/*
 * Universal Interface for Intel High Definition Audio Codec
 *
 * HD audio interface patch for ALC 260/880/882 codecs
 *
 * Copyright (c) 2004 Kailang Yang <kailang@realtek.com.tw>
 *                    PeiSen Hou <pshou@realtek.com.tw>
 *                    Takashi Iwai <tiwai@suse.de>
 *                    Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
 *
 *  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 <sound/driver.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/pci.h>
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"


/* ALC880 board config type */
enum {
	ALC880_3ST,
	ALC880_3ST_DIG,
	ALC880_5ST,
	ALC880_5ST_DIG,
	ALC880_W810,
	ALC880_Z71V,
	ALC880_6ST,
	ALC880_6ST_DIG,
	ALC880_F1734,
	ALC880_ASUS,
	ALC880_ASUS_DIG,
	ALC880_ASUS_W1V,
	ALC880_ASUS_DIG2,
	ALC880_UNIWILL_DIG,
	ALC880_CLEVO,
	ALC880_TCL_S700,
	ALC880_LG,
	ALC880_LG_LW,
#ifdef CONFIG_SND_DEBUG
	ALC880_TEST,
#endif
	ALC880_AUTO,
	ALC880_MODEL_LAST /* last tag */
};

/* ALC260 models */
enum {
	ALC260_BASIC,
	ALC260_HP,
	ALC260_HP_3013,
	ALC260_FUJITSU_S702X,
	ALC260_ACER,
#ifdef CONFIG_SND_DEBUG
	ALC260_TEST,
#endif
	ALC260_AUTO,
	ALC260_MODEL_LAST /* last tag */
};

/* ALC262 models */
enum {
	ALC262_BASIC,
	ALC262_FUJITSU,
	ALC262_HP_BPC,
	ALC262_BENQ_ED8,
	ALC262_AUTO,
	ALC262_MODEL_LAST /* last tag */
};

/* ALC861 models */
enum {
	ALC861_3ST,
	ALC660_3ST,
	ALC861_3ST_DIG,
	ALC861_6ST_DIG,
	ALC861_UNIWILL_M31,
	ALC861_AUTO,
	ALC861_MODEL_LAST,
};

/* ALC882 models */
enum {
	ALC882_3ST_DIG,
	ALC882_6ST_DIG,
	ALC882_ARIMA,
	ALC882_AUTO,
	ALC882_MODEL_LAST,
};

/* ALC883 models */
enum {
	ALC883_3ST_2ch_DIG,
	ALC883_3ST_6ch_DIG,
	ALC883_3ST_6ch,
	ALC883_6ST_DIG,
	ALC888_DEMO_BOARD,
	ALC883_ACER,
	ALC883_AUTO,
	ALC883_MODEL_LAST,
};

/* for GPIO Poll */
#define GPIO_MASK	0x03

struct alc_spec {
	/* codec parameterization */
	struct snd_kcontrol_new *mixers[5];	/* mixer arrays */
	unsigned int num_mixers;

	const struct hda_verb *init_verbs[5];	/* initialization verbs
						 * don't forget NULL
						 * termination!
						 */
	unsigned int num_init_verbs;

	char *stream_name_analog;	/* analog PCM stream */
	struct hda_pcm_stream *stream_analog_playback;
	struct hda_pcm_stream *stream_analog_capture;

	char *stream_name_digital;	/* digital PCM stream */ 
	struct hda_pcm_stream *stream_digital_playback;
	struct hda_pcm_stream *stream_digital_capture;

	/* playback */
	struct hda_multi_out multiout;	/* playback set-up
					 * max_channels, dacs must be set
					 * dig_out_nid and hp_nid are optional
					 */

	/* capture */
	unsigned int num_adc_nids;
	hda_nid_t *adc_nids;
	hda_nid_t dig_in_nid;		/* digital-in NID; optional */

	/* capture source */
	unsigned int num_mux_defs;
	const struct hda_input_mux *input_mux;
	unsigned int cur_mux[3];

	/* channel model */
	const struct hda_channel_mode *channel_mode;
	int num_channel_mode;
	int need_dac_fix;

	/* PCM information */
	struct hda_pcm pcm_rec[3];	/* used in alc_build_pcms() */

	/* 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;
	hda_nid_t private_dac_nids[5];

