diff options
Diffstat (limited to 'sound/pci/hda/patch_realtek.c')
| -rw-r--r-- | sound/pci/hda/patch_realtek.c | 1840 |
1 files changed, 1360 insertions, 480 deletions
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9a47eec5a27b..53b0428abfc2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -102,6 +102,8 @@ enum { /* ALC268 models */ enum { ALC268_3ST, + ALC268_TOSHIBA, + ALC268_ACER, ALC268_AUTO, ALC268_MODEL_LAST /* last tag */ }; @@ -129,6 +131,7 @@ enum { ALC861VD_6ST_DIG, ALC861VD_LENOVO, ALC861VD_DALLAS, + ALC861VD_HP, ALC861VD_AUTO, ALC861VD_MODEL_LAST, }; @@ -140,6 +143,7 @@ enum { ALC662_3ST_6ch, ALC662_5ST_DIG, ALC662_LENOVO_101E, + ALC662_ASUS_EEEPC_P701, ALC662_AUTO, ALC662_MODEL_LAST, }; @@ -152,7 +156,9 @@ enum { ALC882_W2JC, ALC882_TARGA, ALC882_ASUS_A7J, + ALC882_ASUS_A7M, ALC885_MACPRO, + ALC885_MBP3, ALC885_IMAC24, ALC882_AUTO, ALC882_MODEL_LAST, @@ -167,12 +173,14 @@ enum { ALC883_TARGA_DIG, ALC883_TARGA_2ch_DIG, ALC883_ACER, + ALC883_ACER_ASPIRE, ALC883_MEDION, ALC883_MEDION_MD2, ALC883_LAPTOP_EAPD, ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, - ALC888_LENOVO_MS7195_DIG, + ALC888_LENOVO_MS7195_DIG, + ALC883_HAIER_W66, ALC888_6ST_HP, ALC888_3ST_HP, ALC883_AUTO, @@ -239,6 +247,10 @@ struct alc_spec { /* for pin sensing */ unsigned int sense_updated: 1; unsigned int jack_present: 1; + +#ifdef CONFIG_SND_HDA_POWER_SAVE + struct hda_loopback_check loopback; +#endif }; /* @@ -263,6 +275,9 @@ struct alc_config_preset { const struct hda_input_mux *input_mux; void (*unsol_event)(struct hda_codec *, unsigned int); void (*init_hook)(struct hda_codec *); +#ifdef CONFIG_SND_HDA_POWER_SAVE + struct hda_amp_list *loopbacks; +#endif }; @@ -441,8 +456,9 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, change = pinctl != alc_pin_mode_values[val]; if (change) { /* Set pin mode to that requested */ - snd_hda_codec_write(codec,nid,0,AC_VERB_SET_PIN_WIDGET_CONTROL, - alc_pin_mode_values[val]); + 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 @@ -455,19 +471,15 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, * this turns out to be necessary in the future. */ if (val <= 2) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_MUTE); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_IN_UNMUTE(0)); + 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_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_IN_MUTE(0)); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); + 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; @@ -486,15 +498,7 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, * needed for any "production" models. */ #ifdef CONFIG_SND_DEBUG -static int alc_gpio_data_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#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) @@ -527,7 +531,8 @@ static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, gpio_data &= ~mask; else gpio_data |= mask; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_GPIO_DATA, gpio_data); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_GPIO_DATA, gpio_data); return change; } @@ -547,15 +552,7 @@ static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, * necessary. */ #ifdef CONFIG_SND_DEBUG -static int alc_spdif_ctrl_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#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) @@ -588,8 +585,8 @@ static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, ctrl_data &= ~mask; else ctrl_data |= mask; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, - ctrl_data); + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, + ctrl_data); return change; } @@ -638,6 +635,9 @@ static void setup_preset(struct alc_spec *spec, spec->unsol_event = preset->unsol_event; spec->init_hook = preset->init_hook; +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = preset->loopbacks; +#endif } /* Enable GPIO mask and set output */ @@ -662,6 +662,44 @@ static struct hda_verb alc_gpio3_init_verbs[] = { { } }; +static void alc_sku_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + unsigned int mute; + unsigned int present; + unsigned int hp_nid = spec->autocfg.hp_pins[0]; + unsigned int sp_nid = spec->autocfg.speaker_pins[0]; + + /* need to execute and sync at first */ + snd_hda_codec_read(codec, hp_nid, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, hp_nid, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & 0x80000000) != 0; + if (spec->jack_present) { + /* mute internal speaker */ + snd_hda_codec_amp_stereo(codec, sp_nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + } else { + /* unmute internal speaker if necessary */ + mute = snd_hda_codec_amp_read(codec, hp_nid, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, sp_nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } +} + +/* unsolicited event for HP jack sensing */ +static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if (codec->vendor_id == 0x10ec0880) + res >>= 28; + else + res >>= 26; + if (res != ALC880_HP_EVENT) + return; + + alc_sku_automute(codec); +} + /* 32-bit subsystem ID for BIOS loading in HD Audio codec. * 31 ~ 16 : Manufacture ID * 15 ~ 8 : SKU ID @@ -672,13 +710,48 @@ static void alc_subsystem_id(struct hda_codec *codec, unsigned int porta, unsigned int porte, unsigned int portd) { - unsigned int ass, tmp; + unsigned int ass, tmp, i; + unsigned nid; + struct alc_spec *spec = codec->spec; - ass = codec->subsystem_id; - if (!(ass & 1)) + ass = codec->subsystem_id & 0xffff; + if ((ass != codec->bus->pci->subsystem_device) && (ass & 1)) + goto do_sku; + + /* + * 31~30 : port conetcivity + * 29~21 : reserve + * 20 : PCBEEP input + * 19~16 : Check sum (15:1) + * 15~1 : Custom + * 0 : override + */ + nid = 0x1d; + if (codec->vendor_id == 0x10ec0260) + nid = 0x17; + ass = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + if (!(ass & 1) && !(ass & 0x100000)) + return; + if ((ass >> 30) != 1) /* no physical connection */ return; - /* Override */ + /* check sum */ + tmp = 0; + for (i = 1; i < 16; i++) { + if ((ass >> i) && 1) + tmp++; + } + if (((ass >> 16) & 0xf) != tmp) + return; +do_sku: + /* + * 0 : override + * 1 : Swap Jack + * 2 : 0 --> Desktop, 1 --> Laptop + * 3~5 : External Amplifier control + * 7~6 : Reserved + */ tmp = (ass & 0x38) >> 3; /* external Amp control */ switch (tmp) { case 1: @@ -690,38 +763,108 @@ static void alc_subsystem_id(struct hda_codec *codec, case 7: snd_hda_sequence_write(codec, alc_gpio3_init_verbs); break; - case 5: + case 5: /* set EAPD output high */ switch (codec->vendor_id) { - case 0x10ec0862: - case 0x10ec0660: - case 0x10ec0662: + case 0x10ec0260: + snd_hda_codec_write(codec, 0x0f, 0, + AC_VERB_SET_EAPD_BTLENABLE, 2); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_EAPD_BTLENABLE, 2); + break; + case 0x10ec0262: case 0x10ec0267: case 0x10ec0268: + case 0x10ec0269: + case 0x10ec0862: + case 0x10ec0662: snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_EAPD_BTLENABLE, 2); snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_EAPD_BTLENABLE, 2); - return; + break; } - case 6: - if (ass & 4) { /* bit 2 : 0 = Desktop, 1 = Laptop */ - hda_nid_t port = 0; - tmp = (ass & 0x1800) >> 11; - switch (tmp) { - case 0: port = porta; break; - case 1: port = porte; break; - case 2: port = portd; break; - } - if (port) - snd_hda_codec_write(codec, port, 0, - AC_VERB_SET_EAPD_BTLENABLE, - 2); + switch (codec->vendor_id) { + case 0x10ec0260: + snd_hda_codec_write(codec, 0x1a, 0, + AC_VERB_SET_COEF_INDEX, 7); + tmp = snd_hda_codec_read(codec, 0x1a, 0, + AC_VERB_GET_PROC_COEF, 0); + snd_hda_codec_write(codec, 0x1a, 0, + AC_VERB_SET_COEF_INDEX, 7); + snd_hda_codec_write(codec, 0x1a, 0, + AC_VERB_SET_PROC_COEF, + tmp | 0x2010); + break; + case 0x10ec0262: + case 0x10ec0880: + case 0x10ec0882: + case 0x10ec0883: + case 0x10ec0885: + case 0x10ec0888: + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_COEF_INDEX, 7); + tmp = snd_hda_codec_read(codec, 0x20, 0, + AC_VERB_GET_PROC_COEF, 0); + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_COEF_INDEX, 7); + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, + tmp | 0x2010); + break; + case 0x10ec0267: + case 0x10ec0268: + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_COEF_INDEX, 7); + tmp = snd_hda_codec_read(codec, 0x20, 0, + AC_VERB_GET_PROC_COEF, 0); + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_COEF_INDEX, 7); + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, + tmp | 0x3000); + break; } - 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 == 5 ? 