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linux / linux / kernel / git / gregkh / char-misc / refs/tags/v2.6.24-rc6 / . / sound / pci / hda / patch_realtek.c
blob: 1c502789cc1ed001a7360e50425694f3635260ce [file] [log] [blame]
/*
* 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_FUJITSU,
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,
ALC260_WILL,
ALC260_REPLACER_672V,
#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_HP_BPC_D7000_WL,
ALC262_HP_BPC_D7000_WF,
ALC262_BENQ_ED8,
ALC262_SONY_ASSAMD,
ALC262_BENQ_T31,
ALC262_AUTO,
ALC262_MODEL_LAST /* last tag */
};
/* ALC268 models */
enum {
ALC268_3ST,
ALC268_TOSHIBA,
ALC268_ACER,
ALC268_AUTO,
ALC268_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,
};
/* ALC861-VD models */
enum {
ALC660VD_3ST,
ALC660VD_3ST_DIG,
ALC861VD_3ST,
ALC861VD_3ST_DIG,
ALC861VD_6ST_DIG,
ALC861VD_LENOVO,
ALC861VD_DALLAS,
ALC861VD_HP,
ALC861VD_AUTO,
ALC861VD_MODEL_LAST,
};
/* ALC662 models */
enum {
ALC662_3ST_2ch_DIG,
ALC662_3ST_6ch_DIG,
ALC662_3ST_6ch,
ALC662_5ST_DIG,
ALC662_LENOVO_101E,
ALC662_ASUS_EEEPC_P701,
ALC662_AUTO,
ALC662_MODEL_LAST,
};
/* ALC882 models */
enum {
ALC882_3ST_DIG,
ALC882_6ST_DIG,
ALC882_ARIMA,
ALC882_W2JC,
ALC882_TARGA,
ALC882_ASUS_A7J,
ALC882_ASUS_A7M,
ALC885_MACPRO,
ALC885_MBP3,
ALC885_IMAC24,
ALC882_AUTO,
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,
ALC883_ACER,
ALC883_ACER_ASPIRE,
ALC883_MEDION,
ALC883_MEDION_MD2,
ALC883_LAPTOP_EAPD,
ALC883_LENOVO_101E_2ch,
ALC883_LENOVO_NB0763,
ALC888_LENOVO_MS7195_DIG,
ALC883_HAIER_W66,
ALC888_6ST_HP,
ALC888_3ST_HP,
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[AUTO_CFG_MAX_OUTS];
/* 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;
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_loopback_check loopback;
#endif
};
/*
* 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 *);
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_amp_list *loopbacks;
#endif
};
/*
* 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_cache(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
alc_pin_mode_values[val]);
/* Also enable the retasking pin's input/output as required
* for the requested pin mode. Enum values of 2 or less are
* input modes.
*
* Dynamically switching the input/output buffers probably
* reduces noise slightly (particularly on input) so we'll
* do it. However, having both input and output buffers
* enabled simultaneously doesn't seem to be problematic if
* this turns out to be necessary in the future.
*/
if (val <= 2) {
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
HDA_AMP_MUTE, 0);
} else {
snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, 0);
}
}
return change;
}
#define ALC_PIN_MODE(xname, nid, dir) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.info = alc_pin_mode_info, \
.get = alc_pin_mode_get, \
.put = alc_pin_mode_put, \
.private_value = nid | (dir<<16) }
/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged
* together using a mask with more than one bit set. This control is
* currently used only by the ALC260 test model. At this stage they are not
* needed for any "production" models.
*/
#ifdef CONFIG_SND_DEBUG
#define alc_gpio_data_info snd_ctl_boolean_mono_info
static int alc_gpio_data_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_GPIO_DATA, 0x00);
*valp = (val & mask) != 0;
return 0;
}
static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_GPIO_DATA,
0x00);
/* Set/unset the masked GPIO bit(s) as needed */
change = (val == 0 ? 0 : mask) != (gpio_data & mask);
if (val == 0)
gpio_data &= ~mask;
else
gpio_data |= mask;
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_GPIO_DATA, gpio_data);
return change;
}
#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.info = alc_gpio_data_info, \
.get = alc_gpio_data_get, \
.put = alc_gpio_data_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
/* A switch control to allow the enabling of the digital IO pins on the
* ALC260. This is incredibly simplistic; the intention of this control is
* to provide something in the test model allowing digital outputs to be
* identified if present. If models are found which can utilise these
* outputs a more complete mixer control can be devised for those models if
* necessary.
