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android_kernel_motorola_sm6225/sound/pci/hda/patch_realtek.c

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Linux-2.6.12-rc2 Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
2005-04-17 00:20:36 +02:00
/*
* Universal Interface for Intel High Definition Audio Codec
*
* HD audio interface patch for ALC 260/880/882 codecs
*
* Copyright (c) 2004 PeiSen Hou <pshou@realtek.com.tw>
* Takashi Iwai <tiwai@suse.de>
*
* This driver is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This driver is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include <sound/driver.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/pci.h>
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
/* ALC880 board config type */
enum {
ALC880_MINIMAL,
ALC880_3ST,
ALC880_3ST_DIG,
ALC880_5ST,
ALC880_5ST_DIG,
ALC880_W810,
[ALSA] Add ASUS Z71V support Documentation,HDA Codec driver Added the ASUS Z71V (or similar) laptop support. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:31:49 +02:00
ALC880_Z71V,
[ALSA] Add test model for debugging ALC880 devices HDA Codec driver Added 'test' model for testing/debugging the devices with ALC880 codec. This model is built only when CONFIG_SND_DEBUG is set. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:51:12 +02:00
ALC880_TEST,
Linux-2.6.12-rc2 Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
2005-04-17 00:20:36 +02:00
};
struct alc_spec {
/* codec parameterization */
unsigned int front_panel: 1;
snd_kcontrol_new_t* mixers[2];
unsigned int num_mixers;
struct hda_verb *init_verbs;
char* stream_name_analog;
struct hda_pcm_stream *stream_analog_playback;
struct hda_pcm_stream *stream_analog_capture;
char* stream_name_digital;
struct hda_pcm_stream *stream_digital_playback;
struct hda_pcm_stream *stream_digital_capture;
/* playback */
struct hda_multi_out multiout;
/* capture */
unsigned int num_adc_nids;
hda_nid_t *adc_nids;
hda_nid_t dig_in_nid;
/* capture source */
const struct hda_input_mux *input_mux;
unsigned int cur_mux[3];
/* channel model */
const struct alc_channel_mode *channel_mode;
int num_channel_mode;
/* PCM information */
struct hda_pcm pcm_rec[2];
};
/* DAC/ADC assignment */
static hda_nid_t alc880_dac_nids[4] = {
/* front, rear, clfe, rear_surr */
0x02, 0x05, 0x04, 0x03
};
static hda_nid_t alc880_w810_dac_nids[3] = {
/* front, rear/surround, clfe */
0x02, 0x03, 0x04
};
[ALSA] Add ASUS Z71V support Documentation,HDA Codec driver Added the ASUS Z71V (or similar) laptop support. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:31:49 +02:00
static hda_nid_t alc880_z71v_dac_nids[1] = {
/* front only? */
0x02
};
Linux-2.6.12-rc2 Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
2005-04-17 00:20:36 +02:00
static hda_nid_t alc880_adc_nids[3] = {
/* ADC0-2 */
0x07, 0x08, 0x09,
};
#define ALC880_DIGOUT_NID 0x06
#define ALC880_DIGIN_NID 0x0a
static hda_nid_t alc260_dac_nids[1] = {
/* front */
0x02,
};
static hda_nid_t alc260_adc_nids[2] = {
/* ADC0-1 */
0x04, 0x05,
};
#define ALC260_DIGOUT_NID 0x03
#define ALC260_DIGIN_NID 0x06
static struct hda_input_mux alc880_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x3 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
static struct hda_input_mux alc260_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
/*
* input MUX handling
*/
static int alc_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
return snd_hda_input_mux_info(spec->input_mux, uinfo);
}
static int alc_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *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);
return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
spec->adc_nids[adc_idx], &spec->cur_mux[adc_idx]);
}
/*
* channel mode setting
*/
struct alc_channel_mode {
int channels;
const struct hda_verb *sequence;
};
/*
* channel source setting (2/6 channel selection for 3-stack)
*/
/*
* set the path ways for 2 channel output
* need to set the codec line out and mic 1 pin widgets to inputs
*/
static struct hda_verb alc880_threestack_ch2_init[] = {
/* set pin widget 1Ah (line in) for input */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* set pin widget 18h (mic1) for input, for mic also enable the vref */
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* mute the output for Line In PW */
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
/* mute for Mic1 PW */
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
{ } /* end */
};
/*
* 6ch mode
* need to set the codec line out and mic 1 pin widgets to outputs
*/
static struct hda_verb alc880_threestack_ch6_init[] = {
/* set pin widget 1Ah (line in) for output */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* set pin widget 18h (mic1) for output */
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* unmute the output for Line In PW */
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
/* unmute for Mic1 PW */
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
/* for rear channel output using Line In 1
* set select widget connection (nid = 0x12) - to summer node
* for rear NID = 0x0f...offset 3 in connection list
*/
{ 0x12, AC_VERB_SET_CONNECT_SEL, 0x3 },
/* for Mic1 - retask for center/lfe */
/* set select widget connection (nid = 0x10) - to summer node for
* front CLFE NID = 0x0e...offset 2 in connection list
*/
{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x2 },
{ } /* end */
};
static struct alc_channel_mode alc880_threestack_modes[2] = {
{ 2, alc880_threestack_ch2_init },
{ 6, alc880_threestack_ch6_init },
};
/*
* channel source setting (6/8 channel selection for 5-stack)
*/
/* set the path ways for 6 channel output
* need to set the codec line out and mic 1 pin widgets to inputs
*/
static struct hda_verb alc880_fivestack_ch6_init[] = {
/* set pin widget 1Ah (line in) for input */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* mute the output for Line In PW */
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
{ } /* end */
};
/* need to set the codec line out and mic 1 pin widgets to outputs */
static struct hda_verb alc880_fivestack_ch8_init[] = {
/* set pin widget 1Ah (line in) for output */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* unmute the output for Line In PW */
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
/* output for surround channel output using Line In 1 */
/* set select widget connection (nid = 0x12) - to summer node
* for surr_rear NID = 0x0d...offset 1 in connection list
*/
