From dfc0ff62a1d24e987205b41fbf322a4377626481 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 May 2005 14:31:49 +0200 Subject: [PATCH] [ALSA] Add ASUS Z71V support Documentation,HDA Codec driver Added the ASUS Z71V (or similar) laptop support. Signed-off-by: Takashi Iwai --- .../sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_realtek.c | 109 ++++++++++++++++++ 2 files changed, 110 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index ef07506e583c..d49325ed706f 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -636,6 +636,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 5stack 5-jack in back, 2-jack in front 5stack-digout 5-jack in back, 2-jack in front, a SPDIF out w810 3-jack + z71v 3-jack (HP shared SPDIF) CMI9880 minimal 3-jack in back diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 17c5062423ae..c106e1fe01cf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -39,6 +39,7 @@ enum { ALC880_5ST, ALC880_5ST_DIG, ALC880_W810, + ALC880_Z71V, }; struct alc_spec { @@ -90,6 +91,11 @@ static hda_nid_t alc880_w810_dac_nids[3] = { 0x02, 0x03, 0x04 }; +static hda_nid_t alc880_z71v_dac_nids[1] = { + /* front only? */ + 0x02 +}; + static hda_nid_t alc880_adc_nids[3] = { /* ADC0-2 */ 0x07, 0x08, 0x09, @@ -284,6 +290,10 @@ static struct alc_channel_mode alc880_w810_modes[1] = { { 6, NULL } }; +static struct alc_channel_mode alc880_z71v_modes[1] = { + { 2, NULL } +}; + /* */ static int alc880_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) @@ -475,6 +485,35 @@ static snd_kcontrol_new_t alc880_w810_base_mixer[] = { { } /* end */ }; +static snd_kcontrol_new_t alc880_z71v_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, + }, + { } /* end */ +}; + /* */ static int alc_build_controls(struct hda_codec *codec) @@ -719,6 +758,58 @@ static struct hda_verb alc880_w810_init_verbs[] = { { } }; +static struct hda_verb alc880_z71v_init_verbs[] = { + /* front channel selector/amp: input 0: DAC: unmuted, (no volume selection) */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* front channel selector/amp: input 1: capture mix: muted, (no volume selection) */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180}, + /* front channel selector/amp: output 0: unmuted, max volume */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* front out pin: muted, (no volume selection) */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + /* front out pin: NOT headphone enable, out enable, vref disabled */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* headphone channel selector/amp: input 0: DAC: unmuted, (no volume selection) */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* headphone channel selector/amp: input 1: capture mix: muted, (no volume selection) */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180}, + /* headphone channel selector/amp: output 0: unmuted, max volume */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* headphone out pin: muted, (no volume selection) */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + /* headpohne out pin: headphone enable, out enable, vref disabled */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, + + /* Line In pin widget for input */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + /* CD pin widget for input */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + /* Mic1 (rear panel) pin widget for input and vref at 80% */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + /* Mic2 (front panel) pin widget for input and vref at 80% */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + /* unmute amp left and right */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000}, + /* set connection select to line in (default select for this ADC) */ + {0x07, AC_VERB_SET_CONNECT_SEL, 0x02}, + + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) for mixer + * widget(nid=0x0B) to support the input path of analog loopback + */ + /* Note: PASD motherboards uses the Line In 2 as the input for front panel mic (mic 2) */ + /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03*/ + /* unmute CD */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, + /* unmute Line In */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, + /* unmute Mic 1 */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + /* unmute Line In 2 (for PASD boards Mic 2) */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, + + { } +}; + static int alc_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -993,6 +1084,9 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .modelname = "w810", .config = ALC880_W810 }, { .pci_vendor = 0x161f, .pci_device = 0x203d, .config = ALC880_W810 }, + { .modelname = "z71v", .config = ALC880_Z71V }, + { .pci_vendor = 0x1043, .pci_device = 0x1964, .config = ALC880_Z71V }, + {} }; @@ -1023,6 +1117,10 @@ static int patch_alc880(struct hda_codec *codec) spec->mixers[spec->num_mixers] = alc880_five_stack_mixer; spec->num_mixers++; break; + case ALC880_Z71V: + spec->mixers[spec->num_mixers] = alc880_z71v_mixer; + spec->num_mixers++; + break; default: spec->mixers[spec->num_mixers] = alc880_base_mixer; spec->num_mixers++; @@ -1033,6 +1131,7 @@ static int patch_alc880(struct hda_codec *codec) case ALC880_3ST_DIG: case ALC880_5ST_DIG: case ALC880_W810: + case ALC880_Z71V: spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; break; default: @@ -1063,6 +1162,11 @@ static int patch_alc880(struct hda_codec *codec) spec->channel_mode = alc880_w810_modes; spec->num_channel_mode = ARRAY_SIZE(alc880_w810_modes); break; + case ALC880_Z71V: + spec->init_verbs = alc880_z71v_init_verbs; + spec->channel_mode = alc880_z71v_modes; + spec->num_channel_mode = ARRAY_SIZE(alc880_z71v_modes); + break; default: spec->init_verbs = alc880_init_verbs_three_stack; spec->channel_mode = alc880_threestack_modes; @@ -1086,6 +1190,11 @@ static int patch_alc880(struct hda_codec *codec) spec->multiout.dac_nids = alc880_w810_dac_nids; // No dedicated headphone socket - it's shared with built-in speakers. break; + case ALC880_Z71V: + spec->multiout.num_dacs = ARRAY_SIZE(alc880_z71v_dac_nids); + spec->multiout.dac_nids = alc880_z71v_dac_nids; + spec->multiout.hp_nid = 0x03; + break; default: spec->multiout.num_dacs = ARRAY_SIZE(alc880_dac_nids); spec->multiout.dac_nids = alc880_dac_nids; -- GitLab