	/* hooks */
	void (*init_hook)(struct hda_codec *codec);
	void (*unsol_event)(struct hda_codec *codec, unsigned int res);

	/* for pin sensing */
	unsigned int sense_updated: 1;
	unsigned int jack_present: 1;
};

/*
 * configuration template - to be copied to the spec instance
 */
struct alc_config_preset {
	struct snd_kcontrol_new *mixers[5]; /* should be identical size
					     * with spec
					     */
	const struct hda_verb *init_verbs[5];
	unsigned int num_dacs;
	hda_nid_t *dac_nids;
	hda_nid_t dig_out_nid;		/* optional */
	hda_nid_t hp_nid;		/* optional */
	unsigned int num_adc_nids;
	hda_nid_t *adc_nids;
	hda_nid_t dig_in_nid;
	unsigned int num_channel_mode;
	const struct hda_channel_mode *channel_mode;
	int need_dac_fix;
	unsigned int num_mux_defs;
	const struct hda_input_mux *input_mux;
	void (*unsol_event)(struct hda_codec *, unsigned int);
	void (*init_hook)(struct hda_codec *);
};


/*
 * input MUX handling
 */
static int alc_mux_enum_info(struct snd_kcontrol *kcontrol,
			     struct snd_ctl_elem_info *uinfo)
{
	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
	struct alc_spec *spec = codec->spec;
	unsigned int mux_idx = snd_ctl_get_ioffidx(kcontrol, &uinfo->id);
	if (mux_idx >= spec->num_mux_defs)
		mux_idx = 0;
	return snd_hda_input_mux_info(&spec->input_mux[mux_idx], uinfo);
}

static int alc_mux_enum_get(struct snd_kcontrol *kcontrol,
			    struct snd_ctl_elem_value *ucontrol)
{
	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
	struct alc_spec *spec = codec->spec;
	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);

	ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
	return 0;
}

static int alc_mux_enum_put(struct snd_kcontrol *kcontrol,
			    struct snd_ctl_elem_value *ucontrol)
{
	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
	struct alc_spec *spec = codec->spec;
	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
	unsigned int mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
	return snd_hda_input_mux_put(codec, &spec->input_mux[mux_idx], ucontrol,
				     spec->adc_nids[adc_idx],
				     &spec->cur_mux[adc_idx]);
}


/*
 * channel mode setting
 */
static int alc_ch_mode_info(struct snd_kcontrol *kcontrol,
			    struct snd_ctl_elem_info *uinfo)
{
	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
	struct alc_spec *spec = codec->spec;
	return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode,
				    spec->num_channel_mode);
}

static int alc_ch_mode_get(struct snd_kcontrol *kcontrol,
			   struct snd_ctl_elem_value *ucontrol)
{
	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
	struct alc_spec *spec = codec->spec;
	return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
				   spec->num_channel_mode,
				   spec->multiout.max_channels);
}

static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
			   struct snd_ctl_elem_value *ucontrol)
{
	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
	struct alc_spec *spec = codec->spec;
	int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
				      spec->num_channel_mode,
				      &spec->multiout.max_channels);
	if (! err && spec->need_dac_fix)
		spec->multiout.num_dacs = spec->multiout.max_channels / 2;
	return err;
}

/*
 * Control the mode of pin widget settings via the mixer.  "pc" is used
 * instead of "%" to avoid consequences of accidently treating the % as 
 * being part of a format specifier.  Maximum allowed length of a value is
 * 63 characters plus NULL terminator.
 *
 * Note: some retasking pin complexes seem to ignore requests for input
 * states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these
 * are requested.  Therefore order this list so that this behaviour will not
 * cause problems when mixer clients move through the enum sequentially.
 * NIDs 0x0f and 0x10 have been observed to have this behaviour as of
 * March 2006.
 */
static char *alc_pin_mode_names[] = {
	"Mic 50pc bias", "Mic 80pc bias",
	"Line in", "Line out", "Headphone out",
};
static unsigned char alc_pin_mode_values[] = {
	PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP,
};
/* The control can present all 5 options, or it can limit the options based
 * in the pin being assumed to be exclusively an input or an output pin.  In
 * addition, "input" pins may or may not process the mic bias option
 * depending on actual widget capability (NIDs 0x0f and 0x10 don't seem to
 * accept requests for bias as of chip versions up to March 2006) and/or
 * wiring in the computer.
 */
#define ALC_PIN_DIR_IN              0x00
#define ALC_PIN_DIR_OUT             0x01
#define ALC_PIN_DIR_INOUT           0x02
#define ALC_PIN_DIR_IN_NOMICBIAS    0x03
#define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04