{ 0x12, AC_VERB_SET_CONNECT_SEL, 0x1 },
{ } /* end */
};
static struct alc_channel_mode alc880_fivestack_modes[2] = {
{ 6, alc880_fivestack_ch6_init },
{ 8, alc880_fivestack_ch8_init },
};
/*
* channel source setting for W810 system
*
* 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 struct alc_channel_mode alc880_w810_modes[1] = {
{ 6, NULL }
};
[ALSA] Add ASUS Z71V support Documentation,HDA Codec driver Added the ASUS Z71V (or similar) laptop support. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:31:49 +02:00
static struct alc_channel_mode alc880_z71v_modes[1] = {
{ 2, NULL }
};
Linux-2.6.12-rc2 Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
2005-04-17 00:20:36 +02:00
/*
*/
static int alc880_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
snd_assert(spec->channel_mode, return -ENXIO);
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 2;
if (uinfo->value.enumerated.item >= 2)
uinfo->value.enumerated.item = 1;
sprintf(uinfo->value.enumerated.name, "%dch",
spec->channel_mode[uinfo->value.enumerated.item].channels);
return 0;
}
static int alc880_ch_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
snd_assert(spec->channel_mode, return -ENXIO);
ucontrol->value.enumerated.item[0] =
(spec->multiout.max_channels == spec->channel_mode[0].channels) ? 0 : 1;
return 0;
}
static int alc880_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
int mode;
snd_assert(spec->channel_mode, return -ENXIO);
mode = ucontrol->value.enumerated.item[0] ? 1 : 0;
if (spec->multiout.max_channels == spec->channel_mode[mode].channels &&
! codec->in_resume)
return 0;
/* change the current channel setting */
spec->multiout.max_channels = spec->channel_mode[mode].channels;
if (spec->channel_mode[mode].sequence)
snd_hda_sequence_write(codec, spec->channel_mode[mode].sequence);
return 1;
}
/*
*/
/* 3-stack mode
* Pin assignment: Front=0x14, Line-In/Rear=0x1a, Mic/CLFE=0x18, F-Mic=0x1b
* HP=0x19
*/
static snd_kcontrol_new_t alc880_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x18, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x18, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 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_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
* FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.count = 2,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc880_ch_mode_info,
.get = alc880_ch_mode_get,
.put = alc880_ch_mode_put,
},
{ } /* end */
};
/* 5-stack mode
* Pin assignment: Front=0x14, Rear=0x17, CLFE=0x16
* Line-In/Side=0x1a, Mic=0x18, F-Mic=0x1b, HP=0x19
*/
static snd_kcontrol_new_t alc880_five_stack_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x17, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Side Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 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_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
* FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.count = 2,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc880_ch_mode_info,
.get = alc880_ch_mode_get,
.put = alc880_ch_mode_put,
},
{ } /* end */
};
static snd_kcontrol_new_t alc880_w810_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
* FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.count = 3,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{ } /* end */
};
[ALSA] Add ASUS Z71V support Documentation,HDA Codec driver Added the ASUS Z71V (or similar) laptop support. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:31:49 +02:00
static snd_kcontrol_new_t alc880_z71v_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 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", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
* FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.count = 2,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{ } /* end */
};
Linux-2.6.12-rc2 Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
2005-04-17 00:20:36 +02:00
/*
*/
static int alc_build_controls(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err;
int i;
for (i = 0; i < spec->num_mixers; i++) {
err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
if (err < 0)
return err;
}
if (spec->multiout.dig_out_nid) {
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
if (err < 0)
return err;
}
if (spec->dig_in_nid) {
err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
if (err < 0)
return err;
}
return 0;
}
/*
* initialize the codec volumes, etc
*/
static struct hda_verb alc880_init_verbs_three_stack[] = {
/* Line In pin widget for input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* Mic2 (front panel) pin widget for input and vref at 80% */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* unmute amp left and right */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* set connection select to line in (default select for this ADC) */
{0x07, AC_VERB_SET_CONNECT_SEL, 0x02},
/* unmute front mixer amp left (volume = 0) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* unmute rear mixer amp left and right (volume = 0) */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* unmute rear mixer amp left and right (volume = 0) */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* using rear surround as the path for headphone output */
/* unmute rear surround mixer amp left and right (volume = 0) */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* PASD 3 stack boards use the Mic 2 as the headphone output */
/* need to program the selector associated with the Mic 2 pin widget to
* surround path (index 0x01) for headphone output */
{0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
/* mute pin widget amp left and right (no gain on this amp) */
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* need to retask the Mic 2 pin widget to output */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) for mixer widget(nid=0x0B)
* to support the input path of analog loopback
* Note: PASD motherboards uses the Line In 2 as the input for front panel
* mic (mic 2)
*/
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */
/* unmute CD */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
/* unmute Line In */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
/* unmute Mic 1 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
/* unmute Line In 2 (for PASD boards Mic 2) */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
/* Unmute input amps for the line out paths to support the output path of
* analog loopback
