/*
* 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"
#define ALC880_FRONT_EVENT 0x01
#define ALC880_DCVOL_EVENT 0x02
#define ALC880_HP_EVENT 0x04
#define ALC880_MIC_EVENT 0x08
/* 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_UNIWILL,
ALC880_UNIWILL_P53,
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_HIPPO,
ALC262_HIPPO_1,
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_TOSHIBA,
ALC861_ASUS,
ALC861_ASUS_LAPTOP,
ALC861_AUTO,
ALC861_MODEL_LAST,
};
/* ALC882 models */
enum {
ALC882_3ST_DIG,
ALC882_6ST_DIG,
ALC882_ARIMA,
ALC882_AUTO,
ALC885_MACPRO,
ALC882_MODEL_LAST,
};
/* ALC883 models */
enum {
ALC883_3ST_2ch_DIG,
ALC883_3ST_6ch_DIG,
ALC883_3ST_6ch,
ALC883_6ST_DIG,
ALC883_TARGA_DIG,
ALC883_TARGA_2ch_DIG,
ALC888_DEMO_BOARD,
ALC883_ACER,
ALC883_MEDION,
ALC883_LAPTOP_EAPD,
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 >= 0 && 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 */
};
/* Uniwill */
static struct snd_kcontrol_new alc880_uniwill_mixer[] = {
HDA_CODEC_VOLUME("HPhone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("HPhone Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("iSpeaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
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 */
};
static struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = {
HDA_CODEC_VOLUME("HPhone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("HPhone Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("iSpeaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
/*
* 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) */
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Mic2 (as headphone out) for HP output */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Line In pin widget for input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line2 (as front mic) pin widget for input and vref at 80% */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/*
* W810 pin configuration:
* front = 0x14, surround = 0x15, clfe = 0x16, HP = 0x1b
*/
static struct hda_verb alc880_pin_w810_init_verbs[] = {
/* hphone/speaker input selector: front DAC */
{0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{ }
};
/*
* Z71V pin configuration:
* Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?)
*/
static struct hda_verb alc880_pin_z71v_init_verbs[] = {
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/*
* 6-stack pin configuration:
* front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18,
* f-mic = 0x19, line = 0x1a, HP = 0x1b
*/
static struct hda_verb alc880_pin_6stack_init_verbs[] = {
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/*
* Uniwill pin configuration:
* HP = 0x14, InternalSpeaker = 0x15, mic = 0x18, internal mic = 0x19,
* line = 0x1a
*/
static struct hda_verb alc880_uniwill_init_verbs[] = {
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, */
/* {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{ }
};
/*
* Uniwill P53
* HP = 0x14, InternalSpeaker = 0x15, mic = 0x19,
*/
static struct hda_verb alc880_uniwill_p53_init_verbs[] = {
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_DCVOL_EVENT},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc880_uniwill_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_update(codec, 0x15, 0, HDA_OUTPUT, 0,
0x80, present ? 0x80 : 0);
snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
0x80, present ? 0x80 : 0);
snd_hda_codec_amp_update(codec, 0x16, 0, HDA_OUTPUT, 0,
0x80, present ? 0x80 : 0);
snd_hda_codec_amp_update(codec, 0x16, 1, HDA_OUTPUT, 0,
0x80, present ? 0x80 : 0);
present = snd_hda_codec_read(codec, 0x18, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
snd_hda_codec_write(codec, 0x0b, 0, AC_VERB_SET_AMP_GAIN_MUTE,
0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
}
static void alc880_uniwill_unsol_event(struct hda_codec *codec,
unsigned int res)
{
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
if ((res >> 28) == ALC880_HP_EVENT ||
(res >> 28) == ALC880_MIC_EVENT)
alc880_uniwill_automute(codec);
}
static void alc880_uniwill_p53_hp_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_update(codec, 0x15, 0, HDA_INPUT, 0,
0x80, present ? 0x80 : 0);
snd_hda_codec_amp_update(codec, 0x15, 1, HDA_INPUT, 0,
0x80, present ? 0x80 : 0);
}
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) & 0x7f;
snd_hda_codec_amp_update(codec, 0x0c, 0, HDA_OUTPUT, 0,
0x7f, present);
snd_hda_codec_amp_update(codec, 0x0c, 1, HDA_OUTPUT, 0,
0x7f, present);
snd_hda_codec_amp_update(codec, 0x0d, 0, HDA_OUTPUT, 0,
0x7f, present);
snd_hda_codec_amp_update(codec, 0x0d, 1, HDA_OUTPUT, 0,
0x7f, present);
}
static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
unsigned int res)
{
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
if ((res >> 28) == ALC880_HP_EVENT)
alc880_uniwill_p53_hp_automute(codec);
if ((res >> 28) == ALC880_DCVOL_EVENT)
alc880_uniwill_p53_dcvol_automute(codec);
}
/* FIXME! */
/*
* F1734 pin configuration:
* HP = 0x14, speaker-out = 0x15, mic = 0x18
*/
static struct hda_verb alc880_pin_f1734_init_verbs[] = {
{0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
{0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/* FIXME! */
/*
* ASUS pin configuration:
* HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a
*/
static struct hda_verb alc880_pin_asus_init_verbs[] = {
{0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
{0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/* Enable GPIO mask and set output */
static struct hda_verb alc880_gpio1_init_verbs[] = {
{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
{ }
};
/* Enable GPIO mask and set output */
static struct hda_verb alc880_gpio2_init_verbs[] = {
{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
{0x01, AC_VERB_SET_GPIO_DATA, 0x02},
{ }
};
/* Clevo m520g init */
static struct hda_verb alc880_pin_clevo_init_verbs[] = {
/* headphone output */
{0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
/* line-out */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Line-in */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* CD */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mic1 (rear panel) */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mic2 (front panel) */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* headphone */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* change to EAPD mode */
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3060},
{ }
};
static struct hda_verb alc880_pin_tcl_S700_init_verbs[] = {
/* change to EAPD mode */
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3060},
/* Headphone output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Front output*/
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Line In pin widget for input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
/* change to EAPD mode */
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3070},
{ }
};
/*
* LG m1 express dual
*
* Pin assignment:
* Rear Line-In/Out (blue): 0x14
* Build-in Mic-In: 0x15
* Speaker-out: 0x17
* HP-Out (green): 0x1b
* Mic-In/Out (red): 0x19
* SPDIF-Out: 0x1e
*/
/* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */
static hda_nid_t alc880_lg_dac_nids[3] = {
0x05, 0x02, 0x03
};
/* seems analog CD is not working */
static struct hda_input_mux alc880_lg_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x1 },
{ "Line", 0x5 },
{ "Internal Mic", 0x6 },
},
};
/* 2,4,6 channel modes */
static struct hda_verb alc880_lg_ch2_init[] = {
/* set line-in and mic-in to input */
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ }
};
static struct hda_verb alc880_lg_ch4_init[] = {
/* set line-in to out and mic-in to input */
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ }
};
static struct hda_verb alc880_lg_ch6_init[] = {
/* set line-in and mic-in to output */
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
{ }
};
static struct hda_channel_mode alc880_lg_ch_modes[3] = {
{ 2, alc880_lg_ch2_init },
{ 4, alc880_lg_ch4_init },
{ 6, alc880_lg_ch6_init },
};
static struct snd_kcontrol_new alc880_lg_mixer[] = {
/* FIXME: it's not really "master" but front channels */
HDA_CODEC_VOLUME("Master Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x06, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x06, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x07, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x07, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
static struct hda_verb alc880_lg_init_verbs[] = {
/* set capture source to mic-in */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* mute all amp mixer inputs */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)},
/* line-in to input */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* built-in mic */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* speaker-out */
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* mic-in to input */
{0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* HP-out */
{0x13, AC_VERB_SET_CONNECT_SEL, 0x03},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* jack sense */
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc880_lg_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_update(codec, 0x17, 0, HDA_OUTPUT, 0,
0x80, present ? 0x80 : 0);
snd_hda_codec_amp_update(codec, 0x17, 1, HDA_OUTPUT, 0,
0x80, present ? 0x80 : 0);
}
static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res)
{
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
if ((res >> 28) == 0x01)
alc880_lg_automute(codec);
}
/*
* LG LW20
*
* Pin assignment:
* Speaker-out: 0x14
* Mic-In: 0x18
* Built-in Mic-In: 0x19 (?)
* HP-Out: 0x1b
* SPDIF-Out: 0x1e
*/
/* seems analog CD is not working */
static struct hda_input_mux alc880_lg_lw_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x0 },
{ "Internal Mic", 0x1 },
},
};
static struct snd_kcontrol_new alc880_lg_lw_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
{ } /* end */
};
static struct hda_verb alc880_lg_lw_init_verbs[] = {
/* set capture source to mic-in */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)},
/* speaker-out */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* HP-out */
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* mic-in to input */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* built-in mic */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* jack sense */
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc880_lg_lw_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_update(codec, 0x14, 0, HDA_OUTPUT, 0,
0x80, present ? 0x80 : 0);
snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
0x80, present ? 0x80 : 0);
}
static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res)
{
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
if ((res >> 28) == 0x01)
alc880_lg_lw_automute(codec);
}
/*
* Common callbacks
*/
static int alc_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int i;
for (i = 0; i < spec->num_init_verbs; i++)
snd_hda_sequence_write(codec, spec->init_verbs[i]);
if (spec->init_hook)
spec->init_hook(codec);
return 0;
}
static void alc_unsol_event(struct hda_codec *codec, unsigned int res)
{
struct alc_spec *spec = codec->spec;
if (spec->unsol_event)
spec->unsol_event(codec, res);
}
#ifdef CONFIG_PM
/*
* resume
*/
static int alc_resume(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
alc_init(codec);
for (i = 0; i < spec->num_mixers; i++)
snd_hda_resume_ctls(codec, spec->mixers[i]);
if (spec->multiout.dig_out_nid)
snd_hda_resume_spdif_out(codec);
if (spec->dig_in_nid)
snd_hda_resume_spdif_in(codec);
return 0;
}
#endif
/*
* Analog playback callbacks
*/
static int alc880_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
}
static int alc880_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 alc_spec *spec = codec->spec;
return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
stream_tag, format, substream);
}
static int alc880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
}
/*
* Digital out
*/
static int alc880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_dig_open(codec, &spec->multiout);
}
static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_dig_close(codec, &spec->multiout);
}
/*
* Analog capture
*/
static int alc880_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
stream_tag, 0, format);
return 0;
}
static int alc880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
0, 0, 0);
return 0;
}
/*
*/
static struct hda_pcm_stream alc880_pcm_analog_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 8,
/* NID is set in alc_build_pcms */
.ops = {
.open = alc880_playback_pcm_open,
.prepare = alc880_playback_pcm_prepare,
.cleanup = alc880_playback_pcm_cleanup
},
};
static struct hda_pcm_stream alc880_pcm_analog_capture = {
.substreams = 2,
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
.ops = {
.prepare = alc880_capture_pcm_prepare,
.cleanup = alc880_capture_pcm_cleanup
},
};
static struct hda_pcm_stream alc880_pcm_digital_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
.ops = {
.open = alc880_dig_playback_pcm_open,
.close = alc880_dig_playback_pcm_close
},
};
static struct hda_pcm_stream alc880_pcm_digital_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
};
/* Used by alc_build_pcms to flag that a PCM has no playback stream */
static struct hda_pcm_stream alc_pcm_null_playback = {
.substreams = 0,
.channels_min = 0,
.channels_max = 0,
};
static int alc_build_pcms(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct hda_pcm *info = spec->pcm_rec;
int i;
codec->num_pcms = 1;
codec->pcm_info = info;
info->name = spec->stream_name_analog;
if (spec->stream_analog_playback) {
snd_assert(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);
info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
}
if (spec->channel_mode) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = 0;
for (i = 0; i < spec->num_channel_mode; i++) {
if (spec->channel_mode[i].channels > info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->channel_mode[i].channels;
}
}
}
/* SPDIF for stream index #1 */
if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
codec->num_pcms = 2;
info = spec->pcm_rec + 1;
info->name = spec->stream_name_digital;
if (spec->multiout.dig_out_nid &&
spec->stream_digital_playback) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback);
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
}
if (spec->dig_in_nid &&
spec->stream_digital_capture) {
info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_digital_capture);
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
}
}
/* If the use of more than one ADC is requested for the current
* model, configure a second analog capture-only PCM.
*/
/* Additional Analaog capture for index #2 */
if (spec->num_adc_nids > 1 && spec->stream_analog_capture &&
spec->adc_nids) {
codec->num_pcms = 3;
info = spec->pcm_rec + 2;
info->name = spec->stream_name_analog;
/* No playback stream for second PCM */
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = alc_pcm_null_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0;
if (spec->stream_analog_capture) {
info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[1];
}
}
return 0;
}
static void alc_free(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int i;
if (! spec)
return;
if (spec->kctl_alloc) {
for (i = 0; i < spec->num_kctl_used; i++)
kfree(spec->kctl_alloc[i].name);
kfree(spec->kctl_alloc);
}
kfree(spec);
}
/*
*/
static struct hda_codec_ops alc_patch_ops = {
.build_controls = alc_build_controls,
.build_pcms = alc_build_pcms,
.init = alc_init,
.free = alc_free,
.unsol_event = alc_unsol_event,
#ifdef CONFIG_PM
.resume = alc_resume,
#endif
};
/*
* Test configuration for debugging
*
* Almost all inputs/outputs are enabled. I/O pins can be configured via
* enum controls.