/* Info about the pin modes supported by the different pin direction modes. 
 * For each direction the minimum and maximum values are given.
 */
static signed char alc_pin_mode_dir_info[5][2] = {
	{ 0, 2 },    /* ALC_PIN_DIR_IN */
	{ 3, 4 },    /* ALC_PIN_DIR_OUT */
	{ 0, 4 },    /* ALC_PIN_DIR_INOUT */
	{ 2, 2 },    /* ALC_PIN_DIR_IN_NOMICBIAS */
	{ 2, 4 },    /* ALC_PIN_DIR_INOUT_NOMICBIAS */
};
#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0])
#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1])
#define alc_pin_mode_n_items(_dir) \
	(alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1)

static int alc_pin_mode_info(struct snd_kcontrol *kcontrol,
			     struct snd_ctl_elem_info *uinfo)
{
	unsigned int item_num = uinfo->value.enumerated.item;
	unsigned char dir = (kcontrol->private_value >> 16) & 0xff;

	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
	uinfo->count = 1;
	uinfo->value.enumerated.items = alc_pin_mode_n_items(dir);

	if (item_num<alc_pin_mode_min(dir) || item_num>alc_pin_mode_max(dir))
		item_num = alc_pin_mode_min(dir);
	strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]);
	return 0;
}

static int alc_pin_mode_get(struct snd_kcontrol *kcontrol,
			    struct snd_ctl_elem_value *ucontrol)
{
	unsigned int i;
	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
	hda_nid_t nid = kcontrol->private_value & 0xffff;
	unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
	long *valp = ucontrol->value.integer.value;
	unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
						 AC_VERB_GET_PIN_WIDGET_CONTROL,
						 0x00);

	/* Find enumerated value for current pinctl setting */
	i = alc_pin_mode_min(dir);
	while (alc_pin_mode_values[i] != pinctl && i <= alc_pin_mode_max(dir))
		i++;
	*valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
	return 0;
}

static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
			    struct snd_ctl_elem_value *ucontrol)
{
	signed int change;
	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
	hda_nid_t nid = kcontrol->private_value & 0xffff;
	unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
	long val = *ucontrol->value.integer.value;
	unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
						 AC_VERB_GET_PIN_WIDGET_CONTROL,
						 0x00);

	if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir)) 
		val = alc_pin_mode_min(dir);

	change = pinctl != alc_pin_mode_values[val];
	if (change) {
		/* Set pin mode to that requested */
		snd_hda_codec_write(codec,nid,0,AC_VERB_SET_PIN_WIDGET_CONTROL,
				    alc_pin_mode_values[val]);

		/* Also enable the retasking pin's input/output as required 
		 * for the requested pin mode.  Enum values of 2 or less are
		 * input modes.
		 *
		 * Dynamically switching the input/output buffers probably
		 * reduces noise slightly (particularly on input) so we'll
		 * do it.  However, having both input and output buffers
		 * enabled simultaneously doesn't seem to be problematic if
		 * this turns out to be necessary in the future.
		 */
		if (val <= 2) {
			snd_hda_codec_write(codec, nid, 0,
					    AC_VERB_SET_AMP_GAIN_MUTE,
					    AMP_OUT_MUTE);
			snd_hda_codec_write(codec, nid, 0,
					    AC_VERB_SET_AMP_GAIN_MUTE,
					    AMP_IN_UNMUTE(0));
		} else {
			snd_hda_codec_write(codec, nid, 0,
					    AC_VERB_SET_AMP_GAIN_MUTE,
					    AMP_IN_MUTE(0));
			snd_hda_codec_write(codec, nid, 0,
					    AC_VERB_SET_AMP_GAIN_MUTE,
					    AMP_OUT_UNMUTE);
		}
	}
	return change;
}

#define ALC_PIN_MODE(xname, nid, dir) \
	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0,  \
	  .info = alc_pin_mode_info, \
	  .get = alc_pin_mode_get, \
	  .put = alc_pin_mode_put, \
	  .private_value = nid | (dir<<16) }

