From 5a053d012d0576e9306009939ca81a86547ef35a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 25 Jul 2006 14:51:15 +0200 Subject: [ALSA] Add model entry for Clevo m665n laptop Added the proper model entry for Clevo m665n laptop with ALC880 codec. Also, added a model string 'clevo' to enable the clevo-type model option. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 18d105263fe..f4c96aa43be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2156,8 +2156,13 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .modelname = "3stack-digout", .config = ALC880_3ST_DIG }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe308, .config = ALC880_3ST_DIG }, { .pci_subvendor = 0x1025, .pci_subdevice = 0x0070, .config = ALC880_3ST_DIG }, - /* Clevo m520G NB */ - { .pci_subvendor = 0x1558, .pci_subdevice = 0x0520, .config = ALC880_CLEVO }, + + /* Clevo laptops */ + { .modelname = "clevo", .config = ALC880_CLEVO }, + { .pci_subvendor = 0x1558, .pci_subdevice = 0x0520, + .config = ALC880_CLEVO }, /* Clevo m520G NB */ + { .pci_subvendor = 0x1558, .pci_subdevice = 0x0660, + .config = ALC880_CLEVO }, /* Clevo m665n */ /* Back 3 jack plus 1 SPDIF out jack, front 2 jack (Internal add Aux-In)*/ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe305, .config = ALC880_3ST_DIG }, -- cgit v1.2.3-70-g09d2 From 6d177ba7839dd7ed391c2f36b121eb09d1eaee4c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 25 Jul 2006 14:51:15 +0200 Subject: [ALSA] Add hp-bpc model type for HP laptops Added 'hp-bpc' model type for HP xw4400-compatible laptops. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_realtek.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 74ea66d33cf..c595acb3bf8 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -798,6 +798,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ALC262 fujitsu Fujitsu Laptop + hp-bpc HP xw4400/6400/8400/9400 laptops basic fixed pin assignment w/o SPDIF auto auto-config reading BIOS (default) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f4c96aa43be..51f76eef935 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5774,6 +5774,7 @@ static struct hda_board_config alc262_cfg_tbl[] = { { .modelname = "fujitsu", .config = ALC262_FUJITSU }, { .pci_subvendor = 0x10cf, .pci_subdevice = 0x1397, .config = ALC262_FUJITSU }, + { .modelname = "hp-bpc", .config = ALC262_HP_BPC }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x208c, .config = ALC262_HP_BPC }, /* xw4400 */ { .pci_subvendor = 0x103c, .pci_subdevice = 0x3014, -- cgit v1.2.3-70-g09d2 From 304dcaac91f0d26543b31fd7e63726f096c826ee Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 25 Jul 2006 14:51:16 +0200 Subject: [ALSA] Add support of Benq laptop with ALC262 Added the support of Benq laptop with ALC262 codec. A model string 'benq' is added, too. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_realtek.c | 21 +++++++++++++++++++++ 2 files changed, 22 insertions(+) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index c595acb3bf8..885d2ed88fd 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -799,6 +799,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ALC262 fujitsu Fujitsu Laptop hp-bpc HP xw4400/6400/8400/9400 laptops + benq Benq ED8 basic fixed pin assignment w/o SPDIF auto auto-config reading BIOS (default) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 51f76eef935..42c4f90a92b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -79,6 +79,7 @@ enum { ALC262_BASIC, ALC262_FUJITSU, ALC262_HP_BPC, + ALC262_BENQ_ED8, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -5504,6 +5505,13 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { { } /* end */ }; +/* additional init verbs for Benq laptops */ +static struct hda_verb alc262_EAPD_verbs[] = { + {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3070}, + {} +}; + /* add playback controls from the parsed DAC table */ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { @@ -5783,6 +5791,9 @@ static struct hda_board_config alc262_cfg_tbl[] = { .config = ALC262_HP_BPC }, /* xw8400 */ { .pci_subvendor = 0x103c, .pci_subdevice = 0x12fe, .config = ALC262_HP_BPC }, /* xw9400 */ + { .modelname = "benq", .config = ALC262_BENQ_ED8 }, + { .pci_subvendor = 0x17ff, .pci_subdevice = 0x0560, + .config = ALC262_BENQ_ED8 }, { .modelname = "auto", .config = ALC262_AUTO }, {} }; @@ -5820,6 +5831,16 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_HP_capture_source, }, + [ALC262_BENQ_ED8] = { + .mixers = { alc262_base_mixer }, + .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_capture_source, + }, }; static int patch_alc262(struct hda_codec *codec) -- cgit v1.2.3-70-g09d2 From 4b146cb087b4a668511f6c991da1dc40e2e04b0d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Jul 2006 14:42:36 +0200 Subject: [ALSA] Misc fixes for Realtek HD-audio codecs - Added model=arima for Arima W820Di1 with ALC882 codec chip - Added EAPD-control verbs to TCL S700 init verbs - Added missing model strings for Realtek codecs (to be specified via module option explicitly for testing/debugging) Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/ALSA-Configuration.txt | 13 +++++-- sound/pci/hda/patch_realtek.c | 48 +++++++++++++++++++++---- 2 files changed, 52 insertions(+), 9 deletions(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 7344815b855..d0dbc3fb20c 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -778,11 +778,15 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 6stack-digout 6-jack with a SPDIF out w810 3-jack z71v 3-jack (HP shared SPDIF) - asus 3-jack + asus 3-jack (ASUS Mobo) + asus-w1v ASUS W1V + asus-dig ASUS with SPDIF out + asus-dig2 ASUS with SPDIF out (using GPIO2) uniwill 3-jack F1734 2-jack lg LG laptop (m1 express dual) lg-lw LG LW20 laptop + tcl TCL S700 clevo Clevo laptops (m520G, m665n) test for testing/debugging purpose, almost all controls can be adjusted. Appearing only when compiled with @@ -791,6 +795,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ALC260 hp HP machines + hp-3013 HP machines (3013-variant) fujitsu Fujitsu S7020 acer Acer TravelMate basic fixed pin assignment (old default model) @@ -806,18 +811,22 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ALC882/885 3stack-dig 3-jack with SPDIF I/O 6stck-dig 6-jack digital with SPDIF I/O + arima Arima W820Di1 auto auto-config reading BIOS (default) ALC883/888 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack digital with SPDIF I/O - 6stack-dig-demo 6-stack digital for Intel demo board + 3stack-6ch 3-jack 6-channel + 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O + 6stack-dig-demo 6-jack digital for Intel demo board auto