			|| machine_is_smdkv210() || machine_is_smdkc110()) {
 		smdk.num_links = 3;
 		/* Secondary is at offset SAMSUNG_I2S_SECOFF from Primary */
diff --git a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c
index 24846351118..ac828eff1a6 100644
--- a/sound/usb/6fire/control.c
+++ b/sound/usb/6fire/control.c
@@ -65,6 +65,15 @@ init_data[] = {
 	{ 0 } /* TERMINATING ENTRY */
 };
 
+static const int rates_altsetting[] = { 1, 1, 2, 2, 3, 3 };
+/* values to write to soundcard register for all samplerates */
+static const u16 rates_6fire_vl[] = {0x00, 0x01, 0x00, 0x01, 0x00, 0x01};
+static const u16 rates_6fire_vh[] = {0x11, 0x11, 0x10, 0x10, 0x00, 0x00};
+
+enum {
+	DIGITAL_THRU_ONLY_SAMPLERATE = 3
+};
+
 static void usb6fire_control_master_vol_update(struct control_runtime *rt)
 {
 	struct comm_runtime *comm_rt = rt->chip->comm;
@@ -95,6 +104,67 @@ static void usb6fire_control_opt_coax_update(struct control_runtime *rt)
 	}
 }
 
+static int usb6fire_control_set_rate(struct control_runtime *rt, int rate)
+{
+	int ret;
+	struct usb_device *device = rt->chip->dev;
+	struct comm_runtime *comm_rt = rt->chip->comm;
+
+	if (rate < 0 || rate >= CONTROL_N_RATES)
+		return -EINVAL;
+
+	ret = usb_set_interface(device, 1, rates_altsetting[rate]);
+	if (ret < 0)
+		return ret;
+
+	/* set soundcard clock */
+	ret = comm_rt->write16(comm_rt, 0x02, 0x01, rates_6fire_vl[rate],
+			rates_6fire_vh[rate]);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static int usb6fire_control_set_channels(
+	struct control_runtime *rt, int n_analog_out,
+	int n_analog_in, bool spdif_out, bool spdif_in)
+{
+	int ret;
+	struct comm_runtime *comm_rt = rt->chip->comm;
+
+	/* enable analog inputs and outputs
+	 * (one bit per stereo-channel) */
+	ret = comm_rt->write16(comm_rt, 0x02, 0x02,
+			(1 << (n_analog_out / 2)) - 1,
+			(1 << (n_analog_in / 2)) - 1);
+	if (ret < 0)
+		return ret;
+
+	/* disable digital inputs and outputs */
+	/* TODO: use spdif_x to enable/disable digital channels */
+	ret = comm_rt->write16(comm_rt, 0x02, 0x03, 0x00, 0x00);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static int usb6fire_control_streaming_update(struct control_runtime *rt)
+{
+	struct comm_runtime *comm_rt = rt->chip->comm;
+
+	if (comm_rt) {
+		if (!rt->usb_streaming && rt->digital_thru_switch)
+			usb6fire_control_set_rate(rt,
+				DIGITAL_THRU_ONLY_SAMPLERATE);
+		return comm_rt->write16(comm_rt, 0x02, 0x00, 0x00,
+			(rt->usb_streaming ? 0x01 : 0x00) |
+			(rt->digital_thru_switch ? 0x08 : 0x00));
+	}
+	return -EINVAL;
+}
+
 static int usb6fire_control_master_vol_info(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_info *uinfo)
 {
@@ -195,6 +265,28 @@ static int usb6fire_control_opt_coax_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
+static int usb6fire_control_digital_thru_put(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_value *ucontrol)
+{
+	struct control_runtime *rt = snd_kcontrol_chip(kcontrol);
+	int changed = 0;
+
+	if (rt->digital_thru_switch != ucontrol->value.integer.value[0]) {
+		rt->digital_thru_switch = ucontrol->value.integer.value[0];
+		usb6fire_control_streaming_update(rt);
+		changed = 1;
+	}
+	return changed;
+}
+
+static int usb6fire_control_digital_thru_get(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_value *ucontrol)
+{
+	struct control_runtime *rt = snd_kcontrol_chip(kcontrol);
+	ucontrol->value.integer.value[0] = rt->digital_thru_switch;
+	return 0;
+}
+
 static struct __devinitdata snd_kcontrol_new elements[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -223,6 +315,15 @@ static struct __devinitdata snd_kcontrol_new elements[] = {
 		.get = usb6fire_control_opt_coax_get,
 		.put = usb6fire_control_opt_coax_put
 	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Digital Thru Playback Route",
+		.index = 0,
+		.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info = snd_ctl_boolean_mono_info,
+		.get = usb6fire_control_digital_thru_get,
+		.put = usb6fire_control_digital_thru_put
+	},
 	{}
 };
 
@@ -238,6 +339,9 @@ int __devinit usb6fire_control_init(struct sfire_chip *chip)
 		return -ENOMEM;
 
 	rt->chip = chip;
+	rt->update_streaming = usb6fire_control_streaming_update;
+	rt->set_rate = usb6fire_control_set_rate;
+	rt->set_channels = usb6fire_control_set_channels;
 
 	i = 0;
 	while (init_data[i].type) {
@@ -249,6 +353,7 @@ int __devinit usb6fire_control_init(struct sfire_chip *chip)
 	usb6fire_control_opt_coax_update(rt);
 	usb6fire_control_line_phono_update(rt);
 	usb6fire_control_master_vol_update(rt);
+	usb6fire_control_streaming_update(rt);
 
 	i = 0;
 	while (elements[i].name) {
diff --git a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h
index b534c777ab0..8f5aeead2e3 100644
--- a/sound/usb/6fire/control.h
+++ b/sound/usb/6fire/control.h
@@ -21,12 +21,29 @@ enum {
 	CONTROL_MAX_ELEMENTS = 32
 };
 
