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 <linux/init.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/pci.h>
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
#include "hda_beep.h"
#define ALC880_FRONT_EVENT 0x01
#define ALC880_DCVOL_EVENT 0x02
#define ALC880_HP_EVENT 0x04
#define ALC880_MIC_EVENT 0x08
/* ALC880 board config type */
enum {
ALC880_3ST,
ALC880_3ST_DIG,
ALC880_5ST,
ALC880_5ST_DIG,
ALC880_W810,
ALC880_Z71V,
ALC880_6ST,
ALC880_6ST_DIG,
ALC880_F1734,
ALC880_ASUS,
ALC880_ASUS_DIG,
ALC880_ASUS_W1V,
ALC880_ASUS_DIG2,
ALC880_FUJITSU,
ALC880_UNIWILL_DIG,
ALC880_UNIWILL,
ALC880_UNIWILL_P53,
ALC880_CLEVO,
ALC880_TCL_S700,
ALC880_LG,
ALC880_LG_LW,
ALC880_MEDION_RIM,
#ifdef CONFIG_SND_DEBUG
ALC880_TEST,
#endif
ALC880_AUTO,
ALC880_MODEL_LAST /* last tag */
};
/* ALC260 models */
enum {
ALC260_BASIC,
ALC260_HP,
ALC260_HP_DC7600,
ALC260_HP_3013,
ALC260_FUJITSU_S702X,
ALC260_ACER,
ALC260_WILL,
ALC260_REPLACER_672V,
ALC260_FAVORIT100,
#ifdef CONFIG_SND_DEBUG
ALC260_TEST,
#endif
ALC260_AUTO,
ALC260_MODEL_LAST /* last tag */
};
/* ALC262 models */
enum {
ALC262_BASIC,
ALC262_HIPPO,
ALC262_HIPPO_1,
ALC262_FUJITSU,
ALC262_HP_BPC,
ALC262_HP_BPC_D7000_WL,
ALC262_HP_BPC_D7000_WF,
ALC262_HP_TC_T5735,
ALC262_HP_RP5700,
ALC262_BENQ_ED8,
ALC262_SONY_ASSAMD,
ALC262_BENQ_T31,
ALC262_ULTRA,
ALC262_LENOVO_3000,
ALC262_NEC,
ALC262_TOSHIBA_S06,
ALC262_TOSHIBA_RX1,
ALC262_TYAN,
ALC262_AUTO,
ALC262_MODEL_LAST /* last tag */
};
/* ALC268 models */
enum {
ALC267_QUANTA_IL1,
ALC268_3ST,
ALC268_TOSHIBA,
ALC268_ACER,
ALC268_ACER_DMIC,
ALC268_ACER_ASPIRE_ONE,
ALC268_DELL,
ALC268_ZEPTO,
#ifdef CONFIG_SND_DEBUG
ALC268_TEST,
#endif
ALC268_AUTO,
ALC268_MODEL_LAST /* last tag */
};
/* ALC269 models */
enum {
ALC269_BASIC,
ALC269_QUANTA_FL1,
ALC269_ASUS_EEEPC_P703,
ALC269_ASUS_EEEPC_P901,
ALC269_FUJITSU,
ALC269_LIFEBOOK,
ALC269_AUTO,
ALC269_MODEL_LAST /* last tag */
};
/* ALC861 models */
enum {
ALC861_3ST,
ALC660_3ST,
ALC861_3ST_DIG,
ALC861_6ST_DIG,
ALC861_UNIWILL_M31,
ALC861_TOSHIBA,
ALC861_ASUS,
ALC861_ASUS_LAPTOP,
ALC861_AUTO,
ALC861_MODEL_LAST,
};
/* ALC861-VD models */
enum {
ALC660VD_3ST,
ALC660VD_3ST_DIG,
ALC660VD_ASUS_V1S,
ALC861VD_3ST,
ALC861VD_3ST_DIG,
ALC861VD_6ST_DIG,
ALC861VD_LENOVO,
ALC861VD_DALLAS,
ALC861VD_HP,
ALC861VD_AUTO,
ALC861VD_MODEL_LAST,
};
/* ALC662 models */
enum {
ALC662_3ST_2ch_DIG,
ALC662_3ST_6ch_DIG,
ALC662_3ST_6ch,
ALC662_5ST_DIG,
ALC662_LENOVO_101E,
ALC662_ASUS_EEEPC_P701,
ALC662_ASUS_EEEPC_EP20,
ALC663_ASUS_M51VA,
ALC663_ASUS_G71V,
ALC663_ASUS_H13,
ALC663_ASUS_G50V,
ALC662_ECS,
ALC663_ASUS_MODE1,
ALC662_ASUS_MODE2,
ALC663_ASUS_MODE3,
ALC663_ASUS_MODE4,
ALC663_ASUS_MODE5,
ALC663_ASUS_MODE6,
ALC272_DELL,
ALC272_DELL_ZM1,
ALC272_SAMSUNG_NC10,
ALC662_AUTO,
ALC662_MODEL_LAST,
};
/* ALC882 models */
enum {
ALC882_3ST_DIG,
ALC882_6ST_DIG,
ALC882_ARIMA,
ALC882_W2JC,
ALC882_TARGA,
ALC882_ASUS_A7J,
ALC882_ASUS_A7M,
ALC885_MACPRO,
ALC885_MBP3,
ALC885_MB5,
ALC885_IMAC24,
ALC883_3ST_2ch_DIG,
ALC883_3ST_6ch_DIG,
ALC883_3ST_6ch,
ALC883_6ST_DIG,
ALC883_TARGA_DIG,
ALC883_TARGA_2ch_DIG,
ALC883_TARGA_8ch_DIG,
ALC883_ACER,
ALC883_ACER_ASPIRE,
ALC888_ACER_ASPIRE_4930G,
ALC888_ACER_ASPIRE_6530G,
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
ALC888_ACER_ASPIRE_8930G,
ALC888_ACER_ASPIRE_7730G,
ALC883_MEDION,
ALC883_MEDION_MD2,
ALC883_LAPTOP_EAPD,
ALC883_LENOVO_101E_2ch,
ALC883_LENOVO_NB0763,
ALC888_LENOVO_MS7195_DIG,
ALC888_LENOVO_SKY,
ALC883_HAIER_W66,
ALC888_3ST_HP,
ALC888_6ST_DELL,
ALC883_MITAC,
ALC883_CLEVO_M720,
ALC883_FUJITSU_PI2515,
ALC888_FUJITSU_XA3530,
ALC883_3ST_6ch_INTEL,
ALC889A_INTEL,
ALC889_INTEL,
ALC888_ASUS_M90V,
ALC888_ASUS_EEE1601,
ALC889A_MB31,
ALC1200_ASUS_P5Q,
ALC883_SONY_VAIO_TT,
ALC882_AUTO,
ALC882_MODEL_LAST,
};
/* for GPIO Poll */
#define GPIO_MASK 0x03
/* extra amp-initialization sequence types */
enum {
ALC_INIT_NONE,
ALC_INIT_DEFAULT,
ALC_INIT_GPIO1,
ALC_INIT_GPIO2,
ALC_INIT_GPIO3,
};
struct alc_mic_route {
hda_nid_t pin;
unsigned char mux_idx;
unsigned char amix_idx;
};
#define MUX_IDX_UNDEF ((unsigned char)-1)
struct alc_spec {
/* codec parameterization */
struct snd_kcontrol_new *mixers[5]; /* mixer arrays */
unsigned int num_mixers;
struct snd_kcontrol_new *cap_mixer; /* capture mixer */
unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */
const struct hda_verb *init_verbs[5]; /* initialization verbs
* don't forget NULL
* termination!
*/
unsigned int num_init_verbs;
char stream_name_analog[32]; /* analog PCM stream */
struct hda_pcm_stream *stream_analog_playback;
struct hda_pcm_stream *stream_analog_capture;
struct hda_pcm_stream *stream_analog_alt_playback;
struct hda_pcm_stream *stream_analog_alt_capture;
char stream_name_digital[32]; /* digital PCM stream */
struct hda_pcm_stream *stream_digital_playback;
struct hda_pcm_stream *stream_digital_capture;
/* playback */
struct hda_multi_out multiout; /* playback set-up
* max_channels, dacs must be set
* dig_out_nid and hp_nid are optional
*/
hda_nid_t alt_dac_nid;
hda_nid_t slave_dig_outs[3]; /* optional - for auto-parsing */
int dig_out_type;
/* capture */
unsigned int num_adc_nids;
hda_nid_t *adc_nids;
hda_nid_t *capsrc_nids;
hda_nid_t dig_in_nid; /* digital-in NID; optional */
/* capture source */
[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 10:47:09 +00:00
unsigned int num_mux_defs;
const struct hda_input_mux *input_mux;
unsigned int cur_mux[3];
struct alc_mic_route ext_mic;
struct alc_mic_route int_mic;
/* channel model */
const struct hda_channel_mode *channel_mode;
int num_channel_mode;
int need_dac_fix;
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
int const_channel_count;
int ext_channel_count;
/* PCM information */
struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */
/* dynamic controls, init_verbs and input_mux */
struct auto_pin_cfg autocfg;
struct snd_array kctls;
struct hda_input_mux private_imux[3];
hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
hda_nid_t private_adc_nids[AUTO_CFG_MAX_OUTS];
hda_nid_t private_capsrc_nids[AUTO_CFG_MAX_OUTS];
/* hooks */
void (*init_hook)(struct hda_codec *codec);
void (*unsol_event)(struct hda_codec *codec, unsigned int res);
/* for pin sensing */
unsigned int sense_updated: 1;
unsigned int jack_present: 1;
unsigned int master_sw: 1;
unsigned int auto_mic:1;
/* other flags */
unsigned int no_analog :1; /* digital I/O only */
int init_amp;
/* for virtual master */
hda_nid_t vmaster_nid;
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_loopback_check loopback;
#endif
/* for PLL fix */
hda_nid_t pll_nid;
unsigned int pll_coef_idx, pll_coef_bit;
};
/*
* configuration template - to be copied to the spec instance
*/
struct alc_config_preset {
struct snd_kcontrol_new *mixers[5]; /* should be identical size
* with spec
*/
struct snd_kcontrol_new *cap_mixer; /* capture mixer */
const struct hda_verb *init_verbs[5];
unsigned int num_dacs;
hda_nid_t *dac_nids;
hda_nid_t dig_out_nid; /* optional */
hda_nid_t hp_nid; /* optional */
hda_nid_t *slave_dig_outs;
unsigned int num_adc_nids;
hda_nid_t *adc_nids;
hda_nid_t *capsrc_nids;
hda_nid_t dig_in_nid;
unsigned int num_channel_mode;
const struct hda_channel_mode *channel_mode;
int need_dac_fix;
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
int const_channel_count;
[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 10:47:09 +00:00
unsigned int num_mux_defs;
const struct hda_input_mux *input_mux;
void (*unsol_event)(struct hda_codec *, unsigned int);
void (*setup)(struct hda_codec *);
void (*init_hook)(struct hda_codec *);
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_amp_list *loopbacks;
#endif
};
/*
* input MUX handling
*/
static int alc_mux_enum_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
[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 10:47:09 +00:00
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;
const struct hda_input_mux *imux;
unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
unsigned int mux_idx;
hda_nid_t nid = spec->capsrc_nids ?
spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx];
unsigned int type;
mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
imux = &spec->input_mux[mux_idx];
type = get_wcaps_type(get_wcaps(codec, nid));
if (type == AC_WID_AUD_MIX) {
/* Matrix-mixer style (e.g. ALC882) */
unsigned int *cur_val = &spec->cur_mux[adc_idx];
unsigned int i, idx;
idx = ucontrol->value.enumerated.item[0];
if (idx >= imux->num_items)
idx = imux->num_items - 1;
if (*cur_val == idx)
return 0;
for (i = 0; i < imux->num_items; i++) {
unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
imux->items[i].index,
HDA_AMP_MUTE, v);
}
*cur_val = idx;
return 1;
} else {
/* MUX style (e.g. ALC880) */
return snd_hda_input_mux_put(codec, imux, ucontrol, nid,
&spec->cur_mux[adc_idx]);
}
}
/*
* channel mode setting
*/
static int alc_ch_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode,
spec->num_channel_mode);
}
static int alc_ch_mode_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
spec->num_channel_mode,
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
spec->ext_channel_count);
}
static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
spec->num_channel_mode,
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
&spec->ext_channel_count);
if (err >= 0 && !spec->const_channel_count) {
spec->multiout.max_channels = spec->ext_channel_count;
if (spec->need_dac_fix)
spec->multiout.num_dacs = spec->multiout.max_channels / 2;
}
return err;
}
/*
* Control the mode of pin widget settings via the mixer. "pc" is used
* instead of "%" to avoid consequences of accidently treating the % as
* being part of a format specifier. Maximum allowed length of a value is
* 63 characters plus NULL terminator.
*
* Note: some retasking pin complexes seem to ignore requests for input
* states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these
* are requested. Therefore order this list so that this behaviour will not
* cause problems when mixer clients move through the enum sequentially.
[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 10:47:09 +00:00
* 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 10:47:09 +00:00
* 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 10:47:09 +00:00
#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
/* 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 10:47:09 +00:00
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 10:47:09 +00:00
{ 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 (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl)
i++;
*valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
return 0;
}
static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL,
0x00);
if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir))
val = alc_pin_mode_min(dir);
change = pinctl != alc_pin_mode_values[val];
if (change) {
/* Set pin mode to that requested */
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
alc_pin_mode_values[val]);
/* Also enable the retasking pin's input/output as required
* for the requested pin mode. Enum values of 2 or less are
* input modes.
*
* Dynamically switching the input/output buffers probably
[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 10:47:09 +00:00
* 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_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
HDA_AMP_MUTE, 0);
} else {
snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, 0);
}
}
return change;
}
#define ALC_PIN_MODE(xname, nid, dir) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.info = alc_pin_mode_info, \
.get = alc_pin_mode_get, \
.put = alc_pin_mode_put, \
.private_value = nid | (dir<<16) }
/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged
* together using a mask with more than one bit set. This control is
* currently used only by the ALC260 test model. At this stage they are not
* needed for any "production" models.
*/
#ifdef CONFIG_SND_DEBUG
#define alc_gpio_data_info snd_ctl_boolean_mono_info
static int alc_gpio_data_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_GPIO_DATA, 0x00);
*valp = (val & mask) != 0;
return 0;
}
static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_GPIO_DATA,
0x00);
/* Set/unset the masked GPIO bit(s) as needed */
change = (val == 0 ? 0 : mask) != (gpio_data & mask);
if (val == 0)
gpio_data &= ~mask;
else
gpio_data |= mask;
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_GPIO_DATA, gpio_data);
return change;
}
#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.info = alc_gpio_data_info, \
.get = alc_gpio_data_get, \
.put = alc_gpio_data_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
/* A switch control to allow the enabling of the digital IO pins on the
* ALC260. This is incredibly simplistic; the intention of this control is
* to provide something in the test model allowing digital outputs to be
* identified if present. If models are found which can utilise these
* outputs a more complete mixer control can be devised for those models if
* necessary.
*/
#ifdef CONFIG_SND_DEBUG
#define alc_spdif_ctrl_info snd_ctl_boolean_mono_info
static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_DIGI_CONVERT_1, 0x00);
*valp = (val & mask) != 0;
return 0;
}
static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_DIGI_CONVERT_1,
0x00);
/* Set/unset the masked control bit(s) as needed */
change = (val == 0 ? 0 : mask) != (ctrl_data & mask);
if (val==0)
ctrl_data &= ~mask;
else
ctrl_data |= mask;
snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
ctrl_data);
return change;
}
#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.info = alc_spdif_ctrl_info, \
.get = alc_spdif_ctrl_get, \
.put = alc_spdif_ctrl_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
/* A switch control to allow the enabling EAPD digital outputs on the ALC26x.
* Again, this is only used in the ALC26x test models to help identify when
* the EAPD line must be asserted for features to work.
*/
#ifdef CONFIG_SND_DEBUG
#define alc_eapd_ctrl_info snd_ctl_boolean_mono_info
static int alc_eapd_ctrl_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_EAPD_BTLENABLE, 0x00);
*valp = (val & mask) != 0;
return 0;
}
static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_EAPD_BTLENABLE,
0x00);
/* Set/unset the masked control bit(s) as needed */
change = (!val ? 0 : mask) != (ctrl_data & mask);
if (!val)
ctrl_data &= ~mask;
else
ctrl_data |= mask;
snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
ctrl_data);
return change;
}
#define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.info = alc_eapd_ctrl_info, \
.get = alc_eapd_ctrl_get, \
.put = alc_eapd_ctrl_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
/*
* set up the input pin config (depending on the given auto-pin type)
*/
static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid,
int auto_pin_type)
{
unsigned int val = PIN_IN;
if (auto_pin_type <= AUTO_PIN_FRONT_MIC) {
unsigned int pincap;
pincap = snd_hda_query_pin_caps(codec, nid);
pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
if (pincap & AC_PINCAP_VREF_80)
val = PIN_VREF80;
else if (pincap & AC_PINCAP_VREF_50)
val = PIN_VREF50;
else if (pincap & AC_PINCAP_VREF_100)
val = PIN_VREF100;
else if (pincap & AC_PINCAP_VREF_GRD)
val = PIN_VREFGRD;
}
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val);
}
/*
*/
static void add_mixer(struct alc_spec *spec, struct snd_kcontrol_new *mix)
{
if (snd_BUG_ON(spec->num_mixers >= ARRAY_SIZE(spec->mixers)))
return;
spec->mixers[spec->num_mixers++] = mix;
}
static void add_verb(struct alc_spec *spec, const struct hda_verb *verb)
{
if (snd_BUG_ON(spec->num_init_verbs >= ARRAY_SIZE(spec->init_verbs)))
return;
spec->init_verbs[spec->num_init_verbs++] = verb;
}
#ifdef CONFIG_PROC_FS
/*
* hook for proc
*/
static void print_realtek_coef(struct snd_info_buffer *buffer,
struct hda_codec *codec, hda_nid_t nid)
{
int coeff;
if (nid != 0x20)
return;
coeff = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PROC_COEF, 0);
snd_iprintf(buffer, " Processing Coefficient: 0x%02x\n", coeff);
coeff = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_COEF_INDEX, 0);
snd_iprintf(buffer, " Coefficient Index: 0x%02x\n", coeff);
}
#else
#define print_realtek_coef NULL
#endif
/*
* set up from the preset table
*/
static void setup_preset(struct hda_codec *codec,
const struct alc_config_preset *preset)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
add_mixer(spec, preset->mixers[i]);
spec->cap_mixer = preset->cap_mixer;
for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i];
i++)
add_verb(spec, preset->init_verbs[i]);
spec->channel_mode = preset->channel_mode;
spec->num_channel_mode = preset->num_channel_mode;
spec->need_dac_fix = preset->need_dac_fix;
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
spec->const_channel_count = preset->const_channel_count;
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
if (preset->const_channel_count)
spec->multiout.max_channels = preset->const_channel_count;
else
spec->multiout.max_channels = spec->channel_mode[0].channels;
spec->ext_channel_count = spec->channel_mode[0].channels;
spec->multiout.num_dacs = preset->num_dacs;
spec->multiout.dac_nids = preset->dac_nids;
spec->multiout.dig_out_nid = preset->dig_out_nid;
spec->multiout.slave_dig_outs = preset->slave_dig_outs;
spec->multiout.hp_nid = preset->hp_nid;
[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 10:47:09 +00:00
spec->num_mux_defs = preset->num_mux_defs;
if (!spec->num_mux_defs)
[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 10:47:09 +00:00
spec->num_mux_defs = 1;
spec->input_mux = preset->input_mux;
spec->num_adc_nids = preset->num_adc_nids;
spec->adc_nids = preset->adc_nids;
spec->capsrc_nids = preset->capsrc_nids;
spec->dig_in_nid = preset->dig_in_nid;
spec->unsol_event = preset->unsol_event;
spec->init_hook = preset->init_hook;
#ifdef CONFIG_SND_HDA_POWER_SAVE
spec->loopback.amplist = preset->loopbacks;
#endif
if (preset->setup)
preset->setup(codec);
}
/* Enable GPIO mask and set output */
static struct hda_verb alc_gpio1_init_verbs[] = {
{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
{ }
};
static struct hda_verb alc_gpio2_init_verbs[] = {
{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
{0x01, AC_VERB_SET_GPIO_DATA, 0x02},
{ }
};
static struct hda_verb alc_gpio3_init_verbs[] = {
{0x01, AC_VERB_SET_GPIO_MASK, 0x03},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03},
{0x01, AC_VERB_SET_GPIO_DATA, 0x03},
{ }
};
/*
* Fix hardware PLL issue
* On some codecs, the analog PLL gating control must be off while
* the default value is 1.
*/
static void alc_fix_pll(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int val;
if (!spec->pll_nid)
return;
snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX,
spec->pll_coef_idx);
val = snd_hda_codec_read(codec, spec->pll_nid, 0,
AC_VERB_GET_PROC_COEF, 0);
snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX,
spec->pll_coef_idx);
snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_PROC_COEF,
val & ~(1 << spec->pll_coef_bit));
}
static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid,
unsigned int coef_idx, unsigned int coef_bit)
{
struct alc_spec *spec = codec->spec;
spec->pll_nid = nid;
spec->pll_coef_idx = coef_idx;
spec->pll_coef_bit = coef_bit;
alc_fix_pll(codec);
}
static void alc_automute_pin(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int present, pincap;
unsigned int nid = spec->autocfg.hp_pins[0];
int i;
pincap = snd_hda_query_pin_caps(codec, nid);
if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
present = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_SENSE, 0);
spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) {
nid = spec->autocfg.speaker_pins[i];
if (!nid)
break;
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
spec->jack_present ? 0 : PIN_OUT);
}
}
static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
hda_nid_t nid)
{
hda_nid_t conn[HDA_MAX_NUM_INPUTS];
int i, nums;
nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
for (i = 0; i < nums; i++)
if (conn[i] == nid)
return i;
return -1;
}
static void alc_mic_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct alc_mic_route *dead, *alive;
unsigned int present, type;
hda_nid_t cap_nid;
if (!spec->auto_mic)
return;
if (!spec->int_mic.pin || !spec->ext_mic.pin)
return;
if (snd_BUG_ON(!spec->adc_nids))
return;
cap_nid = spec->capsrc_nids ? spec->capsrc_nids[0] : spec->adc_nids[0];
present = snd_hda_codec_read(codec, spec->ext_mic.pin, 0,
AC_VERB_GET_PIN_SENSE, 0);
present &= AC_PINSENSE_PRESENCE;
if (present) {
alive = &spec->ext_mic;
dead = &spec->int_mic;
} else {
alive = &spec->int_mic;
dead = &spec->ext_mic;
}
type = get_wcaps_type(get_wcaps(codec, cap_nid));
if (type == AC_WID_AUD_MIX) {
/* Matrix-mixer style (e.g. ALC882) */
snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT,
alive->mux_idx,
HDA_AMP_MUTE, 0);
snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT,
dead->mux_idx,
HDA_AMP_MUTE, HDA_AMP_MUTE);
} else {
/* MUX style (e.g. ALC880) */
snd_hda_codec_write_cache(codec, cap_nid, 0,
AC_VERB_SET_CONNECT_SEL,
alive->mux_idx);
}
/* FIXME: analog mixer */
}
/* unsolicited event for HP jack sensing */
static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
{
if (codec->vendor_id == 0x10ec0880)
res >>= 28;
else
res >>= 26;
switch (res) {
case ALC880_HP_EVENT:
alc_automute_pin(codec);
break;
case ALC880_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
static void alc_inithook(struct hda_codec *codec)
{
alc_automute_pin(codec);
alc_mic_automute(codec);
}
/* additional initialization for ALC888 variants */
static void alc888_coef_init(struct hda_codec *codec)
{
unsigned int tmp;
snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0);
tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0);
snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7);
if ((tmp & 0xf0) == 0x20)
/* alc888S-VC */
snd_hda_codec_read(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF, 0x830);
else
/* alc888-VB */
snd_hda_codec_read(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF, 0x3030);
}
static void alc889_coef_init(struct hda_codec *codec)
{
unsigned int tmp;
snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7);
tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0);
snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7);
snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp|0x2010);
}
static void alc_auto_init_amp(struct hda_codec *codec, int type)
{
unsigned int tmp;
switch (type) {
case ALC_INIT_GPIO1:
snd_hda_sequence_write(codec, alc_gpio1_init_verbs);
break;
case ALC_INIT_GPIO2:
snd_hda_sequence_write(codec, alc_gpio2_init_verbs);
break;
case ALC_INIT_GPIO3:
snd_hda_sequence_write(codec, alc_gpio3_init_verbs);
break;
case ALC_INIT_DEFAULT:
switch (codec->vendor_id) {
case 0x10ec0260:
snd_hda_codec_write(codec, 0x0f, 0,
AC_VERB_SET_EAPD_BTLENABLE, 2);
snd_hda_codec_write(codec, 0x10, 0,
AC_VERB_SET_EAPD_BTLENABLE, 2);
break;
case 0x10ec0262:
case 0x10ec0267:
case 0x10ec0268:
case 0x10ec0269:
case 0x10ec0272:
case 0x10ec0660:
case 0x10ec0662:
case 0x10ec0663:
case 0x10ec0862:
case 0x10ec0889:
snd_hda_codec_write(codec, 0x14, 0,
AC_VERB_SET_EAPD_BTLENABLE, 2);
snd_hda_codec_write(codec, 0x15, 0,
AC_VERB_SET_EAPD_BTLENABLE, 2);
break;
}
switch (codec->vendor_id) {
case 0x10ec0260:
snd_hda_codec_write(codec, 0x1a, 0,
AC_VERB_SET_COEF_INDEX, 7);
tmp = snd_hda_codec_read(codec, 0x1a, 0,
AC_VERB_GET_PROC_COEF, 0);
snd_hda_codec_write(codec, 0x1a, 0,
AC_VERB_SET_COEF_INDEX, 7);
snd_hda_codec_write(codec, 0x1a, 0,
AC_VERB_SET_PROC_COEF,
tmp | 0x2010);
break;
case 0x10ec0262:
case 0x10ec0880:
case 0x10ec0882:
case 0x10ec0883:
case 0x10ec0885:
case 0x10ec0887:
case 0x10ec0889:
alc889_coef_init(codec);
break;
case 0x10ec0888:
alc888_coef_init(codec);
break;
case 0x10ec0267:
case 0x10ec0268:
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 7);
tmp = snd_hda_codec_read(codec, 0x20, 0,
AC_VERB_GET_PROC_COEF, 0);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 7);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF,
tmp | 0x3000);
break;
}
break;
}
}
static void alc_init_auto_hp(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
if (!spec->autocfg.hp_pins[0])
return;
if (!spec->autocfg.speaker_pins[0]) {
if (spec->autocfg.line_out_pins[0] &&
spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
spec->autocfg.speaker_pins[0] =
spec->autocfg.line_out_pins[0];
else
return;
}
snd_printdd("realtek: Enable HP auto-muting on NID 0x%x\n",
spec->autocfg.hp_pins[0]);
snd_hda_codec_write_cache(codec, spec->autocfg.hp_pins[0], 0,
AC_VERB_SET_UNSOLICITED_ENABLE,
AC_USRSP_EN | ALC880_HP_EVENT);
spec->unsol_event = alc_sku_unsol_event;
}
static void alc_init_auto_mic(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
hda_nid_t fixed, ext;
int i;
/* there must be only two mic inputs exclusively */
for (i = AUTO_PIN_LINE; i < AUTO_PIN_LAST; i++)
if (cfg->input_pins[i])
return;
fixed = ext = 0;
for (i = AUTO_PIN_MIC; i <= AUTO_PIN_FRONT_MIC; i++) {
hda_nid_t nid = cfg->input_pins[i];
unsigned int defcfg;
if (!nid)
return;
defcfg = snd_hda_codec_get_pincfg(codec, nid);
switch (get_defcfg_connect(defcfg)) {
case AC_JACK_PORT_FIXED:
if (fixed)
return; /* already occupied */
fixed = nid;
break;
case AC_JACK_PORT_COMPLEX:
if (ext)
return; /* already occupied */
ext = nid;
break;
default:
return; /* invalid entry */
}
}
if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP))
return; /* no unsol support */
snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n",
ext, fixed);
spec->ext_mic.pin = ext;
spec->int_mic.pin = fixed;
spec->ext_mic.mux_idx = MUX_IDX_UNDEF; /* set later */
spec->int_mic.mux_idx = MUX_IDX_UNDEF; /* set later */
spec->auto_mic = 1;
snd_hda_codec_write_cache(codec, spec->ext_mic.pin, 0,
AC_VERB_SET_UNSOLICITED_ENABLE,
AC_USRSP_EN | ALC880_MIC_EVENT);
spec->unsol_event = alc_sku_unsol_event;
}
/* check subsystem ID and set up device-specific initialization;
* return 1 if initialized, 0 if invalid SSID
*/
/* 32-bit subsystem ID for BIOS loading in HD Audio codec.
