diff options
Diffstat (limited to 'sound/pci/hda/patch_realtek.c')
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 431 |
1 files changed, 276 insertions, 155 deletions
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d22b2606801..30eeb304351 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -224,6 +224,7 @@ enum { ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, + ALC888_ACER_ASPIRE_6530G, ALC888_ACER_ASPIRE_8930G, ALC883_MEDION, ALC883_MEDION_MD2, @@ -249,13 +250,6 @@ enum { ALC883_MODEL_LAST, }; -/* styles of capture selection */ -enum { - CAPT_MUX = 0, /* only mux based */ - CAPT_MIX, /* only mixer based */ - CAPT_1MUX_MIX, /* first mux and other mixers */ -}; - /* for GPIO Poll */ #define GPIO_MASK 0x03 @@ -281,13 +275,13 @@ struct alc_spec { */ unsigned int num_init_verbs; - char stream_name_analog[16]; /* analog PCM stream */ + char stream_name_analog[32]; /* analog PCM stream */ struct hda_pcm_stream *stream_analog_playback; struct hda_pcm_stream *stream_analog_capture; struct hda_pcm_stream *stream_analog_alt_playback; struct hda_pcm_stream *stream_analog_alt_capture; - char stream_name_digital[16]; /* digital PCM stream */ + char stream_name_digital[32]; /* digital PCM stream */ struct hda_pcm_stream *stream_digital_playback; struct hda_pcm_stream *stream_digital_capture; @@ -305,7 +299,6 @@ struct alc_spec { hda_nid_t *adc_nids; hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ - int capture_style; /* capture style (CAPT_*) */ /* capture source */ unsigned int num_mux_defs; @@ -419,12 +412,13 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, unsigned int mux_idx; hda_nid_t nid = spec->capsrc_nids ? spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx]; + unsigned int type; mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; imux = &spec->input_mux[mux_idx]; - if (spec->capture_style && - !(spec->capture_style == CAPT_1MUX_MIX && !adc_idx)) { + type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (type == AC_WID_AUD_MIX) { /* Matrix-mixer style (e.g. ALC882) */ unsigned int *cur_val = &spec->cur_mux[adc_idx]; unsigned int i, idx; @@ -565,7 +559,7 @@ static int alc_pin_mode_get(struct snd_kcontrol *kcontrol, /* Find enumerated value for current pinctl setting */ i = alc_pin_mode_min(dir); - while (alc_pin_mode_values[i] != pinctl && i <= alc_pin_mode_max(dir)) + while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl) i++; *valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir); return 0; @@ -951,12 +945,13 @@ static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid, static void alc_automute_pin(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int present; + unsigned int present, pincap; unsigned int nid = spec->autocfg.hp_pins[0]; int i; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); + pincap = snd_hda_query_pin_caps(codec, nid); + if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ + snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); present = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; @@ -970,7 +965,7 @@ static void alc_automute_pin(struct hda_codec *codec) } } -#if 0 /* it's broken in some acses -- temporarily disabled */ +#if 0 /* it's broken in some cases -- temporarily disabled */ static void alc_mic_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -1170,7 +1165,7 @@ static int alc_subsystem_id(struct hda_codec *codec, /* invalid SSID, check the special NID pin defcfg instead */ /* - * 31~30 : port conetcivity + * 31~30 : port connectivity * 29~21 : reserve * 20 : PCBEEP input * 19~16 : Check sum (15:1) @@ -1398,7 +1393,7 @@ static struct hda_verb alc888_fujitsu_xa3530_verbs[] = { static void alc_automute_amp(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int val, mute; + unsigned int val, mute, pincap; hda_nid_t nid; int i; @@ -1407,6 +1402,10 @@ static void alc_automute_amp(struct hda_codec *codec) nid = spec->autocfg.hp_pins[i]; if (!nid) break; + pincap = snd_hda_query_pin_caps(codec, nid); + if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ + snd_hda_codec_read(codec, nid, 0, + AC_VERB_SET_PIN_SENSE, 0); val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); if (val & AC_PINSENSE_PRESENCE) { @@ -1471,6 +1470,29 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { }; /* + * ALC888 Acer Aspire 6530G model + */ + +static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { +/* Bias voltage on for external mic port */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80}, +/* Front Mic: set to PIN_IN (empty by default) */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, +/* Unselect Front Mic by default in input mixer 3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, +/* Enable