From 4b33c7675d2b0d4a9cb4e38cd73aa1d940f9278d Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Fri, 10 Oct 2008 09:07:23 -0400 Subject: ALSA: hda: add mixers for analog mixer on 92hd75xx codecs Add support for mixers on the analog mixer on some 92hd75xx codecs, along with adding a 'Mixer' entry for it's connection on the dmux. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 50 ++++++++++++++++++++++++++++++------------ 1 file changed, 36 insertions(+), 14 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c461baa83c2..1e7b6c111b2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -322,8 +322,8 @@ static hda_nid_t stac92hd71bxx_mux_nids[2] = { 0x1a, 0x1b }; -static hda_nid_t stac92hd71bxx_dmux_nids[1] = { - 0x1c, +static hda_nid_t stac92hd71bxx_dmux_nids[2] = { + 0x1c, 0x1d, }; static hda_nid_t stac92hd71bxx_smux_nids[2] = { @@ -861,20 +861,18 @@ static struct hda_verb stac92hd71bxx_core_init[] = { { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, }; -#define HD_DISABLE_PORTF 3 +#define HD_DISABLE_PORTF 2 static struct hda_verb stac92hd71bxx_analog_core_init[] = { /* start of config #1 */ /* connect port 0f to audio mixer */ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute right and left channels for node 0x0f */ { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* start of config #2 */ @@ -883,10 +881,6 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = { { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* connect port 0d to audio mixer */ - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, - /* unmute dac0 input in audio mixer */ - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, /* unmute right and left channels for nodes 0x0a, 0xd */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -1107,6 +1101,7 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1119,8 +1114,17 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), */ - HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x3, HDA_INPUT), + + HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x3, HDA_INPUT), + + HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT), + + HDA_CODEC_MUTE("DAC1 Capture Switch", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x17, 0x4, HDA_INPUT), { } /* end */ }; @@ -1649,7 +1653,7 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { static unsigned int ref92hd71bxx_pin_configs[11] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, - 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, + 0x0181302e, 0x01014010, 0x01019020, 0x90a000f0, 0x90a000f0, 0x01452050, 0x01452050, }; @@ -3000,7 +3004,7 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) /* labels for amp mux outputs */ static const char *stac92xx_amp_labels[3] = { - "Front Microphone", "Microphone", "Line In" + "Front Microphone", "Microphone", "Line In", }; /* create amp out controls mux on capable codecs */ @@ -4327,6 +4331,16 @@ static struct hda_codec_ops stac92hd71bxx_patch_ops = { #endif }; +static struct hda_input_mux stac92hd71bxx_dmux = { + .num_items = 4, + .items = { + { "Analog Inputs", 0x00 }, + { "Mixer", 0x01 }, + { "Digital Mic 1", 0x02 }, + { "Digital Mic 2", 0x03 }, + } +}; + static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -4341,6 +4355,8 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pin_nids = stac92hd71bxx_pin_nids; + memcpy(&spec->private_dimux, &stac92hd71bxx_dmux, + sizeof(stac92hd71bxx_dmux)); spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, stac92hd71bxx_models, @@ -4392,6 +4408,7 @@ again: /* no output amps */ spec->num_pwrs = 0; spec->mixer = stac92hd71bxx_analog_mixer; + spec->dinput_mux = &spec->private_dimux; /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; @@ -4409,12 +4426,13 @@ again: spec->num_pwrs = 0; /* fallthru */ default: + spec->dinput_mux = &spec->private_dimux; spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } - spec->aloopback_mask = 0x20; + spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; if (spec->board_config > STAC_92HD71BXX_REF) { @@ -4456,6 +4474,10 @@ again: spec->multiout.num_dacs = 1; spec->multiout.hp_nid = 0x11; spec->multiout.dac_nids = stac92hd71bxx_dac_nids; + if (spec->dinput_mux) + spec->private_dimux.num_items += + spec->num_dmics - + (ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1); err = stac92xx_parse_auto_config(codec, 0x21, 0x23); if (!err) { -- cgit v1.2.3 From 687cb98e893f492932abb3e92660d7d828bd44fb Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Sat, 11 Oct 2008 13:52:43 -0400 Subject: ALSA: hda: corrected invalid mixer values Corrected invalid mixer index values on the 92hd71bxxx codec branch. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1e7b6c111b2..c5906551311 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1114,11 +1114,11 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), */ - HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x1, HDA_INPUT), HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT), HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT), -- cgit v1.2.3 From d21995e3e3acb78e8c48c6631432a3bff191bc46 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Mon, 13 Oct 2008 13:22:45 -0400 Subject: ALSA: hda: fix nid variable warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixed compiler warning with possible uninitialized variable 'nid'. CC [M] /home/mranostay/git/alsa-driver/pci/hda/patch_sigmatel.o /home/mranostay/git/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c: In function ‘stac92xx_parse_auto_config’: /home/mranostay/git/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:2815: warning: ‘nid’ may be used uninitialized in this function Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c5906551311..a2ac7205d45 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2816,7 +2816,7 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; - hda_nid_t nid; + hda_nid_t nid = 0; int i, err; struct sigmatel_spec *spec = codec->spec; -- cgit v1.2.3 From 7fb0d78fb155845812e98ed10605d8f01963ce05 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 15 Oct 2008 11:12:35 +0200 Subject: ALSA: hda - Add auto mic switch in realtek auto-probe mode Add the automatic mic switch via jack sensing in auto-probe mode for Realtek codecs. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 65 +++++++++++++++++++++++++++++++++---------- 1 file changed, 50 insertions(+), 15 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0b6e682c46d..80c3f642007 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -822,6 +822,27 @@ static void alc_sku_automute(struct hda_codec *codec) spec->jack_present ? 0 : PIN_OUT); } +static void alc_mic_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + unsigned int present; + unsigned int mic_nid = spec->autocfg.input_pins[AUTO_PIN_MIC]; + unsigned int fmic_nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC]; + unsigned int mix_nid = spec->capsrc_nids[0]; + unsigned int capsrc_idx_mic, capsrc_idx_fmic; + + capsrc_idx_mic = mic_nid - 0x18; + capsrc_idx_fmic = fmic_nid - 0x18; + present = snd_hda_codec_read(codec, mic_nid, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_write(codec, mix_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (capsrc_idx_mic << 8) | (present ? 0 : 0x80)); + snd_hda_codec_write(codec, mix_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (capsrc_idx_fmic << 8) | (present ? 0x80 : 0)); + snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, capsrc_idx_fmic, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + /* unsolicited event for HP jack sensing */ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) { @@ -829,10 +850,17 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) res >>= 28; else res >>= 26; - if (res != ALC880_HP_EVENT) - return; + if (res == ALC880_HP_EVENT) + alc_sku_automute(codec); + if (res == ALC880_MIC_EVENT) + alc_mic_automute(codec); +} + +static void alc_inithook(struct hda_codec *codec) +{ alc_sku_automute(codec); + alc_mic_automute(codec); } /* additional initialization for ALC888 variants */ @@ -1018,10 +1046,17 @@ do_sku: else return; } + if (spec->autocfg.hp_pins[0]) + snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | ALC880_HP_EVENT); - snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ALC880_HP_EVENT); + if (spec->autocfg.input_pins[AUTO_PIN_MIC] && + spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC]) + snd_hda_codec_write(codec, + spec->autocfg.input_pins[AUTO_PIN_MIC], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | ALC880_MIC_EVENT); spec->unsol_event = alc_sku_unsol_event; } @@ -3808,7 +3843,7 @@ static void alc880_auto_init(struct hda_codec *codec) alc880_auto_init_extra_out(codec); alc880_auto_init_analog_input(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } /* @@ -5219,7 +5254,7 @@ static void alc260_auto_init(struct hda_codec *codec) alc260_auto_init_multi_out(codec); alc260_auto_init_analog_input(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -6629,7 +6664,7 @@ static void alc882_auto_init(struct hda_codec *codec) alc882_auto_init_analog_input(codec); alc882_auto_init_input_src(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } static int patch_alc883(struct hda_codec *codec); /* called in patch_alc882() */ @@ -8758,7 +8793,7 @@ static void alc883_auto_init(struct hda_codec *codec) alc883_auto_init_analog_input(codec); alc883_auto_init_input_src(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } static int patch_alc883(struct hda_codec *codec) @@ -10285,7 +10320,7 @@ static void alc262_auto_init(struct hda_codec *codec) alc262_auto_init_analog_input(codec); alc262_auto_init_input_src(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } /* @@ -11417,7 +11452,7 @@ static void alc268_auto_init(struct hda_codec *codec) alc268_auto_init_mono_speaker_out(codec); alc268_auto_init_analog_input(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } /* @@ -12200,7 +12235,7 @@ static void alc269_auto_init(struct hda_codec *codec) alc269_auto_init_hp_out(codec); alc269_auto_init_analog_input(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } /* @@ -13281,7 +13316,7 @@ static void alc861_auto_init(struct hda_codec *codec) alc861_auto_init_hp_out(codec); alc861_auto_init_analog_input(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -14393,7 +14428,7 @@ static void alc861vd_auto_init(struct hda_codec *codec) alc861vd_auto_init_analog_input(codec); alc861vd_auto_init_input_src(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } static int patch_alc861vd(struct hda_codec *codec) @@ -16223,7 +16258,7 @@ static void alc662_auto_init(struct hda_codec *codec) alc662_auto_init_analog_input(codec); alc662_auto_init_input_src(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } static int patch_alc662(struct hda_codec *codec) -- cgit v1.2.3 From a01c30cb77aa7b80ea08d003783fb9f0470455ee Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 15 Oct 2008 11:14:58 +0200 Subject: ALSA: hda - Fix PCI SSID of ASUS M90V ASUS M90V has PCI SSID 1043:1873. Corrected in the quirk list. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 80c3f642007..87b69acdd6d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8341,8 +8341,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1043, 0x8317, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601), SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), -- cgit v1.2.3 From 4442608d4b0071a00067dcbf64e7362ce08e91a5 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 15 Oct 2008 11:18:05 +0200 Subject: ALSA: hda - Add ALC1200 support Add ALC1200 codec support. Almost compatible with ALC888. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 87b69acdd6d..0e759845c45 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8837,8 +8837,13 @@ static int patch_alc883(struct hda_codec *codec) switch (codec->vendor_id) { case 0x10ec0888: - spec->stream_name_analog = "ALC888 Analog"; - spec->stream_name_digital = "ALC888 Digital"; + if (codec->revision_id == 0x100101) { + spec->stream_name_analog = "ALC1200 Analog"; + spec->stream_name_digital = "ALC1200 Digital"; + } else { + spec->stream_name_analog = "ALC888 Analog"; + spec->stream_name_digital = "ALC888 Digital"; + } break; case 0x10ec0889: spec->stream_name_analog = "ALC889 Analog"; @@ -16359,6 +16364,8 @@ struct hda_codec_preset snd_hda_preset_realtek[] = { .patch = patch_alc882 }, /* should be patch_alc883() in future */ { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, + { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", + .patch = patch_alc883 }, { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 }, {} /* terminator */ }; -- cgit v1.2.3 From a385a52925398e53bedf1a8b30a9a3e002569f27 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 15 Oct 2008 11:20:21 +0200 Subject: ALSA: hda - Add ALC887 support Added ALC887 support. It's almost compatible with ALC883/888. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0e759845c45..011f00aa0ec 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16363,6 +16363,7 @@ struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A", .patch = patch_alc882 }, /* should be patch_alc883() in future */ { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, + { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc883 }, { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", .patch = patch_alc883 }, -- cgit v1.2.3 From 01afd41f55524e8378601dbf33b858d8dd4b3f31 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 15 Oct 2008 11:22:09 +0200 Subject: ALSA: hda - Add support of ALC272 Added the support of ALC272 codec. It's almost compatible with ALC663. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 011f00aa0ec..99123a755a5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16308,6 +16308,9 @@ static int patch_alc662(struct hda_codec *codec) if (codec->vendor_id == 0x10ec0663) { spec->stream_name_analog = "ALC663 Analog"; spec->stream_name_digital = "ALC663 Digital"; + } else if (codec->vendor_id == 0x10ec0272) { + spec->stream_name_analog = "ALC272 Analog"; + spec->stream_name_digital = "ALC272 Digital"; } else { spec->stream_name_analog = "ALC662 Analog"; spec->stream_name_digital = "ALC662 Digital"; @@ -16345,6 +16348,7 @@ struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, { .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 }, { .id = 0x10ec0269, .name = "ALC269", .patch = patch_alc269 }, + { .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, -- cgit v1.2.3 From 80ffe86925a226f513b36d0ce13e049133841970 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 15 Oct 2008 11:23:27 +0200 Subject: ALSA: hda - Fix quirk lists for realtek codecs - Fix Toshiba S06 SSID to 1179:ff7b - Fix ASUS G50V quirk name - Add ASUS N20 quirk Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 99123a755a5..e72707cb60a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10383,7 +10383,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", ALC262_TOSHIBA_RX1), - SND_PCI_QUIRK(0x1179, 0x0268, "Toshiba S06", ALC262_TOSHIBA_S06), + SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), @@ -15707,7 +15707,7 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA), - SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS M51VA", ALC663_ASUS_G50V), + SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V), SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), @@ -15720,6 +15720,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2), -- cgit v1.2.3 From ec4e86ba0662ed85f3b3a38fb220dc51d951da84 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Oct 2008 08:02:41 +0200 Subject: ALSA: hda - Fix PCM type of Nvidia HDMI devices Added the missing PCM type for Nvidia HDMI devices so that they point the right device number. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_nvhdmi.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 1a65775d28e..2eed2c8b98d 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -116,6 +116,7 @@ static int nvhdmi_build_pcms(struct hda_codec *codec) codec->pcm_info = info; info->name = "NVIDIA HDMI"; + info->pcm_type = HDA_PCM_TYPE_HDMI; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = nvhdmi_pcm_digital_playback; return 0; -- cgit v1.2.3 From e78521f3212d5d3931442819cbf0910fe1b28beb Mon Sep 17 00:00:00 2001 From: Mariusz Kozlowski Date: Sun, 19 Oct 2008 10:34:22 +0200 Subject: ALSA: misc typo fixes Fixed typos in disabled codes via #if 0. Signed-off-by: Mariusz Kozlowski Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_main.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index a7d89662acf..88fbf285d2b 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -759,7 +759,6 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | SPCS_GENERATIONSTATUS | 0x00001200 | 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT ); - } #endif return 0; -- cgit v1.2.3