Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (297 commits)
  ALSA: asihpi - Replace with snd_ctl_boolean_mono_info()
  ALSA: asihpi - HPI version 4.08
  ALSA: asihpi - Add volume mute controls
  ALSA: asihpi - Control name updates
  ALSA: asihpi - Use size_t for sizeof result
  ALSA: asihpi - Explicitly include mutex.h
  ALSA: asihpi - Add new node and message defines
  ALSA: asihpi - Make local function static
  ALSA: asihpi - Fix minor typos and spelling
  ALSA: asihpi - Remove unused structures, macros and functions
  ALSA: asihpi - Remove spurious adapter index check
  ALSA: asihpi - Revise snd_pcm_debug_name, get rid of DEBUG_NAME macro
  ALSA: asihpi - DSP code loader API now independent of OS
  ALSA: asihpi - Remove controlex structs and associated special data transfer code
  ALSA: asihpi - Increase request and response buffer sizes
  ALSA: asihpi - Give more meaningful name to hpi request message type
  ALSA: usb-audio - Add quirk for  Roland / BOSS BR-800
  ALSA: hda - Remove a superfluous argument of via_auto_init_output()
  ALSA: hda - Fix indep-HP path (de-)activation for VT1708* codecs
  ALSA: hda - Add documentation for codec-specific mixer controls
  ...

Documentation/DocBook/writing-an-alsa-driver.tmpl

@@ -1164,7 +1164,7 @@
           }
           chip->port = pci_resource_start(pci, 0);
           if (request_irq(pci->irq, snd_mychip_interrupt,
-                          IRQF_SHARED, "My Chip", chip)) {
+                          IRQF_SHARED, KBUILD_MODNAME, chip)) {
                   printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
                   snd_mychip_free(chip);
                   return -EBUSY;
@@ -1197,7 +1197,7 @@
 
   /* pci_driver definition */
   static struct pci_driver driver = {
-          .name = "My Own Chip",
+          .name = KBUILD_MODNAME,
           .id_table = snd_mychip_ids,
           .probe = snd_mychip_probe,
           .remove = __devexit_p(snd_mychip_remove),
@@ -1340,7 +1340,7 @@
           <programlisting>
 <![CDATA[
   if (request_irq(pci->irq, snd_mychip_interrupt,
-                  IRQF_SHARED, "My Chip", chip)) {
+                  IRQF_SHARED, KBUILD_MODNAME, chip)) {
           printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
           snd_mychip_free(chip);
           return -EBUSY;
@@ -1616,7 +1616,7 @@
           <programlisting>
 <![CDATA[
   static struct pci_driver driver = {
-          .name = "My Own Chip",
+          .name = KBUILD_MODNAME,
           .id_table = snd_mychip_ids,
           .probe = snd_mychip_probe,
           .remove = __devexit_p(snd_mychip_remove),
@@ -5816,7 +5816,7 @@ struct _snd_pcm_runtime {
         <programlisting>
 <![CDATA[
   static struct pci_driver driver = {
-          .name = "My Chip",
+          .name = KBUILD_MODNAME,
           .id_table = snd_my_ids,
           .probe = snd_my_probe,
           .remove = __devexit_p(snd_my_remove),

Documentation/sound/alsa/HD-Audio-Controls.txt

@@ -0,0 +1,100 @@
+This file explains the codec-specific mixer controls.
+
+Realtek codecs
+--------------
+
+* Channel Mode
+  This is an enum control to change the surround-channel setup,
+  appears only when the surround channels are available.
+  It gives the number of channels to be used, "2ch", "4ch", "6ch",
+  and "8ch".  According to the configuration, this also controls the
+  jack-retasking of multi-I/O jacks.
+
+* Auto-Mute Mode
+  This is an enum control to change the auto-mute behavior of the
+  headphone and line-out jacks.  If built-in speakers and headphone
+  and/or line-out jacks are available on a machine, this controls
+  appears.
+  When there are only either headphones or line-out jacks, it gives
+  "Disabled" and "Enabled" state.  When enabled, the speaker is muted
+  automatically when a jack is plugged.
+
+  When both headphone and line-out jacks are present, it gives
+  "Disabled", "Speaker Only" and "Line-Out+Speaker".  When
+  speaker-only is chosen, plugging into a headphone or a line-out jack
+  mutes the speakers, but not line-outs.  When line-out+speaker is
+  selected, plugging to a headphone jack mutes both speakers and
+  line-outs.
+
+
+IDT/Sigmatel codecs
+-------------------
+
+* Analog Loopback
+  This control enables/disables the analog-loopback circuit.  This
+  appears only when "loopback" is set to true in a codec hint
+  (see HD-Audio.txt).  Note that on some codecs the analog-loopback
+  and the normal PCM playback are exclusive, i.e. when this is on, you
+  won't hear any PCM stream.
+
+* Swap Center/LFE
+  Swaps the center and LFE channel order.  Normally, the left
+  corresponds to the center and the right to the LFE.  When this is
+  ON, the left to the LFE and the right to the center.
+
+* Headphone as Line Out
+  When this control is ON, treat the headphone jacks as line-out
+  jacks.  That is, the headphone won't auto-mute the other line-outs,
+  and no HP-amp is set to the pins.
+
+* Mic Jack Mode, Line Jack Mode, etc
+  These enum controls the direction and the bias of the input jack
+  pins.  Depending on the jack type, it can set as "Mic In" and "Line 
+  In", for determining the input bias, or it can be set to "Line Out"
+  when the pin is a multi-I/O jack for surround channels.
+
+
+VIA codecs
+----------
+
+* Smart 5.1
+  An enum control to re-task the multi-I/O jacks for surround outputs.
+  When it's ON, the corresponding input jacks (usually a line-in and a
+  mic-in) are switched as the surround and the CLFE output jacks.
+
+* Independent HP
+  When this enum control is enabled, the headphone output is routed
+  from an individual stream (the third PCM such as hw:0,2) instead of
+  the primary stream.  In the case the headphone DAC is shared with a
+  side or a CLFE-channel DAC, the DAC is switched to the headphone
+  automatically.
+
+* Loopback Mixing
+  An enum control to determine whether the analog-loopback route is
+  enabled or not.  When it's enabled, the analog-loopback is mixed to
+  the front-channel.  Also, the same route is used for the headphone
+  and speaker outputs.  As a side-effect, when this mode is set, the
+  individual volume controls will be no longer available for
+  headphones and speakers because there is only one DAC connected to a
+  mixer widget.
+
+* Dynamic Power-Control
+  This control determines whether the dynamic power-control per jack
+  detection is enabled or not.  When enabled, the widgets power state
+  (D0/D3) are changed dynamically depending on the jack plugging
+  state for saving power consumptions.  However, if your system
+  doesn't provide a proper jack-detection, this won't work; in such a
+  case, turn this control OFF.
+
+* Jack Detect
+  This control is provided only for VT1708 codec which gives no proper
+  unsolicited event per jack plug.  When this is on, the driver polls
+  the jack detection so that the headphone auto-mute can work, while 
+  turning this off would reduce the power consumption.
+
+
+Conexant codecs
+---------------
+
+* Auto-Mute Mode
+  See Reatek codecs.

MAINTAINERS

@@ -534,6 +534,8 @@ L:	device-drivers-devel@blackfin.uclinux.org
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 W:	http://wiki.analog.com/
 S:	Supported
+F:	sound/soc/codecs/adau*
+F:	sound/soc/codecs/adav*
 F:	sound/soc/codecs/ad1*
 F:	sound/soc/codecs/ssm*
 

include/linux/pci_ids.h

@@ -1308,6 +1308,7 @@
 #define PCI_SUBDEVICE_ID_CREATIVE_SB08801	0x0041
 #define PCI_SUBDEVICE_ID_CREATIVE_SB08802	0x0042
 #define PCI_SUBDEVICE_ID_CREATIVE_SB08803	0x0043
+#define PCI_SUBDEVICE_ID_CREATIVE_SB1270	0x0062
 #define PCI_SUBDEVICE_ID_CREATIVE_HENDRIX	0x6000
 
 #define PCI_VENDOR_ID_ECTIVA		0x1102 /* duplicate: CREATIVE */

include/sound/rawmidi.h

@@ -27,6 +27,7 @@
 #include <linux/spinlock.h>
 #include <linux/wait.h>
 #include <linux/mutex.h>
+#include <linux/workqueue.h>
 
 #if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
 #include "seq_device.h"
@@ -63,6 +64,7 @@ struct snd_rawmidi_global_ops {
 };
 
 struct snd_rawmidi_runtime {
+	struct snd_rawmidi_substream *substream;
 	unsigned int drain: 1,	/* drain stage */
 		     oss: 1;	/* OSS compatible mode */
 	/* midi stream buffer */
@@ -79,7 +81,7 @@ struct snd_rawmidi_runtime {
 	/* event handler (new bytes, input only) */
 	void (*event)(struct snd_rawmidi_substream *substream);
 	/* defers calls to event [input] or ops->trigger [output] */
-	struct tasklet_struct tasklet;
+	struct work_struct event_work;
 	/* private data */
 	void *private_data;
 	void (*private_free)(struct snd_rawmidi_substream *substream);

include/sound/soc-dai.h

@@ -209,6 +209,10 @@ struct snd_soc_dai_driver {
 	struct snd_soc_pcm_stream capture;
 	struct snd_soc_pcm_stream playback;
 	unsigned int symmetric_rates:1;
+
+	/* probe ordering - for components with runtime dependencies */
+	int probe_order;
+	int remove_order;
 };
 
 /*

include/sound/soc-dapm.h

@@ -348,6 +348,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
 void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
 int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
 			    const struct snd_soc_dapm_route *route, int num);
+int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
+			     const struct snd_soc_dapm_route *route, int num);
 
 /* dapm events */
 int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
@@ -429,6 +431,7 @@ struct snd_soc_dapm_path {
 	/* status */
 	u32 connect:1;	/* source and sink widgets are connected */
 	u32 walked:1;	/* path has been walked */
+	u32 weak:1;	/* path ignored for power management */
 
 	int (*connected)(struct snd_soc_dapm_widget *source,
 			 struct snd_soc_dapm_widget *sink);
@@ -444,6 +447,7 @@ struct snd_soc_dapm_widget {
 	char *name;		/* widget name */
 	char *sname;	/* stream name */
 	struct snd_soc_codec *codec;
+	struct snd_soc_platform *platform;
 	struct list_head list;
 	struct snd_soc_dapm_context *dapm;
 
@@ -507,10 +511,11 @@ struct snd_soc_dapm_context {
 
 	struct device *dev; /* from parent - for debug */
 	struct snd_soc_codec *codec; /* parent codec */
+	struct snd_soc_platform *platform; /* parent platform */
 	struct snd_soc_card *card; /* parent card */
 
 	/* used during DAPM updates */
-	int dev_power;
+	enum snd_soc_bias_level target_bias_level;
 	struct list_head list;
 
 #ifdef CONFIG_DEBUG_FS

include/sound/soc.h

@@ -202,6 +202,16 @@
 #define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \
 	SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
 
+/*
+ * Component probe and remove ordering levels for components with runtime
+ * dependencies.
+ */
+#define SND_SOC_COMP_ORDER_FIRST		-2
+#define SND_SOC_COMP_ORDER_EARLY		-1
+#define SND_SOC_COMP_ORDER_NORMAL		0
+#define SND_SOC_COMP_ORDER_LATE		1
+#define SND_SOC_COMP_ORDER_LAST		2
+
 /*
  * Bias levels
  *
@@ -214,10 +224,10 @@
  * @OFF:     Power Off. No restrictions on transition times.
  */
 enum snd_soc_bias_level {
-	SND_SOC_BIAS_OFF,
-	SND_SOC_BIAS_STANDBY,
-	SND_SOC_BIAS_PREPARE,
-	SND_SOC_BIAS_ON,
+	SND_SOC_BIAS_OFF = 0,
+	SND_SOC_BIAS_STANDBY = 1,
+	SND_SOC_BIAS_PREPARE = 2,
+	SND_SOC_BIAS_ON = 3,
 };
 
 struct snd_jack;
@@ -258,6 +268,11 @@ enum snd_soc_compress_type {
 	SND_SOC_RBTREE_COMPRESSION
 };
 
+enum snd_soc_pcm_subclass {
+	SND_SOC_PCM_CLASS_PCM	= 0,
+	SND_SOC_PCM_CLASS_BE	= 1,
+};
+
 int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 			     unsigned int freq, int dir);
 int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
@@ -297,6 +312,10 @@ int snd_soc_default_readable_register(struct snd_soc_codec *codec,
 				      unsigned int reg);
 int snd_soc_default_writable_register(struct snd_soc_codec *codec,
 				      unsigned int reg);
+int snd_soc_platform_read(struct snd_soc_platform *platform,
+					unsigned int reg);
+int snd_soc_platform_write(struct snd_soc_platform *platform,
+					unsigned int reg, unsigned int val);
 
 /* Utility functions to get clock rates from various things */
 int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
@@ -349,6 +368,8 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
 				  const char *prefix);
 int snd_soc_add_controls(struct snd_soc_codec *codec,
 	const struct snd_kcontrol_new *controls, int num_controls);
+int snd_soc_add_platform_controls(struct snd_soc_platform *platform,
+	const struct snd_kcontrol_new *controls, int num_controls);
 int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_info *uinfo);
 int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
@@ -612,6 +633,10 @@ struct snd_soc_codec_driver {
 
 	void (*seq_notifier)(struct snd_soc_dapm_context *,
 			     enum snd_soc_dapm_type, int);
+
+	/* probe ordering - for components with runtime dependencies */
+	int probe_order;
+	int remove_order;
 };
 
 /* SoC platform interface */
@@ -623,10 +648,17 @@ struct snd_soc_platform_driver {
 	int (*resume)(struct snd_soc_dai *dai);
 
 	/* pcm creation and destruction */
-	int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
-		struct snd_pcm *);
+	int (*pcm_new)(struct snd_soc_pcm_runtime *);
 	void (*pcm_free)(struct snd_pcm *);
 
+	/* Default control and setup, added after probe() is run */
+	const struct snd_kcontrol_new *controls;
+	int num_controls;
+	const struct snd_soc_dapm_widget *dapm_widgets;
+	int num_dapm_widgets;
+	const struct snd_soc_dapm_route *dapm_routes;
+	int num_dapm_routes;
+
 	/*
 	 * For platform caused delay reporting.
 	 * Optional.
@@ -636,6 +668,14 @@ struct snd_soc_platform_driver {
 
 	/* platform stream ops */
 	struct snd_pcm_ops *ops;
+
+	/* probe ordering - for components with runtime dependencies */
+	int probe_order;
+	int remove_order;
+
+	/* platform IO - used for platform DAPM */
+	unsigned int (*read)(struct snd_soc_platform *, unsigned int);
+	int (*write)(struct snd_soc_platform *, unsigned int, unsigned int);
 };
 
 struct snd_soc_platform {
@@ -650,6 +690,8 @@ struct snd_soc_platform {
 	struct snd_soc_card *card;
 	struct list_head list;
 	struct list_head card_list;
+
+	struct snd_soc_dapm_context dapm;
 };
 
 struct snd_soc_dai_link {
@@ -725,8 +767,10 @@ struct snd_soc_card {
 
 	/* callbacks */
 	int (*set_bias_level)(struct snd_soc_card *,
+			      struct snd_soc_dapm_context *dapm,
 			      enum snd_soc_bias_level level);
 	int (*set_bias_level_post)(struct snd_soc_card *,
+				   struct snd_soc_dapm_context *dapm,
 				   enum snd_soc_bias_level level);
 
 	long pmdown_time;
@@ -789,6 +833,9 @@ struct snd_soc_pcm_runtime  {
 	struct device dev;
 	struct snd_soc_card *card;
 	struct snd_soc_dai_link *dai_link;
+	struct mutex pcm_mutex;
+	enum snd_soc_pcm_subclass pcm_subclass;
+	struct snd_pcm_ops ops;
 
 	unsigned int complete:1;
 	unsigned int dev_registered:1;

include/trace/events/asoc.h

@@ -9,6 +9,7 @@
 
 struct snd_soc_jack;
 struct snd_soc_codec;
+struct snd_soc_platform;
 struct snd_soc_card;
 struct snd_soc_dapm_widget;
 
@@ -59,6 +60,50 @@ DEFINE_EVENT(snd_soc_reg, snd_soc_reg_read,
 
 );
 
+DECLARE_EVENT_CLASS(snd_soc_preg,
+
+	TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
+		 unsigned int val),
+
+	TP_ARGS(platform, reg, val),
+
+	TP_STRUCT__entry(
+		__string(	name,		platform->name	)
+		__field(	int,		id		)
+		__field(	unsigned int,	reg		)
+		__field(	unsigned int,	val		)
+	),
+
+	TP_fast_assign(
+		__assign_str(name, platform->name);
+		__entry->id = platform->id;
+		__entry->reg = reg;
+		__entry->val = val;
+	),
+
+	TP_printk("platform=%s.%d reg=%x val=%x", __get_str(name),
+		  (int)__entry->id, (unsigned int)__entry->reg,
+		  (unsigned int)__entry->val)
+);
+
+DEFINE_EVENT(snd_soc_preg, snd_soc_preg_write,
+
+	TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
+		 unsigned int val),
+
+	TP_ARGS(platform, reg, val)
+
+);
+
+DEFINE_EVENT(snd_soc_preg, snd_soc_preg_read,
+
+	TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
+		 unsigned int val),
+
+	TP_ARGS(platform, reg, val)
+
+);
+
 DECLARE_EVENT_CLASS(snd_soc_card,
 
 	TP_PROTO(struct snd_soc_card *card, int val),

sound/core/rawmidi.c

@@ -92,16 +92,12 @@ static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substre
 	       (!substream->append || runtime->avail >= count);
 }
 
-static void snd_rawmidi_input_event_tasklet(unsigned long data)
+static void snd_rawmidi_input_event_work(struct work_struct *work)
 {
-	struct snd_rawmidi_substream *substream = (struct snd_rawmidi_substream *)data;
-	substream->runtime->event(substream);
-}
-
-static void snd_rawmidi_output_trigger_tasklet(unsigned long data)
-{
-	struct snd_rawmidi_substream *substream = (struct snd_rawmidi_substream *)data;
-	substream->ops->trigger(substream, 1);
+	struct snd_rawmidi_runtime *runtime =
+		container_of(work, struct snd_rawmidi_runtime, event_work);
+	if (runtime->event)
+		runtime->event(runtime->substream);
 }
 
 static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream)
@@ -110,16 +106,10 @@ static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream)
 
 	if ((runtime = kzalloc(sizeof(*runtime), GFP_KERNEL)) == NULL)
 		return -ENOMEM;
+	runtime->substream = substream;
 	spin_lock_init(&runtime->lock);
 	init_waitqueue_head(&runtime->sleep);
-	if (substream->stream == SNDRV_RAWMIDI_STREAM_INPUT)
-		tasklet_init(&runtime->tasklet,
-			     snd_rawmidi_input_event_tasklet,
-			     (unsigned long)substream);
-	else
-		tasklet_init(&runtime->tasklet,
-			     snd_rawmidi_output_trigger_tasklet,
-			     (unsigned long)substream);
+	INIT_WORK(&runtime->event_work, snd_rawmidi_input_event_work);
 	runtime->event = NULL;
 	runtime->buffer_size = PAGE_SIZE;
 	runtime->avail_min = 1;
@@ -150,12 +140,7 @@ static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *subs
 {
 	if (!substream->opened)
 		return;
-	if (up) {
-		tasklet_schedule(&substream->runtime->tasklet);
-	} else {
-		tasklet_kill(&substream->runtime->tasklet);
-		substream->ops->trigger(substream, 0);
-	}
+	substream->ops->trigger(substream, up);
 }
 
 static void snd_rawmidi_input_trigger(struct snd_rawmidi_substream *substream, int up)
@@ -163,8 +148,8 @@ static void snd_rawmidi_input_trigger(struct snd_rawmidi_substream *substream, i
 	if (!substream->opened)
 		return;
 	substream->ops->trigger(substream, up);
-	if (!up && substream->runtime->event)
-		tasklet_kill(&substream->runtime->tasklet);
+	if (!up)
+		cancel_work_sync(&substream->runtime->event_work);
 }
 
 int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream)
@@ -641,10 +626,10 @@ int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream,
 		return -EINVAL;
 	}
 	if (params->buffer_size != runtime->buffer_size) {
-		newbuf = kmalloc(params->buffer_size, GFP_KERNEL);
+		newbuf = krealloc(runtime->buffer, params->buffer_size,
+				  GFP_KERNEL);
 		if (!newbuf)
 			return -ENOMEM;
-		kfree(runtime->buffer);
 		runtime->buffer = newbuf;
 		runtime->buffer_size = params->buffer_size;
 		runtime->avail = runtime->buffer_size;
@@ -668,10 +653,10 @@ int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream,
 		return -EINVAL;
 	}
 	if (params->buffer_size != runtime->buffer_size) {
-		newbuf = kmalloc(params->buffer_size, GFP_KERNEL);
+		newbuf = krealloc(runtime->buffer, params->buffer_size,
+				  GFP_KERNEL);
 		if (!newbuf)
 			return -ENOMEM;
-		kfree(runtime->buffer);
 		runtime->buffer = newbuf;
 		runtime->buffer_size = params->buffer_size;
 	}
@@ -926,7 +911,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream,
 	}
 	if (result > 0) {
 		if (runtime->event)
-			tasklet_schedule(&runtime->tasklet);
+			schedule_work(&runtime->event_work);
 		else if (snd_rawmidi_ready(substream))
 			wake_up(&runtime->sleep);
 	}

sound/firewire/speakers.c

@@ -171,7 +171,7 @@ static int fwspk_open(struct snd_pcm_substream *substream)
 
 	err = snd_pcm_hw_constraint_minmax(runtime,
 					   SNDRV_PCM_HW_PARAM_PERIOD_TIME,
-					   5000, 8192000);
+					   5000, UINT_MAX);
 	if (err < 0)
 		return err;
 

sound/pci/ad1889.c

@@ -944,7 +944,7 @@ snd_ad1889_create(struct snd_card *card,
 	spin_lock_init(&chip->lock);	/* only now can we call ad1889_free */
 
 	if (request_irq(pci->irq, snd_ad1889_interrupt,
-			IRQF_SHARED, card->driver, chip)) {
+			IRQF_SHARED, KBUILD_MODNAME, chip)) {
 		printk(KERN_ERR PFX "cannot obtain IRQ %d\n", pci->irq);
 		snd_ad1889_free(chip);
 		return -EBUSY;
@@ -1055,7 +1055,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_ad1889_ids) = {
 MODULE_DEVICE_TABLE(pci, snd_ad1889_ids);
 
 static struct pci_driver ad1889_pci_driver = {
-	.name = "AD1889 Audio",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_ad1889_ids,
 	.probe = snd_ad1889_probe,
 	.remove = __devexit_p(snd_ad1889_remove),

sound/pci/ali5451/ali5451.c

@@ -2090,7 +2090,7 @@ static int __devinit snd_ali_resources(struct snd_ali *codec)
 	codec->port = pci_resource_start(codec->pci, 0);
 
 	if (request_irq(codec->pci->irq, snd_ali_card_interrupt,
-			IRQF_SHARED, "ALI 5451", codec)) {
+			IRQF_SHARED, KBUILD_MODNAME, codec)) {
 		snd_printk(KERN_ERR "Unable to request irq.\n");
 		return -EBUSY;
 	}
@@ -2295,7 +2295,7 @@ static void __devexit snd_ali_remove(struct pci_dev *pci)
 }
 
 static struct pci_driver driver = {
-	.name = "ALI 5451",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_ali_ids,
 	.probe = snd_ali_probe,
 	.remove = __devexit_p(snd_ali_remove),

sound/pci/als300.c

@@ -722,7 +722,7 @@ static int __devinit snd_als300_create(struct snd_card *card,
 		irq_handler = snd_als300_interrupt;
 
 	if (request_irq(pci->irq, irq_handler, IRQF_SHARED,
-			card->shortname, chip)) {
+			KBUILD_MODNAME, chip)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		snd_als300_free(chip);
 		return -EBUSY;
@@ -846,7 +846,7 @@ static int __devinit snd_als300_probe(struct pci_dev *pci,
 }
 
 static struct pci_driver driver = {
-	.name = "ALS300",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_als300_ids,
 	.probe = snd_als300_probe,
 	.remove = __devexit_p(snd_als300_remove),

sound/pci/als4000.c

@@ -1036,7 +1036,7 @@ static int snd_als4000_resume(struct pci_dev *pci)
 
 
 static struct pci_driver driver = {
-	.name = "ALS4000",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_als4000_ids,
 	.probe = snd_card_als4000_probe,
 	.remove = __devexit_p(snd_card_als4000_remove),

sound/pci/asihpi/asihpi.c

@@ -49,19 +49,21 @@ 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)
+static inline void
+snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size)
 {
-	snprintf(name, len, "pcmC%dD%d%c:%d",
+	snprintf(buf, size, "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)
+static inline void
+snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size)
+{
+	*buf = 0;
+}
 #endif
 
 #if defined CONFIG_SND_DEBUG_VERBOSE
@@ -304,7 +306,8 @@ static u16 handle_error(u16 err, int line, char *filename)
 static void print_hwparams(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *p)
 {
-	DEBUG_NAME(substream, name);
+	char name[16];
+	snd_pcm_debug_name(substream, name, sizeof(name));
 	snd_printd("%s HWPARAMS\n", name);
 	snd_printd(" samplerate %d Hz\n", params_rate(p));
 	snd_printd(" channels %d\n", params_channels(p));
@@ -576,8 +579,9 @@ 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);
+	char name[16];
 
+	snd_pcm_debug_name(substream, name, sizeof(name));
 	snd_printdd("%s trigger\n", name);
 
 	switch (cmd) {
@@ -741,7 +745,9 @@ 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);
+	char name[16];
+
+	snd_pcm_debug_name(substream, name, sizeof(name));
 
 	snd_printdd("%s snd_card_asihpi_timer_function\n", name);
 
@@ -1323,10 +1329,12 @@ static const char * const asihpi_src_names[] = {
 	"RF",
 	"Clock",
 	"Bitstream",
-	"Microphone",
-	"Cobranet",
+	"Mic",
+	"Net",
 	"Analog",
 	"Adapter",
+	"RTP",
+	"GPI",
 };
 
 compile_time_assert(
@@ -1341,8 +1349,10 @@ static const char * const asihpi_dst_names[] = {
 	"Digital",
 	"RF",
 	"Speaker",
-	"Cobranet Out",
-	"Analog"
+	"Net",
+	"Analog",
+	"RTP",
+	"GPO",
 };
 
 compile_time_assert(
@@ -1476,11 +1486,40 @@ static int snd_asihpi_volume_put(struct snd_kcontrol *kcontrol,
 
 static const DECLARE_TLV_DB_SCALE(db_scale_100, -10000, VOL_STEP_mB, 0);
 
+#define snd_asihpi_volume_mute_info	snd_ctl_boolean_mono_info
+
+static int snd_asihpi_volume_mute_get(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	u32 h_control = kcontrol->private_value;
+	u32 mute;
+
+	hpi_handle_error(hpi_volume_get_mute(h_control, &mute));
+	ucontrol->value.integer.value[0] = mute ? 0 : 1;
+
+	return 0;
+}
+
+static int snd_asihpi_volume_mute_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	u32 h_control = kcontrol->private_value;
+	int change = 1;
+	/* HPI currently only supports all or none muting of multichannel volume
+	ALSA Switch element has opposite sense to HPI mute: on==unmuted, off=muted
+	*/
+	int mute =  ucontrol->value.integer.value[0] ? 0 : HPI_BITMASK_ALL_CHANNELS;
+	hpi_handle_error(hpi_volume_set_mute(h_control, mute));
+	return change;
+}
+
 static int __devinit snd_asihpi_volume_add(struct snd_card_asihpi *asihpi,
 					struct hpi_control *hpi_ctl)
 {
 	struct snd_card *card = asihpi->card;
 	struct snd_kcontrol_new snd_control;
+	int err;
+	u32 mute;
 
 	asihpi_ctl_init(&snd_control, hpi_ctl, "Volume");
 	snd_control.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
@@ -1490,7 +1529,19 @@ static int __devinit snd_asihpi_volume_add(struct snd_card_asihpi *asihpi,
 	snd_control.put = snd_asihpi_volume_put;
 	snd_control.tlv.p = db_scale_100;
 
-	return ctl_add(card, &snd_control, asihpi);
+	err = ctl_add(card, &snd_control, asihpi);
+	if (err)
+		return err;
+
+	if (hpi_volume_get_mute(hpi_ctl->h_control, &mute) == 0) {
+		asihpi_ctl_init(&snd_control, hpi_ctl, "Switch");
+		snd_control.access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
+		snd_control.info = snd_asihpi_volume_mute_info;
+		snd_control.get = snd_asihpi_volume_mute_get;
+		snd_control.put = snd_asihpi_volume_mute_put;
+		err = ctl_add(card, &snd_control, asihpi);
+	}
+	return err;
 }
 
 /*------------------------------------------------------------
@@ -2923,7 +2974,7 @@ static DEFINE_PCI_DEVICE_TABLE(asihpi_pci_tbl) = {
 MODULE_DEVICE_TABLE(pci, asihpi_pci_tbl);
 
 static struct pci_driver driver = {
-	.name = "asihpi",
+	.name = KBUILD_MODNAME,
 	.id_table = asihpi_pci_tbl,
 	.probe = snd_asihpi_probe,
 	.remove = __devexit_p(snd_asihpi_remove),

sound/pci/asihpi/hpi.h

@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2010  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -42,12 +42,11 @@ i.e 3.05.02 is a development version
 #define HPI_VER_MINOR(v) ((int)((v >> 8) & 0xFF))
 #define HPI_VER_RELEASE(v) ((int)(v & 0xFF))
 
-/* Use single digits for versions less that 10 to avoid octal. */
-#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 6, 0)
-#define HPI_VER_STRING "4.06.00"
+#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 8, 0)
+#define HPI_VER_STRING "4.08.00"
 
 /* Library version as documented in hpi-api-versions.txt */
-#define HPI_LIB_VER  HPI_VERSION_CONSTRUCTOR(9, 0, 0)
+#define HPI_LIB_VER  HPI_VERSION_CONSTRUCTOR(10, 0, 0)
 
 #include <linux/types.h>
 #define HPI_BUILD_EXCLUDE_DEPRECATED
@@ -211,8 +210,12 @@ enum HPI_SOURCENODES {
 	HPI_SOURCENODE_COBRANET = 109,
 	HPI_SOURCENODE_ANALOG = 110,	     /**< analog input node. */
 	HPI_SOURCENODE_ADAPTER = 111,	     /**< adapter node. */
+	/** RTP stream input node - This node is a destination for
+	    packets of RTP audio samples from other devices. */
+	HPI_SOURCENODE_RTP_DESTINATION = 112,
+	HPI_SOURCENODE_GP_IN = 113,	     /**< general purpose input. */
 	/* !!!Update this  AND hpidebug.h if you add a new sourcenode type!!! */
-	HPI_SOURCENODE_LAST_INDEX = 111	     /**< largest ID */
+	HPI_SOURCENODE_LAST_INDEX = 113	     /**< largest ID */
 		/* AX6 max sourcenode types = 15 */
 };
 
@@ -228,7 +231,7 @@ enum HPI_DESTNODES {
 	HPI_DESTNODE_NONE = 200,
 	/** In Stream (Record) node. */
 	HPI_DESTNODE_ISTREAM = 201,
-	HPI_DESTNODE_LINEOUT = 202,	    /**< line out node. */
+	HPI_DESTNODE_LINEOUT = 202,	     /**< line out node. */
 	HPI_DESTNODE_AESEBU_OUT = 203,	     /**< AES/EBU output node. */
 	HPI_DESTNODE_RF = 204,		     /**< RF output node. */
 	HPI_DESTNODE_SPEAKER = 205,	     /**< speaker output node. */
@@ -236,9 +239,12 @@ enum HPI_DESTNODES {
 	    Audio samples from the device are sent out on the Cobranet network.*/
 	HPI_DESTNODE_COBRANET = 206,
 	HPI_DESTNODE_ANALOG = 207,	     /**< analog output node. */
-
+	/** RTP stream output node - This node is a source for
+	    packets of RTP audio samples that are sent to other devices. */
+	HPI_DESTNODE_RTP_SOURCE = 208,
+	HPI_DESTNODE_GP_OUT = 209,	     /**< general purpose output node. */
 	/* !!!Update this AND hpidebug.h if you add a new destnode type!!! */
-	HPI_DESTNODE_LAST_INDEX = 207	     /**< largest ID */
+	HPI_DESTNODE_LAST_INDEX = 209	     /**< largest ID */
 		/* AX6 max destnode types = 15 */
 };
 

sound/pci/asihpi/hpi6000.c

@@ -359,7 +359,7 @@ void HPI_6000(struct hpi_message *phm, struct hpi_response *phr)
 			HPI_ERROR_PROCESSING_MESSAGE);
 
 	switch (phm->type) {
-	case HPI_TYPE_MESSAGE:
+	case HPI_TYPE_REQUEST:
 		switch (phm->object) {
 		case HPI_OBJ_SUBSYSTEM:
 			subsys_message(phm, phr);
@@ -538,7 +538,7 @@ static short create_adapter_obj(struct hpi_adapter_obj *pao,
 
 		HPI_DEBUG_LOG(VERBOSE, "send ADAPTER_GET_INFO\n");
 		memset(&hm, 0, sizeof(hm));
-		hm.type = HPI_TYPE_MESSAGE;
+		hm.type = HPI_TYPE_REQUEST;
 		hm.size = sizeof(struct hpi_message);
 		hm.object = HPI_OBJ_ADAPTER;
 		hm.function = HPI_ADAPTER_GET_INFO;
@@ -946,11 +946,8 @@ static short hpi6000_adapter_boot_load_dsp(struct hpi_adapter_obj *pao,
 		}
 
 		/* write the DSP code down into the DSPs memory */
-		/*HpiDspCode_Open(nBootLoadFamily,&DspCode,pdwOsErrorCode); */
-		dsp_code.ps_dev = pao->pci.pci_dev;
-
-		error = hpi_dsp_code_open(boot_load_family, &dsp_code,
-			pos_error_code);
+		error = hpi_dsp_code_open(boot_load_family, pao->pci.pci_dev,
+			&dsp_code, pos_error_code);
 
 		if (error)
 			return error;

sound/pci/asihpi/hpi6205.c

@@ -373,6 +373,7 @@ static void instream_message(struct hpi_adapter_obj *pao,
 /** Entry point to this HPI backend
  * All calls to the HPI start here
  */
+static
 void _HPI_6205(struct hpi_adapter_obj *pao, struct hpi_message *phm,
 	struct hpi_response *phr)
 {
@@ -392,7 +393,7 @@ void _HPI_6205(struct hpi_adapter_obj *pao, struct hpi_message *phm,
 
 	HPI_DEBUG_LOG(VERBOSE, "start of switch\n");
 	switch (phm->type) {
-	case HPI_TYPE_MESSAGE:
+	case HPI_TYPE_REQUEST:
 		switch (phm->object) {
 		case HPI_OBJ_SUBSYSTEM:
 			subsys_message(pao, phm, phr);
@@ -402,7 +403,6 @@ void _HPI_6205(struct hpi_adapter_obj *pao, struct hpi_message *phm,
 			adapter_message(pao, phm, phr);
 			break;
 
-		case HPI_OBJ_CONTROLEX:
 		case HPI_OBJ_CONTROL:
 			control_message(pao, phm, phr);
 			break;
@@ -634,11 +634,12 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao,
 
 		HPI_DEBUG_LOG(VERBOSE, "init ADAPTER_GET_INFO\n");
 		memset(&hm, 0, sizeof(hm));
-		hm.type = HPI_TYPE_MESSAGE;
+		/* wAdapterIndex == version == 0 */
+		hm.type = HPI_TYPE_REQUEST;
 		hm.size = sizeof(hm);
 		hm.object = HPI_OBJ_ADAPTER;
 		hm.function = HPI_ADAPTER_GET_INFO;
-		hm.adapter_index = 0;
+
 		memset(&hr, 0, sizeof(hr));
 		hr.size = sizeof(hr);
 
@@ -658,9 +659,6 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao,
 			hr.u.ax.info.num_outstreams +
 			hr.u.ax.info.num_instreams;
 
-		hpios_locked_mem_prepare((max_streams * 6) / 10, max_streams,
-			65536, pao->pci.pci_dev);
-
 		HPI_DEBUG_LOG(VERBOSE,
 			"got adapter info type %x index %d serial %d\n",
 			hr.u.ax.info.adapter_type, hr.u.ax.info.adapter_index,
@@ -709,9 +707,6 @@ static void delete_adapter_obj(struct hpi_adapter_obj *pao)
 				[i]);
 			phw->outstream_host_buffer_size[i] = 0;
 		}
-
-	hpios_locked_mem_unprepare(pao->pci.pci_dev);
-
 	kfree(phw);
 }
 
@@ -1371,9 +1366,8 @@ static u16 adapter_boot_load_dsp(struct hpi_adapter_obj *pao,
 			return err;
 
 		/* write the DSP code down into the DSPs memory */
-		dsp_code.ps_dev = pao->pci.pci_dev;
-		err = hpi_dsp_code_open(boot_code_id[dsp], &dsp_code,
-			pos_error_code);
+		err = hpi_dsp_code_open(boot_code_id[dsp], pao->pci.pci_dev,
+			&dsp_code, pos_error_code);
 		if (err)
 			return err;
 
@@ -2084,13 +2078,13 @@ static u16 message_response_sequence(struct hpi_adapter_obj *pao,
 	u16 err = 0;
 
 	message_count++;
-	if (phm->size > sizeof(interface->u)) {
+	if (phm->size > sizeof(interface->u.message_buffer)) {
 		phr->error = HPI_ERROR_MESSAGE_BUFFER_TOO_SMALL;
-		phr->specific_error = sizeof(interface->u);
+		phr->specific_error = sizeof(interface->u.message_buffer);
 		phr->size = sizeof(struct hpi_response_header);
 		HPI_DEBUG_LOG(ERROR,
 			"message len %d too big for buffer %zd \n", phm->size,
-			sizeof(interface->u));
+			sizeof(interface->u.message_buffer));
 		return 0;
 	}
 
@@ -2122,18 +2116,19 @@ static u16 message_response_sequence(struct hpi_adapter_obj *pao,
 
 	/* read the result */
 	if (time_out) {
-		if (interface->u.response_buffer.size <= phr->size)
+		if (interface->u.response_buffer.response.size <= phr->size)
 			memcpy(phr, &interface->u.response_buffer,
-				interface->u.response_buffer.size);
+				interface->u.response_buffer.response.size);
 		else {
 			HPI_DEBUG_LOG(ERROR,
 				"response len %d too big for buffer %d\n",
-				interface->u.response_buffer.size, phr->size);
+				interface->u.response_buffer.response.size,
+				phr->size);
 			memcpy(phr, &interface->u.response_buffer,
 				sizeof(struct hpi_response_header));
 			phr->error = HPI_ERROR_RESPONSE_BUFFER_TOO_SMALL;
 			phr->specific_error =
-				interface->u.response_buffer.size;
+				interface->u.response_buffer.response.size;
 			phr->size = sizeof(struct hpi_response_header);
 		}
 	}
@@ -2202,23 +2197,6 @@ static void hw_message(struct hpi_adapter_obj *pao, struct hpi_message *phm,
 			phm->u.d.u.data.data_size, H620_HIF_GET_DATA);
 		break;
 
-	case HPI_CONTROL_SET_STATE:
-		if (phm->object == HPI_OBJ_CONTROLEX
-			&& phm->u.cx.attribute == HPI_COBRANET_SET_DATA)
-			err = hpi6205_transfer_data(pao,
-				phm->u.cx.u.cobranet_bigdata.pb_data,
-				phm->u.cx.u.cobranet_bigdata.byte_count,
-				H620_HIF_SEND_DATA);
-		break;
-
-	case HPI_CONTROL_GET_STATE:
-		if (phm->object == HPI_OBJ_CONTROLEX
-			&& phm->u.cx.attribute == HPI_COBRANET_GET_DATA)
-			err = hpi6205_transfer_data(pao,
-				phm->u.cx.u.cobranet_bigdata.pb_data,
-				phr->u.cx.u.cobranet_data.byte_count,
-				H620_HIF_GET_DATA);
-		break;
 	}
 	phr->error = err;
 

sound/pci/asihpi/hpi6205.h

@@ -1,7 +1,7 @@
 /*****************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2010  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -70,15 +70,28 @@ The Host located memory buffer that the 6205 will bus master
 in and out of.
 ************************************************************/
 #define HPI6205_SIZEOF_DATA (16*1024)
+
+struct message_buffer_6205 {
+	struct hpi_message message;
+	char data[256];
+};
+
+struct response_buffer_6205 {
+	struct hpi_response response;
+	char data[256];
+};
+
+union buffer_6205 {
+	struct message_buffer_6205 message_buffer;
+	struct response_buffer_6205 response_buffer;
+	u8 b_data[HPI6205_SIZEOF_DATA];
+};
+
 struct bus_master_interface {
 	u32 host_cmd;
 	u32 dsp_ack;
 	u32 transfer_size_in_bytes;
-	union {
-		struct hpi_message_header message_buffer;
-		struct hpi_response_header response_buffer;
-		u8 b_data[HPI6205_SIZEOF_DATA];
-	} u;
+	union buffer_6205 u;
 	struct controlcache_6205 control_cache;
 	struct async_event_buffer_6205 async_buffer;
 	struct hpi_hostbuffer_status

sound/pci/asihpi/hpi_internal.h

@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2010  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -32,12 +32,6 @@ HPI internal definitions
 #include "hpios.h"
 
 /* physical memory allocation */
-void hpios_locked_mem_init(void
-	);
-void hpios_locked_mem_free_all(void
-	);
-#define hpios_locked_mem_prepare(a, b, c, d);
-#define hpios_locked_mem_unprepare(a)
 
 /** Allocate and map an area of locked memory for bus master DMA operations.
 
@@ -226,8 +220,8 @@ enum HPI_CONTROL_ATTRIBUTES {
 
 	HPI_COBRANET_SET = HPI_CTL_ATTR(COBRANET, 1),
 	HPI_COBRANET_GET = HPI_CTL_ATTR(COBRANET, 2),
-	HPI_COBRANET_SET_DATA = HPI_CTL_ATTR(COBRANET, 3),
-	HPI_COBRANET_GET_DATA = HPI_CTL_ATTR(COBRANET, 4),
+	/*HPI_COBRANET_SET_DATA         = HPI_CTL_ATTR(COBRANET, 3), */
+	/*HPI_COBRANET_GET_DATA         = HPI_CTL_ATTR(COBRANET, 4), */
 	HPI_COBRANET_GET_STATUS = HPI_CTL_ATTR(COBRANET, 5),
 	HPI_COBRANET_SEND_PACKET = HPI_CTL_ATTR(COBRANET, 6),
 	HPI_COBRANET_GET_PACKET = HPI_CTL_ATTR(COBRANET, 7),
@@ -364,10 +358,12 @@ Used in DLL to indicate device not present
 #define HPI_ADAPTER_ASI(f)   (f)
 
 enum HPI_MESSAGE_TYPES {
-	HPI_TYPE_MESSAGE = 1,
+	HPI_TYPE_REQUEST = 1,
 	HPI_TYPE_RESPONSE = 2,
 	HPI_TYPE_DATA = 3,
-	HPI_TYPE_SSX2BYPASS_MESSAGE = 4
+	HPI_TYPE_SSX2BYPASS_MESSAGE = 4,
+	HPI_TYPE_COMMAND = 5,
+	HPI_TYPE_NOTIFICATION = 6
 };
 
 enum HPI_OBJECT_TYPES {
@@ -383,7 +379,7 @@ enum HPI_OBJECT_TYPES {
 	HPI_OBJ_WATCHDOG = 10,
 	HPI_OBJ_CLOCK = 11,
 	HPI_OBJ_PROFILE = 12,
-	HPI_OBJ_CONTROLEX = 13,
+	/* HPI_ OBJ_ CONTROLEX  = 13, */
 	HPI_OBJ_ASYNCEVENT = 14
 #define HPI_OBJ_MAXINDEX 14
 };
@@ -608,7 +604,7 @@ struct hpi_data_compat32 {
 #endif
 
 struct hpi_buffer {
-  /** placehoder for backward compatibility (see dwBufferSize) */
+  /** placeholder for backward compatibility (see dwBufferSize) */
 	struct hpi_msg_format reserved;
 	u32 command; /**< HPI_BUFFER_CMD_xxx*/
 	u32 pci_address; /**< PCI physical address of buffer for DSP DMA */
@@ -912,95 +908,13 @@ union hpi_control_union_res {
 		u32 remaining_chars;
 	} chars8;
 	char c_data12[12];
-};
-
-/* HPI_CONTROLX_STRUCTURES */
-
-/* Message */
-
-/** Used for all HMI variables where max length <= 8 bytes
-*/
-struct hpi_controlx_msg_cobranet_data {
-	u32 hmi_address;
-	u32 byte_count;
-	u32 data[2];
-};
-
-/** Used for string data, and for packet bridge
-*/
-struct hpi_controlx_msg_cobranet_bigdata {
-	u32 hmi_address;
-	u32 byte_count;
-	u8 *pb_data;
-#ifndef HPI64BIT
-	u32 padding;
-#endif
-};
-
-/** Used for PADS control reading of string fields.
-*/
-struct hpi_controlx_msg_pad_data {
-	u32 field;
-	u32 byte_count;
-	u8 *pb_data;
-#ifndef HPI64BIT
-	u32 padding;
-#endif
-};
-
-/** Used for generic data
-*/
-
-struct hpi_controlx_msg_generic {
-	u32 param1;
-	u32 param2;
-};
-
-struct hpi_controlx_msg {
-	u16 attribute;		/* control attribute or property */
-	u16 saved_index;
-	union {
-		struct hpi_controlx_msg_cobranet_data cobranet_data;
-		struct hpi_controlx_msg_cobranet_bigdata cobranet_bigdata;
-		struct hpi_controlx_msg_generic generic;
-		struct hpi_controlx_msg_pad_data pad_data;
-		/*struct param_value universal_value; */
-		/* nothing extra to send for status read */
-	} u;
-};
-
-/* Response */
-/**
-*/
-struct hpi_controlx_res_cobranet_data {
-	u32 byte_count;
-	u32 data[2];
-};
-
-struct hpi_controlx_res_cobranet_bigdata {
-	u32 byte_count;
-};
-
-struct hpi_controlx_res_cobranet_status {
-	u32 status;
-	u32 readable_size;
-	u32 writeable_size;
-};
-
-struct hpi_controlx_res_generic {
-	u32 param1;
-	u32 param2;
-};
-
-struct hpi_controlx_res {
 	union {
-		struct hpi_controlx_res_cobranet_bigdata cobranet_bigdata;
-		struct hpi_controlx_res_cobranet_data cobranet_data;
-		struct hpi_controlx_res_cobranet_status cobranet_status;
-		struct hpi_controlx_res_generic generic;
-		/*struct param_info universal_info; */
-		/*struct param_value universal_value; */
-	} u;
+		struct {
+			u32 status;
+			u32 readable_size;
+			u32 writeable_size;
+		} status;
+	} cobranet;
 };
 
 struct hpi_nvmemory_msg {
@@ -1126,7 +1040,6 @@ struct hpi_message {
 		/* identical to struct hpi_control_msg,
 		   but field naming is improved */
 		struct hpi_control_union_msg cu;
-		struct hpi_controlx_msg cx;	/* extended mixer control; */
 		struct hpi_nvmemory_msg n;
 		struct hpi_gpio_msg l;	/* digital i/o */
 		struct hpi_watchdog_msg w;
@@ -1151,7 +1064,7 @@ struct hpi_message {
 	sizeof(struct hpi_message_header) + sizeof(struct hpi_watchdog_msg),\
 	sizeof(struct hpi_message_header) + sizeof(struct hpi_clock_msg),\
 	sizeof(struct hpi_message_header) + sizeof(struct hpi_profile_msg),\
-	sizeof(struct hpi_message_header) + sizeof(struct hpi_controlx_msg),\
+	sizeof(struct hpi_message_header), /* controlx obj removed */ \
 	sizeof(struct hpi_message_header) + sizeof(struct hpi_async_msg) \
 }
 
@@ -1188,7 +1101,6 @@ struct hpi_response {
 		struct hpi_control_res c;	/* mixer control; */
 		/* identical to hpi_control_res, but field naming is improved */
 		union hpi_control_union_res cu;
-		struct hpi_controlx_res cx;	/* extended mixer control; */
 		struct hpi_nvmemory_res n;
 		struct hpi_gpio_res l;	/* digital i/o */
 		struct hpi_watchdog_res w;
@@ -1213,7 +1125,7 @@ struct hpi_response {
 	sizeof(struct hpi_response_header) + sizeof(struct hpi_watchdog_res),\
 	sizeof(struct hpi_response_header) + sizeof(struct hpi_clock_res),\
 	sizeof(struct hpi_response_header) + sizeof(struct hpi_profile_res),\
-	sizeof(struct hpi_response_header) + sizeof(struct hpi_controlx_res),\
+	sizeof(struct hpi_response_header), /* controlx obj removed */ \
 	sizeof(struct hpi_response_header) + sizeof(struct hpi_async_res) \
 }
 
@@ -1308,6 +1220,30 @@ struct hpi_res_adapter_debug_read {
 	u8 bytes[256];
 };
 
+struct hpi_msg_cobranet_hmi {
+	u16 attribute;
+	u16 padding;
+	u32 hmi_address;
+	u32 byte_count;
+};
+
+struct hpi_msg_cobranet_hmiwrite {
+	struct hpi_message_header h;
+	struct hpi_msg_cobranet_hmi p;
+	u8 bytes[256];
+};
+
+struct hpi_msg_cobranet_hmiread {
+	struct hpi_message_header h;
+	struct hpi_msg_cobranet_hmi p;
+};
+
+struct hpi_res_cobranet_hmiread {
+	struct hpi_response_header h;
+	u32 byte_count;
+	u8 bytes[256];
+};
+
 #if 1
 #define hpi_message_header_v1 hpi_message_header
 #define hpi_response_header_v1 hpi_response_header
@@ -1338,7 +1274,6 @@ struct hpi_msg_payload_v0 {
 		union hpi_mixerx_msg mx;
 		struct hpi_control_msg c;
 		struct hpi_control_union_msg cu;
-		struct hpi_controlx_msg cx;
 		struct hpi_nvmemory_msg n;
 		struct hpi_gpio_msg l;
 		struct hpi_watchdog_msg w;
@@ -1358,7 +1293,6 @@ struct hpi_res_payload_v0 {
 		union hpi_mixerx_res mx;
 		struct hpi_control_res c;
 		union hpi_control_union_res cu;
-		struct hpi_controlx_res cx;
 		struct hpi_nvmemory_res n;
 		struct hpi_gpio_res l;
 		struct hpi_watchdog_res w;
@@ -1493,12 +1427,6 @@ struct hpi_control_cache_microphone {
 	char temp_padding[6];
 };
 
-struct hpi_control_cache_generic {
-	struct hpi_control_cache_info i;
-	u32 dw1;
-	u32 dw2;
-};
-
 struct hpi_control_cache_single {
 	union {
 		struct hpi_control_cache_info i;
@@ -1514,7 +1442,6 @@ struct hpi_control_cache_single {
 		struct hpi_control_cache_silencedetector silence;
 		struct hpi_control_cache_sampleclock clk;
 		struct hpi_control_cache_microphone microphone;
-		struct hpi_control_cache_generic generic;
 	} u;
 };
 

sound/pci/asihpi/hpicmn.c

@@ -57,7 +57,7 @@ u16 hpi_validate_response(struct hpi_message *phm, struct hpi_response *phr)
 	}
 
 	if (phr->function != phm->function) {
-		HPI_DEBUG_LOG(ERROR, "header type %d invalid\n",
+		HPI_DEBUG_LOG(ERROR, "header function %d invalid\n",
 			phr->function);
 		return HPI_ERROR_INVALID_RESPONSE;
 	}
@@ -315,8 +315,7 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache,
 	short found = 1;
 	struct hpi_control_cache_info *pI;
 	struct hpi_control_cache_single *pC;
-	struct hpi_control_cache_pad *p_pad;
-
+	size_t response_size;
 	if (!find_control(phm->obj_index, p_cache, &pI)) {
 		HPI_DEBUG_LOG(VERBOSE,
 			"HPICMN find_control() failed for adap %d\n",
@@ -326,11 +325,15 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache,
 
 	phr->error = 0;
 
+	/* set the default response size */
+	response_size =
+		sizeof(struct hpi_response_header) +
+		sizeof(struct hpi_control_res);
+
 	/* pC is the default cached control strucure. May be cast to
 	   something else in the following switch statement.
 	 */
 	pC = (struct hpi_control_cache_single *)pI;
-	p_pad = (struct hpi_control_cache_pad *)pI;
 
 	switch (pI->control_type) {
 
@@ -529,9 +532,7 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache,
 		pI->control_index, pI->control_type, phm->u.c.attribute);
 
 	if (found)
-		phr->size =
-			sizeof(struct hpi_response_header) +
-			sizeof(struct hpi_control_res);
+		phr->size = (u16)response_size;
 
 	return found;
 }
@@ -682,7 +683,7 @@ static void subsys_message(struct hpi_message *phm, struct hpi_response *phr)
 void HPI_COMMON(struct hpi_message *phm, struct hpi_response *phr)
 {
 	switch (phm->type) {
-	case HPI_TYPE_MESSAGE:
+	case HPI_TYPE_REQUEST:
 		switch (phm->object) {
 		case HPI_OBJ_SUBSYSTEM:
 			subsys_message(phm, phr);

sound/pci/asihpi/hpidspcd.c

@@ -1,8 +1,8 @@
 /***********************************************************************/
-/*!
+/**
 
     AudioScience HPI driver
-    Copyright (C) 1997-2010  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -18,90 +18,59 @@
     Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 
 \file
-Functions for reading DSP code to load into DSP
-
-(Linux only:) If DSPCODE_FIRMWARE_LOADER is defined, code is read using
+Functions for reading DSP code using
 hotplug firmware loader from individual dsp code files
-
-If neither of the above is defined, code is read from linked arrays.
-DSPCODE_ARRAY is defined.
-
-HPI_INCLUDE_**** must be defined
-and the appropriate hzz?????.c or hex?????.c linked in
-
- */
+*/
 /***********************************************************************/
 #define SOURCEFILE_NAME "hpidspcd.c"
 #include "hpidspcd.h"
 #include "hpidebug.h"
 
-/**
- Header structure for binary dsp code file (see asidsp.doc)
- This structure must match that used in s2bin.c for generation of asidsp.bin
- */
-
-#ifndef DISABLE_PRAGMA_PACK1
-#pragma pack(push, 1)
-#endif
-
-struct code_header {
-	u32 size;
-	char type[4];
-	u32 adapter;
-	u32 version;
-	u32 crc;
+struct dsp_code_private {
+	/**  Firmware descriptor */
+	const struct firmware *firmware;
+	struct pci_dev *dev;
 };
 
-#ifndef DISABLE_PRAGMA_PACK1
-#pragma pack(pop)
-#endif
-
 #define HPI_VER_DECIMAL ((int)(HPI_VER_MAJOR(HPI_VER) * 10000 + \
 	    HPI_VER_MINOR(HPI_VER) * 100 + HPI_VER_RELEASE(HPI_VER)))
 
-/***********************************************************************/
-#include <linux/pci.h>
 /*-------------------------------------------------------------------*/
-short hpi_dsp_code_open(u32 adapter, struct dsp_code *ps_dsp_code,
-	u32 *pos_error_code)
+short hpi_dsp_code_open(u32 adapter, void *os_data, struct dsp_code *dsp_code,
+	u32 *os_error_code)
 {
-	const struct firmware *ps_firmware = ps_dsp_code->ps_firmware;
+	const struct firmware *firmware;
+	struct pci_dev *dev = os_data;
 	struct code_header header;
 	char fw_name[20];
 	int err;
 
 	sprintf(fw_name, "asihpi/dsp%04x.bin", adapter);
 
-	err = request_firmware(&ps_firmware, fw_name,
-		&ps_dsp_code->ps_dev->dev);
+	err = request_firmware(&firmware, fw_name, &dev->dev);
 
-	if (err != 0) {
-		dev_printk(KERN_ERR, &ps_dsp_code->ps_dev->dev,
+	if (err || !firmware) {
+		dev_printk(KERN_ERR, &dev->dev,
 			"%d, request_firmware failed for  %s\n", err,
 			fw_name);
 		goto error1;
 	}
-	if (ps_firmware->size < sizeof(header)) {
-		dev_printk(KERN_ERR, &ps_dsp_code->ps_dev->dev,
-			"Header size too small %s\n", fw_name);
-		goto error2;
-	}
-	memcpy(&header, ps_firmware->data, sizeof(header));
-	if (header.adapter != adapter) {
-		dev_printk(KERN_ERR, &ps_dsp_code->ps_dev->dev,
-			"Adapter type incorrect %4x != %4x\n", header.adapter,
-			adapter);
+	if (firmware->size < sizeof(header)) {
+		dev_printk(KERN_ERR, &dev->dev, "Header size too small %s\n",
+			fw_name);
 		goto error2;
 	}
-	if (header.size != ps_firmware->size) {
-		dev_printk(KERN_ERR, &ps_dsp_code->ps_dev->dev,
-			"Code size wrong  %d != %ld\n", header.size,
-			(unsigned long)ps_firmware->size);
+	memcpy(&header, firmware->data, sizeof(header));
+
+	if ((header.type != 0x45444F43) ||	/* "CODE" */
+		(header.adapter != adapter)
+		|| (header.size != firmware->size)) {
+		dev_printk(KERN_ERR, &dev->dev, "Invalid firmware file\n");
 		goto error2;
 	}
 
-	if (header.version / 100 != HPI_VER_DECIMAL / 100) {
-		dev_printk(KERN_ERR, &ps_dsp_code->ps_dev->dev,
+	if ((header.version / 100 & ~1) != (HPI_VER_DECIMAL / 100 & ~1)) {
+		dev_printk(KERN_ERR, &dev->dev,
 			"Incompatible firmware version "
 			"DSP image %d != Driver %d\n", header.version,
 			HPI_VER_DECIMAL);
@@ -109,67 +78,70 @@ short hpi_dsp_code_open(u32 adapter, struct dsp_code *ps_dsp_code,
 	}
 
 	if (header.version != HPI_VER_DECIMAL) {
-		dev_printk(KERN_WARNING, &ps_dsp_code->ps_dev->dev,
+		dev_printk(KERN_WARNING, &dev->dev,
 			"Firmware: release version mismatch  DSP image %d != Driver %d\n",
 			header.version, HPI_VER_DECIMAL);
 	}
 
 	HPI_DEBUG_LOG(DEBUG, "dsp code %s opened\n", fw_name);
-	ps_dsp_code->ps_firmware = ps_firmware;
-	ps_dsp_code->block_length = header.size / sizeof(u32);
-	ps_dsp_code->word_count = sizeof(header) / sizeof(u32);
-	ps_dsp_code->version = header.version;
-	ps_dsp_code->crc = header.crc;
+	dsp_code->pvt = kmalloc(sizeof(*dsp_code->pvt), GFP_KERNEL);
+	if (!dsp_code->pvt)
+		return HPI_ERROR_MEMORY_ALLOC;
+
+	dsp_code->pvt->dev = dev;
+	dsp_code->pvt->firmware = firmware;
+	dsp_code->header = header;
+	dsp_code->block_length = header.size / sizeof(u32);
+	dsp_code->word_count = sizeof(header) / sizeof(u32);
 	return 0;
 
 error2:
-	release_firmware(ps_firmware);
+	release_firmware(firmware);
 error1:
-	ps_dsp_code->ps_firmware = NULL;
-	ps_dsp_code->block_length = 0;
+	dsp_code->block_length = 0;
 	return HPI_ERROR_DSP_FILE_NOT_FOUND;
 }
 
 /*-------------------------------------------------------------------*/
-void hpi_dsp_code_close(struct dsp_code *ps_dsp_code)
+void hpi_dsp_code_close(struct dsp_code *dsp_code)
 {
-	if (ps_dsp_code->ps_firmware != NULL) {
+	if (dsp_code->pvt->firmware) {
 		HPI_DEBUG_LOG(DEBUG, "dsp code closed\n");
-		release_firmware(ps_dsp_code->ps_firmware);
-		ps_dsp_code->ps_firmware = NULL;
+		release_firmware(dsp_code->pvt->firmware);
+		dsp_code->pvt->firmware = NULL;
 	}
+	kfree(dsp_code->pvt);
 }
 
 /*-------------------------------------------------------------------*/
-void hpi_dsp_code_rewind(struct dsp_code *ps_dsp_code)
+void hpi_dsp_code_rewind(struct dsp_code *dsp_code)
 {
 	/* Go back to start of  data, after header */
-	ps_dsp_code->word_count = sizeof(struct code_header) / sizeof(u32);
+	dsp_code->word_count = sizeof(struct code_header) / sizeof(u32);
 }
 
 /*-------------------------------------------------------------------*/
-short hpi_dsp_code_read_word(struct dsp_code *ps_dsp_code, u32 *pword)
+short hpi_dsp_code_read_word(struct dsp_code *dsp_code, u32 *pword)
 {
-	if (ps_dsp_code->word_count + 1 > ps_dsp_code->block_length)
+	if (dsp_code->word_count + 1 > dsp_code->block_length)
 		return HPI_ERROR_DSP_FILE_FORMAT;
 
-	*pword = ((u32 *)(ps_dsp_code->ps_firmware->data))[ps_dsp_code->
+	*pword = ((u32 *)(dsp_code->pvt->firmware->data))[dsp_code->
 		word_count];
-	ps_dsp_code->word_count++;
+	dsp_code->word_count++;
 	return 0;
 }
 
 /*-------------------------------------------------------------------*/
 short hpi_dsp_code_read_block(size_t words_requested,
-	struct dsp_code *ps_dsp_code, u32 **ppblock)
+	struct dsp_code *dsp_code, u32 **ppblock)
 {
-	if (ps_dsp_code->word_count + words_requested >
-		ps_dsp_code->block_length)
+	if (dsp_code->word_count + words_requested > dsp_code->block_length)
 		return HPI_ERROR_DSP_FILE_FORMAT;
 
 	*ppblock =
-		((u32 *)(ps_dsp_code->ps_firmware->data)) +
-		ps_dsp_code->word_count;
-	ps_dsp_code->word_count += words_requested;
+		((u32 *)(dsp_code->pvt->firmware->data)) +
+		dsp_code->word_count;
+	dsp_code->word_count += words_requested;
 	return 0;
 }

sound/pci/asihpi/hpidspcd.h

@@ -2,7 +2,7 @@
 /**
 
     AudioScience HPI driver
-    Copyright (C) 1997-2010  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -20,19 +20,6 @@
 \file
 Functions for reading DSP code to load into DSP
 
- hpi_dspcode_defines HPI DSP code loading method
-Define exactly one of these to select how the DSP code is supplied to
-the adapter.
-
-End users writing applications that use the HPI interface do not have to
-use any of the below defines; they are only necessary for building drivers
-
-HPI_DSPCODE_FILE:
-DSP code is supplied as a file that is opened and read from by the driver.
-
-HPI_DSPCODE_FIRMWARE:
-DSP code is read using the hotplug firmware loader module.
-     Only valid when compiling the HPI kernel driver under Linux.
 */
 /***********************************************************************/
 #ifndef _HPIDSPCD_H_
@@ -40,37 +27,56 @@ DSP code is read using the hotplug firmware loader module.
 
