blob: 26c70d6c11aff143345b2e8f2e1efebf2e51c750 [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 <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"
#include "hda_beep.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,
ALC880_MEDION_RIM,
#ifdef CONFIG_SND_DEBUG
ALC880_TEST,
#endif
ALC880_AUTO,
ALC880_MODEL_LAST /* last tag */
};
/* ALC260 models */
enum {
ALC260_BASIC,
ALC260_HP,
ALC260_HP_DC7600,
ALC260_HP_3013,
ALC260_FUJITSU_S702X,
ALC260_ACER,
ALC260_WILL,
ALC260_REPLACER_672V,
ALC260_FAVORIT100,
#ifdef CONFIG_SND_DEBUG
ALC260_TEST,
#endif
ALC260_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_HP_TC_T5735,
ALC262_HP_RP5700,
ALC262_BENQ_ED8,
ALC262_SONY_ASSAMD,
ALC262_BENQ_T31,
ALC262_ULTRA,
ALC262_LENOVO_3000,
ALC262_NEC,
ALC262_TOSHIBA_S06,
ALC262_TOSHIBA_RX1,
ALC262_TYAN,
ALC262_AUTO,
ALC262_MODEL_LAST /* last tag */
};
/* ALC268 models */
enum {
ALC267_QUANTA_IL1,
ALC268_3ST,
ALC268_TOSHIBA,
ALC268_ACER,
ALC268_ACER_DMIC,
ALC268_ACER_ASPIRE_ONE,
ALC268_DELL,
ALC268_ZEPTO,
#ifdef CONFIG_SND_DEBUG
ALC268_TEST,
#endif
ALC268_AUTO,
ALC268_MODEL_LAST /* last tag */
};
/* ALC269 models */
enum {
ALC269_BASIC,
ALC269_QUANTA_FL1,
ALC269_ASUS_EEEPC_P703,
ALC269_ASUS_EEEPC_P901,
ALC269_FUJITSU,
ALC269_LIFEBOOK,
ALC269_AUTO,
ALC269_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,
ALC660VD_ASUS_V1S,
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_ASUS_EEEPC_EP20,
ALC663_ASUS_M51VA,
ALC663_ASUS_G71V,
ALC663_ASUS_H13,
ALC663_ASUS_G50V,
ALC662_ECS,
ALC663_ASUS_MODE1,
ALC662_ASUS_MODE2,
ALC663_ASUS_MODE3,
ALC663_ASUS_MODE4,
ALC663_ASUS_MODE5,
ALC663_ASUS_MODE6,
ALC272_DELL,
ALC272_DELL_ZM1,
ALC272_SAMSUNG_NC10,
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_MB5,
ALC885_IMAC24,
ALC883_3ST_2ch_DIG,
ALC883_3ST_6ch_DIG,
ALC883_3ST_6ch,
ALC883_6ST_DIG,
ALC883_TARGA_DIG,
ALC883_TARGA_2ch_DIG,
ALC883_TARGA_8ch_DIG,
ALC883_ACER,
ALC883_ACER_ASPIRE,
ALC888_ACER_ASPIRE_4930G,
ALC888_ACER_ASPIRE_6530G,
ALC888_ACER_ASPIRE_8930G,
ALC888_ACER_ASPIRE_7730G,
ALC883_MEDION,
ALC883_MEDION_MD2,
ALC883_LAPTOP_EAPD,
ALC883_LENOVO_101E_2ch,
ALC883_LENOVO_NB0763,
ALC888_LENOVO_MS7195_DIG,
ALC888_LENOVO_SKY,
ALC883_HAIER_W66,
ALC888_3ST_HP,
ALC888_6ST_DELL,
ALC883_MITAC,
ALC883_CLEVO_M540R,
ALC883_CLEVO_M720,
ALC883_FUJITSU_PI2515,
ALC888_FUJITSU_XA3530,
ALC883_3ST_6ch_INTEL,
ALC889A_INTEL,
ALC889_INTEL,
ALC888_ASUS_M90V,
ALC888_ASUS_EEE1601,
ALC889A_MB31,
ALC1200_ASUS_P5Q,
ALC883_SONY_VAIO_TT,
ALC882_AUTO,
ALC882_MODEL_LAST,
};
/* for GPIO Poll */
#define GPIO_MASK 0x03
/* extra amp-initialization sequence types */
enum {
ALC_INIT_NONE,
ALC_INIT_DEFAULT,
ALC_INIT_GPIO1,
ALC_INIT_GPIO2,
ALC_INIT_GPIO3,
};
struct alc_mic_route {
hda_nid_t pin;
unsigned char mux_idx;
unsigned char amix_idx;
};
#define MUX_IDX_UNDEF ((unsigned char)-1)
struct alc_spec {
/* codec parameterization */
struct snd_kcontrol_new *mixers[5]; /* mixer arrays */
unsigned int num_mixers;
struct snd_kcontrol_new *cap_mixer; /* capture mixer */
unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */
const struct hda_verb *init_verbs[10]; /* initialization verbs
* don't forget NULL
* termination!