* 31 ~ 16 : Manufacture ID
* 15 ~ 8 : SKU ID
* 7 ~ 0 : Assembly ID
* port-A --> pin 39/41, port-E --> pin 14/15, port-D --> pin 35/36
*/
static int alc_subsystem_id(struct hda_codec *codec,
hda_nid_t porta, hda_nid_t porte,
hda_nid_t portd)
{
unsigned int ass, tmp, i;
unsigned nid;
struct alc_spec *spec = codec->spec;
ass = codec->subsystem_id & 0xffff;
if ((ass != codec->bus->pci->subsystem_device) && (ass & 1))
goto do_sku;
/* invalid SSID, check the special NID pin defcfg instead */
/*
* 31~30 : port connectivity
* 29~21 : reserve
* 20 : PCBEEP input
* 19~16 : Check sum (15:1)
* 15~1 : Custom
* 0 : override
*/
nid = 0x1d;
if (codec->vendor_id == 0x10ec0260)
nid = 0x17;
ass = snd_hda_codec_get_pincfg(codec, nid);
snd_printd("realtek: No valid SSID, "
"checking pincfg 0x%08x for NID 0x%x\n",
ass, nid);
if (!(ass & 1) && !(ass & 0x100000))
return 0;
if ((ass >> 30) != 1) /* no physical connection */
return 0;
/* check sum */
tmp = 0;
for (i = 1; i < 16; i++) {
if ((ass >> i) & 1)
tmp++;
}
if (((ass >> 16) & 0xf) != tmp)
return 0;
do_sku:
snd_printd("realtek: Enabling init ASM_ID=0x%04x CODEC_ID=%08x\n",
ass & 0xffff, codec->vendor_id);
/*
* 0 : override
* 1 : Swap Jack
* 2 : 0 --> Desktop, 1 --> Laptop
* 3~5 : External Amplifier control
* 7~6 : Reserved
*/
tmp = (ass & 0x38) >> 3; /* external Amp control */
switch (tmp) {
case 1:
spec->init_amp = ALC_INIT_GPIO1;
break;
case 3:
spec->init_amp = ALC_INIT_GPIO2;
break;
case 7:
spec->init_amp = ALC_INIT_GPIO3;
break;
case 5:
spec->init_amp = ALC_INIT_DEFAULT;
break;
}
/* is laptop or Desktop and enable the function "Mute internal speaker
* when the external headphone out jack is plugged"
*/
if (!(ass & 0x8000))
return 1;
/*
* 10~8 : Jack location
* 12~11: Headphone out -> 00: PortA, 01: PortE, 02: PortD, 03: Resvered
* 14~13: Resvered
* 15 : 1 --> enable the function "Mute internal speaker
* when the external headphone out jack is plugged"
*/
if (!spec->autocfg.hp_pins[0]) {
tmp = (ass >> 11) & 0x3; /* HP to chassis */
if (tmp == 0)
spec->autocfg.hp_pins[0] = porta;
else if (tmp == 1)
spec->autocfg.hp_pins[0] = porte;
else if (tmp == 2)
spec->autocfg.hp_pins[0] = portd;
else
return 1;
}
alc_init_auto_hp(codec);
alc_init_auto_mic(codec);
return 1;
}
static void alc_ssid_check(struct hda_codec *codec,
hda_nid_t porta, hda_nid_t porte, hda_nid_t portd)
{
if (!alc_subsystem_id(codec, porta, porte, portd)) {
struct alc_spec *spec = codec->spec;
snd_printd("realtek: "
"Enable default setup for auto mode as fallback\n");
spec->init_amp = ALC_INIT_DEFAULT;
alc_init_auto_hp(codec);
alc_init_auto_mic(codec);
}
}
/*
* Fix-up pin default configurations
*/
struct alc_pincfg {
hda_nid_t nid;
u32 val;
};
static void alc_fix_pincfg(struct hda_codec *codec,
const struct snd_pci_quirk *quirk,
const struct alc_pincfg **pinfix)
{
const struct alc_pincfg *cfg;
quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
if (!quirk)
return;
cfg = pinfix[quirk->value];
for (; cfg->nid; cfg++)
snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
}
/*
* ALC888
*/
/*
* 2ch mode
*/
static struct hda_verb alc888_4ST_ch2_intel_init[] = {
/* Mic-in jack as mic in */
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
/* Line-in jack as Line in */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
/* Line-Out as Front */
{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
{ } /* end */
};
/*
* 4ch mode
*/
static struct hda_verb alc888_4ST_ch4_intel_init[] = {
/* Mic-in jack as mic in */
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
/* Line-in jack as Surround */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
/* Line-Out as Front */
{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
{ } /* end */
};
/*
* 6ch mode
*/
static struct hda_verb alc888_4ST_ch6_intel_init[] = {
/* Mic-in jack as CLFE */
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
/* Line-in jack as Surround */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
/* Line-Out as CLFE (workaround because Mic-in is not loud enough) */
{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
{ } /* end */
};
/*
* 8ch mode
*/
static struct hda_verb alc888_4ST_ch8_intel_init[] = {
/* Mic-in jack as CLFE */
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
/* Line-in jack as Surround */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
/* Line-Out as Side */
{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
{ } /* end */
};
static struct hda_channel_mode alc888_4ST_8ch_intel_modes[4] = {
{ 2, alc888_4ST_ch2_intel_init },
{ 4, alc888_4ST_ch4_intel_init },
{ 6, alc888_4ST_ch6_intel_init },
{ 8, alc888_4ST_ch8_intel_init },
};
/*
* ALC888 Fujitsu Siemens Amillo xa3530
*/
static struct hda_verb alc888_fujitsu_xa3530_verbs[] = {
/* Front Mic: set to PIN_IN (empty by default) */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Connect Internal HP to Front */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Connect Bass HP to Front */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Connect Line-Out side jack (SPDIF) to Side */
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
/* Connect Mic jack to CLFE */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x02},
/* Connect Line-in jack to Surround */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
/* Connect HP out jack to Front */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Enable unsolicited event for HP jack and Line-out jack */
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{0x17, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{}
};
static void alc_automute_amp(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int val, mute, pincap;
hda_nid_t nid;
int i;
spec->jack_present = 0;
for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) {
nid = spec->autocfg.hp_pins[i];
if (!nid)
break;
pincap = snd_hda_query_pin_caps(codec, nid);
if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
snd_hda_codec_read(codec, nid, 0,
AC_VERB_SET_PIN_SENSE, 0);
val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_SENSE, 0);
if (val & AC_PINSENSE_PRESENCE) {
spec->jack_present = 1;
break;
}
}
mute = spec->jack_present ? HDA_AMP_MUTE : 0;
/* Toggle internal speakers muting */
for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) {
nid = spec->autocfg.speaker_pins[i];
if (!nid)
break;
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, mute);
}
}
static void alc_automute_amp_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if (codec->vendor_id == 0x10ec0880)
res >>= 28;
else
res >>= 26;
if (res == ALC880_HP_EVENT)
alc_automute_amp(codec);
}
static void alc889_automute_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x17;
spec->autocfg.speaker_pins[3] = 0x19;
spec->autocfg.speaker_pins[4] = 0x1a;
}
static void alc889_intel_init_hook(struct hda_codec *codec)
{
alc889_coef_init(codec);
alc_automute_amp(codec);
}
static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x17; /* line-out */
spec->autocfg.hp_pins[1] = 0x1b; /* hp */
spec->autocfg.speaker_pins[0] = 0x14; /* speaker */
spec->autocfg.speaker_pins[1] = 0x15; /* bass */
}
/*
* ALC888 Acer Aspire 4930G model
*/
static struct hda_verb alc888_acer_aspire_4930g_verbs[] = {
/* Front Mic: set to PIN_IN (empty by default) */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Unselect Front Mic by default in input mixer 3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
/* Enable unsolicited event for HP jack */
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
/* Connect Internal HP to front */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Connect HP out to front */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{ }
};
/*
* ALC888 Acer Aspire 6530G model
*/
static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
/* Bias voltage on for external mic port */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
/* Front Mic: set to PIN_IN (empty by default) */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Unselect Front Mic by default in input mixer 3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
/* Enable unsolicited event for HP jack */
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
/* Enable speaker output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Enable headphone output */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{ }
};
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
/*
* ALC889 Acer Aspire 8930G model
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
*/
static struct hda_verb alc889_acer_aspire_8930g_verbs[] = {
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
/* Front Mic: set to PIN_IN (empty by default) */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Unselect Front Mic by default in input mixer 3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
/* Enable unsolicited event for HP jack */
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
/* Connect Internal Front to Front */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Connect Internal Rear to Rear */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x01},
/* Connect Internal CLFE to CLFE */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
/* Connect HP out to Front */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Enable all DACs */
/* DAC DISABLE/MUTE 1? */
/* setting bits 1-5 disables DAC nids 0x02-0x06 apparently. Init=0x38 */
{0x20, AC_VERB_SET_COEF_INDEX, 0x03},
{0x20, AC_VERB_SET_PROC_COEF, 0x0000},
/* DAC DISABLE/MUTE 2? */
/* some bit here disables the other DACs. Init=0x4900 */
{0x20, AC_VERB_SET_COEF_INDEX, 0x08},
{0x20, AC_VERB_SET_PROC_COEF, 0x0000},
/* Enable amplifiers */
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
{0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
/* DMIC fix
* This laptop has a stereo digital microphone. The mics are only 1cm apart
* which makes the stereo useless. However, either the mic or the ALC889
* makes the signal become a difference/sum signal instead of standard
* stereo, which is annoying. So instead we flip this bit which makes the
* codec replicate the sum signal to both channels, turning it into a
* normal mono mic.
*/
/* DMIC_CONTROL? Init value = 0x0001 */
{0x20, AC_VERB_SET_COEF_INDEX, 0x0b},
{0x20, AC_VERB_SET_PROC_COEF, 0x0003},
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
{ }
};
static struct hda_input_mux alc888_2_capture_sources[2] = {
/* Front mic only available on one ADC */
{
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Front Mic", 0xb },
},
},
{
.num_items = 3,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
}
};
static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = {
/* Interal mic only available on one ADC */
{
.num_items = 5,
.items = {
{ "Ext Mic", 0x0 },
{ "Line In", 0x2 },
{ "CD", 0x4 },
{ "Input Mix", 0xa },
{ "Int Mic", 0xb },
},
},
{
.num_items = 4,
.items = {
{ "Ext Mic", 0x0 },
{ "Line In", 0x2 },
{ "CD", 0x4 },
{ "Input Mix", 0xa },
},
}
};
static struct hda_input_mux alc889_capture_sources[3] = {
/* Digital mic only available on first "ADC" */
{
.num_items = 5,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Front Mic", 0xb },
{ "Input Mix", 0xa },
},
},
{
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Input Mix", 0xa },
},
},
{
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "Input Mix", 0xa },
},
}
};
static struct snd_kcontrol_new alc888_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
}
static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x17;
}
static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x1b;
}
/*
* ALC880 3-stack model
*
* DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e)
* Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18,
* F-Mic = 0x1b, HP = 0x19
*/
static hda_nid_t alc880_dac_nids[4] = {
/* front, rear, clfe, rear_surr */
0x02, 0x05, 0x04, 0x03
};
static hda_nid_t alc880_adc_nids[3] = {
/* ADC0-2 */
0x07, 0x08, 0x09,
};
/* The datasheet says the node 0x07 is connected from inputs,
* but it shows zero connection in the real implementation on some devices.
* Note: this is a 915GAV bug, fixed on 915GLV
*/
static hda_nid_t alc880_adc_nids_alt[2] = {
/* ADC1-2 */
0x08, 0x09,
};
#define ALC880_DIGOUT_NID 0x06
#define ALC880_DIGIN_NID 0x0a
static struct hda_input_mux alc880_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x3 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
/* channel source setting (2/6 channel selection for 3-stack) */
/* 2ch mode */
static struct hda_verb alc880_threestack_ch2_init[] = {
/* set line-in to input, mute it */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
/* set mic-in to input vref 80%, mute it */
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
{ } /* end */
};
/* 6ch mode */
static struct hda_verb alc880_threestack_ch6_init[] = {
/* set line-in to output, unmute it */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
/* set mic-in to output, unmute it */
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ } /* end */
};
static struct hda_channel_mode alc880_threestack_modes[2] = {
{ 2, alc880_threestack_ch2_init },
{ 6, alc880_threestack_ch6_init },
};
static struct snd_kcontrol_new alc880_three_stack_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
/* capture mixer elements */
static int alc_cap_vol_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
int err;
mutex_lock(&codec->control_mutex);
kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0,
HDA_INPUT);
err = snd_hda_mixer_amp_volume_info(kcontrol, uinfo);
mutex_unlock(&codec->control_mutex);
return err;
}
static int alc_cap_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
unsigned int size, unsigned int __user *tlv)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
int err;
mutex_lock(&codec->control_mutex);
kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0,
HDA_INPUT);
err = snd_hda_mixer_amp_tlv(kcontrol, op_flag, size, tlv);
mutex_unlock(&codec->control_mutex);
return err;
}
typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol,
getput_call_t func)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
int err;
mutex_lock(&codec->control_mutex);
kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx],
3, 0, HDA_INPUT);
err = func(kcontrol, ucontrol);
mutex_unlock(&codec->control_mutex);
return err;
}
static int alc_cap_vol_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
snd_hda_mixer_amp_volume_get);
}
static int alc_cap_vol_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
snd_hda_mixer_amp_volume_put);
}
/* capture mixer elements */
#define alc_cap_sw_info snd_ctl_boolean_stereo_info
static int alc_cap_sw_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
snd_hda_mixer_amp_switch_get);
}
static int alc_cap_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
snd_hda_mixer_amp_switch_put);
}
#define _DEFINE_CAPMIX(num) \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = "Capture Switch", \
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
.count = num, \
.info = alc_cap_sw_info, \
.get = alc_cap_sw_get, \
.put = alc_cap_sw_put, \
}, \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = "Capture Volume", \
.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | \
SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK), \
.count = num, \
.info = alc_cap_vol_info, \
.get = alc_cap_vol_get, \
.put = alc_cap_vol_put, \
.tlv = { .c = alc_cap_vol_tlv }, \
}
#define _DEFINE_CAPSRC(num) \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
/* .name = "Capture Source", */ \
.name = "Input Source", \
.count = num, \
.info = alc_mux_enum_info, \
.get = alc_mux_enum_get, \
.put = alc_mux_enum_put, \
}
#define DEFINE_CAPMIX(num) \
static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \
_DEFINE_CAPMIX(num), \
_DEFINE_CAPSRC(num), \
{ } /* end */ \
}
#define DEFINE_CAPMIX_NOSRC(num) \
static struct snd_kcontrol_new alc_capture_mixer_nosrc ## num[] = { \
_DEFINE_CAPMIX(num), \
{ } /* end */ \
}
/* up to three ADCs */
DEFINE_CAPMIX(1);
DEFINE_CAPMIX(2);
DEFINE_CAPMIX(3);
DEFINE_CAPMIX_NOSRC(1);
DEFINE_CAPMIX_NOSRC(2);
DEFINE_CAPMIX_NOSRC(3);
/*
* ALC880 5-stack model
*
* DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d),
* Side = 0x02 (0xd)
* Pin assignment: Front = 0x14, Surr = 0x17, CLFE = 0x16
* Line-In/Side = 0x1a, Mic = 0x18, F-Mic = 0x1b, HP = 0x19
*/
/* additional mixers to alc880_three_stack_mixer */
static struct snd_kcontrol_new alc880_five_stack_mixer[] = {
HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT),
{ } /* end */
};
/* channel source setting (6/8 channel selection for 5-stack) */
/* 6ch mode */
static struct hda_verb alc880_fivestack_ch6_init[] = {
/* set line-in to input, mute it */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
{ } /* end */
};
/* 8ch mode */
static struct hda_verb alc880_fivestack_ch8_init[] = {
/* set line-in to output, unmute it */
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ } /* end */
};
static struct hda_channel_mode alc880_fivestack_modes[2] = {
{ 6, alc880_fivestack_ch6_init },
{ 8, alc880_fivestack_ch8_init },
};
/*
* ALC880 6-stack model
*
* DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e),
* Side = 0x05 (0x0f)
* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, Side = 0x17,
* Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b
*/
static hda_nid_t alc880_6st_dac_nids[4] = {
/* front, rear, clfe, rear_surr */
0x02, 0x03, 0x04, 0x05
};
static struct hda_input_mux alc880_6stack_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
/* fixed 8-channels */
static struct hda_channel_mode alc880_sixstack_modes[1] = {
{ 8, NULL },
};
static struct snd_kcontrol_new alc880_six_stack_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
/*
* ALC880 W810 model
*
* W810 has rear IO for:
* Front (DAC 02)
* Surround (DAC 03)
* Center/LFE (DAC 04)
* Digital out (06)
*
* The system also has a pair of internal speakers, and a headphone jack.
* These are both connected to Line2 on the codec, hence to DAC 02.
*
* There is a variable resistor to control the speaker or headphone
* volume. This is a hardware-only device without a software API.
*
* Plugging headphones in will disable the internal speakers. This is
* implemented in hardware, not via the driver using jack sense. In
* a similar fashion, plugging into the rear socket marked "front" will
* disable both the speakers and headphones.
*
* For input, there's a microphone jack, and an "audio in" jack.
* These may not do anything useful with this driver yet, because I
* haven't setup any initialization verbs for these yet...
*/
static hda_nid_t alc880_w810_dac_nids[3] = {
/* front, rear/surround, clfe */
0x02, 0x03, 0x04
};
/* fixed 6 channels */
static struct hda_channel_mode alc880_w810_modes[1] = {
{ 6, NULL }
};
/* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */
static struct snd_kcontrol_new alc880_w810_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
{ } /* end */
};
/*
* Z710V model
*
* DAC: Front = 0x02 (0x0c), HP = 0x03 (0x0d)
* Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?),
* Line = 0x1a
*/
static hda_nid_t alc880_z71v_dac_nids[1] = {
0x02
};
#define ALC880_Z71V_HP_DAC 0x03
/* fixed 2 channels */
static struct hda_channel_mode alc880_2_jack_modes[1] = {
{ 2, NULL }
};
static struct snd_kcontrol_new alc880_z71v_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
/*
* ALC880 F1734 model
*
* DAC: HP = 0x02 (0x0c), Front = 0x03 (0x0d)
* Pin assignment: HP = 0x14, Front = 0x15, Mic = 0x18
*/
static hda_nid_t alc880_f1734_dac_nids[1] = {
0x03
};
#define ALC880_F1734_HP_DAC 0x02
static struct snd_kcontrol_new alc880_f1734_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct hda_input_mux alc880_f1734_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x1 },
{ "CD", 0x4 },
},
};
/*
* ALC880 ASUS model
*
* DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
* Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
* Mic = 0x18, Line = 0x1a
*/
#define alc880_asus_dac_nids alc880_w810_dac_nids /* identical with w810 */
#define alc880_asus_modes alc880_threestack_modes /* 2/6 channel mode */
static struct snd_kcontrol_new alc880_asus_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
/*
* ALC880 ASUS W1V model
*
* DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
* Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
* Mic = 0x18, Line = 0x1a, Line2 = 0x1b
*/
/* additional mixers to alc880_asus_mixer */
static struct snd_kcontrol_new alc880_asus_w1v_mixer[] = {
HDA_CODEC_VOLUME("Line2 Playback Volume", 0x0b, 0x03, HDA_INPUT),
HDA_CODEC_MUTE("Line2 Playback Switch", 0x0b, 0x03, HDA_INPUT),
{ } /* end */
};
/* TCL S700 */
static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0B, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0B, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0B, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0B, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
{ } /* end */
};
/* Uniwill */
static struct snd_kcontrol_new alc880_uniwill_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
static struct snd_kcontrol_new alc880_fujitsu_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
/*
* virtual master controls
*/
/*
* slave controls for virtual master
*/
static const char *alc_slave_vols[] = {
"Front Playback Volume",
"Surround Playback Volume",
"Center Playback Volume",
"LFE Playback Volume",
"Side Playback Volume",
"Headphone Playback Volume",
"Speaker Playback Volume",
"Mono Playback Volume",
"Line-Out Playback Volume",
"PCM Playback Volume",
NULL,
};
static const char *alc_slave_sws[] = {
"Front Playback Switch",
"Surround Playback Switch",
"Center Playback Switch",
"LFE Playback Switch",
"Side Playback Switch",
"Headphone Playback Switch",
"Speaker Playback Switch",
"Mono Playback Switch",
"IEC958 Playback Switch",
NULL,
};
/*
* build control elements
*/
static void alc_free_kctls(struct hda_codec *codec);
/* additional beep mixers; the actual parameters are overwritten at build */
static struct snd_kcontrol_new alc_beep_mixer[] = {
HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT),
HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_INPUT),
{ } /* end */
};
static int alc_build_controls(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err;
int i;
for (i = 0; i < spec->num_mixers; i++) {
err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
if (err < 0)
return err;
}
if (spec->cap_mixer) {
err = snd_hda_add_new_ctls(codec, spec->cap_mixer);
if (err < 0)
return err;
}
if (spec->multiout.dig_out_nid) {
err = snd_hda_create_spdif_out_ctls(codec,
spec->multiout.dig_out_nid);
if (err < 0)
return err;
if (!spec->no_analog) {
err = snd_hda_create_spdif_share_sw(codec,
&spec->multiout);
if (err < 0)
return err;
spec->multiout.share_spdif = 1;
}
}
if (spec->dig_in_nid) {
err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
if (err < 0)
return err;
}
/* create beep controls if needed */
if (spec->beep_amp) {
struct snd_kcontrol_new *knew;
for (knew = alc_beep_mixer; knew->name; knew++) {
struct snd_kcontrol *kctl;
kctl = snd_ctl_new1(knew, codec);
if (!kctl)
return -ENOMEM;
kctl->private_value = spec->beep_amp;
err = snd_hda_ctl_add(codec, kctl);
if (err < 0)
return err;
}
}
/* if we have no master control, let's create it */
if (!spec->no_analog &&
!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
unsigned int vmaster_tlv[4];
snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
vmaster_tlv, alc_slave_vols);
if (err < 0)
return err;
}
if (!spec->no_analog &&
!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL, alc_slave_sws);
if (err < 0)
return err;
}
alc_free_kctls(codec); /* no longer needed */
return 0;
}
/*
* initialize the codec volumes, etc
*/
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc880_volume_init_verbs[] = {
/*
* Unmute ADC0-2 and set the default input to mic-in
*/
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for front
* panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
/*
* Set up output mixers (0x0c - 0x0f)
*/
/* set vol=0 to output mixers */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ }
};
/*
* 3-stack pin configuration:
* front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
*/
static struct hda_verb alc880_pin_3stack_init_verbs[] = {
/*
* preset connection lists of input pins
* 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
*/
{0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
{0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
/*
* Set pin mode and muting
*/
/* set front pin widgets 0x14 for output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Mic2 (as headphone out) for HP output */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Line In pin widget for input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line2 (as front mic) pin widget for input and vref at 80% */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/*
* 5-stack pin configuration:
* front = 0x14, surround = 0x17, clfe = 0x16, mic = 0x18, HP = 0x19,
* line-in/side = 0x1a, f-mic = 0x1b
*/
static struct hda_verb alc880_pin_5stack_init_verbs[] = {
/*
* preset connection lists of input pins
* 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
*/
{0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/side */
/*
* Set pin mode and muting
*/
/* set pin widgets 0x14-0x17 for output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* unmute pins for output (no gain on this amp) */
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Mic2 (as headphone out) for HP output */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Line In pin widget for input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line2 (as front mic) pin widget for input and vref at 80% */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/*
* W810 pin configuration:
* front = 0x14, surround = 0x15, clfe = 0x16, HP = 0x1b
*/
static struct hda_verb alc880_pin_w810_init_verbs[] = {
/* hphone/speaker input selector: front DAC */
{0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{ }
};
/*
* Z71V pin configuration:
* Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?)
*/
static struct hda_verb alc880_pin_z71v_init_verbs[] = {
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/*
* 6-stack pin configuration:
* front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18,
* f-mic = 0x19, line = 0x1a, HP = 0x1b
*/
static struct hda_verb alc880_pin_6stack_init_verbs[] = {
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/*
* Uniwill pin configuration:
* HP = 0x14, InternalSpeaker = 0x15, mic = 0x18, internal mic = 0x19,
* line = 0x1a
*/
static struct hda_verb alc880_uniwill_init_verbs[] = {
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, */
/* {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{ }
};
/*
* Uniwill P53
* HP = 0x14, InternalSpeaker = 0x15, mic = 0x19,
*/
static struct hda_verb alc880_uniwill_p53_init_verbs[] = {
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_DCVOL_EVENT},
{ }
};
static struct hda_verb alc880_beep_init_verbs[] = {
{ 0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) },
{ }
};
/* auto-toggle front mic */
static void alc880_uniwill_mic_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x18, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
}
static void alc880_uniwill_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x16;
}
static void alc880_uniwill_init_hook(struct hda_codec *codec)
{
alc_automute_amp(codec);
alc880_uniwill_mic_automute(codec);
}
static void alc880_uniwill_unsol_event(struct hda_codec *codec,
unsigned int res)
{
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
switch (res >> 28) {
case ALC880_MIC_EVENT:
alc880_uniwill_mic_automute(codec);
break;
default:
alc_automute_amp_unsol_event(codec, res);
break;
}
}
static void alc880_uniwill_p53_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
}
static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
{
unsigned int present;
present = snd_hda_codec_read(codec, 0x21, 0,
AC_VERB_GET_VOLUME_KNOB_CONTROL, 0);
present &= HDA_AMP_VOLMASK;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0,
HDA_AMP_VOLMASK, present);
snd_hda_codec_amp_stereo(codec, 0x0d, HDA_OUTPUT, 0,
HDA_AMP_VOLMASK, present);
}
static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
unsigned int res)
{
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
if ((res >> 28) == ALC880_DCVOL_EVENT)
alc880_uniwill_p53_dcvol_automute(codec);
else
alc_automute_amp_unsol_event(codec, res);
}
/*
* F1734 pin configuration:
* HP = 0x14, speaker-out = 0x15, mic = 0x18
*/
static struct hda_verb alc880_pin_f1734_init_verbs[] = {
{0x07, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
{0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_DCVOL_EVENT},
{ }
};
/*
* ASUS pin configuration:
* HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a
*/
static struct hda_verb alc880_pin_asus_init_verbs[] = {
{0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
{0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/* Enable GPIO mask and set output */
#define alc880_gpio1_init_verbs alc_gpio1_init_verbs
#define alc880_gpio2_init_verbs alc_gpio2_init_verbs
#define alc880_gpio3_init_verbs alc_gpio3_init_verbs
/* Clevo m520g init */
static struct hda_verb alc880_pin_clevo_init_verbs[] = {
/* headphone output */
{0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
/* line-out */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Line-in */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* CD */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mic1 (rear panel) */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mic2 (front panel) */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* headphone */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* change to EAPD mode */
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3060},
{ }
};
static struct hda_verb alc880_pin_tcl_S700_init_verbs[] = {
/* change to EAPD mode */
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3060},
/* Headphone output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Front output*/
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Line In pin widget for input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
/* change to EAPD mode */
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3070},
{ }
};
/*
* 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[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x06, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x06, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x07, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x07, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
static struct hda_verb alc880_lg_init_verbs[] = {
/* set capture source to mic-in */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* mute all amp mixer inputs */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
/* line-in to input */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* built-in mic */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* speaker-out */
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* mic-in to input */
{0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* HP-out */
{0x13, AC_VERB_SET_CONNECT_SEL, 0x03},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* jack sense */
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc880_lg_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x17;
}
/*
* LG LW20
*
* Pin assignment:
* Speaker-out: 0x14
* Mic-In: 0x18
* Built-in Mic-In: 0x19
* Line-In: 0x1b
* HP-Out: 0x1a
* SPDIF-Out: 0x1e
*/
static struct hda_input_mux alc880_lg_lw_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x0 },
{ "Internal Mic", 0x1 },
{ "Line In", 0x2 },
},
};
#define alc880_lg_lw_modes alc880_threestack_modes
static struct snd_kcontrol_new alc880_lg_lw_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
static struct hda_verb alc880_lg_lw_init_verbs[] = {
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
{0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
/* set capture source to mic-in */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
/* speaker-out */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* HP-out */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* mic-in to input */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* built-in mic */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* jack sense */
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc880_lg_lw_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
}
static struct snd_kcontrol_new alc880_medion_rim_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, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Internal Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct hda_input_mux alc880_medion_rim_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x0 },
{ "Internal Mic", 0x1 },
},
};
static struct hda_verb alc880_medion_rim_init_verbs[] = {
{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* 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_MUTE},
/* Internal Speaker */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3060},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc880_medion_rim_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
alc_automute_amp(codec);
/* toggle EAPD */
if (spec->jack_present)
snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0);
else
snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2);
}
static void alc880_medion_rim_unsol_event(struct hda_codec *codec,
unsigned int res)
{
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
if ((res >> 28) == ALC880_HP_EVENT)
alc880_medion_rim_automute(codec);
}
static void alc880_medion_rim_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x1b;
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
static struct hda_amp_list alc880_loopbacks[] = {
{ 0x0b, HDA_INPUT, 0 },
{ 0x0b, HDA_INPUT, 1 },
{ 0x0b, HDA_INPUT, 2 },
{ 0x0b, HDA_INPUT, 3 },
{ 0x0b, HDA_INPUT, 4 },
{ } /* end */
};
static struct hda_amp_list alc880_lg_loopbacks[] = {
{ 0x0b, HDA_INPUT, 1 },
{ 0x0b, HDA_INPUT, 6 },
{ 0x0b, HDA_INPUT, 7 },
{ } /* end */
};
#endif
/*
* Common callbacks
*/
static int alc_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int i;
alc_fix_pll(codec);
alc_auto_init_amp(codec, spec->init_amp);
for (i = 0; i < spec->num_init_verbs; i++)
snd_hda_sequence_write(codec, spec->init_verbs[i]);
if (spec->init_hook)
spec->init_hook(codec);
return 0;
}
static void alc_unsol_event(struct hda_codec *codec, unsigned int res)
{
struct alc_spec *spec = codec->spec;
if (spec->unsol_event)
spec->unsol_event(codec, res);
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
static int alc_check_power_status(struct hda_codec *codec, hda_nid_t nid)
{
struct alc_spec *spec = codec->spec;
return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
}
#endif
/*
* Analog playback callbacks
*/
static int alc880_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
hinfo);
}
static int alc880_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
stream_tag, format, substream);
}
static int alc880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
}
/*
* Digital out
*/
static int alc880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_dig_open(codec, &spec->multiout);
}
static int alc880_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
return snd_hda_multi_out_dig_prepare(codec, &spec->multiout,
stream_tag, format, substream);
}
static int alc880_dig_playback_pcm_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_dig_cleanup(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_alt_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 + 1],
stream_tag, 0, format);
return 0;
}
static int alc880_alt_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_cleanup_stream(codec,
spec->adc_nids[substream->number + 1]);
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 = 1,
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
};
static struct hda_pcm_stream alc880_pcm_analog_alt_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
};
static struct hda_pcm_stream alc880_pcm_analog_alt_capture = {
.substreams = 2, /* can be overridden */
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
.ops = {
.prepare = alc880_alt_capture_pcm_prepare,
.cleanup = alc880_alt_capture_pcm_cleanup
},
};
static struct hda_pcm_stream alc880_pcm_digital_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
.ops = {
.open = alc880_dig_playback_pcm_open,
.close = alc880_dig_playback_pcm_close,
.prepare = alc880_dig_playback_pcm_prepare,
.cleanup = alc880_dig_playback_pcm_cleanup
},
};
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_stream = {
.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;
if (spec->no_analog)
goto skip_analog;
snprintf(spec->stream_name_analog, sizeof(spec->stream_name_analog),
"%s Analog", codec->chip_name);
info->name = spec->stream_name_analog;
if (spec->stream_analog_playback) {
if (snd_BUG_ON(!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) {
if (snd_BUG_ON(!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;
}
}
}
skip_analog:
/* SPDIF for stream index #1 */
if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
snprintf(spec->stream_name_digital,
sizeof(spec->stream_name_digital),
"%s Digital", codec->chip_name);
codec->num_pcms = 2;
codec->slave_dig_outs = spec->multiout.slave_dig_outs;
info = spec->pcm_rec + 1;
info->name = spec->stream_name_digital;
if (spec->dig_out_type)
info->pcm_type = spec->dig_out_type;
else
info->pcm_type = HDA_PCM_TYPE_SPDIF;
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;
}
/* FIXME: do we need this for all Realtek codec models? */
codec->spdif_status_reset = 1;
}
if (spec->no_analog)
return 0;
/* If the use of more than one ADC is requested for the current
* model, configure a second analog capture-only PCM.
*/
/* Additional Analaog capture for index #2 */
if ((spec->alt_dac_nid && spec->stream_analog_alt_playback) ||
(spec->num_adc_nids > 1 && spec->stream_analog_alt_capture)) {
codec->num_pcms = 3;
info = spec->pcm_rec + 2;
info->name = spec->stream_name_analog;
if (spec->alt_dac_nid) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
*spec->stream_analog_alt_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
spec->alt_dac_nid;
} else {
info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
alc_pcm_null_stream;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0;
}
if (spec->num_adc_nids > 1) {
info->stream[SNDRV_PCM_STREAM_CAPTURE] =
*spec->stream_analog_alt_capture;
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
spec->adc_nids[1];
info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams =
spec->num_adc_nids - 1;
} else {
info->stream[SNDRV_PCM_STREAM_CAPTURE] =
alc_pcm_null_stream;
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = 0;
}
}
return 0;
}
static void alc_free_kctls(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
if (spec->kctls.list) {
struct snd_kcontrol_new *kctl = spec->kctls.list;
int i;
for (i = 0; i < spec->kctls.used; i++)
kfree(kctl[i].name);
}
snd_array_free(&spec->kctls);
}
static void alc_free(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
if (!spec)
return;
alc_free_kctls(codec);
kfree(spec);
snd_hda_detach_beep_device(codec);
}
#ifdef SND_HDA_NEEDS_RESUME
static int alc_resume(struct hda_codec *codec)
{
codec->patch_ops.init(codec);
snd_hda_codec_resume_amp(codec);
snd_hda_codec_resume_cache(codec);
return 0;
}
#endif
/*
*/
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 SND_HDA_NEEDS_RESUME
.resume = alc_resume,
#endif
#ifdef CONFIG_SND_HDA_POWER_SAVE
.check_power_status = alc_check_power_status,
#endif
};
/*
* Test configuration for debugging
*
* Almost all inputs/outputs are enabled. I/O pins can be configured via
* enum controls.
*/
#ifdef CONFIG_SND_DEBUG
static hda_nid_t alc880_test_dac_nids[4] = {
0x02, 0x03, 0x04, 0x05
};
static struct hda_input_mux alc880_test_capture_source = {
.num_items = 7,
.items = {
{ "In-1", 0x0 },
{ "In-2", 0x1 },
{ "In-3", 0x2 },
{ "In-4", 0x3 },
{ "CD", 0x4 },
{ "Front", 0x5 },
{ "Surround", 0x6 },
},
};
static struct hda_channel_mode alc880_test_modes[4] = {
{ 2, NULL },
{ 4, NULL },
{ 6, NULL },
{ 8, NULL },
};
static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = {
"N/A", "Line Out", "HP Out",
"In Hi-Z", "In 50%", "In Grd", "In 80%", "In 100%"
};
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 8;
if (uinfo->value.enumerated.item >= 8)
uinfo->value.enumerated.item = 7;
strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
return 0;
}
static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
unsigned int pin_ctl, item = 0;
pin_ctl = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
if (pin_ctl & AC_PINCTL_OUT_EN) {
if (pin_ctl & AC_PINCTL_HP_EN)
item = 2;
else
item = 1;
} else if (pin_ctl & AC_PINCTL_IN_EN) {
switch (pin_ctl & AC_PINCTL_VREFEN) {
case AC_PINCTL_VREF_HIZ: item = 3; break;
case AC_PINCTL_VREF_50: item = 4; break;
case AC_PINCTL_VREF_GRD: item = 5; break;
case AC_PINCTL_VREF_80: item = 6; break;
case AC_PINCTL_VREF_100: item = 7; break;
}
}
ucontrol->value.enumerated.item[0] = item;
return 0;
}
static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
static unsigned int ctls[] = {
0, AC_PINCTL_OUT_EN, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_50,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_80,
AC_PINCTL_IN_EN | AC_PINCTL_VREF_100,
};
unsigned int old_ctl, new_ctl;
old_ctl = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
new_ctl = ctls[ucontrol->value.enumerated.item[0]];
if (old_ctl != new_ctl) {
int val;
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
new_ctl);
val = ucontrol->value.enumerated.item[0] >= 3 ?
HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, val);
return 1;
}
return 0;
}
static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = {
"Front", "Surround", "CLFE", "Side"
};
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 4;
if (uinfo->value.enumerated.item >= 4)
uinfo->value.enumerated.item = 3;
strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
return 0;
}
static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
unsigned int sel;
sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0);
ucontrol->value.enumerated.item[0] = sel & 3;
return 0;
}
static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
unsigned int sel;
sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3;
if (ucontrol->value.enumerated.item[0] != sel) {
sel = ucontrol->value.enumerated.item[0] & 3;
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_CONNECT_SEL, sel);
return 1;
}
return 0;
}
#define PIN_CTL_TEST(xname,nid) { \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
.info = alc_test_pin_ctl_info, \
.get = alc_test_pin_ctl_get, \
.put = alc_test_pin_ctl_put, \
.private_value = nid \
}
#define PIN_SRC_TEST(xname,nid) { \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
.info = alc_test_pin_src_info, \
.get = alc_test_pin_src_get, \
.put = alc_test_pin_src_put, \
.private_value = nid \
}
static struct snd_kcontrol_new alc880_test_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CLFE Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_BIND_MUTE("CLFE Playback Switch", 0x0e, 2, HDA_INPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
PIN_CTL_TEST("Front Pin Mode", 0x14),
PIN_CTL_TEST("Surround Pin Mode", 0x15),
PIN_CTL_TEST("CLFE Pin Mode", 0x16),
PIN_CTL_TEST("Side Pin Mode", 0x17),
PIN_CTL_TEST("In-1 Pin Mode", 0x18),
PIN_CTL_TEST("In-2 Pin Mode", 0x19),
PIN_CTL_TEST("In-3 Pin Mode", 0x1a),
PIN_CTL_TEST("In-4 Pin Mode", 0x1b),
PIN_SRC_TEST("In-1 Pin Source", 0x18),
PIN_SRC_TEST("In-2 Pin Source", 0x19),
PIN_SRC_TEST("In-3 Pin Source", 0x1a),
PIN_SRC_TEST("In-4 Pin Source", 0x1b),
HDA_CODEC_VOLUME("In-1 Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("In-1 Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("In-2 Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("In-2 Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("In-3 Playback Volume", 0x0b, 0x2, HDA_INPUT),
HDA_CODEC_MUTE("In-3 Playback Switch", 0x0b, 0x2, HDA_INPUT),
HDA_CODEC_VOLUME("In-4 Playback Volume", 0x0b, 0x3, HDA_INPUT),
HDA_CODEC_MUTE("In-4 Playback Switch", 0x0b, 0x3, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x4, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x4, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
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 const char *alc880_models[ALC880_MODEL_LAST] = {
[ALC880_3ST] = "3stack",
[ALC880_TCL_S700] = "tcl",
[ALC880_3ST_DIG] = "3stack-digout",
[ALC880_CLEVO] = "clevo",
[ALC880_5ST] = "5stack",
[ALC880_5ST_DIG] = "5stack-digout",
[ALC880_W810] = "w810",
[ALC880_Z71V] = "z71v",
[ALC880_6ST] = "6stack",
[ALC880_6ST_DIG] = "6stack-digout",
[ALC880_ASUS] = "asus",
[ALC880_ASUS_W1V] = "asus-w1v",
[ALC880_ASUS_DIG] = "asus-dig",
[ALC880_ASUS_DIG2] = "asus-dig2",
[ALC880_UNIWILL_DIG] = "uniwill",
[ALC880_UNIWILL_P53] = "uniwill-p53",
[ALC880_FUJITSU] = "fujitsu",
[ALC880_F1734] = "F1734",
[ALC880_LG] = "lg",
[ALC880_LG_LW] = "lg-lw",
[ALC880_MEDION_RIM] = "medion",
#ifdef CONFIG_SND_DEBUG
[ALC880_TEST] = "test",
#endif
[ALC880_AUTO] = "auto",
};
static struct snd_pci_quirk alc880_cfg_tbl[] = {
SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_W810),
SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_6ST),
SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_3ST_DIG),
SND_PCI_QUIRK(0x1025, 0x0077, "ULI", ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1025, 0x0078, "ULI", ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1025, 0x0087, "ULI", ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG),
SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST),
SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG),
SND_PCI_QUIRK(0x103c, 0x2a09, "HP", ALC880_5ST),
SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V),
SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x1113, "ASUS", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x1123, "ASUS", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x1173, "ASUS", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_Z71V),
/* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */
SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x814e, "ASUS P5GD1 w/SPDIF", ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG),
SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST),
SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST),
SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_ASUS), /* default ASUS */
SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST),
SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST),
SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST),
SND_PCI_QUIRK(0x107b, 0x3033, "Gateway", ALC880_5ST),
SND_PCI_QUIRK(0x107b, 0x4039, "Gateway", ALC880_5ST),
SND_PCI_QUIRK(0x1297, 0xc790, "Shuttle ST20G5", ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG),
SND_PCI_QUIRK(0x1558, 0x0520, "Clevo m520G", ALC880_CLEVO),
SND_PCI_QUIRK(0x1558, 0x0660, "Clevo m655n", ALC880_CLEVO),
SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2),
SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG),
SND_PCI_QUIRK(0x1584, 0x9054, "Uniwlll", ALC880_F1734),
SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL),
SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53),
SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810),
SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_MEDION_RIM),
SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734),
SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU),
SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL),
SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU),
SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW),
SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG),
SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_LG),
SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_LG_LW),
SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700),
SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */
SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG),
SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x8086, 0xd400, "Intel mobo", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x8086, 0xd401, "Intel mobo", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_3ST_DIG),
SND_PCI_QUIRK(0x8086, 0xe224, "Intel mobo", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_3ST_DIG),
SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_3ST_DIG),
SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG),
/* default Intel */
SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_3ST),
SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG),
SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG),
{}
};
/*
* 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,
.need_dac_fix = 1,
.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,
.need_dac_fix = 1,
.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,
.adc_nids = alc880_adc_nids_alt, /* FIXME: correct? */
.num_adc_nids = 1, /* single ADC */
.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_f1734_capture_source,
.unsol_event = alc880_uniwill_p53_unsol_event,
.setup = alc880_uniwill_p53_setup,
.init_hook = alc_automute_amp,
},
[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,
.need_dac_fix = 1,
.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,
.need_dac_fix = 1,
.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,
.need_dac_fix = 1,
.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,
.need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_UNIWILL_DIG] = {
.mixers = { alc880_asus_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,
.need_dac_fix = 1,
.input_mux = &alc880_capture_source,
},
[ALC880_UNIWILL] = {
.mixers = { alc880_uniwill_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_uniwill_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_threestack_modes),
.channel_mode = alc880_threestack_modes,
.need_dac_fix = 1,
.input_mux = &alc880_capture_source,
.unsol_event = alc880_uniwill_unsol_event,
.setup = alc880_uniwill_setup,
.init_hook = alc880_uniwill_init_hook,
},
[ALC880_UNIWILL_P53] = {
.mixers = { alc880_uniwill_p53_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_uniwill_p53_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
.dac_nids = alc880_asus_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc880_w810_modes),
.channel_mode = alc880_threestack_modes,
.input_mux = &alc880_capture_source,
.unsol_event = alc880_uniwill_p53_unsol_event,
.setup = alc880_uniwill_p53_setup,
.init_hook = alc_automute_amp,
},
[ALC880_FUJITSU] = {
.mixers = { alc880_fujitsu_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_uniwill_p53_init_verbs,
alc880_beep_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_2_jack_modes),
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_capture_source,
.unsol_event = alc880_uniwill_p53_unsol_event,
.setup = alc880_uniwill_p53_setup,
.init_hook = alc_automute_amp,
},
[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,
.need_dac_fix = 1,
.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,
.need_dac_fix = 1,
.input_mux = &alc880_lg_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc880_lg_setup,
.init_hook = alc_automute_amp,
#ifdef CONFIG_SND_HDA_POWER_SAVE
.loopbacks = alc880_lg_loopbacks,
#endif
},
[ALC880_LG_LW] = {
.mixers = { alc880_lg_lw_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_lg_lw_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_lg_lw_modes),
.channel_mode = alc880_lg_lw_modes,
.input_mux = &alc880_lg_lw_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc880_lg_lw_setup,
.init_hook = alc_automute_amp,
},
[ALC880_MEDION_RIM] = {
.mixers = { alc880_medion_rim_mixer },
.init_verbs = { alc880_volume_init_verbs,
alc880_medion_rim_init_verbs,
alc_gpio2_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_2_jack_modes),
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_medion_rim_capture_source,
.unsol_event = alc880_medion_rim_unsol_event,
.setup = alc880_medion_rim_setup,
.init_hook = alc880_medion_rim_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
*/
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;
snd_array_init(&spec->kctls, sizeof(*knew), 32);
knew = snd_array_new(&spec->kctls);
if (!knew)
return -ENOMEM;
*knew = alc880_control_templates[type];
knew->name = kstrdup(name, GFP_KERNEL);
if (!knew->name)
return -ENOMEM;
knew->private_value = val;
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_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 */
err = add_control(spec, ALC_CTL_WIDGET_VOL,
"Center Playback Volume",
HDA_COMPOSE_AMP_VAL(nid, 1, 0,
HDA_OUTPUT));
if (err < 0)
return err;
err = add_control(spec, ALC_CTL_WIDGET_VOL,
"LFE Playback Volume",
HDA_COMPOSE_AMP_VAL(nid, 2, 0,
HDA_OUTPUT));
if (err < 0)
return err;
err = add_control(spec, ALC_CTL_BIND_MUTE,
"Center Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 1, 2,
HDA_INPUT));
if (err < 0)
return err;
err = add_control(spec, ALC_CTL_BIND_MUTE,
"LFE Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 2, 2,
HDA_INPUT));
if (err < 0)
return err;
} else {
const char *pfx;
if (cfg->line_outs == 1 &&
cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
pfx = "Speaker";
else
pfx = chname[i];
sprintf(name, "%s Playback Volume", pfx);
err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 0,
HDA_OUTPUT));
if (err < 0)
return err;
sprintf(name, "%s Playback Switch", pfx);
err = add_control(spec, ALC_CTL_BIND_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 2,
HDA_INPUT));
if (err < 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);
err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
if (err < 0)
return err;
sprintf(name, "%s Playback Switch", pfx);
err = add_control(spec, ALC_CTL_BIND_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT));
if (err < 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);
err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
if (err < 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);
err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT));
if (err < 0)
return err;
sprintf(name, "%s Playback Switch", ctlname);
err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT));
if (err < 0)
return err;
return 0;
}
static int alc_is_input_pin(struct hda_codec *codec, hda_nid_t nid)
{
unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
return (pincap & AC_PINCAP_IN) != 0;
}
/* create playback/capture controls for input pins */
static int alc_auto_create_input_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg,
hda_nid_t mixer,
hda_nid_t cap1, hda_nid_t cap2)
{
struct alc_spec *spec = codec->spec;
struct hda_input_mux *imux = &spec->private_imux[0];
int i, err, idx;
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t pin;
pin = cfg->input_pins[i];
if (!alc_is_input_pin(codec, pin))
continue;
if (mixer) {
idx = get_connection_index(codec, mixer, pin);
if (idx >= 0) {
err = new_analog_input(spec, pin,
auto_pin_cfg_labels[i],
idx, mixer);
if (err < 0)
return err;
}
}
if (!cap1)
continue;
idx = get_connection_index(codec, cap1, pin);
if (idx < 0 && cap2)
idx = get_connection_index(codec, cap2, pin);
if (idx >= 0) {
imux->items[imux->num_items].label =
auto_pin_cfg_labels[i];
imux->items[imux->num_items].index = idx;
imux->num_items++;
}
}
return 0;
}
static int alc880_auto_create_input_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x08, 0x09);
}
static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid,
unsigned int pin_type)
{
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
pin_type);
/* unmute pin */
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_UNMUTE);
}
static void alc880_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
int dac_idx)
{
alc_set_pin_output(codec, nid, pin_type);
/* 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 int get_pin_type(int line_out_type)
{
if (line_out_type == AUTO_PIN_HP_OUT)
return PIN_HP;
else
return PIN_OUT;
}
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];
int pin_type = get_pin_type(spec->autocfg.line_out_type);
alc880_auto_set_output_and_unmute(codec, nid, pin_type, 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_pins[0];
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 (alc_is_input_pin(codec, nid)) {
alc_set_input_pin(codec, nid, i);
if (nid != ALC880_PIN_CD_NID &&
(get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
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 i, err;
static hda_nid_t alc880_ignore[] = { 0x1d, 0 };
err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc880_ignore);
if (err < 0)
return err;
if (!spec->autocfg.line_outs)
return 0; /* can't find valid BIOS pin config */
err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
if (err < 0)
return err;
err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
err = alc880_auto_create_extra_out(spec,
spec->autocfg.speaker_pins[0],
"Speaker");
if (err < 0)
return err;
err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
"Headphone");
if (err < 0)
return err;
err = alc880_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
/* check multiple SPDIF-out (for recent codecs) */
for (i = 0; i < spec->autocfg.dig_outs; i++) {
hda_nid_t dig_nid;
err = snd_hda_get_connections(codec,
spec->autocfg.dig_out_pins[i],
&dig_nid, 1);
if (err < 0)
continue;
if (!i)
spec->multiout.dig_out_nid = dig_nid;
else {
spec->multiout.slave_dig_outs = spec->slave_dig_outs;
spec->slave_dig_outs[i - 1] = dig_nid;
if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
break;
}
}
if (spec->autocfg.dig_in_pin)
spec->dig_in_nid = ALC880_DIGIN_NID;
if (spec->kctls.list)
add_mixer(spec, spec->kctls.list);
add_verb(spec, 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 10:47:09 +00:00
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux[0];
alc_ssid_check(codec, 0x15, 0x1b, 0x14);
return 1;
}
/* additional initialization for auto-configuration model */
static void alc880_auto_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
alc880_auto_init_multi_out(codec);
alc880_auto_init_extra_out(codec);
alc880_auto_init_analog_input(codec);
if (spec->unsol_event)
alc_inithook(codec);
}
/* check the ADC/MUX contains all input pins; some ADC/MUX contains only
* one of two digital mic pins, e.g. on ALC272
*/
static void fixup_automic_adc(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < spec->num_adc_nids; i++) {
hda_nid_t cap = spec->capsrc_nids ?
spec->capsrc_nids[i] : spec->adc_nids[i];
int iidx, eidx;
iidx = get_connection_index(codec, cap, spec->int_mic.pin);
if (iidx < 0)
continue;
eidx = get_connection_index(codec, cap, spec->ext_mic.pin);
if (eidx < 0)
continue;
spec->int_mic.mux_idx = iidx;
spec->ext_mic.mux_idx = eidx;
if (spec->capsrc_nids)
spec->capsrc_nids += i;
spec->adc_nids += i;
spec->num_adc_nids = 1;
return;
}
snd_printd(KERN_INFO "hda_codec: %s: "
"No ADC/MUX containing both 0x%x and 0x%x pins\n",
codec->chip_name, spec->int_mic.pin, spec->ext_mic.pin);
spec->auto_mic = 0; /* disable auto-mic to be sure */
}
static void set_capture_mixer(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
static struct snd_kcontrol_new *caps[2][3] = {
{ alc_capture_mixer_nosrc1,
alc_capture_mixer_nosrc2,
alc_capture_mixer_nosrc3 },
{ alc_capture_mixer1,
alc_capture_mixer2,
alc_capture_mixer3 },
};
if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) {
int mux;
if (spec->auto_mic) {
mux = 0;
fixup_automic_adc(codec);
} else if (spec->input_mux && spec->input_mux->num_items > 1)
mux = 1;
else
mux = 0;
spec->cap_mixer = caps[mux][spec->num_adc_nids - 1];
}
}
#define set_beep_amp(spec, nid, idx, dir) \
((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir))
/*
* 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_MODEL_LAST,
alc880_models,
alc880_cfg_tbl);
if (board_config < 0) {
printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
codec->chip_name);
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;
}
}
err = snd_hda_attach_beep_device(codec, 0x1);
if (err < 0) {
alc_free(codec);
return err;
}
if (board_config != ALC880_AUTO)
setup_preset(codec, &alc880_presets[board_config]);
spec->stream_analog_playback = &alc880_pcm_analog_playback;
spec->stream_analog_capture = &alc880_pcm_analog_capture;
spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture;
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]);
/* get type */
wcap = get_wcaps_type(wcap);
if (wcap != AC_WID_AUD_IN) {
spec->adc_nids = alc880_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids_alt);
} else {
spec->adc_nids = alc880_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids);
}
}
set_capture_mixer(codec);
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
spec->vmaster_nid = 0x0c;
codec->patch_ops = alc_patch_ops;
if (board_config == ALC880_AUTO)
spec->init_hook = alc880_auto_init;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
spec->loopback.amplist = alc880_loopbacks;
#endif
codec->proc_widget_hook = print_realtek_coef;
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,
};
/* 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 10:47:09 +00:00
* 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 10:47:09 +00:00
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 10:47:09 +00:00
{
.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 10:47:09 +00:00
/* 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 10:47:09 +00:00
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 },
},
},
};
/* Maxdata Favorit 100XS */
static struct hda_input_mux alc260_favorit100_capture_sources[2] = {
{
.num_items = 2,
.items = {
{ "Line/Mic", 0x0 },
{ "CD", 0x4 },
},
},
{
.num_items = 3,
.items = {
{ "Line/Mic", 0x0 },
{ "CD", 0x4 },
{ "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 */
};
/* update HP, line and mono out pins according to the master switch */
static void alc260_hp_master_update(struct hda_codec *codec,
hda_nid_t hp, hda_nid_t line,
hda_nid_t mono)
{
struct alc_spec *spec = codec->spec;
unsigned int val = spec->master_sw ? PIN_HP : 0;
/* change HP and line-out pins */
snd_hda_codec_write(codec, hp, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
val);
snd_hda_codec_write(codec, line, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
val);
/* mono (speaker) depending on the HP jack sense */
val = (val && !spec->jack_present) ? PIN_OUT : 0;
snd_hda_codec_write(codec, mono, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
val);
}
static int alc260_hp_master_sw_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;
*ucontrol->value.integer.value = spec->master_sw;
return 0;
}
static int alc260_hp_master_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
int val = !!*ucontrol->value.integer.value;
hda_nid_t hp, line, mono;
if (val == spec->master_sw)
return 0;
spec->master_sw = val;
hp = (kcontrol->private_value >> 16) & 0xff;
line = (kcontrol->private_value >> 8) & 0xff;
mono = kcontrol->private_value & 0xff;
alc260_hp_master_update(codec, hp, line, mono);
return 1;
}
static struct snd_kcontrol_new alc260_hp_output_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.info = snd_ctl_boolean_mono_info,
.get = alc260_hp_master_sw_get,
.put = alc260_hp_master_sw_put,
.private_value = (0x0f << 16) | (0x10 << 8) | 0x11
},
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("Speaker Playback Volume", 0x0a, 1, 0x0,
HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2, HDA_INPUT),
{ } /* end */
};
static struct hda_verb alc260_hp_unsol_verbs[] = {
{0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{},
};
static void alc260_hp_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int present;
present = snd_hda_codec_read(codec, 0x10, 0,
AC_VERB_GET_PIN_SENSE, 0);
spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
alc260_hp_master_update(codec, 0x0f, 0x10, 0x11);
}
static void alc260_hp_unsol_event(struct hda_codec *codec, unsigned int res)
{
if ((res >> 26) == ALC880_HP_EVENT)
alc260_hp_automute(codec);
}
static struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.info = snd_ctl_boolean_mono_info,
.get = alc260_hp_master_sw_get,
.put = alc260_hp_master_sw_put,
.private_value = (0x15 << 16) | (0x10 << 8) | 0x11
},
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("Speaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT),
{ } /* end */
};
static struct hda_bind_ctls alc260_dc7600_bind_master_vol = {
.ops = &snd_hda_bind_vol,
.values = {
HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x0a, 3, 0, HDA_OUTPUT),
0
},
};
static struct hda_bind_ctls alc260_dc7600_bind_switch = {
.ops = &snd_hda_bind_sw,
.values = {
HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
0
},
};
static struct snd_kcontrol_new alc260_hp_dc7600_mixer[] = {
HDA_BIND_VOL("Master Playback Volume", &alc260_dc7600_bind_master_vol),
HDA_BIND_SW("LineOut Playback Switch", &alc260_dc7600_bind_switch),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x0f, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x10, 0x0, HDA_OUTPUT),
{ } /* end */
};
static struct hda_verb alc260_hp_3013_unsol_verbs[] = {
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{},
};
static void alc260_hp_3013_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int present;
present = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0);
spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
alc260_hp_master_update(codec, 0x15, 0x10, 0x11);
}
static void alc260_hp_3013_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC880_HP_EVENT)
alc260_hp_3013_automute(codec);
}
static void alc260_hp_3012_automute(struct hda_codec *codec)
{
unsigned int present, bits;
present = snd_hda_codec_read(codec, 0x10, 0,
AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE;
bits = present ? 0 : PIN_OUT;
snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
bits);
snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
bits);
snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
bits);
}
static void alc260_hp_3012_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC880_HP_EVENT)
alc260_hp_3012_automute(codec);
}
/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12,
[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 10:47:09 +00:00
* 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("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("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 10:47:09 +00:00
/* 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.
*
* The C20x Tablet series have a mono internal speaker which is controlled
* via the chip's Mono sum widget and pin complex, so include the necessary
* controls for such models. On models without a "mono speaker" the control
* won't do anything.
[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 10:47:09 +00:00
*/
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 10:47:09 +00:00
ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2,
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("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),
{ } /* end */
};
/* Maxdata Favorit 100XS: one output and one input (0x12) jack
*/
static struct snd_kcontrol_new alc260_favorit100_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
{ } /* end */
};
/* Packard bell V7900 ALC260 pin usage: HP = 0x0f, Mic jack = 0x12,
* Line In jack = 0x14, CD audio = 0x16, pc beep = 0x17.
*/
static struct snd_kcontrol_new alc260_will_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, 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("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
{ } /* end */
};
/* Replacer 672V ALC260 pin usage: Mic jack = 0x12,
* Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f.
*/
static struct snd_kcontrol_new alc260_replacer_672v_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, 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("ATAPI Mic Playback Volume", 0x07, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("ATATI Mic Playback Switch", 0x07, 0x1, HDA_INPUT),
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),
{ } /* 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 analog inputs */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* 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)},
{ }
};
#if 0 /* should be identical with alc260_init_verbs? */
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
*/
/* mute analog inputs */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* 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))},
{ }
};
#endif
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
*/
/* mute analog inputs */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* 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 10:47:09 +00:00
* 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},
/* Some Acers (eg: C20x Tablets) use Mono pin for internal speaker */
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* 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)},
{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 mono pin widget amp output (no equiv mixer ctrl) */
{0x11, 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 10:47:09 +00:00
* 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 10:47:09 +00:00
{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 */
{ }
};
/* Initialisation sequence for Maxdata Favorit 100XS
* (adapted from Acer init verbs).
*/
static struct hda_verb alc260_favorit100_init_verbs[] = {
/* GPIO 0 enables the output 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},
/* Line/Mic input jack is connected to Mic1 pin */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
/* 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)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
{0x14, 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)},
/* 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
* set ADC connection to mic to match ALSA's default state.
*/
{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 */
{ }
};
static struct hda_verb alc260_will_verbs[] = {
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x0b, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
{0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
{0x1a, AC_VERB_SET_PROC_COEF, 0x3040},
{}
};
static struct hda_verb alc260_replacer_672v_verbs[] = {
{0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
{0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
{0x1a, AC_VERB_SET_PROC_COEF, 0x3050},
{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
{0x01, AC_VERB_SET_GPIO_DATA, 0x00},
{0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
/* toggle speaker-output according to the hp-jack state */
static void alc260_replacer_672v_automute(struct hda_codec *codec)
{
unsigned int present;
/* speaker --> GPIO Data 0, hp or spdif --> GPIO data 1 */
present = snd_hda_codec_read(codec, 0x0f, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
if (present) {
snd_hda_codec_write_cache(codec, 0x01, 0,
AC_VERB_SET_GPIO_DATA, 1);
snd_hda_codec_write_cache(codec, 0x0f, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
PIN_HP);
} else {
snd_hda_codec_write_cache(codec, 0x01, 0,
AC_VERB_SET_GPIO_DATA, 0);
snd_hda_codec_write_cache(codec, 0x0f, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
PIN_OUT);
}
}
static void alc260_replacer_672v_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC880_HP_EVENT)
alc260_replacer_672v_automute(codec);
}
static struct hda_verb alc260_hp_dc7600_verbs[] = {
{0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
/* 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 10:47:09 +00:00
/* For testing the ALC260, each input MUX needs its own definition since
* the signal assignments are different. This assumes that the first ADC
[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 10:47:09 +00:00
* 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 10:47:09 +00:00
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 10:47:09 +00:00
/* 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("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),
/* A switch allowing EAPD to be enabled. Some laptops seem to use
* this output to turn on an external amplifier.
*/
ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
{ } /* 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
#define alc260_pcm_analog_playback alc880_pcm_analog_alt_playback
#define alc260_pcm_analog_capture alc880_pcm_analog_capture
#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, int *vol_bits)
{
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 */
if (!(*vol_bits & (1 << nid_vol))) {
/* first control for the volume widget */
snprintf(name, sizeof(name), "%s Playback Volume", pfx);
err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val);
if (err < 0)
return err;
*vol_bits |= (1 << nid_vol);
}
snprintf(name, sizeof(name), "%s Playback Switch", pfx);
err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, sw_val);
if (err < 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;
int vols = 0;
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) {
const char *pfx;
if (!cfg->speaker_pins[0] && !cfg->hp_pins[0])
pfx = "Master";
else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
pfx = "Speaker";
else
pfx = "Front";
err = alc260_add_playback_controls(spec, nid, pfx, &vols);
if (err < 0)
return err;
}
nid = cfg->speaker_pins[0];
if (nid) {
err = alc260_add_playback_controls(spec, nid, "Speaker", &vols);
if (err < 0)
return err;
}
nid = cfg->hp_pins[0];
if (nid) {
err = alc260_add_playback_controls(spec, nid, "Headphone",
&vols);
if (err < 0)
return err;
}
return 0;
}
/* create playback/capture controls for input pins */
static int alc260_auto_create_input_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
return alc_auto_create_input_ctls(codec, cfg, 0x07, 0x04, 0x05);
}
static void alc260_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
int sel_idx)
{
alc_set_pin_output(codec, nid, pin_type);
/* 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) {
int pin_type = get_pin_type(spec->autocfg.line_out_type);
alc260_auto_set_output_and_unmute(codec, nid, pin_type, 0);
}
nid = spec->autocfg.speaker_pins[0];
if (nid)
alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0);
nid = spec->autocfg.hp_pins[0];
if (nid)
alc260_auto_set_output_and_unmute(codec, nid, PIN_HP, 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) {
alc_set_input_pin(codec, nid, i);
if (nid != ALC260_PIN_CD_NID &&
(get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
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 */
/* mute analog inputs */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(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;
int err;
static hda_nid_t alc260_ignore[] = { 0x17, 0 };
err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc260_ignore);
if (err < 0)
return err;
err = alc260_auto_create_multi_out_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
if (!spec->kctls.list)
return 0; /* can't find valid BIOS pin config */
err = alc260_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
spec->multiout.max_channels = 2;
if (spec->autocfg.dig_outs)
spec->multiout.dig_out_nid = ALC260_DIGOUT_NID;
if (spec->kctls.list)
add_mixer(spec, spec->kctls.list);
add_verb(spec, 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 10:47:09 +00:00
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux[0];
alc_ssid_check(codec, 0x10, 0x15, 0x0f);
return 1;
}
/* additional initialization for auto-configuration model */
static void alc260_auto_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
alc260_auto_init_multi_out(codec);
alc260_auto_init_analog_input(codec);
if (spec->unsol_event)
alc_inithook(codec);
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
static struct hda_amp_list alc260_loopbacks[] = {
{ 0x07, HDA_INPUT, 0 },
{ 0x07, HDA_INPUT, 1 },
{ 0x07, HDA_INPUT, 2 },
{ 0x07, HDA_INPUT, 3 },
{ 0x07, HDA_INPUT, 4 },
{ } /* end */
};
#endif
/*
* ALC260 configurations
*/
static const char *alc260_models[ALC260_MODEL_LAST] = {
[ALC260_BASIC] = "basic",
[ALC260_HP] = "hp",
[ALC260_HP_3013] = "hp-3013",
[ALC260_HP_DC7600] = "hp-dc7600",
[ALC260_FUJITSU_S702X] = "fujitsu",
[ALC260_ACER] = "acer",
[ALC260_WILL] = "will",
[ALC260_REPLACER_672V] = "replacer",
[ALC260_FAVORIT100] = "favorit100",
#ifdef CONFIG_SND_DEBUG
[ALC260_TEST] = "test",
#endif
[ALC260_AUTO] = "auto",
};
static struct snd_pci_quirk alc260_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER),
SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013),
SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013),
SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600),
SND_PCI_QUIRK(0x103c, 0x3013, "HP", ALC260_HP_3013),
SND_PCI_QUIRK(0x103c, 0x3014, "HP", ALC260_HP),
SND_PCI_QUIRK(0x103c, 0x3015, "HP", ALC260_HP),
SND_PCI_QUIRK(0x103c, 0x3016, "HP", ALC260_HP),
SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC),
SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC),
SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC),
SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X),
SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC),
SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_REPLACER_672V),
SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_WILL),
{}
};
static struct alc_config_preset alc260_presets[] = {
[ALC260_BASIC] = {
.mixers = { alc260_base_output_mixer,
alc260_input_mixer },
.init_verbs = { alc260_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_adc_nids,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
},
[ALC260_HP] = {
.mixers = { alc260_hp_output_mixer,
alc260_input_mixer },
.init_verbs = { alc260_init_verbs,
alc260_hp_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
.adc_nids = alc260_adc_nids_alt,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
.unsol_event = alc260_hp_unsol_event,
.init_hook = alc260_hp_automute,
},
[ALC260_HP_DC7600] = {
.mixers = { alc260_hp_dc7600_mixer,
alc260_input_mixer },
.init_verbs = { alc260_init_verbs,
alc260_hp_dc7600_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
.adc_nids = alc260_adc_nids_alt,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
.unsol_event = alc260_hp_3012_unsol_event,
.init_hook = alc260_hp_3012_automute,
},
[ALC260_HP_3013] = {
.mixers = { alc260_hp_3013_mixer,
alc260_input_mixer },
.init_verbs = { alc260_hp_3013_init_verbs,
alc260_hp_3013_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
.adc_nids = alc260_adc_nids_alt,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
.unsol_event = alc260_hp_3013_unsol_event,
.init_hook = alc260_hp_3013_automute,
},
[ALC260_FUJITSU_S702X] = {
.mixers = { alc260_fujitsu_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 10:47:09 +00:00
.num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources),
.input_mux = alc260_fujitsu_capture_sources,
},
[ALC260_ACER] = {
.mixers = { alc260_acer_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 10:47:09 +00:00
.num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources),
.input_mux = alc260_acer_capture_sources,
},
[ALC260_FAVORIT100] = {
.mixers = { alc260_favorit100_mixer },
.init_verbs = { alc260_favorit100_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,
.num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources),
.input_mux = alc260_favorit100_capture_sources,
},
[ALC260_WILL] = {
.mixers = { alc260_will_mixer },
.init_verbs = { alc260_init_verbs, alc260_will_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,
.dig_out_nid = ALC260_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
},
[ALC260_REPLACER_672V] = {
.mixers = { alc260_replacer_672v_mixer },
.init_verbs = { alc260_init_verbs, alc260_replacer_672v_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,
.dig_out_nid = ALC260_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
.unsol_event = alc260_replacer_672v_unsol_event,
.init_hook = alc260_replacer_672v_automute,
},
#ifdef CONFIG_SND_DEBUG
[ALC260_TEST] = {
.mixers = { alc260_test_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 10:47:09 +00:00
.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_MODEL_LAST,
alc260_models,
alc260_cfg_tbl);
if (board_config < 0) {
snd_printd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
codec->chip_name);
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;
}
}
err = snd_hda_attach_beep_device(codec, 0x1);
if (err < 0) {
alc_free(codec);
return err;
}
if (board_config != ALC260_AUTO)
setup_preset(codec, &alc260_presets[board_config]);
spec->stream_analog_playback = &alc260_pcm_analog_playback;
spec->stream_analog_capture = &alc260_pcm_analog_capture;
spec->stream_digital_playback = &alc260_pcm_digital_playback;
spec->stream_digital_capture = &alc260_pcm_digital_capture;
if (!spec->adc_nids && spec->input_mux) {
/* check whether NID 0x04 is valid */
unsigned int wcap = get_wcaps(codec, 0x04);
wcap = get_wcaps_type(wcap);
/* get type */
if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
spec->adc_nids = alc260_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt);
} else {
spec->adc_nids = alc260_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids);
}
}
set_capture_mixer(codec);
set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
spec->vmaster_nid = 0x08;
codec->patch_ops = alc_patch_ops;
if (board_config == ALC260_AUTO)
spec->init_hook = alc260_auto_init;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
spec->loopback.amplist = alc260_loopbacks;
#endif
codec->proc_widget_hook = print_realtek_coef;
return 0;
}
/*
* ALC882/883/885/888/889 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
#define ALC883_DIGOUT_NID ALC882_DIGOUT_NID
#define ALC883_DIGIN_NID ALC882_DIGIN_NID
#define ALC1200_DIGOUT_NID 0x10
static struct hda_channel_mode alc882_ch_modes[1] = {
{ 8, NULL }
};
/* DACs */
static hda_nid_t alc882_dac_nids[4] = {
/* front, rear, clfe, rear_surr */
0x02, 0x03, 0x04, 0x05
};
#define alc883_dac_nids alc882_dac_nids
/* ADCs */
#define alc882_adc_nids alc880_adc_nids
#define alc882_adc_nids_alt alc880_adc_nids_alt
#define alc883_adc_nids alc882_adc_nids_alt
static hda_nid_t alc883_adc_nids_alt[1] = { 0x08 };
static hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 };
#define alc889_adc_nids alc880_adc_nids
static hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 };
static hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 };
#define alc883_capsrc_nids alc882_capsrc_nids_alt
static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 };
#define alc889_capsrc_nids alc882_capsrc_nids
/* 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 alc883_capture_source alc882_capture_source
static struct hda_input_mux alc889_capture_source = {
.num_items = 3,
.items = {
{ "Front Mic", 0x0 },
{ "Mic", 0x3 },
{ "Line", 0x2 },
},
};
static struct hda_input_mux mb5_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
static struct hda_input_mux alc883_3stack_6ch_intel = {
.num_items = 4,
.items = {
{ "Mic", 0x1 },
{ "Front Mic", 0x0 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
static struct hda_input_mux alc883_lenovo_101e_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x1 },
{ "Line", 0x2 },
},
};
static struct hda_input_mux alc883_lenovo_nb0763_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "iMic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x0 },
{ "Int Mic", 0x1 },
},
};
static struct hda_input_mux alc883_lenovo_sky_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x4 },
},
};
static struct hda_input_mux alc883_asus_eee1601_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x0 },
{ "Line", 0x2 },
},
};
static struct hda_input_mux alc889A_mb31_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x0 },
/* Front Mic (0x01) unused */
{ "Line", 0x2 },
/* Line 2 (0x03) unused */
/* CD (0x04) unsused? */
},
};
/*
* 2ch mode
*/
static struct hda_channel_mode alc883_3ST_2ch_modes[1] = {
{ 2, NULL }
};
/*
* 2ch mode
*/
static struct hda_verb alc882_3ST_ch2_init[] = {
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ 0x18, 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 },
{ } /* end */
};
/*
* 4ch mode
*/
static struct hda_verb alc882_3ST_ch4_init[] = {
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ 0x18, 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 },
{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
{ } /* end */
};
/*
* 6ch mode
*/
static struct hda_verb alc882_3ST_ch6_init[] = {
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
{ } /* end */
};
static struct hda_channel_mode alc882_3ST_6ch_modes[3] = {
{ 2, alc882_3ST_ch2_init },
{ 4, alc882_3ST_ch4_init },
{ 6, alc882_3ST_ch6_init },
};
#define alc883_3ST_6ch_modes alc882_3ST_6ch_modes
/*
* 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 },
};
/*
* macbook pro ALC885 can switch LineIn to LineOut without losing Mic
*/
/*
* 2ch mode
*/
static struct hda_verb alc885_mbp_ch2_init[] = {
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ } /* end */
};
/*
* 6ch mode
*/
static struct hda_verb alc885_mbp_ch6_init[] = {
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{ } /* end */
};
static struct hda_channel_mode alc885_mbp_6ch_modes[2] = {
{ 2, alc885_mbp_ch2_init },
{ 6, alc885_mbp_ch6_init },
};
/*
* 2ch
* Speakers/Woofer/HP = Front
* LineIn = Input
*/
static struct hda_verb alc885_mb5_ch2_init[] = {
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{ } /* end */
};
/*
* 6ch mode
* Speakers/HP = Front
* Woofer = LFE
* LineIn = Surround
*/
static struct hda_verb alc885_mb5_ch6_init[] = {
{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},
{ } /* end */
};
static struct hda_channel_mode alc885_mb5_6ch_modes[2] = {
{ 2, alc885_mb5_ch2_init },
{ 6, alc885_mb5_ch6_init },
};
/*
* 2ch mode
*/
static struct hda_verb alc883_4ST_ch2_init[] = {
{ 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 },
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
{ } /* end */
};
/*
* 4ch mode
*/
static struct hda_verb alc883_4ST_ch4_init[] = {
{ 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 },
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
{ } /* end */
};
/*
* 6ch mode
*/
static struct hda_verb alc883_4ST_ch6_init[] = {
{ 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_OUT },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
{ } /* end */
};
/*
* 8ch mode
*/
static struct hda_verb alc883_4ST_ch8_init[] = {
{ 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 },
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
{ } /* end */
};
static struct hda_channel_mode alc883_4ST_8ch_modes[4] = {
{ 2, alc883_4ST_ch2_init },
{ 4, alc883_4ST_ch4_init },
{ 6, alc883_4ST_ch6_init },
{ 8, alc883_4ST_ch8_init },
};
/*
* 2ch mode
*/
static struct hda_verb alc883_3ST_ch2_intel_init[] = {
{ 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 },
{ } /* end */
};
/*
* 4ch mode
*/
static struct hda_verb alc883_3ST_ch4_intel_init[] = {
{ 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 },
{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
{ } /* end */
};
/*
* 6ch mode
*/
static struct hda_verb alc883_3ST_ch6_intel_init[] = {
{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 },
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
{ } /* end */
};
static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = {
{ 2, alc883_3ST_ch2_intel_init },
{ 4, alc883_3ST_ch4_intel_init },
{ 6, alc883_3ST_ch6_intel_init },
};
/*
* 2ch mode
*/
static struct hda_verb alc889_ch2_intel_init[] = {
{ 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x19, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x16, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
{ } /* end */
};
/*
* 6ch mode
*/
static struct hda_verb alc889_ch6_intel_init[] = {
{ 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 },
{ 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 },
{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
{ 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 alc889_ch8_intel_init[] = {
{ 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 },
{ 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 },
{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 },
{ 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 alc889_8ch_intel_modes[3] = {
{ 2, alc889_ch2_intel_init },
{ 6, alc889_ch6_intel_init },
{ 8, alc889_ch8_intel_init },
};
/*
* 6ch mode
*/
static struct hda_verb alc883_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 alc883_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 alc883_sixstack_modes[2] = {
{ 6, alc883_sixstack_ch6_init },
{ 8, alc883_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_VOLUME("Mic Boost", 0x18, 0, 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_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc885_mbp3_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
HDA_CODEC_MUTE ("Speaker Playback Switch", 0x14, 0x00, HDA_OUTPUT),
HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
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, 0x00, HDA_INPUT),
HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Line Boost", 0x1a, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc885_mb5_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 0x02, 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, 0x01, HDA_INPUT),
HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x19, 0x00, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc882_w2jc_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc882_targa_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 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("Mic Boost", 0x18, 0, 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("Front Mic Boost", 0x19, 0, HDA_INPUT),
{ } /* end */
};
/* Pin assignment: Front=0x14, HP = 0x15, Front = 0x16, ???
