Merge branch 'x86/urgent' into core/signal

Conflicts:
	arch/x86/kernel/signal_64.c

16 files changed, 4628 insertions, 446 deletions

diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index ab0c726d648e..1980c6d207e7 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -5,6 +5,7 @@ snd-hda-intel-y := hda_intel.o
 snd-hda-intel-y += hda_codec.o
 snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o
 snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
+snd-hda-intel-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o
 snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o
 snd-hda-intel-$(CONFIG_SND_HDA_CODEC_REALTEK) += patch_realtek.o
 snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CMEDIA) += patch_cmedia.o
@@ -14,5 +15,6 @@ snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SI3054) += patch_si3054.o
 snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ATIHDMI) += patch_atihdmi.o
 snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CONEXANT) += patch_conexant.o
 snd-hda-intel-$(CONFIG_SND_HDA_CODEC_VIA) += patch_via.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_NVHDMI) += patch_nvhdmi.o
 
 obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
new file mode 100644
index 000000000000..9b77b3e0fa98
--- /dev/null
+++ b/sound/pci/hda/hda_beep.c
@@ -0,0 +1,134 @@
+/*
+ * Digital Beep Input Interface for HD-audio codec
+ *
+ * Author: Matthew Ranostay <mranostay@embeddedalley.com>
+ * Copyright (c) 2008 Embedded Alley Solutions Inc
+ *
+ *  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/input.h>
+#include <linux/pci.h>
+#include <linux/workqueue.h>
+#include <sound/core.h>
+#include "hda_beep.h"
+
+enum {
+	DIGBEEP_HZ_STEP = 46875,	/* 46.875 Hz */
+	DIGBEEP_HZ_MIN = 93750,		/* 93.750 Hz */
+	DIGBEEP_HZ_MAX = 12000000,	/* 12 KHz */
+};
+
+static void snd_hda_generate_beep(struct work_struct *work)
+{
+	struct hda_beep *beep =
+		container_of(work, struct hda_beep, beep_work);
+	struct hda_codec *codec = beep->codec;
+
+	/* generate tone */
+	snd_hda_codec_write_cache(codec, beep->nid, 0,
+			AC_VERB_SET_BEEP_CONTROL, beep->tone);
+}
+
+static int snd_hda_beep_event(struct input_dev *dev, unsigned int type,
+				unsigned int code, int hz)
+{
+	struct hda_beep *beep = input_get_drvdata(dev);
+
+	switch (code) {
+	case SND_BELL:
+		if (hz)
+			hz = 1000;
+	case SND_TONE:
+		hz *= 1000; /* fixed point */
+		hz = hz - DIGBEEP_HZ_MIN;
+		if (hz < 0)
+			hz = 0; /* turn off PC beep*/
+		else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN))
+			hz = 0xff;
+		else {
+			hz /= DIGBEEP_HZ_STEP;
+			hz++;
+		}
+		break;
+	default:
+		return -1;
+	}
+	beep->tone = hz;
+
+	/* schedule beep event */
+	schedule_work(&beep->beep_work);
+	return 0;
+}
+
+int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
+{
+	struct input_dev *input_dev;
+	struct hda_beep *beep;
+	int err;
+
+	beep = kzalloc(sizeof(*beep), GFP_KERNEL);
+	if (beep == NULL)
+		return -ENOMEM;
+	snprintf(beep->phys, sizeof(beep->phys),
+		"card%d/codec#%d/beep0", codec->bus->card->number, codec->addr);
+	input_dev = input_allocate_device();
+
+	/* setup digital beep device */
+	input_dev->name = "HDA Digital PCBeep";
+	input_dev->phys = beep->phys;
+	input_dev->id.bustype = BUS_PCI;
+
+	input_dev->id.vendor = codec->vendor_id >> 16;
+	input_dev->id.product = codec->vendor_id & 0xffff;
+	input_dev->id.version = 0x01;
+
+	input_dev->evbit[0] = BIT_MASK(EV_SND);
+	input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE);
+	input_dev->event = snd_hda_beep_event;
+	input_dev->dev.parent = &codec->bus->pci->dev;
+	input_set_drvdata(input_dev, beep);
+
+	err = input_register_device(input_dev);
+	if (err < 0) {
+		input_free_device(input_dev);
+		kfree(beep);
+		return err;
+	}
+
+	/* enable linear scale */
+	snd_hda_codec_write(codec, nid, 0,
+		AC_VERB_SET_DIGI_CONVERT_2, 0x01);
+
+	beep->nid = nid;
+	beep->dev = input_dev;
+	beep->codec = codec;
+	codec->beep = beep;
+
+	INIT_WORK(&beep->beep_work, &snd_hda_generate_beep);
+	return 0;
+}
+
+void snd_hda_detach_beep_device(struct hda_codec *codec)
+{
+	struct hda_beep *beep = codec->beep;
+	if (beep) {
+		cancel_work_sync(&beep->beep_work);
+		flush_scheduled_work();
+
+		input_unregister_device(beep->dev);
+		kfree(beep);
+	}
+}
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
new file mode 100644
index 000000000000..de4036e6e710
--- /dev/null
+++ b/sound/pci/hda/hda_beep.h
@@ -0,0 +1,44 @@
+/*
+ * Digital Beep Input Interface for HD-audio codec
+ *
+ * Author: Matthew Ranostay <mranostay@embeddedalley.com>
+ * Copyright (c) 2008 Embedded Alley Solutions Inc
+ *
+ *  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
+ */
+
+#ifndef __SOUND_HDA_BEEP_H
+#define __SOUND_HDA_BEEP_H
+
+#include "hda_codec.h"
+
+/* beep information */
+struct hda_beep {
+	struct input_dev *dev;
+	struct hda_codec *codec;
+	char phys[32];
+	int tone;
+	int nid;
+	struct work_struct beep_work; /* scheduled task for beep event */
+};
+
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+int snd_hda_attach_beep_device(struct hda_codec *codec, int nid);
+void snd_hda_detach_beep_device(struct hda_codec *codec);
+#else
+#define snd_hda_attach_beep_device(...)
+#define snd_hda_detach_beep_device(...)
+#endif
+#endif
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index d2e1093f8e97..6447754ae56e 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -94,6 +94,9 @@ static const struct hda_codec_preset *hda_preset_tables[] = {
 #ifdef CONFIG_SND_HDA_CODEC_VIA
 	snd_hda_preset_via,
 #endif
+#ifdef CONFIG_SND_HDA_CODEC_NVHDMI
+	snd_hda_preset_nvhdmi,
+#endif
 	NULL
 };
 
@@ -211,7 +214,8 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
 	unsigned int shift, num_elems, mask;
 	hda_nid_t prev_nid;
 
-	snd_assert(conn_list && max_conns > 0, return -EINVAL);
+	if (snd_BUG_ON(!conn_list || max_conns <= 0))
+		return -EINVAL;
 
 	parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN);
 	if (parm & AC_CLIST_LONG) {
@@ -313,7 +317,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex)
 }
 
 /*
- * process queueud unsolicited events
+ * process queued unsolicited events
  */
 static void process_unsol_events(struct work_struct *work)
 {
@@ -407,8 +411,10 @@ int __devinit snd_hda_bus_new(struct snd_card *card,
 		.dev_free = snd_hda_bus_dev_free,
 	};
 
-	snd_assert(temp, return -EINVAL);
-	snd_assert(temp->ops.command && temp->ops.get_response, return -EINVAL);
+	if (snd_BUG_ON(!temp))
+		return -EINVAL;
+	if (snd_BUG_ON(!temp->ops.command || !temp->ops.get_response))
+		return -EINVAL;
 
 	if (busp)
 		*busp = NULL;
@@ -585,11 +591,13 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
 				struct hda_codec **codecp)
 {
 	struct hda_codec *codec;
-	char component[13];
+	char component[31];
 	int err;
 
-	snd_assert(bus, return -EINVAL);
-	snd_assert(codec_addr <= HDA_MAX_CODEC_ADDRESS, return -EINVAL);
+	if (snd_BUG_ON(!bus))
+		return -EINVAL;
+	if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS))
+		return -EINVAL;
 
 	if (bus->caddr_tbl[codec_addr]) {
 		snd_printk(KERN_ERR "hda_codec: "
@@ -688,7 +696,7 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
 	snd_hda_create_hwdep(codec);
 #endif
 
-	sprintf(component, "HDA:%08x", codec->vendor_id);
+	sprintf(component, "HDA:%08x,%08x,%08x", codec->vendor_id, codec->subsystem_id, codec->revision_id);
 	snd_component_add(codec->bus->card, component);
 
 	if (codecp)
@@ -956,15 +964,6 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec)
 }
 #endif /* SND_HDA_NEEDS_RESUME */
 
-/*
- * AMP control callbacks
- */
-/* retrieve parameters from private_value */
-#define get_amp_nid(kc)		((kc)->private_value & 0xffff)
-#define get_amp_channels(kc)	(((kc)->private_value >> 16) & 0x3)
-#define get_amp_direction(kc)	(((kc)->private_value >> 18) & 0x1)
-#define get_amp_index(kc)	(((kc)->private_value >> 19) & 0xf)
-
 /* volume */
 int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_info *uinfo)
@@ -1430,6 +1429,29 @@ static unsigned int convert_to_spdif_status(unsigned short val)
 	return sbits;
 }
 
+/* set digital convert verbs both for the given NID and its slaves */
+static void set_dig_out(struct hda_codec *codec, hda_nid_t nid,
+			int verb, int val)
+{
+	hda_nid_t *d;
+
+	snd_hda_codec_write(codec, nid, 0, verb, val);
+	d = codec->slave_dig_outs;
+	if (!d)
+		return;
+	for (; *d; d++)
+		snd_hda_codec_write(codec, *d, 0, verb, val);
+}
+
+static inline void set_dig_out_convert(struct hda_codec *codec, hda_nid_t nid,
+				       int dig1, int dig2)
+{
+	if (dig1 != -1)
+		set_dig_out(codec, nid, AC_VERB_SET_DIGI_CONVERT_1, dig1);
+	if (dig2 != -1)
+		set_dig_out(codec, nid, AC_VERB_SET_DIGI_CONVERT_2, dig2);
+}
+
 static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol,
 				     struct snd_ctl_elem_value *ucontrol)
 {
@@ -1448,14 +1470,8 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol,
 	change = codec->spdif_ctls != val;
 	codec->spdif_ctls = val;
 
-	if (change) {
-		snd_hda_codec_write_cache(codec, nid, 0,
-					  AC_VERB_SET_DIGI_CONVERT_1,
-					  val & 0xff);
-		snd_hda_codec_write_cache(codec, nid, 0,
-					  AC_VERB_SET_DIGI_CONVERT_2,
-					  val >> 8);
-	}
+	if (change)
+		set_dig_out_convert(codec, nid, val & 0xff, (val >> 8) & 0xff);
 
 	mutex_unlock(&codec->spdif_mutex);
 	return change;
@@ -1487,9 +1503,7 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol,
 	change = codec->spdif_ctls != val;
 	if (change) {
 		codec->spdif_ctls = val;
-		snd_hda_codec_write_cache(codec, nid, 0,
-					  AC_VERB_SET_DIGI_CONVERT_1,
-					  val & 0xff);
+		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))
@@ -2236,11 +2250,13 @@ static int __devinit set_pcm_default_values(struct hda_codec *codec,
 	if (info->ops.close == NULL)
 		info->ops.close = hda_pcm_default_open_close;
 	if (info->ops.prepare == NULL) {
-		snd_assert(info->nid, return -EINVAL);
+		if (snd_BUG_ON(!info->nid))
+			return -EINVAL;
 		info->ops.prepare = hda_pcm_default_prepare;
 	}
 	if (info->ops.cleanup == NULL) {
-		snd_assert(info->nid, return -EINVAL);
+		if (snd_BUG_ON(!info->nid))
+			return -EINVAL;
 		info->ops.cleanup = hda_pcm_default_cleanup;
 	}
 	return 0;
@@ -2583,14 +2599,31 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid,
 				 unsigned int stream_tag, unsigned int format)
 {
 	/* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
-	if (codec->spdif_ctls & AC_DIG1_ENABLE)
-		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
-				    codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff);
+	if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
+		set_dig_out_convert(codec, nid, 
+				    codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff,
+				    -1);
 	snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
+	if (codec->slave_dig_outs) {
+		hda_nid_t *d;
+		for (d = codec->slave_dig_outs; *d; d++)
+			snd_hda_codec_setup_stream(codec, *d, stream_tag, 0,
+						   format);
+	}
 	/* turn on again (if needed) */
-	if (codec->spdif_ctls & AC_DIG1_ENABLE)
-		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
-				    codec->spdif_ctls & 0xff);
+	if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
+		set_dig_out_convert(codec, nid,
+				    codec->spdif_ctls & 0xff, -1);
+}
+
+static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid)
+{
+	snd_hda_codec_cleanup_stream(codec, nid);
+	if (codec->slave_dig_outs) {
+		hda_nid_t *d;
+		for (d = codec->slave_dig_outs; *d; d++)
+			snd_hda_codec_cleanup_stream(codec, *d);
+	}
 }
 
 /*
@@ -2602,7 +2635,7 @@ int snd_hda_multi_out_dig_open(struct hda_codec *codec,
 	mutex_lock(&codec->spdif_mutex);
 	if (mout->dig_out_used == HDA_DIG_ANALOG_DUP)
 		/* already opened as analog dup; reset it once */
-		snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid);
+		cleanup_dig_out_stream(codec, mout->dig_out_nid);
 	mout->dig_out_used = HDA_DIG_EXCLUSIVE;
 	mutex_unlock(&codec->spdif_mutex);
 	return 0;
@@ -2697,7 +2730,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
 					     stream_tag, format);
 		} else {
 			mout->dig_out_used = 0;
-			snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid);
+			cleanup_dig_out_stream(codec, mout->dig_out_nid);
 		}
 	}
 	mutex_unlock(&codec->spdif_mutex);
@@ -2748,7 +2781,7 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
 						     mout->extra_out_nid[i]);
 	mutex_lock(&codec->spdif_mutex);
 	if (mout->dig_out_nid && mout->dig_out_used == HDA_DIG_ANALOG_DUP) {
-		snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid);
+		cleanup_dig_out_stream(codec, mout->dig_out_nid);
 		mout->dig_out_used = 0;
 	}
 	mutex_unlock(&codec->spdif_mutex);
@@ -2756,7 +2789,7 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
 }
 
 /*
- * Helper for automatic ping configuration
+ * Helper for automatic pin configuration
  */
 
 static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list)
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index efc682888b31..60468f562400 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -90,6 +90,14 @@ enum {
 #define AC_VERB_GET_CONFIG_DEFAULT		0x0f1c
 /* f20: AFG/MFG */
 #define AC_VERB_GET_SUBSYSTEM_ID		0x0f20
+#define AC_VERB_GET_CVT_CHAN_COUNT		0x0f2d
+#define AC_VERB_GET_HDMI_DIP_SIZE		0x0f2e
+#define AC_VERB_GET_HDMI_ELDD			0x0f2f
+#define AC_VERB_GET_HDMI_DIP_INDEX		0x0f30
+#define AC_VERB_GET_HDMI_DIP_DATA		0x0f31
+#define AC_VERB_GET_HDMI_DIP_XMIT		0x0f32
+#define AC_VERB_GET_HDMI_CP_CTRL		0x0f33
+#define AC_VERB_GET_HDMI_CHAN_SLOT		0x0f34
 
 /*
  * SET verbs
@@ -121,7 +129,14 @@ enum {
 #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1	0x71d
 #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2	0x71e
 #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3	0x71f
+#define AC_VERB_SET_EAPD				0x788
 #define AC_VERB_SET_CODEC_RESET			0x7ff
+#define AC_VERB_SET_CVT_CHAN_COUNT		0x72d
+#define AC_VERB_SET_HDMI_DIP_INDEX		0x730
+#define AC_VERB_SET_HDMI_DIP_DATA		0x731
+#define AC_VERB_SET_HDMI_DIP_XMIT		0x732
+#define AC_VERB_SET_HDMI_CP_CTRL		0x733
+#define AC_VERB_SET_HDMI_CHAN_SLOT		0x734
 
 /*
  * Parameter IDs
@@ -143,6 +158,7 @@ enum {
 #define AC_PAR_GPIO_CAP			0x11
 #define AC_PAR_AMP_OUT_CAP		0x12
 #define AC_PAR_VOL_KNB_CAP		0x13
+#define AC_PAR_HDMI_LPCM_CAP		0x20
 
 /*
  * AC_VERB_PARAMETERS results (32bit)
@@ -171,6 +187,8 @@ enum {
 #define AC_WCAP_DIGITAL			(1<<9)	/* digital I/O */
 #define AC_WCAP_POWER			(1<<10)	/* power control */
 #define AC_WCAP_LR_SWAP			(1<<11)	/* L/R swap */
+#define AC_WCAP_CP_CAPS			(1<<12) /* content protection */
+#define AC_WCAP_CHAN_CNT_EXT		(7<<13)	/* channel count ext */
 #define AC_WCAP_DELAY			(0xf<<16)
 #define AC_WCAP_DELAY_SHIFT		16
 #define AC_WCAP_TYPE			(0xf<<20)
@@ -206,9 +224,20 @@ enum {
 /* Input converter SDI select */
 #define AC_SDI_SELECT			(0xf<<0)
 
-/* Unsolicited response */
+/* Unsolicited response control */
 #define AC_UNSOL_TAG			(0x3f<<0)
 #define AC_UNSOL_ENABLED		(1<<7)
+#define AC_USRSP_EN			AC_UNSOL_ENABLED
+
+/* Unsolicited responses */
+#define AC_UNSOL_RES_TAG		(0x3f<<26)
+#define AC_UNSOL_RES_TAG_SHIFT		26
+#define AC_UNSOL_RES_SUBTAG		(0x1f<<21)
+#define AC_UNSOL_RES_SUBTAG_SHIFT	21
+#define AC_UNSOL_RES_ELDV		(1<<1)	/* ELD Data valid (for HDMI) */
+#define AC_UNSOL_RES_PD			(1<<0)	/* pinsense detect */
+#define AC_UNSOL_RES_CP_STATE		(1<<1)	/* content protection */
+#define AC_UNSOL_RES_CP_READY		(1<<0)	/* content protection */
 
 /* Pin widget capabilies */
 #define AC_PINCAP_IMP_SENSE		(1<<0)	/* impedance sense capable */
@@ -222,6 +251,10 @@ enum {
  *       but is marked reserved in the Intel HDA specification.
  */
 #define AC_PINCAP_LR_SWAP		(1<<7)	/* L/R swap */
+/* Note: The same bit as LR_SWAP is newly defined as HDMI capability
+ *       in HD-audio specification
+ */
+#define AC_PINCAP_HDMI			(1<<7)	/* HDMI pin */
 #define AC_PINCAP_VREF			(0x37<<8)
 #define AC_PINCAP_VREF_SHIFT		8
 #define AC_PINCAP_EAPD			(1<<16)	/* EAPD capable */
@@ -272,6 +305,22 @@ enum {
 #define AC_KNBCAP_NUM_STEPS		(0x7f<<0)
 #define AC_KNBCAP_DELTA			(1<<7)
 
+/* HDMI LPCM capabilities */
+#define AC_LPCMCAP_48K_CP_CHNS		(0x0f<<0) /* max channels w/ CP-on */	
+#define AC_LPCMCAP_48K_NO_CHNS		(0x0f<<4) /* max channels w/o CP-on */
+#define AC_LPCMCAP_48K_20BIT		(1<<8)	/* 20b bitrate supported */
+#define AC_LPCMCAP_48K_24BIT		(1<<9)	/* 24b bitrate supported */
+#define AC_LPCMCAP_96K_CP_CHNS		(0x0f<<10) /* max channels w/ CP-on */	
+#define AC_LPCMCAP_96K_NO_CHNS		(0x0f<<14) /* max channels w/o CP-on */
+#define AC_LPCMCAP_96K_20BIT		(1<<18)	/* 20b bitrate supported */
+#define AC_LPCMCAP_96K_24BIT		(1<<19)	/* 24b bitrate supported */
+#define AC_LPCMCAP_192K_CP_CHNS		(0x0f<<20) /* max channels w/ CP-on */	
+#define AC_LPCMCAP_192K_NO_CHNS		(0x0f<<24) /* max channels w/o CP-on */
+#define AC_LPCMCAP_192K_20BIT		(1<<28)	/* 20b bitrate supported */
+#define AC_LPCMCAP_192K_24BIT		(1<<29)	/* 24b bitrate supported */
+#define AC_LPCMCAP_44K			(1<<30)	/* 44.1kHz support */
+#define AC_LPCMCAP_44K_MS		(1<<31)	/* 44.1kHz-multiplies support */
+
 /*
  * Control Parameters
  */
@@ -317,18 +366,44 @@ enum {
 #define AC_PINCTL_OUT_EN		(1<<6)
 #define AC_PINCTL_HP_EN			(1<<7)
 
-/* Unsolicited response - 8bit */
-#define AC_USRSP_EN			(1<<7)
-
 /* Pin sense - 32bit */
 #define AC_PINSENSE_IMPEDANCE_MASK	(0x7fffffff)
 #define AC_PINSENSE_PRESENCE		(1<<31)
+#define AC_PINSENSE_ELDV		(1<<30)	/* ELD valid (HDMI) */
 
 /* EAPD/BTL enable - 32bit */
 #define AC_EAPDBTL_BALANCED		(1<<0)
 #define AC_EAPDBTL_EAPD			(1<<1)
 #define AC_EAPDBTL_LR_SWAP		(1<<2)
 
+/* HDMI ELD data */
+#define AC_ELDD_ELD_VALID		(1<<31)
+#define AC_ELDD_ELD_DATA		0xff
+
+/* HDMI DIP size */
+#define AC_DIPSIZE_ELD_BUF		(1<<3) /* ELD buf size of packet size */
+#define AC_DIPSIZE_PACK_IDX		(0x07<<0) /* packet index */
+
+/* HDMI DIP index */
+#define AC_DIPIDX_PACK_IDX		(0x07<<5) /* packet idnex */
+#define AC_DIPIDX_BYTE_IDX		(0x1f<<0) /* byte index */
+
+/* HDMI DIP xmit (transmit) control */
+#define AC_DIPXMIT_MASK			(0x3<<6)
+#define AC_DIPXMIT_DISABLE		(0x0<<6) /* disable xmit */
+#define AC_DIPXMIT_ONCE			(0x2<<6) /* xmit once then disable */
+#define AC_DIPXMIT_BEST			(0x3<<6) /* best effort */
+
+/* HDMI content protection (CP) control */
+#define AC_CPCTRL_CES			(1<<9) /* current encryption state */
+#define AC_CPCTRL_READY			(1<<8) /* ready bit */
+#define AC_CPCTRL_SUBTAG		(0x1f<<3) /* subtag for unsol-resp */
+#define AC_CPCTRL_STATE			(3<<0) /* current CP request state */
+
+/* Converter channel <-> HDMI slot mapping */
+#define AC_CVTMAP_HDMI_SLOT		(0xf<<0) /* HDMI slot number */
+#define AC_CVTMAP_CHAN			(0xf<<4) /* converter channel number */
+
 /* configuration default - 32bit */
 #define AC_DEFCFG_SEQUENCE		(0xf<<0)
 #define AC_DEFCFG_DEF_ASSOC		(0xf<<4)
@@ -449,6 +524,7 @@ enum {
  */
 
 struct hda_bus;
+struct hda_beep;
 struct hda_codec;
 struct hda_pcm;
 struct hda_pcm_stream;
@@ -634,6 +710,9 @@ struct hda_codec {
 	/* codec specific info */
 	void *spec;
 
+	/* beep device */
+	struct hda_beep *beep;
+
 	/* widget capabilities cache */
 	unsigned int num_nodes;
 	hda_nid_t start_nid;
@@ -646,9 +725,15 @@ struct hda_codec {
 	unsigned int spdif_status;	/* IEC958 status bits */
 	unsigned short spdif_ctls;	/* SPDIF control bits */
 	unsigned int spdif_in_enable;	/* SPDIF input enable? */
+	hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
 
 	struct snd_hwdep *hwdep;	/* assigned hwdep device */
 
+	/* misc flags */
+	unsigned int spdif_status_reset :1; /* needs to toggle SPDIF for each
+					     * status change
+					     * (e.g. Realtek codecs)
+					     */
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	unsigned int power_on :1;	/* current (global) power-state */
 	unsigned int power_transition :1; /* power-state in transition */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 59e4389c94a4..0ca30894f7c6 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -174,7 +174,8 @@ static int build_afg_tree(struct hda_codec *codec)
 	int i, nodes, err;
 	hda_nid_t nid;
 
-	snd_assert(spec, return -EINVAL);
+	if (snd_BUG_ON(!spec))
+		return -EINVAL;
 
 	spec->def_amp_out_caps = snd_hda_param_read(codec, codec->afg, AC_PAR_AMP_OUT_CAP);
 	spec->def_amp_in_caps = snd_hda_param_read(codec, codec->afg, AC_PAR_AMP_IN_CAP);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 1c53e337ecb2..9f316c1b2790 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -222,9 +222,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
 #define RIRB_INT_OVERRUN	0x04
 #define RIRB_INT_MASK		0x05
 
-/* STATESTS int mask: SD2,SD1,SD0 */
-#define AZX_MAX_CODECS		3
-#define STATESTS_INT_MASK	0x07
+/* STATESTS int mask: S3,SD2,SD1,SD0 */
+#define AZX_MAX_CODECS		4
+#define STATESTS_INT_MASK	0x0f
 
 /* SD_CTL bits */
 #define SD_CTL_STREAM_RESET	0x01	/* stream reset bit */
@@ -286,6 +286,11 @@ enum {
 #define INTEL_SCH_HDA_DEVC      0x78
 #define INTEL_SCH_HDA_DEVC_NOSNOOP       (0x1<<11)
 
+/* Define IN stream 0 FIFO size offset in VIA controller */
+#define VIA_IN_STREAM0_FIFO_SIZE_OFFSET	0x90
+/* Define VIA HD Audio Device ID*/
+#define VIA_HDAC_DEVICE_ID		0x3288
+
 
 /*
  */
@@ -317,6 +322,12 @@ struct azx_dev {
 	unsigned int running :1;
 	unsigned int irq_pending :1;
 	unsigned int irq_ignore :1;
+	/*
+	 * For VIA:
+	 *  A flag to ensure DMA position is 0
+	 *  when link position is not greater than FIFO size
+	 */
+	unsigned int insufficient :1;
 };
 
 /* CORB/RIRB */
@@ -379,6 +390,7 @@ struct azx {
 	unsigned int polling_mode :1;
 	unsigned int msi :1;
 	unsigned int irq_pending_warned :1;
+	unsigned int via_dmapos_patch :1; /* enable DMA-position fix for VIA */
 
 	/* for debugging */
 	unsigned int last_cmd;	/* last issued command (to sync) */
@@ -398,6 +410,7 @@ enum {
 	AZX_DRIVER_ULI,
 	AZX_DRIVER_NVIDIA,
 	AZX_DRIVER_TERA,
+	AZX_NUM_DRIVERS, /* keep this as last entry */
 };
 
 static char *driver_short_names[] __devinitdata = {
@@ -818,6 +831,11 @@ static void azx_int_clear(struct azx *chip)
 /* start a stream */
 static void azx_stream_start(struct azx *chip, struct azx_dev *azx_dev)
 {
+	/*
+	 * Before stream start, initialize parameter
+	 */
+	azx_dev->insufficient = 1;
+
 	/* enable SIE */
 	azx_writeb(chip, INTCTL,
 		   azx_readb(chip, INTCTL) | (1 << azx_dev->index));
@@ -998,7 +1016,6 @@ static int setup_bdle(struct snd_pcm_substream *substream,
 		      struct azx_dev *azx_dev, u32 **bdlp,
 		      int ofs, int size, int with_ioc)
 {
-	struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream);
 	u32 *bdl = *bdlp;
 
 	while (size > 0) {
@@ -1008,14 +1025,12 @@ static int setup_bdle(struct snd_pcm_substream *substream,
 		if (azx_dev->frags >= AZX_MAX_BDL_ENTRIES)
 			return -EINVAL;
 
-		addr = snd_pcm_sgbuf_get_addr(sgbuf, ofs);
+		addr = snd_pcm_sgbuf_get_addr(substream, ofs);
 		/* program the address field of the BDL entry */
 		bdl[0] = cpu_to_le32((u32)addr);
 		bdl[1] = cpu_to_le32(upper_32_bits(addr));
 		/* program the size field of the BDL entry */
-		chunk = PAGE_SIZE - (ofs % PAGE_SIZE);
-		if (size < chunk)
-			chunk = size;
+		chunk = snd_pcm_sgbuf_get_chunk_size(substream, ofs, size);
 		bdl[2] = cpu_to_le32(chunk);
 		/* program the IOC to enable interrupt
 		 * only when the whole fragment is processed
@@ -1151,7 +1166,8 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
 
 	/* enable the position buffer */
 	if (chip->position_fix == POS_FIX_POSBUF ||
-	    chip->position_fix == POS_FIX_AUTO) {
+	    chip->position_fix == POS_FIX_AUTO ||
+	    chip->via_dmapos_patch) {
 		if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE))
 			azx_writel(chip, DPLBASE,
 				(u32)chip->posbuf.addr | ICH6_DPLBASE_ENABLE);
@@ -1169,23 +1185,26 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
  * Codec initialization
  */
 