*/
unsigned int num_init_verbs;
char stream_name_analog[32]; /* analog PCM stream */
struct hda_pcm_stream *stream_analog_playback;
struct hda_pcm_stream *stream_analog_capture;
struct hda_pcm_stream *stream_analog_alt_playback;
struct hda_pcm_stream *stream_analog_alt_capture;
char stream_name_digital[32]; /* 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
*/
hda_nid_t alt_dac_nid;
hda_nid_t slave_dig_outs[3]; /* optional - for auto-parsing */
int dig_out_type;
/* capture */
unsigned int num_adc_nids;
hda_nid_t *adc_nids;
hda_nid_t *capsrc_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];
struct alc_mic_route ext_mic;
struct alc_mic_route int_mic;
/* channel model */
const struct hda_channel_mode *channel_mode;
int num_channel_mode;
int need_dac_fix;
int const_channel_count;
int ext_channel_count;
/* 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;
struct snd_array kctls;
struct hda_input_mux private_imux[3];
hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
hda_nid_t private_adc_nids[AUTO_CFG_MAX_OUTS];
hda_nid_t private_capsrc_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;
unsigned int master_sw: 1;
unsigned int auto_mic:1;
/* other flags */
unsigned int no_analog :1; /* digital I/O only */
int init_amp;
/* for virtual master */
hda_nid_t vmaster_nid;
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_loopback_check loopback;
#endif
/* for PLL fix */
hda_nid_t pll_nid;
unsigned int pll_coef_idx, pll_coef_bit;
};
/*
* 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
*/
struct snd_kcontrol_new *cap_mixer; /* capture mixer */
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 */
hda_nid_t *slave_dig_outs;
unsigned int num_adc_nids;
hda_nid_t *adc_nids;
hda_nid_t *capsrc_nids;
hda_nid_t dig_in_nid;
unsigned int num_channel_mode;
const struct hda_channel_mode *channel_mode;
int need_dac_fix;
int const_channel_count;
unsigned int num_mux_defs;
const struct hda_input_mux *input_mux;
void (*unsol_event)(struct hda_codec *, unsigned int);
void (*setup)(struct hda_codec *);
void (*init_hook)(struct hda_codec *);
#ifdef CONFIG_SND_HDA_POWER_SAVE
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;
if (!spec->input_mux[mux_idx].num_items && mux_idx > 0)
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;
const struct hda_input_mux *imux;
unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
unsigned int mux_idx;
hda_nid_t nid = spec->capsrc_nids ?
spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx];
unsigned int type;
mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
imux = &spec->input_mux[mux_idx];
if (!imux->num_items && mux_idx > 0)
imux = &spec->input_mux[0];
type = get_wcaps_type(get_wcaps(codec, nid));
if (type == AC_WID_AUD_MIX) {
/* Matrix-mixer style (e.g. ALC882) */
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)
return 0;
for (i = 0; i < imux->num_items; i++) {
unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
imux->items[i].index,
HDA_AMP_MUTE, v);
}
*cur_val = idx;
return 1;
} else {
/* MUX style (e.g. ALC880) */
return snd_hda_input_mux_put(codec, imux, ucontrol, nid,
&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->ext_channel_count);
}
static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
spec->num_channel_mode,
&spec->ext_channel_count);
if (err >= 0 && !spec->const_channel_count) {
spec->multiout.max_channels = spec->ext_channel_count;
if (spec->need_dac_fix)
spec->multiout.num_dacs = spec->multiout.max_channels / 2;
}
return err;
}
/*
* Control the mode of pin widget settings via the mixer. "pc" is used
* instead of "%" to avoid consequences of 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 (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl)
i++;
*valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
return 0;
}
static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL,
0x00);
if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir))
val = alc_pin_mode_min(dir);
change = pinctl != alc_pin_mode_values[val];
if (change) {
/* Set pin mode to that requested */
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
alc_pin_mode_values[val]);
/* Also enable the retasking pin's input/output as required
* for the requested pin mode. Enum values of 2 or less are
* input modes.
*
* Dynamically switching the input/output buffers probably
* reduces noise slightly (particularly on input) so we'll
* do it. However, having both input and output buffers
* enabled simultaneously doesn't seem to be problematic if
* this turns out to be necessary in the future.