+enum {
+	CONTROL_RATE_44KHZ,
+	CONTROL_RATE_48KHZ,
+	CONTROL_RATE_88KHZ,
+	CONTROL_RATE_96KHZ,
+	CONTROL_RATE_176KHZ,
+	CONTROL_RATE_192KHZ,
+	CONTROL_N_RATES
+};
+
 struct control_runtime {
+	int (*update_streaming)(struct control_runtime *rt);
+	int (*set_rate)(struct control_runtime *rt, int rate);
+	int (*set_channels)(struct control_runtime *rt, int n_analog_out,
+		int n_analog_in, bool spdif_out, bool spdif_in);
+
 	struct sfire_chip *chip;
 
 	struct snd_kcontrol *element[CONTROL_MAX_ELEMENTS];
 	bool opt_coax_switch;
 	bool line_phono_switch;
+	bool digital_thru_switch;
+	bool usb_streaming;
 	u8 master_vol;
 };
 
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index 86c1a310376..d47beffedb0 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -3,12 +3,6 @@
  *
  * Firmware loader
  *
- * Currently not working for all devices. To be able to use the device
- * in linux, it is also possible to let the windows driver upload the firmware.
- * For that, start the computer in windows and reboot.
- * As long as the device is connected to the power supply, no firmware reload
- * needs to be performed.
- *
  * Author:	Torsten Schenk <torsten.schenk@zoho.com>
  * Created:	Jan 01, 2011
  * Version:	0.3.0
@@ -21,6 +15,7 @@
  */
 
 #include <linux/firmware.h>
+#include <linux/bitrev.h>
 
 #include "firmware.h"
 #include "chip.h"
@@ -33,32 +28,6 @@ enum {
 	FPGA_BUFSIZE = 512, FPGA_EP = 2
 };
 
-static const u8 BIT_REVERSE_TABLE[256] = {
-	0x00, 0x80, 0x40, 0xc0, 0x20, 0xa0, 0x60, 0xe0, 0x10, 0x90, 0x50,
-	0xd0, 0x30, 0xb0, 0x70, 0xf0, 0x08, 0x88, 0x48, 0xc8, 0x28, 0xa8,
-	0x68, 0xe8, 0x18, 0x98, 0x58, 0xd8, 0x38, 0xb8, 0x78, 0xf8, 0x04,
-	0x84, 0x44, 0xc4, 0x24, 0xa4, 0x64, 0xe4, 0x14, 0x94, 0x54, 0xd4,
-	0x34, 0xb4, 0x74, 0xf4, 0x0c, 0x8c, 0x4c, 0xcc, 0x2c, 0xac, 0x6c,
-	0xec, 0x1c, 0x9c, 0x5c, 0xdc, 0x3c, 0xbc, 0x7c, 0xfc, 0x02, 0x82,
-	0x42, 0xc2, 0x22, 0xa2, 0x62, 0xe2, 0x12, 0x92, 0x52, 0xd2, 0x32,
-	0xb2, 0x72, 0xf2, 0x0a, 0x8a, 0x4a, 0xca, 0x2a, 0xaa, 0x6a, 0xea,
-	0x1a, 0x9a, 0x5a, 0xda, 0x3a, 0xba, 0x7a, 0xfa, 0x06, 0x86, 0x46,
-	0xc6, 0x26, 0xa6, 0x66, 0xe6, 0x16, 0x96, 0x56, 0xd6, 0x36, 0xb6,
-	0x76, 0xf6, 0x0e, 0x8e, 0x4e, 0xce, 0x2e, 0xae, 0x6e, 0xee, 0x1e,
-	0x9e, 0x5e, 0xde, 0x3e, 0xbe, 0x7e, 0xfe, 0x01, 0x81, 0x41, 0xc1,
-	0x21, 0xa1, 0x61, 0xe1, 0x11, 0x91, 0x51, 0xd1, 0x31, 0xb1, 0x71,
-	0xf1, 0x09, 0x89, 0x49, 0xc9, 0x29, 0xa9, 0x69, 0xe9, 0x19, 0x99,
-	0x59, 0xd9, 0x39, 0xb9, 0x79, 0xf9, 0x05, 0x85, 0x45, 0xc5, 0x25,
-	0xa5, 0x65, 0xe5, 0x15, 0x95, 0x55, 0xd5, 0x35, 0xb5, 0x75, 0xf5,
-	0x0d, 0x8d, 0x4d, 0xcd, 0x2d, 0xad, 0x6d, 0xed, 0x1d, 0x9d, 0x5d,
-	0xdd, 0x3d, 0xbd, 0x7d, 0xfd, 0x03, 0x83, 0x43, 0xc3, 0x23, 0xa3,
-	0x63, 0xe3, 0x13, 0x93, 0x53, 0xd3, 0x33, 0xb3, 0x73, 0xf3, 0x0b,
-	0x8b, 0x4b, 0xcb, 0x2b, 0xab, 0x6b, 0xeb, 0x1b, 0x9b, 0x5b, 0xdb,
-	0x3b, 0xbb, 0x7b, 0xfb, 0x07, 0x87, 0x47, 0xc7, 0x27, 0xa7, 0x67,
-	0xe7, 0x17, 0x97, 0x57, 0xd7, 0x37, 0xb7, 0x77, 0xf7, 0x0f, 0x8f,
-	0x4f, 0xcf, 0x2f, 0xaf, 0x6f, 0xef, 0x1f, 0x9f, 0x5f, 0xdf, 0x3f,
-	0xbf, 0x7f, 0xff };
-
 /*
  * wMaxPacketSize of pcm endpoints.
  * keep synced with rates_in_packet_size and rates_out_packet_size in pcm.c
@@ -72,6 +41,10 @@ static const u8 ep_w_max_packet_size[] = {
 	0x94, 0x01, 0x5c, 0x02  /* alt 3: 404 EP2 and 604 EP6 (25 fpp) */
 };
 