unsolicited event for HP jack */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, +/* Enable speaker output */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, +/* Enable headphone output */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + { } +}; + +/* * ALC889 Acer Aspire 8930G model */ @@ -1544,6 +1566,29 @@ static struct hda_input_mux alc888_2_capture_sources[2] = { } }; +static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { + /* Interal mic only available on one ADC */ + { + .num_items = 5, + .items = { + { "Ext Mic", 0x0 }, + { "Line In", 0x2 }, + { "CD", 0x4 }, + { "Input Mix", 0xa }, + { "Int Mic", 0xb }, + }, + }, + { + .num_items = 4, + .items = { + { "Ext Mic", 0x0 }, + { "Line In", 0x2 }, + { "CD", 0x4 }, + { "Input Mix", 0xa }, + }, + } +}; + static struct hda_input_mux alc889_capture_sources[3] = { /* Digital mic only available on first "ADC" */ { @@ -1607,6 +1652,17 @@ static void alc888_acer_aspire_4930g_init_hook(struct hda_codec *codec) alc_automute_amp(codec); } +static void alc888_acer_aspire_6530g_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x17; + alc_automute_amp(codec); +} + static void alc889_acer_aspire_8930g_init_hook(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -4449,6 +4505,12 @@ static int alc880_parse_auto_config(struct hda_codec *codec) &dig_nid, 1); if (err < 0) continue; + if (dig_nid > 0x7f) { + printk(KERN_ERR "alc880_auto: invalid dig_nid " + "connection 0x%x for NID 0x%x\n", dig_nid, + spec->autocfg.dig_out_pins[i]); + continue; + } if (!i) spec->multiout.dig_out_nid = dig_nid; else { @@ -6347,7 +6409,7 @@ static struct hda_channel_mode alc882_sixstack_modes[2] = { }; /* - * macbook pro ALC885 can switch LineIn to LineOut without loosing Mic + * macbook pro ALC885 can switch LineIn to LineOut without losing Mic */ /* @@ -6361,9 +6423,9 @@ static struct hda_verb alc885_mbp_ch2_init[] = { }; /* - * 6ch mode + * 4ch mode */ -static struct hda_verb alc885_mbp_ch6_init[] = { +static struct hda_verb alc885_mbp_ch4_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 }, @@ -6372,9 +6434,9 @@ static struct hda_verb alc885_mbp_ch6_init[] = { { } /* end */ }; -static struct hda_channel_mode alc885_mbp_6ch_modes[2] = { +static struct hda_channel_mode alc885_mbp_4ch_modes[2] = { { 2, alc885_mbp_ch2_init }, - { 6, alc885_mbp_ch6_init }, + { 4, alc885_mbp_ch4_init }, }; /* @@ -6435,10 +6497,11 @@ static struct snd_kcontrol_new alc882_base_mixer[] = { }; static struct snd_kcontrol_new alc885_mbp3_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), - HDA_CODEC_MUTE ("Speaker Playback Switch", 0x14, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0e, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0e, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), @@ -6752,14 +6815,18 @@ static struct hda_verb alc885_mbp3_init_verbs[] = { {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)}, + /* HP mixer */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* 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) */ + /* HP Pin: output 0 (0x0e) */ {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_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, {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}, @@ -6863,9 +6930,6 @@ static struct hda_verb alc882_targa_verbs[] = { {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, - {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x03}, { } /* end */ }; @@ -7047,7 +7111,7 @@ static struct hda_verb alc882_auto_init_verbs[] = { #define alc882_loopbacks alc880_loopbacks #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc882_pcm_analog_playback alc880_pcm_analog_playback #define alc882_pcm_analog_capture alc880_pcm_analog_capture #define alc882_pcm_digital_playback alc880_pcm_digital_playback @@ -7136,10 +7200,11 @@ static struct alc_config_preset alc882_presets[] = { .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, .init_verbs = { alc885_mbp3_init_verbs, alc880_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .num_dacs = 2, .dac_nids = alc882_dac_nids, - .channel_mode = alc885_mbp_6ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mbp_6ch_modes), + .hp_nid = 0x04, + .channel_mode = alc885_mbp_4ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes), .