*/
#ifdef CONFIG_SND_DEBUG
#define alc_spdif_ctrl_info snd_ctl_boolean_mono_info
static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_DIGI_CONVERT, 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_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
ctrl_data);
return change;
}
#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.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;
#ifdef CONFIG_SND_HDA_POWER_SAVE
spec->loopback.amplist = preset->loopbacks;
#endif
}
/* Enable GPIO mask and set output */
static struct hda_verb alc_gpio1_init_verbs[] = {
{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
{ }
};
static struct hda_verb alc_gpio2_init_verbs[] = {
{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
{0x01, AC_VERB_SET_GPIO_DATA, 0x02},
{ }
};
static struct hda_verb alc_gpio3_init_verbs[] = {
{0x01, AC_VERB_SET_GPIO_MASK, 0x03},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03},
{0x01, AC_VERB_SET_GPIO_DATA, 0x03},
{ }
};
static void alc_sku_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int mute;
unsigned int present;
unsigned int hp_nid = spec->autocfg.hp_pins[0];
unsigned int sp_nid = spec->autocfg.speaker_pins[0];
/* need to execute and sync at first */
snd_hda_codec_read(codec, hp_nid, 0, AC_VERB_SET_PIN_SENSE, 0);
present = snd_hda_codec_read(codec, hp_nid, 0,
AC_VERB_GET_PIN_SENSE, 0);
spec->jack_present = (present & 0x80000000) != 0;
if (spec->jack_present) {
/* mute internal speaker */
snd_hda_codec_amp_stereo(codec, sp_nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
} else {
/* unmute internal speaker if necessary */
mute = snd_hda_codec_amp_read(codec, hp_nid, 0, HDA_OUTPUT, 0);
snd_hda_codec_amp_stereo(codec, sp_nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, mute);
}
}
/* unsolicited event for HP jack sensing */
static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
{
if (codec->vendor_id == 0x10ec0880)
res >>= 28;
else
res >>= 26;
if (res != ALC880_HP_EVENT)
return;
alc_sku_automute(codec);
}
/* 32-bit subsystem ID for BIOS loading in HD Audio codec.
* 31 ~ 16 : Manufacture ID
* 15 ~ 8 : SKU ID
* 7 ~ 0 : Assembly ID
* port-A --> pin 39/41, port-E --> pin 14/15, port-D --> pin 35/36
*/
static void alc_subsystem_id(struct hda_codec *codec,
unsigned int porta, unsigned int porte,
unsigned int portd)
{
unsigned int ass, tmp, i;
unsigned nid;
struct alc_spec *spec = codec->spec;
ass = codec->subsystem_id & 0xffff;
if ((ass != codec->bus->pci->subsystem_device) && (ass & 1))
goto do_sku;
/*
* 31~30 : port conetcivity
* 29~21 : reserve
* 20 : PCBEEP input
* 19~16 : Check sum (15:1)
* 15~1 : Custom
* 0 : override
*/
nid = 0x1d;
if (codec->vendor_id == 0x10ec0260)
nid = 0x17;
ass = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_CONFIG_DEFAULT, 0);
if (!(ass & 1) && !(ass & 0x100000))
return;
if ((ass >> 30) != 1) /* no physical connection */
return;
/* check sum */
tmp = 0;
for (i = 1; i < 16; i++) {
if ((ass >> i) && 1)
tmp++;
}
if (((ass >> 16) & 0xf) != tmp)
return;
do_sku:
/*
* 0 : override
* 1 : Swap Jack
* 2 : 0 --> Desktop, 1 --> Laptop
* 3~5 : External Amplifier control
* 7~6 : Reserved
*/
tmp = (ass & 0x38) >> 3; /* external Amp control */
switch (tmp) {
case 1:
snd_hda_sequence_write(codec, alc_gpio1_init_verbs);
break;
case 3:
snd_hda_sequence_write(codec, alc_gpio2_init_verbs);
break;
case 7:
snd_hda_sequence_write(codec, alc_gpio3_init_verbs);
break;
case 5: /* set EAPD output high */
switch (codec->vendor_id) {
case 0x10ec0260:
snd_hda_codec_write(codec, 0x0f, 0,
AC_VERB_SET_EAPD_BTLENABLE, 2);
snd_hda_codec_write(codec, 0x10, 0,
AC_VERB_SET_EAPD_BTLENABLE, 2);
break;
case 0x10ec0262:
case 0x10ec0267:
case 0x10ec0268:
case 0x10ec0269:
case 0x10ec0862:
case 0x10ec0662:
snd_hda_codec_write(codec, 0x14, 0,
AC_VERB_SET_EAPD_BTLENABLE, 2);
snd_hda_codec_write(codec, 0x15, 0,
AC_VERB_SET_EAPD_BTLENABLE, 2);
break;
}
switch (codec->vendor_id) {
case 0x10ec0260:
snd_hda_codec_write(codec, 0x1a, 0,
