linux/sound/pci/hda/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 <kailang@realtek.com.tw>
* PeiSen Hou <pshou@realtek.com.tw>
* Takashi Iwai <tiwai@suse.de>
* Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
*
* This driver is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This driver is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include <sound/driver.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/pci.h>
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
/* ALC880 board config type */
enum {
ALC880_3ST,
ALC880_3ST_DIG,
ALC880_5ST,
ALC880_5ST_DIG,
ALC880_W810,
ALC880_Z71V,
ALC880_6ST,
ALC880_6ST_DIG,
ALC880_F1734,
ALC880_ASUS,
ALC880_ASUS_DIG,
ALC880_ASUS_W1V,
ALC880_ASUS_DIG2,
ALC880_UNIWILL_DIG,
ALC880_CLEVO,
ALC880_TCL_S700,
ALC880_LG,
ALC880_LG_LW,
#ifdef CONFIG_SND_DEBUG
ALC880_TEST,
#endif
ALC880_AUTO,
ALC880_MODEL_LAST /* last tag */
};
/* ALC260 models */
enum {
ALC260_BASIC,
ALC260_HP,
ALC260_HP_3013,
ALC260_FUJITSU_S702X,
ALC260_ACER,
#ifdef CONFIG_SND_DEBUG
ALC260_TEST,
#endif
ALC260_AUTO,
ALC260_MODEL_LAST /* last tag */
};
/* ALC262 models */
enum {
ALC262_BASIC,
ALC262_FUJITSU,
ALC262_AUTO,
ALC262_MODEL_LAST /* last tag */
};
/* ALC861 models */
enum {
ALC861_3ST,
ALC861_3ST_DIG,
ALC861_6ST_DIG,
ALC861_AUTO,
ALC861_MODEL_LAST,
};
/* ALC882 models */
enum {
ALC882_3ST_DIG,
ALC882_6ST_DIG,
ALC882_AUTO,
ALC882_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 */
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
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;
/* PCM information */
struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */
/* dynamic controls, init_verbs and input_mux */
struct auto_pin_cfg autocfg;
unsigned int num_kctl_alloc, num_kctl_used;
struct snd_kcontrol_new *kctl_alloc;
struct hda_input_mux private_imux;
hda_nid_t private_dac_nids[5];
/* hooks */
void (*init_hook)(struct hda_codec *codec);
void (*unsol_event)(struct hda_codec *codec, unsigned int res);
/* for pin sensing */
unsigned int sense_updated: 1;
unsigned int jack_present: 1;
};
/*
* configuration template - to be copied to the spec instance
*/
struct alc_config_preset {
struct snd_kcontrol_new *mixers[5]; /* should be identical size with spec */
const struct hda_verb *init_verbs[5];
unsigned int num_dacs;
hda_nid_t *dac_nids;
hda_nid_t dig_out_nid; /* optional */
hda_nid_t hp_nid; /* optional */
unsigned int num_adc_nids;
hda_nid_t *adc_nids;
hda_nid_t dig_in_nid;
unsigned int num_channel_mode;
const struct hda_channel_mode *channel_mode;
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
unsigned int num_mux_defs;
const struct hda_input_mux *input_mux;
void (*unsol_event)(struct hda_codec *, unsigned int);
void (*init_hook)(struct hda_codec *);
};
/*
* input MUX handling
*/
static int alc_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
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);
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
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;
return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
spec->num_channel_mode, &spec->multiout.max_channels);
}
/*
* 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.
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
* 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
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
* 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.
*/
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
#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
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
/* Info about the pin modes supported by the different pin direction modes.
* For each direction the minimum and maximum values are given.
*/
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
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 */
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
{ 2, 2 }, /* ALC_PIN_DIR_IN_NOMICBIAS */
{ 2, 4 }, /* ALC_PIN_DIR_INOUT_NOMICBIAS */
};
#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0])
#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1])
#define alc_pin_mode_n_items(_dir) \
(alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1)
static int alc_pin_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
{
unsigned int item_num = uinfo->value.enumerated.item;
unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = alc_pin_mode_n_items(dir);
if (item_num<alc_pin_mode_min(dir) || item_num>alc_pin_mode_max(dir))
item_num = alc_pin_mode_min(dir);
strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]);
return 0;
}
static int alc_pin_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
unsigned int i;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int pinctl = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00);
/* Find enumerated value for current pinctl setting */
i = alc_pin_mode_min(dir);
while (alc_pin_mode_values[i]!=pinctl && i<=alc_pin_mode_max(dir))
i++;
*valp = i<=alc_pin_mode_max(dir)?i:alc_pin_mode_min(dir);
return 0;
}
static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int pinctl = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00);
if (val<alc_pin_mode_min(dir) || val>alc_pin_mode_max(dir))
val = alc_pin_mode_min(dir);
change = pinctl != alc_pin_mode_values[val];
if (change) {
/* Set pin mode to that requested */
snd_hda_codec_write(codec,nid,0,AC_VERB_SET_PIN_WIDGET_CONTROL,
alc_pin_mode_values[val]);
/* Also enable the retasking pin's input/output as required
* for the requested pin mode. Enum values of 2 or less are
* input modes.
*
* Dynamically switching the input/output buffers probably
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
* reduces noise slightly (particularly on input) so we'll
* do it. However, having both input and output buffers
* enabled simultaneously doesn't seem to be problematic if
* this turns out to be necessary in the future.
*/
if (val <= 2) {
snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_MUTE);
snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_UNMUTE(0));
} else {
snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_MUTE(0));
snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_UNMUTE);
}
}
return change;
}
#define ALC_PIN_MODE(xname, nid, dir) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.info = alc_pin_mode_info, \
.get = alc_pin_mode_get, \
.put = alc_pin_mode_put, \
.private_value = nid | (dir<<16) }
/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged
* together using a mask with more than one bit set. This control is
* currently used only by the ALC260 test model. At this stage they are not
* needed for any "production" models.
*/
#ifdef CONFIG_SND_DEBUG
static int alc_gpio_data_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
static int alc_gpio_data_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int val = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_GPIO_DATA,0x00);
*valp = (val & mask) != 0;
return 0;
}
static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int gpio_data = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_GPIO_DATA,0x00);
/* Set/unset the masked GPIO bit(s) as needed */
change = (val==0?0:mask) != (gpio_data & mask);
if (val==0)
gpio_data &= ~mask;
else
gpio_data |= mask;
snd_hda_codec_write(codec,nid,0,AC_VERB_SET_GPIO_DATA,gpio_data);
return change;
}
#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.info = alc_gpio_data_info, \
.get = alc_gpio_data_get, \
.put = alc_gpio_data_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
/* A switch control to allow the enabling of the digital IO pins on the
* ALC260. This is incredibly simplistic; the intention of this control is
* to provide something in the test model allowing digital outputs to be
* identified if present. If models are found which can utilise these
* outputs a more complete mixer control can be devised for those models if
* necessary.
*/
#ifdef CONFIG_SND_DEBUG
static int alc_spdif_ctrl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int val = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_DIGI_CONVERT,0x00);
*valp = (val & mask) != 0;
return 0;
}
static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int ctrl_data = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_DIGI_CONVERT,0x00);
/* Set/unset the masked control bit(s) as needed */
change = (val==0?0:mask) != (ctrl_data & mask);
if (val==0)
ctrl_data &= ~mask;
else
ctrl_data |= mask;
snd_hda_codec_write(codec,nid,0,AC_VERB_SET_DIGI_CONVERT_1,ctrl_data);
return change;
}
#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.info = alc_spdif_ctrl_info, \
.get = alc_spdif_ctrl_get, \
.put = alc_spdif_ctrl_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
/*
* set up from the preset table
*/
static void setup_preset(struct alc_spec *spec, const struct alc_config_preset *preset)
{
int i;
for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
spec->mixers[spec->num_mixers++] = preset->mixers[i];
for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i]; i++)
spec->init_verbs[spec->num_init_verbs++] = preset->init_verbs[i];
spec->channel_mode = preset->channel_mode;
spec->num_channel_mode = preset->num_channel_mode;
spec->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;
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
spec->num_mux_defs = preset->num_mux_defs;
if (! spec->num_mux_defs)
spec->num_mux_defs = 1;
spec->input_mux = preset->input_mux;
spec->num_adc_nids = preset->num_adc_nids;
spec->adc_nids = preset->adc_nids;
spec->dig_in_nid = preset->dig_in_nid;
spec->unsol_event = preset->unsol_event;
spec->init_hook = preset->init_hook;
}
/*
* ALC880 3-stack model
*
* DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e)
* Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18, F-Mic = 0x1b
* HP = 0x19
*/
static hda_nid_t alc880_dac_nids[4] = {
/* front, rear, clfe, rear_surr */
0x02, 0x05, 0x04, 0x03
};
static hda_nid_t alc880_adc_nids[3] = {
/* ADC0-2 */
0x07, 0x08, 0x09,
};
/* The datasheet says the node 0x07 is connected from inputs,
* but it shows zero connection in the real implementation on some devices.
* Note: this is a 915GAV bug, fixed on 915GLV
*/
static hda_nid_t alc880_adc_nids_alt[2] = {
/* ADC1-2 */
0x08, 0x09,
};
#define ALC880_DIGOUT_NID 0x06
#define ALC880_DIGIN_NID 0x0a
static struct hda_input_mux alc880_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x3 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
/* channel source setting (2/6 channel selection for 3-stack) */
/* 2ch mode */
static struct hda_verb alc880_threestack_ch2_init[] = {
/* set line-in to input, mute it */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
/* set mic-in to input vref 80%, mute it */
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
{ } /* end */
};
/* 6ch mode */
static struct hda_verb alc880_threestack_ch6_init[] = {
/* set line-in to output, unmute it */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
/* set mic-in to output, unmute it */
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ } /* end */
};
static struct hda_channel_mode alc880_threestack_modes[2] = {
{ 2, alc880_threestack_ch2_init },
{ 6, alc880_threestack_ch6_init },
};
static struct snd_kcontrol_new alc880_three_stack_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
/* capture mixer elements */
static struct snd_kcontrol_new alc880_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
* FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.count = 3,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{ } /* end */
};
/* capture mixer elements (in case NID 0x07 not available) */
static struct snd_kcontrol_new alc880_capture_alt_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
* FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.count = 2,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{ } /* end */
};
/*
* ALC880 5-stack model
*
* DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d), Side = 0x02 (0xd)
* Pin assignment: Front = 0x14, Surr = 0x17, CLFE = 0x16
* Line-In/Side = 0x1a, Mic = 0x18, F-Mic = 0x1b, HP = 0x19
*/
/* additional mixers to alc880_three_stack_mixer */
static struct snd_kcontrol_new alc880_five_stack_mixer[] = {
HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT),
{ } /* end */
};
/* channel source setting (6/8 channel selection for 5-stack) */
/* 6ch mode */
static struct hda_verb alc880_fivestack_ch6_init[] = {
/* set line-in to input, mute it */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
{ } /* end */
};
/* 8ch mode */
static struct hda_verb alc880_fivestack_ch8_init[] = {
/* set line-in to output, unmute it */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ } /* end */
};
static struct hda_channel_mode alc880_fivestack_modes[2] = {
{ 6, alc880_fivestack_ch6_init },
{ 8, alc880_fivestack_ch8_init },
};
/*
* ALC880 6-stack model
*
* DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e), Side = 0x05 (0x0f)
* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, Side = 0x17,
* Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b
*/
static hda_nid_t alc880_6st_dac_nids[4] = {
/* front, rear, clfe, rear_surr */
0x02, 0x03, 0x04, 0x05
};
static struct hda_input_mux alc880_6stack_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
/* fixed 8-channels */
static struct hda_channel_mode alc880_sixstack_modes[1] = {
{ 8, NULL },
};
static struct snd_kcontrol_new alc880_six_stack_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
/*
* ALC880 W810 model
*
* W810 has rear IO for:
* Front (DAC 02)
* Surround (DAC 03)
* Center/LFE (DAC 04)
* Digital out (06)
*
* The system also has a pair of internal speakers, and a headphone jack.
* These are both connected to Line2 on the codec, hence to DAC 02.
*
* There is a variable resistor to control the speaker or headphone
* volume. This is a hardware-only device without a software API.
*
* Plugging headphones in will disable the internal speakers. This is
* implemented in hardware, not via the driver using jack sense. In
* a similar fashion, plugging into the rear socket marked "front" will
* disable both the speakers and headphones.
*
* For input, there's a microphone jack, and an "audio in" jack.
* These may not do anything useful with this driver yet, because I
* haven't setup any initialization verbs for these yet...