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, @@ -7185,7 +7250,8 @@ static struct alc_config_preset alc882_presets[] = { }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_targa_verbs}, + .init_verbs = { alc882_init_verbs, alc880_gpio3_init_verbs, + alc882_targa_verbs}, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .dig_out_nid = ALC882_DIGOUT_NID, @@ -7518,7 +7584,6 @@ static int patch_alc882(struct hda_codec *codec) spec->stream_digital_playback = &alc882_pcm_digital_playback; spec->stream_digital_capture = &alc882_pcm_digital_capture; - spec->capture_style = CAPT_MIX; /* matrix-style capture */ if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -8068,7 +8133,7 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_tagra_mixer[] = { +static struct snd_kcontrol_new alc883_targa_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -8088,7 +8153,7 @@ static struct snd_kcontrol_new alc883_tagra_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_tagra_2ch_mixer[] = { +static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -8153,6 +8218,21 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -8417,7 +8497,7 @@ static struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = { { } /* end */ }; -static struct hda_verb alc883_tagra_verbs[] = { +static struct hda_verb alc883_targa_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -8626,8 +8706,8 @@ static void alc883_medion_md2_init_hook(struct hda_codec *codec) } /* toggle speaker-output according to the hp-jack state */ -#define alc883_tagra_init_hook alc882_targa_init_hook -#define alc883_tagra_unsol_event alc882_targa_unsol_event +#define alc883_targa_init_hook alc882_targa_init_hook +#define alc883_targa_unsol_event alc882_targa_unsol_event static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) { @@ -8957,7 +9037,7 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) #define alc883_loopbacks alc880_loopbacks #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc883_pcm_analog_playback alc880_pcm_analog_playback #define alc883_pcm_analog_capture alc880_pcm_analog_capture #define alc883_pcm_analog_alt_capture alc880_pcm_analog_alt_capture @@ -8978,6 +9058,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_ACER] = "acer", [ALC883_ACER_ASPIRE] = "acer-aspire", [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", + [ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g", [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", @@ -9019,9 +9100,9 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", - ALC888_ACER_ASPIRE_4930G), + ALC888_ACER_ASPIRE_6530G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", - ALC888_ACER_ASPIRE_4930G), + ALC888_ACER_ASPIRE_6530G), /* default Acer -- disabled as it causes more problems. * model=auto should work fine now */ @@ -9069,6 +9150,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), @@ -9165,8 +9247,9 @@ static struct alc_config_preset alc883_presets[] = { .input_mux = &alc883_capture_source, }, [ALC883_TARGA_DIG] = { - .mixers = { alc883_tagra_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc883_tagra_verbs}, + .mixers = { alc883_targa_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, + alc883_targa_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, @@ -9174,12 +9257,13 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, - .unsol_event = alc883_tagra_unsol_event, - .init_hook = alc883_tagra_init_hook, + .unsol_event = alc883_targa_unsol_event, + .init_hook = alc883_targa_init_hook, }, [ALC883_TARGA_2ch_DIG] = { - .mixers = { alc883_tagra_2ch_mixer}, - .init_verbs = { alc883_init_verbs, alc883_tagra_verbs}, + .mixers = { alc883_targa_2ch_mixer}, + .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, + alc883_targa_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .adc_nids = alc883_adc_nids_alt, @@ -9188,13 +9272,13 @@ static struct alc_config_preset alc883_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc883_tagra_unsol_event, - .init_hook = alc883_tagra_init_hook, + .unsol_event = alc883_targa_unsol_event, + .init_hook = alc883_targa_init_hook, }, [ALC883_TARGA_8ch_DIG] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, - alc883_tagra_verbs }, + alc883_targa_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), @@ -9206,8 +9290,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_4ST_8ch_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, - .unsol_event = alc883_tagra_unsol_event, - .init_hook = alc883_tagra_init_hook, + .unsol_event = alc883_targa_unsol_event, + .init_hook = alc883_targa_init_hook, }, [ALC883_ACER] = { .mixers = { alc883_base_mixer }, @@ -9255,6 +9339,24 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc888_acer_aspire_4930g_init_hook, }, + [ALC888_ACER_ASPIRE_6530G] = { + .mixers = { alc888_acer_aspire_6530_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, + alc888_acer_aspire_6530g_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), + .adc_nids = alc883_adc_nids_rev, + .capsrc_nids = alc883_capsrc_nids_rev, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .num_mux_defs = + ARRAY_SIZE(alc888_2_capture_sources), + .input_mux = alc888_acer_aspire_6530_sources, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_acer_aspire_6530g_init_hook, + }, [ALC888_ACER_ASPIRE_8930G] = { .mixers = { alc888_base_mixer, alc883_chmode_mixer }, @@ -9361,7 +9463,7 @@ static struct alc_config_preset alc883_presets[] = { .init_hook = alc888_lenovo_ms7195_front_automute, }, [ALC883_HAIER_W66] = { - .mixers = { alc883_tagra_2ch_mixer}, + .mixers = { alc883_targa_2ch_mixer}, .init_verbs = { alc883_init_verbs, alc883_haier_w66_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, @@ -9709,7 +9811,6 @@ static int patch_alc883(struct hda_codec *codec) } if (!spec->capsrc_nids) spec->capsrc_nids = alc883_capsrc_nids; - spec->capture_style = CAPT_MIX; /* matrix-style capture */ spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */ break; case 0x10ec0889: @@ -9719,8 +9820,6 @@ static int patch_alc883(struct hda_codec *codec) } if (!spec->capsrc_nids) spec->capsrc_nids = alc889_capsrc_nids; - spec->capture_style = CAPT_1MUX_MIX; /* 1mux/Nmix-style - capture */ break; default: if (!spec->num_adc_nids) { @@ -9729,7 +9828,6 @@ static int patch_alc883(struct hda_codec *codec) } if (!spec->capsrc_nids) spec->capsrc_nids = alc883_capsrc_nids; - spec->capture_style = CAPT_MIX; /* matrix-style capture */ break; } @@ -10539,6 +10637,18 @@ static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec, alc262_lenovo_3000_automute(codec, 1); } +static int amp_stereo_mute_update(struct hda_codec *codec, hda_nid_t nid, + int dir, int idx, long *valp) +{ + int i, change = 0; + + for (i = 0; i < 2; i++, valp++) + change |= snd_hda_codec_amp_update(codec, nid, i, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE); + return change; +} + /* bind hp and internal speaker mute (with plug check) */ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -10547,13 +10657,8 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change; - change = snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp ? 0 : HDA_AMP_MUTE); - change |= snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp ? 0 : HDA_AMP_MUTE); - + change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp); + change |= amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp); if (change) alc262_fujitsu_automute(codec, 0); return change; @@ -10588,10 +10693,7 @@ static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change; - change = snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp ? 0 : HDA_AMP_MUTE); - + change = amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp); if (change) alc262_lenovo_3000_automute(codec, 0); return change; @@ -10841,9 +10943,27 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, return 0; } -/* identical with ALC880 */ -#define alc262_auto_create_analog_input_ctls \ - alc880_auto_create_analog_input_ctls +static int alc262_auto_create_analog_input_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int err; + + err = alc880_auto_create_analog_input_ctls(spec, cfg); + if (err < 0) + return err; + /* digital-mic input pin is excluded in alc880_auto_create..() + * because it's under 0x18 + */ + if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || + cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { + struct hda_input_mux *imux = &spec->private_imux[0]; + imux->items[imux->num_items].label = "Int Mic"; + imux->items[imux->num_items].index = 0x09; + imux->num_items++; + } + return 0; +} + /* * generic initialization of ADC, input mixers and output mixers @@ -11131,7 +11251,7 @@ static struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = { #define alc262_loopbacks alc880_loopbacks #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc262_pcm_analog_playback alc880_pcm_analog_playback #define alc262_pcm_analog_capture alc880_pcm_analog_capture #define alc262_pcm_digital_playback alc880_pcm_digital_playback @@ -11260,6 +11380,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO), SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */ + SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06), SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", @@ -11467,6 +11588,7 @@ static struct alc_config_preset alc262_presets[] = { .capsrc_nids = alc262_dmic_capsrc_nids, .dac_nids = alc262_dac_nids, .adc_nids = alc262_dmic_adc_nids, /* ADC0 */ + .num_adc_nids = 1, /* single ADC */ .dig_out_nid = ALC262_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, @@ -11568,21 +11690,36 @@ static int patch_alc262(struct hda_codec *codec) spec->stream_digital_playback = &alc262_pcm_digital_playback; spec->stream_digital_capture = &alc262_pcm_digital_capture; - spec->capture_style = CAPT_MIX; if (!spec->adc_nids && spec->input_mux) { - /* check whether NID 0x07 is valid */ - unsigned int wcap = get_wcaps(codec, 0x07); - - /* get type */ - wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wcap != AC_WID_AUD_IN) { - spec->adc_nids = alc262_adc_nids_alt; - spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids_alt); - spec->capsrc_nids = alc262_capsrc_nids_alt; + int i; + /* check whether the digital-mic has to be supported */ + for (i = 0; i < spec->input_mux->num_items; i++) { + if (spec->input_mux->items[i].index >= 9) + break; + } + if (i < spec->input_mux->num_items) { + /* use only ADC0 */ + spec->adc_nids = alc262_dmic_adc_nids; + spec->num_adc_nids = 1; + spec->capsrc_nids = alc262_dmic_capsrc_nids; } else { - spec->adc_nids = alc262_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids); - spec->capsrc_nids = alc262_capsrc_nids; + /* all analog inputs */ + /* check whether NID 0x07 is valid */ + unsigned int wcap = get_wcaps(codec, 0x07); + + /* get type */ + wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (wcap != AC_WID_AUD_IN) { + spec->adc_nids = alc262_adc_nids_alt; + spec->num_adc_nids = + ARRAY_SIZE(alc262_adc_nids_alt); + spec->capsrc_nids = alc262_capsrc_nids_alt; + } else { + spec->adc_nids = alc262_adc_nids; + spec->num_adc_nids = + ARRAY_SIZE(alc262_adc_nids); + spec->capsrc_nids = alc262_capsrc_nids; + } } } if (!spec->cap_mixer && !spec->no_analog) @@ -11727,12 +11864,7 @@ static int alc268_acer_master_sw_put(struct snd_kcontrol *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); + change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp); if (change) alc268_acer_automute(codec, 0); return change; @@ -12286,7 +12418,7 @@ static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec) AC_VERB_SET_AMP_GAIN_MUTE, dac_vol2); } -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc268_pcm_analog_playback alc880_pcm_analog_playback #define alc268_pcm_analog_capture alc880_pcm_analog_capture #define alc268_pcm_analog_alt_capture alc880_pcm_analog_alt_capture @@ -12342,6 +12474,8 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + alc_ssid_check(codec, 0x15, 0x1b, 0x14); + return 1; } @@ -12393,8 +12527,6 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), - SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05", - ALC268_TOSHIBA), SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA), SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), @@ -12402,6 +12534,15 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { {} }; +/* Toshiba laptops have no unique PCI SSID but only codec SSID */ +static struct snd_pci_quirk alc268_ssid_cfg_tbl[] = { + SND_PCI_QUIRK(0x1179, 0xff0a, "TOSHIBA X-200", ALC268_AUTO), + SND_PCI_QUIRK(0x1179, 0xff0e, "TOSHIBA X-200 HDMI", ALC268_AUTO), + SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05", + ALC268_TOSHIBA), + {} +}; + static struct alc_config_preset alc268_presets[] = { [ALC267_QUANTA_IL1] = { .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer }, @@ -12568,6 +12709,10 @@ static int patch_alc268(struct hda_codec *codec) alc268_models, alc268_cfg_tbl); + if (board_config < 0 || board_config >= ALC268_MODEL_LAST) + board_config = snd_hda_check_board_codec_sid_config(codec, + ALC882_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl); + if (board_config < 0 || board_config >= ALC268_MODEL_LAST) { printk(KERN_INFO "hda_codec: Unknown model for %s, " "trying auto-probe from BIOS...\n", codec->chip_name); @@ -12753,20 +12898,11 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { } }; -/* bind volumes of both NID 0x0c and 0x0d */ -static struct hda_bind_ctls alc269_epc_bind_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT), - 0 - }, -}; - static struct snd_kcontrol_new alc269_eeepc_mixer[] = { - HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_BIND_VOL("LineOut Playback Volume", &alc269_epc_bind_vol), - HDA_CODEC_MUTE("LineOut Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -12779,12 +12915,7 @@ static struct snd_kcontrol_new alc269_epc_capture_mixer[] = { }; /* FSC amilo */ -static struct snd_kcontrol_new alc269_fujitsu_mixer[] = { - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_BIND_VOL("PCM Playback Volume", &alc269_epc_bind_vol), - { } /* end */ -}; +#define alc269_fujitsu_mixer alc269_eeepc_mixer static struct hda_verb alc269_quanta_fl1_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -13172,32 +13303,14 @@ static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec, return 0; } -static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec, - const struct auto_pin_cfg *cfg) -{ - int err; - - err = alc880_auto_create_analog_input_ctls(spec, cfg); - if (err < 0) - return err; - /* digital-mic input pin is excluded in alc880_auto_create..