auto-config reading BIOS (default) ALC861/660 3stack 3-jack 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack with SPDIF I/O + 3stack-660 3-jack (for ALC660) auto auto-config reading BIOS (default) CMI9880 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 42c4f90a92b..378e5f111e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -98,6 +98,7 @@ enum { enum { ALC882_3ST_DIG, ALC882_6ST_DIG, + ALC882_ARIMA, ALC882_AUTO, ALC882_MODEL_LAST, }; @@ -1349,6 +1350,10 @@ static struct hda_verb alc880_pin_clevo_init_verbs[] = { }; static struct hda_verb alc880_pin_tcl_S700_init_verbs[] = { + /* change to EAPD mode */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3060}, + /* Headphone output */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, /* Front output*/ @@ -2146,6 +2151,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x107b, .pci_subdevice = 0x4040, .config = ALC880_3ST }, { .pci_subvendor = 0x107b, .pci_subdevice = 0x4041, .config = ALC880_3ST }, /* TCL S700 */ + { .modelname = "tcl", .config = ALC880_TCL_S700 }, { .pci_subvendor = 0x19db, .pci_subdevice = 0x4188, .config = ALC880_TCL_S700 }, /* Back 3 jack, front 2 jack (Internal add Aux-In) */ @@ -2232,8 +2238,11 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1043, .pci_subdevice = 0x1133, .config = ALC880_ASUS }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1123, .config = ALC880_ASUS_DIG }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1143, .config = ALC880_ASUS }, + { .modelname = "asus-w1v", .config = ALC880_ASUS_W1V }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x10b3, .config = ALC880_ASUS_W1V }, + { .modelname = "asus-dig", .config = ALC880_ASUS_DIG }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x8181, .config = ALC880_ASUS_DIG }, /* ASUS P4GPL-X */ + { .modelname = "asus-dig2", .config = ALC880_ASUS_DIG2 }, { .pci_subvendor = 0x1558, .pci_subdevice = 0x5401, .config = ALC880_ASUS_DIG2 }, { .modelname = "uniwill", .config = ALC880_UNIWILL_DIG }, @@ -3906,6 +3915,7 @@ static struct hda_board_config alc260_cfg_tbl[] = { { .pci_subvendor = 0x152d, .pci_subdevice = 0x0729, .config = ALC260_BASIC }, /* CTL Travel Master U553W */ { .modelname = "hp", .config = ALC260_HP }, + { .modelname = "hp-3013", .config = ALC260_HP_3013 }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3011, .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3012, .config = ALC260_HP_3013 }, @@ -4272,6 +4282,13 @@ static struct hda_verb alc882_init_verbs[] = { { } }; +static struct hda_verb alc882_eapd_verbs[] = { + /* change to EAPD mode */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3060}, + { } +}; + /* * generic initialization of ADC, input mixers and output mixers */ @@ -4403,6 +4420,9 @@ static struct hda_board_config alc882_cfg_tbl[] = { .config = ALC882_6ST_DIG }, /* Foxconn */ { .pci_subvendor = 0x1019, .pci_subdevice = 0x6668, .config = ALC882_6ST_DIG }, /* ECS to Intel*/ + { .modelname = "arima", .config = ALC882_ARIMA }, + { .pci_subvendor = 0x161f, .pci_subdevice = 0x2054, + .config = ALC882_ARIMA }, /* Arima W820Di1 */ { .modelname = "auto", .config = ALC882_AUTO }, {} }; @@ -4430,6 +4450,15 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc882_sixstack_modes, .input_mux = &alc882_capture_source, }, + [ALC882_ARIMA] = { + .mixers = { alc882_base_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_init_verbs, alc882_eapd_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), + .channel_mode = alc882_sixstack_modes, + .input_mux = &alc882_capture_source, + }, }; @@ -5005,16 +5034,18 @@ static struct snd_kcontrol_new alc883_capture_mixer[] = { */ static struct hda_board_config alc883_cfg_tbl[] = { { .modelname = "3stack-dig", .config = ALC883_3ST_2ch_DIG }, + { .modelname = "3stack-6ch-dig", .config = ALC883_3ST_6ch_DIG }, + { .pci_subvendor = 0x1019, .pci_subdevice = 0x6668, + .config = ALC883_3ST_6ch_DIG }, /* ECS to Intel*/ + { .modelname = "3stack-6ch", .config = ALC883_3ST_6ch }, + { .pci_subvendor = 0x108e, .pci_subdevice = 0x534d, + .config = ALC883_3ST_6ch }, { .modelname = "6stack-dig", .config = ALC883_6ST_DIG }, - { .modelname = "6stack-dig-demo", .config = ALC888_DEMO_BOARD }, { .pci_subvendor = 0x1462, .pci_subdevice = 0x6668, .config = ALC883_6ST_DIG }, /* MSI */ { .pci_subvendor = 0x105b, .pci_subdevice = 0x6668, .config = ALC883_6ST_DIG }, /* Foxconn */ - { .pci_subvendor = 0x1019, .pci_subdevice = 0x6668, - .config = ALC883_3ST_6ch_DIG }, /* ECS to Intel*/ - { .pci_subvendor = 0x108e, .pci_subdevice = 0x534d, - .config = ALC883_3ST_6ch }, + { .modelname = "6stack-dig-demo", .config = ALC888_DEMO_BOARD }, { .modelname = "auto", .config = ALC883_AUTO }, {} }; @@ -5223,8 +5254,10 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_digital_playback = &alc883_pcm_digital_playback; spec->stream_digital_capture = &alc883_pcm_digital_capture; - spec->adc_nids = alc883_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + if (! spec->adc_nids && spec->input_mux) { + spec->adc_nids = alc883_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + } codec->patch_ops = alc_patch_ops; if (board_config == ALC883_AUTO) @@ -6504,6 +6537,7 @@ static struct hda_board_config alc861_cfg_tbl[] = { { .modelname = "3stack", .config = ALC861_3ST }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xd600, .config = ALC861_3ST }, + { .modelname = "3stack-660", .config = ALC660_3ST }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x81e7, .config = ALC660_3ST }, { .modelname = "3stack-dig", .config = ALC861_3ST_DIG }, -- cgit v1.2.3-70-g09d2 From 4e195a7b78618c89b06547f3140e67a69ec23272 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Jul 2006 14:47:34 +0200 Subject: [ALSA] Fix noisy output with shared channel mode with hd-audio - Fix the wrong initialization of num_dacs when changing the channel mode between 2 and multi-channel modes. It must be evaluated after calling snd_hda_ch_mode_put() - Added the similar check of num_dacs fix in Realtek code. