From ca8f04247eaaec554528279686a514c6ce087bb9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 11:51:19 +0100 Subject: ALSA: hda/realtek - Add the fixup codes for ALC260 model=will The model=will for ALC260 requires the pin 0x0f to be a headphone and some special verbs for the COEF to turn on the amp. Now added these as fixup entries and removed the static model quirk. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - 1 file changed, 1 deletion(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index c8c54544abc..fd09f050c80 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -31,7 +31,6 @@ ALC260 ====== fujitsu Fujitsu S7020 acer Acer TravelMate - will Will laptops (PB V7900) replacer Replacer 672V favorit100 Maxdata Favorit 100XS basic fixed pin assignment (old default model) -- cgit v1.2.3-70-g09d2 From 15317ab21686044f1af96dd329ba809a08f04b89 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 12:02:53 +0100 Subject: ALSA: hda/realtek - Replace ALC260 model=acer with the auto-parser The ALC260 model=acer needs GPIO1 setup. It could be selected well if the codec SSID is set properly by BIOS, but to make sure, enable it forcibly. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - sound/pci/hda/alc260_quirks.c | 146 --------------------------- sound/pci/hda/patch_realtek.c | 7 ++ 3 files changed, 7 insertions(+), 147 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index fd09f050c80..5cc76090f5d 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -30,7 +30,6 @@ ALC880 ALC260 ====== fujitsu Fujitsu S7020 - acer Acer TravelMate replacer Replacer 672V favorit100 Maxdata Favorit 100XS basic fixed pin assignment (old default model) diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c index 79aaae8e0d9..2f1594b3d4b 100644 --- a/sound/pci/hda/alc260_quirks.c +++ b/sound/pci/hda/alc260_quirks.c @@ -8,7 +8,6 @@ enum { ALC260_AUTO, ALC260_BASIC, ALC260_FUJITSU_S702X, - ALC260_ACER, ALC260_REPLACER_672V, ALC260_FAVORIT100, #ifdef CONFIG_SND_DEBUG @@ -181,48 +180,6 @@ static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = { { } /* end */ }; -/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks. Note that current - * versions of the ALC260 don't act on requests to enable mic bias from NID - * 0x0f (used to drive the headphone jack in these laptops). The ALC260 - * datasheet doesn't mention this restriction. At this stage it's not clear - * whether this behaviour is intentional or is a hardware bug in chip - * revisions available in early 2006. Therefore for now allow the - * "Headphone Jack Mode" control to span all choices, but if it turns out - * that the lack of mic bias for this NID is intentional we could change the - * mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS. - * - * In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006 - * don't appear to make the mic bias available from the "line" jack, even - * though the NID used for this jack (0x14) can supply it. The theory is - * that perhaps Acer have included blocking capacitors between the ALC260 - * and the output jack. If this turns out to be the case for all such - * models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT - * to ALC_PIN_DIR_INOUT_NOMICBIAS. - * - * The C20x Tablet series have a mono internal speaker which is controlled - * via the chip's Mono sum widget and pin complex, so include the necessary - * controls for such models. On models without a "mono speaker" the control - * won't do anything. - */ -static const struct snd_kcontrol_new alc260_acer_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), - ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT), - HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, - HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2, - HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), - ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), - HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), - ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), - { } /* end */ -}; - /* Maxdata Favorit 100XS: one output and one input (0x12) jack */ static const struct snd_kcontrol_new alc260_favorit100_mixer[] = { @@ -401,94 +358,6 @@ static const struct hda_verb alc260_fujitsu_init_verbs[] = { { } }; -/* Initialisation sequence for ALC260 as configured in Acer TravelMate and - * similar laptops (adapted from Fujitsu init verbs). - */ -static const struct hda_verb alc260_acer_init_verbs[] = { - /* 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. - */ - {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, - /* Internal speaker/Headphone jack is connected to Line-out pin */ - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Internal microphone/Mic jack is connected to Mic1 pin */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, - /* Line In