-static unsigned int azx_max_codecs[] __devinitdata = {
-	[AZX_DRIVER_ICH] = 4,		/* Some ICH9 boards use SD3 */
-	[AZX_DRIVER_SCH] = 3,
-	[AZX_DRIVER_ATI] = 4,
-	[AZX_DRIVER_ATIHDMI] = 4,
-	[AZX_DRIVER_VIA] = 3,		/* FIXME: correct? */
-	[AZX_DRIVER_SIS] = 3,		/* FIXME: correct? */
-	[AZX_DRIVER_ULI] = 3,		/* FIXME: correct? */
-	[AZX_DRIVER_NVIDIA] = 3,	/* FIXME: correct? */
+/* number of codec slots for each chipset: 0 = default slots (i.e. 4) */
+static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = {
 	[AZX_DRIVER_TERA] = 1,
 };
 
+/* number of slots to probe as default
+ * this can be different from azx_max_codecs[] -- e.g. some boards
+ * report wrongly the non-existing 4th slot availability
+ */
+static unsigned int azx_default_codecs[AZX_NUM_DRIVERS] __devinitdata = {
+	[AZX_DRIVER_ICH] = 3,
+	[AZX_DRIVER_ATI] = 3,
+};
+
 static int __devinit azx_codec_create(struct azx *chip, const char *model,
 				      unsigned int codec_probe_mask)
 {
 	struct hda_bus_template bus_temp;
 	int c, codecs, audio_codecs, err;
+	int def_slots, max_slots;
 
 	memset(&bus_temp, 0, sizeof(bus_temp));
 	bus_temp.private_data = chip;
@@ -1201,8 +1220,17 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model,
 	if (err < 0)
 		return err;
 
+	if (chip->driver_type == AZX_DRIVER_NVIDIA)
+		chip->bus->needs_damn_long_delay = 1;
+
 	codecs = audio_codecs = 0;
-	for (c = 0; c < AZX_MAX_CODECS; c++) {
+	max_slots = azx_max_codecs[chip->driver_type];
+	if (!max_slots)
+		max_slots = AZX_MAX_CODECS;
+	def_slots = azx_default_codecs[chip->driver_type];
+	if (!def_slots)
+		def_slots = max_slots;
+	for (c = 0; c < def_slots; c++) {
 		if ((chip->codec_mask & (1 << c)) & codec_probe_mask) {
 			struct hda_codec *codec;
 			err = snd_hda_codec_new(chip->bus, c, &codec);
@@ -1215,7 +1243,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model,
 	}
 	if (!audio_codecs) {
 		/* probe additional slots if no codec is found */
-		for (; c < azx_max_codecs[chip->driver_type]; c++) {
+		for (; c < max_slots; c++) {
 			if ((chip->codec_mask & (1 << c)) & codec_probe_mask) {
 				err = snd_hda_codec_new(chip->bus, c, NULL);
 				if (err < 0)
@@ -1507,13 +1535,71 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	return 0;
 }
 
+/* get the current DMA position with correction on VIA chips */
+static unsigned int azx_via_get_position(struct azx *chip,
+					 struct azx_dev *azx_dev)
+{
+	unsigned int link_pos, mini_pos, bound_pos;
+	unsigned int mod_link_pos, mod_dma_pos, mod_mini_pos;
+	unsigned int fifo_size;
+
+	link_pos = azx_sd_readl(azx_dev, SD_LPIB);
+	if (azx_dev->index >= 4) {
+		/* Playback, no problem using link position */
+		return link_pos;
+	}
+
+	/* Capture */
+	/* For new chipset,
+	 * use mod to get the DMA position just like old chipset
+	 */
+	mod_dma_pos = le32_to_cpu(*azx_dev->posbuf);
+	mod_dma_pos %= azx_dev->period_bytes;
+
+	/* azx_dev->fifo_size can't get FIFO size of in stream.
+	 * Get from base address + offset.
+	 */
+	fifo_size = readw(chip->remap_addr + VIA_IN_STREAM0_FIFO_SIZE_OFFSET);
+
+	if (azx_dev->insufficient) {
+		/* Link position never gather than FIFO size */
+		if (link_pos <= fifo_size)
+			return 0;
+
+		azx_dev->insufficient = 0;
+	}
+
+	if (link_pos <= fifo_size)
+		mini_pos = azx_dev->bufsize + link_pos - fifo_size;
+	else
+		mini_pos = link_pos - fifo_size;
+
+	/* Find nearest previous boudary */
+	mod_mini_pos = mini_pos % azx_dev->period_bytes;
+	mod_link_pos = link_pos % azx_dev->period_bytes;
+	if (mod_link_pos >= fifo_size)
+		bound_pos = link_pos - mod_link_pos;
+	else if (mod_dma_pos >= mod_mini_pos)
+		bound_pos = mini_pos - mod_mini_pos;
+	else {
+		bound_pos = mini_pos - mod_mini_pos + azx_dev->period_bytes;
+		if (bound_pos >= azx_dev->bufsize)
+			bound_pos = 0;
+	}
+
+	/* Calculate real DMA position we want */
+	return bound_pos + mod_dma_pos;
+}
+
 static unsigned int azx_get_position(struct azx *chip,
 				     struct azx_dev *azx_dev)
 {
 	unsigned int pos;
 
-	if (chip->position_fix == POS_FIX_POSBUF ||
-	    chip->position_fix == POS_FIX_AUTO) {
+	if (chip->via_dmapos_patch)
+		pos = azx_via_get_position(chip, azx_dev);
+	else if (chip->position_fix == POS_FIX_POSBUF ||
+		 chip->position_fix == POS_FIX_AUTO) {
 		/* use the position buffer */
 		pos = le32_to_cpu(*azx_dev->posbuf);
 	} else {
@@ -1559,6 +1645,8 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
 			chip->position_fix = POS_FIX_POSBUF;
 	}
 
+	if (!bdl_pos_adj[chip->dev_index])
+		return 1; /* no delayed ack */
 	if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2)
 		return 0; /* NG - it's below the period boundary */
 	return 1; /* OK, it's fine */
@@ -1646,7 +1734,8 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec,
 	if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams)
 		return 0;
 
-	snd_assert(cpcm->name, return -EINVAL);
+	if (snd_BUG_ON(!cpcm->name))
+		return -EINVAL;
 
 	err = snd_pcm_new(chip->card, cpcm->name, cpcm->device,
 			  cpcm->stream[0].substreams,
@@ -1670,7 +1759,7 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec,
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &azx_pcm_ops);
 	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
 					      snd_dma_pci_data(chip->pci),
-					      1024 * 64, 1024 * 1024);
+					      1024 * 64, 32 * 1024 * 1024);
 	chip->pcm[cpcm->device] = pcm;
 	return 0;
 }
@@ -1946,6 +2035,15 @@ static int __devinit check_position_fix(struct azx *chip, int fix)
 {
 	const struct snd_pci_quirk *q;
 
+	/* Check VIA HD Audio Controller exist */
+	if (chip->pci->vendor == PCI_VENDOR_ID_VIA &&
+	    chip->pci->device == VIA_HDAC_DEVICE_ID) {
+		chip->via_dmapos_patch = 1;
+		/* Use link position directly, avoid any transfer problem. */
+		return POS_FIX_LPIB;
+	}
+	chip->via_dmapos_patch = 0;
+
 	if (fix == POS_FIX_AUTO) {
 		q = snd_pci_quirk_lookup(chip->pci, position_fix_list);
 		if (q) {
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 5c9e578f7f2d..7957fefda730 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -368,12 +368,15 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 #define AMP_OUT_UNMUTE	0xb000
 #define AMP_OUT_ZERO	0xb000
 /* pinctl values */
-#define PIN_IN		0x20
-#define PIN_VREF80	0x24
-#define PIN_VREF50	0x21
-#define PIN_OUT		0x40
-#define PIN_HP		0xc0
-#define PIN_HP_AMP	0x80
+#define PIN_IN			(AC_PINCTL_IN_EN)
+#define PIN_VREFHIZ	(AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ)
+#define PIN_VREF50		(AC_PINCTL_IN_EN | AC_PINCTL_VREF_50)
+#define PIN_VREFGRD	(AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD)
+#define PIN_VREF80		(AC_PINCTL_IN_EN | AC_PINCTL_VREF_80)
+#define PIN_VREF100	(AC_PINCTL_IN_EN | AC_PINCTL_VREF_100)
+#define PIN_OUT		(AC_PINCTL_OUT_EN)
+#define PIN_HP			(AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN)
+#define PIN_HP_AMP		(AC_PINCTL_HP_EN)
 
 /*
  * get widget capabilities
@@ -418,4 +421,13 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
 				 hda_nid_t nid);
 #endif /* CONFIG_SND_HDA_POWER_SAVE */
 
+/*
+ * AMP control callbacks
+ */
+/* retrieve parameters from private_value */
+#define get_amp_nid(kc)		((kc)->private_value & 0xffff)
+#define get_amp_channels(kc)	(((kc)->private_value >> 16) & 0x3)
+#define get_amp_direction(kc)	(((kc)->private_value >> 18) & 0x1)
+#define get_amp_index(kc)	(((kc)->private_value >> 19) & 0xf)
+
 #endif /* __SOUND_HDA_LOCAL_H */
diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h
index 2fdf2358dbc2..dfbcfa88da44 100644
--- a/sound/pci/hda/hda_patch.h
+++ b/sound/pci/hda/hda_patch.h
@@ -18,3 +18,5 @@ extern struct hda_codec_preset snd_hda_preset_atihdmi[];
 extern struct hda_codec_preset snd_hda_preset_conexant[];
 /* VIA codecs */
 extern struct hda_codec_preset snd_hda_preset_via[];
+/* NVIDIA HDMI codecs */
+extern struct hda_codec_preset snd_hda_preset_nvhdmi[];
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 1e5aff5c48d1..743d77922bce 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -216,7 +216,7 @@ static void print_pin_caps(struct snd_info_buffer *buffer,
 	unsigned int caps, val;
 
 	caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
-	snd_iprintf(buffer, "  Pincap 0x08%x:", caps);
+	snd_iprintf(buffer, "  Pincap 0x%08x:", caps);
 	if (caps & AC_PINCAP_IN)
 		snd_iprintf(buffer, " IN");
 	if (caps & AC_PINCAP_OUT)
@@ -229,8 +229,13 @@ static void print_pin_caps(struct snd_info_buffer *buffer,
 		snd_iprintf(buffer, " Detect");
 	if (caps & AC_PINCAP_BALANCE)
 		snd_iprintf(buffer, " Balanced");
-	if (caps & AC_PINCAP_LR_SWAP)
-		snd_iprintf(buffer, " R/L");
+	if (caps & AC_PINCAP_HDMI) {
+		/* Realtek uses this bit as a different meaning */
+		if ((codec->vendor_id >> 16) == 0x10ec)
+			snd_iprintf(buffer, " R/L");
+		else
+			snd_iprintf(buffer, " HDMI");
+	}
 	if (caps & AC_PINCAP_TRIG_REQ)
 		snd_iprintf(buffer, " Trigger");
 	if (caps & AC_PINCAP_IMP_SENSE)
@@ -552,9 +557,15 @@ static void print_codec_info(struct snd_info_entry *entry,
 
 		snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid,
 			    get_wid_type_name(wid_type), wid_caps);
-		if (wid_caps & AC_WCAP_STEREO)
-			snd_iprintf(buffer, " Stereo");
-		else
+		if (wid_caps & AC_WCAP_STEREO) {
+			unsigned int chans;
+			chans = (wid_caps & AC_WCAP_CHAN_CNT_EXT) >> 13;
+			chans = ((chans << 1) | 1) + 1;
+			if (chans == 2)
+				snd_iprintf(buffer, " Stereo");
+			else
+				snd_iprintf(buffer, " %d-Channels", chans);
+		} else
 			snd_iprintf(buffer, " Mono");
 		if (wid_caps & AC_WCAP_DIGITAL)
 			snd_iprintf(buffer, " Digital");
@@ -566,6 +577,8 @@ static void print_codec_info(struct snd_info_entry *entry,
 			snd_iprintf(buffer, " Stripe");
 		if (wid_caps & AC_WCAP_LR_SWAP)
 			snd_iprintf(buffer, " R/L");
+		if (wid_caps & AC_WCAP_CP_CAPS)
+			snd_iprintf(buffer, " CP");
 		snd_iprintf(buffer, "\n");
 
 		/* volume knob is a special widget that always have connection
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e8003d99f0bf..2b00c4afdf97 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1826,9 +1826,14 @@ static hda_nid_t ad1988_capsrc_nids[3] = {
 	0x0c, 0x0d, 0x0e
 };
 
-#define AD1988_SPDIF_OUT	0x02
+#define AD1988_SPDIF_OUT		0x02
+#define AD1988_SPDIF_OUT_HDMI	0x0b
 #define AD1988_SPDIF_IN		0x07
 
+static hda_nid_t ad1989b_slave_dig_outs[2] = {
+	AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI
+};
+
 static struct hda_input_mux ad1988_6stack_capture_source = {
 	.num_items = 5,
 	.items = {
@@ -2143,6 +2148,7 @@ static struct snd_kcontrol_new ad1988_spdif_in_mixers[] = {
 
 static struct snd_kcontrol_new ad1989_spdif_out_mixers[] = {
 	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("HDMI Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
 	{ } /* end */
 };
 
@@ -2207,6 +2213,8 @@ static struct hda_verb ad1988_6stack_init_verbs[] = {
 	{0x34, AC_VERB_SET_CONNECT_SEL, 0x0},
 	/* Analog CD Input */
 	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	/* Analog Mix output amp */
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
 
 	{ }
 };
@@ -2247,8 +2255,12 @@ static struct hda_verb ad1988_spdif_init_verbs[] = {
 
 /* AD1989 has no ADC -> SPDIF route */
 static struct hda_verb ad1989_spdif_init_verbs[] = {
-	/* SPDIF out pin */
+	/* SPDIF-1 out pin */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
+	/* SPDIF-2/HDMI out pin */
+	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
 	{ }
 };
 
@@ -2336,6 +2348,8 @@ static struct hda_verb ad1988_3stack_init_verbs[] = {
 	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Analog Mix output amp */
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
 	{ }
 };
 
@@ -2409,6 +2423,8 @@ static struct hda_verb ad1988_laptop_init_verbs[] = {
 	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Analog Mix output amp */
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
 	{ }
 };
 
@@ -2868,6 +2884,7 @@ static struct snd_pci_quirk ad1988_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG),
 	SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG),
 	SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG),
+	SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG),
 	{}
 };
 
@@ -2975,6 +2992,7 @@ static int patch_ad1988(struct hda_codec *codec)
 				ad1989_spdif_out_mixers;
 			spec->init_verbs[spec->num_init_verbs++] =
 				ad1989_spdif_init_verbs;
+			codec->slave_dig_outs = ad1989b_slave_dig_outs;
 		} else {
 			spec->mixers[spec->num_mixers++] =
 				ad1988_spdif_out_mixers;
@@ -3911,7 +3929,7 @@ static int patch_ad1884a(struct hda_codec *codec)
 
 
 /*
- * AD1882
+ * AD1882 / AD1882A
  *
  * port-A - front hp-out
  * port-B - front mic-in
@@ -3948,6 +3966,18 @@ static struct hda_input_mux ad1882_capture_source = {
 	},
 };
 
+/* list: 0x11, 0x39, 0x3a, 0x3c, 0x18, 0x1f, 0x12, 0x20 */
+static struct hda_input_mux ad1882a_capture_source = {
+	.num_items = 5,
+	.items = {
+		{ "Front Mic", 0x1 },
+		{ "Mic", 0x4},
+		{ "Line", 0x2 },
+		{ "Digital Mic", 0x06 },
+		{ "Mix", 0x7 },
+	},
+};
+
 static struct snd_kcontrol_new ad1882_base_mixers[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
@@ -3957,16 +3987,7 @@ static struct snd_kcontrol_new ad1882_base_mixers[] = {
 	HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT),
+
 	HDA_CODEC_VOLUME("Mic Boost", 0x3c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x39, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Line-In Boost", 0x3a, 0x0, HDA_OUTPUT),
@@ -3999,6 +4020,35 @@ static struct snd_kcontrol_new ad1882_base_mixers[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new ad1882_loopback_mixers[] = {
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
+	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT),
+	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT),
+	{ } /* end */
+};
+
+static struct snd_kcontrol_new ad1882a_loopback_mixers[] = {
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
+	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT),
+	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT),
+	HDA_CODEC_VOLUME("Digital Mic Boost", 0x1f, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
 static struct snd_kcontrol_new ad1882_3stack_mixers[] = {
 	HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x17, 1, 0x0, HDA_OUTPUT),
@@ -4168,9 +4218,16 @@ static int patch_ad1882(struct hda_codec *codec)
 	spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids);
 	spec->adc_nids = ad1882_adc_nids;
 	spec->capsrc_nids = ad1882_capsrc_nids;
-	spec->input_mux = &ad1882_capture_source;
-	spec->num_mixers = 1;
+	if (codec->vendor_id == 0x11d1882)
+		spec->input_mux = &ad1882_capture_source;
+	else
+		spec->input_mux = &ad1882a_capture_source;
+	spec->num_mixers = 2;
 	spec->mixers[0] = ad1882_base_mixers;
+	if (codec->vendor_id == 0x11d1882)
+		spec->mixers[1] = ad1882_loopback_mixers;
+	else
+		spec->mixers[1] = ad1882a_loopback_mixers;
 	spec->num_init_verbs = 1;
 	spec->init_verbs[0] = ad1882_init_verbs;
 	spec->spdif_route = 0;
@@ -4187,8 +4244,8 @@ static int patch_ad1882(struct hda_codec *codec)
 	switch (board_config) {
 	default:
 	case AD1882_3STACK:
-		spec->num_mixers = 2;
-		spec->mixers[1] = ad1882_3stack_mixers;
+		spec->num_mixers = 3;
+		spec->mixers[2] = ad1882_3stack_mixers;
 		spec->channel_mode = ad1882_modes;
 		spec->num_channel_mode = ARRAY_SIZE(ad1882_modes);
 		spec->need_dac_fix = 1;
@@ -4196,8 +4253,8 @@ static int patch_ad1882(struct hda_codec *codec)
 		spec->multiout.num_dacs = 1;
 		break;
 	case AD1882_6STACK:
-		spec->num_mixers = 2;
-		spec->mixers[1] = ad1882_6stack_mixers;
+		spec->num_mixers = 3;
+		spec->mixers[2] = ad1882_6stack_mixers;
 		break;
 	}
 	return 0;
@@ -4220,6 +4277,7 @@ struct hda_codec_preset snd_hda_preset_analog[] = {
 	{ .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },
 	{ .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 },
 	{ .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 },
+	{ .id = 0x11d4882a, .name = "AD1882A", .patch = patch_ad1882 },
 	{ .id = 0x11d4989a, .name = "AD1989A", .patch = patch_ad1988 },
 	{ .id = 0x11d4989b, .name = "AD1989B", .patch = patch_ad1988 },
 	{} /* terminator */
diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c
index 12272508b112..ba61575983fd 100644
--- a/sound/pci/hda/patch_atihdmi.c
+++ b/sound/pci/hda/patch_atihdmi.c
@@ -35,6 +35,9 @@ struct atihdmi_spec {
 	struct hda_pcm pcm_rec;
 };
 
+#define CVT_NID		0x02	/* audio converter */
+#define PIN_NID		0x03	/* HDMI output pin */
+
 static struct hda_verb atihdmi_basic_init[] = {
 	/* enable digital output on pin widget */
 	{ 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
@@ -60,8 +63,9 @@ static int atihdmi_init(struct hda_codec *codec)
 {
 	snd_hda_sequence_write(codec, atihdmi_basic_init);
 	/* SI codec requires to unmute the pin */
-	if (get_wcaps(codec, 0x03) & AC_WCAP_OUT_AMP)
-		snd_hda_codec_write(codec, 0x03, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+	if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP)
+		snd_hda_codec_write(codec, PIN_NID, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE,
 				    AMP_OUT_UNMUTE);
 	return 0;
 }
@@ -92,15 +96,29 @@ static int atihdmi_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 					    struct snd_pcm_substream *substream)
 {
 	struct atihdmi_spec *spec = codec->spec;
-	return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
-					     format, substream);
+	int chans = substream->runtime->channels;
+	int i, err;
+
+	err = snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
+					    format, substream);
+	if (err < 0)
+		return err;
+	snd_hda_codec_write(codec, CVT_NID, 0, AC_VERB_SET_CVT_CHAN_COUNT,
+			    chans - 1);
+	/* FIXME: XXX */
+	for (i = 0; i < chans; i++) {
+		snd_hda_codec_write(codec, CVT_NID, 0,
+				    AC_VERB_SET_HDMI_CHAN_SLOT,
+				    (i << 4) | i);
+	}
+	return 0;
 }
 
 static struct hda_pcm_stream atihdmi_pcm_digital_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
-	.nid = 0x2, /* NID to query formats and rates and setup streams */
+	.nid = CVT_NID, /* NID to query formats and rates and setup streams */
 	.ops = {
 		.open = atihdmi_dig_playback_pcm_open,
 		.close = atihdmi_dig_playback_pcm_close,
@@ -112,6 +130,7 @@ static int atihdmi_build_pcms(struct hda_codec *codec)
 {
 	struct atihdmi_spec *spec = codec->spec;
 	struct hda_pcm *info = &spec->pcm_rec;
+	unsigned int chans;
 
 	codec->num_pcms = 1;
 	codec->pcm_info = info;
@@ -120,6 +139,13 @@ static int atihdmi_build_pcms(struct hda_codec *codec)
 	info->pcm_type = HDA_PCM_TYPE_HDMI;
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = atihdmi_pcm_digital_playback;
 
+	/* FIXME: we must check ELD and change the PCM parameters dynamically
+	 */
+	chans = get_wcaps(codec, CVT_NID);
+	chans = (chans & AC_WCAP_CHAN_CNT_EXT) >> 13;
+	chans = ((chans << 1) | 1) + 1;
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans;
+
 	return 0;
 }
 
@@ -147,9 +173,11 @@ static int patch_atihdmi(struct hda_codec *codec)
 
 	spec->multiout.num_dacs = 0;	  /* no analog */
 	spec->multiout.max_channels = 2;
-	spec->multiout.dig_out_nid = 0x2; /* NID for copying analog to digital,
-					   * seems to be unused in pure-digital
-					   * case. */
+	/* NID for copying analog to digital,
+	 * seems to be unused in pure-digital
+	 * case.
+	 */
+	spec->multiout.dig_out_nid = CVT_NID;
 
 	codec->patch_ops = atihdmi_patch_ops;
 
@@ -164,6 +192,7 @@ struct hda_codec_preset snd_hda_preset_atihdmi[] = {
 	{ .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi },
 	{ .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi },
 	{ .id = 0x1002aa01, .name = "ATI R6xx HDMI", .patch = patch_atihdmi },
+	{ .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_atihdmi },
 	{ .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_atihdmi },
 	{ .id = 0x17e80047, .name = "Chrontel HDMI",  .patch = patch_atihdmi },
 	{} /* terminator */
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
new file mode 100644
index 000000000000..1a65775d28e1
--- /dev/null
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -0,0 +1,164 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * HD audio interface patch for NVIDIA HDMI codecs
+ *
+ * Copyright (c) 2008 NVIDIA Corp.  All rights reserved.
+ * Copyright (c) 2008 Wei Ni <wni@nvidia.com>
+ *
+ *
+ *  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 <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+
+struct nvhdmi_spec {
+	struct hda_multi_out multiout;
+
+	struct hda_pcm pcm_rec;
+};
+
+static struct hda_verb nvhdmi_basic_init[] = {
+	/* enable digital output on pin widget */
+	{ 0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{} /* terminator */
+};
+
+/*
+ * Controls
+ */
+static int nvhdmi_build_controls(struct hda_codec *codec)
+{
+	struct nvhdmi_spec *spec = codec->spec;
+	int err;
+
+	err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
+static int nvhdmi_init(struct hda_codec *codec)
+{
+	snd_hda_sequence_write(codec, nvhdmi_basic_init);
+	return 0;
+}
+
+/*
+ * Digital out
+ */
+static int nvhdmi_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+				     struct hda_codec *codec,
+				     struct snd_pcm_substream *substream)
+{
+	struct nvhdmi_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int nvhdmi_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+				      struct hda_codec *codec,
+				      struct snd_pcm_substream *substream)
+{
+	struct nvhdmi_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+static int nvhdmi_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 nvhdmi_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
+					     format, substream);
+}
+
+static struct hda_pcm_stream nvhdmi_pcm_digital_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.nid = 0x4, /* NID to query formats and rates and setup streams */
+	.rates = SNDRV_PCM_RATE_48000,
+	.maxbps = 16,
+	.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	.ops = {
+		.open = nvhdmi_dig_playback_pcm_open,
+		.close = nvhdmi_dig_playback_pcm_close,
+		.prepare = nvhdmi_dig_playback_pcm_prepare
+	},
+};
+
+static int nvhdmi_build_pcms(struct hda_codec *codec)
+{
+	struct nvhdmi_spec *spec = codec->spec;
+	struct hda_pcm *info = &spec->pcm_rec;
+
+	codec->num_pcms = 1;
+	codec->pcm_info = info;
+
+	info->name = "NVIDIA HDMI";
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = nvhdmi_pcm_digital_playback;
+
+	return 0;
+}
+
+static void nvhdmi_free(struct hda_codec *codec)
+{
+	kfree(codec->spec);
+}
+
+static struct hda_codec_ops nvhdmi_patch_ops = {
+	.build_controls = nvhdmi_build_controls,
+	.build_pcms = nvhdmi_build_pcms,
+	.init = nvhdmi_init,
+	.free = nvhdmi_free,
+};
+
+static int patch_nvhdmi(struct hda_codec *codec)
+{
+	struct nvhdmi_spec *spec;
+
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL)
+		return -ENOMEM;
+
+	codec->spec = spec;
+
+	spec->multiout.num_dacs = 0;	  /* no analog */
+	spec->multiout.max_channels = 2;
+	spec->multiout.dig_out_nid = 0x4; /* NID for copying analog to digital,
+					   * seems to be unused in pure-digital
+					   * case. */
+
+	codec->patch_ops = nvhdmi_patch_ops;
+
+	return 0;
+}
+
+/*
+ * patch entries
+ */
+struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
+	{ .id = 0x10de0002, .name = "NVIDIA MCP78 HDMI", .patch = patch_nvhdmi },
+	{ .id = 0x10de0007, .name = "NVIDIA MCP7A HDMI", .patch = patch_nvhdmi },
+	{} /* terminator */
+};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 66025161bd69..0b6e682c46d0 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -72,6 +72,7 @@ enum {
 enum {
 	ALC260_BASIC,
 	ALC260_HP,
+	ALC260_HP_DC7600,
 	ALC260_HP_3013,
 	ALC260_FUJITSU_S702X,
 	ALC260_ACER,
@@ -100,6 +101,9 @@ enum {
 	ALC262_BENQ_T31,
 	ALC262_ULTRA,
 	ALC262_LENOVO_3000,
+	ALC262_NEC,
+	ALC262_TOSHIBA_S06,
+	ALC262_TOSHIBA_RX1,
 	ALC262_AUTO,
 	ALC262_MODEL_LAST /* last tag */
 };
@@ -110,6 +114,7 @@ enum {
 	ALC268_3ST,
 	ALC268_TOSHIBA,
 	ALC268_ACER,
+	ALC268_ACER_ASPIRE_ONE,
 	ALC268_DELL,
 	ALC268_ZEPTO,
 #ifdef CONFIG_SND_DEBUG
@@ -122,6 +127,7 @@ enum {
 /* ALC269 models */
 enum {
 	ALC269_BASIC,
+	ALC269_QUANTA_FL1,
 	ALC269_ASUS_EEEPC_P703,
 	ALC269_ASUS_EEEPC_P901,
 	ALC269_AUTO,
@@ -169,6 +175,13 @@ enum {
 	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,
 	ALC662_AUTO,
 	ALC662_MODEL_LAST,
 };
@@ -200,18 +213,21 @@ enum {
 	ALC883_ACER,
 	ALC883_ACER_ASPIRE,
 	ALC883_MEDION,
-	ALC883_MEDION_MD2,	
+	ALC883_MEDION_MD2,
 	ALC883_LAPTOP_EAPD,
 	ALC883_LENOVO_101E_2ch,
 	ALC883_LENOVO_NB0763,
 	ALC888_LENOVO_MS7195_DIG,
-	ALC883_HAIER_W66,		
+	ALC888_LENOVO_SKY,
+	ALC883_HAIER_W66,
 	ALC888_3ST_HP,
 	ALC888_6ST_DELL,
 	ALC883_MITAC,
 	ALC883_CLEVO_M720,
 	ALC883_FUJITSU_PI2515,
 	ALC883_3ST_6ch_INTEL,
+	ALC888_ASUS_M90V,
+	ALC888_ASUS_EEE1601,
 	ALC883_AUTO,
 	ALC883_MODEL_LAST,
 };
@@ -398,7 +414,7 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
 
 /*
  * Control the mode of pin widget settings via the mixer.  "pc" is used
- * instead of "%" to avoid consequences of accidently treating the % as 
+ * instead of "%" to avoid consequences of accidently treating the % as
  * being part of a format specifier.  Maximum allowed length of a value is
  * 63 characters plus NULL terminator.
  *
@@ -429,7 +445,7 @@ static unsigned char alc_pin_mode_values[] = {
 #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. 
+/* Info about the pin modes supported by the different pin direction modes.
  * For each direction the minimum and maximum values are given.
  */
 static signed char alc_pin_mode_dir_info[5][2] = {
@@ -502,7 +518,7 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
 					  AC_VERB_SET_PIN_WIDGET_CONTROL,
 					  alc_pin_mode_values[val]);
 