() - * because it's under 0x18 - */ - if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || - cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { - struct hda_input_mux *imux = &spec->private_imux[0]; - imux->items[imux->num_items].label = "Int Mic"; - imux->items[imux->num_items].index = 0x05; - imux->num_items++; - } - return 0; -} +#define alc269_auto_create_analog_input_ctls \ + alc262_auto_create_analog_input_ctls #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc269_loopbacks alc880_loopbacks #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc269_pcm_analog_playback alc880_pcm_analog_playback #define alc269_pcm_analog_capture alc880_pcm_analog_capture #define alc269_pcm_digital_playback alc880_pcm_digital_playback @@ -13268,6 +13381,8 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); + alc_ssid_check(codec, 0x15, 0x1b, 0x14); + return 1; } @@ -13465,6 +13580,8 @@ static int patch_alc269(struct hda_codec *codec) set_capture_mixer(spec); set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); + spec->vmaster_nid = 0x02; + codec->patch_ops = alc_patch_ops; if (board_config == ALC269_AUTO) spec->init_hook = alc269_auto_init; @@ -14059,7 +14176,7 @@ static void alc861_toshiba_unsol_event(struct hda_codec *codec, alc861_toshiba_automute(codec); } -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc861_pcm_analog_playback alc880_pcm_analog_playback #define alc861_pcm_analog_capture alc880_pcm_analog_capture #define alc861_pcm_digital_playback alc880_pcm_digital_playback @@ -14582,7 +14699,7 @@ static hda_nid_t alc861vd_dac_nids[4] = { /* dac_nids for ALC660vd are in a different order - according to * Realtek's driver. - * This should probably tesult in a different mixer for 6stack models + * This should probably result in a different mixer for 6stack models * of ALC660vd codecs, but for now there is only 3stack mixer * - and it is the same as in 861vd. * adc_nids in ALC660vd are (is) the same as in 861vd @@ -15027,7 +15144,7 @@ static void alc861vd_dallas_init_hook(struct hda_codec *codec) #define alc861vd_loopbacks alc880_loopbacks #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc861vd_pcm_analog_playback alc880_pcm_analog_playback #define alc861vd_pcm_analog_capture alc880_pcm_analog_capture #define alc861vd_pcm_digital_playback alc880_pcm_digital_playback @@ -15059,7 +15176,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO), /*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/ - SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS), + SND_PCI_QUIRK(0x1179, 0xff01, "Toshiba A135", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS), SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG), @@ -15206,7 +15323,7 @@ static void alc861vd_auto_init_hp_out(struct hda_codec *codec) hda_nid_t pin; pin = spec->autocfg.hp_pins[0]; - if (pin) /* connect to front and use dac 0 */ + if (pin) /* connect to front and use dac 0 */ alc861vd_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); pin = spec->autocfg.speaker_pins[0]; if (pin) @@ -15479,10 +15596,12 @@ static int patch_alc861vd(struct hda_codec *codec) spec->stream_digital_playback = &alc861vd_pcm_digital_playback; spec->stream_digital_capture = &alc861vd_pcm_digital_capture; - spec->adc_nids = alc861vd_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids); - spec->capsrc_nids = alc861vd_capsrc_nids; - spec->capture_style = CAPT_MIX; + if (!spec->adc_nids) { + spec->adc_nids = alc861vd_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids); + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc861vd_capsrc_nids; set_capture_mixer(spec); set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); @@ -16669,7 +16788,7 @@ static struct snd_kcontrol_new alc272_nc10_mixer[] = { #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc662_pcm_analog_playback alc880_pcm_analog_playback #define alc662_pcm_analog_capture alc880_pcm_analog_capture #define alc662_pcm_digital_playback alc880_pcm_digital_playback @@ -17399,10 +17518,12 @@ static int patch_alc662(struct hda_codec *codec) spec->stream_digital_playback = &alc662_pcm_digital_playback; spec->stream_digital_capture = &alc662_pcm_digital_capture; - spec->adc_nids = alc662_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids); - spec->capsrc_nids = alc662_capsrc_nids; - spec->capture_style = CAPT_MIX; + if (!spec->adc_nids) { + spec->adc_nids = alc662_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids); + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc662_capsrc_nids; if (!spec->cap_mixer) set_capture_mixer(spec); |