*/
static hda_nid_t alc880_w810_dac_nids[3] = {
/* front, rear/surround, clfe */
0x02, 0x03, 0x04
};
/* fixed 6 channels */
static struct hda_channel_mode alc880_w810_modes[1] = {
{ 6, NULL }
};
/* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */
static struct snd_kcontrol_new alc880_w810_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
{ } /* end */
};
/*
* Z710V model
*
* DAC: Front = 0x02 (0x0c), HP = 0x03 (0x0d)
* Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?), Line = 0x1a
*/
static hda_nid_t alc880_z71v_dac_nids[1] = {
0x02
};
#define ALC880_Z71V_HP_DAC 0x03
/* fixed 2 channels */
static struct hda_channel_mode alc880_2_jack_modes[1] = {
{ 2, NULL }
};
static struct snd_kcontrol_new alc880_z71v_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
/* FIXME! */
/*
* ALC880 F1734 model
*
* DAC: HP = 0x02 (0x0c), Front = 0x03 (0x0d)
* Pin assignment: HP = 0x14, Front = 0x15, Mic = 0x18
*/
static hda_nid_t alc880_f1734_dac_nids[1] = {
0x03
};
#define ALC880_F1734_HP_DAC 0x02
static struct snd_kcontrol_new alc880_f1734_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Internal Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
/* FIXME! */
/*
* ALC880 ASUS model
*
* DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
* Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
* Mic = 0x18, Line = 0x1a
*/
#define alc880_asus_dac_nids alc880_w810_dac_nids /* identical with w810 */
#define alc880_asus_modes alc880_threestack_modes /* 2/6 channel mode */
static struct snd_kcontrol_new alc880_asus_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
/* FIXME! */
/*
* ALC880 ASUS W1V model
*
* DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
* Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
* Mic = 0x18, Line = 0x1a, Line2 = 0x1b
*/
/* additional mixers to alc880_asus_mixer */
static struct snd_kcontrol_new alc880_asus_w1v_mixer[] = {
HDA_CODEC_VOLUME("Line2 Playback Volume", 0x0b, 0x03, HDA_INPUT),
HDA_CODEC_MUTE("Line2 Playback Switch", 0x0b, 0x03, HDA_INPUT),
{ } /* end */
};
/* additional mixers to alc880_asus_mixer */
static struct snd_kcontrol_new alc880_pcbeep_mixer[] = {
HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
{ } /* end */
};
/* TCL S700 */
static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0B, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0B, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0B, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0B, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
* FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.count = 1,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{ } /* end */
};
/*
* build control elements
*/
static int alc_build_controls(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err;
int i;
for (i = 0; i < spec->num_mixers; i++) {
err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
if (err < 0)
return err;
}
if (spec->multiout.dig_out_nid) {
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
if (err < 0)
return err;
}
if (spec->dig_in_nid) {
err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
if (err < 0)
return err;
}
return 0;
}
/*
* initialize the codec volumes, etc
*/
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc880_volume_init_verbs[] = {
/*
* Unmute ADC0-2 and set the default input to mic-in
*/
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for front panel
* mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
/*
* Set up output mixers (0x0c - 0x0f)
*/
/* set vol=0 to output mixers */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ }
};
/*
* 3-stack pin configuration:
* front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
*/
static struct hda_verb alc880_pin_3stack_init_verbs[] = {
/*
* preset connection lists of input pins
* 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
*/
{0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
{0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
/*
* Set pin mode and muting
*/
/* set front pin widgets 0x14 for output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Mic2 (as headphone out) for HP output */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Line In pin widget for input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line2 (as front mic) pin widget for input and vref at 80% */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/*
* 5-stack pin configuration:
* front = 0x14, surround = 0x17, clfe = 0x16, mic = 0x18, HP = 0x19,
* line-in/side = 0x1a, f-mic = 0x1b
*/
static struct hda_verb alc880_pin_5stack_init_verbs[] = {
/*
* preset connection lists of input pins
* 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
*/
{0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/side */
/*
* Set pin mode and muting
*/
/* set pin widgets 0x14-0x17 for output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* unmute pins for output (no gain on this amp) */
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Mic2 (as headphone out) for HP output */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Line In pin widget for input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line2 (as front mic) pin widget for input and vref at 80% */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/*
* W810 pin configuration:
* front = 0x14, surround = 0x15, clfe = 0x16, HP = 0x1b
*/
static struct hda_verb alc880_pin_w810_init_verbs[] = {
/* hphone/speaker input selector: front DAC */
{0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{ }
};
/*
* Z71V pin configuration:
* Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?)
*/
static struct hda_verb alc880_pin_z71v_init_verbs[] = {
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/*
* 6-stack pin configuration:
* front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18, f-mic = 0x19,
* line = 0x1a, HP = 0x1b
*/
static struct hda_verb alc880_pin_6stack_init_verbs[] = {
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/* FIXME! */
/*
* F1734 pin configuration:
* HP = 0x14, speaker-out = 0x15, mic = 0x18
*/
static struct hda_verb alc880_pin_f1734_init_verbs[] = {
{0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
{0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/* FIXME! */
/*
* ASUS pin configuration:
* HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a
*/
static struct hda_verb alc880_pin_asus_init_verbs[] = {
{0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
{0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/* Enable GPIO mask and set output */
static struct hda_verb alc880_gpio1_init_verbs[] = {
{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
{ }
};
/* Enable GPIO mask and set output */
static struct hda_verb alc880_gpio2_init_verbs[] = {
{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
{0x01, AC_VERB_SET_GPIO_DATA, 0x02},
{ }
};
/* Clevo m520g init */
static struct hda_verb alc880_pin_clevo_init_verbs[] = {
/* headphone output */
{0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
/* line-out */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Line-in */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* CD */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mic1 (rear panel) */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mic2 (front panel) */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* headphone */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* change to EAPD mode */
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3060},
{ }
};
static struct hda_verb alc880_pin_tcl_S700_init_verbs[] = {
/* Headphone output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Front output*/
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Line In pin widget for input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
/* change to EAPD mode */
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3070},
{ }
};
/*
* LG m1 express dual
*
* Pin assignment:
* Rear Line-In/Out (blue): 0x14
* Build-in Mic-In: 0x15
* Speaker-out: 0x17
* HP-Out (green): 0x1b
* Mic-In/Out (red): 0x19
* SPDIF-Out: 0x1e
*/
/* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */
static hda_nid_t alc880_lg_dac_nids[3] = {
0x05, 0x02, 0x03
};
/* seems analog CD is not working */
static struct hda_input_mux alc880_lg_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x1 },
{ "Line", 0x5 },
{ "Internal Mic", 0x6 },
},
};
/* 2,4,6 channel modes */
static struct hda_verb alc880_lg_ch2_init[] = {
/* set line-in and mic-in to input */
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ }
};
static struct hda_verb alc880_lg_ch4_init[] = {
/* set line-in to out and mic-in to input */
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ }
};
static struct hda_verb alc880_lg_ch6_init[] = {
/* set line-in and mic-in to output */
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
{ }
};
static struct hda_channel_mode alc880_lg_ch_modes[3] = {
{ 2, alc880_lg_ch2_init },
{ 4, alc880_lg_ch4_init },
{ 6, alc880_lg_ch6_init },
};
static struct snd_kcontrol_new alc880_lg_mixer[] = {
/* FIXME: it's not really "master" but front channels */
HDA_CODEC_VOLUME("Master Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x06, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x06, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x07, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x07, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
static struct hda_verb alc880_lg_init_verbs[] = {
/* set capture source to mic-in */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* mute all amp mixer inputs */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)},
/* line-in to input */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* built-in mic */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* speaker-out */
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* mic-in to input */
{0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* HP-out */
{0x13, AC_VERB_SET_CONNECT_SEL, 0x03},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* jack sense */
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc880_lg_automute(struct hda_codec *codec)
{
unsigned int present;
present = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
snd_hda_codec_amp_update(codec, 0x17, 0, HDA_OUTPUT, 0,
0x80, present ? 0x80 : 0);
snd_hda_codec_amp_update(codec, 0x17, 1, HDA_OUTPUT, 0,
0x80, present ? 0x80 : 0);
}
static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res)
{
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
if ((res >> 28) == 0x01)
alc880_lg_automute(codec);
}
/*
* LG LW20
*
* Pin assignment:
* Speaker-out: 0x14
* Mic-In: 0x18
* Built-in Mic-In: 0x19 (?)
* HP-Out: 0x1b
* SPDIF-Out: 0x1e
*/
/* seems analog CD is not working */
static struct hda_input_mux alc880_lg_lw_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x0 },
{ "Internal Mic", 0x1 },
},
};
static struct snd_kcontrol_new alc880_lg_lw_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
{ } /* end */
};
static struct hda_verb alc880_lg_lw_init_verbs[] = {
/* set capture source to mic-in */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)},
/* speaker-out */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* HP-out */
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* mic-in to input */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* built-in mic */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* jack sense */
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc880_lg_lw_automute(struct hda_codec *codec)
{
unsigned int present;
present = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
0x80, present ? 0x80 : 0);
snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
0x80, present ? 0x80 : 0);
}
static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res)
{
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
if ((res >> 28) == 0x01)
alc880_lg_lw_automute(codec);
}
/*
* Common callbacks
*/
static int alc_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int i;
for (i = 0; i < spec->num_init_verbs; i++)
snd_hda_sequence_write(codec, spec->init_verbs[i]);
if (spec->init_hook)
spec->init_hook(codec);
return 0;
}
static void alc_unsol_event(struct hda_codec *codec, unsigned int res)
{
struct alc_spec *spec = codec->spec;
if (spec->unsol_event)
spec->unsol_event(codec, res);
}
#ifdef CONFIG_PM
/*
* resume
*/
static int alc_resume(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
alc_init(codec);
for (i = 0; i < spec->num_mixers; i++)
snd_hda_resume_ctls(codec, spec->mixers[i]);
if (spec->multiout.dig_out_nid)
snd_hda_resume_spdif_out(codec);
if (spec->dig_in_nid)
snd_hda_resume_spdif_in(codec);
return 0;
}
#endif
/*
* Analog playback callbacks
*/
static int alc880_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
}
static int alc880_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
format, substream);
}
static int alc880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
}
/*
* Digital out
*/
static int alc880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_dig_open(codec, &spec->multiout);
}
static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_dig_close(codec, &spec->multiout);
}
/*
* Analog capture
*/
static int alc880_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
stream_tag, 0, format);
return 0;
}
static int alc880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], 0, 0, 0);
return 0;
}
/*
*/
static struct hda_pcm_stream alc880_pcm_analog_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 8,
/* NID is set in alc_build_pcms */
.ops = {
.open = alc880_playback_pcm_open,
.prepare = alc880_playback_pcm_prepare,
.cleanup = alc880_playback_pcm_cleanup
},
};
static struct hda_pcm_stream alc880_pcm_analog_capture = {
.substreams = 2,
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
.ops = {
.prepare = alc880_capture_pcm_prepare,
.cleanup = alc880_capture_pcm_cleanup
},
};
static struct hda_pcm_stream alc880_pcm_digital_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
.ops = {
.open = alc880_dig_playback_pcm_open,
.close = alc880_dig_playback_pcm_close
},
};
static struct hda_pcm_stream alc880_pcm_digital_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
};
/* Used by alc_build_pcms to flag that a PCM has no playback stream */
static struct hda_pcm_stream alc_pcm_null_playback = {
.substreams = 0,
.channels_min = 0,
.channels_max = 0,
};
static int alc_build_pcms(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct hda_pcm *info = spec->pcm_rec;
int i;
codec->num_pcms = 1;
codec->pcm_info = info;
info->name = spec->stream_name_analog;
if (spec->stream_analog_playback) {
snd_assert(spec->multiout.dac_nids, return -EINVAL);
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback);
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
}
if (spec->stream_analog_capture) {
snd_assert(spec->adc_nids, return -EINVAL);
info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
}
if (spec->channel_mode) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = 0;
for (i = 0; i < spec->num_channel_mode; i++) {
if (spec->channel_mode[i].channels > info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->channel_mode[i].channels;
}
}
}
/* If the use of more than one ADC is requested for the current
* model, configure a second analog capture-only PCM.
*/
if (spec->num_adc_nids > 1) {
codec->num_pcms++;
info++;
info->name = spec->stream_name_analog;
/* No playback stream for second PCM */
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = alc_pcm_null_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0;
if (spec->stream_analog_capture) {
snd_assert(spec->adc_nids, return -EINVAL);
info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[1];
}
}
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 &&
spec->stream_digital_playback) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback);
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
}
if (spec->dig_in_nid &&
spec->stream_digital_capture) {
info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_digital_capture);
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
}
}
return 0;
}
static void alc_free(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int i;
if (! spec)
return;
if (spec->kctl_alloc) {
for (i = 0; i < spec->num_kctl_used; i++)
kfree(spec->kctl_alloc[i].name);
kfree(spec->kctl_alloc);
}
kfree(spec);
}
/*
*/
static struct hda_codec_ops alc_patch_ops = {
.build_controls = alc_build_controls,
.build_pcms = alc_build_pcms,
.init = alc_init,
.free = alc_free,
.unsol_event = alc_unsol_event,
#ifdef CONFIG_PM
.resume = alc_resume,
#endif
};
/*
* Test configuration for debugging
*
* Almost all inputs/outputs are enabled. I/O pins can be configured via
* enum controls.