*/
#ifdef CONFIG_SND_DEBUG
static hda_nid_t alc880_test_dac_nids[4] = {
0x02, 0x03, 0x04, 0x05
};
static struct hda_input_mux alc880_test_capture_source = {
.num_items = 7,
.items = {
{ "In-1", 0x0 },
{ "In-2", 0x1 },
{ "In-3", 0x2 },
{ "In-4", 0x3 },
{ "CD", 0x4 },
{ "Front", 0x5 },
{ "Surround", 0x6 },
},
};
static struct hda_channel_mode alc880_test_modes[4] = {
{ 2, NULL },
{ 4, NULL },
{ 6, NULL },
{ 8, NULL },
};
static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = {
"N/A", "Line Out", "HP Out",
"In Hi-Z", "In 50%", "In Grd", "In 80%", "In 100%"
};
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 8;
if (uinfo->value.enumerated.item >= 8)
uinfo->value.enumerated.item = 7;
strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
return 0;
}
static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
unsigned int pin_ctl, item = 0;
pin_ctl = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
if (pin_ctl & AC_PINCTL_OUT_EN) {
if (pin_ctl & AC_PINCTL_HP_EN)
item = 2;
else
item = 1;
} else if (pin_ctl & AC_PINCTL_IN_EN) {
switch (pin_ctl & AC_PINCTL_VREFEN) {
case AC_PINCTL_VREF_HIZ: item = 3; break;
case AC_PINCTL_VREF_50: item = 4; break;
case AC_PINCTL_VREF_GRD: item = 5; break;
case AC_PINCTL_VREF_80: item = 6; break;
case AC_PINCTL_VREF_100: item = 7; break;
}
}
ucontrol->value.enumerated.item[0] = item;
return 0;
}
static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
static unsigned int ctls[] = {
0, AC_PINCTL_OUT_EN, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_50,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_80,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_100,
};
unsigned int old_ctl, new_ctl;
old_ctl = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
new_ctl = ctls[ucontrol->value.enumerated.item[0]];
if (old_ctl != new_ctl) {
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, new_ctl);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
(ucontrol->value.enumerated.item[0] >= 3 ?
0xb080 : 0xb000));
return 1;
}
return 0;
}
static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = {
"Front", "Surround", "CLFE", "Side"
};
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 4;
if (uinfo->value.enumerated.item >= 4)
uinfo->value.enumerated.item = 3;
strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
return 0;
}
static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
unsigned int sel;
sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0);
ucontrol->value.enumerated.item[0] = sel & 3;
return 0;
}
static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
unsigned int sel;
sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3;
if (ucontrol->value.enumerated.item[0] != sel) {
sel = ucontrol->value.enumerated.item[0] & 3;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, sel);
return 1;
}
return 0;
}
#define PIN_CTL_TEST(xname,nid) { \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
.info = alc_test_pin_ctl_info, \
.get = alc_test_pin_ctl_get, \
.put = alc_test_pin_ctl_put, \
.private_value = nid \
}
#define PIN_SRC_TEST(xname,nid) { \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
.info = alc_test_pin_src_info, \
.get = alc_test_pin_src_get, \
.put = alc_test_pin_src_put, \
.private_value = nid \
}
static struct snd_kcontrol_new alc880_test_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CLFE Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_BIND_MUTE("CLFE Playback Switch", 0x0e, 2, HDA_INPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
PIN_CTL_TEST("Front Pin Mode", 0x14),
PIN_CTL_TEST("Surround Pin Mode", 0x15),
PIN_CTL_TEST("CLFE Pin Mode", 0x16),
PIN_CTL_TEST("Side Pin Mode", 0x17),
PIN_CTL_TEST("In-1 Pin Mode", 0x18),
PIN_CTL_TEST("In-2 Pin Mode", 0x19),
PIN_CTL_TEST("In-3 Pin Mode", 0x1a),
PIN_CTL_TEST("In-4 Pin Mode", 0x1b),
PIN_SRC_TEST("In-1 Pin Source", 0x18),
PIN_SRC_TEST("In-2 Pin Source", 0x19),
PIN_SRC_TEST("In-3 Pin Source", 0x1a),
PIN_SRC_TEST("In-4 Pin Source", 0x1b),
HDA_CODEC_VOLUME("In-1 Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("In-1 Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("In-2 Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("In-2 Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("In-3 Playback Volume", 0x0b, 0x2, HDA_INPUT),
HDA_CODEC_MUTE("In-3 Playback Switch", 0x0b, 0x2, HDA_INPUT),
HDA_CODEC_VOLUME("In-4 Playback Volume", 0x0b, 0x3, HDA_INPUT),
HDA_CODEC_MUTE("In-4 Playback Switch", 0x0b, 0x3, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x4, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x4, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
static struct hda_verb alc880_test_init_verbs[] = {
/* Unmute inputs of 0x0c - 0x0f */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Vol output for 0x0c-0x0f */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Set output pins 0x14-0x17 */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* Unmute output pins 0x14-0x17 */
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Set input pins 0x18-0x1c */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Mute input pins 0x18-0x1b */
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* ADC set up */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Analog input/passthru */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
{ }
};
#endif
/*
*/
static const char *alc880_models[ALC880_MODEL_LAST] = {
[ALC880_3ST] = "3stack",
[ALC880_TCL_S700] = "tcl",
[ALC880_3ST_DIG] = "3stack-digout",
[ALC880_CLEVO] = "clevo",
[ALC880_5ST] = "5stack",
[ALC880_5ST_DIG] = "5stack-digout",
[ALC880_W810] = "w810",
[ALC880_Z71V] = "z71v",
[ALC880_6ST] = "6stack",
[ALC880_6ST_DIG] = "6stack-digout",
[ALC880_ASUS] = "asus",
[ALC880_ASUS_W1V] = "asus-w1v",
[ALC880_ASUS_DIG] = "asus-dig",
[ALC880_ASUS_DIG2] = "asus-dig2",
[ALC880_UNIWILL_DIG] = "uniwill",
[ALC880_F1734] = "F1734",
[ALC880_LG] = "lg",
[ALC880_LG_LW] = "lg-lw",
#ifdef CONFIG_SND_DEBUG
[ALC880_TEST] = "test",
#endif
[ALC880_AUTO] = "auto",
};
static struct snd_pci_quirk alc880_cfg_tbl[] = {
/* Broken BIOS configuration */
SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG),
SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_6ST),
SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_W810),
SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_3ST_DIG),
SND_PCI_QUIRK(0x1025, 0x0077, "ULI", ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1025, 0x0078, "ULI", ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1025, 0x0087, "ULI", ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG),
SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST),
SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG),
SND_PCI_QUIRK(0x103c, 0x2a09, "HP", ALC880_5ST),
SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V),
SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x1113, "ASUS", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x1123, "ASUS", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x1173, "ASUS", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_Z71V),
/* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */
SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x814e, "ASUS", ALC880_5ST),
SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST),
SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST),
SND_PCI_QUIRK(0x1043, 0, "ASUS", ALC880_ASUS),
SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST),
SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST),
SND_PCI_QUIRK(0x107b, 0x3033, "Gateway", ALC880_5ST),
SND_PCI_QUIRK(0x107b, 0x4039, "Gateway", ALC880_5ST),
SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST),
SND_PCI_QUIRK(0x1558, 0x0520, "Clevo m520G", ALC880_CLEVO),
SND_PCI_QUIRK(0x1558, 0x0660, "Clevo m655n", ALC880_CLEVO),
SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810),
SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700),
SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG),
SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1297, 0xc790, "Shuttle ST20G5", ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2),
SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG),
SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL),
SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53),
SND_PCI_QUIRK(0x1584, 0x9054, "Uniwlll", ALC880_F1734),
SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL),
SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734),
SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG),
SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_LG),
SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW),
SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_LG_LW),
SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_3ST_DIG),
SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_3ST_DIG),
SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_3ST_DIG),
SND_PCI_QUIRK(0x8086, 0xd400, "Intel mobo", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x8086, 0xd401, "Intel mobo", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x8086, 0xe224, "Intel mobo", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x8086, 0, "Intel mobo", ALC880_3ST),
{}
};
/*
* ALC880 codec presets
*/
static struct alc_config_preset alc880_presets[] = {
[ALC880_3ST] = {
.mixers = { alc880_three_stack_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_dac_nids),
.dac_nids = alc880_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
.channel_mode = alc880_threestack_modes,
.need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_3ST_DIG] = {
.mixers = { alc880_three_stack_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_dac_nids),
.dac_nids = alc880_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
.channel_mode = alc880_threestack_modes,
.need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_TCL_S700] = {
.mixers = { alc880_tcl_s700_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_pin_tcl_S700_init_verbs,
alc880_gpio2_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_dac_nids),
.dac_nids = alc880_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_5ST] = {
.mixers = { alc880_three_stack_mixer, alc880_five_stack_mixer},
.init_verbs = { alc880_volume_init_verbs, alc880_pin_5stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_dac_nids),
.dac_nids = alc880_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
.channel_mode = alc880_fivestack_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_5ST_DIG] = {
.mixers = { alc880_three_stack_mixer, alc880_five_stack_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_5stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_dac_nids),
.dac_nids = alc880_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
.channel_mode = alc880_fivestack_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_6ST] = {
.mixers = { alc880_six_stack_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_6stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
.dac_nids = alc880_6st_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
.channel_mode = alc880_sixstack_modes,
.input_mux = &alc880_6stack_capture_source,
},
[ALC880_6ST_DIG] = {
.mixers = { alc880_six_stack_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_6stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
.dac_nids = alc880_6st_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
.channel_mode = alc880_sixstack_modes,
.input_mux = &alc880_6stack_capture_source,
},
[ALC880_W810] = {
.mixers = { alc880_w810_base_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_w810_init_verbs,
alc880_gpio2_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_w810_dac_nids),
.dac_nids = alc880_w810_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_w810_modes),
.channel_mode = alc880_w810_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_Z71V] = {
.mixers = { alc880_z71v_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_z71v_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_z71v_dac_nids),
.dac_nids = alc880_z71v_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_F1734] = {
.mixers = { alc880_f1734_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_f1734_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_f1734_dac_nids),
.dac_nids = alc880_f1734_dac_nids,
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_ASUS] = {
.mixers = { alc880_asus_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
.dac_nids = alc880_asus_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
.channel_mode = alc880_asus_modes,
.need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_ASUS_DIG] = {
.mixers = { alc880_asus_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
.dac_nids = alc880_asus_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
.channel_mode = alc880_asus_modes,
.need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_ASUS_DIG2] = {
.mixers = { alc880_asus_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs,
alc880_gpio2_init_verbs }, /* use GPIO2 */
.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
.dac_nids = alc880_asus_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
.channel_mode = alc880_asus_modes,
.need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_ASUS_W1V] = {
.mixers = { alc880_asus_mixer, alc880_asus_w1v_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
.dac_nids = alc880_asus_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
.channel_mode = alc880_asus_modes,
.need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_UNIWILL_DIG] = {
.mixers = { alc880_asus_mixer, alc880_pcbeep_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_pin_asus_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
.dac_nids = alc880_asus_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
.channel_mode = alc880_asus_modes,
.need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_UNIWILL] = {
.mixers = { alc880_uniwill_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_uniwill_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
.dac_nids = alc880_asus_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
.channel_mode = alc880_threestack_modes,
.need_dac_fix = 1,
.input_mux = &alc880_capture_source,
.unsol_event = alc880_uniwill_unsol_event,
.init_hook = alc880_uniwill_automute,
},
[ALC880_UNIWILL_P53] = {
.mixers = { alc880_uniwill_p53_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_uniwill_p53_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
.dac_nids = alc880_asus_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc880_w810_modes),
.channel_mode = alc880_w810_modes,
.input_mux = &alc880_capture_source,
.unsol_event = alc880_uniwill_p53_unsol_event,
.init_hook = alc880_uniwill_p53_hp_automute,
},
[ALC880_CLEVO] = {
.mixers = { alc880_three_stack_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_pin_clevo_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_dac_nids),
.dac_nids = alc880_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
.channel_mode = alc880_threestack_modes,
.need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_LG] = {
.mixers = { alc880_lg_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_lg_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_lg_dac_nids),
.dac_nids = alc880_lg_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes),
.channel_mode = alc880_lg_ch_modes,
.need_dac_fix = 1,
.input_mux = &alc880_lg_capture_source,
.unsol_event = alc880_lg_unsol_event,
.init_hook = alc880_lg_automute,
},
[ALC880_LG_LW] = {
.mixers = { alc880_lg_lw_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_lg_lw_init_verbs },