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 8 +++++--- sound/pci/hda/patch_realtek.c | 27 ++++++++++++++++++++++++--- 2 files changed, 29 insertions(+), 6 deletions(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 8955397cca6..077f1ce01ee 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1647,10 +1647,12 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *spec = codec->spec; - if (spec->need_dac_fix) + int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, + spec->num_channel_mode, + &spec->multiout.max_channels); + if (! err && spec->need_dac_fix) spec->multiout.num_dacs = spec->multiout.max_channels / 2; - return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, &spec->multiout.max_channels); + return err; } /* 6-stack mode */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 378e5f111e3..991f1079116 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -155,6 +155,7 @@ struct alc_spec { /* channel model */ const struct hda_channel_mode *channel_mode; int num_channel_mode; + int need_dac_fix; /* PCM information */ struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ @@ -192,6 +193,7 @@ struct alc_config_preset { hda_nid_t dig_in_nid; unsigned int num_channel_mode; const struct hda_channel_mode *channel_mode; + int need_dac_fix; unsigned int num_mux_defs; const struct hda_input_mux *input_mux; void (*unsol_event)(struct hda_codec *, unsigned int); @@ -264,9 +266,12 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - &spec->multiout.max_channels); + int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, + spec->num_channel_mode, + &spec->multiout.max_channels); + if (! err && spec->need_dac_fix) + spec->multiout.num_dacs = spec->multiout.max_channels / 2; + return err; } /* @@ -546,6 +551,7 @@ static void setup_preset(struct alc_spec *spec, spec->channel_mode = preset->channel_mode; spec->num_channel_mode = preset->num_channel_mode; + spec->need_dac_fix = preset->need_dac_fix; spec->multiout.max_channels = spec->channel_mode[0].channels; @@ -2278,6 +2284,7 @@ static struct alc_config_preset alc880_presets[] = { .dac_nids = alc880_dac_nids, .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), .channel_mode = alc880_threestack_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_3ST_DIG] = { @@ -2288,6 +2295,7 @@ static struct alc_config_preset alc880_presets[] = { .dig_out_nid = ALC880_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), .channel_mode = alc880_threestack_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_TCL_S700] = { @@ -2380,6 +2388,7 @@ static struct alc_config_preset alc880_presets[] = { .dac_nids = alc880_asus_dac_nids, .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), .channel_mode = alc880_asus_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_ASUS_DIG] = { @@ -2391,6 +2400,7 @@ static struct alc_config_preset alc880_presets[] = { .dig_out_nid = ALC880_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), .channel_mode = alc880_asus_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_ASUS_DIG2] = { @@ -2402,6 +2412,7 @@ static struct alc_config_preset alc880_presets[] = { .dig_out_nid = ALC880_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), .channel_mode = alc880_asus_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_ASUS_W1V] = { @@ -2413,6 +2424,7 @@ static struct alc_config_preset alc880_presets[] = { .dig_out_nid = ALC880_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), .channel_mode = alc880_asus_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_UNIWILL_DIG] = { @@ -2423,6 +2435,7 @@ static struct alc_config_preset alc880_presets[] = { .dig_out_nid = ALC880_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), .channel_mode = alc880_asus_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_CLEVO] = { @@ -2434,6 +2447,7 @@ static struct alc_config_preset alc880_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), .channel_mode = alc880_threestack_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_LG] = { @@ -2445,6 +2459,7 @@ static struct alc_config_preset alc880_presets[] = { .dig_out_nid = ALC880_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes), .channel_mode = alc880_lg_ch_modes, + .need_dac_fix = 1, .input_mux = &alc880_lg_capture_source, .unsol_event = alc880_lg_unsol_event, .init_hook = alc880_lg_automute, @@ -4437,6 +4452,7 @@ static struct alc_config_preset alc882_presets[] = { .dig_in_nid = ALC882_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), .channel_mode = alc882_ch_modes, + .need_dac_fix = 1, .input_mux = &alc882_capture_source, }, [ALC882_6ST_DIG] = { @@ -5075,6 +5091,7 @@ static struct alc_config_preset alc883_presets[] = { .dig_in_nid = ALC883_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, .input_mux = &alc883_capture_source, }, [ALC883_3ST_6ch] = { @@ -5086,6 +5103,7 @@ static struct alc_config_preset alc883_presets[] = { .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, .input_mux = &alc883_capture_source, }, [ALC883_6ST_DIG] = { @@ -6554,6 +6572,7 @@ static struct alc_config_preset alc861_presets[] = { .dac_nids = alc861_dac_nids, .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes), .channel_mode = alc861_threestack_modes, + .need_dac_fix = 1, .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), .adc_nids = alc861_adc_nids, .input_mux = &alc861_capture_source, @@ -6566,6 +6585,7 @@ static struct alc_config_preset alc861_presets[] = { .dig_out_nid = ALC861_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes), .channel_mode = alc861_threestack_modes, + .need_dac_fix = 1, .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), .adc_nids = alc861_adc_nids, .input_mux = &alc861_capture_source, @@ -6589,6 +6609,7 @@ static struct alc_config_preset alc861_presets[] = { .dac_nids = alc660_dac_nids, .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes), .channel_mode = alc861_threestack_modes, + .need_dac_fix = 1, .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), .adc_nids = alc861_adc_nids, .input_mux = &alc861_capture_source, -- cgit v1.2.3-70-g09d2 From 25b6c43b3d6258f3e87244eeb2b9347dc5e83c40 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Aug 2006 13:01:14 +0200 Subject: [ALSA] Fix the preselected model for HP machine Fixed the preselected model for a HP machine with SSID 103c:3010 to use hp-3013 (ALSA bug#2157). Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 991f1079116..ac561a5d866 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3931,7 +3931,7 @@ static struct hda_board_config alc260_cfg_tbl[] = { .config = ALC260_BASIC }, /* CTL Travel Master U553W */ { .modelname = "hp", .config = ALC260_HP }, { .modelname = "hp-3013", .config = ALC260_HP_3013 }, - { .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP }, + { .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP_3013 }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3011, .