 #include "hpi_internal.h"
 
-#ifndef DISABLE_PRAGMA_PACK1
-#pragma pack(push, 1)
-#endif
+/** Code header version is decimal encoded e.g. 4.06.10 is 40601 */
+#define HPI_VER_DECIMAL ((int)(HPI_VER_MAJOR(HPI_VER) * 10000 + \
+HPI_VER_MINOR(HPI_VER) * 100 + HPI_VER_RELEASE(HPI_VER)))
+
+/** Header structure for dsp firmware file
+ This structure must match that used in s2bin.c for generation of asidsp.bin
+ */
+/*#ifndef DISABLE_PRAGMA_PACK1 */
+/*#pragma pack(push, 1) */
+/*#endif */
+struct code_header {
+	/** Size in bytes including header */
+	u32 size;
+	/** File type tag "CODE" == 0x45444F43 */
+	u32 type;
+	/** Adapter model number */
+	u32 adapter;
+	/** Firmware version*/
+	u32 version;
+	/** Data checksum */
+	u32 checksum;
+};
+/*#ifndef DISABLE_PRAGMA_PACK1 */
+/*#pragma pack(pop) */
+/*#endif */
+
+/*? Don't need the pragmas? */
+compile_time_assert((sizeof(struct code_header) == 20), code_header_size);
 
 /** Descriptor for dspcode from firmware loader */
 struct dsp_code {
-	/**  Firmware descriptor */
-	const struct firmware *ps_firmware;
-	struct pci_dev *ps_dev;
+	/** copy of  file header */
+	struct code_header header;
 	/** Expected number of words in the whole dsp code,INCL header */
-	long int block_length;
+	u32 block_length;
 	/** Number of words read so far */
-	long int word_count;
-	/** Version read from dsp code file */
-	u32 version;
-	/** CRC read from dsp code file */
-	u32 crc;
-};
+	u32 word_count;
 
-#ifndef DISABLE_PRAGMA_PACK1
-#pragma pack(pop)
-#endif
+	/** internal state of DSP code reader */
+	struct dsp_code_private *pvt;
+};
 
-/** Prepare *psDspCode to refer to the requuested adapter.
- Searches the file, or selects the appropriate linked array
+/** Prepare *psDspCode to refer to the requested adapter's firmware.
+Code file name is obtained from HpiOs_GetDspCodePath
 
 \return 0 for success, or error code if requested code is not available
 */
 short hpi_dsp_code_open(
 	/** Code identifier, usually adapter family */
-	u32 adapter,
+	u32 adapter, void *pci_dev,
 	/** Pointer to DSP code control structure */
 	struct dsp_code *ps_dsp_code,
 	/** Pointer to dword to receive OS specific error code */

sound/pci/asihpi/hpifunc.c

@@ -1663,68 +1663,64 @@ u16 hpi_channel_mode_get(u32 h_control, u16 *mode)
 u16 hpi_cobranet_hmi_write(u32 h_control, u32 hmi_address, u32 byte_count,
 	u8 *pb_data)
 {
-	struct hpi_message hm;
-	struct hpi_response hr;
+	struct hpi_msg_cobranet_hmiwrite hm;
+	struct hpi_response_header hr;
 
-	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROLEX,
-		HPI_CONTROL_SET_STATE);
-	if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index))
-		return HPI_ERROR_INVALID_HANDLE;
+	hpi_init_message_responseV1(&hm.h, sizeof(hm), &hr, sizeof(hr),
+		HPI_OBJ_CONTROL, HPI_CONTROL_SET_STATE);
 
-	hm.u.cx.u.cobranet_data.byte_count = byte_count;
-	hm.u.cx.u.cobranet_data.hmi_address = hmi_address;
+	if (hpi_handle_indexes(h_control, &hm.h.adapter_index,
+			&hm.h.obj_index))
+		return HPI_ERROR_INVALID_HANDLE;
 
-	if (byte_count <= 8) {
-		memcpy(hm.u.cx.u.cobranet_data.data, pb_data, byte_count);
-		hm.u.cx.attribute = HPI_COBRANET_SET;
-	} else {
-		hm.u.cx.u.cobranet_bigdata.pb_data = pb_data;
-		hm.u.cx.attribute = HPI_COBRANET_SET_DATA;
-	}
+	if (byte_count > sizeof(hm.bytes))
+		return HPI_ERROR_MESSAGE_BUFFER_TOO_SMALL;
 
-	hpi_send_recv(&hm, &hr);
+	hm.p.attribute = HPI_COBRANET_SET;
+	hm.p.byte_count = byte_count;
+	hm.p.hmi_address = hmi_address;
+	memcpy(hm.bytes, pb_data, byte_count);
+	hm.h.size = (u16)(sizeof(hm.h) + sizeof(hm.p) + byte_count);
 
+	hpi_send_recvV1(&hm.h, &hr);
 	return hr.error;
 }
 
 u16 hpi_cobranet_hmi_read(u32 h_control, u32 hmi_address, u32 max_byte_count,
 	u32 *pbyte_count, u8 *pb_data)
 {
-	struct hpi_message hm;
-	struct hpi_response hr;
+	struct hpi_msg_cobranet_hmiread hm;
+	struct hpi_res_cobranet_hmiread hr;
 
-	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROLEX,
-		HPI_CONTROL_GET_STATE);
-	if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index))
+	hpi_init_message_responseV1(&hm.h, sizeof(hm), &hr.h, sizeof(hr),
+		HPI_OBJ_CONTROL, HPI_CONTROL_GET_STATE);
+
+	if (hpi_handle_indexes(h_control, &hm.h.adapter_index,
+			&hm.h.obj_index))
 		return HPI_ERROR_INVALID_HANDLE;
 
-	hm.u.cx.u.cobranet_data.byte_count = max_byte_count;
-	hm.u.cx.u.cobranet_data.hmi_address = hmi_address;
+	if (max_byte_count > sizeof(hr.bytes))
+		return HPI_ERROR_RESPONSE_BUFFER_TOO_SMALL;
 
-	if (max_byte_count <= 8) {
-		hm.u.cx.attribute = HPI_COBRANET_GET;
-	} else {
-		hm.u.cx.u.cobranet_bigdata.pb_data = pb_data;
-		hm.u.cx.attribute = HPI_COBRANET_GET_DATA;
-	}
+	hm.p.attribute = HPI_COBRANET_GET;
+	hm.p.byte_count = max_byte_count;
+	hm.p.hmi_address = hmi_address;
 
-	hpi_send_recv(&hm, &hr);
-	if (!hr.error && pb_data) {
+	hpi_send_recvV1(&hm.h, &hr.h);
 
-		*pbyte_count = hr.u.cx.u.cobranet_data.byte_count;
+	if (!hr.h.error && pb_data) {
+		if (hr.byte_count > sizeof(hr.bytes))
 
-		if (*pbyte_count < max_byte_count)
-			max_byte_count = *pbyte_count;
+			return HPI_ERROR_RESPONSE_BUFFER_TOO_SMALL;
 
-		if (hm.u.cx.attribute == HPI_COBRANET_GET) {
-			memcpy(pb_data, hr.u.cx.u.cobranet_data.data,
-				max_byte_count);
-		} else {
+		*pbyte_count = hr.byte_count;
 
-		}
+		if (hr.byte_count < max_byte_count)
+			max_byte_count = *pbyte_count;
 
+		memcpy(pb_data, hr.bytes, max_byte_count);
 	}
-	return hr.error;
+	return hr.h.error;
 }
 
 u16 hpi_cobranet_hmi_get_status(u32 h_control, u32 *pstatus,
@@ -1733,23 +1729,23 @@ u16 hpi_cobranet_hmi_get_status(u32 h_control, u32 *pstatus,
 	struct hpi_message hm;
 	struct hpi_response hr;
 
-	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROLEX,
+	hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
 		HPI_CONTROL_GET_STATE);
 	if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index))
 		return HPI_ERROR_INVALID_HANDLE;
 
-	hm.u.cx.attribute = HPI_COBRANET_GET_STATUS;
+	hm.u.c.attribute = HPI_COBRANET_GET_STATUS;
 
 	hpi_send_recv(&hm, &hr);
 	if (!hr.error) {
 		if (pstatus)
-			*pstatus = hr.u.cx.u.cobranet_status.status;
+			*pstatus = hr.u.cu.cobranet.status.status;
 		if (preadable_size)
 			*preadable_size =
-				hr.u.cx.u.cobranet_status.readable_size;
+				hr.u.cu.cobranet.status.readable_size;
 		if (pwriteable_size)
 			*pwriteable_size =
-				hr.u.cx.u.cobranet_status.writeable_size;
+				hr.u.cu.cobranet.status.writeable_size;
 	}
 	return hr.error;
 }

sound/pci/asihpi/hpimsginit.c

@@ -46,7 +46,7 @@ static void hpi_init_message(struct hpi_message *phm, u16 object,
 	if (gwSSX2_bypass)
 		phm->type = HPI_TYPE_SSX2BYPASS_MESSAGE;
 	else
-		phm->type = HPI_TYPE_MESSAGE;
+		phm->type = HPI_TYPE_REQUEST;
 	phm->object = object;
 	phm->function = function;
 	phm->version = 0;
@@ -89,7 +89,7 @@ static void hpi_init_messageV1(struct hpi_message_header *phm, u16 size,
 	memset(phm, 0, sizeof(*phm));
 	if ((object > 0) && (object <= HPI_OBJ_MAXINDEX)) {
 		phm->size = size;
-		phm->type = HPI_TYPE_MESSAGE;
+		phm->type = HPI_TYPE_REQUEST;
 		phm->object = object;
 		phm->function = function;
 		phm->version = 1;

sound/pci/asihpi/hpimsgx.c

@@ -16,7 +16,7 @@
     along with this program; if not, write to the Free Software
     Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 
-Extended Message Function With Response Cacheing
+Extended Message Function With Response Caching
 
 (C) Copyright AudioScience Inc. 2002
 *****************************************************************************/
@@ -186,7 +186,6 @@ static void subsys_message(struct hpi_message *phm, struct hpi_response *phr,
 		/* Initialize this module's internal state */
 		hpios_msgxlock_init(&msgx_lock);
 		memset(&hpi_entry_points, 0, sizeof(hpi_entry_points));
-		hpios_locked_mem_init();
 		/* Init subsys_findadapters response to no-adapters */
 		HPIMSGX__reset(HPIMSGX_ALLADAPTERS);
 		hpi_init_response(phr, HPI_OBJ_SUBSYSTEM,
@@ -197,7 +196,6 @@ static void subsys_message(struct hpi_message *phm, struct hpi_response *phr,
 	case HPI_SUBSYS_DRIVER_UNLOAD:
 		HPI_COMMON(phm, phr);
 		HPIMSGX__cleanup(HPIMSGX_ALLADAPTERS, h_owner);
-		hpios_locked_mem_free_all();
 		hpi_init_response(phr, HPI_OBJ_SUBSYSTEM,
 			HPI_SUBSYS_DRIVER_UNLOAD, 0);
 		return;
@@ -315,7 +313,7 @@ void hpi_send_recv_ex(struct hpi_message *phm, struct hpi_response *phr,
 {
 	HPI_DEBUG_MESSAGE(DEBUG, phm);
 
-	if (phm->type != HPI_TYPE_MESSAGE) {
+	if (phm->type != HPI_TYPE_REQUEST) {
 		hpi_init_response(phr, phm->object, phm->function,
 			HPI_ERROR_INVALID_TYPE);
 		return;

sound/pci/asihpi/hpioctl.c

@@ -1,7 +1,7 @@
 /*******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2010  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -157,11 +157,6 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
 		goto out;
 	}
 
-	if (hm->h.adapter_index >= HPI_MAX_ADAPTERS) {
-		err = -EINVAL;
-		goto out;
-	}
-
 	switch (hm->h.function) {
 	case HPI_SUBSYS_CREATE_ADAPTER:
 	case HPI_ADAPTER_DELETE:
@@ -187,7 +182,6 @@ 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 ((adapter > HPI_MAX_ADAPTERS) || (!pa->type)) {
 			hpi_init_response(&hr->r0, HPI_OBJ_ADAPTER,
@@ -203,6 +197,8 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
 			goto out;
 		}
 
+		pa = &adapters[adapter];
+
 		if (mutex_lock_interruptible(&adapters[adapter].mutex)) {
 			err = -EINTR;
 			goto out;

sound/pci/asihpi/hpios.c

@@ -39,10 +39,6 @@ void hpios_delay_micro_seconds(u32 num_micro_sec)
 
 }
 
-void hpios_locked_mem_init(void)
-{
-}
-
 /** Allocated an area of locked memory for bus master DMA operations.
 
 On error, return -ENOMEM, and *pMemArea.size = 0
@@ -85,7 +81,3 @@ u16 hpios_locked_mem_free(struct consistent_dma_area *p_mem_area)
 		return 1;
 	}
 }
-
-void hpios_locked_mem_free_all(void)
-{
-}

sound/pci/asihpi/hpios.h

@@ -38,6 +38,7 @@ HPI Operating System Specific macros for Linux Kernel driver
 #include <linux/firmware.h>
 #include <linux/interrupt.h>
 #include <linux/pci.h>
+#include <linux/mutex.h>
 
 #define HPI_NO_OS_FILE_OPS
 

sound/pci/atiixp.c

@@ -1624,7 +1624,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card,
 	}
 
 	if (request_irq(pci->irq, snd_atiixp_interrupt, IRQF_SHARED,
-			card->shortname, chip)) {
+			KBUILD_MODNAME, chip)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		snd_atiixp_free(chip);
 		return -EBUSY;
@@ -1701,7 +1701,7 @@ static void __devexit snd_atiixp_remove(struct pci_dev *pci)
 }
 
 static struct pci_driver driver = {
-	.name = "ATI IXP AC97 controller",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_atiixp_ids,
 	.probe = snd_atiixp_probe,
 	.remove = __devexit_p(snd_atiixp_remove),

sound/pci/atiixp_modem.c

@@ -1260,7 +1260,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card,
 	}
 
 	if (request_irq(pci->irq, snd_atiixp_interrupt, IRQF_SHARED,
-			card->shortname, chip)) {
+			KBUILD_MODNAME, chip)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		snd_atiixp_free(chip);
 		return -EBUSY;
@@ -1332,7 +1332,7 @@ static void __devexit snd_atiixp_remove(struct pci_dev *pci)
 }
 
 static struct pci_driver driver = {
-	.name = "ATI IXP MC97 controller",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_atiixp_ids,
 	.probe = snd_atiixp_probe,
 	.remove = __devexit_p(snd_atiixp_remove),

sound/pci/au88x0/au88x0.c

@@ -196,7 +196,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip)
 	}
 
 	if ((err = request_irq(pci->irq, vortex_interrupt,
-	                       IRQF_SHARED, CARD_NAME_SHORT,
+			       IRQF_SHARED, KBUILD_MODNAME,
 	                       chip)) != 0) {
 		printk(KERN_ERR "cannot grab irq\n");
 		goto irq_out;
@@ -375,7 +375,7 @@ static void __devexit snd_vortex_remove(struct pci_dev *pci)
 
 // pci_driver definition
 static struct pci_driver driver = {
-	.name = CARD_NAME_SHORT,
+	.name = KBUILD_MODNAME,
 	.id_table = snd_vortex_ids,
 	.probe = snd_vortex_probe,
 	.remove = __devexit_p(snd_vortex_remove),

sound/pci/aw2/aw2-alsa.c

@@ -171,7 +171,7 @@ MODULE_DEVICE_TABLE(pci, snd_aw2_ids);
 
 /* pci_driver definition */
 static struct pci_driver driver = {
-	.name = "Emagic Audiowerk 2",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_aw2_ids,
 	.probe = snd_aw2_probe,
 	.remove = __devexit_p(snd_aw2_remove),
@@ -317,7 +317,7 @@ static int __devinit snd_aw2_create(struct snd_card *card,
 	snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt);
 
 	if (request_irq(pci->irq, snd_aw2_saa7146_interrupt,
-			IRQF_SHARED, "Audiowerk2", chip)) {
+			IRQF_SHARED, KBUILD_MODNAME, chip)) {
 		printk(KERN_ERR "aw2: Cannot grab irq %d\n", pci->irq);
 
 		iounmap(chip->iobase_virt);

sound/pci/azt3328.c

@@ -2559,7 +2559,7 @@ snd_azf3328_create(struct snd_card *card,
 	codec_setup->name = "I2S_OUT";
 
 	if (request_irq(pci->irq, snd_azf3328_interrupt,
-			IRQF_SHARED, card->shortname, chip)) {
+			IRQF_SHARED, KBUILD_MODNAME, chip)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		err = -EBUSY;
 		goto out_err;
@@ -2860,7 +2860,7 @@ snd_azf3328_resume(struct pci_dev *pci)
 
 
 static struct pci_driver driver = {
-	.name = "AZF3328",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_azf3328_ids,
 	.probe = snd_azf3328_probe,
 	.remove = __devexit_p(snd_azf3328_remove),

sound/pci/bt87x.c

@@ -760,7 +760,7 @@ static int __devinit snd_bt87x_create(struct snd_card *card,
 	snd_bt87x_writel(chip, REG_INT_STAT, MY_INTERRUPTS);
 
 	err = request_irq(pci->irq, snd_bt87x_interrupt, IRQF_SHARED,
-			  "Bt87x audio", chip);
+			  KBUILD_MODNAME, chip);
 	if (err < 0) {
 		snd_printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
 		goto fail;
@@ -965,7 +965,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_default_ids) = {
 };
 
 static struct pci_driver driver = {
-	.name = "Bt87x",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_bt87x_ids,
 	.probe = snd_bt87x_probe,
 	.remove = __devexit_p(snd_bt87x_remove),

sound/pci/ca0106/ca0106_main.c

@@ -1666,7 +1666,7 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card,
 	}
 
 	if (request_irq(pci->irq, snd_ca0106_interrupt,
-			IRQF_SHARED, "snd_ca0106", chip)) {
+			IRQF_SHARED, KBUILD_MODNAME, chip)) {
 		snd_ca0106_free(chip);
 		printk(KERN_ERR "cannot grab irq\n");
 		return -EBUSY;
@@ -1933,7 +1933,7 @@ MODULE_DEVICE_TABLE(pci, snd_ca0106_ids);
 
 // pci_driver definition
 static struct pci_driver driver = {
-	.name = "CA0106",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_ca0106_ids,
 	.probe = snd_ca0106_probe,
 	.remove = __devexit_p(snd_ca0106_remove),

sound/pci/cmipci.c

@@ -3053,7 +3053,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
 	cm->iobase = pci_resource_start(pci, 0);
 
 	if (request_irq(pci->irq, snd_cmipci_interrupt,
-			IRQF_SHARED, card->driver, cm)) {
+			IRQF_SHARED, KBUILD_MODNAME, cm)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		snd_cmipci_free(cm);
 		return -EBUSY;
@@ -3398,7 +3398,7 @@ static int snd_cmipci_resume(struct pci_dev *pci)
 #endif /* CONFIG_PM */
 
 static struct pci_driver driver = {
-	.name = "C-Media PCI",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_cmipci_ids,
 	.probe = snd_cmipci_probe,
 	.remove = __devexit_p(snd_cmipci_remove),

sound/pci/cs4281.c

@@ -1382,7 +1382,7 @@ static int __devinit snd_cs4281_create(struct snd_card *card,
 	}
 	
 	if (request_irq(pci->irq, snd_cs4281_interrupt, IRQF_SHARED,
-			"CS4281", chip)) {
+			KBUILD_MODNAME, chip)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		snd_cs4281_free(chip);
 		return -ENOMEM;
@@ -2085,7 +2085,7 @@ static int cs4281_resume(struct pci_dev *pci)
 #endif /* CONFIG_PM */
 
 static struct pci_driver driver = {
-	.name = "CS4281",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_cs4281_ids,
 	.probe = snd_cs4281_probe,
 	.remove = __devexit_p(snd_cs4281_remove),

sound/pci/cs46xx/cs46xx.c

@@ -162,7 +162,7 @@ static void __devexit snd_card_cs46xx_remove(struct pci_dev *pci)
 }
 
 static struct pci_driver driver = {
-	.name = "Sound Fusion CS46xx",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_cs46xx_ids,
 	.probe = snd_card_cs46xx_probe,
 	.remove = __devexit_p(snd_card_cs46xx_remove),

sound/pci/cs46xx/cs46xx_lib.c

@@ -3835,7 +3835,7 @@ int __devinit snd_cs46xx_create(struct snd_card *card,
 	}
 
 	if (request_irq(pci->irq, snd_cs46xx_interrupt, IRQF_SHARED,
-			"CS46XX", chip)) {
+			KBUILD_MODNAME, chip)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		snd_cs46xx_free(chip);
 		return -EBUSY;

sound/pci/cs5530.c

@@ -285,7 +285,7 @@ static int __devinit snd_cs5530_probe(struct pci_dev *pci,
 }
 
 static struct pci_driver driver = {
-	.name = "CS5530_Audio",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_cs5530_ids,
 	.probe = snd_cs5530_probe,
 	.remove = __devexit_p(snd_cs5530_remove),

sound/pci/cs5535audio/cs5535audio.c

@@ -311,7 +311,7 @@ static int __devinit snd_cs5535audio_create(struct snd_card *card,
 	cs5535au->port = pci_resource_start(pci, 0);
 
 	if (request_irq(pci->irq, snd_cs5535audio_interrupt,
-			IRQF_SHARED, "CS5535 Audio", cs5535au)) {
+			IRQF_SHARED, KBUILD_MODNAME, cs5535au)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		err = -EBUSY;
 		goto sndfail;
@@ -395,7 +395,7 @@ static void __devexit snd_cs5535audio_remove(struct pci_dev *pci)
 }
 
 static struct pci_driver driver = {
-	.name = DRIVER_NAME,
+	.name = KBUILD_MODNAME,
 	.id_table = snd_cs5535audio_ids,
 	.probe = snd_cs5535audio_probe,
 	.remove = __devexit_p(snd_cs5535audio_remove),

sound/pci/ctxfi/ct20k2reg.h

@@ -55,6 +55,7 @@
 /* GPIO Registers */
 #define GPIO_DATA           0x1B7020
 #define GPIO_CTRL           0x1B7024
+#define GPIO_EXT_DATA       0x1B70A0
 
 /* Virtual memory registers */
 #define VMEM_PTPAL          0x1C6300 /* 0x1C6300 + (16 * Chn) */

sound/pci/ctxfi/ctatc.c

@@ -18,7 +18,6 @@
 #include "ctatc.h"
 #include "ctpcm.h"
 #include "ctmixer.h"
-#include "cthardware.h"
 #include "ctsrc.h"
 #include "ctamixer.h"
 #include "ctdaio.h"
@@ -30,7 +29,6 @@
 #include <sound/asoundef.h>
 
 #define MONO_SUM_SCALE	0x19a8	/* 2^(-0.5) in 14-bit floating format */
-#define DAIONUM		7
 #define MAX_MULTI_CHN	8
 
 #define IEC958_DEFAULT_CON ((IEC958_AES0_NONAUDIO \
@@ -53,6 +51,8 @@ static struct snd_pci_quirk __devinitdata subsys_20k1_list[] = {
 static struct snd_pci_quirk __devinitdata subsys_20k2_list[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB0760,
 		      "SB0760", CTSB0760),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB1270,
+		      "SB1270", CTSB1270),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB08801,
 		      "SB0880", CTSB0880),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB08802,
@@ -75,6 +75,7 @@ static const char *ct_subsys_name[NUM_CTCARDS] = {
 	[CTSB0760]	= "SB076x",
 	[CTHENDRIX]	= "Hendrix",
 	[CTSB0880]	= "SB0880",
+	[CTSB1270]      = "SB1270",
 	[CT20K2_UNKNOWN] = "Unknown",
 };
 
@@ -459,12 +460,12 @@ static void setup_src_node_conf(struct ct_atc *atc, struct ct_atc_pcm *apcm,
 				apcm->substream->runtime->rate);
 	*n_srcc = 0;
 
-	if (1 == atc->msr) {
+	if (1 == atc->msr) { /* FIXME: do we really need SRC here if pitch==1 */
 		*n_srcc = apcm->substream->runtime->channels;
 		conf[0].pitch = pitch;
 		conf[0].mix_msr = conf[0].imp_msr = conf[0].msr = 1;
 		conf[0].vo = 1;
-	} else if (2 == atc->msr) {
+	} else if (2 <= atc->msr) {
 		if (0x8000000 < pitch) {
 			/* Need two-stage SRCs, SRCIMPs and
 			 * AMIXERs for converting format */
@@ -970,11 +971,39 @@ static int atc_select_mic_in(struct ct_atc *atc)
 	return 0;
 }
 
-static int atc_have_digit_io_switch(struct ct_atc *atc)
+static struct capabilities atc_capabilities(struct ct_atc *atc)
 {
 	struct hw *hw = atc->hw;
 
-	return hw->have_digit_io_switch(hw);
+	return hw->capabilities(hw);
+}
+
+static int atc_output_switch_get(struct ct_atc *atc)
+{
+	struct hw *hw = atc->hw;
+
+	return hw->output_switch_get(hw);
+}
+
+static int atc_output_switch_put(struct ct_atc *atc, int position)
+{
+	struct hw *hw = atc->hw;
+
+	return hw->output_switch_put(hw, position);
+}
+
+static int atc_mic_source_switch_get(struct ct_atc *atc)
+{
+	struct hw *hw = atc->hw;
+
+	return hw->mic_source_switch_get(hw);
+}
+
+static int atc_mic_source_switch_put(struct ct_atc *atc, int position)
+{
+	struct hw *hw = atc->hw;
+
+	return hw->mic_source_switch_put(hw, position);
 }
 
 static int atc_select_digit_io(struct ct_atc *atc)
@@ -1045,6 +1074,11 @@ static int atc_line_in_unmute(struct ct_atc *atc, unsigned char state)
 	return atc_daio_unmute(atc, state, LINEIM);
 }
 
+static int atc_mic_unmute(struct ct_atc *atc, unsigned char state)
+{
+	return atc_daio_unmute(atc, state, MIC);
+}
+
 static int atc_spdif_out_unmute(struct ct_atc *atc, unsigned char state)
 {
 	return atc_daio_unmute(atc, state, SPDIFOO);
@@ -1331,17 +1365,20 @@ static int atc_get_resources(struct ct_atc *atc)
 	struct srcimp_mgr *srcimp_mgr;
 	struct sum_desc sum_dsc = {0};
 	struct sum_mgr *sum_mgr;
-	int err, i;
+	int err, i, num_srcs, num_daios;
 
-	atc->daios = kzalloc(sizeof(void *)*(DAIONUM), GFP_KERNEL);
+	num_daios = ((atc->model == CTSB1270) ? 8 : 7);
+	num_srcs = ((atc->model == CTSB1270) ? 6 : 4);
+
+	atc->daios = kzalloc(sizeof(void *)*num_daios, GFP_KERNEL);
 	if (!atc->daios)
 		return -ENOMEM;
 
-	atc->srcs = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL);
+	atc->srcs = kzalloc(sizeof(void *)*num_srcs, GFP_KERNEL);
 	if (!atc->srcs)
 		return -ENOMEM;
 
-	atc->srcimps = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL);
+	atc->srcimps = kzalloc(sizeof(void *)*num_srcs, GFP_KERNEL);
 	if (!atc->srcimps)
 		return -ENOMEM;
 
@@ -1351,8 +1388,9 @@ static int atc_get_resources(struct ct_atc *atc)
 
 	daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO];
 	da_desc.msr = atc->msr;
-	for (i = 0, atc->n_daio = 0; i < DAIONUM-1; i++) {
-		da_desc.type = i;
+	for (i = 0, atc->n_daio = 0; i < num_daios; i++) {
+		da_desc.type = (atc->model != CTSB073X) ? i :
+			     ((i == SPDIFIO) ? SPDIFI1 : i);
 		err = daio_mgr->get_daio(daio_mgr, &da_desc,
 					(struct daio **)&atc->daios[i]);
 		if (err) {
@@ -1362,23 +1400,12 @@ static int atc_get_resources(struct ct_atc *atc)
 		}
 		atc->n_daio++;
 	}
-	if (atc->model == CTSB073X)
-		da_desc.type = SPDIFI1;
-	else
-		da_desc.type = SPDIFIO;
-	err = daio_mgr->get_daio(daio_mgr, &da_desc,
-				(struct daio **)&atc->daios[i]);
-	if (err) {
-		printk(KERN_ERR "ctxfi: Failed to get S/PDIF-in resource!!!\n");
-		return err;
-	}
-	atc->n_daio++;
 
 	src_mgr = atc->rsc_mgrs[SRC];
 	src_dsc.multi = 1;
 	src_dsc.msr = atc->msr;
 	src_dsc.mode = ARCRW;
-	for (i = 0, atc->n_src = 0; i < (2*2); i++) {
+	for (i = 0, atc->n_src = 0; i < num_srcs; i++) {
 		err = src_mgr->get_src(src_mgr, &src_dsc,
 					(struct src **)&atc->srcs[i]);
 		if (err)
@@ -1388,8 +1415,8 @@ static int atc_get_resources(struct ct_atc *atc)
 	}
 
 	srcimp_mgr = atc->rsc_mgrs[SRCIMP];
-	srcimp_dsc.msr = 8; /* SRCIMPs for S/PDIFIn SRT */
-	for (i = 0, atc->n_srcimp = 0; i < (2*1); i++) {
+	srcimp_dsc.msr = 8;
+	for (i = 0, atc->n_srcimp = 0; i < num_srcs; i++) {
 		err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc,
 					(struct srcimp **)&atc->srcimps[i]);
 		if (err)
@@ -1397,15 +1424,6 @@ static int atc_get_resources(struct ct_atc *atc)
 
 		atc->n_srcimp++;
 	}
-	srcimp_dsc.msr = 8; /* SRCIMPs for LINE/MICIn SRT */
-	for (i = 0; i < (2*1); i++) {
-		err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc,
-				(struct srcimp **)&atc->srcimps[2*1+i]);
-		if (err)
-			return err;
-
-		atc->n_srcimp++;
-	}
 
 	sum_mgr = atc->rsc_mgrs[SUM];
 	sum_dsc.msr = atc->msr;
@@ -1488,6 +1506,18 @@ static void atc_connect_resources(struct ct_atc *atc)
 	src = atc->srcs[3];
 	mixer->set_input_right(mixer, MIX_LINE_IN, &src->rsc);
 
+	if (atc->model == CTSB1270) {
+		/* Titanium HD has a dedicated ADC for the Mic. */
+		dai = container_of(atc->daios[MIC], struct dai, daio);
+		atc_connect_dai(atc->rsc_mgrs[SRC], dai,
+			(struct src **)&atc->srcs[4],
+			(struct srcimp **)&atc->srcimps[4]);
+		src = atc->srcs[4];
+		mixer->set_input_left(mixer, MIX_MIC_IN, &src->rsc);
+		src = atc->srcs[5];
+		mixer->set_input_right(mixer, MIX_MIC_IN, &src->rsc);
+	}
+
 	dai = container_of(atc->daios[SPDIFIO], struct dai, daio);
 	atc_connect_dai(atc->rsc_mgrs[SRC], dai,
 			(struct src **)&atc->srcs[0],
@@ -1606,12 +1636,17 @@ static struct ct_atc atc_preset __devinitdata = {
 	.line_clfe_unmute = atc_line_clfe_unmute,
 	.line_rear_unmute = atc_line_rear_unmute,
 	.line_in_unmute = atc_line_in_unmute,
+	.mic_unmute = atc_mic_unmute,
 	.spdif_out_unmute = atc_spdif_out_unmute,
 	.spdif_in_unmute = atc_spdif_in_unmute,
 	.spdif_out_get_status = atc_spdif_out_get_status,
 	.spdif_out_set_status = atc_spdif_out_set_status,
 	.spdif_out_passthru = atc_spdif_out_passthru,
-	.have_digit_io_switch = atc_have_digit_io_switch,
+	.capabilities = atc_capabilities,
+	.output_switch_get = atc_output_switch_get,
+	.output_switch_put = atc_output_switch_put,
+	.mic_source_switch_get = atc_mic_source_switch_get,
+	.mic_source_switch_put = atc_mic_source_switch_put,
 #ifdef CONFIG_PM
 	.suspend = atc_suspend,
 	.resume = atc_resume,

sound/pci/ctxfi/ctatc.h

@@ -25,6 +25,7 @@
 #include <sound/core.h>
 
 #include "ctvmem.h"
+#include "cthardware.h"
 #include "ctresource.h"
 
 enum CTALSADEVS {		/* Types of alsa devices */
@@ -115,12 +116,17 @@ struct ct_atc {
 	int (*line_clfe_unmute)(struct ct_atc *atc, unsigned char state);
 	int (*line_rear_unmute)(struct ct_atc *atc, unsigned char state);
 	int (*line_in_unmute)(struct ct_atc *atc, unsigned char state);
+	int (*mic_unmute)(struct ct_atc *atc, unsigned char state);
 	int (*spdif_out_unmute)(struct ct_atc *atc, unsigned char state);
 	int (*spdif_in_unmute)(struct ct_atc *atc, unsigned char state);
 	int (*spdif_out_get_status)(struct ct_atc *atc, unsigned int *status);
 	int (*spdif_out_set_status)(struct ct_atc *atc, unsigned int status);
 	int (*spdif_out_passthru)(struct ct_atc *atc, unsigned char state);
-	int (*have_digit_io_switch)(struct ct_atc *atc);
+	struct capabilities (*capabilities)(struct ct_atc *atc);
+	int (*output_switch_get)(struct ct_atc *atc);
+	int (*output_switch_put)(struct ct_atc *atc, int position);
+	int (*mic_source_switch_get)(struct ct_atc *atc);
+	int (*mic_source_switch_put)(struct ct_atc *atc, int position);
 
 	/* Don't touch! Used for internal object. */
 	void *rsc_mgrs[NUM_RSCTYP]; /* chip resource managers */

sound/pci/ctxfi/ctdaio.c

@@ -22,20 +22,9 @@
 #include <linux/slab.h>
 #include <linux/kernel.h>
 
-#define DAIO_RESOURCE_NUM	NUM_DAIOTYP
 #define DAIO_OUT_MAX		SPDIFOO
 
-union daio_usage {
-	struct {
-		unsigned short lineo1:1;
-		unsigned short lineo2:1;
-		unsigned short lineo3:1;
-		unsigned short lineo4:1;
-		unsigned short spdifoo:1;
-		unsigned short lineim:1;
-		unsigned short spdifio:1;
-		unsigned short spdifi1:1;
-	} bf;
+struct daio_usage {
 	unsigned short data;
 };
 
@@ -61,6 +50,7 @@ struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = {
 	[LINEO3] = {.left = 0x50, .right = 0x51},
 	[LINEO4] = {.left = 0x70, .right = 0x71},
 	[LINEIM] = {.left = 0x45, .right = 0xc5},
+	[MIC]	 = {.left = 0x55, .right = 0xd5},
 	[SPDIFOO] = {.left = 0x00, .right = 0x01},
 	[SPDIFIO] = {.left = 0x05, .right = 0x85},
 };
@@ -138,6 +128,7 @@ static unsigned int daio_device_index(enum DAIOTYP type, struct hw *hw)
 		case LINEO3:	return 5;
 		case LINEO4:	return 6;
 		case LINEIM:	return 4;
+		case MIC:	return 5;
 		default:	return -EINVAL;
 		}
 	default:
@@ -519,17 +510,17 @@ static int dai_rsc_uninit(struct dai *dai)
 
 static int daio_mgr_get_rsc(struct rsc_mgr *mgr, enum DAIOTYP type)
 {
-	if (((union daio_usage *)mgr->rscs)->data & (0x1 << type))
+	if (((struct daio_usage *)mgr->rscs)->data & (0x1 << type))
 		return -ENOENT;
 
-	((union daio_usage *)mgr->rscs)->data |= (0x1 << type);
+	((struct daio_usage *)mgr->rscs)->data |= (0x1 << type);
 
 	return 0;
 }
 
 static int daio_mgr_put_rsc(struct rsc_mgr *mgr, enum DAIOTYP type)
 {
-	((union daio_usage *)mgr->rscs)->data &= ~(0x1 << type);
+	((struct daio_usage *)mgr->rscs)->data &= ~(0x1 << type);
 
 	return 0;
 }
@@ -712,7 +703,7 @@ int daio_mgr_create(void *hw, struct daio_mgr **rdaio_mgr)
 	if (!daio_mgr)
 		return -ENOMEM;
 
-	err = rsc_mgr_init(&daio_mgr->mgr, DAIO, DAIO_RESOURCE_NUM, hw);
+	err = rsc_mgr_init(&daio_mgr->mgr, DAIO, NUM_DAIOTYP, hw);
 	if (err)
 		goto error1;
 

sound/pci/ctxfi/ctdaio.h

@@ -33,6 +33,7 @@ enum DAIOTYP {
 	SPDIFOO,	/* S/PDIF Out (Flexijack/Optical) */
 	LINEIM,
 	SPDIFIO,	/* S/PDIF In (Flexijack/Optical) on the card */
+	MIC,		/* Dedicated mic on Titanium HD */
 	SPDIFI1,	/* S/PDIF In on internal Drive Bay */
 	NUM_DAIOTYP
 };

sound/pci/ctxfi/cthardware.h

@@ -39,6 +39,7 @@ enum CTCARDS {
 	CT20K2_MODEL_FIRST = CTSB0760,
 	CTHENDRIX,
 	CTSB0880,
+	CTSB1270,
 	CT20K2_UNKNOWN,
 	NUM_CTCARDS		/* This should always be the last */
 };
@@ -60,6 +61,13 @@ struct card_conf {
 	unsigned int msr;	/* master sample rate in rsrs */
 };
 
+struct capabilities {
+	unsigned int digit_io_switch:1;
+	unsigned int dedicated_mic:1;
+	unsigned int output_switch:1;
+	unsigned int mic_source_switch:1;
+};
+
 struct hw {
 	int (*card_init)(struct hw *hw, struct card_conf *info);
 	int (*card_stop)(struct hw *hw);
@@ -70,7 +78,11 @@ struct hw {
 #endif
 	int (*is_adc_source_selected)(struct hw *hw, enum ADCSRC source);
 	int (*select_adc_source)(struct hw *hw, enum ADCSRC source);
-	int (*have_digit_io_switch)(struct hw *hw);
+	struct capabilities (*capabilities)(struct hw *hw);
+	int (*output_switch_get)(struct hw *hw);
+	int (*output_switch_put)(struct hw *hw, int position);
+	int (*mic_source_switch_get)(struct hw *hw);
+	int (*mic_source_switch_put)(struct hw *hw, int position);
 
 	/* SRC operations */
 	int (*src_rsc_get_ctrl_blk)(void **rblk);

sound/pci/ctxfi/cthw20k1.c

@@ -1777,10 +1777,17 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info)
 		return adc_init_SBx(hw, info->input, info->mic20db);
 }
 
-static int hw_have_digit_io_switch(struct hw *hw)
+static struct capabilities hw_capabilities(struct hw *hw)
 {
+	struct capabilities cap;
+
 	/* SB073x and Vista compatible cards have no digit IO switch */
-	return !(hw->model == CTSB073X || hw->model == CTUAA);
+	cap.digit_io_switch = !(hw->model == CTSB073X || hw->model == CTUAA);
+	cap.dedicated_mic = 0;
+	cap.output_switch = 0;
+	cap.mic_source_switch = 0;
+
+	return cap;
 }
 
 #define CTLBITS(a, b, c, d)	(((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
@@ -1933,7 +1940,7 @@ static int hw_card_start(struct hw *hw)
 
 	if (hw->irq < 0) {
 		err = request_irq(pci->irq, ct_20k1_interrupt, IRQF_SHARED,
-				  "ctxfi", hw);
+				  KBUILD_MODNAME, hw);
 		if (err < 0) {
 			printk(KERN_ERR "XFi: Cannot get irq %d\n", pci->irq);
 			goto error2;
@@ -2172,7 +2179,7 @@ static struct hw ct20k1_preset __devinitdata = {
 	.pll_init = hw_pll_init,
 	.is_adc_source_selected = hw_is_adc_input_selected,
 	.select_adc_source = hw_adc_input_select,
-	.have_digit_io_switch = hw_have_digit_io_switch,
+	.capabilities = hw_capabilities,
 #ifdef CONFIG_PM
 	.suspend = hw_suspend,
 	.resume = hw_resume,

sound/pci/ctxfi/cthw20k2.c

@@ -8,7 +8,7 @@
  * @File	cthw20k2.c
  *
  * @Brief
- * This file contains the implementation of hardware access methord for 20k2.
+ * This file contains the implementation of hardware access method for 20k2.
  *
  * @Author	Liu Chun
  * @Date 	May 14 2008
@@ -38,6 +38,8 @@ struct hw20k2 {
 	unsigned char dev_id;
 	unsigned char addr_size;
 	unsigned char data_size;
+
+	int mic_source;
 };
 
 static u32 hw_read_20kx(struct hw *hw, u32 reg);
@@ -1163,7 +1165,12 @@ static int hw_daio_init(struct hw *hw, const struct daio_conf *info)
 		hw_write_20kx(hw, AUDIO_IO_TX_BLRCLK, 0x01010101);
 		hw_write_20kx(hw, AUDIO_IO_RX_BLRCLK, 0);
 	} else if (2 == info->msr) {
-		hw_write_20kx(hw, AUDIO_IO_MCLK, 0x11111111);
+		if (hw->model != CTSB1270) {
+			hw_write_20kx(hw, AUDIO_IO_MCLK, 0x11111111);
+		} else {
+			/* PCM4220 on Titanium HD is different. */
+			hw_write_20kx(hw, AUDIO_IO_MCLK, 0x11011111);
+		}
 		/* Specify all playing 96khz
 		 * EA [0]	- Enabled
 		 * RTA [4:5]	- 96kHz
@@ -1175,6 +1182,10 @@ static int hw_daio_init(struct hw *hw, const struct daio_conf *info)
 		 * RTD [28:29]	- 96kHz */
 		hw_write_20kx(hw, AUDIO_IO_TX_BLRCLK, 0x11111111);
 		hw_write_20kx(hw, AUDIO_IO_RX_BLRCLK, 0);
+	} else if ((4 == info->msr) && (hw->model == CTSB1270)) {
+		hw_write_20kx(hw, AUDIO_IO_MCLK, 0x21011111);
+		hw_write_20kx(hw, AUDIO_IO_TX_BLRCLK, 0x21212121);
+		hw_write_20kx(hw, AUDIO_IO_RX_BLRCLK, 0);
 	} else {
 		printk(KERN_ALERT "ctxfi: ERROR!!! Invalid sampling rate!!!\n");
 		return -EINVAL;
@@ -1182,6 +1193,8 @@ static int hw_daio_init(struct hw *hw, const struct daio_conf *info)
 
 	for (i = 0; i < 8; i++) {
 		if (i <= 3) {
+			/* This comment looks wrong since loop is over 4  */
+			/* channels and emu20k2 supports 4 spdif IOs.     */
 			/* 1st 3 channels are SPDIFs (SB0960) */
 			if (i == 3)
 				data = 0x1001001;
@@ -1206,12 +1219,16 @@ static int hw_daio_init(struct hw *hw, const struct daio_conf *info)
 
 			hw_write_20kx(hw, AUDIO_IO_TX_CSTAT_H+(0x40*i), 0x0B);
 		} else {
+			/* Again, loop is over 4 channels not 5. */
 			/* Next 5 channels are I2S (SB0960) */
 			data = 0x11;
 			hw_write_20kx(hw, AUDIO_IO_RX_CTL+(0x40*i), data);
 			if (2 == info->msr) {
 				/* Four channels per sample period */
 				data |= 0x1000;
+			} else if (4 == info->msr) {
+				/* FIXME: check this against the chip spec */
+				data |= 0x2000;
 			}
 			hw_write_20kx(hw, AUDIO_IO_TX_CTL+(0x40*i), data);
 		}
@@ -1299,21 +1316,18 @@ static int hw_pll_init(struct hw *hw, unsigned int rsr)
 
 	pllenb = 0xB;
 	hw_write_20kx(hw, PLL_ENB, pllenb);
-	pllctl = 0x20D00000;
-	set_field(&pllctl, PLLCTL_FD, 16 - 4);
+	pllctl = 0x20C00000;
+	set_field(&pllctl, PLLCTL_B, 0);
+	set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 4 : 147 - 4);
+	set_field(&pllctl, PLLCTL_RD, 48000 == rsr ? 1 - 1 : 10 - 1);
 	hw_write_20kx(hw, PLL_CTL, pllctl);
 	mdelay(40);
+
 	pllctl = hw_read_20kx(hw, PLL_CTL);
-	set_field(&pllctl, PLLCTL_B, 0);
-	if (48000 == rsr) {
-		set_field(&pllctl, PLLCTL_FD, 16 - 2);
-		set_field(&pllctl, PLLCTL_RD, 1 - 1); /* 3000*16/1 = 48000 */
-	} else { /* 44100 */
-		set_field(&pllctl, PLLCTL_FD, 147 - 2);
-		set_field(&pllctl, PLLCTL_RD, 10 - 1); /* 3000*147/10 = 44100 */
-	}
+	set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 2 : 147 - 2);
 	hw_write_20kx(hw, PLL_CTL, pllctl);
 	mdelay(40);
+
 	for (i = 0; i < 1000; i++) {
 		pllstat = hw_read_20kx(hw, PLL_STAT);
 		if (get_field(pllstat, PLLSTAT_PD))
@@ -1557,7 +1571,7 @@ static int hw20k2_i2c_write(struct hw *hw, u16 addr, u32 data)
 
 	hw_write_20kx(hw, I2C_IF_STATUS, i2c_status);
 	hw20k2_i2c_wait_data_ready(hw);
-	/* Dummy write to trigger the write oprtation */
+	/* Dummy write to trigger the write operation */
 	hw_write_20kx(hw, I2C_IF_WDATA, 0);
 	hw20k2_i2c_wait_data_ready(hw);
 
@@ -1568,6 +1582,30 @@ static int hw20k2_i2c_write(struct hw *hw, u16 addr, u32 data)
 	return 0;
 }
 
+static void hw_dac_stop(struct hw *hw)
+{
+	u32 data;
+	data = hw_read_20kx(hw, GPIO_DATA);
+	data &= 0xFFFFFFFD;
+	hw_write_20kx(hw, GPIO_DATA, data);
+	mdelay(10);
+}
+
+static void hw_dac_start(struct hw *hw)
+{
+	u32 data;
+	data = hw_read_20kx(hw, GPIO_DATA);
+	data |= 0x2;
+	hw_write_20kx(hw, GPIO_DATA, data);
+	mdelay(50);
+}
+
+static void hw_dac_reset(struct hw *hw)
+{
+	hw_dac_stop(hw);
+	hw_dac_start(hw);
+}
+
 static int hw_dac_init(struct hw *hw, const struct dac_conf *info)
 {
 	int err;
@@ -1594,6 +1632,21 @@ static int hw_dac_init(struct hw *hw, const struct dac_conf *info)
 				   0x00000000   /* Vol Control B4 */
 				 };
 
+	if (hw->model == CTSB1270) {
+		hw_dac_stop(hw);
+		data = hw_read_20kx(hw, GPIO_DATA);
+		data &= ~0x0600;
+		if (1 == info->msr)
+			data |= 0x0000; /* Single Speed Mode 0-50kHz */
+		else if (2 == info->msr)
+			data |= 0x0200; /* Double Speed Mode 50-100kHz */
+		else
+			data |= 0x0600; /* Quad Speed Mode 100-200kHz */
+		hw_write_20kx(hw, GPIO_DATA, data);
+		hw_dac_start(hw);
+		return 0;
+	}
+
 	/* Set DAC reset bit as output */
 	data = hw_read_20kx(hw, GPIO_CTRL);
 	data |= 0x02;
@@ -1606,22 +1659,8 @@ static int hw_dac_init(struct hw *hw, const struct dac_conf *info)
 	for (i = 0; i < 2; i++) {
 		/* Reset DAC twice just in-case the chip
 		 * didn't initialized properly */
-		data = hw_read_20kx(hw, GPIO_DATA);
-		/* GPIO data bit 1 */
-		data &= 0xFFFFFFFD;
-		hw_write_20kx(hw, GPIO_DATA, data);
-		mdelay(10);
-		data |= 0x2;
-		hw_write_20kx(hw, GPIO_DATA, data);
-		mdelay(50);
-
-		/* Reset the 2nd time */
-		data &= 0xFFFFFFFD;
-		hw_write_20kx(hw, GPIO_DATA, data);
-		mdelay(10);
-		data |= 0x2;
-		hw_write_20kx(hw, GPIO_DATA, data);
-		mdelay(50);
+		hw_dac_reset(hw);
+		hw_dac_reset(hw);
 
 		if (hw20k2_i2c_read(hw, CS4382_MC1,  &cs_read.mode_control_1))
 			continue;
@@ -1725,7 +1764,11 @@ End:
 static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type)
 {
 	u32 data;
-
+	if (hw->model == CTSB1270) {
+		/* Titanium HD has two ADC chips, one for line in and one */
+		/* for MIC. We don't need to switch the ADC input. */
+		return 1;
+	}
 	data = hw_read_20kx(hw, GPIO_DATA);
 	switch (type) {
 	case ADC_MICIN:
@@ -1742,35 +1785,47 @@ static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type)
 
 #define MIC_BOOST_0DB 0xCF
 #define MIC_BOOST_STEPS_PER_DB 2
-#define MIC_BOOST_20DB (MIC_BOOST_0DB + 20 * MIC_BOOST_STEPS_PER_DB)
+
+static void hw_wm8775_input_select(struct hw *hw, u8 input, s8 gain_in_db)
+{
+	u32 adcmc, gain;
+
+	if (input > 3)
+		input = 3;
+
+	adcmc = ((u32)1 << input) | 0x100; /* Link L+R gain... */
+
+	hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, adcmc),
+				MAKE_WM8775_DATA(adcmc));
+
+	if (gain_in_db < -103)
+		gain_in_db = -103;
+	if (gain_in_db > 24)
+		gain_in_db = 24;
+
+	gain = gain_in_db * MIC_BOOST_STEPS_PER_DB + MIC_BOOST_0DB;
+
+	hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, gain),
+				MAKE_WM8775_DATA(gain));
+	/* ...so there should be no need for the following. */
+	hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, gain),
+				MAKE_WM8775_DATA(gain));
+}
 
 static int hw_adc_input_select(struct hw *hw, enum ADCSRC type)
 {
 	u32 data;
-
 	data = hw_read_20kx(hw, GPIO_DATA);
 	switch (type) {
 	case ADC_MICIN:
 		data |= (0x1 << 14);
 		hw_write_20kx(hw, GPIO_DATA, data);
-		hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101),
-				MAKE_WM8775_DATA(0x101)); /* Mic-in */
-		hw20k2_i2c_write(hw,
-				MAKE_WM8775_ADDR(WM8775_AADCL, MIC_BOOST_20DB),
-				MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */
-		hw20k2_i2c_write(hw,
-				MAKE_WM8775_ADDR(WM8775_AADCR, MIC_BOOST_20DB),
-				MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */
+		hw_wm8775_input_select(hw, 0, 20); /* Mic, 20dB */
 		break;
 	case ADC_LINEIN:
 		data &= ~(0x1 << 14);
 		hw_write_20kx(hw, GPIO_DATA, data);
-		hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x102),
-				MAKE_WM8775_DATA(0x102)); /* Line-in */
-		hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xCF),
-				MAKE_WM8775_DATA(0xCF)); /* No boost */
-		hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xCF),
-				MAKE_WM8775_DATA(0xCF)); /* No boost */
+		hw_wm8775_input_select(hw, 1, 0); /* Line-in, 0dB */
 		break;
 	default:
 		break;
@@ -1782,7 +1837,7 @@ static int hw_adc_input_select(struct hw *hw, enum ADCSRC type)
 static int hw_adc_init(struct hw *hw, const struct adc_conf *info)
 {
 	int err;
-	u32 mux = 2, data, ctl;
+	u32 data, ctl;
 