* Front Mic=0x18, Line In = 0x1a, Line In = 0x1b, CD = 0x1c
*/
static struct snd_kcontrol_new alc882_asus_a7j_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Mobile Front Playback Switch", 0x16, 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("Mobile Line Playback Volume", 0x0b, 0x03, HDA_INPUT),
HDA_CODEC_MUTE("Mobile Line Playback Switch", 0x0b, 0x03, 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("Mic Boost", 0x18, 0, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc882_asus_a7m_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
static 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_base_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)},
/* mute analog input loopbacks */
{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)},
/* 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 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)},
/* 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},
{ }
};
static struct hda_verb alc882_adc1_init_verbs[] = {
/* 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)},
/* ADC1: mute amp left and right */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
{ }
};
static struct hda_verb alc882_eapd_verbs[] = {
/* change to EAPD mode */
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3060},
{ }
};
static struct hda_verb alc889_eapd_verbs[] = {
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
static struct hda_verb alc_hp15_unsol_verbs[] = {
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{}
};
static struct hda_verb alc885_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)},
/* mute analog input loopbacks */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
/* Front HP Pin: output 0 (0x0c) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
/* 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) */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x19, 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% */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Front Mic pin: input vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, 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},
/* Mixer elements: 0x18, , 0x1a, 0x1b */
/* Input mixer1 */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
/* ADC2: mute amp left and right */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
/* ADC3: mute amp left and right */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{ }
};
static struct hda_verb alc885_init_input_verbs[] = {
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{ }
};
/* Unmute Selector 24h and set the default input to front mic */
static struct hda_verb alc889_init_input_verbs[] = {
{0x24, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{ }
};
#define alc883_init_verbs alc882_base_init_verbs
/* Mac Pro test */
static struct snd_kcontrol_new alc882_macpro_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x18, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
/* FIXME: this looks suspicious...
HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x02, HDA_INPUT),
*/
{ } /* end */
};
static struct hda_verb alc882_macpro_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)},
/* Front Pin: output 0 (0x0c) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
/* 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},
/* Speaker: output */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x04},
/* Headphone output (output 0 - 0x0c) */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
/* 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},
{ }
};
/* Macbook 5,1 */
static struct hda_verb alc885_mb5_init_verbs[] = {
/* DACs */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{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},
/* Front mixer */
{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)},
/* Surround 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)},
/* LFE 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)},
/* HP 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 (0x0c) */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
/* LFE Pin (0x0e) */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x02},
/* HP Pin (0x0f) */
{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, 0x03},
/* 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 */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{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)},
{ }
};
/* Macbook Pro rev3 */
static struct hda_verb alc885_mbp3_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)},
/* 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},
/* HP Pin: output 0 (0x0d) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
/* 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: use output 1 when in LineOut mode */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, 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, 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},
{ }
};
/* iMac 24 mixer. */
static struct snd_kcontrol_new alc885_imac24_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x00, HDA_INPUT),
{ } /* end */
};
/* iMac 24 init verbs. */
static struct hda_verb alc885_imac24_init_verbs[] = {
/* Internal speakers: output 0 (0x0c) */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Internal speakers: output 0 (0x0c) */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Headphone: output 0 (0x0c) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
/* Front Mic: input vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{ }
};
/* Toggle speaker-output according to the hp-jack state */
static void alc885_imac24_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x18;
spec->autocfg.speaker_pins[1] = 0x1a;
}
static void alc885_mbp3_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
}
static struct hda_verb alc882_targa_verbs[] = {
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ } /* end */
};
/* toggle speaker-output according to the hp-jack state */
static void alc882_targa_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
alc_automute_amp(codec);
snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
spec->jack_present ? 1 : 3);
}
static void alc882_targa_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x1b;
}
static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res)
{
if ((res >> 26) == ALC880_HP_EVENT)
alc882_targa_automute(codec);
}
static struct hda_verb alc882_asus_a7j_verbs[] = {
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
{0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{ } /* end */
};
static struct hda_verb alc882_asus_a7m_verbs[] = {
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
{0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{ } /* end */
};
static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted)
{
unsigned int gpiostate, gpiomask, gpiodir;
gpiostate = snd_hda_codec_read(codec, codec->afg, 0,
AC_VERB_GET_GPIO_DATA, 0);
if (!muted)
gpiostate |= (1 << pin);
else
gpiostate &= ~(1 << pin);
gpiomask = snd_hda_codec_read(codec, codec->afg, 0,
AC_VERB_GET_GPIO_MASK, 0);
gpiomask |= (1 << pin);
gpiodir = snd_hda_codec_read(codec, codec->afg, 0,
AC_VERB_GET_GPIO_DIRECTION, 0);
gpiodir |= (1 << pin);
snd_hda_codec_write(codec, codec->afg, 0,
AC_VERB_SET_GPIO_MASK, gpiomask);
snd_hda_codec_write(codec, codec->afg, 0,
AC_VERB_SET_GPIO_DIRECTION, gpiodir);
msleep(1);
snd_hda_codec_write(codec, codec->afg, 0,
AC_VERB_SET_GPIO_DATA, gpiostate);
}
/* set up GPIO at initialization */
static void alc885_macpro_init_hook(struct hda_codec *codec)
{
alc882_gpio_mute(codec, 0, 0);
alc882_gpio_mute(codec, 1, 0);
}
/* set up GPIO and update auto-muting at initialization */
static void alc885_imac24_init_hook(struct hda_codec *codec)
{
alc885_macpro_init_hook(codec);
alc_automute_amp(codec);
}
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc883_auto_init_verbs[] = {
/*
* Unmute ADC0-2 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)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for
* front panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* 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 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))},
{ }
};
/* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */
static struct hda_verb alc889A_mb31_ch2_init[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */
{ } /* end */
};
/* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */
static struct hda_verb alc889A_mb31_ch4_init[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
{ } /* end */
};
/* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */
static struct hda_verb alc889A_mb31_ch5_init[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as rear */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */
{ } /* end */
};
/* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */
static struct hda_verb alc889A_mb31_ch6_init[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as front */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Subwoofer off */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
{ } /* end */
};
static struct hda_channel_mode alc889A_mb31_6ch_modes[4] = {
{ 2, alc889A_mb31_ch2_init },
{ 4, alc889A_mb31_ch4_init },
{ 5, alc889A_mb31_ch5_init },
{ 6, alc889A_mb31_ch6_init },
};
static struct hda_verb alc883_medion_eapd_verbs[] = {
/* eanable EAPD on medion laptop */
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3070},
{ }
};
#define alc883_base_mixer alc882_base_mixer
static struct snd_kcontrol_new alc883_mitac_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 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", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc883_clevo_m720_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 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_VOLUME("Mic Boost", 0x18, 0, 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_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc883_3ST_6ch_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),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc883_3ST_6ch_intel_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", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc885_8ch_intel_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("Speaker Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
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, 0x3, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x1b, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc883_fivestack_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),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc883_targa_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
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_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front 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("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT),
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("iMic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("iMic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc883_medion_md2_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("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_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc888_lenovo_sky_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", 0x0e, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 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("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc889A_mb31_mixer[] = {
/* Output mixers */
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x00,
HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x02, HDA_INPUT),
/* Output switches */
HDA_CODEC_MUTE("Enable Speaker", 0x14, 0x00, HDA_OUTPUT),
HDA_CODEC_MUTE("Enable Headphones", 0x15, 0x00, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Enable LFE", 0x16, 2, 0x00, HDA_OUTPUT),
/* Boost mixers */
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Line Boost", 0x1a, 0x00, HDA_INPUT),
/* Input mixers */
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc883_vaiott_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct hda_bind_ctls alc883_bind_cap_vol = {
.ops = &snd_hda_bind_vol,
.values = {
HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
0
},
};
static struct hda_bind_ctls alc883_bind_cap_switch = {
.ops = &snd_hda_bind_sw,
.values = {
HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
0
},
};
static struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = {
HDA_BIND_VOL("Capture Volume", &alc883_bind_cap_vol),
HDA_BIND_SW("Capture Switch", &alc883_bind_cap_switch),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* .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 struct snd_kcontrol_new alc883_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 */
};
/* toggle speaker-output according to the hp-jack state */
static void alc883_mitac_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x17;
}
/* auto-toggle front mic */
/*
static void alc883_mitac_mic_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x18, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
}
*/
static struct hda_verb alc883_mitac_verbs[] = {
/* HP */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Subwoofer */
{0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* enable unsolicited event */
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
/* {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, */
{ } /* end */
};
static struct hda_verb alc883_clevo_m720_verbs[] = {
/* HP */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Int speaker */
{0x14, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* enable unsolicited event */
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
{ } /* end */
};
static struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = {
/* HP */
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* Subwoofer */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* enable unsolicited event */
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ } /* end */
};
static struct hda_verb alc883_targa_verbs[] = {
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* Connect Line-Out side jack (SPDIF) to Side */
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
/* Connect Mic jack to CLFE */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_CONNECT_SEL, 0x02},
/* Connect Line-in jack to Surround */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
/* Connect HP out jack to Front */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ } /* end */
};
static struct hda_verb alc883_lenovo_101e_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_FRONT_EVENT|AC_USRSP_EN},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT|AC_USRSP_EN},
{ } /* end */
};
static struct hda_verb alc883_lenovo_nb0763_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{ } /* end */
};
static struct hda_verb alc888_lenovo_ms7195_verbs[] = {
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_FRONT_EVENT | AC_USRSP_EN},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ } /* end */
};
static struct hda_verb alc883_haier_w66_verbs[] = {
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{ } /* end */
};
static struct hda_verb alc888_lenovo_sky_verbs[] = {
{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)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ } /* end */
};
static struct hda_verb alc888_6st_dell_verbs[] = {
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ }
};
static struct hda_verb alc883_vaiott_verbs[] = {
/* HP */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* enable unsolicited event */
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ } /* end */
};
static void alc888_3st_hp_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x18;
}
static struct hda_verb alc888_3st_hp_verbs[] = {
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */
{0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */
{0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ } /* end */
};
/*
* 2ch mode
*/
static struct hda_verb alc888_3st_hp_2ch_init[] = {
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
{ } /* end */
};
/*
* 4ch mode
*/
static struct hda_verb alc888_3st_hp_4ch_init[] = {
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
{ 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, 0x01 },
{ } /* end */
};
/*
* 6ch mode
*/
static struct hda_verb alc888_3st_hp_6ch_init[] = {
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
{ 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, 0x01 },
{ } /* end */
};
static struct hda_channel_mode alc888_3st_hp_modes[3] = {
{ 2, alc888_3st_hp_2ch_init },
{ 4, alc888_3st_hp_4ch_init },
{ 6, alc888_3st_hp_6ch_init },
};
/* toggle front-jack and RCA according to the hp-jack state */
static void alc888_lenovo_ms7195_front_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_stereo(codec, 0x14, HDA_OUTPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
/* toggle RCA according to the front-jack state */
static void alc888_lenovo_ms7195_rca_automute(struct hda_codec *codec)
{
unsigned int present;
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
static void alc883_lenovo_ms7195_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC880_HP_EVENT)
alc888_lenovo_ms7195_front_automute(codec);
if ((res >> 26) == ALC880_FRONT_EVENT)
alc888_lenovo_ms7195_rca_automute(codec);
}
static struct hda_verb alc883_medion_md2_verbs[] = {
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ } /* end */
};
/* toggle speaker-output according to the hp-jack state */
static void alc883_medion_md2_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
}
/* toggle speaker-output according to the hp-jack state */
#define alc883_targa_init_hook alc882_targa_init_hook
#define alc883_targa_unsol_event alc882_targa_unsol_event
static void alc883_clevo_m720_mic_automute(struct hda_codec *codec)
{
unsigned int present;
present = snd_hda_codec_read(codec, 0x18, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
static void alc883_clevo_m720_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
}
static void alc883_clevo_m720_init_hook(struct hda_codec *codec)
{
alc_automute_amp(codec);
alc883_clevo_m720_mic_automute(codec);
}
static void alc883_clevo_m720_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC880_MIC_EVENT:
alc883_clevo_m720_mic_automute(codec);
break;
default:
alc_automute_amp_unsol_event(codec, res);
break;
}
}
/* toggle speaker-output according to the hp-jack state */
static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
}
static void alc883_haier_w66_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
}
static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0)
& AC_PINSENSE_PRESENCE;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
}
static void alc883_lenovo_101e_all_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
}
static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC880_HP_EVENT)
alc883_lenovo_101e_all_automute(codec);
if ((res >> 26) == ALC880_FRONT_EVENT)
alc883_lenovo_101e_ispeaker_automute(codec);
}
/* toggle speaker-output according to the hp-jack state */
static void alc883_acer_aspire_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
spec->autocfg.speaker_pins[1] = 0x16;
}
static struct hda_verb alc883_acer_eapd_verbs[] = {
/* HP Pin: output 0 (0x0c) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Front Pin: output 0 (0x0c) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
/* eanable EAPD on medion laptop */
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3050},
/* enable unsolicited event */
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ }
};
static struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ } /* end */
};
static void alc888_6st_dell_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x15;
spec->autocfg.speaker_pins[2] = 0x16;
spec->autocfg.speaker_pins[3] = 0x17;
}
static void alc888_lenovo_sky_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x15;
spec->autocfg.speaker_pins[2] = 0x16;
spec->autocfg.speaker_pins[3] = 0x17;
spec->autocfg.speaker_pins[4] = 0x1a;
}
static void alc883_vaiott_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x17;
}
static struct hda_verb alc888_asus_m90v_verbs[] = {
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* enable unsolicited event */
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
{ } /* end */
};
static void alc883_mode2_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x15;
spec->autocfg.speaker_pins[2] = 0x16;
spec->ext_mic.pin = 0x18;
spec->int_mic.pin = 0x19;
spec->ext_mic.mux_idx = 0;
spec->int_mic.mux_idx = 1;
spec->auto_mic = 1;
}
static struct hda_verb alc888_asus_eee1601_verbs[] = {
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x20, AC_VERB_SET_COEF_INDEX, 0x0b},
{0x20, AC_VERB_SET_PROC_COEF, 0x0838},
/* enable unsolicited event */
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ } /* end */
};
static void alc883_eee1601_inithook(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x1b;
alc_automute_pin(codec);
}
static struct hda_verb alc889A_mb31_verbs[] = {
/* Init rear pin (used as headphone output) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, /* Apple Headphones */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Connect to front */
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
/* Init line pin (used as output in 4ch and 6ch mode) */
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect to CLFE */
/* Init line 2 pin (used as headphone out by default) */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Use as input */
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Mute output */
{ } /* end */
};
/* Mute speakers according to the headphone jack state */
static void alc889A_mb31_automute(struct hda_codec *codec)
{
unsigned int present;
/* Mute only in 2ch or 4ch mode */
if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0)
== 0x00) {
present = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE;
snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
}
static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
{
if ((res >> 26) == ALC880_HP_EVENT)
alc889A_mb31_automute(codec);
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
#define alc882_loopbacks alc880_loopbacks
#endif
/* pcm configuration: identical 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
static hda_nid_t alc883_slave_dig_outs[] = {
ALC1200_DIGOUT_NID, 0,
};
static hda_nid_t alc1200_slave_dig_outs[] = {
ALC883_DIGOUT_NID, 0,
};
/*
* configuration and preset
*/
static const char *alc882_models[ALC882_MODEL_LAST] = {
[ALC882_3ST_DIG] = "3stack-dig",
[ALC882_6ST_DIG] = "6stack-dig",
[ALC882_ARIMA] = "arima",
[ALC882_W2JC] = "w2jc",
[ALC882_TARGA] = "targa",
[ALC882_ASUS_A7J] = "asus-a7j",
[ALC882_ASUS_A7M] = "asus-a7m",
[ALC885_MACPRO] = "macpro",
[ALC885_MB5] = "mb5",
[ALC885_MBP3] = "mbp3",
[ALC885_IMAC24] = "imac24",
[ALC883_3ST_2ch_DIG] = "3stack-2ch-dig",
[ALC883_3ST_6ch_DIG] = "3stack-6ch-dig",
[ALC883_3ST_6ch] = "3stack-6ch",
[ALC883_6ST_DIG] = "alc883-6stack-dig",
[ALC883_TARGA_DIG] = "targa-dig",
[ALC883_TARGA_2ch_DIG] = "targa-2ch-dig",
[ALC883_TARGA_8ch_DIG] = "targa-8ch-dig",
[ALC883_ACER] = "acer",
[ALC883_ACER_ASPIRE] = "acer-aspire",
[ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g",
[ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g",
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
[ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g",
[ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g",
[ALC883_MEDION] = "medion",
[ALC883_MEDION_MD2] = "medion-md2",
[ALC883_LAPTOP_EAPD] = "laptop-eapd",
[ALC883_LENOVO_101E_2ch] = "lenovo-101e",
[ALC883_LENOVO_NB0763] = "lenovo-nb0763",
[ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig",
[ALC888_LENOVO_SKY] = "lenovo-sky",
[ALC883_HAIER_W66] = "haier-w66",
[ALC888_3ST_HP] = "3stack-hp",
[ALC888_6ST_DELL] = "6stack-dell",
[ALC883_MITAC] = "mitac",
[ALC883_CLEVO_M720] = "clevo-m720",
[ALC883_FUJITSU_PI2515] = "fujitsu-pi2515",
[ALC888_FUJITSU_XA3530] = "fujitsu-xa3530",
[ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel",
[ALC889A_INTEL] = "intel-alc889a",
[ALC889_INTEL] = "intel-x58",
[ALC1200_ASUS_P5Q] = "asus-p5q",
[ALC889A_MB31] = "mb31",
[ALC883_SONY_VAIO_TT] = "sony-vaio-tt",
[ALC882_AUTO] = "auto",
};
static struct snd_pci_quirk alc882_cfg_tbl[] = {
SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G",
ALC888_ACER_ASPIRE_4930G),
SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G",
ALC888_ACER_ASPIRE_4930G),
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G",
ALC888_ACER_ASPIRE_8930G),
SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G",
ALC888_ACER_ASPIRE_8930G),
SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC882_AUTO),
SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC882_AUTO),
SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
ALC888_ACER_ASPIRE_6530G),
SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
ALC888_ACER_ASPIRE_6530G),
SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
ALC888_ACER_ASPIRE_7730G),
/* default Acer -- disabled as it causes more problems.
* model=auto should work fine now
*/
/* SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), */
SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP),
SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP),
SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J),
SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J),
SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M),
SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V),
SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC),
SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q),
SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601),
SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT),
SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC),
SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC),
SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL),
SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch),
SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */
SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3fc1, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3fdf, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x42cd, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x6510, "MSI GX620", ALC883_TARGA_8ch_DIG),
SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7260, "MSI 7260", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),
SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD),
SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
/* SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), */
SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1100, "FSC AMILO Xi/Pi25xx",
ALC883_FUJITSU_PI2515),
SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1130, "Fujitsu AMILO Xa35xx",
ALC888_FUJITSU_XA3530),
SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch),
SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763),
SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),
SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
SND_PCI_QUIRK(0x17aa, 0x101d, "Lenovo Sky", ALC888_LENOVO_SKY),
SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG),
SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL),
SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL),
SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC),
SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC889_INTEL),
SND_PCI_QUIRK(0x8086, 0x0021, "Intel IbexPeak", ALC889A_INTEL),
SND_PCI_QUIRK(0x8086, 0x3b56, "Intel IbexPeak", ALC889A_INTEL),
SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch),
{}
};
/* codec SSID table for Intel Mac */
static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_MACPRO),
SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_IMAC24),
SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_IMAC24),
SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31),
SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24),
SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
/* FIXME: HP jack sense seems not working for MBP 5,1, so apparently
* no perfect solution yet
*/
SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5),
{} /* terminator */
};
static struct alc_config_preset alc882_presets[] = {
[ALC882_3ST_DIG] = {
.mixers = { alc882_base_mixer },
.init_verbs = { alc882_base_init_verbs,
alc882_adc1_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
.channel_mode = alc882_ch_modes,
.need_dac_fix = 1,
.input_mux = &alc882_capture_source,
},
[ALC882_6ST_DIG] = {
.mixers = { alc882_base_mixer, alc882_chmode_mixer },
.init_verbs = { alc882_base_init_verbs,
alc882_adc1_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
.channel_mode = alc882_sixstack_modes,
.input_mux = &alc882_capture_source,
},
[ALC882_ARIMA] = {
.mixers = { alc882_base_mixer, alc882_chmode_mixer },
.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
alc882_eapd_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
.channel_mode = alc882_sixstack_modes,
.input_mux = &alc882_capture_source,
},
[ALC882_W2JC] = {
.mixers = { alc882_w2jc_mixer, alc882_chmode_mixer },
.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
alc882_eapd_verbs, alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
.channel_mode = alc880_threestack_modes,
.need_dac_fix = 1,
.input_mux = &alc882_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
},
[ALC885_MBP3] = {
.mixers = { alc885_mbp3_mixer, alc882_chmode_mixer },
.init_verbs = { alc885_mbp3_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.channel_mode = alc885_mbp_6ch_modes,
.num_channel_mode = ARRAY_SIZE(alc885_mbp_6ch_modes),
.input_mux = &alc882_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc885_mbp3_setup,
.init_hook = alc_automute_amp,
},
[ALC885_MB5] = {
.mixers = { alc885_mb5_mixer, alc882_chmode_mixer },
.init_verbs = { alc885_mb5_init_verbs,
alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.channel_mode = alc885_mb5_6ch_modes,
.num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes),
.input_mux = &mb5_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
},
[ALC885_MACPRO] = {
.mixers = { alc882_macpro_mixer },
.init_verbs = { alc882_macpro_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
.channel_mode = alc882_ch_modes,
.input_mux = &alc882_capture_source,
.init_hook = alc885_macpro_init_hook,
},
[ALC885_IMAC24] = {
.mixers = { alc885_imac24_mixer },
.init_verbs = { alc885_imac24_init_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
.channel_mode = alc882_ch_modes,
.input_mux = &alc882_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc885_imac24_setup,
.init_hook = alc885_imac24_init_hook,
},
[ALC882_TARGA] = {
.mixers = { alc882_targa_mixer, alc882_chmode_mixer },
.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
alc880_gpio3_init_verbs, alc882_targa_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,
.capsrc_nids = alc882_capsrc_nids,
.num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
.channel_mode = alc882_3ST_6ch_modes,
.need_dac_fix = 1,
.input_mux = &alc882_capture_source,
.unsol_event = alc882_targa_unsol_event,
.setup = alc882_targa_setup,
.init_hook = alc882_targa_automute,
},
[ALC882_ASUS_A7J] = {
.mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer },
.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
alc882_asus_a7j_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,
.capsrc_nids = alc882_capsrc_nids,
.num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
.channel_mode = alc882_3ST_6ch_modes,
.need_dac_fix = 1,
.input_mux = &alc882_capture_source,
},
[ALC882_ASUS_A7M] = {
.mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer },
.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
alc882_eapd_verbs, alc880_gpio1_init_verbs,
alc882_asus_a7m_verbs },
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.dig_out_nid = ALC882_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
.channel_mode = alc880_threestack_modes,
.need_dac_fix = 1,
.input_mux = &alc882_capture_source,
},
[ALC883_3ST_2ch_DIG] = {
.mixers = { alc883_3ST_2ch_mixer },
.init_verbs = { alc883_init_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
},
[ALC883_3ST_6ch_DIG] = {
.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
},
[ALC883_3ST_6ch] = {
.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
},
[ALC883_3ST_6ch_INTEL] = {
.mixers = { alc883_3ST_6ch_intel_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.dig_in_nid = ALC883_DIGIN_NID,
.slave_dig_outs = alc883_slave_dig_outs,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes),
.channel_mode = alc883_3ST_6ch_intel_modes,
.need_dac_fix = 1,
.input_mux = &alc883_3stack_6ch_intel,
},
[ALC889A_INTEL] = {
.mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer },
.init_verbs = { alc885_init_verbs, alc885_init_input_verbs,
alc_hp15_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
.adc_nids = alc889_adc_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.dig_in_nid = ALC883_DIGIN_NID,
.slave_dig_outs = alc883_slave_dig_outs,
.num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes),
.channel_mode = alc889_8ch_intel_modes,
.capsrc_nids = alc889_capsrc_nids,
.input_mux = &alc889_capture_source,
.setup = alc889_automute_setup,
.init_hook = alc_automute_amp,
.unsol_event = alc_automute_amp_unsol_event,
.need_dac_fix = 1,
},
[ALC889_INTEL] = {
.mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer },
.init_verbs = { alc885_init_verbs, alc889_init_input_verbs,
alc889_eapd_verbs, alc_hp15_unsol_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
.adc_nids = alc889_adc_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.dig_in_nid = ALC883_DIGIN_NID,
.slave_dig_outs = alc883_slave_dig_outs,
.num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes),
.channel_mode = alc889_8ch_intel_modes,
.capsrc_nids = alc889_capsrc_nids,
.input_mux = &alc889_capture_source,
.setup = alc889_automute_setup,
.init_hook = alc889_intel_init_hook,
.unsol_event = alc_automute_amp_unsol_event,
.need_dac_fix = 1,
},
[ALC883_6ST_DIG] = {
.mixers = { alc883_base_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
.channel_mode = alc883_sixstack_modes,
.input_mux = &alc883_capture_source,
},
[ALC883_TARGA_DIG] = {
.mixers = { alc883_targa_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
alc883_targa_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_targa_unsol_event,
.setup = alc882_targa_setup,
.init_hook = alc882_targa_automute,
},
[ALC883_TARGA_2ch_DIG] = {
.mixers = { alc883_targa_2ch_mixer},
.init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
alc883_targa_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.adc_nids = alc883_adc_nids_alt,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_targa_unsol_event,
.setup = alc882_targa_setup,
.init_hook = alc882_targa_automute,
},
[ALC883_TARGA_8ch_DIG] = {
.mixers = { alc883_base_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
alc883_targa_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
.adc_nids = alc883_adc_nids_rev,
.capsrc_nids = alc883_capsrc_nids_rev,
.dig_out_nid = ALC883_DIGOUT_NID,
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_4ST_8ch_modes),
.channel_mode = alc883_4ST_8ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_targa_unsol_event,
.setup = alc882_targa_setup,
.init_hook = alc882_targa_automute,
},
[ALC883_ACER] = {
.mixers = { alc883_base_mixer },
/* 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.