-		/* Also enable the retasking pin's input/output as required 
+		/* Also enable the retasking pin's input/output as required
 		 * for the requested pin mode.  Enum values of 2 or less are
 		 * input modes.
 		 *
@@ -707,7 +723,7 @@ static void setup_preset(struct alc_spec *spec,
 	     i++)
 		spec->init_verbs[spec->num_init_verbs++] =
 			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;
@@ -718,7 +734,7 @@ static void setup_preset(struct alc_spec *spec,
 	spec->multiout.dac_nids = preset->dac_nids;
 	spec->multiout.dig_out_nid = preset->dig_out_nid;
 	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;
@@ -855,7 +871,7 @@ static void alc_subsystem_id(struct hda_codec *codec,
 	if ((ass != codec->bus->pci->subsystem_device) && (ass & 1))
 		goto do_sku;
 
-	/*	
+	/*
 	 * 31~30	: port conetcivity
 	 * 29~21	: reserve
 	 * 20		: PCBEEP input
@@ -946,7 +962,7 @@ do_sku:
 			tmp = snd_hda_codec_read(codec, 0x20, 0,
 						 AC_VERB_GET_PROC_COEF, 0);
 			snd_hda_codec_write(codec, 0x20, 0,
-					    AC_VERB_SET_COEF_INDEX, 7);	
+					    AC_VERB_SET_COEF_INDEX, 7);
 			snd_hda_codec_write(codec, 0x20, 0,
 					    AC_VERB_SET_PROC_COEF,
 					    tmp | 0x2010);
@@ -961,7 +977,7 @@ do_sku:
 			tmp = snd_hda_codec_read(codec, 0x20, 0,
 						 AC_VERB_GET_PROC_COEF, 0);
 			snd_hda_codec_write(codec, 0x20, 0,
-					    AC_VERB_SET_COEF_INDEX, 7);	
+					    AC_VERB_SET_COEF_INDEX, 7);
 			snd_hda_codec_write(codec, 0x20, 0,
 					    AC_VERB_SET_PROC_COEF,
 					    tmp | 0x3000);
@@ -970,7 +986,7 @@ do_sku:
 	default:
 		break;
 	}
-	
+
 	/* is laptop or Desktop and enable the function "Mute internal speaker
 	 * when the external headphone out jack is plugged"
 	 */
@@ -1006,6 +1022,7 @@ do_sku:
 	snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0,
 			    AC_VERB_SET_UNSOLICITED_ENABLE,
 			    AC_USRSP_EN | ALC880_HP_EVENT);
+
 	spec->unsol_event = alc_sku_unsol_event;
 }
 
@@ -1296,7 +1313,7 @@ static struct snd_kcontrol_new alc880_six_stack_mixer[] = {
  *
  * 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.
  *
@@ -1824,7 +1841,7 @@ static struct hda_verb alc880_pin_6stack_init_verbs[] = {
 	{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},
-	
+
 	{ }
 };
 
@@ -1869,7 +1886,7 @@ static struct hda_verb alc880_uniwill_init_verbs[] = {
 
 /*
 * Uniwill P53
-* HP = 0x14, InternalSpeaker = 0x15, mic = 0x19, 
+* HP = 0x14, InternalSpeaker = 0x15, mic = 0x19,
  */
 static struct hda_verb alc880_uniwill_p53_init_verbs[] = {
 	{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
@@ -1968,7 +1985,7 @@ static void alc880_uniwill_p53_hp_automute(struct hda_codec *codec)
 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;
@@ -2050,7 +2067,7 @@ static struct hda_verb alc880_pin_asus_init_verbs[] = {
 	{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},
-	
+
 	{ }
 };
 
@@ -2632,12 +2649,14 @@ static int alc_build_pcms(struct hda_codec *codec)
 
 	info->name = spec->stream_name_analog;
 	if (spec->stream_analog_playback) {
-		snd_assert(spec->multiout.dac_nids, return -EINVAL);
+		if (snd_BUG_ON(!spec->multiout.dac_nids))
+			return -EINVAL;
 		info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback);
 		info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
 	}
 	if (spec->stream_analog_capture) {
-		snd_assert(spec->adc_nids, return -EINVAL);
+		if (snd_BUG_ON(!spec->adc_nids))
+			return -EINVAL;
 		info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
 		info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
 	}
@@ -2667,6 +2686,8 @@ static int alc_build_pcms(struct hda_codec *codec)
 			info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_digital_capture);
 			info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
 		}
+		/* FIXME: do we need this for all Realtek codec models? */
+		codec->spdif_status_reset = 1;
 	}
 
 	/* If the use of more than one ADC is requested for the current
@@ -3683,7 +3704,7 @@ static void alc880_auto_init_multi_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	int i;
-	
+
 	alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
 	for (i = 0; i < spec->autocfg.line_outs; i++) {
 		hda_nid_t nid = spec->autocfg.line_out_pins[i];
@@ -4124,6 +4145,33 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
 	{ } /* end */
 };
 
+static 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 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 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 struct hda_verb alc260_hp_3013_unsol_verbs[] = {
 	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
 	{},
@@ -4147,7 +4195,30 @@ static void alc260_hp_3013_unsol_event(struct hda_codec *codec,
 		alc260_hp_3013_automute(codec);
 }
 
-/* Fujitsu S702x series laptops.  ALC260 pin usage: Mic/Line jack = 0x12, 
+static void alc260_hp_3012_automute(struct hda_codec *codec)
+{
+	unsigned int present, bits;
+
+	present = snd_hda_codec_read(codec, 0x10, 0,
+			AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE;
+
+	bits = present ? 0 : PIN_OUT;
+	snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			    bits);
+	snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			    bits);
+	snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			    bits);
+}
+
+static void alc260_hp_3012_unsol_event(struct hda_codec *codec,
+				       unsigned int res)
+{
+	if ((res >> 26) == ALC880_HP_EVENT)
+		alc260_hp_3012_automute(codec);
+}
+
+/* Fujitsu S702x series laptops.  ALC260 pin usage: Mic/Line jack = 0x12,
  * HP jack = 0x14, CD audio =  0x16, internal speaker = 0x10.
  */
 static struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
@@ -4478,7 +4549,7 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = {
 	{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 
+	/* Ensure Line1 pin widget takes its input from the OUT1 sum bus
 	 * when acting as an output.
 	 */
 	{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
@@ -4503,14 +4574,14 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = {
 	 * stage.
 	 */
 	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Unmute input buffer of pin widget used for Line-in (no equiv 
+	/* 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 
+	/* Set ADC connection select to match default mixer setting - line
 	 * in (on mic1 pin)
 	 */
 	{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -4564,7 +4635,7 @@ static struct hda_verb alc260_acer_init_verbs[] = {
 	{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 
+	/* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
 	 * bus when acting as outputs.
 	 */
 	{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
@@ -4675,6 +4746,20 @@ static void alc260_replacer_672v_unsol_event(struct hda_codec *codec,
                 alc260_replacer_672v_automute(codec);
 }
 
+static 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 | ALC880_HP_EVENT},
+	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{}
+};
+
 /* Test configuration for debugging, modelled after the ALC880 test
  * configuration.
  */
@@ -4686,7 +4771,7 @@ static 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 
+ * the signal assignments are different.  This assumes that the first ADC
  * is NID 0x04.
  */
 static struct hda_input_mux alc260_test_capture_sources[2] = {
@@ -4769,7 +4854,7 @@ static struct snd_kcontrol_new alc260_test_mixer[] = {
 
 	/* 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. 
+	 * 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),
@@ -4805,7 +4890,7 @@ static struct hda_verb alc260_test_init_verbs[] = {
 	{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 
+	/* 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},
@@ -4897,7 +4982,7 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
 		sw_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
 	} else
 		return 0; /* N/A */
-	
+
 	snprintf(name, sizeof(name), "%s Playback Volume", pfx);
 	err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val);
 	if (err < 0)
@@ -5003,7 +5088,7 @@ static void alc260_auto_init_multi_out(struct hda_codec *codec)
 		int pin_type = get_pin_type(spec->autocfg.line_out_type);
 		alc260_auto_set_output_and_unmute(codec, nid, pin_type, 0);
 	}
-	
+
 	nid = spec->autocfg.speaker_pins[0];
 	if (nid)
 		alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0);
@@ -5045,7 +5130,7 @@ static struct hda_verb alc260_volume_init_verbs[] = {
 	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x05, 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
@@ -5074,7 +5159,7 @@ static struct hda_verb alc260_volume_init_verbs[] = {
 	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	
+
 	{ }
 };
 
@@ -5155,6 +5240,7 @@ static const char *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",
@@ -5172,7 +5258,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013),
 	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_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),
@@ -5218,6 +5304,22 @@ static struct alc_config_preset alc260_presets[] = {
 		.unsol_event = alc260_hp_unsol_event,
 		.init_hook = alc260_hp_automute,
 	},
+	[ALC260_HP_DC7600] = {
+		.mixers = { alc260_hp_dc7600_mixer,
+			    alc260_input_mixer,
+			    alc260_capture_alt_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_hp_adc_nids),
+		.adc_nids = alc260_hp_adc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc260_modes),
+		.channel_mode = alc260_modes,
+		.input_mux = &alc260_capture_source,
+		.unsol_event = alc260_hp_3012_unsol_event,
+		.init_hook = alc260_hp_3012_automute,
+	},
 	[ALC260_HP_3013] = {
 		.mixers = { alc260_hp_3013_mixer,
 			    alc260_input_mixer,
@@ -5933,7 +6035,7 @@ static struct hda_verb alc882_targa_verbs[] = {
 
 	{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 */
@@ -5949,7 +6051,7 @@ static struct hda_verb alc882_targa_verbs[] = {
 static void alc882_targa_automute(struct hda_codec *codec)
 {
  	unsigned int present;
- 
+
  	present = snd_hda_codec_read(codec, 0x14, 0,
 				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
 	snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
@@ -5975,7 +6077,7 @@ static struct hda_verb alc882_asus_a7j_verbs[] = {
 	{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 */
@@ -5993,7 +6095,7 @@ static struct hda_verb alc882_asus_a7m_verbs[] = {
 	{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 */
@@ -6319,7 +6421,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.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_init_verbs, alc882_eapd_verbs,
@@ -6332,14 +6434,14 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc880_threestack_modes,
 		.need_dac_fix = 1,
 		.input_mux = &alc882_capture_source,
-	},	
+	},
 };
 
 
 /*
  * Pin config fixes
  */
-enum { 
+enum {
 	PINFIX_ABIT_AW9D_MAX
 };
 
@@ -6554,16 +6656,19 @@ static int patch_alc882(struct hda_codec *codec)
 			board_config = ALC885_MACPRO;
 			break;
 		case 0x106b1000: /* iMac 24 */
+		case 0x106b2800: /* AppleTV */
 			board_config = ALC885_IMAC24;
 			break;
 		case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */
+		case 0x106b00a4: /* MacbookPro4,1 */
 		case 0x106b2c00: /* Macbook Pro rev3 */
 		case 0x106b3600: /* Macbook 3.1 */
 			board_config = ALC885_MBP3;
 			break;
 		default:
 			/* ALC889A is handled better as ALC888-compatible */
-			if (codec->revision_id == 0x100103) {
+			if (codec->revision_id == 0x100101 ||
+			    codec->revision_id == 0x100103) {
 				alc_free(codec);
 				return patch_alc883(codec);
 			}
@@ -6718,6 +6823,23 @@ static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = {
 	},
 };
 
+static struct hda_input_mux alc883_lenovo_sky_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x1 },
+		{ "Line", 0x4 },
+	},
+};
+
+static struct hda_input_mux alc883_asus_eee1601_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Line", 0x2 },
+	},
+};
+
 #define alc883_mux_enum_info alc_mux_enum_info
 #define alc883_mux_enum_get alc_mux_enum_get
 /* ALC883 has the ALC882-type input selection */
@@ -7032,13 +7154,11 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
 	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		/* .name = "Capture Source", */
 		.name = "Input Source",
-		.count = 2,
+		.count = 1,
 		.info = alc883_mux_enum_info,
 		.get = alc883_mux_enum_get,
 		.put = alc883_mux_enum_put,
@@ -7256,7 +7376,7 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = {
 		.put = alc883_mux_enum_put,
 	},
 	{ } /* end */
-};	
+};
 
 static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -7283,6 +7403,87 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
 	{ } /* end */
 };
 
+static 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_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x1a, 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", 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", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		/* .name = "Capture Source", */
+		.name = "Input Source",
+		.count = 2,
+		.info = alc883_mux_enum_info,
+		.get = alc883_mux_enum_get,
+		.put = alc883_mux_enum_put,
+	},
+	{ } /* end */
+};
+
+static 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 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 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", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	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 = alc883_mux_enum_info,
+		.get = alc883_mux_enum_get,
+		.put = alc883_mux_enum_put,
+	},
+	{ } /* end */
+};
+
 static struct snd_kcontrol_new alc883_chmode_mixer[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -7296,7 +7497,7 @@ static struct snd_kcontrol_new alc883_chmode_mixer[] = {
 
 static struct hda_verb alc883_init_verbs[] = {
 	/* ADC1: mute amp left and right */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
 	/* ADC2: mute amp left and right */
 	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
@@ -7361,14 +7562,14 @@ static struct hda_verb alc883_init_verbs[] = {
 	/* 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)},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{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_UNMUTE(1)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 	{ }
 };
 
@@ -7468,7 +7669,7 @@ static struct hda_verb alc883_tagra_verbs[] = {
 
 	{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 */
@@ -7518,6 +7719,18 @@ static struct hda_verb alc883_haier_w66_verbs[] = {
 	{ } /* end */
 };
 
+static 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, ALC880_HP_EVENT | AC_USRSP_EN},
+	{ } /* end */
+};
+
 static 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) */
@@ -7555,7 +7768,7 @@ static struct hda_channel_mode alc888_3st_hp_modes[2] = {
 static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec)
 {
  	unsigned int present;
- 
+
  	present = snd_hda_codec_read(codec, 0x1b, 0,
 				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
 	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
@@ -7568,7 +7781,7 @@ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec)
 static void alc888_lenovo_ms7195_rca_automute(struct hda_codec *codec)
 {
  	unsigned int present;
- 
+
  	present = snd_hda_codec_read(codec, 0x14, 0,
 				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
 	snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
@@ -7598,7 +7811,7 @@ static struct hda_verb alc883_medion_md2_verbs[] = {
 static void alc883_medion_md2_automute(struct hda_codec *codec)
 {
  	unsigned int present;
- 
+
  	present = snd_hda_codec_read(codec, 0x14, 0,
 				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
 	snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
@@ -7753,7 +7966,7 @@ static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec,
 static void alc883_acer_aspire_automute(struct hda_codec *codec)
 {
  	unsigned int present;
- 
+
  	present = snd_hda_codec_read(codec, 0x14, 0,
 				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
 	snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
@@ -7790,7 +8003,7 @@ static struct hda_verb alc883_acer_eapd_verbs[] = {
 static void alc888_6st_dell_front_automute(struct hda_codec *codec)
 {
  	unsigned int present;
- 
+
  	present = snd_hda_codec_read(codec, 0x1b, 0,
 				AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
 	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
@@ -7814,6 +8027,50 @@ static void alc888_6st_dell_unsol_event(struct hda_codec *codec,
 	}
 }
 
+static void alc888_lenovo_sky_front_automute(struct hda_codec *codec)
+{
+	unsigned int mute;
+	unsigned int present;
+
+	snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
+	present = snd_hda_codec_read(codec, 0x1b, 0,
+				     AC_VERB_GET_PIN_SENSE, 0);
+	present = (present & 0x80000000) != 0;
+	if (present) {
+		/* mute internal speaker */
+		snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, HDA_AMP_MUTE);
+		snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, HDA_AMP_MUTE);
+		snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, HDA_AMP_MUTE);
+		snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, HDA_AMP_MUTE);
+		snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, HDA_AMP_MUTE);
+	} else {
+		/* unmute internal speaker if necessary */
+		mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0);
+		snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, mute);
+		snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, mute);
+		snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, mute);
+		snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, mute);
+		snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, mute);
+	}
+}
+
+static void alc883_lenovo_sky_unsol_event(struct hda_codec *codec,
+					     unsigned int res)
+{
+	if ((res >> 26) == ALC880_HP_EVENT)
+		alc888_lenovo_sky_front_automute(codec);
+}
+
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
@@ -7898,6 +8155,105 @@ static struct snd_kcontrol_new alc883_capture_mixer[] = {
 	{ } /* end */
 };
 
+static 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, ALC880_HP_EVENT | AC_USRSP_EN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
+	{ } /* end */
+};
+
+static void alc883_nb_mic_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+
+	present = snd_hda_codec_read(codec, 0x18, 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+			    0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
+	snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+			    0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
+}
+
+static void alc883_M90V_speaker_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned char bits;
+
+	present = snd_hda_codec_read(codec, 0x1b, 0,
+				     AC_VERB_GET_PIN_SENSE, 0)
+		& AC_PINSENSE_PRESENCE;
+	bits = present ? 0 : PIN_OUT;
+	snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			    bits);
+	snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			    bits);
+	snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			    bits);
+}
+
+static void alc883_mode2_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
+		alc883_M90V_speaker_automute(codec);
+		break;
+	case ALC880_MIC_EVENT:
+		alc883_nb_mic_automute(codec);
+		break;
+	}
+}
+
+static void alc883_mode2_inithook(struct hda_codec *codec)
+{
+	alc883_M90V_speaker_automute(codec);
+	alc883_nb_mic_automute(codec);
+}
+
+static 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, ALC880_HP_EVENT | AC_USRSP_EN},
+	{ } /* end */
+};
+
+static void alc883_eee1601_speaker_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned char bits;
+
+	present = snd_hda_codec_read(codec, 0x14, 0,
+				     AC_VERB_GET_PIN_SENSE, 0)
+		& AC_PINSENSE_PRESENCE;
+	bits = present ? 0 : PIN_OUT;
+	snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			    bits);
+}
+
+static void alc883_eee1601_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
+		alc883_eee1601_speaker_automute(codec);
+		break;
+	}
+}
+
+static void alc883_eee1601_inithook(struct hda_codec *codec)
+{
+	alc883_eee1601_speaker_automute(codec);
+}
+
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 #define alc883_loopbacks	alc880_loopbacks
 #endif
@@ -7927,6 +8283,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
 	[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",
@@ -7942,7 +8299,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", 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, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */
 	SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
 	SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
@@ -7950,10 +8307,13 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x1043, 0x8317, "Asus M90V", ALC888_ASUS_M90V),
+	SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601),
 	SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
 	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, "MSI", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
@@ -7989,6 +8349,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	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, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
 	SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
@@ -8128,7 +8489,7 @@ static struct alc_config_preset alc883_presets[] = {
 		.input_mux = &alc883_capture_source,
 		.unsol_event = alc883_medion_md2_unsol_event,
 		.init_hook = alc883_medion_md2_automute,
-	},	
+	},
 	[ALC883_LAPTOP_EAPD] = {
 		.mixers = { alc883_base_mixer },
 		.init_verbs = { alc883_init_verbs, alc882_eapd_verbs },
@@ -8245,6 +8606,49 @@ static struct alc_config_preset alc883_presets[] = {
 		.unsol_event = alc883_2ch_fujitsu_pi2515_unsol_event,
 		.init_hook = alc883_2ch_fujitsu_pi2515_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_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+		.adc_nids = alc883_adc_nids,
+		.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 = alc883_lenovo_sky_unsol_event,
+		.init_hook = alc888_lenovo_sky_front_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 = alc883_mode2_unsol_event,
+		.init_hook = alc883_mode2_inithook,
+	},
+	[ALC888_ASUS_EEE1601] = {
+		.mixers = { alc883_asus_eee1601_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 = alc883_eee1601_unsol_event,
+		.init_hook = alc883_eee1601_inithook,
+	},
 };
 
 
@@ -8452,6 +8856,13 @@ static int patch_alc883(struct hda_codec *codec)
 #define alc262_modes		alc260_modes
 #define alc262_capture_source	alc882_capture_source
 
+static hda_nid_t alc262_dmic_adc_nids[1] = {
+	/* ADC0 */
+	0x09
+};
+
+static hda_nid_t alc262_dmic_capsrc_nids[1] = { 0x22 };
+
 static 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),
@@ -8833,10 +9244,10 @@ static struct hda_verb alc262_init_verbs[] = {
 	{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 */
@@ -8858,6 +9269,12 @@ static struct hda_verb alc262_init_verbs[] = {
 	{ }
 };
 
+static struct hda_verb alc262_eapd_verbs[] = {
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{ }
+};
+
 static struct hda_verb alc262_hippo_unsol_verbs[] = {
 	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
@@ -8884,6 +9301,91 @@ static struct hda_verb alc262_sony_unsol_verbs[] = {
 	{}
 };
 
+static struct hda_input_mux alc262_dmic_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Int DMic", 0x9 },
+		{ "Mic", 0x0 },
+	},
+};
+
+static 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),
+	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		/* The multiple "Capture Source" controls confuse alsamixer
+		 * So call somewhat different..
+		 */
+		/* .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 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 | ALC880_MIC_EVENT},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{}
+};
+
+static void alc262_dmic_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+
+	present = snd_hda_codec_read(codec, 0x18, 0,
+					AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	snd_hda_codec_write(codec, 0x22, 0,
+				AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x09);
+}
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc262_toshiba_s06_speaker_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned char bits;
+
+	present = snd_hda_codec_read(codec, 0x15, 0,
+					AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	bits = present ? 0 : PIN_OUT;
+	snd_hda_codec_write(codec, 0x14, 0,
+					AC_VERB_SET_PIN_WIDGET_CONTROL, bits);
+}
+
+
+
+/* unsolicited event for HP jack sensing */
+static void alc262_toshiba_s06_unsol_event(struct hda_codec *codec,
+				       unsigned int res)
+{
+	if ((res >> 26) == ALC880_HP_EVENT)
+		alc262_toshiba_s06_speaker_automute(codec);
+	if ((res >> 26) == ALC880_MIC_EVENT)
+		alc262_dmic_automute(codec);
+
+}
+
+static void alc262_toshiba_s06_init_hook(struct hda_codec *codec)
+{
+	alc262_toshiba_s06_speaker_automute(codec);
+	alc262_dmic_automute(codec);
+}
+
 /* mute/unmute internal speaker according to the hp jack and mute state */
 static void alc262_hippo_automute(struct hda_codec *codec)
 {
@@ -8948,6 +9450,41 @@ static void alc262_hippo1_unsol_event(struct hda_codec *codec,
 }
 
 /*
+ * nec model
+ *  0x15 = headphone
+ *  0x16 = internal speaker
+ *  0x18 = external mic
+ */
+
+static 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", 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 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 | ALC880_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
@@ -9179,6 +9716,25 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = {
+	HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Master Playback Switch",
+		.info = snd_hda_mixer_amp_switch_info,
+		.get = snd_hda_mixer_amp_switch_get,
+		.put = alc262_sony_master_sw_put,
+		.private_value = HDA_COMPOSE_AMP_VAL(0x15, 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", 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", 0x19, 0, HDA_INPUT),
+	{ } /* end */
+};
+
 /* additional init verbs for Benq laptops */
 static struct hda_verb alc262_EAPD_verbs[] = {
 	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
@@ -9427,7 +9983,7 @@ static struct hda_verb alc262_volume_init_verbs[] = {
 	{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)},
@@ -9482,7 +10038,7 @@ static struct hda_verb alc262_HP_BPC_init_verbs[] = {
 	{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)
 	 */
@@ -9643,6 +10199,24 @@ static struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = {
 	{ }
 };
 
+static 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 | ALC880_HP_EVENT},
+	{}
+};
+
+
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 #define alc262_loopbacks	alc880_loopbacks
 #endif
@@ -9729,13 +10303,17 @@ static const char *alc262_models[ALC262_MODEL_LAST] = {
 	[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_AUTO]		= "auto",
 };
 
 static 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(0x103c, 0x12fe, "HP xw9400", ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x12ff, "HP xw4550", ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC),
@@ -9764,7 +10342,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
-		      ALC262_SONY_ASSAMD),
+		      ALC262_TOSHIBA_RX1),
+	SND_PCI_QUIRK(0x1179, 0x0268, "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(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA),
@@ -9918,7 +10497,7 @@ static struct alc_config_preset alc262_presets[] = {
 		.input_mux = &alc262_capture_source,
 		.unsol_event = alc262_hippo_unsol_event,
 		.init_hook = alc262_hippo_automute,
-	},	
+	},
 	[ALC262_ULTRA] = {
 		.mixers = { alc262_ultra_mixer, alc262_ultra_capture_mixer },
 		.init_verbs = { alc262_ultra_verbs },
@@ -9946,6 +10525,43 @@ static struct alc_config_preset alc262_presets[] = {
 		.input_mux = &alc262_fujitsu_capture_source,
 		.unsol_event = alc262_lenovo_3000_unsol_event,
 	},
+	[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 */
+		.dig_out_nid = ALC262_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_dmic_capture_source,
+		.unsol_event = alc262_toshiba_s06_unsol_event,
+		.init_hook = alc262_toshiba_s06_init_hook,
+	},
+	[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 = alc262_hippo_unsol_event,
+		.init_hook = alc262_hippo_automute,
+	},
 };
 
 static int patch_alc262(struct hda_codec *codec)
@@ -10004,7 +10620,7 @@ static int patch_alc262(struct hda_codec *codec)
 	spec->stream_name_analog = "ALC262 Analog";
 	spec->stream_analog_playback = &alc262_pcm_analog_playback;
 	spec->stream_analog_capture = &alc262_pcm_analog_capture;
-		
+
 	spec->stream_name_digital = "ALC262 Digital";
 	spec->stream_digital_playback = &alc262_pcm_digital_playback;
 	spec->stream_digital_capture = &alc262_pcm_digital_capture;
@@ -10040,7 +10656,7 @@ static int patch_alc262(struct hda_codec *codec)
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc262_loopbacks;
 #endif
-		
+
 	return 0;
 }
 
@@ -10049,7 +10665,7 @@ static int patch_alc262(struct hda_codec *codec)
  */
 #define ALC268_DIGOUT_NID	ALC880_DIGOUT_NID
 #define alc268_modes		alc260_modes
-	
+
 static hda_nid_t alc268_dac_nids[2] = {
 	/* front, hp */
 	0x02, 0x03
@@ -10109,6 +10725,14 @@ static struct hda_verb alc268_toshiba_verbs[] = {
 	{ } /* end */
 };
 
+static struct hda_input_mux alc268_acer_lc_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "i-Mic", 0x6 },
+		{ "E-Mic", 0x0 },
+	},
+};
+
 /* Acer specific */
 /* bind volumes of both NID 0x02 and 0x03 */
 static struct hda_bind_ctls alc268_acer_bind_master_vol = {
@@ -10161,6 +10785,21 @@ static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
+static 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",
+		.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 Boost Capture Volume", 0x18, 0, HDA_INPUT),
+	{ }
+};
+
 static struct snd_kcontrol_new alc268_acer_mixer[] = {
 	/* output mixer control */
 	HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
@@ -10178,6 +10817,16 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = {
 	{ }
 };
 
+static struct hda_verb alc268_acer_aspire_one_verbs[] = {
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+	{0x23, AC_VERB_SET_CONNECT_SEL, 0x06},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, 0xa017},
+	{ }
+};
+
 static 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},
@@ -10185,7 +10834,6 @@ static struct hda_verb alc268_acer_verbs[] = {
 	{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, ALC880_HP_EVENT | AC_USRSP_EN},
 	{ }
 };
@@ -10212,6 +10860,47 @@ static void alc268_acer_init_hook(struct hda_codec *codec)
 	alc268_acer_automute(codec, 1);
 }
 
+/* toggle speaker-output according to the hp-jack state */
+static void alc268_aspire_one_speaker_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned char bits;
+
+	present = snd_hda_codec_read(codec, 0x15, 0,
+				AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	bits = present ? AMP_IN_MUTE(0) : 0;
+	snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0,
+				AMP_IN_MUTE(0), bits);
+	snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 1,
+				AMP_IN_MUTE(0), bits);
+}
+
+
+static void alc268_acer_mic_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+
+	present = snd_hda_codec_read(codec, 0x18, 0,
+				AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_CONNECT_SEL,
+			    present ? 0x0 : 0x6);
+}
+
+static void alc268_acer_lc_unsol_event(struct hda_codec *codec,
+				    unsigned int res)
+{
+	if ((res >> 26) == ALC880_HP_EVENT)
+		alc268_aspire_one_speaker_automute(codec);
+	if ((res >> 26) == ALC880_MIC_EVENT)
+		alc268_acer_mic_automute(codec);
+}
+
+static void alc268_acer_lc_init_hook(struct hda_codec *codec)
+{
+	alc268_aspire_one_speaker_automute(codec);
+	alc268_acer_mic_automute(codec);
+}
+
 static struct snd_kcontrol_new alc268_dell_mixer[] = {
 	/* output mixer control */
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
@@ -10360,7 +11049,7 @@ static struct hda_verb alc268_base_init_verbs[] = {
 	{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},
@@ -10559,7 +11248,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
 
 	nid = cfg->line_out_pins[0];
 	if (nid)
-		alc268_new_analog_output(spec, nid, "Front", 0);	
+		alc268_new_analog_output(spec, nid, "Front", 0);
 
 	nid = cfg->speaker_pins[0];
 	if (nid == 0x1d) {
@@ -10581,7 +11270,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
 		if (err < 0)
 			return err;
 	}
-	return 0;	
+	return 0;
 }
 