/* A switch control for ALC260 GPIO pins.  Multiple GPIOs can be ganged
 * together using a mask with more than one bit set.  This control is
 * currently used only by the ALC260 test model.  At this stage they are not
 * needed for any "production" models.
 */
#ifdef CONFIG_SND_DEBUG
static int alc_gpio_data_info(struct snd_kcontrol *kcontrol,
			      struct snd_ctl_elem_info *uinfo)
{
	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
	uinfo->count = 1;
	uinfo->value.integer.min = 0;
	uinfo->value.integer.max = 1;
	return 0;
}                                
static int alc_gpio_data_get(struct snd_kcontrol *kcontrol,
			     struct snd_ctl_elem_value *ucontrol)
{
	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
	hda_nid_t nid = kcontrol->private_value & 0xffff;
	unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
	long *valp = ucontrol->value.integer.value;
	unsigned int val = snd_hda_codec_read(codec, nid, 0,
					      AC_VERB_GET_GPIO_DATA, 0x00);

	*valp = (val & mask) != 0;
	return 0;
}
static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
			     struct snd_ctl_elem_value *ucontrol)
{
	signed int change;
	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
	hda_nid_t nid = kcontrol->private_value & 0xffff;
	unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
	long val = *ucontrol->value.integer.value;
	unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0,
						    AC_VERB_GET_GPIO_DATA,
						    0x00);

	/* Set/unset the masked GPIO bit(s) as needed */
	change = (val == 0 ? 0 : mask) != (gpio_data & mask);
	if (val == 0)
		gpio_data &= ~mask;
	else
		gpio_data |= mask;
	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_GPIO_DATA, gpio_data);

	return change;
}
#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \
	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0,  \
	  .info = alc_gpio_data_info, \
	  .get = alc_gpio_data_get, \
	  .put = alc_gpio_data_put, \
	  .private_value = nid | (mask<<16) }
#endif   /* CONFIG_SND_DEBUG */

/* A switch control to allow the enabling of the digital IO pins on the
 * ALC260.  This is incredibly simplistic; the intention of this control is
 * to provide something in the test model allowing digital outputs to be
 * identified if present.  If models are found which can utilise these
 * outputs a more complete mixer control can be devised for those models if
 * necessary.
 */
#ifdef CONFIG_SND_DEBUG
static int alc_spdif_ctrl_info(struct snd_kcontrol *kcontrol,
			       struct snd_ctl_elem_info *uinfo)
{
	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
	uinfo->count = 1;
	uinfo->value.integer.min = 0;
	uinfo->value.integer.max = 1;
	return 0;
}                                
static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol,
			      struct snd_ctl_elem_value *ucontrol)
{
	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
	hda_nid_t nid = kcontrol->private_value & 0xffff;
	unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
	long *valp = ucontrol->value.integer.value;
	unsigned int val = snd_hda_codec_read(codec, nid, 0,
					      AC_VERB_GET_DIGI_CONVERT, 0x00);

	*valp = (val & mask) != 0;
	return 0;
}
static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol,
			      struct snd_ctl_elem_value *ucontrol)
{
	signed int change;
	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
	hda_nid_t nid = kcontrol->private_value & 0xffff;
	unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
	long val = *ucontrol->value.integer.value;
	unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
						    AC_VERB_GET_DIGI_CONVERT,
						    0x00);

	/* Set/unset the masked control bit(s) as needed */
	change = (val == 0 ? 0 : mask) != (ctrl_data & mask);
	if (val==0)
		ctrl_data &= ~mask;
	else
		ctrl_data |= mask;
	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
			    ctrl_data);

	return change;
}
#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \
	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0,  \
	  .info = alc_spdif_ctrl_info, \
	  .get = alc_spdif_ctrl_get, \
	  .put = alc_spdif_ctrl_put, \
	  .private_value = nid | (mask<<16) }
#endif   /* CONFIG_SND_DEBUG */