AC_VERB_SET_COEF_INDEX, 7);
tmp = snd_hda_codec_read(codec, 0x1a, 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 | 0x2010);
break;
case 0x10ec0262:
case 0x10ec0880:
case 0x10ec0882:
case 0x10ec0883:
case 0x10ec0885:
case 0x10ec0888:
snd_hda_codec_write(codec, 0x20, 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, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 7);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF,
tmp | 0x2010);
break;
case 0x10ec0267:
case 0x10ec0268:
snd_hda_codec_write(codec, 0x20, 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, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 7);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF,
tmp | 0x3000);
break;
}
default:
break;
}
/* is laptop and enable the function "Mute internal speaker
* when the external headphone out jack is plugged"
*/
if (!(ass & 0x4) || !(ass & 0x8000))
return;
/*
* 10~8 : Jack location
* 12~11: Headphone out -> 00: PortA, 01: PortE, 02: PortD, 03: Resvered
* 14~13: Resvered
* 15 : 1 --> enable the function "Mute internal speaker
* when the external headphone out jack is plugged"
*/
if (!spec->autocfg.speaker_pins[0]) {
if (spec->multiout.dac_nids[0])
spec->autocfg.speaker_pins[0] =
spec->multiout.dac_nids[0];
else
return;
}
if (!spec->autocfg.hp_pins[0]) {
tmp = (ass >> 11) & 0x3; /* HP to chassis */
if (tmp == 0)
spec->autocfg.hp_pins[0] = porta;
else if (tmp == 1)
spec->autocfg.hp_pins[0] = porte;
else if (tmp == 2)
spec->autocfg.hp_pins[0] = portd;
else
return;
}
snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0,
AC_VERB_SET_UNSOLICITED_ENABLE,
AC_USRSP_EN | ALC880_HP_EVENT);
spec->unsol_event = alc_sku_unsol_event;
spec->init_hook = alc_sku_automute;
}
/*
* Fix-up pin default configurations
*/
struct alc_pincfg {
hda_nid_t nid;
u32 val;
};
static void alc_fix_pincfg(struct hda_codec *codec,
const struct snd_pci_quirk *quirk,
const struct alc_pincfg **pinfix)
{
const struct alc_pincfg *cfg;
quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
if (!quirk)
return;
cfg = pinfix[quirk->value];
for (; cfg->nid; cfg++) {
int i;
u32 val = cfg->val;
for (i = 0; i < 4; i++) {
snd_hda_codec_write(codec, cfg->nid, 0,
AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i,
val & 0xff);
val >>= 8;
}
}
}
/*
* 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_fujitsu_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* 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_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
/*
* Set up output mixers (0x0c - 0x0f)
*/
/* set vol=0 to output mixers */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ }
};
/*
* 3-stack pin configuration:
* front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
*/
static 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},
{ }
};
static struct hda_verb alc880_beep_init_verbs[] = {
{ 0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) },
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc880_uniwill_hp_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
}
/* auto-toggle front mic */
static void alc880_uniwill_mic_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x18, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
}
static void alc880_uniwill_automute(struct hda_codec *codec)
{
alc880_uniwill_hp_automute(codec);
alc880_uniwill_mic_automute(codec);
}
static void alc880_uniwill_unsol_event(struct hda_codec *codec,
unsigned int res)
{
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
switch (res >> 28) {
case ALC880_HP_EVENT:
alc880_uniwill_hp_automute(codec);
break;
case ALC880_MIC_EVENT:
alc880_uniwill_mic_automute(codec);
break;
}
}
static void alc880_uniwill_p53_hp_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x15, HDA_INPUT, 0, HDA_AMP_MUTE, bits);
}
static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
{
unsigned int present;
present = snd_hda_codec_read(codec, 0x21, 0,
AC_VERB_GET_VOLUME_KNOB_CONTROL, 0);