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3012, .config = ALC260_HP_3013 }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3013, .config = ALC260_HP_3013 }, -- cgit v1.2.3-70-g09d2 From f5a5ffad072ec3c1fd636174c30f0ba52fe0259f Mon Sep 17 00:00:00 2001 From: Danny Tholen Date: Tue, 8 Aug 2006 18:59:07 +0200 Subject: [ALSA] [snd-hda-intel] fix sound on some Asus W6A chips This patch adds support in ALSA snd-hda-intel driver for Asus W6A motherboard as reported in MDV Bugzilla #19962 (see http://qa.mandriva.com/show_bug.cgi?id=19962) Signed-off-by: Danny Tholen Signed-off-by: Thomas Backlund Signed-off-by: Thierry Vignaud Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ac561a5d866..a9175731676 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2240,6 +2240,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1043, .pci_subdevice = 0x1113, .config = ALC880_ASUS_DIG }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1173, .config = ALC880_ASUS_DIG }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1993, .config = ALC880_ASUS }, + { .pci_subvendor = 0x1043, .pci_subdevice = 0x10c2, .config = ALC880_ASUS_DIG }, /* Asus W6A */ { .pci_subvendor = 0x1043, .pci_subdevice = 0x10c3, .config = ALC880_ASUS_DIG }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1133, .config = ALC880_ASUS }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1123, .config = ALC880_ASUS_DIG }, -- cgit v1.2.3-70-g09d2 From 22309c3e0c8911865cad0aa94f53a9afadaad7ee Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Aug 2006 16:57:28 +0200 Subject: [ALSA] Added model for Uniwill laptop with ALC861 Added a new model 'uniwill-m31' for Uniwill laptops with ALC861 codec chip. The patch is taken from ALSA bug#2035, and modifeid. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_realtek.c | 137 ++++++++++++++++++++++++ 2 files changed, 138 insertions(+) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index d0dbc3fb20c..74be228596a 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -827,6 +827,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack with SPDIF I/O 3stack-660 3-jack (for ALC660) + uniwill-m31 Uniwill M31 laptop auto auto-config reading BIOS (default) CMI9880 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a9175731676..f857e963ff4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -90,6 +90,7 @@ enum { ALC660_3ST, ALC861_3ST_DIG, ALC861_6ST_DIG, + ALC861_UNIWILL_M31, ALC861_AUTO, ALC861_MODEL_LAST, }; @@ -6021,6 +6022,23 @@ static struct hda_channel_mode alc861_threestack_modes[2] = { { 2, alc861_threestack_ch2_init }, { 6, alc861_threestack_ch6_init }, }; +/* Set mic1 as input and unmute the mixer */ +static struct hda_verb alc861_uniwill_m31_ch2_init[] = { + { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/ + { } /* end */ +}; +/* Set mic1 as output and mute mixer */ +static struct hda_verb alc861_uniwill_m31_ch4_init[] = { + { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/ + { } /* end */ +}; + +static struct hda_channel_mode alc861_uniwill_m31_modes[2] = { + { 2, alc861_uniwill_m31_ch2_init }, + { 4, alc861_uniwill_m31_ch4_init }, +}; /* patch-ALC861 */ @@ -6099,6 +6117,47 @@ static struct snd_kcontrol_new alc861_3ST_mixer[] = { }, { } /* end */ }; +static struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = { + /* output mixer control */ + HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT), + /*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */ + + /* Input mixer control */ + /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */ + HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT), + + /* Capture mixer control */ + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .count = 1, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + .private_value = ARRAY_SIZE(alc861_uniwill_m31_modes), + }, + { } /* end */ +}; /* * generic initialization of ADC, input mixers and output mixers @@ -6227,6 +6286,67 @@ static struct hda_verb alc861_threestack_init_verbs[] = { {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, { } }; + +static struct hda_verb alc861_uniwill_m31_init_verbs[] = { + /* + * Unmute ADC0 and set the default input to mic-in + */ + /* port-A for surround (rear panel) */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, + /* port-B for mic-in (rear panel) with vref */ + { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + /* port-C for line-in (rear panel) */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, + /* port-D for Front */ + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 }, + /* port-E for HP out (front panel) */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, // this has to be set to VREF80 + /* route front PCM to HP */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x01 }, + /* port-F for mic-in (front panel) with vref */ + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + /* port-G for CLFE (rear panel) */ + { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, + /* port-H for side (rear panel) */ + { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, + /* CD-in */ + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, + /* route front mic to ADC1*/ + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + /* Unmute DAC0~3 & spdif out*/ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Unmute Mixer 14 (mic) 1c (Line in)*/ + {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* Unmute Stereo Mixer 15 */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c }, //Output 0~12 step + + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, // hp used DAC 3 (Front) + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + { } +}; + /* * generic initialization of ADC, input mixers and output mixers */ @@ -6561,6 +6681,9 @@ static struct hda_board_config alc861_cfg_tbl[] = { .config = ALC660_3ST }, { .modelname = "3stack-dig", .config = ALC861_3ST_DIG }, { .modelname = "6stack-dig", .config = ALC861_6ST_DIG }, + { .modelname = "uniwill-m31", .config = ALC861_UNIWILL_M31}, + { .pci_subvendor = 0x1584, .pci_subdevice = 0x9072, + .config = ALC861_UNIWILL_M31 }, { .modelname = "auto", .config = ALC861_AUTO }, {} }; @@ -6615,6 +6738,20 @@ static struct alc_config_preset alc861_presets[] = { .adc_nids = alc861_adc_nids, .input_mux = &alc861_capture_source, }, + [ALC861_UNIWILL_M31] = { + .mixers = { alc861_uniwill_m31_mixer }, + .init_verbs = { alc861_uniwill_m31_init_verbs }, + .num_dacs = ARRAY_SIZE(alc861_dac_nids), + .dac_nids = alc861_dac_nids, + .dig_out_nid = ALC861_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc861_uniwill_m31_modes), + .channel_mode = alc861_uniwill_m31_modes, + .need_dac_fix = 1, + .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), + .adc_nids = alc861_adc_nids, + .input_mux = &alc861_capture_source, + }, + }; -- cgit v1.2.3-70-g09d2 From c256652466127872f1b2e510431dc25524ba40ba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Aug 2006 18:21:36 +0200 Subject: [ALSA] Add missing TLV callbacks for HD-audio codecs Added missing TLV callbacks for HD-audio codec supports. Also cleaned up the tlv callback for ad1986a (no mutex is needed there). Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 16 ++-------------- sound/pci/hda/patch_realtek.c | 1 + sound/pci/hda/patch_sigmatel.c | 1 + 3 files changed, 4 insertions(+), 14 deletions(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 043256c67d1..71abc2aa61a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -452,19 +452,6 @@ static int ad1986a_pcm_amp_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl return change; } -static int ad1986a_pcm_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, - unsigned int size, unsigned int __user *_tlv) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *ad = codec->spec; - - mutex_lock(&ad->amp_mutex); - snd_hda_mixer_amp_tlv(kcontrol, op_flag, size, _tlv); - mutex_unlock(&ad->amp_mutex); - return 0; -} - - #define ad1986a_pcm_amp_sw_info snd_hda_mixer_amp_switch_info static int ad1986a_pcm_amp_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -507,7 +494,7 @@ static struct snd_kcontrol_new ad1986a_mixers[] = { .info = ad1986a_pcm_amp_vol_info, .get = ad1986a_pcm_amp_vol_get, .put = ad1986a_pcm_amp_vol_put, - .tlv = { .c = ad1986a_pcm_amp_tlv }, + .tlv = { .c = snd_hda_mixer_amp_tlv }, .private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT) }, { @@ -654,6 +641,7 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { .info = snd_hda_mixer_amp_volume_info, .get = snd_hda_mixer_amp_volume_get, .put = ad1986a_laptop_master_vol_put, + .tlv = { .c = snd_hda_mixer_amp_tlv }, .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), }, { diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f857e963ff4..79d361260b2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5540,6 +5540,7 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { .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), }, { diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7eaf755b014..887b52e96ec 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1528,6 +1528,7 @@ static struct snd_kcontrol_new vaio_mixer[] = { .info = snd_hda_mixer_amp_volume_info, .get = snd_hda_mixer_amp_volume_get, .put = vaio_master_vol_put, + .tlv = { .c = snd_hda_mixer_amp_tlv }, .private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), }, { -- cgit v1.2.3-70-g09d2 From 2aaeee8bd1cf51b6ed7c751a8472cb77f3ddc642 Mon Sep 17 00:00:00 2001 From: Tobin Davis Date: Mon, 21 Aug 2006 19:01:12 +0200 Subject: [ALSA] hda-codec - add missing device ids This patch adds missing device ids for Intel 915 and D102GGC motherboards. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 79d361260b2..53aa57f5a1a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2143,7 +2143,10 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x8086, .pci_subdevice = 0xe20f, .config = ALC880_3ST }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe210, .config = ALC880_3ST }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe211, .config = ALC880_3ST }, + { .pci_subvendor = 0x8086, .pci_subdevice = 0xe212, .config = ALC880_3ST }, + { .pci_subvendor = 0x8086, .pci_subdevice = 0xe213, .config = ALC880_3ST }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe214, .config = ALC880_3ST }, + { .pci_subvendor = 0x8086, .pci_subdevice = 0xe234, .config = ALC880_3ST }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe302, .config = ALC880_3ST }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe303, .config = ALC880_3ST }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe304, .config = ALC880_3ST }, @@ -5058,6 +5061,8 @@ static struct hda_board_config alc883_cfg_tbl[] = { { .modelname = "3stack-6ch", .config = ALC883_3ST_6ch }, { .pci_subvendor = 0x108e, .pci_subdevice = 0x534d, .config = ALC883_3ST_6ch }, + { .pci_subvendor = 0x8086, .pci_subdevice = 0xd601, + .config = ALC883_3ST_6ch }, /* D102GGC */ { .modelname = "6stack-dig", .config = ALC883_6ST_DIG }, { .pci_subvendor = 0x1462, .pci_subdevice = 0x6668, .config = ALC883_6ST_DIG }, /* MSI */ -- cgit v1.2.3-70-g09d2 From bab282b912baf372d8f705357946ef691b621899 Mon Sep 17 00:00:00 2001 From: Vladimir Avdonin Date: Tue, 22 Aug 2006 13:31:58 +0200 Subject: [ALSA] hda-codec - Fix for Acer laptops with ALC883 codec Patch enables the internal speaker on acer laptops with ALC883. Signed-off-by: Vladimir Avdonin Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_realtek.c | 21 +++++++++++++++++++++ 2 files changed, 22 insertions(+) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index d7e95f14456..504ebbceafb 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -820,6 +820,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-6ch 3-jack 6-channel 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O 6stack-dig-demo 6-jack digital for Intel demo board + acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) auto auto-config reading BIOS (default) ALC861/660 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 53aa57f5a1a..65903812b30 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -111,6 +111,7 @@ enum { ALC883_3ST_6ch, ALC883_6ST_DIG, ALC888_DEMO_BOARD, + ALC883_ACER, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -5069,6 +5070,9 @@ static struct hda_board_config alc883_cfg_tbl[] = { { .pci_subvendor = 0x105b, .pci_subdevice = 0x6668, .config = ALC883_6ST_DIG }, /* Foxconn */ { .modelname = "6stack-dig-demo", .config = ALC888_DEMO_BOARD }, + { .modelname = "acer", .config = ALC883_ACER }, + { .pci_subvendor = 0x1025, .pci_subdevice = 0/*0x0102*/, + .config = ALC883_ACER }, { .modelname = "auto", .config = ALC883_AUTO }, {} }; @@ -5139,6 +5143,23 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, }, + [ALC883_ACER] = { + .mixers = { alc883_base_mixer, + alc883_chmode_mixer }, + /* On TravelMate laptops, GPIO 0 enables the internal speaker + * and the headphone jack. Turn this on and rely on the + * standard mute methods whenever the user wants to turn + * these outputs off. + */ + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_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, + }, }; -- cgit v1.2.3-70-g09d2 From cd417d4fe89638a2848980cb389b9781d4913173 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Sep 2006 16:03:11 +0200 Subject: [ALSA] hda-codec - Add support for LG LW25 laptop Added the support for LG LW25 laptop with ALC880 codec. It's the same codec model as LG LW20 (model=lg-lw). Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/ALSA-Configuration.txt | 2 +- sound/pci/hda/patch_realtek.c | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index a788dd7bc79..1b749947233 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -785,7 +785,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. uniwill 3-jack F1734 2-jack lg LG laptop (m1 express dual) - lg-lw LG LW20 laptop + lg-lw LG LW20/LW25 laptop tcl TCL S700 clevo Clevo laptops (m520G, m665n) test for testing/debugging purpose, almost all controls can be diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 65903812b30..d037051b66b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2270,6 +2270,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .modelname = "lg-lw", .config = ALC880_LG_LW }, { .pci_subvendor = 0x1854, .pci_subdevice = 0x0018, .config = ALC880_LG_LW }, + { .pci_subvendor = 0x1854, .pci_subdevice = 0x0077, .config = ALC880_LG_LW }, #ifdef CONFIG_SND_DEBUG { .modelname = "test", .config = ALC880_TEST }, -- cgit v1.2.3-70-g09d2 From e08a007d1041e0bc3df6b855043d8efde91851aa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Sep 2006 17:52:14 +0200 Subject: [ALSA] hda-codec - Fix SPDIF device number of ALC codecs Assign the SPDIF always to the secondary device (dev#1) to keep the same configuration. Move the optional capture device to the third device (dev#2). hda_intel now just ignores the NULL entries in the pcm arrays from codecs. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 10 ++++++++-- sound/pci/hda/patch_realtek.c | 38 ++++++++++++++++++++------------------ 2 files changed, 28 insertions(+), 20 deletions(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bfd74a526b8..6309e0c67e6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1242,7 +1242,12 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec, struct snd_pcm *pcm; struct azx_pcm *apcm; - snd_assert(cpcm->stream[0].substreams || cpcm->stream[1].substreams, return -EINVAL); + /* if no substreams are defined for both playback and capture, + * it's just a placeholder. ignore it. + */ + if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams) + return 0; + snd_assert(cpcm->name, return -EINVAL); err = snd_pcm_new(chip->card, cpcm->name, pcm_dev, @@ -1268,7 +1273,8 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec, snd_dma_pci_data(chip->pci), 1024 * 64, 1024 * 128); chip->pcm[pcm_dev] = pcm; - chip->pcm_devs = pcm_dev + 1; + if (chip->pcm_devs < pcm_dev + 1) + chip->pcm_devs = pcm_dev + 1; return 0; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d037051b66b..ba9e050e201 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1796,25 +1796,9 @@ static int alc_build_pcms(struct hda_codec *codec) } } - /* If the use of more than one ADC is requested for the current - * model, configure a second analog capture-only PCM. - */ - if (spec->num_adc_nids > 1) { - codec->num_pcms++; - info++; - info->name = spec->stream_name_analog; - /* No playback stream for second PCM */ - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = alc_pcm_null_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0; - if (spec->stream_analog_capture) { - snd_assert(spec->adc_nids, return -EINVAL); - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[1]; - } - } - + /* SPDIF for stream index #1 */ if (spec->multiout.dig_out_nid || spec->dig_in_nid) { - codec->num_pcms++; + codec->num_pcms = 2; info++; info->name = spec->stream_name_digital; if (spec->multiout.dig_out_nid && @@ -1829,6 +1813,24 @@ static int alc_build_pcms(struct hda_codec *codec) } } + /* If the use of more than one ADC is requested for the current + * model, configure a second analog capture-only PCM. + */ + /* Additional Analaog capture for index #2 */ + if (spec->num_adc_nids > 1 && spec->stream_analog_capture && + spec->adc_nids) { + codec->num_pcms = 3; + info++; + info->name = spec->stream_name_analog; + /* No playback stream for second PCM */ + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = alc_pcm_null_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0; + if (spec->stream_analog_capture) { + info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[1]; + } + } + return 0; } -- cgit v1.2.3-70-g09d2 From eb06ed8f4c2440558ebf465e8baeac6367d90201 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Sep 2006 17:10:27 +0200 Subject: [ALSA] hda-codec - Support multiple headphone pins Some machines have multiple headpohne pins (usually on the lpatop and on the docking station) while the current hda-codec driver assumes a single headphone pin. Now it supports multiple hp pins (at least for detection). The sigmatel 92xx code supports this new multiple hp pins. It detects all hp pins for auto-muting, too. Also, the driver checks speaker pins in addition. In some cases, all line-out, speaker and hp-pins coexist. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_codec.c | 23 +++-- sound/pci/hda/hda_local.h | 3 +- sound/pci/hda/patch_analog.c | 4 +- sound/pci/hda/patch_realtek.c | 18 ++-- sound/pci/hda/patch_sigmatel.c | 202 +++++++++++++++++++++++++++-------------- 5 files changed, 164 insertions(+), 86 deletions(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8b2c080c85a..07360996caa 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2012,7 +2012,7 @@ static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list) * in the order of front, rear, CLFE, side, ... * * If more extra outputs (speaker and headphone) are found, the pins are - * assisnged to hp_pin and speaker_pins[], respectively. If no line-out jack + * assisnged to hp_pins[] and speaker_pins[], respectively. If no line-out jack * is detected, one of speaker of HP pins is assigned as the primary * output, i.e. to line_out_pins[0]. So, line_outs is always positive * if any analog output exists. @@ -2074,7 +2074,10 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c cfg->speaker_outs++; break; case AC_JACK_HP_OUT: - cfg->hp_pin = nid; + if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins)) + continue; + cfg->hp_pins[cfg->hp_outs] = nid; + cfg->hp_outs++; break; case AC_JACK_MIC_IN: if (loc == AC_JACK_LOC_FRONT) @@ -2147,8 +2150,10 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c cfg->speaker_outs, cfg->speaker_pins[0], cfg->speaker_pins[1], cfg->speaker_pins[2], cfg->speaker_pins[3], cfg->speaker_pins[4]); - snd_printd(" hp=0x%x, dig_out=0x%x, din_in=0x%x\n", - cfg->hp_pin, cfg->dig_out_pin, cfg->dig_in_pin); + snd_printd(" hp_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", + cfg->hp_outs, cfg->hp_pins[0], + cfg->hp_pins[1], cfg->hp_pins[2], + cfg->hp_pins[3], cfg->hp_pins[4]); snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x," " cd=0x%x, aux=0x%x\n", cfg->input_pins[AUTO_PIN_MIC], @@ -2169,10 +2174,12 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c sizeof(cfg->speaker_pins)); cfg->speaker_outs = 0; memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins)); - } else if (cfg->hp_pin) { - cfg->line_outs = 1; - cfg->line_out_pins[0] = cfg->hp_pin; - cfg->hp_pin = 0; + } else if (cfg->hp_outs) { + cfg->line_outs = cfg->hp_outs; + memcpy(cfg->line_out_pins, cfg->hp_pins, + sizeof(cfg->hp_pins)); + cfg->hp_outs = 0; + memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins)); } } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index ff24266fe35..f9416c36396 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -229,7 +229,8 @@ struct auto_pin_cfg { hda_nid_t line_out_pins[5]; /* sorted in the order of Front/Surr/CLFE/Side */ int speaker_outs; hda_nid_t speaker_pins[5]; - hda_nid_t hp_pin; + int hp_outs; + hda_nid_t hp_pins[5]; hda_nid_t input_pins[AUTO_PIN_LAST]; hda_nid_t dig_out_pin; hda_nid_t dig_in_pin; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 71abc2aa61a..511df07fa2a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2471,7 +2471,7 @@ static void ad1988_auto_init_extra_out(struct hda_codec *codec) pin = spec->autocfg.speaker_pins[0]; if (pin) /* connect to front */ ad1988_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); - pin = spec->autocfg.hp_pin; + pin = spec->autocfg.hp_pins[0]; if (pin) /* connect to front */ ad1988_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); } @@ -2523,7 +2523,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) (err = ad1988_auto_create_extra_out(codec, spec->autocfg.speaker_pins[0], "Speaker")) < 0 || - (err = ad1988_auto_create_extra_out(codec, spec->autocfg.hp_pin, + (err = ad1988_auto_create_extra_out(codec, spec->autocfg.hp_pins[0], "Headphone")) < 0 || (err = ad1988_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) return err; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ba9e050e201..d08d2e399c8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2753,7 +2753,7 @@ static void alc880_auto_init_extra_out(struct hda_codec *codec) pin = spec->autocfg.speaker_pins[0]; if (pin) /* connect to front */ alc880_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); - pin = spec->autocfg.hp_pin; + pin = spec->autocfg.hp_pins[0]; if (pin) /* connect to front */ alc880_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); } @@ -2794,7 +2794,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) (err = alc880_auto_create_extra_out(spec, spec->autocfg.speaker_pins[0], "Speaker")) < 0 || - (err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pin, + (err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0], "Headphone")) < 0 || (err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) return err; @@ -3736,7 +3736,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, return err; } - nid = cfg->hp_pin; + nid = cfg->hp_pins[0]; if (nid) { err = alc260_add_playback_controls(spec, nid, "Headphone"); if (err < 0) @@ -3806,7 +3806,7 @@ static void alc260_auto_init_multi_out(struct hda_codec *codec) if (nid) alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0); - nid = spec->autocfg.hp_pin; + nid = spec->autocfg.hp_pins[0]; if (nid) alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0); } @@ -4526,7 +4526,7 @@ static void alc882_auto_init_hp_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; hda_nid_t pin; - pin = spec->autocfg.hp_pin; + pin = spec->autocfg.hp_pins[0]; if (pin) /* connect to front */ alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); /* use dac 0 */ } @@ -5207,7 +5207,7 @@ static void alc883_auto_init_hp_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; hda_nid_t pin; - pin = spec->autocfg.hp_pin; + pin = spec->autocfg.hp_pins[0]; if (pin) /* connect to front */ /* use dac 0 */ alc883_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); @@ -5630,7 +5630,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct return err; } } - nid = cfg->hp_pin; + nid = cfg->hp_pins[0]; if (nid) { /* spec->multiout.hp_nid = 2; */ if (nid == 0x16) { @@ -6630,7 +6630,7 @@ static void alc861_auto_init_hp_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; hda_nid_t pin; - pin = spec->autocfg.hp_pin; + pin = spec->autocfg.hp_pins[0]; if (pin) /* connect to front */ alc861_auto_set_output_and_unmute(codec, pin, PIN_HP, spec->multiout.dac_nids[0]); } @@ -6665,7 +6665,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) if ((err = alc861_auto_fill_dac_nids(spec, &spec->autocfg)) < 0 || (err = alc861_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || - (err = alc861_auto_create_hp_ctls(spec, spec->autocfg.hp_pin)) < 0 || + (err = alc861_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0])) < 0 || (err = alc861_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) return err; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index bcbbe111ab9..7cc06426520 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1011,11 +1011,29 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, return 0; } +/* create volume control/switch for the given prefx type */ +static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_t nid, int chs) +{ + char name[32]; + int err; + + sprintf(name, "%s Playback Volume", pfx); + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", pfx); + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); + if (err < 0) + return err; + return 0; +} + /* add playback controls from the parsed DAC table */ static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec, const struct auto_pin_cfg *cfg) { - char name[32]; static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; @@ -1030,26 +1048,15 @@ static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec, if (i == 2) { /* Center/LFE */ - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "Center Playback Volume", - HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0) + err = create_controls(spec, "Center", nid, 1); + if (err < 0) return err; - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "LFE Playback Volume", - HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0) - return err; - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "Center Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0) - return err; - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "LFE Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0) + err = create_controls(spec, "LFE", nid, 2); + if (err < 0) return err; } else { - sprintf(name, "%s Playback Volume", chname[i]); - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) - return err; - sprintf(name, "%s Playback Switch", chname[i]); - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) + err = create_controls(spec, chname[i], nid, 