+static const u8 known_fw_versions[][4] = {
+	{ 0x03, 0x01, 0x0b, 0x00 }
+};
+
 struct ihex_record {
 	u16 address;
 	u8 len;
@@ -340,7 +313,7 @@ static int usb6fire_fw_fpga_upload(
 
 	while (c != end) {
 		for (i = 0; c != end && i < FPGA_BUFSIZE; i++, c++)
-			buffer[i] = BIT_REVERSE_TABLE[(u8) *c];
+			buffer[i] = byte_rev_table[(u8) *c];
 
 		ret = usb6fire_fw_fpga_write(device, buffer, i);
 		if (ret < 0) {
@@ -363,6 +336,25 @@ static int usb6fire_fw_fpga_upload(
 	return 0;
 }
 
+/* check, if the firmware version the devices has currently loaded
+ * is known by this driver. 'version' needs to have 4 bytes version
+ * info data. */
+static int usb6fire_fw_check(u8 *version)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(known_fw_versions); i++)
+		if (!memcmp(version, known_fw_versions + i, 4))
+			return 0;
+
+	snd_printk(KERN_ERR PREFIX "invalid fimware version in device: "
+			"%02x %02x %02x %02x. "
+			"please reconnect to power. if this failure "
+			"still happens, check your firmware installation.",
+			version[0], version[1], version[2], version[3]);
+	return -EINVAL;
+}
+
 int usb6fire_fw_init(struct usb_interface *intf)
 {
 	int i;
@@ -378,9 +370,7 @@ int usb6fire_fw_init(struct usb_interface *intf)
 				"firmware state.\n");
 		return ret;
 	}
-	if (buffer[0] != 0xeb || buffer[1] != 0xaa || buffer[2] != 0x55
-			|| buffer[4] != 0x03 || buffer[5] != 0x01 || buffer[7]
-			!= 0x00) {
+	if (buffer[0] != 0xeb || buffer[1] != 0xaa || buffer[2] != 0x55) {
 		snd_printk(KERN_ERR PREFIX "unknown device firmware state "
 				"received from device: ");
 		for (i = 0; i < 8; i++)
@@ -389,7 +379,7 @@ int usb6fire_fw_init(struct usb_interface *intf)
 		return -EIO;
 	}
 	/* do we need fpga loader ezusb firmware? */
-	if (buffer[3] == 0x01 && buffer[6] == 0x19) {
+	if (buffer[3] == 0x01) {
 		ret = usb6fire_fw_ezusb_upload(intf,
 				"6fire/dmx6firel2.ihx", 0, NULL, 0);
 		if (ret < 0)
@@ -397,7 +387,10 @@ int usb6fire_fw_init(struct usb_interface *intf)
 		return FW_NOT_READY;
 	}
 	/* do we need fpga firmware and application ezusb firmware? */
-	else if (buffer[3] == 0x02 && buffer[6] == 0x0b) {
+	else if (buffer[3] == 0x02) {
+		ret = usb6fire_fw_check(buffer + 4);
+		if (ret < 0)
+			return ret;
 		ret = usb6fire_fw_fpga_upload(intf, "6fire/dmx6firecf.bin");
 		if (ret < 0)
 			return ret;
@@ -410,8 +403,8 @@ int usb6fire_fw_init(struct usb_interface *intf)
 		return FW_NOT_READY;
 	}
 	/* all fw loaded? */
-	else if (buffer[3] == 0x03 && buffer[6] == 0x0b)
-		return 0;
+	else if (buffer[3] == 0x03)
+		return usb6fire_fw_check(buffer + 4);
 	/* unknown data? */
 	else {
 		snd_printk(KERN_ERR PREFIX "unknown device firmware state "
diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c
index ba62c7468ba..b137b25865c 100644
--- a/sound/usb/6fire/pcm.c
+++ b/sound/usb/6fire/pcm.c
@@ -17,26 +17,23 @@
 #include "pcm.h"
 #include "chip.h"
 #include "comm.h"
+#include "control.h"
 
 enum {
 	OUT_N_CHANNELS = 6, IN_N_CHANNELS = 4
 };
 
 /* keep next two synced with
- * FW_EP_W_MAX_PACKET_SIZE[] and RATES_MAX_PACKET_SIZE */
+ * FW_EP_W_MAX_PACKET_SIZE[] and RATES_MAX_PACKET_SIZE
+ * and CONTROL_RATE_XXX in control.h */
 static const int rates_in_packet_size[] = { 228, 228, 420, 420, 404, 404 };
 static const int rates_out_packet_size[] = { 228, 228, 420, 420, 604, 604 };
 static const int rates[] = { 44100, 48000, 88200, 96000, 176400, 192000 };
-static const int rates_altsetting[] = { 1, 1, 2, 2, 3, 3 };
 static const int rates_alsaid[] = {
 	SNDRV_PCM_RATE_44100, SNDRV_PCM_RATE_48000,
 	SNDRV_PCM_RATE_88200, SNDRV_PCM_RATE_96000,
 	SNDRV_PCM_RATE_176400, SNDRV_PCM_RATE_192000 };
 
-/* values to write to soundcard register for all samplerates */
-static const u16 rates_6fire_vl[] = {0x00, 0x01, 0x00, 0x01, 0x00, 0x01};
-static const u16 rates_6fire_vh[] = {0x11, 0x11, 0x10, 0x10, 0x00, 0x00};
-
 enum { /* settings for pcm */
 	OUT_EP = 6, IN_EP = 2, MAX_BUFSIZE = 128 * 1024
 };
@@ -48,15 +45,6 @@ enum { /* pcm streaming states */
 	STREAM_STOPPING
 };
 
-enum { /* pcm sample rates (also index into RATES_XXX[]) */
-	RATE_44KHZ,
-	RATE_48KHZ,
-	RATE_88KHZ,
-	RATE_96KHZ,
-	RATE_176KHZ,
-	RATE_192KHZ
-};
-
 static const struct snd_pcm_hardware pcm_hw = {
 	.info = SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_INTERLEAVED |
@@ -64,7 +52,7 @@ static const struct snd_pcm_hardware pcm_hw = {
 		SNDRV_PCM_INFO_MMAP_VALID |
 		SNDRV_PCM_INFO_BATCH,
 