jack is connected to Line1 pin */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Some Acers (eg: C20x Tablets) use Mono pin for internal speaker */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Ensure all other unused pins are disabled and muted. */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Disable digital (SPDIF) pins */ - {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, - {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum - * bus when acting as outputs. - */ - {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, - - /* Start with output sum widgets muted and their output gains at min */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* Unmute Line-out pin widget amp left and right - * (no equiv mixer ctrl) - */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute mono pin widget amp output (no equiv mixer ctrl) */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute Mic1 and Line1 pin widget input buffers since they start as - * inputs. If the pin mode is changed by the user the pin mode control - * will take care of enabling the pin's input/output buffers as needed. - * Therefore there's no need to enable the input buffer at this - * stage. - */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute capture amp left and right */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Set ADC connection select to match default mixer setting - mic - * (on mic1 pin) - */ - {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Do similar with the second ADC: mute capture input amp and - * set ADC connection to mic to match ALSA's default state. - */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ - - { } -}; - /* Initialisation sequence for Maxdata Favorit 100XS * (adapted from Acer init verbs). */ @@ -822,7 +691,6 @@ static const struct hda_verb alc260_test_init_verbs[] = { static const char * const alc260_models[ALC260_MODEL_LAST] = { [ALC260_BASIC] = "basic", [ALC260_FUJITSU_S702X] = "fujitsu", - [ALC260_ACER] = "acer", [ALC260_REPLACER_672V] = "replacer", [ALC260_FAVORIT100] = "favorit100", #ifdef CONFIG_SND_DEBUG @@ -832,8 +700,6 @@ static const char * const alc260_models[ALC260_MODEL_LAST] = { }; static const struct snd_pci_quirk alc260_cfg_tbl[] = { - SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER), - SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100), SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC), SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC), @@ -869,18 +735,6 @@ static const struct alc_config_preset alc260_presets[] = { .num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources), .input_mux = alc260_fujitsu_capture_sources, }, - [ALC260_ACER] = { - .mixers = { alc260_acer_mixer }, - .init_verbs = { alc260_acer_init_verbs }, - .num_dacs = ARRAY_SIZE(alc260_dac_nids), - .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), - .adc_nids = alc260_dual_adc_nids, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources), - .input_mux = alc260_acer_capture_sources, - }, [ALC260_FAVORIT100] = { .mixers = { alc260_favorit100_mixer }, .init_verbs = { alc260_favorit100_init_verbs }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f5f37103623..95ef722e407 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4210,6 +4210,7 @@ enum { ALC260_FIXUP_HP_DC5750, ALC260_FIXUP_HP_PIN_0F, ALC260_FIXUP_COEF, + ALC260_FIXUP_GPIO1, }; static const struct alc_fixup alc260_fixups[] = { @@ -4237,10 +4238,16 @@ static const struct alc_fixup alc260_fixups[] = { .chained = true, .chain_id = ALC260_FIXUP_HP_PIN_0F, }, + [ALC260_FIXUP_GPIO1] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = alc_gpio1_init_verbs, + }, }; static const struct snd_pci_quirk alc260_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x1025, 0x007f, "Acer Aspire 9500", ALC260_FIXUP_COEF), + SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF), {} -- cgit v1.2.3-70-g09d2 From 20f7d928fa6e51ca81648946ead6244c58a0b4c0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 12:35:16 +0100 Subject: ALSA: hda/realtek - Replace ALC260 model=replacer with the auto-parser The support for Replacer 627V in the auto-parser needs the unique unsol event handling: although the machine has a single output pin 0x0f, it's used for both the headphone and the speaker, and the driver needs to toggle the output route via GPIO 1. In addition, it needs a special COEF setup with 0x3050. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - sound/pci/hda/alc260_quirks.c | 76 ---------------------------- sound/pci/hda/patch_realtek.c | 44 ++++++++++++++++ 3 files changed, 44 insertions(+), 77 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 5cc76090f5d..870cb1a2247 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -30,7 +30,6 @@ ALC880 ALC260 ====== fujitsu Fujitsu S7020 - replacer Replacer 672V favorit100 Maxdata Favorit 100XS basic fixed pin assignment (old default model) test for testing/debugging purpose, almost all controls can diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c index 2f1594b3d4b..55da43dddf3 100644 --- a/sound/pci/hda/alc260_quirks.c +++ b/sound/pci/hda/alc260_quirks.c @@ -8,7 +8,6 @@ enum { ALC260_AUTO, ALC260_BASIC, ALC260_FUJITSU_S702X, - ALC260_REPLACER_672V, ALC260_FAVORIT100, #ifdef CONFIG_SND_DEBUG ALC260_TEST, @@ -192,23 +191,6 @@ static const struct snd_kcontrol_new alc260_favorit100_mixer[] = { { } /* end */ }; -/* Replacer 672V ALC260 pin usage: Mic jack = 0x12, - * Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f. - */ -static const struct snd_kcontrol_new alc260_replacer_672v_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), - ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), - HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x07, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("ATATI Mic Playback Switch", 0x07, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), - ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), - { } /* end */ -}; - /* * initialization verbs */ @@ -441,48 +423,6 @@ static const struct hda_verb alc260_favorit100_init_verbs[] = { { } }; -static const struct hda_verb alc260_replacer_672v_verbs[] = { - {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, - {0x1a, AC_VERB_SET_COEF_INDEX, 0x07}, - {0x1a, AC_VERB_SET_PROC_COEF, 0x3050}, - - {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, - - {0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -/* toggle speaker-output according to the hp-jack state */ -static void alc260_replacer_672v_automute(struct hda_codec *codec) -{ - unsigned int present; - - /* speaker --> GPIO Data 0, hp or spdif --> GPIO data 1 */ - present = snd_hda_jack_detect(codec, 0x0f); - if (present) { - snd_hda_codec_write_cache(codec, 0x01, 0, - AC_VERB_SET_GPIO_DATA, 1); - snd_hda_codec_write_cache(codec, 0x0f, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_HP); - } else { - snd_hda_codec_write_cache(codec, 0x01, 0, - AC_VERB_SET_GPIO_DATA, 0); - snd_hda_codec_write_cache(codec, 0x0f, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_OUT); - } -} - -static void alc260_replacer_672v_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC_HP_EVENT) - alc260_replacer_672v_automute(codec); -} - static const struct hda_verb alc260_hp_dc7600_verbs[] = { {0x05, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -691,7 +631,6 @@ static const struct hda_verb alc260_test_init_verbs[] = { static const char * const alc260_models[ALC260_MODEL_LAST] = { [ALC260_BASIC] = "basic", [ALC260_FUJITSU_S702X] = "fujitsu", - [ALC260_REPLACER_672V] = "replacer", [ALC260_FAVORIT100] = "favorit100", #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = "test", @@ -706,7 +645,6 @@ static const struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC), SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X), SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC), - SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_REPLACER_672V), {} }; @@ -747,20 +685,6 @@ static const struct alc_config_preset alc260_presets[] = { .num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources), .input_mux = alc260_favorit100_capture_sources, }, - [ALC260_REPLACER_672V] = { - .mixers = { alc260_replacer_672v_mixer }, - .init_verbs = { alc260_init_verbs, alc260_replacer_672v_verbs }, - .num_dacs = ARRAY_SIZE(alc260_dac_nids), - .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_adc_nids), - .adc_nids = alc260_adc_nids, - .dig_out_nid = ALC260_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .input_mux = &alc260_capture_source, - .unsol_event = alc260_replacer_672v_unsol_event, - .init_hook = alc260_replacer_672v_automute, - }, #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = { .mixers = { alc260_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 95ef722e407..cfa6ad75834 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4211,8 +4211,35 @@ enum { ALC260_FIXUP_HP_PIN_0F, ALC260_FIXUP_COEF, ALC260_FIXUP_GPIO1, + ALC260_FIXUP_GPIO1_TOGGLE, + ALC260_FIXUP_REPLACER, }; +static void alc260_gpio1_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->hp_jack_present); +} + +static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == ALC_FIXUP_ACT_PROBE) { + /* although the machine has only one output pin, we need to + * toggle GPIO1 according to the jack state + */ + spec->automute_hook = alc260_gpio1_automute; + spec->detect_hp = 1; + spec->automute_speaker = 1; + spec->autocfg.hp_pins[0] = 0x0f; /* copy it for automute */ + snd_hda_jack_detect_enable(codec, 0x0f, ALC_HP_EVENT); + spec->unsol_event = alc_sku_unsol_event; + add_verb(codec->spec, alc_gpio1_init_verbs); + } +} + static