 /* create playback/capture controls for input pins */
@@ -10602,7 +11291,7 @@ static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec,
 		case 0x1a:
 			idx1 = 2;	/* Line In */
 			break;
-		case 0x1c:	
+		case 0x1c:
 			idx1 = 3;	/* CD */
 			break;
 		case 0x12:
@@ -10614,7 +11303,7 @@ static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec,
 		}
 		imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
 		imux->items[imux->num_items].index = idx1;
-		imux->num_items++;	
+		imux->num_items++;
 	}
 	return 0;
 }
@@ -10644,11 +11333,11 @@ static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec)
 	}
 
 	dac_vol1 = dac_vol2 = 0xb000 | 0x40;	/* set max volume  */
-	if (line_nid == 0x14)	
+	if (line_nid == 0x14)
 		dac_vol2 = AMP_OUT_ZERO;
 	else if (line_nid == 0x15)
 		dac_vol1 = AMP_OUT_ZERO;
-	if (hp_nid == 0x14)	
+	if (hp_nid == 0x14)
 		dac_vol2 = AMP_OUT_ZERO;
 	else if (hp_nid == 0x15)
 		dac_vol1 = AMP_OUT_ZERO;
@@ -10739,6 +11428,7 @@ static const char *alc268_models[ALC268_MODEL_LAST] = {
 	[ALC268_3ST]		= "3stack",
 	[ALC268_TOSHIBA]	= "toshiba",
 	[ALC268_ACER]		= "acer",
+	[ALC268_ACER_ASPIRE_ONE]	= "acer-aspire",
 	[ALC268_DELL]		= "dell",
 	[ALC268_ZEPTO]		= "zepto",
 #ifdef CONFIG_SND_DEBUG
@@ -10753,11 +11443,14 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
 	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(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA),
 	SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
 	SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
 	SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),
+	SND_PCI_QUIRK(0x1179, 0xff64, "TOSHIBA L305", ALC268_TOSHIBA),
 	SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
 	SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
 	SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
@@ -10830,6 +11523,23 @@ static struct alc_config_preset alc268_presets[] = {
 		.unsol_event = alc268_acer_unsol_event,
 		.init_hook = alc268_acer_init_hook,
 	},
+	[ALC268_ACER_ASPIRE_ONE] = {
+		.mixers = { alc268_acer_aspire_one_mixer,
+				alc268_capture_alt_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,
+		.input_mux = &alc268_acer_lc_capture_source,
+		.unsol_event = alc268_acer_lc_unsol_event,
+		.init_hook = alc268_acer_lc_init_hook,
+	},
 	[ALC268_DELL] = {
 		.mixers = { alc268_dell_mixer, alc268_beep_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
@@ -10974,7 +11684,7 @@ static int patch_alc268(struct hda_codec *codec)
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC268_AUTO)
 		spec->init_hook = alc268_auto_init;
-		
+
 	return 0;
 }
 
@@ -10990,6 +11700,14 @@ static hda_nid_t alc269_adc_nids[1] = {
 	0x08,
 };
 
+static hda_nid_t alc269_capsrc_nids[1] = {
+	0x23,
+};
+
+/* NOTE: ADC2 (0x07) is connected from a recording *MIXER* (0x24),
+ *       not a mux!
+ */
+
 static struct hda_input_mux alc269_eeepc_dmic_capture_source = {
 	.num_items = 2,
 	.items = {
@@ -11016,6 +11734,8 @@ static struct snd_kcontrol_new alc269_base_mixer[] = {
 	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("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 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),
@@ -11025,6 +11745,28 @@ static struct snd_kcontrol_new alc269_base_mixer[] = {
 	{ } /* end */
 };
 
+static 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",
+		.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", 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", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	{ }
+};
+
 /* bind volumes of both NID 0x0c and 0x0d */
 static struct hda_bind_ctls alc269_epc_bind_vol = {
 	.ops = &snd_hda_bind_vol,
@@ -11068,75 +11810,72 @@ static struct snd_kcontrol_new alc269_epc_capture_mixer[] = {
 	{ } /* end */
 };
 
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static 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)},
+/* beep control */
+static struct snd_kcontrol_new alc269_beep_mixer[] = {
+	HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT),
+	{ } /* end */
+};
 
-	/* Mute input amps (PCBeep, 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)},
+static 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, ALC880_HP_EVENT | AC_USRSP_EN},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{ }
+};
 
-	/*
-	 * Set up output mixers (0x0c - 0x0e)
-	 */
-	/* 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},
+/* toggle speaker-output according to the hp-jack state */
+static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned char bits;
 
-	/* 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)},
+	present = snd_hda_codec_read(codec, 0x15, 0,
+			AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	bits = present ? AMP_IN_MUTE(0) : 0;
+	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
+			AMP_IN_MUTE(0), bits);
+	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
+			AMP_IN_MUTE(0), bits);
 
-	{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},
+	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);
 
-	{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_MUTE},
-	{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},
+	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);
+}
 
-	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+static void alc269_quanta_fl1_mic_automute(struct hda_codec *codec)
+{
+	unsigned int present;
 
-	/* FIXME: use matrix-type input source selection */
-	/* Mixer elements: 0x18, 19, 1a, 1b, 1d, 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(1)},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	present = snd_hda_codec_read(codec, 0x18, 0,
+				AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	snd_hda_codec_write(codec, 0x23, 0,
+			    AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x1);
+}
 
-	/* set EAPD */
-	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
-	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
-	{ }
-};
+static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec,
+				    unsigned int res)
+{
+	if ((res >> 26) == ALC880_HP_EVENT)
+		alc269_quanta_fl1_speaker_automute(codec);
+	if ((res >> 26) == ALC880_MIC_EVENT)
+		alc269_quanta_fl1_mic_automute(codec);
+}
+
+static void alc269_quanta_fl1_init_hook(struct hda_codec *codec)
+{
+	alc269_quanta_fl1_speaker_automute(codec);
+	alc269_quanta_fl1_mic_automute(codec);
+}
 
 static struct hda_verb alc269_eeepc_dmic_init_verbs[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
@@ -11163,42 +11902,42 @@ static struct hda_verb alc269_eeepc_amic_init_verbs[] = {
 static void alc269_speaker_automute(struct hda_codec *codec)
 {
 	unsigned int present;
-	unsigned int bits;
+	unsigned char bits;
 
 	present = snd_hda_codec_read(codec, 0x15, 0,
-				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+				AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
 	bits = present ? AMP_IN_MUTE(0) : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				 AMP_IN_MUTE(0), bits);
+				AMP_IN_MUTE(0), bits);
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				 AMP_IN_MUTE(0), bits);
+				AMP_IN_MUTE(0), bits);
 }
 
 static void alc269_eeepc_dmic_automute(struct hda_codec *codec)
 {
 	unsigned int present;
 
-	present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0)
-		& AC_PINSENSE_PRESENCE;
-	snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_CONNECT_SEL,
-			    present ? 0 : 5);
+	present = snd_hda_codec_read(codec, 0x18, 0,
+				AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	snd_hda_codec_write(codec, 0x23, 0,
+				AC_VERB_SET_CONNECT_SEL,  (present ? 0 : 5));
 }
 
 static void alc269_eeepc_amic_automute(struct hda_codec *codec)
 {
 	unsigned int present;
 
-	present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0)
-		& AC_PINSENSE_PRESENCE;
+	present = snd_hda_codec_read(codec, 0x18, 0,
+				AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
 	snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-			    present ? AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0));
+				0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
 	snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-			    present ? AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1));
+				0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
 }
 
 /* unsolicited event for HP jack sensing */
 static void alc269_eeepc_dmic_unsol_event(struct hda_codec *codec,
-					  unsigned int res)
+				     unsigned int res)
 {
 	if ((res >> 26) == ALC880_HP_EVENT)
 		alc269_speaker_automute(codec);
@@ -11215,7 +11954,7 @@ static void alc269_eeepc_dmic_inithook(struct hda_codec *codec)
 
 /* unsolicited event for HP jack sensing */
 static void alc269_eeepc_amic_unsol_event(struct hda_codec *codec,
-					  unsigned int res)
+				     unsigned int res)
 {
 	if ((res >> 26) == ALC880_HP_EVENT)
 		alc269_speaker_automute(codec);
@@ -11230,6 +11969,76 @@ static void alc269_eeepc_amic_inithook(struct hda_codec *codec)
 	alc269_eeepc_amic_automute(codec);
 }
 
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc269_init_verbs[] = {
+	/*
+	 * Unmute ADC0 and set the default input to mic-in
+	 */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+	/* Mute input amps (PCBeep, 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 */
+	{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},
+	{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},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* FIXME: use matrix-type input source selection */
+	/* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
+	/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+	{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)},
+
+	/* set EAPD */
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{ }
+};
+
 /* add playback controls from the parsed DAC table */
 static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec,
 					     const struct auto_pin_cfg *cfg)
@@ -11330,7 +12139,7 @@ static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec,
 static int alc269_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	int err;
+	int i, err;
 	static hda_nid_t alc269_ignore[] = { 0x1d, 0 };
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
@@ -11353,9 +12162,20 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	if (spec->kctl_alloc)
 		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
 
+	/* create a beep mixer control if the pin 0x1d isn't assigned */
+	for (i = 0; i < ARRAY_SIZE(spec->autocfg.input_pins); i++)
+		if (spec->autocfg.input_pins[i] == 0x1d)
+			break;
+	if (i >= ARRAY_SIZE(spec->autocfg.input_pins))
+		spec->mixers[spec->num_mixers++] = alc269_beep_mixer;
+
 	spec->init_verbs[spec->num_init_verbs++] = alc269_init_verbs;
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
+	/* set default input source */
+	snd_hda_codec_write_cache(codec, alc269_capsrc_nids[0],
+				  0, AC_VERB_SET_CONNECT_SEL,
+				  spec->input_mux->items[0].index);
 
 	err = alc_auto_add_mic_boost(codec);
 	if (err < 0)
@@ -11387,14 +12207,20 @@ static void alc269_auto_init(struct hda_codec *codec)
  * configuration and preset
  */
 static const char *alc269_models[ALC269_MODEL_LAST] = {
-	[ALC269_BASIC]		= "basic",
+	[ALC269_BASIC]			= "basic",
+	[ALC269_QUANTA_FL1]		= "quanta",
+	[ALC269_ASUS_EEEPC_P703]	= "eeepc-p703",
+	[ALC269_ASUS_EEEPC_P901]	= "eeepc-p901"
 };
 
 static struct snd_pci_quirk alc269_cfg_tbl[] = {
+	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1),
 	SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
 		      ALC269_ASUS_EEEPC_P703),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
 		      ALC269_ASUS_EEEPC_P901),
+	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
+		      ALC269_ASUS_EEEPC_P901),
 	{}
 };
 
@@ -11409,6 +12235,18 @@ static struct alc_config_preset alc269_presets[] = {
 		.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,
+		.init_hook = alc269_quanta_fl1_init_hook,
+	},
 	[ALC269_ASUS_EEEPC_P703] = {
 		.mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer },
 		.init_verbs = { alc269_init_verbs,
@@ -11488,6 +12326,7 @@ static int patch_alc269(struct hda_codec *codec)
 
 	spec->adc_nids = alc269_adc_nids;
 	spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
+	spec->capsrc_nids = alc269_capsrc_nids;
 
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC269_AUTO)
@@ -11689,7 +12528,7 @@ static struct snd_kcontrol_new alc861_toshiba_mixer[] = {
 	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),
-	
+
         /*Capture mixer control */
 	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
@@ -11832,20 +12671,20 @@ static struct hda_verb alc861_base_init_verbs[] = {
 	/* 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)},
@@ -11901,13 +12740,13 @@ static struct hda_verb alc861_threestack_init_verbs[] = {
 	{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)},
@@ -11963,13 +12802,13 @@ static struct hda_verb alc861_uniwill_m31_init_verbs[] = {
 	{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)},
@@ -12034,7 +12873,7 @@ static struct hda_verb alc861_asus_init_verbs[] = {
         {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)},
@@ -12071,20 +12910,20 @@ static struct hda_verb alc861_auto_init_verbs[] = {
 	 */
 	/* {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_MUTE},
 	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{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)},
@@ -12659,7 +13498,7 @@ static int patch_alc861(struct hda_codec *codec)
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc861_loopbacks;
 #endif
-		
+
 	return 0;
 }
 
@@ -12913,7 +13752,7 @@ static struct snd_kcontrol_new alc861vd_hp_mixer[] = {
 	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 */
 };
 
@@ -13058,7 +13897,7 @@ static struct hda_verb alc861vd_lenovo_unsol_verbs[] = {
 	{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 | ALC880_HP_EVENT},
-	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},	
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
 	{}
 };
 
@@ -13120,7 +13959,7 @@ static struct hda_verb alc861vd_dallas_verbs[] = {
 	{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},
@@ -13145,7 +13984,7 @@ static struct hda_verb alc861vd_dallas_verbs[] = {
 	{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},	
+	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
 
 	{ } /* end */
@@ -13304,7 +14143,7 @@ static struct alc_config_preset alc861vd_presets[] = {
 		.input_mux = &alc861vd_hp_capture_source,
 		.unsol_event = alc861vd_dallas_unsol_event,
 		.init_hook = alc861vd_dallas_automute,
-	},		
+	},
 };
 
 /*
@@ -13883,13 +14722,120 @@ static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = {
 	{ } /* end */
 };
 
+static 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 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 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 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 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 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 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 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 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 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 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),
-	HDA_CODEC_MUTE("DMic Playback Switch", 0x23, 0x9, HDA_INPUT),
+	{ } /* end */
+};
+
+static 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 */
 };
 
@@ -14074,14 +15020,81 @@ static struct hda_verb alc663_auto_init_verbs[] = {
 };
 
 static 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, 0x00},	/* Headphone */
+	{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 | ALC880_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{}
+};
 
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+static 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 | ALC880_MIC_EVENT},
+	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{}
+};
 
+static 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 | ALC880_MIC_EVENT},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{}
+};
+
+static 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 | ALC880_MIC_EVENT},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{}
+};
+
+static 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 | ALC880_MIC_EVENT},
 	{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{}
+};
+
+static 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 | ALC880_MIC_EVENT},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
 	{}
 };
 
@@ -14110,6 +15123,14 @@ static struct hda_verb alc663_g50v_init_verbs[] = {
 	{}
 };
 
+static struct hda_verb alc662_ecs_init_verbs[] = {
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{}
+};
+
 /* capture mixer elements */
 static struct snd_kcontrol_new alc662_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
@@ -14129,6 +15150,12 @@ static struct snd_kcontrol_new alc662_capture_mixer[] = {
 	{ } /* end */
 };
 
+static 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 void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
 {
 	unsigned int present;
@@ -14209,12 +15236,12 @@ static void alc662_eeepc_ep20_automute(struct hda_codec *codec)
 	if (present) {
 		/* mute internal speaker */
 		snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
-					 HDA_AMP_MUTE, HDA_AMP_MUTE);
+					HDA_AMP_MUTE, HDA_AMP_MUTE);
 	} else {
 		/* unmute internal speaker if necessary */
 		mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0);
 		snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
-					 HDA_AMP_MUTE, mute);
+					HDA_AMP_MUTE, mute);
 	}
 }
 
@@ -14237,11 +15264,108 @@ static void alc663_m51va_speaker_automute(struct hda_codec *codec)
 	unsigned char bits;
 
 	present = snd_hda_codec_read(codec, 0x21, 0,
-				     AC_VERB_GET_PIN_SENSE, 0)
-		& AC_PINSENSE_PRESENCE;
+			AC_VERB_GET_PIN_SENSE, 0)
+			& AC_PINSENSE_PRESENCE;
 	bits = present ? HDA_AMP_MUTE : 0;
-	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
+	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
+				AMP_IN_MUTE(0), bits);
+	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
+				AMP_IN_MUTE(0), bits);
+}
+
+static void alc663_21jd_two_speaker_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned char bits;
+
+	present = snd_hda_codec_read(codec, 0x21, 0,
+			AC_VERB_GET_PIN_SENSE, 0)
+			& AC_PINSENSE_PRESENCE;
+	bits = present ? HDA_AMP_MUTE : 0;
+	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
+				AMP_IN_MUTE(0), bits);
+	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
+				AMP_IN_MUTE(0), bits);
+	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
+				AMP_IN_MUTE(0), bits);
+	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
+				AMP_IN_MUTE(0), bits);
+}
+
+static void alc663_15jd_two_speaker_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned char bits;
+
+	present = snd_hda_codec_read(codec, 0x15, 0,
+			AC_VERB_GET_PIN_SENSE, 0)
+			& AC_PINSENSE_PRESENCE;
+	bits = present ? HDA_AMP_MUTE : 0;
+	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
+				AMP_IN_MUTE(0), bits);
+	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
+				AMP_IN_MUTE(0), bits);
+	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
+				AMP_IN_MUTE(0), bits);
+	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
+				AMP_IN_MUTE(0), bits);
+}
+
+static void alc662_f5z_speaker_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned char bits;
+
+	present = snd_hda_codec_read(codec, 0x1b, 0,
+			AC_VERB_GET_PIN_SENSE, 0)
+			& AC_PINSENSE_PRESENCE;
+	bits = present ? 0 : PIN_OUT;
+	snd_hda_codec_write(codec, 0x14, 0,
+			 AC_VERB_SET_PIN_WIDGET_CONTROL, bits);
+}
+
+static void alc663_two_hp_m1_speaker_automute(struct hda_codec *codec)
+{
+	unsigned int present1, present2;
+
+	present1 = snd_hda_codec_read(codec, 0x21, 0,
+			AC_VERB_GET_PIN_SENSE, 0)
+			& AC_PINSENSE_PRESENCE;
+	present2 = snd_hda_codec_read(codec, 0x15, 0,
+			AC_VERB_GET_PIN_SENSE, 0)
+			& AC_PINSENSE_PRESENCE;
+
+	if (present1 || present2) {
+		snd_hda_codec_write_cache(codec, 0x14, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+	} else {
+		snd_hda_codec_write_cache(codec, 0x14, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+	}
+}
+
+static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec)
+{
+	unsigned int present1, present2;
+
+	present1 = snd_hda_codec_read(codec, 0x1b, 0,
+				AC_VERB_GET_PIN_SENSE, 0)
+				& AC_PINSENSE_PRESENCE;
+	present2 = snd_hda_codec_read(codec, 0x15, 0,
+				AC_VERB_GET_PIN_SENSE, 0)
+				& AC_PINSENSE_PRESENCE;
+
+	if (present1 || present2) {
+		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
+				AMP_IN_MUTE(0), AMP_IN_MUTE(0));
+		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
+				AMP_IN_MUTE(0), AMP_IN_MUTE(0));
+	} else {
+		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
+				AMP_IN_MUTE(0), 0);
+		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
+				AMP_IN_MUTE(0), 0);
+	}
 }
 
 static void alc663_m51va_mic_automute(struct hda_codec *codec)
@@ -14249,16 +15373,16 @@ static void alc663_m51va_mic_automute(struct hda_codec *codec)
 	unsigned int present;
 
 	present = snd_hda_codec_read(codec, 0x18, 0,
-				     AC_VERB_GET_PIN_SENSE, 0)
-		& AC_PINSENSE_PRESENCE;
+			AC_VERB_GET_PIN_SENSE, 0)
+			& AC_PINSENSE_PRESENCE;
 	snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-			    0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
+			0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
 	snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-			    0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
+			0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
 	snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-			    0x7000 | (0x09 << 8) | (present ? 0x80 : 0));
+			0x7000 | (0x09 << 8) | (present ? 0x80 : 0));
 	snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-			    0x7000 | (0x09 << 8) | (present ? 0x80 : 0));
+			0x7000 | (0x09 << 8) | (present ? 0x80 : 0));
 }
 
 static void alc663_m51va_unsol_event(struct hda_codec *codec,
@@ -14280,6 +15404,121 @@ static void alc663_m51va_inithook(struct hda_codec *codec)
 	alc663_m51va_mic_automute(codec);
 }
 
+/* ***************** Mode1 ******************************/
+static void alc663_mode1_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
+		alc663_m51va_speaker_automute(codec);
+		break;
+	case ALC880_MIC_EVENT:
+		alc662_eeepc_mic_automute(codec);
+		break;
+	}
+}
+
+static void alc663_mode1_inithook(struct hda_codec *codec)
+{
+	alc663_m51va_speaker_automute(codec);
+	alc662_eeepc_mic_automute(codec);
+}
+/* ***************** Mode2 ******************************/
+static void alc662_mode2_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
+		alc662_f5z_speaker_automute(codec);
+		break;
+	case ALC880_MIC_EVENT:
+		alc662_eeepc_mic_automute(codec);
+		break;
+	}
+}
+
+static void alc662_mode2_inithook(struct hda_codec *codec)
+{
+	alc662_f5z_speaker_automute(codec);
+	alc662_eeepc_mic_automute(codec);
+}
+/* ***************** Mode3 ******************************/
+static void alc663_mode3_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
+		alc663_two_hp_m1_speaker_automute(codec);
+		break;
+	case ALC880_MIC_EVENT:
+		alc662_eeepc_mic_automute(codec);
+		break;
+	}
+}
+
+static void alc663_mode3_inithook(struct hda_codec *codec)
+{
+	alc663_two_hp_m1_speaker_automute(codec);
+	alc662_eeepc_mic_automute(codec);
+}
+/* ***************** Mode4 ******************************/
+static void alc663_mode4_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
+		alc663_21jd_two_speaker_automute(codec);
+		break;
+	case ALC880_MIC_EVENT:
+		alc662_eeepc_mic_automute(codec);
+		break;
+	}
+}
+
+static void alc663_mode4_inithook(struct hda_codec *codec)
+{
+	alc663_21jd_two_speaker_automute(codec);
+	alc662_eeepc_mic_automute(codec);
+}
+/* ***************** Mode5 ******************************/
+static void alc663_mode5_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
+		alc663_15jd_two_speaker_automute(codec);
+		break;
+	case ALC880_MIC_EVENT:
+		alc662_eeepc_mic_automute(codec);
+		break;
+	}
+}
+
+static void alc663_mode5_inithook(struct hda_codec *codec)
+{
+	alc663_15jd_two_speaker_automute(codec);
+	alc662_eeepc_mic_automute(codec);
+}
+/* ***************** Mode6 ******************************/
+static void alc663_mode6_unsol_event(struct hda_codec *codec,
+					   unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
+		alc663_two_hp_m2_speaker_automute(codec);
+		break;
+	case ALC880_MIC_EVENT:
+		alc662_eeepc_mic_automute(codec);
+		break;
+	}
+}
+
+static void alc663_mode6_inithook(struct hda_codec *codec)
+{
+	alc663_two_hp_m2_speaker_automute(codec);
+	alc662_eeepc_mic_automute(codec);
+}
+
 static void alc663_g71v_hp_automute(struct hda_codec *codec)
 {
 	unsigned int present;
@@ -14350,6 +15589,46 @@ static void alc663_g50v_inithook(struct hda_codec *codec)
 	alc662_eeepc_mic_automute(codec);
 }
 
+/* bind hp and internal speaker mute (with plug check) */
+static int alc662_ecs_master_sw_put(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	long *valp = ucontrol->value.integer.value;
+	int change;
+
+	change = snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0,
+					  HDA_AMP_MUTE,
+					  valp[0] ? 0 : HDA_AMP_MUTE);
+	change |= snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0,
+					   HDA_AMP_MUTE,
+					   valp[1] ? 0 : HDA_AMP_MUTE);
+	if (change)
+		alc262_hippo1_automute(codec);
+	return change;
+}
+
+static struct snd_kcontrol_new alc662_ecs_mixer[] = {
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Master Playback Switch",
+		.info = snd_hda_mixer_amp_switch_info,
+		.get = snd_hda_mixer_amp_switch_get,
+		.put = alc662_ecs_master_sw_put,
+		.private_value = HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+	},
+
+	HDA_CODEC_VOLUME("e-Mic/LineIn Boost", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("e-Mic/LineIn Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("e-Mic/LineIn Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("i-Mic Boost", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 #define alc662_loopbacks	alc880_loopbacks
 #endif
@@ -14372,21 +15651,67 @@ static const char *alc662_models[ALC662_MODEL_LAST] = {
 	[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",
 	[ALC662_AUTO]		= "auto",
 };
 
 static struct snd_pci_quirk alc662_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS G71V", ALC663_ASUS_G71V),
 	SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA),
 	SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS M51VA", ALC663_ASUS_G50V),
 	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(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4),
+	SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5),
+	SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6),
+	SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
+		      ALC662_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
+	SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS),
+	SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
+	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
+		      ALC662_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0",
+					ALC662_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1854, 0x2000, "ASUS H13-2000", ALC663_ASUS_H13),
 	SND_PCI_QUIRK(0x1854, 0x2001, "ASUS H13-2001", ALC663_ASUS_H13),
 	SND_PCI_QUIRK(0x1854, 0x2002, "ASUS H13-2002", ALC663_ASUS_H13),
@@ -14477,6 +15802,18 @@ static struct alc_config_preset alc662_presets[] = {
 		.unsol_event = alc662_eeepc_ep20_unsol_event,
 		.init_hook = alc662_eeepc_ep20_inithook,
 	},
+	[ALC662_ECS] = {
+		.mixers = { alc662_ecs_mixer, alc662_capture_mixer },
+		.init_verbs = { alc662_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,
+		.input_mux = &alc662_eeepc_capture_source,
+		.unsol_event = alc662_eeepc_unsol_event,
+		.init_hook = alc662_eeepc_inithook,
+	},
 	[ALC663_ASUS_M51VA] = {
 		.mixers = { alc663_m51va_mixer, alc662_capture_mixer},
 		.init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs },
@@ -14524,6 +15861,91 @@ static struct alc_config_preset alc662_presets[] = {
 		.unsol_event = alc663_g50v_unsol_event,
 		.init_hook = alc663_g50v_inithook,
 	},
+	[ALC663_ASUS_MODE1] = {
+		.mixers = { alc663_m51va_mixer, alc662_auto_capture_mixer },
+		.init_verbs = { alc662_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,
+		.input_mux = &alc662_eeepc_capture_source,
+		.unsol_event = alc663_mode1_unsol_event,
+		.init_hook = alc663_mode1_inithook,
+	},
+	[ALC662_ASUS_MODE2] = {
+		.mixers = { alc662_1bjd_mixer, alc662_auto_capture_mixer },
+		.init_verbs = { alc662_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,
+		.input_mux = &alc662_eeepc_capture_source,
+		.unsol_event = alc662_mode2_unsol_event,
+		.init_hook = alc662_mode2_inithook,
+	},
+	[ALC663_ASUS_MODE3] = {
+		.mixers = { alc663_two_hp_m1_mixer, alc662_auto_capture_mixer },
+		.init_verbs = { alc662_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,
+		.input_mux = &alc662_eeepc_capture_source,
+		.unsol_event = alc663_mode3_unsol_event,
+		.init_hook = alc663_mode3_inithook,
+	},
+	[ALC663_ASUS_MODE4] = {
+		.mixers = { alc663_asus_21jd_clfe_mixer,
+				alc662_auto_capture_mixer},
+		.init_verbs = { alc662_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,
+		.input_mux = &alc662_eeepc_capture_source,
+		.unsol_event = alc663_mode4_unsol_event,
+		.init_hook = alc663_mode4_inithook,
+	},
+	[ALC663_ASUS_MODE5] = {
+		.mixers = { alc663_asus_15jd_clfe_mixer,
+				alc662_auto_capture_mixer },
+		.init_verbs = { alc662_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,
+		.input_mux = &alc662_eeepc_capture_source,
+		.unsol_event = alc663_mode5_unsol_event,
+		.init_hook = alc663_mode5_inithook,
+	},
+	[ALC663_ASUS_MODE6] = {
+		.mixers = { alc663_two_hp_m2_mixer, alc662_auto_capture_mixer },
+		.init_verbs = { alc662_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,
+		.input_mux = &alc662_eeepc_capture_source,
+		.unsol_event = alc663_mode6_unsol_event,
+		.init_hook = alc663_mode6_inithook,
+	},
 };
 
 
@@ -14560,15 +15982,15 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
 							      HDA_OUTPUT));
 			if (err < 0)
 				return err;
-			err = add_control(spec, ALC_CTL_BIND_MUTE,
+			err = add_control(spec, ALC_CTL_WIDGET_MUTE,
 					  "Center Playback Switch",
-					  HDA_COMPOSE_AMP_VAL(nid, 1, 2,
+					  HDA_COMPOSE_AMP_VAL(0x0e, 1, 0,
 							      HDA_INPUT));
 			if (err < 0)
 				return err;
-			err = add_control(spec, ALC_CTL_BIND_MUTE,
+			err = add_control(spec, ALC_CTL_WIDGET_MUTE,
 					  "LFE Playback Switch",
-					  HDA_COMPOSE_AMP_VAL(nid, 2, 2,
+					  HDA_COMPOSE_AMP_VAL(0x0e, 2, 0,
 							      HDA_INPUT));
 			if (err < 0)
 				return err;
@@ -14580,9 +16002,9 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
 			if (err < 0)
 				return err;
 			sprintf(name, "%s Playback Switch", chname[i]);
-			err = add_control(spec, ALC_CTL_BIND_MUTE, name,
-					  HDA_COMPOSE_AMP_VAL(nid, 3, 2,
-							      HDA_INPUT));
+			err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+				HDA_COMPOSE_AMP_VAL(alc880_idx_to_mixer(i),
+						    3, 0, HDA_INPUT));
 			if (err < 0)
 				return err;
 		}
@@ -14777,7 +16199,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
-	
+
 	spec->init_verbs[spec->num_init_verbs++] = alc662_auto_init_verbs;
 	if (codec->vendor_id == 0x10ec0663)
 		spec->init_verbs[spec->num_init_verbs++] =
@@ -14896,6 +16318,8 @@ struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
 	{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
 	{ .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 },
+	{ .id = 0x10ec0885, .rev = 0x100101, .name = "ALC889A",
+	  .patch = patch_alc882 }, /* should be patch_alc883() in future */
 	{ .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
 	  .patch = patch_alc882 }, /* should be patch_alc883() in future */
 	{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index ad994fcab725..c461baa83c2a 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -33,10 +33,12 @@
 #include "hda_codec.h"
 #include "hda_local.h"
 #include "hda_patch.h"
+#include "hda_beep.h"
 