-	.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	.formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE,
 
 	.rates = SNDRV_PCM_RATE_44100 |
 		SNDRV_PCM_RATE_48000 |
@@ -87,57 +75,34 @@ static const struct snd_pcm_hardware pcm_hw = {
 static int usb6fire_pcm_set_rate(struct pcm_runtime *rt)
 {
 	int ret;
-	struct usb_device *device = rt->chip->dev;
-	struct comm_runtime *comm_rt = rt->chip->comm;
+	struct control_runtime *ctrl_rt = rt->chip->control;
 
-	if (rt->rate >= ARRAY_SIZE(rates))
-		return -EINVAL;
-	/* disable streaming */
-	ret = comm_rt->write16(comm_rt, 0x02, 0x00, 0x00, 0x00);
+	ctrl_rt->usb_streaming = false;
+	ret = ctrl_rt->update_streaming(ctrl_rt);
 	if (ret < 0) {
 		snd_printk(KERN_ERR PREFIX "error stopping streaming while "
 				"setting samplerate %d.\n", rates[rt->rate]);
 		return ret;
 	}
 
-	ret = usb_set_interface(device, 1, rates_altsetting[rt->rate]);
-	if (ret < 0) {
-		snd_printk(KERN_ERR PREFIX "error setting interface "
-				"altsetting %d for samplerate %d.\n",
-				rates_altsetting[rt->rate], rates[rt->rate]);
-		return ret;
-	}
-
-	/* set soundcard clock */
-	ret = comm_rt->write16(comm_rt, 0x02, 0x01, rates_6fire_vl[rt->rate],
-			rates_6fire_vh[rt->rate]);
+	ret = ctrl_rt->set_rate(ctrl_rt, rt->rate);
 	if (ret < 0) {
 		snd_printk(KERN_ERR PREFIX "error setting samplerate %d.\n",
 				rates[rt->rate]);
 		return ret;
 	}
 
-	/* enable analog inputs and outputs
-	 * (one bit per stereo-channel) */
-	ret = comm_rt->write16(comm_rt, 0x02, 0x02,
-			(1 << (OUT_N_CHANNELS / 2)) - 1,
-			(1 << (IN_N_CHANNELS / 2)) - 1);
+	ret = ctrl_rt->set_channels(ctrl_rt, OUT_N_CHANNELS, IN_N_CHANNELS,
+			false, false);
 	if (ret < 0) {
-		snd_printk(KERN_ERR PREFIX "error initializing analog channels "
+		snd_printk(KERN_ERR PREFIX "error initializing channels "
 				"while setting samplerate %d.\n",
 				rates[rt->rate]);
 		return ret;
 	}
-	/* disable digital inputs and outputs */
-	ret = comm_rt->write16(comm_rt, 0x02, 0x03, 0x00, 0x00);
-	if (ret < 0) {
-		snd_printk(KERN_ERR PREFIX "error initializing digital "
-				"channels while setting samplerate %d.\n",
-				rates[rt->rate]);
-		return ret;
-	}
 
-	ret = comm_rt->write16(comm_rt, 0x02, 0x00, 0x00, 0x01);
+	ctrl_rt->usb_streaming = true;
+	ret = ctrl_rt->update_streaming(ctrl_rt);
 	if (ret < 0) {
 		snd_printk(KERN_ERR PREFIX "error starting streaming while "
 				"setting samplerate %d.\n", rates[rt->rate]);
@@ -168,12 +133,15 @@ static struct pcm_substream *usb6fire_pcm_get_substream(
 static void usb6fire_pcm_stream_stop(struct pcm_runtime *rt)
 {
 	int i;
+	struct control_runtime *ctrl_rt = rt->chip->control;
 
 	if (rt->stream_state != STREAM_DISABLED) {
 		for (i = 0; i < PCM_N_URBS; i++) {
 			usb_kill_urb(&rt->in_urbs[i].instance);
 			usb_kill_urb(&rt->out_urbs[i].instance);
 		}
+		ctrl_rt->usb_streaming = false;
+		ctrl_rt->update_streaming(ctrl_rt);
 		rt->stream_state = STREAM_DISABLED;
 	}
 }
@@ -228,7 +196,7 @@ static void usb6fire_pcm_capture(struct pcm_substream *sub, struct pcm_urb *urb)
 	unsigned int total_length = 0;
 	struct pcm_runtime *rt = snd_pcm_substream_chip(sub->instance);
 	struct snd_pcm_runtime *alsa_rt = sub->instance->runtime;
-	u32 *src = (u32 *) urb->buffer;
+	u32 *src = NULL;
 	u32 *dest = (u32 *) (alsa_rt->dma_area + sub->dma_off
 			* (alsa_rt->frame_bits >> 3));
 	u32 *dest_end = (u32 *) (alsa_rt->dma_area + alsa_rt->buffer_size
@@ -244,7 +212,12 @@ static void usb6fire_pcm_capture(struct pcm_substream *sub, struct pcm_urb *urb)
 		else
 			frame_count = 0;
 
-		src = (u32 *) (urb->buffer + total_length);
+		if (alsa_rt->format == SNDRV_PCM_FORMAT_S24_LE)
+			src = (u32 *) (urb->buffer + total_length);
+		else if (alsa_rt->format == SNDRV_PCM_FORMAT_S32_LE)
+			src = (u32 *) (urb->buffer - 1 + total_length);
+		else
+			return;
 		src++; /* skip leading 4 bytes of every packet */
 		total_length += urb->packets[i].length;
 		for (frame = 0; frame < frame_count; frame++) {
@@ -274,9 +247,18 @@ static void usb6fire_pcm_playback(struct pcm_substream *sub,
 			* (alsa_rt->frame_bits >> 3));
 	u32 *src_end = (u32 *) (alsa_rt->dma_area + alsa_rt->buffer_size
 			* (alsa_rt->frame_bits >> 3));
-	u32 *dest = (u32 *) urb->buffer;
+	u32 *dest;
 	int bytes_per_frame = alsa_rt->channels << 2;
 