 	/*  Set ADC reset bit as output */
 	data = hw_read_20kx(hw, GPIO_CTRL);
@@ -1796,19 +1851,42 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info)
 		goto error;
 	}
 
-	/* Make ADC in normal operation */
+	/* Reset the ADC (reset is active low). */
 	data = hw_read_20kx(hw, GPIO_DATA);
 	data &= ~(0x1 << 15);
+	hw_write_20kx(hw, GPIO_DATA, data);
+
+	if (hw->model == CTSB1270) {
+		/* Set up the PCM4220 ADC on Titanium HD */
+		data &= ~0x0C;
+		if (1 == info->msr)
+			data |= 0x00; /* Single Speed Mode 32-50kHz */
+		else if (2 == info->msr)
+			data |= 0x08; /* Double Speed Mode 50-108kHz */
+		else
+			data |= 0x04; /* Quad Speed Mode 108kHz-216kHz */
+		hw_write_20kx(hw, GPIO_DATA, data);
+	}
+
 	mdelay(10);
+	/* Return the ADC to normal operation. */
 	data |= (0x1 << 15);
 	hw_write_20kx(hw, GPIO_DATA, data);
 	mdelay(50);
 
+	/* I2C write to register offset 0x0B to set ADC LRCLK polarity */
+	/* invert bit, interface format to I2S, word length to 24-bit, */
+	/* enable ADC high pass filter. Fixes bug 5323?		*/
+	hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_IC, 0x26),
+			 MAKE_WM8775_DATA(0x26));
+
 	/* Set the master mode (256fs) */
 	if (1 == info->msr) {
+		/* slave mode, 128x oversampling 256fs */
 		hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x02),
 						MAKE_WM8775_DATA(0x02));
-	} else if (2 == info->msr) {
+	} else if ((2 == info->msr) || (4 == info->msr)) {
+		/* slave mode, 64x oversampling, 256fs */
 		hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x0A),
 						MAKE_WM8775_DATA(0x0A));
 	} else {
@@ -1818,55 +1896,113 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info)
 		goto error;
 	}
 
-	/* Configure GPIO bit 14 change to line-in/mic-in */
-	ctl = hw_read_20kx(hw, GPIO_CTRL);
-	ctl |= 0x1 << 14;
-	hw_write_20kx(hw, GPIO_CTRL, ctl);
-
-	/* Check using Mic-in or Line-in */
-	data = hw_read_20kx(hw, GPIO_DATA);
-
-	if (mux == 1) {
-		/* Configures GPIO data to select Mic-in */
-		data |= 0x1 << 14;
-		hw_write_20kx(hw, GPIO_DATA, data);
-
-		hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101),
-				MAKE_WM8775_DATA(0x101)); /* Mic-in */
-		hw20k2_i2c_write(hw,
-				MAKE_WM8775_ADDR(WM8775_AADCL, MIC_BOOST_20DB),
-				MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */
-		hw20k2_i2c_write(hw,
-				MAKE_WM8775_ADDR(WM8775_AADCR, MIC_BOOST_20DB),
-				MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */
-	} else if (mux == 2) {
-		/* Configures GPIO data to select Line-in */
-		data &= ~(0x1 << 14);
-		hw_write_20kx(hw, GPIO_DATA, data);
-
-		/* Setup ADC */
-		hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x102),
-				MAKE_WM8775_DATA(0x102)); /* Line-in */
-		hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xCF),
-				MAKE_WM8775_DATA(0xCF)); /* No boost */
-		hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xCF),
-				MAKE_WM8775_DATA(0xCF)); /* No boost */
+	if (hw->model != CTSB1270) {
+		/* Configure GPIO bit 14 change to line-in/mic-in */
+		ctl = hw_read_20kx(hw, GPIO_CTRL);
+		ctl |= 0x1 << 14;
+		hw_write_20kx(hw, GPIO_CTRL, ctl);
+		hw_adc_input_select(hw, ADC_LINEIN);
 	} else {
-		printk(KERN_ALERT "ctxfi: ERROR!!! Invalid input mux!!!\n");
-		err = -EINVAL;
-		goto error;
+		hw_wm8775_input_select(hw, 0, 0);
 	}
 
 	return 0;
-
 error:
 	hw20k2_i2c_uninit(hw);
 	return err;
 }
 
-static int hw_have_digit_io_switch(struct hw *hw)
+static struct capabilities hw_capabilities(struct hw *hw)
 {
-	return 0;
+	struct capabilities cap;
+
+	cap.digit_io_switch = 0;
+	cap.dedicated_mic = hw->model == CTSB1270;
+	cap.output_switch = hw->model == CTSB1270;
+	cap.mic_source_switch = hw->model == CTSB1270;
+
+	return cap;
+}
+
+static int hw_output_switch_get(struct hw *hw)
+{
+	u32 data = hw_read_20kx(hw, GPIO_EXT_DATA);
+
+	switch (data & 0x30) {
+	case 0x00:
+	     return 0;
+	case 0x10:
+	     return 1;
+	case 0x20:
+	     return 2;
+	default:
+	     return 3;
+	}
+}
+
+static int hw_output_switch_put(struct hw *hw, int position)
+{
+	u32 data;
+
+	if (position == hw_output_switch_get(hw))
+		return 0;
+
+	/* Mute line and headphones (intended for anti-pop). */
+	data = hw_read_20kx(hw, GPIO_DATA);
+	data |= (0x03 << 11);
+	hw_write_20kx(hw, GPIO_DATA, data);
+
+	data = hw_read_20kx(hw, GPIO_EXT_DATA) & ~0x30;
+	switch (position) {
+	case 0:
+		break;
+	case 1:
+		data |= 0x10;
+		break;
+	default:
+		data |= 0x20;
+	}
+	hw_write_20kx(hw, GPIO_EXT_DATA, data);
+
+	/* Unmute line and headphones. */
+	data = hw_read_20kx(hw, GPIO_DATA);
+	data &= ~(0x03 << 11);
+	hw_write_20kx(hw, GPIO_DATA, data);
+
+	return 1;
+}
+
+static int hw_mic_source_switch_get(struct hw *hw)
+{
+	struct hw20k2 *hw20k2 = (struct hw20k2 *)hw;
+
+	return hw20k2->mic_source;
+}
+
+static int hw_mic_source_switch_put(struct hw *hw, int position)
+{
+	struct hw20k2 *hw20k2 = (struct hw20k2 *)hw;
+
+	if (position == hw20k2->mic_source)
+		return 0;
+
+	switch (position) {
+	case 0:
+		hw_wm8775_input_select(hw, 0, 0); /* Mic, 0dB */
+		break;
+	case 1:
+		hw_wm8775_input_select(hw, 1, 0); /* FP Mic, 0dB */
+		break;
+	case 2:
+		hw_wm8775_input_select(hw, 3, 0); /* Aux Ext, 0dB */
+		break;
+	default:
+		return 0;
+	}
+
+	hw20k2->mic_source = position;
+
+	return 1;
 }
 
 static irqreturn_t ct_20k2_interrupt(int irq, void *dev_id)
@@ -1925,7 +2061,7 @@ static int hw_card_start(struct hw *hw)
 
 	if (hw->irq < 0) {
 		err = request_irq(pci->irq, ct_20k2_interrupt, IRQF_SHARED,
-				  "ctxfi", hw);
+				  KBUILD_MODNAME, hw);
 		if (err < 0) {
 			printk(KERN_ERR "XFi: Cannot get irq %d\n", pci->irq);
 			goto error2;
@@ -2023,13 +2159,16 @@ static int hw_card_init(struct hw *hw, struct card_conf *info)
 	/* Reset all SRC pending interrupts */
 	hw_write_20kx(hw, SRC_IP, 0);
 
-	/* TODO: detect the card ID and configure GPIO accordingly. */
-	/* Configures GPIO (0xD802 0x98028) */
-	/*hw_write_20kx(hw, GPIO_CTRL, 0x7F07);*/
-	/* Configures GPIO (SB0880) */
-	/*hw_write_20kx(hw, GPIO_CTRL, 0xFF07);*/
-	hw_write_20kx(hw, GPIO_CTRL, 0xD802);
-
+	if (hw->model != CTSB1270) {
+		/* TODO: detect the card ID and configure GPIO accordingly. */
+		/* Configures GPIO (0xD802 0x98028) */
+		/*hw_write_20kx(hw, GPIO_CTRL, 0x7F07);*/
+		/* Configures GPIO (SB0880) */
+		/*hw_write_20kx(hw, GPIO_CTRL, 0xFF07);*/
+		hw_write_20kx(hw, GPIO_CTRL, 0xD802);
+	} else {
+		hw_write_20kx(hw, GPIO_CTRL, 0x9E5F);
+	}
 	/* Enable audio ring */
 	hw_write_20kx(hw, MIXER_AR_ENABLE, 0x01);
 
@@ -2106,7 +2245,11 @@ static struct hw ct20k2_preset __devinitdata = {
 	.pll_init = hw_pll_init,
 	.is_adc_source_selected = hw_is_adc_input_selected,
 	.select_adc_source = hw_adc_input_select,
-	.have_digit_io_switch = hw_have_digit_io_switch,
+	.capabilities = hw_capabilities,
+	.output_switch_get = hw_output_switch_get,
+	.output_switch_put = hw_output_switch_put,
+	.mic_source_switch_get = hw_mic_source_switch_get,
+	.mic_source_switch_put = hw_mic_source_switch_put,
 #ifdef CONFIG_PM
 	.suspend = hw_suspend,
 	.resume = hw_resume,

sound/pci/ctxfi/ctmixer.c

@@ -86,9 +86,7 @@ enum CTALSA_MIXER_CTL {
 	MIXER_LINEIN_C_S,
 	MIXER_MIC_C_S,
 	MIXER_SPDIFI_C_S,
-	MIXER_LINEIN_P_S,
 	MIXER_SPDIFO_P_S,
-	MIXER_SPDIFI_P_S,
 	MIXER_WAVEF_P_S,
 	MIXER_WAVER_P_S,
 	MIXER_WAVEC_P_S,
@@ -137,11 +135,11 @@ ct_kcontrol_init_table[NUM_CTALSA_MIXERS] = {
 	},
 	[MIXER_LINEIN_P] = {
 		.ctl = 1,
-		.name = "Line-in Playback Volume",
+		.name = "Line Playback Volume",
 	},
 	[MIXER_LINEIN_C] = {
 		.ctl = 1,
-		.name = "Line-in Capture Volume",
+		.name = "Line Capture Volume",
 	},
 	[MIXER_MIC_P] = {
 		.ctl = 1,
@@ -153,15 +151,15 @@ ct_kcontrol_init_table[NUM_CTALSA_MIXERS] = {
 	},
 	[MIXER_SPDIFI_P] = {
 		.ctl = 1,
-		.name = "S/PDIF-in Playback Volume",
+		.name = "IEC958 Playback Volume",
 	},
 	[MIXER_SPDIFI_C] = {
 		.ctl = 1,
-		.name = "S/PDIF-in Capture Volume",
+		.name = "IEC958 Capture Volume",
 	},
 	[MIXER_SPDIFO_P] = {
 		.ctl = 1,
-		.name = "S/PDIF-out Playback Volume",
+		.name = "Digital Playback Volume",
 	},
 	[MIXER_WAVEF_P] = {
 		.ctl = 1,
@@ -179,14 +177,13 @@ ct_kcontrol_init_table[NUM_CTALSA_MIXERS] = {
 		.ctl = 1,
 		.name = "Surround Playback Volume",
 	},
-
 	[MIXER_PCM_C_S] = {
 		.ctl = 1,
 		.name = "PCM Capture Switch",
 	},
 	[MIXER_LINEIN_C_S] = {
 		.ctl = 1,
-		.name = "Line-in Capture Switch",
+		.name = "Line Capture Switch",
 	},
 	[MIXER_MIC_C_S] = {
 		.ctl = 1,
@@ -194,19 +191,11 @@ ct_kcontrol_init_table[NUM_CTALSA_MIXERS] = {
 	},
 	[MIXER_SPDIFI_C_S] = {
 		.ctl = 1,
-		.name = "S/PDIF-in Capture Switch",
-	},
-	[MIXER_LINEIN_P_S] = {
-		.ctl = 1,
-		.name = "Line-in Playback Switch",
+		.name = "IEC958 Capture Switch",
 	},
 	[MIXER_SPDIFO_P_S] = {
 		.ctl = 1,
-		.name = "S/PDIF-out Playback Switch",
-	},
-	[MIXER_SPDIFI_P_S] = {
-		.ctl = 1,
-		.name = "S/PDIF-in Playback Switch",
+		.name = "Digital Playback Switch",
 	},
 	[MIXER_WAVEF_P_S] = {
 		.ctl = 1,
@@ -236,6 +225,8 @@ ct_mixer_recording_select(struct ct_mixer *mixer, enum CT_AMIXER_CTL type);
 static void
 ct_mixer_recording_unselect(struct ct_mixer *mixer, enum CT_AMIXER_CTL type);
 
+/* FIXME: this static looks like it would fail if more than one card was */
+/* installed. */
 static struct snd_kcontrol *kctls[2] = {NULL};
 
 static enum CT_AMIXER_CTL get_amixer_index(enum CTALSA_MIXER_CTL alsa_index)
@@ -420,6 +411,77 @@ static struct snd_kcontrol_new vol_ctl = {
 	.tlv		= { .p =  ct_vol_db_scale },
 };
 
+static int output_switch_info(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_info *info)
+{
+	static const char *const names[3] = {
+	  "FP Headphones", "Headphones", "Speakers"
+	};
+
+	return snd_ctl_enum_info(info, 1, 3, names);
+}
+
+static int output_switch_get(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
+	ucontrol->value.enumerated.item[0] = atc->output_switch_get(atc);
+	return 0;
+}
+
+static int output_switch_put(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
+	if (ucontrol->value.enumerated.item[0] > 2)
+		return -EINVAL;
+	return atc->output_switch_put(atc, ucontrol->value.enumerated.item[0]);
+}
+
+static struct snd_kcontrol_new output_ctl = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Analog Output Playback Enum",
+	.info = output_switch_info,
+	.get = output_switch_get,
+	.put = output_switch_put,
+};
+
+static int mic_source_switch_info(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_info *info)
+{
+	static const char *const names[3] = {
+	  "Mic", "FP Mic", "Aux"
+	};
+
+	return snd_ctl_enum_info(info, 1, 3, names);
+}
+
+static int mic_source_switch_get(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
+	ucontrol->value.enumerated.item[0] = atc->mic_source_switch_get(atc);
+	return 0;
+}
+
+static int mic_source_switch_put(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
+	if (ucontrol->value.enumerated.item[0] > 2)
+		return -EINVAL;
+	return atc->mic_source_switch_put(atc,
+					ucontrol->value.enumerated.item[0]);
+}
+
+static struct snd_kcontrol_new mic_source_ctl = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Mic Source Capture Enum",
+	.info = mic_source_switch_info,
+	.get = mic_source_switch_get,
+	.put = mic_source_switch_put,
+};
+
 static void
 do_line_mic_switch(struct ct_atc *atc, enum CTALSA_MIXER_CTL type)
 {
@@ -465,6 +527,7 @@ do_digit_io_switch(struct ct_atc *atc, int state)
 static void do_switch(struct ct_atc *atc, enum CTALSA_MIXER_CTL type, int state)
 {
 	struct ct_mixer *mixer = atc->mixer;
+	struct capabilities cap = atc->capabilities(atc);
 
 	/* Do changes in mixer. */
 	if ((SWH_CAPTURE_START <= type) && (SWH_CAPTURE_END >= type)) {
@@ -477,8 +540,17 @@ static void do_switch(struct ct_atc *atc, enum CTALSA_MIXER_CTL type, int state)
 		}
 	}
 	/* Do changes out of mixer. */
-	if (state && (MIXER_LINEIN_C_S == type || MIXER_MIC_C_S == type))
-		do_line_mic_switch(atc, type);
+	if (!cap.dedicated_mic &&
+	    (MIXER_LINEIN_C_S == type || MIXER_MIC_C_S == type)) {
+		if (state)
+			do_line_mic_switch(atc, type);
+		atc->line_in_unmute(atc, state);
+	} else if (cap.dedicated_mic && (MIXER_LINEIN_C_S == type))
+		atc->line_in_unmute(atc, state);
+	else if (cap.dedicated_mic && (MIXER_MIC_C_S == type))
+		atc->mic_unmute(atc, state);
+	else if (MIXER_SPDIFI_C_S == type)
+		atc->spdif_in_unmute(atc, state);
 	else if (MIXER_WAVEF_P_S == type)
 		atc->line_front_unmute(atc, state);
 	else if (MIXER_WAVES_P_S == type)
@@ -487,12 +559,8 @@ static void do_switch(struct ct_atc *atc, enum CTALSA_MIXER_CTL type, int state)
 		atc->line_clfe_unmute(atc, state);
 	else if (MIXER_WAVER_P_S == type)
 		atc->line_rear_unmute(atc, state);
-	else if (MIXER_LINEIN_P_S == type)
-		atc->line_in_unmute(atc, state);
 	else if (MIXER_SPDIFO_P_S == type)
 		atc->spdif_out_unmute(atc, state);
-	else if (MIXER_SPDIFI_P_S == type)
-		atc->spdif_in_unmute(atc, state);
 	else if (MIXER_DIGITAL_IO_S == type)
 		do_digit_io_switch(atc, state);
 
@@ -671,6 +739,7 @@ static int ct_mixer_kcontrols_create(struct ct_mixer *mixer)
 {
 	enum CTALSA_MIXER_CTL type;
 	struct ct_atc *atc = mixer->atc;
+	struct capabilities cap = atc->capabilities(atc);
 	int err;
 
 	/* Create snd kcontrol instances on demand */
@@ -684,8 +753,8 @@ static int ct_mixer_kcontrols_create(struct ct_mixer *mixer)
 		}
 	}
 
-	ct_kcontrol_init_table[MIXER_DIGITAL_IO_S].ctl =
-					atc->have_digit_io_switch(atc);
+	ct_kcontrol_init_table[MIXER_DIGITAL_IO_S].ctl = cap.digit_io_switch;
+
 	for (type = SWH_MIXER_START; type <= SWH_MIXER_END; type++) {
 		if (ct_kcontrol_init_table[type].ctl) {
 			swh_ctl.name = ct_kcontrol_init_table[type].name;
@@ -708,6 +777,17 @@ static int ct_mixer_kcontrols_create(struct ct_mixer *mixer)
 	if (err)
 		return err;
 
+	if (cap.output_switch) {
+		err = ct_mixer_kcontrol_new(mixer, &output_ctl);
+		if (err)
+			return err;
+	}
+
+	if (cap.mic_source_switch) {
+		err = ct_mixer_kcontrol_new(mixer, &mic_source_ctl);
+		if (err)
+			return err;
+	}
 	atc->line_front_unmute(atc, 1);
 	set_switch_state(mixer, MIXER_WAVEF_P_S, 1);
 	atc->line_surround_unmute(atc, 0);
@@ -719,13 +799,12 @@ static int ct_mixer_kcontrols_create(struct ct_mixer *mixer)
 	atc->spdif_out_unmute(atc, 0);
 	set_switch_state(mixer, MIXER_SPDIFO_P_S, 0);
 	atc->line_in_unmute(atc, 0);
-	set_switch_state(mixer, MIXER_LINEIN_P_S, 0);
+	if (cap.dedicated_mic)
+		atc->mic_unmute(atc, 0);
 	atc->spdif_in_unmute(atc, 0);
-	set_switch_state(mixer, MIXER_SPDIFI_P_S, 0);
-
-	set_switch_state(mixer, MIXER_PCM_C_S, 1);
-	set_switch_state(mixer, MIXER_LINEIN_C_S, 1);
-	set_switch_state(mixer, MIXER_SPDIFI_C_S, 1);
+	set_switch_state(mixer, MIXER_PCM_C_S, 0);
+	set_switch_state(mixer, MIXER_LINEIN_C_S, 0);
+	set_switch_state(mixer, MIXER_SPDIFI_C_S, 0);
 
 	return 0;
 }

sound/pci/ctxfi/xfi.c

@@ -80,11 +80,11 @@ ct_card_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		       "are 48000 and 44100, Value 48000 is assumed.\n");
 		reference_rate = 48000;
 	}
-	if ((multiple != 1) && (multiple != 2)) {
+	if ((multiple != 1) && (multiple != 2) && (multiple != 4)) {
 		printk(KERN_ERR "ctxfi: Invalid multiple value %u!!!\n",
 		       multiple);
 		printk(KERN_ERR "ctxfi: The valid values for multiple are "
-		       "1 and 2, Value 2 is assumed.\n");
+		       "1, 2 and 4, Value 2 is assumed.\n");
 		multiple = 2;
 	}
 	err = ct_atc_create(card, pci, reference_rate, multiple,
@@ -143,7 +143,7 @@ static int ct_card_resume(struct pci_dev *pci)
 #endif
 
 static struct pci_driver ct_driver = {
-	.name = "SB-XFi",
+	.name = KBUILD_MODNAME,
 	.id_table = ct_pci_dev_ids,
 	.probe = ct_card_probe,
 	.remove = __devexit_p(ct_card_remove),

sound/pci/echoaudio/echoaudio.c

@@ -1995,7 +1995,7 @@ static __devinit int snd_echo_create(struct snd_card *card,
 		ioremap_nocache(chip->dsp_registers_phys, sz);
 
 	if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED,
-			ECHOCARD_NAME, chip)) {
+			KBUILD_MODNAME, chip)) {
 		snd_echo_free(chip);
 		snd_printk(KERN_ERR "cannot grab irq\n");
 		return -EBUSY;
@@ -2286,7 +2286,7 @@ static int snd_echo_resume(struct pci_dev *pci)
 	kfree(commpage_bak);
 
 	if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED,
-			ECHOCARD_NAME, chip)) {
+			KBUILD_MODNAME, chip)) {
 		snd_echo_free(chip);
 		snd_printk(KERN_ERR "cannot grab irq\n");
 		return -EBUSY;
@@ -2327,7 +2327,7 @@ static void __devexit snd_echo_remove(struct pci_dev *pci)
 
 /* pci_driver definition */
 static struct pci_driver driver = {
-	.name = "Echoaudio " ECHOCARD_NAME,
+	.name = KBUILD_MODNAME,
 	.id_table = snd_echo_ids,
 	.probe = snd_echo_probe,
 	.remove = __devexit_p(snd_echo_remove),

sound/pci/emu10k1/emu10k1.c

@@ -264,7 +264,7 @@ static int snd_emu10k1_resume(struct pci_dev *pci)
 #endif
 
 static struct pci_driver driver = {
-	.name = "EMU10K1_Audigy",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_emu10k1_ids,
 	.probe = snd_card_emu10k1_probe,
 	.remove = __devexit_p(snd_card_emu10k1_remove),

sound/pci/emu10k1/emu10k1_main.c

@@ -1912,7 +1912,7 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
 
 	/* irq handler must be registered after I/O ports are activated */
 	if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED,
-			"EMU10K1", emu)) {
+			KBUILD_MODNAME, emu)) {
 		err = -EBUSY;
 		goto error;
 	}

sound/pci/emu10k1/emu10k1x.c

@@ -925,7 +925,7 @@ static int __devinit snd_emu10k1x_create(struct snd_card *card,
 	}
 
 	if (request_irq(pci->irq, snd_emu10k1x_interrupt,
-			IRQF_SHARED, "EMU10K1X", chip)) {
+			IRQF_SHARED, KBUILD_MODNAME, chip)) {
 		snd_printk(KERN_ERR "emu10k1x: cannot grab irq %d\n", pci->irq);
 		snd_emu10k1x_free(chip);
 		return -EBUSY;
@@ -1613,7 +1613,7 @@ MODULE_DEVICE_TABLE(pci, snd_emu10k1x_ids);
 
 // pci_driver definition
 static struct pci_driver driver = {
-	.name = "EMU10K1X",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_emu10k1x_ids,
 	.probe = snd_emu10k1x_probe,
 	.remove = __devexit_p(snd_emu10k1x_remove),

sound/pci/ens1370.c

@@ -2120,7 +2120,7 @@ static int __devinit snd_ensoniq_create(struct snd_card *card,
 	}
 	ensoniq->port = pci_resource_start(pci, 0);
 	if (request_irq(pci->irq, snd_audiopci_interrupt, IRQF_SHARED,
-			"Ensoniq AudioPCI", ensoniq)) {
+			KBUILD_MODNAME, ensoniq)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		snd_ensoniq_free(ensoniq);
 		return -EBUSY;
@@ -2489,7 +2489,7 @@ static void __devexit snd_audiopci_remove(struct pci_dev *pci)
 }
 
 static struct pci_driver driver = {
-	.name = DRIVER_NAME,
+	.name = KBUILD_MODNAME,
 	.id_table = snd_audiopci_ids,
 	.probe = snd_audiopci_probe,
 	.remove = __devexit_p(snd_audiopci_remove),

sound/pci/es1938.c

@@ -1514,7 +1514,7 @@ static int es1938_resume(struct pci_dev *pci)
 	}
 
 	if (request_irq(pci->irq, snd_es1938_interrupt,
-			IRQF_SHARED, "ES1938", chip)) {
+			IRQF_SHARED, KBUILD_MODNAME, chip)) {
 		printk(KERN_ERR "es1938: unable to grab IRQ %d, "
 		       "disabling device\n", pci->irq);
 		snd_card_disconnect(card);
@@ -1636,7 +1636,7 @@ static int __devinit snd_es1938_create(struct snd_card *card,
 	chip->mpu_port = pci_resource_start(pci, 3);
 	chip->game_port = pci_resource_start(pci, 4);
 	if (request_irq(pci->irq, snd_es1938_interrupt, IRQF_SHARED,
-			"ES1938", chip)) {
+			KBUILD_MODNAME, chip)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		snd_es1938_free(chip);
 		return -EBUSY;
@@ -1882,7 +1882,7 @@ static void __devexit snd_es1938_remove(struct pci_dev *pci)
 }
 
 static struct pci_driver driver = {
-	.name = "ESS ES1938 (Solo-1)",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_es1938_ids,
 	.probe = snd_es1938_probe,
 	.remove = __devexit_p(snd_es1938_remove),

sound/pci/es1968.c

@@ -554,9 +554,8 @@ struct es1968 {
 #else
 	struct snd_kcontrol *master_switch; /* for h/w volume control */
 	struct snd_kcontrol *master_volume;
-	spinlock_t ac97_lock;
-	struct tasklet_struct hwvol_tq;
 #endif
+	struct work_struct hwvol_work;
 
 #ifdef CONFIG_SND_ES1968_RADIO
 	struct snd_tea575x tea;
@@ -646,38 +645,23 @@ static int snd_es1968_ac97_wait_poll(struct es1968 *chip)
 static void snd_es1968_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val)
 {
 	struct es1968 *chip = ac97->private_data;
-#ifndef CONFIG_SND_ES1968_INPUT
-	unsigned long flags;
-#endif
 
 	snd_es1968_ac97_wait(chip);
 
 	/* Write the bus */
-#ifndef CONFIG_SND_ES1968_INPUT
-	spin_lock_irqsave(&chip->ac97_lock, flags);
-#endif
 	outw(val, chip->io_port + ESM_AC97_DATA);
 	/*msleep(1);*/
 	outb(reg, chip->io_port + ESM_AC97_INDEX);
 	/*msleep(1);*/
-#ifndef CONFIG_SND_ES1968_INPUT
-	spin_unlock_irqrestore(&chip->ac97_lock, flags);
-#endif
 }
 
 static unsigned short snd_es1968_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
 {
 	u16 data = 0;
 	struct es1968 *chip = ac97->private_data;
-#ifndef CONFIG_SND_ES1968_INPUT
-	unsigned long flags;
-#endif
 
 	snd_es1968_ac97_wait(chip);
 
-#ifndef CONFIG_SND_ES1968_INPUT
-	spin_lock_irqsave(&chip->ac97_lock, flags);
-#endif
 	outb(reg | 0x80, chip->io_port + ESM_AC97_INDEX);
 	/*msleep(1);*/
 
@@ -685,9 +669,6 @@ static unsigned short snd_es1968_ac97_read(struct snd_ac97 *ac97, unsigned short
 		data = inw(chip->io_port + ESM_AC97_DATA);
 		/*msleep(1);*/
 	}
-#ifndef CONFIG_SND_ES1968_INPUT
-	spin_unlock_irqrestore(&chip->ac97_lock, flags);
-#endif
 
 	return data;
 }
@@ -1904,13 +1885,10 @@ static void snd_es1968_update_pcm(struct es1968 *chip, struct esschan *es)
    (without wrap around) in response to volume button presses and then
    generating an interrupt. The pair of counters is stored in bits 1-3 and 5-7
    of a byte wide register. The meaning of bits 0 and 4 is unknown. */
-static void es1968_update_hw_volume(unsigned long private_data)
+static void es1968_update_hw_volume(struct work_struct *work)
 {
-	struct es1968 *chip = (struct es1968 *) private_data;
+	struct es1968 *chip = container_of(work, struct es1968, hwvol_work);
 	int x, val;
-#ifndef CONFIG_SND_ES1968_INPUT
-	unsigned long flags;
-#endif
 
 	/* Figure out which volume control button was pushed,
 	   based on differences from the default register
@@ -1929,18 +1907,11 @@ static void es1968_update_hw_volume(unsigned long private_data)
 	if (! chip->master_switch || ! chip->master_volume)
 		return;
 
-	/* FIXME: we can't call snd_ac97_* functions since here is in tasklet. */
-	spin_lock_irqsave(&chip->ac97_lock, flags);
-	val = chip->ac97->regs[AC97_MASTER];
+	val = snd_ac97_read(chip->ac97, AC97_MASTER);
 	switch (x) {
 	case 0x88:
 		/* mute */
 		val ^= 0x8000;
-		chip->ac97->regs[AC97_MASTER] = val;
-		outw(val, chip->io_port + ESM_AC97_DATA);
-		outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX);
-		snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
-			       &chip->master_switch->id);
 		break;
 	case 0xaa:
 		/* volume up */
@@ -1948,11 +1919,6 @@ static void es1968_update_hw_volume(unsigned long private_data)
 			val--;
 		if ((val & 0x7f00) > 0)
 			val -= 0x0100;
-		chip->ac97->regs[AC97_MASTER] = val;
-		outw(val, chip->io_port + ESM_AC97_DATA);
-		outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX);
-		snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
-			       &chip->master_volume->id);
 		break;
 	case 0x66:
 		/* volume down */
@@ -1960,14 +1926,11 @@ static void es1968_update_hw_volume(unsigned long private_data)
 			val++;
 		if ((val & 0x7f00) < 0x1f00)
 			val += 0x0100;
-		chip->ac97->regs[AC97_MASTER] = val;
-		outw(val, chip->io_port + ESM_AC97_DATA);
-		outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX);
-		snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
-			       &chip->master_volume->id);
 		break;
 	}
-	spin_unlock_irqrestore(&chip->ac97_lock, flags);
+	if (snd_ac97_update(chip->ac97, AC97_MASTER, val))
+		snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+			       &chip->master_volume->id);
 #else
 	if (!chip->input_dev)
 		return;
@@ -2013,11 +1976,7 @@ static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id)
 	outw(inw(chip->io_port + 4) & 1, chip->io_port + 4);
 
 	if (event & ESM_HWVOL_IRQ)
-#ifdef CONFIG_SND_ES1968_INPUT
-		es1968_update_hw_volume((unsigned long)chip);
-#else
-		tasklet_schedule(&chip->hwvol_tq); /* we'll do this later */
-#endif
+		schedule_work(&chip->hwvol_work);
 
 	/* else ack 'em all, i imagine */
 	outb(0xFF, chip->io_port + 0x1A);
@@ -2426,6 +2385,7 @@ static int es1968_suspend(struct pci_dev *pci, pm_message_t state)
 		return 0;
 
 	chip->in_suspend = 1;
+	cancel_work_sync(&chip->hwvol_work);
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
 	snd_pcm_suspend_all(chip->pcm);
 	snd_ac97_suspend(chip->ac97);
@@ -2638,6 +2598,7 @@ static struct snd_tea575x_ops snd_es1968_tea_ops = {
 
 static int snd_es1968_free(struct es1968 *chip)
 {
+	cancel_work_sync(&chip->hwvol_work);
 #ifdef CONFIG_SND_ES1968_INPUT
 	if (chip->input_dev)
 		input_unregister_device(chip->input_dev);
@@ -2728,10 +2689,7 @@ static int __devinit snd_es1968_create(struct snd_card *card,
 	INIT_LIST_HEAD(&chip->buf_list);
 	INIT_LIST_HEAD(&chip->substream_list);
 	mutex_init(&chip->memory_mutex);
-#ifndef CONFIG_SND_ES1968_INPUT
-	spin_lock_init(&chip->ac97_lock);
-	tasklet_init(&chip->hwvol_tq, es1968_update_hw_volume, (unsigned long)chip);
-#endif
+	INIT_WORK(&chip->hwvol_work, es1968_update_hw_volume);
 	chip->card = card;
 	chip->pci = pci;
 	chip->irq = -1;
@@ -2746,7 +2704,7 @@ static int __devinit snd_es1968_create(struct snd_card *card,
 	}
 	chip->io_port = pci_resource_start(pci, 0);
 	if (request_irq(pci->irq, snd_es1968_interrupt, IRQF_SHARED,
-			"ESS Maestro", chip)) {
+			KBUILD_MODNAME, chip)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		snd_es1968_free(chip);
 		return -EBUSY;
@@ -2925,7 +2883,7 @@ static void __devexit snd_es1968_remove(struct pci_dev *pci)
 }
 
 static struct pci_driver driver = {
-	.name = "ES1968 (ESS Maestro)",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_es1968_ids,
 	.probe = snd_es1968_probe,
 	.remove = __devexit_p(snd_es1968_remove),

sound/pci/fm801.c

@@ -1199,7 +1199,7 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 	chip->port = pci_resource_start(pci, 0);
 	if ((tea575x_tuner & TUNER_ONLY) == 0) {
 		if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_SHARED,
-				"FM801", chip)) {
+				KBUILD_MODNAME, chip)) {
 			snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq);
 			snd_fm801_free(chip);
 			return -EBUSY;
@@ -1394,7 +1394,7 @@ static int snd_fm801_resume(struct pci_dev *pci)
 #endif
 
 static struct pci_driver driver = {
-	.name = "FM801",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_fm801_ids,
 	.probe = snd_card_fm801_probe,
 	.remove = __devexit_p(snd_card_fm801_remove),

sound/pci/hda/Kconfig

@@ -14,6 +14,19 @@ menuconfig SND_HDA_INTEL
 
 if SND_HDA_INTEL
 
+config SND_HDA_PREALLOC_SIZE
+	int "Pre-allocated buffer size for HD-audio driver"
+	range 0 32768
+	default 64
+	help
+	  Specifies the default pre-allocated buffer-size in kB for the
+	  HD-audio driver.  A larger buffer (e.g. 2048) is preferred
+	  for systems using PulseAudio.  The default 64 is chosen just
+	  for compatibility reasons.
+
+	  Note that the pre-allocation size can be changed dynamically
+	  via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too.
+
 config SND_HDA_HWDEP
 	bool "Build hwdep interface for HD-audio driver"
 	select SND_HWDEP
@@ -83,6 +96,19 @@ config SND_HDA_CODEC_REALTEK
 	  snd-hda-codec-realtek.
 	  This module is automatically loaded at probing.
 
+config SND_HDA_ENABLE_REALTEK_QUIRKS
+	bool "Build static quirks for Realtek codecs"
+	depends on SND_HDA_CODEC_REALTEK
+	default y
+	help
+	  Say Y here to build the static quirks codes for Realtek codecs.
+	  If you need the "model" preset that the default BIOS auto-parser
+	  can't handle, turn this option on.
+
+	  If your device works with model=auto option, basically you don't
+	  need the quirk code.  By turning this off, you can reduce the
+	  module size quite a lot.
+
 config SND_HDA_CODEC_ANALOG
 	bool "Build Analog Device HD-audio codec support"
 	default y
@@ -171,6 +197,19 @@ config SND_HDA_CODEC_CA0110
 	  snd-hda-codec-ca0110.
 	  This module is automatically loaded at probing.
 
+config SND_HDA_CODEC_CA0132
+	bool "Build Creative CA0132 codec support"
+	depends on SND_HDA_INTEL
+	default y
+	help
+	  Say Y here to include Creative CA0132 codec support in
+	  snd-hda-intel driver.
+
+	  When the HD-audio driver is built as a module, the codec
+	  support code is also built as another module,
+	  snd-hda-codec-ca0132.
+	  This module is automatically loaded at probing.
+
 config SND_HDA_CODEC_CMEDIA
 	bool "Build C-Media HD-audio codec support"
 	default y

sound/pci/hda/Makefile

@@ -13,6 +13,7 @@ snd-hda-codec-idt-objs :=	patch_sigmatel.o
 snd-hda-codec-si3054-objs :=	patch_si3054.o
 snd-hda-codec-cirrus-objs :=	patch_cirrus.o
 snd-hda-codec-ca0110-objs :=	patch_ca0110.o
+snd-hda-codec-ca0132-objs :=	patch_ca0132.o
 snd-hda-codec-conexant-objs :=	patch_conexant.o
 snd-hda-codec-via-objs :=	patch_via.o
 snd-hda-codec-hdmi-objs :=	patch_hdmi.o hda_eld.o
@@ -42,6 +43,9 @@ endif
 ifdef CONFIG_SND_HDA_CODEC_CA0110
 obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o
 endif
+ifdef CONFIG_SND_HDA_CODEC_CA0132
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0132.o
+endif
 ifdef CONFIG_SND_HDA_CODEC_CONEXANT
 obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o
 endif

sound/pci/hda/alc260_quirks.c

@@ -0,0 +1,1272 @@
+/*
+ * ALC260 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC260 models */
+enum {
+	ALC260_AUTO,
+	ALC260_BASIC,
+	ALC260_HP,
+	ALC260_HP_DC7600,
+	ALC260_HP_3013,
+	ALC260_FUJITSU_S702X,
+	ALC260_ACER,
+	ALC260_WILL,
+	ALC260_REPLACER_672V,
+	ALC260_FAVORIT100,
+#ifdef CONFIG_SND_DEBUG
+	ALC260_TEST,
+#endif
+	ALC260_MODEL_LAST /* last tag */
+};
+
+static const hda_nid_t alc260_dac_nids[1] = {
+	/* front */
+	0x02,
+};
+
+static const hda_nid_t alc260_adc_nids[1] = {
+	/* ADC0 */
+	0x04,
+};
+
+static const hda_nid_t alc260_adc_nids_alt[1] = {
+	/* ADC1 */
+	0x05,
+};
+
+/* NIDs used when simultaneous access to both ADCs makes sense.  Note that
+ * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC.
+ */
+static const hda_nid_t alc260_dual_adc_nids[2] = {
+	/* ADC0, ADC1 */
+	0x04, 0x05
+};
+
+#define ALC260_DIGOUT_NID	0x03
+#define ALC260_DIGIN_NID	0x06
+
+static const struct hda_input_mux alc260_capture_source = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x1 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
+/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack,
+ * headphone jack and the internal CD lines since these are the only pins at
+ * which audio can appear.  For flexibility, also allow the option of
+ * recording the mixer output on the second ADC (ADC0 doesn't have a
+ * connection to the mixer output).
+ */
+static const struct hda_input_mux alc260_fujitsu_capture_sources[2] = {
+	{
+		.num_items = 3,
+		.items = {
+			{ "Mic/Line", 0x0 },
+			{ "CD", 0x4 },
+			{ "Headphone", 0x2 },
+		},
+	},
+	{
+		.num_items = 4,
+		.items = {
+			{ "Mic/Line", 0x0 },
+			{ "CD", 0x4 },
+			{ "Headphone", 0x2 },
+			{ "Mixer", 0x5 },
+		},
+	},
+
+};
+
+/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to
+ * the Fujitsu S702x, but jacks are marked differently.
+ */
+static const struct hda_input_mux alc260_acer_capture_sources[2] = {
+	{
+		.num_items = 4,
+		.items = {
+			{ "Mic", 0x0 },
+			{ "Line", 0x2 },
+			{ "CD", 0x4 },
+			{ "Headphone", 0x5 },
+		},
+	},
+	{
+		.num_items = 5,
+		.items = {
+			{ "Mic", 0x0 },
+			{ "Line", 0x2 },
+			{ "CD", 0x4 },
+			{ "Headphone", 0x6 },
+			{ "Mixer", 0x5 },
+		},
+	},
+};
+
+/* Maxdata Favorit 100XS */
+static const struct hda_input_mux alc260_favorit100_capture_sources[2] = {
+	{
+		.num_items = 2,
+		.items = {
+			{ "Line/Mic", 0x0 },
+			{ "CD", 0x4 },
+		},
+	},
+	{
+		.num_items = 3,
+		.items = {
+			{ "Line/Mic", 0x0 },
+			{ "CD", 0x4 },
+			{ "Mixer", 0x5 },
+		},
+	},
+};
+
+/*
+ * This is just place-holder, so there's something for alc_build_pcms to look
+ * at when it calculates the maximum number of channels. ALC260 has no mixer
+ * element which allows changing the channel mode, so the verb list is
+ * never used.
+ */
+static const struct hda_channel_mode alc260_modes[1] = {
+	{ 2, NULL },
+};
+
+
+/* Mixer combinations
+ *
+ * basic: base_output + input + pc_beep + capture
+ * HP: base_output + input + capture_alt
+ * HP_3013: hp_3013 + input + capture
+ * fujitsu: fujitsu + capture
+ * acer: acer + capture
+ */
+
+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),
+	HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT),
+	{ } /* end */
+};
+
+/* update HP, line and mono out pins according to the master switch */
+static void alc260_hp_master_update(struct hda_codec *codec)
+{
+	update_speakers(codec);
+}
+
+static int alc260_hp_master_sw_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.integer.value = !spec->master_mute;
+	return 0;
+}
+
+static int alc260_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_mute)
+		return 0;
+	spec->master_mute = val;
+	alc260_hp_master_update(codec);
+	return 1;
+}
+
+static const struct snd_kcontrol_new alc260_hp_output_mixer[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Master Playback Switch",
+		.subdevice = HDA_SUBDEV_NID_FLAG | 0x11,
+		.info = snd_ctl_boolean_mono_info,
+		.get = alc260_hp_master_sw_get,
+		.put = alc260_hp_master_sw_put,
+	},
+	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),
+	HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
+			      HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2, HDA_INPUT),
+	{ } /* end */
+};
+
+static const struct hda_verb alc260_hp_unsol_verbs[] = {
+	{0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{},
+};
+
+static void alc260_hp_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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 const struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Master Playback Switch",
+		.subdevice = HDA_SUBDEV_NID_FLAG | 0x11,
+		.info = snd_ctl_boolean_mono_info,
+		.get = alc260_hp_master_sw_get,
+		.put = alc260_hp_master_sw_put,
+	},
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Aux-In Playback Volume", 0x07, 0x06, HDA_INPUT),
+	HDA_CODEC_MUTE("Aux-In Playback Switch", 0x07, 0x06, HDA_INPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT),
+	{ } /* end */
+};
+
+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),
+		HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT),
+		HDA_COMPOSE_AMP_VAL(0x0a, 3, 0, HDA_OUTPUT),
+		0
+	},
+};
+
+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),
+		HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+		0
+	},
+};
+
+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),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x10, 0x0, HDA_OUTPUT),
+	{ } /* end */
+};
+
+static const struct hda_verb alc260_hp_3013_unsol_verbs[] = {
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{},
+};
+
+static void alc260_hp_3012_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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 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),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT),
+	ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN),
+	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT),
+	{ } /* end */
+};
+
+/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks.  Note that current
+ * versions of the ALC260 don't act on requests to enable mic bias from NID
+ * 0x0f (used to drive the headphone jack in these laptops).  The ALC260
+ * datasheet doesn't mention this restriction.  At this stage it's not clear
+ * whether this behaviour is intentional or is a hardware bug in chip
+ * revisions available in early 2006.  Therefore for now allow the
+ * "Headphone Jack Mode" control to span all choices, but if it turns out
+ * that the lack of mic bias for this NID is intentional we could change the
+ * mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
+ *
+ * In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006
+ * don't appear to make the mic bias available from the "line" jack, even
+ * though the NID used for this jack (0x14) can supply it.  The theory is
+ * that perhaps Acer have included blocking capacitors between the ALC260
+ * and the output jack.  If this turns out to be the case for all such
+ * models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT
+ * to ALC_PIN_DIR_INOUT_NOMICBIAS.
+ *
+ * The C20x Tablet series have a mono internal speaker which is controlled
+ * via the chip's Mono sum widget and pin complex, so include the necessary
+ * controls for such models.  On models without a "mono speaker" the control
+ * won't do anything.
+ */
+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),
+	HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
+			      HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2,
+			   HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+	ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
+	ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
+	{ } /* end */
+};
+
+/* Maxdata Favorit 100XS: one output and one input (0x12) jack
+ */
+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),
+	HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+	ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
+	{ } /* end */
+};
+
+/* Packard bell V7900  ALC260 pin usage: HP = 0x0f, Mic jack = 0x12,
+ * Line In jack = 0x14, CD audio =  0x16, pc beep = 0x17.
+ */
+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),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+	ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
+	ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+	{ } /* end */
+};
+
+/* Replacer 672V ALC260 pin usage: Mic jack = 0x12,
+ * Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f.
+ */
+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),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+	ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
+	HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x07, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("ATATI Mic Playback Switch", 0x07, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
+	ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
+	{ } /* end */
+};
+
+/*
+ * initialization 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 */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	/* Mic1 (rear panel) pin widget for input and vref at 80% */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	/* Mic2 (front panel) pin widget for input and vref at 80% */
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	/* LINE-2 is used for line-out in rear */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	/* select line-out */
+	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* LINE-OUT pin */
+	{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	/* enable HP */
+	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	/* enable Mono */
+	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	/* mute capture amp left and right */
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	/* set connection select to line in (default select for this ADC) */
+	{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
+	/* mute capture amp left and right */
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	/* set connection select to line in (default select for this ADC) */
+	{0x05, AC_VERB_SET_CONNECT_SEL, 0x02},
+	/* set vol=0 Line-Out mixer amp left and right */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	/* unmute pin widget amp left and right (no gain on this amp) */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* set vol=0 HP mixer amp left and right */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	/* unmute pin widget amp left and right (no gain on this amp) */
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* set vol=0 Mono mixer amp left and right */
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	/* unmute pin widget amp left and right (no gain on this amp) */
+	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* unmute LINE-2 out pin */
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
+	 * Line In 2 = 0x03
+	 */
+	/* mute analog inputs */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
+	/* mute Front out path */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* mute Headphone out path */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* mute Mono out path */
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{ }
+};
+
+#if 0 /* should be identical with alc260_init_verbs? */
+static const struct hda_verb alc260_hp_init_verbs[] = {
+	/* Headphone and output */
+	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+	/* mono output */
+	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	/* Mic1 (rear panel) pin widget for input and vref at 80% */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	/* Mic2 (front panel) pin widget for input and vref at 80% */
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	/* Line In pin widget for input */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	/* Line-2 pin widget for output */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	/* CD pin widget for input */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	/* unmute amp left and right */
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+	/* set connection select to line in (default select for this ADC) */
+	{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
+	/* unmute Line-Out mixer amp left and right (volume = 0) */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* mute pin widget amp left and right (no gain on this amp) */
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	/* unmute HP mixer amp left and right (volume = 0) */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* mute pin widget amp left and right (no gain on this amp) */
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
+	 * Line In 2 = 0x03
+	 */
+	/* mute analog inputs */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
+	/* Unmute Front out path */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	/* Unmute Headphone out path */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	/* Unmute Mono out path */
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	{ }
+};
+#endif
+
+static const struct hda_verb alc260_hp_3013_init_verbs[] = {
+	/* Line out and output */
+	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	/* mono output */
+	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	/* Mic1 (rear panel) pin widget for input and vref at 80% */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	/* Mic2 (front panel) pin widget for input and vref at 80% */
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	/* Line In pin widget for input */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	/* Headphone pin widget for output */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+	/* CD pin widget for input */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	/* unmute amp left and right */
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+	/* set connection select to line in (default select for this ADC) */
+	{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
+	/* unmute Line-Out mixer amp left and right (volume = 0) */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* mute pin widget amp left and right (no gain on this amp) */
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	/* unmute HP mixer amp left and right (volume = 0) */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* mute pin widget amp left and right (no gain on this amp) */
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
+	 * Line In 2 = 0x03
+	 */
+	/* mute analog inputs */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
+	/* Unmute Front out path */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	/* Unmute Headphone out path */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	/* Unmute Mono out path */
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	{ }
+};
+
+/* Initialisation sequence for ALC260 as configured in Fujitsu S702x
+ * laptops.  ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD
+ * audio = 0x16, internal speaker = 0x10.
+ */
+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 */
+	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	/* Headphone/Line-out jack connects to Line1 pin; make it an output */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	/* Mic/Line-in jack is connected to mic1 pin, so make it an input */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	/* Ensure all other unused pins are disabled and muted. */
+	{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+
+	/* Disable digital (SPDIF) pins */
+	{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
+	{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+	/* Ensure Line1 pin widget takes its input from the OUT1 sum bus
+	 * when acting as an output.
+	 */
+	{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
+
+	/* Start with output sum widgets muted and their output gains at min */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+	/* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Unmute Line1 pin widget output buffer since it starts as an output.
+	 * If the pin mode is changed by the user the pin mode control will
+	 * take care of enabling the pin's input/output buffers as needed.
+	 * Therefore there's no need to enable the input buffer at this
+	 * stage.
+	 */
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Unmute input buffer of pin widget used for Line-in (no equiv
+	 * mixer ctrl)
+	 */
+	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+	/* Mute capture amp left and right */
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	/* Set ADC connection select to match default mixer setting - line
+	 * in (on mic1 pin)
+	 */
+	{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Do the same for the second ADC: mute capture input amp and
+	 * set ADC connection to line in (on mic1 pin)
+	 */
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Mute all inputs to mixer widget (even unconnected ones) */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+
+	{ }
+};
+
+/* Initialisation sequence for ALC260 as configured in Acer TravelMate and
+ * similar laptops (adapted from Fujitsu 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.
+	 */
+	{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
+	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+	{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
+	/* Internal speaker/Headphone jack is connected to Line-out pin */
+	{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	/* Internal microphone/Mic jack is connected to Mic1 pin */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+	/* Line In jack is connected to Line1 pin */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	/* Some Acers (eg: C20x Tablets) use Mono pin for internal speaker */
+	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	/* Ensure all other unused pins are disabled and muted. */
+	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	/* Disable digital (SPDIF) pins */
+	{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
+	{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+	/* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
+	 * bus when acting as outputs.
+	 */
+	{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
+	{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
+
+	/* Start with output sum widgets muted and their output gains at min */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+	/* Unmute Line-out pin widget amp left and right
+	 * (no equiv mixer ctrl)
+	 */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Unmute mono pin widget amp output (no equiv mixer ctrl) */
+	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Unmute Mic1 and Line1 pin widget input buffers since they start as
+	 * inputs. If the pin mode is changed by the user the pin mode control
+	 * will take care of enabling the pin's input/output buffers as needed.
+	 * Therefore there's no need to enable the input buffer at this
+	 * stage.
+	 */
+	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+	/* Mute capture amp left and right */
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	/* Set ADC connection select to match default mixer setting - mic
+	 * (on mic1 pin)
+	 */
+	{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Do similar with the second ADC: mute capture input amp and
+	 * set ADC connection to mic to match ALSA's default state.
+	 */
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Mute all inputs to mixer widget (even unconnected ones) */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+
+	{ }
+};
+
+/* Initialisation sequence for Maxdata Favorit 100XS
+ * (adapted from Acer 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.
+	 */
+	{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
+	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+	{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
+	/* Line/Mic input jack is connected to Mic1 pin */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+	/* Ensure all other unused pins are disabled and muted. */
+	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	/* Disable digital (SPDIF) pins */
+	{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
+	{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+	/* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
+	 * bus when acting as outputs.
+	 */
+	{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
+	{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
+
+	/* Start with output sum widgets muted and their output gains at min */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+	/* Unmute Line-out pin widget amp left and right
+	 * (no equiv mixer ctrl)
+	 */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Unmute Mic1 and Line1 pin widget input buffers since they start as
+	 * inputs. If the pin mode is changed by the user the pin mode control
+	 * will take care of enabling the pin's input/output buffers as needed.
+	 * Therefore there's no need to enable the input buffer at this
+	 * stage.
+	 */
+	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+	/* Mute capture amp left and right */
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	/* Set ADC connection select to match default mixer setting - mic
+	 * (on mic1 pin)
+	 */
+	{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Do similar with the second ADC: mute capture input amp and
+	 * set ADC connection to mic to match ALSA's default state.
+	 */
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Mute all inputs to mixer widget (even unconnected ones) */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+
+	{ }
+};
+
+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},
+	{0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
+	{0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
+	{0x1a, AC_VERB_SET_PROC_COEF, 0x3040},
+	{}
+};
+
+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},
+
+	{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
+	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+	{0x01, AC_VERB_SET_GPIO_DATA, 0x00},
+
+	{0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc260_replacer_672v_automute(struct hda_codec *codec)
+{
+        unsigned int present;
+
+	/* speaker --> GPIO Data 0, hp or spdif --> GPIO data 1 */
+	present = snd_hda_jack_detect(codec, 0x0f);
+	if (present) {
+		snd_hda_codec_write_cache(codec, 0x01, 0,
+					  AC_VERB_SET_GPIO_DATA, 1);
+		snd_hda_codec_write_cache(codec, 0x0f, 0,
+					  AC_VERB_SET_PIN_WIDGET_CONTROL,
+					  PIN_HP);
+	} else {
+		snd_hda_codec_write_cache(codec, 0x01, 0,
+					  AC_VERB_SET_GPIO_DATA, 0);
+		snd_hda_codec_write_cache(codec, 0x0f, 0,
+					  AC_VERB_SET_PIN_WIDGET_CONTROL,
+					  PIN_OUT);
+	}
+}
+
+static void alc260_replacer_672v_unsol_event(struct hda_codec *codec,
+                                       unsigned int res)
+{
+        if ((res >> 26) == ALC_HP_EVENT)
+                alc260_replacer_672v_automute(codec);
+}
+
+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},
+	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+/* Test configuration for debugging, modelled after the ALC880 test
+ * configuration.
+ */
+#ifdef CONFIG_SND_DEBUG
+static const hda_nid_t alc260_test_dac_nids[1] = {
+	0x02,
+};
+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 const struct hda_input_mux alc260_test_capture_sources[2] = {
+	{
+		.num_items = 7,
+		.items = {
+			{ "MIC1 pin", 0x0 },
+			{ "MIC2 pin", 0x1 },
+			{ "LINE1 pin", 0x2 },
+			{ "LINE2 pin", 0x3 },
+			{ "CD pin", 0x4 },
+			{ "LINE-OUT pin", 0x5 },
+			{ "HP-OUT pin", 0x6 },
+		},
+        },
+	{
+		.num_items = 8,
+		.items = {
+			{ "MIC1 pin", 0x0 },
+			{ "MIC2 pin", 0x1 },
+			{ "LINE1 pin", 0x2 },
+			{ "LINE2 pin", 0x3 },
+			{ "CD pin", 0x4 },
+			{ "Mixer", 0x5 },
+			{ "LINE-OUT pin", 0x6 },
+			{ "HP-OUT pin", 0x7 },
+		},
+        },
+};
+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),
+	HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT),
+
+	/* Modes for retasking pin widgets
+	 * Note: the ALC260 doesn't seem to act on requests to enable mic
+         * bias from NIDs 0x0f and 0x10.  The ALC260 datasheet doesn't
+         * mention this restriction.  At this stage it's not clear whether
+         * this behaviour is intentional or is a hardware bug in chip
+         * revisions available at least up until early 2006.  Therefore for
+         * now allow the "HP-OUT" and "LINE-OUT" Mode controls to span all
+         * choices, but if it turns out that the lack of mic bias for these
+         * NIDs is intentional we could change their modes from
+         * ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
+	 */
+	ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT),
+	ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT),
+	ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT),
+	ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT),
+	ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT),
+	ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT),
+
+	/* Loopback mixer controls */
+	HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT),
+	HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT),
+	HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT),
+	HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT),
+	HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT),
+	HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT),
+	HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT),
+
+	/* Controls for GPIO pins, assuming they are configured as outputs */
+	ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
+	ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
+	ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
+	ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
+
+	/* Switches to allow the digital IO pins to be enabled.  The datasheet
+	 * is ambigious as to which NID is which; testing on laptops which
+	 * make this output available should provide clarification.
+	 */
+	ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01),
+	ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01),
+
+	/* A switch allowing EAPD to be enabled.  Some laptops seem to use
+	 * this output to turn on an external amplifier.
+	 */
+	ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
+	ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
+
+	{ } /* end */
+};
+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},
+	{0x01, AC_VERB_SET_GPIO_MASK, 0x0f},
+
+	/* Enable retasking pins as output, initially without power amp */
+	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x0f, 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},
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+	/* Disable digital (SPDIF) pins initially, but users can enable
+	 * them via a mixer switch.  In the case of SPDIF-out, this initverb
+	 * payload also sets the generation to 0, output to be in "consumer"
+	 * PCM format, copyright asserted, no pre-emphasis and no validity
+	 * control.
+	 */
+	{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
+	{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+	/* Ensure mic1, mic2, line1 and line2 pin widgets take input from the
+	 * OUT1 sum bus when acting as an output.
+	 */
+	{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
+	{0x0c, AC_VERB_SET_CONNECT_SEL, 0},
+	{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
+	{0x0e, AC_VERB_SET_CONNECT_SEL, 0},
+
+	/* Start with output sum widgets muted and their output gains at min */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+	/* Unmute retasking pin widget output buffers since the default
+	 * state appears to be output.  As the pin mode is changed by the
+	 * user the pin mode control will take care of enabling the pin's
+	 * input/output buffers as needed.
+	 */
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Also unmute the mono-out pin widget */
+	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* Mute capture amp left and right */
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	/* Set ADC connection select to match default mixer setting (mic1
+	 * pin)
+	 */
+	{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Do the same for the second ADC: mute capture input amp and
+	 * set ADC connection to mic1 pin
+	 */
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Mute all inputs to mixer widget (even unconnected ones) */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+
+	{ }
+};
+#endif
+
+/*
+ * ALC260 configurations
+ */
+static const char * const alc260_models[ALC260_MODEL_LAST] = {
+	[ALC260_BASIC]		= "basic",
+	[ALC260_HP]		= "hp",
+	[ALC260_HP_3013]	= "hp-3013",
+	[ALC260_HP_DC7600]	= "hp-dc7600",
+	[ALC260_FUJITSU_S702X]	= "fujitsu",
+	[ALC260_ACER]		= "acer",
+	[ALC260_WILL]		= "will",
+	[ALC260_REPLACER_672V]	= "replacer",
+	[ALC260_FAVORIT100]	= "favorit100",
+#ifdef CONFIG_SND_DEBUG
+	[ALC260_TEST]		= "test",
+#endif
+	[ALC260_AUTO]		= "auto",
+};
+
+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),
+	SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
+	SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
+	SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_AUTO), /* no quirk */
+	SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
+	SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013),
+	SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600),
+	SND_PCI_QUIRK(0x103c, 0x3013, "HP", ALC260_HP_3013),
+	SND_PCI_QUIRK(0x103c, 0x3014, "HP", ALC260_HP),
+	SND_PCI_QUIRK(0x103c, 0x3015, "HP", ALC260_HP),
+	SND_PCI_QUIRK(0x103c, 0x3016, "HP", ALC260_HP),
+	SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC),
+	SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC),
+	SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC),
+	SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X),
+	SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC),
+	SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_REPLACER_672V),
+	SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_WILL),
+	{}
+};
+
+static const struct alc_config_preset alc260_presets[] = {
+	[ALC260_BASIC] = {
+		.mixers = { alc260_base_output_mixer,
+			    alc260_input_mixer },
+		.init_verbs = { alc260_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
+		.dac_nids = alc260_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
+		.adc_nids = alc260_dual_adc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc260_modes),
+		.channel_mode = alc260_modes,
+		.input_mux = &alc260_capture_source,
+	},
+	[ALC260_HP] = {
+		.mixers = { alc260_hp_output_mixer,
+			    alc260_input_mixer },
+		.init_verbs = { alc260_init_verbs,
+				alc260_hp_unsol_verbs },
+		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
+		.dac_nids = alc260_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
+		.adc_nids = alc260_adc_nids_alt,
+		.num_channel_mode = ARRAY_SIZE(alc260_modes),
+		.channel_mode = alc260_modes,
+		.input_mux = &alc260_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc260_hp_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC260_HP_DC7600] = {
+		.mixers = { alc260_hp_dc7600_mixer,
+			    alc260_input_mixer },
+		.init_verbs = { alc260_init_verbs,
+				alc260_hp_dc7600_verbs },
+		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
+		.dac_nids = alc260_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
+		.adc_nids = alc260_adc_nids_alt,
+		.num_channel_mode = ARRAY_SIZE(alc260_modes),
+		.channel_mode = alc260_modes,
+		.input_mux = &alc260_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc260_hp_3012_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC260_HP_3013] = {
+		.mixers = { alc260_hp_3013_mixer,
+			    alc260_input_mixer },
+		.init_verbs = { alc260_hp_3013_init_verbs,
+				alc260_hp_3013_unsol_verbs },
+		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
+		.dac_nids = alc260_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
+		.adc_nids = alc260_adc_nids_alt,
+		.num_channel_mode = ARRAY_SIZE(alc260_modes),
+		.channel_mode = alc260_modes,
+		.input_mux = &alc260_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc260_hp_3013_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC260_FUJITSU_S702X] = {
+		.mixers = { alc260_fujitsu_mixer },
+		.init_verbs = { alc260_fujitsu_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
+		.dac_nids = alc260_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
+		.adc_nids = alc260_dual_adc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc260_modes),
+		.channel_mode = alc260_modes,
+		.num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources),
+		.input_mux = alc260_fujitsu_capture_sources,
+	},
+	[ALC260_ACER] = {
+		.mixers = { alc260_acer_mixer },
+		.init_verbs = { alc260_acer_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
+		.dac_nids = alc260_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
+		.adc_nids = alc260_dual_adc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc260_modes),
+		.channel_mode = alc260_modes,
+		.num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources),
+		.input_mux = alc260_acer_capture_sources,
+	},
+	[ALC260_FAVORIT100] = {
+		.mixers = { alc260_favorit100_mixer },
+		.init_verbs = { alc260_favorit100_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
+		.dac_nids = alc260_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
+		.adc_nids = alc260_dual_adc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc260_modes),
+		.channel_mode = alc260_modes,
+		.num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources),
+		.input_mux = alc260_favorit100_capture_sources,
+	},
+	[ALC260_WILL] = {
+		.mixers = { alc260_will_mixer },
+		.init_verbs = { alc260_init_verbs, alc260_will_verbs },
+		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
+		.dac_nids = alc260_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
+		.adc_nids = alc260_adc_nids,
+		.dig_out_nid = ALC260_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc260_modes),
+		.channel_mode = alc260_modes,
+		.input_mux = &alc260_capture_source,
+	},
+	[ALC260_REPLACER_672V] = {
+		.mixers = { alc260_replacer_672v_mixer },
+		.init_verbs = { alc260_init_verbs, alc260_replacer_672v_verbs },
+		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
+		.dac_nids = alc260_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
+		.adc_nids = alc260_adc_nids,
+		.dig_out_nid = ALC260_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc260_modes),
+		.channel_mode = alc260_modes,
+		.input_mux = &alc260_capture_source,
+		.unsol_event = alc260_replacer_672v_unsol_event,
+		.init_hook = alc260_replacer_672v_automute,
+	},
+#ifdef CONFIG_SND_DEBUG
+	[ALC260_TEST] = {
+		.mixers = { alc260_test_mixer },
+		.init_verbs = { alc260_test_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc260_test_dac_nids),
+		.dac_nids = alc260_test_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids),
+		.adc_nids = alc260_test_adc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc260_modes),
+		.channel_mode = alc260_modes,
+		.num_mux_defs = ARRAY_SIZE(alc260_test_capture_sources),
+		.input_mux = alc260_test_capture_sources,
+	},
+#endif
+};
+