/*
 * set up from the preset table
 */
static void setup_preset(struct alc_spec *spec,
			 const struct alc_config_preset *preset)
{
	int i;

	for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
		spec->mixers[spec->num_mixers++] = preset->mixers[i];
	for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i];
	     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;

	spec->multiout.max_channels = spec->channel_mode[0].channels;

	spec->multiout.num_dacs = preset->num_dacs;
	spec->multiout.dac_nids = preset->dac_nids;
	spec->multiout.dig_out_nid = preset->dig_out_nid;
	spec->multiout.hp_nid = preset->hp_nid;
	
	spec->num_mux_defs = preset->num_mux_defs;
	if (! spec->num_mux_defs)
		spec->num_mux_defs = 1;
	spec->input_mux = preset->input_mux;

	spec->num_adc_nids = preset->num_adc_nids;
	spec->adc_nids = preset->adc_nids;
	spec->dig_in_nid = preset->dig_in_nid;

	spec->unsol_event = preset->unsol_event;
	spec->init_hook = preset->init_hook;
}

/*
 * ALC880 3-stack model
 *
 * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e)
 * Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18,
 *                 F-Mic = 0x1b, HP = 0x19
 */

static hda_nid_t alc880_dac_nids[4] = {
	/* front, rear, clfe, rear_surr */
	0x02, 0x05, 0x04, 0x03
};

static hda_nid_t alc880_adc_nids[3] = {
	/* ADC0-2 */
	0x07, 0x08, 0x09,
};

/* The datasheet says the node 0x07 is connected from inputs,
 * but it shows zero connection in the real implementation on some devices.
 * Note: this is a 915GAV bug, fixed on 915GLV
 */
static hda_nid_t alc880_adc_nids_alt[2] = {
	/* ADC1-2 */
	0x08, 0x09,
};

#define ALC880_DIGOUT_NID	0x06
#define ALC880_DIGIN_NID	0x0a

static struct hda_input_mux alc880_capture_source = {
	.num_items = 4,
	.items = {
		{ "Mic", 0x0 },
		{ "Front Mic", 0x3 },
		{ "Line", 0x2 },
		{ "CD", 0x4 },
	},
};

/* channel source setting (2/6 channel selection for 3-stack) */
/* 2ch mode */
static struct hda_verb alc880_threestack_ch2_init[] = {
	/* set line-in to input, mute it */
	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
	/* set mic-in to input vref 80%, mute it */
	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
	{ } /* end */
};

/* 6ch mode */
static struct hda_verb alc880_threestack_ch6_init[] = {
	/* set line-in to output, unmute it */
	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
	/* set mic-in to output, unmute it */
	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
	{ } /* end */
};

static struct hda_channel_mode alc880_threestack_modes[2] = {
	{ 2, alc880_threestack_ch2_init },
	{ 6, alc880_threestack_ch6_init },
};

static struct snd_kcontrol_new alc880_three_stack_mixer[] = {
	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
	HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
	{
		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
		.name = "Channel Mode",
		.info = alc_ch_mode_info,
		.get = alc_ch_mode_get,
		.put = alc_ch_mode_put,
	},
	{ } /* end */
};

/* capture mixer elements */
static struct snd_kcontrol_new alc880_capture_mixer[] = {
	HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
	HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
	HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
	HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
	{
		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
		/* The multiple "Capture Source" controls confuse alsamixer
		 * So call somewhat different..
		 * FIXME: the controls appear in the "playback" view!
		 */
		/* .name = "Capture Source", */
		.name = "Input Source",
		.count = 3,
		.info = alc_mux_enum_info,
		.get = alc_mux_enum_get,
		.put = alc_mux_enum_put,
	},
	{ } /* end */
};

/* capture mixer elements (in case NID 0x07 not available) */
static struct snd_kcontrol_new alc880_capture_alt_mixer[] = {
	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,
		/* The multiple "Capture Source" controls confuse alsamixer
		 * So call somewhat different..
		 * FIXME: the controls appear in the "playback" view!
		 */
		/* .name = "Capture Source", */
		.name = "Input Source",
		.count = 2,
		.info = alc_mux_enum_info,
		.get = alc_mux_enum_get,
		.put = alc_mux_enum_put,
	},
	{ } /* end */
};



/*
 * ALC880 5-stack model
 *
 * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d),
 *      Side = 0x02 (0xd)
 * Pin assignment: Front = 0x14, Surr = 0x17, CLFE = 0x16
 *                 Line-In/Side = 0x1a, Mic = 0x18, F-Mic = 0x1b, HP = 0x19
 */

/* additional mixers to alc880_three_stack_mixer */
static struct snd_kcontrol_new alc880_five_stack_mixer[] = {
	HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT),
	{ } /* end */
};