+	if (alsa_rt->format == SNDRV_PCM_FORMAT_S32_LE)
+		dest = (u32 *) (urb->buffer - 1);
+	else if (alsa_rt->format == SNDRV_PCM_FORMAT_S24_LE)
+		dest = (u32 *) (urb->buffer);
+	else {
+		snd_printk(KERN_ERR PREFIX "Unknown sample format.");
+		return;
+	}
+
 	for (i = 0; i < PCM_N_PACKETS_PER_URB; i++) {
 		/* at least 4 header bytes for valid packet.
 		 * after that: 32 bits per sample for analog channels */
@@ -456,7 +438,7 @@ static int usb6fire_pcm_close(struct snd_pcm_substream *alsa_sub)
 		/* all substreams closed? if so, stop streaming */
 		if (!rt->playback.instance && !rt->capture.instance) {
 			usb6fire_pcm_stream_stop(rt);
-			rt->rate = -1;
+			rt->rate = ARRAY_SIZE(rates);
 		}
 	}
 	mutex_unlock(&rt->stream_mutex);
@@ -480,7 +462,6 @@ static int usb6fire_pcm_prepare(struct snd_pcm_substream *alsa_sub)
 	struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub);
 	struct pcm_substream *sub = usb6fire_pcm_get_substream(alsa_sub);
 	struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime;
-	int i;
 	int ret;
 
 	if (rt->panic)
@@ -493,12 +474,10 @@ static int usb6fire_pcm_prepare(struct snd_pcm_substream *alsa_sub)
 	sub->period_off = 0;
 
 	if (rt->stream_state == STREAM_DISABLED) {
-		for (i = 0; i < ARRAY_SIZE(rates); i++)
-			if (alsa_rt->rate == rates[i]) {
-				rt->rate = i;
+		for (rt->rate = 0; rt->rate < ARRAY_SIZE(rates); rt->rate++)
+			if (alsa_rt->rate == rates[rt->rate])
 				break;
-			}
-		if (i == ARRAY_SIZE(rates)) {
+		if (rt->rate == ARRAY_SIZE(rates)) {
 			mutex_unlock(&rt->stream_mutex);
 			snd_printk("invalid rate %d in prepare.\n",
 					alsa_rt->rate);
@@ -613,7 +592,7 @@ int __devinit usb6fire_pcm_init(struct sfire_chip *chip)
 
 	rt->chip = chip;
 	rt->stream_state = STREAM_DISABLED;
-	rt->rate = -1;
+	rt->rate = ARRAY_SIZE(rates);
 	init_waitqueue_head(&rt->stream_wait_queue);
 	mutex_init(&rt->stream_mutex);
 
diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig
index 97724d8fa9f..8beb77563da 100644
--- a/sound/usb/Kconfig
+++ b/sound/usb/Kconfig
@@ -100,19 +100,17 @@ config SND_USB_US122L
 
 config SND_USB_6FIRE
         tristate "TerraTec DMX 6Fire USB"
-        depends on EXPERIMENTAL
         select FW_LOADER
+        select BITREVERSE
         select SND_RAWMIDI
         select SND_PCM
         help
           Say Y here to include support for TerraTec 6fire DMX USB interface.
 
           You will need firmware files in order to be able to use the device
-          after it has been coldstarted. This driver currently does not support
-          firmware loading for all devices. If you own such a device,
-          you could start windows and let the windows driver upload
-          the firmware. As long as you do not unplug your device from power,
-          it should be usable.
+          after it has been coldstarted. An install script for the firmware
+          and further help can be found at
+          http://sixfireusb.sourceforge.net
 
 endif	# SND_USB
 
diff --git a/sound/usb/card.c b/sound/usb/card.c
index a90662af2d6..220c6167dd8 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -48,6 +48,7 @@
 #include <linux/usb/audio.h>
 #include <linux/usb/audio-v2.h>
 
+#include <sound/control.h>
 #include <sound/core.h>
 #include <sound/info.h>
 #include <sound/pcm.h>
@@ -492,14 +493,6 @@ static void *snd_usb_audio_probe(struct usb_device *dev,
 		}
 	}
 
-	chip->txfr_quirk = 0;
-	err = 1; /* continue */
-	if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) {
-		/* need some special handlings */
-		if ((err = snd_usb_create_quirk(chip, intf, &usb_audio_driver, quirk)) < 0)
-			goto __error;
-	}
-
 	/*
 	 * For devices with more than one control interface, we assume the
 	 * first contains the audio controls. We might need a more specific
@@ -508,6 +501,14 @@ static void *snd_usb_audio_probe(struct usb_device *dev,
 	if (!chip->ctrl_intf)
 		chip->ctrl_intf = alts;
 