*/
#ifdef CONFIG_SND_DEBUG
static hda_nid_t alc880_test_dac_nids[4] = {
0x02, 0x03, 0x04, 0x05
};
static struct hda_input_mux alc880_test_capture_source = {
.num_items = 7,
.items = {
{ "In-1", 0x0 },
{ "In-2", 0x1 },
{ "In-3", 0x2 },
{ "In-4", 0x3 },
{ "CD", 0x4 },
{ "Front", 0x5 },
{ "Surround", 0x6 },
},
};
static struct hda_channel_mode alc880_test_modes[4] = {
{ 2, NULL },
{ 4, NULL },
{ 6, NULL },
{ 8, NULL },
};
static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = {
"N/A", "Line Out", "HP Out",
"In Hi-Z", "In 50%", "In Grd", "In 80%", "In 100%"
};
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 8;
if (uinfo->value.enumerated.item >= 8)
uinfo->value.enumerated.item = 7;
strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
return 0;
}
static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
unsigned int pin_ctl, item = 0;
pin_ctl = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
if (pin_ctl & AC_PINCTL_OUT_EN) {
if (pin_ctl & AC_PINCTL_HP_EN)
item = 2;
else
item = 1;
} else if (pin_ctl & AC_PINCTL_IN_EN) {
switch (pin_ctl & AC_PINCTL_VREFEN) {
case AC_PINCTL_VREF_HIZ: item = 3; break;
case AC_PINCTL_VREF_50: item = 4; break;
case AC_PINCTL_VREF_GRD: item = 5; break;
case AC_PINCTL_VREF_80: item = 6; break;
case AC_PINCTL_VREF_100: item = 7; break;
}
}
ucontrol->value.enumerated.item[0] = item;
return 0;
}
static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
static unsigned int ctls[] = {
0, AC_PINCTL_OUT_EN, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_50,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_80,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_100,
};
unsigned int old_ctl, new_ctl;
old_ctl = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
new_ctl = ctls[ucontrol->value.enumerated.item[0]];
if (old_ctl != new_ctl) {
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, new_ctl);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
ucontrol->value.enumerated.item[0] >= 3 ? 0xb080 : 0xb000);
return 1;
}
return 0;
}
static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = {
"Front", "Surround", "CLFE", "Side"
};
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 4;
if (uinfo->value.enumerated.item >= 4)
uinfo->value.enumerated.item = 3;
strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
return 0;
}
static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
unsigned int sel;
sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0);
ucontrol->value.enumerated.item[0] = sel & 3;
return 0;
}
static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
unsigned int sel;
sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3;
if (ucontrol->value.enumerated.item[0] != sel) {
sel = ucontrol->value.enumerated.item[0] & 3;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, sel);
return 1;
}
return 0;
}
#define PIN_CTL_TEST(xname,nid) { \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
.info = alc_test_pin_ctl_info, \
.get = alc_test_pin_ctl_get, \
.put = alc_test_pin_ctl_put, \
.private_value = nid \
}
#define PIN_SRC_TEST(xname,nid) { \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
.info = alc_test_pin_src_info, \
.get = alc_test_pin_src_get, \
.put = alc_test_pin_src_put, \
.private_value = nid \
}
static struct snd_kcontrol_new alc880_test_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CLFE Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_BIND_MUTE("CLFE Playback Switch", 0x0e, 2, HDA_INPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
PIN_CTL_TEST("Front Pin Mode", 0x14),
PIN_CTL_TEST("Surround Pin Mode", 0x15),
PIN_CTL_TEST("CLFE Pin Mode", 0x16),
PIN_CTL_TEST("Side Pin Mode", 0x17),
PIN_CTL_TEST("In-1 Pin Mode", 0x18),
PIN_CTL_TEST("In-2 Pin Mode", 0x19),
PIN_CTL_TEST("In-3 Pin Mode", 0x1a),
PIN_CTL_TEST("In-4 Pin Mode", 0x1b),
PIN_SRC_TEST("In-1 Pin Source", 0x18),
PIN_SRC_TEST("In-2 Pin Source", 0x19),
PIN_SRC_TEST("In-3 Pin Source", 0x1a),
PIN_SRC_TEST("In-4 Pin Source", 0x1b),
HDA_CODEC_VOLUME("In-1 Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("In-1 Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("In-2 Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("In-2 Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("In-3 Playback Volume", 0x0b, 0x2, HDA_INPUT),
HDA_CODEC_MUTE("In-3 Playback Switch", 0x0b, 0x2, HDA_INPUT),
HDA_CODEC_VOLUME("In-4 Playback Volume", 0x0b, 0x3, HDA_INPUT),
HDA_CODEC_MUTE("In-4 Playback Switch", 0x0b, 0x3, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x4, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x4, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
static struct hda_verb alc880_test_init_verbs[] = {
/* Unmute inputs of 0x0c - 0x0f */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Vol output for 0x0c-0x0f */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Set output pins 0x14-0x17 */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* Unmute output pins 0x14-0x17 */
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Set input pins 0x18-0x1c */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Mute input pins 0x18-0x1b */
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* ADC set up */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Analog input/passthru */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
{ }
};
#endif
/*
*/
static struct hda_board_config alc880_cfg_tbl[] = {
/* Back 3 jack, front 2 jack */
{ .modelname = "3stack", .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe200, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe201, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe202, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe203, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe204, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe205, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe206, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe207, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe208, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe209, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe20a, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe20b, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe20c, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe20d, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe20e, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe20f, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe210, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe211, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe214, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe302, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe303, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe304, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe306, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe307, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe404, .config = ALC880_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xa101, .config = ALC880_3ST },
{ .pci_subvendor = 0x107b, .pci_subdevice = 0x3031, .config = ALC880_3ST },
{ .pci_subvendor = 0x107b, .pci_subdevice = 0x4036, .config = ALC880_3ST },
{ .pci_subvendor = 0x107b, .pci_subdevice = 0x4037, .config = ALC880_3ST },
{ .pci_subvendor = 0x107b, .pci_subdevice = 0x4038, .config = ALC880_3ST },
{ .pci_subvendor = 0x107b, .pci_subdevice = 0x4040, .config = ALC880_3ST },
{ .pci_subvendor = 0x107b, .pci_subdevice = 0x4041, .config = ALC880_3ST },
/* TCL S700 */
{ .pci_subvendor = 0x19db, .pci_subdevice = 0x4188, .config = ALC880_TCL_S700 },
/* Back 3 jack, front 2 jack (Internal add Aux-In) */
{ .pci_subvendor = 0x1025, .pci_subdevice = 0xe310, .config = ALC880_3ST },
{ .pci_subvendor = 0x104d, .pci_subdevice = 0x81d6, .config = ALC880_3ST },
{ .pci_subvendor = 0x104d, .pci_subdevice = 0x81a0, .config = ALC880_3ST },
/* Back 3 jack plus 1 SPDIF out jack, front 2 jack */
{ .modelname = "3stack-digout", .config = ALC880_3ST_DIG },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe308, .config = ALC880_3ST_DIG },
{ .pci_subvendor = 0x1025, .pci_subdevice = 0x0070, .config = ALC880_3ST_DIG },
/* Clevo m520G NB */
{ .pci_subvendor = 0x1558, .pci_subdevice = 0x0520, .config = ALC880_CLEVO },
/* Back 3 jack plus 1 SPDIF out jack, front 2 jack (Internal add Aux-In)*/
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe305, .config = ALC880_3ST_DIG },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xd402, .config = ALC880_3ST_DIG },
{ .pci_subvendor = 0x1025, .pci_subdevice = 0xe309, .config = ALC880_3ST_DIG },
/* Back 5 jack, front 2 jack */
{ .modelname = "5stack", .config = ALC880_5ST },
{ .pci_subvendor = 0x107b, .pci_subdevice = 0x3033, .config = ALC880_5ST },
{ .pci_subvendor = 0x107b, .pci_subdevice = 0x4039, .config = ALC880_5ST },
{ .pci_subvendor = 0x107b, .pci_subdevice = 0x3032, .config = ALC880_5ST },
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x2a09, .config = ALC880_5ST },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x814e, .config = ALC880_5ST },
/* Back 5 jack plus 1 SPDIF out jack, front 2 jack */
{ .modelname = "5stack-digout", .config = ALC880_5ST_DIG },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe224, .config = ALC880_5ST_DIG },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe400, .config = ALC880_5ST_DIG },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe401, .config = ALC880_5ST_DIG },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xe402, .config = ALC880_5ST_DIG },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xd400, .config = ALC880_5ST_DIG },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xd401, .config = ALC880_5ST_DIG },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xa100, .config = ALC880_5ST_DIG },
{ .pci_subvendor = 0x1565, .pci_subdevice = 0x8202, .config = ALC880_5ST_DIG },
{ .pci_subvendor = 0x1019, .pci_subdevice = 0xa880, .config = ALC880_5ST_DIG },
{ .pci_subvendor = 0xa0a0, .pci_subdevice = 0x0560,
.config = ALC880_5ST_DIG }, /* Aopen i915GMm-HFS */
/* { .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_5ST_DIG }, */ /* conflict with 6stack */
{ .pci_subvendor = 0x1695, .pci_subdevice = 0x400d, .config = ALC880_5ST_DIG },
/* note subvendor = 0 below */
/* { .pci_subvendor = 0x0000, .pci_subdevice = 0x8086, .config = ALC880_5ST_DIG }, */
{ .modelname = "w810", .config = ALC880_W810 },
{ .pci_subvendor = 0x161f, .pci_subdevice = 0x203d, .config = ALC880_W810 },
{ .modelname = "z71v", .config = ALC880_Z71V },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_Z71V },
{ .modelname = "6stack", .config = ALC880_6ST },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x8196, .config = ALC880_6ST }, /* ASUS P5GD1-HVM */
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x81b4, .config = ALC880_6ST },
{ .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_6ST }, /* Acer APFV */
{ .pci_subvendor = 0x1458, .pci_subdevice = 0xa102, .config = ALC880_6ST }, /* Gigabyte K8N51 */
{ .modelname = "6stack-digout", .config = ALC880_6ST_DIG },
{ .pci_subvendor = 0x2668, .pci_subdevice = 0x8086, .config = ALC880_6ST_DIG },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0x2668, .config = ALC880_6ST_DIG },
{ .pci_subvendor = 0x1462, .pci_subdevice = 0x1150, .config = ALC880_6ST_DIG },
{ .pci_subvendor = 0xe803, .pci_subdevice = 0x1019, .config = ALC880_6ST_DIG },
{ .pci_subvendor = 0x1039, .pci_subdevice = 0x1234, .config = ALC880_6ST_DIG },
{ .pci_subvendor = 0x1025, .pci_subdevice = 0x0077, .config = ALC880_6ST_DIG },
{ .pci_subvendor = 0x1025, .pci_subdevice = 0x0078, .config = ALC880_6ST_DIG },
{ .pci_subvendor = 0x1025, .pci_subdevice = 0x0087, .config = ALC880_6ST_DIG },
{ .pci_subvendor = 0x1297, .pci_subdevice = 0xc790, .config = ALC880_6ST_DIG }, /* Shuttle ST20G5 */
{ .pci_subvendor = 0x1509, .pci_subdevice = 0x925d, .config = ALC880_6ST_DIG }, /* FIC P4M-915GD1 */
{ .pci_subvendor = 0x1695, .pci_subdevice = 0x4012, .config = ALC880_5ST_DIG }, /* Epox EP-5LDA+ GLi */
{ .modelname = "asus", .config = ALC880_ASUS },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_ASUS_DIG },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1973, .config = ALC880_ASUS_DIG },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x19b3, .config = ALC880_ASUS_DIG },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1113, .config = ALC880_ASUS_DIG },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1173, .config = ALC880_ASUS_DIG },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1993, .config = ALC880_ASUS },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x10c3, .config = ALC880_ASUS_DIG },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1133, .config = ALC880_ASUS },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1123, .config = ALC880_ASUS_DIG },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1143, .config = ALC880_ASUS },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x10b3, .config = ALC880_ASUS_W1V },
{ .pci_subvendor = 0x1043, .pci_subdevice = 0x8181, .config = ALC880_ASUS_DIG }, /* ASUS P4GPL-X */
{ .pci_subvendor = 0x1558, .pci_subdevice = 0x5401, .config = ALC880_ASUS_DIG2 },
{ .modelname = "uniwill", .config = ALC880_UNIWILL_DIG },
{ .pci_subvendor = 0x1584, .pci_subdevice = 0x9050, .config = ALC880_UNIWILL_DIG },
{ .modelname = "F1734", .config = ALC880_F1734 },
{ .pci_subvendor = 0x1734, .pci_subdevice = 0x107c, .config = ALC880_F1734 },
{ .pci_subvendor = 0x1584, .pci_subdevice = 0x9054, .config = ALC880_F1734 },
{ .modelname = "lg", .config = ALC880_LG },
{ .pci_subvendor = 0x1854, .pci_subdevice = 0x003b, .config = ALC880_LG },
{ .modelname = "lg-lw", .config = ALC880_LG_LW },
{ .pci_subvendor = 0x1854, .pci_subdevice = 0x0018, .config = ALC880_LG_LW },
#ifdef CONFIG_SND_DEBUG
{ .modelname = "test", .config = ALC880_TEST },
#endif
{ .modelname = "auto", .config = ALC880_AUTO },
{}
};
/*
* ALC880 codec presets
*/
static struct alc_config_preset alc880_presets[] = {
[ALC880_3ST] = {
.mixers = { alc880_three_stack_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_dac_nids),
.dac_nids = alc880_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
.channel_mode = alc880_threestack_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_3ST_DIG] = {
.mixers = { alc880_three_stack_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_dac_nids),
.dac_nids = alc880_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
.channel_mode = alc880_threestack_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_TCL_S700] = {
.mixers = { alc880_tcl_s700_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_pin_tcl_S700_init_verbs,
alc880_gpio2_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_dac_nids),
.dac_nids = alc880_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_5ST] = {
.mixers = { alc880_three_stack_mixer, alc880_five_stack_mixer},
.init_verbs = { alc880_volume_init_verbs, alc880_pin_5stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_dac_nids),
.dac_nids = alc880_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
.channel_mode = alc880_fivestack_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_5ST_DIG] = {
.mixers = { alc880_three_stack_mixer, alc880_five_stack_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_5stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_dac_nids),
.dac_nids = alc880_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