.num_dacs = 1,
.dac_nids = alc880_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_lg_lw_capture_source,
.unsol_event = alc880_lg_lw_unsol_event,
.init_hook = alc880_lg_lw_automute,
},
#ifdef CONFIG_SND_DEBUG
[ALC880_TEST] = {
.mixers = { alc880_test_mixer },
.init_verbs = { alc880_test_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_test_dac_nids),
.dac_nids = alc880_test_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_test_modes),
.channel_mode = alc880_test_modes,
.input_mux = &alc880_test_capture_source,
},
#endif
};
/*
* Automatic parse of I/O pins from the BIOS configuration
*/
#define NUM_CONTROL_ALLOC 32
#define NUM_VERB_ALLOC 32
enum {
ALC_CTL_WIDGET_VOL,
ALC_CTL_WIDGET_MUTE,
ALC_CTL_BIND_MUTE,
};
static struct snd_kcontrol_new alc880_control_templates[] = {
HDA_CODEC_VOLUME(NULL, 0, 0, 0),
HDA_CODEC_MUTE(NULL, 0, 0, 0),
HDA_BIND_MUTE(NULL, 0, 0, 0),
};
/* add dynamic controls */
static int add_control(struct alc_spec *spec, int type, const char *name, unsigned long val)
{
struct snd_kcontrol_new *knew;
if (spec->num_kctl_used >= spec->num_kctl_alloc) {
int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC;
knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); /* array + terminator */
if (! knew)
return -ENOMEM;
if (spec->kctl_alloc) {
memcpy(knew, spec->kctl_alloc, sizeof(*knew) * spec->num_kctl_alloc);
kfree(spec->kctl_alloc);
}
spec->kctl_alloc = knew;
spec->num_kctl_alloc = num;
}
knew = &spec->kctl_alloc[spec->num_kctl_used];
*knew = alc880_control_templates[type];
knew->name = kstrdup(name, GFP_KERNEL);
if (! knew->name)
return -ENOMEM;
knew->private_value = val;
spec->num_kctl_used++;
return 0;
}
#define alc880_is_fixed_pin(nid) ((nid) >= 0x14 && (nid) <= 0x17)
#define alc880_fixed_pin_idx(nid) ((nid) - 0x14)
#define alc880_is_multi_pin(nid) ((nid) >= 0x18)
#define alc880_multi_pin_idx(nid) ((nid) - 0x18)
#define alc880_is_input_pin(nid) ((nid) >= 0x18)
#define alc880_input_pin_idx(nid) ((nid) - 0x18)
#define alc880_idx_to_dac(nid) ((nid) + 0x02)
#define alc880_dac_to_idx(nid) ((nid) - 0x02)
#define alc880_idx_to_mixer(nid) ((nid) + 0x0c)
#define alc880_idx_to_selector(nid) ((nid) + 0x10)
#define ALC880_PIN_CD_NID 0x1c
/* fill in the dac_nids table from the parsed pin configuration */
static int alc880_auto_fill_dac_nids(struct alc_spec *spec, const struct auto_pin_cfg *cfg)
{
hda_nid_t nid;
int assigned[4];
int i, j;
memset(assigned, 0, sizeof(assigned));
spec->multiout.dac_nids = spec->private_dac_nids;
/* check the pins hardwired to audio widget */
for (i = 0; i < cfg->line_outs; i++) {
nid = cfg->line_out_pins[i];
if (alc880_is_fixed_pin(nid)) {
int idx = alc880_fixed_pin_idx(nid);
spec->multiout.dac_nids[i] = alc880_idx_to_dac(idx);
assigned[idx] = 1;
}
}
/* left pins can be connect to any audio widget */
for (i = 0; i < cfg->line_outs; i++) {
nid = cfg->line_out_pins[i];
if (alc880_is_fixed_pin(nid))
continue;
/* search for an empty channel */
for (j = 0; j < cfg->line_outs; j++) {
if (! assigned[j]) {
spec->multiout.dac_nids[i] = alc880_idx_to_dac(j);
assigned[j] = 1;
break;
}
}
}
spec->multiout.num_dacs = cfg->line_outs;
return 0;
}
/* add playback controls from the parsed DAC table */
static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
{
char name[32];
static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" };
hda_nid_t nid;
int i, err;
for (i = 0; i < cfg->line_outs; i++) {
if (! spec->multiout.dac_nids[i])
continue;
nid = alc880_idx_to_mixer(alc880_dac_to_idx(spec->multiout.dac_nids[i]));
if (i == 2) {
/* Center/LFE */
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Center Playback Volume",
HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0)
return err;
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "LFE Playback Volume",
HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0)
return err;
if ((err = add_control(spec, ALC_CTL_BIND_MUTE, "Center Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 1, 2, HDA_INPUT))) < 0)
return err;
if ((err = add_control(spec, ALC_CTL_BIND_MUTE, "LFE Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 2, 2, HDA_INPUT))) < 0)
return err;
} else {
sprintf(name, "%s Playback Volume", chname[i]);
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
return err;
sprintf(name, "%s Playback Switch", chname[i]);
if ((err = add_control(spec, ALC_CTL_BIND_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT))) < 0)
return err;
}
}
return 0;
}
/* add playback controls for speaker and HP outputs */
static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
const char *pfx)
{
hda_nid_t nid;
int err;
char name[32];
if (! pin)
return 0;
if (alc880_is_fixed_pin(pin)) {
nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
/* specify the DAC as the extra output */
if (! spec->multiout.hp_nid)
spec->multiout.hp_nid = nid;
else
spec->multiout.extra_out_nid[0] = nid;
/* control HP volume/switch on the output mixer amp */
nid = alc880_idx_to_mixer(alc880_fixed_pin_idx(pin));
sprintf(name, "%s Playback Volume", pfx);
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
return err;
sprintf(name, "%s Playback Switch", pfx);
if ((err = add_control(spec, ALC_CTL_BIND_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT))) < 0)
return err;
} else if (alc880_is_multi_pin(pin)) {
/* set manual connection */
/* we have only a switch on HP-out PIN */
sprintf(name, "%s Playback Switch", pfx);
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT))) < 0)
return err;
}
return 0;
}
/* create input playback/capture controls for the given pin */
static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, const char *ctlname,
int idx, hda_nid_t mix_nid)
{
char name[32];
int err;
sprintf(name, "%s Playback Volume", ctlname);
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT))) < 0)
return err;
sprintf(name, "%s Playback Switch", ctlname);
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT))) < 0)
return err;
return 0;
}
/* create playback/capture controls for input pins */
static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
{
struct hda_input_mux *imux = &spec->private_imux;
int i, err, idx;
for (i = 0; i < AUTO_PIN_LAST; i++) {
if (alc880_is_input_pin(cfg->input_pins[i])) {
idx = alc880_input_pin_idx(cfg->input_pins[i]);
err = new_analog_input(spec, cfg->input_pins[i],
auto_pin_cfg_labels[i],
idx, 0x0b);
if (err < 0)
return err;
imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
imux->items[imux->num_items].index = alc880_input_pin_idx(cfg->input_pins[i]);
imux->num_items++;
}
}
return 0;
}
static void alc880_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
int dac_idx)
{
/* set as output */
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
/* need the manual connection? */
if (alc880_is_multi_pin(nid)) {
struct alc_spec *spec = codec->spec;
int idx = alc880_multi_pin_idx(nid);
snd_hda_codec_write(codec, alc880_idx_to_selector(idx), 0,
AC_VERB_SET_CONNECT_SEL,
alc880_dac_to_idx(spec->multiout.dac_nids[dac_idx]));
}
}
static void alc880_auto_init_multi_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < spec->autocfg.line_outs; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
alc880_auto_set_output_and_unmute(codec, nid, PIN_OUT, i);
}
}
static void alc880_auto_init_extra_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t pin;
pin = spec->autocfg.speaker_pins[0];
if (pin) /* connect to front */
alc880_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
pin = spec->autocfg.hp_pins[0];
if (pin) /* connect to front */
alc880_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
}
static void alc880_auto_init_analog_input(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = spec->autocfg.input_pins[i];
if (alc880_is_input_pin(nid)) {
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
i <= AUTO_PIN_FRONT_MIC ? PIN_VREF80 : PIN_IN);
if (nid != ALC880_PIN_CD_NID)
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_MUTE);
}
}
}
/* parse the BIOS configuration and set up the alc_spec */
/* return 1 if successful, 0 if the proper config is not found, or a negative error code */
static int alc880_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err;
static hda_nid_t alc880_ignore[] = { 0x1d, 0 };
if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc880_ignore)) < 0)
return err;
if (! spec->autocfg.line_outs)
return 0; /* can't find valid BIOS pin config */
if ((err = alc880_auto_fill_dac_nids(spec, &spec->autocfg)) < 0 ||
(err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 ||
(err = alc880_auto_create_extra_out(spec,
spec->autocfg.speaker_pins[0],
"Speaker")) < 0 ||
(err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
"Headphone")) < 0 ||
(err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
if (spec->autocfg.dig_out_pin)
spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
if (spec->autocfg.dig_in_pin)
spec->dig_in_nid = ALC880_DIGIN_NID;
if (spec->kctl_alloc)
spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
spec->init_verbs[spec->num_init_verbs++] = alc880_volume_init_verbs;
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux;
return 1;
}
/* additional initialization for auto-configuration model */
static void alc880_auto_init(struct hda_codec *codec)
{
alc880_auto_init_multi_out(codec);
alc880_auto_init_extra_out(codec);
alc880_auto_init_analog_input(codec);
}
/*
* OK, here we have finally the patch for ALC880
*/
static int patch_alc880(struct hda_codec *codec)
{
struct alc_spec *spec;
int board_config;
int err;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
board_config = snd_hda_check_board_config(codec, ALC880_MODEL_LAST,
alc880_models,
alc880_cfg_tbl);
if (board_config < 0) {
printk(KERN_INFO "hda_codec: Unknown model for ALC880, "
"trying auto-probe from BIOS...\n");
board_config = ALC880_AUTO;
}
if (board_config == ALC880_AUTO) {
/* automatic parse from the BIOS config */
err = alc880_parse_auto_config(codec);
if (err < 0) {
alc_free(codec);
return err;
} else if (! err) {
printk(KERN_INFO
"hda_codec: Cannot set up configuration "
"from BIOS. Using 3-stack mode...\n");
board_config = ALC880_3ST;
}
}
if (board_config != ALC880_AUTO)
setup_preset(spec, &alc880_presets[board_config]);
spec->stream_name_analog = "ALC880 Analog";
spec->stream_analog_playback = &alc880_pcm_analog_playback;
spec->stream_analog_capture = &alc880_pcm_analog_capture;
spec->stream_name_digital = "ALC880 Digital";
spec->stream_digital_playback = &alc880_pcm_digital_playback;
spec->stream_digital_capture = &alc880_pcm_digital_capture;
if (! spec->adc_nids && spec->input_mux) {
/* check whether NID 0x07 is valid */
unsigned int wcap = get_wcaps(codec, alc880_adc_nids[0]);
wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */
if (wcap != AC_WID_AUD_IN) {
spec->adc_nids = alc880_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids_alt);
spec->mixers[spec->num_mixers] = alc880_capture_alt_mixer;
spec->num_mixers++;
} else {
spec->adc_nids = alc880_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids);
spec->mixers[spec->num_mixers] = alc880_capture_mixer;
spec->num_mixers++;
}
}
codec->patch_ops = alc_patch_ops;
if (board_config == ALC880_AUTO)
spec->init_hook = alc880_auto_init;
return 0;
}
/*
* ALC260 support
*/
static hda_nid_t alc260_dac_nids[1] = {
/* front */
0x02,
};
static hda_nid_t alc260_adc_nids[1] = {
/* ADC0 */
0x04,
};
static hda_nid_t alc260_adc_nids_alt[1] = {
/* ADC1 */
0x05,
};
static hda_nid_t alc260_hp_adc_nids[2] = {
/* ADC1, 0 */
0x05, 0x04
};
/* NIDs used when simultaneous access to both ADCs makes sense. Note that
* alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC.
*/
static hda_nid_t alc260_dual_adc_nids[2] = {
/* ADC0, ADC1 */
0x04, 0x05
};
#define ALC260_DIGOUT_NID 0x03
#define ALC260_DIGIN_NID 0x06
static struct hda_input_mux alc260_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack,
* headphone jack and the internal CD lines since these are the only pins at
* which audio can appear. For flexibility, also allow the option of
* recording the mixer output on the second ADC (ADC0 doesn't have a
* connection to the mixer output).
*/
static struct hda_input_mux alc260_fujitsu_capture_sources[2] = {
{
.num_items = 3,
.items = {
{ "Mic/Line", 0x0 },
{ "CD", 0x4 },
{ "Headphone", 0x2 },
},
},
{
.num_items = 4,
.items = {
{ "Mic/Line", 0x0 },
{ "CD", 0x4 },
{ "Headphone", 0x2 },
{ "Mixer", 0x5 },
},
},
};
/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to
* the Fujitsu S702x, but jacks are marked differently.
*/
static struct hda_input_mux alc260_acer_capture_sources[2] = {
{
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Headphone", 0x5 },
},
},
{
.num_items = 5,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Headphone", 0x6 },
{ "Mixer", 0x5 },
},
},
};
/*
* This is just place-holder, so there's something for alc_build_pcms to look
* at when it calculates the maximum number of channels. ALC260 has no mixer
* element which allows changing the channel mode, so the verb list is
* never used.
*/
static struct hda_channel_mode alc260_modes[1] = {
{ 2, NULL },
};
/* Mixer combinations
*
* basic: base_output + input + pc_beep + capture
* HP: base_output + input + capture_alt
* HP_3013: hp_3013 + input + capture
* fujitsu: fujitsu + capture
* acer: acer + capture
*/
static struct snd_kcontrol_new alc260_base_output_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc260_input_mixer[] = {
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc260_pc_beep_mixer[] = {
HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Aux-In Playback Volume", 0x07, 0x06, HDA_INPUT),
HDA_CODEC_MUTE("Aux-In Playback Switch", 0x07, 0x06, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("iSpeaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("iSpeaker Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT),
{ } /* end */
};
/* 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[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT),
ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT),
ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN),
HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Internal Speaker Playback Switch", 0x09, 2, HDA_INPUT),
{ } /* end */
};
/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks. Note that current
* versions of the ALC260 don't act on requests to enable mic bias from NID
* 0x0f (used to drive the headphone jack in these laptops). The ALC260
* datasheet doesn't mention this restriction. At this stage it's not clear
* whether this behaviour is intentional or is a hardware bug in chip
* revisions available in early 2006. Therefore for now allow the
* "Headphone Jack Mode" control to span all choices, but if it turns out
* that the lack of mic bias for this NID is intentional we could change the
* mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
*
* In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006
* don't appear to make the mic bias available from the "line" jack, even
* though the NID used for this jack (0x14) can supply it. The theory is
* that perhaps Acer have included blocking capacitors between the ALC260
* and the output jack. If this turns out to be the case for all such
* models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT
* to ALC_PIN_DIR_INOUT_NOMICBIAS.
*
* The C20x Tablet series have a mono internal speaker which is controlled
* via the chip's Mono sum widget and pin complex, so include the necessary
* controls for such models. On models without a "mono speaker" the control
* won't do anything.