+	chip->txfr_quirk = 0;
+	err = 1; /* continue */
+	if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) {
+		/* need some special handlings */
+		if ((err = snd_usb_create_quirk(chip, intf, &usb_audio_driver, quirk)) < 0)
+			goto __error;
+	}
+
 	if (err > 0) {
 		/* create normal USB audio interfaces */
 		if (snd_usb_create_streams(chip, ifnum) < 0 ||
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 7754a103454..075195e8661 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -104,6 +104,15 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id)
 	int err;
 	unsigned char data;
 	struct usb_device *dev = chip->dev;
+	struct uac_clock_source_descriptor *cs_desc =
+		snd_usb_find_clock_source(chip->ctrl_intf, source_id);
+
+	if (!cs_desc)
+		return 0;
+
+	/* If a clock source can't tell us whether it's valid, we assume it is */
+	if (!uac2_control_is_readable(cs_desc->bmControls, UAC2_CS_CONTROL_CLOCK_VALID))
+		return 1;
 
 	err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
 			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
@@ -114,7 +123,7 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id)
 	if (err < 0) {
 		snd_printk(KERN_WARNING "%s(): cannot get clock validity for id %d\n",
 			   __func__, source_id);
-		return err;
+		return 0;
 	}
 
 	return !!data;
diff --git a/sound/usb/debug.h b/sound/usb/debug.h
index 343ec2d9ee6..58030176f00 100644
--- a/sound/usb/debug.h
+++ b/sound/usb/debug.h
@@ -8,7 +8,7 @@
 #ifdef HW_CONST_DEBUG
 #define hwc_debug(fmt, args...) printk(KERN_DEBUG fmt, ##args)
 #else
-#define hwc_debug(fmt, args...) /**/
+#define hwc_debug(fmt, args...) do { } while(0)
 #endif
 
 #endif /* __USBAUDIO_DEBUG_H */
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 5b792d2c806..8d042dce0d1 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -30,6 +30,7 @@
 #include "helper.h"
 #include "debug.h"
 #include "clock.h"
+#include "format.h"
 
 /*
  * parse the audio format type I descriptor
@@ -176,9 +177,11 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
 			if (!rate)
 				continue;
 			/* C-Media CM6501 mislabels its 96 kHz altsetting */
+			/* Terratec Aureon 7.1 USB C-Media 6206, too */
 			if (rate == 48000 && nr_rates == 1 &&
 			    (chip->usb_id == USB_ID(0x0d8c, 0x0201) ||
-			     chip->usb_id == USB_ID(0x0d8c, 0x0102)) &&
+			     chip->usb_id == USB_ID(0x0d8c, 0x0102) ||
+			     chip->usb_id == USB_ID(0x0ccd, 0x00b1)) &&
 			    fp->altsetting == 5 && fp->maxpacksize == 392)
 				rate = 96000;
 			/* Creative VF0470 Live Cam reports 16 kHz instead of 8kHz */
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 6ec33b62e6c..c22fa76e363 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -86,16 +86,6 @@ struct mixer_build {
 	const struct usbmix_selector_map *selector_map;
 };
 
-enum {
-	USB_MIXER_BOOLEAN,
-	USB_MIXER_INV_BOOLEAN,
-	USB_MIXER_S8,
-	USB_MIXER_U8,
-	USB_MIXER_S16,
-	USB_MIXER_U16,
-};
-
-
 /*E-mu 0202/0404/0204 eXtension Unit(XU) control*/
 enum {
 	USB_XU_CLOCK_RATE 		= 0xe301,
@@ -535,20 +525,21 @@ static int check_matrix_bitmap(unsigned char *bmap, int ich, int och, int num_ou
  * if failed, give up and free the control instance.
  */
 
-static int add_control_to_empty(struct mixer_build *state, struct snd_kcontrol *kctl)
+int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer,
+			      struct snd_kcontrol *kctl)
 {
 	struct usb_mixer_elem_info *cval = kctl->private_data;
 	int err;
 
-	while (snd_ctl_find_id(state->chip->card, &kctl->id))
+	while (snd_ctl_find_id(mixer->chip->card, &kctl->id))
 		kctl->id.index++;
-	if ((err = snd_ctl_add(state->chip->card, kctl)) < 0) {
+	if ((err = snd_ctl_add(mixer->chip->card, kctl)) < 0) {
 		snd_printd(KERN_ERR "cannot add control (err = %d)\n", err);
 		return err;
 	}
 	cval->elem_id = &kctl->id;
-	cval->next_id_elem = state->mixer->id_elems[cval->id];
-	state->mixer->id_elems[cval->id] = cval;
+	cval->next_id_elem = mixer->id_elems[cval->id];
+	mixer->id_elems[cval->id] = cval;
 	return 0;
 }
 
@@ -984,6 +975,9 @@ static struct snd_kcontrol_new usb_feature_unit_ctl_ro = {
 	.put = NULL,
 };
 
+/* This symbol is exported in order to allow the mixer quirks to
+ * hook up to the standard feature unit control mechanism */
+struct snd_kcontrol_new *snd_usb_feature_unit_ctl = &usb_feature_unit_ctl;
 
 /*
  * build a feature control
@@ -1097,11 +1091,13 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 		append_ctl_name(kctl, control == UAC_FU_MUTE ?
 				" Switch" : " Volume");
 		if (control == UAC_FU_VOLUME) {
-			kctl->tlv.c = mixer_vol_tlv;
-			kctl->vd[0].access |= 
-				SNDRV_CTL_ELEM_ACCESS_TLV_READ |
-				SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK;
 			check_mapped_dB(map, cval);
+			if (cval->dBmin < cval->dBmax) {
+				kctl->tlv.c = mixer_vol_tlv;
+				kctl->vd[0].access |= 
+					SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+					SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK;
+			}
 		}
 		break;
 
@@ -1174,7 +1170,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 
 	snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n",
 		    cval->id, kctl->id.name, cval->channels, cval->min, cval->max, cval->res);
-	add_control_to_empty(state, kctl);
+	snd_usb_mixer_add_control(state->mixer, kctl);
 }
 
 
@@ -1338,7 +1334,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state,
 
 	snd_printdd(KERN_INFO "[%d] MU [%s] ch = %d, val = %d/%d\n",
 		    cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
-	add_control_to_empty(state, kctl);
+	snd_usb_mixer_add_control(state->mixer, kctl);
 }
 