.channel_mode = alc880_fivestack_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_6ST] = {
.mixers = { alc880_six_stack_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_6stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
.dac_nids = alc880_6st_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
.channel_mode = alc880_sixstack_modes,
.input_mux = &alc880_6stack_capture_source,
},
[ALC880_6ST_DIG] = {
.mixers = { alc880_six_stack_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_6stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
.dac_nids = alc880_6st_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
.channel_mode = alc880_sixstack_modes,
.input_mux = &alc880_6stack_capture_source,
},
[ALC880_W810] = {
.mixers = { alc880_w810_base_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_w810_init_verbs,
alc880_gpio2_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_w810_dac_nids),
.dac_nids = alc880_w810_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_w810_modes),
.channel_mode = alc880_w810_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_Z71V] = {
.mixers = { alc880_z71v_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_z71v_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_z71v_dac_nids),
.dac_nids = alc880_z71v_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_F1734] = {
.mixers = { alc880_f1734_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_f1734_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_f1734_dac_nids),
.dac_nids = alc880_f1734_dac_nids,
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_ASUS] = {
.mixers = { alc880_asus_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
.dac_nids = alc880_asus_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
.channel_mode = alc880_asus_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_ASUS_DIG] = {
.mixers = { alc880_asus_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
.dac_nids = alc880_asus_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
.channel_mode = alc880_asus_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_ASUS_DIG2] = {
.mixers = { alc880_asus_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs,
alc880_gpio2_init_verbs }, /* use GPIO2 */
.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
.dac_nids = alc880_asus_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
.channel_mode = alc880_asus_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_ASUS_W1V] = {
.mixers = { alc880_asus_mixer, alc880_asus_w1v_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
.dac_nids = alc880_asus_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
.channel_mode = alc880_asus_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_UNIWILL_DIG] = {
.mixers = { alc880_asus_mixer, alc880_pcbeep_mixer },
.init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
.dac_nids = alc880_asus_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
.channel_mode = alc880_asus_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_CLEVO] = {
.mixers = { alc880_three_stack_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_pin_clevo_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_dac_nids),
.dac_nids = alc880_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
.channel_mode = alc880_threestack_modes,
.input_mux = &alc880_capture_source,
},
[ALC880_LG] = {
.mixers = { alc880_lg_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_lg_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_lg_dac_nids),
.dac_nids = alc880_lg_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes),
.channel_mode = alc880_lg_ch_modes,
.input_mux = &alc880_lg_capture_source,
.unsol_event = alc880_lg_unsol_event,
.init_hook = alc880_lg_automute,
},
[ALC880_LG_LW] = {
.mixers = { alc880_lg_lw_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_lg_lw_init_verbs },
.num_dacs = 1,
.dac_nids = alc880_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_lg_lw_capture_source,
.unsol_event = alc880_lg_lw_unsol_event,
.init_hook = alc880_lg_lw_automute,
},
#ifdef CONFIG_SND_DEBUG
[ALC880_TEST] = {
.mixers = { alc880_test_mixer },
.init_verbs = { alc880_test_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_test_dac_nids),
.dac_nids = alc880_test_dac_nids,
.dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_test_modes),
.channel_mode = alc880_test_modes,
.input_mux = &alc880_test_capture_source,
},
#endif
};
/*
* Automatic parse of I/O pins from the BIOS configuration
*/
#define NUM_CONTROL_ALLOC 32
#define NUM_VERB_ALLOC 32
enum {
ALC_CTL_WIDGET_VOL,
ALC_CTL_WIDGET_MUTE,
ALC_CTL_BIND_MUTE,
};
static struct snd_kcontrol_new alc880_control_templates[] = {
HDA_CODEC_VOLUME(NULL, 0, 0, 0),
HDA_CODEC_MUTE(NULL, 0, 0, 0),
HDA_BIND_MUTE(NULL, 0, 0, 0),
};
/* add dynamic controls */
static int add_control(struct alc_spec *spec, int type, const char *name, unsigned long val)
{
struct snd_kcontrol_new *knew;
if (spec->num_kctl_used >= spec->num_kctl_alloc) {
int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC;
knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); /* array + terminator */
if (! knew)
return -ENOMEM;
if (spec->kctl_alloc) {
memcpy(knew, spec->kctl_alloc, sizeof(*knew) * spec->num_kctl_alloc);
kfree(spec->kctl_alloc);
}
spec->kctl_alloc = knew;
spec->num_kctl_alloc = num;
}
knew = &spec->kctl_alloc[spec->num_kctl_used];
*knew = alc880_control_templates[type];
knew->name = kstrdup(name, GFP_KERNEL);
if (! knew->name)
return -ENOMEM;
knew->private_value = val;
spec->num_kctl_used++;
return 0;
}
#define alc880_is_fixed_pin(nid) ((nid) >= 0x14 && (nid) <= 0x17)
#define alc880_fixed_pin_idx(nid) ((nid) - 0x14)
#define alc880_is_multi_pin(nid) ((nid) >= 0x18)
#define alc880_multi_pin_idx(nid) ((nid) - 0x18)
#define alc880_is_input_pin(nid) ((nid) >= 0x18)
#define alc880_input_pin_idx(nid) ((nid) - 0x18)
#define alc880_idx_to_dac(nid) ((nid) + 0x02)
#define alc880_dac_to_idx(nid) ((nid) - 0x02)
#define alc880_idx_to_mixer(nid) ((nid) + 0x0c)
#define alc880_idx_to_selector(nid) ((nid) + 0x10)
#define ALC880_PIN_CD_NID 0x1c
/* fill in the dac_nids table from the parsed pin configuration */
static int alc880_auto_fill_dac_nids(struct alc_spec *spec, const struct auto_pin_cfg *cfg)
{
hda_nid_t nid;
int assigned[4];
int i, j;
memset(assigned, 0, sizeof(assigned));
spec->multiout.dac_nids = spec->private_dac_nids;
/* check the pins hardwired to audio widget */
for (i = 0; i < cfg->line_outs; i++) {
nid = cfg->line_out_pins[i];
if (alc880_is_fixed_pin(nid)) {
int idx = alc880_fixed_pin_idx(nid);
spec->multiout.dac_nids[i] = alc880_idx_to_dac(idx);
assigned[idx] = 1;
}
}
/* left pins can be connect to any audio widget */
for (i = 0; i < cfg->line_outs; i++) {
nid = cfg->line_out_pins[i];
if (alc880_is_fixed_pin(nid))
continue;
/* search for an empty channel */
for (j = 0; j < cfg->line_outs; j++) {
if (! assigned[j]) {
spec->multiout.dac_nids[i] = alc880_idx_to_dac(j);
assigned[j] = 1;
break;
}
}
}
spec->multiout.num_dacs = cfg->line_outs;
return 0;
}
/* add playback controls from the parsed DAC table */
static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
{
char name[32];
static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" };
hda_nid_t nid;
int i, err;
for (i = 0; i < cfg->line_outs; i++) {
if (! spec->multiout.dac_nids[i])
continue;
nid = alc880_idx_to_mixer(alc880_dac_to_idx(spec->multiout.dac_nids[i]));
if (i == 2) {
/* Center/LFE */
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Center Playback Volume",
HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0)
return err;
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "LFE Playback Volume",
HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0)
return err;
if ((err = add_control(spec, ALC_CTL_BIND_MUTE, "Center Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 1, 2, HDA_INPUT))) < 0)
return err;
if ((err = add_control(spec, ALC_CTL_BIND_MUTE, "LFE Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 2, 2, HDA_INPUT))) < 0)
return err;
} else {
sprintf(name, "%s Playback Volume", chname[i]);
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
return err;
sprintf(name, "%s Playback Switch", chname[i]);
if ((err = add_control(spec, ALC_CTL_BIND_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT))) < 0)
return err;
}
}
return 0;
}
/* add playback controls for speaker and HP outputs */
static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
const char *pfx)
{
hda_nid_t nid;
int err;
char name[32];
if (! pin)
return 0;
if (alc880_is_fixed_pin(pin)) {
nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
/* specify the DAC as the extra output */
if (! spec->multiout.hp_nid)
spec->multiout.hp_nid = nid;
else
spec->multiout.extra_out_nid[0] = nid;
/* control HP volume/switch on the output mixer amp */
nid = alc880_idx_to_mixer(alc880_fixed_pin_idx(pin));
sprintf(name, "%s Playback Volume", pfx);
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
return err;
sprintf(name, "%s Playback Switch", pfx);
if ((err = add_control(spec, ALC_CTL_BIND_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT))) < 0)
return err;
} else if (alc880_is_multi_pin(pin)) {
/* set manual connection */
/* we have only a switch on HP-out PIN */
sprintf(name, "%s Playback Switch", pfx);
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT))) < 0)
return err;
}
return 0;
}
/* create input playback/capture controls for the given pin */
static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, const char *ctlname,
int idx, hda_nid_t mix_nid)
{
char name[32];
int err;
sprintf(name, "%s Playback Volume", ctlname);
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT))) < 0)
return err;
sprintf(name, "%s Playback Switch", ctlname);
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT))) < 0)
return err;
return 0;
}
/* create playback/capture controls for input pins */
static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
{
struct hda_input_mux *imux = &spec->private_imux;
int i, err, idx;
for (i = 0; i < AUTO_PIN_LAST; i++) {
if (alc880_is_input_pin(cfg->input_pins[i])) {
idx = alc880_input_pin_idx(cfg->input_pins[i]);
err = new_analog_input(spec, cfg->input_pins[i],
auto_pin_cfg_labels[i],
idx, 0x0b);
if (err < 0)
return err;
imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
imux->items[imux->num_items].index = alc880_input_pin_idx(cfg->input_pins[i]);
imux->num_items++;
}
}
return 0;
}
static void alc880_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
int dac_idx)
{
/* set as output */
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
/* need the manual connection? */
if (alc880_is_multi_pin(nid)) {
struct alc_spec *spec = codec->spec;
int idx = alc880_multi_pin_idx(nid);
snd_hda_codec_write(codec, alc880_idx_to_selector(idx), 0,
AC_VERB_SET_CONNECT_SEL,
alc880_dac_to_idx(spec->multiout.dac_nids[dac_idx]));
}
}
static void alc880_auto_init_multi_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < spec->autocfg.line_outs; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
alc880_auto_set_output_and_unmute(codec, nid, PIN_OUT, i);
}
}
static void alc880_auto_init_extra_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t pin;
pin = spec->autocfg.speaker_pins[0];
if (pin) /* connect to front */
alc880_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
pin = spec->autocfg.hp_pin;
if (pin) /* connect to front */
alc880_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
}
static void alc880_auto_init_analog_input(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = spec->autocfg.input_pins[i];
if (alc880_is_input_pin(nid)) {
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
i <= AUTO_PIN_FRONT_MIC ? PIN_VREF80 : PIN_IN);
if (nid != ALC880_PIN_CD_NID)
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_MUTE);
}
}
}
/* parse the BIOS configuration and set up the alc_spec */
/* return 1 if successful, 0 if the proper config is not found, or a negative error code */
static int alc880_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err;
static hda_nid_t alc880_ignore[] = { 0x1d, 0 };
if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc880_ignore)) < 0)
return err;
if (! spec->autocfg.line_outs)
return 0; /* can't find valid BIOS pin config */
if ((err = alc880_auto_fill_dac_nids(spec, &spec->autocfg)) < 0 ||
(err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 ||
(err = alc880_auto_create_extra_out(spec,
spec->autocfg.speaker_pins[0],
"Speaker")) < 0 ||
(err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pin,
"Headphone")) < 0 ||
(err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
if (spec->autocfg.dig_out_pin)
spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
if (spec->autocfg.dig_in_pin)
spec->dig_in_nid = ALC880_DIGIN_NID;
if (spec->kctl_alloc)
spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
spec->init_verbs[spec->num_init_verbs++] = alc880_volume_init_verbs;
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux;
return 1;
}
/* additional initialization for auto-configuration model */
static void alc880_auto_init(struct hda_codec *codec)
{
alc880_auto_init_multi_out(codec);
alc880_auto_init_extra_out(codec);
alc880_auto_init_analog_input(codec);
}
/*
* OK, here we have finally the patch for ALC880
*/
static int patch_alc880(struct hda_codec *codec)
{
struct alc_spec *spec;
int board_config;
int err;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
board_config = snd_hda_check_board_config(codec, alc880_cfg_tbl);
if (board_config < 0 || board_config >= ALC880_MODEL_LAST) {
printk(KERN_INFO "hda_codec: Unknown model for ALC880, trying auto-probe from BIOS...\n");
board_config = ALC880_AUTO;
}
if (board_config == ALC880_AUTO) {
/* automatic parse from the BIOS config */
err = alc880_parse_auto_config(codec);
if (err < 0) {
alc_free(codec);
return err;
} else if (! err) {
printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using 3-stack mode...\n");
board_config = ALC880_3ST;
}
}
if (board_config != ALC880_AUTO)
setup_preset(spec, &alc880_presets[board_config]);
spec->stream_name_analog = "ALC880 Analog";
spec->stream_analog_playback = &alc880_pcm_analog_playback;
spec->stream_analog_capture = &alc880_pcm_analog_capture;
spec->stream_name_digital = "ALC880 Digital";
spec->stream_digital_playback = &alc880_pcm_digital_playback;
spec->stream_digital_capture = &alc880_pcm_digital_capture;
if (! spec->adc_nids && spec->input_mux) {
/* check whether NID 0x07 is valid */
unsigned int wcap = get_wcaps(codec, alc880_adc_nids[0]);
wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */
if (wcap != AC_WID_AUD_IN) {
spec->adc_nids = alc880_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids_alt);
spec->mixers[spec->num_mixers] = alc880_capture_alt_mixer;
spec->num_mixers++;
} else {
spec->adc_nids = alc880_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids);
spec->mixers[spec->num_mixers] = alc880_capture_mixer;
spec->num_mixers++;
}
}
codec->patch_ops = alc_patch_ops;
if (board_config == ALC880_AUTO)
spec->init_hook = alc880_auto_init;
return 0;
}
/*
* ALC260 support
*/
static hda_nid_t alc260_dac_nids[1] = {
/* front */
0x02,
};
static hda_nid_t alc260_adc_nids[1] = {
/* ADC0 */
0x04,
};
static hda_nid_t alc260_adc_nids_alt[1] = {
/* ADC1 */
0x05,
};
static hda_nid_t alc260_hp_adc_nids[2] = {
/* ADC1, 0 */
0x05, 0x04
};
/* NIDs used when simultaneous access to both ADCs makes sense. Note that
* alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC.