3); + if (err < 0) return err; } } @@ -1065,39 +1072,85 @@ static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec, return 0; } -/* add playback controls for HP output */ -static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin_cfg *cfg) +static int check_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) { - struct sigmatel_spec *spec = codec->spec; - hda_nid_t pin = cfg->hp_pin; - hda_nid_t nid; - int i, err; - unsigned int wid_caps; + int i; - if (! pin) - return 0; + for (i = 0; i < spec->multiout.num_dacs; i++) { + if (spec->multiout.dac_nids[i] == nid) + return 1; + } + if (spec->multiout.hp_nid == nid) + return 1; + return 0; +} - wid_caps = get_wcaps(codec, pin); - if (wid_caps & AC_WCAP_UNSOL_CAP) - spec->hp_detect = 1; +static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) +{ + if (!spec->multiout.hp_nid) + spec->multiout.hp_nid = nid; + else if (spec->multiout.num_dacs > 4) { + printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid); + return 1; + } else { + spec->multiout.dac_nids[spec->multiout.num_dacs] = nid; + spec->multiout.num_dacs++; + } + return 0; +} - nid = snd_hda_codec_read(codec, pin, 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff; - for (i = 0; i < cfg->line_outs; i++) { - if (! spec->multiout.dac_nids[i]) +/* add playback controls for Speaker and HP outputs */ +static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, + struct auto_pin_cfg *cfg) +{ + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid; + int i, old_num_dacs, err; + + old_num_dacs = spec->multiout.num_dacs; + for (i = 0; i < cfg->hp_outs; i++) { + unsigned int wid_caps = get_wcaps(codec, cfg->hp_pins[i]); + if (wid_caps & AC_WCAP_UNSOL_CAP) + spec->hp_detect = 1; + nid = snd_hda_codec_read(codec, cfg->hp_pins[i], 0, + AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (check_in_dac_nids(spec, nid)) + nid = 0; + if (! nid) continue; - if (spec->multiout.dac_nids[i] == nid) - return 0; + add_spec_dacs(spec, nid); + } + for (i = 0; i < cfg->speaker_outs; i++) { + nid = snd_hda_codec_read(codec, cfg->speaker_pins[0], 0, + AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (check_in_dac_nids(spec, nid)) + nid = 0; + if (check_in_dac_nids(spec, nid)) + nid = 0; + if (! nid) + continue; + add_spec_dacs(spec, nid); } - spec->multiout.hp_nid = nid; - - /* control HP volume/switch on the output mixer amp */ - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "Headphone Playback Volume", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) - return err; - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "Headphone Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) - return err; + for (i = old_num_dacs; i < spec->multiout.num_dacs; i++) { + static const char *pfxs[] = { + "Speaker", "External Speaker", "Speaker2", + }; + err = create_controls(spec, pfxs[i - old_num_dacs], + spec->multiout.dac_nids[i], 3); + if (err < 0) + return err; + } + if (spec->multiout.hp_nid) { + const char *pfx; + if (old_num_dacs == spec->multiout.num_dacs) + pfx = "Master"; + else + pfx = "Headphone"; + err = create_controls(spec, pfx, spec->multiout.hp_nid, 3); + if (err < 0) + return err; + } return 0; } @@ -1160,11 +1213,20 @@ static void stac92xx_auto_init_multi_out(struct hda_codec *codec) static void stac92xx_auto_init_hp_out(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - hda_nid_t pin; + int i; - pin = spec->autocfg.hp_pin; - if (pin) /* connect to front */ - stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); + for (i = 0; i < spec->autocfg.hp_outs; i++) { + hda_nid_t pin; + pin = spec->autocfg.hp_pins[i]; + if (pin) /* connect to front */ + stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); + } + for (i = 0; i < spec->autocfg.speaker_outs; i++) { + hda_nid_t pin; + pin = spec->autocfg.speaker_pins[i]; + if (pin) /* connect to front */ + stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN); + } } static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in) @@ -1210,7 +1272,7 @@ static int stac9200_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; - hda_nid_t pin = cfg->hp_pin; + hda_nid_t pin = cfg->hp_pins[0]; unsigned int wid_caps; if (! pin) @@ -1266,16 +1328,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, } if (lfe_pin) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, - "LFE Playback Volume", - HDA_COMPOSE_AMP_VAL(lfe_pin, 1, 0, - HDA_OUTPUT)); - if (err < 0) - return err; - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, - "LFE Playback Switch", - HDA_COMPOSE_AMP_VAL(lfe_pin, 1, 0, - HDA_OUTPUT)); + err = create_controls(spec, "LFE", lfe_pin, 1); if (err < 0) return err; } @@ -1363,9 +1416,11 @@ static int stac92xx_init(struct hda_codec *codec) /* set up pins */ if (spec->hp_detect) { /* Enable unsolicited responses on the HP widget */ - snd_hda_codec_write(codec, cfg->hp_pin, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - STAC_UNSOL_ENABLE); + for (i = 0; i < cfg->hp_outs; i++) + if (get_wcaps(codec, cfg->hp_pins[i]) & AC_WCAP_UNSOL_CAP) + snd_hda_codec_write(codec, cfg->hp_pins[i], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + STAC_UNSOL_ENABLE); /* fake event to set up pins */ codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); /* enable the headphones by default. If/when unsol_event detection works, this will be ignored */ @@ -1447,21 +1502,36 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) if ((res >> 26) != STAC_HP_EVENT) return; - presence = snd_hda_codec_read(codec, cfg->hp_pin, 0, - AC_VERB_GET_PIN_SENSE, 0x00) >> 31; + presence = 0; + for (i = 0; i < cfg->hp_outs; i++) { + int p = snd_hda_codec_read(codec, cfg->hp_pins[i], 0, + AC_VERB_GET_PIN_SENSE, 0x00); + if (p & (1 << 31)) + presence++; + } if (presence) { /* disable lineouts, enable hp */ for (i = 0; i < cfg->line_outs; i++) stac92xx_reset_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); - stac92xx_set_pinctl(codec, cfg->hp_pin, AC_PINCTL_OUT_EN); + for (i = 0; i < cfg->speaker_outs; i++) + stac92xx_reset_pinctl(codec, cfg->speaker_pins[i], + AC_PINCTL_OUT_EN); + for (i = 0; i < cfg->hp_outs; i++) + stac92xx_set_pinctl(codec, cfg->hp_pins[i], + AC_PINCTL_OUT_EN); } else { /* enable lineouts, disable hp */ for (i = 0; i < cfg->line_outs; i++) stac92xx_set_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); - stac92xx_reset_pinctl(codec, cfg->hp_pin, AC_PINCTL_OUT_EN); + for (i = 0; i < cfg->speaker_outs; i++) + stac92xx_set_pinctl(codec, cfg->speaker_pins[i], + AC_PINCTL_OUT_EN); + for (i = 0; i < cfg->hp_outs; i++) + stac92xx_reset_pinctl(codec, cfg->hp_pins[i], + AC_PINCTL_OUT_EN); } } -- cgit v1.2.3-70-g09d2