*/
if (val <= 2) {
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
HDA_AMP_MUTE, 0);
} else {
snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, 0);
}
}
return change;
}
#define ALC_PIN_MODE(xname, nid, dir) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.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_1, 0x00);
*valp = (val & mask) != 0;
return 0;
}
static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_DIGI_CONVERT_1,
0x00);
/* Set/unset the masked control bit(s) as needed */
change = (val == 0 ? 0 : mask) != (ctrl_data & mask);
if (val==0)
ctrl_data &= ~mask;
else
ctrl_data |= mask;
snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
ctrl_data);
return change;
}
#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.info = alc_spdif_ctrl_info, \
.get = alc_spdif_ctrl_get, \
.put = alc_spdif_ctrl_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
/* A switch control to allow the enabling EAPD digital outputs on the ALC26x.
* Again, this is only used in the ALC26x test models to help identify when
* the EAPD line must be asserted for features to work.
*/
#ifdef CONFIG_SND_DEBUG
#define alc_eapd_ctrl_info snd_ctl_boolean_mono_info
static int alc_eapd_ctrl_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_EAPD_BTLENABLE, 0x00);
*valp = (val & mask) != 0;
return 0;
}
static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_EAPD_BTLENABLE,
0x00);
/* Set/unset the masked control bit(s) as needed */
change = (!val ? 0 : mask) != (ctrl_data & mask);
if (!val)
ctrl_data &= ~mask;
else
ctrl_data |= mask;
snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
ctrl_data);
return change;
}
#define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.info = alc_eapd_ctrl_info, \
.get = alc_eapd_ctrl_get, \
.put = alc_eapd_ctrl_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
/*
* set up the input pin config (depending on the given auto-pin type)
*/
static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid,
int auto_pin_type)
{
unsigned int val = PIN_IN;
if (auto_pin_type <= AUTO_PIN_FRONT_MIC) {
unsigned int pincap;
pincap = snd_hda_query_pin_caps(codec, nid);
pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
if (pincap & AC_PINCAP_VREF_80)
val = PIN_VREF80;
else if (pincap & AC_PINCAP_VREF_50)
val = PIN_VREF50;
else if (pincap & AC_PINCAP_VREF_100)
val = PIN_VREF100;
else if (pincap & AC_PINCAP_VREF_GRD)
val = PIN_VREFGRD;
}
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val);
}
/*
*/
static void add_mixer(struct alc_spec *spec, struct snd_kcontrol_new *mix)
{
if (snd_BUG_ON(spec->num_mixers >= ARRAY_SIZE(spec->mixers)))
return;
spec->mixers[spec->num_mixers++] = mix;
}
static void add_verb(struct alc_spec *spec, const struct hda_verb *verb)
{
if (snd_BUG_ON(spec->num_init_verbs >= ARRAY_SIZE(spec->init_verbs)))
return;
spec->init_verbs[spec->num_init_verbs++] = verb;
}
#ifdef CONFIG_PROC_FS
/*
* hook for proc
*/
static void print_realtek_coef(struct snd_info_buffer *buffer,
struct hda_codec *codec, hda_nid_t nid)
{
int coeff;
if (nid != 0x20)
return;
coeff = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PROC_COEF, 0);
snd_iprintf(buffer, " Processing Coefficient: 0x%02x\n", coeff);
coeff = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_COEF_INDEX, 0);
snd_iprintf(buffer, " Coefficient Index: 0x%02x\n", coeff);
}
#else
#define print_realtek_coef NULL
#endif
/*
* set up from the preset table
*/
static void setup_preset(struct hda_codec *codec,
const struct alc_config_preset *preset)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
add_mixer(spec, preset->mixers[i]);
spec->cap_mixer = preset->cap_mixer;
for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i];
i++)
add_verb(spec, preset->init_verbs[i]);
spec->channel_mode = preset->channel_mode;
spec->num_channel_mode = preset->num_channel_mode;
spec->need_dac_fix = preset->need_dac_fix;
spec->const_channel_count = preset->const_channel_count;
if (preset->const_channel_count)
spec->multiout.max_channels = preset->const_channel_count;
else
spec->multiout.max_channels = spec->channel_mode[0].channels;
spec->ext_channel_count = spec->channel_mode[0].channels;
spec->multiout.num_dacs = preset->num_dacs;
spec->multiout.dac_nids = preset->dac_nids;
spec->multiout.dig_out_nid = preset->dig_out_nid;
spec->multiout.slave_dig_outs = preset->slave_dig_outs;
spec->multiout.hp_nid = preset->hp_nid;
spec->num_mux_defs = preset->num_mux_defs;
if (!spec->num_mux_defs)
spec->num_mux_defs = 1;
spec->input_mux = preset->input_mux;
spec->num_adc_nids = preset->num_adc_nids;
spec->adc_nids = preset->adc_nids;
spec->capsrc_nids = preset->capsrc_nids;
spec->dig_in_nid = preset->dig_in_nid;
spec->unsol_event = preset->unsol_event;
spec->init_hook = preset->init_hook;
#ifdef CONFIG_SND_HDA_POWER_SAVE
spec->loopback.amplist = preset->loopbacks;
#endif
if (preset->setup)
preset->setup(codec);
}
/* 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},
{ }
};
/*
* Fix hardware PLL issue
* On some codecs, the analog PLL gating control must be off while
* the default value is 1.