*/
static hda_nid_t alc260_dual_adc_nids[2] = {
/* ADC0, ADC1 */
0x04, 0x05
};
#define ALC260_DIGOUT_NID 0x03
#define ALC260_DIGIN_NID 0x06
static struct hda_input_mux alc260_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack,
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
* headphone jack and the internal CD lines since these are the only pins at
* which audio can appear. For flexibility, also allow the option of
* recording the mixer output on the second ADC (ADC0 doesn't have a
* connection to the mixer output).
*/
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
static struct hda_input_mux alc260_fujitsu_capture_sources[2] = {
{
.num_items = 3,
.items = {
{ "Mic/Line", 0x0 },
{ "CD", 0x4 },
{ "Headphone", 0x2 },
},
},
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
{
.num_items = 4,
.items = {
{ "Mic/Line", 0x0 },
{ "CD", 0x4 },
{ "Headphone", 0x2 },
{ "Mixer", 0x5 },
},
},
};
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to
* the Fujitsu S702x, but jacks are marked differently.
*/
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
static struct hda_input_mux alc260_acer_capture_sources[2] = {
{
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Headphone", 0x5 },
},
},
{
.num_items = 5,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Headphone", 0x6 },
{ "Mixer", 0x5 },
},
},
};
/*
* This is just place-holder, so there's something for alc_build_pcms to look
* at when it calculates the maximum number of channels. ALC260 has no mixer
* element which allows changing the channel mode, so the verb list is
* never used.
*/
static struct hda_channel_mode alc260_modes[1] = {
{ 2, NULL },
};
/* Mixer combinations
*
* basic: base_output + input + pc_beep + capture
* HP: base_output + input + capture_alt
* HP_3013: hp_3013 + input + capture
* fujitsu: fujitsu + capture
* acer: acer + capture
*/
static struct snd_kcontrol_new alc260_base_output_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc260_input_mixer[] = {
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc260_pc_beep_mixer[] = {
HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Aux-In Playback Volume", 0x07, 0x06, HDA_INPUT),
HDA_CODEC_MUTE("Aux-In Playback Switch", 0x07, 0x06, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("iSpeaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("iSpeaker Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT),
{ } /* end */
};
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12,
* HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10.
*/
static struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT),
ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT),
ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN),
HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Internal Speaker Playback Switch", 0x09, 2, HDA_INPUT),
{ } /* end */
};
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks. Note that current
* versions of the ALC260 don't act on requests to enable mic bias from NID
* 0x0f (used to drive the headphone jack in these laptops). The ALC260
* datasheet doesn't mention this restriction. At this stage it's not clear
* whether this behaviour is intentional or is a hardware bug in chip
* revisions available in early 2006. Therefore for now allow the
* "Headphone Jack Mode" control to span all choices, but if it turns out
* that the lack of mic bias for this NID is intentional we could change the
* mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
*
* In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006
* don't appear to make the mic bias available from the "line" jack, even
* though the NID used for this jack (0x14) can supply it. The theory is
* that perhaps Acer have included blocking capacitors between the ALC260
* and the output jack. If this turns out to be the case for all such
* models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT
* to ALC_PIN_DIR_INOUT_NOMICBIAS.
*/
static struct snd_kcontrol_new alc260_acer_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
{ } /* end */
};
/* capture mixer elements */
static struct snd_kcontrol_new alc260_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x04, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x05, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x05, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
* FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.count = 2,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{ } /* end */
};
static struct snd_kcontrol_new alc260_capture_alt_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x05, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x05, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
* FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.count = 1,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{ } /* end */
};
/*
* initialization verbs
*/
static struct hda_verb alc260_init_verbs[] = {
/* Line In pin widget for input */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* CD pin widget for input */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
/* Mic2 (front panel) pin widget for input and vref at 80% */
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
/* LINE-2 is used for line-out in rear */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* select line-out */
{0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
/* LINE-OUT pin */
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* enable HP */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* enable Mono */
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* set connection select to line in (default select for this ADC) */
{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
/* mute capture amp left and right */
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* set connection select to line in (default select for this ADC) */
{0x05, AC_VERB_SET_CONNECT_SEL, 0x02},
/* set vol=0 Line-Out mixer amp left and right */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* unmute pin widget amp left and right (no gain on this amp) */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* set vol=0 HP mixer amp left and right */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* unmute pin widget amp left and right (no gain on this amp) */
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* set vol=0 Mono mixer amp left and right */
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* unmute pin widget amp left and right (no gain on this amp) */
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* unmute LINE-2 out pin */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */
/* mute CD */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
/* mute Line In */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
/* mute Mic */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
/* mute Front out path */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* mute Headphone out path */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* mute Mono out path */
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ }
};
static struct hda_verb alc260_hp_init_verbs[] = {
/* Headphone and output */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
/* mono output */
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* Mic2 (front panel) pin widget for input and vref at 80% */
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* Line In pin widget for input */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* Line-2 pin widget for output */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* CD pin widget for input */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* unmute amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* set connection select to line in (default select for this ADC) */
{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
/* unmute Line-Out mixer amp left and right (volume = 0) */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* unmute HP mixer amp left and right (volume = 0) */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */
/* unmute CD */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
/* unmute Line In */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
/* unmute Mic */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
/* Unmute Front out path */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Unmute Headphone out path */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Unmute Mono out path */
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{ }
};
static struct hda_verb alc260_hp_3013_init_verbs[] = {
/* Line out and output */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* mono output */
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* Mic2 (front panel) pin widget for input and vref at 80% */
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
/* Line In pin widget for input */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* Headphone pin widget for output */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
/* CD pin widget for input */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* unmute amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
/* set connection select to line in (default select for this ADC) */
{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
/* unmute Line-Out mixer amp left and right (volume = 0) */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* unmute HP mixer amp left and right (volume = 0) */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
/* mute pin widget amp left and right (no gain on this amp) */
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */
/* unmute CD */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
/* unmute Line In */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
/* unmute Mic */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
/* Unmute Front out path */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Unmute Headphone out path */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
/* Unmute Mono out path */
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{ }
};
/* Initialisation sequence for ALC260 as configured in Fujitsu S702x
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
* laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD
* audio = 0x16, internal speaker = 0x10.
*/
static struct hda_verb alc260_fujitsu_init_verbs[] = {
/* Disable all GPIOs */
{0x01, AC_VERB_SET_GPIO_MASK, 0},
/* Internal speaker is connected to headphone pin */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Headphone/Line-out jack connects to Line1 pin; make it an output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* Mic/Line-in jack is connected to mic1 pin, so make it an input */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Ensure all other unused pins are disabled and muted. */
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Disable digital (SPDIF) pins */
{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
/* Ensure Line1 pin widget takes its input from the OUT1 sum bus
* when acting as an output.
*/
{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
/* Start with output sum widgets muted and their output gains at min */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Line1 pin widget output buffer since it starts as an output.
* If the pin mode is changed by the user the pin mode control will
* take care of enabling the pin's input/output buffers as needed.
* Therefore there's no need to enable the input buffer at this
* stage.
*/
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute input buffer of pin widget used for Line-in (no equiv
* mixer ctrl)
*/
{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Set ADC connection select to match default mixer setting - line
* in (on mic1 pin)
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Do the same for the second ADC: mute capture input amp and
* set ADC connection to line in (on mic1 pin)
*/
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mute all inputs to mixer widget (even unconnected ones) */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
{ }
};
/* Initialisation sequence for ALC260 as configured in Acer TravelMate and
* similar laptops (adapted from Fujitsu init verbs).
*/
static struct hda_verb alc260_acer_init_verbs[] = {
/* On TravelMate laptops, GPIO 0 enables the internal speaker and
* the headphone jack. Turn this on and rely on the standard mute
* methods whenever the user wants to turn these outputs off.
*/
{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
/* Internal speaker/Headphone jack is connected to Line-out pin */
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Internal microphone/Mic jack is connected to Mic1 pin */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
/* Line In jack is connected to Line1 pin */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Ensure all other unused pins are disabled and muted. */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Disable digital (SPDIF) pins */
{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
/* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
* bus when acting as outputs.
*/
{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
/* Start with output sum widgets muted and their output gains at min */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Unmute Line-out pin widget amp left and right (no equiv mixer ctrl) */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mic1 and Line1 pin widget input buffers since they start as
* inputs. If the pin mode is changed by the user the pin mode control
* will take care of enabling the pin's input/output buffers as needed.
* Therefore there's no need to enable the input buffer at this
* stage.
*/
{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Set ADC connection select to match default mixer setting - mic
* (on mic1 pin)
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Do similar with the second ADC: mute capture input amp and
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
* set ADC connection to mic to match ALSA's default state.
*/
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mute all inputs to mixer widget (even unconnected ones) */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
{ }
};
/* Test configuration for debugging, modelled after the ALC880 test
* configuration.
*/
#ifdef CONFIG_SND_DEBUG
static hda_nid_t alc260_test_dac_nids[1] = {
0x02,
};
static hda_nid_t alc260_test_adc_nids[2] = {
0x04, 0x05,
};
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
/* For testing the ALC260, each input MUX needs its own definition since
* the signal assignments are different. This assumes that the first ADC
* is NID 0x04.
*/
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
static struct hda_input_mux alc260_test_capture_sources[2] = {
{
.num_items = 7,
.items = {
{ "MIC1 pin", 0x0 },
{ "MIC2 pin", 0x1 },
{ "LINE1 pin", 0x2 },
{ "LINE2 pin", 0x3 },
{ "CD pin", 0x4 },
{ "LINE-OUT pin", 0x5 },
{ "HP-OUT pin", 0x6 },
},
},
{
.num_items = 8,
.items = {
{ "MIC1 pin", 0x0 },
{ "MIC2 pin", 0x1 },
{ "LINE1 pin", 0x2 },
{ "LINE2 pin", 0x3 },
{ "CD pin", 0x4 },
{ "Mixer", 0x5 },
{ "LINE-OUT pin", 0x6 },
{ "HP-OUT pin", 0x7 },
},
},
};
static struct snd_kcontrol_new alc260_test_mixer[] = {
/* Output driver widgets */
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT),
HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT),
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
/* Modes for retasking pin widgets
* Note: the ALC260 doesn't seem to act on requests to enable mic
* bias from NIDs 0x0f and 0x10. The ALC260 datasheet doesn't
* mention this restriction. At this stage it's not clear whether
* this behaviour is intentional or is a hardware bug in chip
* revisions available at least up until early 2006. Therefore for
* now allow the "HP-OUT" and "LINE-OUT" Mode controls to span all
* choices, but if it turns out that the lack of mic bias for these
* NIDs is intentional we could change their modes from
* ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
*/
ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT),
/* Loopback mixer controls */
HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT),
HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT),
HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT),
HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT),
HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT),
HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT),
/* Controls for GPIO pins, assuming they are configured as outputs */
ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
/* Switches to allow the digital IO pins to be enabled. The datasheet
* is ambigious as to which NID is which; testing on laptops which
* make this output available should provide clarification.
*/
ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01),
ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01),
{ } /* end */
};
static struct hda_verb alc260_test_init_verbs[] = {
/* Enable all GPIOs as outputs with an initial value of 0 */
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f},
{0x01, AC_VERB_SET_GPIO_DATA, 0x00},
{0x01, AC_VERB_SET_GPIO_MASK, 0x0f},
/* Enable retasking pins as output, initially without power amp */
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* Disable digital (SPDIF) pins initially, but users can enable
* them via a mixer switch. In the case of SPDIF-out, this initverb
* payload also sets the generation to 0, output to be in "consumer"
* PCM format, copyright asserted, no pre-emphasis and no validity
* control.
*/
{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
/* Ensure mic1, mic2, line1 and line2 pin widgets take input from the
* OUT1 sum bus when acting as an output.
*/
{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
{0x0c, AC_VERB_SET_CONNECT_SEL, 0},
{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
{0x0e, AC_VERB_SET_CONNECT_SEL, 0},
/* Start with output sum widgets muted and their output gains at min */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Unmute retasking pin widget output buffers since the default
* state appears to be output. As the pin mode is changed by the
* user the pin mode control will take care of enabling the pin's
* input/output buffers as needed.