/* channel source setting (6/8 channel selection for 5-stack) */
/* 6ch mode */
static struct hda_verb alc880_fivestack_ch6_init[] = {
	/* set line-in to input, mute it */
	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
	{ } /* end */
};

/* 8ch mode */
static struct hda_verb alc880_fivestack_ch8_init[] = {
	/* set line-in to output, unmute it */
	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
	{ } /* end */
};

static struct hda_channel_mode alc880_fivestack_modes[2] = {
	{ 6, alc880_fivestack_ch6_init },
	{ 8, alc880_fivestack_ch8_init },
};


/*
 * ALC880 6-stack model
 *
 * DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e),
 *      Side = 0x05 (0x0f)
 * Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, Side = 0x17,
 *   Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b
 */

static hda_nid_t alc880_6st_dac_nids[4] = {
	/* front, rear, clfe, rear_surr */
	0x02, 0x03, 0x04, 0x05
};	

static struct hda_input_mux alc880_6stack_capture_source = {
	.num_items = 4,
	.items = {
		{ "Mic", 0x0 },
		{ "Front Mic", 0x1 },
		{ "Line", 0x2 },
		{ "CD", 0x4 },
	},
};

/* fixed 8-channels */
static struct hda_channel_mode alc880_sixstack_modes[1] = {
	{ 8, NULL },
};

static struct snd_kcontrol_new alc880_six_stack_mixer[] = {
	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
	HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
	{
		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
		.name = "Channel Mode",
		.info = alc_ch_mode_info,
		.get = alc_ch_mode_get,
		.put = alc_ch_mode_put,
	},
	{ } /* end */
};


/*
 * ALC880 W810 model
 *
 * W810 has rear IO for:
 * Front (DAC 02)
 * Surround (DAC 03)
 * Center/LFE (DAC 04)
 * Digital out (06)
 *
 * The system also has a pair of internal speakers, and a headphone jack.
 * These are both connected to Line2 on the codec, hence to DAC 02.
 * 
 * There is a variable resistor to control the speaker or headphone
 * volume. This is a hardware-only device without a software API.
 *
 * Plugging headphones in will disable the internal speakers. This is
 * implemented in hardware, not via the driver using jack sense. In
 * a similar fashion, plugging into the rear socket marked "front" will
 * disable both the speakers and headphones.
 *
 * For input, there's a microphone jack, and an "audio in" jack.
 * These may not do anything useful with this driver yet, because I
 * haven't setup any initialization verbs for these yet...
 */

static hda_nid_t alc880_w810_dac_nids[3] = {
	/* front, rear/surround, clfe */
	0x02, 0x03, 0x04
};

/* fixed 6 channels */
static struct hda_channel_mode alc880_w810_modes[1] = {
	{ 6, NULL }
};

/* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */
static struct snd_kcontrol_new alc880_w810_base_mixer[] = {
	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
	{ } /* end */
};


/*
 * Z710V model
 *
 * DAC: Front = 0x02 (0x0c), HP = 0x03 (0x0d)
 * Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?),
 *                 Line = 0x1a
 */

static hda_nid_t alc880_z71v_dac_nids[1] = {
	0x02
};
#define ALC880_Z71V_HP_DAC	0x03

/* fixed 2 channels */
static struct hda_channel_mode alc880_2_jack_modes[1] = {
	{ 2, NULL }
};

static struct snd_kcontrol_new alc880_z71v_mixer[] = {
	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
	{ } /* end */
};


/* FIXME! */
/*
 * ALC880 F1734 model
 *
 * DAC: HP = 0x02 (0x0c), Front = 0x03 (0x0d)
 * Pin assignment: HP = 0x14, Front = 0x15, Mic = 0x18
 */

static hda_nid_t alc880_f1734_dac_nids[1] = {
	0x03
};
#define ALC880_F1734_HP_DAC	0x02

static struct snd_kcontrol_new alc880_f1734_mixer[] = {
	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
	HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE("Internal Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
	{ } /* end */
};


/* FIXME! */
/*
 * ALC880 ASUS model
 *
 * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
 * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
 *  Mic = 0x18, Line = 0x1a
 */