0x3040 : 0x3050)); + default: break; } + + /* is laptop and enable the function "Mute internal speaker + * when the external headphone out jack is plugged" + */ + if (!(ass & 0x4) || !(ass & 0x8000)) + return; + /* + * 10~8 : Jack location + * 12~11: Headphone out -> 00: PortA, 01: PortE, 02: PortD, 03: Resvered + * 14~13: Resvered + * 15 : 1 --> enable the function "Mute internal speaker + * when the external headphone out jack is plugged" + */ + if (!spec->autocfg.speaker_pins[0]) { + if (spec->multiout.dac_nids[0]) + spec->autocfg.speaker_pins[0] = + spec->multiout.dac_nids[0]; + else + return; + } + + if (!spec->autocfg.hp_pins[0]) { + tmp = (ass >> 11) & 0x3; /* HP to chassis */ + if (tmp == 0) + spec->autocfg.hp_pins[0] = porta; + else if (tmp == 1) + spec->autocfg.hp_pins[0] = porte; + else if (tmp == 2) + spec->autocfg.hp_pins[0] = portd; + else + return; + } + + snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | ALC880_HP_EVENT); + spec->unsol_event = alc_sku_unsol_event; + spec->init_hook = alc_sku_automute; } /* @@ -1304,11 +1447,13 @@ static struct hda_verb alc880_volume_init_verbs[] = { * panel mic (mic 2) */ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + {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) @@ -1568,15 +1713,11 @@ static void alc880_uniwill_hp_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0x80 : 0; - snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x16, 0, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x16, 1, HDA_OUTPUT, 0, - 0x80, bits); + 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, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); } /* auto-toggle front mic */ @@ -1587,11 +1728,8 @@ static void alc880_uniwill_mic_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0x80 : 0; - snd_hda_codec_amp_update(codec, 0x0b, 0, HDA_INPUT, 1, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x0b, 1, HDA_INPUT, 1, - 0x80, bits); + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); } static void alc880_uniwill_automute(struct hda_codec *codec) @@ -1623,11 +1761,8 @@ static void alc880_uniwill_p53_hp_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0x80 : 0; - snd_hda_codec_amp_update(codec, 0x15, 0, HDA_INPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x15, 1, HDA_INPUT, 0, - 0x80, bits); + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x15, HDA_INPUT, 0, HDA_AMP_MUTE, bits); } static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec) @@ -1635,19 +1770,14 @@ 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) & 0x7f; - - snd_hda_codec_amp_update(codec, 0x0c, 0, HDA_OUTPUT, 0, - 0x7f, present); - snd_hda_codec_amp_update(codec, 0x0c, 1, HDA_OUTPUT, 0, - 0x7f, present); - - snd_hda_codec_amp_update(codec, 0x0d, 0, HDA_OUTPUT, 0, - 0x7f, present); - snd_hda_codec_amp_update(codec, 0x0d, 1, HDA_OUTPUT, 0, - 0x7f, present); - + 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) { @@ -1868,8 +1998,8 @@ static struct hda_verb alc880_lg_init_verbs[] = { {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* mute all amp mixer inputs */ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)}, + {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}, @@ -1900,11 +2030,9 @@ static void alc880_lg_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0x80 : 0; - snd_hda_codec_amp_update(codec, 0x17, 0, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x17, 1, HDA_OUTPUT, 0, - 0x80, bits); + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); } static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res) @@ -1973,7 +2101,7 @@ static struct hda_verb alc880_lg_lw_init_verbs[] = { {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)}, + {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}, @@ -1999,11 +2127,9 @@ static void alc880_lg_lw_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0x80 : 0; - snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, - 0x80, bits); + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); } static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res) @@ -2015,6 +2141,24 @@ static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res) alc880_lg_lw_automute(codec); } +#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 */ @@ -2041,24 +2185,11 @@ static void alc_unsol_event(struct hda_codec *codec, unsigned int res) spec->unsol_event(codec, res); } -#ifdef CONFIG_PM -/* - * resume - */ -static int alc_resume(struct hda_codec *codec) +#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; - int i; - - alc_init(codec); - for (i = 0; i < spec->num_mixers; i++) - snd_hda_resume_ctls(codec, spec->mixers[i]); - if (spec->multiout.dig_out_nid) - snd_hda_resume_spdif_out(codec); - if (spec->dig_in_nid) - snd_hda_resume_spdif_in(codec); - - return 0; + return snd_hda_check_amp_list_power(codec, &spec->loopback, nid); } #endif @@ -2293,8 +2424,8 @@ static struct hda_codec_ops alc_patch_ops = { .init = alc_init, .free = alc_free, .unsol_event = alc_unsol_event, -#ifdef CONFIG_PM - .resume = alc_resume, +#ifdef CONFIG_SND_HDA_POWER_SAVE + .check_power_status = alc_check_power_status, #endif }; @@ -2392,11 +2523,14 @@ static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); new_ctl = ctls[ucontrol->value.enumerated.item[0]]; if (old_ctl != new_ctl) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, new_ctl); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - (ucontrol->value.enumerated.item[0] >= 3 ? - 0xb080 : 0xb000)); + 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; @@ -2439,7 +2573,8 @@ static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol, sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3; if (ucontrol->value.enumerated.item[0] != sel) { sel = ucontrol->value.enumerated.item[0] & 3; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, sel); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, sel); return 1; } return 0; @@ -2885,6 +3020,7 @@ static struct alc_config_preset alc880_presets[] = { 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, @@ -2916,6 +3052,9 @@ static struct alc_config_preset alc880_presets[] = { .input_mux = &alc880_lg_capture_source, .unsol_event = alc880_lg_unsol_event, .init_hook = alc880_lg_automute, +#ifdef CONFIG_SND_HDA_POWER_SAVE + .loopbacks = alc880_lg_loopbacks, +#endif }, [ALC880_LG_LW] = { .mixers = { alc880_lg_lw_mixer }, @@ -3399,6 +3538,10 @@ static int