*/
static void alc_fix_pll(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int val;
if (!spec->pll_nid)
return;
snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX,
spec->pll_coef_idx);
val = snd_hda_codec_read(codec, spec->pll_nid, 0,
AC_VERB_GET_PROC_COEF, 0);
snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX,
spec->pll_coef_idx);
snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_PROC_COEF,
val & ~(1 << spec->pll_coef_bit));
}
static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid,
unsigned int coef_idx, unsigned int coef_bit)
{
struct alc_spec *spec = codec->spec;
spec->pll_nid = nid;
spec->pll_coef_idx = coef_idx;
spec->pll_coef_bit = coef_bit;
alc_fix_pll(codec);
}
static void alc_automute_pin(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int present, pincap;
unsigned int nid = spec->autocfg.hp_pins[0];
int i;
if (!nid)
return;
pincap = snd_hda_query_pin_caps(codec, nid);
if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
present = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_SENSE, 0);
spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) {
nid = spec->autocfg.speaker_pins[i];
if (!nid)
break;
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
spec->jack_present ? 0 : PIN_OUT);
}
}
static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
hda_nid_t nid)
{
hda_nid_t conn[HDA_MAX_NUM_INPUTS];
int i, nums;
nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
for (i = 0; i < nums; i++)
if (conn[i] == nid)
return i;
return -1;
}
static void alc_mic_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct alc_mic_route *dead, *alive;
unsigned int present, type;
hda_nid_t cap_nid;
if (!spec->auto_mic)
return;
if (!spec->int_mic.pin || !spec->ext_mic.pin)
return;
if (snd_BUG_ON(!spec->adc_nids))
return;
cap_nid = spec->capsrc_nids ? spec->capsrc_nids[0] : spec->adc_nids[0];
present = snd_hda_codec_read(codec, spec->ext_mic.pin, 0,
AC_VERB_GET_PIN_SENSE, 0);
present &= AC_PINSENSE_PRESENCE;
if (present) {
alive = &spec->ext_mic;
dead = &spec->int_mic;
} else {
alive = &spec->int_mic;
dead = &spec->ext_mic;
}
type = get_wcaps_type(get_wcaps(codec, cap_nid));
if (type == AC_WID_AUD_MIX) {
/* Matrix-mixer style (e.g. ALC882) */
snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT,
alive->mux_idx,
HDA_AMP_MUTE, 0);
snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT,
dead->mux_idx,
HDA_AMP_MUTE, HDA_AMP_MUTE);
} else {
/* MUX style (e.g. ALC880) */
snd_hda_codec_write_cache(codec, cap_nid, 0,
AC_VERB_SET_CONNECT_SEL,
alive->mux_idx);
}
/* FIXME: analog mixer */
}
/* 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;
switch (res) {
case ALC880_HP_EVENT:
alc_automute_pin(codec);
break;
case ALC880_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
static void alc_inithook(struct hda_codec *codec)
{
alc_automute_pin(codec);
alc_mic_automute(codec);
}
/* additional initialization for ALC888 variants */
static void alc888_coef_init(struct hda_codec *codec)
{
unsigned int tmp;
snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0);
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);
if ((tmp & 0xf0) == 0x20)
/* alc888S-VC */
snd_hda_codec_read(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF, 0x830);
else
/* alc888-VB */
snd_hda_codec_read(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF, 0x3030);
}
static void alc889_coef_init(struct hda_codec *codec)
{
unsigned int tmp;
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);
}
static void alc_auto_init_amp(struct hda_codec *codec, int type)
{
unsigned int tmp;
switch (type) {
case ALC_INIT_GPIO1:
snd_hda_sequence_write(codec, alc_gpio1_init_verbs);
break;
case ALC_INIT_GPIO2:
snd_hda_sequence_write(codec, alc_gpio2_init_verbs);
break;
case ALC_INIT_GPIO3:
snd_hda_sequence_write(codec, alc_gpio3_init_verbs);
break;
case ALC_INIT_DEFAULT:
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 0x10ec0272:
case 0x10ec0660:
case 0x10ec0662:
case 0x10ec0663:
case 0x10ec0862:
case 0x10ec0889:
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 0x10ec0887:
case 0x10ec0889:
alc889_coef_init(codec);
break;
case 0x10ec0888:
alc888_coef_init(codec);
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;
}
break;
}
}
static void alc_init_auto_hp(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
if (!spec->autocfg.hp_pins[0])
return;
if (!spec->autocfg.speaker_pins[0]) {
if (spec->autocfg.line_out_pins[0] &&
spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
spec->autocfg.speaker_pins[0] =
spec->autocfg.line_out_pins[0];
else
return;
}
snd_printdd("realtek: Enable HP auto-muting on NID 0x%x\n",
spec->autocfg.hp_pins[0]);
snd_hda_codec_write_cache(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;
}
static void alc_init_auto_mic(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
hda_nid_t fixed, ext;
int i;
/* there must be only two mic inputs exclusively */
for (i = AUTO_PIN_LINE; i < AUTO_PIN_LAST; i++)
if (cfg->input_pins[i])
return;
fixed = ext = 0;
for (i = AUTO_PIN_MIC; i <= AUTO_PIN_FRONT_MIC; i++) {
hda_nid_t nid = cfg->input_pins[i];
unsigned int defcfg;
if (!nid)
return;
defcfg = snd_hda_codec_get_pincfg(codec, nid);