 
@@ -1639,7 +1635,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw
 
 		snd_printdd(KERN_INFO "[%d] PU [%s] ch = %d, val = %d/%d\n",
 			    cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
-		if ((err = add_control_to_empty(state, kctl)) < 0)
+		if ((err = snd_usb_mixer_add_control(state->mixer, kctl)) < 0)
 			return err;
 	}
 	return 0;
@@ -1856,7 +1852,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void
 
 	snd_printdd(KERN_INFO "[%d] SU [%s] items = %d\n",
 		    cval->id, kctl->id.name, desc->bNrInPins);
-	if ((err = add_control_to_empty(state, kctl)) < 0)
+	if ((err = snd_usb_mixer_add_control(state->mixer, kctl)) < 0)
 		return err;
 
 	return 0;
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index b4a2c8165e4..ae1a14dcfe8 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -24,7 +24,16 @@ struct usb_mixer_interface {
 	u8 xonar_u1_status;
 };
 
-#define MAX_CHANNELS	10	/* max logical channels */
+#define MAX_CHANNELS	16	/* max logical channels */
+
+enum {
+	USB_MIXER_BOOLEAN,
+	USB_MIXER_INV_BOOLEAN,
+	USB_MIXER_S8,
+	USB_MIXER_U8,
+	USB_MIXER_S16,
+	USB_MIXER_U16,
+};
 
 struct usb_mixer_elem_info {
 	struct usb_mixer_interface *mixer;
@@ -55,4 +64,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval,
 void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer);
 int snd_usb_mixer_activate(struct usb_mixer_interface *mixer);
 
+int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer,
+			      struct snd_kcontrol *kctl);
+
 #endif /* __USBMIXER_H */
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 73dcc8256bc..3d0f4873112 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -40,6 +40,8 @@
 #include "mixer_quirks.h"
 #include "helper.h"
 
+extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl;
+
 /*
  * Sound Blaster remote control configuration
  *
@@ -61,6 +63,7 @@ static const struct rc_config {
 	{ USB_ID(0x041e, 0x3020), 2, 1, 6, 6,  18, 0x0013 }, /* Audigy 2 NX  */
 	{ USB_ID(0x041e, 0x3040), 2, 2, 6, 6,  2,  0x6e91 }, /* Live! 24-bit */
 	{ USB_ID(0x041e, 0x3042), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 */
+	{ USB_ID(0x041e, 0x30df), 0, 1, 1, 1,  1,  0x000d }, /* Usb X-Fi S51 Pro */
 	{ USB_ID(0x041e, 0x3048), 2, 2, 6, 6,  2,  0x6e91 }, /* Toshiba SB0500 */
 };
 
@@ -188,6 +191,12 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 			      usb_sndctrlpipe(mixer->chip->dev, 0), 0x24,
 			      USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
 			      !value, 0, NULL, 0, 100);
+	/* USB X-Fi S51 Pro */
+	if (mixer->chip->usb_id == USB_ID(0x041e, 0x30df))
+		err = snd_usb_ctl_msg(mixer->chip->dev,
+			      usb_sndctrlpipe(mixer->chip->dev, 0), 0x24,
+			      USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
+			      !value, 0, NULL, 0, 100);
 	else
 		err = snd_usb_ctl_msg(mixer->chip->dev,
 			      usb_sndctrlpipe(mixer->chip->dev, 0), 0x24,
@@ -234,9 +243,13 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer)
 		/* USB X-Fi S51 doesn't have a CMSS LED */
 		if ((mixer->chip->usb_id == USB_ID(0x041e, 0x3042)) && i == 0)
 			continue;
+		/* USB X-Fi S51 Pro doesn't have one either */
+		if ((mixer->chip->usb_id == USB_ID(0x041e, 0x30df)) && i == 0)
+			continue;
 		if (i > 1 && /* Live24ext has 2 LEDs only */
 			(mixer->chip->usb_id == USB_ID(0x041e, 0x3040) ||
 			 mixer->chip->usb_id == USB_ID(0x041e, 0x3042) ||
+			 mixer->chip->usb_id == USB_ID(0x041e, 0x30df) ||
 			 mixer->chip->usb_id == USB_ID(0x041e, 0x3048)))
 			break; 
 		err = snd_ctl_add(mixer->chip->card,
@@ -481,6 +494,69 @@ static int snd_nativeinstruments_create_mixer(struct usb_mixer_interface *mixer,
 	return err;
 }
 
+/* M-Audio FastTrack Ultra quirks */
+
+/* private_free callback */
+static void usb_mixer_elem_free(struct snd_kcontrol *kctl)
+{
+	kfree(kctl->private_data);
+	kctl->private_data = NULL;
+}
+
+static int snd_maudio_ftu_create_ctl(struct usb_mixer_interface *mixer,
+				     int in, int out, const char *name)
+{
+	struct usb_mixer_elem_info *cval;
+	struct snd_kcontrol *kctl;
+
+	cval = kzalloc(sizeof(*cval), GFP_KERNEL);
+	if (!cval)
+		return -ENOMEM;
+
+	cval->id = 5;
+	cval->mixer = mixer;
+	cval->val_type = USB_MIXER_S16;
+	cval->channels = 1;
+	cval->control = out + 1;
+	cval->cmask = 1 << in;
+
+	kctl = snd_ctl_new1(snd_usb_feature_unit_ctl, cval);
+	if (!kctl) {
+		kfree(cval);
+		return -ENOMEM;
+	}
+
+	snprintf(kctl->id.name, sizeof(kctl->id.name), name);
+	kctl->private_free = usb_mixer_elem_free;
+	return snd_usb_mixer_add_control(mixer, kctl);
+}
+
+static int snd_maudio_ftu_create_mixer(struct usb_mixer_interface *mixer)
+{
+	char name[64];
+	int in, out, err;
+
+	for (out = 0; out < 8; out++) {
+		for (in = 0; in < 8; in++) {
+			snprintf(name, sizeof(name),
+				 "AIn%d - Out%d Capture Volume", in  + 1, out + 1);
+			err = snd_maudio_ftu_create_ctl(mixer, in, out, name);
+			if (err < 0)
+				return err;
+		}
+
+		for (in = 8; in < 16; in++) {
+			snprintf(name, sizeof(name),
+				 "DIn%d - Out%d Playback Volume", in - 7, out + 1);
+			err = snd_maudio_ftu_create_ctl(mixer, in, out, name);
+			if (err < 0)
+				return err;
+		}
+	}
+
+	return 0;
+}
+
 void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
 			       unsigned char samplerate_id)
 {
@@ -512,6 +588,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 	case USB_ID(0x041e, 0x3020):
 	case USB_ID(0x041e, 0x3040):
 	case USB_ID(0x041e, 0x3042):
+	case USB_ID(0x041e, 0x30df):
 	case USB_ID(0x041e, 0x3048):
 		err = snd_audigy2nx_controls_create(mixer);
 		if (err < 0)
@@ -521,6 +598,11 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 					      snd_audigy2nx_proc_read);
 		break;
 