sound/pci/hda/alc262_quirks.c

@@ -0,0 +1,1353 @@
+/*
+ * ALC262 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC262 models */
+enum {
+	ALC262_AUTO,
+	ALC262_BASIC,
+	ALC262_HIPPO,
+	ALC262_HIPPO_1,
+	ALC262_FUJITSU,
+	ALC262_HP_BPC,
+	ALC262_HP_BPC_D7000_WL,
+	ALC262_HP_BPC_D7000_WF,
+	ALC262_HP_TC_T5735,
+	ALC262_HP_RP5700,
+	ALC262_BENQ_ED8,
+	ALC262_SONY_ASSAMD,
+	ALC262_BENQ_T31,
+	ALC262_ULTRA,
+	ALC262_LENOVO_3000,
+	ALC262_NEC,
+	ALC262_TOSHIBA_S06,
+	ALC262_TOSHIBA_RX1,
+	ALC262_TYAN,
+	ALC262_MODEL_LAST /* last tag */
+};
+
+#define ALC262_DIGOUT_NID	ALC880_DIGOUT_NID
+#define ALC262_DIGIN_NID	ALC880_DIGIN_NID
+
+#define alc262_dac_nids		alc260_dac_nids
+#define alc262_adc_nids		alc882_adc_nids
+#define alc262_adc_nids_alt	alc882_adc_nids_alt
+#define alc262_capsrc_nids	alc882_capsrc_nids
+#define alc262_capsrc_nids_alt	alc882_capsrc_nids_alt
+
+#define alc262_modes		alc260_modes
+#define alc262_capture_source	alc882_capture_source
+
+static const hda_nid_t alc262_dmic_adc_nids[1] = {
+	/* ADC0 */
+	0x09
+};
+
+static const hda_nid_t alc262_dmic_capsrc_nids[1] = { 0x22 };
+
+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),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
+	{ } /* end */
+};
+
+/* update HP, line and mono-out pins according to the master switch */
+#define alc262_hp_master_update		alc260_hp_master_update
+
+static void alc262_hp_bpc_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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
+#define alc262_hp_master_sw_put		alc260_hp_master_sw_put
+
+#define ALC262_HP_MASTER_SWITCH					\
+	{							\
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,		\
+		.name = "Master Playback Switch",		\
+		.info = snd_ctl_boolean_mono_info,		\
+		.get = alc262_hp_master_sw_get,			\
+		.put = alc262_hp_master_sw_put,			\
+	}, \
+	{							\
+		.iface = NID_MAPPING,				\
+		.name = "Master Playback Switch",		\
+		.private_value = 0x15 | (0x16 << 8) | (0x1b << 16),	\
+	}
+
+
+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),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
+			      HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
+			    HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT),
+	HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
+			      HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
+			    HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x1a, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	{ } /* end */
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc262_hp_t5735_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_PIN;
+}
+
+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),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	{ } /* end */
+};
+
+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},
+
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{ }
+};
+
+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),
+	HDA_CODEC_MUTE("Speaker Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	{ } /* end */
+};
+
+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},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
+	{}
+};
+
+static const struct hda_input_mux alc262_hp_rp5700_capture_source = {
+	.num_items = 1,
+	.items = {
+		{ "Line", 0x1 },
+	},
+};
+
+/* bind hp and internal speaker mute (with plug check) as master switch */
+#define alc262_hippo_master_update	alc262_hp_master_update
+#define alc262_hippo_master_sw_get	alc262_hp_master_sw_get
+#define alc262_hippo_master_sw_put	alc262_hp_master_sw_put
+
+#define ALC262_HIPPO_MASTER_SWITCH				\
+	{							\
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,		\
+		.name = "Master Playback Switch",		\
+		.info = snd_ctl_boolean_mono_info,		\
+		.get = alc262_hippo_master_sw_get,		\
+		.put = alc262_hippo_master_sw_put,		\
+	},							\
+	{							\
+		.iface = NID_MAPPING,				\
+		.name = "Master Playback Switch",		\
+		.subdevice = SUBDEV_HP(0) | (SUBDEV_LINE(0) << 8) | \
+			     (SUBDEV_SPEAKER(0) << 16), \
+	}
+
+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),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	{ } /* end */
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+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)
+{
+	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 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),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE("Aux Playback Switch", 0x0b, 0x06, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	{ } /* end */
+};
+
+static const struct hda_verb alc262_tyan_verbs[] = {
+	/* Headphone automute */
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* P11 AUX_IN, white 4-pin connector */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, 0xe1},
+	{0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, 0x93},
+	{0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0x19},
+
+	{}
+};
+
+/* unsolicited event for HP jack sensing */
+static void alc262_tyan_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x1b;
+	spec->autocfg.speaker_pins[0] = 0x15;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+
+#define alc262_capture_mixer		alc882_capture_mixer
+#define alc262_capture_alt_mixer	alc882_capture_alt_mixer
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static const struct hda_verb alc262_init_verbs[] = {
+	/*
+	 * Unmute ADC0-2 and set the default input to mic-in
+	 */
+	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x09, 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
+	 * Note: PASD motherboards uses the Line In 2 as the input for
+	 * front panel mic (mic 2)
+	 */
+	/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+	/*
+	 * Set up output mixers (0x0c - 0x0e)
+	 */
+	/* set vol=0 to output mixers */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	/* set up input amps for analog loopback */
+	/* Amp Indices: DAC = 0, mixer = 1 */
+	{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)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+
+	/* FIXME: use matrix-type input source selection */
+	/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+	/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+	/* Input mixer2 */
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+	/* Input mixer3 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+
+	{ }
+};
+
+static const struct hda_verb alc262_eapd_verbs[] = {
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{ }
+};
+
+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},
+
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{}
+};
+
+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
+
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{}
+};
+
+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),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+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},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x22, AC_VERB_SET_CONNECT_SEL, 0x09},
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+static void alc262_toshiba_s06_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->ext_mic_pin = 0x18;
+	spec->int_mic_pin = 0x12;
+	spec->auto_mic = 1;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
+/*
+ * nec model
+ *  0x15 = headphone
+ *  0x16 = internal speaker
+ *  0x18 = external mic
+ */
+
+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),
+
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	{ } /* end */
+};
+
+static const struct hda_verb alc262_nec_verbs[] = {
+	/* Unmute Speaker */
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* Headphone */
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+	/* External mic to headphone */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	/* External mic to speaker */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{}
+};
+
+/*
+ * fujitsu model
+ *  0x14 = headphone/spdif-out, 0x15 = internal speaker,
+ *  0x1b = port replicator headphone out
+ */
+
+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},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{}
+};
+
+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 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 const struct hda_input_mux alc262_fujitsu_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Internal Mic", 0x1 },
+		{ "CD", 0x4 },
+	},
+};
+
+static const struct hda_input_mux alc262_HP_capture_source = {
+	.num_items = 5,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x1 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+		{ "AUX IN", 0x6 },
+	},
+};
+
+static const struct hda_input_mux alc262_HP_D7000_capture_source = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x2 },
+		{ "Line", 0x1 },
+		{ "CD", 0x4 },
+	},
+};
+
+static void alc262_fujitsu_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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 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),
+		HDA_COMPOSE_AMP_VAL(0x0d, 3, 0, HDA_OUTPUT),
+		0
+	},
+};
+
+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_NID_FLAG | 0x14,
+		.info = snd_ctl_boolean_mono_info,
+		.get = alc262_hp_master_sw_get,
+		.put = alc262_hp_master_sw_put,
+	},
+	{
+		.iface = NID_MAPPING,
+		.name = "Master Playback Switch",
+		.private_value = 0x1b,
+	},
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+static void alc262_lenovo_3000_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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 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_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),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	{ } /* end */
+};
+
+/* additional init verbs for Benq laptops */
+static const struct hda_verb alc262_EAPD_verbs[] = {
+	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+	{0x20, AC_VERB_SET_PROC_COEF,  0x3070},
+	{}
+};
+
+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},
+
+	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+	{0x20, AC_VERB_SET_PROC_COEF,  0x3050},
+	{}
+};
+
+/* Samsung Q1 Ultra Vista model setup */
+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),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Headphone Mic Boost Volume", 0x15, 0, HDA_INPUT),
+	{ } /* end */
+};
+
+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)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	/* speaker */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* HP */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	/* internal mic */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* ADC, choose mic */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(8)},
+	{}
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc262_ultra_automute(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	unsigned int mute;
+
+	mute = 0;
+	/* auto-mute only when HP is used as HP */
+	if (!spec->cur_mux[0]) {
+		spec->jack_present = snd_hda_jack_detect(codec, 0x15);
+		if (spec->jack_present)
+			mute = HDA_AMP_MUTE;
+	}
+	/* mute/unmute internal speaker */
+	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, mute);
+	/* mute/unmute HP */
+	snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, mute ? 0 : HDA_AMP_MUTE);
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc262_ultra_unsol_event(struct hda_codec *codec,
+				       unsigned int res)
+{
+	if ((res >> 26) != ALC_HP_EVENT)
+		return;
+	alc262_ultra_automute(codec);
+}
+
+static const struct hda_input_mux alc262_ultra_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x1 },
+		{ "Headphone", 0x7 },
+	},
+};
+
+static int alc262_ultra_mux_enum_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 ret;
+
+	ret = alc_mux_enum_put(kcontrol, ucontrol);
+	if (!ret)
+		return 0;
+	/* reprogram the HP pin as mic or HP according to the input source */
+	snd_hda_codec_write_cache(codec, 0x15, 0,
+				  AC_VERB_SET_PIN_WIDGET_CONTROL,
+				  spec->cur_mux[0] ? PIN_VREF80 : PIN_HP);
+	alc262_ultra_automute(codec); /* mute/unmute HP */
+	return ret;
+}
+
+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),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Capture Source",
+		.info = alc_mux_enum_info,
+		.get = alc_mux_enum_get,
+		.put = alc262_ultra_mux_enum_put,
+	},
+	{
+		.iface = NID_MAPPING,
+		.name = "Capture Source",
+		.private_value = 0x15,
+	},
+	{ } /* end */
+};
+
+static const struct hda_verb alc262_HP_BPC_init_verbs[] = {
+	/*
+	 * Unmute ADC0-2 and set the default input to mic-in
+	 */
+	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x09, 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
+	 * Note: PASD motherboards uses the Line In 2 as the input for
+	 * front panel mic (mic 2)
+	 */
+	/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+
+	/*
+	 * Set up output mixers (0x0c - 0x0e)
+	 */
+	/* set vol=0 to output mixers */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+	/* set up input amps for analog loopback */
+	/* Amp Indices: DAC = 0, mixer = 1 */
+	{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)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+        {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+        {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+
+
+	/* FIXME: use matrix-type input source selection */
+	/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 0b, 12 */
+	/* Input mixer1: only unmute Mic */
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
+	/* Input mixer2 */
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
+	/* Input mixer3 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
+
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+
+	{ }
+};
+
+static const struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = {
+	/*
+	 * Unmute ADC0-2 and set the default input to mic-in
+	 */
+	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x09, 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
+	 * Note: PASD motherboards uses the Line In 2 as the input for front
+	 * panel mic (mic 2)
+	 */
+	/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+	/*
+	 * Set up output mixers (0x0c - 0x0e)
+	 */
+	/* set vol=0 to output mixers */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+	/* set up input amps for analog loopback */
+	/* Amp Indices: DAC = 0, mixer = 1 */
+	{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)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },	/* HP */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },	/* Mono */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },	/* rear MIC */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },	/* Line in */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },	/* Front MIC */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },	/* Line out */
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },	/* CD in */
+
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+
+	/* {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 }, */
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
+	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+
+	/* FIXME: use matrix-type input source selection */
+	/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+	/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, /*rear MIC*/
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, /*Line in*/
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, /*F MIC*/
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, /*Front*/
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, /*CD*/
+        /* {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))},  */
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))}, /*HP*/
+	/* Input mixer2 */
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+        /* {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))},
+	/* Input mixer3 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+        /* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))},
+
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+
+	{ }
+};
+
+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 },
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x01},
+
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },	/* MIC jack */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },	/* Front MIC */
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) },
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) },
+
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },	/* HP  jack */
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+/*
+ * configuration and preset
+ */
+static const char * const alc262_models[ALC262_MODEL_LAST] = {
+	[ALC262_BASIC]		= "basic",
+	[ALC262_HIPPO]		= "hippo",
+	[ALC262_HIPPO_1]	= "hippo_1",
+	[ALC262_FUJITSU]	= "fujitsu",
+	[ALC262_HP_BPC]		= "hp-bpc",
+	[ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000",
+	[ALC262_HP_TC_T5735]	= "hp-tc-t5735",
+	[ALC262_HP_RP5700]	= "hp-rp5700",
+	[ALC262_BENQ_ED8]	= "benq",
+	[ALC262_BENQ_T31]	= "benq-t31",
+	[ALC262_SONY_ASSAMD]	= "sony-assamd",
+	[ALC262_TOSHIBA_S06]	= "toshiba-s06",
+	[ALC262_TOSHIBA_RX1]	= "toshiba-rx1",
+	[ALC262_ULTRA]		= "ultra",
+	[ALC262_LENOVO_3000]	= "lenovo-3000",
+	[ALC262_NEC]		= "nec",
+	[ALC262_TYAN]		= "tyan",
+	[ALC262_AUTO]		= "auto",
+};
+
+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",
+			   ALC262_HP_BPC),
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series",
+			   ALC262_HP_BPC),
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series",
+			   ALC262_HP_BPC),
+	SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200",
+			   ALC262_AUTO),
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
+			   ALC262_HP_BPC),
+	SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
+	SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF),
+	SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL),
+	SND_PCI_QUIRK(0x103c, 0x2803, "HP D7000", ALC262_HP_BPC_D7000_WF),
+	SND_PCI_QUIRK(0x103c, 0x2804, "HP D7000", ALC262_HP_BPC_D7000_WL),
+	SND_PCI_QUIRK(0x103c, 0x2805, "HP D7000", ALC262_HP_BPC_D7000_WF),
+	SND_PCI_QUIRK(0x103c, 0x2806, "HP D7000", ALC262_HP_BPC_D7000_WL),
+	SND_PCI_QUIRK(0x103c, 0x2807, "HP D7000", ALC262_HP_BPC_D7000_WF),
+	SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC),
+	SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC),
+	SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC),
+	SND_PCI_QUIRK(0x103c, 0x302f, "HP Thin Client T5735",
+		      ALC262_HP_TC_T5735),
+	SND_PCI_QUIRK(0x103c, 0x2817, "HP RP5700", ALC262_HP_RP5700),
+	SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD),
+	SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO),
+	SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
+	SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
+	SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06),
+	SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO),
+	SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO),
+#if 0 /* disable the quirk since model=auto works better in recent versions */
+	SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
+			   ALC262_SONY_ASSAMD),
+#endif
+	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
+		      ALC262_TOSHIBA_RX1),
+	SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06),
+	SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
+	SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU),
+	SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN),
+	SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1",
+			   ALC262_ULTRA),
+	SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO),
+	SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000),
+	SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8),
+	SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31),
+	SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1),
+	{}
+};
+
+static const struct alc_config_preset alc262_presets[] = {
+	[ALC262_BASIC] = {
+		.mixers = { alc262_base_mixer },
+		.init_verbs = { alc262_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_capture_source,
+	},
+	[ALC262_HIPPO] = {
+		.mixers = { alc262_hippo_mixer },
+		.init_verbs = { alc262_init_verbs, alc_hp15_unsol_verbs},
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x03,
+		.dig_out_nid = ALC262_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_hippo_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC262_HIPPO_1] = {
+		.mixers = { alc262_hippo1_mixer },
+		.init_verbs = { alc262_init_verbs, alc262_hippo1_unsol_verbs},
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x02,
+		.dig_out_nid = ALC262_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_hippo1_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC262_FUJITSU] = {
+		.mixers = { alc262_fujitsu_mixer },
+		.init_verbs = { alc262_init_verbs, alc262_EAPD_verbs,
+				alc262_fujitsu_unsol_verbs },
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x03,
+		.dig_out_nid = ALC262_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_fujitsu_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_fujitsu_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC262_HP_BPC] = {
+		.mixers = { alc262_HP_BPC_mixer },
+		.init_verbs = { alc262_HP_BPC_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_HP_capture_source,
+		.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 },
+		.init_verbs = { alc262_HP_BPC_WildWest_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_HP_D7000_capture_source,
+		.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,
+			    alc262_HP_BPC_WildWest_option_mixer },
+		.init_verbs = { alc262_HP_BPC_WildWest_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_HP_D7000_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_hp_wildwest_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC262_HP_TC_T5735] = {
+		.mixers = { alc262_hp_t5735_mixer },
+		.init_verbs = { alc262_init_verbs, alc262_hp_t5735_verbs },
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_hp_t5735_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC262_HP_RP5700] = {
+		.mixers = { alc262_hp_rp5700_mixer },
+		.init_verbs = { alc262_init_verbs, alc262_hp_rp5700_verbs },
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_hp_rp5700_capture_source,
+        },
+	[ALC262_BENQ_ED8] = {
+		.mixers = { alc262_base_mixer },
+		.init_verbs = { alc262_init_verbs, alc262_EAPD_verbs },
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_capture_source,
+	},
+	[ALC262_SONY_ASSAMD] = {
+		.mixers = { alc262_sony_mixer },
+		.init_verbs = { alc262_init_verbs, alc262_sony_unsol_verbs},
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x02,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_hippo_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC262_BENQ_T31] = {
+		.mixers = { alc262_benq_t31_mixer },
+		.init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs,
+				alc_hp15_unsol_verbs },
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_hippo_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC262_ULTRA] = {
+		.mixers = { alc262_ultra_mixer },
+		.cap_mixer = alc262_ultra_capture_mixer,
+		.init_verbs = { alc262_ultra_verbs },
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_ultra_capture_source,
+		.adc_nids = alc262_adc_nids, /* ADC0 */
+		.capsrc_nids = alc262_capsrc_nids,
+		.num_adc_nids = 1, /* single ADC */
+		.unsol_event = alc262_ultra_unsol_event,
+		.init_hook = alc262_ultra_automute,
+	},
+	[ALC262_LENOVO_3000] = {
+		.mixers = { alc262_lenovo_3000_mixer },
+		.init_verbs = { alc262_init_verbs, alc262_EAPD_verbs,
+				alc262_lenovo_3000_unsol_verbs,
+				alc262_lenovo_3000_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x03,
+		.dig_out_nid = ALC262_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_fujitsu_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_lenovo_3000_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC262_NEC] = {
+		.mixers = { alc262_nec_mixer },
+		.init_verbs = { alc262_nec_verbs },
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_capture_source,
+	},
+	[ALC262_TOSHIBA_S06] = {
+		.mixers = { alc262_toshiba_s06_mixer },
+		.init_verbs = { alc262_init_verbs, alc262_toshiba_s06_verbs,
+							alc262_eapd_verbs },
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.capsrc_nids = alc262_dmic_capsrc_nids,
+		.dac_nids = alc262_dac_nids,
+		.adc_nids = alc262_dmic_adc_nids, /* ADC0 */
+		.num_adc_nids = 1, /* single ADC */
+		.dig_out_nid = ALC262_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_toshiba_s06_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC262_TOSHIBA_RX1] = {
+		.mixers = { alc262_toshiba_rx1_mixer },
+		.init_verbs = { alc262_init_verbs, alc262_toshiba_rx1_unsol_verbs },
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_hippo_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC262_TYAN] = {
+		.mixers = { alc262_tyan_mixer },
+		.init_verbs = { alc262_init_verbs, alc262_tyan_verbs},
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x02,
+		.dig_out_nid = ALC262_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_tyan_setup,
+		.init_hook = alc_hp_automute,
+	},
+};
+

sound/pci/hda/alc268_quirks.c

@@ -0,0 +1,636 @@
+/*
+ * ALC267/ALC268 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC268 models */
+enum {
+	ALC268_AUTO,
+	ALC267_QUANTA_IL1,
+	ALC268_3ST,
+	ALC268_TOSHIBA,
+	ALC268_ACER,
+	ALC268_ACER_DMIC,
+	ALC268_ACER_ASPIRE_ONE,
+	ALC268_DELL,
+	ALC268_ZEPTO,
+#ifdef CONFIG_SND_DEBUG
+	ALC268_TEST,
+#endif
+	ALC268_MODEL_LAST /* last tag */
+};
+
+/*
+ *  ALC268 channel source setting (2 channel)
+ */
+#define ALC268_DIGOUT_NID	ALC880_DIGOUT_NID
+#define alc268_modes		alc260_modes
+
+static const hda_nid_t alc268_dac_nids[2] = {
+	/* front, hp */
+	0x02, 0x03
+};
+
+static const hda_nid_t alc268_adc_nids[2] = {
+	/* ADC0-1 */
+	0x08, 0x07
+};
+
+static const hda_nid_t alc268_adc_nids_alt[1] = {
+	/* ADC0 */
+	0x08
+};
+
+static const hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 };
+
+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),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
+	{ }
+};
+
+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),
+	ALC262_HIPPO_MASTER_SWITCH,
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
+	{ }
+};
+
+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 const struct hda_verb alc268_toshiba_verbs[] = {
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{ } /* end */
+};
+
+/* Acer specific */
+/* bind volumes of both NID 0x02 and 0x03 */
+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),
+		HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
+		0
+	},
+};
+
+static void alc268_acer_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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
+
+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_NID_FLAG | 0x15,
+		.info = snd_ctl_boolean_mono_info,
+		.get = alc268_acer_master_sw_get,
+		.put = alc268_acer_master_sw_put,
+	},
+	HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x18, 0, HDA_INPUT),
+	{ }
+};
+
+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_NID_FLAG | 0x14,
+		.info = snd_ctl_boolean_mono_info,
+		.get = alc268_acer_master_sw_get,
+		.put = alc268_acer_master_sw_put,
+	},
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
+	{ }
+};
+
+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_NID_FLAG | 0x14,
+		.info = snd_ctl_boolean_mono_info,
+		.get = alc268_acer_master_sw_get,
+		.put = alc268_acer_master_sw_put,
+	},
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
+	{ }
+};
+
+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, ALC_HP_EVENT | AC_USRSP_EN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x23, AC_VERB_SET_CONNECT_SEL, 0x06},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, 0xa017},
+	{ }
+};
+
+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},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{ }
+};
+
+/* unsolicited event for HP jack sensing */
+#define alc268_toshiba_setup		alc262_hippo_setup
+
+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 = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
+	spec->ext_mic_pin = 0x18;
+	spec->int_mic_pin = 0x12;
+	spec->auto_mic = 1;
+}
+
+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),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	{ }
+};
+
+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, ALC_HP_EVENT | AC_USRSP_EN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
+	{ }
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc268_dell_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->ext_mic_pin = 0x18;
+	spec->int_mic_pin = 0x19;
+	spec->auto_mic = 1;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
+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),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Capture Volume", 0x23, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Mic Capture Switch", 0x23, 2, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	{ }
+};
+
+static const struct hda_verb alc267_quanta_il1_verbs[] = {
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
+	{ }
+};
+
+static void alc267_quanta_il1_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->ext_mic_pin = 0x18;
+	spec->int_mic_pin = 0x19;
+	spec->auto_mic = 1;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+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},
+
+	/*
+	 * Set up output mixers (0x0c - 0x0e)
+	 */
+	/* set vol=0 to output mixers */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+	/* set PCBEEP vol = 0, mute connections */
+	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+	/* Unmute Selector 23h,24h and set the default input to mic-in */
+
+	{0x23, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x24, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	{ }
+};
+
+/* only for model=test */
+#ifdef CONFIG_SND_DEBUG
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+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},
+
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{ }
+};
+#endif /* CONFIG_SND_DEBUG */
+
+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 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 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),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT),
+	_DEFINE_CAPSRC(2),
+	{ } /* end */
+};
+
+static const struct hda_input_mux alc268_capture_source = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x1 },
+		{ "Line", 0x2 },
+		{ "CD", 0x3 },
+	},
+};
+
+static const struct hda_input_mux alc268_acer_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Internal Mic", 0x1 },
+		{ "Line", 0x2 },
+	},
+};
+
+static const struct hda_input_mux alc268_acer_dmic_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Internal Mic", 0x6 },
+		{ "Line", 0x2 },
+	},
+};
+
+#ifdef CONFIG_SND_DEBUG
+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),
+	HDA_BIND_MUTE_MONO("Mono sum Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE("LINE-OUT sum Playback Switch", 0x0f, 2, HDA_INPUT),
+	HDA_BIND_MUTE("HP-OUT sum Playback Switch", 0x10, 2, HDA_INPUT),
+	HDA_BIND_MUTE("LINE-OUT Playback Switch", 0x14, 2, HDA_OUTPUT),
+	HDA_BIND_MUTE("HP-OUT Playback Switch", 0x15, 2, HDA_OUTPUT),
+	HDA_BIND_MUTE("Mono Playback Switch", 0x16, 2, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("MIC1 Capture Volume", 0x18, 0x0, HDA_INPUT),
+	HDA_BIND_MUTE("MIC1 Capture Switch", 0x18, 2, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("MIC2 Capture Volume", 0x19, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("LINE1 Capture Volume", 0x1a, 0x0, HDA_INPUT),
+	HDA_BIND_MUTE("LINE1 Capture Switch", 0x1a, 2, HDA_OUTPUT),
+	/* The below appears problematic on some hardwares */
+	/*HDA_CODEC_VOLUME("PCBEEP Playback Volume", 0x1d, 0x0, HDA_INPUT),*/
+	HDA_CODEC_VOLUME("PCM-IN1 Capture Volume", 0x23, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("PCM-IN1 Capture Switch", 0x23, 2, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("PCM-IN2 Capture Volume", 0x24, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("PCM-IN2 Capture Switch", 0x24, 2, HDA_OUTPUT),
+
+	/* Modes for retasking pin widgets */
+	ALC_PIN_MODE("LINE-OUT pin mode", 0x14, ALC_PIN_DIR_INOUT),
+	ALC_PIN_MODE("HP-OUT pin mode", 0x15, ALC_PIN_DIR_INOUT),
+	ALC_PIN_MODE("MIC1 pin mode", 0x18, ALC_PIN_DIR_INOUT),
+	ALC_PIN_MODE("LINE1 pin mode", 0x1a, ALC_PIN_DIR_INOUT),
+
+	/* Controls for GPIO pins, assuming they are configured as outputs */
+	ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
+	ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
+	ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
+	ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
+
+	/* Switches to allow the digital SPDIF output pin to be enabled.
+	 * The ALC268 does not have an SPDIF input.
+	 */
+	ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x06, 0x01),
+
+	/* A switch allowing EAPD to be enabled.  Some laptops seem to use
+	 * this output to turn on an external amplifier.
+	 */
+	ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
+	ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
+
+	{ } /* end */
+};
+#endif
+
+/*
+ * configuration and preset
+ */
+static const char * const alc268_models[ALC268_MODEL_LAST] = {
+	[ALC267_QUANTA_IL1]	= "quanta-il1",
+	[ALC268_3ST]		= "3stack",
+	[ALC268_TOSHIBA]	= "toshiba",
+	[ALC268_ACER]		= "acer",
+	[ALC268_ACER_DMIC]	= "acer-dmic",
+	[ALC268_ACER_ASPIRE_ONE]	= "acer-aspire",
+	[ALC268_DELL]		= "dell",
+	[ALC268_ZEPTO]		= "zepto",
+#ifdef CONFIG_SND_DEBUG
+	[ALC268_TEST]		= "test",
+#endif
+	[ALC268_AUTO]		= "auto",
+};
+
+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),
+	SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER),
+	SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER),
+	SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
+						ALC268_ACER_ASPIRE_ONE),
+	SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
+	SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron 910", ALC268_AUTO),
+	SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0,
+			"Dell Inspiron Mini9/Vostro A90", ALC268_DELL),
+	/* almost compatible with toshiba but with optional digital outs;
+	 * auto-probing seems working fine
+	 */
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP TX25xx series",
+			   ALC268_AUTO),
+	SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
+	SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
+	SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
+	SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
+	{}
+};
+
+/* Toshiba laptops have no unique PCI SSID but only codec SSID */
+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",
+			   ALC268_TOSHIBA),
+	{}
+};
+
+static const struct alc_config_preset alc268_presets[] = {
+	[ALC267_QUANTA_IL1] = {
+		.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
+			    alc268_capture_nosrc_mixer },
+		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+				alc267_quanta_il1_verbs },
+		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
+		.dac_nids = alc268_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+		.adc_nids = alc268_adc_nids_alt,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc268_modes),
+		.channel_mode = alc268_modes,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc267_quanta_il1_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC268_3ST] = {
+		.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
+			    alc268_beep_mixer },
+		.init_verbs = { alc268_base_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
+		.dac_nids = alc268_dac_nids,
+                .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+                .adc_nids = alc268_adc_nids_alt,
+		.capsrc_nids = alc268_capsrc_nids,
+		.hp_nid = 0x03,
+		.dig_out_nid = ALC268_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc268_modes),
+		.channel_mode = alc268_modes,
+		.input_mux = &alc268_capture_source,
+	},
+	[ALC268_TOSHIBA] = {
+		.mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer,
+			    alc268_beep_mixer },
+		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+				alc268_toshiba_verbs },
+		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
+		.dac_nids = alc268_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+		.adc_nids = alc268_adc_nids_alt,
+		.capsrc_nids = alc268_capsrc_nids,
+		.hp_nid = 0x03,
+		.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 = alc_inithook,
+	},
+	[ALC268_ACER] = {
+		.mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
+			    alc268_beep_mixer },
+		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+				alc268_acer_verbs },
+		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
+		.dac_nids = alc268_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+		.adc_nids = alc268_adc_nids_alt,
+		.capsrc_nids = alc268_capsrc_nids,
+		.hp_nid = 0x02,
+		.num_channel_mode = ARRAY_SIZE(alc268_modes),
+		.channel_mode = alc268_modes,
+		.input_mux = &alc268_acer_capture_source,
+		.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,
+			    alc268_beep_mixer },
+		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+				alc268_acer_verbs },
+		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
+		.dac_nids = alc268_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+		.adc_nids = alc268_adc_nids_alt,
+		.capsrc_nids = alc268_capsrc_nids,
+		.hp_nid = 0x02,
+		.num_channel_mode = ARRAY_SIZE(alc268_modes),
+		.channel_mode = alc268_modes,
+		.input_mux = &alc268_acer_dmic_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc268_acer_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC268_ACER_ASPIRE_ONE] = {
+		.mixers = { alc268_acer_aspire_one_mixer,
+			    alc268_beep_mixer,
+			    alc268_capture_nosrc_mixer },
+		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+				alc268_acer_aspire_one_verbs },
+		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
+		.dac_nids = alc268_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+		.adc_nids = alc268_adc_nids_alt,
+		.capsrc_nids = alc268_capsrc_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc268_modes),
+		.channel_mode = alc268_modes,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc268_acer_lc_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC268_DELL] = {
+		.mixers = { alc268_dell_mixer, alc268_beep_mixer,
+			    alc268_capture_nosrc_mixer },
+		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+				alc268_dell_verbs },
+		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
+		.dac_nids = alc268_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+		.adc_nids = alc268_adc_nids_alt,
+		.capsrc_nids = alc268_capsrc_nids,
+		.hp_nid = 0x02,
+		.num_channel_mode = ARRAY_SIZE(alc268_modes),
+		.channel_mode = alc268_modes,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc268_dell_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC268_ZEPTO] = {
+		.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
+			    alc268_beep_mixer },
+		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+				alc268_toshiba_verbs },
+		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
+		.dac_nids = alc268_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+		.adc_nids = alc268_adc_nids_alt,
+		.capsrc_nids = alc268_capsrc_nids,
+		.hp_nid = 0x03,
+		.dig_out_nid = ALC268_DIGOUT_NID,
+		.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 = alc_inithook,
+	},
+#ifdef CONFIG_SND_DEBUG
+	[ALC268_TEST] = {
+		.mixers = { alc268_test_mixer, alc268_capture_mixer },
+		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+				alc268_volume_init_verbs,
+				alc268_beep_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
+		.dac_nids = alc268_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+		.adc_nids = alc268_adc_nids_alt,
+		.capsrc_nids = alc268_capsrc_nids,
+		.hp_nid = 0x03,
+		.dig_out_nid = ALC268_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc268_modes),
+		.channel_mode = alc268_modes,
+		.input_mux = &alc268_capture_source,
+	},
+#endif
+};
+

sound/pci/hda/alc269_quirks.c

@@ -0,0 +1,681 @@
+/*
+ * ALC269/ALC270/ALC275/ALC276 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC269 models */
+enum {
+	ALC269_AUTO,
+	ALC269_BASIC,
+	ALC269_QUANTA_FL1,
+	ALC269_AMIC,
+	ALC269_DMIC,
+	ALC269VB_AMIC,
+	ALC269VB_DMIC,
+	ALC269_FUJITSU,
+	ALC269_LIFEBOOK,
+	ALC271_ACER,
+	ALC269_MODEL_LAST /* last tag */
+};
+
+/*
+ *  ALC269 channel source setting (2 channel)
+ */
+#define ALC269_DIGOUT_NID	ALC880_DIGOUT_NID
+
+#define alc269_dac_nids		alc260_dac_nids
+
+static const hda_nid_t alc269_adc_nids[1] = {
+	/* ADC1 */
+	0x08,
+};
+
+static const hda_nid_t alc269_capsrc_nids[1] = {
+	0x23,
+};
+
+static const hda_nid_t alc269vb_adc_nids[1] = {
+	/* ADC1 */
+	0x09,
+};
+
+static const hda_nid_t alc269vb_capsrc_nids[1] = {
+	0x22,
+};
+
+#define alc269_modes		alc260_modes
+#define alc269_capture_source	alc880_lg_lw_capture_source
+
+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),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
+	{ } /* end */
+};
+
+static const struct snd_kcontrol_new alc269_quanta_fl1_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,
+		.put = alc268_acer_master_sw_put,
+		.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+	},
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	{ }
+};
+
+static const struct snd_kcontrol_new alc269_lifebook_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,
+		.put = alc268_acer_master_sw_put,
+		.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+	},
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT),
+	HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT),
+	HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x1b, 0, HDA_INPUT),
+	{ }
+};
+
+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),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+	{ } /* end */
+};
+
+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 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),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	{ } /* end */
+};
+
+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 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),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	{ } /* end */
+};
+
+/* FSC amilo */
+#define alc269_fujitsu_mixer	alc269_laptop_mixer
+
+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},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{ }
+};
+
+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},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{ }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
+{
+	alc_hp_automute(codec);
+
+	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)
+{
+	unsigned int present_laptop;
+	unsigned int present_dock;
+
+	present_laptop	= snd_hda_jack_detect(codec, 0x18);
+	present_dock	= snd_hda_jack_detect(codec, 0x1b);
+
+	/* Laptop mic port overrides dock mic port, design decision */
+	if (present_dock)
+		snd_hda_codec_write(codec, 0x23, 0,
+				AC_VERB_SET_CONNECT_SEL, 0x3);
+	if (present_laptop)
+		snd_hda_codec_write(codec, 0x23, 0,
+				AC_VERB_SET_CONNECT_SEL, 0x0);
+	if (!present_dock && !present_laptop)
+		snd_hda_codec_write(codec, 0x23, 0,
+				AC_VERB_SET_CONNECT_SEL, 0x1);
+}
+
+static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec,
+				    unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC_HP_EVENT:
+		alc269_quanta_fl1_speaker_automute(codec);
+		break;
+	case ALC_MIC_EVENT:
+		alc_mic_automute(codec);
+		break;
+	}
+}
+
+static void alc269_lifebook_unsol_event(struct hda_codec *codec,
+					unsigned int res)
+{
+	if ((res >> 26) == ALC_HP_EVENT)
+		alc269_lifebook_speaker_automute(codec);
+	if ((res >> 26) == ALC_MIC_EVENT)
+		alc269_lifebook_mic_autoswitch(codec);
+}
+
+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->int_mic_pin = 0x19;
+	spec->auto_mic = 1;
+}
+
+static void alc269_quanta_fl1_init_hook(struct hda_codec *codec)
+{
+	alc269_quanta_fl1_speaker_automute(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 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 },
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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 },
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x701b | (0x00 << 8))},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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 },
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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 },
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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},
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{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},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x22, AC_VERB_SET_CONNECT_SEL, 6},
+	{ }
+};
+
+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->int_mic_pin = 0x19;
+	spec->auto_mic = 1;
+}
+
+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->int_mic_pin = 0x12;
+	spec->auto_mic = 1;
+}
+
+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->int_mic_pin = 0x19;
+	spec->auto_mic = 1;
+}
+
+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->int_mic_pin = 0x12;
+	spec->auto_mic = 1;
+}
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static const struct hda_verb alc269_init_verbs[] = {
+	/*
+	 * Unmute ADC0 and set the default input to mic-in
+	 */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+	/*
+	 * Set up output mixers (0x02 - 0x03)
+	 */
+	/* set vol=0 to output mixers */
+	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+	/* set up input amps for analog loopback */
+	/* Amp Indices: DAC = 0, mixer = 1 */
+	{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)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* FIXME: use Mux-type input source selection */
+	/* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
+	/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+	{0x23, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* set EAPD */
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{ }
+};
+
+static const struct hda_verb alc269vb_init_verbs[] = {
+	/*
+	 * Unmute ADC0 and set the default input to mic-in
+	 */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+	/*
+	 * Set up output mixers (0x02 - 0x03)
+	 */
+	/* set vol=0 to output mixers */
+	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+	/* set up input amps for analog loopback */
+	/* Amp Indices: DAC = 0, mixer = 1 */
+	{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)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* FIXME: use Mux-type input source selection */
+	/* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
+	/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+	{0x22, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* set EAPD */
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{ }
+};
+
+/*
+ * configuration and preset
+ */
+static const char * const alc269_models[ALC269_MODEL_LAST] = {
+	[ALC269_BASIC]			= "basic",
+	[ALC269_QUANTA_FL1]		= "quanta",
+	[ALC269_AMIC]			= "laptop-amic",
+	[ALC269_DMIC]			= "laptop-dmic",
+	[ALC269_FUJITSU]		= "fujitsu",
+	[ALC269_LIFEBOOK]		= "lifebook",
+	[ALC269_AUTO]			= "auto",
+};
+
+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",
+		      ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269VB_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269VB_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269VB_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269VB_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x11b3, "ASUS K52DR", ALC269VB_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269VB_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80Jt", ALC269VB_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82JV", ALC269VB_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1593, "ASUS N51Vn", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
+		      ALC269_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
+		      ALC269_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC),
+	SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO),
+	SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
+	SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC),
+	SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU),
+	SND_PCI_QUIRK(0x17aa, 0x3be9, "Quanta Wistron", ALC269_AMIC),
+	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_AMIC),
+	SND_PCI_QUIRK(0x17ff, 0x059a, "Quanta EL3", ALC269_DMIC),
+	SND_PCI_QUIRK(0x17ff, 0x059b, "Quanta JR1", ALC269_DMIC),
+	{}
+};
+
+static const struct alc_config_preset alc269_presets[] = {
+	[ALC269_BASIC] = {
+		.mixers = { alc269_base_mixer },
+		.init_verbs = { alc269_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
+		.dac_nids = alc269_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc269_modes),
+		.channel_mode = alc269_modes,
+		.input_mux = &alc269_capture_source,
+	},
+	[ALC269_QUANTA_FL1] = {
+		.mixers = { alc269_quanta_fl1_mixer },
+		.init_verbs = { alc269_init_verbs, alc269_quanta_fl1_verbs },
+		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
+		.dac_nids = alc269_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc269_modes),
+		.channel_mode = alc269_modes,
+		.input_mux = &alc269_capture_source,
+		.unsol_event = alc269_quanta_fl1_unsol_event,
+		.setup = alc269_quanta_fl1_setup,
+		.init_hook = alc269_quanta_fl1_init_hook,
+	},
+	[ALC269_AMIC] = {
+		.mixers = { alc269_laptop_mixer },
+		.cap_mixer = alc269_laptop_analog_capture_mixer,
+		.init_verbs = { alc269_init_verbs,
+				alc269_laptop_amic_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
+		.dac_nids = alc269_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc269_modes),
+		.channel_mode = alc269_modes,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc269_laptop_amic_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC269_DMIC] = {
+		.mixers = { alc269_laptop_mixer },
+		.cap_mixer = alc269_laptop_digital_capture_mixer,
+		.init_verbs = { alc269_init_verbs,
+				alc269_laptop_dmic_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
+		.dac_nids = alc269_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc269_modes),
+		.channel_mode = alc269_modes,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc269_laptop_dmic_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC269VB_AMIC] = {
+		.mixers = { alc269vb_laptop_mixer },
+		.cap_mixer = alc269vb_laptop_analog_capture_mixer,
+		.init_verbs = { alc269vb_init_verbs,
+				alc269vb_laptop_amic_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
+		.dac_nids = alc269_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc269_modes),
+		.channel_mode = alc269_modes,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc269vb_laptop_amic_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC269VB_DMIC] = {
+		.mixers = { alc269vb_laptop_mixer },
+		.cap_mixer = alc269vb_laptop_digital_capture_mixer,
+		.init_verbs = { alc269vb_init_verbs,
+				alc269vb_laptop_dmic_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
+		.dac_nids = alc269_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc269_modes),
+		.channel_mode = alc269_modes,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc269vb_laptop_dmic_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC269_FUJITSU] = {
+		.mixers = { alc269_fujitsu_mixer },
+		.cap_mixer = alc269_laptop_digital_capture_mixer,
+		.init_verbs = { alc269_init_verbs,
+				alc269_laptop_dmic_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
+		.dac_nids = alc269_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc269_modes),
+		.channel_mode = alc269_modes,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc269_laptop_dmic_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC269_LIFEBOOK] = {
+		.mixers = { alc269_lifebook_mixer },
+		.init_verbs = { alc269_init_verbs, alc269_lifebook_verbs },
+		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
+		.dac_nids = alc269_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc269_modes),
+		.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] = {
+		.mixers = { alc269_asus_mixer },
+		.cap_mixer = alc269vb_laptop_digital_capture_mixer,
+		.init_verbs = { alc269_init_verbs, alc271_acer_dmic_verbs },
+		.num_dacs = ARRAY_SIZE(alc269_dac_nids),
+		.dac_nids = alc269_dac_nids,
+		.adc_nids = alc262_dmic_adc_nids,
+		.num_adc_nids = ARRAY_SIZE(alc262_dmic_adc_nids),
+		.capsrc_nids = alc262_dmic_capsrc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc269_modes),
+		.channel_mode = alc269_modes,
+		.input_mux = &alc269_capture_source,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc269vb_laptop_dmic_setup,
+		.init_hook = alc_inithook,
+	},
+};
+