*/
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Also unmute the mono-out pin widget */
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mute capture amp left and right */
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* Set ADC connection select to match default mixer setting (mic1
* pin)
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Do the same for the second ADC: mute capture input amp and
* set ADC connection to mic1 pin
*/
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mute all inputs to mixer widget (even unconnected ones) */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
{ }
};
#endif
static struct hda_pcm_stream alc260_pcm_analog_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
};
static struct hda_pcm_stream alc260_pcm_analog_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
};
#define alc260_pcm_digital_playback alc880_pcm_digital_playback
#define alc260_pcm_digital_capture alc880_pcm_digital_capture
/*
* for BIOS auto-configuration
*/
static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
const char *pfx)
{
hda_nid_t nid_vol;
unsigned long vol_val, sw_val;
char name[32];
int err;
if (nid >= 0x0f && nid < 0x11) {
nid_vol = nid - 0x7;
vol_val = HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT);
sw_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
} else if (nid == 0x11) {
nid_vol = nid - 0x7;
vol_val = HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT);
sw_val = HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT);
} else if (nid >= 0x12 && nid <= 0x15) {
nid_vol = 0x08;
vol_val = HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT);
sw_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
} else
return 0; /* N/A */
snprintf(name, sizeof(name), "%s Playback Volume", pfx);
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val)) < 0)
return err;
snprintf(name, sizeof(name), "%s Playback Switch", pfx);
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, sw_val)) < 0)
return err;
return 1;
}
/* add playback controls from the parsed DAC table */
static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
{
hda_nid_t nid;
int err;
spec->multiout.num_dacs = 1;
spec->multiout.dac_nids = spec->private_dac_nids;
spec->multiout.dac_nids[0] = 0x02;
nid = cfg->line_out_pins[0];
if (nid) {
err = alc260_add_playback_controls(spec, nid, "Front");
if (err < 0)
return err;
}
nid = cfg->speaker_pins[0];
if (nid) {
err = alc260_add_playback_controls(spec, nid, "Speaker");
if (err < 0)
return err;
}
nid = cfg->hp_pin;
if (nid) {
err = alc260_add_playback_controls(spec, nid, "Headphone");
if (err < 0)
return err;
}
return 0;
}
/* create playback/capture controls for input pins */
static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
{
struct hda_input_mux *imux = &spec->private_imux;
int i, err, idx;
for (i = 0; i < AUTO_PIN_LAST; i++) {
if (cfg->input_pins[i] >= 0x12) {
idx = cfg->input_pins[i] - 0x12;
err = new_analog_input(spec, cfg->input_pins[i],
auto_pin_cfg_labels[i], idx, 0x07);
if (err < 0)
return err;
imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
imux->items[imux->num_items].index = idx;
imux->num_items++;
}
if ((cfg->input_pins[i] >= 0x0f) && (cfg->input_pins[i] <= 0x10)){
idx = cfg->input_pins[i] - 0x09;
err = new_analog_input(spec, cfg->input_pins[i],
auto_pin_cfg_labels[i], idx, 0x07);
if (err < 0)
return err;
imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
imux->items[imux->num_items].index = idx;
imux->num_items++;
}
}
return 0;
}
static void alc260_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
int sel_idx)
{
/* set as output */
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
/* need the manual connection? */
if (nid >= 0x12) {
int idx = nid - 0x12;
snd_hda_codec_write(codec, idx + 0x0b, 0,
AC_VERB_SET_CONNECT_SEL, sel_idx);
}
}
static void alc260_auto_init_multi_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t nid;
nid = spec->autocfg.line_out_pins[0];
if (nid)
alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0);
nid = spec->autocfg.speaker_pins[0];
if (nid)
alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0);
nid = spec->autocfg.hp_pin;
if (nid)
alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0);
}
#define ALC260_PIN_CD_NID 0x16
static void alc260_auto_init_analog_input(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = spec->autocfg.input_pins[i];
if (nid >= 0x12) {
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
i <= AUTO_PIN_FRONT_MIC ? PIN_VREF80 : PIN_IN);
if (nid != ALC260_PIN_CD_NID)
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_MUTE);
}
}
}
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc260_volume_init_verbs[] = {
/*
* Unmute ADC0-1 and set the default input to mic-in
*/
{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for front panel
* mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
/*
* Set up output mixers (0x08 - 0x0a)
*/
/* set vol=0 to output mixers */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{ }
};
static int alc260_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int wcap;
int err;
static hda_nid_t alc260_ignore[] = { 0x17, 0 };
if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc260_ignore)) < 0)
return err;
if ((err = alc260_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0)
return err;
if (! spec->kctl_alloc)
return 0; /* can't find valid BIOS pin config */
if ((err = alc260_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
return err;
spec->multiout.max_channels = 2;
if (spec->autocfg.dig_out_pin)
spec->multiout.dig_out_nid = ALC260_DIGOUT_NID;
if (spec->kctl_alloc)
spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
spec->init_verbs[spec->num_init_verbs++] = alc260_volume_init_verbs;
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux;
/* check whether NID 0x04 is valid */
wcap = get_wcaps(codec, 0x04);
wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */
if (wcap != AC_WID_AUD_IN) {
spec->adc_nids = alc260_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt);
spec->mixers[spec->num_mixers] = alc260_capture_alt_mixer;
} else {
spec->adc_nids = alc260_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids);
spec->mixers[spec->num_mixers] = alc260_capture_mixer;
}
spec->num_mixers++;
return 1;
}
/* additional initialization for auto-configuration model */
static void alc260_auto_init(struct hda_codec *codec)
{
alc260_auto_init_multi_out(codec);
alc260_auto_init_analog_input(codec);
}
/*
* ALC260 configurations
*/
static struct hda_board_config alc260_cfg_tbl[] = {
{ .modelname = "basic", .config = ALC260_BASIC },
{ .pci_subvendor = 0x104d, .pci_subdevice = 0x81bb,
.config = ALC260_BASIC }, /* Sony VAIO */
{ .pci_subvendor = 0x104d, .pci_subdevice = 0x81cd,
.config = ALC260_BASIC }, /* Sony VAIO */
{ .pci_subvendor = 0x152d, .pci_subdevice = 0x0729,
.config = ALC260_BASIC }, /* CTL Travel Master U553W */
{ .modelname = "hp", .config = ALC260_HP },
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP },
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3011, .config = ALC260_HP },
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3012, .config = ALC260_HP },
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3013, .config = ALC260_HP_3013 },
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3014, .config = ALC260_HP },
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3015, .config = ALC260_HP },
{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3016, .config = ALC260_HP },
{ .modelname = "fujitsu", .config = ALC260_FUJITSU_S702X },
{ .pci_subvendor = 0x10cf, .pci_subdevice = 0x1326, .config = ALC260_FUJITSU_S702X },
{ .modelname = "acer", .config = ALC260_ACER },
{ .pci_subvendor = 0x1025, .pci_subdevice = 0x008f, .config = ALC260_ACER },
#ifdef CONFIG_SND_DEBUG
{ .modelname = "test", .config = ALC260_TEST },
#endif
{ .modelname = "auto", .config = ALC260_AUTO },
{}
};
static struct alc_config_preset alc260_presets[] = {
[ALC260_BASIC] = {
.mixers = { alc260_base_output_mixer,
alc260_input_mixer,
alc260_pc_beep_mixer,
alc260_capture_mixer },
.init_verbs = { alc260_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
.adc_nids = alc260_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
},
[ALC260_HP] = {
.mixers = { alc260_base_output_mixer,
alc260_input_mixer,
alc260_capture_alt_mixer },
.init_verbs = { alc260_hp_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids),
.adc_nids = alc260_hp_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
},
[ALC260_HP_3013] = {
.mixers = { alc260_hp_3013_mixer,
alc260_input_mixer,
alc260_capture_alt_mixer },
.init_verbs = { alc260_hp_3013_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids),
.adc_nids = alc260_hp_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
},
[ALC260_FUJITSU_S702X] = {
.mixers = { alc260_fujitsu_mixer,
alc260_capture_mixer },
.init_verbs = { alc260_fujitsu_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
.adc_nids = alc260_dual_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
.num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources),
.input_mux = alc260_fujitsu_capture_sources,
},
[ALC260_ACER] = {
.mixers = { alc260_acer_mixer,
alc260_capture_mixer },
.init_verbs = { alc260_acer_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
.adc_nids = alc260_dual_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
.num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources),
.input_mux = alc260_acer_capture_sources,
},
#ifdef CONFIG_SND_DEBUG
[ALC260_TEST] = {
.mixers = { alc260_test_mixer,
alc260_capture_mixer },
.init_verbs = { alc260_test_init_verbs },
.num_dacs = ARRAY_SIZE(alc260_test_dac_nids),
.dac_nids = alc260_test_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids),
.adc_nids = alc260_test_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
.num_mux_defs = ARRAY_SIZE(alc260_test_capture_sources),
.input_mux = alc260_test_capture_sources,
},
#endif
};
static int patch_alc260(struct hda_codec *codec)
{
struct alc_spec *spec;
int err, board_config;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
board_config = snd_hda_check_board_config(codec, alc260_cfg_tbl);
if (board_config < 0 || board_config >= ALC260_MODEL_LAST) {
snd_printd(KERN_INFO "hda_codec: Unknown model for ALC260\n");
board_config = ALC260_AUTO;
}
if (board_config == ALC260_AUTO) {
/* automatic parse from the BIOS config */
err = alc260_parse_auto_config(codec);
if (err < 0) {
alc_free(codec);
return err;
} else if (! err) {
printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using base mode...\n");
board_config = ALC260_BASIC;
}
}
if (board_config != ALC260_AUTO)
setup_preset(spec, &alc260_presets[board_config]);
spec->stream_name_analog = "ALC260 Analog";
spec->stream_analog_playback = &alc260_pcm_analog_playback;
spec->stream_analog_capture = &alc260_pcm_analog_capture;
spec->stream_name_digital = "ALC260 Digital";
spec->stream_digital_playback = &alc260_pcm_digital_playback;
spec->stream_digital_capture = &alc260_pcm_digital_capture;
codec->patch_ops = alc_patch_ops;
if (board_config == ALC260_AUTO)
spec->init_hook = alc260_auto_init;
return 0;
}
/*
* ALC882 support
*
* ALC882 is almost identical with ALC880 but has cleaner and more flexible
* configuration. Each pin widget can choose any input DACs and a mixer.
* Each ADC is connected from a mixer of all inputs. This makes possible
* 6-channel independent captures.
*
* In addition, an independent DAC for the multi-playback (not used in this
* driver yet).
*/
#define ALC882_DIGOUT_NID 0x06
#define ALC882_DIGIN_NID 0x0a
static struct hda_channel_mode alc882_ch_modes[1] = {
{ 8, NULL }
};
static hda_nid_t alc882_dac_nids[4] = {
/* front, rear, clfe, rear_surr */
0x02, 0x03, 0x04, 0x05
};
/* identical with ALC880 */
#define alc882_adc_nids alc880_adc_nids
#define alc882_adc_nids_alt alc880_adc_nids_alt
/* input MUX */
/* FIXME: should be a matrix-type input source selection */
static struct hda_input_mux alc882_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
#define alc882_mux_enum_info alc_mux_enum_info
#define alc882_mux_enum_get alc_mux_enum_get
static int alc882_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
const struct hda_input_mux *imux = spec->input_mux;
unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 };
hda_nid_t nid = capture_mixers[adc_idx];
unsigned int *cur_val = &spec->cur_mux[adc_idx];
unsigned int i, idx;
idx = ucontrol->value.enumerated.item[0];
if (idx >= imux->num_items)
idx = imux->num_items - 1;
if (*cur_val == idx && ! codec->in_resume)
return 0;
for (i = 0; i < imux->num_items; i++) {
unsigned int v = (i == idx) ? 0x7000 : 0x7080;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
v | (imux->items[i].index << 8));
}
*cur_val = idx;
return 1;
}
/*
* 6ch mode
*/
static struct hda_verb alc882_sixstack_ch6_init[] = {
{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ } /* end */
};
/*
* 8ch mode
*/
static struct hda_verb alc882_sixstack_ch8_init[] = {
{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ } /* end */
};
static struct hda_channel_mode alc882_sixstack_modes[2] = {
{ 6, alc882_sixstack_ch6_init },
{ 8, alc882_sixstack_ch8_init },
};
/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
* Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
*/
static struct snd_kcontrol_new alc882_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
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 snd_kcontrol_new alc882_chmode_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
static struct hda_verb alc882_init_verbs[] = {
/* Front mixer: unmute input/output amp left and right (volume = 0) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Rear mixer */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* CLFE mixer */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Side mixer */
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Front Pin: output 0 (0x0c) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Rear Pin: output 1 (0x0d) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
/* CLFE Pin: output 2 (0x0e) */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
/* Side Pin: output 3 (0x0f) */
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
/* Mic (rear) pin: input vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Front Mic pin: input vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line In pin: input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line-2 In: Headphone output (output 0 - 0x0c) */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* ADC1: mute amp left and right */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
/* ADC2: mute amp left and right */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
/* ADC3: mute amp left and right */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{ }
};
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc882_auto_init_verbs[] = {
/*
* Unmute ADC0-2 and set the default input to mic-in
*/
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for front panel
* mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
/*
* Set up output mixers (0x0c - 0x0f)
*/
/* set vol=0 to output mixers */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
{ }
};
/* capture mixer elements */
static struct snd_kcontrol_new alc882_capture_alt_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
* FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.count = 2,
.info = alc882_mux_enum_info,
.get = alc882_mux_enum_get,
.put = alc882_mux_enum_put,
},
{ } /* end */
};
static struct snd_kcontrol_new alc882_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
* FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.count = 3,
.info = alc882_mux_enum_info,
.get = alc882_mux_enum_get,
.put = alc882_mux_enum_put,
},
{ } /* end */
};
/* pcm configuration: identiacal with ALC880 */
#define alc882_pcm_analog_playback alc880_pcm_analog_playback
#define alc882_pcm_analog_capture alc880_pcm_analog_capture
#define alc882_pcm_digital_playback alc880_pcm_digital_playback
#define alc882_pcm_digital_capture alc880_pcm_digital_capture
/*
* configuration and preset
*/
static struct hda_board_config alc882_cfg_tbl[] = {
{ .modelname = "3stack-dig", .config = ALC882_3ST_DIG },