* the mixers on the output path has 2 inputs, one from the DAC and one
* from the mixer
*/
/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
/* Unmute Front out path */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Unmute Surround (used as HP) out path */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Unmute C/LFE out path */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, /* mute */
/* Unmute rear Surround out path */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{ }
};
static struct hda_verb alc880_init_verbs_five_stack[] = {
/* Line In pin widget for input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* Mic2 (front panel) pin widget for input and vref at 80% */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* unmute amp left and right */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* set connection select to line in (default select for this ADC) */
{0x07, AC_VERB_SET_CONNECT_SEL, 0x02},
/* unmute front mixer amp left and right (volume = 0) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* five rear and clfe */
/* unmute rear mixer amp left and right (volume = 0) */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* unmute clfe mixer amp left and right (volume = 0) */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* using rear surround as the path for headphone output */
/* unmute rear surround mixer amp left and right (volume = 0) */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* PASD 3 stack boards use the Mic 2 as the headphone output */
/* need to program the selector associated with the Mic 2 pin widget to
* surround path (index 0x01) for headphone output
*/
{0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
/* mute pin widget amp left and right (no gain on this amp) */
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* need to retask the Mic 2 pin widget to output */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) for mixer
* widget(nid=0x0B) to support the input path of analog loopback
*/
/* Note: PASD motherboards uses the Line In 2 as the input for front panel mic (mic 2) */
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03*/
/* unmute CD */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
/* unmute Line In */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
/* unmute Mic 1 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
/* unmute Line In 2 (for PASD boards Mic 2) */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
/* Unmute input amps for the line out paths to support the output path of
* analog loopback
* the mixers on the output path has 2 inputs, one from the DAC and
* one from the mixer
*/
/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
/* Unmute Front out path */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Unmute Surround (used as HP) out path */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Unmute C/LFE out path */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, /* mute */
/* Unmute rear Surround out path */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{ }
};
static struct hda_verb alc880_w810_init_verbs[] = {
/* front channel selector/amp: input 0: DAC: unmuted, (no volume selection) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* front channel selector/amp: input 1: capture mix: muted, (no volume selection) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180},
/* front channel selector/amp: output 0: unmuted, max volume */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* front out pin: muted, (no volume selection) */
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
/* front out pin: NOT headphone enable, out enable, vref disabled */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* surround channel selector/amp: input 0: DAC: unmuted, (no volume selection) */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* surround channel selector/amp: input 1: capture mix: muted, (no volume selection) */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180},
/* surround channel selector/amp: output 0: unmuted, max volume */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* surround out pin: muted, (no volume selection) */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
/* surround out pin: NOT headphone enable, out enable, vref disabled */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* c/lfe channel selector/amp: input 0: DAC: unmuted, (no volume selection) */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* c/lfe channel selector/amp: input 1: capture mix: muted, (no volume selection) */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180},
/* c/lfe channel selector/amp: output 0: unmuted, max volume */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* c/lfe out pin: muted, (no volume selection) */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
/* c/lfe out pin: NOT headphone enable, out enable, vref disabled */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* hphone/speaker input selector: front DAC */
{0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
/* hphone/speaker out pin: muted, (no volume selection) */
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
/* hphone/speaker out pin: NOT headphone enable, out enable, vref disabled */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
{ }
};
[ALSA] Add ASUS Z71V support Documentation,HDA Codec driver Added the ASUS Z71V (or similar) laptop support. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:31:49 +02:00
static struct hda_verb alc880_z71v_init_verbs[] = {
/* front channel selector/amp: input 0: DAC: unmuted, (no volume selection) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* front channel selector/amp: input 1: capture mix: muted, (no volume selection) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180},
/* front channel selector/amp: output 0: unmuted, max volume */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* front out pin: muted, (no volume selection) */
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
/* front out pin: NOT headphone enable, out enable, vref disabled */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* headphone channel selector/amp: input 0: DAC: unmuted, (no volume selection) */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* headphone channel selector/amp: input 1: capture mix: muted, (no volume selection) */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180},