present &= HDA_AMP_VOLMASK;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0,
HDA_AMP_VOLMASK, present);
snd_hda_codec_amp_stereo(codec, 0x0d, HDA_OUTPUT, 0,
HDA_AMP_VOLMASK, present);
}
static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
unsigned int res)
{
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
if ((res >> 28) == 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 */
#define alc880_gpio1_init_verbs alc_gpio1_init_verbs
#define alc880_gpio2_init_verbs alc_gpio2_init_verbs
/* 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_MUTE(6)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
/* line-in to input */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* built-in mic */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* speaker-out */
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* mic-in to input */
{0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* HP-out */
{0x13, AC_VERB_SET_CONNECT_SEL, 0x03},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* jack sense */
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc880_lg_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
}
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
* Line-In: 0x1b
* HP-Out: 0x1a
* SPDIF-Out: 0x1e
*/
static struct hda_input_mux alc880_lg_lw_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x0 },
{ "Internal Mic", 0x1 },
{ "Line In", 0x2 },
},
};
#define alc880_lg_lw_modes alc880_threestack_modes
static struct snd_kcontrol_new alc880_lg_lw_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
static struct hda_verb alc880_lg_lw_init_verbs[] = {
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
{0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
/* set capture source to mic-in */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
/* speaker-out */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* HP-out */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* mic-in to input */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* built-in mic */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* jack sense */
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc880_lg_lw_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
}
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);
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
static struct hda_amp_list alc880_loopbacks[] = {
{ 0x0b, HDA_INPUT, 0 },
{ 0x0b, HDA_INPUT, 1 },
{ 0x0b, HDA_INPUT, 2 },
{ 0x0b, HDA_INPUT, 3 },
{ 0x0b, HDA_INPUT, 4 },
{ } /* end */
};
static struct hda_amp_list alc880_lg_loopbacks[] = {
{ 0x0b, HDA_INPUT, 1 },
{ 0x0b, HDA_INPUT, 6 },
{ 0x0b, HDA_INPUT, 7 },
{ } /* end */
};
#endif
/*
* 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_SND_HDA_POWER_SAVE
static int alc_check_power_status(struct hda_codec *codec, hda_nid_t nid)
{
struct alc_spec *spec = codec->spec;
return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
}
#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_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_dig_prepare(codec, &spec->multiout,
stream_tag, format, substream);
}
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,
.prepare = alc880_dig_playback_pcm_prepare
},
};
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_SND_HDA_POWER_SAVE
.check_power_status = alc_check_power_status,
#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) {
int val;
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
new_ctl);
val = ucontrol->value.enumerated.item[0] >= 3 ?
HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, val);
return 1;
}
return 0;
}
static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static 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_cache(codec, nid, 0,
AC_VERB_SET_CONNECT_SEL, sel);
return 1;
}
return 0;
}
#define PIN_CTL_TEST(xname,nid) { \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
.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 */
};