*/
.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
},
[ALC883_ACER_ASPIRE] = {
.mixers = { alc883_acer_aspire_mixer },
.init_verbs = { alc883_init_verbs, alc883_acer_eapd_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc883_acer_aspire_setup,
.init_hook = alc_automute_amp,
},
[ALC888_ACER_ASPIRE_4930G] = {
.mixers = { alc888_base_mixer,
alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
alc888_acer_aspire_4930g_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
.adc_nids = alc883_adc_nids_rev,
.capsrc_nids = alc883_capsrc_nids_rev,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
.num_mux_defs =
ARRAY_SIZE(alc888_2_capture_sources),
.input_mux = alc888_2_capture_sources,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc888_acer_aspire_4930g_setup,
.init_hook = alc_automute_amp,
},
[ALC888_ACER_ASPIRE_6530G] = {
.mixers = { alc888_acer_aspire_6530_mixer },
.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
alc888_acer_aspire_6530g_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
.adc_nids = alc883_adc_nids_rev,
.capsrc_nids = alc883_capsrc_nids_rev,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.num_mux_defs =
ARRAY_SIZE(alc888_2_capture_sources),
.input_mux = alc888_acer_aspire_6530_sources,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc888_acer_aspire_6530g_setup,
.init_hook = alc_automute_amp,
},
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
[ALC888_ACER_ASPIRE_8930G] = {
.mixers = { alc888_base_mixer,
alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
alc889_acer_aspire_8930g_verbs },
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
.adc_nids = alc889_adc_nids,
.capsrc_nids = alc889_capsrc_nids,
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
.const_channel_count = 6,
.num_mux_defs =
ARRAY_SIZE(alc889_capture_sources),
.input_mux = alc889_capture_sources,
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc889_acer_aspire_8930g_setup,
.init_hook = alc_automute_amp,
ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 08:54:19 +00:00
},
[ALC888_ACER_ASPIRE_7730G] = {
.mixers = { alc883_3ST_6ch_mixer,
alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
alc888_acer_aspire_7730G_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
.adc_nids = alc883_adc_nids_rev,
.capsrc_nids = alc883_capsrc_nids_rev,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
.const_channel_count = 6,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc888_acer_aspire_6530g_setup,
.init_hook = alc_automute_amp,
},
[ALC883_MEDION] = {
.mixers = { alc883_fivestack_mixer,
alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs,
alc883_medion_eapd_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.adc_nids = alc883_adc_nids_alt,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
.channel_mode = alc883_sixstack_modes,
.input_mux = &alc883_capture_source,
},
[ALC883_MEDION_MD2] = {
.mixers = { alc883_medion_md2_mixer},
.init_verbs = { alc883_init_verbs, alc883_medion_md2_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc883_medion_md2_setup,
.init_hook = alc_automute_amp,
},
[ALC883_LAPTOP_EAPD] = {
.mixers = { alc883_base_mixer },
.init_verbs = { alc883_init_verbs, alc882_eapd_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
},
[ALC883_CLEVO_M720] = {
.mixers = { alc883_clevo_m720_mixer },
.init_verbs = { alc883_init_verbs, alc883_clevo_m720_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_clevo_m720_unsol_event,
.setup = alc883_clevo_m720_setup,
.init_hook = alc883_clevo_m720_init_hook,
},
[ALC883_LENOVO_101E_2ch] = {
.mixers = { alc883_lenovo_101e_2ch_mixer},
.init_verbs = { alc883_init_verbs, alc883_lenovo_101e_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.adc_nids = alc883_adc_nids_alt,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_lenovo_101e_capture_source,
.unsol_event = alc883_lenovo_101e_unsol_event,
.init_hook = alc883_lenovo_101e_all_automute,
},
[ALC883_LENOVO_NB0763] = {
.mixers = { alc883_lenovo_nb0763_mixer },
.init_verbs = { alc883_init_verbs, alc883_lenovo_nb0763_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_lenovo_nb0763_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc883_medion_md2_setup,
.init_hook = alc_automute_amp,
},
[ALC888_LENOVO_MS7195_DIG] = {
.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc888_lenovo_ms7195_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_lenovo_ms7195_unsol_event,
.init_hook = alc888_lenovo_ms7195_front_automute,
},
[ALC883_HAIER_W66] = {
.mixers = { alc883_targa_2ch_mixer},
.init_verbs = { alc883_init_verbs, alc883_haier_w66_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc883_haier_w66_setup,
.init_hook = alc_automute_amp,
},
[ALC888_3ST_HP] = {
.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc888_3st_hp_modes),
.channel_mode = alc888_3st_hp_modes,
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc888_3st_hp_setup,
.init_hook = alc_automute_amp,
},
[ALC888_6ST_DELL] = {
.mixers = { alc883_base_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc888_6st_dell_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
.channel_mode = alc883_sixstack_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc888_6st_dell_setup,
.init_hook = alc_automute_amp,
},
[ALC883_MITAC] = {
.mixers = { alc883_mitac_mixer },
.init_verbs = { alc883_init_verbs, alc883_mitac_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc883_mitac_setup,
.init_hook = alc_automute_amp,
},
[ALC883_FUJITSU_PI2515] = {
.mixers = { alc883_2ch_fujitsu_pi2515_mixer },
.init_verbs = { alc883_init_verbs,
alc883_2ch_fujitsu_pi2515_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_fujitsu_pi2515_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc883_2ch_fujitsu_pi2515_setup,
.init_hook = alc_automute_amp,
},
[ALC888_FUJITSU_XA3530] = {
.mixers = { alc888_base_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs,
alc888_fujitsu_xa3530_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
.adc_nids = alc883_adc_nids_rev,
.capsrc_nids = alc883_capsrc_nids_rev,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc888_4ST_8ch_intel_modes),
.channel_mode = alc888_4ST_8ch_intel_modes,
.num_mux_defs =
ARRAY_SIZE(alc888_2_capture_sources),
.input_mux = alc888_2_capture_sources,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc888_fujitsu_xa3530_setup,
.init_hook = alc_automute_amp,
},
[ALC888_LENOVO_SKY] = {
.mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc888_lenovo_sky_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
.channel_mode = alc883_sixstack_modes,
.need_dac_fix = 1,
.input_mux = &alc883_lenovo_sky_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc888_lenovo_sky_setup,
.init_hook = alc_automute_amp,
},
[ALC888_ASUS_M90V] = {
.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc888_asus_m90v_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_fujitsu_pi2515_capture_source,
.unsol_event = alc_sku_unsol_event,
.setup = alc883_mode2_setup,
.init_hook = alc_inithook,
},
[ALC888_ASUS_EEE1601] = {
.mixers = { alc883_asus_eee1601_mixer },
.cap_mixer = alc883_asus_eee1601_cap_mixer,
.init_verbs = { alc883_init_verbs, alc888_asus_eee1601_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.dig_in_nid = ALC883_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_asus_eee1601_capture_source,
.unsol_event = alc_sku_unsol_event,
.init_hook = alc883_eee1601_inithook,
},
[ALC1200_ASUS_P5Q] = {
.mixers = { alc883_base_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC1200_DIGOUT_NID,
.dig_in_nid = ALC883_DIGIN_NID,
.slave_dig_outs = alc1200_slave_dig_outs,
.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
.channel_mode = alc883_sixstack_modes,
.input_mux = &alc883_capture_source,
},
[ALC889A_MB31] = {
.mixers = { alc889A_mb31_mixer, alc883_chmode_mixer},
.init_verbs = { alc883_init_verbs, alc889A_mb31_verbs,
alc880_gpio1_init_verbs },
.adc_nids = alc883_adc_nids,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
.dac_nids = alc883_dac_nids,
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.channel_mode = alc889A_mb31_6ch_modes,
.num_channel_mode = ARRAY_SIZE(alc889A_mb31_6ch_modes),
.input_mux = &alc889A_mb31_capture_source,
.dig_out_nid = ALC883_DIGOUT_NID,
.unsol_event = alc889A_mb31_unsol_event,
.init_hook = alc889A_mb31_automute,
},
[ALC883_SONY_VAIO_TT] = {
.mixers = { alc883_vaiott_mixer },
.init_verbs = { alc883_init_verbs, alc883_vaiott_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc883_vaiott_setup,
.init_hook = alc_automute_amp,
},
};
/*
* Pin config fixes
*/
enum {
PINFIX_ABIT_AW9D_MAX
};
static struct alc_pincfg alc882_abit_aw9d_pinfix[] = {
{ 0x15, 0x01080104 }, /* side */
{ 0x16, 0x01011012 }, /* rear */
{ 0x17, 0x01016011 }, /* clfe */
{ }
};
static const struct alc_pincfg *alc882_pin_fixes[] = {
[PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix,
};
static struct snd_pci_quirk alc882_pinfix_tbl[] = {
SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
{}
};
/*
* BIOS auto configuration
*/
static int alc882_auto_create_input_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x23, 0x22);
}
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;
alc_set_pin_output(codec, nid, pin_type);
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_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];
int pin_type = get_pin_type(spec->autocfg.line_out_type);
if (nid)
alc882_auto_set_output_and_unmute(codec, nid, pin_type,
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_pins[0];
if (pin) /* connect to front */
/* use dac 0 */
alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
pin = spec->autocfg.speaker_pins[0];
if (pin)
alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
}
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 (!nid)
continue;
alc_set_input_pin(codec, nid, i);
if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_MUTE);
}
}
static void alc882_auto_init_input_src(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int c;
for (c = 0; c < spec->num_adc_nids; c++) {
hda_nid_t conn_list[HDA_MAX_NUM_INPUTS];
hda_nid_t nid = spec->capsrc_nids[c];
unsigned int mux_idx;
const struct hda_input_mux *imux;
int conns, mute, idx, item;
conns = snd_hda_get_connections(codec, nid, conn_list,
ARRAY_SIZE(conn_list));
if (conns < 0)
continue;
mux_idx = c >= spec->num_mux_defs ? 0 : c;
imux = &spec->input_mux[mux_idx];
for (idx = 0; idx < conns; idx++) {
/* if the current connection is the selected one,
* unmute it as default - otherwise mute it
*/
mute = AMP_IN_MUTE(idx);
for (item = 0; item < imux->num_items; item++) {
if (imux->items[item].index == idx) {
if (spec->cur_mux[c] == item)
mute = AMP_IN_UNMUTE(idx);
break;
}
}
/* check if we have a selector or mixer
* we could check for the widget type instead, but
* just check for Amp-In presence (in case of mixer
* without amp-in there is something wrong, this
* function shouldn't be used or capsrc nid is wrong)
*/
if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
mute);
else if (mute != AMP_IN_MUTE(idx))
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_CONNECT_SEL,
idx);
}
}
}
/* add mic boosts if needed */
static int alc_auto_add_mic_boost(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err;
hda_nid_t nid;
nid = spec->autocfg.input_pins[AUTO_PIN_MIC];
if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) {
err = add_control(spec, ALC_CTL_WIDGET_VOL,
"Mic Boost",
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
if (err < 0)
return err;
}
nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC];
if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) {
err = add_control(spec, ALC_CTL_WIDGET_VOL,
"Front Mic Boost",
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
if (err < 0)
return err;
}
return 0;
}
/* almost identical with ALC880 parser... */
static int alc882_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
static hda_nid_t alc882_ignore[] = { 0x1d, 0 };
int i, err;
err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc882_ignore);
if (err < 0)
return err;
if (!spec->autocfg.line_outs)
return 0; /* can't find valid BIOS pin config */
err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
if (err < 0)
return err;
err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
err = alc880_auto_create_extra_out(spec,
spec->autocfg.speaker_pins[0],
"Speaker");
if (err < 0)
return err;
err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
"Headphone");
if (err < 0)
return err;
err = alc882_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
/* check multiple SPDIF-out (for recent codecs) */
for (i = 0; i < spec->autocfg.dig_outs; i++) {
hda_nid_t dig_nid;
err = snd_hda_get_connections(codec,
spec->autocfg.dig_out_pins[i],
&dig_nid, 1);
if (err < 0)
continue;
if (!i)
spec->multiout.dig_out_nid = dig_nid;
else {
spec->multiout.slave_dig_outs = spec->slave_dig_outs;
spec->slave_dig_outs[i - 1] = dig_nid;
if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
break;
}
}
if (spec->autocfg.dig_in_pin)
spec->dig_in_nid = ALC880_DIGIN_NID;
if (spec->kctls.list)
add_mixer(spec, spec->kctls.list);
add_verb(spec, alc883_auto_init_verbs);
/* if ADC 0x07 is available, initialize it, too */
if (get_wcaps_type(get_wcaps(codec, 0x07)) == AC_WID_AUD_IN)
add_verb(spec, alc882_adc1_init_verbs);
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux[0];
alc_ssid_check(codec, 0x15, 0x1b, 0x14);
err = alc_auto_add_mic_boost(codec);
if (err < 0)
return err;
return 1; /* config found */
}
/* additional initialization for auto-configuration model */
static void alc882_auto_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
alc882_auto_init_multi_out(codec);
alc882_auto_init_hp_out(codec);
alc882_auto_init_analog_input(codec);
alc882_auto_init_input_src(codec);
if (spec->unsol_event)
alc_inithook(codec);
}
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;
switch (codec->vendor_id) {
case 0x10ec0882:
case 0x10ec0885:
break;
default:
/* ALC883 and variants */
alc_fix_pll_init(codec, 0x20, 0x0a, 10);
break;
}
board_config = snd_hda_check_board_config(codec, ALC882_MODEL_LAST,
alc882_models,
alc882_cfg_tbl);
if (board_config < 0 || board_config >= ALC882_MODEL_LAST)
board_config = snd_hda_check_board_codec_sid_config(codec,
ALC882_MODEL_LAST, alc882_models, alc882_ssid_cfg_tbl);
if (board_config < 0 || board_config >= ALC882_MODEL_LAST) {
printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
codec->chip_name);
board_config = ALC882_AUTO;
}
alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes);
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;
}
}
err = snd_hda_attach_beep_device(codec, 0x1);
if (err < 0) {
alc_free(codec);
return err;
}
if (board_config != ALC882_AUTO)
setup_preset(codec, &alc882_presets[board_config]);
spec->stream_analog_playback = &alc882_pcm_analog_playback;
spec->stream_analog_capture = &alc882_pcm_analog_capture;
/* FIXME: setup DAC5 */
/*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/
spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture;
spec->stream_digital_playback = &alc882_pcm_digital_playback;
spec->stream_digital_capture = &alc882_pcm_digital_capture;
if (codec->vendor_id == 0x10ec0888)
spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */
if (!spec->adc_nids && spec->input_mux) {
int i;
spec->num_adc_nids = 0;
for (i = 0; i < ARRAY_SIZE(alc882_adc_nids); i++) {
hda_nid_t cap;
hda_nid_t nid = alc882_adc_nids[i];
unsigned int wcap = get_wcaps(codec, nid);
/* get type */
wcap = get_wcaps_type(wcap);
if (wcap != AC_WID_AUD_IN)
continue;
spec->private_adc_nids[spec->num_adc_nids] = nid;
err = snd_hda_get_connections(codec, nid, &cap, 1);
if (err < 0)
continue;
spec->private_capsrc_nids[spec->num_adc_nids] = cap;
spec->num_adc_nids++;
}
spec->adc_nids = spec->private_adc_nids;
spec->capsrc_nids = spec->private_capsrc_nids;
}
set_capture_mixer(codec);
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
spec->vmaster_nid = 0x0c;
codec->patch_ops = alc_patch_ops;
if (board_config == ALC882_AUTO)
spec->init_hook = alc882_auto_init;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
spec->loopback.amplist = alc882_loopbacks;
#endif
codec->proc_widget_hook = print_realtek_coef;
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_capsrc_nids alc882_capsrc_nids
#define alc262_capsrc_nids_alt alc882_capsrc_nids_alt
#define alc262_modes alc260_modes
#define alc262_capture_source alc882_capture_source
static hda_nid_t alc262_dmic_adc_nids[1] = {
/* ADC0 */
0x09
};
static hda_nid_t alc262_dmic_capsrc_nids[1] = { 0x22 };
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("Mic Boost", 0x18, 0, 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("Front Mic Boost", 0x19, 0, 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 */
};
/* update HP, line and mono-out pins according to the master switch */
static void alc262_hp_master_update(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int val = spec->master_sw;
/* HP & line-out */
snd_hda_codec_write_cache(codec, 0x1b, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
val ? PIN_HP : 0);
snd_hda_codec_write_cache(codec, 0x15, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
val ? PIN_HP : 0);
/* mono (speaker) depending on the HP jack sense */
val = val && !spec->jack_present;
snd_hda_codec_write_cache(codec, 0x16, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
val ? PIN_OUT : 0);
}
static void alc262_hp_bpc_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int presence;
presence = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0);
spec->jack_present = !!(presence & AC_PINSENSE_PRESENCE);
alc262_hp_master_update(codec);
}
static void alc262_hp_bpc_unsol_event(struct hda_codec *codec, unsigned int res)
{
if ((res >> 26) != ALC880_HP_EVENT)
return;
alc262_hp_bpc_automute(codec);
}
static void alc262_hp_wildwest_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int presence;
presence = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0);
spec->jack_present = !!(presence & AC_PINSENSE_PRESENCE);
alc262_hp_master_update(codec);
}
static void alc262_hp_wildwest_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) != ALC880_HP_EVENT)
return;
alc262_hp_wildwest_automute(codec);
}
#define alc262_hp_master_sw_get alc260_hp_master_sw_get
static int alc262_hp_master_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
int val = !!*ucontrol->value.integer.value;
if (val == spec->master_sw)
return 0;
spec->master_sw = val;
alc262_hp_master_update(codec);
return 1;
}
#define ALC262_HP_MASTER_SWITCH \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = "Master Playback Switch", \
.info = snd_ctl_boolean_mono_info, \
.get = alc262_hp_master_sw_get, \
.put = alc262_hp_master_sw_put, \
}
static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
ALC262_HP_MASTER_SWITCH,
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, 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("Front Mic Boost", 0x19, 0, 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("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT),
HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = {
ALC262_HP_MASTER_SWITCH,
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
HDA_OUTPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost", 0x1a, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = {
HDA_CODEC_VOLUME("Rear Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Rear Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Rear Mic Boost", 0x18, 0, HDA_INPUT),
{ } /* end */
};
/* mute/unmute internal speaker according to the hp jack and mute state */
static void alc262_hp_t5735_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */
}
static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
{ } /* end */
};
static struct hda_verb alc262_hp_t5735_verbs[] = {
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ }
};
static struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x16, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
{ } /* end */
};
static struct hda_verb alc262_hp_rp5700_verbs[] = {
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
{}
};
static struct hda_input_mux alc262_hp_rp5700_capture_source = {
.num_items = 1,
.items = {
{ "Line", 0x1 },
},
};
/* bind hp and internal speaker mute (with plug check) as master switch */
static void alc262_hippo_master_update(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t hp_nid = spec->autocfg.hp_pins[0];
hda_nid_t line_nid = spec->autocfg.line_out_pins[0];
hda_nid_t speaker_nid = spec->autocfg.speaker_pins[0];
unsigned int mute;
/* HP */
mute = spec->master_sw ? 0 : HDA_AMP_MUTE;
snd_hda_codec_amp_stereo(codec, hp_nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, mute);
/* mute internal speaker per jack sense */
if (spec->jack_present)
mute = HDA_AMP_MUTE;
if (line_nid)
snd_hda_codec_amp_stereo(codec, line_nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, mute);
if (speaker_nid && speaker_nid != line_nid)
snd_hda_codec_amp_stereo(codec, speaker_nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, mute);
}
#define alc262_hippo_master_sw_get alc262_hp_master_sw_get
static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
int val = !!*ucontrol->value.integer.value;
if (val == spec->master_sw)
return 0;
spec->master_sw = val;
alc262_hippo_master_update(codec);
return 1;
}
#define ALC262_HIPPO_MASTER_SWITCH \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = "Master Playback Switch", \
.info = snd_ctl_boolean_mono_info, \
.get = alc262_hippo_master_sw_get, \
.put = alc262_hippo_master_sw_put, \
}
static struct snd_kcontrol_new alc262_hippo_mixer[] = {
ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 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("Mic Boost", 0x18, 0, 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("Front Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc262_hippo1_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
ALC262_HIPPO_MASTER_SWITCH,
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("Mic Boost", 0x18, 0, 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("Front Mic Boost", 0x19, 0, HDA_INPUT),
{ } /* end */
};
/* mute/unmute internal speaker according to the hp jack and mute state */
static void alc262_hippo_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t hp_nid = spec->autocfg.hp_pins[0];
unsigned int present;
/* need to execute and sync at first */
snd_hda_codec_read(codec, hp_nid, 0, AC_VERB_SET_PIN_SENSE, 0);
present = snd_hda_codec_read(codec, hp_nid, 0,
AC_VERB_GET_PIN_SENSE, 0);
spec->jack_present = (present & 0x80000000) != 0;
alc262_hippo_master_update(codec);
}
static void alc262_hippo_unsol_event(struct hda_codec *codec, unsigned int res)
{
if ((res >> 26) != ALC880_HP_EVENT)
return;
alc262_hippo_automute(codec);
}
static void alc262_hippo_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
}
static void alc262_hippo1_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
}
static struct snd_kcontrol_new alc262_sony_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc262_benq_t31_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc262_tyan_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Aux Playback Volume", 0x0b, 0x06, HDA_INPUT),
HDA_CODEC_MUTE("Aux Playback Switch", 0x0b, 0x06, 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("Mic Boost", 0x18, 0, 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("Front Mic Boost", 0x19, 0, HDA_INPUT),
{ } /* end */
};
static struct hda_verb alc262_tyan_verbs[] = {
/* Headphone automute */
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
/* P11 AUX_IN, white 4-pin connector */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, 0xe1},
{0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, 0x93},
{0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0x19},
{}
};
/* unsolicited event for HP jack sensing */
static void alc262_tyan_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x15;
}
#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)},
/* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for
* front panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* 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))},
{ }
};
static struct hda_verb alc262_eapd_verbs[] = {
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
static struct hda_verb alc262_hippo1_unsol_verbs[] = {
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{}
};
static struct hda_verb alc262_sony_unsol_verbs[] = {
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, // Front Mic
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{}
};
static struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
static struct hda_verb alc262_toshiba_s06_verbs[] = {
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x22, AC_VERB_SET_CONNECT_SEL, 0x09},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
static void alc262_toshiba_s06_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->ext_mic.pin = 0x18;
spec->ext_mic.mux_idx = 0;
spec->int_mic.pin = 0x12;
spec->int_mic.mux_idx = 9;
spec->auto_mic = 1;
}
/*
* nec model
* 0x15 = headphone
* 0x16 = internal speaker
* 0x18 = external mic
*/
static struct snd_kcontrol_new alc262_nec_mixer[] = {
HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 0, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
{ } /* end */
};
static struct hda_verb alc262_nec_verbs[] = {
/* Unmute Speaker */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Headphone */
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* External mic to headphone */
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* External mic to speaker */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{}
};
/*
* fujitsu model
* 0x14 = headphone/spdif-out, 0x15 = internal speaker,
* 0x1b = port replicator headphone out
*/
#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},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{}
};
static struct hda_verb alc262_lenovo_3000_unsol_verbs[] = {
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{}
};
static struct hda_input_mux alc262_fujitsu_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x0 },
{ "Int Mic", 0x1 },
{ "CD", 0x4 },
},
};
static struct hda_input_mux alc262_HP_capture_source = {
.num_items = 5,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "AUX IN", 0x6 },
},
};
static struct hda_input_mux alc262_HP_D7000_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x2 },
{ "Line", 0x1 },
{ "CD", 0x4 },
},
};
/* mute/unmute internal speaker according to the hp jacks 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);
/* check laptop HP jack */
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0);
/* need to execute and sync at first */
snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
/* check docking HP jack */
present |= snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0);
if (present & AC_PINSENSE_PRESENCE)
spec->jack_present = 1;
else
spec->jack_present = 0;
spec->sense_updated = 1;
}
/* unmute internal speaker only if both HPs are unplugged and
* master switch is on
*/
if (spec->jack_present)
mute = HDA_AMP_MUTE;
else
mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0);
snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
HDA_AMP_MUTE, mute);
}
/* 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);
}
static void alc262_fujitsu_init_hook(struct hda_codec *codec)
{
alc262_fujitsu_automute(codec, 1);
}
/* bind volumes of both NID 0x0c and 0x0d */
static struct hda_bind_ctls alc262_fujitsu_bind_master_vol = {
.ops = &snd_hda_bind_vol,
.values = {
HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x0d, 3, 0, HDA_OUTPUT),
0
},
};
/* mute/unmute internal speaker according to the hp jack and mute state */
static void alc262_lenovo_3000_automute(struct hda_codec *codec, int force)
{
struct alc_spec *spec = codec->spec;
unsigned int mute;
if (force || !spec->sense_updated) {
unsigned int present_int_hp;
/* need to execute and sync at first */
snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
present_int_hp = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0);
spec->jack_present = (present_int_hp & 0x80000000) != 0;
spec->sense_updated = 1;
}
if (spec->jack_present) {
/* mute internal speaker */
snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
} else {
/* unmute internal speaker if necessary */
mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0);
snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
HDA_AMP_MUTE, mute);
snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
HDA_AMP_MUTE, mute);
}
}
/* unsolicited event for HP jack sensing */
static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) != ALC_HP_EVENT)
return;
alc262_lenovo_3000_automute(codec, 1);
}
static int amp_stereo_mute_update(struct hda_codec *codec, hda_nid_t nid,
int dir, int idx, long *valp)
{
int i, change = 0;
for (i = 0; i < 2; i++, valp++)
change |= snd_hda_codec_amp_update(codec, nid, i, dir, idx,
HDA_AMP_MUTE,
*valp ? 0 : HDA_AMP_MUTE);
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 = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
change |= amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
if (change)
alc262_fujitsu_automute(codec, 0);
return change;
}
static struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
{
.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),
HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
/* bind hp and internal speaker mute (with plug check) */
static int alc262_lenovo_3000_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 = amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
if (change)
alc262_lenovo_3000_automute(codec, 0);
return change;
}
static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = {
HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
{
.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_lenovo_3000_master_sw_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x1b, 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),
HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = {
HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, 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("Front Mic Boost", 0x19, 0, HDA_INPUT),
{ } /* end */
};
/* additional init verbs for Benq laptops */
static struct hda_verb alc262_EAPD_verbs[] = {
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3070},
{}
};
static struct hda_verb alc262_benq_t31_EAPD_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
{0x20, AC_VERB_SET_PROC_COEF, 0x3050},
{}
};
/* Samsung Q1 Ultra Vista model setup */
static struct snd_kcontrol_new alc262_ultra_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, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Mic Boost", 0x15, 0, HDA_INPUT),
{ } /* end */
};
static struct hda_verb alc262_ultra_verbs[] = {
/* output mixer */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* speaker */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* HP */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
/* internal mic */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* ADC, choose mic */
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(8)},
{}
};
/* mute/unmute internal speaker according to the hp jack and mute state */
static void alc262_ultra_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int mute;
mute = 0;
/* auto-mute only when HP is used as HP */
if (!spec->cur_mux[0]) {
unsigned int present;
/* need to execute and sync at first */
snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0);
present = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0);
spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
if (spec->jack_present)
mute = HDA_AMP_MUTE;
}
/* mute/unmute internal speaker */
snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
HDA_AMP_MUTE, mute);
/* mute/unmute HP */
snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
HDA_AMP_MUTE, mute ? 0 : HDA_AMP_MUTE);
}
/* unsolicited event for HP jack sensing */
static void alc262_ultra_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) != ALC880_HP_EVENT)
return;
alc262_ultra_automute(codec);
}
static struct hda_input_mux alc262_ultra_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x1 },
{ "Headphone", 0x7 },
},
};
static int alc262_ultra_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;
int ret;
ret = alc_mux_enum_put(kcontrol, ucontrol);
if (!ret)
return 0;
/* reprogram the HP pin as mic or HP according to the input source */
snd_hda_codec_write_cache(codec, 0x15, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
spec->cur_mux[0] ? PIN_VREF80 : PIN_HP);
alc262_ultra_automute(codec); /* mute/unmute HP */
return ret;
}
static struct snd_kcontrol_new alc262_ultra_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Capture Source",
.info = alc_mux_enum_info,
.get = alc_mux_enum_get,
.put = alc262_ultra_mux_enum_put,
},
{ } /* end */
};
/* We use two mixers depending on the output pin; 0x16 is a mono output
* and thus it's bound with a different mixer.
* This function returns which mixer amp should be used.