/* headphone channel selector/amp: output 0: unmuted, max volume */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* headphone out pin: muted, (no volume selection) */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
/* headpohne out pin: headphone enable, out enable, vref disabled */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
/* Line In pin widget for input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* Mic2 (front panel) pin widget for input and vref at 80% */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* unmute amp left and right */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* set connection select to line in (default select for this ADC) */
{0x07, AC_VERB_SET_CONNECT_SEL, 0x02},
/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) for mixer
* widget(nid=0x0B) to support the input path of analog loopback
*/
/* Note: PASD motherboards uses the Line In 2 as the input for front panel mic (mic 2) */
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03*/
/* unmute CD */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
/* unmute Line In */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
/* unmute Mic 1 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
/* unmute Line In 2 (for PASD boards Mic 2) */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
{ }
};
Linux-2.6.12-rc2 Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
2005-04-17 00:20:36 +02:00
static int alc_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
snd_hda_sequence_write(codec, spec->init_verbs);
return 0;
}
#ifdef CONFIG_PM
/*
* resume
*/
static int alc_resume(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
alc_init(codec);
for (i = 0; i < spec->num_mixers; i++) {
snd_hda_resume_ctls(codec, spec->mixers[i]);
}
if (spec->multiout.dig_out_nid)
snd_hda_resume_spdif_out(codec);
if (spec->dig_in_nid)
snd_hda_resume_spdif_in(codec);
return 0;
}
#endif
/*
* Analog playback callbacks
*/
static int alc880_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
snd_pcm_substream_t *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
}
static int alc880_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
snd_pcm_substream_t *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
format, substream);
}
static int alc880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
snd_pcm_substream_t *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
}
/*
* Digital out
*/
static int alc880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
snd_pcm_substream_t *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_dig_open(codec, &spec->multiout);
}
static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
snd_pcm_substream_t *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_dig_close(codec, &spec->multiout);
}
/*
* Analog capture
*/
static int alc880_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
snd_pcm_substream_t *substream)
{
struct alc_spec *spec = codec->spec;
snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
stream_tag, 0, format);
return 0;
}
static int alc880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
snd_pcm_substream_t *substream)
{
struct alc_spec *spec = codec->spec;
snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], 0, 0, 0);
return 0;
}
/*
*/
static struct hda_pcm_stream alc880_pcm_analog_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 8,
.nid = 0x02, /* NID to query formats and rates */
.ops = {
.open = alc880_playback_pcm_open,
.prepare = alc880_playback_pcm_prepare,
.cleanup = alc880_playback_pcm_cleanup
},
};
static struct hda_pcm_stream alc880_pcm_analog_capture = {
.substreams = 2,
.channels_min = 2,
.channels_max = 2,
.nid = 0x07, /* NID to query formats and rates */
.ops = {
.prepare = alc880_capture_pcm_prepare,
.cleanup = alc880_capture_pcm_cleanup
},
};
static struct hda_pcm_stream alc880_pcm_digital_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
.ops = {
.open = alc880_dig_playback_pcm_open,
.close = alc880_dig_playback_pcm_close
},
};
static struct hda_pcm_stream alc880_pcm_digital_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
};
static int alc_build_pcms(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct hda_pcm *info = spec->pcm_rec;
int i;
codec->num_pcms = 1;
codec->pcm_info = info;
info->name = spec->stream_name_analog;
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback);
info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = 0;
for (i = 0; i < spec->num_channel_mode; i++) {
if (spec->channel_mode[i].channels > info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->channel_mode[i].channels;
}
}
if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
codec->num_pcms++;
info++;
info->name = spec->stream_name_digital;
if (spec->multiout.dig_out_nid) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback);
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
}
if (spec->dig_in_nid) {
info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_digital_capture);
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
}
}
return 0;
}
static void alc_free(struct hda_codec *codec)
{
kfree(codec->spec);
}
/*
*/
static struct hda_codec_ops alc_patch_ops = {
.build_controls = alc_build_controls,
.build_pcms = alc_build_pcms,
.init = alc_init,
.free = alc_free,
#ifdef CONFIG_PM
.resume = alc_resume,
#endif
};
[ALSA] Add test model for debugging ALC880 devices HDA Codec driver Added 'test' model for testing/debugging the devices with ALC880 codec. This model is built only when CONFIG_SND_DEBUG is set. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:51:12 +02:00
/*
* Test configuration for debugging
*
* Almost all inputs/outputs are enabled. I/O pins can be configured via
* enum controls.
*/
#ifdef CONFIG_SND_DEBUG
static hda_nid_t alc880_test_dac_nids[4] = {
0x02, 0x03, 0x04, 0x05
};
static struct hda_input_mux alc880_test_capture_source = {
.num_items = 5,
.items = {
{ "In-1", 0x0 },
{ "In-2", 0x1 },
{ "In-3", 0x2 },
{ "In-4", 0x3 },
{ "CD", 0x4 },
},
};
static struct alc_channel_mode alc880_test_modes[2] = {
{ 2, NULL },
{ 6, NULL },
};
static int alc_test_pin_ctl_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
{
static char *texts[] = {
"N/A", "Line Out", "HP Out",
"In Hi-Z", "In 50%", "In Grd", "In 80%", "In 100%"
};
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 8;
if (uinfo->value.enumerated.item >= 8)
uinfo->value.enumerated.item = 7;
strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
return 0;
}