*/
static struct snd_kcontrol_new alc260_acer_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME_MONO("Mono Speaker Playback Volume", 0x0a, 1, 0x0,
HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Mono Speaker Playback Switch", 0x0a, 1, 2,
HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
{ } /* end */
};
/* capture mixer elements */
static struct snd_kcontrol_new alc260_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x04, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x05, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x05, 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 */
};
static struct snd_kcontrol_new alc260_capture_alt_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x05, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x05, 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 */
};
/*
* initialization verbs
*/
static struct hda_verb alc260_init_verbs[] = {
/* Line In pin widget for input */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* CD pin widget for input */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
/* Mic2 (front panel) pin widget for input and vref at 80% */
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
/* LINE-2 is used for line-out in rear */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* select line-out */
{0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
/* LINE-OUT pin */
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* enable HP */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* enable Mono */
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* set connection select to line in (default select for this ADC) */
{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
/* mute capture amp left and right */
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* set connection select to line in (default select for this ADC) */
{0x05, AC_VERB_SET_CONNECT_SEL, 0x02},
/* set vol=0 Line-Out mixer amp left and right */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* unmute pin widget amp left and right (no gain on this amp) */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* set vol=0 HP mixer amp left and right */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* unmute pin widget amp left and right (no gain on this amp) */
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* set vol=0 Mono mixer amp left and right */
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* unmute pin widget amp left and right (no gain on this amp) */
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* unmute LINE-2 out pin */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */
/* mute CD */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
/* mute Line In */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
/* mute Mic */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
/* mute Front out path */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* mute Headphone out path */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* mute Mono out path */
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ }
};
#if 0 /* should be identical with alc260_init_verbs? */
static struct hda_verb alc260_hp_init_verbs[] = {
/* Headphone and output */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
/* mono output */
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* Mic2 (front panel) pin widget for input and vref at 80% */
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* Line In pin widget for input */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* Line-2 pin widget for output */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* CD pin widget for input */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* unmute amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* set connection select to line in (default select for this ADC) */
{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
/* unmute Line-Out mixer amp left and right (volume = 0) */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* unmute HP mixer amp left and right (volume = 0) */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */
/* unmute CD */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
/* unmute Line In */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
/* unmute Mic */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
/* Unmute Front out path */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Unmute Headphone out path */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Unmute Mono out path */
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{ }
};
#endif
static struct hda_verb alc260_hp_3013_init_verbs[] = {
/* Line out and output */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* mono output */
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* Mic2 (front panel) pin widget for input and vref at 80% */
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* Line In pin widget for input */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* Headphone pin widget for output */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
/* CD pin widget for input */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* unmute amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* set connection select to line in (default select for this ADC) */
{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
/* unmute Line-Out mixer amp left and right (volume = 0) */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* unmute HP mixer amp left and right (volume = 0) */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */
/* unmute CD */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
/* unmute Line In */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
/* unmute Mic */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
/* Unmute Front out path */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Unmute Headphone out path */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Unmute Mono out path */
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{ }
};
/* Initialisation sequence for ALC260 as configured in Fujitsu S702x
* laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD
* audio = 0x16, internal speaker = 0x10.
*/
static struct hda_verb alc260_fujitsu_init_verbs[] = {
/* Disable all GPIOs */
{0x01, AC_VERB_SET_GPIO_MASK, 0},
/* Internal speaker is connected to headphone pin */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Headphone/Line-out jack connects to Line1 pin; make it an output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* Mic/Line-in jack is connected to mic1 pin, so make it an input */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Ensure all other unused pins are disabled and muted. */
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Disable digital (SPDIF) pins */
{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
/* Ensure Line1 pin widget takes its input from the OUT1 sum bus
* when acting as an output.
*/
{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
/* Start with output sum widgets muted and their output gains at min */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Line1 pin widget output buffer since it starts as an output.
* If the pin mode is changed by the user the pin mode control will
* take care of enabling the pin's input/output buffers as needed.
* Therefore there's no need to enable the input buffer at this
* stage.
*/
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute input buffer of pin widget used for Line-in (no equiv
* mixer ctrl)
*/
{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Set ADC connection select to match default mixer setting - line
* in (on mic1 pin)
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Do the same for the second ADC: mute capture input amp and
* set ADC connection to line in (on mic1 pin)
*/
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mute all inputs to mixer widget (even unconnected ones) */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
{ }
};
/* Initialisation sequence for ALC260 as configured in Acer TravelMate and
* similar laptops (adapted from Fujitsu init verbs).
*/
static struct hda_verb alc260_acer_init_verbs[] = {
/* On TravelMate laptops, GPIO 0 enables the internal speaker and
* the headphone jack. Turn this on and rely on the standard mute
* methods whenever the user wants to turn these outputs off.
*/
{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
/* Internal speaker/Headphone jack is connected to Line-out pin */
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Internal microphone/Mic jack is connected to Mic1 pin */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
/* Line In jack is connected to Line1 pin */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Some Acers (eg: C20x Tablets) use Mono pin for internal speaker */
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Ensure all other unused pins are disabled and muted. */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Disable digital (SPDIF) pins */
{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
/* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
* bus when acting as outputs.
*/
{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
/* Start with output sum widgets muted and their output gains at min */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Unmute Line-out pin widget amp left and right (no equiv mixer ctrl) */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute mono pin widget amp output (no equiv mixer ctrl) */
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mic1 and Line1 pin widget input buffers since they start as
* inputs. If the pin mode is changed by the user the pin mode control
* will take care of enabling the pin's input/output buffers as needed.
* Therefore there's no need to enable the input buffer at this
* stage.
*/
{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Set ADC connection select to match default mixer setting - mic
* (on mic1 pin)
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Do similar with the second ADC: mute capture input amp and
* set ADC connection to mic to match ALSA's default state.
*/
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mute all inputs to mixer widget (even unconnected ones) */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
{ }
};
/* Test configuration for debugging, modelled after the ALC880 test
* configuration.
*/
#ifdef CONFIG_SND_DEBUG
static hda_nid_t alc260_test_dac_nids[1] = {
0x02,
};
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
* is NID 0x04.
*/
static struct hda_input_mux alc260_test_capture_sources[2] = {
{
.num_items = 7,
.items = {
{ "MIC1 pin", 0x0 },
{ "MIC2 pin", 0x1 },
{ "LINE1 pin", 0x2 },
{ "LINE2 pin", 0x3 },
{ "CD pin", 0x4 },
{ "LINE-OUT pin", 0x5 },
{ "HP-OUT pin", 0x6 },
},
},
{
.num_items = 8,
.items = {
{ "MIC1 pin", 0x0 },
{ "MIC2 pin", 0x1 },
{ "LINE1 pin", 0x2 },
{ "LINE2 pin", 0x3 },
{ "CD pin", 0x4 },
{ "Mixer", 0x5 },
{ "LINE-OUT pin", 0x6 },
{ "HP-OUT pin", 0x7 },
},
},
};
static struct snd_kcontrol_new alc260_test_mixer[] = {
/* Output driver widgets */
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT),
HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT),
/* Modes for retasking pin widgets
* Note: the ALC260 doesn't seem to act on requests to enable mic
* bias from NIDs 0x0f and 0x10. The ALC260 datasheet doesn't
* mention this restriction. At this stage it's not clear whether
* this behaviour is intentional or is a hardware bug in chip
* revisions available at least up until early 2006. Therefore for
* now allow the "HP-OUT" and "LINE-OUT" Mode controls to span all
* choices, but if it turns out that the lack of mic bias for these
* NIDs is intentional we could change their modes from
* ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
*/
ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT),
/* Loopback mixer controls */
HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT),
HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT),
HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT),
HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT),
HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT),
HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT),
/* Controls for GPIO pins, assuming they are configured as outputs */
ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
/* Switches to allow the digital IO pins to be enabled. The datasheet
* is ambigious as to which NID is which; testing on laptops which
* make this output available should provide clarification.
*/
ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01),
ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01),
{ } /* end */
};
static struct hda_verb alc260_test_init_verbs[] = {
/* Enable all GPIOs as outputs with an initial value of 0 */
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f},
{0x01, AC_VERB_SET_GPIO_DATA, 0x00},
{0x01, AC_VERB_SET_GPIO_MASK, 0x0f},
/* Enable retasking pins as output, initially without power amp */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* Disable digital (SPDIF) pins initially, but users can enable
* them via a mixer switch. In the case of SPDIF-out, this initverb
* payload also sets the generation to 0, output to be in "consumer"
* PCM format, copyright asserted, no pre-emphasis and no validity
* control.
*/
{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
/* Ensure mic1, mic2, line1 and line2 pin widgets take input from the
* OUT1 sum bus when acting as an output.
*/
{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
{0x0c, AC_VERB_SET_CONNECT_SEL, 0},
{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
{0x0e, AC_VERB_SET_CONNECT_SEL, 0},
/* Start with output sum widgets muted and their output gains at min */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Unmute retasking pin widget output buffers since the default
* state appears to be output. As the pin mode is changed by the
* user the pin mode control will take care of enabling the pin's
* input/output buffers as needed.
*/
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Also unmute the mono-out pin widget */
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Set ADC connection select to match default mixer setting (mic1
* pin)
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Do the same for the second ADC: mute capture input amp and
* set ADC connection to mic1 pin
*/
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mute all inputs to mixer widget (even unconnected ones) */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
{ }
};
#endif
static struct hda_pcm_stream alc260_pcm_analog_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
};
static struct hda_pcm_stream alc260_pcm_analog_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
};
#define alc260_pcm_digital_playback alc880_pcm_digital_playback
#define alc260_pcm_digital_capture alc880_pcm_digital_capture
/*
* for BIOS auto-configuration
*/
static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
const char *pfx)
{
hda_nid_t nid_vol;
unsigned long vol_val, sw_val;
char name[32];
int err;
if (nid >= 0x0f && nid < 0x11) {
nid_vol = nid - 0x7;
vol_val = HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT);
sw_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
} else if (nid == 0x11) {
nid_vol = nid - 0x7;
vol_val = HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT);
sw_val = HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT);
} else if (nid >= 0x12 && nid <= 0x15) {
nid_vol = 0x08;
vol_val = HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT);
sw_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
} else
return 0; /* N/A */
snprintf(name, sizeof(name), "%s Playback Volume", pfx);
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val)) < 0)
return err;
snprintf(name, sizeof(name), "%s Playback Switch", pfx);
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, sw_val)) < 0)
return err;
return 1;
}
/* add playback controls from the parsed DAC table */
static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
{
hda_nid_t nid;
int err;
spec->multiout.num_dacs = 1;
spec->multiout.dac_nids = spec->private_dac_nids;
spec->multiout.dac_nids[0] = 0x02;
nid = cfg->line_out_pins[0];
if (nid) {
err = alc260_add_playback_controls(spec, nid, "Front");
if (err < 0)
return err;
}
nid = cfg->speaker_pins[0];
if (nid) {
err = alc260_add_playback_controls(spec, nid, "Speaker");
if (err < 0)
return err;
}
nid = cfg->hp_pins[0];
if (nid) {
err = alc260_add_playback_controls(spec, nid, "Headphone");
if (err < 0)
return err;
}
return 0;
}
/* create playback/capture controls for input pins */
static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
{
struct hda_input_mux *imux = &spec->private_imux;
int i, err, idx;
for (i = 0; i < AUTO_PIN_LAST; i++) {
if (cfg->input_pins[i] >= 0x12) {
idx = cfg->input_pins[i] - 0x12;
err = new_analog_input(spec, cfg->input_pins[i],
auto_pin_cfg_labels[i], idx, 0x07);
if (err < 0)
return err;
imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
imux->items[imux->num_items].index = idx;
imux->num_items++;
}
if ((cfg->input_pins[i] >= 0x0f) && (cfg->input_pins[i] <= 0x10)){
idx = cfg->input_pins[i] - 0x09;
err = new_analog_input(spec, cfg->input_pins[i],
auto_pin_cfg_labels[i], idx, 0x07);
if (err < 0)
return err;
imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
imux->items[imux->num_items].index = idx;
imux->num_items++;
}
}
return 0;
}
static void alc260_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
int sel_idx)
{
/* set as output */
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
/* need the manual connection? */
if (nid >= 0x12) {
int idx = nid - 0x12;
snd_hda_codec_write(codec, idx + 0x0b, 0,
AC_VERB_SET_CONNECT_SEL, sel_idx);
}
}
static void alc260_auto_init_multi_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t nid;
nid = spec->autocfg.line_out_pins[0];
if (nid)
alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0);
nid = spec->autocfg.speaker_pins[0];
if (nid)
alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0);
nid = spec->autocfg.hp_pins[0];
if (nid)
alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0);
}
#define ALC260_PIN_CD_NID 0x16
static void alc260_auto_init_analog_input(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = spec->autocfg.input_pins[i];
if (nid >= 0x12) {
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
i <= AUTO_PIN_FRONT_MIC ? PIN_VREF80 : PIN_IN);
if (nid != ALC260_PIN_CD_NID)
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_MUTE);
}
}
}
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc260_volume_init_verbs[] = {
/*
* Unmute ADC0-1 and set the default input to mic-in
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
{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 front panel
* mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
/*
* Set up output mixers (0x08 - 0x0a)
*/
/* set vol=0 to output mixers */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{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)},
{ }
};
static int alc260_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int wcap;
int err;
static hda_nid_t alc260_ignore[] = { 0x17, 0 };
if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc260_ignore)) < 0)
return err;
if ((err = alc260_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0)
return err;
if (! spec->kctl_alloc)
return 0; /* can't find valid BIOS pin config */
if ((err = alc260_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
return err;
spec->multiout.max_channels = 2;
if (spec->autocfg.dig_out_pin)
spec->multiout.dig_out_nid = ALC260_DIGOUT_NID;
if (spec->kctl_alloc)
spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
spec->init_verbs[spec->num_init_verbs++] = alc260_volume_init_verbs;
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux;
/* check whether NID 0x04 is valid */
wcap = get_wcaps(codec, 0x04);
wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */
if (wcap != AC_WID_AUD_IN) {
spec->adc_nids = alc260_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt);
spec->mixers[spec->num_mixers] = alc260_capture_alt_mixer;
} else {
spec->adc_nids = alc260_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids);
spec->mixers[spec->num_mixers] = alc260_capture_mixer;
}
spec->num_mixers++;
return 1;
}
/* additional initialization for auto-configuration model */
static void alc260_auto_init(struct hda_codec *codec)
{
alc260_auto_init_multi_out(codec);
alc260_auto_init_analog_input(codec);
}
/*
* ALC260 configurations
*/
static const char *alc260_models[ALC260_MODEL_LAST] = {
[ALC260_BASIC] = "basic",
[ALC260_HP] = "hp",
[ALC260_HP_3013] = "hp-3013",
[ALC260_FUJITSU_S702X] = "fujitsu",
[ALC260_ACER] = "acer",
#ifdef CONFIG_SND_DEBUG
[ALC260_TEST] = "test",
#endif
[ALC260_AUTO] = "auto",
};
static struct snd_pci_quirk alc260_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER),
SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP),
SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_3013),
SND_PCI_QUIRK(0x103c, 0x3013, "HP", ALC260_HP_3013),
SND_PCI_QUIRK(0x103c, 0x3014, "HP", ALC260_HP),
SND_PCI_QUIRK(0x103c, 0x3015, "HP", ALC260_HP),
SND_PCI_QUIRK(0x103c, 0x3016, "HP", ALC260_HP),
SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC),
SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC),
SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC),
SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X),
SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC),
{}
};
static struct alc_config_preset alc260_presets[] = {
[ALC260_BASIC] = {
.mixers = { alc260_base_output_mixer,
alc260_input_mixer,
alc260_pc_beep_mixer,
alc260_capture_mixer },
.init_verbs = { alc260_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
.adc_nids = alc260_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
},
[ALC260_HP] = {
.mixers = { alc260_base_output_mixer,
alc260_input_mixer,
alc260_capture_alt_mixer },
.init_verbs = { alc260_init_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,
},
[ALC260_HP_3013] = {
.mixers = { alc260_hp_3013_mixer,
alc260_input_mixer,
alc260_capture_alt_mixer },
.init_verbs = { alc260_hp_3013_init_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,
},
[ALC260_FUJITSU_S702X] = {
.mixers = { alc260_fujitsu_mixer,
alc260_capture_mixer },
.init_verbs = { alc260_fujitsu_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
.adc_nids = alc260_dual_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources),
.input_mux = alc260_fujitsu_capture_sources,
},
[ALC260_ACER] = {
.mixers = { alc260_acer_mixer,
alc260_capture_mixer },
.init_verbs = { alc260_acer_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
.adc_nids = alc260_dual_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources),
.input_mux = alc260_acer_capture_sources,
},
#ifdef CONFIG_SND_DEBUG
[ALC260_TEST] = {
.mixers = { alc260_test_mixer,
alc260_capture_mixer },
.init_verbs = { alc260_test_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_test_dac_nids),
.dac_nids = alc260_test_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids),
.adc_nids = alc260_test_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.num_mux_defs = ARRAY_SIZE(alc260_test_capture_sources),
.input_mux = alc260_test_capture_sources,
},
#endif
};
static int patch_alc260(struct hda_codec *codec)
{
struct alc_spec *spec;
int err, board_config;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
board_config = snd_hda_check_board_config(codec, ALC260_MODEL_LAST,
alc260_models,
alc260_cfg_tbl);
if (board_config < 0) {
snd_printd(KERN_INFO "hda_codec: Unknown model for ALC260, "
"trying auto-probe from BIOS...\n");
board_config = ALC260_AUTO;
}
if (board_config == ALC260_AUTO) {
/* automatic parse from the BIOS config */
err = alc260_parse_auto_config(codec);
if (err < 0) {
alc_free(codec);
return err;
} else if (! err) {
printk(KERN_INFO
"hda_codec: Cannot set up configuration "
"from BIOS. Using base mode...\n");
board_config = ALC260_BASIC;
}
}
if (board_config != ALC260_AUTO)
setup_preset(spec, &alc260_presets[board_config]);
spec->stream_name_analog = "ALC260 Analog";
spec->stream_analog_playback = &alc260_pcm_analog_playback;
spec->stream_analog_capture = &alc260_pcm_analog_capture;
spec->stream_name_digital = "ALC260 Digital";
spec->stream_digital_playback = &alc260_pcm_digital_playback;
spec->stream_digital_capture = &alc260_pcm_digital_capture;
codec->patch_ops = alc_patch_ops;
if (board_config == ALC260_AUTO)
spec->init_hook = alc260_auto_init;
return 0;
}
/*
* ALC882 support
*
* ALC882 is almost identical with ALC880 but has cleaner and more flexible
* configuration. Each pin widget can choose any input DACs and a mixer.