*/
static int alc262_check_volbit(hda_nid_t nid)
{
if (!nid)
return 0;
else if (nid == 0x16)
return 2;
else
return 1;
}
static int alc262_add_out_vol_ctl(struct alc_spec *spec, hda_nid_t nid,
const char *pfx, int *vbits)
{
char name[32];
unsigned long val;
int vbit;
vbit = alc262_check_volbit(nid);
if (!vbit)
return 0;
if (*vbits & vbit) /* a volume control for this mixer already there */
return 0;
*vbits |= vbit;
snprintf(name, sizeof(name), "%s Playback Volume", pfx);
if (vbit == 2)
val = HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_OUTPUT);
else
val = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT);
return add_control(spec, ALC_CTL_WIDGET_VOL, name, val);
}
static int alc262_add_out_sw_ctl(struct alc_spec *spec, hda_nid_t nid,
const char *pfx)
{
char name[32];
unsigned long val;
if (!nid)
return 0;
snprintf(name, sizeof(name), "%s Playback Switch", pfx);
if (nid == 0x16)
val = HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT);
else
val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
return add_control(spec, ALC_CTL_WIDGET_MUTE, name, val);
}
/* 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)
{
const char *pfx;
int vbits;
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;
if (!cfg->speaker_pins[0] && !cfg->hp_pins[0])
pfx = "Master";
else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
pfx = "Speaker";
else
pfx = "Front";
err = alc262_add_out_sw_ctl(spec, cfg->line_out_pins[0], pfx);
if (err < 0)
return err;
err = alc262_add_out_sw_ctl(spec, cfg->speaker_pins[0], "Speaker");
if (err < 0)
return err;
err = alc262_add_out_sw_ctl(spec, cfg->hp_pins[0], "Headphone");
if (err < 0)
return err;
vbits = alc262_check_volbit(cfg->line_out_pins[0]) |
alc262_check_volbit(cfg->speaker_pins[0]) |
alc262_check_volbit(cfg->hp_pins[0]);
if (vbits == 1 || vbits == 2)
pfx = "Master"; /* only one mixer is used */
else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
pfx = "Speaker";
else
pfx = "Front";
vbits = 0;
err = alc262_add_out_vol_ctl(spec, cfg->line_out_pins[0], pfx, &vbits);
if (err < 0)
return err;
err = alc262_add_out_vol_ctl(spec, cfg->speaker_pins[0], "Speaker",
&vbits);
if (err < 0)
return err;
err = alc262_add_out_vol_ctl(spec, cfg->hp_pins[0], "Headphone",
&vbits);
if (err < 0)
return err;
return 0;
}
#define alc262_auto_create_input_ctls \
alc880_auto_create_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)},
/* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for
* front panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* 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))},
{ }
};
static struct hda_verb alc262_HP_BPC_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)},
/* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for
* front panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
/*
* 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)},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x19, 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, 0x7000 },
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 0b, 12 */
/* Input mixer1: only unmute Mic */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ }
};
static struct hda_verb alc262_HP_BPC_WildWest_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)},
/* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for front
* panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
/*
* Set up output mixers (0x0c - 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)},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Mono */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* rear MIC */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* Line in */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Front MIC */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Line out */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD in */
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
/* {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 }, */
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
/* 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))}, /*rear MIC*/
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, /*Line in*/
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, /*F MIC*/
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, /*Front*/
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, /*CD*/
/* {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))}, /*HP*/
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
/* {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
/* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ }
};
static struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Front Speaker */
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{0x14, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* MIC jack */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Front MIC */
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) },
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) },
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP jack */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
#ifdef CONFIG_SND_HDA_POWER_SAVE
#define alc262_loopbacks alc880_loopbacks
#endif
/* pcm configuration: identical 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 };
err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc262_ignore);
if (err < 0)
return err;
if (!spec->autocfg.line_outs) {
if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
spec->multiout.max_channels = 2;
spec->no_analog = 1;
goto dig_only;
}
return 0; /* can't find valid BIOS pin config */
}
err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
err = alc262_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
dig_only:
if (spec->autocfg.dig_outs) {
spec->multiout.dig_out_nid = ALC262_DIGOUT_NID;
spec->dig_out_type = spec->autocfg.dig_out_type[0];
}
if (spec->autocfg.dig_in_pin)
spec->dig_in_nid = ALC262_DIGIN_NID;
if (spec->kctls.list)
add_mixer(spec, spec->kctls.list);
add_verb(spec, 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 10:47:09 +00:00
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux[0];
err = alc_auto_add_mic_boost(codec);
if (err < 0)
return err;
alc_ssid_check(codec, 0x15, 0x14, 0x1b);
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
#define alc262_auto_init_input_src alc882_auto_init_input_src
/* init callback for auto-configuration model -- overriding the default init */
static void alc262_auto_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
alc262_auto_init_multi_out(codec);
alc262_auto_init_hp_out(codec);
alc262_auto_init_analog_input(codec);
alc262_auto_init_input_src(codec);
if (spec->unsol_event)
alc_inithook(codec);
}
/*
* configuration and preset
*/
static const char *alc262_models[ALC262_MODEL_LAST] = {
[ALC262_BASIC] = "basic",
[ALC262_HIPPO] = "hippo",
[ALC262_HIPPO_1] = "hippo_1",
[ALC262_FUJITSU] = "fujitsu",
[ALC262_HP_BPC] = "hp-bpc",
[ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000",
[ALC262_HP_TC_T5735] = "hp-tc-t5735",
[ALC262_HP_RP5700] = "hp-rp5700",
[ALC262_BENQ_ED8] = "benq",
[ALC262_BENQ_T31] = "benq-t31",
[ALC262_SONY_ASSAMD] = "sony-assamd",
[ALC262_TOSHIBA_S06] = "toshiba-s06",
[ALC262_TOSHIBA_RX1] = "toshiba-rx1",
[ALC262_ULTRA] = "ultra",
[ALC262_LENOVO_3000] = "lenovo-3000",
[ALC262_NEC] = "nec",
[ALC262_TYAN] = "tyan",
[ALC262_AUTO] = "auto",
};
static struct snd_pci_quirk alc262_cfg_tbl[] = {
SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO),
SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC),
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1200, "HP xw series",
ALC262_HP_BPC),
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series",
ALC262_HP_BPC),
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF),
SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL),
SND_PCI_QUIRK(0x103c, 0x2803, "HP D7000", ALC262_HP_BPC_D7000_WF),
SND_PCI_QUIRK(0x103c, 0x2804, "HP D7000", ALC262_HP_BPC_D7000_WL),
SND_PCI_QUIRK(0x103c, 0x2805, "HP D7000", ALC262_HP_BPC_D7000_WF),
SND_PCI_QUIRK(0x103c, 0x2806, "HP D7000", ALC262_HP_BPC_D7000_WL),
SND_PCI_QUIRK(0x103c, 0x2807, "HP D7000", ALC262_HP_BPC_D7000_WF),
SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x302f, "HP Thin Client T5735",
ALC262_HP_TC_T5735),
SND_PCI_QUIRK(0x103c, 0x2817, "HP RP5700", ALC262_HP_RP5700),
SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO),
SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06),
SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
ALC262_TOSHIBA_RX1),
SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06),
SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU),
SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN),
SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1",
ALC262_ULTRA),
SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO),
SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000),
SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8),
SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31),
SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1),
{}
};
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_HIPPO] = {
.mixers = { alc262_hippo_mixer },
.init_verbs = { alc262_init_verbs, alc_hp15_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_capture_source,
.unsol_event = alc262_hippo_unsol_event,
.setup = alc262_hippo_setup,
.init_hook = alc262_hippo_automute,
},
[ALC262_HIPPO_1] = {
.mixers = { alc262_hippo1_mixer },
.init_verbs = { alc262_init_verbs, alc262_hippo1_unsol_verbs},
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x02,
.dig_out_nid = ALC262_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo_unsol_event,
.setup = alc262_hippo1_setup,
.init_hook = alc262_hippo_automute,
},
[ALC262_FUJITSU] = {
.mixers = { alc262_fujitsu_mixer },
.init_verbs = { alc262_init_verbs, alc262_EAPD_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,
.init_hook = alc262_fujitsu_init_hook,
},
[ALC262_HP_BPC] = {
.mixers = { alc262_HP_BPC_mixer },
.init_verbs = { alc262_HP_BPC_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_HP_capture_source,
.unsol_event = alc262_hp_bpc_unsol_event,
.init_hook = alc262_hp_bpc_automute,
},
[ALC262_HP_BPC_D7000_WF] = {
.mixers = { alc262_HP_BPC_WildWest_mixer },
.init_verbs = { alc262_HP_BPC_WildWest_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_HP_D7000_capture_source,
.unsol_event = alc262_hp_wildwest_unsol_event,
.init_hook = alc262_hp_wildwest_automute,
},
[ALC262_HP_BPC_D7000_WL] = {
.mixers = { alc262_HP_BPC_WildWest_mixer,
alc262_HP_BPC_WildWest_option_mixer },
.init_verbs = { alc262_HP_BPC_WildWest_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_HP_D7000_capture_source,
.unsol_event = alc262_hp_wildwest_unsol_event,
.init_hook = alc262_hp_wildwest_automute,
},
[ALC262_HP_TC_T5735] = {
.mixers = { alc262_hp_t5735_mixer },
.init_verbs = { alc262_init_verbs, alc262_hp_t5735_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,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc262_hp_t5735_setup,
.init_hook = alc_automute_amp,
},
[ALC262_HP_RP5700] = {
.mixers = { alc262_hp_rp5700_mixer },
.init_verbs = { alc262_init_verbs, alc262_hp_rp5700_verbs },
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_hp_rp5700_capture_source,
},
[ALC262_BENQ_ED8] = {
.mixers = { alc262_base_mixer },
.init_verbs = { alc262_init_verbs, alc262_EAPD_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_SONY_ASSAMD] = {
.mixers = { alc262_sony_mixer },
.init_verbs = { alc262_init_verbs, alc262_sony_unsol_verbs},
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo_unsol_event,
.setup = alc262_hippo_setup,
.init_hook = alc262_hippo_automute,
},
[ALC262_BENQ_T31] = {
.mixers = { alc262_benq_t31_mixer },
.init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs,
alc_hp15_unsol_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,
.unsol_event = alc262_hippo_unsol_event,
.setup = alc262_hippo_setup,
.init_hook = alc262_hippo_automute,
},
[ALC262_ULTRA] = {
.mixers = { alc262_ultra_mixer },
.cap_mixer = alc262_ultra_capture_mixer,
.init_verbs = { alc262_ultra_verbs },
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_ultra_capture_source,
.adc_nids = alc262_adc_nids, /* ADC0 */
.capsrc_nids = alc262_capsrc_nids,
.num_adc_nids = 1, /* single ADC */
.unsol_event = alc262_ultra_unsol_event,
.init_hook = alc262_ultra_automute,
},
[ALC262_LENOVO_3000] = {
.mixers = { alc262_lenovo_3000_mixer },
.init_verbs = { alc262_init_verbs, alc262_EAPD_verbs,
alc262_lenovo_3000_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_lenovo_3000_unsol_event,
},
[ALC262_NEC] = {
.mixers = { alc262_nec_mixer },
.init_verbs = { alc262_nec_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_TOSHIBA_S06] = {
.mixers = { alc262_toshiba_s06_mixer },
.init_verbs = { alc262_init_verbs, alc262_toshiba_s06_verbs,
alc262_eapd_verbs },
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.capsrc_nids = alc262_dmic_capsrc_nids,
.dac_nids = alc262_dac_nids,
.adc_nids = alc262_dmic_adc_nids, /* ADC0 */
.num_adc_nids = 1, /* single ADC */
.dig_out_nid = ALC262_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc262_toshiba_s06_setup,
.init_hook = alc_inithook,
},
[ALC262_TOSHIBA_RX1] = {
.mixers = { alc262_toshiba_rx1_mixer },
.init_verbs = { alc262_init_verbs, alc262_toshiba_rx1_unsol_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,
.unsol_event = alc262_hippo_unsol_event,
.setup = alc262_hippo_setup,
.init_hook = alc262_hippo_automute,
},
[ALC262_TYAN] = {
.mixers = { alc262_tyan_mixer },
.init_verbs = { alc262_init_verbs, alc262_tyan_verbs},
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x02,
.dig_out_nid = ALC262_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc262_tyan_setup,
.init_hook = alc_automute_amp,
},
};
static int patch_alc262(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;
#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
alc_fix_pll_init(codec, 0x20, 0x0a, 10);
board_config = snd_hda_check_board_config(codec, ALC262_MODEL_LAST,
alc262_models,
alc262_cfg_tbl);
if (board_config < 0) {
printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
codec->chip_name);
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 (!spec->no_analog) {
err = snd_hda_attach_beep_device(codec, 0x1);
if (err < 0) {
alc_free(codec);
return err;
}
}
if (board_config != ALC262_AUTO)
setup_preset(codec, &alc262_presets[board_config]);
spec->stream_analog_playback = &alc262_pcm_analog_playback;
spec->stream_analog_capture = &alc262_pcm_analog_capture;
spec->stream_digital_playback = &alc262_pcm_digital_playback;
spec->stream_digital_capture = &alc262_pcm_digital_capture;
if (!spec->adc_nids && spec->input_mux) {
int i;
/* check whether the digital-mic has to be supported */
for (i = 0; i < spec->input_mux->num_items; i++) {
if (spec->input_mux->items[i].index >= 9)
break;
}
if (i < spec->input_mux->num_items) {
/* use only ADC0 */
spec->adc_nids = alc262_dmic_adc_nids;
spec->num_adc_nids = 1;
spec->capsrc_nids = alc262_dmic_capsrc_nids;
} else {
/* all analog inputs */
/* check whether NID 0x07 is valid */
unsigned int wcap = get_wcaps(codec, 0x07);
/* get type */
wcap = get_wcaps_type(wcap);
if (wcap != AC_WID_AUD_IN) {
spec->adc_nids = alc262_adc_nids_alt;
spec->num_adc_nids =
ARRAY_SIZE(alc262_adc_nids_alt);
spec->capsrc_nids = alc262_capsrc_nids_alt;
} else {
spec->adc_nids = alc262_adc_nids;
spec->num_adc_nids =
ARRAY_SIZE(alc262_adc_nids);
spec->capsrc_nids = alc262_capsrc_nids;
}
}
}
if (!spec->cap_mixer && !spec->no_analog)
set_capture_mixer(codec);
if (!spec->no_analog)
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
spec->vmaster_nid = 0x0c;
codec->patch_ops = alc_patch_ops;
if (board_config == ALC262_AUTO)
spec->init_hook = alc262_auto_init;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
spec->loopback.amplist = alc262_loopbacks;
#endif
codec->proc_widget_hook = print_realtek_coef;
return 0;
}
/*
* ALC268 channel source setting (2 channel)
*/
#define ALC268_DIGOUT_NID ALC880_DIGOUT_NID
#define alc268_modes alc260_modes
static hda_nid_t alc268_dac_nids[2] = {
/* front, hp */
0x02, 0x03
};
static hda_nid_t alc268_adc_nids[2] = {
/* ADC0-1 */
0x08, 0x07
};
static hda_nid_t alc268_adc_nids_alt[1] = {
/* ADC0 */
0x08
};
static hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 };
static struct snd_kcontrol_new alc268_base_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT),
{ }
};
static struct snd_kcontrol_new alc268_toshiba_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT),
{ }
};
/* bind Beep switches of both NID 0x0f and 0x10 */
static struct hda_bind_ctls alc268_bind_beep_sw = {
.ops = &snd_hda_bind_sw,
.values = {
HDA_COMPOSE_AMP_VAL(0x0f, 3, 1, HDA_INPUT),
HDA_COMPOSE_AMP_VAL(0x10, 3, 1, HDA_INPUT),
0
},
};
static struct snd_kcontrol_new alc268_beep_mixer[] = {
HDA_CODEC_VOLUME("Beep Playback Volume", 0x1d, 0x0, HDA_INPUT),
HDA_BIND_SW("Beep Playback Switch", &alc268_bind_beep_sw),
{ }
};
static struct hda_verb alc268_eapd_verbs[] = {
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
/* Toshiba specific */
static struct hda_verb alc268_toshiba_verbs[] = {
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ } /* end */
};
/* Acer specific */
/* bind volumes of both NID 0x02 and 0x03 */
static struct hda_bind_ctls alc268_acer_bind_master_vol = {
.ops = &snd_hda_bind_vol,
.values = {
HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
0
},
};
/* mute/unmute internal speaker according to the hp jack and mute state */
static void alc268_acer_automute(struct hda_codec *codec, int force)
{
struct alc_spec *spec = codec->spec;
unsigned int mute;
if (force || !spec->sense_updated) {
unsigned int present;
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 = HDA_AMP_MUTE; /* mute internal speaker */
else /* unmute internal speaker if necessary */
mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0);
snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
HDA_AMP_MUTE, mute);
}
/* bind hp and internal speaker mute (with plug check) */
static int alc268_acer_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 = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
if (change)
alc268_acer_automute(codec, 0);
return change;
}
static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.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 = alc268_acer_master_sw_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
},
HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x18, 0, HDA_INPUT),
{ }
};
static struct snd_kcontrol_new alc268_acer_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.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 = alc268_acer_master_sw_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
},
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT),
{ }
};
static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.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 = alc268_acer_master_sw_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
},
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT),
{ }
};
static struct hda_verb alc268_acer_aspire_one_verbs[] = {
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x23, AC_VERB_SET_CONNECT_SEL, 0x06},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, 0xa017},
{ }
};
static struct hda_verb alc268_acer_verbs[] = {
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* internal dmic? */
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ }
};
/* unsolicited event for HP jack sensing */
#define alc268_toshiba_unsol_event alc262_hippo_unsol_event
#define alc268_toshiba_setup alc262_hippo_setup
#define alc268_toshiba_automute alc262_hippo_automute
static void alc268_acer_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) != ALC880_HP_EVENT)
return;
alc268_acer_automute(codec, 1);
}
static void alc268_acer_init_hook(struct hda_codec *codec)
{
alc268_acer_automute(codec, 1);
}
/* toggle speaker-output according to the hp-jack state */
static void alc268_aspire_one_speaker_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? AMP_IN_MUTE(0) : 0;
snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 1,
AMP_IN_MUTE(0), bits);
}
static void alc268_acer_lc_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC880_HP_EVENT:
alc268_aspire_one_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
static void alc268_acer_lc_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->ext_mic.pin = 0x18;
spec->ext_mic.mux_idx = 0;
spec->int_mic.pin = 0x12;
spec->int_mic.mux_idx = 6;
spec->auto_mic = 1;
}
static void alc268_acer_lc_init_hook(struct hda_codec *codec)
{
alc268_aspire_one_speaker_automute(codec);
alc_mic_automute(codec);
}
static struct snd_kcontrol_new alc268_dell_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT),
{ }
};
static struct hda_verb alc268_dell_verbs[] = {
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
{ }
};
/* mute/unmute internal speaker according to the hp jack and mute state */
static void alc268_dell_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->ext_mic.pin = 0x18;
spec->ext_mic.mux_idx = 0;
spec->int_mic.pin = 0x19;
spec->int_mic.mux_idx = 1;
spec->auto_mic = 1;
}
static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Mic Capture Switch", 0x23, 2, HDA_OUTPUT),
HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
{ }
};
static struct hda_verb alc267_quanta_il1_verbs[] = {
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
{ }
};
static void alc267_quanta_il1_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->ext_mic.pin = 0x18;
spec->ext_mic.mux_idx = 0;
spec->int_mic.pin = 0x19;
spec->int_mic.mux_idx = 1;
spec->auto_mic = 1;
}
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc268_base_init_verbs[] = {
/* Unmute DAC0-1 and set vol = 0 */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/*
* Set up output mixers (0x0c - 0x0e)
*/
/* set vol=0 to output mixers */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{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},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* set PCBEEP vol = 0, mute connections */
{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Unmute Selector 23h,24h and set the default input to mic-in */
{0x23, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x24, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{ }
};
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc268_volume_init_verbs[] = {
/* set output DAC */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* set PCBEEP vol = 0, mute connections */
{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ }
};
static struct snd_kcontrol_new alc268_capture_nosrc_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc268_capture_alt_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
_DEFINE_CAPSRC(1),
{ } /* end */
};
static struct snd_kcontrol_new alc268_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT),
_DEFINE_CAPSRC(2),
{ } /* end */
};
static struct hda_input_mux alc268_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x3 },
},
};
static struct hda_input_mux alc268_acer_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x0 },
{ "Internal Mic", 0x1 },
{ "Line", 0x2 },
},
};
static struct hda_input_mux alc268_acer_dmic_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x0 },
{ "Internal Mic", 0x6 },
{ "Line", 0x2 },
},
};
#ifdef CONFIG_SND_DEBUG
static struct snd_kcontrol_new alc268_test_mixer[] = {
/* Volume widgets */
HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Mono sum Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE("LINE-OUT sum Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_BIND_MUTE("HP-OUT sum Playback Switch", 0x10, 2, HDA_INPUT),
HDA_BIND_MUTE("LINE-OUT Playback Switch", 0x14, 2, HDA_OUTPUT),
HDA_BIND_MUTE("HP-OUT Playback Switch", 0x15, 2, HDA_OUTPUT),
HDA_BIND_MUTE("Mono Playback Switch", 0x16, 2, HDA_OUTPUT),
HDA_CODEC_VOLUME("MIC1 Capture Volume", 0x18, 0x0, HDA_INPUT),
HDA_BIND_MUTE("MIC1 Capture Switch", 0x18, 2, HDA_OUTPUT),
HDA_CODEC_VOLUME("MIC2 Capture Volume", 0x19, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("LINE1 Capture Volume", 0x1a, 0x0, HDA_INPUT),
HDA_BIND_MUTE("LINE1 Capture Switch", 0x1a, 2, HDA_OUTPUT),
/* The below appears problematic on some hardwares */
/*HDA_CODEC_VOLUME("PCBEEP Playback Volume", 0x1d, 0x0, HDA_INPUT),*/
HDA_CODEC_VOLUME("PCM-IN1 Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("PCM-IN1 Capture Switch", 0x23, 2, HDA_OUTPUT),
HDA_CODEC_VOLUME("PCM-IN2 Capture Volume", 0x24, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("PCM-IN2 Capture Switch", 0x24, 2, HDA_OUTPUT),
/* Modes for retasking pin widgets */
ALC_PIN_MODE("LINE-OUT pin mode", 0x14, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("HP-OUT pin mode", 0x15, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("MIC1 pin mode", 0x18, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("LINE1 pin mode", 0x1a, ALC_PIN_DIR_INOUT),
/* 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 SPDIF output pin to be enabled.
* The ALC268 does not have an SPDIF input.
*/
ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x06, 0x01),
/* A switch allowing EAPD to be enabled. Some laptops seem to use
* this output to turn on an external amplifier.
*/
ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
{ } /* end */
};
#endif
/* create input playback/capture controls for the given pin */
static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
const char *ctlname, int idx)
{
char name[32];
hda_nid_t dac;
int err;
sprintf(name, "%s Playback Volume", ctlname);
switch (nid) {
case 0x14:
case 0x16:
dac = 0x02;
break;
case 0x15:
dac = 0x03;
break;
default:
return 0;
}
if (spec->multiout.dac_nids[0] != dac &&
spec->multiout.dac_nids[1] != dac) {
err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(dac, 3, idx,
HDA_OUTPUT));
if (err < 0)
return err;
spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
}
sprintf(name, "%s Playback Switch", ctlname);
if (nid != 0x16)
err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT));
else /* mono */
err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid, 2, idx, HDA_OUTPUT));
if (err < 0)
return err;
return 0;
}
/* add playback controls from the parsed DAC table */
static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
{
hda_nid_t nid;
int err;
spec->multiout.dac_nids = spec->private_dac_nids;
nid = cfg->line_out_pins[0];
if (nid) {
const char *name;
if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
name = "Speaker";
else
name = "Front";
err = alc268_new_analog_output(spec, nid, name, 0);
if (err < 0)
return err;
}
nid = cfg->speaker_pins[0];
if (nid == 0x1d) {
err = add_control(spec, ALC_CTL_WIDGET_VOL,
"Speaker Playback Volume",
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
if (err < 0)
return err;
} else {
err = alc268_new_analog_output(spec, nid, "Speaker", 0);
if (err < 0)
return err;
}
nid = cfg->hp_pins[0];
if (nid) {
err = alc268_new_analog_output(spec, nid, "Headphone", 0);
if (err < 0)
return err;
}
nid = cfg->line_out_pins[1] | cfg->line_out_pins[2];
if (nid == 0x16) {
err = add_control(spec, ALC_CTL_WIDGET_MUTE,
"Mono Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT));
if (err < 0)
return err;
}
return 0;
}
/* create playback/capture controls for input pins */
static int alc268_auto_create_input_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
return alc_auto_create_input_ctls(codec, cfg, 0, 0x23, 0x24);
}
static void alc268_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type)
{
int idx;
alc_set_pin_output(codec, nid, pin_type);
if (nid == 0x14 || nid == 0x16)
idx = 0;
else
idx = 1;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
}
static void alc268_auto_init_multi_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t nid = spec->autocfg.line_out_pins[0];
if (nid) {
int pin_type = get_pin_type(spec->autocfg.line_out_type);
alc268_auto_set_output_and_unmute(codec, nid, pin_type);
}
}
static void alc268_auto_init_hp_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t pin;
pin = spec->autocfg.hp_pins[0];
if (pin)
alc268_auto_set_output_and_unmute(codec, pin, PIN_HP);
pin = spec->autocfg.speaker_pins[0];
if (pin)
alc268_auto_set_output_and_unmute(codec, pin, PIN_OUT);
}
static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t speaker_nid = spec->autocfg.speaker_pins[0];
hda_nid_t hp_nid = spec->autocfg.hp_pins[0];
hda_nid_t line_nid = spec->autocfg.line_out_pins[0];
unsigned int dac_vol1, dac_vol2;
if (line_nid == 0x1d || speaker_nid == 0x1d) {
snd_hda_codec_write(codec, speaker_nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
/* mute mixer inputs from 0x1d */
snd_hda_codec_write(codec, 0x0f, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_UNMUTE(1));
snd_hda_codec_write(codec, 0x10, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_UNMUTE(1));
} else {
/* unmute mixer inputs from 0x1d */
snd_hda_codec_write(codec, 0x0f, 0,
AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1));
snd_hda_codec_write(codec, 0x10, 0,
AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1));
}
dac_vol1 = dac_vol2 = 0xb000 | 0x40; /* set max volume */
if (line_nid == 0x14)
dac_vol2 = AMP_OUT_ZERO;
else if (line_nid == 0x15)
dac_vol1 = AMP_OUT_ZERO;
if (hp_nid == 0x14)
dac_vol2 = AMP_OUT_ZERO;
else if (hp_nid == 0x15)
dac_vol1 = AMP_OUT_ZERO;
if (line_nid != 0x16 || hp_nid != 0x16 ||
spec->autocfg.line_out_pins[1] != 0x16 ||
spec->autocfg.line_out_pins[2] != 0x16)
dac_vol1 = dac_vol2 = AMP_OUT_ZERO;
snd_hda_codec_write(codec, 0x02, 0,
AC_VERB_SET_AMP_GAIN_MUTE, dac_vol1);
snd_hda_codec_write(codec, 0x03, 0,
AC_VERB_SET_AMP_GAIN_MUTE, dac_vol2);
}
/* pcm configuration: identical with ALC880 */
#define alc268_pcm_analog_playback alc880_pcm_analog_playback
#define alc268_pcm_analog_capture alc880_pcm_analog_capture
#define alc268_pcm_analog_alt_capture alc880_pcm_analog_alt_capture
#define alc268_pcm_digital_playback alc880_pcm_digital_playback
/*
* BIOS auto configuration
*/
static int alc268_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err;
static hda_nid_t alc268_ignore[] = { 0 };
err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc268_ignore);
if (err < 0)
return err;
if (!spec->autocfg.line_outs) {
if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
spec->multiout.max_channels = 2;
spec->no_analog = 1;
goto dig_only;
}
return 0; /* can't find valid BIOS pin config */
}
err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
err = alc268_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
spec->multiout.max_channels = 2;
dig_only:
/* digital only support output */
if (spec->autocfg.dig_outs) {
spec->multiout.dig_out_nid = ALC268_DIGOUT_NID;
spec->dig_out_type = spec->autocfg.dig_out_type[0];
}
if (spec->kctls.list)
add_mixer(spec, spec->kctls.list);
if (!spec->no_analog && spec->autocfg.speaker_pins[0] != 0x1d)
add_mixer(spec, alc268_beep_mixer);
add_verb(spec, alc268_volume_init_verbs);
spec->num_mux_defs = 2;
spec->input_mux = &spec->private_imux[0];
err = alc_auto_add_mic_boost(codec);
if (err < 0)
return err;
alc_ssid_check(codec, 0x15, 0x1b, 0x14);
return 1;
}
#define alc268_auto_init_analog_input alc882_auto_init_analog_input
/* init callback for auto-configuration model -- overriding the default init */
static void alc268_auto_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
alc268_auto_init_multi_out(codec);
alc268_auto_init_hp_out(codec);
alc268_auto_init_mono_speaker_out(codec);
alc268_auto_init_analog_input(codec);
if (spec->unsol_event)
alc_inithook(codec);
}
/*
* configuration and preset
*/
static const char *alc268_models[ALC268_MODEL_LAST] = {
[ALC267_QUANTA_IL1] = "quanta-il1",
[ALC268_3ST] = "3stack",
[ALC268_TOSHIBA] = "toshiba",
[ALC268_ACER] = "acer",
[ALC268_ACER_DMIC] = "acer-dmic",
[ALC268_ACER_ASPIRE_ONE] = "acer-aspire",
[ALC268_DELL] = "dell",
[ALC268_ZEPTO] = "zepto",
#ifdef CONFIG_SND_DEBUG
[ALC268_TEST] = "test",
#endif
[ALC268_AUTO] = "auto",
};
static struct snd_pci_quirk alc268_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x011e, "Acer Aspire 5720z", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
ALC268_ACER_ASPIRE_ONE),
SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL),
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP TX25xx series",
ALC268_TOSHIBA),
SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
SND_PCI_QUIRK(0x1854, 0x1775, "LG R510", ALC268_DELL),
{}
};
/* Toshiba laptops have no unique PCI SSID but only codec SSID */
static struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
SND_PCI_QUIRK(0x1179, 0xff0a, "TOSHIBA X-200", ALC268_AUTO),
SND_PCI_QUIRK(0x1179, 0xff0e, "TOSHIBA X-200 HDMI", ALC268_AUTO),
SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05",
ALC268_TOSHIBA),
{}
};
static struct alc_config_preset alc268_presets[] = {
[ALC267_QUANTA_IL1] = {
.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
alc268_capture_nosrc_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc267_quanta_il1_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc267_quanta_il1_setup,
.init_hook = alc_inithook,
},
[ALC268_3ST] = {
.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.dig_out_nid = ALC268_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
},
[ALC268_TOSHIBA] = {
.mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_toshiba_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
.unsol_event = alc268_toshiba_unsol_event,
.setup = alc268_toshiba_setup,
.init_hook = alc268_toshiba_automute,
},
[ALC268_ACER] = {
.mixers = { alc268_acer_mixer, alc268_capture_nosrc_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_acer_capture_source,
.unsol_event = alc268_acer_unsol_event,
.init_hook = alc268_acer_init_hook,
},
[ALC268_ACER_DMIC] = {
.mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_acer_dmic_capture_source,
.unsol_event = alc268_acer_unsol_event,
.init_hook = alc268_acer_init_hook,
},
[ALC268_ACER_ASPIRE_ONE] = {
.mixers = { alc268_acer_aspire_one_mixer,
alc268_beep_mixer,
alc268_capture_nosrc_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_aspire_one_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.unsol_event = alc268_acer_lc_unsol_event,
.setup = alc268_acer_lc_setup,
.init_hook = alc268_acer_lc_init_hook,
},
[ALC268_DELL] = {
.mixers = { alc268_dell_mixer, alc268_beep_mixer,
alc268_capture_nosrc_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_dell_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc268_dell_setup,
.init_hook = alc_inithook,
},
[ALC268_ZEPTO] = {
.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_toshiba_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.dig_out_nid = ALC268_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
.setup = alc268_toshiba_setup,
.init_hook = alc268_toshiba_automute,
},
#ifdef CONFIG_SND_DEBUG
[ALC268_TEST] = {
.mixers = { alc268_test_mixer, alc268_capture_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_volume_init_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.dig_out_nid = ALC268_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
},
#endif
};
static int patch_alc268(struct hda_codec *codec)
{
struct alc_spec *spec;
int board_config;
int i, has_beep, err;
spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
board_config = snd_hda_check_board_config(codec, ALC268_MODEL_LAST,
alc268_models,
alc268_cfg_tbl);
if (board_config < 0 || board_config >= ALC268_MODEL_LAST)
board_config = snd_hda_check_board_codec_sid_config(codec,
ALC882_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl);
if (board_config < 0 || board_config >= ALC268_MODEL_LAST) {
printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
codec->chip_name);
board_config = ALC268_AUTO;
}
if (board_config == ALC268_AUTO) {
/* automatic parse from the BIOS config */
err = alc268_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 = ALC268_3ST;
}
}
if (board_config != ALC268_AUTO)
setup_preset(codec, &alc268_presets[board_config]);
spec->stream_analog_playback = &alc268_pcm_analog_playback;
spec->stream_analog_capture = &alc268_pcm_analog_capture;
spec->stream_analog_alt_capture = &alc268_pcm_analog_alt_capture;
spec->stream_digital_playback = &alc268_pcm_digital_playback;
has_beep = 0;
for (i = 0; i < spec->num_mixers; i++) {
if (spec->mixers[i] == alc268_beep_mixer) {
has_beep = 1;
break;
}
}
if (has_beep) {
err = snd_hda_attach_beep_device(codec, 0x1);
if (err < 0) {
alc_free(codec);
return err;
}
if (!query_amp_caps(codec, 0x1d, HDA_INPUT))
/* override the amp caps for beep generator */
snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT,
(0x0c << AC_AMPCAP_OFFSET_SHIFT) |
(0x0c << AC_AMPCAP_NUM_STEPS_SHIFT) |
(0x07 << AC_AMPCAP_STEP_SIZE_SHIFT) |
(0 << AC_AMPCAP_MUTE_SHIFT));
}
if (!spec->no_analog && !spec->adc_nids && spec->input_mux) {
/* check whether NID 0x07 is valid */
unsigned int wcap = get_wcaps(codec, 0x07);
int i;
/* get type */
wcap = get_wcaps_type(wcap);
if (spec->auto_mic ||
wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
spec->adc_nids = alc268_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt);
if (spec->auto_mic || spec->input_mux->num_items == 1)
add_mixer(spec, alc268_capture_nosrc_mixer);
else
add_mixer(spec, alc268_capture_alt_mixer);
} else {
spec->adc_nids = alc268_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids);
add_mixer(spec, alc268_capture_mixer);
}
spec->capsrc_nids = alc268_capsrc_nids;
/* set default input source */
for (i = 0; i < spec->num_adc_nids; i++)
snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i],
0, AC_VERB_SET_CONNECT_SEL,
i < spec->num_mux_defs ?
spec->input_mux[i].items[0].index :
spec->input_mux->items[0].index);
}
spec->vmaster_nid = 0x02;
codec->patch_ops = alc_patch_ops;
if (board_config == ALC268_AUTO)
spec->init_hook = alc268_auto_init;
codec->proc_widget_hook = print_realtek_coef;
return 0;
}
/*
* ALC269 channel source setting (2 channel)
*/
#define ALC269_DIGOUT_NID ALC880_DIGOUT_NID
#define alc269_dac_nids alc260_dac_nids
static hda_nid_t alc269_adc_nids[1] = {
/* ADC1 */
0x08,
};
static hda_nid_t alc269_capsrc_nids[1] = {
0x23,
};
/* NOTE: ADC2 (0x07) is connected from a recording *MIXER* (0x24),
* not a mux!