 #define NUM_CONTROL_ALLOC	32
 #define STAC_PWR_EVENT		0x20
 #define STAC_HP_EVENT		0x30
+#define STAC_VREF_EVENT		0x40
 
 enum {
 	STAC_REF,
@@ -71,9 +73,15 @@ enum {
 };
 
 enum {
+	STAC_92HD83XXX_REF,
+	STAC_92HD83XXX_MODELS
+};
+
+enum {
 	STAC_92HD71BXX_REF,
 	STAC_DELL_M4_1,
 	STAC_DELL_M4_2,
+	STAC_HP_M4,
 	STAC_92HD71BXX_MODELS
 };
 
@@ -104,6 +112,7 @@ enum {
 	STAC_MACBOOK_PRO_V2,
 	STAC_IMAC_INTEL,
 	STAC_IMAC_INTEL_20,
+	STAC_ECS_202,
 	STAC_922X_DELL_D81,
 	STAC_922X_DELL_D82,
 	STAC_922X_DELL_M81,
@@ -130,6 +139,7 @@ struct sigmatel_spec {
 	unsigned int mic_switch: 1;
 	unsigned int alt_switch: 1;
 	unsigned int hp_detect: 1;
+	unsigned int spdif_mute: 1;
 
 	/* gpio lines */
 	unsigned int eapd_mask;
@@ -138,17 +148,22 @@ struct sigmatel_spec {
 	unsigned int gpio_data;
 	unsigned int gpio_mute;
 
+	/* stream */
+	unsigned int stream_delay;
+
 	/* analog loopback */
 	unsigned char aloopback_mask;
 	unsigned char aloopback_shift;
 
 	/* power management */
 	unsigned int num_pwrs;
+	unsigned int *pwr_mapping;
 	hda_nid_t *pwr_nids;
 	hda_nid_t *dac_list;
 
 	/* playback */
 	struct hda_input_mux *mono_mux;
+	struct hda_input_mux *amp_mux;
 	unsigned int cur_mmux;
 	struct hda_multi_out multiout;
 	hda_nid_t dac_nids[5];
@@ -162,8 +177,14 @@ struct sigmatel_spec {
 	unsigned int num_dmics;
 	hda_nid_t *dmux_nids;
 	unsigned int num_dmuxes;
+	hda_nid_t *smux_nids;
+	unsigned int num_smuxes;
+	const char **spdif_labels;
+
 	hda_nid_t dig_in_nid;
 	hda_nid_t mono_nid;
+	hda_nid_t anabeep_nid;
+	hda_nid_t digbeep_nid;
 
 	/* pin widgets */
 	hda_nid_t *pin_nids;
@@ -180,6 +201,12 @@ struct sigmatel_spec {
 	unsigned int cur_dmux[2];
 	struct hda_input_mux *input_mux;
 	unsigned int cur_mux[3];
+	struct hda_input_mux *sinput_mux;
+	unsigned int cur_smux[2];
+	unsigned int cur_amux;
+	hda_nid_t *amp_nids;
+	unsigned int num_amps;
+	unsigned int powerdown_adcs;
 
 	/* i/o switches */
 	unsigned int io_switch[2];
@@ -195,6 +222,8 @@ struct sigmatel_spec {
 	struct snd_kcontrol_new *kctl_alloc;
 	struct hda_input_mux private_dimux;
 	struct hda_input_mux private_imux;
+	struct hda_input_mux private_smux;
+	struct hda_input_mux private_amp_mux;
 	struct hda_input_mux private_mono_mux;
 };
 
@@ -215,10 +244,19 @@ static hda_nid_t stac92hd73xx_pwr_nids[8] = {
 	0x0f, 0x10, 0x11
 };
 
+static hda_nid_t stac92hd73xx_slave_dig_outs[2] = {
+	0x26, 0,
+};
+
 static hda_nid_t stac92hd73xx_adc_nids[2] = {
 	0x1a, 0x1b
 };
 
+#define DELL_M6_AMP 2
+static hda_nid_t stac92hd73xx_amp_nids[3] = {
+	0x0b, 0x0c, 0x0e
+};
+
 #define STAC92HD73XX_NUM_DMICS	2
 static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = {
 	0x13, 0x14, 0
@@ -237,6 +275,41 @@ static hda_nid_t stac92hd73xx_dmux_nids[2] = {
 	0x20, 0x21,
 };
 
+static hda_nid_t stac92hd73xx_smux_nids[2] = {
+	0x22, 0x23,
+};
+
+#define STAC92HD83XXX_NUM_DMICS	2
+static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = {
+	0x11, 0x12, 0
+};
+
+#define STAC92HD81_DAC_COUNT 2
+#define STAC92HD83_DAC_COUNT 3
+static hda_nid_t stac92hd83xxx_dac_nids[STAC92HD73_DAC_COUNT] = {
+	0x13, 0x14, 0x22,
+};
+
+static hda_nid_t stac92hd83xxx_dmux_nids[2] = {
+	0x17, 0x18,
+};
+
+static hda_nid_t stac92hd83xxx_adc_nids[2] = {
+	0x15, 0x16,
+};
+
+static hda_nid_t stac92hd83xxx_pwr_nids[4] = {
+	0xa, 0xb, 0xd, 0xe,
+};
+
+static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = {
+	0x1e, 0,
+};
+
+static unsigned int stac92hd83xxx_pwr_mapping[4] = {
+	0x03, 0x0c, 0x10, 0x40,
+};
+
 static hda_nid_t stac92hd71bxx_pwr_nids[3] = {
 	0x0a, 0x0d, 0x0f
 };
@@ -253,6 +326,10 @@ static hda_nid_t stac92hd71bxx_dmux_nids[1] = {
 	0x1c,
 };
 
+static hda_nid_t stac92hd71bxx_smux_nids[2] = {
+	0x24, 0x25,
+};
+
 static hda_nid_t stac92hd71bxx_dac_nids[1] = {
 	0x10, /*0x11, */
 };
@@ -262,6 +339,10 @@ static hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = {
 	0x18, 0x19, 0
 };
 
+static hda_nid_t stac92hd71bxx_slave_dig_outs[2] = {
+	0x22, 0
+};
+
 static hda_nid_t stac925x_adc_nids[1] = {
         0x03,
 };
@@ -299,6 +380,10 @@ static hda_nid_t stac927x_mux_nids[3] = {
         0x15, 0x16, 0x17
 };
 
+static hda_nid_t stac927x_smux_nids[1] = {
+	0x21,
+};
+
 static hda_nid_t stac927x_dac_nids[6] = {
 	0x02, 0x03, 0x04, 0x05, 0x06, 0
 };
@@ -312,6 +397,11 @@ static hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = {
 	0x13, 0x14, 0
 };
 
+static const char *stac927x_spdif_labels[5] = {
+	"Digital Playback", "ADAT", "Analog Mux 1",
+	"Analog Mux 2", "Analog Mux 3"
+};
+
 static hda_nid_t stac9205_adc_nids[2] = {
         0x12, 0x13
 };
@@ -324,6 +414,10 @@ static hda_nid_t stac9205_dmux_nids[1] = {
 	0x1d,
 };
 
+static hda_nid_t stac9205_smux_nids[1] = {
+	0x21,
+};
+
 #define STAC9205_NUM_DMICS	2
 static hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = {
         0x17, 0x18, 0
@@ -347,12 +441,18 @@ static hda_nid_t stac922x_pin_nids[10] = {
 static hda_nid_t stac92hd73xx_pin_nids[13] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
 	0x0f, 0x10, 0x11, 0x12, 0x13,
-	0x14, 0x1e, 0x22
+	0x14, 0x22, 0x23
 };
 
-static hda_nid_t stac92hd71bxx_pin_nids[10] = {
+static hda_nid_t stac92hd83xxx_pin_nids[14] = {
+	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
+	0x0f, 0x10, 0x11, 0x12, 0x13,
+	0x1d, 0x1e, 0x1f, 0x20
+};
+static hda_nid_t stac92hd71bxx_pin_nids[11] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
 	0x0f, 0x14, 0x18, 0x19, 0x1e,
+	0x1f,
 };
 
 static hda_nid_t stac927x_pin_nids[14] = {
@@ -367,6 +467,34 @@ static hda_nid_t stac9205_pin_nids[12] = {
 	0x21, 0x22,
 };
 
+#define stac92xx_amp_volume_info snd_hda_mixer_amp_volume_info
+
+static int stac92xx_amp_volume_get(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct sigmatel_spec *spec = codec->spec;
+	hda_nid_t nid = spec->amp_nids[spec->cur_amux];
+
+	kcontrol->private_value ^= get_amp_nid(kcontrol);
+	kcontrol->private_value |= nid;
+
+	return snd_hda_mixer_amp_volume_get(kcontrol, ucontrol);
+}
+
+static int stac92xx_amp_volume_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct sigmatel_spec *spec = codec->spec;
+	hda_nid_t nid = spec->amp_nids[spec->cur_amux];
+
+	kcontrol->private_value ^= get_amp_nid(kcontrol);
+	kcontrol->private_value |= nid;
+
+	return snd_hda_mixer_amp_volume_put(kcontrol, ucontrol);
+}
+
 static int stac92xx_dmux_enum_info(struct snd_kcontrol *kcontrol,
 				   struct snd_ctl_elem_info *uinfo)
 {
@@ -397,6 +525,58 @@ static int stac92xx_dmux_enum_put(struct snd_kcontrol *kcontrol,
 			spec->dmux_nids[dmux_idx], &spec->cur_dmux[dmux_idx]);
 }
 
+static int stac92xx_smux_enum_info(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_info *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct sigmatel_spec *spec = codec->spec;
+	return snd_hda_input_mux_info(spec->sinput_mux, uinfo);
+}
+
+static int stac92xx_smux_enum_get(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct sigmatel_spec *spec = codec->spec;
+	unsigned int smux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+	ucontrol->value.enumerated.item[0] = spec->cur_smux[smux_idx];
+	return 0;
+}
+
+static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct sigmatel_spec *spec = codec->spec;
+	struct hda_input_mux *smux = &spec->private_smux;
+	unsigned int smux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+	int err, val;
+	hda_nid_t nid;
+
+	err = snd_hda_input_mux_put(codec, spec->sinput_mux, ucontrol,
+			spec->smux_nids[smux_idx], &spec->cur_smux[smux_idx]);
+	if (err < 0)
+		return err;
+
+	if (spec->spdif_mute) {
+		if (smux_idx == 0)
+			nid = spec->multiout.dig_out_nid;
+		else
+			nid = codec->slave_dig_outs[smux_idx - 1];
+		if (spec->cur_smux[smux_idx] == smux->num_items - 1)
+			val = AMP_OUT_MUTE;
+		if (smux_idx == 0)
+			nid = spec->multiout.dig_out_nid;
+		else
+			nid = codec->slave_dig_outs[smux_idx - 1];
+		/* un/mute SPDIF out */
+		snd_hda_codec_write_cache(codec, nid, 0,
+			AC_VERB_SET_AMP_GAIN_MUTE, val);
+	}
+	return 0;
+}
+
 static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
@@ -452,6 +632,41 @@ static int stac92xx_mono_mux_enum_put(struct snd_kcontrol *kcontrol,
 				     spec->mono_nid, &spec->cur_mmux);
 }
 
+static int stac92xx_amp_mux_enum_info(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct sigmatel_spec *spec = codec->spec;
+	return snd_hda_input_mux_info(spec->amp_mux, uinfo);
+}
+
+static int stac92xx_amp_mux_enum_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct sigmatel_spec *spec = codec->spec;
+
+	ucontrol->value.enumerated.item[0] = spec->cur_amux;
+	return 0;
+}
+
+static int stac92xx_amp_mux_enum_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct sigmatel_spec *spec = codec->spec;
+	struct snd_kcontrol *ctl =
+		snd_hda_find_mixer_ctl(codec, "Amp Capture Volume");
+	if (!ctl)
+		return -EINVAL;
+
+	snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE |
+		SNDRV_CTL_EVENT_MASK_INFO, &ctl->id);
+
+	return snd_hda_input_mux_put(codec, spec->amp_mux, ucontrol,
+				     0, &spec->cur_amux);
+}
+
 #define stac92xx_aloopback_info snd_ctl_boolean_mono_info
 
 static int stac92xx_aloopback_get(struct snd_kcontrol *kcontrol,
@@ -546,8 +761,8 @@ static struct hda_verb dell_eq_core_init[] = {
 	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec},
 	/* setup audio connections */
 	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02},
+	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x02},
+	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x01},
 	/* setup adcs to point to mixer */
 	{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
 	{ 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
@@ -628,6 +843,19 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = {
 	{}
 };
 
+static struct hda_verb stac92hd83xxx_core_init[] = {
+	/* start of config #1 */
+	{ 0xe, AC_VERB_SET_CONNECT_SEL, 0x3},
+
+	/* start of config #2 */
+	{ 0xa, AC_VERB_SET_CONNECT_SEL, 0x0},
+	{ 0xb, AC_VERB_SET_CONNECT_SEL, 0x0},
+	{ 0xd, AC_VERB_SET_CONNECT_SEL, 0x1},
+
+	/* power state controls amps */
+	{ 0x01, AC_VERB_SET_EAPD, 1 << 2},
+};
+
 static struct hda_verb stac92hd71bxx_core_init[] = {
 	/* set master volume and direct control */
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@@ -690,12 +918,16 @@ static struct hda_verb d965_core_init[] = {
 static struct hda_verb stac927x_core_init[] = {
 	/* set master volume and direct control */	
 	{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+	/* enable analog pc beep path */
+	{ 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5},
 	{}
 };
 
 static struct hda_verb stac9205_core_init[] = {
 	/* set master volume and direct control */	
 	{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+	/* enable analog pc beep path */
+	{ 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5},
 	{}
 };
 
@@ -709,6 +941,31 @@ static struct hda_verb stac9205_core_init[] = {
 		.put = stac92xx_mono_mux_enum_put, \
 	}
 
+#define STAC_AMP_MUX \
+	{ \
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+		.name = "Amp Selector Capture Switch", \
+		.count = 1, \
+		.info = stac92xx_amp_mux_enum_info, \
+		.get = stac92xx_amp_mux_enum_get, \
+		.put = stac92xx_amp_mux_enum_put, \
+	}
+
+#define STAC_AMP_VOL(xname, nid, chs, idx, dir) \
+	{ \
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+		.name = xname, \
+		.index = 0, \
+		.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+			SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+			SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
+		.info = stac92xx_amp_volume_info, \
+		.get = stac92xx_amp_volume_get, \
+		.put = stac92xx_amp_volume_put, \
+		.tlv = { .c = snd_hda_mixer_amp_tlv }, \
+		.private_value = HDA_COMPOSE_AMP_VAL(nid, chs, idx, dir) \
+	}
+
 #define STAC_INPUT_SOURCE(cnt) \
 	{ \
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
@@ -736,33 +993,36 @@ static struct snd_kcontrol_new stac9200_mixer[] = {
 	STAC_INPUT_SOURCE(1),
 	HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Mux Volume", 0x0c, 0, HDA_OUTPUT),
 	{ } /* end */
 };
 
+#define DELL_M6_MIXER 6
 static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = {
-	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3),
-
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT),
-
+	/* start of config #1 */
 	HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT),
-
 	HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT),
 	HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT),
 
+	HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT),
+
+	/* start of config #2 */
+	HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT),
+
 	HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT),
 	HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT),
+	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3),
+
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),
+
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT),
+
 	{ } /* end */
 };
 
@@ -818,19 +1078,46 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = {
 	{ } /* end */
 };
 
+
+static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_OUTPUT),
+
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_OUTPUT),
+
+	HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x1, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x2, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x2, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x3, HDA_INPUT),
+
+	/*
+	HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x4, HDA_INPUT),
+	*/
+	{ } /* end */
+};
+
 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
 	STAC_INPUT_SOURCE(2),
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT),
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT),
-
+	/* analog pc-beep replaced with digital beep support */
+	/*
 	HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT),
 	HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT),
+	*/
 
 	HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT),
 	HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT),
@@ -843,11 +1130,9 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT),
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT),
 	{ } /* end */
 };
 
@@ -855,7 +1140,6 @@ static struct snd_kcontrol_new stac925x_mixer[] = {
 	STAC_INPUT_SOURCE(1),
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT),
 	{ } /* end */
 };
 
@@ -865,12 +1149,9 @@ static struct snd_kcontrol_new stac9205_mixer[] = {
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x19, 0x0, HDA_OUTPUT),
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1c, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1e, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x1A, 0x0, HDA_OUTPUT),
-
 	{ } /* end */
 };
 
@@ -879,11 +1160,9 @@ static struct snd_kcontrol_new stac922x_mixer[] = {
 	STAC_INPUT_SOURCE(2),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x12, 0x0, HDA_OUTPUT),
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x13, 0x0, HDA_OUTPUT),
 	{ } /* end */
 };
 
@@ -894,15 +1173,12 @@ static struct snd_kcontrol_new stac927x_mixer[] = {
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x15, 0x0, HDA_OUTPUT),
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x19, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x16, 0x0, HDA_OUTPUT),
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x2, 0x1A, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x2, 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x2, 0x17, 0x0, HDA_OUTPUT),
 	{ } /* end */
 };
 
@@ -915,6 +1191,15 @@ static struct snd_kcontrol_new stac_dmux_mixer = {
 	.put = stac92xx_dmux_enum_put,
 };
 
+static struct snd_kcontrol_new stac_smux_mixer = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "IEC958 Playback Source",
+	/* count set later */
+	.info = stac92xx_smux_enum_info,
+	.get = stac92xx_smux_enum_get,
+	.put = stac92xx_smux_enum_put,
+};
+
 static const char *slave_vols[] = {
 	"Front Playback Volume",
 	"Surround Playback Volume",
@@ -966,6 +1251,22 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 	}
+	if (spec->num_smuxes > 0) {
+		int wcaps = get_wcaps(codec, spec->multiout.dig_out_nid);
+		struct hda_input_mux *smux = &spec->private_smux;
+		/* check for mute support on SPDIF out */
+		if (wcaps & AC_WCAP_OUT_AMP) {
+			smux->items[smux->num_items].label = "Off";
+			smux->items[smux->num_items].index = 0;
+			smux->num_items++;
+			spec->spdif_mute = 1;
+		}
+		stac_smux_mixer.count = spec->num_smuxes;
+		err = snd_ctl_add(codec->bus->card,
+				  snd_ctl_new1(&stac_smux_mixer, codec));
+		if (err < 0)
+			return err;
+	}
 
 	if (spec->multiout.dig_out_nid) {
 		err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
@@ -977,7 +1278,7 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 			return err;
 		spec->multiout.share_spdif = 1;
 	}
-	if (spec->dig_in_nid) {
+	if (spec->dig_in_nid && (!spec->gpio_dir & 0x01)) {
 		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
 		if (err < 0)
 			return err;
@@ -1325,40 +1626,65 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 	{} /* terminator */
 };
 
-static unsigned int ref92hd71bxx_pin_configs[10] = {
+static unsigned int ref92hd83xxx_pin_configs[14] = {
+	0x02214030, 0x02211010, 0x02a19020, 0x02170130,
+	0x01014050, 0x01819040, 0x01014020, 0x90a3014e,
+	0x40f000f0, 0x40f000f0, 0x40f000f0, 0x40f000f0,
+	0x01451160, 0x98560170,
+};
+
+static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
+	[STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs,
+};
+
+static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
+	[STAC_92HD83XXX_REF] = "ref",
+};
+
+static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
+	/* SigmaTel reference board */
+	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
+		      "DFI LanParty", STAC_92HD71BXX_REF),
+};
+
+static unsigned int ref92hd71bxx_pin_configs[11] = {
 	0x02214030, 0x02a19040, 0x01a19020, 0x01014010,
 	0x0181302e, 0x01114010, 0x01019020, 0x90a000f0,
-	0x90a000f0, 0x01452050,
+	0x90a000f0, 0x01452050, 0x01452050,
 };
 
-static unsigned int dell_m4_1_pin_configs[10] = {
+static unsigned int dell_m4_1_pin_configs[11] = {
 	0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110,
 	0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0,
-	0x40f000f0, 0x4f0000f0,
+	0x40f000f0, 0x4f0000f0, 0x4f0000f0,
 };
 
-static unsigned int dell_m4_2_pin_configs[10] = {
+static unsigned int dell_m4_2_pin_configs[11] = {
 	0x0421101f, 0x04a11221, 0x90a70330, 0x90170110,
 	0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0,
-	0x40f000f0, 0x044413b0,
+	0x40f000f0, 0x044413b0, 0x044413b0,
 };
 
 static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = {
 	[STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs,
 	[STAC_DELL_M4_1]	= dell_m4_1_pin_configs,
 	[STAC_DELL_M4_2]	= dell_m4_2_pin_configs,
+	[STAC_HP_M4]		= NULL,
 };
 
 static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = {
 	[STAC_92HD71BXX_REF] = "ref",
 	[STAC_DELL_M4_1] = "dell-m4-1",
 	[STAC_DELL_M4_2] = "dell-m4-2",
+	[STAC_HP_M4] = "hp-m4",
 };
 
 static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_92HD71BXX_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a,
+				"unknown HP", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
 				"unknown Dell", STAC_DELL_M4_1),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234,
@@ -1477,6 +1803,11 @@ static unsigned int intel_mac_v5_pin_configs[10] = {
 	0x400000fc, 0x400000fb,
 };
 
+static unsigned int ecs202_pin_configs[10] = {
+	0x0221401f, 0x02a19020, 0x01a19020, 0x01114010,
+	0x408000f0, 0x01813022, 0x074510a0, 0x40c400f1,
+	0x9037012e, 0x40e000f2,
+};
 
 static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = {
 	[STAC_D945_REF] = ref922x_pin_configs,
@@ -1495,6 +1826,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = {
 	[STAC_MACBOOK_PRO_V2] = intel_mac_v3_pin_configs,
 	[STAC_IMAC_INTEL] = intel_mac_v2_pin_configs,
 	[STAC_IMAC_INTEL_20] = intel_mac_v3_pin_configs,
+	[STAC_ECS_202] = ecs202_pin_configs,
 	[STAC_922X_DELL_D81] = dell_922x_d81_pin_configs,
 	[STAC_922X_DELL_D82] = dell_922x_d82_pin_configs,	
 	[STAC_922X_DELL_M81] = dell_922x_m81_pin_configs,
@@ -1518,6 +1850,7 @@ static const char *stac922x_models[STAC_922X_MODELS] = {
 	[STAC_MACBOOK_PRO_V2]	= "macbook-pro",
 	[STAC_IMAC_INTEL] = "imac-intel",
 	[STAC_IMAC_INTEL_20] = "imac-intel-20",
+	[STAC_ECS_202] = "ecs202",
 	[STAC_922X_DELL_D81] = "dell-d81",
 	[STAC_922X_DELL_D82] = "dell-d82",
 	[STAC_922X_DELL_M81] = "dell-m81",
@@ -1604,6 +1937,33 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = {
 		      "unknown Dell", STAC_922X_DELL_D81),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d7,
 		      "Dell XPS M1210", STAC_922X_DELL_M82),
+	/* ECS/PC Chips boards */
+	SND_PCI_QUIRK(0x1019, 0x2144,
+		      "ECS/PC chips", STAC_ECS_202),
+	SND_PCI_QUIRK(0x1019, 0x2608,
+		      "ECS/PC chips", STAC_ECS_202),
+	SND_PCI_QUIRK(0x1019, 0x2633,
+		      "ECS/PC chips P17G/1333", STAC_ECS_202),
+	SND_PCI_QUIRK(0x1019, 0x2811,
+		      "ECS/PC chips", STAC_ECS_202),
+	SND_PCI_QUIRK(0x1019, 0x2812,
+		      "ECS/PC chips", STAC_ECS_202),
+	SND_PCI_QUIRK(0x1019, 0x2813,
+		      "ECS/PC chips", STAC_ECS_202),
+	SND_PCI_QUIRK(0x1019, 0x2814,
+		      "ECS/PC chips", STAC_ECS_202),
+	SND_PCI_QUIRK(0x1019, 0x2815,
+		      "ECS/PC chips", STAC_ECS_202),
+	SND_PCI_QUIRK(0x1019, 0x2816,
+		      "ECS/PC chips", STAC_ECS_202),
+	SND_PCI_QUIRK(0x1019, 0x2817,
+		      "ECS/PC chips", STAC_ECS_202),
+	SND_PCI_QUIRK(0x1019, 0x2818,
+		      "ECS/PC chips", STAC_ECS_202),
+	SND_PCI_QUIRK(0x1019, 0x2819,
+		      "ECS/PC chips", STAC_ECS_202),
+	SND_PCI_QUIRK(0x1019, 0x2820,
+		      "ECS/PC chips", STAC_ECS_202),
 	{} /* terminator */
 };
 
@@ -1683,8 +2043,8 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
 	/* Dell 3 stack systems with verb table in BIOS */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f3, "Dell Inspiron 1420", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x0227, "Dell Vostro 1400  ", STAC_DELL_BIOS),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x022f, "Dell     ", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x022e, "Dell     ", STAC_DELL_BIOS),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x022f, "Dell Inspiron 1525", STAC_DELL_3ST),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x0242, "Dell     ", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x0243, "Dell     ", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x02ff, "Dell     ", STAC_DELL_BIOS),
@@ -1867,6 +2227,8 @@ static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo,
 				      struct snd_pcm_substream *substream)
 {
 	struct sigmatel_spec *spec = codec->spec;
+	if (spec->stream_delay)
+		msleep(spec->stream_delay);
 	return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
 					     hinfo);
 }
@@ -1930,9 +2292,14 @@ static int stac92xx_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
 					struct snd_pcm_substream *substream)
 {
 	struct sigmatel_spec *spec = codec->spec;
+	hda_nid_t nid = spec->adc_nids[substream->number];
 
-	snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
-                                   stream_tag, 0, format);
+	if (spec->powerdown_adcs) {
+		msleep(40);
+		snd_hda_codec_write_cache(codec, nid, 0,
+			AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+	}
+	snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
 	return 0;
 }
 
@@ -1941,8 +2308,12 @@ static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 					struct snd_pcm_substream *substream)
 {
 	struct sigmatel_spec *spec = codec->spec;
+	hda_nid_t nid = spec->adc_nids[substream->number];
 
-	snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
+	snd_hda_codec_cleanup_stream(codec, nid);
+	if (spec->powerdown_adcs)
+		snd_hda_codec_write_cache(codec, nid, 0,
+			AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
 	return 0;
 }
 
@@ -2193,6 +2564,8 @@ enum {
 	STAC_CTL_WIDGET_VOL,
 	STAC_CTL_WIDGET_MUTE,
 	STAC_CTL_WIDGET_MONO_MUX,
+	STAC_CTL_WIDGET_AMP_MUX,
+	STAC_CTL_WIDGET_AMP_VOL,
 	STAC_CTL_WIDGET_HP_SWITCH,
 	STAC_CTL_WIDGET_IO_SWITCH,
 	STAC_CTL_WIDGET_CLFE_SWITCH
@@ -2202,13 +2575,16 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = {
 	HDA_CODEC_VOLUME(NULL, 0, 0, 0),
 	HDA_CODEC_MUTE(NULL, 0, 0, 0),
 	STAC_MONO_MUX,
+	STAC_AMP_MUX,
+	STAC_AMP_VOL(NULL, 0, 0, 0, 0),
 	STAC_CODEC_HP_SWITCH(NULL),
 	STAC_CODEC_IO_SWITCH(NULL, 0),
 	STAC_CODEC_CLFE_SWITCH(NULL, 0),
 };
 
 /* add dynamic controls */
-static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char *name, unsigned long val)
+static int stac92xx_add_control_idx(struct sigmatel_spec *spec, int type,
+		int idx, const char *name, unsigned long val)
 {
 	struct snd_kcontrol_new *knew;
 
@@ -2228,6 +2604,7 @@ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char
 
 	knew = &spec->kctl_alloc[spec->num_kctl_used];
 	*knew = stac92xx_control_templates[type];
+	knew->index = idx;
 	knew->name = kstrdup(name, GFP_KERNEL);
 	if (! knew->name)
 		return -ENOMEM;
@@ -2236,6 +2613,14 @@ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char
 	return 0;
 }
 
+
+/* add dynamic controls */
+static int stac92xx_add_control(struct sigmatel_spec *spec, int type,
+		const char *name, unsigned long val)
+{
+	return stac92xx_add_control_idx(spec, type, 0, name, val);
+}
+
 /* flag inputs as additional dynamic lineouts */
 static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cfg *cfg)
 {
@@ -2467,6 +2852,10 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
 		}
 	}
 
+	if ((spec->multiout.num_dacs - cfg->line_outs) > 0 &&
+			cfg->hp_outs && !spec->multiout.hp_nid)
+		spec->multiout.hp_nid = nid;
+
 	if (cfg->hp_outs > 1) {
 		err = stac92xx_add_control(spec,
 			STAC_CTL_WIDGET_HP_SWITCH,
@@ -2579,8 +2968,8 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec,
 }
 