* Each ADC is connected from a mixer of all inputs. This makes possible
* 6-channel independent captures.
*
* In addition, an independent DAC for the multi-playback (not used in this
* driver yet).
*/
#define ALC882_DIGOUT_NID 0x06
#define ALC882_DIGIN_NID 0x0a
static struct hda_channel_mode alc882_ch_modes[1] = {
{ 8, NULL }
};
static hda_nid_t alc882_dac_nids[4] = {
/* front, rear, clfe, rear_surr */
0x02, 0x03, 0x04, 0x05
};
/* identical with ALC880 */
#define alc882_adc_nids alc880_adc_nids
#define alc882_adc_nids_alt alc880_adc_nids_alt
/* input MUX */
/* FIXME: should be a matrix-type input source selection */
static struct hda_input_mux alc882_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
#define alc882_mux_enum_info alc_mux_enum_info
#define alc882_mux_enum_get alc_mux_enum_get
static int alc882_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;
const struct hda_input_mux *imux = spec->input_mux;
unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 };
hda_nid_t nid = capture_mixers[adc_idx];
unsigned int *cur_val = &spec->cur_mux[adc_idx];
unsigned int i, idx;
idx = ucontrol->value.enumerated.item[0];
if (idx >= imux->num_items)
idx = imux->num_items - 1;
if (*cur_val == idx && ! codec->in_resume)
return 0;
for (i = 0; i < imux->num_items; i++) {
unsigned int v = (i == idx) ? 0x7000 : 0x7080;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
v | (imux->items[i].index << 8));
}
*cur_val = idx;
return 1;
}
/*
* 6ch mode
*/
static struct hda_verb alc882_sixstack_ch6_init[] = {
{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ } /* end */
};
/*
* 8ch mode
*/
static struct hda_verb alc882_sixstack_ch8_init[] = {
{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ } /* end */
};
static struct hda_channel_mode alc882_sixstack_modes[2] = {
{ 6, alc882_sixstack_ch6_init },
{ 8, alc882_sixstack_ch8_init },
};
/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
* Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
*/
static struct snd_kcontrol_new alc882_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_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),
{ } /* end */
};
static struct snd_kcontrol_new alc882_chmode_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
static struct hda_verb alc882_init_verbs[] = {
/* Front mixer: unmute input/output amp left and right (volume = 0) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Rear mixer */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* CLFE mixer */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Side mixer */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Front Pin: output 0 (0x0c) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Rear Pin: output 1 (0x0d) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
/* CLFE Pin: output 2 (0x0e) */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
/* Side Pin: output 3 (0x0f) */
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
/* Mic (rear) pin: input vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Front Mic pin: input vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line In pin: input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line-2 In: Headphone output (output 0 - 0x0c) */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* ADC1: mute amp left and right */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
/* ADC2: mute amp left and right */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
/* ADC3: mute amp left and right */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{ }
};
static struct hda_verb alc882_eapd_verbs[] = {
/* change to EAPD mode */
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3060},
{ }
};
/* Mac Pro test */
static struct snd_kcontrol_new alc882_macpro_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x18, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x02, HDA_INPUT),
{ } /* end */
};
static struct hda_verb alc882_macpro_init_verbs[] = {
/* Front mixer: unmute input/output amp left and right (volume = 0) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Front Pin: output 0 (0x0c) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Front Mic pin: input vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Speaker: output */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x04},
/* Headphone output (output 0 - 0x0c) */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* ADC1: mute amp left and right */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
/* ADC2: mute amp left and right */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
/* ADC3: mute amp left and right */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{ }
};
static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted)
{
unsigned int gpiostate, gpiomask, gpiodir;
gpiostate = snd_hda_codec_read(codec, codec->afg, 0,
AC_VERB_GET_GPIO_DATA, 0);
if (!muted)
gpiostate |= (1 << pin);
else
gpiostate &= ~(1 << pin);
gpiomask = snd_hda_codec_read(codec, codec->afg, 0,
AC_VERB_GET_GPIO_MASK, 0);
gpiomask |= (1 << pin);
gpiodir = snd_hda_codec_read(codec, codec->afg, 0,
AC_VERB_GET_GPIO_DIRECTION, 0);
gpiodir |= (1 << pin);
snd_hda_codec_write(codec, codec->afg, 0,
AC_VERB_SET_GPIO_MASK, gpiomask);
snd_hda_codec_write(codec, codec->afg, 0,
AC_VERB_SET_GPIO_DIRECTION, gpiodir);
msleep(1);
snd_hda_codec_write(codec, codec->afg, 0,
AC_VERB_SET_GPIO_DATA, gpiostate);
}
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc882_auto_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_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
{ }
};
/* capture mixer elements */
static struct snd_kcontrol_new alc882_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 = alc882_mux_enum_info,
.get = alc882_mux_enum_get,
.put = alc882_mux_enum_put,
},
{ } /* end */
};
static struct snd_kcontrol_new alc882_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 = alc882_mux_enum_info,
.get = alc882_mux_enum_get,
.put = alc882_mux_enum_put,
},
{ } /* end */
};
/* pcm configuration: identiacal with ALC880 */
#define alc882_pcm_analog_playback alc880_pcm_analog_playback
#define alc882_pcm_analog_capture alc880_pcm_analog_capture
#define alc882_pcm_digital_playback alc880_pcm_digital_playback
#define alc882_pcm_digital_capture alc880_pcm_digital_capture
/*
* configuration and preset
*/
static const char *alc882_models[ALC882_MODEL_LAST] = {
[ALC882_3ST_DIG] = "3stack-dig",
[ALC882_6ST_DIG] = "6stack-dig",
[ALC882_ARIMA] = "arima",
[ALC885_MACPRO] = "macpro",
[ALC882_AUTO] = "auto",
};
static struct snd_pci_quirk alc882_cfg_tbl[] = {
SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA),
{}
};
static struct alc_config_preset alc882_presets[] = {
[ALC882_3ST_DIG] = {
.mixers = { alc882_base_mixer },
.init_verbs = { alc882_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
.channel_mode = alc882_ch_modes,
.need_dac_fix = 1,
.input_mux = &alc882_capture_source,
},
[ALC882_6ST_DIG] = {
.mixers = { alc882_base_mixer, alc882_chmode_mixer },
.init_verbs = { alc882_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
.channel_mode = alc882_sixstack_modes,
.input_mux = &alc882_capture_source,
},
[ALC882_ARIMA] = {
.mixers = { alc882_base_mixer, alc882_chmode_mixer },
.init_verbs = { alc882_init_verbs, alc882_eapd_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
.channel_mode = alc882_sixstack_modes,
.input_mux = &alc882_capture_source,
},
[ALC885_MACPRO] = {
.mixers = { alc882_macpro_mixer },
.init_verbs = { alc882_macpro_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
.channel_mode = alc882_ch_modes,
.input_mux = &alc882_capture_source,
},
};
/*
* BIOS auto configuration
*/
static void alc882_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
int dac_idx)
{
/* set as output */
struct alc_spec *spec = codec->spec;
int idx;
if (spec->multiout.dac_nids[dac_idx] == 0x25)
idx = 4;
else
idx = spec->multiout.dac_nids[dac_idx] - 2;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
}
static void alc882_auto_init_multi_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i <= HDA_SIDE; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
if (nid)
alc882_auto_set_output_and_unmute(codec, nid, PIN_OUT, i);
}
}
static void alc882_auto_init_hp_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t pin;
pin = spec->autocfg.hp_pins[0];
if (pin) /* connect to front */
alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); /* use dac 0 */
}
#define alc882_is_input_pin(nid) alc880_is_input_pin(nid)
#define ALC882_PIN_CD_NID ALC880_PIN_CD_NID
static void alc882_auto_init_analog_input(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = spec->autocfg.input_pins[i];
if (alc882_is_input_pin(nid)) {
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
i <= AUTO_PIN_FRONT_MIC ? PIN_VREF80 : PIN_IN);
if (nid != ALC882_PIN_CD_NID)
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_MUTE);
}
}
}
/* almost identical with ALC880 parser... */
static int alc882_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err = alc880_parse_auto_config(codec);
if (err < 0)
return err;
else if (err > 0)
/* hack - override the init verbs */
spec->init_verbs[0] = alc882_auto_init_verbs;
return err;
}
/* additional initialization for auto-configuration model */
static void alc882_auto_init(struct hda_codec *codec)
{
alc882_auto_init_multi_out(codec);
alc882_auto_init_hp_out(codec);
alc882_auto_init_analog_input(codec);
}
static int patch_alc882(struct hda_codec *codec)
{
struct alc_spec *spec;
int err, board_config;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
board_config = snd_hda_check_board_config(codec, ALC882_MODEL_LAST,
alc882_models,
alc882_cfg_tbl);
if (board_config < 0 || board_config >= ALC882_MODEL_LAST) {
printk(KERN_INFO "hda_codec: Unknown model for ALC882, "
"trying auto-probe from BIOS...\n");
board_config = ALC882_AUTO;
}
if (board_config == ALC882_AUTO) {
/* automatic parse from the BIOS config */
err = alc882_parse_auto_config(codec);
if (err < 0) {
alc_free(codec);
return err;
} else if (! err) {
printk(KERN_INFO
"hda_codec: Cannot set up configuration "
"from BIOS. Using base mode...\n");
board_config = ALC882_3ST_DIG;
}
}
if (board_config != ALC882_AUTO)
setup_preset(spec, &alc882_presets[board_config]);
if (board_config == ALC885_MACPRO) {
alc882_gpio_mute(codec, 0, 0);
alc882_gpio_mute(codec, 1, 0);
}
spec->stream_name_analog = "ALC882 Analog";
spec->stream_analog_playback = &alc882_pcm_analog_playback;
spec->stream_analog_capture = &alc882_pcm_analog_capture;
spec->stream_name_digital = "ALC882 Digital";
spec->stream_digital_playback = &alc882_pcm_digital_playback;
spec->stream_digital_capture = &alc882_pcm_digital_capture;
if (! spec->adc_nids && spec->input_mux) {
/* check whether NID 0x07 is valid */
unsigned int wcap = get_wcaps(codec, 0x07);
wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */
if (wcap != AC_WID_AUD_IN) {
spec->adc_nids = alc882_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids_alt);
spec->mixers[spec->num_mixers] = alc882_capture_alt_mixer;
spec->num_mixers++;
} else {
spec->adc_nids = alc882_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids);
spec->mixers[spec->num_mixers] = alc882_capture_mixer;
spec->num_mixers++;
}
}
codec->patch_ops = alc_patch_ops;
if (board_config == ALC882_AUTO)
spec->init_hook = alc882_auto_init;
return 0;
}
/*
* ALC883 support
*
* ALC883 is almost identical with ALC880 but has cleaner and more flexible
* configuration. Each pin widget can choose any input DACs and a mixer.