patch_alc880(struct hda_codec *codec) 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 return 0; } @@ -3747,12 +3890,12 @@ static struct hda_verb alc260_init_verbs[] = { /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & * Line In 2 = 0x03 */ - /* mute CD */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, - /* mute Line In */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - /* mute Mic */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + /* 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)}, @@ -3797,12 +3940,12 @@ static struct hda_verb alc260_hp_init_verbs[] = { /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & * Line In 2 = 0x03 */ - /* unmute CD */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, - /* unmute Line In */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, - /* unmute Mic */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + /* 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))}, @@ -3847,12 +3990,12 @@ static struct hda_verb alc260_hp_3013_init_verbs[] = { /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & * Line In 2 = 0x03 */ - /* unmute CD */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, - /* unmute Line In */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, - /* unmute Mic */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + /* 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))}, @@ -4069,13 +4212,17 @@ static void alc260_replacer_672v_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x0f, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; if (present) { - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 1); - snd_hda_codec_write(codec, 0x0f, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + 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(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0); - snd_hda_codec_write(codec, 0x0f, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + 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); } } @@ -4470,11 +4617,12 @@ static struct hda_verb alc260_volume_init_verbs[] = { * front panel mic (mic 2) */ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + /* 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) @@ -4551,6 +4699,17 @@ static void alc260_auto_init(struct hda_codec *codec) alc260_auto_init_analog_input(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 */ @@ -4750,6 +4909,10 @@ static int patch_alc260(struct hda_codec *codec) 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 return 0; } @@ -4812,12 +4975,13 @@ static int alc882_mux_enum_put(struct snd_kcontrol *kcontrol, idx = ucontrol->value.enumerated.item[0]; if (idx >= imux->num_items) idx = imux->num_items - 1; - if (*cur_val == idx && !codec->in_resume) + if (*cur_val == idx) return 0; for (i = 0; i < imux->num_items; i++) { - unsigned int v = (i == idx) ? 0x7000 : 0x7080; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - v | (imux->items[i].index << 8)); + 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; @@ -4879,6 +5043,38 @@ static struct hda_channel_mode alc882_sixstack_modes[2] = { { 8, alc882_sixstack_ch8_init }, }; +/* + * macbook pro ALC885 can switch LineIn to LineOut without loosing 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 }, +}; + + /* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b */ @@ -4909,6 +5105,19 @@ static struct snd_kcontrol_new alc882_base_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc885_mbp3_mixer[] = { + HDA_CODEC_VOLUME("Master Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Master Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_MUTE ("Speaker Switch", 0x14, 0x00, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line Out Volume", 0x0d,0x00, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line In Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE ("Line In 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 In Boost", 0x1a, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 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), @@ -4934,8 +5143,10 @@ static struct snd_kcontrol_new alc882_targa_mixer[] = { 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 */ }; @@ -4955,6 +5166,23 @@ static struct snd_kcontrol_new alc882_asus_a7j_mixer[] = { 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), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -5119,6 +5347,66 @@ static struct hda_verb alc882_macpro_init_verbs[] = { { } }; +/* 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), @@ -5154,14 +5442,10 @@ static void alc885_imac24_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_update(codec, 0x18, 0, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); - snd_hda_codec_amp_update(codec, 0x18, 1, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); - snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); - snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } /* Processes unsolicited events. */ @@ -5173,6 +5457,27 @@ static void alc885_imac24_unsol_event(struct hda_codec *codec, alc885_imac24_automute(codec); } +static void alc885_mbp3_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x15, 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 ? 0 : HDA_AMP_MUTE); + +} +static void alc885_mbp3_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_mbp3_automute(codec); +} + + 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)}, @@ -5198,11 +5503,10 @@ static void alc882_targa_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); - snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); - snd_hda_codec_write(codec, 1, 0, AC_VERB_SET_GPIO_DATA, present ? 1 : 3); + snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA, + present ? 1 : 3); } static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res) @@ -5233,6 +5537,24 @@ static struct hda_verb alc882_asus_a7j_verbs[] = { { } /* 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; @@ -5265,6 +5587,20 @@ static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted) 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); + alc885_imac24_automute(codec); +} + /* * generic initialization of ADC, input mixers and output mixers */ @@ -5279,17 +5615,17 @@ static struct hda_verb alc882_auto_init_verbs[] = { {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 + /* 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_UNMUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + {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) @@ -5378,6 +5714,10 @@ static struct snd_kcontrol_new alc882_capture_mixer[] = { { } /* end */ }; +#ifdef CONFIG_SND_HDA_POWER_SAVE +#define alc882_loopbacks alc880_loopbacks +#endif + /* pcm configuration: identiacal with ALC880 */ #define alc882_pcm_analog_playback alc880_pcm_analog_playback #define alc882_pcm_analog_capture alc880_pcm_analog_capture @@ -5392,7 +5732,11 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [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_MBP3] = "mbp3", [ALC885_IMAC24] = "imac24", [ALC882_AUTO] = "auto", }; @@ -5404,6 +5748,8 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), 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, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC), @@ -5455,6 +5801,20 @@ static struct alc_config_preset alc882_presets[] = { .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 = alc885_mbp3_unsol_event, + .init_hook = alc885_mbp3_automute, + }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, .init_verbs = { alc882_macpro_init_verbs }, @@ -5465,6 +5825,7 @@ static struct alc_config_preset alc882_presets[] = { .