 /* labels for mono mux outputs */
-static const char *stac92xx_mono_labels[3] = {
-	"DAC0", "DAC1", "Mixer"
+static const char *stac92xx_mono_labels[4] = {
+	"DAC0", "DAC1", "Mixer", "DAC2"
 };
 
 /* create mono mux for mono out on capable codecs */
@@ -2609,6 +2998,116 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec)
 				"Mono Mux", spec->mono_nid);
 }
 
+/* labels for amp mux outputs */
+static const char *stac92xx_amp_labels[3] = {
+	"Front Microphone", "Microphone", "Line In"
+};
+
+/* create amp out controls mux on capable codecs */
+static int stac92xx_auto_create_amp_output_ctls(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	struct hda_input_mux *amp_mux = &spec->private_amp_mux;
+	int i, err;
+
+	for (i = 0; i < spec->num_amps; i++) {
+		amp_mux->items[amp_mux->num_items].label =
+					stac92xx_amp_labels[i];
+		amp_mux->items[amp_mux->num_items].index = i;
+		amp_mux->num_items++;
+	}
+
+	if (spec->num_amps > 1) {
+		err = stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_MUX,
+			"Amp Selector Capture Switch", 0);
+		if (err < 0)
+			return err;
+	}
+	return stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_VOL,
+		"Amp Capture Volume",
+		HDA_COMPOSE_AMP_VAL(spec->amp_nids[0], 3, 0, HDA_INPUT));
+}
+
+
+/* create PC beep volume controls */
+static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec,
+						hda_nid_t nid)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	u32 caps = query_amp_caps(codec, nid, HDA_OUTPUT);
+	int err;
+
+	/* check for mute support for the the amp */
+	if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) {
+		err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE,
+			"PC Beep Playback Switch",
+			HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT));
+			if (err < 0)
+				return err;
+	}
+
+	/* check to see if there is volume support for the amp */
+	if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) {
+		err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL,
+			"PC Beep Playback Volume",
+			HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT));
+			if (err < 0)
+				return err;
+	}
+	return 0;
+}
+
+static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	int wcaps, nid, i, err = 0;
+
+	for (i = 0; i < spec->num_muxes; i++) {
+		nid = spec->mux_nids[i];
+		wcaps = get_wcaps(codec, nid);
+
+		if (wcaps & AC_WCAP_OUT_AMP) {
+			err = stac92xx_add_control_idx(spec,
+				STAC_CTL_WIDGET_VOL, i, "Mux Capture Volume",
+				HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
+			if (err < 0)
+				return err;
+		}
+	}
+	return 0;
+};
+
+static const char *stac92xx_spdif_labels[3] = {
+	"Digital Playback", "Analog Mux 1", "Analog Mux 2",
+};
+
+static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	struct hda_input_mux *spdif_mux = &spec->private_smux;
+	const char **labels = spec->spdif_labels;
+	int i, num_cons;
+	hda_nid_t con_lst[HDA_MAX_NUM_INPUTS];
+
+	num_cons = snd_hda_get_connections(codec,
+				spec->smux_nids[0],
+				con_lst,
+				HDA_MAX_NUM_INPUTS);
+	if (!num_cons)
+		return -EINVAL;
+
+	if (!labels)
+		labels = stac92xx_spdif_labels;
+
+	for (i = 0; i < num_cons; i++) {
+		spdif_mux->items[spdif_mux->num_items].label = labels[i];
+		spdif_mux->items[spdif_mux->num_items].index = i;
+		spdif_mux->num_items++;
+	}
+
+	return 0;
+}
+
 /* labels for dmic mux inputs */
 static const char *stac92xx_dmic_labels[5] = {
 	"Analog Inputs", "Digital Mic 1", "Digital Mic 2",
@@ -2656,16 +3155,19 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 			}
 		continue;
 found:
-		wcaps = get_wcaps(codec, nid);
+		wcaps = get_wcaps(codec, nid) &
+			(AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
 
-		if (wcaps & AC_WCAP_OUT_AMP) {
+		if (wcaps) {
 			sprintf(name, "%s Capture Volume",
 				stac92xx_dmic_labels[dimux->num_items]);
 
 			err = stac92xx_add_control(spec,
 				STAC_CTL_WIDGET_VOL,
 				name,
-				HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
+				HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+				(wcaps & AC_WCAP_OUT_AMP) ?
+				HDA_OUTPUT : HDA_INPUT));
 			if (err < 0)
 				return err;
 		}
@@ -2789,8 +3291,8 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		hp_speaker_swap = 1;
 	}
 	if (spec->autocfg.mono_out_pin) {
-		int dir = (get_wcaps(codec, spec->autocfg.mono_out_pin)
-				& AC_WCAP_OUT_AMP) ? HDA_OUTPUT : HDA_INPUT;
+		int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) &
+			(AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
 		u32 caps = query_amp_caps(codec,
 				spec->autocfg.mono_out_pin, dir);
 		hda_nid_t conn_list[1];
@@ -2812,21 +3314,26 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 						!(wcaps & AC_WCAP_LR_SWAP))
 					spec->mono_nid = conn_list[0];
 		}
-		/* all mono outs have a least a mute/unmute switch */
-		err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE,
-			"Mono Playback Switch",
-			HDA_COMPOSE_AMP_VAL(spec->autocfg.mono_out_pin,
-					1, 0, dir));
-		if (err < 0)
-			return err;
-		/* check to see if there is volume support for the amp */
-		if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) {
-			err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL,
-				"Mono Playback Volume",
-				HDA_COMPOSE_AMP_VAL(spec->autocfg.mono_out_pin,
-					1, 0, dir));
+		if (dir) {
+			hda_nid_t nid = spec->autocfg.mono_out_pin;
+
+			/* most mono outs have a least a mute/unmute switch */
+			dir = (dir & AC_WCAP_OUT_AMP) ? HDA_OUTPUT : HDA_INPUT;
+			err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE,
+				"Mono Playback Switch",
+				HDA_COMPOSE_AMP_VAL(nid, 1, 0, dir));
 			if (err < 0)
 				return err;
+			/* check for volume support for the amp */
+			if ((caps & AC_AMPCAP_NUM_STEPS)
+					>> AC_AMPCAP_NUM_STEPS_SHIFT) {
+				err = stac92xx_add_control(spec,
+					STAC_CTL_WIDGET_VOL,
+					"Mono Playback Volume",
+				HDA_COMPOSE_AMP_VAL(nid, 1, 0, dir));
+				if (err < 0)
+					return err;
+			}
 		}
 
 		stac92xx_auto_set_pinctl(codec, spec->autocfg.mono_out_pin,
@@ -2844,6 +3351,28 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 	if (err < 0)
 		return err;
 
+	/* setup analog beep controls */
+	if (spec->anabeep_nid > 0) {
+		err = stac92xx_auto_create_beep_ctls(codec,
+			spec->anabeep_nid);
+		if (err < 0)
+			return err;
+	}
+
+	/* setup digital beep controls and input device */
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+	if (spec->digbeep_nid > 0) {
+		hda_nid_t nid = spec->digbeep_nid;
+
+		err = stac92xx_auto_create_beep_ctls(codec, nid);
+		if (err < 0)
+			return err;
+		err = snd_hda_attach_beep_device(codec, nid);
+		if (err < 0)
+			return err;
+	}
+#endif
+
 	if (hp_speaker_swap == 1) {
 		/* Restore the hp_outs and line_outs */
 		memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins,
@@ -2872,11 +3401,25 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		if (err < 0)
 			return err;
 	}
-
-	if (spec->num_dmics > 0)
+	if (spec->num_amps > 0) {
+		err = stac92xx_auto_create_amp_output_ctls(codec);
+		if (err < 0)
+			return err;
+	}
+	if (spec->num_dmics > 0 && !spec->dinput_mux)
 		if ((err = stac92xx_auto_create_dmic_input_ctls(codec,
 						&spec->autocfg)) < 0)
 			return err;
+	if (spec->num_muxes > 0) {
+		err = stac92xx_auto_create_mux_input_ctls(codec);
+		if (err < 0)
+			return err;
+	}
+	if (spec->num_smuxes > 0) {
+		err = stac92xx_auto_create_spdif_mux_ctls(codec);
+		if (err < 0)
+			return err;
+	}
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 	if (spec->multiout.max_channels > 2)
@@ -2884,17 +3427,17 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 
 	if (spec->autocfg.dig_out_pin)
 		spec->multiout.dig_out_nid = dig_out;
-	if (spec->autocfg.dig_in_pin)
+	if (dig_in && spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = dig_in;
 
 	if (spec->kctl_alloc)
 		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
 
 	spec->input_mux = &spec->private_imux;
-	if (!spec->dinput_mux)
-		spec->dinput_mux = &spec->private_dimux;
+	spec->dinput_mux = &spec->private_dimux;
+	spec->sinput_mux = &spec->private_smux;
 	spec->mono_mux = &spec->private_mono_mux;
-
+	spec->amp_mux = &spec->private_amp_mux;
 	return 1;
 }
 
@@ -3074,6 +3617,12 @@ static int stac92xx_init(struct hda_codec *codec)
 
 	snd_hda_sequence_write(codec, spec->init);
 
+	/* power down adcs initially */
+	if (spec->powerdown_adcs)
+		for (i = 0; i < spec->num_adcs; i++)
+			snd_hda_codec_write_cache(codec,
+				spec->adc_nids[i], 0,
+				AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
 	/* set up pins */
 	if (spec->hp_detect) {
 		/* Enable unsolicited responses on the HP widget */
@@ -3095,7 +3644,12 @@ static int stac92xx_init(struct hda_codec *codec)
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = cfg->input_pins[i];
 		if (nid) {
-			unsigned int pinctl = AC_PINCTL_IN_EN;
+			unsigned int pinctl = snd_hda_codec_read(codec, nid,
+				0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+			/* if PINCTL already set then skip */
+			if (pinctl & AC_PINCAP_IN)
+				continue;
+			pinctl = AC_PINCTL_IN_EN;
 			if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC)
 				pinctl |= stac92xx_get_vref(codec, nid);
 			stac92xx_auto_set_pinctl(codec, nid, pinctl);
@@ -3158,6 +3712,7 @@ static void stac92xx_free(struct hda_codec *codec)
 		kfree(spec->bios_pin_configs);
 
 	kfree(spec);
+	snd_hda_detach_beep_device(codec);
 }
 
 static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid,
@@ -3279,7 +3834,12 @@ static void stac92xx_pin_sense(struct hda_codec *codec, int idx)
 	val = snd_hda_codec_read(codec, codec->afg, 0, 0x0fec, 0x0)
 							& 0x000000ff;
 	presence = get_hp_pin_presence(codec, nid);
-	idx = 1 << idx;
+
+	/* several codecs have two power down bits */
+	if (spec->pwr_mapping)
+		idx = spec->pwr_mapping[idx];
+	else
+		idx = 1 << idx;
 
 	if (presence)
 		val &= ~idx;
@@ -3295,13 +3855,22 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
 	struct sigmatel_spec *spec = codec->spec;
 	int idx = res >> 26 & 0x0f;
 
-	switch ((res >> 26) & 0x30) {
+	switch ((res >> 26) & 0x70) {
 	case STAC_HP_EVENT:
 		stac92xx_hp_detect(codec, res);
 		/* fallthru */
 	case STAC_PWR_EVENT:
 		if (spec->num_pwrs > 0)
 			stac92xx_pin_sense(codec, idx);
+		break;
+	case STAC_VREF_EVENT: {
+		int data = snd_hda_codec_read(codec, codec->afg, 0,
+			AC_VERB_GET_GPIO_DATA, 0);
+		/* toggle VREF state based on GPIOx status */
+		snd_hda_codec_write(codec, codec->afg, 0, 0x7e0,
+			!!(data & (1 << idx)));
+		break;
+		}
 	}
 }
 
@@ -3478,9 +4047,9 @@ static struct hda_input_mux stac92hd73xx_dmux = {
 	.num_items = 4,
 	.items = {
 		{ "Analog Inputs", 0x0b },
-		{ "CD", 0x08 },
 		{ "Digital Mic 1", 0x09 },
 		{ "Digital Mic 2", 0x0a },
+		{ "CD", 0x08 },
 	}
 };
 
@@ -3495,6 +4064,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
 		return -ENOMEM;
 
 	codec->spec = spec;
+	codec->slave_dig_outs = stac92hd73xx_slave_dig_outs;
 	spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids);
 	spec->pin_nids = stac92hd73xx_pin_nids;
 	spec->board_config = snd_hda_check_board_config(codec,
@@ -3527,17 +4097,14 @@ again:
 
 	switch (spec->multiout.num_dacs) {
 	case 0x3: /* 6 Channel */
-		spec->multiout.hp_nid = 0x17;
 		spec->mixer = stac92hd73xx_6ch_mixer;
 		spec->init = stac92hd73xx_6ch_core_init;
 		break;
 	case 0x4: /* 8 Channel */
-		spec->multiout.hp_nid = 0x18;
 		spec->mixer = stac92hd73xx_8ch_mixer;
 		spec->init = stac92hd73xx_8ch_core_init;
 		break;
 	case 0x5: /* 10 Channel */
-		spec->multiout.hp_nid = 0x19;
 		spec->mixer = stac92hd73xx_10ch_mixer;
 		spec->init = stac92hd73xx_10ch_core_init;
 	};
@@ -3546,27 +4113,34 @@ again:
 	spec->aloopback_mask = 0x01;
 	spec->aloopback_shift = 8;
 
+	spec->digbeep_nid = 0x1c;
 	spec->mux_nids = stac92hd73xx_mux_nids;
 	spec->adc_nids = stac92hd73xx_adc_nids;
 	spec->dmic_nids = stac92hd73xx_dmic_nids;
 	spec->dmux_nids = stac92hd73xx_dmux_nids;
+	spec->smux_nids = stac92hd73xx_smux_nids;
+	spec->amp_nids = stac92hd73xx_amp_nids;
+	spec->num_amps = ARRAY_SIZE(stac92hd73xx_amp_nids);
 
 	spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids);
 	spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids);
 	spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids);
-	spec->dinput_mux = &stac92hd73xx_dmux;
-	/* GPIO0 High = Enable EAPD */
-	spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1;
-	spec->gpio_data = 0x01;
+	memcpy(&spec->private_dimux, &stac92hd73xx_dmux,
+			sizeof(stac92hd73xx_dmux));
 
 	switch (spec->board_config) {
 	case STAC_DELL_M6:
 		spec->init = dell_eq_core_init;
+		spec->num_smuxes = 0;
+		spec->mixer = &stac92hd73xx_6ch_mixer[DELL_M6_MIXER];
+		spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP];
+		spec->num_amps = 1;
 		switch (codec->subsystem_id) {
 		case 0x1028025e: /* Analog Mics */
 		case 0x1028025f:
 			stac92xx_set_config_reg(codec, 0x0b, 0x90A70170);
 			spec->num_dmics = 0;
+			spec->private_dimux.num_items = 1;
 			break;
 		case 0x10280271: /* Digital Mics */
 		case 0x10280272:
@@ -3576,23 +4150,32 @@ again:
 		case 0x10280255:
 			stac92xx_set_config_reg(codec, 0x13, 0x90A60160);
 			spec->num_dmics = 1;
+			spec->private_dimux.num_items = 2;
 			break;
 		case 0x10280256: /* Both */
 		case 0x10280057:
 			stac92xx_set_config_reg(codec, 0x0b, 0x90A70170);
 			stac92xx_set_config_reg(codec, 0x13, 0x90A60160);
 			spec->num_dmics = 1;
+			spec->private_dimux.num_items = 2;
 			break;
 		}
 		break;
 	default:
 		spec->num_dmics = STAC92HD73XX_NUM_DMICS;
+		spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids);
 	}
+	if (spec->board_config > STAC_92HD73XX_REF) {
+		/* GPIO0 High = Enable EAPD */
+		spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1;
+		spec->gpio_data = 0x01;
+	}
+	spec->dinput_mux = &spec->private_dimux;
 
 	spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids);
 	spec->pwr_nids = stac92hd73xx_pwr_nids;
 
-	err = stac92xx_parse_auto_config(codec, 0x22, 0x24);
+	err = stac92xx_parse_auto_config(codec, 0x25, 0x27);
 
 	if (!err) {
 		if (spec->board_config < 0) {
@@ -3614,6 +4197,136 @@ again:
 	return 0;
 }
 
+static struct hda_input_mux stac92hd83xxx_dmux = {
+	.num_items = 3,
+	.items = {
+		{ "Analog Inputs", 0x03 },
+		{ "Digital Mic 1", 0x04 },
+		{ "Digital Mic 2", 0x05 },
+	}
+};
+
+static int patch_stac92hd83xxx(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec;
+	int err;
+
+	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL)
+		return -ENOMEM;
+
+	codec->spec = spec;
+	codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs;
+	spec->mono_nid = 0x19;
+	spec->digbeep_nid = 0x21;
+	spec->dmic_nids = stac92hd83xxx_dmic_nids;
+	spec->dmux_nids = stac92hd83xxx_dmux_nids;
+	spec->adc_nids = stac92hd83xxx_adc_nids;
+	spec->pwr_nids = stac92hd83xxx_pwr_nids;
+	spec->pwr_mapping = stac92hd83xxx_pwr_mapping;
+	spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids);
+	spec->multiout.dac_nids = stac92hd83xxx_dac_nids;
+
+	spec->init = stac92hd83xxx_core_init;
+	switch (codec->vendor_id) {
+	case 0x111d7605:
+		spec->multiout.num_dacs = STAC92HD81_DAC_COUNT;
+		break;
+	default:
+		spec->num_pwrs--;
+		spec->init++; /* switch to config #2 */
+		spec->multiout.num_dacs = STAC92HD83_DAC_COUNT;
+	}
+
+	spec->mixer = stac92hd83xxx_mixer;
+	spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids);
+	spec->num_dmuxes = ARRAY_SIZE(stac92hd83xxx_dmux_nids);
+	spec->num_adcs = ARRAY_SIZE(stac92hd83xxx_adc_nids);
+	spec->num_dmics = STAC92HD83XXX_NUM_DMICS;
+	spec->dinput_mux = &stac92hd83xxx_dmux;
+	spec->pin_nids = stac92hd83xxx_pin_nids;
+	spec->board_config = snd_hda_check_board_config(codec,
+							STAC_92HD83XXX_MODELS,
+							stac92hd83xxx_models,
+							stac92hd83xxx_cfg_tbl);
+again:
+	if (spec->board_config < 0) {
+		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
+			" STAC92HD83XXX, using BIOS defaults\n");
+		err = stac92xx_save_bios_config_regs(codec);
+		if (err < 0) {
+			stac92xx_free(codec);
+			return err;
+		}
+		spec->pin_configs = spec->bios_pin_configs;
+	} else {
+		spec->pin_configs = stac92hd83xxx_brd_tbl[spec->board_config];
+		stac92xx_set_config_regs(codec);
+	}
+
+	err = stac92xx_parse_auto_config(codec, 0x1d, 0);
+	if (!err) {
+		if (spec->board_config < 0) {
+			printk(KERN_WARNING "hda_codec: No auto-config is "
+			       "available, default to model=ref\n");
+			spec->board_config = STAC_92HD83XXX_REF;
+			goto again;
+		}
+		err = -EINVAL;
+	}
+
+	if (err < 0) {
+		stac92xx_free(codec);
+		return err;
+	}
+
+	codec->patch_ops = stac92xx_patch_ops;
+
+	return 0;
+}
+
+#ifdef SND_HDA_NEEDS_RESUME
+static void stac92hd71xx_set_power_state(struct hda_codec *codec, int pwr)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	int i;
+	snd_hda_codec_write_cache(codec, codec->afg, 0,
+		AC_VERB_SET_POWER_STATE, pwr);
+
+	msleep(1);
+	for (i = 0; i < spec->num_adcs; i++) {
+		snd_hda_codec_write_cache(codec,
+			spec->adc_nids[i], 0,
+			AC_VERB_SET_POWER_STATE, pwr);
+	}
+};
+
+static int stac92hd71xx_resume(struct hda_codec *codec)
+{
+	stac92hd71xx_set_power_state(codec, AC_PWRST_D0);
+	return stac92xx_resume(codec);
+}
+
+static int stac92hd71xx_suspend(struct hda_codec *codec, pm_message_t state)
+{
+	stac92hd71xx_set_power_state(codec, AC_PWRST_D3);
+	return 0;
+};
+
+#endif
+
+static struct hda_codec_ops stac92hd71bxx_patch_ops = {
+	.build_controls = stac92xx_build_controls,
+	.build_pcms = stac92xx_build_pcms,
+	.init = stac92xx_init,
+	.free = stac92xx_free,
+	.unsol_event = stac92xx_unsol_event,
+#ifdef SND_HDA_NEEDS_RESUME
+	.resume = stac92hd71xx_resume,
+	.suspend = stac92hd71xx_suspend,
+#endif
+};
+
 static int patch_stac92hd71bxx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
@@ -3624,6 +4337,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 		return -ENOMEM;
 
 	codec->spec = spec;
+	codec->patch_ops = stac92xx_patch_ops;
 	spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids);
 	spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids);
 	spec->pin_nids = stac92hd71bxx_pin_nids;
@@ -3653,8 +4367,28 @@ again:
 	case 0x111d76b5:
 		spec->mixer = stac92hd71bxx_mixer;
 		spec->init = stac92hd71bxx_core_init;
+		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 		break;
 	case 0x111d7608: /* 5 Port with Analog Mixer */
+		switch (codec->subsystem_id) {
+		case 0x103c361a:
+			/* Enable VREF power saving on GPIO1 detect */
+			snd_hda_codec_write(codec, codec->afg, 0,
+				AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02);
+			snd_hda_codec_write_cache(codec, codec->afg, 0,
+					AC_VERB_SET_UNSOLICITED_ENABLE,
+					(AC_USRSP_EN | STAC_VREF_EVENT | 0x01));
+			spec->gpio_mask |= 0x02;
+			break;
+		}
+		if ((codec->revision_id & 0xf) == 0 ||
+				(codec->revision_id & 0xf) == 1) {
+#ifdef SND_HDA_NEEDS_RESUME
+			codec->patch_ops = stac92hd71bxx_patch_ops;
+#endif
+			spec->stream_delay = 40; /* 40 milliseconds */
+		}
+
 		/* no output amps */
 		spec->num_pwrs = 0;
 		spec->mixer = stac92hd71bxx_analog_mixer;
@@ -3664,32 +4398,60 @@ again:
 		stac92xx_set_config_reg(codec, 0xf, 0x40f000f0);
 		break;
 	case 0x111d7603: /* 6 Port with Analog Mixer */
+		if ((codec->revision_id & 0xf) == 1) {
+#ifdef SND_HDA_NEEDS_RESUME
+			codec->patch_ops = stac92hd71bxx_patch_ops;
+#endif
+			spec->stream_delay = 40; /* 40 milliseconds */
+		}
+
 		/* no output amps */
 		spec->num_pwrs = 0;
 		/* fallthru */
 	default:
 		spec->mixer = stac92hd71bxx_analog_mixer;
 		spec->init = stac92hd71bxx_analog_core_init;
+		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 	}
 
 	spec->aloopback_mask = 0x20;
 	spec->aloopback_shift = 0;
 
-	/* GPIO0 High = EAPD */
-	spec->gpio_mask = 0x01;
-	spec->gpio_dir = 0x01;
-	spec->gpio_data = 0x01;
+	if (spec->board_config > STAC_92HD71BXX_REF) {
+		/* GPIO0 = EAPD */
+		spec->gpio_mask = 0x01;
+		spec->gpio_dir = 0x01;
+		spec->gpio_data = 0x01;
+	}
 
+	spec->powerdown_adcs = 1;
+	spec->digbeep_nid = 0x26;
 	spec->mux_nids = stac92hd71bxx_mux_nids;
 	spec->adc_nids = stac92hd71bxx_adc_nids;
 	spec->dmic_nids = stac92hd71bxx_dmic_nids;
 	spec->dmux_nids = stac92hd71bxx_dmux_nids;
+	spec->smux_nids = stac92hd71bxx_smux_nids;
 	spec->pwr_nids = stac92hd71bxx_pwr_nids;
 
 	spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids);
 	spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids);
-	spec->num_dmics = STAC92HD71BXX_NUM_DMICS;
-	spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
+
+	switch (spec->board_config) {
+	case STAC_HP_M4:
+		spec->num_dmics = 0;
+		spec->num_smuxes = 0;
+		spec->num_dmuxes = 0;
+
+		/* enable internal microphone */
+		stac92xx_set_config_reg(codec, 0x0e, 0x01813040);
+		stac92xx_auto_set_pinctl(codec, 0x0e,
+			AC_PINCTL_IN_EN | AC_PINCTL_VREF_80);
+		break;
+	default:
+		spec->num_dmics = STAC92HD71BXX_NUM_DMICS;
+		spec->num_smuxes = ARRAY_SIZE(stac92hd71bxx_smux_nids);
+		spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
+	};
 
 	spec->multiout.num_dacs = 1;
 	spec->multiout.hp_nid = 0x11;
@@ -3711,8 +4473,6 @@ again:
 		return err;
 	}
 
-	codec->patch_ops = stac92xx_patch_ops;
-
 	return 0;
 };
 
@@ -3854,10 +4614,14 @@ static int patch_stac927x(struct hda_codec *codec)
 		stac92xx_set_config_regs(codec);
 	}
 
+	spec->digbeep_nid = 0x23;
 	spec->adc_nids = stac927x_adc_nids;
 	spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids);
 	spec->mux_nids = stac927x_mux_nids;
 	spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids);
+	spec->smux_nids = stac927x_smux_nids;
+	spec->num_smuxes = ARRAY_SIZE(stac927x_smux_nids);
+	spec->spdif_labels = stac927x_spdif_labels;
 	spec->dac_list = stac927x_dac_nids;
 	spec->multiout.dac_nids = spec->dac_nids;
 
@@ -3900,9 +4664,11 @@ static int patch_stac927x(struct hda_codec *codec)
 		spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids);
 		break;
 	default:
-		/* GPIO0 High = Enable EAPD */
-		spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1;
-		spec->gpio_data = 0x01;
+		if (spec->board_config > STAC_D965_REF) {
+			/* GPIO0 High = Enable EAPD */
+			spec->eapd_mask = spec->gpio_mask = 0x01;
+			spec->gpio_dir = spec->gpio_data = 0x01;
+		}
 		spec->num_dmics = 0;
 
 		spec->init = stac927x_core_init;
@@ -3974,10 +4740,13 @@ static int patch_stac9205(struct hda_codec *codec)
 		stac92xx_set_config_regs(codec);
 	}
 
+	spec->digbeep_nid = 0x23;
 	spec->adc_nids = stac9205_adc_nids;
 	spec->num_adcs = ARRAY_SIZE(stac9205_adc_nids);
 	spec->mux_nids = stac9205_mux_nids;
 	spec->num_muxes = ARRAY_SIZE(stac9205_mux_nids);
+	spec->smux_nids = stac9205_smux_nids;
+	spec->num_smuxes = ARRAY_SIZE(stac9205_smux_nids);
 	spec->dmic_nids = stac9205_dmic_nids;
 	spec->num_dmics = STAC9205_NUM_DMICS;
 	spec->dmux_nids = stac9205_dmux_nids;
@@ -4013,6 +4782,9 @@ static int patch_stac9205(struct hda_codec *codec)
 		 */
 		spec->gpio_data = 0x01;
 		break;
+	case STAC_9205_REF:
+		/* SPDIF-In enabled */
+		break;
 	default:
 		/* GPIO0 High = EAPD */
 		spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1;
@@ -4332,6 +5104,8 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = {
  	{ .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 },
  	{ .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 },
 	{ .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx},
+	{ .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx},
+	{ .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx},
 	{ .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx },
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index e7e43524f8c7..63e4871e5d8f 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1,10 +1,10 @@
 /*
  * Universal Interface for Intel High Definition Audio Codec
  *
- * HD audio interface patch for VIA VT1708 codec
+ * HD audio interface patch for VIA VT1702/VT1708/VT1709 codec
  *
- * Copyright (c) 2006 Lydia Wang <lydiawang@viatech.com>
- *                    Takashi Iwai <tiwai@suse.de>
+ * Copyright (c) 2006-2008 Lydia Wang <lydiawang@viatech.com>
+ *			   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
@@ -29,6 +29,13 @@
 /* 2006-09-08  Lydia Wang  Fix internal loopback recording source select bug */
 /* 2007-09-12  Lydia Wang  Add EAPD enable during driver initialization      */
 /* 2007-09-17  Lydia Wang  Add VT1708B codec support                        */
+/* 2007-11-14  Lydia Wang  Add VT1708A codec HP and CD pin connect config    */
+/* 2008-02-03  Lydia Wang  Fix Rear channels and Back channels inverse issue */
+/* 2008-03-06  Lydia Wang  Add VT1702 codec and VT1708S codec support        */
+/* 2008-04-09  Lydia Wang  Add mute front speaker when HP plugin             */
+/* 2008-04-09  Lydia Wang  Add Independent HP feature                        */
+/* 2008-05-28  Lydia Wang  Add second S/PDIF Out support for VT1702	     */
+/* 2008-09-15  Logan Li    Add VT1708S Mic Boost workaround/backdoor	     */
 /*                                                                           */
 /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */
 
@@ -37,6 +44,7 @@
 #include <linux/delay.h>
 #include <linux/slab.h>
 #include <sound/core.h>
+#include <sound/asoundef.h>
 #include "hda_codec.h"
 #include "hda_local.h"
 #include "hda_patch.h"
@@ -53,6 +61,8 @@
 #define VT1708_DIGOUT_NID	0x14
 #define VT1708_DIGIN_NID	0x16
 #define VT1708_DIGIN_PIN	0x26
+#define VT1708_HP_PIN_NID	0x20
+#define VT1708_CD_PIN_NID	0x24
 