* Each ADC is connected from a mixer of all inputs. This makes possible
* 6-channel independent captures.
*
* In addition, an independent DAC for the multi-playback (not used in this
* driver yet).
*/
#define ALC883_DIGOUT_NID 0x06
#define ALC883_DIGIN_NID 0x0a
static hda_nid_t alc883_dac_nids[4] = {
/* front, rear, clfe, rear_surr */
0x02, 0x04, 0x03, 0x05
};
static hda_nid_t alc883_adc_nids[2] = {
/* ADC1-2 */
0x08, 0x09,
};
/* input MUX */
/* FIXME: should be a matrix-type input source selection */
static struct hda_input_mux alc883_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
#define alc883_mux_enum_info alc_mux_enum_info
#define alc883_mux_enum_get alc_mux_enum_get
static int alc883_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;
const struct hda_input_mux *imux = spec->input_mux;
unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 };
hda_nid_t nid = capture_mixers[adc_idx];
unsigned int *cur_val = &spec->cur_mux[adc_idx];
unsigned int i, idx;
idx = ucontrol->value.enumerated.item[0];
if (idx >= imux->num_items)
idx = imux->num_items - 1;
if (*cur_val == idx && ! codec->in_resume)
return 0;
for (i = 0; i < imux->num_items; i++) {
unsigned int v = (i == idx) ? 0x7000 : 0x7080;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
v | (imux->items[i].index << 8));
}
*cur_val = idx;
return 1;
}
/*
* 2ch mode
*/
static struct hda_channel_mode alc883_3ST_2ch_modes[1] = {
{ 2, NULL }
};
/*
* 2ch mode
*/
static struct hda_verb alc883_3ST_ch2_init[] = {
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
{ } /* end */
};
/*
* 6ch mode
*/
static struct hda_verb alc883_3ST_ch6_init[] = {
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
{ } /* end */
};
static struct hda_channel_mode alc883_3ST_6ch_modes[2] = {
{ 2, alc883_3ST_ch2_init },
{ 6, alc883_3ST_ch6_init },
};
/*
* 6ch mode
*/
static struct hda_verb alc883_sixstack_ch6_init[] = {
{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ } /* end */
};
/*
* 8ch mode
*/
static struct hda_verb alc883_sixstack_ch8_init[] = {
{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ } /* end */
};
static struct hda_channel_mode alc883_sixstack_modes[2] = {
{ 6, alc883_sixstack_ch6_init },
{ 8, alc883_sixstack_ch8_init },
};
static struct hda_verb alc883_medion_eapd_verbs[] = {
/* eanable EAPD on medion laptop */
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3070},
{ }
};
/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
* Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
*/
static struct snd_kcontrol_new alc883_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_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),
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 snd_kcontrol_new alc883_3ST_2ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_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),
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 snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_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),
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 snd_kcontrol_new alc883_fivestack_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
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_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
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),
{
.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_tagra_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_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),
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 snd_kcontrol_new alc883_tagra_2ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_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),
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 snd_kcontrol_new alc883_chmode_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
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_CONNECT_SEL, 0x00},
/* ADC2: mute amp left and right */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Front mixer: unmute input/output amp left and right (volume = 0) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Rear mixer */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* CLFE mixer */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Side mixer */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{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)},
/* Front Pin: output 0 (0x0c) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Rear Pin: output 1 (0x0d) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
/* CLFE Pin: output 2 (0x0e) */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
/* Side Pin: output 3 (0x0f) */
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
/* Mic (rear) pin: input vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Front Mic pin: input vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line In pin: input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line-2 In: Headphone output (output 0 - 0x0c) */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{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)},
/* 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)},
{ }
};
static struct hda_verb alc883_tagra_verbs[] = {
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{0x01, AC_VERB_SET_GPIO_MASK, 0x03},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03},
{0x01, AC_VERB_SET_GPIO_DATA, 0x03},
{ } /* end */
};
/* toggle speaker-output according to the hp-jack state */
static void alc883_tagra_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_update(codec, 0x1b, 0, HDA_OUTPUT, 0,
0x80, present ? 0x80 : 0);
snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0,
0x80, present ? 0x80 : 0);
snd_hda_codec_write(codec, 1, 0, AC_VERB_SET_GPIO_DATA, present ? 1 : 3);
}
static void alc883_tagra_unsol_event(struct hda_codec *codec, unsigned int res)
{
if ((res >> 26) == ALC880_HP_EVENT)
alc883_tagra_automute(codec);
}
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc883_auto_init_verbs[] = {
/*
* Unmute ADC0-2 and set the default input to mic-in
*/
{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_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer1 */
{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(3)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
/* Input mixer2 */
{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(3)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
{ }
};
/* capture mixer elements */
static struct snd_kcontrol_new alc883_capture_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 = alc882_mux_enum_info,
.get = alc882_mux_enum_get,
.put = alc882_mux_enum_put,
},
{ } /* end */
};
/* pcm configuration: identiacal with ALC880 */
#define alc883_pcm_analog_playback alc880_pcm_analog_playback
#define alc883_pcm_analog_capture alc880_pcm_analog_capture
#define alc883_pcm_digital_playback alc880_pcm_digital_playback
#define alc883_pcm_digital_capture alc880_pcm_digital_capture
/*
* configuration and preset
*/
static const char *alc883_models[ALC883_MODEL_LAST] = {
[ALC883_3ST_2ch_DIG] = "3stack-dig",
[ALC883_3ST_6ch_DIG] = "3stack-6ch-dig",
[ALC883_3ST_6ch] = "3stack-6ch",
[ALC883_6ST_DIG] = "6stack-dig",
[ALC883_TARGA_DIG] = "targa-dig",
[ALC883_TARGA_2ch_DIG] = "targa-2ch-dig",
[ALC888_DEMO_BOARD] = "6stack-dig-demo",
[ALC883_ACER] = "acer",
[ALC883_MEDION] = "medion",
[ALC883_LAPTOP_EAPD] = "laptop-eapd",
[ALC883_AUTO] = "auto",
};
static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG),
SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch),
SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD),
SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3fc1, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER),
SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch),
{}
};
static struct alc_config_preset alc883_presets[] = {
[ALC883_3ST_2ch_DIG] = {
.mixers = { alc883_3ST_2ch_mixer },
.init_verbs = { alc883_init_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
.adc_nids = alc883_adc_nids,
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
},
[ALC883_3ST_6ch_DIG] = {
.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
.adc_nids = alc883_adc_nids,
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
},
[ALC883_3ST_6ch] = {
.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
.adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
},
[ALC883_6ST_DIG] = {
.mixers = { alc883_base_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
.adc_nids = alc883_adc_nids,
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
.channel_mode = alc883_sixstack_modes,
.input_mux = &alc883_capture_source,
},
[ALC883_TARGA_DIG] = {
.mixers = { alc883_tagra_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc883_tagra_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
.adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_tagra_unsol_event,
.init_hook = alc883_tagra_automute,
},
[ALC883_TARGA_2ch_DIG] = {
.mixers = { alc883_tagra_2ch_mixer},
.init_verbs = { alc883_init_verbs, alc883_tagra_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
.adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_tagra_unsol_event,
.init_hook = alc883_tagra_automute,
},
[ALC888_DEMO_BOARD] = {
.mixers = { alc883_base_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
.adc_nids = alc883_adc_nids,
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
.channel_mode = alc883_sixstack_modes,
.input_mux = &alc883_capture_source,
},
[ALC883_ACER] = {
.mixers = { alc883_base_mixer,
alc883_chmode_mixer },
/* On TravelMate laptops, GPIO 0 enables the internal speaker
* and the headphone jack. Turn this on and rely on the
* standard mute methods whenever the user wants to turn
* these outputs off.
*/
.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
.adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
},
[ALC883_MEDION] = {
.mixers = { alc883_fivestack_mixer,
alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs,
alc883_medion_eapd_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.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,
.input_mux = &alc883_capture_source,
},
[ALC883_LAPTOP_EAPD] = {
.mixers = { alc883_base_mixer,
alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc882_eapd_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
.adc_nids = alc883_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
},
};
/*
* BIOS auto configuration
*/
static void alc883_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
int dac_idx)
{
/* set as output */
struct alc_spec *spec = codec->spec;
int idx;
if (spec->multiout.dac_nids[dac_idx] == 0x25)
idx = 4;
else
idx = spec->multiout.dac_nids[dac_idx] - 2;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
pin_type);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_UNMUTE);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
}
static void alc883_auto_init_multi_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i <= HDA_SIDE; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
if (nid)
alc883_auto_set_output_and_unmute(codec, nid, PIN_OUT, i);
}
}
static void alc883_auto_init_hp_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t pin;
pin = spec->autocfg.hp_pins[0];
if (pin) /* connect to front */
/* use dac 0 */
alc883_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
}
#define alc883_is_input_pin(nid) alc880_is_input_pin(nid)
#define ALC883_PIN_CD_NID ALC880_PIN_CD_NID
static void alc883_auto_init_analog_input(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = spec->autocfg.input_pins[i];
if (alc883_is_input_pin(nid)) {
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
(i <= AUTO_PIN_FRONT_MIC ?
PIN_VREF80 : PIN_IN));
if (nid != ALC883_PIN_CD_NID)
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_MUTE);
}
}
}
/* almost identical with ALC880 parser... */
static int alc883_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err = alc880_parse_auto_config(codec);
if (err < 0)
return err;
else if (err > 0)
/* hack - override the init verbs */
spec->init_verbs[0] = alc883_auto_init_verbs;
spec->mixers[spec->num_mixers] = alc883_capture_mixer;
spec->num_mixers++;
return err;
}
/* additional initialization for auto-configuration model */
static void alc883_auto_init(struct hda_codec *codec)
{
alc883_auto_init_multi_out(codec);
alc883_auto_init_hp_out(codec);
alc883_auto_init_analog_input(codec);
}
static int patch_alc883(struct hda_codec *codec)
{
struct alc_spec *spec;
int err, board_config;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
board_config = snd_hda_check_board_config(codec, ALC883_MODEL_LAST,
alc883_models,
alc883_cfg_tbl);
if (board_config < 0) {
printk(KERN_INFO "hda_codec: Unknown model for ALC883, "
"trying auto-probe from BIOS...\n");
board_config = ALC883_AUTO;
}
if (board_config == ALC883_AUTO) {
/* automatic parse from the BIOS config */
err = alc883_parse_auto_config(codec);
if (err < 0) {
alc_free(codec);
return err;
} else if (! err) {
printk(KERN_INFO
"hda_codec: Cannot set up configuration "
"from BIOS. Using base mode...\n");
board_config = ALC883_3ST_2ch_DIG;
}
}
if (board_config != ALC883_AUTO)
setup_preset(spec, &alc883_presets[board_config]);
spec->stream_name_analog = "ALC883 Analog";
spec->stream_analog_playback = &alc883_pcm_analog_playback;
spec->stream_analog_capture = &alc883_pcm_analog_capture;
spec->stream_name_digital = "ALC883 Digital";
spec->stream_digital_playback = &alc883_pcm_digital_playback;
spec->stream_digital_capture = &alc883_pcm_digital_capture;
if (! spec->adc_nids && spec->input_mux) {
spec->adc_nids = alc883_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
}
codec->patch_ops = alc_patch_ops;
if (board_config == ALC883_AUTO)
spec->init_hook = alc883_auto_init;
return 0;
}
/*
* ALC262 support
*/
#define ALC262_DIGOUT_NID ALC880_DIGOUT_NID
#define ALC262_DIGIN_NID ALC880_DIGIN_NID
#define alc262_dac_nids alc260_dac_nids
#define alc262_adc_nids alc882_adc_nids
#define alc262_adc_nids_alt alc882_adc_nids_alt
#define alc262_modes alc260_modes
#define alc262_capture_source alc882_capture_source
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),
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, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
/* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("PC Beelp Playback Switch", 0x0b, 0x05, HDA_INPUT), */
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc262_hippo1_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
/* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("PC Beelp Playback Switch", 0x0b, 0x05, HDA_INPUT), */
/*HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),*/
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT),
HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT),
HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT),
{ } /* end */
};
#define alc262_capture_mixer alc882_capture_mixer
#define alc262_capture_alt_mixer alc882_capture_alt_mixer
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc262_init_verbs[] = {
/*
* Unmute ADC0-2 and set the default input to mic-in
*/
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* 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 - 0x0e)
*/
/* set vol=0 to output mixers */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
{ }
};
static 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},
{}
};
static struct hda_verb alc262_hippo1_unsol_verbs[] = {
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{}
};
/* mute/unmute internal speaker according to the hp jack and mute state */
static void alc262_hippo_automute(struct hda_codec *codec, int force)
{
struct alc_spec *spec = codec->spec;
unsigned int mute;
if (force || ! spec->sense_updated) {
unsigned int present;
/* need to execute and sync at first */
snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0);
present = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0);
spec->jack_present = (present & 0x80000000) != 0;
spec->sense_updated = 1;
}
if (spec->jack_present) {
/* mute internal speaker */
snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
0x80, 0x80);
snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
0x80, 0x80);
} else {
/* unmute internal speaker if necessary */
mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0);
snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
0x80, mute & 0x80);
mute = snd_hda_codec_amp_read(codec, 0x15, 1, HDA_OUTPUT, 0);
snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
0x80, mute & 0x80);
}
}
/* unsolicited event for HP jack sensing */
static void alc262_hippo_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) != ALC880_HP_EVENT)
return;
alc262_hippo_automute(codec, 1);
}
static void alc262_hippo1_automute(struct hda_codec *codec, int force)
{
struct alc_spec *spec = codec->spec;
unsigned int mute;
if (force || ! spec->sense_updated) {
unsigned int present;
/* need to execute and sync at first */
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);
spec->jack_present = (present & 0x80000000) != 0;
spec->sense_updated = 1;
}
if (spec->jack_present) {
/* mute internal speaker */
snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
0x80, 0x80);
snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
0x80, 0x80);
} else {
/* unmute internal speaker if necessary */
mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0);
snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
0x80, mute & 0x80);
mute = snd_hda_codec_amp_read(codec, 0x1b, 1, HDA_OUTPUT, 0);
snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
0x80, mute & 0x80);
}
}
/* unsolicited event for HP jack sensing */
static void alc262_hippo1_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) != ALC880_HP_EVENT)
return;
alc262_hippo1_automute(codec, 1);
}
/*
* fujitsu model
* 0x14 = headphone/spdif-out, 0x15 = internal speaker
*/
#define ALC_HP_EVENT 0x37
static struct hda_verb alc262_fujitsu_unsol_verbs[] = {
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{}
};
static struct hda_input_mux alc262_fujitsu_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x0 },
{ "CD", 0x4 },
},
};
static struct hda_input_mux alc262_HP_capture_source = {
.num_items = 5,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x3 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "AUX IN", 0x6 },
},
};
/* mute/unmute internal speaker according to the hp jack and mute state */
static void alc262_fujitsu_automute(struct hda_codec *codec, int force)
{
struct alc_spec *spec = codec->spec;
unsigned int mute;
if (force || ! spec->sense_updated) {
unsigned int present;
/* need to execute and sync at first */
snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0);
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0);
spec->jack_present = (present & 0x80000000) != 0;
spec->sense_updated = 1;
}
if (spec->jack_present) {
/* mute internal speaker */
snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
0x80, 0x80);
snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
0x80, 0x80);
} else {
/* unmute internal speaker if necessary */
mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0);
snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
0x80, mute & 0x80);
mute = snd_hda_codec_amp_read(codec, 0x14, 1, HDA_OUTPUT, 0);
snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
0x80, mute & 0x80);
}
}
/* unsolicited event for HP jack sensing */
static void alc262_fujitsu_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) != ALC_HP_EVENT)
return;
alc262_fujitsu_automute(codec, 1);
}
/* bind volumes of both NID 0x0c and 0x0d */
static int alc262_fujitsu_master_vol_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, 0x0c, 0, HDA_OUTPUT, 0,
0x7f, valp[0] & 0x7f);
change |= snd_hda_codec_amp_update(codec, 0x0c, 1, HDA_OUTPUT, 0,
0x7f, valp[1] & 0x7f);
snd_hda_codec_amp_update(codec, 0x0d, 0, HDA_OUTPUT, 0,
0x7f, valp[0] & 0x7f);
snd_hda_codec_amp_update(codec, 0x0d, 1, HDA_OUTPUT, 0,
0x7f, valp[1] & 0x7f);
return change;
}
/* bind hp and internal speaker mute (with plug check) */
static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
long *valp = ucontrol->value.integer.value;
int change;
change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
0x80, valp[0] ? 0 : 0x80);
change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
0x80, valp[1] ? 0 : 0x80);
if (change || codec->in_resume)
alc262_fujitsu_automute(codec, codec->in_resume);
return change;
}
static struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Volume",
.info = snd_hda_mixer_amp_volume_info,
.get = snd_hda_mixer_amp_volume_get,
.put = alc262_fujitsu_master_vol_put,
.tlv = { .c = snd_hda_mixer_amp_tlv },
.private_value = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, 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 = alc262_fujitsu_master_sw_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, 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 Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
/* additional init verbs for Benq laptops */
static struct hda_verb alc262_EAPD_verbs[] = {
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3070},
{}
};
/* add playback controls from the parsed DAC table */
static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg)
{
hda_nid_t nid;
int err;
spec->multiout.num_dacs = 1; /* only use one dac */
spec->multiout.dac_nids = spec->private_dac_nids;
spec->multiout.dac_nids[0] = 2;
nid = cfg->line_out_pins[0];
if (nid) {
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Front Playback Volume",
HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT))) < 0)
return err;
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Front Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
return err;
}
nid = cfg->speaker_pins[0];
if (nid) {
if (nid == 0x16) {
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Speaker Playback Volume",
HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_OUTPUT))) < 0)
return err;
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Speaker Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0)
return err;
} else {
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Speaker Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
return err;
}
}
nid = cfg->hp_pins[0];
if (nid) {
/* spec->multiout.hp_nid = 2; */
if (nid == 0x16) {
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Headphone Playback Volume",
HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_OUTPUT))) < 0)
return err;
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Headphone Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0)
return err;
} else {
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Headphone Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
return err;
}
}
return 0;
}
/* identical with ALC880 */
#define alc262_auto_create_analog_input_ctls alc880_auto_create_analog_input_ctls
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc262_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},
/* 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)},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
{ }
};
static struct hda_verb alc262_HP_BPC_init_verbs[] = {
/*
* Unmute ADC0-2 and set the default input to mic-in
*/
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* 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)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)},
/*
* Set up output mixers (0x0c - 0x0e)
*/
/* set vol=0 to output mixers */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
{ }
};
/* pcm configuration: identiacal with ALC880 */
#define alc262_pcm_analog_playback alc880_pcm_analog_playback
#define alc262_pcm_analog_capture alc880_pcm_analog_capture
#define alc262_pcm_digital_playback alc880_pcm_digital_playback
#define alc262_pcm_digital_capture alc880_pcm_digital_capture
/*
* BIOS auto configuration
*/
static int alc262_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err;
static hda_nid_t alc262_ignore[] = { 0x1d, 0 };
if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc262_ignore)) < 0)
return err;
if (! spec->autocfg.line_outs)
return 0; /* can't find valid BIOS pin config */
if ((err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 ||
(err = alc262_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
if (spec->autocfg.dig_out_pin)
spec->multiout.dig_out_nid = ALC262_DIGOUT_NID;
if (spec->autocfg.dig_in_pin)
spec->dig_in_nid = ALC262_DIGIN_NID;
if (spec->kctl_alloc)
spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
spec->init_verbs[spec->num_init_verbs++] = alc262_volume_init_verbs;
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux;
return 1;
}
#define alc262_auto_init_multi_out alc882_auto_init_multi_out
#define alc262_auto_init_hp_out alc882_auto_init_hp_out
#define alc262_auto_init_analog_input alc882_auto_init_analog_input
/* init callback for auto-configuration model -- overriding the default init */
static void alc262_auto_init(struct hda_codec *codec)
{
alc262_auto_init_multi_out(codec);
alc262_auto_init_hp_out(codec);
alc262_auto_init_analog_input(codec);
}
/*
* configuration and preset
*/
static const char *alc262_models[ALC262_MODEL_LAST] = {
[ALC262_BASIC] = "basic",
[ALC262_HIPPO] = "hippo",
[ALC262_HIPPO_1] = "hippo_1",
[ALC262_FUJITSU] = "fujitsu",
[ALC262_HP_BPC] = "hp-bpc",
[ALC262_BENQ_ED8] = "benq",
[ALC262_AUTO] = "auto",
};
static struct snd_pci_quirk alc262_cfg_tbl[] = {
SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO),
SND_PCI_QUIRK(0x103c, 0x12fe, "HP xw9400", ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x2801, "HP q954", ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC),
SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO),
SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1),
SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8),
{}
};
static struct alc_config_preset alc262_presets[] = {
[ALC262_BASIC] = {
.mixers = { alc262_base_mixer },
.init_verbs = { alc262_init_verbs },
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
},
[ALC262_HIPPO] = {
.mixers = { alc262_base_mixer },
.init_verbs = { alc262_init_verbs, alc262_hippo_unsol_verbs},
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x03,
.dig_out_nid = ALC262_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo_unsol_event,
},
[ALC262_HIPPO_1] = {
.mixers = { alc262_hippo1_mixer },
.init_verbs = { alc262_init_verbs, alc262_hippo1_unsol_verbs},
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x02,
.dig_out_nid = ALC262_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo1_unsol_event,
},
[ALC262_FUJITSU] = {
.mixers = { alc262_fujitsu_mixer },
.init_verbs = { alc262_init_verbs, alc262_fujitsu_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x03,
.dig_out_nid = ALC262_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_fujitsu_capture_source,
.unsol_event = alc262_fujitsu_unsol_event,
},
[ALC262_HP_BPC] = {
.mixers = { alc262_HP_BPC_mixer },
.init_verbs = { alc262_HP_BPC_init_verbs },
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_HP_capture_source,
},
[ALC262_BENQ_ED8] = {
.mixers = { alc262_base_mixer },
.init_verbs = { alc262_init_verbs, alc262_EAPD_verbs },
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
},
};
static int patch_alc262(struct hda_codec *codec)
{
struct alc_spec *spec;
int board_config;
int err;
spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
#if 0
/* pshou 07/11/05 set a zero PCM sample to DAC when FIFO is under-run */
{
int tmp;
snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_COEF_INDEX, 7);
tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0);
snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_COEF_INDEX, 7);
snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PROC_COEF, tmp | 0x80);
}
#endif
board_config = snd_hda_check_board_config(codec, ALC262_MODEL_LAST,
alc262_models,
alc262_cfg_tbl);
if (board_config < 0) {
printk(KERN_INFO "hda_codec: Unknown model for ALC262, "
"trying auto-probe from BIOS...\n");
board_config = ALC262_AUTO;
}
if (board_config == ALC262_AUTO) {
/* automatic parse from the BIOS config */
err = alc262_parse_auto_config(codec);
if (err < 0) {
alc_free(codec);
return err;
} else if (! err) {
printk(KERN_INFO
"hda_codec: Cannot set up configuration "
"from BIOS. Using base mode...\n");
board_config = ALC262_BASIC;
}
}
if (board_config != ALC262_AUTO)
setup_preset(spec, &alc262_presets[board_config]);
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;
if (! spec->adc_nids && spec->input_mux) {
/* check whether NID 0x07 is valid */
unsigned int wcap = get_wcaps(codec, 0x07);
wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */
if (wcap != AC_WID_AUD_IN) {
spec->adc_nids = alc262_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids_alt);
spec->mixers[spec->num_mixers] = alc262_capture_alt_mixer;
spec->num_mixers++;
} else {
spec->adc_nids = alc262_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids);
spec->mixers[spec->num_mixers] = alc262_capture_mixer;
spec->num_mixers++;
}
}
codec->patch_ops = alc_patch_ops;
if (board_config == ALC262_AUTO)
spec->init_hook = alc262_auto_init;
return 0;
}
/*
* ALC861 channel source setting (2/6 channel selection for 3-stack)
*/
/*
* set the path ways for 2 channel output
* need to set the codec line out and mic 1 pin widgets to inputs
*/
static struct hda_verb alc861_threestack_ch2_init[] = {
/* set pin widget 1Ah (line in) for input */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* set pin widget 18h (mic1/2) for input, for mic also enable the vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
#endif
{ } /* end */
};
/*
* 6ch mode
* need to set the codec line out and mic 1 pin widgets to outputs
*/
static struct hda_verb alc861_threestack_ch6_init[] = {
/* set pin widget 1Ah (line in) for output (Back Surround)*/
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* set pin widget 18h (mic1) for output (CLFE)*/
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
#endif
{ } /* end */
};
static struct hda_channel_mode alc861_threestack_modes[2] = {
{ 2, alc861_threestack_ch2_init },
{ 6, alc861_threestack_ch6_init },
};
/* Set mic1 as input and unmute the mixer */
static struct hda_verb alc861_uniwill_m31_ch2_init[] = {
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
{ } /* end */
};
/* Set mic1 as output and mute mixer */
static struct hda_verb alc861_uniwill_m31_ch4_init[] = {
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
{ } /* end */
};
static struct hda_channel_mode alc861_uniwill_m31_modes[2] = {
{ 2, alc861_uniwill_m31_ch2_init },
{ 4, alc861_uniwill_m31_ch4_init },
};
/* Set mic1 and line-in as input and unmute the mixer */
static struct hda_verb alc861_asus_ch2_init[] = {
/* set pin widget 1Ah (line in) for input */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* set pin widget 18h (mic1/2) for input, for mic also enable the vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
#endif
{ } /* end */
};
/* Set mic1 nad line-in as output and mute mixer */
static struct hda_verb alc861_asus_ch6_init[] = {
/* set pin widget 1Ah (line in) for output (Back Surround)*/
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* { 0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
/* set pin widget 18h (mic1) for output (CLFE)*/
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
{ 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
#endif
{ } /* end */
};
static struct hda_channel_mode alc861_asus_modes[2] = {
{ 2, alc861_asus_ch2_init },
{ 6, alc861_asus_ch6_init },
};
/* patch-ALC861 */
static struct snd_kcontrol_new alc861_base_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
/*Input mixer control */
/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
/* Capture mixer control */
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Capture Source",
.count = 1,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{ } /* end */
};
static struct snd_kcontrol_new alc861_3ST_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
/*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
/* Input mixer control */
/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
/* Capture mixer control */
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Capture Source",
.count = 1,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
.private_value = ARRAY_SIZE(alc861_threestack_modes),
},
{ } /* end */
};
static struct snd_kcontrol_new alc861_toshiba_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Master Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
/*Capture mixer control */
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Capture Source",
.count = 1,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{ } /* end */
};
static struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
/*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
/* Input mixer control */
/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
/* Capture mixer control */
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Capture Source",
.count = 1,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
.private_value = ARRAY_SIZE(alc861_uniwill_m31_modes),
},
{ } /* end */
};
static struct snd_kcontrol_new alc861_asus_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
/* Input mixer control */
HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_OUTPUT), /* was HDA_INPUT (why?) */
/* Capture mixer control */
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Capture Source",
.count = 1,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
.private_value = ARRAY_SIZE(alc861_asus_modes),
},
{ }
};
/* additional mixer */
static struct snd_kcontrol_new alc861_asus_laptop_mixer[] = {
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("PC Beep Playback Switch", 0x23, 0x0, HDA_OUTPUT),
{ }
};
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc861_base_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
/* port-A for surround (rear panel) */
{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0e, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-B for mic-in (rear panel) with vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-C for line-in (rear panel) */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* port-D for Front */
{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-E for HP out (front panel) */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-F for mic-in (front panel) with vref */
{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-G for CLFE (rear panel) */
{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x1f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-H for side (rear panel) */
{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* CD-in */
{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* route front mic to ADC1*/
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute DAC0~3 & spdif out*/
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mixer 14 (mic) 1c (Line in)*/
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Unmute Stereo Mixer 15 */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c }, //Output 0~12 step
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, // hp used DAC 3 (Front)
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
static struct hda_verb alc861_threestack_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
/* port-A for surround (rear panel) */
{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* port-B for mic-in (rear panel) with vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-C for line-in (rear panel) */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* port-D for Front */
{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-E for HP out (front panel) */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-F for mic-in (front panel) with vref */
{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-G for CLFE (rear panel) */
{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* port-H for side (rear panel) */
{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* CD-in */
{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* route front mic to ADC1*/
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute DAC0~3 & spdif out*/
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mixer 14 (mic) 1c (Line in)*/
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Unmute Stereo Mixer 15 */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c }, //Output 0~12 step
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, // hp used DAC 3 (Front)
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
static struct hda_verb alc861_uniwill_m31_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
/* port-A for surround (rear panel) */
{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* port-B for mic-in (rear panel) with vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-C for line-in (rear panel) */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* port-D for Front */
{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-E for HP out (front panel) */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, // this has to be set to VREF80
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-F for mic-in (front panel) with vref */
{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-G for CLFE (rear panel) */
{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* port-H for side (rear panel) */
{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* CD-in */
{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* route front mic to ADC1*/
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute DAC0~3 & spdif out*/
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mixer 14 (mic) 1c (Line in)*/
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Unmute Stereo Mixer 15 */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c }, //Output 0~12 step
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, // hp used DAC 3 (Front)
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
static struct hda_verb alc861_asus_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
/* port-A for surround (rear panel) | according to codec#0 this is the HP jack*/
{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, /* was 0x00 */
/* route front PCM to HP */
{ 0x0e, AC_VERB_SET_CONNECT_SEL, 0x01 },
/* port-B for mic-in (rear panel) with vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-C for line-in (rear panel) */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* port-D for Front */
{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-E for HP out (front panel) */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, /* this has to be set to VREF80 */
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-F for mic-in (front panel) with vref */
{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-G for CLFE (rear panel) */
{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* port-H for side (rear panel) */
{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* CD-in */
{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* route front mic to ADC1*/
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute DAC0~3 & spdif out*/
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mixer 14 (mic) 1c (Line in)*/
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Unmute Stereo Mixer 15 */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c }, /* Output 0~12 step */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, /* hp used DAC 3 (Front) */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
/* additional init verbs for ASUS laptops */
static struct hda_verb alc861_asus_laptop_init_verbs[] = {
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x45 }, /* HP-out */
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2) }, /* mute line-in */
{ }
};
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc861_auto_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
// {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)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, // set Mic 1
{ }
};
static struct hda_verb alc861_toshiba_init_verbs[] = {
{0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc861_toshiba_automute(struct hda_codec *codec)
{
unsigned int present;
present = snd_hda_codec_read(codec, 0x0f, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
snd_hda_codec_amp_update(codec, 0x16, 0, HDA_INPUT, 0,
0x80, present ? 0x80 : 0);
snd_hda_codec_amp_update(codec, 0x16, 1, HDA_INPUT, 0,
0x80, present ? 0x80 : 0);
snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_INPUT, 3,
0x80, present ? 0 : 0x80);
snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_INPUT, 3,
0x80, present ? 0 : 0x80);
}
static void alc861_toshiba_unsol_event(struct hda_codec *codec,
unsigned int res)
{
/* Looks like the unsol event is incompatible with the standard
* definition. 6bit tag is placed at 26 bit!