*/
#define alc269_modes alc260_modes
#define alc269_capture_source alc880_lg_lw_capture_source
static struct snd_kcontrol_new alc269_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
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("Mic Boost", 0x18, 0, 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("Front Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.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 = alc268_acer_master_sw_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
},
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT),
{ }
};
static struct snd_kcontrol_new alc269_lifebook_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.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 = alc268_acer_master_sw_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
},
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT),
HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT),
HDA_CODEC_VOLUME("Dock Mic Boost", 0x1b, 0, HDA_INPUT),
{ }
};
static struct snd_kcontrol_new alc269_eeepc_mixer[] = {
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
{ } /* end */
};
/* capture mixer elements */
static struct snd_kcontrol_new alc269_epc_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
{ } /* end */
};
/* FSC amilo */
#define alc269_fujitsu_mixer alc269_eeepc_mixer
static struct hda_verb alc269_quanta_fl1_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
static struct hda_verb alc269_lifebook_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? AMP_IN_MUTE(0) : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
AMP_IN_MUTE(0), bits);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 0x0c);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF, 0x680);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 0x0c);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF, 0x480);
}
/* toggle speaker-output according to the hp-jacks state */
static void alc269_lifebook_speaker_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
/* Check laptop headphone socket */
present = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
/* Check port replicator headphone socket */
present |= snd_hda_codec_read(codec, 0x1a, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? AMP_IN_MUTE(0) : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
AMP_IN_MUTE(0), bits);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 0x0c);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF, 0x680);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 0x0c);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF, 0x480);
}
static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec)
{
unsigned int present_laptop;
unsigned int present_dock;
present_laptop = snd_hda_codec_read(codec, 0x18, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
present_dock = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
/* Laptop mic port overrides dock mic port, design decision */
if (present_dock)
snd_hda_codec_write(codec, 0x23, 0,
AC_VERB_SET_CONNECT_SEL, 0x3);
if (present_laptop)
snd_hda_codec_write(codec, 0x23, 0,
AC_VERB_SET_CONNECT_SEL, 0x0);
if (!present_dock && !present_laptop)
snd_hda_codec_write(codec, 0x23, 0,
AC_VERB_SET_CONNECT_SEL, 0x1);
}
static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC880_HP_EVENT:
alc269_quanta_fl1_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
static void alc269_lifebook_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC880_HP_EVENT)
alc269_lifebook_speaker_automute(codec);
if ((res >> 26) == ALC880_MIC_EVENT)
alc269_lifebook_mic_autoswitch(codec);
}
static void alc269_quanta_fl1_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->ext_mic.pin = 0x18;
spec->ext_mic.mux_idx = 0;
spec->int_mic.pin = 0x19;
spec->int_mic.mux_idx = 1;
spec->auto_mic = 1;
}
static void alc269_quanta_fl1_init_hook(struct hda_codec *codec)
{
alc269_quanta_fl1_speaker_automute(codec);
alc_mic_automute(codec);
}
static void alc269_lifebook_init_hook(struct hda_codec *codec)
{
alc269_lifebook_speaker_automute(codec);
alc269_lifebook_mic_autoswitch(codec);
}
static struct hda_verb alc269_eeepc_dmic_init_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x23, AC_VERB_SET_CONNECT_SEL, 0x05},
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
static struct hda_verb alc269_eeepc_amic_init_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x23, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x701b | (0x00 << 8))},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
/* toggle speaker-output according to the hp-jack state */
static void alc269_speaker_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? AMP_IN_MUTE(0) : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
AMP_IN_MUTE(0), bits);
}
/* unsolicited event for HP jack sensing */
static void alc269_eeepc_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC880_HP_EVENT:
alc269_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
static void alc269_eeepc_dmic_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->ext_mic.pin = 0x18;
spec->ext_mic.mux_idx = 0;
spec->int_mic.pin = 0x12;
spec->int_mic.mux_idx = 5;
spec->auto_mic = 1;
}
static void alc269_eeepc_amic_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->ext_mic.pin = 0x18;
spec->ext_mic.mux_idx = 0;
spec->int_mic.pin = 0x19;
spec->int_mic.mux_idx = 1;
spec->auto_mic = 1;
}
static void alc269_eeepc_inithook(struct hda_codec *codec)
{
alc269_speaker_automute(codec);
alc_mic_automute(codec);
}
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc269_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Mute input amps (PCBeep, Line In, Mic 1 & Mic 2) of the
* analog-loopback mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for
* front panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* Set up output mixers (0x0c - 0x0e)
*/
/* set vol=0 to output mixers */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, 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, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{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},
{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_MUTE},
{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},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
/* set EAPD */
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
#define alc269_auto_create_multi_out_ctls \
alc268_auto_create_multi_out_ctls
#define alc269_auto_create_input_ctls \
alc268_auto_create_input_ctls
#ifdef CONFIG_SND_HDA_POWER_SAVE
#define alc269_loopbacks alc880_loopbacks
#endif
/* pcm configuration: identical with ALC880 */
#define alc269_pcm_analog_playback alc880_pcm_analog_playback
#define alc269_pcm_analog_capture alc880_pcm_analog_capture
#define alc269_pcm_digital_playback alc880_pcm_digital_playback
#define alc269_pcm_digital_capture alc880_pcm_digital_capture
static struct hda_pcm_stream alc269_44k_pcm_analog_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 8,
.rates = SNDRV_PCM_RATE_44100, /* fixed rate */
/* 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 alc269_44k_pcm_analog_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_44100, /* fixed rate */
/* NID is set in alc_build_pcms */
};
/*
* BIOS auto configuration
*/
static int alc269_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err;
static hda_nid_t alc269_ignore[] = { 0x1d, 0 };
err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc269_ignore);
if (err < 0)
return err;
err = alc269_auto_create_multi_out_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
err = alc269_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
if (spec->autocfg.dig_outs)
spec->multiout.dig_out_nid = ALC269_DIGOUT_NID;
if (spec->kctls.list)
add_mixer(spec, spec->kctls.list);
add_verb(spec, alc269_init_verbs);
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux[0];
/* set default input source */
snd_hda_codec_write_cache(codec, alc269_capsrc_nids[0],
0, AC_VERB_SET_CONNECT_SEL,
spec->input_mux->items[0].index);
err = alc_auto_add_mic_boost(codec);
if (err < 0)
return err;
if (!spec->cap_mixer && !spec->no_analog)
set_capture_mixer(codec);
alc_ssid_check(codec, 0x15, 0x1b, 0x14);
return 1;
}
#define alc269_auto_init_multi_out alc268_auto_init_multi_out
#define alc269_auto_init_hp_out alc268_auto_init_hp_out
#define alc269_auto_init_analog_input alc882_auto_init_analog_input
/* init callback for auto-configuration model -- overriding the default init */
static void alc269_auto_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
alc269_auto_init_multi_out(codec);
alc269_auto_init_hp_out(codec);
alc269_auto_init_analog_input(codec);
if (spec->unsol_event)
alc_inithook(codec);
}
/*
* configuration and preset
*/
static const char *alc269_models[ALC269_MODEL_LAST] = {
[ALC269_BASIC] = "basic",
[ALC269_QUANTA_FL1] = "quanta",
[ALC269_ASUS_EEEPC_P703] = "eeepc-p703",
[ALC269_ASUS_EEEPC_P901] = "eeepc-p901",
[ALC269_FUJITSU] = "fujitsu",
[ALC269_LIFEBOOK] = "lifebook"
};
static struct snd_pci_quirk alc269_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1),
SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
ALC269_ASUS_EEEPC_P703),
SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_ASUS_EEEPC_P703),
SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_ASUS_EEEPC_P703),
SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_ASUS_EEEPC_P703),
SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_ASUS_EEEPC_P703),
SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_ASUS_EEEPC_P703),
SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_ASUS_EEEPC_P703),
SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
ALC269_ASUS_EEEPC_P901),
SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
ALC269_ASUS_EEEPC_P901),
SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_ASUS_EEEPC_P901),
SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU),
SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
{}
};
static struct alc_config_preset alc269_presets[] = {
[ALC269_BASIC] = {
.mixers = { alc269_base_mixer },
.init_verbs = { alc269_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.input_mux = &alc269_capture_source,
},
[ALC269_QUANTA_FL1] = {
.mixers = { alc269_quanta_fl1_mixer },
.init_verbs = { alc269_init_verbs, alc269_quanta_fl1_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.input_mux = &alc269_capture_source,
.unsol_event = alc269_quanta_fl1_unsol_event,
.setup = alc269_quanta_fl1_setup,
.init_hook = alc269_quanta_fl1_init_hook,
},
[ALC269_ASUS_EEEPC_P703] = {
.mixers = { alc269_eeepc_mixer },
.cap_mixer = alc269_epc_capture_mixer,
.init_verbs = { alc269_init_verbs,
alc269_eeepc_amic_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.unsol_event = alc269_eeepc_unsol_event,
.setup = alc269_eeepc_amic_setup,
.init_hook = alc269_eeepc_inithook,
},
[ALC269_ASUS_EEEPC_P901] = {
.mixers = { alc269_eeepc_mixer },
.cap_mixer = alc269_epc_capture_mixer,
.init_verbs = { alc269_init_verbs,
alc269_eeepc_dmic_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.unsol_event = alc269_eeepc_unsol_event,
.setup = alc269_eeepc_dmic_setup,
.init_hook = alc269_eeepc_inithook,
},
[ALC269_FUJITSU] = {
.mixers = { alc269_fujitsu_mixer },
.cap_mixer = alc269_epc_capture_mixer,
.init_verbs = { alc269_init_verbs,
alc269_eeepc_dmic_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.unsol_event = alc269_eeepc_unsol_event,
.setup = alc269_eeepc_dmic_setup,
.init_hook = alc269_eeepc_inithook,
},
[ALC269_LIFEBOOK] = {
.mixers = { alc269_lifebook_mixer },
.init_verbs = { alc269_init_verbs, alc269_lifebook_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.input_mux = &alc269_capture_source,
.unsol_event = alc269_lifebook_unsol_event,
.init_hook = alc269_lifebook_init_hook,
},
};
static int patch_alc269(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;
alc_fix_pll_init(codec, 0x20, 0x04, 15);
board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST,
alc269_models,
alc269_cfg_tbl);
if (board_config < 0) {
printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
codec->chip_name);
board_config = ALC269_AUTO;
}
if (board_config == ALC269_AUTO) {
/* automatic parse from the BIOS config */
err = alc269_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 = ALC269_BASIC;
}
}
err = snd_hda_attach_beep_device(codec, 0x1);
if (err < 0) {
alc_free(codec);
return err;
}
if (board_config != ALC269_AUTO)
setup_preset(codec, &alc269_presets[board_config]);
if (codec->subsystem_id == 0x17aa3bf8) {
/* Due to a hardware problem on Lenovo Ideadpad, we need to
* fix the sample rate of analog I/O to 44.1kHz
*/
spec->stream_analog_playback = &alc269_44k_pcm_analog_playback;
spec->stream_analog_capture = &alc269_44k_pcm_analog_capture;
} else {
spec->stream_analog_playback = &alc269_pcm_analog_playback;
spec->stream_analog_capture = &alc269_pcm_analog_capture;
}
spec->stream_digital_playback = &alc269_pcm_digital_playback;
spec->stream_digital_capture = &alc269_pcm_digital_capture;
spec->adc_nids = alc269_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
spec->capsrc_nids = alc269_capsrc_nids;
if (!spec->cap_mixer)
set_capture_mixer(codec);
set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
spec->vmaster_nid = 0x02;
codec->patch_ops = alc_patch_ops;
if (board_config == ALC269_AUTO)
spec->init_hook = alc269_auto_init;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
spec->loopback.amplist = alc269_loopbacks;
#endif
codec->proc_widget_hook = print_realtek_coef;
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 },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
#endif
{ } /* 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 },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
#endif
{ } /* end */
};
static struct hda_channel_mode alc861_threestack_modes[2] = {
{ 2, alc861_threestack_ch2_init },
{ 6, alc861_threestack_ch6_init },
};
/* Set mic1 as input and unmute the mixer */
static struct hda_verb alc861_uniwill_m31_ch2_init[] = {
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
{ } /* end */
};
/* Set mic1 as output and mute mixer */
static struct hda_verb alc861_uniwill_m31_ch4_init[] = {
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
{ } /* end */
};
static struct hda_channel_mode alc861_uniwill_m31_modes[2] = {
{ 2, alc861_uniwill_m31_ch2_init },
{ 4, alc861_uniwill_m31_ch4_init },
};
/* Set mic1 and line-in as input and unmute the mixer */
static struct hda_verb alc861_asus_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 },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
#endif
{ } /* end */
};
/* Set mic1 nad line-in as output and mute mixer */
static struct hda_verb alc861_asus_ch6_init[] = {
/* set pin widget 1Ah (line in) for output (Back Surround)*/
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* { 0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
/* set pin widget 18h (mic1) for output (CLFE)*/
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
{ 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
#endif
{ } /* end */
};
static struct hda_channel_mode alc861_asus_modes[2] = {
{ 2, alc861_asus_ch2_init },
{ 6, alc861_asus_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),
{ } /* 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),
{
.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 */
};
static struct snd_kcontrol_new alc861_toshiba_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Master Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc861_uniwill_m31_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),
{
.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_uniwill_m31_modes),
},
{ } /* end */
};
static struct snd_kcontrol_new alc861_asus_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_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,
.private_value = ARRAY_SIZE(alc861_asus_modes),
},
{ }
};
/* additional mixer */
static struct snd_kcontrol_new alc861_asus_laptop_mixer[] = {
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
{ }
};
/*
* 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, 0x00 },
/* 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)},
/* hp used DAC 3 (Front) */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{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, 0x00 },
/* 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)},
/* hp used DAC 3 (Front) */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
static struct hda_verb alc861_uniwill_m31_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) */
/* this has to be set to VREF80 */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* 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)},
/* hp used DAC 3 (Front) */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
static struct hda_verb alc861_asus_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
/* port-A for surround (rear panel)
* according to codec#0 this is the HP jack
*/
{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, /* was 0x00 */
/* route front PCM to HP */
{ 0x0e, AC_VERB_SET_CONNECT_SEL, 0x01 },
/* 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) */
/* this has to be set to VREF80 */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* 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 },
/* port-H for side (rear panel) */
{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* 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)},
/* hp used DAC 3 (Front) */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
/* additional init verbs for ASUS laptops */
static struct hda_verb alc861_asus_laptop_init_verbs[] = {
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x45 }, /* HP-out */
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2) }, /* mute line-in */
{ }
};
/*
* 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_MUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(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_MUTE(2)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(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_MUTE(2)},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, /* set Mic 1 */
{ }
};
static struct hda_verb alc861_toshiba_init_verbs[] = {
{0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc861_toshiba_automute(struct hda_codec *codec)
{
unsigned int present;
present = snd_hda_codec_read(codec, 0x0f, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
snd_hda_codec_amp_stereo(codec, 0x16, HDA_INPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 3,
HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE);
}
static void alc861_toshiba_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC880_HP_EVENT)
alc861_toshiba_automute(codec);
}
/* pcm configuration: identical 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 alc660_dac_nids[3] = {
/* front, clfe, surround */
0x03, 0x05, 0x06
};
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 },
},
};
static hda_nid_t alc861_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
{
struct alc_spec *spec = codec->spec;
hda_nid_t mix, srcs[5];
int i, j, num;
if (snd_hda_get_connections(codec, pin, &mix, 1) != 1)
return 0;
num = snd_hda_get_connections(codec, mix, srcs, ARRAY_SIZE(srcs));
if (num < 0)
return 0;
for (i = 0; i < num; i++) {
unsigned int type;
type = get_wcaps_type(get_wcaps(codec, srcs[i]));
if (type != AC_WID_AUD_OUT)
continue;
for (j = 0; j < spec->multiout.num_dacs; j++)
if (spec->multiout.dac_nids[j] == srcs[i])
break;
if (j >= spec->multiout.num_dacs)
return srcs[i];
}
return 0;
}
/* fill in the dac_nids table from the parsed pin configuration */
static int alc861_auto_fill_dac_nids(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
struct alc_spec *spec = codec->spec;
int i;
hda_nid_t nid, dac;
spec->multiout.dac_nids = spec->private_dac_nids;
for (i = 0; i < cfg->line_outs; i++) {
nid = cfg->line_out_pins[i];
dac = alc861_look_for_dac(codec, nid);
if (!dac)
continue;
spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
}
return 0;
}
static int alc861_create_out_sw(struct hda_codec *codec, const char *pfx,
hda_nid_t nid, unsigned int chs)
{
char name[32];
snprintf(name, sizeof(name), "%s Playback Switch", pfx);
return add_control(codec->spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
}
/* add playback controls from the parsed DAC table */
static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
struct alc_spec *spec = codec->spec;
static const char *chname[4] = {
"Front", "Surround", NULL /*CLFE*/, "Side"
};
hda_nid_t nid;
int i, err;
if (cfg->line_outs == 1) {
const char *pfx = NULL;
if (!cfg->hp_outs)
pfx = "Master";
else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
pfx = "Speaker";
if (pfx) {
nid = spec->multiout.dac_nids[0];
return alc861_create_out_sw(codec, pfx, nid, 3);
}
}
for (i = 0; i < cfg->line_outs; i++) {
nid = spec->multiout.dac_nids[i];
if (!nid)
continue;
if (i == 2) {
/* Center/LFE */
err = alc861_create_out_sw(codec, "Center", nid, 1);
if (err < 0)
return err;
err = alc861_create_out_sw(codec, "LFE", nid, 2);
if (err < 0)
return err;
} else {
err = alc861_create_out_sw(codec, chname[i], nid, 3);
if (err < 0)
return err;
}
}
return 0;
}
static int alc861_auto_create_hp_ctls(struct hda_codec *codec, hda_nid_t pin)
{
struct alc_spec *spec = codec->spec;
int err;
hda_nid_t nid;
if (!pin)
return 0;
if ((pin >= 0x0b && pin <= 0x10) || pin == 0x1f || pin == 0x20) {
nid = alc861_look_for_dac(codec, pin);
if (nid) {
err = alc861_create_out_sw(codec, "Headphone", nid, 3);
if (err < 0)
return err;
spec->multiout.hp_nid = nid;
}
}
return 0;
}
/* create playback/capture controls for input pins */
static int alc861_auto_create_input_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x08, 0);
}
static void alc861_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid,
int pin_type, hda_nid_t dac)
{
hda_nid_t mix, srcs[5];
int i, num;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
pin_type);
snd_hda_codec_write(codec, dac, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_UNMUTE);
if (snd_hda_get_connections(codec, nid, &mix, 1) != 1)
return;
num = snd_hda_get_connections(codec, mix, srcs, ARRAY_SIZE(srcs));
if (num < 0)
return;
for (i = 0; i < num; i++) {
unsigned int mute;
if (srcs[i] == dac || srcs[i] == 0x15)
mute = AMP_IN_UNMUTE(i);
else
mute = AMP_IN_MUTE(i);
snd_hda_codec_write(codec, mix, 0, AC_VERB_SET_AMP_GAIN_MUTE,
mute);
}
}
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];
int pin_type = get_pin_type(spec->autocfg.line_out_type);
if (nid)
alc861_auto_set_output_and_unmute(codec, nid, pin_type,
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_pins[0];
if (pin)
alc861_auto_set_output_and_unmute(codec, pin, PIN_HP,
spec->multiout.hp_nid);
pin = spec->autocfg.speaker_pins[0];
if (pin)
alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT,
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)
alc_set_input_pin(codec, nid, i);
}
}
/* 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 };
err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc861_ignore);
if (err < 0)
return err;
if (!spec->autocfg.line_outs)
return 0; /* can't find valid BIOS pin config */
err = alc861_auto_fill_dac_nids(codec, &spec->autocfg);
if (err < 0)
return err;
err = alc861_auto_create_multi_out_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
err = alc861_auto_create_hp_ctls(codec, spec->autocfg.hp_pins[0]);
if (err < 0)
return err;
err = alc861_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
if (spec->autocfg.dig_outs)
spec->multiout.dig_out_nid = ALC861_DIGOUT_NID;
if (spec->kctls.list)
add_mixer(spec, spec->kctls.list);
add_verb(spec, 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 10:47:09 +00:00
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux[0];
spec->adc_nids = alc861_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids);
set_capture_mixer(codec);
alc_ssid_check(codec, 0x0e, 0x0f, 0x0b);
return 1;
}
/* additional initialization for auto-configuration model */
static void alc861_auto_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
alc861_auto_init_multi_out(codec);
alc861_auto_init_hp_out(codec);
alc861_auto_init_analog_input(codec);
if (spec->unsol_event)
alc_inithook(codec);
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
static struct hda_amp_list alc861_loopbacks[] = {
{ 0x15, HDA_INPUT, 0 },
{ 0x15, HDA_INPUT, 1 },
{ 0x15, HDA_INPUT, 2 },
{ 0x15, HDA_INPUT, 3 },
{ } /* end */
};
#endif
/*
* configuration and preset
*/
static const char *alc861_models[ALC861_MODEL_LAST] = {
[ALC861_3ST] = "3stack",
[ALC660_3ST] = "3stack-660",
[ALC861_3ST_DIG] = "3stack-dig",
[ALC861_6ST_DIG] = "6stack-dig",
[ALC861_UNIWILL_M31] = "uniwill-m31",
[ALC861_TOSHIBA] = "toshiba",
[ALC861_ASUS] = "asus",
[ALC861_ASUS_LAPTOP] = "asus-laptop",
[ALC861_AUTO] = "auto",
};
static struct snd_pci_quirk alc861_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC861_3ST),
SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP),
SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP),
SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS),
SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP),
SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG),
SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA),
/* FIXME: the entry below breaks Toshiba A100 (model=auto works!)
* Any other models that need this preset?
*/
/* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */
SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST),
SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST),
SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31),
SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31),
SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP),
/* FIXME: the below seems conflict */
/* SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), */
SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST),
SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST),
{}
};
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,
.need_dac_fix = 1,
.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,
.need_dac_fix = 1,
.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,
},
[ALC660_3ST] = {
.mixers = { alc861_3ST_mixer },
.init_verbs = { alc861_threestack_init_verbs },
.num_dacs = ARRAY_SIZE(alc660_dac_nids),
.dac_nids = alc660_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
.channel_mode = alc861_threestack_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_UNIWILL_M31] = {
.mixers = { alc861_uniwill_m31_mixer },
.init_verbs = { alc861_uniwill_m31_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_uniwill_m31_modes),
.channel_mode = alc861_uniwill_m31_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_TOSHIBA] = {
.mixers = { alc861_toshiba_mixer },
.init_verbs = { alc861_base_init_verbs,
alc861_toshiba_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
.unsol_event = alc861_toshiba_unsol_event,
.init_hook = alc861_toshiba_automute,
},
[ALC861_ASUS] = {
.mixers = { alc861_asus_mixer },
.init_verbs = { alc861_asus_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_asus_modes),
.channel_mode = alc861_asus_modes,
.need_dac_fix = 1,
.hp_nid = 0x06,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_ASUS_LAPTOP] = {
.mixers = { alc861_toshiba_mixer, alc861_asus_laptop_mixer },
.init_verbs = { alc861_asus_init_verbs,
alc861_asus_laptop_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(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.need_dac_fix = 1,
.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 = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
board_config = snd_hda_check_board_config(codec, ALC861_MODEL_LAST,
alc861_models,
alc861_cfg_tbl);
if (board_config < 0) {
printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
codec->chip_name);
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;
}
}
err = snd_hda_attach_beep_device(codec, 0x23);
if (err < 0) {
alc_free(codec);
return err;
}
if (board_config != ALC861_AUTO)
setup_preset(codec, &alc861_presets[board_config]);
spec->stream_analog_playback = &alc861_pcm_analog_playback;
spec->stream_analog_capture = &alc861_pcm_analog_capture;
spec->stream_digital_playback = &alc861_pcm_digital_playback;
spec->stream_digital_capture = &alc861_pcm_digital_capture;
set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
spec->vmaster_nid = 0x03;
codec->patch_ops = alc_patch_ops;
if (board_config == ALC861_AUTO)
spec->init_hook = alc861_auto_init;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
spec->loopback.amplist = alc861_loopbacks;
#endif
codec->proc_widget_hook = print_realtek_coef;
return 0;
}
/*
* ALC861-VD support
*
* Based on ALC882
*
* In addition, an independent DAC
*/
#define ALC861VD_DIGOUT_NID 0x06
static hda_nid_t alc861vd_dac_nids[4] = {
/* front, surr, clfe, side surr */
0x02, 0x03, 0x04, 0x05
};
/* dac_nids for ALC660vd are in a different order - according to
* Realtek's driver.
* This should probably result in a different mixer for 6stack models
* of ALC660vd codecs, but for now there is only 3stack mixer
* - and it is the same as in 861vd.
* adc_nids in ALC660vd are (is) the same as in 861vd
*/
static hda_nid_t alc660vd_dac_nids[3] = {
/* front, rear, clfe, rear_surr */
0x02, 0x04, 0x03
};
static hda_nid_t alc861vd_adc_nids[1] = {
/* ADC0 */
0x09,
};
static hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 };
/* input MUX */
/* FIXME: should be a matrix-type input source selection */
static struct hda_input_mux alc861vd_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
static struct hda_input_mux alc861vd_dallas_capture_source = {
.num_items = 2,
.items = {
{ "Ext Mic", 0x0 },
{ "Int Mic", 0x1 },
},
};
static struct hda_input_mux alc861vd_hp_capture_source = {
.num_items = 2,
.items = {
{ "Front Mic", 0x0 },
{ "ATAPI Mic", 0x1 },
},
};
/*
* 2ch mode
*/
static struct hda_channel_mode alc861vd_3stack_2ch_modes[1] = {
{ 2, NULL }
};
/*
* 6ch mode
*/
static struct hda_verb alc861vd_6stack_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 alc861vd_6stack_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 alc861vd_6stack_modes[2] = {
{ 6, alc861vd_6stack_ch6_init },
{ 8, alc861vd_6stack_ch8_init },
};
static struct snd_kcontrol_new alc861vd_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 */
};
/* 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 alc861vd_6st_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0,
HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 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", 0x05, 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("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),
HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, 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("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc861vd_3st_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
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),
HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, 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("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc861vd_lenovo_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
/*HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),*/
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
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),
HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, 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("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
{ } /* end */
};
/* Pin assignment: Speaker=0x14, HP = 0x15,
* Ext Mic=0x18, Int Mic = 0x19, CD = 0x1c, PC Beep = 0x1d
*/
static struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
/* Pin assignment: Speaker=0x14, Line-out = 0x15,
* Front Mic=0x18, ATAPI Mic = 0x19,
*/
static struct snd_kcontrol_new alc861vd_hp_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc861vd_volume_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
{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
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Capture mixer: unmute Mic, F-Mic, Line, CD inputs */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
/*
* Set up output mixers (0x02 - 0x05)
*/
/* set vol=0 to output mixers */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x05, 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 alc861vd_3stack_init_verbs[] = {
/*
* 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},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* 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},
{ }
};
/*
* 6-stack pin configuration:
*/
static struct hda_verb alc861vd_6stack_init_verbs[] = {
/*
* 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},
{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},
{ }
};
static struct hda_verb alc861vd_eapd_verbs[] = {
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
static struct hda_verb alc660vd_eapd_verbs[] = {
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
static struct hda_verb alc861vd_lenovo_unsol_verbs[] = {
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{}
};
static void alc861vd_lenovo_mic_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x18, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1,
HDA_AMP_MUTE, bits);
}
static void alc861vd_lenovo_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
}
static void alc861vd_lenovo_init_hook(struct hda_codec *codec)
{
alc_automute_amp(codec);
alc861vd_lenovo_mic_automute(codec);
}
static void alc861vd_lenovo_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC880_MIC_EVENT:
alc861vd_lenovo_mic_automute(codec);
break;
default:
alc_automute_amp_unsol_event(codec, res);
break;
}
}
static struct hda_verb alc861vd_dallas_verbs[] = {
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{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_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)},
{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_VREF50},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{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)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{ } /* end */
};
/* toggle speaker-output according to the hp-jack state */
static void alc861vd_dallas_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
#define alc861vd_loopbacks alc880_loopbacks
#endif
/* pcm configuration: identical with ALC880 */
#define alc861vd_pcm_analog_playback alc880_pcm_analog_playback
#define alc861vd_pcm_analog_capture alc880_pcm_analog_capture
#define alc861vd_pcm_digital_playback alc880_pcm_digital_playback
#define alc861vd_pcm_digital_capture alc880_pcm_digital_capture
/*
* configuration and preset
*/
static const char *alc861vd_models[ALC861VD_MODEL_LAST] = {
[ALC660VD_3ST] = "3stack-660",
[ALC660VD_3ST_DIG] = "3stack-660-digout",
[ALC660VD_ASUS_V1S] = "asus-v1s",
[ALC861VD_3ST] = "3stack",
[ALC861VD_3ST_DIG] = "3stack-digout",
[ALC861VD_6ST_DIG] = "6stack-digout",
[ALC861VD_LENOVO] = "lenovo",
[ALC861VD_DALLAS] = "dallas",
[ALC861VD_HP] = "hp",
[ALC861VD_AUTO] = "auto",
};
static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),
SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),
SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
/*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/
SND_PCI_QUIRK(0x1179, 0xff01, "Toshiba A135", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS),
SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG),
{}
};
static struct alc_config_preset alc861vd_presets[] = {
[ALC660VD_3ST] = {
.mixers = { alc861vd_3st_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
.dac_nids = alc660vd_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
},
[ALC660VD_3ST_DIG] = {
.mixers = { alc861vd_3st_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
.dac_nids = alc660vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
},
[ALC861VD_3ST] = {
.mixers = { alc861vd_3st_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
},
[ALC861VD_3ST_DIG] = {
.mixers = { alc861vd_3st_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
},
[ALC861VD_6ST_DIG] = {
.mixers = { alc861vd_6st_mixer, alc861vd_chmode_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_6stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861vd_6stack_modes),
.channel_mode = alc861vd_6stack_modes,
.input_mux = &alc861vd_capture_source,
},
[ALC861VD_LENOVO] = {
.mixers = { alc861vd_lenovo_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs,
alc861vd_eapd_verbs,
alc861vd_lenovo_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
.dac_nids = alc660vd_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
.unsol_event = alc861vd_lenovo_unsol_event,
.setup = alc861vd_lenovo_setup,
.init_hook = alc861vd_lenovo_init_hook,
},
[ALC861VD_DALLAS] = {
.mixers = { alc861vd_dallas_mixer },
.init_verbs = { alc861vd_dallas_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_dallas_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc861vd_dallas_setup,
.init_hook = alc_automute_amp,
},
[ALC861VD_HP] = {
.mixers = { alc861vd_hp_mixer },
.init_verbs = { alc861vd_dallas_verbs, alc861vd_eapd_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_hp_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc861vd_dallas_setup,
.init_hook = alc_automute_amp,
},
[ALC660VD_ASUS_V1S] = {
.mixers = { alc861vd_lenovo_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs,
alc861vd_eapd_verbs,
alc861vd_lenovo_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
.dac_nids = alc660vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
.unsol_event = alc861vd_lenovo_unsol_event,
.setup = alc861vd_lenovo_setup,
.init_hook = alc861vd_lenovo_init_hook,
},
};
/*
* BIOS auto configuration
*/
static int alc861vd_auto_create_input_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0);
}
static void alc861vd_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type, int dac_idx)
{
alc_set_pin_output(codec, nid, pin_type);
}
static void alc861vd_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];
int pin_type = get_pin_type(spec->autocfg.line_out_type);
if (nid)
alc861vd_auto_set_output_and_unmute(codec, nid,
pin_type, i);
}
}
static void alc861vd_auto_init_hp_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t pin;
pin = spec->autocfg.hp_pins[0];
if (pin) /* connect to front and use dac 0 */
alc861vd_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
pin = spec->autocfg.speaker_pins[0];
if (pin)
alc861vd_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
}
#define ALC861VD_PIN_CD_NID ALC880_PIN_CD_NID
static void alc861vd_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 (alc_is_input_pin(codec, nid)) {
alc_set_input_pin(codec, nid, i);
if (nid != ALC861VD_PIN_CD_NID &&
(get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_MUTE);
}
}
}
#define alc861vd_auto_init_input_src alc882_auto_init_input_src
#define alc861vd_idx_to_mixer_vol(nid) ((nid) + 0x02)
#define alc861vd_idx_to_mixer_switch(nid) ((nid) + 0x0c)
/* add playback controls from the parsed DAC table */
/* Based on ALC880 version. But ALC861VD has separate,
* different NIDs for mute/unmute switch and volume control */
static int alc861vd_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", "CLFE", "Side"};
hda_nid_t nid_v, nid_s;
int i, err;
for (i = 0; i < cfg->line_outs; i++) {
if (!spec->multiout.dac_nids[i])
continue;
nid_v = alc861vd_idx_to_mixer_vol(
alc880_dac_to_idx(
spec->multiout.dac_nids[i]));
nid_s = alc861vd_idx_to_mixer_switch(
alc880_dac_to_idx(
spec->multiout.dac_nids[i]));
if (i == 2) {
/* Center/LFE */
err = add_control(spec, ALC_CTL_WIDGET_VOL,
"Center Playback Volume",
HDA_COMPOSE_AMP_VAL(nid_v, 1, 0,
HDA_OUTPUT));
if (err < 0)
return err;
err = add_control(spec, ALC_CTL_WIDGET_VOL,
"LFE Playback Volume",
HDA_COMPOSE_AMP_VAL(nid_v, 2, 0,
HDA_OUTPUT));
if (err < 0)
return err;
err = add_control(spec, ALC_CTL_BIND_MUTE,
"Center Playback Switch",
HDA_COMPOSE_AMP_VAL(nid_s, 1, 2,
HDA_INPUT));
if (err < 0)
return err;
err = add_control(spec, ALC_CTL_BIND_MUTE,
"LFE Playback Switch",
HDA_COMPOSE_AMP_VAL(nid_s, 2, 2,
HDA_INPUT));
if (err < 0)
return err;
} else {
const char *pfx;
if (cfg->line_outs == 1 &&
cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
if (!cfg->hp_pins)
pfx = "Speaker";
else
pfx = "PCM";
} else
pfx = chname[i];
sprintf(name, "%s Playback Volume", pfx);
err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(nid_v, 3, 0,
HDA_OUTPUT));
if (err < 0)
return err;
if (cfg->line_outs == 1 &&
cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
pfx = "Speaker";
sprintf(name, "%s Playback Switch", pfx);
err = add_control(spec, ALC_CTL_BIND_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid_s, 3, 2,
HDA_INPUT));
if (err < 0)
return err;
}
}
return 0;
}
/* add playback controls for speaker and HP outputs */
/* Based on ALC880 version. But ALC861VD has separate,
* different NIDs for mute/unmute switch and volume control */
static int alc861vd_auto_create_extra_out(struct alc_spec *spec,
hda_nid_t pin, const char *pfx)
{
hda_nid_t nid_v, nid_s;
int err;
char name[32];
if (!pin)
return 0;