switch (get_defcfg_connect(defcfg)) {
case AC_JACK_PORT_FIXED:
if (fixed)
return; /* already occupied */
fixed = nid;
break;
case AC_JACK_PORT_COMPLEX:
if (ext)
return; /* already occupied */
ext = nid;
break;
default:
return; /* invalid entry */
}
}
if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP))
return; /* no unsol support */
snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n",
ext, fixed);
spec->ext_mic.pin = ext;
spec->int_mic.pin = fixed;
spec->ext_mic.mux_idx = MUX_IDX_UNDEF; /* set later */
spec->int_mic.mux_idx = MUX_IDX_UNDEF; /* set later */
spec->auto_mic = 1;
snd_hda_codec_write_cache(codec, spec->ext_mic.pin, 0,
AC_VERB_SET_UNSOLICITED_ENABLE,
AC_USRSP_EN | ALC880_MIC_EVENT);
spec->unsol_event = alc_sku_unsol_event;
}
/* check subsystem ID and set up device-specific initialization;
* return 1 if initialized, 0 if invalid SSID
*/
/* 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 int alc_subsystem_id(struct hda_codec *codec,
hda_nid_t porta, hda_nid_t porte,
hda_nid_t 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;
/* invalid SSID, check the special NID pin defcfg instead */
/*
* 31~30 : port connectivity
* 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_get_pincfg(codec, nid);
snd_printd("realtek: No valid SSID, "
"checking pincfg 0x%08x for NID 0x%x\n",
ass, nid);
if (!(ass & 1) && !(ass & 0x100000))
return 0;
if ((ass >> 30) != 1) /* no physical connection */
return 0;
/* check sum */
tmp = 0;
for (i = 1; i < 16; i++) {
if ((ass >> i) & 1)
tmp++;
}
if (((ass >> 16) & 0xf) != tmp)
return 0;
do_sku:
snd_printd("realtek: Enabling init ASM_ID=0x%04x CODEC_ID=%08x\n",
ass & 0xffff, codec->vendor_id);
/*
* 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:
spec->init_amp = ALC_INIT_GPIO1;
break;
case 3:
spec->init_amp = ALC_INIT_GPIO2;
break;
case 7:
spec->init_amp = ALC_INIT_GPIO3;
break;
case 5:
spec->init_amp = ALC_INIT_DEFAULT;
break;
}
/* is laptop or Desktop and enable the function "Mute internal speaker
* when the external headphone out jack is plugged"
*/
if (!(ass & 0x8000))
return 1;
/*
* 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.hp_pins[0]) {
hda_nid_t nid;
tmp = (ass >> 11) & 0x3; /* HP to chassis */
if (tmp == 0)
nid = porta;
else if (tmp == 1)
nid = porte;
else if (tmp == 2)
nid = portd;
else
return 1;
for (i = 0; i < spec->autocfg.line_outs; i++)
if (spec->autocfg.line_out_pins[i] == nid)
return 1;
spec->autocfg.hp_pins[0] = nid;
}
alc_init_auto_hp(codec);
alc_init_auto_mic(codec);
return 1;
}
static void alc_ssid_check(struct hda_codec *codec,
hda_nid_t porta, hda_nid_t porte, hda_nid_t portd)
{
if (!alc_subsystem_id(codec, porta, porte, portd)) {
struct alc_spec *spec = codec->spec;
snd_printd("realtek: "
"Enable default setup for auto mode as fallback\n");
spec->init_amp = ALC_INIT_DEFAULT;
alc_init_auto_hp(codec);
alc_init_auto_mic(codec);
}
}
/*
* Fix-up pin default configurations and add default verbs
*/
struct alc_pincfg {
hda_nid_t nid;
u32 val;
};
struct alc_fixup {
const struct alc_pincfg *pins;
const struct hda_verb *verbs;
};
static void alc_pick_fixup(struct hda_codec *codec,
const struct snd_pci_quirk *quirk,
const struct alc_fixup *fix)
{
const struct alc_pincfg *cfg;
quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
if (!quirk)
return;
fix += quirk->value;
cfg = fix->pins;
if (cfg) {
for (; cfg->nid; cfg++)
snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
}
if (fix->verbs)
add_verb(codec->spec, fix->verbs);
}
/*
* ALC888
*/
/*
* 2ch mode
*/
static struct hda_verb alc888_4ST_ch2_intel_init[] = {
/* Mic-in jack as mic in */
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
/* Line-in jack as Line in */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
/* Line-Out as Front */
{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
{ } /* end */
};
/*
* 4ch mode
*/
static struct hda_verb alc888_4ST_ch4_intel_init[] = {
/* Mic-in jack as mic in */
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
/* Line-in jack as Surround */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
/* Line-Out as Front */
{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
{ } /* end */
};
/*
* 6ch mode
*/
static struct hda_verb alc888_4ST_ch6_intel_init[] = {
/* Mic-in jack as CLFE */
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
/* Line-in jack as Surround */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
/* Line-Out as CLFE (workaround because Mic-in is not loud enough) */
{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
{ } /* end */
};
/*
* 8ch mode
*/
static struct hda_verb alc888_4ST_ch8_intel_init[] = {
/* Mic-in jack as CLFE */
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
/* Line-in jack as Surround */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
/* Line-Out as Side */
{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
{ } /* end */
};
static struct hda_channel_mode alc888_4ST_8ch_intel_modes[4] = {
{ 2, alc888_4ST_ch2_intel_init },
{ 4, alc888_4ST_ch4_intel_init },
{ 6, alc888_4ST_ch6_intel_init },
{ 8, alc888_4ST_ch8_intel_init },
};
/*
* ALC888 Fujitsu Siemens Amillo xa3530
*/
static struct hda_verb alc888_fujitsu_xa3530_verbs[] = {