static int alc_test_pin_ctl_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
unsigned int pin_ctl, item = 0;
pin_ctl = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
if (pin_ctl & AC_PINCTL_OUT_EN) {
if (pin_ctl & AC_PINCTL_HP_EN)
item = 2;
else
item = 1;
} else if (pin_ctl & AC_PINCTL_IN_EN) {
switch (pin_ctl & AC_PINCTL_VREFEN) {
case AC_PINCTL_VREF_HIZ: item = 3; break;
case AC_PINCTL_VREF_50: item = 4; break;
case AC_PINCTL_VREF_GRD: item = 5; break;
case AC_PINCTL_VREF_80: item = 6; break;
case AC_PINCTL_VREF_100: item = 7; break;
}
}
ucontrol->value.enumerated.item[0] = item;
return 0;
}
static int alc_test_pin_ctl_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
static unsigned int ctls[] = {
0, AC_PINCTL_OUT_EN, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_50,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_80,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_100,
};
unsigned int old_ctl, new_ctl;
old_ctl = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
new_ctl = ctls[ucontrol->value.enumerated.item[0]];
if (old_ctl != new_ctl) {
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, new_ctl);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
ucontrol->value.enumerated.item[0] >= 3 ? 0xb080 : 0xb000);
return 1;
}
return 0;
}
static int alc_test_pin_src_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
{
static char *texts[] = {
"Front", "Surround", "CLFE", "Side"
};
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 4;
if (uinfo->value.enumerated.item >= 4)
uinfo->value.enumerated.item = 3;
strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
return 0;
}
static int alc_test_pin_src_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
unsigned int sel;
sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0);
ucontrol->value.enumerated.item[0] = sel & 3;
return 0;
}
static int alc_test_pin_src_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
unsigned int sel;
sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3;
if (ucontrol->value.enumerated.item[0] != sel) {
sel = ucontrol->value.enumerated.item[0] & 3;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, sel);
return 1;
}
return 0;
}
#define PIN_CTL_TEST(xname,nid) { \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
.info = alc_test_pin_ctl_info, \
.get = alc_test_pin_ctl_get, \
.put = alc_test_pin_ctl_put, \
.private_value = nid \
}
#define PIN_SRC_TEST(xname,nid) { \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
.info = alc_test_pin_src_info, \
.get = alc_test_pin_src_get, \
.put = alc_test_pin_src_put, \
.private_value = nid \
}
static snd_kcontrol_new_t alc880_test_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CLFE Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
PIN_CTL_TEST("Front Pin Mode", 0x14),
PIN_CTL_TEST("Surround Pin Mode", 0x15),
PIN_CTL_TEST("CLFE Pin Mode", 0x16),
PIN_CTL_TEST("Side Pin Mode", 0x17),
PIN_CTL_TEST("In-1 Pin Mode", 0x18),
PIN_CTL_TEST("In-2 Pin Mode", 0x19),
PIN_CTL_TEST("In-3 Pin Mode", 0x1a),
PIN_CTL_TEST("In-4 Pin Mode", 0x1b),
PIN_SRC_TEST("In-1 Pin Source", 0x18),
PIN_SRC_TEST("In-2 Pin Source", 0x19),
PIN_SRC_TEST("In-3 Pin Source", 0x1a),
PIN_SRC_TEST("In-4 Pin Source", 0x1b),
HDA_CODEC_VOLUME("In-1 Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("In-1 Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("In-2 Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("In-2 Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("In-3 Playback Volume", 0x0b, 0x2, HDA_INPUT),
HDA_CODEC_MUTE("In-3 Playback Switch", 0x0b, 0x2, HDA_INPUT),
HDA_CODEC_VOLUME("In-4 Playback Volume", 0x0b, 0x3, HDA_INPUT),
HDA_CODEC_MUTE("In-4 Playback Switch", 0x0b, 0x3, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x4, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x4, HDA_INPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Input Source",
.count = 2,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc880_ch_mode_info,
.get = alc880_ch_mode_get,
.put = alc880_ch_mode_put,
},
{ } /* end */
};
static struct hda_verb alc880_test_init_verbs[] = {
/* Unmute inputs of 0x0c - 0x0f */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7100},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7100},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0x7100},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0x7100},
/* Vol output for 0x0c-0x0f */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* Set output pins 0x14-0x17 */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* Unmute output pins 0x14-0x17 */
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
/* Set input pins 0x18-0x1c */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, /* vref 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* Mute input pins 0x18-0x1b */
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
{ }
};
#endif
Linux-2.6.12-rc2 Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
2005-04-17 00:20:36 +02:00
/*
*/
static struct hda_board_config alc880_cfg_tbl[] = {
/* Back 3 jack, front 2 jack */
{ .modelname = "3stack", .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe200, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe201, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe202, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe203, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe204, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe205, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe206, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe207, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe208, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe209, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe20a, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe20b, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe20c, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe20d, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe20e, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe20f, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe210, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe211, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe214, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe302, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe303, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe304, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe306, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe307, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xe404, .config = ALC880_3ST },