#define alc880_asus_dac_nids	alc880_w810_dac_nids	/* identical with w810 */
#define alc880_asus_modes	alc880_threestack_modes	/* 2/6 channel mode */

static struct snd_kcontrol_new alc880_asus_mixer[] = {
	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
	{
		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
		.name = "Channel Mode",
		.info = alc_ch_mode_info,
		.get = alc_ch_mode_get,
		.put = alc_ch_mode_put,
	},
	{ } /* end */
};

/* FIXME! */
/*
 * ALC880 ASUS W1V model
 *
 * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
 * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
 *  Mic = 0x18, Line = 0x1a, Line2 = 0x1b
 */

/* additional mixers to alc880_asus_mixer */
static struct snd_kcontrol_new alc880_asus_w1v_mixer[] = {
	HDA_CODEC_VOLUME("Line2 Playback Volume", 0x0b, 0x03, HDA_INPUT),
	HDA_CODEC_MUTE("Line2 Playback Switch", 0x0b, 0x03, HDA_INPUT),
	{ } /* end */
};

/* additional mixers to alc880_asus_mixer */
static struct snd_kcontrol_new alc880_pcbeep_mixer[] = {
	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
	{ } /* end */
};

/* TCL S700 */
static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = {
	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
	HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
	HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
	HDA_CODEC_VOLUME("CD Playback Volume", 0x0B, 0x04, HDA_INPUT),
	HDA_CODEC_MUTE("CD Playback Switch", 0x0B, 0x04, HDA_INPUT),
	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0B, 0x0, HDA_INPUT),
	HDA_CODEC_MUTE("Mic Playback Switch", 0x0B, 0x0, HDA_INPUT),
	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
	{
		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
		/* The multiple "Capture Source" controls confuse alsamixer
		 * So call somewhat different..
		 * FIXME: the controls appear in the "playback" view!
		 */
		/* .name = "Capture Source", */
		.name = "Input Source",
		.count = 1,
		.info = alc_mux_enum_info,
		.get = alc_mux_enum_get,
		.put = alc_mux_enum_put,
	},
	{ } /* end */
};

/*
 * build control elements
 */
static int alc_build_controls(struct hda_codec *codec)
{
	struct alc_spec *spec = codec->spec;
	int err;
	int i;

	for (i = 0; i < spec->num_mixers; i++) {
		err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
		if (err < 0)
			return err;
	}

	if (spec->multiout.dig_out_nid) {
		err = snd_hda_create_spdif_out_ctls(codec,
						    spec->multiout.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);
		if (err < 0)
			return err;
	}
	return 0;
}


/*
 * initialize the codec volumes, etc
 */

/*
 * generic initialization of ADC, input mixers and output mixers
 */
static struct hda_verb alc880_volume_init_verbs[] = {
	/*
	 * Unmute ADC0-2 and set the default input to mic-in
	 */
	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},

	/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
	 * mixer widget
	 * Note: PASD motherboards uses the Line In 2 as the input for front
	 * panel mic (mic 2)
	 */
	/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},

	/*
	 * Set up output mixers (0x0c - 0x0f)
	 */
	/* set vol=0 to output mixers */
	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
	/* set up input amps for analog loopback */
	/* Amp Indices: DAC = 0, mixer = 1 */
	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},

	{ }
};

/*
 * 3-stack pin configuration:
 * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
 */
static struct hda_verb alc880_pin_3stack_init_verbs[] = {
	/*
	 * preset connection lists of input pins
	 * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
	 */
	{0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
	{0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
	{0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */

	/*
	 * Set pin mode and muting
	 */
	/* set front pin widgets 0x14 for output */
	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
	/* Mic1 (rear panel) pin widget for input and vref at 80% */
	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
	/* Mic2 (as headphone out) for HP output */
	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
	/* Line In pin widget for input */
	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
	/* Line2 (as front mic) pin widget for input and vref at 80% */
	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
	/* CD pin widget for input */
	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},

	{ }
};

/*
 * 5-stack pin configuration:
 * front = 0x14, surround = 0x17, clfe = 0x16, mic = 0x18, HP = 0x19,
 * line-in/side = 0x1a, f-mic = 0x1b
 */
static struct hda_verb alc880_pin_5stack_init_verbs[] = {
	/*
	 * preset connection lists of input pins
	 * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
	 */
	{0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
	{0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/side */

	/*
	 * Set pin mode and muting
	 */
	/* set pin widgets 0x14-0x17 for output */
	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
	/* unmute pins for output (no gain on this amp) */