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 }, @@ -5477,7 +5838,7 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc882_ch_modes, .input_mux = &alc882_capture_source, .unsol_event = alc885_imac24_unsol_event, - .init_hook = alc885_imac24_automute, + .init_hook = alc885_imac24_init_hook, }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer, @@ -5509,6 +5870,19 @@ static struct alc_config_preset alc882_presets[] = { .need_dac_fix = 1, .input_mux = &alc882_capture_source, }, + [ALC882_ASUS_A7M] = { + .mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_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, + }, }; @@ -5608,6 +5982,32 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) } } +/* 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) { + 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) { + 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) { @@ -5616,10 +6016,17 @@ static int alc882_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - else if (err > 0) - /* hack - override the init verbs */ - spec->init_verbs[0] = alc882_auto_init_verbs; - return err; + else if (!err) + return 0; /* no config found */ + + err = alc_auto_add_mic_boost(codec); + if (err < 0) + return err; + + /* hack - override the init verbs */ + spec->init_verbs[0] = alc882_auto_init_verbs; + + return 1; /* config found */ } /* additional initialization for auto-configuration model */ @@ -5654,6 +6061,9 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b1000: /* iMac 24 */ board_config = ALC885_IMAC24; break; + case 0x106b2c00: /* Macbook Pro rev3 */ + board_config = ALC885_MBP3; + break; default: printk(KERN_INFO "hda_codec: Unknown model for ALC882, " "trying auto-probe from BIOS...\n"); @@ -5680,11 +6090,6 @@ static int patch_alc882(struct hda_codec *codec) if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); - if (board_config == ALC885_MACPRO || board_config == ALC885_IMAC24) { - alc882_gpio_mute(codec, 0, 0); - alc882_gpio_mute(codec, 1, 0); - } - spec->stream_name_analog = "ALC882 Analog"; spec->stream_analog_playback = &alc882_pcm_analog_playback; spec->stream_analog_capture = &alc882_pcm_analog_capture; @@ -5715,6 +6120,10 @@ static int patch_alc882(struct hda_codec *codec) 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 return 0; } @@ -5792,12 +6201,13 @@ static int alc883_mux_enum_put(struct snd_kcontrol *kcontrol, idx = ucontrol->value.enumerated.item[0]; if (idx >= imux->num_items) idx = imux->num_items - 1; - if (*cur_val == idx && !codec->in_resume) + if (*cur_val == idx) return 0; for (i = 0; i < imux->num_items; i++) { - unsigned int v = (i == idx) ? 0x7000 : 0x7080; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - v | (imux->items[i].index << 8)); + 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; @@ -5822,6 +6232,18 @@ static struct hda_verb alc883_3ST_ch2_init[] = { }; /* + * 4ch mode + */ +static struct hda_verb alc883_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 alc883_3ST_ch6_init[] = { @@ -5834,8 +6256,9 @@ static struct hda_verb alc883_3ST_ch6_init[] = { { } /* end */ }; -static struct hda_channel_mode alc883_3ST_6ch_modes[2] = { +static struct hda_channel_mode alc883_3ST_6ch_modes[3] = { { 2, alc883_3ST_ch2_init }, + { 4, alc883_3ST_ch4_init }, { 6, alc883_3ST_ch6_init }, }; @@ -6235,6 +6658,31 @@ static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { { } /* 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), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -6270,11 +6718,12 @@ static struct hda_verb alc883_init_verbs[] = { {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + /* 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}, @@ -6366,6 +6815,19 @@ static struct hda_verb alc888_lenovo_ms7195_verbs[] = { { } /* 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_6st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 2 (0x0e) */ @@ -6409,15 +6871,10 @@ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); - snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); - snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); - snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); - + 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 */ @@ -6427,12 +6884,10 @@ static void alc888_lenovo_ms7195_rca_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); - snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); - + 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) { @@ -6459,10 +6914,8 @@ static void alc883_medion_md2_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); - snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } static void alc883_medion_md2_unsol_event(struct hda_codec *codec, @@ -6480,13 +6933,11 @@ static void alc883_tagra_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0x80 : 0; - snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_write(codec, 1, 0, AC_VERB_SET_GPIO_DATA, - present ? 1 : 3); + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA, + present ? 1 : 3); } static void alc883_tagra_unsol_event(struct hda_codec *codec, unsigned int res) @@ -6495,6 +6946,25 @@ static void alc883_tagra_unsol_event(struct hda_codec *codec, unsigned int res) alc883_tagra_automute(codec); } +static void alc883_haier_w66_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 ? 0x80 : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + 0x80, bits); +} + +static void alc883_haier_w66_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc883_haier_w66_automute(codec); +} + static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec) { unsigned int present; @@ -6502,11 +6972,9 @@ static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0x80 : 0; - snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, - 0x80, bits); + 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) @@ -6516,15 +6984,11 @@ static void alc883_lenovo_101e_all_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0x80 : 0; - snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, - 0x80, bits); + 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, @@ -6536,6 +7000,44 @@ static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec, alc883_lenovo_101e_ispeaker_automute(codec); } +/* toggle speaker-output according to the hp-jack state */ +static void alc883_acer_aspire_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); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + +static void alc883_acer_aspire_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc883_acer_aspire_automute(codec); +} + +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}, + { } +}; + /* * generic initialization of ADC, input mixers and output mixers */ @@ -6548,17 +7050,17 @@ static struct hda_verb alc883_auto_init_verbs[] = { {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 + /* 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_UNMUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + {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) @@ -6621,6 +7123,10 @@ static struct snd_kcontrol_new alc883_capture_mixer[] = { { } /* end */ }; +#ifdef CONFIG_SND_HDA_POWER_SAVE +#define alc883_loopbacks alc880_loopbacks +#endif + /* pcm configuration: identiacal with ALC880 */ #define alc883_pcm_analog_playback alc880_pcm_analog_playback #define alc883_pcm_analog_capture alc880_pcm_analog_capture @@ -6638,12 +7144,14 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_TARGA_DIG] = "targa-dig", [ALC883_TARGA_2ch_DIG] = "targa-2ch-dig", [ALC883_ACER] = "acer", + [ALC883_ACER_ASPIRE] = "acer-aspire", [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", + [ALC883_HAIER_W66] = "haier-w66", [ALC888_6ST_HP] = "6stack-hp", [ALC888_3ST_HP] = "3stack-hp", [ALC883_AUTO] = "auto", @@ -6669,10 +7177,14 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x3fcc, "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, 0x3fdf, "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, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", 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, 0, "Acer laptop", ALC883_ACER), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), @@ -6685,6 +7197,10 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), + SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), + SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763), + SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), + SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), {} }; @@ -6771,8 +7287,7 @@ static struct alc_config_preset alc883_presets[] = { .init_hook = alc883_tagra_automute, }, [ALC883_ACER] = { - .mixers = { alc883_base_mixer, - alc883_chmode_mixer }, + .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 @@ -6787,6 +7302,20 @@ static struct alc_config_preset alc883_presets[] = { .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_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc883_acer_aspire_unsol_event, + .init_hook = alc883_acer_aspire_automute, + }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer, alc883_chmode_mixer }, @@ -6815,8 +7344,7 @@ static struct alc_config_preset alc883_presets[] = { .init_hook = alc883_medion_md2_automute, }, [ALC883_LAPTOP_EAPD] = { - .mixers = { alc883_base_mixer, - alc883_chmode_mixer }, + .mixers = { alc883_base_mixer }, .init_verbs = { alc883_init_verbs, alc882_eapd_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, @@ -6867,6 +7395,20 @@ static struct alc_config_preset alc883_presets[] = { .input_mux = &alc883_capture_source, .unsol_event = alc883_lenovo_ms7195_unsol_event, .init_hook = alc888_lenovo_ms7195_front_automute, + }, + [ALC883_HAIER_W66] = { + .mixers = { alc883_tagra_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_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc883_haier_w66_unsol_event, + .init_hook = alc883_haier_w66_automute, }, [ALC888_6ST_HP] = { .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer }, @@ -6977,12 +7519,19 @@ static int alc883_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - else if (err > 0) - /* hack - override the init verbs */ - spec->init_verbs[0] = alc883_auto_init_verbs; + else if (!err) + return 0; /* no config found */ + + err = alc_auto_add_mic_boost(codec); + if (err < 0) + return err; + + /* hack - override the init verbs */ + spec->init_verbs[0] = alc883_auto_init_verbs; spec->mixers[spec->num_mixers] = alc883_capture_mixer; spec->num_mixers++; - return err; + + return 1; /* config found */ } /* additional initialization for auto-configuration model */ @@ -7046,6 +7595,10 @@ static int patch_alc883(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC883_AUTO) spec->init_hook = alc883_auto_init; +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!spec->loopback.amplist) + spec->loopback.amplist = alc883_loopbacks; +#endif return 0; } @@ -7156,9 +7709,46 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = { { } /* end */ }; +/* bind hp and internal speaker mute (with plug check) */ +static int alc262_sony_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 hp mute */ + change = snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, + HDA_AMP_MUTE, + valp[0] ? 0 : HDA_AMP_MUTE); + change |= snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, + HDA_AMP_MUTE, + valp[1] ? 0 : HDA_AMP_MUTE); + if (change) { + /* change speaker according to HP jack state */ + struct alc_spec *spec = codec->spec; + unsigned int mute; + if (spec->jack_present) + mute = HDA_AMP_MUTE; + else + mute = snd_hda_codec_amp_read(codec, 0x15, 0, + HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } + return change; +} + static struct snd_kcontrol_new alc262_sony_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc262_sony_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x15, 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("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), @@ -7194,17 +7784,17 @@ static struct hda_verb alc262_init_verbs[] = { {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 + /* 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_UNMUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + {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) @@ -7285,34 +7875,26 @@ static struct hda_verb alc262_sony_unsol_verbs[] = { }; /* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc262_hippo_automute(struct hda_codec *codec, int force) +static void alc262_hippo_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; unsigned int mute; + unsigned int present; - if (force || !spec->sense_updated) { - 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 & 0x80000000) != 0; - spec->sense_updated = 1; - } + /* 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 & 0x80000000) != 0; if (spec->jack_present) { /* mute internal speaker */ - snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, - 0x80, 0x80); - snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, - 0x80, 0x80); + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); } else { /* unmute internal speaker if necessary */ mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, - 0x80, mute & 0x80); - mute = snd_hda_codec_amp_read(codec, 0x15, 1, HDA_OUTPUT, 0); - snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, - 0x80, mute & 0x80); + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); } } @@ -7322,37 +7904,27 @@ static void alc262_hippo_unsol_event(struct hda_codec *codec, { if ((res >> 26) != ALC880_HP_EVENT) return; - alc262_hippo_automute(codec, 1); + alc262_hippo_automute(codec); } -static void alc262_hippo1_automute(struct hda_codec *codec, int force) +static void alc262_hippo1_automute(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; unsigned int mute; + unsigned int present; - if (force || !spec->sense_updated) { - unsigned int present; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; - spec->sense_updated = 1; - } - if (spec->jack_present) { + snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0); + present = (present & 0x80000000) != 0; + if (present) { /* mute internal speaker */ - snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, - 0x80, 0x80); - snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, - 0x80, 0x80); + snd_hda_codec_amp_stereo(codec, 0x14, 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_update(codec, 0x14, 0, HDA_OUTPUT, 0, - 0x80, mute & 0x80); - mute = snd_hda_codec_amp_read(codec, 0x1b, 1, HDA_OUTPUT, 0); - snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, - 0x80, mute & 0x80); + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); } } @@ -7362,7 +7934,7 @@ static void alc262_hippo1_unsol_event(struct hda_codec *codec, { if ((res >> 26) != ALC880_HP_EVENT) return; - alc262_hippo1_automute(codec, 1); + alc262_hippo1_automute(codec); } /* @@ -7379,9 +7951,10 @@ static struct hda_verb