sound/pci/hda/alc662_quirks.c

@@ -0,0 +1,1408 @@
+/*
+ * ALC662/ALC663/ALC665/ALC670 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC662 models */
+enum {
+	ALC662_AUTO,
+	ALC662_3ST_2ch_DIG,
+	ALC662_3ST_6ch_DIG,
+	ALC662_3ST_6ch,
+	ALC662_5ST_DIG,
+	ALC662_LENOVO_101E,
+	ALC662_ASUS_EEEPC_P701,
+	ALC662_ASUS_EEEPC_EP20,
+	ALC663_ASUS_M51VA,
+	ALC663_ASUS_G71V,
+	ALC663_ASUS_H13,
+	ALC663_ASUS_G50V,
+	ALC662_ECS,
+	ALC663_ASUS_MODE1,
+	ALC662_ASUS_MODE2,
+	ALC663_ASUS_MODE3,
+	ALC663_ASUS_MODE4,
+	ALC663_ASUS_MODE5,
+	ALC663_ASUS_MODE6,
+	ALC663_ASUS_MODE7,
+	ALC663_ASUS_MODE8,
+	ALC272_DELL,
+	ALC272_DELL_ZM1,
+	ALC272_SAMSUNG_NC10,
+	ALC662_MODEL_LAST,
+};
+
+#define ALC662_DIGOUT_NID	0x06
+#define ALC662_DIGIN_NID	0x0a
+
+static const hda_nid_t alc662_dac_nids[3] = {
+	/* front, rear, clfe */
+	0x02, 0x03, 0x04
+};
+
+static const hda_nid_t alc272_dac_nids[2] = {
+	0x02, 0x03
+};
+
+static const hda_nid_t alc662_adc_nids[2] = {
+	/* ADC1-2 */
+	0x09, 0x08
+};
+
+static const hda_nid_t alc272_adc_nids[1] = {
+	/* ADC1-2 */
+	0x08,
+};
+
+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 const struct hda_input_mux alc662_capture_source = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x1 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
+static const struct hda_input_mux alc662_lenovo_101e_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x1 },
+		{ "Line", 0x2 },
+	},
+};
+
+static const struct hda_input_mux alc663_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x1 },
+		{ "Line", 0x2 },
+	},
+};
+
+#if 0 /* set to 1 for testing other input sources below */
+static const struct hda_input_mux alc272_nc10_capture_source = {
+	.num_items = 16,
+	.items = {
+		{ "Autoselect Mic", 0x0 },
+		{ "Internal Mic", 0x1 },
+		{ "In-0x02", 0x2 },
+		{ "In-0x03", 0x3 },
+		{ "In-0x04", 0x4 },
+		{ "In-0x05", 0x5 },
+		{ "In-0x06", 0x6 },
+		{ "In-0x07", 0x7 },
+		{ "In-0x08", 0x8 },
+		{ "In-0x09", 0x9 },
+		{ "In-0x0a", 0x0a },
+		{ "In-0x0b", 0x0b },
+		{ "In-0x0c", 0x0c },
+		{ "In-0x0d", 0x0d },
+		{ "In-0x0e", 0x0e },
+		{ "In-0x0f", 0x0f },
+	},
+};
+#endif
+
+/*
+ * 2ch mode
+ */
+static const struct hda_channel_mode alc662_3ST_2ch_modes[1] = {
+	{ 2, NULL }
+};
+
+/*
+ * 2ch mode
+ */
+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 },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+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 },
+	{ 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 },
+	{ } /* end */
+};
+
+static const struct hda_channel_mode alc662_3ST_6ch_modes[2] = {
+	{ 2, alc662_3ST_ch2_init },
+	{ 6, alc662_3ST_ch6_init },
+};
+
+/*
+ * 2ch mode
+ */
+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 },
+	{ } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+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 const struct hda_channel_mode alc662_5stack_modes[2] = {
+	{ 2, alc662_sixstack_ch6_init },
+	{ 6, alc662_sixstack_ch8_init },
+};
+
+/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
+ *                 Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
+ */
+
+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),
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x3, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+
+	/*Input mixer control */
+	HDA_CODEC_VOLUME("CD Playback Volume", 0xb, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0xb, 0x4, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0xb, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0xb, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0xb, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0xb, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0xb, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0xb, 0x01, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_BIND_MUTE("Speaker Playback Switch", 0x03, 2, HDA_INPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+static const struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+	ALC262_HIPPO_MASTER_SWITCH,
+
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("MuteCtrl Playback Switch", 0x0c, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+		HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
+		0
+	},
+};
+
+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),
+		HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+		0
+	},
+};
+
+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),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+		HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+		HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+		0
+	},
+};
+
+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),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+	{ } /* end */
+};
+
+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),
+		HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+		HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+		0
+	},
+};
+
+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),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+		HDA_COMPOSE_AMP_VAL(0x04, 3, 0, HDA_OUTPUT),
+		0
+	},
+};
+
+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),
+		HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_OUTPUT),
+		0
+	},
+};
+
+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),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+		HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+		HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
+		HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+		HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+		0
+	},
+};
+
+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),
+		HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
+		0
+	},
+};
+
+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),
+	HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("IntMic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("IntMic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+
+static const struct snd_kcontrol_new alc662_chmode_mixer[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+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},
+
+	{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)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+	/* Front Pin: output 0 (0x0c) */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* Rear Pin: output 1 (0x0d) */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* CLFE Pin: output 2 (0x0e) */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* Mic (rear) pin: input vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Front Mic pin: input vref at 80% */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Line In pin: input */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Line-2 In: Headphone output (output 0 - 0x0c) */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* CD pin widget for input */
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	/* FIXME: use matrix-type input source selection */
+	/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+	/* Input mixer */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+	{ }
+};
+
+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 const struct hda_verb alc662_sue_init_verbs[] = {
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_FRONT_EVENT},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_HP_EVENT},
+	{}
+};
+
+static const struct hda_verb alc662_eeepc_sue_init_verbs[] = {
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+/* Set Unsolicited Event*/
+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 | ALC_HP_EVENT},
+	{}
+};
+
+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},
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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},
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},	/* Headphone */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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},
+	{0x21, AC_VERB_SET_CONNECT_SEL, 0x0},	/* Headphone */
+	{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, 0x0},	/* Headphone */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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},
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
+	{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 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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 */
+
+	{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 */
+
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_FRONT_EVENT},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_HP_EVENT},
+	{}
+};
+
+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 */
+
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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 | ALC_MIC_EVENT},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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},
+	{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},
+	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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},
+	{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},
+	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{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 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1b, 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},
+	{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Headphone */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{}
+};
+
+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 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_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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)
+{
+	struct alc_spec *spec = codec->spec;
+
+	alc262_hippo1_setup(codec);
+	spec->ext_mic_pin = 0x18;
+	spec->int_mic_pin = 0x19;
+	spec->auto_mic = 1;
+}
+
+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;
+	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->int_mic_pin = 0x12;
+	spec->auto_mic = 1;
+}
+
+/* ***************** Mode1 ******************************/
+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->int_mic_pin = 0x19;
+	spec->auto_mic = 1;
+}
+
+/* ***************** Mode2 ******************************/
+static void alc662_mode2_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_PIN;
+	spec->ext_mic_pin = 0x18;
+	spec->int_mic_pin = 0x19;
+	spec->auto_mic = 1;
+}
+
+/* ***************** Mode3 ******************************/
+static void alc663_mode3_setup(struct hda_codec *codec)
+{
+	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->int_mic_pin = 0x19;
+	spec->auto_mic = 1;
+}
+
+/* ***************** Mode4 ******************************/
+static void alc663_mode4_setup(struct hda_codec *codec)
+{
+	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->int_mic_pin = 0x19;
+	spec->auto_mic = 1;
+}
+
+/* ***************** Mode5 ******************************/
+static void alc663_mode5_setup(struct hda_codec *codec)
+{
+	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->int_mic_pin = 0x19;
+	spec->auto_mic = 1;
+}
+
+/* ***************** Mode6 ******************************/
+static void alc663_mode6_setup(struct hda_codec *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->int_mic_pin = 0x19;
+	spec->auto_mic = 1;
+}
+
+/* ***************** Mode7 ******************************/
+static void alc663_mode7_setup(struct hda_codec *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->int_mic_pin = 0x19;
+	spec->auto_mic = 1;
+}
+
+/* ***************** Mode8 ******************************/
+static void alc663_mode8_setup(struct hda_codec *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->int_mic_pin = 0x12;
+	spec->auto_mic = 1;
+}
+
+static void alc663_g71v_setup(struct hda_codec *codec)
+{
+	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->int_mic_pin = 0x12;
+	spec->auto_mic = 1;
+}
+
+#define alc663_g50v_setup	alc663_m51va_setup
+
+static const struct snd_kcontrol_new alc662_ecs_mixer[] = {
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+	ALC262_HIPPO_MASTER_SWITCH,
+
+	HDA_CODEC_VOLUME("Mic/LineIn Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic/LineIn Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic/LineIn Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	{ } /* end */
+};
+
+
+/*
+ * configuration and preset
+ */
+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]	= "5stack-dig",
+	[ALC662_LENOVO_101E]	= "lenovo-101e",
+	[ALC662_ASUS_EEEPC_P701] = "eeepc-p701",
+	[ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20",
+	[ALC662_ECS] = "ecs",
+	[ALC663_ASUS_M51VA] = "m51va",
+	[ALC663_ASUS_G71V] = "g71v",
+	[ALC663_ASUS_H13] = "h13",
+	[ALC663_ASUS_G50V] = "g50v",
+	[ALC663_ASUS_MODE1] = "asus-mode1",
+	[ALC662_ASUS_MODE2] = "asus-mode2",
+	[ALC663_ASUS_MODE3] = "asus-mode3",
+	[ALC663_ASUS_MODE4] = "asus-mode4",
+	[ALC663_ASUS_MODE5] = "asus-mode5",
+	[ALC663_ASUS_MODE6] = "asus-mode6",
+	[ALC663_ASUS_MODE7] = "asus-mode7",
+	[ALC663_ASUS_MODE8] = "asus-mode8",
+	[ALC272_DELL]		= "dell",
+	[ALC272_DELL_ZM1]	= "dell-zm1",
+	[ALC272_SAMSUNG_NC10]	= "samsung-nc10",
+	[ALC662_AUTO]		= "auto",
+};
+
+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),
+	SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1303, "ASUS G60J", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1333, "ASUS G60Jx", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x13e3, "ASUS N71JA", ALC663_ASUS_MODE7),
+	SND_PCI_QUIRK(0x1043, 0x1463, "ASUS N71", ALC663_ASUS_MODE7),
+	SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G72", ALC663_ASUS_MODE8),
+	SND_PCI_QUIRK(0x1043, 0x1563, "ASUS N90", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x15d3, "ASUS N50SF F50SF", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS K40C K50C", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1733, "ASUS N81De", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1793, "ASUS F50GX", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x17b3, "ASUS F70SL", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x17c3, "ASUS UX20", ALC663_ASUS_M51VA),
+	SND_PCI_QUIRK(0x1043, 0x17f3, "ASUS X58LE", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5),
+	SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1853, "ASUS F50Z", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA),
+	/*SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),*/
+	SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x18b3, "ASUS N80Vc", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x18c3, "ASUS VX5", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS N81Te", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x18f3, "ASUS N505Tp", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1943, "ASUS Vx3V", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x1983, "ASUS N5051A", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V),
+	/*SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),*/
+	SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x19d3, "ASUS NB", ALC663_ASUS_M51VA),
+	SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4),
+	SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701),
+	SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20),
+	SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
+	SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
+		      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),
+	SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
+	SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0",
+					ALC662_3ST_6ch_DIG),
+	SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x",
+			   ALC663_ASUS_H13),
+	SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E),
+	{}
+};
+
+static const struct alc_config_preset alc662_presets[] = {
+	[ALC662_3ST_2ch_DIG] = {
+		.mixers = { alc662_3ST_2ch_mixer },
+		.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,
+		.dig_in_nid = ALC662_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+		.channel_mode = alc662_3ST_2ch_modes,
+		.input_mux = &alc662_capture_source,
+	},
+	[ALC662_3ST_6ch_DIG] = {
+		.mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
+		.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,
+		.dig_in_nid = ALC662_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
+		.channel_mode = alc662_3ST_6ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc662_capture_source,
+	},
+	[ALC662_3ST_6ch] = {
+		.mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
+		.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),
+		.channel_mode = alc662_3ST_6ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc662_capture_source,
+	},
+	[ALC662_5ST_DIG] = {
+		.mixers = { alc662_base_mixer, alc662_chmode_mixer },
+		.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,
+		.dig_in_nid = ALC662_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc662_5stack_modes),
+		.channel_mode = alc662_5stack_modes,
+		.input_mux = &alc662_capture_source,
+	},
+	[ALC662_LENOVO_101E] = {
+		.mixers = { alc662_lenovo_101e_mixer },
+		.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 = 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 = alc_sku_unsol_event,
+		.setup = alc662_eeepc_setup,
+		.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 = alc_sku_unsol_event,
+		.setup = alc662_eeepc_ep20_setup,
+		.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 = alc_sku_unsol_event,
+		.setup = alc662_eeepc_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC663_ASUS_M51VA] = {
+		.mixers = { alc663_m51va_mixer },
+		.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 = alc_sku_unsol_event,
+		.setup = alc663_m51va_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC663_ASUS_G71V] = {
+		.mixers = { alc663_g71v_mixer },
+		.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 = alc_sku_unsol_event,
+		.setup = alc663_g71v_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC663_ASUS_H13] = {
+		.mixers = { alc663_m51va_mixer },
+		.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,
+		.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,
+				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 = alc_sku_unsol_event,
+		.setup = alc663_g50v_setup,
+		.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,
+		.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 = alc_sku_unsol_event,
+		.setup = alc663_mode1_setup,
+		.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 = alc_sku_unsol_event,
+		.setup = alc662_mode2_setup,
+		.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,
+		.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 = alc_sku_unsol_event,
+		.setup = alc663_mode3_setup,
+		.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,
+		.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 = alc_sku_unsol_event,
+		.setup = alc663_mode4_setup,
+		.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,
+		.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 = alc_sku_unsol_event,
+		.setup = alc663_mode5_setup,
+		.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,
+		.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 = alc_sku_unsol_event,
+		.setup = alc663_mode6_setup,
+		.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,
+		.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 = alc_sku_unsol_event,
+		.setup = alc663_mode7_setup,
+		.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,
+		.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 = alc_sku_unsol_event,
+		.setup = alc663_mode8_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC272_DELL] = {
+		.mixers = { alc663_m51va_mixer },
+		.cap_mixer = alc272_auto_capture_mixer,
+		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
+				alc272_dell_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc272_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 = alc_sku_unsol_event,
+		.setup = alc663_m51va_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC272_DELL_ZM1] = {
+		.mixers = { alc663_m51va_mixer },
+		.cap_mixer = alc662_auto_capture_mixer,
+		.init_verbs = { alc662_init_verbs,
+				alc662_eapd_init_verbs,
+				alc272_dell_zm1_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc272_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 = alc_sku_unsol_event,
+		.setup = alc663_m51va_setup,
+		.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 = alc_sku_unsol_event,
+		.setup = alc663_mode4_setup,
+		.init_hook = alc_inithook,
+	},
+};
+
+

sound/pci/hda/alc680_quirks.c

@@ -0,0 +1,222 @@
+/*
+ * ALC680 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC680 models */
+enum {
+	ALC680_AUTO,
+	ALC680_BASE,
+	ALC680_MODEL_LAST,
+};
+
+#define ALC680_DIGIN_NID	ALC880_DIGIN_NID
+#define ALC680_DIGOUT_NID	ALC880_DIGOUT_NID
+#define alc680_modes		alc260_modes
+
+static const hda_nid_t alc680_dac_nids[3] = {
+	/* Lout1, Lout2, hp */
+	0x02, 0x03, 0x04
+};
+
+static const hda_nid_t alc680_adc_nids[3] = {
+	/* ADC0-2 */
+	/* DMIC, MIC, Line-in*/
+	0x07, 0x08, 0x09
+};
+
+/*
+ * Analog capture ADC cgange
+ */
+static hda_nid_t alc680_get_cur_adc(struct hda_codec *codec)
+{
+	static hda_nid_t pins[] = {0x18, 0x19};
+	static hda_nid_t adcs[] = {0x08, 0x09};
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(pins); i++) {
+		if (!is_jack_detectable(codec, pins[i]))
+			continue;
+		if (snd_hda_jack_detect(codec, pins[i]))
+			return adcs[i];
+	}
+	return 0x07;
+}
+
+static void alc680_rec_autoswitch(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t nid = alc680_get_cur_adc(codec);
+	if (spec->cur_adc && nid != spec->cur_adc) {
+		__snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1);
+		spec->cur_adc = nid;
+		snd_hda_codec_setup_stream(codec, nid,
+					   spec->cur_adc_stream_tag, 0,
+					   spec->cur_adc_format);
+	}
+}
+
+static int alc680_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 alc_spec *spec = codec->spec;
+	hda_nid_t nid = alc680_get_cur_adc(codec);
+
+	spec->cur_adc = nid;
+	spec->cur_adc_stream_tag = stream_tag;
+	spec->cur_adc_format = format;
+	snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
+	return 0;
+}
+
+static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				      struct hda_codec *codec,
+				      struct snd_pcm_substream *substream)
+{
+	struct alc_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 alc680_pcm_analog_auto_capture = {
+	.substreams = 1, /* can be overridden */
+	.channels_min = 2,
+	.channels_max = 2,
+	/* NID is set in alc_build_pcms */
+	.ops = {
+		.prepare = alc680_capture_pcm_prepare,
+		.cleanup = alc680_capture_pcm_cleanup
+	},
+};
+
+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),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x12, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line In Boost Volume", 0x19, 0, HDA_INPUT),
+	{ }
+};
+
+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),
+		HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+		HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+		0
+	},
+};
+
+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),
+		HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+		HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+		0
+	},
+};
+
+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 */
+};
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+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},
+
+	{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_OUT},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+	{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT   | AC_USRSP_EN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT  | AC_USRSP_EN},
+	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT  | AC_USRSP_EN},
+
+	{ }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc680_base_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x16;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[1] = 0x15;
+	spec->autocfg.num_inputs = 2;
+	spec->autocfg.inputs[0].pin = 0x18;
+	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) == ALC_HP_EVENT)
+		alc_hp_automute(codec);
+	if ((res >> 26) == ALC_MIC_EVENT)
+		alc680_rec_autoswitch(codec);
+}
+
+static void alc680_inithook(struct hda_codec *codec)
+{
+	alc_hp_automute(codec);
+	alc680_rec_autoswitch(codec);
+}
+
+/*
+ * configuration and preset
+ */
+static const char * const alc680_models[ALC680_MODEL_LAST] = {
+	[ALC680_BASE]		= "base",
+	[ALC680_AUTO]		= "auto",
+};
+
+static const struct snd_pci_quirk alc680_cfg_tbl[] = {
+	SND_PCI_QUIRK(0x1043, 0x12f3, "ASUS NX90", ALC680_BASE),
+	{}
+};
+
+static const struct alc_config_preset alc680_presets[] = {
+	[ALC680_BASE] = {
+		.mixers = { alc680_base_mixer },
+		.cap_mixer =  alc680_master_capture_mixer,
+		.init_verbs = { alc680_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc680_dac_nids),
+		.dac_nids = alc680_dac_nids,
+		.dig_out_nid = ALC680_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc680_modes),
+		.channel_mode = alc680_modes,
+		.unsol_event = alc680_unsol_event,
+		.setup = alc680_base_setup,
+		.init_hook = alc680_inithook,
+
+	},
+};

sound/pci/hda/alc861_quirks.c

@@ -0,0 +1,725 @@
+/*
+ * ALC660/ALC861 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC861 models */
+enum {
+	ALC861_AUTO,
+	ALC861_3ST,
+	ALC660_3ST,
+	ALC861_3ST_DIG,
+	ALC861_6ST_DIG,
+	ALC861_UNIWILL_M31,
+	ALC861_TOSHIBA,
+	ALC861_ASUS,
+	ALC861_ASUS_LAPTOP,
+	ALC861_MODEL_LAST,
+};
+
+/*
+ *  ALC861 channel source setting (2/6 channel selection for 3-stack)
+ */
+
+/*
+ * set the path ways for 2 channel output
+ * need to set the codec line out and mic 1 pin widgets to inputs
+ */
+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
+	 * the vref
+	 */
+	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+
+	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
+#if 0
+	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
+	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
+#endif
+	{ } /* end */
+};
+/*
+ * 6ch mode
+ * need to set the codec line out and mic 1 pin widgets to outputs
+ */
+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)*/
+	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+
+	{ 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
+
+	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
+#if 0
+	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
+	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
+#endif
+	{ } /* end */
+};
+
+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 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 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 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 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
+	 * the vref
+	 */
+	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+
+	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
+#if 0
+	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
+	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
+#endif
+	{ } /* end */
+};
+/* Set mic1 nad line-in as output and mute mixer */
+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 }, */
+	/* set pin widget 18h (mic1) for output (CLFE)*/
+	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	/* { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
+	{ 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
+
+	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
+#if 0
+	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
+	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
+#endif
+	{ } /* end */
+};
+
+static const struct hda_channel_mode alc861_asus_modes[2] = {
+	{ 2, alc861_asus_ch2_init },
+	{ 6, alc861_asus_ch6_init },
+};
+
+/* patch-ALC861 */
+
+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),
+	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
+
+        /*Input mixer control */
+	/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+	   HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
+
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
+	/*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
+
+	/* Input mixer control */
+	/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+	   HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
+
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+                .private_value = ARRAY_SIZE(alc861_threestack_modes),
+	},
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
+	/*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
+
+	/* Input mixer control */
+	/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+	   HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
+
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+                .private_value = ARRAY_SIZE(alc861_uniwill_m31_modes),
+	},
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
+
+	/* Input mixer control */
+	HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_OUTPUT),
+
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+                .private_value = ARRAY_SIZE(alc861_asus_modes),
+	},
+	{ }
+};
+
+/* additional 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),
+	{ }
+};
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static const struct hda_verb alc861_base_init_verbs[] = {
+	/*
+	 * Unmute ADC0 and set the default input to mic-in
+	 */
+	/* port-A for surround (rear panel) */
+	{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	{ 0x0e, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	/* port-B for mic-in (rear panel) with vref */
+	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* port-C for line-in (rear panel) */
+	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+	/* port-D for Front */
+	{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	/* port-E for HP out (front panel) */
+	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+	/* route front PCM to HP */
+	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	/* port-F for mic-in (front panel) with vref */
+	{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* port-G for CLFE (rear panel) */
+	{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	{ 0x1f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	/* port-H for side (rear panel) */
+	{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	/* CD-in */
+	{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+	/* route front mic to ADC1*/
+	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+	/* Unmute DAC0~3 & spdif out*/
+	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* Unmute Mixer 14 (mic) 1c (Line in)*/
+	{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+	/* Unmute Stereo Mixer 15 */
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
+
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	/* hp used DAC 3 (Front) */
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+        {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+
+	{ }
+};
+
+static const struct hda_verb alc861_threestack_init_verbs[] = {
+	/*
+	 * Unmute ADC0 and set the default input to mic-in
+	 */
+	/* port-A for surround (rear panel) */
+	{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+	/* port-B for mic-in (rear panel) with vref */
+	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* port-C for line-in (rear panel) */
+	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+	/* port-D for Front */
+	{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	/* port-E for HP out (front panel) */
+	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+	/* route front PCM to HP */
+	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	/* port-F for mic-in (front panel) with vref */
+	{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* port-G for CLFE (rear panel) */
+	{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+	/* port-H for side (rear panel) */
+	{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+	/* CD-in */
+	{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+	/* route front mic to ADC1*/
+	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	/* Unmute DAC0~3 & spdif out*/
+	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* Unmute Mixer 14 (mic) 1c (Line in)*/
+	{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+	/* Unmute Stereo Mixer 15 */
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
+
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	/* hp used DAC 3 (Front) */
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+        {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+	{ }
+};
+
+static const struct hda_verb alc861_uniwill_m31_init_verbs[] = {
+	/*
+	 * Unmute ADC0 and set the default input to mic-in
+	 */
+	/* port-A for surround (rear panel) */
+	{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+	/* port-B for mic-in (rear panel) with vref */
+	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* port-C for line-in (rear panel) */
+	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+	/* port-D for Front */
+	{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	/* port-E for HP out (front panel) */
+	/* this has to be set to VREF80 */
+	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* route front PCM to HP */
+	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	/* port-F for mic-in (front panel) with vref */
+	{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* port-G for CLFE (rear panel) */
+	{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+	/* port-H for side (rear panel) */
+	{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+	/* CD-in */
+	{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+	/* route front mic to ADC1*/
+	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	/* Unmute DAC0~3 & spdif out*/
+	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* Unmute Mixer 14 (mic) 1c (Line in)*/
+	{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+	/* Unmute Stereo Mixer 15 */
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
+
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	/* hp used DAC 3 (Front) */
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+        {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+	{ }
+};
+
+static const struct hda_verb alc861_asus_init_verbs[] = {
+	/*
+	 * Unmute ADC0 and set the default input to mic-in
+	 */
+	/* port-A for surround (rear panel)
+	 * according to codec#0 this is the HP jack
+	 */
+	{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, /* was 0x00 */
+	/* route front PCM to HP */
+	{ 0x0e, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	/* port-B for mic-in (rear panel) with vref */
+	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* port-C for line-in (rear panel) */
+	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+	/* port-D for Front */
+	{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	/* port-E for HP out (front panel) */
+	/* this has to be set to VREF80 */
+	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* route front PCM to HP */
+	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	/* port-F for mic-in (front panel) with vref */
+	{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* port-G for CLFE (rear panel) */
+	{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	/* port-H for side (rear panel) */
+	{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	/* CD-in */
+	{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+	/* route front mic to ADC1*/
+	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	/* Unmute DAC0~3 & spdif out*/
+	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Unmute Mixer 14 (mic) 1c (Line in)*/
+	{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+	/* Unmute Stereo Mixer 15 */
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
+
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	/* hp used DAC 3 (Front) */
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+	{ }
+};
+
+/* additional init verbs for ASUS laptops */
+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 */
+	{ }
+};
+
+static const struct hda_verb alc861_toshiba_init_verbs[] = {
+	{0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+
+	{ }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc861_toshiba_automute(struct hda_codec *codec)
+{
+	unsigned int present = snd_hda_jack_detect(codec, 0x0f);
+
+	snd_hda_codec_amp_stereo(codec, 0x16, HDA_INPUT, 0,
+				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+	snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 3,
+				 HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE);
+}
+
+static void alc861_toshiba_unsol_event(struct hda_codec *codec,
+				       unsigned int res)
+{
+	if ((res >> 26) == ALC_HP_EVENT)
+		alc861_toshiba_automute(codec);
+}
+
+#define ALC861_DIGOUT_NID	0x07
+
+static const struct hda_channel_mode alc861_8ch_modes[1] = {
+	{ 8, NULL }
+};
+
+static const hda_nid_t alc861_dac_nids[4] = {
+	/* front, surround, clfe, side */
+	0x03, 0x06, 0x05, 0x04
+};
+
+static const hda_nid_t alc660_dac_nids[3] = {
+	/* front, clfe, surround */
+	0x03, 0x05, 0x06
+};
+
+static const hda_nid_t alc861_adc_nids[1] = {
+	/* ADC0-2 */
+	0x08,
+};
+
+static const struct hda_input_mux alc861_capture_source = {
+	.num_items = 5,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x3 },
+		{ "Line", 0x1 },
+		{ "CD", 0x4 },
+		{ "Mixer", 0x5 },
+	},
+};
+
+/*
+ * configuration and preset
+ */
+static const char * const alc861_models[ALC861_MODEL_LAST] = {
+	[ALC861_3ST]		= "3stack",
+	[ALC660_3ST]		= "3stack-660",
+	[ALC861_3ST_DIG]	= "3stack-dig",
+	[ALC861_6ST_DIG]	= "6stack-dig",
+	[ALC861_UNIWILL_M31]	= "uniwill-m31",
+	[ALC861_TOSHIBA]	= "toshiba",
+	[ALC861_ASUS]		= "asus",
+	[ALC861_ASUS_LAPTOP]	= "asus-laptop",
+	[ALC861_AUTO]		= "auto",
+};
+
+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),
+	SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS),
+	SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP),
+	SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG),
+	SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA),
+	/* FIXME: the entry below breaks Toshiba A100 (model=auto works!)
+	 *        Any other models that need this preset?
+	 */
+	/* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */
+	SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST),
+	SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST),
+	SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31),
+	SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31),
+	SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP),
+	/* FIXME: the below seems conflict */
+	/* SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), */
+	SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST),
+	SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST),
+	{}
+};
+
+static const struct alc_config_preset alc861_presets[] = {
+	[ALC861_3ST] = {
+		.mixers = { alc861_3ST_mixer },
+		.init_verbs = { alc861_threestack_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc861_dac_nids),
+		.dac_nids = alc861_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
+		.channel_mode = alc861_threestack_modes,
+		.need_dac_fix = 1,
+		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+		.adc_nids = alc861_adc_nids,
+		.input_mux = &alc861_capture_source,
+	},
+	[ALC861_3ST_DIG] = {
+		.mixers = { alc861_base_mixer },
+		.init_verbs = { alc861_threestack_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc861_dac_nids),
+		.dac_nids = alc861_dac_nids,
+		.dig_out_nid = ALC861_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
+		.channel_mode = alc861_threestack_modes,
+		.need_dac_fix = 1,
+		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+		.adc_nids = alc861_adc_nids,
+		.input_mux = &alc861_capture_source,
+	},
+	[ALC861_6ST_DIG] = {
+		.mixers = { alc861_base_mixer },
+		.init_verbs = { alc861_base_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc861_dac_nids),
+		.dac_nids = alc861_dac_nids,
+		.dig_out_nid = ALC861_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc861_8ch_modes),
+		.channel_mode = alc861_8ch_modes,
+		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+		.adc_nids = alc861_adc_nids,
+		.input_mux = &alc861_capture_source,
+	},
+	[ALC660_3ST] = {
+		.mixers = { alc861_3ST_mixer },
+		.init_verbs = { alc861_threestack_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc660_dac_nids),
+		.dac_nids = alc660_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
+		.channel_mode = alc861_threestack_modes,
+		.need_dac_fix = 1,
+		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+		.adc_nids = alc861_adc_nids,
+		.input_mux = &alc861_capture_source,
+	},
+	[ALC861_UNIWILL_M31] = {
+		.mixers = { alc861_uniwill_m31_mixer },
+		.init_verbs = { alc861_uniwill_m31_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc861_dac_nids),
+		.dac_nids = alc861_dac_nids,
+		.dig_out_nid = ALC861_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc861_uniwill_m31_modes),
+		.channel_mode = alc861_uniwill_m31_modes,
+		.need_dac_fix = 1,
+		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+		.adc_nids = alc861_adc_nids,
+		.input_mux = &alc861_capture_source,
+	},
+	[ALC861_TOSHIBA] = {
+		.mixers = { alc861_toshiba_mixer },
+		.init_verbs = { alc861_base_init_verbs,
+				alc861_toshiba_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc861_dac_nids),
+		.dac_nids = alc861_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+		.adc_nids = alc861_adc_nids,
+		.input_mux = &alc861_capture_source,
+		.unsol_event = alc861_toshiba_unsol_event,
+		.init_hook = alc861_toshiba_automute,
+	},
+	[ALC861_ASUS] = {
+		.mixers = { alc861_asus_mixer },
+		.init_verbs = { alc861_asus_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc861_dac_nids),
+		.dac_nids = alc861_dac_nids,
+		.dig_out_nid = ALC861_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc861_asus_modes),
+		.channel_mode = alc861_asus_modes,
+		.need_dac_fix = 1,
+		.hp_nid = 0x06,
+		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+		.adc_nids = alc861_adc_nids,
+		.input_mux = &alc861_capture_source,
+	},
+	[ALC861_ASUS_LAPTOP] = {
+		.mixers = { alc861_toshiba_mixer, alc861_asus_laptop_mixer },
+		.init_verbs = { alc861_asus_init_verbs,
+				alc861_asus_laptop_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc861_dac_nids),
+		.dac_nids = alc861_dac_nids,
+		.dig_out_nid = ALC861_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.need_dac_fix = 1,
+		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+		.adc_nids = alc861_adc_nids,
+		.input_mux = &alc861_capture_source,
+	},
+};
+

sound/pci/hda/alc861vd_quirks.c

@@ -0,0 +1,605 @@
+/*
+ * ALC660-VD/ALC861-VD quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC861-VD models */
+enum {
+	ALC861VD_AUTO,
+	ALC660VD_3ST,
+	ALC660VD_3ST_DIG,
+	ALC660VD_ASUS_V1S,
+	ALC861VD_3ST,
+	ALC861VD_3ST_DIG,
+	ALC861VD_6ST_DIG,
+	ALC861VD_LENOVO,
+	ALC861VD_DALLAS,
+	ALC861VD_HP,
+	ALC861VD_MODEL_LAST,
+};
+
+#define ALC861VD_DIGOUT_NID	0x06
+
+static const hda_nid_t alc861vd_dac_nids[4] = {
+	/* front, surr, clfe, side surr */
+	0x02, 0x03, 0x04, 0x05
+};
+
+/* dac_nids for ALC660vd are in a different order - according to
+ * Realtek's driver.
+ * This should probably result in a different mixer for 6stack models
+ * of ALC660vd codecs, but for now there is only 3stack mixer
+ * - and it is the same as in 861vd.
+ * adc_nids in ALC660vd are (is) the same as in 861vd
+ */
+static const hda_nid_t alc660vd_dac_nids[3] = {
+	/* front, rear, clfe, rear_surr */
+	0x02, 0x04, 0x03
+};
+
+static const hda_nid_t alc861vd_adc_nids[1] = {
+	/* ADC0 */
+	0x09,
+};
+
+static const hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 };
+
+/* input MUX */
+/* FIXME: should be a matrix-type input source selection */
+static const struct hda_input_mux alc861vd_capture_source = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x1 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
+static const struct hda_input_mux alc861vd_dallas_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Internal Mic", 0x1 },
+	},
+};
+
+static const struct hda_input_mux alc861vd_hp_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Front Mic", 0x0 },
+		{ "ATAPI Mic", 0x1 },
+	},
+};
+
+/*
+ * 2ch mode
+ */
+static const struct hda_channel_mode alc861vd_3stack_2ch_modes[1] = {
+	{ 2, NULL }
+};
+
+/*
+ * 6ch mode
+ */
+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 },
+	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+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 },
+	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ } /* end */
+};
+
+static const struct hda_channel_mode alc861vd_6stack_modes[2] = {
+	{ 6, alc861vd_6stack_ch6_init },
+	{ 8, alc861vd_6stack_ch8_init },
+};
+
+static const struct snd_kcontrol_new alc861vd_chmode_mixer[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
+ *                 Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
+ */
+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),
+
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0,
+				HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0,
+				HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Side Playback Volume", 0x05, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+
+	{ } /* end */
+};
+
+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),
+
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+
+	{ } /* end */
+};
+
+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),
+
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+
+	{ } /* end */
+};
+
+/* Pin assignment: Speaker=0x14, HP = 0x15,
+ *                 Mic=0x18, Internal Mic = 0x19, CD = 0x1c, PC Beep = 0x1d
+ */
+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),
+	HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+/* Pin assignment: Speaker=0x14, Line-out = 0x15,
+ *                 Front Mic=0x18, ATAPI Mic = 0x19,
+ */
+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),
+	HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+	{ } /* end */
+};
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static const struct hda_verb alc861vd_volume_init_verbs[] = {
+	/*
+	 * Unmute ADC0 and set the default input to mic-in
+	 */
+	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+	/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of
+	 * the analog-loopback mixer widget
+	 */
+	/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+	/* Capture mixer: unmute Mic, F-Mic, Line, CD inputs */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+	/*
+	 * Set up output mixers (0x02 - 0x05)
+	 */
+	/* set vol=0 to output mixers */
+	{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},
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+	/* set up input amps for analog loopback */
+	/* Amp Indices: DAC = 0, mixer = 1 */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+	{ }
+};
+
+/*
+ * 3-stack pin configuration:
+ * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
+ */
+static const struct hda_verb alc861vd_3stack_init_verbs[] = {
+	/*
+	 * Set pin mode and muting
+	 */
+	/* set front pin widgets 0x14 for output */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Mic (rear) pin: input vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Front Mic pin: input vref at 80% */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Line In pin: input */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Line-2 In: Headphone output (output 0 - 0x0c) */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* CD pin widget for input */
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	{ }
+};
+
+/*
+ * 6-stack pin configuration:
+ */
+static const struct hda_verb alc861vd_6stack_init_verbs[] = {
+	/*
+	 * Set pin mode and muting
+	 */
+	/* set front pin widgets 0x14 for output */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Rear Pin: output 1 (0x0d) */
+	{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},
+	/* CLFE Pin: output 2 (0x0e) */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+	/* Side Pin: output 3 (0x0f) */
+	{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},
+
+	/* Mic (rear) pin: input vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Front Mic pin: input vref at 80% */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Line In pin: input */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Line-2 In: Headphone output (output 0 - 0x0c) */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* CD pin widget for input */
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	{ }
+};
+
+static const struct hda_verb alc861vd_eapd_verbs[] = {
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{ }
+};
+
+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)},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+	{}
+};
+
+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_hp_automute(codec);
+	alc88x_simple_mic_automute(codec);
+}
+
+static void alc861vd_lenovo_unsol_event(struct hda_codec *codec,
+					unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC_MIC_EVENT:
+		alc88x_simple_mic_automute(codec);
+		break;
+	default:
+		alc_sku_unsol_event(codec, res);
+		break;
+	}
+}
+
+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},
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+	{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_MUTE(0)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+	{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_OUT},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{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_VREF50},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+
+	{ } /* end */
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc861vd_dallas_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;
+}
+
+/*
+ * configuration and preset
+ */
+static const char * const alc861vd_models[ALC861VD_MODEL_LAST] = {
+	[ALC660VD_3ST]		= "3stack-660",
+	[ALC660VD_3ST_DIG]	= "3stack-660-digout",
+	[ALC660VD_ASUS_V1S]	= "asus-v1s",
+	[ALC861VD_3ST]		= "3stack",
+	[ALC861VD_3ST_DIG]	= "3stack-digout",
+	[ALC861VD_6ST_DIG]	= "6stack-digout",
+	[ALC861VD_LENOVO]	= "lenovo",
+	[ALC861VD_DALLAS]	= "dallas",
+	[ALC861VD_HP]		= "hp",
+	[ALC861VD_AUTO]		= "auto",
+};
+
+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),
+	/*SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),*/ /* auto */
+	SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S),
+	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
+	SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
+	SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
+	/*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/
+	SND_PCI_QUIRK(0x1179, 0xff01, "Toshiba A135", ALC861VD_LENOVO),
+	SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO),
+	SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS),
+	SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
+	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", ALC861VD_LENOVO),
+	SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG),
+	{}
+};
+
+static const struct alc_config_preset alc861vd_presets[] = {
+	[ALC660VD_3ST] = {
+		.mixers = { alc861vd_3st_mixer },
+		.init_verbs = { alc861vd_volume_init_verbs,
+				 alc861vd_3stack_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
+		.dac_nids = alc660vd_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+		.channel_mode = alc861vd_3stack_2ch_modes,
+		.input_mux = &alc861vd_capture_source,
+	},
+	[ALC660VD_3ST_DIG] = {
+		.mixers = { alc861vd_3st_mixer },
+		.init_verbs = { alc861vd_volume_init_verbs,
+				 alc861vd_3stack_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
+		.dac_nids = alc660vd_dac_nids,
+		.dig_out_nid = ALC861VD_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+		.channel_mode = alc861vd_3stack_2ch_modes,
+		.input_mux = &alc861vd_capture_source,
+	},
+	[ALC861VD_3ST] = {
+		.mixers = { alc861vd_3st_mixer },
+		.init_verbs = { alc861vd_volume_init_verbs,
+				 alc861vd_3stack_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+		.dac_nids = alc861vd_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+		.channel_mode = alc861vd_3stack_2ch_modes,
+		.input_mux = &alc861vd_capture_source,
+	},
+	[ALC861VD_3ST_DIG] = {
+		.mixers = { alc861vd_3st_mixer },
+		.init_verbs = { alc861vd_volume_init_verbs,
+		 		 alc861vd_3stack_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+		.dac_nids = alc861vd_dac_nids,
+		.dig_out_nid = ALC861VD_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+		.channel_mode = alc861vd_3stack_2ch_modes,
+		.input_mux = &alc861vd_capture_source,
+	},
+	[ALC861VD_6ST_DIG] = {
+		.mixers = { alc861vd_6st_mixer, alc861vd_chmode_mixer },
+		.init_verbs = { alc861vd_volume_init_verbs,
+				alc861vd_6stack_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+		.dac_nids = alc861vd_dac_nids,
+		.dig_out_nid = ALC861VD_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc861vd_6stack_modes),
+		.channel_mode = alc861vd_6stack_modes,
+		.input_mux = &alc861vd_capture_source,
+	},
+	[ALC861VD_LENOVO] = {
+		.mixers = { alc861vd_lenovo_mixer },
+		.init_verbs = { alc861vd_volume_init_verbs,
+				alc861vd_3stack_init_verbs,
+				alc861vd_eapd_verbs,
+				alc861vd_lenovo_unsol_verbs },
+		.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
+		.dac_nids = alc660vd_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+		.channel_mode = alc861vd_3stack_2ch_modes,
+		.input_mux = &alc861vd_capture_source,
+		.unsol_event = alc861vd_lenovo_unsol_event,
+		.setup = alc861vd_lenovo_setup,
+		.init_hook = alc861vd_lenovo_init_hook,
+	},
+	[ALC861VD_DALLAS] = {
+		.mixers = { alc861vd_dallas_mixer },
+		.init_verbs = { alc861vd_dallas_verbs },
+		.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+		.dac_nids = alc861vd_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+		.channel_mode = alc861vd_3stack_2ch_modes,
+		.input_mux = &alc861vd_dallas_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc861vd_dallas_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC861VD_HP] = {
+		.mixers = { alc861vd_hp_mixer },
+		.init_verbs = { alc861vd_dallas_verbs, alc861vd_eapd_verbs },
+		.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+		.dac_nids = alc861vd_dac_nids,
+		.dig_out_nid = ALC861VD_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+		.channel_mode = alc861vd_3stack_2ch_modes,
+		.input_mux = &alc861vd_hp_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc861vd_dallas_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC660VD_ASUS_V1S] = {
+		.mixers = { alc861vd_lenovo_mixer },
+		.init_verbs = { alc861vd_volume_init_verbs,
+				alc861vd_3stack_init_verbs,
+				alc861vd_eapd_verbs,
+				alc861vd_lenovo_unsol_verbs },
+		.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
+		.dac_nids = alc660vd_dac_nids,
+		.dig_out_nid = ALC861VD_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+		.channel_mode = alc861vd_3stack_2ch_modes,
+		.input_mux = &alc861vd_capture_source,
+		.unsol_event = alc861vd_lenovo_unsol_event,
+		.setup = alc861vd_lenovo_setup,
+		.init_hook = alc861vd_lenovo_init_hook,
+	},
+};
+

sound/pci/hda/alc880_quirks.c

@@ -0,0 +1,1898 @@
+/*
+ * ALC880 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC880 board config type */
+enum {
+	ALC880_AUTO,
+	ALC880_3ST,
+	ALC880_3ST_DIG,
+	ALC880_5ST,
+	ALC880_5ST_DIG,
+	ALC880_W810,
+	ALC880_Z71V,
+	ALC880_6ST,
+	ALC880_6ST_DIG,
+	ALC880_F1734,
+	ALC880_ASUS,
+	ALC880_ASUS_DIG,
+	ALC880_ASUS_W1V,
+	ALC880_ASUS_DIG2,
+	ALC880_FUJITSU,
+	ALC880_UNIWILL_DIG,
+	ALC880_UNIWILL,
+	ALC880_UNIWILL_P53,
+	ALC880_CLEVO,
+	ALC880_TCL_S700,
+	ALC880_LG,
+	ALC880_LG_LW,
+	ALC880_MEDION_RIM,
+#ifdef CONFIG_SND_DEBUG
+	ALC880_TEST,
+#endif
+	ALC880_MODEL_LAST /* last tag */
+};
+
+/*
+ * ALC880 3-stack model
+ *
+ * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e)
+ * Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18,
+ *                 F-Mic = 0x1b, HP = 0x19
+ */
+
+static const hda_nid_t alc880_dac_nids[4] = {
+	/* front, rear, clfe, rear_surr */
+	0x02, 0x05, 0x04, 0x03
+};
+
+static const hda_nid_t alc880_adc_nids[3] = {
+	/* ADC0-2 */
+	0x07, 0x08, 0x09,
+};
+
+/* The datasheet says the node 0x07 is connected from inputs,
+ * but it shows zero connection in the real implementation on some devices.
+ * Note: this is a 915GAV bug, fixed on 915GLV
+ */
+static const hda_nid_t alc880_adc_nids_alt[2] = {
+	/* ADC1-2 */
+	0x08, 0x09,
+};
+
+#define ALC880_DIGOUT_NID	0x06
+#define ALC880_DIGIN_NID	0x0a
+#define ALC880_PIN_CD_NID	0x1c
+
+static const struct hda_input_mux alc880_capture_source = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x3 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
+/* channel source setting (2/6 channel selection for 3-stack) */
+/* 2ch mode */
+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 },
+	/* set mic-in to input vref 80%, mute it */
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ } /* end */
+};
+
+/* 6ch mode */
+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 },
+	/* set mic-in to output, unmute it */
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ } /* end */
+};
+
+static const struct hda_channel_mode alc880_threestack_modes[2] = {
+	{ 2, alc880_threestack_ch2_init },
+	{ 6, alc880_threestack_ch6_init },
+};
+
+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),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+/*
+ * ALC880 5-stack model
+ *
+ * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d),
+ *      Side = 0x02 (0xd)
+ * Pin assignment: Front = 0x14, Surr = 0x17, CLFE = 0x16
+ *                 Line-In/Side = 0x1a, Mic = 0x18, F-Mic = 0x1b, HP = 0x19
+ */
+
+/* additional mixers to alc880_three_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 */
+};
+
+/* channel source setting (6/8 channel selection for 5-stack) */
+/* 6ch mode */
+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 },
+	{ } /* end */
+};
+
+/* 8ch mode */
+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 const struct hda_channel_mode alc880_fivestack_modes[2] = {
+	{ 6, alc880_fivestack_ch6_init },
+	{ 8, alc880_fivestack_ch8_init },
+};
+
+
+/*
+ * ALC880 6-stack model
+ *
+ * DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e),
+ *      Side = 0x05 (0x0f)
+ * Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, Side = 0x17,
+ *   Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b
+ */
+
+static const hda_nid_t alc880_6st_dac_nids[4] = {
+	/* front, rear, clfe, rear_surr */
+	0x02, 0x03, 0x04, 0x05
+};
+
+static const struct hda_input_mux alc880_6stack_capture_source = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x1 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
+/* fixed 8-channels */
+static const struct hda_channel_mode alc880_sixstack_modes[1] = {
+	{ 8, NULL },
+};
+
+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),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+
+/*
+ * ALC880 W810 model
+ *
+ * W810 has rear IO for:
+ * Front (DAC 02)
+ * Surround (DAC 03)
+ * Center/LFE (DAC 04)
+ * Digital out (06)
+ *
+ * The system also has a pair of internal speakers, and a headphone jack.
+ * These are both connected to Line2 on the codec, hence to DAC 02.
+ *
+ * There is a variable resistor to control the speaker or headphone
+ * volume. This is a hardware-only device without a software API.
+ *
+ * Plugging headphones in will disable the internal speakers. This is
+ * implemented in hardware, not via the driver using jack sense. In
+ * a similar fashion, plugging into the rear socket marked "front" will
+ * disable both the speakers and headphones.
+ *
+ * For input, there's a microphone jack, and an "audio in" jack.
+ * These may not do anything useful with this driver yet, because I
+ * haven't setup any initialization verbs for these yet...
+ */
+
+static const hda_nid_t alc880_w810_dac_nids[3] = {
+	/* front, rear/surround, clfe */
+	0x02, 0x03, 0x04
+};
+
+/* fixed 6 channels */
+static const struct hda_channel_mode alc880_w810_modes[1] = {
+	{ 6, NULL }
+};
+
+/* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */
+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),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	{ } /* end */
+};
+
+
+/*
+ * Z710V model
+ *
+ * DAC: Front = 0x02 (0x0c), HP = 0x03 (0x0d)
+ * Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?),
+ *                 Line = 0x1a
+ */
+
+static const hda_nid_t alc880_z71v_dac_nids[1] = {
+	0x02
+};
+#define ALC880_Z71V_HP_DAC	0x03
+
+/* fixed 2 channels */
+static const struct hda_channel_mode alc880_2_jack_modes[1] = {
+	{ 2, NULL }
+};
+
+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),
+	HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+
+/*
+ * ALC880 F1734 model
+ *
+ * DAC: HP = 0x02 (0x0c), Front = 0x03 (0x0d)
+ * Pin assignment: HP = 0x14, Front = 0x15, Mic = 0x18
+ */
+
+static const hda_nid_t alc880_f1734_dac_nids[1] = {
+	0x03
+};
+#define ALC880_F1734_HP_DAC	0x02
+
+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),
+	HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+static const struct hda_input_mux alc880_f1734_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x1 },
+		{ "CD", 0x4 },
+	},
+};
+
+
+/*
+ * ALC880 ASUS model
+ *
+ * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
+ * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
+ *  Mic = 0x18, Line = 0x1a
+ */
+
+#define alc880_asus_dac_nids	alc880_w810_dac_nids	/* identical with w810 */
+#define alc880_asus_modes	alc880_threestack_modes	/* 2/6 channel mode */
+
+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),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+/*
+ * ALC880 ASUS W1V model
+ *
+ * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
+ * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
+ *  Mic = 0x18, Line = 0x1a, Line2 = 0x1b
+ */
+
+/* additional mixers to alc880_asus_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 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),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0B, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0B, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0B, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0B, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+/* Uniwill */
+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),
+	HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+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),
+	HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+/*
+ * initialize the codec volumes, etc
+ */
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static const struct hda_verb alc880_volume_init_verbs[] = {
+	/*
+	 * Unmute ADC0-2 and set the default input to mic-in
+	 */
+	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+	/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+	 * mixer widget
+	 * Note: PASD motherboards uses the Line In 2 as the input for front
+	 * panel mic (mic 2)
+	 */
+	/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+
+	/*
+	 * Set up output mixers (0x0c - 0x0f)
+	 */
+	/* set vol=0 to output mixers */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	/* set up input amps for analog loopback */
+	/* Amp Indices: DAC = 0, mixer = 1 */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+	{ }
+};
+
+/*
+ * 3-stack pin configuration:
+ * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
+ */
+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
+	 */
+	{0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+	{0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+	{0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
+
+	/*
+	 * Set pin mode and muting
+	 */
+	/* set front pin widgets 0x14 for output */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Mic1 (rear panel) pin widget for input and vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Mic2 (as headphone out) for HP output */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Line In pin widget for input */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Line2 (as front mic) pin widget for input and vref at 80% */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* CD pin widget for input */
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	{ }
+};
+
+/*
+ * 5-stack pin configuration:
+ * front = 0x14, surround = 0x17, clfe = 0x16, mic = 0x18, HP = 0x19,
+ * line-in/side = 0x1a, f-mic = 0x1b
+ */
+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
+	 */
+	{0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+	{0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/side */
+
+	/*
+	 * Set pin mode and muting
+	 */
+	/* set pin widgets 0x14-0x17 for output */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	/* unmute pins for output (no gain on this amp) */
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* Mic1 (rear panel) pin widget for input and vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Mic2 (as headphone out) for HP output */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Line In pin widget for input */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Line2 (as front mic) pin widget for input and vref at 80% */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* CD pin widget for input */
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	{ }
+};
+
+/*
+ * W810 pin configuration:
+ * front = 0x14, surround = 0x15, clfe = 0x16, HP = 0x1b
+ */
+static const struct hda_verb alc880_pin_w810_init_verbs[] = {
+	/* hphone/speaker input selector: front DAC */
+	{0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
+
+	{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_OUT},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+	{ }
+};
+
+/*
+ * Z71V pin configuration:
+ * Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?)
+ */
+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},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	{ }
+};
+
+/*
+ * 6-stack pin configuration:
+ * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18,
+ * f-mic = 0x19, line = 0x1a, HP = 0x1b
+ */
+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},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{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},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{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},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	{ }
+};
+
+/*
+ * Uniwill pin configuration:
+ * HP = 0x14, InternalSpeaker = 0x15, mic = 0x18, internal mic = 0x19,
+ * line = 0x1a
+ */
+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},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{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},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, */
+	/* {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+
+	{ }
+};
+
+/*
+* Uniwill P53
+* HP = 0x14, InternalSpeaker = 0x15, mic = 0x19,
+ */
+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},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{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},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_DCVOL_EVENT},
+
+	{ }
+};
+
+static const struct hda_verb alc880_beep_init_verbs[] = {
+	{ 0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) },
+	{ }
+};
+
+static void alc880_uniwill_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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_hp_automute(codec);
+	alc88x_simple_mic_automute(codec);
+}
+
+static void alc880_uniwill_unsol_event(struct hda_codec *codec,
+				       unsigned int res)
+{
+	/* Looks like the unsol event is incompatible with the standard
+	 * definition.  4bit tag is placed at 28 bit!
+	 */
+	switch (res >> 28) {
+	case ALC_MIC_EVENT:
+		alc88x_simple_mic_automute(codec);
+		break;
+	default:
+		alc_sku_unsol_event(codec, res);
+		break;
+	}
+}
+
+static void alc880_uniwill_p53_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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)
+{
+	unsigned int present;
+
+	present = snd_hda_codec_read(codec, 0x21, 0,
+				     AC_VERB_GET_VOLUME_KNOB_CONTROL, 0);
+	present &= HDA_AMP_VOLMASK;
+	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0,
+				 HDA_AMP_VOLMASK, present);
+	snd_hda_codec_amp_stereo(codec, 0x0d, HDA_OUTPUT, 0,
+				 HDA_AMP_VOLMASK, present);
+}
+
+static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	/* Looks like the unsol event is incompatible with the standard
+	 * definition.  4bit tag is placed at 28 bit!
+	 */
+	if ((res >> 28) == ALC_DCVOL_EVENT)
+		alc880_uniwill_p53_dcvol_automute(codec);
+	else
+		alc_sku_unsol_event(codec, res);
+}
+
+/*
+ * F1734 pin configuration:
+ * HP = 0x14, speaker-out = 0x15, mic = 0x18
+ */
+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},
+	{0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_HP_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_DCVOL_EVENT},
+
+	{ }
+};
+
+/*
+ * ASUS pin configuration:
+ * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a
+ */
+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},
+	{0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	{ }
+};
+
+/* Enable GPIO mask and set output */
+#define alc880_gpio1_init_verbs	alc_gpio1_init_verbs
+#define alc880_gpio2_init_verbs	alc_gpio2_init_verbs
+#define alc880_gpio3_init_verbs	alc_gpio3_init_verbs
+
+/* Clevo m520g init */
+static const struct hda_verb alc880_pin_clevo_init_verbs[] = {
+	/* headphone output */
+	{0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
+	/* line-out */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Line-in */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* CD */
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Mic1 (rear panel) */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Mic2 (front panel) */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* headphone */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+        /* change to EAPD mode */
+	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+	{0x20, AC_VERB_SET_PROC_COEF,  0x3060},
+
+	{ }
+};
+
+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},
+
+	/* Headphone output */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	/* Front output*/
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Line In pin widget for input */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	/* CD pin widget for input */
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	/* Mic1 (rear panel) pin widget for input and vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+
+	/* change to EAPD mode */
+	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+	{0x20, AC_VERB_SET_PROC_COEF,  0x3070},
+
+	{ }
+};
+
+/*
+ * LG m1 express dual
+ *
+ * Pin assignment:
+ *   Rear Line-In/Out (blue): 0x14
+ *   Build-in Mic-In: 0x15
+ *   Speaker-out: 0x17
+ *   HP-Out (green): 0x1b
+ *   Mic-In/Out (red): 0x19
+ *   SPDIF-Out: 0x1e
+ */
+
+/* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */
+static const hda_nid_t alc880_lg_dac_nids[3] = {
+	0x05, 0x02, 0x03
+};
+
+/* seems analog CD is not working */
+static const struct hda_input_mux alc880_lg_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Mic", 0x1 },
+		{ "Line", 0x5 },
+		{ "Internal Mic", 0x6 },
+	},
+};
+
+/* 2,4,6 channel modes */
+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 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 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 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 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),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x06, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x06, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x07, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x07, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+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)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	/* mute all amp mixer inputs */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+	/* line-in to input */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* built-in mic */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* speaker-out */
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* mic-in to input */
+	{0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* HP-out */
+	{0x13, AC_VERB_SET_CONNECT_SEL, 0x03},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* jack sense */
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{ }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc880_lg_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x1b;
+	spec->autocfg.speaker_pins[0] = 0x17;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+/*
+ * LG LW20
+ *
+ * Pin assignment:
+ *   Speaker-out: 0x14
+ *   Mic-In: 0x18
+ *   Built-in Mic-In: 0x19
+ *   Line-In: 0x1b
+ *   HP-Out: 0x1a
+ *   SPDIF-Out: 0x1e
+ */
+
+static const struct hda_input_mux alc880_lg_lw_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Internal Mic", 0x1 },
+		{ "Line In", 0x2 },
+	},
+};
+
+#define alc880_lg_lw_modes alc880_threestack_modes
+
+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),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+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 */
+
+	/* set capture source to mic-in */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+	/* speaker-out */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* HP-out */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* mic-in to input */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* built-in mic */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* jack sense */
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{ }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc880_lg_lw_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 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),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+static const struct hda_input_mux alc880_medion_rim_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Internal Mic", 0x1 },
+	},
+};
+
+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},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* Mic1 (rear panel) pin widget for input and vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Mic2 (as headphone out) for HP output */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Internal Speaker */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+	{0x20, AC_VERB_SET_PROC_COEF,  0x3060},
+
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{ }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc880_medion_rim_automute(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	alc_hp_automute(codec);
+	/* toggle EAPD */
+	if (spec->jack_present)
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0);
+	else
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2);
+}
+
+static void alc880_medion_rim_unsol_event(struct hda_codec *codec,
+					  unsigned int res)
+{
+	/* Looks like the unsol event is incompatible with the standard
+	 * definition.  4bit tag is placed at 28 bit!
+	 */
+	if ((res >> 28) == ALC_HP_EVENT)
+		alc880_medion_rim_automute(codec);
+}
+
+static void alc880_medion_rim_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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 const struct hda_amp_list alc880_lg_loopbacks[] = {
+	{ 0x0b, HDA_INPUT, 1 },
+	{ 0x0b, HDA_INPUT, 6 },
+	{ 0x0b, HDA_INPUT, 7 },
+	{ } /* end */
+};
+#endif
+
+/*
+ * Test configuration for debugging
+ *
+ * Almost all inputs/outputs are enabled.  I/O pins can be configured via
+ * enum controls.
+ */
+#ifdef CONFIG_SND_DEBUG
+static const hda_nid_t alc880_test_dac_nids[4] = {
+	0x02, 0x03, 0x04, 0x05
+};
+
+static const struct hda_input_mux alc880_test_capture_source = {
+	.num_items = 7,
+	.items = {
+		{ "In-1", 0x0 },
+		{ "In-2", 0x1 },
+		{ "In-3", 0x2 },
+		{ "In-4", 0x3 },
+		{ "CD", 0x4 },
+		{ "Front", 0x5 },
+		{ "Surround", 0x6 },
+	},
+};
+
+static const struct hda_channel_mode alc880_test_modes[4] = {
+	{ 2, NULL },
+	{ 4, NULL },
+	{ 6, NULL },
+	{ 8, NULL },
+};
+
+static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	static const char * const texts[] = {
+		"N/A", "Line Out", "HP Out",
+		"In Hi-Z", "In 50%", "In Grd", "In 80%", "In 100%"
+	};
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = 8;
+	if (uinfo->value.enumerated.item >= 8)
+		uinfo->value.enumerated.item = 7;
+	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+
+static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+	unsigned int pin_ctl, item = 0;
+
+	pin_ctl = snd_hda_codec_read(codec, nid, 0,
+				     AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+	if (pin_ctl & AC_PINCTL_OUT_EN) {
+		if (pin_ctl & AC_PINCTL_HP_EN)
+			item = 2;
+		else
+			item = 1;
+	} else if (pin_ctl & AC_PINCTL_IN_EN) {
+		switch (pin_ctl & AC_PINCTL_VREFEN) {
+		case AC_PINCTL_VREF_HIZ: item = 3; break;
+		case AC_PINCTL_VREF_50:  item = 4; break;
+		case AC_PINCTL_VREF_GRD: item = 5; break;
+		case AC_PINCTL_VREF_80:  item = 6; break;
+		case AC_PINCTL_VREF_100: item = 7; break;
+		}
+	}
+	ucontrol->value.enumerated.item[0] = item;
+	return 0;
+}
+
+static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+	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,
+		AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD,
+		AC_PINCTL_IN_EN | AC_PINCTL_VREF_80,
+		AC_PINCTL_IN_EN | AC_PINCTL_VREF_100,
+	};
+	unsigned int old_ctl, new_ctl;
+
+	old_ctl = snd_hda_codec_read(codec, nid, 0,
+				     AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+	new_ctl = ctls[ucontrol->value.enumerated.item[0]];
+	if (old_ctl != new_ctl) {
+		int val;
+		snd_hda_codec_write_cache(codec, nid, 0,
+					  AC_VERB_SET_PIN_WIDGET_CONTROL,
+					  new_ctl);
+		val = ucontrol->value.enumerated.item[0] >= 3 ?
+			HDA_AMP_MUTE : 0;
+		snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, val);
+		return 1;
+	}
+	return 0;
+}
+
+static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	static const char * const texts[] = {
+		"Front", "Surround", "CLFE", "Side"
+	};
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = 4;
+	if (uinfo->value.enumerated.item >= 4)
+		uinfo->value.enumerated.item = 3;
+	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+
+static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+	unsigned int sel;
+
+	sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0);
+	ucontrol->value.enumerated.item[0] = sel & 3;
+	return 0;
+}
+
+static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+	unsigned int sel;
+
+	sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3;
+	if (ucontrol->value.enumerated.item[0] != sel) {
+		sel = ucontrol->value.enumerated.item[0] & 3;
+		snd_hda_codec_write_cache(codec, nid, 0,
+					  AC_VERB_SET_CONNECT_SEL, sel);
+		return 1;
+	}
+	return 0;
+}
+
+#define PIN_CTL_TEST(xname,nid) {			\
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,	\
+			.name = xname,		       \
+			.subdevice = HDA_SUBDEV_NID_FLAG | nid, \
+			.info = alc_test_pin_ctl_info, \
+			.get = alc_test_pin_ctl_get,   \
+			.put = alc_test_pin_ctl_put,   \
+			.private_value = nid	       \
+			}
+
+#define PIN_SRC_TEST(xname,nid) {			\
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,	\
+			.name = xname,		       \
+			.subdevice = HDA_SUBDEV_NID_FLAG | nid, \
+			.info = alc_test_pin_src_info, \
+			.get = alc_test_pin_src_get,   \
+			.put = alc_test_pin_src_put,   \
+			.private_value = nid	       \
+			}
+
+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),
+	HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_BIND_MUTE("CLFE Playback Switch", 0x0e, 2, HDA_INPUT),
+	HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+	PIN_CTL_TEST("Front Pin Mode", 0x14),
+	PIN_CTL_TEST("Surround Pin Mode", 0x15),
+	PIN_CTL_TEST("CLFE Pin Mode", 0x16),
+	PIN_CTL_TEST("Side Pin Mode", 0x17),
+	PIN_CTL_TEST("In-1 Pin Mode", 0x18),
+	PIN_CTL_TEST("In-2 Pin Mode", 0x19),
+	PIN_CTL_TEST("In-3 Pin Mode", 0x1a),
+	PIN_CTL_TEST("In-4 Pin Mode", 0x1b),
+	PIN_SRC_TEST("In-1 Pin Source", 0x18),
+	PIN_SRC_TEST("In-2 Pin Source", 0x19),
+	PIN_SRC_TEST("In-3 Pin Source", 0x1a),
+	PIN_SRC_TEST("In-4 Pin Source", 0x1b),
+	HDA_CODEC_VOLUME("In-1 Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("In-1 Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("In-2 Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("In-2 Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("In-3 Playback Volume", 0x0b, 0x2, HDA_INPUT),
+	HDA_CODEC_MUTE("In-3 Playback Switch", 0x0b, 0x2, HDA_INPUT),
+	HDA_CODEC_VOLUME("In-4 Playback Volume", 0x0b, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("In-4 Playback Switch", 0x0b, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x4, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+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)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	/* Vol output for 0x0c-0x0f */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	/* Set output pins 0x14-0x17 */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	/* Unmute output pins 0x14-0x17 */
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Set input pins 0x18-0x1c */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	/* Mute input pins 0x18-0x1b */
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* ADC set up */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* Analog input/passthru */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	{ }
+};
+#endif
+
+/*
+ */
+
+static const char * const alc880_models[ALC880_MODEL_LAST] = {
+	[ALC880_3ST]		= "3stack",
+	[ALC880_TCL_S700]	= "tcl",
+	[ALC880_3ST_DIG]	= "3stack-digout",
+	[ALC880_CLEVO]		= "clevo",
+	[ALC880_5ST]		= "5stack",
+	[ALC880_5ST_DIG]	= "5stack-digout",
+	[ALC880_W810]		= "w810",
+	[ALC880_Z71V]		= "z71v",
+	[ALC880_6ST]		= "6stack",
+	[ALC880_6ST_DIG]	= "6stack-digout",
+	[ALC880_ASUS]		= "asus",
+	[ALC880_ASUS_W1V]	= "asus-w1v",
+	[ALC880_ASUS_DIG]	= "asus-dig",
+	[ALC880_ASUS_DIG2]	= "asus-dig2",
+	[ALC880_UNIWILL_DIG]	= "uniwill",
+	[ALC880_UNIWILL_P53]	= "uniwill-p53",
+	[ALC880_FUJITSU]	= "fujitsu",
+	[ALC880_F1734]		= "F1734",
+	[ALC880_LG]		= "lg",
+	[ALC880_LG_LW]		= "lg-lw",
+	[ALC880_MEDION_RIM]	= "medion",
+#ifdef CONFIG_SND_DEBUG
+	[ALC880_TEST]		= "test",
+#endif
+	[ALC880_AUTO]		= "auto",
+};
+
+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),
+	SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_3ST_DIG),
+	SND_PCI_QUIRK(0x1025, 0x0077, "ULI", ALC880_6ST_DIG),
+	SND_PCI_QUIRK(0x1025, 0x0078, "ULI", ALC880_6ST_DIG),
+	SND_PCI_QUIRK(0x1025, 0x0087, "ULI", ALC880_6ST_DIG),
+	SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG),
+	SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST),
+	SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG),
+	SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V),
+	SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG),
+	SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG),
+	SND_PCI_QUIRK(0x1043, 0x1113, "ASUS", ALC880_ASUS_DIG),
+	SND_PCI_QUIRK(0x1043, 0x1123, "ASUS", ALC880_ASUS_DIG),
+	SND_PCI_QUIRK(0x1043, 0x1173, "ASUS", ALC880_ASUS_DIG),
+	SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_Z71V),
+	/* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */
+	SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG),
+	SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS", ALC880_ASUS_DIG),
+	SND_PCI_QUIRK(0x1043, 0x814e, "ASUS P5GD1 w/SPDIF", ALC880_6ST_DIG),
+	SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG),
+	SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST),
+	SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST),
+	SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_ASUS), /* default ASUS */
+	SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST),
+	SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST),
+	SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST),
+	SND_PCI_QUIRK(0x107b, 0x3033, "Gateway", ALC880_5ST),
+	SND_PCI_QUIRK(0x107b, 0x4039, "Gateway", ALC880_5ST),
+	SND_PCI_QUIRK(0x1297, 0xc790, "Shuttle ST20G5", ALC880_6ST_DIG),
+	SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_6ST_DIG),
+	SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_6ST_DIG),
+	SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG),
+	SND_PCI_QUIRK(0x1558, 0x0520, "Clevo m520G", ALC880_CLEVO),
+	SND_PCI_QUIRK(0x1558, 0x0660, "Clevo m655n", ALC880_CLEVO),
+	SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2),
+	SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG),
+	SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG),
+	SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_F1734),
+	SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL),
+	SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53),
+	SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810),
+	SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_MEDION_RIM),
+	SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG),
+	SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG),
+	SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734),
+	SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU),
+	SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734),
+	SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU),
+	SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW),
+	SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG),
+	SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_LG),
+	SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_LG),
+	SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_LG_LW),
+	SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700),
+	SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */
+	SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG),
+	SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG),
+	SND_PCI_QUIRK(0x8086, 0xd400, "Intel mobo", ALC880_5ST_DIG),
+	SND_PCI_QUIRK(0x8086, 0xd401, "Intel mobo", ALC880_5ST_DIG),
+	SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_3ST_DIG),
+	SND_PCI_QUIRK(0x8086, 0xe224, "Intel mobo", ALC880_5ST_DIG),
+	SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_3ST_DIG),
+	SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_3ST_DIG),
+	SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG),
+	SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG),
+	SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG),
+	/* default Intel */
+	SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_3ST),
+	SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG),
+	SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG),
+	{}
+};
+
+/*
+ * ALC880 codec presets
+ */
+static const struct alc_config_preset alc880_presets[] = {
+	[ALC880_3ST] = {
+		.mixers = { alc880_three_stack_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_3stack_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_dac_nids),
+		.dac_nids = alc880_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+		.channel_mode = alc880_threestack_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc880_capture_source,
+	},
+	[ALC880_3ST_DIG] = {
+		.mixers = { alc880_three_stack_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_3stack_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_dac_nids),
+		.dac_nids = alc880_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+		.channel_mode = alc880_threestack_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc880_capture_source,
+	},
+	[ALC880_TCL_S700] = {
+		.mixers = { alc880_tcl_s700_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_tcl_S700_init_verbs,
+				alc880_gpio2_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_dac_nids),
+		.dac_nids = alc880_dac_nids,
+		.adc_nids = alc880_adc_nids_alt, /* FIXME: correct? */
+		.num_adc_nids = 1, /* single ADC */
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+		.channel_mode = alc880_2_jack_modes,
+		.input_mux = &alc880_capture_source,
+	},
+	[ALC880_5ST] = {
+		.mixers = { alc880_three_stack_mixer,
+			    alc880_five_stack_mixer},
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_5stack_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_dac_nids),
+		.dac_nids = alc880_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
+		.channel_mode = alc880_fivestack_modes,
+		.input_mux = &alc880_capture_source,
+	},
+	[ALC880_5ST_DIG] = {
+		.mixers = { alc880_three_stack_mixer,
+			    alc880_five_stack_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_5stack_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_dac_nids),
+		.dac_nids = alc880_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
+		.channel_mode = alc880_fivestack_modes,
+		.input_mux = &alc880_capture_source,
+	},
+	[ALC880_6ST] = {
+		.mixers = { alc880_six_stack_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_6stack_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
+		.dac_nids = alc880_6st_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
+		.channel_mode = alc880_sixstack_modes,
+		.input_mux = &alc880_6stack_capture_source,
+	},
+	[ALC880_6ST_DIG] = {
+		.mixers = { alc880_six_stack_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_6stack_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
+		.dac_nids = alc880_6st_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
+		.channel_mode = alc880_sixstack_modes,
+		.input_mux = &alc880_6stack_capture_source,
+	},
+	[ALC880_W810] = {
+		.mixers = { alc880_w810_base_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_w810_init_verbs,
+				alc880_gpio2_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_w810_dac_nids),
+		.dac_nids = alc880_w810_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_w810_modes),
+		.channel_mode = alc880_w810_modes,
+		.input_mux = &alc880_capture_source,
+	},
+	[ALC880_Z71V] = {
+		.mixers = { alc880_z71v_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_z71v_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_z71v_dac_nids),
+		.dac_nids = alc880_z71v_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+		.channel_mode = alc880_2_jack_modes,
+		.input_mux = &alc880_capture_source,
+	},
+	[ALC880_F1734] = {
+		.mixers = { alc880_f1734_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_f1734_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_f1734_dac_nids),
+		.dac_nids = alc880_f1734_dac_nids,
+		.hp_nid = 0x02,
+		.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+		.channel_mode = alc880_2_jack_modes,
+		.input_mux = &alc880_f1734_capture_source,
+		.unsol_event = alc880_uniwill_p53_unsol_event,
+		.setup = alc880_uniwill_p53_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC880_ASUS] = {
+		.mixers = { alc880_asus_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_asus_init_verbs,
+				alc880_gpio1_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+		.dac_nids = alc880_asus_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+		.channel_mode = alc880_asus_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc880_capture_source,
+	},
+	[ALC880_ASUS_DIG] = {
+		.mixers = { alc880_asus_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_asus_init_verbs,
+				alc880_gpio1_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+		.dac_nids = alc880_asus_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+		.channel_mode = alc880_asus_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc880_capture_source,
+	},
+	[ALC880_ASUS_DIG2] = {
+		.mixers = { alc880_asus_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_asus_init_verbs,
+				alc880_gpio2_init_verbs }, /* use GPIO2 */
+		.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+		.dac_nids = alc880_asus_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+		.channel_mode = alc880_asus_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc880_capture_source,
+	},
+	[ALC880_ASUS_W1V] = {
+		.mixers = { alc880_asus_mixer, alc880_asus_w1v_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_asus_init_verbs,
+				alc880_gpio1_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+		.dac_nids = alc880_asus_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+		.channel_mode = alc880_asus_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc880_capture_source,
+	},
+	[ALC880_UNIWILL_DIG] = {
+		.mixers = { alc880_asus_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_asus_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+		.dac_nids = alc880_asus_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+		.channel_mode = alc880_asus_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc880_capture_source,
+	},
+	[ALC880_UNIWILL] = {
+		.mixers = { alc880_uniwill_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_uniwill_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+		.dac_nids = alc880_asus_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+		.channel_mode = alc880_threestack_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc880_capture_source,
+		.unsol_event = alc880_uniwill_unsol_event,
+		.setup = alc880_uniwill_setup,
+		.init_hook = alc880_uniwill_init_hook,
+	},
+	[ALC880_UNIWILL_P53] = {
+		.mixers = { alc880_uniwill_p53_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_uniwill_p53_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+		.dac_nids = alc880_asus_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc880_w810_modes),
+		.channel_mode = alc880_threestack_modes,
+		.input_mux = &alc880_capture_source,
+		.unsol_event = alc880_uniwill_p53_unsol_event,
+		.setup = alc880_uniwill_p53_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC880_FUJITSU] = {
+		.mixers = { alc880_fujitsu_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_uniwill_p53_init_verbs,
+	       			alc880_beep_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_dac_nids),
+		.dac_nids = alc880_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+		.channel_mode = alc880_2_jack_modes,
+		.input_mux = &alc880_capture_source,
+		.unsol_event = alc880_uniwill_p53_unsol_event,
+		.setup = alc880_uniwill_p53_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC880_CLEVO] = {
+		.mixers = { alc880_three_stack_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_pin_clevo_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_dac_nids),
+		.dac_nids = alc880_dac_nids,
+		.hp_nid = 0x03,
+		.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+		.channel_mode = alc880_threestack_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc880_capture_source,
+	},
+	[ALC880_LG] = {
+		.mixers = { alc880_lg_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_lg_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_lg_dac_nids),
+		.dac_nids = alc880_lg_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes),
+		.channel_mode = alc880_lg_ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc880_lg_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc880_lg_setup,
+		.init_hook = alc_hp_automute,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+		.loopbacks = alc880_lg_loopbacks,
+#endif
+	},
+	[ALC880_LG_LW] = {
+		.mixers = { alc880_lg_lw_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_lg_lw_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_dac_nids),
+		.dac_nids = alc880_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.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_sku_unsol_event,
+		.setup = alc880_lg_lw_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC880_MEDION_RIM] = {
+		.mixers = { alc880_medion_rim_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_medion_rim_init_verbs,
+				alc_gpio2_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_dac_nids),
+		.dac_nids = alc880_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+		.channel_mode = alc880_2_jack_modes,
+		.input_mux = &alc880_medion_rim_capture_source,
+		.unsol_event = alc880_medion_rim_unsol_event,
+		.setup = alc880_medion_rim_setup,
+		.init_hook = alc880_medion_rim_automute,
+	},
+#ifdef CONFIG_SND_DEBUG
+	[ALC880_TEST] = {
+		.mixers = { alc880_test_mixer },
+		.init_verbs = { alc880_test_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_test_dac_nids),
+		.dac_nids = alc880_test_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_test_modes),
+		.channel_mode = alc880_test_modes,
+		.input_mux = &alc880_test_capture_source,
+	},
+#endif
+};
+