{ .modelname = "6stack-dig", .config = ALC882_6ST_DIG },
{ .pci_subvendor = 0x1462, .pci_subdevice = 0x6668, .config = ALC882_6ST_DIG }, /* MSI */
{ .pci_subvendor = 0x105b, .pci_subdevice = 0x6668, .config = ALC882_6ST_DIG }, /* Foxconn */
{ .pci_subvendor = 0x1019, .pci_subdevice = 0x6668, .config = ALC882_6ST_DIG }, /* ECS */
{ .modelname = "auto", .config = ALC882_AUTO },
{}
};
static struct alc_config_preset alc882_presets[] = {
[ALC882_3ST_DIG] = {
.mixers = { alc882_base_mixer },
.init_verbs = { alc882_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.dig_out_nid = ALC882_DIGOUT_NID,
.num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
.adc_nids = alc882_adc_nids,
.dig_in_nid = ALC882_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
.channel_mode = alc882_ch_modes,
.input_mux = &alc882_capture_source,
},
[ALC882_6ST_DIG] = {
.mixers = { alc882_base_mixer, alc882_chmode_mixer },
.init_verbs = { alc882_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.dig_out_nid = ALC882_DIGOUT_NID,
.num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
.adc_nids = alc882_adc_nids,
.dig_in_nid = ALC882_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
.channel_mode = alc882_sixstack_modes,
.input_mux = &alc882_capture_source,
},
};
/*
* BIOS auto configuration
*/
static void alc882_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
int dac_idx)
{
/* set as output */
struct alc_spec *spec = codec->spec;
int idx;
if (spec->multiout.dac_nids[dac_idx] == 0x25)
idx = 4;
else
idx = spec->multiout.dac_nids[dac_idx] - 2;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
}
static void alc882_auto_init_multi_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i <= HDA_SIDE; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
if (nid)
alc882_auto_set_output_and_unmute(codec, nid, PIN_OUT, i);
}
}
static void alc882_auto_init_hp_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t pin;
pin = spec->autocfg.hp_pin;
if (pin) /* connect to front */
alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); /* use dac 0 */
}
#define alc882_is_input_pin(nid) alc880_is_input_pin(nid)
#define ALC882_PIN_CD_NID ALC880_PIN_CD_NID
static void alc882_auto_init_analog_input(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = spec->autocfg.input_pins[i];
if (alc882_is_input_pin(nid)) {
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
i <= AUTO_PIN_FRONT_MIC ? PIN_VREF80 : PIN_IN);
if (nid != ALC882_PIN_CD_NID)
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_MUTE);
}
}
}
/* almost identical with ALC880 parser... */
static int alc882_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err = alc880_parse_auto_config(codec);
if (err < 0)
return err;
else if (err > 0)
/* hack - override the init verbs */
spec->init_verbs[0] = alc882_auto_init_verbs;
return err;
}
/* additional initialization for auto-configuration model */
static void alc882_auto_init(struct hda_codec *codec)
{
alc882_auto_init_multi_out(codec);
alc882_auto_init_hp_out(codec);
alc882_auto_init_analog_input(codec);
}
/*
* ALC882 Headphone poll in 3.5.1a or 3.5.2
*/
static int patch_alc882(struct hda_codec *codec)
{
struct alc_spec *spec;
int err, board_config;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
board_config = snd_hda_check_board_config(codec, alc882_cfg_tbl);
if (board_config < 0 || board_config >= ALC882_MODEL_LAST) {
printk(KERN_INFO "hda_codec: Unknown model for ALC882, trying auto-probe from BIOS...\n");
board_config = ALC882_AUTO;
}
if (board_config == ALC882_AUTO) {
/* automatic parse from the BIOS config */
err = alc882_parse_auto_config(codec);
if (err < 0) {
alc_free(codec);
return err;
} else if (! err) {
printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using base mode...\n");
board_config = ALC882_3ST_DIG;
}
}
if (board_config != ALC882_AUTO)
setup_preset(spec, &alc882_presets[board_config]);
spec->stream_name_analog = "ALC882 Analog";
spec->stream_analog_playback = &alc882_pcm_analog_playback;
spec->stream_analog_capture = &alc882_pcm_analog_capture;
spec->stream_name_digital = "ALC882 Digital";
spec->stream_digital_playback = &alc882_pcm_digital_playback;
spec->stream_digital_capture = &alc882_pcm_digital_capture;
if (! spec->adc_nids && spec->input_mux) {
/* check whether NID 0x07 is valid */
unsigned int wcap = get_wcaps(codec, 0x07);
wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */
if (wcap != AC_WID_AUD_IN) {
spec->adc_nids = alc882_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids_alt);
spec->mixers[spec->num_mixers] = alc882_capture_alt_mixer;
spec->num_mixers++;
} else {
spec->adc_nids = alc882_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids);
spec->mixers[spec->num_mixers] = alc882_capture_mixer;
spec->num_mixers++;
}
}
codec->patch_ops = alc_patch_ops;
if (board_config == ALC882_AUTO)
spec->init_hook = alc882_auto_init;
return 0;
}
/*
* ALC262 support
*/
#define ALC262_DIGOUT_NID ALC880_DIGOUT_NID
#define ALC262_DIGIN_NID ALC880_DIGIN_NID
#define alc262_dac_nids alc260_dac_nids
#define alc262_adc_nids alc882_adc_nids
#define alc262_adc_nids_alt alc882_adc_nids_alt
#define alc262_modes alc260_modes
#define alc262_capture_source alc882_capture_source
static struct snd_kcontrol_new alc262_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
/* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("PC Beelp Playback Switch", 0x0b, 0x05, HDA_INPUT), */
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
{ } /* end */
};
#define alc262_capture_mixer alc882_capture_mixer
#define alc262_capture_alt_mixer alc882_capture_alt_mixer
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc262_init_verbs[] = {
/*
* Unmute ADC0-2 and set the default input to mic-in
*/
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for front panel
* mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
/*
* Set up output mixers (0x0c - 0x0e)
*/
/* set vol=0 to output mixers */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
{ }
};
/*
* fujitsu model
* 0x14 = headphone/spdif-out, 0x15 = internal speaker
*/
#define ALC_HP_EVENT 0x37
static struct hda_verb alc262_fujitsu_unsol_verbs[] = {
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{}
};
static struct hda_input_mux alc262_fujitsu_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x0 },
{ "CD", 0x4 },
},
};
/* mute/unmute internal speaker according to the hp jack and mute state */
static void alc262_fujitsu_automute(struct hda_codec *codec, int force)
{
struct alc_spec *spec = codec->spec;
unsigned int mute;
if (force || ! spec->sense_updated) {
unsigned int present;
/* need to execute and sync at first */
snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0);
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0);
spec->jack_present = (present & 0x80000000) != 0;
spec->sense_updated = 1;
}
if (spec->jack_present) {
/* mute internal speaker */
snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
0x80, 0x80);
snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
0x80, 0x80);
} else {
/* unmute internal speaker if necessary */
mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0);
snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
0x80, mute & 0x80);
mute = snd_hda_codec_amp_read(codec, 0x14, 1, HDA_OUTPUT, 0);
snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
0x80, mute & 0x80);
}
}
/* unsolicited event for HP jack sensing */
static void alc262_fujitsu_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) != ALC_HP_EVENT)
return;
alc262_fujitsu_automute(codec, 1);
}
/* bind volumes of both NID 0x0c and 0x0d */
static int alc262_fujitsu_master_vol_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
long *valp = ucontrol->value.integer.value;
int change;
change = snd_hda_codec_amp_update(codec, 0x0c, 0, HDA_OUTPUT, 0,
0x7f, valp[0] & 0x7f);
change |= snd_hda_codec_amp_update(codec, 0x0c, 1, HDA_OUTPUT, 0,
0x7f, valp[1] & 0x7f);
snd_hda_codec_amp_update(codec, 0x0d, 0, HDA_OUTPUT, 0,
0x7f, valp[0] & 0x7f);
snd_hda_codec_amp_update(codec, 0x0d, 1, HDA_OUTPUT, 0,
0x7f, valp[1] & 0x7f);
return change;
}
/* bind hp and internal speaker mute (with plug check) */
static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
long *valp = ucontrol->value.integer.value;
int change;
change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
0x80, valp[0] ? 0 : 0x80);
change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
0x80, valp[1] ? 0 : 0x80);
if (change || codec->in_resume)
alc262_fujitsu_automute(codec, codec->in_resume);
return change;
}
static struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Volume",
.info = snd_hda_mixer_amp_volume_info,
.get = snd_hda_mixer_amp_volume_get,
.put = alc262_fujitsu_master_vol_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT),
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.info = snd_hda_mixer_amp_switch_info,
.get = snd_hda_mixer_amp_switch_get,
.put = alc262_fujitsu_master_sw_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
},
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
/* add playback controls from the parsed DAC table */
static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg)
{
hda_nid_t nid;
int err;
spec->multiout.num_dacs = 1; /* only use one dac */
spec->multiout.dac_nids = spec->private_dac_nids;
spec->multiout.dac_nids[0] = 2;
nid = cfg->line_out_pins[0];
if (nid) {
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Front Playback Volume",
HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT))) < 0)
return err;
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Front Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
return err;
}
nid = cfg->speaker_pins[0];
if (nid) {
if (nid == 0x16) {
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Speaker Playback Volume",
HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_OUTPUT))) < 0)
return err;
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Speaker Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0)
return err;
} else {
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Speaker Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
return err;
}
}
nid = cfg->hp_pin;
if (nid) {
/* spec->multiout.hp_nid = 2; */
if (nid == 0x16) {
if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Headphone Playback Volume",
HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_OUTPUT))) < 0)
return err;
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Headphone Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0)
return err;
} else {
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Headphone Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
return err;
}
}
return 0;
}
/* identical with ALC880 */
#define alc262_auto_create_analog_input_ctls alc880_auto_create_analog_input_ctls
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc262_volume_init_verbs[] = {
/*
* Unmute ADC0-2 and set the default input to mic-in
*/
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for front panel
* mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
/*
* Set up output mixers (0x0c - 0x0f)
*/
/* set vol=0 to output mixers */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
{ }
};
/* pcm configuration: identiacal with ALC880 */
#define alc262_pcm_analog_playback alc880_pcm_analog_playback
#define alc262_pcm_analog_capture alc880_pcm_analog_capture
#define alc262_pcm_digital_playback alc880_pcm_digital_playback
#define alc262_pcm_digital_capture alc880_pcm_digital_capture
/*
* BIOS auto configuration
*/
static int alc262_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err;
static hda_nid_t alc262_ignore[] = { 0x1d, 0 };
if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc262_ignore)) < 0)
return err;
if (! spec->autocfg.line_outs)
return 0; /* can't find valid BIOS pin config */
if ((err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 ||
(err = alc262_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
if (spec->autocfg.dig_out_pin)
spec->multiout.dig_out_nid = ALC262_DIGOUT_NID;
if (spec->autocfg.dig_in_pin)
spec->dig_in_nid = ALC262_DIGIN_NID;
if (spec->kctl_alloc)
spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
spec->init_verbs[spec->num_init_verbs++] = alc262_volume_init_verbs;
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux;
return 1;
}
#define alc262_auto_init_multi_out alc882_auto_init_multi_out
#define alc262_auto_init_hp_out alc882_auto_init_hp_out
#define alc262_auto_init_analog_input alc882_auto_init_analog_input
/* init callback for auto-configuration model -- overriding the default init */
static void alc262_auto_init(struct hda_codec *codec)
{
alc262_auto_init_multi_out(codec);
alc262_auto_init_hp_out(codec);
alc262_auto_init_analog_input(codec);
}
/*
* configuration and preset
*/
static struct hda_board_config alc262_cfg_tbl[] = {
{ .modelname = "basic", .config = ALC262_BASIC },
{ .modelname = "fujitsu", .config = ALC262_FUJITSU },
{ .pci_subvendor = 0x10cf, .pci_subdevice = 0x1397, .config = ALC262_FUJITSU },
{ .modelname = "auto", .config = ALC262_AUTO },
{}
};
static struct alc_config_preset alc262_presets[] = {
[ALC262_BASIC] = {
.mixers = { alc262_base_mixer },
.init_verbs = { alc262_init_verbs },
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
},
[ALC262_FUJITSU] = {
.mixers = { alc262_fujitsu_mixer },
.init_verbs = { alc262_init_verbs, alc262_fujitsu_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x03,
.dig_out_nid = ALC262_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_fujitsu_capture_source,
.unsol_event = alc262_fujitsu_unsol_event,
},
};
static int patch_alc262(struct hda_codec *codec)
{
struct alc_spec *spec;
int board_config;
int err;
spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
#if 0
/* pshou 07/11/05 set a zero PCM sample to DAC when FIFO is under-run */
{
int tmp;
snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_COEF_INDEX, 7);
tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0);
snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_COEF_INDEX, 7);
snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PROC_COEF, tmp | 0x80);
}
#endif
board_config = snd_hda_check_board_config(codec, alc262_cfg_tbl);
if (board_config < 0 || board_config >= ALC262_MODEL_LAST) {
printk(KERN_INFO "hda_codec: Unknown model for ALC262, trying auto-probe from BIOS...\n");
board_config = ALC262_AUTO;
}
if (board_config == ALC262_AUTO) {
/* automatic parse from the BIOS config */
err = alc262_parse_auto_config(codec);
if (err < 0) {
alc_free(codec);
return err;
} else if (! err) {
printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using base mode...\n");
board_config = ALC262_BASIC;
}
}
if (board_config != ALC262_AUTO)
setup_preset(spec, &alc262_presets[board_config]);
spec->stream_name_analog = "ALC262 Analog";
spec->stream_analog_playback = &alc262_pcm_analog_playback;
spec->stream_analog_capture = &alc262_pcm_analog_capture;
spec->stream_name_digital = "ALC262 Digital";
spec->stream_digital_playback = &alc262_pcm_digital_playback;
spec->stream_digital_capture = &alc262_pcm_digital_capture;
if (! spec->adc_nids && spec->input_mux) {
/* check whether NID 0x07 is valid */
unsigned int wcap = get_wcaps(codec, 0x07);
wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */
if (wcap != AC_WID_AUD_IN) {
spec->adc_nids = alc262_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids_alt);
spec->mixers[spec->num_mixers] = alc262_capture_alt_mixer;
spec->num_mixers++;
} else {
spec->adc_nids = alc262_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids);
spec->mixers[spec->num_mixers] = alc262_capture_mixer;
spec->num_mixers++;
}
}
codec->patch_ops = alc_patch_ops;
if (board_config == ALC262_AUTO)
spec->init_hook = alc262_auto_init;
return 0;
}
/*
* ALC861 channel source setting (2/6 channel selection for 3-stack)
*/
/*
* set the path ways for 2 channel output
* need to set the codec line out and mic 1 pin widgets to inputs
*/
static struct hda_verb alc861_threestack_ch2_init[] = {
/* set pin widget 1Ah (line in) for input */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* set pin widget 18h (mic1/2) for input, for mic also enable the vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, //mic
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, //line in