alc262_fujitsu_unsol_verbs[] = { }; static struct hda_input_mux alc262_fujitsu_capture_source = { - .num_items = 2, + .num_items = 3, .items = { { "Mic", 0x0 }, + { "Int Mic", 0x1 }, { "CD", 0x4 }, }, }; @@ -7390,13 +7963,23 @@ static struct hda_input_mux alc262_HP_capture_source = { .num_items = 5, .items = { { "Mic", 0x0 }, - { "Front Mic", 0x3 }, + { "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 jack and mute state */ static void alc262_fujitsu_automute(struct hda_codec *codec, int force) { @@ -7414,18 +7997,13 @@ static void alc262_fujitsu_automute(struct hda_codec *codec, int force) } if (spec->jack_present) { /* mute internal speaker */ - snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, - 0x80, 0x80); - snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, - 0x80, 0x80); + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); } else { /* unmute internal speaker if necessary */ mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, - 0x80, mute & 0x80); - mute = snd_hda_codec_amp_read(codec, 0x14, 1, HDA_OUTPUT, 0); - snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, - 0x80, mute & 0x80); + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); } } @@ -7439,23 +8017,14 @@ static void alc262_fujitsu_unsol_event(struct hda_codec *codec, } /* bind volumes of both NID 0x0c and 0x0d */ -static int alc262_fujitsu_master_vol_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - long *valp = ucontrol->value.integer.value; - int change; - - change = snd_hda_codec_amp_update(codec, 0x0c, 0, HDA_OUTPUT, 0, - 0x7f, valp[0] & 0x7f); - change |= snd_hda_codec_amp_update(codec, 0x0c, 1, HDA_OUTPUT, 0, - 0x7f, valp[1] & 0x7f); - snd_hda_codec_amp_update(codec, 0x0d, 0, HDA_OUTPUT, 0, - 0x7f, valp[0] & 0x7f); - snd_hda_codec_amp_update(codec, 0x0d, 1, HDA_OUTPUT, 0, - 0x7f, valp[1] & 0x7f); - return change; -} +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 + }, +}; /* bind hp and internal speaker mute (with plug check) */ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, @@ -7466,24 +8035,18 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, int change; change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, - 0x80, valp[0] ? 0 : 0x80); + HDA_AMP_MUTE, + valp[0] ? 0 : HDA_AMP_MUTE); change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, - 0x80, valp[1] ? 0 : 0x80); - if (change || codec->in_resume) - alc262_fujitsu_automute(codec, codec->in_resume); + HDA_AMP_MUTE, + valp[1] ? 0 : HDA_AMP_MUTE); + if (change) + alc262_fujitsu_automute(codec, 0); return change; } static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Volume", - .info = snd_hda_mixer_amp_volume_info, - .get = snd_hda_mixer_amp_volume_get, - .put = alc262_fujitsu_master_vol_put, - .tlv = { .c = snd_hda_mixer_amp_tlv }, - .private_value = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT), - }, + HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", @@ -7497,6 +8060,9 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { 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 */ }; @@ -7611,17 +8177,17 @@ static struct hda_verb alc262_volume_init_verbs[] = { {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 + /* 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_UNMUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + {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) @@ -7672,19 +8238,19 @@ static struct hda_verb alc262_HP_BPC_init_verbs[] = { {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 + /* 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_UNMUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_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) @@ -7759,20 +8325,20 @@ static struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = { {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 + /* 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_UNMUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)}, + {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) */ @@ -7842,6 +8408,10 @@ static struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = { { } }; +#ifdef CONFIG_SND_HDA_POWER_SAVE +#define alc262_loopbacks alc880_loopbacks +#endif + /* pcm configuration: identiacal with ALC880 */ #define alc262_pcm_analog_playback alc880_pcm_analog_playback #define alc262_pcm_analog_capture alc880_pcm_analog_capture @@ -7884,6 +8454,10 @@ static int alc262_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; + err = alc_auto_add_mic_boost(codec); + if (err < 0) + return err; + return 1; } @@ -7939,6 +8513,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), + SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), @@ -7967,6 +8542,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, + .init_hook = alc262_hippo_automute, }, [ALC262_HIPPO_1] = { .mixers = { alc262_hippo1_mixer }, @@ -7979,10 +8555,12 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo1_unsol_event, + .init_hook = alc262_hippo1_automute, }, [ALC262_FUJITSU] = { .mixers = { alc262_fujitsu_mixer }, - .init_verbs = { alc262_init_verbs, alc262_fujitsu_unsol_verbs }, + .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, @@ -8010,7 +8588,7 @@ static struct alc_config_preset alc262_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, - .input_mux = &alc262_HP_capture_source, + .input_mux = &alc262_HP_D7000_capture_source, }, [ALC262_HP_BPC_D7000_WL] = { .mixers = { alc262_HP_BPC_WildWest_mixer, @@ -8021,7 +8599,7 @@ static struct alc_config_preset alc262_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, - .input_mux = &alc262_HP_capture_source, + .input_mux = &alc262_HP_D7000_capture_source, }, [ALC262_BENQ_ED8] = { .mixers = { alc262_base_mixer }, @@ -8043,6 +8621,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, + .init_hook = alc262_hippo_automute, }, [ALC262_BENQ_T31] = { .mixers = { alc262_benq_t31_mixer }, @@ -8054,6 +8633,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, + .init_hook = alc262_hippo_automute, }, }; @@ -8139,6 +8719,10 @@ static int patch_alc262(struct hda_codec *codec) 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 return 0; } @@ -8170,9 +8754,125 @@ static struct snd_kcontrol_new alc268_base_mixer[] = { 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 hda_verb alc268_eapd_verbs[] = { + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, + { } +}; + +/* Toshiba specific */ +#define alc268_toshiba_automute alc262_hippo_automute + +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 = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, + HDA_AMP_MUTE, + valp[0] ? 0 : HDA_AMP_MUTE); + change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, + HDA_AMP_MUTE, + valp[1] ? 0 : HDA_AMP_MUTE); + if (change) + alc268_acer_automute(codec, 0); + return change; +} + +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 hda_verb alc268_acer_verbs[] = { + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + { } +}; + +/* unsolicited event for HP jack sensing */ +static void alc268_toshiba_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC880_HP_EVENT) + return; + alc268_toshiba_automute(codec); +} + +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); +} + /* * generic initialization of ADC, input mixers and output mixers */ @@ -8282,14 +8982,16 @@ static int alc268_mux_enum_put(struct snd_kcontrol *kcontrol, idx = ucontrol->value.enumerated.item[0]; if (idx >= imux->num_items) idx = imux->num_items - 1; - if (*cur_val == idx && !codec->in_resume) + if (*cur_val == idx) return 0; for (i = 0; i < imux->num_items; i++) { - unsigned int v = (i == idx) ? 