/* Front Mic: set to PIN_IN (empty by default) */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Connect Internal HP to Front */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Connect Bass HP to Front */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Connect Line-Out side jack (SPDIF) to Side */
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
/* Connect Mic jack to CLFE */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x02},
/* Connect Line-in jack to Surround */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
/* Connect HP out jack to Front */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Enable unsolicited event for HP jack and Line-out jack */
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{0x17, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{}
};
static void alc_automute_amp(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int val, mute, pincap;
hda_nid_t nid;
int i;
spec->jack_present = 0;
for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) {
nid = spec->autocfg.hp_pins[i];
if (!nid)
break;
pincap = snd_hda_query_pin_caps(codec, nid);
if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
snd_hda_codec_read(codec, nid, 0,
AC_VERB_SET_PIN_SENSE, 0);
val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_SENSE, 0);
if (val & AC_PINSENSE_PRESENCE) {
spec->jack_present = 1;
break;
}
}
mute = spec->jack_present ? HDA_AMP_MUTE : 0;
/* Toggle internal speakers muting */
for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) {
nid = spec->autocfg.speaker_pins[i];
if (!nid)
break;
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, mute);
}
}
static void alc_automute_amp_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if (codec->vendor_id == 0x10ec0880)
res >>= 28;
else
res >>= 26;
if (res == ALC880_HP_EVENT)
alc_automute_amp(codec);
}
static void alc889_automute_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x17;
spec->autocfg.speaker_pins[3] = 0x19;
spec->autocfg.speaker_pins[4] = 0x1a;
}
static void alc889_intel_init_hook(struct hda_codec *codec)
{
alc889_coef_init(codec);
alc_automute_amp(codec);
}
static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x17; /* line-out */
spec->autocfg.hp_pins[1] = 0x1b; /* hp */
spec->autocfg.speaker_pins[0] = 0x14; /* speaker */
spec->autocfg.speaker_pins[1] = 0x15; /* bass */
}
/*
* ALC888 Acer Aspire 4930G model
*/
static struct hda_verb alc888_acer_aspire_4930g_verbs[] = {
/* Front Mic: set to PIN_IN (empty by default) */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Unselect Front Mic by default in input mixer 3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
/* Enable unsolicited event for HP jack */
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
/* Connect Internal HP to front */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Connect HP out to front */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{ }
};
/*
* ALC888 Acer Aspire 6530G model
*/
static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
/* Bias voltage on for external mic port */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
/* Front Mic: set to PIN_IN (empty by default) */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Unselect Front Mic by default in input mixer 3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
/* Enable unsolicited event for HP jack */
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
/* Enable speaker output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Enable headphone output */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{ }
};
/*
* ALC889 Acer Aspire 8930G model
*/
static struct hda_verb alc889_acer_aspire_8930g_verbs[] = {
/* Front Mic: set to PIN_IN (empty by default) */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Unselect Front Mic by default in input mixer 3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
/* Enable unsolicited event for HP jack */
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
/* Connect Internal Front to Front */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Connect Internal Rear to Rear */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x01},
/* Connect Internal CLFE to CLFE */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
/* Connect HP out to Front */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Enable all DACs */
/* DAC DISABLE/MUTE 1? */
/* setting bits 1-5 disables DAC nids 0x02-0x06 apparently. Init=0x38 */
{0x20, AC_VERB_SET_COEF_INDEX, 0x03},
{0x20, AC_VERB_SET_PROC_COEF, 0x0000},
/* DAC DISABLE/MUTE 2? */
/* some bit here disables the other DACs. Init=0x4900 */
{0x20, AC_VERB_SET_COEF_INDEX, 0x08},
{0x20, AC_VERB_SET_PROC_COEF, 0x0000},
/* Enable amplifiers */
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
{0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
/* DMIC fix
* This laptop has a stereo digital microphone. The mics are only 1cm apart
* which makes the stereo useless. However, either the mic or the ALC889
* makes the signal become a difference/sum signal instead of standard
* stereo, which is annoying. So instead we flip this bit which makes the
* codec replicate the sum signal to both channels, turning it into a
* normal mono mic.