 #define VT1709_HP_DAC_NID	0x28
 #define VT1709_DIGOUT_NID	0x13
@@ -64,12 +74,64 @@
 #define VT1708B_DIGIN_NID	0x15
 #define VT1708B_DIGIN_PIN	0x21
 
+#define VT1708S_HP_NID		0x25
+#define VT1708S_DIGOUT_NID	0x12
+
+#define VT1702_HP_NID		0x17
+#define VT1702_DIGOUT_NID	0x11
+
 #define IS_VT1708_VENDORID(x)		((x) >= 0x11061708 && (x) <= 0x1106170b)
 #define IS_VT1709_10CH_VENDORID(x)	((x) >= 0x1106e710 && (x) <= 0x1106e713)
 #define IS_VT1709_6CH_VENDORID(x)	((x) >= 0x1106e714 && (x) <= 0x1106e717)
 #define IS_VT1708B_8CH_VENDORID(x)	((x) >= 0x1106e720 && (x) <= 0x1106e723)
 #define IS_VT1708B_4CH_VENDORID(x)	((x) >= 0x1106e724 && (x) <= 0x1106e727)
+#define IS_VT1708S_VENDORID(x)		((x) >= 0x11060397 && (x) <= 0x11067397)
+#define IS_VT1702_VENDORID(x)		((x) >= 0x11060398 && (x) <= 0x11067398)
+
+enum VIA_HDA_CODEC {
+	UNKNOWN = -1,
+	VT1708,
+	VT1709_10CH,
+	VT1709_6CH,
+	VT1708B_8CH,
+	VT1708B_4CH,
+	VT1708S,
+	VT1702,
+	CODEC_TYPES,
+};
+
+static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id)
+{
+	u16 ven_id = vendor_id >> 16;
+	u16 dev_id = vendor_id & 0xffff;
+	enum VIA_HDA_CODEC codec_type;
+
+	/* get codec type */
+	if (ven_id != 0x1106)
+		codec_type = UNKNOWN;
+	else if (dev_id >= 0x1708 && dev_id <= 0x170b)
+		codec_type = VT1708;
+	else if (dev_id >= 0xe710 && dev_id <= 0xe713)
+		codec_type = VT1709_10CH;
+	else if (dev_id >= 0xe714 && dev_id <= 0xe717)
+		codec_type = VT1709_6CH;
+	else if (dev_id >= 0xe720 && dev_id <= 0xe723)
+		codec_type = VT1708B_8CH;
+	else if (dev_id >= 0xe724 && dev_id <= 0xe727)
+		codec_type = VT1708B_4CH;
+	else if ((dev_id & 0xfff) == 0x397
+		 && (dev_id >> 12) < 8)
+		codec_type = VT1708S;
+	else if ((dev_id & 0xfff) == 0x398
+		 && (dev_id >> 12) < 8)
+		codec_type = VT1702;
+	else
+		codec_type = UNKNOWN;
+	return codec_type;
+};
 
+#define VIA_HP_EVENT		0x01
+#define VIA_GPIO_EVENT		0x02
 
 enum {
 	VIA_CTL_WIDGET_VOL,
@@ -77,12 +139,54 @@ enum {
 };
 
 enum {
-	AUTO_SEQ_FRONT,
+	AUTO_SEQ_FRONT = 0,
 	AUTO_SEQ_SURROUND,
 	AUTO_SEQ_CENLFE,
 	AUTO_SEQ_SIDE
 };
 
+#define get_amp_nid(kc)	((kc)->private_value & 0xffff)
+
+/* Some VT1708S based boards gets the micboost setting wrong, so we have
+ * to apply some brute-force and re-write the TLV's by software. */
+static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+			 unsigned int size, unsigned int __user *_tlv)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = get_amp_nid(kcontrol);
+
+	if (get_codec_type(codec->vendor_id) == VT1708S
+	    && (nid == 0x1a || nid == 0x1e)) {
+		if (size < 4 * sizeof(unsigned int))
+			return -ENOMEM;
+		if (put_user(1, _tlv))	/* SNDRV_CTL_TLVT_DB_SCALE */
+			return -EFAULT;
+		if (put_user(2 * sizeof(unsigned int), _tlv + 1))
+			return -EFAULT;
+		if (put_user(0, _tlv + 2)) /* offset = 0 */
+			return -EFAULT;
+		if (put_user(1000, _tlv + 3)) /* step size = 10 dB */
+			return -EFAULT;
+	}
+	return 0;
+}
+
+static int mic_boost_volume_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = get_amp_nid(kcontrol);
+
+	if (get_codec_type(codec->vendor_id) == VT1708S
+	    && (nid == 0x1a || nid == 0x1e)) {
+		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+		uinfo->count = 2;
+		uinfo->value.integer.min = 0;
+		uinfo->value.integer.max = 3;
+	}
+	return 0;
+}
+
 static struct snd_kcontrol_new vt1708_control_templates[] = {
 	HDA_CODEC_VOLUME(NULL, 0, 0, 0),
 	HDA_CODEC_MUTE(NULL, 0, 0, 0),
@@ -94,7 +198,8 @@ struct via_spec {
 	struct snd_kcontrol_new *mixers[3];
 	unsigned int num_mixers;
 
-	struct hda_verb *init_verbs;
+	struct hda_verb *init_verbs[5];
+	unsigned int num_iverbs;
 
 	char *stream_name_analog;
 	struct hda_pcm_stream *stream_analog_playback;
@@ -106,6 +211,7 @@ struct via_spec {
 
 	/* playback */
 	struct hda_multi_out multiout;
+	hda_nid_t extra_dig_out_nid;
 
 	/* capture */
 	unsigned int num_adc_nids;
@@ -117,15 +223,19 @@ struct via_spec {
 	unsigned int cur_mux[3];
 
 	/* PCM information */
-	struct hda_pcm pcm_rec[2];
+	struct hda_pcm pcm_rec[3];
 
 	/* dynamic controls, init_verbs and input_mux */
 	struct auto_pin_cfg autocfg;
 	unsigned int num_kctl_alloc, num_kctl_used;
 	struct snd_kcontrol_new *kctl_alloc;
-	struct hda_input_mux private_imux;
+	struct hda_input_mux private_imux[2];
 	hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
 
+	/* HP mode source */
+	const struct hda_input_mux *hp_mux;
+	unsigned int hp_independent_mode;
+
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	struct hda_loopback_check loopback;
 #endif
@@ -146,6 +256,16 @@ static hda_nid_t vt1708B_adc_nids[2] = {
 	0x13, 0x14
 };
 
+static hda_nid_t vt1708S_adc_nids[2] = {
+	/* ADC1-2 */
+	0x13, 0x14
+};
+
+static hda_nid_t vt1702_adc_nids[3] = {
+	/* ADC1-2 */
+	0x12, 0x20, 0x1F
+};
+
 /* add dynamic controls */
 static int via_add_control(struct via_spec *spec, int type, const char *name,
 			   unsigned long val)
@@ -283,19 +403,108 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol,
 		return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
 					     0x18, &spec->cur_mux[adc_idx]);
 	else if ((IS_VT1709_10CH_VENDORID(vendor_id) ||
-		  IS_VT1709_6CH_VENDORID(vendor_id)) && adc_idx == 0)
+		  IS_VT1709_6CH_VENDORID(vendor_id)) && (adc_idx == 0))
 		return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
 					     0x19, &spec->cur_mux[adc_idx]);
 	else if ((IS_VT1708B_8CH_VENDORID(vendor_id) ||
-		  IS_VT1708B_4CH_VENDORID(vendor_id)) && adc_idx == 0)
+		  IS_VT1708B_4CH_VENDORID(vendor_id)) && (adc_idx == 0))
 		return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
 					     0x17, &spec->cur_mux[adc_idx]);
+	else if (IS_VT1702_VENDORID(vendor_id) && (adc_idx == 0))
+		return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
+					     0x13, &spec->cur_mux[adc_idx]);
 	else
 		return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
 					     spec->adc_nids[adc_idx],
 					     &spec->cur_mux[adc_idx]);
 }
 
+static int via_independent_hp_info(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_info *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct via_spec *spec = codec->spec;
+	return snd_hda_input_mux_info(spec->hp_mux, uinfo);
+}
+
+static int via_independent_hp_get(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct via_spec *spec = codec->spec;
+	hda_nid_t nid = spec->autocfg.hp_pins[0];
+	unsigned int pinsel = snd_hda_codec_read(codec, nid, 0,
+						 AC_VERB_GET_CONNECT_SEL,
+						 0x00);
+
+	ucontrol->value.enumerated.item[0] = pinsel;
+
+	return 0;
+}
+
+static int via_independent_hp_put(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct via_spec *spec = codec->spec;
+	hda_nid_t nid = spec->autocfg.hp_pins[0];
+	unsigned int pinsel = ucontrol->value.enumerated.item[0];
+	unsigned int con_nid = snd_hda_codec_read(codec, nid, 0,
+					 AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
+
+	if (con_nid == spec->multiout.hp_nid) {
+		if (pinsel == 0) {
+			if (!spec->hp_independent_mode) {
+				if (spec->multiout.num_dacs > 1)
+					spec->multiout.num_dacs -= 1;
+				spec->hp_independent_mode = 1;
+			}
+		} else if (pinsel == 1) {
+		       if (spec->hp_independent_mode) {
+				if (spec->multiout.num_dacs > 1)
+					spec->multiout.num_dacs += 1;
+				spec->hp_independent_mode = 0;
+		       }
+		}
+	} else {
+		if (pinsel == 0) {
+			if (spec->hp_independent_mode) {
+				if (spec->multiout.num_dacs > 1)
+					spec->multiout.num_dacs += 1;
+				spec->hp_independent_mode = 0;
+			}
+		} else if (pinsel == 1) {
+		       if (!spec->hp_independent_mode) {
+				if (spec->multiout.num_dacs > 1)
+					spec->multiout.num_dacs -= 1;
+				spec->hp_independent_mode = 1;
+		       }
+		}
+	}
+	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL,
+			    pinsel);
+
+	if (spec->multiout.hp_nid &&
+	    spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT])
+			snd_hda_codec_setup_stream(codec,
+						   spec->multiout.hp_nid,
+						   0, 0, 0);
+
+	return 0;
+}
+
+static struct snd_kcontrol_new via_hp_mixer[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Independent HP",
+		.count = 1,
+		.info = via_independent_hp_info,
+		.get = via_independent_hp_get,
+		.put = via_independent_hp_put,
+	},
+	{ } /* end */
+};
+
 /* capture mixer elements */
 static struct snd_kcontrol_new vt1708_capture_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT),
@@ -380,6 +589,138 @@ static int via_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
 	return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
 }
 
+
+static void playback_multi_pcm_prep_0(struct hda_codec *codec,
+				      unsigned int stream_tag,
+				      unsigned int format,
+				      struct snd_pcm_substream *substream)
+{
+	struct via_spec *spec = codec->spec;
+	struct hda_multi_out *mout = &spec->multiout;
+	hda_nid_t *nids = mout->dac_nids;
+	int chs = substream->runtime->channels;
+	int i;
+
+	mutex_lock(&codec->spdif_mutex);
+	if (mout->dig_out_nid && mout->dig_out_used != HDA_DIG_EXCLUSIVE) {
+		if (chs == 2 &&
+		    snd_hda_is_supported_format(codec, mout->dig_out_nid,
+						format) &&
+		    !(codec->spdif_status & IEC958_AES0_NONAUDIO)) {
+			mout->dig_out_used = HDA_DIG_ANALOG_DUP;
+			/* turn off SPDIF once; otherwise the IEC958 bits won't
+			 * be updated */
+			if (codec->spdif_ctls & AC_DIG1_ENABLE)
+				snd_hda_codec_write(codec, mout->dig_out_nid, 0,
+						    AC_VERB_SET_DIGI_CONVERT_1,
+						    codec->spdif_ctls &
+							~AC_DIG1_ENABLE & 0xff);
+			snd_hda_codec_setup_stream(codec, mout->dig_out_nid,
+						   stream_tag, 0, format);
+			/* turn on again (if needed) */
+			if (codec->spdif_ctls & AC_DIG1_ENABLE)
+				snd_hda_codec_write(codec, mout->dig_out_nid, 0,
+						    AC_VERB_SET_DIGI_CONVERT_1,
+						    codec->spdif_ctls & 0xff);
+		} else {
+			mout->dig_out_used = 0;
+			snd_hda_codec_setup_stream(codec, mout->dig_out_nid,
+						   0, 0, 0);
+		}
+	}
+	mutex_unlock(&codec->spdif_mutex);
+
+	/* front */
+	snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag,
+				   0, format);
+
+	if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] &&
+	    !spec->hp_independent_mode)
+		/* headphone out will just decode front left/right (stereo) */
+		snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag,
+					   0, format);
+
+	/* extra outputs copied from front */
+	for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++)
+		if (mout->extra_out_nid[i])
+			snd_hda_codec_setup_stream(codec,
+						   mout->extra_out_nid[i],
+						   stream_tag, 0, format);
+
+	/* surrounds */
+	for (i = 1; i < mout->num_dacs; i++) {
+		if (chs >= (i + 1) * 2) /* independent out */
+			snd_hda_codec_setup_stream(codec, nids[i], stream_tag,
+						   i * 2, format);
+		else /* copy front */
+			snd_hda_codec_setup_stream(codec, nids[i], stream_tag,
+						   0, format);
+	}
+}
+
+static int via_playback_multi_pcm_prepare(struct hda_pcm_stream *hinfo,
+					  struct hda_codec *codec,
+					  unsigned int stream_tag,
+					  unsigned int format,
+					  struct snd_pcm_substream *substream)
+{
+	struct via_spec *spec = codec->spec;
+	struct hda_multi_out *mout = &spec->multiout;
+	hda_nid_t *nids = mout->dac_nids;
+
+	if (substream->number == 0)
+		playback_multi_pcm_prep_0(codec, stream_tag, format,
+					  substream);
+	else {
+		if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] &&
+		    spec->hp_independent_mode)
+			snd_hda_codec_setup_stream(codec, mout->hp_nid,
+						   stream_tag, 0, format);
+	}
+
+	return 0;
+}
+
+static int via_playback_multi_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				    struct hda_codec *codec,
+				    struct snd_pcm_substream *substream)
+{
+	struct via_spec *spec = codec->spec;
+	struct hda_multi_out *mout = &spec->multiout;
+	hda_nid_t *nids = mout->dac_nids;
+	int i;
+
+	if (substream->number == 0) {
+		for (i = 0; i < mout->num_dacs; i++)
+			snd_hda_codec_setup_stream(codec, nids[i], 0, 0, 0);
+
+		if (mout->hp_nid && !spec->hp_independent_mode)
+			snd_hda_codec_setup_stream(codec, mout->hp_nid,
+						   0, 0, 0);
+
+		for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++)
+			if (mout->extra_out_nid[i])
+				snd_hda_codec_setup_stream(codec,
+							mout->extra_out_nid[i],
+							0, 0, 0);
+		mutex_lock(&codec->spdif_mutex);
+		if (mout->dig_out_nid &&
+		    mout->dig_out_used == HDA_DIG_ANALOG_DUP) {
+			snd_hda_codec_setup_stream(codec, mout->dig_out_nid,
+						   0, 0, 0);
+			mout->dig_out_used = 0;
+		}
+		mutex_unlock(&codec->spdif_mutex);
+	} else {
+		if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] &&
+		    spec->hp_independent_mode)
+			snd_hda_codec_setup_stream(codec, mout->hp_nid,
+						   0, 0, 0);
+	}
+
+	return 0;
+}
+
 /*
  * Digital out
  */
@@ -399,6 +740,21 @@ static int via_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
 	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
 }
 
+/* setup SPDIF output stream */
+static void setup_dig_playback_stream(struct hda_codec *codec, hda_nid_t nid,
+				 unsigned int stream_tag, unsigned int format)
+{
+	/* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
+	if (codec->spdif_ctls & AC_DIG1_ENABLE)
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
+				    codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff);
+	snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
+	/* turn on again (if needed) */
+	if (codec->spdif_ctls & AC_DIG1_ENABLE)
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
+				    codec->spdif_ctls & 0xff);
+}
+
 static int via_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 					struct hda_codec *codec,
 					unsigned int stream_tag,
@@ -406,8 +762,20 @@ static int via_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 					struct snd_pcm_substream *substream)
 {
 	struct via_spec *spec = codec->spec;
-	return snd_hda_multi_out_dig_prepare(codec, &spec->multiout,
-					     stream_tag, format, substream);
+	hda_nid_t nid;
+
+	/* 1st or 2nd S/PDIF */
+	if (substream->number == 0)
+		nid = spec->multiout.dig_out_nid;
+	else if (substream->number == 1)
+		nid = spec->extra_dig_out_nid;
+	else
+		return -1;
+
+	mutex_lock(&codec->spdif_mutex);
+	setup_dig_playback_stream(codec, nid, stream_tag, format);
+	mutex_unlock(&codec->spdif_mutex);
+	return 0;
 }
 
 /*
@@ -436,14 +804,14 @@ static int via_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 }
 
 static struct hda_pcm_stream vt1708_pcm_analog_playback = {
-	.substreams = 1,
+	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 8,
 	.nid = 0x10, /* NID to query formats and rates */
 	.ops = {
 		.open = via_playback_pcm_open,
-		.prepare = via_playback_pcm_prepare,
-		.cleanup = via_playback_pcm_cleanup
+		.prepare = via_playback_multi_pcm_prepare,
+		.cleanup = via_playback_multi_pcm_cleanup
 	},
 };
 
@@ -515,6 +883,13 @@ static int via_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 		spec->multiout.share_spdif = 1;
+
+		if (spec->extra_dig_out_nid) {
+			err = snd_hda_create_spdif_out_ctls(codec,
+						    spec->extra_dig_out_nid);
+			if (err < 0)
+				return err;
+		}
 	}
 	if (spec->dig_in_nid) {
 		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
@@ -580,10 +955,89 @@ static void via_free(struct hda_codec *codec)
 	kfree(codec->spec);
 }
 
+/* mute internal speaker if HP is plugged */
+static void via_hp_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	struct via_spec *spec = codec->spec;
+
+	present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	snd_hda_codec_amp_stereo(codec, spec->autocfg.line_out_pins[0],
+				 HDA_OUTPUT, 0, HDA_AMP_MUTE,
+				 present ? HDA_AMP_MUTE : 0);
+}
+
+static void via_gpio_control(struct hda_codec *codec)
+{
+	unsigned int gpio_data;
+	unsigned int vol_counter;
+	unsigned int vol;
+	unsigned int master_vol;
+
+	struct via_spec *spec = codec->spec;
+
+	gpio_data = snd_hda_codec_read(codec, codec->afg, 0,
+				       AC_VERB_GET_GPIO_DATA, 0) & 0x03;
+
+	vol_counter = (snd_hda_codec_read(codec, codec->afg, 0,
+					  0xF84, 0) & 0x3F0000) >> 16;
+
+	vol = vol_counter & 0x1F;
+	master_vol = snd_hda_codec_read(codec, 0x1A, 0,
+					AC_VERB_GET_AMP_GAIN_MUTE,
+					AC_AMP_GET_INPUT);
+
+	if (gpio_data == 0x02) {
+		/* unmute line out */
+		snd_hda_codec_amp_stereo(codec, spec->autocfg.line_out_pins[0],
+					 HDA_OUTPUT, 0, HDA_AMP_MUTE, 0);
+
+		if (vol_counter & 0x20) {
+			/* decrease volume */
+			if (vol > master_vol)
+				vol = master_vol;
+			snd_hda_codec_amp_stereo(codec, 0x1A, HDA_INPUT,
+						 0, HDA_AMP_VOLMASK,
+						 master_vol-vol);
+		} else {
+			/* increase volume */
+			snd_hda_codec_amp_stereo(codec, 0x1A, HDA_INPUT, 0,
+					 HDA_AMP_VOLMASK,
+					 ((master_vol+vol) > 0x2A) ? 0x2A :
+					  (master_vol+vol));
+		}
+	} else if (!(gpio_data & 0x02)) {
+		/* mute line out */
+		snd_hda_codec_amp_stereo(codec,
+					 spec->autocfg.line_out_pins[0],
+					 HDA_OUTPUT, 0, HDA_AMP_MUTE,
+					 HDA_AMP_MUTE);
+	}
+}
+
+/* unsolicited event for jack sensing */
+static void via_unsol_event(struct hda_codec *codec,
+				  unsigned int res)
+{
+	res >>= 26;
+	if (res == VIA_HP_EVENT)
+		via_hp_automute(codec);
+	else if (res == VIA_GPIO_EVENT)
+		via_gpio_control(codec);
+}
+
+static hda_nid_t slave_dig_outs[] = {
+	0,
+};
+
 static int via_init(struct hda_codec *codec)
 {
 	struct via_spec *spec = codec->spec;
-	snd_hda_sequence_write(codec, spec->init_verbs);
+	int i;
+	for (i = 0; i < spec->num_iverbs; i++)
+		snd_hda_sequence_write(codec, spec->init_verbs[i]);
+
 	/* Lydia Add for EAPD enable */
 	if (!spec->dig_in_nid) { /* No Digital In connection */
 		if (IS_VT1708_VENDORID(codec->vendor_id)) {
@@ -611,6 +1065,9 @@ static int via_init(struct hda_codec *codec)
 		snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0,
 				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN);
 
+	/* no slave outs */
+	codec->slave_dig_outs = slave_dig_outs;
+
  	return 0;
 }
 
@@ -657,10 +1114,10 @@ static int vt1708_auto_fill_dac_nids(struct via_spec *spec,
 				spec->multiout.dac_nids[i] = 0x12;
 				break;
 			case AUTO_SEQ_SURROUND:
-				spec->multiout.dac_nids[i] = 0x13;
+				spec->multiout.dac_nids[i] = 0x11;
 				break;
 			case AUTO_SEQ_SIDE:
-				spec->multiout.dac_nids[i] = 0x11;
+				spec->multiout.dac_nids[i] = 0x13;
 				break;
 			}
 		}
@@ -685,7 +1142,7 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec,
 			continue;
 		
 		if (i != AUTO_SEQ_FRONT)
-			nid_vol = 0x1b - i + 1;
+			nid_vol = 0x18 + i;
 
 		if (i == AUTO_SEQ_CENLFE) {
 			/* Center/LFE */
@@ -760,6 +1217,24 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec,
 	return 0;
 }
 
+static void create_hp_imux(struct via_spec *spec)
+{
+	int i;
+	struct hda_input_mux *imux = &spec->private_imux[1];
+	static const char *texts[] = { "OFF", "ON", NULL};
+
+	/* for hp mode select */
+	i = 0;
+	while (texts[i] != NULL) {
+		imux->items[imux->num_items].label =  texts[i];
+		imux->items[imux->num_items].index = i;
+		imux->num_items++;
+		i++;
+	}
+
+	spec->hp_mux = &spec->private_imux[1];
+}
+
 static int vt1708_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
 {
 	int err;
@@ -780,6 +1255,8 @@ static int vt1708_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
 	if (err < 0)
 		return err;
 
+	create_hp_imux(spec);
+
 	return 0;
 }
 
@@ -790,7 +1267,7 @@ static int vt1708_auto_create_analog_input_ctls(struct via_spec *spec,
 	static char *labels[] = {
 		"Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL
 	};
-	struct hda_input_mux *imux = &spec->private_imux;
+	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, err, idx = 0;
 
 	/* for internal loopback recording select */
@@ -840,11 +1317,36 @@ static struct hda_amp_list vt1708_loopbacks[] = {
 };
 #endif
 
+static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid)
+{
+	unsigned int def_conf;
+	unsigned char seqassoc;
+
+	def_conf = snd_hda_codec_read(codec, nid, 0,
+				      AC_VERB_GET_CONFIG_DEFAULT, 0);
+	seqassoc = (unsigned char) get_defcfg_association(def_conf);
+	seqassoc = (seqassoc << 4) | get_defcfg_sequence(def_conf);
+	if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) {
+		if (seqassoc == 0xff) {
+			def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30));
+			snd_hda_codec_write(codec, nid, 0,
+					    AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
+					    def_conf >> 24);
+		}
+	}
+
+	return;
+}
+
 static int vt1708_parse_auto_config(struct hda_codec *codec)
 {
 	struct via_spec *spec = codec->spec;
 	int err;
 
+	/* Add HP and CD pin config connect bit re-config action */
+	vt1708_set_pinconfig_connect(codec, VT1708_HP_PIN_NID);
+	vt1708_set_pinconfig_connect(codec, VT1708_CD_PIN_NID);
+
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
 	if (err < 0)
 		return err;
@@ -874,9 +1376,12 @@ static int vt1708_parse_auto_config(struct hda_codec *codec)
 	if (spec->kctl_alloc)
 		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
 
-	spec->init_verbs = vt1708_volume_init_verbs;	
+	spec->init_verbs[spec->num_iverbs++] = vt1708_volume_init_verbs;
+
+	spec->input_mux = &spec->private_imux[0];
 
-	spec->input_mux = &spec->private_imux;
+	if (spec->hp_mux)
+		spec->mixers[spec->num_mixers++] = via_hp_mixer;
 
 	return 1;
 }
@@ -897,7 +1402,7 @@ static int patch_vt1708(struct hda_codec *codec)
 	int err;
 
 	/* create a codec specific record */
-	spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
 		return -ENOMEM;
 
@@ -966,6 +1471,11 @@ static struct snd_kcontrol_new vt1709_capture_mixer[] = {
 	{ } /* end */
 };
 
+static struct hda_verb vt1709_uniwill_init_verbs[] = {
+	{0x20, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT},
+	{ }
+};
+
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
@@ -1090,11 +1600,11 @@ static int vt1709_auto_fill_dac_nids(struct via_spec *spec,
 					break;
 				case AUTO_SEQ_SURROUND:
 					/* AOW3 */
-					spec->multiout.dac_nids[i] = 0x27;
+					spec->multiout.dac_nids[i] = 0x11;
 					break;
 				case AUTO_SEQ_SIDE:
 					/* AOW1 */
-					spec->multiout.dac_nids[i] = 0x11;
+					spec->multiout.dac_nids[i] = 0x27;
 					break;
 				default:
 					break;
@@ -1203,26 +1713,26 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec,
 		} else if (i == AUTO_SEQ_SURROUND) {
 			sprintf(name, "%s Playback Volume", chname[i]);
 			err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
-					      HDA_COMPOSE_AMP_VAL(0x29, 3, 0,
+					      HDA_COMPOSE_AMP_VAL(0x1a, 3, 0,
 								  HDA_OUTPUT));
 			if (err < 0)
 				return err;
 			sprintf(name, "%s Playback Switch", chname[i]);
 			err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
-					      HDA_COMPOSE_AMP_VAL(0x29, 3, 0,
+					      HDA_COMPOSE_AMP_VAL(0x1a, 3, 0,
 								  HDA_OUTPUT));
 			if (err < 0)
 				return err;
 		} else if (i == AUTO_SEQ_SIDE) {
 			sprintf(name, "%s Playback Volume", chname[i]);
 			err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
-					      HDA_COMPOSE_AMP_VAL(0x1a, 3, 0,
+					      HDA_COMPOSE_AMP_VAL(0x29, 3, 0,
 								  HDA_OUTPUT));
 			if (err < 0)
 				return err;
 			sprintf(name, "%s Playback Switch", chname[i]);
 			err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
-					      HDA_COMPOSE_AMP_VAL(0x1a, 3, 0,
+					      HDA_COMPOSE_AMP_VAL(0x29, 3, 0,
 								  HDA_OUTPUT));
 			if (err < 0)
 				return err;
@@ -1265,7 +1775,7 @@ static int vt1709_auto_create_analog_input_ctls(struct via_spec *spec,
 	static char *labels[] = {
 		"Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL
 	};
-	struct hda_input_mux *imux = &spec->private_imux;
+	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, err, idx = 0;
 
 	/* for internal loopback recording select */
@@ -1339,7 +1849,10 @@ static int vt1709_parse_auto_config(struct hda_codec *codec)
 	if (spec->kctl_alloc)
 		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
 
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
+
+	if (spec->hp_mux)
+		spec->mixers[spec->num_mixers++] = via_hp_mixer;
 
 	return 1;
 }
@@ -1360,7 +1873,7 @@ static int patch_vt1709_10ch(struct hda_codec *codec)
 	int err;
 
 	/* create a codec specific record */
-	spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
 		return -ENOMEM;
 
@@ -1375,7 +1888,8 @@ static int patch_vt1709_10ch(struct hda_codec *codec)
 		       "Using genenic mode...\n");
 	}
 
-	spec->init_verbs = vt1709_10ch_volume_init_verbs;	
+	spec->init_verbs[spec->num_iverbs++] = vt1709_10ch_volume_init_verbs;
+	spec->init_verbs[spec->num_iverbs++] = vt1709_uniwill_init_verbs;
 
 	spec->stream_name_analog = "VT1709 Analog";
 	spec->stream_analog_playback = &vt1709_10ch_pcm_analog_playback;
@@ -1396,6 +1910,7 @@ static int patch_vt1709_10ch(struct hda_codec *codec)
 	codec->patch_ops = via_patch_ops;
 
 	codec->patch_ops.init = via_auto_init;
+	codec->patch_ops.unsol_event = via_unsol_event;
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	spec->loopback.amplist = vt1709_loopbacks;
 #endif
@@ -1451,7 +1966,7 @@ static int patch_vt1709_6ch(struct hda_codec *codec)
 	int err;
 