0x7000 : 0x7080; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - v | (imux->items[i].index << 8)); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, - idx ); + 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); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, + idx ); } *cur_val = idx; return 1; @@ -8530,6 +9232,10 @@ static int alc268_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; + err = alc_auto_add_mic_boost(codec); + if (err < 0) + return err; + return 1; } @@ -8551,11 +9257,19 @@ static void alc268_auto_init(struct hda_codec *codec) */ static const char *alc268_models[ALC268_MODEL_LAST] = { [ALC268_3ST] = "3stack", + [ALC268_TOSHIBA] = "toshiba", + [ALC268_ACER] = "acer", [ALC268_AUTO] = "auto", }; static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), + SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), + SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA), + SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), + SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER), + SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER), + SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), {} }; @@ -8573,6 +9287,37 @@ static struct alc_config_preset alc268_presets[] = { .channel_mode = alc268_modes, .input_mux = &alc268_capture_source, }, + [ALC268_TOSHIBA] = { + .mixers = { alc268_base_mixer, alc268_capture_alt_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, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc268_modes), + .channel_mode = alc268_modes, + .input_mux = &alc268_capture_source, + .input_mux = &alc268_capture_source, + .unsol_event = alc268_toshiba_unsol_event, + .init_hook = alc268_toshiba_automute, + }, + [ALC268_ACER] = { + .mixers = { alc268_acer_mixer, alc268_capture_alt_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, + .hp_nid = 0x02, + .num_channel_mode = ARRAY_SIZE(alc268_modes), + .channel_mode = alc268_modes, + .input_mux = &alc268_capture_source, + .unsol_event = alc268_acer_unsol_event, + .init_hook = alc268_acer_init_hook, + }, }; static int patch_alc268(struct hda_codec *codec) @@ -9279,14 +10024,10 @@ static void alc861_toshiba_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x0f, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_update(codec, 0x16, 0, HDA_INPUT, 0, - 0x80, present ? 0x80 : 0); - snd_hda_codec_amp_update(codec, 0x16, 1, HDA_INPUT, 0, - 0x80, present ? 0x80 : 0); - snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_INPUT, 3, - 0x80, present ? 0 : 0x80); - snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_INPUT, 3, - 0x80, present ? 0 : 0x80); + 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, @@ -9599,6 +10340,16 @@ static void alc861_auto_init(struct hda_codec *codec) alc861_auto_init_analog_input(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 @@ -9796,6 +10547,10 @@ static int patch_alc861(struct hda_codec *codec) 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 return 0; } @@ -9852,6 +10607,14 @@ static struct hda_input_mux alc861vd_dallas_capture_source = { }, }; +static struct hda_input_mux alc861vd_hp_capture_source = { + .num_items = 2, + .items = { + { "Front Mic", 0x0 }, + { "ATAPI Mic", 0x1 }, + }, +}; + #define alc861vd_mux_enum_info alc_mux_enum_info #define alc861vd_mux_enum_get alc_mux_enum_get @@ -9870,12 +10633,13 @@ static int alc861vd_mux_enum_put(struct snd_kcontrol *kcontrol, idx = ucontrol->value.enumerated.item[0]; if (idx >= imux->num_items) idx = imux->num_items - 1; - if (*cur_val == idx && !codec->in_resume) + if (*cur_val == idx) return 0; for (i = 0; i < imux->num_items; i++) { - unsigned int v = (i == idx) ? 0x7000 : 0x7080; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - v | (imux->items[i].index << 8)); + 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; @@ -10049,17 +10813,22 @@ static struct snd_kcontrol_new alc861vd_dallas_mixer[] = { HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, - }, + { } /* 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 */ }; @@ -10077,11 +10846,11 @@ static struct hda_verb alc861vd_volume_init_verbs[] = { * 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_UNMUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + {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)}, @@ -10210,11 +10979,9 @@ static void alc861vd_lenovo_hp_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0x80 : 0; - snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, - 0x80, bits); + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); } static void alc861vd_lenovo_mic_automute(struct hda_codec *codec) @@ -10224,11 +10991,9 @@ static void alc861vd_lenovo_mic_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0x80 : 0; - snd_hda_codec_amp_update(codec, 0x0b, 0, HDA_INPUT, 1, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x0b, 1, HDA_INPUT, 1, - 0x80, bits); + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, + HDA_AMP_MUTE, bits); } static void alc861vd_lenovo_automute(struct hda_codec *codec) @@ -10302,10 +11067,8 @@ static void alc861vd_dallas_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); - snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int res) @@ -10314,6 +11077,10 @@ static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int re alc861vd_dallas_automute(codec); } +#ifdef CONFIG_SND_HDA_POWER_SAVE +#define alc861vd_loopbacks alc880_loopbacks +#endif + /* pcm configuration: identiacal with ALC880 */ #define alc861vd_pcm_analog_playback alc880_pcm_analog_playback #define alc861vd_pcm_analog_capture alc880_pcm_analog_capture @@ -10325,12 +11092,13 @@ static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int re */ static const char *alc861vd_models[ALC861VD_MODEL_LAST] = { [ALC660VD_3ST] = "3stack-660", - [ALC660VD_3ST_DIG]= "3stack-660-digout", + [ALC660VD_3ST_DIG] = "3stack-660-digout", [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", }; @@ -10341,11 +11109,15 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST), - SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS), + /*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/ SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO), + SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO), + SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG), + SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG), + SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP), {} }; @@ -10435,7 +11207,21 @@ static struct alc_config_preset alc861vd_presets[] = { .input_mux = &alc861vd_dallas_capture_source, .unsol_event = alc861vd_dallas_unsol_event, .init_hook = alc861vd_dallas_automute, - }, + }, + [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, + .