{ .pci_vendor = 0x8086, .pci_device = 0xa101, .config = ALC880_3ST },
{ .pci_vendor = 0x107b, .pci_device = 0x3031, .config = ALC880_3ST },
{ .pci_vendor = 0x107b, .pci_device = 0x4036, .config = ALC880_3ST },
{ .pci_vendor = 0x107b, .pci_device = 0x4037, .config = ALC880_3ST },
{ .pci_vendor = 0x107b, .pci_device = 0x4038, .config = ALC880_3ST },
{ .pci_vendor = 0x107b, .pci_device = 0x4040, .config = ALC880_3ST },
{ .pci_vendor = 0x107b, .pci_device = 0x4041, .config = ALC880_3ST },
/* Back 3 jack, front 2 jack (Internal add Aux-In) */
{ .pci_vendor = 0x1025, .pci_device = 0xe310, .config = ALC880_3ST },
/* Back 3 jack plus 1 SPDIF out jack, front 2 jack */
{ .modelname = "3stack-digout", .config = ALC880_3ST_DIG },
{ .pci_vendor = 0x8086, .pci_device = 0xe308, .config = ALC880_3ST_DIG },
/* Back 3 jack plus 1 SPDIF out jack, front 2 jack (Internal add Aux-In)*/
{ .pci_vendor = 0x8086, .pci_device = 0xe305, .config = ALC880_3ST_DIG },
{ .pci_vendor = 0x8086, .pci_device = 0xd402, .config = ALC880_3ST_DIG },
{ .pci_vendor = 0x1025, .pci_device = 0xe309, .config = ALC880_3ST_DIG },
/* Back 5 jack, front 2 jack */
{ .modelname = "5stack", .config = ALC880_5ST },
{ .pci_vendor = 0x107b, .pci_device = 0x3033, .config = ALC880_5ST },
{ .pci_vendor = 0x107b, .pci_device = 0x4039, .config = ALC880_5ST },
{ .pci_vendor = 0x107b, .pci_device = 0x3032, .config = ALC880_5ST },
{ .pci_vendor = 0x103c, .pci_device = 0x2a09, .config = ALC880_5ST },
/* Back 5 jack plus 1 SPDIF out jack, front 2 jack */
{ .modelname = "5stack-digout", .config = ALC880_5ST_DIG },
{ .pci_vendor = 0x8086, .pci_device = 0xe224, .config = ALC880_5ST_DIG },
{ .pci_vendor = 0x8086, .pci_device = 0xe400, .config = ALC880_5ST_DIG },
{ .pci_vendor = 0x8086, .pci_device = 0xe401, .config = ALC880_5ST_DIG },
{ .pci_vendor = 0x8086, .pci_device = 0xe402, .config = ALC880_5ST_DIG },
{ .pci_vendor = 0x8086, .pci_device = 0xd400, .config = ALC880_5ST_DIG },
{ .pci_vendor = 0x8086, .pci_device = 0xd401, .config = ALC880_5ST_DIG },
{ .pci_vendor = 0x8086, .pci_device = 0xa100, .config = ALC880_5ST_DIG },
{ .pci_vendor = 0x1565, .pci_device = 0x8202, .config = ALC880_5ST_DIG },
{ .modelname = "w810", .config = ALC880_W810 },
{ .pci_vendor = 0x161f, .pci_device = 0x203d, .config = ALC880_W810 },
[ALSA] Add ASUS Z71V support Documentation,HDA Codec driver Added the ASUS Z71V (or similar) laptop support. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:31:49 +02:00
{ .modelname = "z71v", .config = ALC880_Z71V },
{ .pci_vendor = 0x1043, .pci_device = 0x1964, .config = ALC880_Z71V },
[ALSA] Add test model for debugging ALC880 devices HDA Codec driver Added 'test' model for testing/debugging the devices with ALC880 codec. This model is built only when CONFIG_SND_DEBUG is set. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:51:12 +02:00
#ifdef CONFIG_SND_DEBUG
{ .modelname = "test", .config = ALC880_TEST },
#endif
Linux-2.6.12-rc2 Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
2005-04-17 00:20:36 +02:00
{}
};
static int patch_alc880(struct hda_codec *codec)
{
struct alc_spec *spec;
int board_config;
spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
board_config = snd_hda_check_board_config(codec, alc880_cfg_tbl);
if (board_config < 0) {
snd_printd(KERN_INFO "hda_codec: Unknown model for ALC880\n");
board_config = ALC880_MINIMAL;
}
switch (board_config) {
case ALC880_W810:
spec->mixers[spec->num_mixers] = alc880_w810_base_mixer;
spec->num_mixers++;
break;
case ALC880_5ST:
case ALC880_5ST_DIG:
spec->mixers[spec->num_mixers] = alc880_five_stack_mixer;
spec->num_mixers++;
break;
[ALSA] Add ASUS Z71V support Documentation,HDA Codec driver Added the ASUS Z71V (or similar) laptop support. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:31:49 +02:00
case ALC880_Z71V:
spec->mixers[spec->num_mixers] = alc880_z71v_mixer;
spec->num_mixers++;
break;
[ALSA] Add test model for debugging ALC880 devices HDA Codec driver Added 'test' model for testing/debugging the devices with ALC880 codec. This model is built only when CONFIG_SND_DEBUG is set. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:51:12 +02:00
#ifdef CONFIG_SND_DEBUG
case ALC880_TEST:
spec->mixers[spec->num_mixers] = alc880_test_mixer;
spec->num_mixers++;
break;
#endif
Linux-2.6.12-rc2 Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
2005-04-17 00:20:36 +02:00
default:
spec->mixers[spec->num_mixers] = alc880_base_mixer;
spec->num_mixers++;
break;
}
switch (board_config) {
case ALC880_3ST_DIG:
case ALC880_5ST_DIG:
case ALC880_W810:
[ALSA] Add ASUS Z71V support Documentation,HDA Codec driver Added the ASUS Z71V (or similar) laptop support. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:31:49 +02:00
case ALC880_Z71V:
[ALSA] Add test model for debugging ALC880 devices HDA Codec driver Added 'test' model for testing/debugging the devices with ALC880 codec. This model is built only when CONFIG_SND_DEBUG is set. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:51:12 +02:00
case ALC880_TEST:
Linux-2.6.12-rc2 Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
2005-04-17 00:20:36 +02:00
spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
break;
default:
break;
}
switch (board_config) {
case ALC880_3ST:
case ALC880_3ST_DIG:
case ALC880_5ST:
case ALC880_5ST_DIG:
case ALC880_W810:
spec->front_panel = 1;
break;
default:
break;
}
switch (board_config) {
case ALC880_5ST:
case ALC880_5ST_DIG:
spec->init_verbs = alc880_init_verbs_five_stack;
spec->channel_mode = alc880_fivestack_modes;
spec->num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes);
break;
case ALC880_W810:
spec->init_verbs = alc880_w810_init_verbs;
spec->channel_mode = alc880_w810_modes;
spec->num_channel_mode = ARRAY_SIZE(alc880_w810_modes);
break;
[ALSA] Add ASUS Z71V support Documentation,HDA Codec driver Added the ASUS Z71V (or similar) laptop support. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:31:49 +02:00
case ALC880_Z71V:
spec->init_verbs = alc880_z71v_init_verbs;
spec->channel_mode = alc880_z71v_modes;
spec->num_channel_mode = ARRAY_SIZE(alc880_z71v_modes);
break;
[ALSA] Add test model for debugging ALC880 devices HDA Codec driver Added 'test' model for testing/debugging the devices with ALC880 codec. This model is built only when CONFIG_SND_DEBUG is set. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:51:12 +02:00
#ifdef CONFIG_SND_DEBUG
case ALC880_TEST:
spec->init_verbs = alc880_test_init_verbs;
spec->channel_mode = alc880_test_modes;
spec->num_channel_mode = ARRAY_SIZE(alc880_test_modes);
break;
#endif
Linux-2.6.12-rc2 Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
2005-04-17 00:20:36 +02:00
default:
spec->init_verbs = alc880_init_verbs_three_stack;
spec->channel_mode = alc880_threestack_modes;
spec->num_channel_mode = ARRAY_SIZE(alc880_threestack_modes);
break;
}
spec->stream_name_analog = "ALC880 Analog";
spec->stream_analog_playback = &alc880_pcm_analog_playback;
spec->stream_analog_capture = &alc880_pcm_analog_capture;
spec->stream_name_digital = "ALC880 Digital";
spec->stream_digital_playback = &alc880_pcm_digital_playback;
spec->stream_digital_capture = &alc880_pcm_digital_capture;
spec->multiout.max_channels = spec->channel_mode[0].channels;
switch (board_config) {
case ALC880_W810:
spec->multiout.num_dacs = ARRAY_SIZE(alc880_w810_dac_nids);
spec->multiout.dac_nids = alc880_w810_dac_nids;
// No dedicated headphone socket - it's shared with built-in speakers.
break;
[ALSA] Add ASUS Z71V support Documentation,HDA Codec driver Added the ASUS Z71V (or similar) laptop support. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:31:49 +02:00
case ALC880_Z71V:
spec->multiout.num_dacs = ARRAY_SIZE(alc880_z71v_dac_nids);
spec->multiout.dac_nids = alc880_z71v_dac_nids;
spec->multiout.hp_nid = 0x03;
break;
[ALSA] Add test model for debugging ALC880 devices HDA Codec driver Added 'test' model for testing/debugging the devices with ALC880 codec. This model is built only when CONFIG_SND_DEBUG is set. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:51:12 +02:00
#ifdef CONFIG_SND_DEBUG
case ALC880_TEST:
spec->multiout.num_dacs = ARRAY_SIZE(alc880_test_dac_nids);
spec->multiout.dac_nids = alc880_test_dac_nids;
spec->input_mux = &alc880_test_capture_source;
break;
#endif
Linux-2.6.12-rc2 Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
2005-04-17 00:20:36 +02:00
default:
spec->multiout.num_dacs = ARRAY_SIZE(alc880_dac_nids);
spec->multiout.dac_nids = alc880_dac_nids;
spec->multiout.hp_nid = 0x03; /* rear-surround NID */
break;
}
[ALSA] Add test model for debugging ALC880 devices HDA Codec driver Added 'test' model for testing/debugging the devices with ALC880 codec. This model is built only when CONFIG_SND_DEBUG is set. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2005-05-12 14:51:12 +02:00
if (! spec->input_mux)
spec->input_mux = &alc880_capture_source;
Linux-2.6.12-rc2 Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
2005-04-17 00:20:36 +02:00
spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids);
spec->adc_nids = alc880_adc_nids;
codec->patch_ops = alc_patch_ops;
return 0;
}
/*
* ALC260 support
*/
/*
* This is just place-holder, so there's something for alc_build_pcms to look
* at when it calculates the maximum number of channels. ALC260 has no mixer
* element which allows changing the channel mode, so the verb list is
* never used.
*/
static struct alc_channel_mode alc260_modes[1] = {
{ 2, NULL },
};
snd_kcontrol_new_t alc260_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
/* use LINE2 for the output */
/* HDA_CODEC_MUTE("Front Playback Switch", 0x0f, 0x0, HDA_OUTPUT), */
HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x10, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x04, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Capture Source",
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{ } /* end */
};
static struct hda_verb alc260_init_verbs[] = {
/* Line In pin widget for input */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* CD pin widget for input */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* Mic2 (front panel) pin widget for input and vref at 80% */
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* LINE-2 is used for line-out in rear */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* select line-out */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
/* LINE-OUT pin */
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* enable HP */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* enable Mono */
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* unmute amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* set connection select to line in (default select for this ADC) */
{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
/* unmute Line-Out mixer amp left and right (volume = 0) */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* unmute HP mixer amp left and right (volume = 0) */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
/* unmute Mono mixer amp left and right (volume = 0) */
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
/* mute LINE-2 out */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */
/* unmute CD */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
/* unmute Line In */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
/* unmute Mic */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
/* Unmute Front out path */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Unmute Headphone out path */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Unmute Mono out path */
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{ }
};
static struct hda_pcm_stream alc260_pcm_analog_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
.nid = 0x2,
};
static struct hda_pcm_stream alc260_pcm_analog_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
.nid = 0x4,
};
static int patch_alc260(struct hda_codec *codec)
{
struct alc_spec *spec;
spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
spec->mixers[spec->num_mixers] = alc260_base_mixer;
spec->num_mixers++;
spec->init_verbs = alc260_init_verbs;
spec->channel_mode = alc260_modes;
spec->num_channel_mode = ARRAY_SIZE(alc260_modes);
spec->stream_name_analog = "ALC260 Analog";
spec->stream_analog_playback = &alc260_pcm_analog_playback;
spec->stream_analog_capture = &alc260_pcm_analog_capture;
spec->multiout.max_channels = spec->channel_mode[0].channels;
spec->multiout.num_dacs = ARRAY_SIZE(alc260_dac_nids);
spec->multiout.dac_nids = alc260_dac_nids;
spec->input_mux = &alc260_capture_source;
spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids);
spec->adc_nids = alc260_adc_nids;
codec->patch_ops = alc_patch_ops;
return 0;
}
/*
* ALC882 support
*
* ALC882 is almost identical with ALC880 but has cleaner and more flexible
* configuration. Each pin widget can choose any input DACs and a mixer.