sound/pci/hda/alc882_quirks.c

@@ -0,0 +1,3755 @@
+/*
+ * ALC882/ALC883/ALC888/ALC889 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC882 models */
+enum {
+	ALC882_AUTO,
+	ALC882_3ST_DIG,
+	ALC882_6ST_DIG,
+	ALC882_ARIMA,
+	ALC882_W2JC,
+	ALC882_TARGA,
+	ALC882_ASUS_A7J,
+	ALC882_ASUS_A7M,
+	ALC885_MACPRO,
+	ALC885_MBA21,
+	ALC885_MBP3,
+	ALC885_MB5,
+	ALC885_MACMINI3,
+	ALC885_IMAC24,
+	ALC885_IMAC91,
+	ALC883_3ST_2ch_DIG,
+	ALC883_3ST_6ch_DIG,
+	ALC883_3ST_6ch,
+	ALC883_6ST_DIG,
+	ALC883_TARGA_DIG,
+	ALC883_TARGA_2ch_DIG,
+	ALC883_TARGA_8ch_DIG,
+	ALC883_ACER,
+	ALC883_ACER_ASPIRE,
+	ALC888_ACER_ASPIRE_4930G,
+	ALC888_ACER_ASPIRE_6530G,
+	ALC888_ACER_ASPIRE_8930G,
+	ALC888_ACER_ASPIRE_7730G,
+	ALC883_MEDION,
+	ALC883_MEDION_WIM2160,
+	ALC883_LAPTOP_EAPD,
+	ALC883_LENOVO_101E_2ch,
+	ALC883_LENOVO_NB0763,
+	ALC888_LENOVO_MS7195_DIG,
+	ALC888_LENOVO_SKY,
+	ALC883_HAIER_W66,
+	ALC888_3ST_HP,
+	ALC888_6ST_DELL,
+	ALC883_MITAC,
+	ALC883_CLEVO_M540R,
+	ALC883_CLEVO_M720,
+	ALC883_FUJITSU_PI2515,
+	ALC888_FUJITSU_XA3530,
+	ALC883_3ST_6ch_INTEL,
+	ALC889A_INTEL,
+	ALC889_INTEL,
+	ALC888_ASUS_M90V,
+	ALC888_ASUS_EEE1601,
+	ALC889A_MB31,
+	ALC1200_ASUS_P5Q,
+	ALC883_SONY_VAIO_TT,
+	ALC882_MODEL_LAST,
+};
+
+/*
+ * 2ch mode
+ */
+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 },
+/* Line-in jack as Line in */
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+/* Line-Out as Front */
+	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{ } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+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 },
+/* Line-in jack as Surround */
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+/* Line-Out as Front */
+	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{ } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+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 },
+/* Line-in jack as Surround */
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+/* Line-Out as CLFE (workaround because Mic-in is not loud enough) */
+	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+	{ } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+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 },
+/* Line-in jack as Surround */
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+/* Line-Out as Side */
+	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+	{ } /* end */
+};
+
+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 },
+	{ 8, alc888_4ST_ch8_intel_init },
+};
+
+/*
+ * ALC888 Fujitsu Siemens Amillo xa3530
+ */
+
+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 */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Connect Bass HP to Front */
+	{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, 0x00},
+/* Connect Line-Out side jack (SPDIF) to Side */
+	{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},
+/* Connect Mic jack to CLFE */
+	{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},
+/* Connect Line-in jack to Surround */
+	{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},
+/* Connect HP out jack to Front */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Enable unsolicited event for HP jack and Line-out jack */
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{0x17, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{}
+};
+
+static void alc889_automute_setup(struct hda_codec *codec)
+{
+	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->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_hp_automute(codec);
+}
+
+static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x17; /* line-out */
+	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 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 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
+/* Enable unsolicited event for HP jack */
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+/* Connect Internal HP to front */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Connect HP out to front */
+	{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, 0x00},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{ }
+};
+
+/*
+ * ALC888 Acer Aspire 6530G model
+ */
+
+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)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+/* 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) */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+/* Unselect Front Mic by default in input mixer 3 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
+/* Enable unsolicited event for HP jack */
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+/* Enable speaker output */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+/* Enable headphone output */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{ }
+};
+
+/*
+ *ALC888 Acer Aspire 7730G model
+ */
+
+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) */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+/* Unselect Front Mic by default in input mixer 3 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
+/* Enable unsolicited event for HP jack */
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+/* Enable speaker output */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+/* Enable headphone output */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+/*Enable internal subwoofer */
+	{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, 0x02},
+	{0x17, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{ }
+};
+
+/*
+ * ALC889 Acer Aspire 8930G model
+ */
+
+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 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
+/* Enable unsolicited event for HP jack */
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+/* Connect Internal Front to Front */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Connect Internal Rear to Rear */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x01},
+/* Connect Internal CLFE to CLFE */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+/* Connect HP out to Front */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Enable all DACs */
+/*  DAC DISABLE/MUTE 1? */
+/*  setting bits 1-5 disables DAC nids 0x02-0x06 apparently. Init=0x38 */
+	{0x20, AC_VERB_SET_COEF_INDEX, 0x03},
+	{0x20, AC_VERB_SET_PROC_COEF, 0x0000},
+/*  DAC DISABLE/MUTE 2? */
+/*  some bit here disables the other DACs. Init=0x4900 */
+	{0x20, AC_VERB_SET_COEF_INDEX, 0x08},
+	{0x20, AC_VERB_SET_PROC_COEF, 0x0000},
+/* DMIC fix
+ * This laptop has a stereo digital microphone. The mics are only 1cm apart
+ * which makes the stereo useless. However, either the mic or the ALC889
+ * makes the signal become a difference/sum signal instead of standard
+ * stereo, which is annoying. So instead we flip this bit which makes the
+ * codec replicate the sum signal to both channels, turning it into a
+ * normal mono mic.
+ */
+/*  DMIC_CONTROL? Init value = 0x0001 */
+	{0x20, AC_VERB_SET_COEF_INDEX, 0x0b},
+	{0x20, AC_VERB_SET_PROC_COEF, 0x0003},
+	{ }
+};
+
+static const struct hda_input_mux alc888_2_capture_sources[2] = {
+	/* Front mic only available on one ADC */
+	{
+		.num_items = 4,
+		.items = {
+			{ "Mic", 0x0 },
+			{ "Line", 0x2 },
+			{ "CD", 0x4 },
+			{ "Front Mic", 0xb },
+		},
+	},
+	{
+		.num_items = 3,
+		.items = {
+			{ "Mic", 0x0 },
+			{ "Line", 0x2 },
+			{ "CD", 0x4 },
+		},
+	}
+};
+
+static const struct hda_input_mux alc888_acer_aspire_6530_sources[2] = {
+	/* Interal mic only available on one ADC */
+	{
+		.num_items = 5,
+		.items = {
+			{ "Mic", 0x0 },
+			{ "Line In", 0x2 },
+			{ "CD", 0x4 },
+			{ "Input Mix", 0xa },
+			{ "Internal Mic", 0xb },
+		},
+	},
+	{
+		.num_items = 4,
+		.items = {
+			{ "Mic", 0x0 },
+			{ "Line In", 0x2 },
+			{ "CD", 0x4 },
+			{ "Input Mix", 0xa },
+		},
+	}
+};
+
+static const struct hda_input_mux alc889_capture_sources[3] = {
+	/* Digital mic only available on first "ADC" */
+	{
+		.num_items = 5,
+		.items = {
+			{ "Mic", 0x0 },
+			{ "Line", 0x2 },
+			{ "CD", 0x4 },
+			{ "Front Mic", 0xb },
+			{ "Input Mix", 0xa },
+		},
+	},
+	{
+		.num_items = 4,
+		.items = {
+			{ "Mic", 0x0 },
+			{ "Line", 0x2 },
+			{ "CD", 0x4 },
+			{ "Input Mix", 0xa },
+		},
+	},
+	{
+		.num_items = 4,
+		.items = {
+			{ "Mic", 0x0 },
+			{ "Line", 0x2 },
+			{ "CD", 0x4 },
+			{ "Input Mix", 0xa },
+		},
+	}
+};
+
+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),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+		HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+		HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Internal LFE Playback Volume", 0x0f, 1, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Internal LFE Playback Switch", 0x0f, 1, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+		HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+
+static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec)
+{
+	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->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)
+{
+	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->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)
+{
+	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->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)
+{
+	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->autocfg.speaker_pins[2] = 0x1b;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+#define ALC882_DIGOUT_NID	0x06
+#define ALC882_DIGIN_NID	0x0a
+#define ALC883_DIGOUT_NID	ALC882_DIGOUT_NID
+#define ALC883_DIGIN_NID	ALC882_DIGIN_NID
+#define ALC1200_DIGOUT_NID	0x10
+
+
+static const struct hda_channel_mode alc882_ch_modes[1] = {
+	{ 8, NULL }
+};
+
+/* DACs */
+static const hda_nid_t alc882_dac_nids[4] = {
+	/* front, rear, clfe, rear_surr */
+	0x02, 0x03, 0x04, 0x05
+};
+#define alc883_dac_nids		alc882_dac_nids
+
+/* ADCs */
+#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 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 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 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 const struct hda_input_mux alc882_capture_source = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x1 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
+#define alc883_capture_source	alc882_capture_source
+
+static const struct hda_input_mux alc889_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Front Mic", 0x0 },
+		{ "Mic", 0x3 },
+		{ "Line", 0x2 },
+	},
+};
+
+static const struct hda_input_mux mb5_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Mic", 0x1 },
+		{ "Line", 0x7 },
+		{ "CD", 0x4 },
+	},
+};
+
+static const struct hda_input_mux macmini3_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
+static const struct hda_input_mux alc883_3stack_6ch_intel = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x1 },
+		{ "Front Mic", 0x0 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
+static const struct hda_input_mux alc883_lenovo_101e_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x1 },
+		{ "Line", 0x2 },
+	},
+};
+
+static const struct hda_input_mux alc883_lenovo_nb0763_capture_source = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Internal Mic", 0x1 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
+static const struct hda_input_mux alc883_fujitsu_pi2515_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Internal Mic", 0x1 },
+	},
+};
+
+static const struct hda_input_mux alc883_lenovo_sky_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x1 },
+		{ "Line", 0x4 },
+	},
+};
+
+static const struct hda_input_mux alc883_asus_eee1601_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Line", 0x2 },
+	},
+};
+
+static const struct hda_input_mux alc889A_mb31_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x0 },
+		/* Front Mic (0x01) unused */
+		{ "Line", 0x2 },
+		/* Line 2 (0x03) unused */
+		/* CD (0x04) unused? */
+	},
+};
+
+static const struct hda_input_mux alc889A_imac91_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x01 },
+		{ "Line", 0x2 }, /* Not sure! */
+	},
+};
+
+/*
+ * 2ch mode
+ */
+static const struct hda_channel_mode alc883_3ST_2ch_modes[1] = {
+	{ 2, NULL }
+};
+
+/*
+ * 2ch mode
+ */
+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 },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+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 },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+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 },
+	{ 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 },
+	{ } /* end */
+};
+
+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 },
+};
+
+#define alc883_3ST_6ch_modes	alc882_3ST_6ch_modes
+
+/*
+ * 2ch mode
+ */
+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 },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+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 },
+	{ 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 },
+	{ } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+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 },
+	{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+	{ 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 },
+	{ } /* end */
+};
+
+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 },
+};
+
+
+/*
+ * 6ch mode
+ */
+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 },
+	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+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 },
+	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ } /* end */
+};
+
+static const struct hda_channel_mode alc882_sixstack_modes[2] = {
+	{ 6, alc882_sixstack_ch6_init },
+	{ 8, alc882_sixstack_ch8_init },
+};
+
+
+/* Macbook Air 2,1 */
+
+static const struct hda_channel_mode alc885_mba21_ch_modes[1] = {
+      { 2, NULL },
+};
+
+/*
+ * macbook pro ALC885 can switch LineIn to LineOut without losing Mic
+ */
+
+/*
+ * 2ch mode
+ */
+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)},
+	{ } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+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 },
+	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{ } /* end */
+};
+
+static const struct hda_channel_mode alc885_mbp_4ch_modes[2] = {
+	{ 2, alc885_mbp_ch2_init },
+	{ 4, alc885_mbp_ch4_init },
+};
+
+/*
+ * 2ch
+ * Speakers/Woofer/HP = Front
+ * LineIn = Input
+ */
+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 */
+};
+
+/*
+ * 6ch mode
+ * Speakers/HP = Front
+ * Woofer = LFE
+ * LineIn = Surround
+ */
+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 const struct hda_channel_mode alc885_mb5_6ch_modes[2] = {
+	{ 2, alc885_mb5_ch2_init },
+	{ 6, alc885_mb5_ch6_init },
+};
+
+#define alc885_macmini3_6ch_modes	alc885_mb5_6ch_modes
+
+/*
+ * 2ch mode
+ */
+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 },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+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 },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ 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 },
+	{ } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+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 },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+	{ 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 },
+	{ } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+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 },
+	{ 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 },
+	{ 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 },
+	{ } /* end */
+};
+
+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 },
+	{ 8, alc883_4ST_ch8_init },
+};
+
+
+/*
+ * 2ch mode
+ */
+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 },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+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 },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+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 },
+	{ 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 },
+	{ } /* end */
+};
+
+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 },
+};
+
+/*
+ * 2ch mode
+ */
+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 },
+	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+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 },
+	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+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 },
+	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
+	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ } /* end */
+};
+
+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 },
+};
+
+/*
+ * 6ch mode
+ */
+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 },
+	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+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 },
+	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ } /* end */
+};
+
+static const struct hda_channel_mode alc883_sixstack_modes[2] = {
+	{ 6, alc883_sixstack_ch6_init },
+	{ 8, alc883_sixstack_ch8_init },
+};
+
+
+/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
+ *                 Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
+ */
+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),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+/* Macbook Air 2,1 same control for HP and internal Speaker */
+
+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 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),
+	HDA_BIND_MUTE   ("Headphone Playback Switch", 0x0e, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
+	HDA_CODEC_MUTE  ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_BIND_MUTE   ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
+	HDA_BIND_MUTE   ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
+	HDA_BIND_MUTE   ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
+	HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE  ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0x00, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_BIND_MUTE   ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
+	HDA_BIND_MUTE   ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
+	HDA_BIND_MUTE   ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
+	HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT),
+	{ } /* end */
+};
+
+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 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),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	{ } /* end */
+};
+
+/* Pin assignment: Front=0x14, HP = 0x15, Front = 0x16, ???
+ *                 Front Mic=0x18, Line In = 0x1a, Line In = 0x1b, CD = 0x1c
+ */
+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),
+	HDA_CODEC_MUTE("Mobile Front Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mobile Line Playback Volume", 0x0b, 0x03, HDA_INPUT),
+	HDA_CODEC_MUTE("Mobile Line Playback Switch", 0x0b, 0x03, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+static const struct snd_kcontrol_new alc882_chmode_mixer[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+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)},
+	/* Rear mixer */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* CLFE mixer */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* Side mixer */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+	/* Front Pin: output 0 (0x0c) */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* Rear Pin: output 1 (0x0d) */
+	{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},
+	/* CLFE Pin: output 2 (0x0e) */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+	/* Side Pin: output 3 (0x0f) */
+	{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},
+	/* Mic (rear) pin: input vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Front Mic pin: input vref at 80% */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Line In pin: input */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Line-2 In: Headphone output (output 0 - 0x0c) */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* CD pin widget for input */
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+	/* FIXME: use matrix-type input source selection */
+	/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+	/* Input mixer2 */
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	/* Input mixer3 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	/* ADC2: mute amp left and right */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* ADC3: mute amp left and right */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	{ }
+};
+
+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)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	/* ADC1: mute amp left and right */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{ }
+};
+
+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 const struct hda_verb alc889_eapd_verbs[] = {
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{ }
+};
+
+static const struct hda_verb alc_hp15_unsol_verbs[] = {
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{}
+};
+
+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)},
+	/* Rear mixer */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	/* CLFE mixer */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	/* Side mixer */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+	/* Front HP Pin: output 0 (0x0c) */
+	{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, 0x00},
+	/* Front Pin: output 0 (0x0c) */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* Rear Pin: output 1 (0x0d) */
+	{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, 0x01},
+	/* CLFE Pin: output 2 (0x0e) */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+	/* Side Pin: output 3 (0x0f) */
+	{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},
+	/* Mic (rear) pin: input vref at 80% */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Front Mic pin: input vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Line In pin: input */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+	/* Mixer elements: 0x18, , 0x1a, 0x1b */
+	/* Input mixer1 */
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	/* Input mixer2 */
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	/* Input mixer3 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	/* ADC2: mute amp left and right */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	/* ADC3: mute amp left and right */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+
+	{ }
+};
+
+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)},
+	{ }
+};
+
+
+/* Unmute Selector 24h and set the default input to front mic */
+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},
+	{ }
+};
+
+
+#define alc883_init_verbs	alc882_base_init_verbs
+
+/* Mac Pro test */
+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),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	/* FIXME: this looks suspicious...
+	HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	*/
+	{ } /* end */
+};
+
+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)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* Front Pin: output 0 (0x0c) */
+	{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, 0x00},
+	/* Front Mic pin: input vref at 80% */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Speaker:  output */
+	{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, 0x04},
+	/* Headphone output (output 0 - 0x0c) */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* FIXME: use matrix-type input source selection */
+	/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+	/* 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)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	/* Input mixer2 */
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	/* Input mixer3 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	/* ADC1: mute amp left and right */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* ADC2: mute amp left and right */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* ADC3: mute amp left and right */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	{ }
+};
+
+/* Macbook 5,1 */
+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},
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Front mixer */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* Surround mixer */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* LFE mixer */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* HP mixer */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* Front Pin (0x0c) */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* LFE Pin (0x0e) */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x02},
+	/* HP Pin (0x0f) */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	/* Front Mic pin: input vref at 80% */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Line In pin */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)},
+	{ }
+};
+
+/* Macmini 3,1 */
+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},
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Front mixer */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* Surround mixer */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* LFE mixer */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* HP mixer */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* Front Pin (0x0c) */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* LFE Pin (0x0e) */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x02},
+	/* HP Pin (0x0f) */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	/* Line In pin */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	{ }
+};
+
+
+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)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	/*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},
+	{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* HP Pin: output 0 (0x0e) */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)},
+	/* Line in (is hp when jack connected)*/
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+	{ }
+ };
+
+
+/* Macbook Pro rev3 */
+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)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* Rear mixer */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* HP mixer */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* Front Pin: output 0 (0x0c) */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* HP Pin: output 0 (0x0e) */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x02},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	/* Mic (rear) pin: input vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Front Mic pin: input vref at 80% */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Line In pin: use output 1 when in LineOut mode */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
+
+	/* FIXME: use matrix-type input source selection */
+	/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+	/* 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)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	/* Input mixer2 */
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	/* Input mixer3 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	/* ADC1: mute amp left and right */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* ADC2: mute amp left and right */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* ADC3: mute amp left and right */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	{ }
+};
+
+/* iMac 9,1 */
+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},
+	{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* HP Pin: Rear */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)},
+	/* Line in Rear */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Front Mic pin: input vref at 80% */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Rear mixer */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* Line-Out 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)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	/* 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	/* 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	/* 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{ }
+};
+
+/* iMac 24 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 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},
+	{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* Internal speakers: output 0 (0x0c) */
+	{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, 0x00},
+	/* Headphone: output 0 (0x0c) */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	/* Front Mic: input vref at 80% */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{ }
+};
+
+/* Toggle speaker-output according to the hp-jack state */
+static void alc885_imac24_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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
+#define alc885_macmini3_setup	alc885_imac24_setup
+
+/* Macbook Air 2,1 */
+static void alc885_mba21_setup(struct hda_codec *codec)
+{
+       struct alc_spec *spec = codec->spec;
+
+       spec->autocfg.hp_pins[0] = 0x14;
+       spec->autocfg.speaker_pins[0] = 0x18;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+
+
+static void alc885_mbp3_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 alc885_imac91_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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 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)},
+
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+	{0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{ } /* end */
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc882_targa_automute(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	alc_hp_automute(codec);
+	snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
+				  spec->jack_present ? 1 : 3);
+}
+
+static void alc882_targa_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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)
+{
+	if ((res >> 26) == ALC_HP_EVENT)
+		alc882_targa_automute(codec);
+}
+
+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)},
+
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+	{0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
+
+	{0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+	{ } /* end */
+};
+
+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)},
+
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+	{0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
+
+	{0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ 	{ } /* end */
+};
+
+static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted)
+{
+	unsigned int gpiostate, gpiomask, gpiodir;
+
+	gpiostate = snd_hda_codec_read(codec, codec->afg, 0,
+				       AC_VERB_GET_GPIO_DATA, 0);
+
+	if (!muted)
+		gpiostate |= (1 << pin);
+	else
+		gpiostate &= ~(1 << pin);
+
+	gpiomask = snd_hda_codec_read(codec, codec->afg, 0,
+				      AC_VERB_GET_GPIO_MASK, 0);
+	gpiomask |= (1 << pin);
+
+	gpiodir = snd_hda_codec_read(codec, codec->afg, 0,
+				     AC_VERB_GET_GPIO_DIRECTION, 0);
+	gpiodir |= (1 << pin);
+
+
+	snd_hda_codec_write(codec, codec->afg, 0,
+			    AC_VERB_SET_GPIO_MASK, gpiomask);
+	snd_hda_codec_write(codec, codec->afg, 0,
+			    AC_VERB_SET_GPIO_DIRECTION, gpiodir);
+
+	msleep(1);
+
+	snd_hda_codec_write(codec, codec->afg, 0,
+			    AC_VERB_SET_GPIO_DATA, gpiostate);
+}
+
+/* set up GPIO at initialization */
+static void alc885_macpro_init_hook(struct hda_codec *codec)
+{
+	alc882_gpio_mute(codec, 0, 0);
+	alc882_gpio_mute(codec, 1, 0);
+}
+
+/* set up GPIO and update auto-muting at initialization */
+static void alc885_imac24_init_hook(struct hda_codec *codec)
+{
+	alc885_macpro_init_hook(codec);
+	alc_hp_automute(codec);
+}
+
+/* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */
+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 */
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},   /* Line off */
+	{ } /* end */
+};
+
+/* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */
+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 */
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
+	{ } /* end */
+};
+
+/* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */
+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 */
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},   /* Line off */
+	{ } /* end */
+};
+
+/* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */
+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 */
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
+	{ } /* end */
+};
+
+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 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},
+	{ }
+};
+
+#define alc883_base_mixer	alc882_base_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),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+			      HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+			      HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Speaker Playback Switch", 0x0f, 2, HDA_INPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x1b, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x08, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+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)},
+
+	/* Set speaker pin to front mixer */
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Init headphone pin */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+
+	{ } /* end */
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_medion_wim2160_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x1a;
+	spec->autocfg.speaker_pins[0] = 0x15;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+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),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume",
+						0x0d, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x00,
+		HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x00, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x02, HDA_INPUT),
+	/* Output switches */
+	HDA_CODEC_MUTE("Enable Speaker", 0x14, 0x00, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Enable Headphones", 0x15, 0x00, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("Enable LFE", 0x16, 2, 0x00, HDA_OUTPUT),
+	/* Boost mixers */
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT),
+	/* Input mixers */
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+		HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+		0
+	},
+};
+
+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),
+		HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+		0
+	},
+};
+
+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),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+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),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		/* .name = "Capture Source", */
+		.name = "Input Source",
+		.count = 1,
+		.info = alc_mux_enum_info,
+		.get = alc_mux_enum_get,
+		.put = alc_mux_enum_put,
+	},
+	{ } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_chmode_mixer[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_mitac_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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 const struct hda_verb alc883_mitac_verbs[] = {
+	/* HP */
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	/* Subwoofer */
+	{0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+	/* enable unsolicited event */
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	/* {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN}, */
+
+	{ } /* end */
+};
+
+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},
+	/* Int speaker */
+	/*{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},*/
+
+	/* enable unsolicited event */
+	/*
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
+	*/
+
+	{ } /* end */
+};
+
+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},
+	/* Int speaker */
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+	/* enable unsolicited event */
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
+
+	{ } /* end */
+};
+
+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},
+	/* Subwoofer */
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+	/* enable unsolicited event */
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+
+	{ } /* end */
+};
+
+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)},
+
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+/* Connect Line-Out side jack (SPDIF) to Side */
+	{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},
+/* Connect Mic jack to CLFE */
+	{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},
+/* Connect Line-in jack to Surround */
+	{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},
+/* Connect HP out jack to Front */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+
+	{ } /* end */
+};
+
+static const struct hda_verb alc883_lenovo_101e_verbs[] = {
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_FRONT_EVENT|AC_USRSP_EN},
+        {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT|AC_USRSP_EN},
+	{ } /* end */
+};
+
+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, ALC_HP_EVENT | AC_USRSP_EN},
+        {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{ } /* end */
+};
+
+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},
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_FRONT_EVENT | AC_USRSP_EN},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT    | AC_USRSP_EN},
+	{ } /* end */
+};
+
+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)},
+
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{ } /* end */
+};
+
+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)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{ } /* end */
+};
+
+static const struct hda_verb alc888_6st_dell_verbs[] = {
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{ }
+};
+
+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},
+
+	/* enable unsolicited event */
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+
+	{ } /* end */
+};
+
+static void alc888_3st_hp_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x1b;
+	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 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) */
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{ } /* end */
+};
+
+/*
+ * 2ch mode
+ */
+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 },
+	{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+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 },
+	{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+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 },
+	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
+};
+
+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 },
+};
+
+static void alc888_lenovo_ms7195_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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 */
+static void alc883_lenovo_nb0763_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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 */
+#define alc883_targa_init_hook		alc882_targa_init_hook
+#define alc883_targa_unsol_event	alc882_targa_unsol_event
+
+static void alc883_clevo_m720_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 alc883_clevo_m720_init_hook(struct hda_codec *codec)
+{
+	alc_hp_automute(codec);
+	alc88x_simple_mic_automute(codec);
+}
+
+static void alc883_clevo_m720_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC_MIC_EVENT:
+		alc88x_simple_mic_automute(codec);
+		break;
+	default:
+		alc_sku_unsol_event(codec, res);
+		break;
+	}
+}
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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)
+{
+	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 alc883_lenovo_101e_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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 */
+static void alc883_acer_aspire_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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 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},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* Front Pin: output 0 (0x0c) */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
+        /* eanable EAPD on medion laptop */
+	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+	{0x20, AC_VERB_SET_PROC_COEF, 0x3050},
+	/* enable unsolicited event */
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{ }
+};
+
+static void alc888_6st_dell_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x1b;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	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)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x1b;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[1] = 0x15;
+	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)
+{
+	struct alc_spec *spec = codec->spec;
+
+	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 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)},
+	/* enable unsolicited event */
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
+	{ } /* end */
+};
+
+static void alc883_mode2_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x1b;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[1] = 0x15;
+	spec->autocfg.speaker_pins[2] = 0x16;
+	spec->ext_mic_pin = 0x18;
+	spec->int_mic_pin = 0x19;
+	spec->auto_mic = 1;
+	spec->automute = 1;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+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)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x20, AC_VERB_SET_COEF_INDEX, 0x0b},
+	{0x20, AC_VERB_SET_PROC_COEF,  0x0838},
+	/* enable unsolicited event */
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	{ } /* end */
+};
+
+static void alc883_eee1601_inithook(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[0] = 0x1b;
+	alc_hp_automute(codec);
+}
+
+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 */
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+	/* Init line pin (used as output in 4ch and 6ch mode) */
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x02},           /* Connect to CLFE */
+	/* Init line 2 pin (used as headphone out by default) */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},  /* Use as input */
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Mute output */
+	{ } /* end */
+};
+
+/* Mute speakers according to the headphone jack state */
+static void alc889A_mb31_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+
+	/* Mute only in 2ch or 4ch mode */
+	if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0)
+	    == 0x00) {
+		present = snd_hda_jack_detect(codec, 0x15);
+		snd_hda_codec_amp_stereo(codec, 0x14,  HDA_OUTPUT, 0,
+			HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+		snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+			HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+	}
+}
+
+static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	if ((res >> 26) == ALC_HP_EVENT)
+		alc889A_mb31_automute(codec);
+}
+
+static const hda_nid_t alc883_slave_dig_outs[] = {
+	ALC1200_DIGOUT_NID, 0,
+};
+
+static const hda_nid_t alc1200_slave_dig_outs[] = {
+	ALC883_DIGOUT_NID, 0,
+};
+
+/*
+ * configuration and preset
+ */
+static const char * const alc882_models[ALC882_MODEL_LAST] = {
+	[ALC882_3ST_DIG]	= "3stack-dig",
+	[ALC882_6ST_DIG]	= "6stack-dig",
+	[ALC882_ARIMA]		= "arima",
+	[ALC882_W2JC]		= "w2jc",
+	[ALC882_TARGA]		= "targa",
+	[ALC882_ASUS_A7J]	= "asus-a7j",
+	[ALC882_ASUS_A7M]	= "asus-a7m",
+	[ALC885_MACPRO]		= "macpro",
+	[ALC885_MB5]		= "mb5",
+	[ALC885_MACMINI3]	= "macmini3",
+	[ALC885_MBA21]		= "mba21",
+	[ALC885_MBP3]		= "mbp3",
+	[ALC885_IMAC24]		= "imac24",
+	[ALC885_IMAC91]		= "imac91",
+	[ALC883_3ST_2ch_DIG]	= "3stack-2ch-dig",
+	[ALC883_3ST_6ch_DIG]	= "3stack-6ch-dig",
+	[ALC883_3ST_6ch]	= "3stack-6ch",
+	[ALC883_6ST_DIG]	= "alc883-6stack-dig",
+	[ALC883_TARGA_DIG]	= "targa-dig",
+	[ALC883_TARGA_2ch_DIG]	= "targa-2ch-dig",
+	[ALC883_TARGA_8ch_DIG]	= "targa-8ch-dig",
+	[ALC883_ACER]		= "acer",
+	[ALC883_ACER_ASPIRE]	= "acer-aspire",
+	[ALC888_ACER_ASPIRE_4930G]	= "acer-aspire-4930g",
+	[ALC888_ACER_ASPIRE_6530G]	= "acer-aspire-6530g",
+	[ALC888_ACER_ASPIRE_8930G]	= "acer-aspire-8930g",
+	[ALC888_ACER_ASPIRE_7730G]	= "acer-aspire-7730g",
+	[ALC883_MEDION]		= "medion",
+	[ALC883_MEDION_WIM2160]	= "medion-wim2160",
+	[ALC883_LAPTOP_EAPD]	= "laptop-eapd",
+	[ALC883_LENOVO_101E_2ch] = "lenovo-101e",
+	[ALC883_LENOVO_NB0763]	= "lenovo-nb0763",
+	[ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig",
+	[ALC888_LENOVO_SKY] = "lenovo-sky",
+	[ALC883_HAIER_W66] 	= "haier-w66",
+	[ALC888_3ST_HP]		= "3stack-hp",
+	[ALC888_6ST_DELL]	= "6stack-dell",
+	[ALC883_MITAC]		= "mitac",
+	[ALC883_CLEVO_M540R]	= "clevo-m540r",
+	[ALC883_CLEVO_M720]	= "clevo-m720",
+	[ALC883_FUJITSU_PI2515] = "fujitsu-pi2515",
+	[ALC888_FUJITSU_XA3530] = "fujitsu-xa3530",
+	[ALC883_3ST_6ch_INTEL]	= "3stack-6ch-intel",
+	[ALC889A_INTEL]		= "intel-alc889a",
+	[ALC889_INTEL]		= "intel-x58",
+	[ALC1200_ASUS_P5Q]	= "asus-p5q",
+	[ALC889A_MB31]		= "mb31",
+	[ALC883_SONY_VAIO_TT]	= "sony-vaio-tt",
+	[ALC882_AUTO]		= "auto",
+};
+
+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),
+	SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE),
+	SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_ACER_ASPIRE),
+	SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
+	SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE),
+	SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE),
+	SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G",
+		ALC888_ACER_ASPIRE_4930G),
+	SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G",
+		ALC888_ACER_ASPIRE_4930G),
+	SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G",
+		ALC888_ACER_ASPIRE_8930G),
+	SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G",
+		ALC888_ACER_ASPIRE_8930G),
+	SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC882_AUTO),
+	SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC882_AUTO),
+	SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
+		ALC888_ACER_ASPIRE_6530G),
+	SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
+		ALC888_ACER_ASPIRE_6530G),
+	SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
+		ALC888_ACER_ASPIRE_7730G),
+	/* default Acer -- disabled as it causes more problems.
+	 *    model=auto should work fine now
+	 */
+	/* SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), */
+
+	SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
+
+	SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavilion", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
+	SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
+	SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP),
+	SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP),
+
+	SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J),
+	SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J),
+	SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M),
+	SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V),
+	SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC),
+	SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG),
+	SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG),
+	SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q),
+	SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601),
+
+	SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT),
+	SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG),
+	SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC),
+	SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC),
+	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),
+	SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG),
+	SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8  */
+	SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC882_AUTO),
+	SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
+	SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG),
+	SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG),
+	SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG),
+	SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG),
+	SND_PCI_QUIRK(0x1462, 0x3fc1, "MSI", ALC883_TARGA_DIG),
+	SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG),
+	SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG),
+	SND_PCI_QUIRK(0x1462, 0x3fdf, "MSI", ALC883_TARGA_DIG),
+	SND_PCI_QUIRK(0x1462, 0x42cd, "MSI", ALC883_TARGA_DIG),
+	SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG),
+	SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG),
+	SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG),
+	SND_PCI_QUIRK(0x1462, 0x4570, "MSI Wind Top AE2220", ALC883_TARGA_DIG),
+	SND_PCI_QUIRK(0x1462, 0x6510, "MSI GX620", ALC883_TARGA_8ch_DIG),
+	SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x1462, 0x7260, "MSI 7260", ALC883_TARGA_DIG),
+	SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x1462, 0x7437, "MSI NetOn AP1900", ALC883_TARGA_DIG),
+	SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
+	SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG),
+
+	SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x1558, 0x0571, "Clevo laptop M570U", ALC883_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
+	SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),
+	SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R),
+	SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD),
+	SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
+	/* SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), */
+	SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
+	SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1100, "FSC AMILO Xi/Pi25xx",
+		      ALC883_FUJITSU_PI2515),
+	SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1130, "Fujitsu AMILO Xa35xx",
+		ALC888_FUJITSU_XA3530),
+	SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch),
+	SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763),
+	SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),
+	SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
+	SND_PCI_QUIRK(0x17aa, 0x101d, "Lenovo Sky", ALC888_LENOVO_SKY),
+	SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG),
+	SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
+
+	SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL),
+	SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL),
+	SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC),
+	SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC889_INTEL),
+	SND_PCI_QUIRK(0x8086, 0x0021, "Intel IbexPeak", ALC889A_INTEL),
+	SND_PCI_QUIRK(0x8086, 0x3b56, "Intel IbexPeak", ALC889A_INTEL),
+	SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC882_6ST_DIG),
+
+	{}
+};
+
+/* codec SSID table for Intel Mac */
+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),
+	SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_MACPRO),
+	SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_IMAC24),
+	SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_IMAC24),
+	SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3),
+	SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889A_MB31),
+	SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_ASUS_A7M),
+	SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC885_MBP3),
+	SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC885_MBA21),
+	SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31),
+	SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
+	SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24),
+	SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC885_IMAC91),
+	SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
+	SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC885_MB5),
+	/* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2,
+	 * so apparently no perfect solution yet
+	 */
+	SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5),
+	SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5),
+	SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC885_MACMINI3),
+	{} /* terminator */
+};
+
+static const struct alc_config_preset alc882_presets[] = {
+	[ALC882_3ST_DIG] = {
+		.mixers = { alc882_base_mixer },
+		.init_verbs = { alc882_base_init_verbs,
+				alc882_adc1_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.dig_in_nid = ALC882_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+		.channel_mode = alc882_ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc882_capture_source,
+	},
+	[ALC882_6ST_DIG] = {
+		.mixers = { alc882_base_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc882_base_init_verbs,
+				alc882_adc1_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.dig_in_nid = ALC882_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
+		.channel_mode = alc882_sixstack_modes,
+		.input_mux = &alc882_capture_source,
+	},
+	[ALC882_ARIMA] = {
+		.mixers = { alc882_base_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+				alc882_eapd_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
+		.channel_mode = alc882_sixstack_modes,
+		.input_mux = &alc882_capture_source,
+	},
+	[ALC882_W2JC] = {
+		.mixers = { alc882_w2jc_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+				alc882_eapd_verbs, alc880_gpio1_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+		.channel_mode = alc880_threestack_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc882_capture_source,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+	},
+	   [ALC885_MBA21] = {
+			.mixers = { alc885_mba21_mixer },
+			.init_verbs = { alc885_mba21_init_verbs, alc880_gpio1_init_verbs },
+			.num_dacs = 2,
+			.dac_nids = alc882_dac_nids,
+			.channel_mode = alc885_mba21_ch_modes,
+			.num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
+			.input_mux = &alc882_capture_source,
+			.unsol_event = alc_sku_unsol_event,
+			.setup = alc885_mba21_setup,
+			.init_hook = alc_hp_automute,
+       },
+	[ALC885_MBP3] = {
+		.mixers = { alc885_mbp3_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc885_mbp3_init_verbs,
+				alc880_gpio1_init_verbs },
+		.num_dacs = 2,
+		.dac_nids = alc882_dac_nids,
+		.hp_nid = 0x04,
+		.channel_mode = alc885_mbp_4ch_modes,
+		.num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes),
+		.input_mux = &alc882_capture_source,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.dig_in_nid = ALC882_DIGIN_NID,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc885_mbp3_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC885_MB5] = {
+		.mixers = { alc885_mb5_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc885_mb5_init_verbs,
+				alc880_gpio1_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.channel_mode = alc885_mb5_6ch_modes,
+		.num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes),
+		.input_mux = &mb5_capture_source,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.dig_in_nid = ALC882_DIGIN_NID,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc885_mb5_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC885_MACMINI3] = {
+		.mixers = { alc885_macmini3_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc885_macmini3_init_verbs,
+				alc880_gpio1_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.channel_mode = alc885_macmini3_6ch_modes,
+		.num_channel_mode = ARRAY_SIZE(alc885_macmini3_6ch_modes),
+		.input_mux = &macmini3_capture_source,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.dig_in_nid = ALC882_DIGIN_NID,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc885_macmini3_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC885_MACPRO] = {
+		.mixers = { alc882_macpro_mixer },
+		.init_verbs = { alc882_macpro_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.dig_in_nid = ALC882_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+		.channel_mode = alc882_ch_modes,
+		.input_mux = &alc882_capture_source,
+		.init_hook = alc885_macpro_init_hook,
+	},
+	[ALC885_IMAC24] = {
+		.mixers = { alc885_imac24_mixer },
+		.init_verbs = { alc885_imac24_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.dig_in_nid = ALC882_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+		.channel_mode = alc882_ch_modes,
+		.input_mux = &alc882_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc885_imac24_setup,
+		.init_hook = alc885_imac24_init_hook,
+	},
+	[ALC885_IMAC91] = {
+		.mixers = {alc885_imac91_mixer},
+		.init_verbs = { alc885_imac91_init_verbs,
+				alc880_gpio1_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.channel_mode = alc885_mba21_ch_modes,
+		.num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
+		.input_mux = &alc889A_imac91_capture_source,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.dig_in_nid = ALC882_DIGIN_NID,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc885_imac91_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC882_TARGA] = {
+		.mixers = { alc882_targa_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+				alc880_gpio3_init_verbs, alc882_targa_verbs},
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
+		.adc_nids = alc882_adc_nids,
+		.capsrc_nids = alc882_capsrc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
+		.channel_mode = alc882_3ST_6ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc882_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc882_targa_setup,
+		.init_hook = alc882_targa_automute,
+	},
+	[ALC882_ASUS_A7J] = {
+		.mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+				alc882_asus_a7j_verbs},
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
+		.adc_nids = alc882_adc_nids,
+		.capsrc_nids = alc882_capsrc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
+		.channel_mode = alc882_3ST_6ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc882_capture_source,
+	},
+	[ALC882_ASUS_A7M] = {
+		.mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+				alc882_eapd_verbs, alc880_gpio1_init_verbs,
+				alc882_asus_a7m_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+		.channel_mode = alc880_threestack_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc882_capture_source,
+	},
+	[ALC883_3ST_2ch_DIG] = {
+		.mixers = { alc883_3ST_2ch_mixer },
+		.init_verbs = { alc883_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.input_mux = &alc883_capture_source,
+	},
+	[ALC883_3ST_6ch_DIG] = {
+		.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+		.channel_mode = alc883_3ST_6ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc883_capture_source,
+	},
+	[ALC883_3ST_6ch] = {
+		.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+		.channel_mode = alc883_3ST_6ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc883_capture_source,
+	},
+	[ALC883_3ST_6ch_INTEL] = {
+		.mixers = { alc883_3ST_6ch_intel_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.slave_dig_outs = alc883_slave_dig_outs,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes),
+		.channel_mode = alc883_3ST_6ch_intel_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc883_3stack_6ch_intel,
+	},
+	[ALC889A_INTEL] = {
+		.mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc885_init_verbs, alc885_init_input_verbs,
+				alc_hp15_unsol_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
+		.adc_nids = alc889_adc_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.slave_dig_outs = alc883_slave_dig_outs,
+		.num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes),
+		.channel_mode = alc889_8ch_intel_modes,
+		.capsrc_nids = alc889_capsrc_nids,
+		.input_mux = &alc889_capture_source,
+		.setup = alc889_automute_setup,
+		.init_hook = alc_hp_automute,
+		.unsol_event = alc_sku_unsol_event,
+		.need_dac_fix = 1,
+	},
+	[ALC889_INTEL] = {
+		.mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc885_init_verbs, alc889_init_input_verbs,
+				alc889_eapd_verbs, alc_hp15_unsol_verbs},
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
+		.adc_nids = alc889_adc_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.slave_dig_outs = alc883_slave_dig_outs,
+		.num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes),
+		.channel_mode = alc889_8ch_intel_modes,
+		.capsrc_nids = alc889_capsrc_nids,
+		.input_mux = &alc889_capture_source,
+		.setup = alc889_automute_setup,
+		.init_hook = alc889_intel_init_hook,
+		.unsol_event = alc_sku_unsol_event,
+		.need_dac_fix = 1,
+	},
+	[ALC883_6ST_DIG] = {
+		.mixers = { alc883_base_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+		.channel_mode = alc883_sixstack_modes,
+		.input_mux = &alc883_capture_source,
+	},
+	[ALC883_TARGA_DIG] = {
+		.mixers = { alc883_targa_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
+				alc883_targa_verbs},
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+		.channel_mode = alc883_3ST_6ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc883_targa_unsol_event,
+		.setup = alc882_targa_setup,
+		.init_hook = alc882_targa_automute,
+	},
+	[ALC883_TARGA_2ch_DIG] = {
+		.mixers = { alc883_targa_2ch_mixer},
+		.init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
+				alc883_targa_verbs},
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.adc_nids = alc883_adc_nids_alt,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
+		.capsrc_nids = alc883_capsrc_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc883_targa_unsol_event,
+		.setup = alc882_targa_setup,
+		.init_hook = alc882_targa_automute,
+	},
+	[ALC883_TARGA_8ch_DIG] = {
+		.mixers = { alc883_targa_mixer, alc883_targa_8ch_mixer,
+			    alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
+				alc883_targa_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+		.adc_nids = alc883_adc_nids_rev,
+		.capsrc_nids = alc883_capsrc_nids_rev,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_4ST_8ch_modes),
+		.channel_mode = alc883_4ST_8ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc883_targa_unsol_event,
+		.setup = alc882_targa_setup,
+		.init_hook = alc882_targa_automute,
+	},
+	[ALC883_ACER] = {
+		.mixers = { alc883_base_mixer },
+		/* 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.
+		 */
+		.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.input_mux = &alc883_capture_source,
+	},
+	[ALC883_ACER_ASPIRE] = {
+		.mixers = { alc883_acer_aspire_mixer },
+		.init_verbs = { alc883_init_verbs, alc883_acer_eapd_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc883_acer_aspire_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC888_ACER_ASPIRE_4930G] = {
+		.mixers = { alc888_acer_aspire_4930g_mixer,
+				alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
+				alc888_acer_aspire_4930g_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+		.adc_nids = alc883_adc_nids_rev,
+		.capsrc_nids = alc883_capsrc_nids_rev,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+		.channel_mode = alc883_3ST_6ch_modes,
+		.need_dac_fix = 1,
+		.const_channel_count = 6,
+		.num_mux_defs =
+			ARRAY_SIZE(alc888_2_capture_sources),
+		.input_mux = alc888_2_capture_sources,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc888_acer_aspire_4930g_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC888_ACER_ASPIRE_6530G] = {
+		.mixers = { alc888_acer_aspire_6530_mixer },
+		.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
+				alc888_acer_aspire_6530g_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+		.adc_nids = alc883_adc_nids_rev,
+		.capsrc_nids = alc883_capsrc_nids_rev,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.num_mux_defs =
+			ARRAY_SIZE(alc888_2_capture_sources),
+		.input_mux = alc888_acer_aspire_6530_sources,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc888_acer_aspire_6530g_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC888_ACER_ASPIRE_8930G] = {
+		.mixers = { alc889_acer_aspire_8930g_mixer,
+				alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
+				alc889_acer_aspire_8930g_verbs,
+				alc889_eapd_verbs},
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
+		.adc_nids = alc889_adc_nids,
+		.capsrc_nids = alc889_capsrc_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+		.channel_mode = alc883_3ST_6ch_modes,
+		.need_dac_fix = 1,
+		.const_channel_count = 6,
+		.num_mux_defs =
+			ARRAY_SIZE(alc889_capture_sources),
+		.input_mux = alc889_capture_sources,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc889_acer_aspire_8930g_setup,
+		.init_hook = alc_hp_automute,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+		.power_hook = alc_power_eapd,
+#endif
+	},
+	[ALC888_ACER_ASPIRE_7730G] = {
+		.mixers = { alc883_3ST_6ch_mixer,
+				alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
+				alc888_acer_aspire_7730G_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+		.adc_nids = alc883_adc_nids_rev,
+		.capsrc_nids = alc883_capsrc_nids_rev,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+		.channel_mode = alc883_3ST_6ch_modes,
+		.need_dac_fix = 1,
+		.const_channel_count = 6,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc888_acer_aspire_7730g_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC883_MEDION] = {
+		.mixers = { alc883_fivestack_mixer,
+			    alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs,
+				alc883_medion_eapd_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.adc_nids = alc883_adc_nids_alt,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
+		.capsrc_nids = alc883_capsrc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+		.channel_mode = alc883_sixstack_modes,
+		.input_mux = &alc883_capture_source,
+	},
+	[ALC883_MEDION_WIM2160] = {
+		.mixers = { alc883_medion_wim2160_mixer },
+		.init_verbs = { alc883_init_verbs, alc883_medion_wim2160_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+		.adc_nids = alc883_adc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc883_medion_wim2160_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC883_LAPTOP_EAPD] = {
+		.mixers = { alc883_base_mixer },
+		.init_verbs = { alc883_init_verbs, alc882_eapd_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.input_mux = &alc883_capture_source,
+	},
+	[ALC883_CLEVO_M540R] = {
+		.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs, alc883_clevo_m540r_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_clevo_modes),
+		.channel_mode = alc883_3ST_6ch_clevo_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc883_capture_source,
+		/* This machine has the hardware HP auto-muting, thus
+		 * we need no software mute via unsol event
+		 */
+	},
+	[ALC883_CLEVO_M720] = {
+		.mixers = { alc883_clevo_m720_mixer },
+		.init_verbs = { alc883_init_verbs, alc883_clevo_m720_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc883_clevo_m720_unsol_event,
+		.setup = alc883_clevo_m720_setup,
+		.init_hook = alc883_clevo_m720_init_hook,
+	},
+	[ALC883_LENOVO_101E_2ch] = {
+		.mixers = { alc883_lenovo_101e_2ch_mixer},
+		.init_verbs = { alc883_init_verbs, alc883_lenovo_101e_verbs},
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.adc_nids = alc883_adc_nids_alt,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
+		.capsrc_nids = alc883_capsrc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.input_mux = &alc883_lenovo_101e_capture_source,
+		.setup = alc883_lenovo_101e_setup,
+		.unsol_event = alc_sku_unsol_event,
+		.init_hook = alc_inithook,
+	},
+	[ALC883_LENOVO_NB0763] = {
+		.mixers = { alc883_lenovo_nb0763_mixer },
+		.init_verbs = { alc883_init_verbs, alc883_lenovo_nb0763_verbs},
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc883_lenovo_nb0763_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc883_lenovo_nb0763_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC888_LENOVO_MS7195_DIG] = {
+		.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs, alc888_lenovo_ms7195_verbs},
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+		.channel_mode = alc883_3ST_6ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc888_lenovo_ms7195_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC883_HAIER_W66] = {
+		.mixers = { alc883_targa_2ch_mixer},
+		.init_verbs = { alc883_init_verbs, alc883_haier_w66_verbs},
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc883_haier_w66_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC888_3ST_HP] = {
+		.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc888_3st_hp_modes),
+		.channel_mode = alc888_3st_hp_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc888_3st_hp_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC888_6ST_DELL] = {
+		.mixers = { alc883_base_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs, alc888_6st_dell_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+		.channel_mode = alc883_sixstack_modes,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc888_6st_dell_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC883_MITAC] = {
+		.mixers = { alc883_mitac_mixer },
+		.init_verbs = { alc883_init_verbs, alc883_mitac_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc883_mitac_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC883_FUJITSU_PI2515] = {
+		.mixers = { alc883_2ch_fujitsu_pi2515_mixer },
+		.init_verbs = { alc883_init_verbs,
+				alc883_2ch_fujitsu_pi2515_verbs},
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.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_sku_unsol_event,
+		.setup = alc883_2ch_fujitsu_pi2515_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC888_FUJITSU_XA3530] = {
+		.mixers = { alc888_base_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs,
+			alc888_fujitsu_xa3530_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+		.adc_nids = alc883_adc_nids_rev,
+		.capsrc_nids = alc883_capsrc_nids_rev,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc888_4ST_8ch_intel_modes),
+		.channel_mode = alc888_4ST_8ch_intel_modes,
+		.num_mux_defs =
+			ARRAY_SIZE(alc888_2_capture_sources),
+		.input_mux = alc888_2_capture_sources,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc888_fujitsu_xa3530_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC888_LENOVO_SKY] = {
+		.mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs, alc888_lenovo_sky_verbs},
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+		.channel_mode = alc883_sixstack_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc883_lenovo_sky_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc888_lenovo_sky_setup,
+		.init_hook = alc_hp_automute,
+	},
+	[ALC888_ASUS_M90V] = {
+		.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs, alc888_asus_m90v_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+		.channel_mode = alc883_3ST_6ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc883_fujitsu_pi2515_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc883_mode2_setup,
+		.init_hook = alc_inithook,
+	},
+	[ALC888_ASUS_EEE1601] = {
+		.mixers = { alc883_asus_eee1601_mixer },
+		.cap_mixer = alc883_asus_eee1601_cap_mixer,
+		.init_verbs = { alc883_init_verbs, alc888_asus_eee1601_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc883_asus_eee1601_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.init_hook = alc883_eee1601_inithook,
+	},
+	[ALC1200_ASUS_P5Q] = {
+		.mixers = { alc883_base_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC1200_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.slave_dig_outs = alc1200_slave_dig_outs,
+		.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+		.channel_mode = alc883_sixstack_modes,
+		.input_mux = &alc883_capture_source,
+	},
+	[ALC889A_MB31] = {
+		.mixers = { alc889A_mb31_mixer, alc883_chmode_mixer},
+		.init_verbs = { alc883_init_verbs, alc889A_mb31_verbs,
+			alc880_gpio1_init_verbs },
+		.adc_nids = alc883_adc_nids,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+		.capsrc_nids = alc883_capsrc_nids,
+		.dac_nids = alc883_dac_nids,
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.channel_mode = alc889A_mb31_6ch_modes,
+		.num_channel_mode = ARRAY_SIZE(alc889A_mb31_6ch_modes),
+		.input_mux = &alc889A_mb31_capture_source,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.unsol_event = alc889A_mb31_unsol_event,
+		.init_hook = alc889A_mb31_automute,
+	},
+	[ALC883_SONY_VAIO_TT] = {
+		.mixers = { alc883_vaiott_mixer },
+		.init_verbs = { alc883_init_verbs, alc883_vaiott_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc883_vaiott_setup,
+		.init_hook = alc_hp_automute,
+	},
+};
+
+