if (alc880_is_fixed_pin(pin)) {
nid_v = 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_v;
else
spec->multiout.extra_out_nid[0] = nid_v;
/* control HP volume/switch on the output mixer amp */
nid_v = alc861vd_idx_to_mixer_vol(
alc880_fixed_pin_idx(pin));
nid_s = alc861vd_idx_to_mixer_switch(
alc880_fixed_pin_idx(pin));
sprintf(name, "%s Playback Volume", pfx);
err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(nid_v, 3, 0, HDA_OUTPUT));
if (err < 0)
return err;
sprintf(name, "%s Playback Switch", pfx);
err = add_control(spec, ALC_CTL_BIND_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid_s, 3, 2, HDA_INPUT));
if (err < 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);
err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
if (err < 0)
return err;
}
return 0;
}
/* 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
* Based on ALC880 version - had to change it to override
* alc880_auto_create_extra_out and alc880_auto_create_multi_out_ctls */
static int alc861vd_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err;
static hda_nid_t alc861vd_ignore[] = { 0x1d, 0 };
err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc861vd_ignore);
if (err < 0)
return err;
if (!spec->autocfg.line_outs)
return 0; /* can't find valid BIOS pin config */
err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
if (err < 0)
return err;
err = alc861vd_auto_create_multi_out_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
err = alc861vd_auto_create_extra_out(spec,
spec->autocfg.speaker_pins[0],
"Speaker");
if (err < 0)
return err;
err = alc861vd_auto_create_extra_out(spec,
spec->autocfg.hp_pins[0],
"Headphone");
if (err < 0)
return err;
err = alc861vd_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
if (spec->autocfg.dig_outs)
spec->multiout.dig_out_nid = ALC861VD_DIGOUT_NID;
if (spec->kctls.list)
add_mixer(spec, spec->kctls.list);
add_verb(spec, alc861vd_volume_init_verbs);
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux[0];
err = alc_auto_add_mic_boost(codec);
if (err < 0)
return err;
alc_ssid_check(codec, 0x15, 0x1b, 0x14);
return 1;
}
/* additional initialization for auto-configuration model */
static void alc861vd_auto_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
alc861vd_auto_init_multi_out(codec);
alc861vd_auto_init_hp_out(codec);
alc861vd_auto_init_analog_input(codec);
alc861vd_auto_init_input_src(codec);
if (spec->unsol_event)
alc_inithook(codec);
}
static int patch_alc861vd(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, ALC861VD_MODEL_LAST,
alc861vd_models,
alc861vd_cfg_tbl);
if (board_config < 0 || board_config >= ALC861VD_MODEL_LAST) {
printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
codec->chip_name);
board_config = ALC861VD_AUTO;
}
if (board_config == ALC861VD_AUTO) {
/* automatic parse from the BIOS config */
err = alc861vd_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 = ALC861VD_3ST;
}
}
err = snd_hda_attach_beep_device(codec, 0x23);
if (err < 0) {
alc_free(codec);
return err;
}
if (board_config != ALC861VD_AUTO)
setup_preset(codec, &alc861vd_presets[board_config]);
if (codec->vendor_id == 0x10ec0660) {
/* always turn on EAPD */
add_verb(spec, alc660vd_eapd_verbs);
}
spec->stream_analog_playback = &alc861vd_pcm_analog_playback;
spec->stream_analog_capture = &alc861vd_pcm_analog_capture;
spec->stream_digital_playback = &alc861vd_pcm_digital_playback;
spec->stream_digital_capture = &alc861vd_pcm_digital_capture;
if (!spec->adc_nids) {
spec->adc_nids = alc861vd_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids);
}
if (!spec->capsrc_nids)
spec->capsrc_nids = alc861vd_capsrc_nids;
set_capture_mixer(codec);
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
spec->vmaster_nid = 0x02;
codec->patch_ops = alc_patch_ops;
if (board_config == ALC861VD_AUTO)
spec->init_hook = alc861vd_auto_init;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
spec->loopback.amplist = alc861vd_loopbacks;
#endif
codec->proc_widget_hook = print_realtek_coef;
return 0;
}
/*
* ALC662 support
*
* ALC662 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 ALC662_DIGOUT_NID 0x06
#define ALC662_DIGIN_NID 0x0a
static hda_nid_t alc662_dac_nids[4] = {
/* front, rear, clfe, rear_surr */
0x02, 0x03, 0x04
};
static hda_nid_t alc272_dac_nids[2] = {
0x02, 0x03
};
static hda_nid_t alc662_adc_nids[2] = {
/* ADC1-2 */
0x09, 0x08
};
static hda_nid_t alc272_adc_nids[1] = {
/* ADC1-2 */
0x08,
};
static hda_nid_t alc662_capsrc_nids[2] = { 0x22, 0x23 };
static hda_nid_t alc272_capsrc_nids[1] = { 0x23 };
/* input MUX */
/* FIXME: should be a matrix-type input source selection */
static struct hda_input_mux alc662_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
static struct hda_input_mux alc662_lenovo_101e_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x1 },
{ "Line", 0x2 },
},
};
static struct hda_input_mux alc663_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
},
};
#if 0 /* set to 1 for testing other input sources below */
static struct hda_input_mux alc272_nc10_capture_source = {
.num_items = 16,
.items = {
{ "Autoselect Mic", 0x0 },
{ "Internal Mic", 0x1 },
{ "In-0x02", 0x2 },
{ "In-0x03", 0x3 },
{ "In-0x04", 0x4 },
{ "In-0x05", 0x5 },
{ "In-0x06", 0x6 },
{ "In-0x07", 0x7 },
{ "In-0x08", 0x8 },
{ "In-0x09", 0x9 },
{ "In-0x0a", 0x0a },
{ "In-0x0b", 0x0b },
{ "In-0x0c", 0x0c },
{ "In-0x0d", 0x0d },
{ "In-0x0e", 0x0e },
{ "In-0x0f", 0x0f },
},
};
#endif
/*
* 2ch mode
*/
static struct hda_channel_mode alc662_3ST_2ch_modes[1] = {
{ 2, NULL }
};
/*
* 2ch mode
*/
static struct hda_verb alc662_3ST_ch2_init[] = {
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ 0x18, 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 },
{ } /* end */
};
/*
* 6ch mode
*/
static struct hda_verb alc662_3ST_ch6_init[] = {
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
{ } /* end */
};
static struct hda_channel_mode alc662_3ST_6ch_modes[2] = {
{ 2, alc662_3ST_ch2_init },
{ 6, alc662_3ST_ch6_init },
};
/*
* 2ch mode
*/
static struct hda_verb alc662_sixstack_ch6_init[] = {
{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ } /* end */
};
/*
* 6ch mode
*/
static struct hda_verb alc662_sixstack_ch8_init[] = {
{ 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 alc662_5stack_modes[2] = {
{ 2, alc662_sixstack_ch6_init },
{ 6, alc662_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 alc662_base_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x3, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
/*Input mixer control */
HDA_CODEC_VOLUME("CD Playback Volume", 0xb, 0x4, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0xb, 0x4, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0xb, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0xb, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0xb, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0xb, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0xb, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0xb, 0x01, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, 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),
{ } /* end */
};
static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, 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),
{ } /* end */
};
static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x03, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
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, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_VOLUME("e-Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("e-Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("e-Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("i-Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = {
ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("MuteCtrl Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
static struct hda_bind_ctls alc663_asus_bind_master_vol = {
.ops = &snd_hda_bind_vol,
.values = {
HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
0
},
};
static struct hda_bind_ctls alc663_asus_one_bind_switch = {
.ops = &snd_hda_bind_sw,
.values = {
HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
0
},
};
static struct snd_kcontrol_new alc663_m51va_mixer[] = {
HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
HDA_BIND_SW("Master Playback Switch", &alc663_asus_one_bind_switch),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
static struct hda_bind_ctls alc663_asus_tree_bind_switch = {
.ops = &snd_hda_bind_sw,
.values = {
HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
0
},
};
static struct snd_kcontrol_new alc663_two_hp_m1_mixer[] = {
HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
HDA_BIND_SW("Master Playback Switch", &alc663_asus_tree_bind_switch),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct hda_bind_ctls alc663_asus_four_bind_switch = {
.ops = &snd_hda_bind_sw,
.values = {
HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
0
},
};
static struct snd_kcontrol_new alc663_two_hp_m2_mixer[] = {
HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
HDA_BIND_SW("Master Playback Switch", &alc663_asus_four_bind_switch),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc662_1bjd_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct hda_bind_ctls alc663_asus_two_bind_master_vol = {
.ops = &snd_hda_bind_vol,
.values = {
HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x04, 3, 0, HDA_OUTPUT),
0
},
};
static struct hda_bind_ctls alc663_asus_two_bind_switch = {
.ops = &snd_hda_bind_sw,
.values = {
HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_OUTPUT),
0
},
};
static struct snd_kcontrol_new alc663_asus_21jd_clfe_mixer[] = {
HDA_BIND_VOL("Master Playback Volume",
&alc663_asus_two_bind_master_vol),
HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc663_asus_15jd_clfe_mixer[] = {
HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc663_g71v_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Front Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc663_g50v_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc662_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 alc662_init_verbs[] = {
/* ADC: mute amp left and right */
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Front mixer: unmute input/output amp left and right (volume = 0) */
{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)},
{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)},
/* Front Pin: output 0 (0x0c) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Rear Pin: output 1 (0x0d) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* CLFE Pin: output 2 (0x0e) */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* 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 mixer */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* always trun on EAPD */
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
static struct hda_verb alc662_sue_init_verbs[] = {
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_FRONT_EVENT},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT},
{}
};
static struct hda_verb alc662_eeepc_sue_init_verbs[] = {
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
/* Set Unsolicited Event*/
static struct hda_verb alc662_eeepc_ep20_sue_init_verbs[] = {
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc662_auto_init_verbs[] = {
/*
* Unmute ADC and set the default input to mic-in
*/
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for front
* panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* Set up output mixers (0x0c - 0x0f)
*/
/* set vol=0 to output mixers */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x04, 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 mixer */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{ }
};
/* additional verbs for ALC663 */
static struct hda_verb alc663_auto_init_verbs[] = {
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{ }
};
static struct hda_verb alc663_m51va_init_verbs[] = {
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
static struct hda_verb alc663_21jd_amic_init_verbs[] = {
{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
static struct hda_verb alc662_1bjd_amic_init_verbs[] = {
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{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}, /* Headphone */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
static struct hda_verb alc663_15jd_amic_init_verbs[] = {
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
static struct hda_verb alc663_two_hp_amic_m1_init_verbs[] = {
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x21, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Headphone */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Headphone */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
static struct hda_verb alc663_two_hp_amic_m2_init_verbs[] = {
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{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, 0x01}, /* Headphone */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
static struct hda_verb alc663_g71v_init_verbs[] = {
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */
/* {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, */ /* Headphone */
{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_FRONT_EVENT},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_MIC_EVENT},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT},
{}
};
static struct hda_verb alc663_g50v_init_verbs[] = {
{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
static struct hda_verb alc662_ecs_init_verbs[] = {
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
static struct hda_verb alc272_dell_zm1_init_verbs[] = {
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
static struct hda_verb alc272_dell_init_verbs[] = {
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{}
};
static struct snd_kcontrol_new alc662_auto_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc272_auto_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
{ } /* end */
};
static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
}
static void alc662_lenovo_101e_all_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
}
static void alc662_lenovo_101e_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC880_HP_EVENT)
alc662_lenovo_101e_all_automute(codec);
if ((res >> 26) == ALC880_FRONT_EVENT)
alc662_lenovo_101e_ispeaker_automute(codec);
}
/* unsolicited event for HP jack sensing */
static void alc662_eeepc_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC880_MIC_EVENT)
alc_mic_automute(codec);
else
alc262_hippo_unsol_event(codec, res);
}
static void alc662_eeepc_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
alc262_hippo1_setup(codec);
spec->ext_mic.pin = 0x18;
spec->ext_mic.mux_idx = 0;
spec->int_mic.pin = 0x19;
spec->int_mic.mux_idx = 1;
spec->auto_mic = 1;
}
static void alc662_eeepc_inithook(struct hda_codec *codec)
{
alc262_hippo_automute(codec);
alc_mic_automute(codec);
}
static void alc662_eeepc_ep20_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x1b;
}
#define alc662_eeepc_ep20_inithook alc262_hippo_master_update
static void alc663_m51va_speaker_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x21, 0,
AC_VERB_GET_PIN_SENSE, 0)
& AC_PINSENSE_PRESENCE;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
AMP_IN_MUTE(0), bits);
}
static void alc663_21jd_two_speaker_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x21, 0,
AC_VERB_GET_PIN_SENSE, 0)
& AC_PINSENSE_PRESENCE;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
AMP_IN_MUTE(0), bits);
snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
AMP_IN_MUTE(0), bits);
}
static void alc663_15jd_two_speaker_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0)
& AC_PINSENSE_PRESENCE;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
AMP_IN_MUTE(0), bits);
snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
AMP_IN_MUTE(0), bits);
}
static void alc662_f5z_speaker_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0)
& AC_PINSENSE_PRESENCE;
bits = present ? 0 : PIN_OUT;
snd_hda_codec_write(codec, 0x14, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, bits);
}
static void alc663_two_hp_m1_speaker_automute(struct hda_codec *codec)
{
unsigned int present1, present2;
present1 = snd_hda_codec_read(codec, 0x21, 0,
AC_VERB_GET_PIN_SENSE, 0)
& AC_PINSENSE_PRESENCE;
present2 = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0)
& AC_PINSENSE_PRESENCE;
if (present1 || present2) {
snd_hda_codec_write_cache(codec, 0x14, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
} else {
snd_hda_codec_write_cache(codec, 0x14, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
}
}
static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec)
{
unsigned int present1, present2;
present1 = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0)
& AC_PINSENSE_PRESENCE;
present2 = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0)
& AC_PINSENSE_PRESENCE;
if (present1 || present2) {
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), AMP_IN_MUTE(0));
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
AMP_IN_MUTE(0), AMP_IN_MUTE(0));
} else {
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), 0);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
AMP_IN_MUTE(0), 0);
}
}
static void alc663_m51va_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC880_HP_EVENT:
alc663_m51va_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
static void alc663_m51va_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->ext_mic.pin = 0x18;
spec->ext_mic.mux_idx = 0;
spec->int_mic.pin = 0x12;
spec->int_mic.mux_idx = 1;
spec->auto_mic = 1;
}
static void alc663_m51va_inithook(struct hda_codec *codec)
{
alc663_m51va_speaker_automute(codec);
alc_mic_automute(codec);
}
/* ***************** Mode1 ******************************/
#define alc663_mode1_unsol_event alc663_m51va_unsol_event
#define alc663_mode1_setup alc663_m51va_setup
#define alc663_mode1_inithook alc663_m51va_inithook
/* ***************** Mode2 ******************************/
static void alc662_mode2_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC880_HP_EVENT:
alc662_f5z_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
#define alc662_mode2_setup alc663_m51va_setup
static void alc662_mode2_inithook(struct hda_codec *codec)
{
alc662_f5z_speaker_automute(codec);
alc_mic_automute(codec);
}
/* ***************** Mode3 ******************************/
static void alc663_mode3_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC880_HP_EVENT:
alc663_two_hp_m1_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
#define alc663_mode3_setup alc663_m51va_setup
static void alc663_mode3_inithook(struct hda_codec *codec)
{
alc663_two_hp_m1_speaker_automute(codec);
alc_mic_automute(codec);
}
/* ***************** Mode4 ******************************/
static void alc663_mode4_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC880_HP_EVENT:
alc663_21jd_two_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
#define alc663_mode4_setup alc663_m51va_setup
static void alc663_mode4_inithook(struct hda_codec *codec)
{
alc663_21jd_two_speaker_automute(codec);
alc_mic_automute(codec);
}
/* ***************** Mode5 ******************************/
static void alc663_mode5_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC880_HP_EVENT:
alc663_15jd_two_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
#define alc663_mode5_setup alc663_m51va_setup
static void alc663_mode5_inithook(struct hda_codec *codec)
{
alc663_15jd_two_speaker_automute(codec);
alc_mic_automute(codec);
}
/* ***************** Mode6 ******************************/
static void alc663_mode6_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC880_HP_EVENT:
alc663_two_hp_m2_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
#define alc663_mode6_setup alc663_m51va_setup
static void alc663_mode6_inithook(struct hda_codec *codec)
{
alc663_two_hp_m2_speaker_automute(codec);
alc_mic_automute(codec);
}
static void alc663_g71v_hp_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x21, 0,
AC_VERB_GET_PIN_SENSE, 0)
& AC_PINSENSE_PRESENCE;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
}
static void alc663_g71v_front_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0)
& AC_PINSENSE_PRESENCE;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
}
static void alc663_g71v_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC880_HP_EVENT:
alc663_g71v_hp_automute(codec);
break;
case ALC880_FRONT_EVENT:
alc663_g71v_front_automute(codec);
break;
case ALC880_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
#define alc663_g71v_setup alc663_m51va_setup
static void alc663_g71v_inithook(struct hda_codec *codec)
{
alc663_g71v_front_automute(codec);
alc663_g71v_hp_automute(codec);
alc_mic_automute(codec);
}
static void alc663_g50v_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC880_HP_EVENT:
alc663_m51va_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
#define alc663_g50v_setup alc663_m51va_setup
static void alc663_g50v_inithook(struct hda_codec *codec)
{
alc663_m51va_speaker_automute(codec);
alc_mic_automute(codec);
}
static struct snd_kcontrol_new alc662_ecs_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_VOLUME("e-Mic/LineIn Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("e-Mic/LineIn Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("e-Mic/LineIn Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("i-Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
static struct snd_kcontrol_new alc272_nc10_mixer[] = {
/* Master Playback automatically created from Speaker and Headphone */
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
{ } /* end */
};
#ifdef CONFIG_SND_HDA_POWER_SAVE
#define alc662_loopbacks alc880_loopbacks
#endif
/* pcm configuration: identical with ALC880 */
#define alc662_pcm_analog_playback alc880_pcm_analog_playback
#define alc662_pcm_analog_capture alc880_pcm_analog_capture
#define alc662_pcm_digital_playback alc880_pcm_digital_playback
#define alc662_pcm_digital_capture alc880_pcm_digital_capture
/*
* configuration and preset
*/
static const char *alc662_models[ALC662_MODEL_LAST] = {
[ALC662_3ST_2ch_DIG] = "3stack-dig",
[ALC662_3ST_6ch_DIG] = "3stack-6ch-dig",
[ALC662_3ST_6ch] = "3stack-6ch",
[ALC662_5ST_DIG] = "6stack-dig",
[ALC662_LENOVO_101E] = "lenovo-101e",
[ALC662_ASUS_EEEPC_P701] = "eeepc-p701",
[ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20",
[ALC662_ECS] = "ecs",
[ALC663_ASUS_M51VA] = "m51va",
[ALC663_ASUS_G71V] = "g71v",
[ALC663_ASUS_H13] = "h13",
[ALC663_ASUS_G50V] = "g50v",
[ALC663_ASUS_MODE1] = "asus-mode1",
[ALC662_ASUS_MODE2] = "asus-mode2",
[ALC663_ASUS_MODE3] = "asus-mode3",
[ALC663_ASUS_MODE4] = "asus-mode4",
[ALC663_ASUS_MODE5] = "asus-mode5",
[ALC663_ASUS_MODE6] = "asus-mode6",
[ALC272_DELL] = "dell",
[ALC272_DELL_ZM1] = "dell-zm1",
[ALC272_SAMSUNG_NC10] = "samsung-nc10",
[ALC662_AUTO] = "auto",
};
static struct snd_pci_quirk alc662_cfg_tbl[] = {
SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS),
SND_PCI_QUIRK(0x1028, 0x02d6, "DELL", ALC272_DELL),
SND_PCI_QUIRK(0x1028, 0x02f4, "DELL ZM1", ALC272_DELL_ZM1),
SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3),
SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3),
SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6),
SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6),
SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x17b3, "ASUS F70SL", ALC663_ASUS_MODE3),
SND_PCI_QUIRK(0x1043, 0x17c3, "ASUS UX20", ALC663_ASUS_M51VA),
SND_PCI_QUIRK(0x1043, 0x17f3, "ASUS X58LE", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5),
SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6),
SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x1853, "ASUS F50Z", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA),
/*SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),*/
SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3),
SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3),
SND_PCI_QUIRK(0x1043, 0x18b3, "ASUS N80Vc", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS N81Te", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x18f3, "ASUS N505Tp", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x1943, "ASUS Vx3V", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3),
SND_PCI_QUIRK(0x1043, 0x1983, "ASUS N5051A", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V),
/*SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),*/
SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x19d3, "ASUS NB", ALC663_ASUS_M51VA),
SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4),
SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701),
SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20),
SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA),
SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0",
ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x",
ALC663_ASUS_H13),
{}
};
static struct alc_config_preset alc662_presets[] = {
[ALC662_3ST_2ch_DIG] = {
.mixers = { alc662_3ST_2ch_mixer },
.init_verbs = { alc662_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.dig_out_nid = ALC662_DIGOUT_NID,
.dig_in_nid = ALC662_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
.input_mux = &alc662_capture_source,
},
[ALC662_3ST_6ch_DIG] = {
.mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
.init_verbs = { alc662_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.dig_out_nid = ALC662_DIGOUT_NID,
.dig_in_nid = ALC662_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
.channel_mode = alc662_3ST_6ch_modes,
.need_dac_fix = 1,
.input_mux = &alc662_capture_source,
},
[ALC662_3ST_6ch] = {
.mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
.init_verbs = { alc662_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
.channel_mode = alc662_3ST_6ch_modes,
.need_dac_fix = 1,
.input_mux = &alc662_capture_source,
},
[ALC662_5ST_DIG] = {
.mixers = { alc662_base_mixer, alc662_chmode_mixer },
.init_verbs = { alc662_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.dig_out_nid = ALC662_DIGOUT_NID,
.dig_in_nid = ALC662_DIGIN_NID,
.num_channel_mode = ARRAY_SIZE(alc662_5stack_modes),
.channel_mode = alc662_5stack_modes,
.input_mux = &alc662_capture_source,
},
[ALC662_LENOVO_101E] = {
.mixers = { alc662_lenovo_101e_mixer },
.init_verbs = { alc662_init_verbs, alc662_sue_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
.input_mux = &alc662_lenovo_101e_capture_source,
.unsol_event = alc662_lenovo_101e_unsol_event,
.init_hook = alc662_lenovo_101e_all_automute,
},
[ALC662_ASUS_EEEPC_P701] = {
.mixers = { alc662_eeepc_p701_mixer },
.init_verbs = { alc662_init_verbs,
alc662_eeepc_sue_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
.unsol_event = alc662_eeepc_unsol_event,
.setup = alc662_eeepc_setup,
.init_hook = alc662_eeepc_inithook,
},
[ALC662_ASUS_EEEPC_EP20] = {
.mixers = { alc662_eeepc_ep20_mixer,
alc662_chmode_mixer },
.init_verbs = { alc662_init_verbs,
alc662_eeepc_ep20_sue_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
.channel_mode = alc662_3ST_6ch_modes,
.input_mux = &alc662_lenovo_101e_capture_source,
.unsol_event = alc662_eeepc_unsol_event,
.setup = alc662_eeepc_ep20_setup,
.init_hook = alc662_eeepc_ep20_inithook,
},
[ALC662_ECS] = {
.mixers = { alc662_ecs_mixer },
.init_verbs = { alc662_init_verbs,
alc662_ecs_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
.unsol_event = alc662_eeepc_unsol_event,
.setup = alc662_eeepc_setup,
.init_hook = alc662_eeepc_inithook,
},
[ALC663_ASUS_M51VA] = {
.mixers = { alc663_m51va_mixer },
.init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
.unsol_event = alc663_m51va_unsol_event,
.setup = alc663_m51va_setup,
.init_hook = alc663_m51va_inithook,
},
[ALC663_ASUS_G71V] = {
.mixers = { alc663_g71v_mixer },
.init_verbs = { alc662_init_verbs, alc663_g71v_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
.unsol_event = alc663_g71v_unsol_event,
.setup = alc663_g71v_setup,
.init_hook = alc663_g71v_inithook,
},
[ALC663_ASUS_H13] = {
.mixers = { alc663_m51va_mixer },
.init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
.unsol_event = alc663_m51va_unsol_event,
.init_hook = alc663_m51va_inithook,
},
[ALC663_ASUS_G50V] = {
.mixers = { alc663_g50v_mixer },
.init_verbs = { alc662_init_verbs, alc663_g50v_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
.channel_mode = alc662_3ST_6ch_modes,
.input_mux = &alc663_capture_source,
.unsol_event = alc663_g50v_unsol_event,
.setup = alc663_g50v_setup,
.init_hook = alc663_g50v_inithook,
},
[ALC663_ASUS_MODE1] = {
.mixers = { alc663_m51va_mixer },
.cap_mixer = alc662_auto_capture_mixer,
.init_verbs = { alc662_init_verbs,
alc663_21jd_amic_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.hp_nid = 0x03,
.dac_nids = alc662_dac_nids,
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
.unsol_event = alc663_mode1_unsol_event,
.setup = alc663_mode1_setup,
.init_hook = alc663_mode1_inithook,
},
[ALC662_ASUS_MODE2] = {
.mixers = { alc662_1bjd_mixer },
.cap_mixer = alc662_auto_capture_mixer,
.init_verbs = { alc662_init_verbs,
alc662_1bjd_amic_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
.unsol_event = alc662_mode2_unsol_event,
.setup = alc662_mode2_setup,
.init_hook = alc662_mode2_inithook,
},
[ALC663_ASUS_MODE3] = {
.mixers = { alc663_two_hp_m1_mixer },
.cap_mixer = alc662_auto_capture_mixer,
.init_verbs = { alc662_init_verbs,
alc663_two_hp_amic_m1_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.hp_nid = 0x03,
.dac_nids = alc662_dac_nids,
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
.unsol_event = alc663_mode3_unsol_event,
.setup = alc663_mode3_setup,
.init_hook = alc663_mode3_inithook,
},
[ALC663_ASUS_MODE4] = {
.mixers = { alc663_asus_21jd_clfe_mixer },
.cap_mixer = alc662_auto_capture_mixer,
.init_verbs = { alc662_init_verbs,
alc663_21jd_amic_init_verbs},
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.hp_nid = 0x03,
.dac_nids = alc662_dac_nids,
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
.unsol_event = alc663_mode4_unsol_event,
.setup = alc663_mode4_setup,
.init_hook = alc663_mode4_inithook,
},
[ALC663_ASUS_MODE5] = {
.mixers = { alc663_asus_15jd_clfe_mixer },
.cap_mixer = alc662_auto_capture_mixer,
.init_verbs = { alc662_init_verbs,
alc663_15jd_amic_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.hp_nid = 0x03,
.dac_nids = alc662_dac_nids,
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
.unsol_event = alc663_mode5_unsol_event,
.setup = alc663_mode5_setup,
.init_hook = alc663_mode5_inithook,
},
[ALC663_ASUS_MODE6] = {
.mixers = { alc663_two_hp_m2_mixer },
.cap_mixer = alc662_auto_capture_mixer,
.init_verbs = { alc662_init_verbs,
alc663_two_hp_amic_m2_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.hp_nid = 0x03,
.dac_nids = alc662_dac_nids,
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
.unsol_event = alc663_mode6_unsol_event,
.setup = alc663_mode6_setup,
.init_hook = alc663_mode6_inithook,
},
[ALC272_DELL] = {
.mixers = { alc663_m51va_mixer },
.cap_mixer = alc272_auto_capture_mixer,
.init_verbs = { alc662_init_verbs, alc272_dell_init_verbs },
.num_dacs = ARRAY_SIZE(alc272_dac_nids),
.dac_nids = alc662_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.adc_nids = alc272_adc_nids,
.num_adc_nids = ARRAY_SIZE(alc272_adc_nids),
.capsrc_nids = alc272_capsrc_nids,
.channel_mode = alc662_3ST_2ch_modes,
.unsol_event = alc663_m51va_unsol_event,
.setup = alc663_m51va_setup,
.init_hook = alc663_m51va_inithook,
},
[ALC272_DELL_ZM1] = {
.mixers = { alc663_m51va_mixer },
.cap_mixer = alc662_auto_capture_mixer,
.init_verbs = { alc662_init_verbs, alc272_dell_zm1_init_verbs },
.num_dacs = ARRAY_SIZE(alc272_dac_nids),
.dac_nids = alc662_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.adc_nids = alc662_adc_nids,
.num_adc_nids = 1,
.capsrc_nids = alc662_capsrc_nids,
.channel_mode = alc662_3ST_2ch_modes,
.unsol_event = alc663_m51va_unsol_event,
.setup = alc663_m51va_setup,
.init_hook = alc663_m51va_inithook,
},
[ALC272_SAMSUNG_NC10] = {
.mixers = { alc272_nc10_mixer },
.init_verbs = { alc662_init_verbs,
alc663_21jd_amic_init_verbs },
.num_dacs = ARRAY_SIZE(alc272_dac_nids),
.dac_nids = alc272_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
/*.input_mux = &alc272_nc10_capture_source,*/
.unsol_event = alc663_mode4_unsol_event,
.setup = alc663_mode4_setup,
.init_hook = alc663_mode4_inithook,
},
};
/*
* BIOS auto configuration
*/
/* add playback controls from the parsed DAC table */
static int alc662_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_dac(i);
if (i == 2) {
/* Center/LFE */
err = add_control(spec, ALC_CTL_WIDGET_VOL,
"Center Playback Volume",
HDA_COMPOSE_AMP_VAL(nid, 1, 0,
HDA_OUTPUT));
if (err < 0)
return err;
err = add_control(spec, ALC_CTL_WIDGET_VOL,
"LFE Playback Volume",
HDA_COMPOSE_AMP_VAL(nid, 2, 0,
HDA_OUTPUT));
if (err < 0)
return err;
err = add_control(spec, ALC_CTL_WIDGET_MUTE,
"Center Playback Switch",
HDA_COMPOSE_AMP_VAL(0x0e, 1, 0,
HDA_INPUT));
if (err < 0)
return err;
err = add_control(spec, ALC_CTL_WIDGET_MUTE,
"LFE Playback Switch",
HDA_COMPOSE_AMP_VAL(0x0e, 2, 0,
HDA_INPUT));
if (err < 0)
return err;
} else {
const char *pfx;
if (cfg->line_outs == 1 &&
cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
if (!cfg->hp_pins)
pfx = "Speaker";
else
pfx = "PCM";
} else
pfx = chname[i];
sprintf(name, "%s Playback Volume", pfx);
err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 0,
HDA_OUTPUT));
if (err < 0)
return err;
if (cfg->line_outs == 1 &&
cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
pfx = "Speaker";
sprintf(name, "%s Playback Switch", pfx);
err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(alc880_idx_to_mixer(i),
3, 0, HDA_INPUT));
if (err < 0)
return err;
}
}
return 0;
}
/* add playback controls for speaker and HP outputs */
static int alc662_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 (pin == 0x17) {
/* ALC663 has a mono output pin on 0x17 */
sprintf(name, "%s Playback Switch", pfx);
err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(pin, 2, 0, HDA_OUTPUT));
return err;
}
if (alc880_is_fixed_pin(pin)) {
nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
/* printk(KERN_DEBUG "DAC nid=%x\n",nid); */
/* 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_dac(alc880_fixed_pin_idx(pin));
sprintf(name, "%s Playback Volume", pfx);
err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
if (err < 0)
return err;
sprintf(name, "%s Playback Switch", pfx);
err = add_control(spec, ALC_CTL_BIND_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT));
if (err < 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);
err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
if (err < 0)
return err;
}
return 0;
}
/* create playback/capture controls for input pins */
#define alc662_auto_create_input_ctls \
alc880_auto_create_input_ctls
static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
int dac_idx)
{
alc_set_pin_output(codec, nid, pin_type);
/* 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 alc662_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];
int pin_type = get_pin_type(spec->autocfg.line_out_type);
if (nid)
alc662_auto_set_output_and_unmute(codec, nid, pin_type,
i);
}
}
static void alc662_auto_init_hp_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t pin;
pin = spec->autocfg.hp_pins[0];
if (pin) /* connect to front */
/* use dac 0 */
alc662_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
pin = spec->autocfg.speaker_pins[0];
if (pin)
alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
}
#define ALC662_PIN_CD_NID ALC880_PIN_CD_NID
static void alc662_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 (alc_is_input_pin(codec, nid)) {
alc_set_input_pin(codec, nid, i);
if (nid != ALC662_PIN_CD_NID &&
(get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_MUTE);
}
}
}
#define alc662_auto_init_input_src alc882_auto_init_input_src
static int alc662_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int err;
static hda_nid_t alc662_ignore[] = { 0x1d, 0 };
err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc662_ignore);
if (err < 0)
return err;
if (!spec->autocfg.line_outs)
return 0; /* can't find valid BIOS pin config */
err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
if (err < 0)
return err;
err = alc662_auto_create_multi_out_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
err = alc662_auto_create_extra_out(spec,
spec->autocfg.speaker_pins[0],
"Speaker");
if (err < 0)
return err;
err = alc662_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
"Headphone");
if (err < 0)
return err;
err = alc662_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
if (spec->autocfg.dig_outs)
spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
if (spec->kctls.list)
add_mixer(spec, spec->kctls.list);
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux[0];
add_verb(spec, alc662_auto_init_verbs);
if (codec->vendor_id == 0x10ec0663)
add_verb(spec, alc663_auto_init_verbs);
err = alc_auto_add_mic_boost(codec);
if (err < 0)
return err;
alc_ssid_check(codec, 0x15, 0x1b, 0x14);
return 1;
}
/* additional initialization for auto-configuration model */
static void alc662_auto_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
alc662_auto_init_multi_out(codec);
alc662_auto_init_hp_out(codec);
alc662_auto_init_analog_input(codec);
alc662_auto_init_input_src(codec);
if (spec->unsol_event)
alc_inithook(codec);
}
static int patch_alc662(struct hda_codec *codec)
{
struct alc_spec *spec;
int err, board_config;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (!spec)
return -ENOMEM;
codec->spec = spec;
alc_fix_pll_init(codec, 0x20, 0x04, 15);
board_config = snd_hda_check_board_config(codec, ALC662_MODEL_LAST,
alc662_models,
alc662_cfg_tbl);
if (board_config < 0) {
printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
codec->chip_name);
board_config = ALC662_AUTO;
}
if (board_config == ALC662_AUTO) {
/* automatic parse from the BIOS config */
err = alc662_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 = ALC662_3ST_2ch_DIG;
}
}
err = snd_hda_attach_beep_device(codec, 0x1);
if (err < 0) {
alc_free(codec);
return err;
}
if (board_config != ALC662_AUTO)
setup_preset(codec, &alc662_presets[board_config]);
spec->stream_analog_playback = &alc662_pcm_analog_playback;
spec->stream_analog_capture = &alc662_pcm_analog_capture;
spec->stream_digital_playback = &alc662_pcm_digital_playback;
spec->stream_digital_capture = &alc662_pcm_digital_capture;
if (!spec->adc_nids) {
spec->adc_nids = alc662_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
}
if (!spec->capsrc_nids)
spec->capsrc_nids = alc662_capsrc_nids;
if (!spec->cap_mixer)
set_capture_mixer(codec);
if (codec->vendor_id == 0x10ec0662)
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
else
set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
spec->vmaster_nid = 0x02;
codec->patch_ops = alc_patch_ops;
if (board_config == ALC662_AUTO)
spec->init_hook = alc662_auto_init;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
spec->loopback.amplist = alc662_loopbacks;
#endif
codec->proc_widget_hook = print_realtek_coef;
return 0;
}
/*
* patch entries
*/
static struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
{ .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
{ .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 },
{ .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 },
{ .id = 0x10ec0269, .name = "ALC269", .patch = patch_alc269 },
{ .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 },
{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
.patch = patch_alc861 },
{ .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
{ .id = 0x10ec0861, .name = "ALC861", .patch = patch_alc861 },
{ .id = 0x10ec0862, .name = "ALC861-VD", .patch = patch_alc861vd },
{ .id = 0x10ec0662, .rev = 0x100002, .name = "ALC662 rev2",
.patch = patch_alc882 },
{ .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1",
.patch = patch_alc662 },
{ .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 },
{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
{ .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 },
{ .id = 0x10ec0885, .rev = 0x100101, .name = "ALC889A",
.patch = patch_alc882 },
{ .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
.patch = patch_alc882 },
{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
{ .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 },
{ .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
.patch = patch_alc882 },
{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc882 },
{ .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 },
{} /* terminator */
};
MODULE_ALIAS("snd-hda-codec-id:10ec*");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Realtek HD-audio codec");
static struct hda_codec_preset_list realtek_list = {
.preset = snd_hda_preset_realtek,
.owner = THIS_MODULE,
};
static int __init patch_realtek_init(void)
{
return snd_hda_add_codec_preset(&realtek_list);
}
static void __exit patch_realtek_exit(void)
{
snd_hda_delete_codec_preset(&realtek_list);
}
module_init(patch_realtek_init)
module_exit(patch_realtek_exit)