const struct alc_fixup alc260_fixups[] = { [ALC260_FIXUP_HP_DC5750] = { .type = ALC_FIXUP_PINS, @@ -4242,6 +4269,22 @@ static const struct alc_fixup alc260_fixups[] = { .type = ALC_FIXUP_VERBS, .v.verbs = alc_gpio1_init_verbs, }, + [ALC260_FIXUP_GPIO1_TOGGLE] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc260_fixup_gpio1_toggle, + .chained = true, + .chain_id = ALC260_FIXUP_HP_PIN_0F, + }, + [ALC260_FIXUP_REPLACER] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x3050 }, + { } + }, + .chained = true, + .chain_id = ALC260_FIXUP_GPIO1_TOGGLE, + }, }; static const struct snd_pci_quirk alc260_fixup_tbl[] = { @@ -4249,6 +4292,7 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x007f, "Acer Aspire 9500", ALC260_FIXUP_COEF), SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), + SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_FIXUP_REPLACER), SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF), {} }; -- cgit v1.2.3-70-g09d2 From b1f58085a9c01e8ffab954fd77a45f1143edf34d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 12:45:03 +0100 Subject: ALSA: hda/realtek - Drop model=favorit100 for ALC260 It's working with the auto-parser just with the standard GPIO 1 setup. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - sound/pci/hda/alc260_quirks.c | 129 --------------------------- sound/pci/hda/patch_realtek.c | 1 + 3 files changed, 1 insertion(+), 130 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 870cb1a2247..e63d5e2ed47 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -30,7 +30,6 @@ ALC880 ALC260 ====== fujitsu Fujitsu S7020 - favorit100 Maxdata Favorit 100XS basic fixed pin assignment (old default model) test for testing/debugging purpose, almost all controls can adjusted. Appearing only when compiled with diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c index 55da43dddf3..94e7a270c5a 100644 --- a/sound/pci/hda/alc260_quirks.c +++ b/sound/pci/hda/alc260_quirks.c @@ -8,7 +8,6 @@ enum { ALC260_AUTO, ALC260_BASIC, ALC260_FUJITSU_S702X, - ALC260_FAVORIT100, #ifdef CONFIG_SND_DEBUG ALC260_TEST, #endif @@ -103,25 +102,6 @@ static const struct hda_input_mux alc260_acer_capture_sources[2] = { }, }; -/* Maxdata Favorit 100XS */ -static const struct hda_input_mux alc260_favorit100_capture_sources[2] = { - { - .num_items = 2, - .items = { - { "Line/Mic", 0x0 }, - { "CD", 0x4 }, - }, - }, - { - .num_items = 3, - .items = { - { "Line/Mic", 0x0 }, - { "CD", 0x4 }, - { "Mixer", 0x5 }, - }, - }, -}; - /* * This is just place-holder, so there's something for alc_build_pcms to look * at when it calculates the maximum number of channels. ALC260 has no mixer @@ -179,18 +159,6 @@ static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = { { } /* end */ }; -/* Maxdata Favorit 100XS: one output and one input (0x12) jack - */ -static const struct snd_kcontrol_new alc260_favorit100_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), - ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT), - HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT), - ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), - { } /* end */ -}; - /* * initialization verbs */ @@ -340,89 +308,6 @@ static const struct hda_verb alc260_fujitsu_init_verbs[] = { { } }; -/* Initialisation sequence for Maxdata Favorit 100XS - * (adapted from Acer init verbs). - */ -static const struct hda_verb alc260_favorit100_init_verbs[] = { - /* GPIO 0 enables the output jack. - * Turn this on and rely on the standard mute - * methods whenever the user wants to turn these outputs off. - */ - {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, - /* Line/Mic input jack is connected to Mic1 pin */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, - /* Ensure all other unused pins are disabled and muted. */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Disable digital (SPDIF) pins */ - {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, - {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum - * bus when acting as outputs. - */ - {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, - - /* Start with output sum widgets muted and their output gains at min */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* Unmute Line-out pin widget amp left and right - * (no equiv mixer ctrl) - */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute Mic1 and Line1 pin widget input buffers since they start as - * inputs. If the pin mode is changed by the user the pin mode control - * will take care of enabling the pin's input/output buffers as needed. - * Therefore there's no need to enable the input buffer at this - * stage. - */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute capture amp left and right */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Set ADC connection select to match default