*/
if ((res >> 26) == ALC880_HP_EVENT)
alc861_toshiba_automute(codec);
}
/* pcm configuration: identiacal with ALC880 */
#define alc861_pcm_analog_playback alc880_pcm_analog_playback
#define alc861_pcm_analog_capture alc880_pcm_analog_capture
#define alc861_pcm_digital_playback alc880_pcm_digital_playback
#define alc861_pcm_digital_capture alc880_pcm_digital_capture
#define ALC861_DIGOUT_NID 0x07
static struct hda_channel_mode alc861_8ch_modes[1] = {
{ 8, NULL }
};
static hda_nid_t alc861_dac_nids[4] = {
/* front, surround, clfe, side */
0x03, 0x06, 0x05, 0x04
};
static hda_nid_t alc660_dac_nids[3] = {
/* front, clfe, surround */
0x03, 0x05, 0x06
};
static hda_nid_t alc861_adc_nids[1] = {
/* ADC0-2 */
0x08,
};
static struct hda_input_mux alc861_capture_source = {
.num_items = 5,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x3 },
{ "Line", 0x1 },
{ "CD", 0x4 },
{ "Mixer", 0x5 },
},
};
/* fill in the dac_nids table from the parsed pin configuration */
static int alc861_auto_fill_dac_nids(struct alc_spec *spec, const struct auto_pin_cfg *cfg)
{
int i;
hda_nid_t nid;
spec->multiout.dac_nids = spec->private_dac_nids;
for (i = 0; i < cfg->line_outs; i++) {
nid = cfg->line_out_pins[i];
if (nid) {
if (i >= ARRAY_SIZE(alc861_dac_nids))
continue;
spec->multiout.dac_nids[i] = alc861_dac_nids[i];
}
}
spec->multiout.num_dacs = cfg->line_outs;
return 0;
}
/* add playback controls from the parsed DAC table */
static int alc861_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
{
char name[32];
static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" };
hda_nid_t nid;
int i, idx, err;
for (i = 0; i < cfg->line_outs; i++) {
nid = spec->multiout.dac_nids[i];
if (! nid)
continue;
if (nid == 0x05) {
/* Center/LFE */
if ((err = add_control(spec, ALC_CTL_BIND_MUTE, "Center Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0)
return err;
if ((err = add_control(spec, ALC_CTL_BIND_MUTE, "LFE Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0)
return err;
} else {
for (idx = 0; idx < ARRAY_SIZE(alc861_dac_nids) - 1; idx++)
if (nid == alc861_dac_nids[idx])
break;
sprintf(name, "%s Playback Switch", chname[idx]);
if ((err = add_control(spec, ALC_CTL_BIND_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
return err;
}
}
return 0;
}
static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin)
{
int err;
hda_nid_t nid;
if (! pin)
return 0;
if ((pin >= 0x0b && pin <= 0x10) || pin == 0x1f || pin == 0x20) {
nid = 0x03;
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Headphone Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
return err;
spec->multiout.hp_nid = nid;
}
return 0;
}
/* create playback/capture controls for input pins */
static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg)
{
struct hda_input_mux *imux = &spec->private_imux;
int i, err, idx, idx1;
for (i = 0; i < AUTO_PIN_LAST; i++) {
switch(cfg->input_pins[i]) {
case 0x0c:
idx1 = 1;
idx = 2; // Line In
break;
case 0x0f:
idx1 = 2;
idx = 2; // Line In
break;
case 0x0d:
idx1 = 0;
idx = 1; // Mic In
break;
case 0x10:
idx1 = 3;
idx = 1; // Mic In
break;
case 0x11:
idx1 = 4;
idx = 0; // CD
break;
default:
continue;
}
err = new_analog_input(spec, cfg->input_pins[i],
auto_pin_cfg_labels[i], idx, 0x15);
if (err < 0)
return err;
imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
imux->items[imux->num_items].index = idx1;
imux->num_items++;
}
return 0;
}
static struct snd_kcontrol_new alc861_capture_mixer[] = {
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 */
};
static void alc861_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid,
int pin_type, int dac_idx)
{
/* set as output */
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
snd_hda_codec_write(codec, dac_idx, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
}
static void alc861_auto_init_multi_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < spec->autocfg.line_outs; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
if (nid)
alc861_auto_set_output_and_unmute(codec, nid, PIN_OUT, spec->multiout.dac_nids[i]);
}
}
static void alc861_auto_init_hp_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t pin;
pin = spec->autocfg.hp_pins[0];
if (pin) /* connect to front */
alc861_auto_set_output_and_unmute(codec, pin, PIN_HP, spec->multiout.dac_nids[0]);
}
static void alc861_auto_init_analog_input(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = spec->autocfg.input_pins[i];
if ((nid>=0x0c) && (nid <=0x11)) {
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
i <= AUTO_PIN_FRONT_MIC ? PIN_VREF80 : PIN_IN);
}
}
}
/* parse the BIOS configuration and set up the alc_spec */
/* return 1 if successful, 0 if the proper config is not found, or a negative error code */
static int alc861_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err;
static hda_nid_t alc861_ignore[] = { 0x1d, 0 };
if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc861_ignore)) < 0)
return err;
if (! spec->autocfg.line_outs)
return 0; /* can't find valid BIOS pin config */
if ((err = alc861_auto_fill_dac_nids(spec, &spec->autocfg)) < 0 ||
(err = alc861_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 ||
(err = alc861_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0])) < 0 ||
(err = alc861_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
if (spec->autocfg.dig_out_pin)
spec->multiout.dig_out_nid = ALC861_DIGOUT_NID;
if (spec->kctl_alloc)
spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
spec->init_verbs[spec->num_init_verbs++] = alc861_auto_init_verbs;
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux;
spec->adc_nids = alc861_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids);
spec->mixers[spec->num_mixers] = alc861_capture_mixer;
spec->num_mixers++;
return 1;
}
/* additional initialization for auto-configuration model */
static void alc861_auto_init(struct hda_codec *codec)
{
alc861_auto_init_multi_out(codec);
alc861_auto_init_hp_out(codec);
alc861_auto_init_analog_input(codec);
}
/*
* configuration and preset
*/
static const char *alc861_models[ALC861_MODEL_LAST] = {
[ALC861_3ST] = "3stack",
[ALC660_3ST] = "3stack-660",
[ALC861_3ST_DIG] = "3stack-dig",
[ALC861_6ST_DIG] = "6stack-dig",
[ALC861_UNIWILL_M31] = "uniwill-m31",
[ALC861_TOSHIBA] = "toshiba",
[ALC861_ASUS] = "asus",
[ALC861_ASUS_LAPTOP] = "asus-laptop",
[ALC861_AUTO] = "auto",
};
static struct snd_pci_quirk alc861_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP),
SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP),
SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660_3ST),
SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA),
SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31),
SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST),
{}
};
static struct alc_config_preset alc861_presets[] = {
[ALC861_3ST] = {
.mixers = { alc861_3ST_mixer },
.init_verbs = { alc861_threestack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
.channel_mode = alc861_threestack_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_3ST_DIG] = {
.mixers = { alc861_base_mixer },
.init_verbs = { alc861_threestack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
.channel_mode = alc861_threestack_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_6ST_DIG] = {
.mixers = { alc861_base_mixer },
.init_verbs = { alc861_base_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861_8ch_modes),
.channel_mode = alc861_8ch_modes,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC660_3ST] = {
.mixers = { alc861_3ST_mixer },
.init_verbs = { alc861_threestack_init_verbs },
.num_dacs = ARRAY_SIZE(alc660_dac_nids),
.dac_nids = alc660_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
.channel_mode = alc861_threestack_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_UNIWILL_M31] = {
.mixers = { alc861_uniwill_m31_mixer },
.init_verbs = { alc861_uniwill_m31_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861_uniwill_m31_modes),
.channel_mode = alc861_uniwill_m31_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_TOSHIBA] = {
.mixers = { alc861_toshiba_mixer },
.init_verbs = { alc861_base_init_verbs, alc861_toshiba_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
.unsol_event = alc861_toshiba_unsol_event,
.init_hook = alc861_toshiba_automute,
},
[ALC861_ASUS] = {
.mixers = { alc861_asus_mixer },
.init_verbs = { alc861_asus_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861_asus_modes),
.channel_mode = alc861_asus_modes,
.need_dac_fix = 1,
.hp_nid = 0x06,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_ASUS_LAPTOP] = {
.mixers = { alc861_toshiba_mixer, alc861_asus_laptop_mixer },
.init_verbs = { alc861_asus_init_verbs,
alc861_asus_laptop_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
};
static int patch_alc861(struct hda_codec *codec)
{
struct alc_spec *spec;
int board_config;
int err;
spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
board_config = snd_hda_check_board_config(codec, ALC861_MODEL_LAST,
alc861_models,
alc861_cfg_tbl);
if (board_config < 0) {
printk(KERN_INFO "hda_codec: Unknown model for ALC861, "
"trying auto-probe from BIOS...\n");
board_config = ALC861_AUTO;
}
if (board_config == ALC861_AUTO) {
/* automatic parse from the BIOS config */
err = alc861_parse_auto_config(codec);
if (err < 0) {
alc_free(codec);
return err;
} else if (! err) {
printk(KERN_INFO
"hda_codec: Cannot set up configuration "
"from BIOS. Using base mode...\n");
board_config = ALC861_3ST_DIG;
}
}
if (board_config != ALC861_AUTO)
setup_preset(spec, &alc861_presets[board_config]);
spec->stream_name_analog = "ALC861 Analog";
spec->stream_analog_playback = &alc861_pcm_analog_playback;
spec->stream_analog_capture = &alc861_pcm_analog_capture;
spec->stream_name_digital = "ALC861 Digital";
spec->stream_digital_playback = &alc861_pcm_digital_playback;
spec->stream_digital_capture = &alc861_pcm_digital_capture;
codec->patch_ops = alc_patch_ops;
if (board_config == ALC861_AUTO)
spec->init_hook = alc861_auto_init;
return 0;
}
/*
* patch entries
*/
struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
{ .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
{ .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 },
{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 },
{ .id = 0x10ec0861, .rev = 0x100300, .name = "ALC861",
.patch = patch_alc861 },
{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
.patch = patch_alc861 },
{ .id = 0x10ec0660, .name = "ALC660", .patch = patch_alc861 },
{} /* terminator */
};