{ } /* end */
};
/*
* 6ch mode
* need to set the codec line out and mic 1 pin widgets to outputs
*/
static struct hda_verb alc861_threestack_ch6_init[] = {
/* set pin widget 1Ah (line in) for output (Back Surround)*/
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* set pin widget 18h (mic1) for output (CLFE)*/
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, //mic
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, //line in
{ } /* end */
};
static struct hda_channel_mode alc861_threestack_modes[2] = {
{ 2, alc861_threestack_ch2_init },
{ 6, alc861_threestack_ch6_init },
};
/* patch-ALC861 */
static struct snd_kcontrol_new alc861_base_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
/*Input mixer control */
/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
/* Capture mixer control */
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Capture Source",
.count = 1,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{ } /* end */
};
static struct snd_kcontrol_new alc861_3ST_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
/*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
/* Input mixer control */
/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
/* Capture mixer control */
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Capture Source",
.count = 1,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
.private_value = ARRAY_SIZE(alc861_threestack_modes),
},
{ } /* end */
};
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc861_base_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
/* port-A for surround (rear panel) */
{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0e, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-B for mic-in (rear panel) with vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-C for line-in (rear panel) */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* port-D for Front */
{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-E for HP out (front panel) */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x01 },
/* port-F for mic-in (front panel) with vref */
{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-G for CLFE (rear panel) */
{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x1f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-H for side (rear panel) */
{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* CD-in */
{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* route front mic to ADC1*/
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute DAC0~3 & spdif out*/
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mixer 14 (mic) 1c (Line in)*/
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Unmute Stereo Mixer 15 */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c }, //Output 0~12 step
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, // hp used DAC 3 (Front)
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
static struct hda_verb alc861_threestack_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
/* port-A for surround (rear panel) */
{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* port-B for mic-in (rear panel) with vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-C for line-in (rear panel) */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* port-D for Front */
{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-E for HP out (front panel) */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x01 },
/* port-F for mic-in (front panel) with vref */
{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-G for CLFE (rear panel) */
{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* port-H for side (rear panel) */
{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* CD-in */
{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* route front mic to ADC1*/
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute DAC0~3 & spdif out*/
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mixer 14 (mic) 1c (Line in)*/
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Unmute Stereo Mixer 15 */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c }, //Output 0~12 step
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, // hp used DAC 3 (Front)
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc861_auto_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
// {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute DAC0~3 & spdif out*/
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mixer 14 (mic) 1c (Line in)*/
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Unmute Stereo Mixer 15 */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, // set Mic 1
{ }
};
/* pcm configuration: identiacal with ALC880 */
#define alc861_pcm_analog_playback alc880_pcm_analog_playback
#define alc861_pcm_analog_capture alc880_pcm_analog_capture
#define alc861_pcm_digital_playback alc880_pcm_digital_playback
#define alc861_pcm_digital_capture alc880_pcm_digital_capture
#define ALC861_DIGOUT_NID 0x07
static struct hda_channel_mode alc861_8ch_modes[1] = {
{ 8, NULL }
};
static hda_nid_t alc861_dac_nids[4] = {
/* front, surround, clfe, side */
0x03, 0x06, 0x05, 0x04
};
static hda_nid_t alc861_adc_nids[1] = {
/* ADC0-2 */
0x08,
};
static struct hda_input_mux alc861_capture_source = {
.num_items = 5,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x3 },
{ "Line", 0x1 },
{ "CD", 0x4 },
{ "Mixer", 0x5 },
},
};
/* fill in the dac_nids table from the parsed pin configuration */
static int alc861_auto_fill_dac_nids(struct alc_spec *spec, const struct auto_pin_cfg *cfg)
{
int i;
hda_nid_t nid;
spec->multiout.dac_nids = spec->private_dac_nids;
for (i = 0; i < cfg->line_outs; i++) {
nid = cfg->line_out_pins[i];
if (nid) {
if (i >= ARRAY_SIZE(alc861_dac_nids))
continue;
spec->multiout.dac_nids[i] = alc861_dac_nids[i];
}
}
spec->multiout.num_dacs = cfg->line_outs;
return 0;
}
/* add playback controls from the parsed DAC table */
static int alc861_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
{
char name[32];
static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" };
hda_nid_t nid;
int i, idx, err;
for (i = 0; i < cfg->line_outs; i++) {
nid = spec->multiout.dac_nids[i];
if (! nid)
continue;
if (nid == 0x05) {
/* Center/LFE */
if ((err = add_control(spec, ALC_CTL_BIND_MUTE, "Center Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0)
return err;
if ((err = add_control(spec, ALC_CTL_BIND_MUTE, "LFE Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0)
return err;
} else {
for (idx = 0; idx < ARRAY_SIZE(alc861_dac_nids) - 1; idx++)
if (nid == alc861_dac_nids[idx])
break;
sprintf(name, "%s Playback Switch", chname[idx]);
if ((err = add_control(spec, ALC_CTL_BIND_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
return err;
}
}
return 0;
}
static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin)
{
int err;
hda_nid_t nid;
if (! pin)
return 0;
if ((pin >= 0x0b && pin <= 0x10) || pin == 0x1f || pin == 0x20) {
nid = 0x03;
if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Headphone Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
return err;
spec->multiout.hp_nid = nid;
}
return 0;
}
/* create playback/capture controls for input pins */
static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg)
{
struct hda_input_mux *imux = &spec->private_imux;
int i, err, idx, idx1;
for (i = 0; i < AUTO_PIN_LAST; i++) {
switch(cfg->input_pins[i]) {
case 0x0c:
idx1 = 1;
idx = 2; // Line In
break;
case 0x0f:
idx1 = 2;
idx = 2; // Line In
break;
case 0x0d:
idx1 = 0;
idx = 1; // Mic In
break;
case 0x10:
idx1 = 3;
idx = 1; // Mic In
break;
case 0x11:
idx1 = 4;
idx = 0; // CD
break;
default:
continue;
}
err = new_analog_input(spec, cfg->input_pins[i],
auto_pin_cfg_labels[i], idx, 0x15);
if (err < 0)
return err;
imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
imux->items[imux->num_items].index = idx1;
imux->num_items++;
}
return 0;
}
static struct snd_kcontrol_new alc861_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
*FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.count = 1,
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc_mux_enum_put,
},
{ } /* end */
};
static void alc861_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid,
int pin_type, int dac_idx)
{
/* set as output */
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
snd_hda_codec_write(codec, dac_idx, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
}
static void alc861_auto_init_multi_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < spec->autocfg.line_outs; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
if (nid)
alc861_auto_set_output_and_unmute(codec, nid, PIN_OUT, spec->multiout.dac_nids[i]);
}
}
static void alc861_auto_init_hp_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t pin;
pin = spec->autocfg.hp_pin;
if (pin) /* connect to front */
alc861_auto_set_output_and_unmute(codec, pin, PIN_HP, spec->multiout.dac_nids[0]);
}
static void alc861_auto_init_analog_input(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = spec->autocfg.input_pins[i];
if ((nid>=0x0c) && (nid <=0x11)) {
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
i <= AUTO_PIN_FRONT_MIC ? PIN_VREF80 : PIN_IN);
}
}
}
/* parse the BIOS configuration and set up the alc_spec */
/* return 1 if successful, 0 if the proper config is not found, or a negative error code */
static int alc861_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err;
static hda_nid_t alc861_ignore[] = { 0x1d, 0 };
if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc861_ignore)) < 0)
return err;
if (! spec->autocfg.line_outs)
return 0; /* can't find valid BIOS pin config */
if ((err = alc861_auto_fill_dac_nids(spec, &spec->autocfg)) < 0 ||
(err = alc861_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 ||
(err = alc861_auto_create_hp_ctls(spec, spec->autocfg.hp_pin)) < 0 ||
(err = alc861_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
if (spec->autocfg.dig_out_pin)
spec->multiout.dig_out_nid = ALC861_DIGOUT_NID;
if (spec->kctl_alloc)
spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
spec->init_verbs[spec->num_init_verbs++] = alc861_auto_init_verbs;
[ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions The following patch relative to CVS from 20060324 adds the following features to the Realtek HDA codec. 1) Define two new pin modes: ALC_PIN_DIR_IN_NOMICBIAS and ALC_PIN_DIR_INOUT_NOMICBIAS. These can be used with jack mode switch definitions in mixers to prevent the user being offered the mic bias options if the hardware doesn't support it. 2) Add the ability to have different input mux definitions for different ADCs. This is needed because the ALC260 chip uses different mux layouts for the two onboard ADCs. A new field (num_mux_defs) was added to the alc_spec and alc_config_preset structures to support this. 3) Adjust numerous comments to make them consistent with the above changes. 4) Utilise the new multi-mux definition functionality for the ALC260 fujitsu model to allow recording of the mixer output. 5) Utilise the new multi-mux definition functionality for the ALC260 test model to make the mux selections a little less confusing. 6) Allow the headphone jack of the ALC260 acer model to be retasked in the mixer. 6) Utilise the new multi-mux definition functionality for the ALC260 acer model to give access to the mixer output and the retasked headphone jack. At this stage the *_NOMICBIAS modes are not used. We have reports that the "Line" jack of at least some Acer models doesn't pass the bias out, and we also know that NIDs 0x0f and 0x10 don't seem to accept the mic bias requests at all. However, I feel we need to collect more evidence on both counts before committing to the use of *_NOMICBIAS. In the case of the Acers, it's not clear whether this issue (probably caused by the inclusion of DC blocking capacitors) affects all Acer models or just a small number. With the issue with NIDs 0x0f and 0x10 it's unclear whether this is a hardware bug which will be addressed in later chip revisions or if it's an intentional restriction. The datasheet makes no mention of the restriction so at this stage I'm inclined to consider it a hardware bug. Comments in the source reflect this reasoning. On a similar theme, the headphone jack of the Fujitsu S7020 also doesn't appear to pass mic bias voltage. I'm still investigating this however. With the ability to retask the headphone jack, owners of ALC260-based Acer laptops should now be able to record 4 channels of audio if they desire. The multiple mux definitions allow this jack to be presented from both ADCs (since this mux input is one of those which differs between the muxes). This patch has been tested on a Fujitsu S7020 laptop and appears to behave itself both for the "test" and "fujitsu" models. Definitions using only a single mux specification also work. Other ALC chips should be fine but I cannot test these myself. The "auto" modes should also continue to function but again I have not verified this. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2006-03-28 18:47:09 +08:00
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux;
spec->adc_nids = alc861_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids);
spec->mixers[spec->num_mixers] = alc861_capture_mixer;
spec->num_mixers++;
return 1;
}
/* additional initialization for auto-configuration model */
static void alc861_auto_init(struct hda_codec *codec)
{
alc861_auto_init_multi_out(codec);
alc861_auto_init_hp_out(codec);
alc861_auto_init_analog_input(codec);
}
/*
* configuration and preset
*/
static struct hda_board_config alc861_cfg_tbl[] = {
{ .modelname = "3stack", .config = ALC861_3ST },
{ .pci_subvendor = 0x8086, .pci_subdevice = 0xd600, .config = ALC861_3ST },
{ .modelname = "3stack-dig", .config = ALC861_3ST_DIG },
{ .modelname = "6stack-dig", .config = ALC861_6ST_DIG },
{ .modelname = "auto", .config = ALC861_AUTO },
{}
};
static struct alc_config_preset alc861_presets[] = {
[ALC861_3ST] = {
.mixers = { alc861_3ST_mixer },
.init_verbs = { alc861_threestack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
.channel_mode = alc861_threestack_modes,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_3ST_DIG] = {
.mixers = { alc861_base_mixer },
.init_verbs = { alc861_threestack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
.channel_mode = alc861_threestack_modes,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_6ST_DIG] = {
.mixers = { alc861_base_mixer },
.init_verbs = { alc861_base_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861_8ch_modes),
.channel_mode = alc861_8ch_modes,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
};
static int patch_alc861(struct hda_codec *codec)
{
struct alc_spec *spec;
int board_config;
int err;
spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
board_config = snd_hda_check_board_config(codec, alc861_cfg_tbl);
if (board_config < 0 || board_config >= ALC861_MODEL_LAST) {
printk(KERN_INFO "hda_codec: Unknown model for ALC861, trying auto-probe from BIOS...\n");
board_config = ALC861_AUTO;
}
if (board_config == ALC861_AUTO) {
/* automatic parse from the BIOS config */
err = alc861_parse_auto_config(codec);
if (err < 0) {
alc_free(codec);
return err;
} else if (! err) {
printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using base mode...\n");
board_config = ALC861_3ST_DIG;
}
}
if (board_config != ALC861_AUTO)
setup_preset(spec, &alc861_presets[board_config]);
spec->stream_name_analog = "ALC861 Analog";
spec->stream_analog_playback = &alc861_pcm_analog_playback;
spec->stream_analog_capture = &alc861_pcm_analog_capture;
spec->stream_name_digital = "ALC861 Digital";
spec->stream_digital_playback = &alc861_pcm_digital_playback;
spec->stream_digital_capture = &alc861_pcm_digital_capture;
codec->patch_ops = alc_patch_ops;
if (board_config == ALC861_AUTO)
spec->init_hook = alc861_auto_init;
return 0;
}
/*
* patch entries
*/
struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
{ .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
{ .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 },
{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
{ .id = 0x10ec0861, .name = "ALC861", .patch = patch_alc861 },
{} /* terminator */
};