sound/pci/hda/alc_quirks.c

@@ -0,0 +1,467 @@
+/*
+ * Common codes for Realtek codec quirks
+ * included by patch_realtek.c
+ */
+
+/*
+ * configuration template - to be copied to the spec instance
+ */
+struct alc_config_preset {
+	const struct snd_kcontrol_new *mixers[5]; /* should be identical size
+					     * with spec
+					     */
+	const struct snd_kcontrol_new *cap_mixer; /* capture mixer */
+	const struct hda_verb *init_verbs[5];
+	unsigned int num_dacs;
+	const hda_nid_t *dac_nids;
+	hda_nid_t dig_out_nid;		/* optional */
+	hda_nid_t hp_nid;		/* optional */
+	const hda_nid_t *slave_dig_outs;
+	unsigned int num_adc_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;
+	int need_dac_fix;
+	int const_channel_count;
+	unsigned int num_mux_defs;
+	const struct hda_input_mux *input_mux;
+	void (*unsol_event)(struct hda_codec *, unsigned int);
+	void (*setup)(struct hda_codec *);
+	void (*init_hook)(struct hda_codec *);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+	const struct hda_amp_list *loopbacks;
+	void (*power_hook)(struct hda_codec *codec);
+#endif
+};
+
+/*
+ * channel mode setting
+ */
+static int alc_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;
+	return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode,
+				    spec->num_channel_mode);
+}
+
+static int alc_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;
+	return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
+				   spec->num_channel_mode,
+				   spec->ext_channel_count);
+}
+
+static int alc_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 err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
+				      spec->num_channel_mode,
+				      &spec->ext_channel_count);
+	if (err >= 0 && !spec->const_channel_count) {
+		spec->multiout.max_channels = spec->ext_channel_count;
+		if (spec->need_dac_fix)
+			spec->multiout.num_dacs = spec->multiout.max_channels / 2;
+	}
+	return err;
+}
+
+/*
+ * Control the mode of pin widget settings via the mixer.  "pc" is used
+ * instead of "%" to avoid consequences of accidentally treating the % as
+ * being part of a format specifier.  Maximum allowed length of a value is
+ * 63 characters plus NULL terminator.
+ *
+ * Note: some retasking pin complexes seem to ignore requests for input
+ * states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these
+ * are requested.  Therefore order this list so that this behaviour will not
+ * cause problems when mixer clients move through the enum sequentially.
+ * NIDs 0x0f and 0x10 have been observed to have this behaviour as of
+ * March 2006.
+ */
+static const char * const alc_pin_mode_names[] = {
+	"Mic 50pc bias", "Mic 80pc bias",
+	"Line in", "Line out", "Headphone out",
+};
+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
+ * in the pin being assumed to be exclusively an input or an output pin.  In
+ * addition, "input" pins may or may not process the mic bias option
+ * depending on actual widget capability (NIDs 0x0f and 0x10 don't seem to
+ * accept requests for bias as of chip versions up to March 2006) and/or
+ * wiring in the computer.
+ */
+#define ALC_PIN_DIR_IN              0x00
+#define ALC_PIN_DIR_OUT             0x01
+#define ALC_PIN_DIR_INOUT           0x02
+#define ALC_PIN_DIR_IN_NOMICBIAS    0x03
+#define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04
+
+/* Info about the pin modes supported by the different pin direction modes.
+ * For each direction the minimum and maximum values are given.
+ */
+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 */
+	{ 2, 2 },    /* ALC_PIN_DIR_IN_NOMICBIAS */
+	{ 2, 4 },    /* ALC_PIN_DIR_INOUT_NOMICBIAS */
+};
+#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0])
+#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1])
+#define alc_pin_mode_n_items(_dir) \
+	(alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1)
+
+static int alc_pin_mode_info(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_info *uinfo)
+{
+	unsigned int item_num = uinfo->value.enumerated.item;
+	unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = alc_pin_mode_n_items(dir);
+
+	if (item_num<alc_pin_mode_min(dir) || item_num>alc_pin_mode_max(dir))
+		item_num = alc_pin_mode_min(dir);
+	strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]);
+	return 0;
+}
+
+static int alc_pin_mode_get(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	unsigned int i;
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = kcontrol->private_value & 0xffff;
+	unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
+	long *valp = ucontrol->value.integer.value;
+	unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
+						 AC_VERB_GET_PIN_WIDGET_CONTROL,
+						 0x00);
+
+	/* Find enumerated value for current pinctl setting */
+	i = alc_pin_mode_min(dir);
+	while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl)
+		i++;
+	*valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
+	return 0;
+}
+
+static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	signed int change;
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = kcontrol->private_value & 0xffff;
+	unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
+	long val = *ucontrol->value.integer.value;
+	unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
+						 AC_VERB_GET_PIN_WIDGET_CONTROL,
+						 0x00);
+
+	if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir))
+		val = alc_pin_mode_min(dir);
+
+	change = pinctl != alc_pin_mode_values[val];
+	if (change) {
+		/* Set pin mode to that requested */
+		snd_hda_codec_write_cache(codec, nid, 0,
+					  AC_VERB_SET_PIN_WIDGET_CONTROL,
+					  alc_pin_mode_values[val]);
+
+		/* Also enable the retasking pin's input/output as required
+		 * for the requested pin mode.  Enum values of 2 or less are
+		 * input modes.
+		 *
+		 * Dynamically switching the input/output buffers probably
+		 * reduces noise slightly (particularly on input) so we'll
+		 * do it.  However, having both input and output buffers
+		 * enabled simultaneously doesn't seem to be problematic if
+		 * this turns out to be necessary in the future.
+		 */
+		if (val <= 2) {
+			snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+						 HDA_AMP_MUTE, HDA_AMP_MUTE);
+			snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
+						 HDA_AMP_MUTE, 0);
+		} else {
+			snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
+						 HDA_AMP_MUTE, HDA_AMP_MUTE);
+			snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+						 HDA_AMP_MUTE, 0);
+		}
+	}
+	return change;
+}
+
+#define ALC_PIN_MODE(xname, nid, dir) \
+	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0,  \
+	  .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
+	  .info = alc_pin_mode_info, \
+	  .get = alc_pin_mode_get, \
+	  .put = alc_pin_mode_put, \
+	  .private_value = nid | (dir<<16) }
+
+/* A switch control for ALC260 GPIO pins.  Multiple GPIOs can be ganged
+ * together using a mask with more than one bit set.  This control is
+ * currently used only by the ALC260 test model.  At this stage they are not
+ * needed for any "production" models.
+ */
+#ifdef CONFIG_SND_DEBUG
+#define alc_gpio_data_info	snd_ctl_boolean_mono_info
+
+static int alc_gpio_data_get(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = kcontrol->private_value & 0xffff;
+	unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+	long *valp = ucontrol->value.integer.value;
+	unsigned int val = snd_hda_codec_read(codec, nid, 0,
+					      AC_VERB_GET_GPIO_DATA, 0x00);
+
+	*valp = (val & mask) != 0;
+	return 0;
+}
+static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	signed int change;
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = kcontrol->private_value & 0xffff;
+	unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+	long val = *ucontrol->value.integer.value;
+	unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0,
+						    AC_VERB_GET_GPIO_DATA,
+						    0x00);
+
+	/* Set/unset the masked GPIO bit(s) as needed */
+	change = (val == 0 ? 0 : mask) != (gpio_data & mask);
+	if (val == 0)
+		gpio_data &= ~mask;
+	else
+		gpio_data |= mask;
+	snd_hda_codec_write_cache(codec, nid, 0,
+				  AC_VERB_SET_GPIO_DATA, gpio_data);
+
+	return change;
+}
+#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \
+	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0,  \
+	  .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
+	  .info = alc_gpio_data_info, \
+	  .get = alc_gpio_data_get, \
+	  .put = alc_gpio_data_put, \
+	  .private_value = nid | (mask<<16) }
+#endif   /* CONFIG_SND_DEBUG */
+
+/* A switch control to allow the enabling of the digital IO pins on the
+ * ALC260.  This is incredibly simplistic; the intention of this control is
+ * to provide something in the test model allowing digital outputs to be
+ * identified if present.  If models are found which can utilise these
+ * outputs a more complete mixer control can be devised for those models if
+ * necessary.
+ */
+#ifdef CONFIG_SND_DEBUG
+#define alc_spdif_ctrl_info	snd_ctl_boolean_mono_info
+
+static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = kcontrol->private_value & 0xffff;
+	unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+	long *valp = ucontrol->value.integer.value;
+	unsigned int val = snd_hda_codec_read(codec, nid, 0,
+					      AC_VERB_GET_DIGI_CONVERT_1, 0x00);
+
+	*valp = (val & mask) != 0;
+	return 0;
+}
+static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	signed int change;
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = kcontrol->private_value & 0xffff;
+	unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+	long val = *ucontrol->value.integer.value;
+	unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
+						    AC_VERB_GET_DIGI_CONVERT_1,
+						    0x00);
+
+	/* Set/unset the masked control bit(s) as needed */
+	change = (val == 0 ? 0 : mask) != (ctrl_data & mask);
+	if (val==0)
+		ctrl_data &= ~mask;
+	else
+		ctrl_data |= mask;
+	snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
+				  ctrl_data);
+
+	return change;
+}
+#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \
+	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0,  \
+	  .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
+	  .info = alc_spdif_ctrl_info, \
+	  .get = alc_spdif_ctrl_get, \
+	  .put = alc_spdif_ctrl_put, \
+	  .private_value = nid | (mask<<16) }
+#endif   /* CONFIG_SND_DEBUG */
+
+/* A switch control to allow the enabling EAPD digital outputs on the ALC26x.
+ * Again, this is only used in the ALC26x test models to help identify when
+ * the EAPD line must be asserted for features to work.
+ */
+#ifdef CONFIG_SND_DEBUG
+#define alc_eapd_ctrl_info	snd_ctl_boolean_mono_info
+
+static int alc_eapd_ctrl_get(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = kcontrol->private_value & 0xffff;
+	unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+	long *valp = ucontrol->value.integer.value;
+	unsigned int val = snd_hda_codec_read(codec, nid, 0,
+					      AC_VERB_GET_EAPD_BTLENABLE, 0x00);
+
+	*valp = (val & mask) != 0;
+	return 0;
+}
+
+static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	int change;
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = kcontrol->private_value & 0xffff;
+	unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+	long val = *ucontrol->value.integer.value;
+	unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
+						    AC_VERB_GET_EAPD_BTLENABLE,
+						    0x00);
+
+	/* Set/unset the masked control bit(s) as needed */
+	change = (!val ? 0 : mask) != (ctrl_data & mask);
+	if (!val)
+		ctrl_data &= ~mask;
+	else
+		ctrl_data |= mask;
+	snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
+				  ctrl_data);
+
+	return change;
+}
+
+#define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \
+	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0,  \
+	  .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
+	  .info = alc_eapd_ctrl_info, \
+	  .get = alc_eapd_ctrl_get, \
+	  .put = alc_eapd_ctrl_put, \
+	  .private_value = nid | (mask<<16) }
+#endif   /* CONFIG_SND_DEBUG */
+
+static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+
+	if (!cfg->line_outs) {
+		while (cfg->line_outs < AUTO_CFG_MAX_OUTS &&
+		       cfg->line_out_pins[cfg->line_outs])
+			cfg->line_outs++;
+	}
+	if (!cfg->speaker_outs) {
+		while (cfg->speaker_outs < AUTO_CFG_MAX_OUTS &&
+		       cfg->speaker_pins[cfg->speaker_outs])
+			cfg->speaker_outs++;
+	}
+	if (!cfg->hp_outs) {
+		while (cfg->hp_outs < AUTO_CFG_MAX_OUTS &&
+		       cfg->hp_pins[cfg->hp_outs])
+			cfg->hp_outs++;
+	}
+}
+
+/*
+ * set up from the preset table
+ */
+static void setup_preset(struct hda_codec *codec,
+			 const struct alc_config_preset *preset)
+{
+	struct alc_spec *spec = codec->spec;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
+		add_mixer(spec, preset->mixers[i]);
+	spec->cap_mixer = preset->cap_mixer;
+	for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i];
+	     i++)
+		add_verb(spec, preset->init_verbs[i]);
+
+	spec->channel_mode = preset->channel_mode;
+	spec->num_channel_mode = preset->num_channel_mode;
+	spec->need_dac_fix = preset->need_dac_fix;
+	spec->const_channel_count = preset->const_channel_count;
+
+	if (preset->const_channel_count)
+		spec->multiout.max_channels = preset->const_channel_count;
+	else
+		spec->multiout.max_channels = spec->channel_mode[0].channels;
+	spec->ext_channel_count = spec->channel_mode[0].channels;
+
+	spec->multiout.num_dacs = preset->num_dacs;
+	spec->multiout.dac_nids = preset->dac_nids;
+	spec->multiout.dig_out_nid = preset->dig_out_nid;
+	spec->multiout.slave_dig_outs = preset->slave_dig_outs;
+	spec->multiout.hp_nid = preset->hp_nid;
+
+	spec->num_mux_defs = preset->num_mux_defs;
+	if (!spec->num_mux_defs)
+		spec->num_mux_defs = 1;
+	spec->input_mux = preset->input_mux;
+
+	spec->num_adc_nids = preset->num_adc_nids;
+	spec->adc_nids = preset->adc_nids;
+	spec->capsrc_nids = preset->capsrc_nids;
+	spec->dig_in_nid = preset->dig_in_nid;
+
+	spec->unsol_event = preset->unsol_event;
+	spec->init_hook = preset->init_hook;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+	spec->power_hook = preset->power_hook;
+	spec->loopback.amplist = preset->loopbacks;
+#endif
+
+	if (preset->setup)
+		preset->setup(codec);
+
+	alc_fixup_autocfg_pin_nums(codec);
+}
+
+
+/* auto-toggle front mic */
+static void alc88x_simple_mic_automute(struct hda_codec *codec)
+{
+ 	unsigned int present;
+	unsigned char bits;
+
+	present = snd_hda_jack_detect(codec, 0x18);
+	bits = present ? HDA_AMP_MUTE : 0;
+	snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
+}
+

sound/pci/hda/hda_codec.c

@@ -243,7 +243,8 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
 {
 	unsigned cmd = make_codec_cmd(codec, nid, direct, verb, parm);
 	unsigned int res;
-	codec_exec_verb(codec, cmd, &res);
+	if (codec_exec_verb(codec, cmd, &res))
+		return -1;
 	return res;
 }
 EXPORT_SYMBOL_HDA(snd_hda_codec_read);
@@ -307,63 +308,107 @@ 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);
+/* look up the cached results */
+static hda_nid_t *lookup_conn_list(struct snd_array *array, hda_nid_t nid)
+{
+	int i, len;
+	for (i = 0; i < array->used; ) {
+		hda_nid_t *p = snd_array_elem(array, i);
+		if (nid == *p)
+			return p;
+		len = p[1];
+		i += len + 2;
+	}
+	return NULL;
+}
 
 /**
- * snd_hda_get_connections - get connection list
+ * snd_hda_get_conn_list - get connection list
  * @codec: the HDA codec
  * @nid: NID to parse
- * @conn_list: connection list array
- * @max_conns: max. number of connections to store
+ * @listp: the pointer to store NID list
  *
  * Parses the connection list of the given widget and stores the list
  * of NIDs.
  *
  * 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)
+int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
+			  const hda_nid_t **listp)
 {
 	struct snd_array *array = &codec->conn_lists;
-	int i, len, old_used;
+	int len, err;
 	hda_nid_t list[HDA_MAX_CONNECTIONS];
+	hda_nid_t *p;
+	bool added = false;
 
-	/* 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;
+ again:
+	/* if the connection-list is already cached, read it */
+	p = lookup_conn_list(array, nid);
+	if (p) {
+		if (listp)
+			*listp = p + 2;
+		return p[1];
 	}
+	if (snd_BUG_ON(added))
+		return -EINVAL;
 
-	len = _hda_get_connections(codec, nid, list, HDA_MAX_CONNECTIONS);
+	/* read the connection and add to the cache */
+	len = snd_hda_get_raw_connections(codec, nid, list, HDA_MAX_CONNECTIONS);
 	if (len < 0)
 		return len;
+	err = snd_hda_override_conn_list(codec, nid, len, list);
+	if (err < 0)
+		return err;
+	added = true;
+	goto again;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_conn_list);
 
-	/* 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;
+/**
+ * snd_hda_get_connections - copy connection list
+ * @codec: the HDA codec
+ * @nid: NID to parse
+ * @conn_list: connection list array
+ * @max_conns: max. number of connections to store
+ *
+ * Parses the connection list of the given widget and stores the list
+ * of NIDs.
+ *
+ * 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)
+{
+	const hda_nid_t *list;
+	int len = snd_hda_get_conn_list(codec, nid, &list);
 
-	return copy_conn_list(nid, conn_list, max_conns, list, len);
-		
- error_add:
-	array->used = old_used;
-	return -ENOMEM;
+	if (len <= 0)
+		return len;
+	if (len > max_conns) {
+		snd_printk(KERN_ERR "hda_codec: "
+			   "Too many connections %d for NID 0x%x\n",
+			   len, nid);
+		return -EINVAL;
+	}
+	memcpy(conn_list, list, len * sizeof(hda_nid_t));
+	return len;
 }
 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)
+/**
+ * snd_hda_get_raw_connections - copy connection list without cache
+ * @codec: the HDA codec
+ * @nid: NID to parse
+ * @conn_list: connection list array
+ * @max_conns: max. number of connections to store
+ *
+ * Like snd_hda_get_connections(), copy the connection list but without
+ * checking through the connection-list cache.
+ * Currently called only from hda_proc.c, so not exported.
+ */
+int snd_hda_get_raw_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;
@@ -376,11 +421,8 @@ static int _hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
 
 	wcaps = get_wcaps(codec, nid);
 	if (!(wcaps & AC_WCAP_CONN_LIST) &&
-	    get_wcaps_type(wcaps) != AC_WID_VOL_KNB) {
-		snd_printk(KERN_WARNING "hda_codec: "
-			   "connection list not available for 0x%x\n", nid);
-		return -EINVAL;
-	}
+	    get_wcaps_type(wcaps) != AC_WID_VOL_KNB)
+		return 0;
 
 	parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN);
 	if (parm & AC_CLIST_LONG) {
@@ -470,18 +512,77 @@ static bool add_conn_list(struct snd_array *array, hda_nid_t 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)
+/**
+ * snd_hda_override_conn_list - add/modify the connection-list to cache
+ * @codec: the HDA codec
+ * @nid: NID to parse
+ * @len: number of connection list entries
+ * @list: the list of connection entries
+ *
+ * Add or modify the given connection-list to the cache.  If the corresponding
+ * cache already exists, invalidate it and append a new one.
+ *
+ * Returns zero or a negative error code.
+ */
+int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int len,
+			       const hda_nid_t *list)
 {
-	if (len > max_dst) {
-		snd_printk(KERN_ERR "hda_codec: "
-			   "Too many connections %d for NID 0x%x\n",
-			   len, nid);
-		return -EINVAL;
+	struct snd_array *array = &codec->conn_lists;
+	hda_nid_t *p;
+	int i, old_used;
+
+	p = lookup_conn_list(array, nid);
+	if (p)
+		*p = -1; /* invalidate the old entry */
+
+	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 0;
+
+ error_add:
+	array->used = old_used;
+	return -ENOMEM;
+}
+EXPORT_SYMBOL_HDA(snd_hda_override_conn_list);
+
+/**
+ * snd_hda_get_conn_index - get the connection index of the given NID
+ * @codec: the HDA codec
+ * @mux: NID containing the list
+ * @nid: NID to select
+ * @recursive: 1 when searching NID recursively, otherwise 0
+ *
+ * Parses the connection list of the widget @mux and checks whether the
+ * widget @nid is present.  If it is, return the connection index.
+ * Otherwise it returns -1.
+ */
+int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
+			   hda_nid_t nid, int recursive)
+{
+	hda_nid_t conn[HDA_MAX_NUM_INPUTS];
+	int i, nums;
+
+	nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
+	for (i = 0; i < nums; i++)
+		if (conn[i] == nid)
+			return i;
+	if (!recursive)
+		return -1;
+	if (recursive > 5) {
+		snd_printd("hda_codec: too deep connection for 0x%x\n", nid);
+		return -1;
 	}
-	memcpy(dst, src, len * sizeof(hda_nid_t));
-	return len;
+	recursive++;
+	for (i = 0; i < nums; i++)
+		if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0)
+			return i;
+	return -1;
 }
+EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
 
 /**
  * snd_hda_queue_unsol_event - add an unsolicited event to queue
@@ -1083,6 +1184,7 @@ static void snd_hda_codec_free(struct hda_codec *codec)
 	snd_array_free(&codec->mixers);
 	snd_array_free(&codec->nids);
 	snd_array_free(&codec->conn_lists);
+	snd_array_free(&codec->spdif_out);
 	codec->bus->caddr_tbl[codec->addr] = NULL;
 	if (codec->patch_ops.free)
 		codec->patch_ops.free(codec);
@@ -1144,6 +1246,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus,
 	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);
+	snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16);
 	if (codec->bus->modelname) {
 		codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL);
 		if (!codec->modelname) {
@@ -2555,11 +2658,13 @@ static int snd_hda_spdif_default_get(struct snd_kcontrol *kcontrol,
 				     struct snd_ctl_elem_value *ucontrol)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	int idx = kcontrol->private_value;
+	struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
 
-	ucontrol->value.iec958.status[0] = codec->spdif_status & 0xff;
-	ucontrol->value.iec958.status[1] = (codec->spdif_status >> 8) & 0xff;
-	ucontrol->value.iec958.status[2] = (codec->spdif_status >> 16) & 0xff;
-	ucontrol->value.iec958.status[3] = (codec->spdif_status >> 24) & 0xff;
+	ucontrol->value.iec958.status[0] = spdif->status & 0xff;
+	ucontrol->value.iec958.status[1] = (spdif->status >> 8) & 0xff;
+	ucontrol->value.iec958.status[2] = (spdif->status >> 16) & 0xff;
+	ucontrol->value.iec958.status[3] = (spdif->status >> 24) & 0xff;
 
 	return 0;
 }
@@ -2644,23 +2749,23 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol,
 				     struct snd_ctl_elem_value *ucontrol)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	hda_nid_t nid = kcontrol->private_value;
+	int idx = kcontrol->private_value;
+	struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+	hda_nid_t nid = spdif->nid;
 	unsigned short val;
 	int change;
 
 	mutex_lock(&codec->spdif_mutex);
-	codec->spdif_status = ucontrol->value.iec958.status[0] |
+	spdif->status = ucontrol->value.iec958.status[0] |
 		((unsigned int)ucontrol->value.iec958.status[1] << 8) |
 		((unsigned int)ucontrol->value.iec958.status[2] << 16) |
 		((unsigned int)ucontrol->value.iec958.status[3] << 24);
-	val = convert_from_spdif_status(codec->spdif_status);
-	val |= codec->spdif_ctls & 1;
-	change = codec->spdif_ctls != val;
-	codec->spdif_ctls = val;
-
-	if (change)
+	val = convert_from_spdif_status(spdif->status);
+	val |= spdif->ctls & 1;
+	change = spdif->ctls != val;
+	spdif->ctls = val;
+	if (change && nid != (u16)-1)
 		set_dig_out_convert(codec, nid, val & 0xff, (val >> 8) & 0xff);
-
 	mutex_unlock(&codec->spdif_mutex);
 	return change;
 }
@@ -2671,33 +2776,42 @@ static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol,
 					struct snd_ctl_elem_value *ucontrol)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	int idx = kcontrol->private_value;
+	struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
 
-	ucontrol->value.integer.value[0] = codec->spdif_ctls & AC_DIG1_ENABLE;
+	ucontrol->value.integer.value[0] = spdif->ctls & AC_DIG1_ENABLE;
 	return 0;
 }
 
+static inline void set_spdif_ctls(struct hda_codec *codec, hda_nid_t nid,
+				  int dig1, int dig2)
+{
+	set_dig_out_convert(codec, nid, dig1, dig2);
+	/* unmute amp switch (if any) */
+	if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) &&
+	    (dig1 & AC_DIG1_ENABLE))
+		snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+					    HDA_AMP_MUTE, 0);
+}
+
 static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol,
 					struct snd_ctl_elem_value *ucontrol)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	hda_nid_t nid = kcontrol->private_value;
+	int idx = kcontrol->private_value;
+	struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+	hda_nid_t nid = spdif->nid;
 	unsigned short val;
 	int change;
 
 	mutex_lock(&codec->spdif_mutex);
-	val = codec->spdif_ctls & ~AC_DIG1_ENABLE;
+	val = spdif->ctls & ~AC_DIG1_ENABLE;
 	if (ucontrol->value.integer.value[0])
 		val |= AC_DIG1_ENABLE;
-	change = codec->spdif_ctls != val;
-	if (change) {
-		codec->spdif_ctls = val;
-		set_dig_out_convert(codec, nid, val & 0xff, -1);
-		/* unmute amp switch (if any) */
-		if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) &&
-		    (val & AC_DIG1_ENABLE))
-			snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
-						 HDA_AMP_MUTE, 0);
-	}
+	change = spdif->ctls != val;
+	spdif->ctls = val;
+	if (change && nid != (u16)-1)
+		set_spdif_ctls(codec, nid, val & 0xff, -1);
 	mutex_unlock(&codec->spdif_mutex);
 	return change;
 }
@@ -2744,36 +2858,79 @@ static struct snd_kcontrol_new dig_mixes[] = {
  *
  * Returns 0 if successful, or a negative error code.
  */
-int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid)
+int snd_hda_create_spdif_out_ctls(struct hda_codec *codec,
+				  hda_nid_t associated_nid,
+				  hda_nid_t cvt_nid)
 {
 	int err;
 	struct snd_kcontrol *kctl;
 	struct snd_kcontrol_new *dig_mix;
 	int idx;
+	struct hda_spdif_out *spdif;
 
 	idx = find_empty_mixer_ctl_idx(codec, "IEC958 Playback Switch");
 	if (idx < 0) {
 		printk(KERN_ERR "hda_codec: too many IEC958 outputs\n");
 		return -EBUSY;
 	}
+	spdif = snd_array_new(&codec->spdif_out);
 	for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) {
 		kctl = snd_ctl_new1(dig_mix, codec);
 		if (!kctl)
 			return -ENOMEM;
 		kctl->id.index = idx;
-		kctl->private_value = nid;
-		err = snd_hda_ctl_add(codec, nid, kctl);
+		kctl->private_value = codec->spdif_out.used - 1;
+		err = snd_hda_ctl_add(codec, associated_nid, kctl);
 		if (err < 0)
 			return err;
 	}
-	codec->spdif_ctls =
-		snd_hda_codec_read(codec, nid, 0,
-				   AC_VERB_GET_DIGI_CONVERT_1, 0);
-	codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls);
+	spdif->nid = cvt_nid;
+	spdif->ctls = snd_hda_codec_read(codec, cvt_nid, 0,
+					 AC_VERB_GET_DIGI_CONVERT_1, 0);
+	spdif->status = convert_to_spdif_status(spdif->ctls);
 	return 0;
 }
 EXPORT_SYMBOL_HDA(snd_hda_create_spdif_out_ctls);
 
+struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
+					       hda_nid_t nid)
+{
+	int i;
+	for (i = 0; i < codec->spdif_out.used; i++) {
+		struct hda_spdif_out *spdif =
+				snd_array_elem(&codec->spdif_out, i);
+		if (spdif->nid == nid)
+			return spdif;
+	}
+	return NULL;
+}
+EXPORT_SYMBOL_HDA(snd_hda_spdif_out_of_nid);
+
+void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx)
+{
+	struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+
+	mutex_lock(&codec->spdif_mutex);
+	spdif->nid = (u16)-1;
+	mutex_unlock(&codec->spdif_mutex);
+}
+EXPORT_SYMBOL_HDA(snd_hda_spdif_ctls_unassign);
+
+void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid)
+{
+	struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+	unsigned short val;
+
+	mutex_lock(&codec->spdif_mutex);
+	if (spdif->nid != nid) {
+		spdif->nid = nid;
+		val = spdif->ctls;
+		set_spdif_ctls(codec, nid, val & 0xff, (val >> 8) & 0xff);
+	}
+	mutex_unlock(&codec->spdif_mutex);
+}
+EXPORT_SYMBOL_HDA(snd_hda_spdif_ctls_assign);
+
 /*
  * SPDIF sharing with analog output
  */
@@ -3356,7 +3513,7 @@ static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid)
  *
  * Returns 0 if successful, otherwise a negative error code.
  */
-static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
+int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 				u32 *ratesp, u64 *formatsp, unsigned int *bpsp)
 {
 	unsigned int i, val, wcaps;
@@ -3448,6 +3605,7 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 
 	return 0;
 }
+EXPORT_SYMBOL_HDA(snd_hda_query_supported_pcm);
 
 /**
  * snd_hda_is_supported_format - Check the validity of the format
@@ -4177,10 +4335,12 @@ EXPORT_SYMBOL_HDA(snd_hda_input_mux_put);
 static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid,
 				 unsigned int stream_tag, unsigned int format)
 {
+	struct hda_spdif_out *spdif = snd_hda_spdif_out_of_nid(codec, nid);
+
 	/* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
-	if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
+	if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE))
 		set_dig_out_convert(codec, nid,
-				    codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff,
+				    spdif->ctls & ~AC_DIG1_ENABLE & 0xff,
 				    -1);
 	snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
 	if (codec->slave_dig_outs) {
@@ -4190,9 +4350,9 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid,
 						   format);
 	}
 	/* turn on again (if needed) */
-	if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
+	if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE))
 		set_dig_out_convert(codec, nid,
-				    codec->spdif_ctls & 0xff, -1);
+				    spdif->ctls & 0xff, -1);
 }
 
 static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid)
@@ -4348,6 +4508,8 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
 {
 	const hda_nid_t *nids = mout->dac_nids;
 	int chs = substream->runtime->channels;
+	struct hda_spdif_out *spdif =
+			snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid);
 	int i;
 
 	mutex_lock(&codec->spdif_mutex);
@@ -4356,7 +4518,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
 		if (chs == 2 &&
 		    snd_hda_is_supported_format(codec, mout->dig_out_nid,
 						format) &&
-		    !(codec->spdif_status & IEC958_AES0_NONAUDIO)) {
+		    !(spdif->status & IEC958_AES0_NONAUDIO)) {
 			mout->dig_out_used = HDA_DIG_ANALOG_DUP;
 			setup_dig_out_stream(codec, mout->dig_out_nid,
 					     stream_tag, format);
@@ -4528,7 +4690,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 		unsigned int wid_caps = get_wcaps(codec, nid);
 		unsigned int wid_type = get_wcaps_type(wid_caps);
 		unsigned int def_conf;
-		short assoc, loc;
+		short assoc, loc, conn, dev;
 
 		/* read all default configuration for pin complex */
 		if (wid_type != AC_WID_PIN)
@@ -4538,10 +4700,19 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 			continue;
 
 		def_conf = snd_hda_codec_get_pincfg(codec, nid);
-		if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
+		conn = get_defcfg_connect(def_conf);
+		if (conn == AC_JACK_PORT_NONE)
 			continue;
 		loc = get_defcfg_location(def_conf);
-		switch (get_defcfg_device(def_conf)) {
+		dev = get_defcfg_device(def_conf);
+
+		/* workaround for buggy BIOS setups */
+		if (dev == AC_JACK_LINE_OUT) {
+			if (conn == AC_JACK_PORT_FIXED)
+				dev = AC_JACK_SPEAKER;
+		}
+
+		switch (dev) {
 		case AC_JACK_LINE_OUT:
 			seq = get_defcfg_sequence(def_conf);
 			assoc = get_defcfg_association(def_conf);
@@ -4957,17 +5128,15 @@ void *snd_array_new(struct snd_array *array)
 {
 	if (array->used >= array->alloced) {
 		int num = array->alloced + array->alloc_align;
+		int size = (num + 1) * array->elem_size;
+		int oldsize = array->alloced * array->elem_size;
 		void *nlist;
 		if (snd_BUG_ON(num >= 4096))
 			return NULL;
-		nlist = kcalloc(num + 1, array->elem_size, GFP_KERNEL);
+		nlist = krealloc(array->list, size, GFP_KERNEL);
 		if (!nlist)
 			return NULL;
-		if (array->list) {
-			memcpy(nlist, array->list,
-			       array->elem_size * array->alloced);
-			kfree(array->list);
-		}
+		memset(nlist + oldsize, 0, size - oldsize);
 		array->list = nlist;
 		array->alloced = num;
 	}

sound/pci/hda/hda_codec.h

@@ -829,8 +829,7 @@ struct hda_codec {
 
 	struct mutex spdif_mutex;
 	struct mutex control_mutex;
-	unsigned int spdif_status;	/* IEC958 status bits */
-	unsigned short spdif_ctls;	/* SPDIF control bits */
+	struct snd_array spdif_out;
 	unsigned int spdif_in_enable;	/* SPDIF input enable? */
 	const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
 	struct snd_array init_pins;	/* initial (BIOS) pin configurations */
@@ -904,6 +903,16 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
 			  hda_nid_t *start_id);
 int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
 			    hda_nid_t *conn_list, int max_conns);
+int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
+			    hda_nid_t *conn_list, int max_conns);
+int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
+			  const hda_nid_t **listp);
+int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums,
+			  const hda_nid_t *list);
+int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
+			   hda_nid_t nid, int recursive);
+int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
+				u32 *ratesp, u64 *formatsp, unsigned int *bpsp);
 
 struct hda_verb {
 	hda_nid_t nid;
@@ -947,6 +956,17 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list,
 		       hda_nid_t nid, unsigned int cfg); /* for hwdep */
 void snd_hda_shutup_pins(struct hda_codec *codec);
 
+/* SPDIF controls */
+struct hda_spdif_out {
+	hda_nid_t nid;		/* Converter nid values relate to */
+	unsigned int status;	/* IEC958 status bits */
+	unsigned short ctls;	/* SPDIF control bits */
+};
+struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
+					       hda_nid_t nid);
+void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx);
+void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid);
+
 /*
  * Mixer
  */
@@ -997,17 +1017,15 @@ int snd_hda_suspend(struct hda_bus *bus);
 int snd_hda_resume(struct hda_bus *bus);
 #endif
 
-#ifdef CONFIG_SND_HDA_POWER_SAVE
 static inline
 int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid)
 {
+#ifdef CONFIG_SND_HDA_POWER_SAVE
 	if (codec->patch_ops.check_power_status)
 		return codec->patch_ops.check_power_status(codec, nid);
+#endif
 	return 0;
 }
-#else	
-#define hda_call_check_power_status(codec, nid)		0
-#endif
 
 /*
  * get widget information

sound/pci/hda/hda_eld.c

@@ -580,43 +580,45 @@ void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld)
 #endif /* CONFIG_PROC_FS */
 
 /* update PCM info based on ELD */
-void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm,
-			      struct hda_pcm_stream *codec_pars)
+void snd_hdmi_eld_update_pcm_info(struct hdmi_eld *eld,
+			      struct hda_pcm_stream *hinfo)
 {
+	u32 rates;
+	u64 formats;
+	unsigned int maxbps;
+	unsigned int channels_max;
 	int i;
 
 	/* assume basic audio support (the basic audio flag is not in ELD;
 	 * however, all audio capable sinks are required to support basic
 	 * audio) */
-	pcm->rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000;
-	pcm->formats = SNDRV_PCM_FMTBIT_S16_LE;
-	pcm->maxbps = 16;
-	pcm->channels_max = 2;
+	rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
+		SNDRV_PCM_RATE_48000;
+	formats = SNDRV_PCM_FMTBIT_S16_LE;
+	maxbps = 16;
+	channels_max = 2;
 	for (i = 0; i < eld->sad_count; i++) {
 		struct cea_sad *a = &eld->sad[i];
-		pcm->rates |= a->rates;
-		if (a->channels > pcm->channels_max)
-			pcm->channels_max = a->channels;
+		rates |= a->rates;
+		if (a->channels > channels_max)
+			channels_max = a->channels;
 		if (a->format == AUDIO_CODING_TYPE_LPCM) {
 			if (a->sample_bits & AC_SUPPCM_BITS_20) {
-				pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE;
-				if (pcm->maxbps < 20)
-					pcm->maxbps = 20;
+				formats |= SNDRV_PCM_FMTBIT_S32_LE;
+				if (maxbps < 20)
+					maxbps = 20;
 			}
 			if (a->sample_bits & AC_SUPPCM_BITS_24) {
-				pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE;
-				if (pcm->maxbps < 24)
-					pcm->maxbps = 24;
+				formats |= SNDRV_PCM_FMTBIT_S32_LE;
+				if (maxbps < 24)
+					maxbps = 24;
 			}
 		}
 	}
 
-	if (!codec_pars)
-		return;
-
 	/* restrict the parameters by the values the codec provides */
-	pcm->rates &= codec_pars->rates;
-	pcm->formats &= codec_pars->formats;
-	pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max);
-	pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps);
+	hinfo->rates &= rates;
+	hinfo->formats &= formats;
+	hinfo->maxbps = min(hinfo->maxbps, maxbps);
+	hinfo->channels_max = min(hinfo->channels_max, channels_max);
 }

sound/pci/hda/hda_intel.c

@@ -177,7 +177,8 @@ MODULE_DESCRIPTION("Intel HDA driver");
 #define ICH6_REG_INTCTL			0x20
 #define ICH6_REG_INTSTS			0x24
 #define ICH6_REG_WALLCLK		0x30	/* 24Mhz source */
-#define ICH6_REG_SYNC			0x34	
+#define ICH6_REG_OLD_SSYNC		0x34	/* SSYNC for old ICH */
+#define ICH6_REG_SSYNC			0x38
 #define ICH6_REG_CORBLBASE		0x40
 #define ICH6_REG_CORBUBASE		0x44
 #define ICH6_REG_CORBWP			0x48
@@ -479,6 +480,7 @@ enum {
 #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 */
+#define AZX_DCAPS_OLD_SSYNC	(1 << 20)	/* Old SSYNC reg for ICH */
 
 /* quirks for ATI SB / AMD Hudson */
 #define AZX_DCAPS_PRESET_ATI_SB \
@@ -1706,13 +1708,16 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	unsigned int bufsize, period_bytes, format_val, stream_tag;
 	int err;
+	struct hda_spdif_out *spdif =
+		snd_hda_spdif_out_of_nid(apcm->codec, hinfo->nid);
+	unsigned short ctls = spdif ? spdif->ctls : 0;
 
 	azx_stream_reset(chip, azx_dev);
 	format_val = snd_hda_calc_stream_format(runtime->rate,
 						runtime->channels,
 						runtime->format,
 						hinfo->maxbps,
-						apcm->codec->spdif_ctls);
+						ctls);
 	if (!format_val) {
 		snd_printk(KERN_ERR SFX
 			   "invalid format_val, rate=%d, ch=%d, format=%d\n",
@@ -1792,7 +1797,11 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	spin_lock(&chip->reg_lock);
 	if (nsync > 1) {
 		/* first, set SYNC bits of corresponding streams */
-		azx_writel(chip, SYNC, azx_readl(chip, SYNC) | sbits);
+		if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC)
+			azx_writel(chip, OLD_SSYNC,
+				   azx_readl(chip, OLD_SSYNC) | sbits);
+		else
+			azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) | sbits);
 	}
 	snd_pcm_group_for_each_entry(s, substream) {
 		if (s->pcm->card != substream->pcm->card)
@@ -1848,7 +1857,11 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	if (nsync > 1) {
 		spin_lock(&chip->reg_lock);
 		/* reset SYNC bits */
-		azx_writel(chip, SYNC, azx_readl(chip, SYNC) & ~sbits);
+		if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC)
+			azx_writel(chip, OLD_SSYNC,
+				   azx_readl(chip, OLD_SSYNC) & ~sbits);
+		else
+			azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) & ~sbits);
 		spin_unlock(&chip->reg_lock);
 	}
 	return 0;
@@ -1863,7 +1876,7 @@ static unsigned int azx_via_get_position(struct azx *chip,
 	unsigned int fifo_size;
 
 	link_pos = azx_sd_readl(azx_dev, SD_LPIB);
-	if (azx_dev->index >= 4) {
+	if (azx_dev->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		/* Playback, no problem using link position */
 		return link_pos;
 	}
@@ -1927,6 +1940,17 @@ static unsigned int azx_get_position(struct azx *chip,
 	default:
 		/* use the position buffer */
 		pos = le32_to_cpu(*azx_dev->posbuf);
+		if (chip->position_fix[stream] == POS_FIX_AUTO) {
+			if (!pos || pos == (u32)-1) {
+				printk(KERN_WARNING
+				       "hda-intel: Invalid position buffer, "
+				       "using LPIB read method instead.\n");
+				chip->position_fix[stream] = POS_FIX_LPIB;
+				pos = azx_sd_readl(azx_dev, SD_LPIB);
+			} else
+				chip->position_fix[stream] = POS_FIX_POSBUF;
+		}
+		break;
 	}
 
 	if (pos >= azx_dev->bufsize)
@@ -1964,16 +1988,6 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
 
 	stream = azx_dev->substream->stream;
 	pos = azx_get_position(chip, azx_dev);
-	if (chip->position_fix[stream] == POS_FIX_AUTO) {
-		if (!pos) {
-			printk(KERN_WARNING
-			       "hda-intel: Invalid position buffer, "
-			       "using LPIB read method instead.\n");
-			chip->position_fix[stream] = POS_FIX_LPIB;
-			pos = azx_get_position(chip, azx_dev);
-		} else
-			chip->position_fix[stream] = POS_FIX_POSBUF;
-	}
 
 	if (WARN_ONCE(!azx_dev->period_bytes,
 		      "hda-intel: zero azx_dev->period_bytes"))
@@ -2061,6 +2075,8 @@ static void azx_pcm_free(struct snd_pcm *pcm)
 	}
 }
 
+#define MAX_PREALLOC_SIZE	(32 * 1024 * 1024)
+
 static int
 azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec,
 		      struct hda_pcm *cpcm)
@@ -2069,6 +2085,7 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec,
 	struct snd_pcm *pcm;
 	struct azx_pcm *apcm;
 	int pcm_dev = cpcm->device;
+	unsigned int size;
 	int s, err;
 
 	if (pcm_dev >= HDA_MAX_PCMS) {
@@ -2104,9 +2121,12 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec,
 			snd_pcm_set_ops(pcm, s, &azx_pcm_ops);
 	}
 	/* buffer pre-allocation */
+	size = CONFIG_SND_HDA_PREALLOC_SIZE * 1024;
+	if (size > MAX_PREALLOC_SIZE)
+		size = MAX_PREALLOC_SIZE;
 	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
 					      snd_dma_pci_data(chip->pci),
-					      1024 * 64, 32 * 1024 * 1024);
+					      size, MAX_PREALLOC_SIZE);
 	return 0;
 }
 
@@ -2149,7 +2169,7 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect)
 {
 	if (request_irq(chip->pci->irq, azx_interrupt,
 			chip->msi ? 0 : IRQF_SHARED,
-			"hda_intel", chip)) {
+			KBUILD_MODNAME, chip)) {
 		printk(KERN_ERR "hda-intel: unable to grab IRQ %d, "
 		       "disabling device\n", chip->pci->irq);
 		if (do_disconnect)
@@ -2347,28 +2367,20 @@ static int azx_dev_free(struct snd_device *device)
  * white/black-listing for position_fix
  */
 static struct snd_pci_quirk position_fix_list[] __devinitdata = {
-	SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB),
-	SND_PCI_QUIRK(0x1025, 0x026f, "Acer Aspire 5538", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB),
-	SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB),
-	SND_PCI_QUIRK(0x1028, 0x0470, "Dell Inspiron 1120", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
-	SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
-	SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
-	SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1849, 0x0888, "775Dual-VSTA", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB),
-	SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB),
 	{}
 };
 
@@ -2815,6 +2827,22 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	/* SCH */
 	{ PCI_DEVICE(0x8086, 0x811b),
 	  .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP },
+	{ PCI_DEVICE(0x8086, 0x2668),
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ICH6 */
+	{ PCI_DEVICE(0x8086, 0x27d8),
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ICH7 */
+	{ PCI_DEVICE(0x8086, 0x269a),
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ESB2 */
+	{ PCI_DEVICE(0x8086, 0x284b),
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ICH8 */
+	{ PCI_DEVICE(0x8086, 0x293e),
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ICH9 */
+	{ PCI_DEVICE(0x8086, 0x293f),
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ICH9 */
+	{ PCI_DEVICE(0x8086, 0x3a3e),
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ICH10 */
+	{ PCI_DEVICE(0x8086, 0x3a6e),
+	  .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC },  /* ICH10 */
 	/* Generic Intel */
 	{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID),
 	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
@@ -2908,7 +2936,7 @@ MODULE_DEVICE_TABLE(pci, azx_ids);
 
 /* pci_driver definition */
 static struct pci_driver driver = {
-	.name = "HDA Intel",
+	.name = KBUILD_MODNAME,
 	.id_table = azx_ids,
 	.probe = azx_probe,
 	.remove = __devexit_p(azx_remove),

sound/pci/hda/hda_local.h

@@ -212,7 +212,9 @@ int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 /*
  * SPDIF I/O
  */
-int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid);
+int snd_hda_create_spdif_out_ctls(struct hda_codec *codec,
+				  hda_nid_t associated_nid,
+				  hda_nid_t cvt_nid);
 int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid);
 
 /*
@@ -563,7 +565,6 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
  * power-management
  */
 
-#ifdef CONFIG_SND_HDA_POWER_SAVE
 void snd_hda_schedule_power_save(struct hda_codec *codec);
 
 struct hda_amp_list {
@@ -580,7 +581,6 @@ struct hda_loopback_check {
 int snd_hda_check_amp_list_power(struct hda_codec *codec,
 				 struct hda_loopback_check *check,
 				 hda_nid_t nid);
-#endif /* CONFIG_SND_HDA_POWER_SAVE */
 
 /*
  * AMP control callbacks
@@ -639,8 +639,8 @@ struct hdmi_eld {
 int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid);
 int snd_hdmi_get_eld(struct hdmi_eld *, struct hda_codec *, hda_nid_t);
 void snd_hdmi_show_eld(struct hdmi_eld *eld);
-void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm,
-			      struct hda_pcm_stream *codec_pars);
+void snd_hdmi_eld_update_pcm_info(struct hdmi_eld *eld,
+			      struct hda_pcm_stream *hinfo);
 