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), + .dig_out_nid = ALC861VD_DIGOUT_NID, + .adc_nids = alc861vd_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), + .channel_mode = alc861vd_3stack_2ch_modes, + .input_mux = &alc861vd_hp_capture_source, + .unsol_event = alc861vd_dallas_unsol_event, + .init_hook = alc861vd_dallas_automute, + }, }; /* @@ -10668,6 +11454,10 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; + err = alc_auto_add_mic_boost(codec); + if (err < 0) + return err; + return 1; } @@ -10735,6 +11525,10 @@ static int patch_alc861vd(struct hda_codec *codec) 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 return 0; } @@ -10782,6 +11576,15 @@ static struct hda_input_mux alc662_lenovo_101e_capture_source = { { "Line", 0x2 }, }, }; + +static struct hda_input_mux alc662_eeepc_capture_source = { + .num_items = 2, + .items = { + { "i-Mic", 0x1 }, + { "e-Mic", 0x0 }, + }, +}; + #define alc662_mux_enum_info alc_mux_enum_info #define alc662_mux_enum_get alc_mux_enum_get @@ -10792,7 +11595,7 @@ static int alc662_mux_enum_put(struct snd_kcontrol *kcontrol, struct alc_spec *spec = codec->spec; const struct hda_input_mux *imux = spec->input_mux; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 }; + static hda_nid_t capture_mixers[2] = { 0x23, 0x22 }; hda_nid_t nid = capture_mixers[adc_idx]; unsigned int *cur_val = &spec->cur_mux[adc_idx]; unsigned int i, idx; @@ -10800,12 +11603,13 @@ static int alc662_mux_enum_put(struct snd_kcontrol *kcontrol, idx = ucontrol->value.enumerated.item[0]; if (idx >= imux->num_items) idx = imux->num_items - 1; - if (*cur_val == idx && !codec->in_resume) + if (*cur_val == idx) return 0; for (i = 0; i < imux->num_items; i++) { - unsigned int v = (i == idx) ? 0x7000 : 0x7080; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - v | (imux->items[i].index << 8)); + 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; @@ -10997,6 +11801,22 @@ static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = { + HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("LineOut Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("LineOut Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + + 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_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -11014,18 +11834,18 @@ static struct hda_verb alc662_init_verbs[] = { {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_UNMUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + {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)}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(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)}, /* Front Pin: output 0 (0x0c) */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -11062,13 +11882,24 @@ static struct hda_verb alc662_init_verbs[] = { {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)}, + + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, { } }; 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}, + {} }; /* @@ -11087,11 +11918,11 @@ static struct hda_verb alc662_auto_init_verbs[] = { * panel mic (mic 2) */ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + {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) @@ -11103,23 +11934,19 @@ static struct hda_verb alc662_auto_init_verbs[] = { /* set up input amps for analog loopback */ /* Amp Indices: DAC = 0, mixer = 1 */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(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)}, - {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(3)},*/ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, - + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { } }; @@ -11150,11 +11977,9 @@ static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0x80 : 0; - snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, - 0x80, bits); + 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) @@ -11164,15 +11989,11 @@ static void alc662_lenovo_101e_all_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0x80 : 0; - snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, - 0x80, bits); - snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, - 0x80, bits); + 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, @@ -11184,6 +12005,43 @@ static void alc662_lenovo_101e_unsol_event(struct hda_codec *codec, alc662_lenovo_101e_ispeaker_automute(codec); } +static void alc662_eeepc_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_write(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); + snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); + snd_hda_codec_write(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x01 << 8) | (present ? 0x80 : 0)); + snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x01 << 8) | (present ? 0x80 : 0)); +} + +/* unsolicited event for HP jack sensing */ +static void alc662_eeepc_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc262_hippo1_automute( codec ); + + if ((res >> 26) == ALC880_MIC_EVENT) + alc662_eeepc_mic_automute(codec); +} + +static void alc662_eeepc_inithook(struct hda_codec *codec) +{ + alc262_hippo1_automute( codec ); + alc662_eeepc_mic_automute(codec); +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +#define alc662_loopbacks alc880_loopbacks +#endif + /* pcm configuration: identiacal with ALC880 */ #define alc662_pcm_analog_playback alc880_pcm_analog_playback @@ -11205,12 +12063,13 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), + SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), {} }; static struct alc_config_preset alc662_presets[] = { [ALC662_3ST_2ch_DIG] = { - .mixers = { alc662_3ST_2ch_mixer }, + .mixers = { alc662_3ST_2ch_mixer, alc662_capture_mixer }, .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -11223,7 +12082,8 @@ static struct alc_config_preset alc662_presets[] = { .input_mux = &alc662_capture_source, }, [ALC662_3ST_6ch_DIG] = { - .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer }, + .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer, + alc662_capture_mixer }, .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -11237,7 +12097,8 @@ static struct alc_config_preset alc662_presets[] = { .input_mux = &alc662_capture_source, }, [ALC662_3ST_6ch] = { - .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer }, + .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer, + alc662_capture_mixer }, .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -11249,7 +12110,8 @@ static struct alc_config_preset alc662_presets[] = { .input_mux = &alc662_capture_source, }, [ALC662_5ST_DIG] = { - .mixers = { alc662_base_mixer, alc662_chmode_mixer }, + .mixers = { alc662_base_mixer, alc662_chmode_mixer, + alc662_capture_mixer }, .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -11262,7 +12124,7 @@ static struct alc_config_preset alc662_presets[] = { .input_mux = &alc662_capture_source, }, [ALC662_LENOVO_101E] = { - .mixers = { alc662_lenovo_101e_mixer }, + .mixers = { alc662_lenovo_101e_mixer, alc662_capture_mixer }, .init_verbs = { alc662_init_verbs, alc662_sue_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -11274,6 +12136,20 @@ static struct alc_config_preset alc662_presets[] = { .unsol_event = alc662_lenovo_101e_unsol_event, .init_hook = alc662_lenovo_101e_all_automute, }, + [ALC662_ASUS_EEEPC_P701] = { + .mixers = { alc662_eeepc_p701_mixer, alc662_capture_mixer }, + .init_verbs = { alc662_init_verbs, + alc662_eeepc_sue_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), + .adc_nids = alc662_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc662_eeepc_capture_source, + .unsol_event = alc662_eeepc_unsol_event, + .init_hook = alc662_eeepc_inithook, + }, }; @@ -11296,7 +12172,7 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec, for (i = 0; i < cfg->line_outs; i++) { if (!spec->multiout.dac_nids[i]) continue; - nid = alc880_idx_to_mixer(i); + nid = alc880_idx_to_dac(i); if (i == 2) { /* Center/LFE */ err = add_control(spec, ALC_CTL_WIDGET_VOL, @@ -11586,6 +12462,10 @@ static int patch_alc662(struct hda_codec *codec) 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 return 0; } |