*/
/* DMIC_CONTROL? Init value = 0x0001 */
{0x20, AC_VERB_SET_COEF_INDEX, 0x0b},
{0x20, AC_VERB_SET_PROC_COEF, 0x0003},
{ }
};
static struct hda_input_mux alc888_2_capture_sources[2] = {
/* Front mic only available on one ADC */
{
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Front Mic", 0xb },
},
},
{
.num_items = 3,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
}
};
static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = {
/* Interal mic only available on one ADC */
{
.num_items = 5,
.items = {
{ "Ext Mic", 0x0 },
{ "Line In", 0x2 },
{ "CD", 0x4 },
{ "Input Mix", 0xa },
{ "Int Mic", 0xb },
},
},
{
.num_items = 4,
.items = {
{ "Ext Mic", 0x0 },
{ "Line In", 0x2 },
{ "CD", 0x4 },
{ "Input Mix", 0xa },
},
}
};
static struct hda_input_mux alc889_capture_sources[3] = {
/* Digital mic only available on first "ADC" */
{
.num_items = 5,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Front Mic", 0xb },
{ "Input Mix", 0xa },
},
},
{
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Input Mix", 0xa },
},
},
{
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Input Mix", 0xa },
},
}
};
static struct snd_kcontrol_new alc888_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
}
static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x17;
}
static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x1b;
}
/*
* 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_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 int alc_cap_vol_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;
int err;
mutex_lock(&codec->control_mutex);
kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0,
HDA_INPUT);
err = snd_hda_mixer_amp_volume_info(kcontrol, uinfo);
mutex_unlock(&codec->control_mutex);
return err;
}
static int alc_cap_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
unsigned int size, unsigned int __user *tlv)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
int err;
mutex_lock(&codec->control_mutex);
kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0,
HDA_INPUT);
err = snd_hda_mixer_amp_tlv(kcontrol, op_flag, size, tlv);
mutex_unlock(&codec->control_mutex);
return err;
}
typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol,
getput_call_t func)
{
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);
int err;
mutex_lock(&codec->control_mutex);
kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx],
3, 0, HDA_INPUT);
err = func(kcontrol, ucontrol);
mutex_unlock(&codec->control_mutex);
return err;
}
static int alc_cap_vol_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
snd_hda_mixer_amp_volume_get);
}
static int alc_cap_vol_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
snd_hda_mixer_amp_volume_put);
}
/* capture mixer elements */
#define alc_cap_sw_info snd_ctl_boolean_stereo_info
static int alc_cap_sw_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
snd_hda_mixer_amp_switch_get);
}
static int alc_cap_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
snd_hda_mixer_amp_switch_put);
}
#define _DEFINE_CAPMIX(num) \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = "Capture Switch", \
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
.count = num, \
.info = alc_cap_sw_info, \
.get = alc_cap_sw_get, \
.put = alc_cap_sw_put, \
}, \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = "Capture Volume", \
.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | \
SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK), \
.count = num, \
.info = alc_cap_vol_info, \
.get = alc_cap_vol_get, \
.put = alc_cap_vol_put, \
.tlv = { .c = alc_cap_vol_tlv }, \
}
#define _DEFINE_CAPSRC(num) \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
/* .name = "Capture Source", */ \
.name = "Input Source", \
.count = num, \
.info = alc_mux_enum_info, \
.get = alc_mux_enum_get, \
.put = alc_mux_enum_put, \
}
#define DEFINE_CAPMIX(num) \
static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \
_DEFINE_CAPMIX(num), \
_DEFINE_CAPSRC(num), \
{ } /* end */ \
}
#define DEFINE_CAPMIX_NOSRC(num) \
static struct snd_kcontrol_new alc_capture_mixer_nosrc ## num[] = { \
_DEFINE_CAPMIX(num), \
{ } /* end */ \
}
/* up to three ADCs */
DEFINE_CAPMIX(1);
DEFINE_CAPMIX(2);
DEFINE_CAPMIX(3);
DEFINE_CAPMIX_NOSRC(1);
DEFINE_CAPMIX_NOSRC(2);
DEFINE_CAPMIX_NOSRC(3);
/*
* 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),
{
.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 */
};
/*
* 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("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct hda_input_mux alc880_f1734_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x1 },
{ "CD", 0x4 },
},
};
/*
* ALC880 ASUS model
*
* DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
* Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
* Mic = 0x18, Line = 0x1a
*/
#define alc880_asus_dac_nids alc880_w810_dac_nids /* identical with w810 */
#define alc880_asus_modes alc880_threestack_modes /* 2/6 channel mode */
static struct snd_kcontrol_new alc880_asus_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
/*
* ALC880 ASUS W1V model
*
* DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
* Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
* Mic = 0x18, Line = 0x1a, Line2 = 0x1b
*/
/* additional mixers to alc880_asus_mixer */
static struct snd_kcontrol_new alc880_asus_w1v_mixer[] = {
HDA_CODEC_VOLUME("Line2 Playback Volume", 0x0b, 0x03, HDA_INPUT),
HDA_CODEC_MUTE("Line2 Playback Switch", 0x0b, 0x03, HDA_INPUT),
{ } /* end */
};
/* TCL S700 */
static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0B, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0B, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0B, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0B, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
{ } /* end */
};
/* Uniwill */
static struct snd_kcontrol_new alc880_uniwill_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
static 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("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
/*
* virtual master controls
*/
/*
* slave controls for virtual master
*/
static const char *alc_slave_vols[] = {
"Front Playback Volume",
"Surround Playback Volume",
"Center Playback Volume",
"LFE Playback Volume",
"Side Playback Volume",
"Headphone Playback Volume",
"Speaker Playback Volume",
"Mono Playback Volume",
"Line-Out Playback Volume",
"PCM Playback Volume",
NULL,
};
static const char *alc_slave_sws[] = {
"Front Playback Switch",
"Surround Playback Switch",
"Center Playback Switch",
"LFE Playback Switch",
"Side Playback Switch",
"Headphone Playback Switch",
"Speaker Playback Switch",
"Mono Playback Switch",
"IEC958 Playback Switch",
"Line-Out Playback Switch",
"PCM Playback Switch",
NULL,
};
/*
* build control elements
*/
static void alc_free_kctls(struct hda_codec *codec);
/* additional beep mixers; the actual parameters are overwritten at build */
static struct snd_kcontrol_new alc_beep_mixer[] = {
HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT),
HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_INPUT),
{ } /* end */
};
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->cap_mixer) {
err = snd_hda_add_new_ctls(codec, spec->cap_mixer);
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->no_analog) {
err = snd_hda_create_spdif_share_sw(codec,
&spec->multiout);
if (err < 0)
return err;
spec->multiout.share_spdif = 1;
}
}
if (spec->dig_in_nid) {
err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
if (err < 0)
return err;
}
/* create beep controls if needed */
if (spec->beep_amp) {
struct snd_kcontrol_new *knew;
for (knew = alc_beep_mixer; knew->name; knew++) {
struct snd_kcontrol *kctl;
kctl = snd_ctl_new1(knew, codec);
if (!kctl)
return -ENOMEM;
kctl->private_value = spec->beep_amp;
err = snd_hda_ctl_add(codec, kctl);
if (err < 0)
return err;
}
}
/* if we have no master control, let's create it */
if (!spec->no_analog &&
!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
unsigned int vmaster_tlv[4];
snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
vmaster_tlv, alc_slave_vols);
if (err < 0)
return err;
}
if (!spec->no_analog &&
!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL, alc_slave_sws);
if (err < 0)
return err;
}
alc_free_kctls(codec); /* no longer needed */
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) },
{ }
};
/* 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_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x16;
}
static void alc880_uniwill_init_hook(struct hda_codec *codec)
{
alc_automute_amp(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_MIC_EVENT:
alc880_uniwill_mic_automute(codec);
break;
default:
alc_automute_amp_unsol_event(codec, res);
break;
}
}
static void alc880_uniwill_p53_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
}
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_DCVOL_EVENT)
alc880_uniwill_p53_dcvol_automute(codec);
else
alc_automute_amp_unsol_event(codec, res);
}
/*
* F1734 pin configuration:
* HP = 0x14, speaker-out = 0x15, mic = 0x18
*/
static struct hda_verb alc880_pin_f1734_init_verbs[] = {
{0x07, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
{0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_DCVOL_EVENT},
{ }
};
/*
* 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
#define alc880_gpio3_init_verbs alc_gpio3_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},