 	/* create a codec specific record */
-	spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
 		return -ENOMEM;
 
@@ -1466,7 +1981,8 @@ static int patch_vt1709_6ch(struct hda_codec *codec)
 		       "Using genenic mode...\n");
 	}
 
-	spec->init_verbs = vt1709_6ch_volume_init_verbs;	
+	spec->init_verbs[spec->num_iverbs++] = vt1709_6ch_volume_init_verbs;
+	spec->init_verbs[spec->num_iverbs++] = vt1709_uniwill_init_verbs;
 
 	spec->stream_name_analog = "VT1709 Analog";
 	spec->stream_analog_playback = &vt1709_6ch_pcm_analog_playback;
@@ -1487,6 +2003,7 @@ static int patch_vt1709_6ch(struct hda_codec *codec)
 	codec->patch_ops = via_patch_ops;
 
 	codec->patch_ops.init = via_auto_init;
+	codec->patch_ops.unsol_event = via_unsol_event;
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	spec->loopback.amplist = vt1709_loopbacks;
 #endif
@@ -1586,27 +2103,32 @@ static struct hda_verb vt1708B_4ch_volume_init_verbs[] = {
 	{ }
 };
 
+static struct hda_verb vt1708B_uniwill_init_verbs[] = {
+	{0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT},
+	{ }
+};
+
 static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = {
-	.substreams = 1,
+	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 8,
 	.nid = 0x10, /* NID to query formats and rates */
 	.ops = {
 		.open = via_playback_pcm_open,
-		.prepare = via_playback_pcm_prepare,
-		.cleanup = via_playback_pcm_cleanup
+		.prepare = via_playback_multi_pcm_prepare,
+		.cleanup = via_playback_multi_pcm_cleanup
 	},
 };
 
 static struct hda_pcm_stream vt1708B_4ch_pcm_analog_playback = {
-	.substreams = 1,
+	.substreams = 2,
 	.channels_min = 2,
 	.channels_max = 4,
 	.nid = 0x10, /* NID to query formats and rates */
 	.ops = {
 		.open = via_playback_pcm_open,
-		.prepare = via_playback_pcm_prepare,
-		.cleanup = via_playback_pcm_cleanup
+		.prepare = via_playback_multi_pcm_prepare,
+		.cleanup = via_playback_multi_pcm_cleanup
 	},
 };
 
@@ -1662,10 +2184,10 @@ static int vt1708B_auto_fill_dac_nids(struct via_spec *spec,
 				spec->multiout.dac_nids[i] = 0x24;
 				break;
 			case AUTO_SEQ_SURROUND:
-				spec->multiout.dac_nids[i] = 0x25;
+				spec->multiout.dac_nids[i] = 0x11;
 				break;
 			case AUTO_SEQ_SIDE:
-				spec->multiout.dac_nids[i] = 0x11;
+				spec->multiout.dac_nids[i] = 0x25;
 				break;
 			}
 		}
@@ -1680,7 +2202,7 @@ static int vt1708B_auto_create_multi_out_ctls(struct via_spec *spec,
 {
 	char name[32];
 	static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" };
-	hda_nid_t nid_vols[] = {0x16, 0x27, 0x26, 0x18};
+	hda_nid_t nid_vols[] = {0x16, 0x18, 0x26, 0x27};
 	hda_nid_t nid, nid_vol = 0;
 	int i, err;
 
@@ -1785,6 +2307,8 @@ static int vt1708B_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
 	if (err < 0)
 		return err;
 
+	create_hp_imux(spec);
+
 	return 0;
 }
 
@@ -1795,7 +2319,7 @@ static int vt1708B_auto_create_analog_input_ctls(struct via_spec *spec,
 	static char *labels[] = {
 		"Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL
 	};
-	struct hda_input_mux *imux = &spec->private_imux;
+	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, err, idx = 0;
 
 	/* for internal loopback recording select */
@@ -1869,7 +2393,10 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec)
 	if (spec->kctl_alloc)
 		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
 
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
+
+	if (spec->hp_mux)
+		spec->mixers[spec->num_mixers++] = via_hp_mixer;
 
 	return 1;
 }
@@ -1890,7 +2417,7 @@ static int patch_vt1708B_8ch(struct hda_codec *codec)
 	int err;
 
 	/* create a codec specific record */
-	spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
 		return -ENOMEM;
 
@@ -1906,7 +2433,8 @@ static int patch_vt1708B_8ch(struct hda_codec *codec)
 		       "from BIOS.  Using genenic mode...\n");
 	}
 
-	spec->init_verbs = vt1708B_8ch_volume_init_verbs;
+	spec->init_verbs[spec->num_iverbs++] = vt1708B_8ch_volume_init_verbs;
+	spec->init_verbs[spec->num_iverbs++] = vt1708B_uniwill_init_verbs;
 
 	spec->stream_name_analog = "VT1708B Analog";
 	spec->stream_analog_playback = &vt1708B_8ch_pcm_analog_playback;
@@ -1926,6 +2454,7 @@ static int patch_vt1708B_8ch(struct hda_codec *codec)
 	codec->patch_ops = via_patch_ops;
 
 	codec->patch_ops.init = via_auto_init;
+	codec->patch_ops.unsol_event = via_unsol_event;
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	spec->loopback.amplist = vt1708B_loopbacks;
 #endif
@@ -1939,7 +2468,7 @@ static int patch_vt1708B_4ch(struct hda_codec *codec)
 	int err;
 
 	/* create a codec specific record */
-	spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
 		return -ENOMEM;
 
@@ -1955,7 +2484,8 @@ static int patch_vt1708B_4ch(struct hda_codec *codec)
 		       "from BIOS.  Using genenic mode...\n");
 	}
 
-	spec->init_verbs = vt1708B_4ch_volume_init_verbs;
+	spec->init_verbs[spec->num_iverbs++] = vt1708B_4ch_volume_init_verbs;
+	spec->init_verbs[spec->num_iverbs++] = vt1708B_uniwill_init_verbs;
 
 	spec->stream_name_analog = "VT1708B Analog";
 	spec->stream_analog_playback = &vt1708B_4ch_pcm_analog_playback;
@@ -1975,6 +2505,7 @@ static int patch_vt1708B_4ch(struct hda_codec *codec)
 	codec->patch_ops = via_patch_ops;
 
 	codec->patch_ops.init = via_auto_init;
+	codec->patch_ops.unsol_event = via_unsol_event;
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	spec->loopback.amplist = vt1708B_loopbacks;
 #endif
@@ -1982,6 +2513,752 @@ static int patch_vt1708B_4ch(struct hda_codec *codec)
 	return 0;
 }
 
+/* Patch for VT1708S */
+
+/* VT1708S software backdoor based override for buggy hardware micboost
+ * setting */
+#define MIC_BOOST_VOLUME(xname, nid) {				\
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,		\
+	.name = xname,					\
+	.index = 0,					\
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |	\
+	SNDRV_CTL_ELEM_ACCESS_TLV_READ |		\
+	SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK,		\
+	.info = mic_boost_volume_info,			\
+	.get = snd_hda_mixer_amp_volume_get,		\
+	.put = snd_hda_mixer_amp_volume_put,		\
+	.tlv = { .c = mic_boost_tlv },			\
+	.private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT) }
+
+/* capture mixer elements */
+static struct snd_kcontrol_new vt1708S_capture_mixer[] = {
+	HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT),
+	MIC_BOOST_VOLUME("Mic Boost Capture Volume", 0x1A),
+	MIC_BOOST_VOLUME("Front Mic Boost Capture Volume", 0x1E),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		/* The multiple "Capture Source" controls confuse alsamixer
+		 * So call somewhat different..
+		 */
+		/* .name = "Capture Source", */
+		.name = "Input Source",
+		.count = 1,
+		.info = via_mux_enum_info,
+		.get = via_mux_enum_get,
+		.put = via_mux_enum_put,
+	},
+	{ } /* end */
+};
+
+static struct hda_verb vt1708S_volume_init_verbs[] = {
+	/* Unmute ADC0-1 and set the default input to mic-in */
+	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+	/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the
+	 * analog-loopback mixer widget */
+	/* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+	/* Setup default input of PW4 to MW0 */
+	{0x1d, AC_VERB_SET_CONNECT_SEL, 0x0},
+	/* PW9, PW10  Output enable */
+	{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	/* Enable Mic Boost Volume backdoor */
+	{0x1, 0xf98, 0x1},
+	{ }
+};
+
+static struct hda_verb vt1708S_uniwill_init_verbs[] = {
+	{0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT},
+	{ }
+};
+
+static struct hda_pcm_stream vt1708S_pcm_analog_playback = {
+	.substreams = 2,
+	.channels_min = 2,
+	.channels_max = 8,
+	.nid = 0x10, /* NID to query formats and rates */
+	.ops = {
+		.open = via_playback_pcm_open,
+		.prepare = via_playback_pcm_prepare,
+		.cleanup = via_playback_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream vt1708S_pcm_analog_capture = {
+	.substreams = 2,
+	.channels_min = 2,
+	.channels_max = 2,
+	.nid = 0x13, /* NID to query formats and rates */
+	.ops = {
+		.prepare = via_capture_pcm_prepare,
+		.cleanup = via_capture_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream vt1708S_pcm_digital_playback = {
+	.substreams = 2,
+	.channels_min = 2,
+	.channels_max = 2,
+	/* NID is set in via_build_pcms */
+	.ops = {
+		.open = via_dig_playback_pcm_open,
+		.close = via_dig_playback_pcm_close,
+		.prepare = via_dig_playback_pcm_prepare
+	},
+};
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int vt1708S_auto_fill_dac_nids(struct via_spec *spec,
+				     const struct auto_pin_cfg *cfg)
+{
+	int i;
+	hda_nid_t nid;
+
+	spec->multiout.num_dacs = cfg->line_outs;
+
+	spec->multiout.dac_nids = spec->private_dac_nids;
+
+	for (i = 0; i < 4; i++) {
+		nid = cfg->line_out_pins[i];
+		if (nid) {
+			/* config dac list */
+			switch (i) {
+			case AUTO_SEQ_FRONT:
+				spec->multiout.dac_nids[i] = 0x10;
+				break;
+			case AUTO_SEQ_CENLFE:
+				spec->multiout.dac_nids[i] = 0x24;
+				break;
+			case AUTO_SEQ_SURROUND:
+				spec->multiout.dac_nids[i] = 0x11;
+				break;
+			case AUTO_SEQ_SIDE:
+				spec->multiout.dac_nids[i] = 0x25;
+				break;
+			}
+		}
+	}
+
+	return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int vt1708S_auto_create_multi_out_ctls(struct via_spec *spec,
+					     const struct auto_pin_cfg *cfg)
+{
+	char name[32];
+	static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" };
+	hda_nid_t nid_vols[] = {0x10, 0x11, 0x24, 0x25};
+	hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x26, 0x27};
+	hda_nid_t nid, nid_vol, nid_mute;
+	int i, err;
+
+	for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
+		nid = cfg->line_out_pins[i];
+
+		if (!nid)
+			continue;
+
+		nid_vol = nid_vols[i];
+		nid_mute = nid_mutes[i];
+
+		if (i == AUTO_SEQ_CENLFE) {
+			/* Center/LFE */
+			err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+					      "Center Playback Volume",
+					      HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
+								  HDA_OUTPUT));
+			if (err < 0)
+				return err;
+			err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+					      "LFE Playback Volume",
+					      HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
+								  HDA_OUTPUT));
+			if (err < 0)
+				return err;
+			err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+					      "Center Playback Switch",
+					      HDA_COMPOSE_AMP_VAL(nid_mute,
+								  1, 0,
+								  HDA_OUTPUT));
+			if (err < 0)
+				return err;
+			err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+					      "LFE Playback Switch",
+					      HDA_COMPOSE_AMP_VAL(nid_mute,
+								  2, 0,
+								  HDA_OUTPUT));
+			if (err < 0)
+				return err;
+		} else if (i == AUTO_SEQ_FRONT) {
+			/* add control to mixer index 0 */
+			err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+					      "Master Front Playback Volume",
+					      HDA_COMPOSE_AMP_VAL(0x16, 3, 0,
+								  HDA_INPUT));
+			if (err < 0)
+				return err;
+			err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+					      "Master Front Playback Switch",
+					      HDA_COMPOSE_AMP_VAL(0x16, 3, 0,
+								  HDA_INPUT));
+			if (err < 0)
+				return err;
+
+			/* Front */
+			sprintf(name, "%s Playback Volume", chname[i]);
+			err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+					      HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
+								  HDA_OUTPUT));
+			if (err < 0)
+				return err;
+			sprintf(name, "%s Playback Switch", chname[i]);
+			err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
+					      HDA_COMPOSE_AMP_VAL(nid_mute,
+								  3, 0,
+								  HDA_OUTPUT));
+			if (err < 0)
+				return err;
+		} else {
+			sprintf(name, "%s Playback Volume", chname[i]);
+			err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+					      HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
+								  HDA_OUTPUT));
+			if (err < 0)
+				return err;
+			sprintf(name, "%s Playback Switch", chname[i]);
+			err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
+					      HDA_COMPOSE_AMP_VAL(nid_mute,
+								  3, 0,
+								  HDA_OUTPUT));
+			if (err < 0)
+				return err;
+		}
+	}
+
+	return 0;
+}
+
+static int vt1708S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
+{
+	int err;
+
+	if (!pin)
+		return 0;
+
+	spec->multiout.hp_nid = VT1708S_HP_NID; /* AOW3 */
+
+	err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+			      "Headphone Playback Volume",
+			      HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT));
+	if (err < 0)
+		return err;
+
+	err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+			      "Headphone Playback Switch",
+			      HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+	if (err < 0)
+		return err;
+
+	create_hp_imux(spec);
+
+	return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int vt1708S_auto_create_analog_input_ctls(struct via_spec *spec,
+						const struct auto_pin_cfg *cfg)
+{
+	static char *labels[] = {
+		"Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL
+	};
+	struct hda_input_mux *imux = &spec->private_imux[0];
+	int i, err, idx = 0;
+
+	/* for internal loopback recording select */
+	imux->items[imux->num_items].label = "Stereo Mixer";
+	imux->items[imux->num_items].index = 5;
+	imux->num_items++;
+
+	for (i = 0; i < AUTO_PIN_LAST; i++) {
+		if (!cfg->input_pins[i])
+			continue;
+
+		switch (cfg->input_pins[i]) {
+		case 0x1a: /* Mic */
+			idx = 2;
+			break;
+
+		case 0x1b: /* Line In */
+			idx = 3;
+			break;
+
+		case 0x1e: /* Front Mic */
+			idx = 4;
+			break;
+
+		case 0x1f: /* CD */
+			idx = 1;
+			break;
+		}
+		err = via_new_analog_input(spec, cfg->input_pins[i], labels[i],
+					   idx, 0x16);
+		if (err < 0)
+			return err;
+		imux->items[imux->num_items].label = labels[i];
+		imux->items[imux->num_items].index = idx-1;
+		imux->num_items++;
+	}
+	return 0;
+}
+
+static int vt1708S_parse_auto_config(struct hda_codec *codec)
+{
+	struct via_spec *spec = codec->spec;
+	int err;
+	static hda_nid_t vt1708s_ignore[] = {0x21, 0};
+
+	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+					   vt1708s_ignore);
+	if (err < 0)
+		return err;
+	err = vt1708S_auto_fill_dac_nids(spec, &spec->autocfg);
+	if (err < 0)
+		return err;
+	if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
+		return 0; /* can't find valid BIOS pin config */
+
+	err = vt1708S_auto_create_multi_out_ctls(spec, &spec->autocfg);
+	if (err < 0)
+		return err;
+	err = vt1708S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
+	if (err < 0)
+		return err;
+	err = vt1708S_auto_create_analog_input_ctls(spec, &spec->autocfg);
+	if (err < 0)
+		return err;
+
+	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+	if (spec->autocfg.dig_out_pin)
+		spec->multiout.dig_out_nid = VT1708S_DIGOUT_NID;
+
+	spec->extra_dig_out_nid = 0x15;
+
+	if (spec->kctl_alloc)
+		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+	spec->input_mux = &spec->private_imux[0];
+
+	if (spec->hp_mux)
+		spec->mixers[spec->num_mixers++] = via_hp_mixer;
+
+	return 1;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list vt1708S_loopbacks[] = {
+	{ 0x16, HDA_INPUT, 1 },
+	{ 0x16, HDA_INPUT, 2 },
+	{ 0x16, HDA_INPUT, 3 },
+	{ 0x16, HDA_INPUT, 4 },
+	{ } /* end */
+};
+#endif
+
+static int patch_vt1708S(struct hda_codec *codec)
+{
+	struct via_spec *spec;
+	int err;
+
+	/* create a codec specific record */
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL)
+		return -ENOMEM;
+
+	codec->spec = spec;
+
+	/* automatic parse from the BIOS config */
+	err = vt1708S_parse_auto_config(codec);
+	if (err < 0) {
+		via_free(codec);
+		return err;
+	} else if (!err) {
+		printk(KERN_INFO "hda_codec: Cannot set up configuration "
+		       "from BIOS.  Using genenic mode...\n");
+	}
+
+	spec->init_verbs[spec->num_iverbs++] = vt1708S_volume_init_verbs;
+	spec->init_verbs[spec->num_iverbs++] = vt1708S_uniwill_init_verbs;
+
+	spec->stream_name_analog = "VT1708S Analog";
+	spec->stream_analog_playback = &vt1708S_pcm_analog_playback;
+	spec->stream_analog_capture = &vt1708S_pcm_analog_capture;
+
+	spec->stream_name_digital = "VT1708S Digital";
+	spec->stream_digital_playback = &vt1708S_pcm_digital_playback;
+
+	if (!spec->adc_nids && spec->input_mux) {
+		spec->adc_nids = vt1708S_adc_nids;
+		spec->num_adc_nids = ARRAY_SIZE(vt1708S_adc_nids);
+		spec->mixers[spec->num_mixers] = vt1708S_capture_mixer;
+		spec->num_mixers++;
+	}
+
+	codec->patch_ops = via_patch_ops;
+
+	codec->patch_ops.init = via_auto_init;
+	codec->patch_ops.unsol_event = via_unsol_event;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+	spec->loopback.amplist = vt1708S_loopbacks;
+#endif
+
+	return 0;
+}
+
+/* Patch for VT1702 */
+
+/* capture mixer elements */
+static struct snd_kcontrol_new vt1702_capture_mixer[] = {
+	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x20, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x20, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x1F, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x1F, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Digital Mic Boost Capture Volume", 0x1E, 0x0,
+			 HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		/* The multiple "Capture Source" controls confuse alsamixer
+		 * So call somewhat different..
+		 */
+		/* .name = "Capture Source", */
+		.name = "Input Source",
+		.count = 1,
+		.info = via_mux_enum_info,
+		.get = via_mux_enum_get,
+		.put = via_mux_enum_put,
+	},
+	{ } /* end */
+};
+
+static struct hda_verb vt1702_volume_init_verbs[] = {
+	/*
+	 * Unmute ADC0-1 and set the default input to mic-in
+	 */
+	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x1F, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x20, 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 = 1, Line = 1, Mic2 = 3 */
+	{0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+	{0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+	{0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+	/* Setup default input of PW4 to MW0 */
+	{0x17, AC_VERB_SET_CONNECT_SEL, 0x1},
+	/* PW6 PW7 Output enable */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	{0x1C, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	{ }
+};
+
+static struct hda_verb vt1702_uniwill_init_verbs[] = {
+	{0x01, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_GPIO_EVENT},
+	{0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT},
+	{ }
+};
+
+static struct hda_pcm_stream vt1702_pcm_analog_playback = {
+	.substreams = 2,
+	.channels_min = 2,
+	.channels_max = 2,
+	.nid = 0x10, /* NID to query formats and rates */
+	.ops = {
+		.open = via_playback_pcm_open,
+		.prepare = via_playback_multi_pcm_prepare,
+		.cleanup = via_playback_multi_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream vt1702_pcm_analog_capture = {
+	.substreams = 3,
+	.channels_min = 2,
+	.channels_max = 2,
+	.nid = 0x12, /* NID to query formats and rates */
+	.ops = {
+		.prepare = via_capture_pcm_prepare,
+		.cleanup = via_capture_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream vt1702_pcm_digital_playback = {
+	.substreams = 2,
+	.channels_min = 2,
+	.channels_max = 2,
+	/* NID is set in via_build_pcms */
+	.ops = {
+		.open = via_dig_playback_pcm_open,
+		.close = via_dig_playback_pcm_close,
+		.prepare = via_dig_playback_pcm_prepare
+	},
+};
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int vt1702_auto_fill_dac_nids(struct via_spec *spec,
+				     const struct auto_pin_cfg *cfg)
+{
+	spec->multiout.num_dacs = 1;
+	spec->multiout.dac_nids = spec->private_dac_nids;
+
+	if (cfg->line_out_pins[0]) {
+		/* config dac list */
+		spec->multiout.dac_nids[0] = 0x10;
+	}
+
+	return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int vt1702_auto_create_line_out_ctls(struct via_spec *spec,
+					     const struct auto_pin_cfg *cfg)
+{
+	int err;
+
+	if (!cfg->line_out_pins[0])
+		return -1;
+
+	/* add control to mixer index 0 */
+	err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+			      "Master Front Playback Volume",
+			      HDA_COMPOSE_AMP_VAL(0x1A, 3, 0, HDA_INPUT));
+	if (err < 0)
+		return err;
+	err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+			      "Master Front Playback Switch",
+			      HDA_COMPOSE_AMP_VAL(0x1A, 3, 0, HDA_INPUT));
+	if (err < 0)
+		return err;
+
+	/* Front */
+	err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+			      "Front Playback Volume",
+			      HDA_COMPOSE_AMP_VAL(0x10, 3, 0, HDA_OUTPUT));
+	if (err < 0)
+		return err;
+	err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+			      "Front Playback Switch",
+			      HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_OUTPUT));
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
+static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
+{
+	int err;
+
+	if (!pin)
+		return 0;
+
+	spec->multiout.hp_nid = 0x1D;
+
+	err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+			      "Headphone Playback Volume",
+			      HDA_COMPOSE_AMP_VAL(0x1D, 3, 0, HDA_OUTPUT));
+	if (err < 0)
+		return err;
+
+	err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+			      "Headphone Playback Switch",
+			      HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+	if (err < 0)
+		return err;
+
+	create_hp_imux(spec);
+
+	return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int vt1702_auto_create_analog_input_ctls(struct via_spec *spec,
+						const struct auto_pin_cfg *cfg)
+{
+	static char *labels[] = {
+		"Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL
+	};
+	struct hda_input_mux *imux = &spec->private_imux[0];
+	int i, err, idx = 0;
+
+	/* for internal loopback recording select */
+	imux->items[imux->num_items].label = "Stereo Mixer";
+	imux->items[imux->num_items].index = 3;
+	imux->num_items++;
+
+	for (i = 0; i < AUTO_PIN_LAST; i++) {
+		if (!cfg->input_pins[i])
+			continue;
+
+		switch (cfg->input_pins[i]) {
+		case 0x14: /* Mic */
+			idx = 1;
+			break;
+
+		case 0x15: /* Line In */
+			idx = 2;
+			break;
+
+		case 0x18: /* Front Mic */
+			idx = 3;
+			break;
+		}
+		err = via_new_analog_input(spec, cfg->input_pins[i],
+					   labels[i], idx, 0x1A);
+		if (err < 0)
+			return err;
+		imux->items[imux->num_items].label = labels[i];
+		imux->items[imux->num_items].index = idx-1;
+		imux->num_items++;
+	}
+	return 0;
+}
+
+static int vt1702_parse_auto_config(struct hda_codec *codec)
+{
+	struct via_spec *spec = codec->spec;
+	int err;
+	static hda_nid_t vt1702_ignore[] = {0x1C, 0};
+
+	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+					   vt1702_ignore);
+	if (err < 0)
+		return err;
+	err = vt1702_auto_fill_dac_nids(spec, &spec->autocfg);
+	if (err < 0)
+		return err;
+	if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
+		return 0; /* can't find valid BIOS pin config */
+
+	err = vt1702_auto_create_line_out_ctls(spec, &spec->autocfg);
+	if (err < 0)
+		return err;
+	err = vt1702_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
+	if (err < 0)
+		return err;
+	err = vt1702_auto_create_analog_input_ctls(spec, &spec->autocfg);
+	if (err < 0)
+		return err;
+
+	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+	if (spec->autocfg.dig_out_pin)
+		spec->multiout.dig_out_nid = VT1702_DIGOUT_NID;
+
+	spec->extra_dig_out_nid = 0x1B;
+
+	if (spec->kctl_alloc)
+		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+	spec->input_mux = &spec->private_imux[0];
+
+	if (spec->hp_mux)
+		spec->mixers[spec->num_mixers++] = via_hp_mixer;
+
+	return 1;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list vt1702_loopbacks[] = {
+	{ 0x1A, HDA_INPUT, 1 },
+	{ 0x1A, HDA_INPUT, 2 },
+	{ 0x1A, HDA_INPUT, 3 },
+	{ 0x1A, HDA_INPUT, 4 },
+	{ } /* end */
+};
+#endif
+
+static int patch_vt1702(struct hda_codec *codec)
+{
+	struct via_spec *spec;
+	int err;
+	unsigned int response;
+	unsigned char control;
+
+	/* create a codec specific record */
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL)
+		return -ENOMEM;
+
+	codec->spec = spec;
+
+	/* automatic parse from the BIOS config */
+	err = vt1702_parse_auto_config(codec);
+	if (err < 0) {
+		via_free(codec);
+		return err;
+	} else if (!err) {
+		printk(KERN_INFO "hda_codec: Cannot set up configuration "
+		       "from BIOS.  Using genenic mode...\n");
+	}
+
+	spec->init_verbs[spec->num_iverbs++] = vt1702_volume_init_verbs;
+	spec->init_verbs[spec->num_iverbs++] = vt1702_uniwill_init_verbs;
+
+	spec->stream_name_analog = "VT1702 Analog";
+	spec->stream_analog_playback = &vt1702_pcm_analog_playback;
+	spec->stream_analog_capture = &vt1702_pcm_analog_capture;
+
+	spec->stream_name_digital = "VT1702 Digital";
+	spec->stream_digital_playback = &vt1702_pcm_digital_playback;
+
+	if (!spec->adc_nids && spec->input_mux) {
+		spec->adc_nids = vt1702_adc_nids;
+		spec->num_adc_nids = ARRAY_SIZE(vt1702_adc_nids);
+		spec->mixers[spec->num_mixers] = vt1702_capture_mixer;
+		spec->num_mixers++;
+	}
+
+	codec->patch_ops = via_patch_ops;
+
+	codec->patch_ops.init = via_auto_init;
+	codec->patch_ops.unsol_event = via_unsol_event;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+	spec->loopback.amplist = vt1702_loopbacks;
+#endif
+
+	/* Open backdoor */
+	response = snd_hda_codec_read(codec, codec->afg, 0, 0xF8C, 0);
+	control = (unsigned char)(response & 0xff);
+	control |= 0x3;
+	snd_hda_codec_write(codec,  codec->afg, 0, 0xF88, control);
+
+	/* Enable GPIO 0&1 for volume&mute control */
+	/* Enable GPIO 2 for DMIC-DATA */
+	response = snd_hda_codec_read(codec, codec->afg, 0, 0xF84, 0);
+	control = (unsigned char)((response >> 16) & 0x3f);
+	snd_hda_codec_write(codec,  codec->afg, 0, 0xF82, control);
+
+	return 0;
+}
+
 /*
  * patch entries
  */
@@ -2022,5 +3299,37 @@ struct hda_codec_preset snd_hda_preset_via[] = {
 	  .patch = patch_vt1708B_4ch},
 	{ .id = 0x1106E727, .name = "VIA VT1708B 4-Ch",
 	  .patch = patch_vt1708B_4ch},
+	{ .id = 0x11060397, .name = "VIA VT1708S",
+	  .patch = patch_vt1708S},
+	{ .id = 0x11061397, .name = "VIA VT1708S",
+	  .patch = patch_vt1708S},
+	{ .id = 0x11062397, .name = "VIA VT1708S",
+	  .patch = patch_vt1708S},
+	{ .id = 0x11063397, .name = "VIA VT1708S",
+	  .patch = patch_vt1708S},
+	{ .id = 0x11064397, .name = "VIA VT1708S",
+	  .patch = patch_vt1708S},
+	{ .id = 0x11065397, .name = "VIA VT1708S",
+	  .patch = patch_vt1708S},
+	{ .id = 0x11066397, .name = "VIA VT1708S",
+	  .patch = patch_vt1708S},
+	{ .id = 0x11067397, .name = "VIA VT1708S",
+	  .patch = patch_vt1708S},
+	{ .id = 0x11060398, .name = "VIA VT1702",
+	  .patch = patch_vt1702},
+	{ .id = 0x11061398, .name = "VIA VT1702",
+	  .patch = patch_vt1702},
+	{ .id = 0x11062398, .name = "VIA VT1702",
+	  .patch = patch_vt1702},
+	{ .id = 0x11063398, .name = "VIA VT1702",
+	  .patch = patch_vt1702},
+	{ .id = 0x11064398, .name = "VIA VT1702",
+	  .patch = patch_vt1702},
+	{ .id = 0x11065398, .name = "VIA VT1702",
+	  .patch = patch_vt1702},
+	{ .id = 0x11066398, .name = "VIA VT1702",
+	  .patch = patch_vt1702},
+	{ .id = 0x11067398, .name = "VIA VT1702",
+	  .patch = patch_vt1702},
 	{} /* terminator */
 };