* Each ADC is connected from a mixer of all inputs. This makes possible
* 6-channel independent captures.
*
* In addition, an independent DAC for the multi-playback (not used in this
* driver yet).
*/
static struct alc_channel_mode alc882_ch_modes[1] = {
{ 8, NULL }
};
static hda_nid_t alc882_dac_nids[4] = {
/* front, rear, clfe, rear_surr */
0x02, 0x03, 0x04, 0x05
};
static hda_nid_t alc882_adc_nids[3] = {
/* ADC0-2 */
0x07, 0x08, 0x09,
};
/* input MUX */
/* FIXME: should be a matrix-type input source selection */
static struct hda_input_mux alc882_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
#define alc882_mux_enum_info alc_mux_enum_info
#define alc882_mux_enum_get alc_mux_enum_get
static int alc882_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
const struct hda_input_mux *imux = spec->input_mux;
unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 };
hda_nid_t nid = capture_mixers[adc_idx];
unsigned int *cur_val = &spec->cur_mux[adc_idx];
unsigned int i, idx;
idx = ucontrol->value.enumerated.item[0];
if (idx >= imux->num_items)
idx = imux->num_items - 1;
if (*cur_val == idx && ! codec->in_resume)
return 0;
for (i = 0; i < imux->num_items; i++) {
unsigned int v = (i == idx) ? 0x7000 : 0x7080;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
v | (imux->items[i].index << 8));
}
*cur_val = idx;
return 1;
}
/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
* Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
*/
static snd_kcontrol_new_t alc882_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Side Playback Switch", 0x17, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* .name = "Capture Source", */
.name = "Input Source",
.count = 3,
.info = alc882_mux_enum_info,
.get = alc882_mux_enum_get,
.put = alc882_mux_enum_put,
},
{ } /* end */
};
static struct hda_verb alc882_init_verbs[] = {
/* Front mixer: unmute input/output amp left and right (volume = 0) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
/* Rear mixer */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
/* CLFE mixer */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
/* Side mixer */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
/* Front Pin: to output mode */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* Front Pin: mute amp left and right (no volume) */
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* select Front mixer (0x0c, index 0) */
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Rear Pin */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* Rear Pin: mute amp left and right (no volume) */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* select Rear mixer (0x0d, index 1) */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
/* CLFE Pin */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* CLFE Pin: mute amp left and right (no volume) */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* select CLFE mixer (0x0e, index 2) */
{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
/* Side Pin */
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* Side Pin: mute amp left and right (no volume) */
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* select Side mixer (0x0f, index 3) */
{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
/* Headphone Pin */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* Headphone Pin: mute amp left and right (no volume) */
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* select Front mixer (0x0c, index 0) */
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mic (rear) pin widget for input and vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* Front Mic pin widget for input and vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* Line In pin widget for input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* ADC1: unmute amp left and right */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* ADC2: unmute amp left and right */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* ADC3: unmute amp left and right */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* Unmute front loopback */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Unmute rear loopback */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Mute CLFE loopback */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
/* Unmute side loopback */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{ }
};
static int patch_alc882(struct hda_codec *codec)
{
struct alc_spec *spec;
spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
spec->mixers[spec->num_mixers] = alc882_base_mixer;
spec->num_mixers++;
spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
spec->dig_in_nid = ALC880_DIGIN_NID;
spec->front_panel = 1;
spec->init_verbs = alc882_init_verbs;
spec->channel_mode = alc882_ch_modes;
spec->num_channel_mode = ARRAY_SIZE(alc882_ch_modes);
spec->stream_name_analog = "ALC882 Analog";
spec->stream_analog_playback = &alc880_pcm_analog_playback;
spec->stream_analog_capture = &alc880_pcm_analog_capture;
spec->stream_name_digital = "ALC882 Digital";
spec->stream_digital_playback = &alc880_pcm_digital_playback;
spec->stream_digital_capture = &alc880_pcm_digital_capture;
spec->multiout.max_channels = spec->channel_mode[0].channels;
spec->multiout.num_dacs = ARRAY_SIZE(alc882_dac_nids);
spec->multiout.dac_nids = alc882_dac_nids;
spec->input_mux = &alc882_capture_source;
spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids);
spec->adc_nids = alc882_adc_nids;
codec->patch_ops = alc_patch_ops;
return 0;
}
/*
* patch entries
*/
struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
{} /* terminator */
};
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