+	case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra */
+	case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */
+		err = snd_maudio_ftu_create_mixer(mixer);
+		break;
+
 	case USB_ID(0x0b05, 0x1739):
 	case USB_ID(0x0b05, 0x1743):
 		err = snd_xonar_u1_controls_create(mixer);
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index c66d3f64dcf..0b2ae8e1c02 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1651,6 +1651,32 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
+{
+	USB_DEVICE(0x0582, 0x0127),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		/* .vendor_name = "Roland", */
+		/* .product_name = "GR-55", */
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_MIDI_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 
 /* Guillemot devices */
 {
@@ -1953,7 +1979,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
-	USB_DEVICE(0x0763, 0x2080),
+	USB_DEVICE_VENDOR_SPEC(0x0763, 0x2080),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
 		/* .vendor_name = "M-Audio", */
 		/* .product_name = "Fast Track Ultra", */
@@ -1962,7 +1988,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		.data = & (const struct snd_usb_audio_quirk[]) {
 			{
 				.ifnum = 0,
-				.type = QUIRK_IGNORE_INTERFACE
+				.type = QUIRK_AUDIO_STANDARD_MIXER,
 			},
 			{
 				.ifnum = 1,
@@ -2020,7 +2046,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
-	USB_DEVICE(0x0763, 0x2081),
+	USB_DEVICE_VENDOR_SPEC(0x0763, 0x2081),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
 		/* .vendor_name = "M-Audio", */
 		/* .product_name = "Fast Track Ultra 8R", */
@@ -2029,7 +2055,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		.data = & (const struct snd_usb_audio_quirk[]) {
 			{
 				.ifnum = 0,
-				.type = QUIRK_IGNORE_INTERFACE
+				.type = QUIRK_AUDIO_STANDARD_MIXER,
 			},
 			{
 				.ifnum = 1,
@@ -2179,6 +2205,17 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 
+/* KORG devices */
+{
+	USB_DEVICE_VENDOR_SPEC(0x0944, 0x0200),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.vendor_name = "KORG, Inc.",
+		/* .product_name = "PANDORA PX5D", */
+		.ifnum = 3,
+		.type = QUIRK_MIDI_STANDARD_INTERFACE,
+	}
+},
+
 /* AKAI devices */
 {
 	USB_DEVICE(0x09e8, 0x0062),
@@ -2332,6 +2369,12 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 
 /* Native Instruments MK2 series */
 {
+	/* Komplete Audio 6 */
+	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+	.idVendor = 0x17cc,
+	.idProduct = 0x1000,
+},
+{
 	/* Traktor Audio 6 */
 	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
 	.idVendor = 0x17cc,
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index ec07e62e53f..2e969cbb393 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -19,6 +19,7 @@
 #include <linux/usb.h>
 #include <linux/usb/audio.h>
 
+#include <sound/control.h>
 #include <sound/core.h>
 #include <sound/info.h>
 #include <sound/pcm.h>
@@ -263,6 +264,20 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
 }
 
 /*
+ * Create a standard mixer for the specified interface.
+ */
+static int create_standard_mixer_quirk(struct snd_usb_audio *chip,
+				       struct usb_interface *iface,
+				       struct usb_driver *driver,
+				       const struct snd_usb_audio_quirk *quirk)
+{
+	if (quirk->ifnum < 0)
+		return 0;
+
+	return snd_usb_create_mixer(chip, quirk->ifnum, 0);
+}
+
+/*
  * audio-interface quirks
  *
  * returns zero if no standard audio/MIDI parsing is needed.
@@ -294,7 +309,8 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
 		[QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk,
 		[QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk,
 		[QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk,
-		[QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk
+		[QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk,
+		[QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk,
 	};
 
 	if (quirk->type < QUIRK_TYPE_COUNT) {
@@ -533,12 +549,14 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev,
 
 	case USB_ID(0x0d8c, 0x0102):
 		/* C-Media CM6206 / CM106-Like Sound Device */
+	case USB_ID(0x0ccd, 0x00b1): /* Terratec Aureon 7.1 USB */
 		return snd_usb_cm6206_boot_quirk(dev);
 
 	case USB_ID(0x133e, 0x0815):
 		/* Access Music VirusTI Desktop */
 		return snd_usb_accessmusic_boot_quirk(dev);
 
+	case USB_ID(0x17cc, 0x1000): /* Komplete Audio 6 */
 	case USB_ID(0x17cc, 0x1010): /* Traktor Audio 6 */
 	case USB_ID(0x17cc, 0x1020): /* Traktor Audio 10 */
 		return snd_usb_nativeinstruments_boot_quirk(dev);
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 32f2a97f2f1..1e79986b577 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -84,6 +84,7 @@ enum quirk_type {
 	QUIRK_AUDIO_FIXED_ENDPOINT,
 	QUIRK_AUDIO_EDIROL_UAXX,
 	QUIRK_AUDIO_ALIGN_TRANSFER,
+	QUIRK_AUDIO_STANDARD_MIXER,
 
 	QUIRK_TYPE_COUNT
 };