 #ifdef CONFIG_PROC_FS
 int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld,

sound/pci/hda/hda_proc.c

@@ -636,7 +636,7 @@ static void print_codec_info(struct snd_info_entry *entry,
 			wid_caps |= AC_WCAP_CONN_LIST;
 
 		if (wid_caps & AC_WCAP_CONN_LIST)
-			conn_len = snd_hda_get_connections(codec, nid, conn,
+			conn_len = snd_hda_get_raw_connections(codec, nid, conn,
 							   HDA_MAX_CONNECTIONS);
 
 		if (wid_caps & AC_WCAP_IN_AMP) {

sound/pci/hda/patch_analog.c

@@ -213,7 +213,9 @@ static int ad198x_build_controls(struct hda_codec *codec)
 			return err;
 	}
 	if (spec->multiout.dig_out_nid) {
-		err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+		err = snd_hda_create_spdif_out_ctls(codec,
+						    spec->multiout.dig_out_nid,
+						    spec->multiout.dig_out_nid);
 		if (err < 0)
 			return err;
 		err = snd_hda_create_spdif_share_sw(codec,
@@ -1920,7 +1922,8 @@ static int patch_ad1981(struct hda_codec *codec)
 		spec->mixers[0] = ad1981_hp_mixers;
 		spec->num_init_verbs = 2;
 		spec->init_verbs[1] = ad1981_hp_init_verbs;
-		spec->multiout.dig_out_nid = 0;
+		if (!is_jack_available(codec, 0x0a))
+			spec->multiout.dig_out_nid = 0;
 		spec->input_mux = &ad1981_hp_capture_source;
 
 		codec->patch_ops.init = ad1981_hp_init;

sound/pci/hda/patch_ca0110.c

@@ -240,7 +240,8 @@ static int ca0110_build_controls(struct hda_codec *codec)
 	}
 
 	if (spec->dig_out) {
-		err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out);
+		err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out,
+						    spec->dig_out);
 		if (err < 0)
 			return err;
 		err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);

sound/pci/hda/patch_ca0132.c

@@ -0,0 +1,1097 @@
+/*
+ * HD audio interface patch for Creative CA0132 chip
+ *
+ * Copyright (c) 2011, Creative Technology Ltd.
+ *
+ * Based on patch_ca0110.c
+ * Copyright (c) 2008 Takashi Iwai <tiwai@suse.de>
+ *
+ *  This driver 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 driver is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+
+#define WIDGET_CHIP_CTRL      0x15
+#define WIDGET_DSP_CTRL       0x16
+
+#define WUH_MEM_CONNID        10
+#define DSP_MEM_CONNID        16
+
+enum hda_cmd_vendor_io {
+	/* for DspIO node */
+	VENDOR_DSPIO_SCP_WRITE_DATA_LOW      = 0x000,
+	VENDOR_DSPIO_SCP_WRITE_DATA_HIGH     = 0x100,
+
+	VENDOR_DSPIO_STATUS                  = 0xF01,
+	VENDOR_DSPIO_SCP_POST_READ_DATA      = 0x702,
+	VENDOR_DSPIO_SCP_READ_DATA           = 0xF02,
+	VENDOR_DSPIO_DSP_INIT                = 0x703,
+	VENDOR_DSPIO_SCP_POST_COUNT_QUERY    = 0x704,
+	VENDOR_DSPIO_SCP_READ_COUNT          = 0xF04,
+
+	/* for ChipIO node */
+	VENDOR_CHIPIO_ADDRESS_LOW            = 0x000,
+	VENDOR_CHIPIO_ADDRESS_HIGH           = 0x100,
+	VENDOR_CHIPIO_STREAM_FORMAT          = 0x200,
+	VENDOR_CHIPIO_DATA_LOW               = 0x300,
+	VENDOR_CHIPIO_DATA_HIGH              = 0x400,
+
+	VENDOR_CHIPIO_GET_PARAMETER          = 0xF00,
+	VENDOR_CHIPIO_STATUS                 = 0xF01,
+	VENDOR_CHIPIO_HIC_POST_READ          = 0x702,
+	VENDOR_CHIPIO_HIC_READ_DATA          = 0xF03,
+
+	VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE   = 0x70A,
+
+	VENDOR_CHIPIO_PLL_PMU_WRITE          = 0x70C,
+	VENDOR_CHIPIO_PLL_PMU_READ           = 0xF0C,
+	VENDOR_CHIPIO_8051_ADDRESS_LOW       = 0x70D,
+	VENDOR_CHIPIO_8051_ADDRESS_HIGH      = 0x70E,
+	VENDOR_CHIPIO_FLAG_SET               = 0x70F,
+	VENDOR_CHIPIO_FLAGS_GET              = 0xF0F,
+	VENDOR_CHIPIO_PARAMETER_SET          = 0x710,
+	VENDOR_CHIPIO_PARAMETER_GET          = 0xF10,
+
+	VENDOR_CHIPIO_PORT_ALLOC_CONFIG_SET  = 0x711,
+	VENDOR_CHIPIO_PORT_ALLOC_SET         = 0x712,
+	VENDOR_CHIPIO_PORT_ALLOC_GET         = 0xF12,
+	VENDOR_CHIPIO_PORT_FREE_SET          = 0x713,
+
+	VENDOR_CHIPIO_PARAMETER_EX_ID_GET    = 0xF17,
+	VENDOR_CHIPIO_PARAMETER_EX_ID_SET    = 0x717,
+	VENDOR_CHIPIO_PARAMETER_EX_VALUE_GET = 0xF18,
+	VENDOR_CHIPIO_PARAMETER_EX_VALUE_SET = 0x718
+};
+
+/*
+ *  Control flag IDs
+ */
+enum control_flag_id {
+	/* Connection manager stream setup is bypassed/enabled */
+	CONTROL_FLAG_C_MGR                  = 0,
+	/* DSP DMA is bypassed/enabled */
+	CONTROL_FLAG_DMA                    = 1,
+	/* 8051 'idle' mode is disabled/enabled */
+	CONTROL_FLAG_IDLE_ENABLE            = 2,
+	/* Tracker for the SPDIF-in path is bypassed/enabled */
+	CONTROL_FLAG_TRACKER                = 3,
+	/* DigitalOut to Spdif2Out connection is disabled/enabled */
+	CONTROL_FLAG_SPDIF2OUT              = 4,
+	/* Digital Microphone is disabled/enabled */
+	CONTROL_FLAG_DMIC                   = 5,
+	/* ADC_B rate is 48 kHz/96 kHz */
+	CONTROL_FLAG_ADC_B_96KHZ            = 6,
+	/* ADC_C rate is 48 kHz/96 kHz */
+	CONTROL_FLAG_ADC_C_96KHZ            = 7,
+	/* DAC rate is 48 kHz/96 kHz (affects all DACs) */
+	CONTROL_FLAG_DAC_96KHZ              = 8,
+	/* DSP rate is 48 kHz/96 kHz */
+	CONTROL_FLAG_DSP_96KHZ              = 9,
+	/* SRC clock is 98 MHz/196 MHz (196 MHz forces rate to 96 KHz) */
+	CONTROL_FLAG_SRC_CLOCK_196MHZ       = 10,
+	/* SRC rate is 48 kHz/96 kHz (48 kHz disabled when clock is 196 MHz) */
+	CONTROL_FLAG_SRC_RATE_96KHZ         = 11,
+	/* Decode Loop (DSP->SRC->DSP) is disabled/enabled */
+	CONTROL_FLAG_DECODE_LOOP            = 12,
+	/* De-emphasis filter on DAC-1 disabled/enabled */
+	CONTROL_FLAG_DAC1_DEEMPHASIS        = 13,
+	/* De-emphasis filter on DAC-2 disabled/enabled */
+	CONTROL_FLAG_DAC2_DEEMPHASIS        = 14,
+	/* De-emphasis filter on DAC-3 disabled/enabled */
+	CONTROL_FLAG_DAC3_DEEMPHASIS        = 15,
+	/* High-pass filter on ADC_B disabled/enabled */
+	CONTROL_FLAG_ADC_B_HIGH_PASS        = 16,
+	/* High-pass filter on ADC_C disabled/enabled */
+	CONTROL_FLAG_ADC_C_HIGH_PASS        = 17,
+	/* Common mode on Port_A disabled/enabled */
+	CONTROL_FLAG_PORT_A_COMMON_MODE     = 18,
+	/* Common mode on Port_D disabled/enabled */
+	CONTROL_FLAG_PORT_D_COMMON_MODE     = 19,
+	/* Impedance for ramp generator on Port_A 16 Ohm/10K Ohm */
+	CONTROL_FLAG_PORT_A_10KOHM_LOAD     = 20,
+	/* Impedance for ramp generator on Port_D, 16 Ohm/10K Ohm */
+	CONTROL_FLAG_PORT_D_10K0HM_LOAD     = 21,
+	/* ASI rate is 48kHz/96kHz */
+	CONTROL_FLAG_ASI_96KHZ              = 22,
+	/* DAC power settings able to control attached ports no/yes */
+	CONTROL_FLAG_DACS_CONTROL_PORTS     = 23,
+	/* Clock Stop OK reporting is disabled/enabled */
+	CONTROL_FLAG_CONTROL_STOP_OK_ENABLE = 24,
+	/* Number of control flags */
+	CONTROL_FLAGS_MAX = (CONTROL_FLAG_CONTROL_STOP_OK_ENABLE+1)
+};
+
+/*
+ * Control parameter IDs
+ */
+enum control_parameter_id {
+	/* 0: force HDA, 1: allow DSP if HDA Spdif1Out stream is idle */
+	CONTROL_PARAM_SPDIF1_SOURCE            = 2,
+
+	/* Stream Control */
+
+	/* Select stream with the given ID */
+	CONTROL_PARAM_STREAM_ID                = 24,
+	/* Source connection point for the selected stream */
+	CONTROL_PARAM_STREAM_SOURCE_CONN_POINT = 25,
+	/* Destination connection point for the selected stream */
+	CONTROL_PARAM_STREAM_DEST_CONN_POINT   = 26,
+	/* Number of audio channels in the selected stream */
+	CONTROL_PARAM_STREAMS_CHANNELS         = 27,
+	/*Enable control for the selected stream */
+	CONTROL_PARAM_STREAM_CONTROL           = 28,
+
+	/* Connection Point Control */
+
+	/* Select connection point with the given ID */
+	CONTROL_PARAM_CONN_POINT_ID            = 29,
+	/* Connection point sample rate */
+	CONTROL_PARAM_CONN_POINT_SAMPLE_RATE   = 30,
+
+	/* Node Control */
+
+	/* Select HDA node with the given ID */
+	CONTROL_PARAM_NODE_ID                  = 31
+};
+
+/*
+ *  Dsp Io Status codes
+ */
+enum hda_vendor_status_dspio {
+	/* Success */
+	VENDOR_STATUS_DSPIO_OK                       = 0x00,
+	/* Busy, unable to accept new command, the host must retry */
+	VENDOR_STATUS_DSPIO_BUSY                     = 0x01,
+	/* SCP command queue is full */
+	VENDOR_STATUS_DSPIO_SCP_COMMAND_QUEUE_FULL   = 0x02,
+	/* SCP response queue is empty */
+	VENDOR_STATUS_DSPIO_SCP_RESPONSE_QUEUE_EMPTY = 0x03
+};
+
+/*
+ *  Chip Io Status codes
+ */
+enum hda_vendor_status_chipio {
+	/* Success */
+	VENDOR_STATUS_CHIPIO_OK   = 0x00,
+	/* Busy, unable to accept new command, the host must retry */
+	VENDOR_STATUS_CHIPIO_BUSY = 0x01
+};
+
+/*
+ *  CA0132 sample rate
+ */
+enum ca0132_sample_rate {
+	SR_6_000        = 0x00,
+	SR_8_000        = 0x01,
+	SR_9_600        = 0x02,
+	SR_11_025       = 0x03,
+	SR_16_000       = 0x04,
+	SR_22_050       = 0x05,
+	SR_24_000       = 0x06,
+	SR_32_000       = 0x07,
+	SR_44_100       = 0x08,
+	SR_48_000       = 0x09,
+	SR_88_200       = 0x0A,
+	SR_96_000       = 0x0B,
+	SR_144_000      = 0x0C,
+	SR_176_400      = 0x0D,
+	SR_192_000      = 0x0E,
+	SR_384_000      = 0x0F,
+
+	SR_COUNT        = 0x10,
+
+	SR_RATE_UNKNOWN = 0x1F
+};
+
+/*
+ *  Scp Helper function
+ */
+enum get_set {
+	IS_SET = 0,
+	IS_GET = 1,
+};
+
+/*
+ * Duplicated from ca0110 codec
+ */
+
+static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
+{
+	if (pin) {
+		snd_hda_codec_write(codec, pin, 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+		if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
+			snd_hda_codec_write(codec, pin, 0,
+					    AC_VERB_SET_AMP_GAIN_MUTE,
+					    AMP_OUT_UNMUTE);
+	}
+	if (dac)
+		snd_hda_codec_write(codec, dac, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO);
+}
+
+static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
+{
+	if (pin) {
+		snd_hda_codec_write(codec, pin, 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL,
+				    PIN_VREF80);
+		if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP)
+			snd_hda_codec_write(codec, pin, 0,
+					    AC_VERB_SET_AMP_GAIN_MUTE,
+					    AMP_IN_UNMUTE(0));
+	}
+	if (adc)
+		snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+				    AMP_IN_UNMUTE(0));
+}
+
+static char *dirstr[2] = { "Playback", "Capture" };
+
+static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
+		       int chan, int dir)
+{
+	char namestr[44];
+	int type = dir ? HDA_INPUT : HDA_OUTPUT;
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type);
+	sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]);
+	return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
+}
+
+static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
+		       int chan, int dir)
+{
+	char namestr[44];
+	int type = dir ? HDA_INPUT : HDA_OUTPUT;
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type);
+	sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]);
+	return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
+}
+
+#define add_out_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 0)
+#define add_out_volume(codec, nid, pfx) _add_volume(codec, nid, pfx, 3, 0)
+#define add_in_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 1)
+#define add_in_volume(codec, nid, pfx) _add_volume(codec, nid, pfx, 3, 1)
+#define add_mono_switch(codec, nid, pfx, chan) \
+	_add_switch(codec, nid, pfx, chan, 0)
+#define add_mono_volume(codec, nid, pfx, chan) \
+	_add_volume(codec, nid, pfx, chan, 0)
+#define add_in_mono_switch(codec, nid, pfx, chan) \
+	_add_switch(codec, nid, pfx, chan, 1)
+#define add_in_mono_volume(codec, nid, pfx, chan) \
+	_add_volume(codec, nid, pfx, chan, 1)
+
+
+/*
+ * CA0132 specific
+ */
+
+struct ca0132_spec {
+	struct auto_pin_cfg autocfg;
+	struct hda_multi_out multiout;
+	hda_nid_t out_pins[AUTO_CFG_MAX_OUTS];
+	hda_nid_t dacs[AUTO_CFG_MAX_OUTS];
+	hda_nid_t hp_dac;
+	hda_nid_t input_pins[AUTO_PIN_LAST];
+	hda_nid_t adcs[AUTO_PIN_LAST];
+	hda_nid_t dig_out;
+	hda_nid_t dig_in;
+	unsigned int num_inputs;
+	long curr_hp_switch;
+	long curr_hp_volume[2];
+	long curr_speaker_switch;
+	struct mutex chipio_mutex;
+	const char *input_labels[AUTO_PIN_LAST];
+	struct hda_pcm pcm_rec[2]; /* PCM information */
+};
+
+/* Chip access helper function */
+static int chipio_send(struct hda_codec *codec,
+		       unsigned int reg,
+		       unsigned int data)
+{
+	unsigned int res;
+	int retry = 50;
+
+	/* send bits of data specified by reg */
+	do {
+		res = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0,
+					 reg, data);
+		if (res == VENDOR_STATUS_CHIPIO_OK)
+			return 0;
+	} while (--retry);
+	return -EIO;
+}
+
+/*
+ * Write chip address through the vendor widget -- NOT protected by the Mutex!
+ */
+static int chipio_write_address(struct hda_codec *codec,
+				unsigned int chip_addx)
+{
+	int res;
+
+	/* send low 16 bits of the address */
+	res = chipio_send(codec, VENDOR_CHIPIO_ADDRESS_LOW,
+			  chip_addx & 0xffff);
+
+	if (res != -EIO) {
+		/* send high 16 bits of the address */
+		res = chipio_send(codec, VENDOR_CHIPIO_ADDRESS_HIGH,
+				  chip_addx >> 16);
+	}
+
+	return res;
+}
+
+/*
+ * Write data through the vendor widget -- NOT protected by the Mutex!
+ */
+
+static int chipio_write_data(struct hda_codec *codec, unsigned int data)
+{
+	int res;
+
+	/* send low 16 bits of the data */
+	res = chipio_send(codec, VENDOR_CHIPIO_DATA_LOW, data & 0xffff);
+
+	if (res != -EIO) {
+		/* send high 16 bits of the data */
+		res = chipio_send(codec, VENDOR_CHIPIO_DATA_HIGH,
+				  data >> 16);
+	}
+
+	return res;
+}
+
+/*
+ * Read data through the vendor widget -- NOT protected by the Mutex!
+ */
+static int chipio_read_data(struct hda_codec *codec, unsigned int *data)
+{
+	int res;
+
+	/* post read */
+	res = chipio_send(codec, VENDOR_CHIPIO_HIC_POST_READ, 0);
+
+	if (res != -EIO) {
+		/* read status */
+		res = chipio_send(codec, VENDOR_CHIPIO_STATUS, 0);
+	}
+
+	if (res != -EIO) {
+		/* read data */
+		*data = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0,
+					   VENDOR_CHIPIO_HIC_READ_DATA,
+					   0);
+	}
+
+	return res;
+}
+
+/*
+ * Write given value to the given address through the chip I/O widget.
+ * protected by the Mutex
+ */
+static int chipio_write(struct hda_codec *codec,
+		unsigned int chip_addx, const unsigned int data)
+{
+	struct ca0132_spec *spec = codec->spec;
+	int err;
+
+	mutex_lock(&spec->chipio_mutex);
+
+	/* write the address, and if successful proceed to write data */
+	err = chipio_write_address(codec, chip_addx);
+	if (err < 0)
+		goto exit;
+
+	err = chipio_write_data(codec, data);
+	if (err < 0)
+		goto exit;
+
+exit:
+	mutex_unlock(&spec->chipio_mutex);
+	return err;
+}
+
+/*
+ * Read the given address through the chip I/O widget
+ * protected by the Mutex
+ */
+static int chipio_read(struct hda_codec *codec,
+		unsigned int chip_addx, unsigned int *data)
+{
+	struct ca0132_spec *spec = codec->spec;
+	int err;
+
+	mutex_lock(&spec->chipio_mutex);
+
+	/* write the address, and if successful proceed to write data */
+	err = chipio_write_address(codec, chip_addx);
+	if (err < 0)
+		goto exit;
+
+	err = chipio_read_data(codec, data);
+	if (err < 0)
+		goto exit;
+
+exit:
+	mutex_unlock(&spec->chipio_mutex);
+	return err;
+}
+
+/*
+ * PCM stuffs
+ */
+static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid,
+				 u32 stream_tag,
+				 int channel_id, int format)
+{
+	unsigned int oldval, newval;
+
+	if (!nid)
+		return;
+
+	snd_printdd("ca0132_setup_stream: "
+		"NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n",
+		nid, stream_tag, channel_id, format);
+
+	/* update the format-id if changed */
+	oldval = snd_hda_codec_read(codec, nid, 0,
+				    AC_VERB_GET_STREAM_FORMAT,
+				    0);
+	if (oldval != format) {
+		msleep(20);
+		snd_hda_codec_write(codec, nid, 0,
+				    AC_VERB_SET_STREAM_FORMAT,
+				    format);
+	}
+
+	oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0);
+	newval = (stream_tag << 4) | channel_id;
+	if (oldval != newval) {
+		snd_hda_codec_write(codec, nid, 0,
+				    AC_VERB_SET_CHANNEL_STREAMID,
+				    newval);
+	}
+}
+
+static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid)
+{
+	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0);
+	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0);
+}
+
+/*
+ * PCM callbacks
+ */
+static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+			struct hda_codec *codec,
+			unsigned int stream_tag,
+			unsigned int format,
+			struct snd_pcm_substream *substream)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format);
+
+	return 0;
+}
+
+static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+			struct hda_codec *codec,
+			struct snd_pcm_substream *substream)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	ca0132_cleanup_stream(codec, spec->dacs[0]);
+
+	return 0;
+}
+
+/*
+ * Digital out
+ */
+static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+			struct hda_codec *codec,
+			unsigned int stream_tag,
+			unsigned int format,
+			struct snd_pcm_substream *substream)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	ca0132_setup_stream(codec, spec->dig_out, stream_tag, 0, format);
+
+	return 0;
+}
+
+static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+			struct hda_codec *codec,
+			struct snd_pcm_substream *substream)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	ca0132_cleanup_stream(codec, spec->dig_out);
+
+	return 0;
+}
+
+/*
+ * Analog capture
+ */
+static int ca0132_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 ca0132_spec *spec = codec->spec;
+
+	ca0132_setup_stream(codec, spec->adcs[substream->number],
+			     stream_tag, 0, format);
+
+	return 0;
+}
+
+static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+			struct hda_codec *codec,
+			struct snd_pcm_substream *substream)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	ca0132_cleanup_stream(codec, spec->adcs[substream->number]);
+
+	return 0;
+}
+
+/*
+ * Digital capture
+ */
+static int ca0132_dig_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 ca0132_spec *spec = codec->spec;
+
+	ca0132_setup_stream(codec, spec->dig_in, stream_tag, 0, format);
+
+	return 0;
+}
+
+static int ca0132_dig_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+			struct hda_codec *codec,
+			struct snd_pcm_substream *substream)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	ca0132_cleanup_stream(codec, spec->dig_in);
+
+	return 0;
+}
+
+/*
+ */
+static struct hda_pcm_stream ca0132_pcm_analog_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.ops = {
+		.prepare = ca0132_playback_pcm_prepare,
+		.cleanup = ca0132_playback_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream ca0132_pcm_analog_capture = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.ops = {
+		.prepare = ca0132_capture_pcm_prepare,
+		.cleanup = ca0132_capture_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream ca0132_pcm_digital_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.ops = {
+		.prepare = ca0132_dig_playback_pcm_prepare,
+		.cleanup = ca0132_dig_playback_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream ca0132_pcm_digital_capture = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.ops = {
+		.prepare = ca0132_dig_capture_pcm_prepare,
+		.cleanup = ca0132_dig_capture_pcm_cleanup
+	},
+};
+
+static int ca0132_build_pcms(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	struct hda_pcm *info = spec->pcm_rec;
+
+	codec->pcm_info = info;
+	codec->num_pcms = 0;
+
+	info->name = "CA0132 Analog";
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0132_pcm_analog_playback;
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0];
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
+		spec->multiout.max_channels;
+	info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture;
+	info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_inputs;
+	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0];
+	codec->num_pcms++;
+
+	if (!spec->dig_out && !spec->dig_in)
+		return 0;
+
+	info++;
+	info->name = "CA0132 Digital";
+	info->pcm_type = HDA_PCM_TYPE_SPDIF;
+	if (spec->dig_out) {
+		info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+			ca0132_pcm_digital_playback;
+		info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dig_out;
+	}
+	if (spec->dig_in) {
+		info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+			ca0132_pcm_digital_capture;
+		info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in;
+	}
+	codec->num_pcms++;
+
+	return 0;
+}
+
+#define REG_CODEC_MUTE		0x18b014
+#define REG_CODEC_HP_VOL_L	0x18b070
+#define REG_CODEC_HP_VOL_R	0x18b074
+
+static int ca0132_hp_switch_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+	long *valp = ucontrol->value.integer.value;
+
+	*valp = spec->curr_hp_switch;
+	return 0;
+}
+
+static int ca0132_hp_switch_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+	long *valp = ucontrol->value.integer.value;
+	unsigned int data;
+	int err;
+
+	/* any change? */
+	if (spec->curr_hp_switch == *valp)
+		return 0;
+
+	snd_hda_power_up(codec);
+
+	err = chipio_read(codec, REG_CODEC_MUTE, &data);
+	if (err < 0)
+		return err;
+
+	/* *valp 0 is mute, 1 is unmute */
+	data = (data & 0x7f) | (*valp ? 0 : 0x80);
+	chipio_write(codec, REG_CODEC_MUTE, data);
+	if (err < 0)
+		return err;
+
+	spec->curr_hp_switch = *valp;
+
+	snd_hda_power_down(codec);
+	return 1;
+}
+
+static int ca0132_speaker_switch_get(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+	long *valp = ucontrol->value.integer.value;
+
+	*valp = spec->curr_speaker_switch;
+	return 0;
+}
+
+static int ca0132_speaker_switch_put(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+	long *valp = ucontrol->value.integer.value;
+	unsigned int data;
+	int err;
+
+	/* any change? */
+	if (spec->curr_speaker_switch == *valp)
+		return 0;
+
+	snd_hda_power_up(codec);
+
+	err = chipio_read(codec, REG_CODEC_MUTE, &data);
+	if (err < 0)
+		return err;
+
+	/* *valp 0 is mute, 1 is unmute */
+	data = (data & 0xef) | (*valp ? 0 : 0x10);
+	chipio_write(codec, REG_CODEC_MUTE, data);
+	if (err < 0)
+		return err;
+
+	spec->curr_speaker_switch = *valp;
+
+	snd_hda_power_down(codec);
+	return 1;
+}
+
+static int ca0132_hp_volume_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+	long *valp = ucontrol->value.integer.value;
+
+	*valp++ = spec->curr_hp_volume[0];
+	*valp = spec->curr_hp_volume[1];
+	return 0;
+}
+
+static int ca0132_hp_volume_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+	long *valp = ucontrol->value.integer.value;
+	long left_vol, right_vol;
+	unsigned int data;
+	int val;
+	int err;
+
+	left_vol = *valp++;
+	right_vol = *valp;
+
+	/* any change? */
+	if ((spec->curr_hp_volume[0] == left_vol) &&
+		(spec->curr_hp_volume[1] == right_vol))
+		return 0;
+
+	snd_hda_power_up(codec);
+
+	err = chipio_read(codec, REG_CODEC_HP_VOL_L, &data);
+	if (err < 0)
+		return err;
+
+	val = 31 - left_vol;
+	data = (data & 0xe0) | val;
+	chipio_write(codec, REG_CODEC_HP_VOL_L, data);
+	if (err < 0)
+		return err;
+
+	val = 31 - right_vol;
+	data = (data & 0xe0) | val;
+	chipio_write(codec, REG_CODEC_HP_VOL_R, data);
+	if (err < 0)
+		return err;
+
+	spec->curr_hp_volume[0] = left_vol;
+	spec->curr_hp_volume[1] = right_vol;
+
+	snd_hda_power_down(codec);
+	return 1;
+}
+
+static int add_hp_switch(struct hda_codec *codec, hda_nid_t nid)
+{
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_MUTE_MONO("Headphone Playback Switch",
+				     nid, 1, 0, HDA_OUTPUT);
+	knew.get = ca0132_hp_switch_get;
+	knew.put = ca0132_hp_switch_put;
+	return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
+}
+
+static int add_hp_volume(struct hda_codec *codec, hda_nid_t nid)
+{
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_VOLUME_MONO("Headphone Playback Volume",
+				       nid, 3, 0, HDA_OUTPUT);
+	knew.get = ca0132_hp_volume_get;
+	knew.put = ca0132_hp_volume_put;
+	return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
+}
+
+static int add_speaker_switch(struct hda_codec *codec, hda_nid_t nid)
+{
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_MUTE_MONO("Speaker Playback Switch",
+				     nid, 1, 0, HDA_OUTPUT);
+	knew.get = ca0132_speaker_switch_get;
+	knew.put = ca0132_speaker_switch_put;
+	return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
+}
+
+static void ca0132_fix_hp_caps(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	unsigned int caps;
+
+	/* set mute-capable, 1db step, 32 steps, ofs 6 */
+	caps = 0x80031f06;
+	snd_hda_override_amp_caps(codec, cfg->hp_pins[0], HDA_OUTPUT, caps);
+}
+
+static int ca0132_build_controls(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i, err;
+
+	if (spec->multiout.num_dacs) {
+		err = add_speaker_switch(codec, spec->out_pins[0]);
+		if (err < 0)
+			return err;
+	}
+
+	if (cfg->hp_outs) {
+		ca0132_fix_hp_caps(codec);
+		err = add_hp_switch(codec, cfg->hp_pins[0]);
+		if (err < 0)
+			return err;
+		err = add_hp_volume(codec, cfg->hp_pins[0]);
+		if (err < 0)
+			return err;
+	}
+
+	for (i = 0; i < spec->num_inputs; i++) {
+		const char *label = spec->input_labels[i];
+
+		err = add_in_switch(codec, spec->adcs[i], label);
+		if (err < 0)
+			return err;
+		err = add_in_volume(codec, spec->adcs[i], label);
+		if (err < 0)
+			return err;
+		if (cfg->inputs[i].type == AUTO_PIN_MIC) {
+			/* add Mic-Boost */
+			err = add_in_mono_volume(codec, spec->input_pins[i],
+						 "Mic Boost", 1);
+			if (err < 0)
+				return err;
+		}
+	}
+
+	if (spec->dig_out) {
+		err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out,
+						    spec->dig_out);
+		if (err < 0)
+			return err;
+		err = add_out_volume(codec, spec->dig_out, "IEC958");
+		if (err < 0)
+			return err;
+	}
+
+	if (spec->dig_in) {
+		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
+		if (err < 0)
+			return err;
+		err = add_in_volume(codec, spec->dig_in, "IEC958");
+	}
+	return 0;
+}
+
+
+static void ca0132_set_ct_ext(struct hda_codec *codec, int enable)
+{
+	/* Set Creative extension */
+	snd_printdd("SET CREATIVE EXTENSION\n");
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE,
+			    enable);
+	msleep(20);
+}
+
+
+static void ca0132_config(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+
+	/* line-outs */
+	cfg->line_outs = 1;
+	cfg->line_out_pins[0] = 0x0b; /* front */
+	cfg->line_out_type = AUTO_PIN_LINE_OUT;
+
+	spec->dacs[0] = 0x02;
+	spec->out_pins[0] = 0x0b;
+	spec->multiout.dac_nids = spec->dacs;
+	spec->multiout.num_dacs = 1;
+	spec->multiout.max_channels = 2;
+
+	/* headphone */
+	cfg->hp_outs = 1;
+	cfg->hp_pins[0] = 0x0f;
+
+	spec->hp_dac = 0;
+	spec->multiout.hp_nid = 0;
+
+	/* inputs */
+	cfg->num_inputs = 2;  /* Mic-in and line-in */
+	cfg->inputs[0].pin = 0x12;
+	cfg->inputs[0].type = AUTO_PIN_MIC;
+	cfg->inputs[1].pin = 0x11;
+	cfg->inputs[1].type = AUTO_PIN_LINE_IN;
+
+	/* Mic-in */
+	spec->input_pins[0] = 0x12;
+	spec->input_labels[0] = "Mic-In";
+	spec->adcs[0] = 0x07;
+
+	/* Line-In */
+	spec->input_pins[1] = 0x11;
+	spec->input_labels[1] = "Line-In";
+	spec->adcs[1] = 0x08;
+	spec->num_inputs = 2;
+}
+
+static void ca0132_init_chip(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	mutex_init(&spec->chipio_mutex);
+}
+
+static void ca0132_exit_chip(struct hda_codec *codec)
+{
+	/* put any chip cleanup stuffs here. */
+}
+
+static int ca0132_init(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i;
+
+	for (i = 0; i < spec->multiout.num_dacs; i++) {
+		init_output(codec, spec->out_pins[i],
+			    spec->multiout.dac_nids[i]);
+	}
+	init_output(codec, cfg->hp_pins[0], spec->hp_dac);
+	init_output(codec, cfg->dig_out_pins[0], spec->dig_out);
+
+	for (i = 0; i < spec->num_inputs; i++)
+		init_input(codec, spec->input_pins[i], spec->adcs[i]);
+
+	init_input(codec, cfg->dig_in_pin, spec->dig_in);
+
+	ca0132_set_ct_ext(codec, 1);
+
+	return 0;
+}
+
+
+static void ca0132_free(struct hda_codec *codec)
+{
+	ca0132_set_ct_ext(codec, 0);
+	ca0132_exit_chip(codec);
+	kfree(codec->spec);
+}
+
+static struct hda_codec_ops ca0132_patch_ops = {
+	.build_controls = ca0132_build_controls,
+	.build_pcms = ca0132_build_pcms,
+	.init = ca0132_init,
+	.free = ca0132_free,
+};
+
+
+
+static int patch_ca0132(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec;
+
+	snd_printdd("patch_ca0132\n");
+
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (!spec)
+		return -ENOMEM;
+	codec->spec = spec;
+
+	ca0132_init_chip(codec);
+
+	ca0132_config(codec);
+
+	codec->patch_ops = ca0132_patch_ops;
+
+	return 0;
+}
+
+/*
+ * patch entries
+ */
+static struct hda_codec_preset snd_hda_preset_ca0132[] = {
+	{ .id = 0x11020011, .name = "CA0132",     .patch = patch_ca0132 },
+	{} /* terminator */
+};
+
+MODULE_ALIAS("snd-hda-codec-id:11020011");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Creative CA0132, CA0132 HD-audio codec");
+
+static struct hda_codec_preset_list ca0132_list = {
+	.preset = snd_hda_preset_ca0132,
+	.owner = THIS_MODULE,
+};
+
+static int __init patch_ca0132_init(void)
+{
+	return snd_hda_add_codec_preset(&ca0132_list);
+}
+
+static void __exit patch_ca0132_exit(void)
+{
+	snd_hda_delete_codec_preset(&ca0132_list);
+}
+
+module_init(patch_ca0132_init)
+module_exit(patch_ca0132_exit)

sound/pci/hda/patch_cirrus.c

@@ -346,21 +346,15 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
 
 	nid = codec->start_nid;
 	for (i = 0; i < codec->num_nodes; i++, nid++) {
-		hda_nid_t pins[2];
 		unsigned int type;
-		int j, nums;
+		int idx;
 		type = get_wcaps_type(get_wcaps(codec, nid));
 		if (type != AC_WID_AUD_IN)
 			continue;
-		nums = snd_hda_get_connections(codec, nid, pins,
-					       ARRAY_SIZE(pins));
-		if (nums <= 0)
-			continue;
-		for (j = 0; j < nums; j++) {
-			if (pins[j] == pin) {
-				*idxp = j;
-				return nid;
-			}
+		idx = snd_hda_get_conn_index(codec, nid, pin, 0);
+		if (idx >= 0) {
+			*idxp = idx;
+			return nid;
 		}
 	}
 	return 0;
@@ -821,7 +815,8 @@ static int build_digital_output(struct hda_codec *codec)
 	if (!spec->multiout.dig_out_nid)
 		return 0;
 
-	err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+	err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid,
+					    spec->multiout.dig_out_nid);
 	if (err < 0)
 		return err;
 	err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);

sound/pci/hda/patch_cmedia.c

@@ -327,7 +327,9 @@ static int cmi9880_build_controls(struct hda_codec *codec)
 			return err;
 	}
 	if (spec->multiout.dig_out_nid) {
-		err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+		err = snd_hda_create_spdif_out_ctls(codec,
+						    spec->multiout.dig_out_nid,
+						    spec->multiout.dig_out_nid);
 		if (err < 0)
 			return err;
 		err = snd_hda_create_spdif_share_sw(codec,
@@ -396,12 +398,11 @@ static int cmi9880_fill_multi_init(struct hda_codec *codec, const struct auto_pi
 {
 	struct cmi_spec *spec = codec->spec;
 	hda_nid_t nid;
-	int i, j, k, len;
+	int i, j, k;
 
 	/* clear the table, only one c-media dac assumed here */
 	memset(spec->multi_init, 0, sizeof(spec->multi_init));
 	for (j = 0, i = 0; i < cfg->line_outs; i++) {
-		hda_nid_t conn[4];
 		nid = cfg->line_out_pins[i];
 		/* set as output */
 		spec->multi_init[j].nid = nid;
@@ -414,12 +415,10 @@ static int cmi9880_fill_multi_init(struct hda_codec *codec, const struct auto_pi
 			spec->multi_init[j].verb = AC_VERB_SET_CONNECT_SEL;
 			spec->multi_init[j].param = 0;
 			/* find the index in connect list */
-			len = snd_hda_get_connections(codec, nid, conn, 4);
-			for (k = 0; k < len; k++)
-				if (conn[k] == spec->dac_nids[i]) {
-					spec->multi_init[j].param = k;
-					break;
-				}
+			k = snd_hda_get_conn_index(codec, nid,
+						   spec->dac_nids[i], 0);
+			if (k >= 0)
+				spec->multi_init[j].param = k;
 			j++;
 		}
 	}

sound/pci/hda/patch_conexant.c

@@ -155,6 +155,10 @@ struct conexant_spec {
 	unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */
 
 	unsigned int beep_amp;
+
+	/* extra EAPD pins */
+	unsigned int num_eapds;
+	hda_nid_t eapds[4];
 };
 
 static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -510,6 +514,7 @@ static int conexant_build_controls(struct hda_codec *codec)
 	}
 	if (spec->multiout.dig_out_nid) {
 		err = snd_hda_create_spdif_out_ctls(codec,
+						    spec->multiout.dig_out_nid,
 						    spec->multiout.dig_out_nid);
 		if (err < 0)
 			return err;
@@ -1123,10 +1128,8 @@ static int patch_cxt5045(struct hda_codec *codec)
 	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
+		board_config = CXT5045_AUTO; /* model=auto as default */
 	if (board_config == CXT5045_AUTO)
 		return patch_conexant_auto(codec);
 
@@ -1564,10 +1567,8 @@ static int patch_cxt5047(struct hda_codec *codec)
 	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
+		board_config = CXT5047_AUTO; /* model=auto as default */
 	if (board_config == CXT5047_AUTO)
 		return patch_conexant_auto(codec);
 
@@ -1993,10 +1994,8 @@ static int patch_cxt5051(struct hda_codec *codec)
 	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
+		board_config = CXT5051_AUTO; /* model=auto as default */
 	if (board_config == CXT5051_AUTO)
 		return patch_conexant_auto(codec);
 
@@ -3114,10 +3113,8 @@ static int patch_cxt5066(struct hda_codec *codec)
 
 	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
+		board_config = CXT5066_AUTO; /* model=auto as default */
 	if (board_config == CXT5066_AUTO)
 		return patch_conexant_auto(codec);
 
@@ -3308,19 +3305,8 @@ static const struct hda_pcm_stream cx_auto_pcm_analog_capture = {
 
 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,
-				hda_nid_t nid)
-{
-	hda_nid_t conn[HDA_MAX_NUM_INPUTS];
-	int i, nums;
-
-	nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
-	for (i = 0; i < nums; i++)
-		if (conn[i] == nid)
-			return i;
-	return -1;
-}
+#define get_connection_index(codec, mux, nid)\
+	snd_hda_get_conn_index(codec, mux, nid, 0)
 
 /* get an unassigned DAC from the given list.
  * Return the nid if found and reduce the DAC list, or return zero if
@@ -3919,6 +3905,38 @@ static void cx_auto_parse_beep(struct hda_codec *codec)
 #define cx_auto_parse_beep(codec)
 #endif
 
+static bool found_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
+{
+	int i;
+	for (i = 0; i < nums; i++)
+		if (list[i] == nid)
+			return true;
+	return false;
+}
+
+/* parse extra-EAPD that aren't assigned to any pins */
+static void cx_auto_parse_eapd(struct hda_codec *codec)
+{
+	struct conexant_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	hda_nid_t nid, end_nid;
+
+	end_nid = codec->start_nid + codec->num_nodes;
+	for (nid = codec->start_nid; nid < end_nid; nid++) {
+		if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
+			continue;
+		if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD))
+			continue;
+		if (found_in_nid_list(nid, cfg->line_out_pins, cfg->line_outs) ||
+		    found_in_nid_list(nid, cfg->hp_pins, cfg->hp_outs) ||
+		    found_in_nid_list(nid, cfg->speaker_pins, cfg->speaker_outs))
+			continue;
+		spec->eapds[spec->num_eapds++] = nid;
+		if (spec->num_eapds >= ARRAY_SIZE(spec->eapds))
+			break;
+	}
+}
+
 static int cx_auto_parse_auto_config(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
@@ -3932,6 +3950,7 @@ static int cx_auto_parse_auto_config(struct hda_codec *codec)
 	cx_auto_parse_input(codec);
 	cx_auto_parse_digital(codec);
 	cx_auto_parse_beep(codec);
+	cx_auto_parse_eapd(codec);
 	return 0;
 }
 
@@ -4019,6 +4038,8 @@ static void cx_auto_init_output(struct hda_codec *codec)
 		}
 	}
 	cx_auto_update_speakers(codec);
+	/* turn on/off extra EAPDs, too */
+	cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
 }
 
 static void cx_auto_init_input(struct hda_codec *codec)

sound/pci/hda/patch_sigmatel.c

@@ -1112,7 +1112,9 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 	}
 
 	if (spec->multiout.dig_out_nid) {
-		err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+		err = snd_hda_create_spdif_out_ctls(codec,
+						    spec->multiout.dig_out_nid,
+						    spec->multiout.dig_out_nid);
 		if (err < 0)
 			return err;
 		err = snd_hda_create_spdif_share_sw(codec,
@@ -3406,30 +3408,9 @@ static hda_nid_t get_connected_node(struct hda_codec *codec, hda_nid_t mux,
 	return 0;
 }
 
-static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
-				hda_nid_t nid)
-{
-	hda_nid_t conn[HDA_MAX_NUM_INPUTS];
-	int i, nums;
-
-	if (!(get_wcaps(codec, mux) & AC_WCAP_CONN_LIST))
-		return -1;
-
-	nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
-	for (i = 0; i < nums; i++)
-		if (conn[i] == nid)
-			return i;
-
-	for (i = 0; i < nums; i++) {
-		unsigned int wid_caps = get_wcaps(codec, conn[i]);
-		unsigned int wid_type = get_wcaps_type(wid_caps);
-
-		if (wid_type != AC_WID_PIN && wid_type != AC_WID_AUD_MIX)
-			if (get_connection_index(codec, conn[i], nid) >= 0)
-				return i;
-	}
-	return -1;
-}
+/* look for NID recursively */
+#define get_connection_index(codec, mux, nid) \
+	snd_hda_get_conn_index(codec, mux, nid, 1)
 
 /* create a volume assigned to the given pin (only if supported) */
 /* return 1 if the volume control is created */

sound/pci/ice1712/ice1712.c

@@ -2607,7 +2607,7 @@ static int __devinit snd_ice1712_create(struct snd_card *card,
 	ice->profi_port = pci_resource_start(pci, 3);
 
 	if (request_irq(pci->irq, snd_ice1712_interrupt, IRQF_SHARED,
-			"ICE1712", ice)) {
+			KBUILD_MODNAME, ice)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		snd_ice1712_free(ice);
 		return -EIO;
@@ -2802,7 +2802,7 @@ static void __devexit snd_ice1712_remove(struct pci_dev *pci)
 }
 
 static struct pci_driver driver = {
-	.name = "ICE1712",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_ice1712_ids,
 	.probe = snd_ice1712_probe,
 	.remove = __devexit_p(snd_ice1712_remove),

sound/pci/ice1712/ice1724.c

@@ -2509,7 +2509,7 @@ static int __devinit snd_vt1724_create(struct snd_card *card,
 	ice->profi_port = pci_resource_start(pci, 1);
 
 	if (request_irq(pci->irq, snd_vt1724_interrupt,
-			IRQF_SHARED, "ICE1724", ice)) {
+			IRQF_SHARED, KBUILD_MODNAME, ice)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		snd_vt1724_free(ice);
 		return -EIO;
@@ -2802,7 +2802,7 @@ static int snd_vt1724_resume(struct pci_dev *pci)
 #endif
 
 static struct pci_driver driver = {
-	.name = "ICE1724",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_vt1724_ids,
 	.probe = snd_vt1724_probe,
 	.remove = __devexit_p(snd_vt1724_remove),

sound/pci/intel8x0.c

@@ -1882,6 +1882,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
 		.name = "Dell Inspiron 6000",
 		.type = AC97_TUNE_HP_MUTE_LED /* cf. Malone #41015 */
 	},
+	{
+		.subvendor = 0x1028,
+		.subdevice = 0x0189,
+		.name = "Dell Inspiron 9300",
+		.type = AC97_TUNE_HP_MUTE_LED
+	},
 	{
 		.subvendor = 0x1028,
 		.subdevice = 0x0191,
@@ -2647,7 +2653,7 @@ static int intel8x0_resume(struct pci_dev *pci)
 	pci_set_master(pci);
 	snd_intel8x0_chip_init(chip, 0);
 	if (request_irq(pci->irq, snd_intel8x0_interrupt,
-			IRQF_SHARED, card->shortname, chip)) {
+			IRQF_SHARED, KBUILD_MODNAME, chip)) {
 		printk(KERN_ERR "intel8x0: unable to grab IRQ %d, "
 		       "disabling device\n", pci->irq);
 		snd_card_disconnect(card);
@@ -3106,7 +3112,7 @@ static int __devinit snd_intel8x0_create(struct snd_card *card,
 
 	/* request irq after initializaing int_sta_mask, etc */
 	if (request_irq(pci->irq, snd_intel8x0_interrupt,
-			IRQF_SHARED, card->shortname, chip)) {
+			IRQF_SHARED, KBUILD_MODNAME, chip)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		snd_intel8x0_free(chip);
 		return -EBUSY;
@@ -3266,7 +3272,7 @@ static void __devexit snd_intel8x0_remove(struct pci_dev *pci)
 }
 
 static struct pci_driver driver = {
-	.name = "Intel ICH",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_intel8x0_ids,
 	.probe = snd_intel8x0_probe,
 	.remove = __devexit_p(snd_intel8x0_remove),

sound/pci/intel8x0m.c

@@ -1047,7 +1047,7 @@ static int intel8x0m_resume(struct pci_dev *pci)
 	}
 	pci_set_master(pci);
 	if (request_irq(pci->irq, snd_intel8x0m_interrupt,
-			IRQF_SHARED, card->shortname, chip)) {
+			IRQF_SHARED, KBUILD_MODNAME, chip)) {
 		printk(KERN_ERR "intel8x0m: unable to grab IRQ %d, "
 		       "disabling device\n", pci->irq);
 		snd_card_disconnect(card);
@@ -1174,7 +1174,7 @@ static int __devinit snd_intel8x0m_create(struct snd_card *card,
 
  port_inited:
 	if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED,
-			card->shortname, chip)) {
+			KBUILD_MODNAME, chip)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		snd_intel8x0m_free(chip);
 		return -EBUSY;
@@ -1325,7 +1325,7 @@ static void __devexit snd_intel8x0m_remove(struct pci_dev *pci)
 }
 
 static struct pci_driver driver = {
-	.name = "Intel ICH Modem",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_intel8x0m_ids,
 	.probe = snd_intel8x0m_probe,
 	.remove = __devexit_p(snd_intel8x0m_remove),

sound/pci/korg1212/korg1212.c

@@ -2241,7 +2241,7 @@ static int __devinit snd_korg1212_create(struct snd_card *card, struct pci_dev *
 
         err = request_irq(pci->irq, snd_korg1212_interrupt,
                           IRQF_SHARED,
-                          "korg1212", korg1212);
+                          KBUILD_MODNAME, korg1212);
 
         if (err) {
 		snd_printk(KERN_ERR "korg1212: unable to grab IRQ %d\n", pci->irq);
@@ -2477,7 +2477,7 @@ static void __devexit snd_korg1212_remove(struct pci_dev *pci)
 }
 
 static struct pci_driver driver = {
-	.name = "korg1212",
+	.name = KBUILD_MODNAME,
 	.id_table = snd_korg1212_ids,
 	.probe = snd_korg1212_probe,
 	.remove = __devexit_p(snd_korg1212_remove),

sound/pci/lola/lola.c

@@ -648,7 +648,7 @@ static int __devinit lola_create(struct snd_card *card, struct pci_dev *pci,
 		goto errout;
 
 	if (request_irq(pci->irq, lola_interrupt, IRQF_SHARED,
-			DRVNAME, chip)) {
+			KBUILD_MODNAME, chip)) {
 		printk(KERN_ERR SFX "unable to grab IRQ %d\n", pci->irq);
 		err = -EBUSY;
 		goto errout;
@@ -771,7 +771,7 @@ MODULE_DEVICE_TABLE(pci, lola_ids);
 
 /* pci_driver definition */
 static struct pci_driver driver = {
-	.name = DRVNAME,
+	.name = KBUILD_MODNAME,
 	.id_table = lola_ids,
 	.probe = lola_probe,
 	.remove = __devexit_p(lola_remove),

sound/pci/lola/lola.h

@@ -480,7 +480,7 @@ struct lola {
 
 /* 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)
+#define LOLA_MIXER_DEST_REC_OUTPUT_SEPARATION(res)  ((res >> 7) & 0x1f)
 
 int lola_codec_write(struct lola *chip, unsigned int nid, unsigned int verb,
 		     unsigned int data, unsigned int extdata);

sound/pci/lola/lola_mixer.c

@@ -144,40 +144,61 @@ int __devinit lola_init_mixer_widget(struct lola *chip, int nid)
 	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
+	/* mixer matrix may 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-------------------------------
+		LOLA_MIXER_DEST_REC_OUTPUT_SEPARATION(val);
+
+	/* example : MixerMatrix of LoLa881 (LoLa16161 uses unused zones)
+	 * +-+  0-------8------16-------8------16
+	 * | |  |       |       |       |       |
+	 * |s|  | INPUT |       | INPUT |       |
+	 * | |->|  ->   |unused |  ->   |unused |
+	 * |r|  |CAPTURE|       | OUTPUT|       |
+	 * | |  |  MIX  |       |  MIX  |       |
+	 * |c|  8--------------------------------
+	 * | |  |       |       |       |       |
+	 * | |  |       |       |       |       |
+	 * |g|  |unused |unused |unused |unused |
+	 * | |  |       |       |       |       |
+	 * |a|  |       |       |       |       |
+	 * | |  16-------------------------------
+	 * |i|  |       |       |       |       |
+	 * | |  | PLAYBK|       | PLAYBK|       |
+	 * |n|->|  ->   |unused |  ->   |unused |
+	 * | |  |CAPTURE|       | OUTPUT|       |
+	 * | |  |  MIX  |       |  MIX  |       |
+	 * |a|  8--------------------------------
+	 * |r|  |       |       |       |       |
+	 * |r|  |       |       |       |       |
+	 * |a|  |unused |unused |unused |unused |
+	 * |y|  |       |       |       |       |
+	 * | |  |       |       |       |       |
+	 * +++  16--|---------------|------------
+	 *      +---V---------------V-----------+
+	 *      |  dest_mix_gain_enable array   |
+	 *      +-------------------------------+
+	 */
+	/* example : MixerMatrix of LoLa280
+	 * +-+  0-------8-2
+	 * | |  |       | |
+	 * |s|  | INPUT | |     INPUT
+	 * |r|->|  ->   | |      ->
+	 * |c|  |CAPTURE| | <-  OUTPUT
+	 * | |  |  MIX  | |      MIX
+	 * |g|  8----------
+	 * |a|  |       | |
+	 * |i|  | PLAYBK| |     PLAYBACK
+	 * |n|->|  ->   | |      ->
+	 * | |  |CAPTURE| | <-  OUTPUT
+	 * |a|  |  MIX  | |      MIX
+	 * |r|  8---|----|-
+	 * |r|  +---V----V-------------------+
+	 * |a|  | dest_mix_gain_enable array |
+	 * |y|  +----------------------------+
 	 */
 	if (chip->mixer.src_stream_out_ofs > MAX_AUDIO_INOUT_COUNT ||
 	    chip->mixer.dest_phys_out_ofs > MAX_STREAM_IN_COUNT) {
@@ -192,6 +213,9 @@ int __devinit lola_init_mixer_widget(struct lola *chip, int nid)
 		(((1U << chip->mixer.dest_phys_outs) - 1)
 		 << chip->mixer.dest_phys_out_ofs);
 
+	snd_printdd("Mixer src_mask=%x, dest_mask=%x\n",
+		    chip->mixer.src_mask, chip->mixer.dest_mask);
+
 	return 0;
 }
 
@@ -202,12 +226,19 @@ static int lola_mixer_set_src_gain(struct lola *chip, unsigned int id,
 
 	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);
+	/* test if values unchanged */
+	if ((val == oldval) &&
+	    (gain == readw(&chip->mixer.array->src_gain[id])))
+		return 0;
+
+	snd_printdd("lola_mixer_set_src_gain (id=%d, gain=%d) enable=%x\n",
+			id, gain, val);
+	writew(gain, &chip->mixer.array->src_gain[id]);
 	writel(val, &chip->mixer.array->src_gain_enable);
 	lola_codec_flush(chip);
 	/* inform micro-controller about the new source gain */
@@ -269,6 +300,7 @@ static int lola_mixer_set_mapping_gain(struct lola *chip,
 				src, dest);
 }
 
+#if 0 /* not used */
 static int lola_mixer_set_dest_gains(struct lola *chip, unsigned int id,
 				     unsigned int mask, unsigned short *gains)
 {
@@ -289,6 +321,7 @@ static int lola_mixer_set_dest_gains(struct lola *chip, unsigned int id,
 	return lola_codec_write(chip, chip->mixer.nid,
 				LOLA_VERB_SET_DESTINATION_GAIN, id, 0);
 }
+#endif /* not used */
 
 /*
  */
@@ -376,6 +409,8 @@ static int set_analog_volume(struct lola *chip, int dir,
 		return 0;
 	if (external_call)
 		lola_codec_flush(chip);
+	snd_printdd("set_analog_volume (dir=%d idx=%d, volume=%d)\n",
+			dir, idx, val);
 	err = lola_codec_write(chip, pin->nid,
 			       LOLA_VERB_SET_AMP_GAIN_MUTE, val, 0);
 	if (err < 0)
@@ -427,23 +462,40 @@ static int init_mixer_values(struct lola *chip)
 {
 	int i;
 
-	/* all src on */
+	/* all sample rate converters on */
 	lola_set_src_config(chip, (1 << chip->pin[CAPT].num_pins) - 1, false);
 
-	/* clear all matrix */
+	/* clear all mixer matrix settings */
 	memset_io(chip->mixer.array, 0, sizeof(*chip->mixer.array));
-	/* set src gain to 0dB */
+	/* inform firmware about all updated matrix columns - capture part */
+	for (i = 0; i < chip->mixer.dest_stream_ins; i++)
+		lola_codec_write(chip, chip->mixer.nid,
+				 LOLA_VERB_SET_DESTINATION_GAIN,
+				 i, 0);
+	/* inform firmware about all updated matrix columns - output part */
+	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);
+
+	/* set all digital input source (master) gains to 0dB */
 	for (i = 0; i < chip->mixer.src_phys_ins; i++)
 		lola_mixer_set_src_gain(chip, i, 336, true); /* 0dB */
+
+	/* set all digital playback source (master) gains to 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 */
+	/* set gain value 0dB diagonally in matrix - part INPUT -> CAPTURE */
 	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);
 	}
+	/* set gain value 0dB diagonally in matrix , part PLAYBACK -> OUTPUT
+	 * (LoLa280 : playback channel 0,2,4,6 linked to output channel 0)
+	 * (LoLa280 : playback channel 1,3,5,7 linked to output channel 1)
+	 */
 	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 +
@@ -693,6 +745,7 @@ static int __devinit create_src_gain_mixer(struct lola *chip,
 			   snd_ctl_new1(&lola_src_gain_mixer, chip));
 }
 
+#if 0 /* not used */
 /*
  * destination gain (matrix-like) mixer
  */
@@ -781,6 +834,7 @@ static int __devinit create_dest_gain_mixer(struct lola *chip,
 	return snd_ctl_add(chip->card,
 			  snd_ctl_new1(&lola_dest_gain_mixer, chip));
 }
+#endif /* not used */
 
 /*
  */
@@ -798,14 +852,16 @@ int __devinit lola_create_mixer(struct lola *chip)
 	if (err < 0)
 		return err;
 	err = create_src_gain_mixer(chip, chip->mixer.src_phys_ins, 0,
-				    "Line Source Gain Volume");
+				    "Digital Capture 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");
+				    "Digital Playback Volume");
 	if (err < 0)
 		return err;
+#if 0
+/* FIXME: buggy mixer matrix handling */
 	err = create_dest_gain_mixer(chip,
 				     chip->mixer.src_phys_ins, 0,
 				     chip->mixer.dest_stream_ins, 0,
@@ -834,6 +890,6 @@ int __devinit lola_create_mixer(struct lola *chip)
 				     "Stream Playback Volume");
 	if (err < 0)
 		return err;
-
+#endif /* FIXME */
 	return init_mixer_values(chip);
 }

sound/pci/lx6464es/lx6464es.c

@@ -762,7 +762,6 @@ static int lx_set_granularity(struct lx6464es *chip, u32 gran)
 static int __devinit lx_init_dsp(struct lx6464es *chip)
 {
 	int err;
-	u8 mac_address[6];
 	int i;
 
 	snd_printdd("->lx_init_dsp\n");
@@ -787,11 +786,11 @@ static int __devinit lx_init_dsp(struct lx6464es *chip)
 	/** \todo the mac address should be ready by not, but it isn't,
 	 *  so we wait for it */
 	for (i = 0; i != 1000; ++i) {
-		err = lx_dsp_get_mac(chip, mac_address);
+		err = lx_dsp_get_mac(chip);
 		if (err)
 			return err;
-		if (mac_address[0] || mac_address[1] || mac_address[2] ||
-		    mac_address[3] || mac_address[4] || mac_address[5])
+		if (chip->mac_address[0] || chip->mac_address[1] || chip->mac_address[2] ||
+		    chip->mac_address[3] || chip->mac_address[4] || chip->mac_address[5])
 			goto mac_ready;
 		msleep(1);
 	}
@@ -800,8 +799,8 @@ static int __devinit lx_init_dsp(struct lx6464es *chip)
 mac_ready:
 	snd_printd(LXP "mac address ready read after: %dms\n", i);
 	snd_printk(LXP "mac address: %02X.%02X.%02X.%02X.%02X.%02X\n",
-		   mac_address[0], mac_address[1], mac_address[2],
-		   mac_address[3], mac_address[4], mac_address[5]);
+		   chip->mac_address[0], chip->mac_address[1], chip->mac_address[2],
+		   chip->mac_address[3], chip->mac_address[4], chip->mac_address[5]);
 
 	err = lx_init_get_version_features(chip);
 	if (err)
@@ -1031,7 +1030,7 @@ static int __devinit snd_lx6464es_create(struct snd_card *card,
 	chip->port_dsp_bar = pci_ioremap_bar(pci, 2);
 
 	err = request_irq(pci->irq, lx_interrupt, IRQF_SHARED,
-			  card_name, chip);
+			  KBUILD_MODNAME, chip);
 	if (err) {
 		snd_printk(KERN_ERR LXP "unable to grab IRQ %d\n", pci->irq);
 		goto request_irq_failed;
@@ -1108,8 +1107,14 @@ static int __devinit snd_lx6464es_probe(struct pci_dev *pci,
 		goto out_free;
 	}
 
-	strcpy(card->driver, "lx6464es");
-	strcpy(card->shortname, "Digigram LX6464ES");
+	strcpy(card->driver, "LX6464ES");
+	sprintf(card->id, "LX6464ES_%02X%02X%02X",
+		chip->mac_address[3], chip->mac_address[4], chip->mac_address[5]);
+
+	sprintf(card->shortname, "LX6464ES %02X.%02X.%02X.%02X.%02X.%02X",
+		chip->mac_address[0], chip->mac_address[1], chip->mac_address[2],
+		chip->mac_address[3], chip->mac_address[4], chip->mac_address[5]);
+
 	sprintf(card->longname, "%s at 0x%lx, 0x%p, irq %i",
 		card->shortname, chip->port_plx,
 		chip->port_dsp_bar, chip->irq);
@@ -1137,7 +1142,7 @@ static void __devexit snd_lx6464es_remove(struct pci_dev *pci)
 
 
 static struct pci_driver driver = {
-	.name =     "Digigram LX6464ES",
+	.name =     KBUILD_MODNAME,
 	.id_table = snd_lx6464es_ids,
 	.probe =    snd_lx6464es_probe,
 	.remove = __devexit_p(snd_lx6464es_remove),

sound/pci/lx6464es/lx6464es.h

@@ -69,6 +69,8 @@ struct lx6464es {
 	struct pci_dev         *pci;
 	int			irq;
 
+	u8			mac_address[6];
+
 	spinlock_t		lock;        /* interrupt spinlock */
 	struct mutex            setup_mutex; /* mutex used in hw_params, open
 					      * and close */