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patch_realtek.c
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/*
* Universal Interface for Intel High Definition Audio Codec
*
* HD audio interface patch for ALC 260/880/882 codecs
*
* Copyright (c) 2004 Kailang Yang <[email protected]>
* PeiSen Hou <[email protected]>
* Takashi Iwai <[email protected]>
* Jonathan Woithe <[email protected]>
*
* 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[5];
/* hooks */
void (*init_hook)(struct hda_codec *codec);
void (*unsol_event)(struct hda_codec *codec, unsigned int res);
/* for pin sensing */
unsigned int sense_updated: 1;
unsigned int jack_present: 1;
#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 */