mixer setting - mic - * (on mic1 pin) - */ - {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Do similar with the second ADC: mute capture input amp and - * set ADC connection to mic to match ALSA's default state. - */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ - - { } -}; - static const struct hda_verb alc260_hp_dc7600_verbs[] = { {0x05, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -631,7 +516,6 @@ static const struct hda_verb alc260_test_init_verbs[] = { static const char * const alc260_models[ALC260_MODEL_LAST] = { [ALC260_BASIC] = "basic", [ALC260_FUJITSU_S702X] = "fujitsu", - [ALC260_FAVORIT100] = "favorit100", #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = "test", #endif @@ -639,7 +523,6 @@ static const char * const alc260_models[ALC260_MODEL_LAST] = { }; static const struct snd_pci_quirk alc260_cfg_tbl[] = { - SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100), SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC), SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC), SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC), @@ -673,18 +556,6 @@ static const struct alc_config_preset alc260_presets[] = { .num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources), .input_mux = alc260_fujitsu_capture_sources, }, - [ALC260_FAVORIT100] = { - .mixers = { alc260_favorit100_mixer }, - .init_verbs = { alc260_favorit100_init_verbs }, - .num_dacs = ARRAY_SIZE(alc260_dac_nids), - .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), - .adc_nids = alc260_dual_adc_nids, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources), - .input_mux = alc260_favorit100_capture_sources, - }, #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = { .mixers = { alc260_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index db1d8c888da..0d81eeb563c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4300,6 +4300,7 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900), + SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_FIXUP_REPLACER), SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF), {} -- cgit v1.2.3-70-g09d2 From c29b3f6dd7798964d77199af4925be72a3a48349 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 12:47:36 +0100 Subject: ALSA: hda/realtek - Drop model=fujitsu from ALC260 static quirks The model works with the auto-parser as is, thus now good to drop. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - sound/pci/hda/alc260_quirks.c | 142 --------------------------- 2 files changed, 143 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index e63d5e2ed47..53703392053 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -29,7 +29,6 @@ ALC880 ALC260 ====== - fujitsu Fujitsu S7020 basic fixed pin assignment (old default model) test for testing/debugging purpose, almost all controls can adjusted. Appearing only when compiled with diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c index 94e7a270c5a..305341f892c 100644 --- a/sound/pci/hda/alc260_quirks.c +++ b/sound/pci/hda/alc260_quirks.c @@ -7,7 +7,6 @@ enum { ALC260_AUTO, ALC260_BASIC, - ALC260_FUJITSU_S702X, #ifdef CONFIG_SND_DEBUG ALC260_TEST, #endif @@ -50,33 +49,6 @@ static const struct hda_input_mux alc260_capture_source = { }, }; -/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack, - * headphone jack and the internal CD lines since these are the only pins at - * which audio can appear. For flexibility, also allow the option of - * recording the mixer output on the second ADC (ADC0 doesn't have a - * connection to the mixer output). - */ -static const struct hda_input_mux alc260_fujitsu_capture_sources[2] = { - { - .num_items = 3, - .items = { - { "Mic/Line", 0x0 }, - { "CD", 0x4 }, - { "Headphone", 0x2 }, - }, - }, - { - .num_items = 4, - .items = { - { "Mic/Line", 0x0 }, - { "CD", 0x4 }, - { "Headphone", 0x2 }, - { "Mixer", 0x5 }, - }, - }, - -}; - /* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to * the Fujitsu S702x, but jacks are marked differently. */ @@ -142,23 +114,6 @@ static const struct snd_kcontrol_new alc260_input_mixer[] = { { } /* end */ }; -/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12, - * HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10. - */ -static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT), - ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT), - ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT), - { } /* end */ -}; - /* * initialization verbs */ @@ -225,89 +180,6 @@ static const struct hda_verb alc260_init_verbs[] = { { } }; -/* Initialisation sequence for ALC260 as configured in Fujitsu S702x - * laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD - * audio = 0x16, internal speaker = 0x10. - */ -static const struct hda_verb alc260_fujitsu_init_verbs[] = { - /* Disable all GPIOs */ - {0x01, AC_VERB_SET_GPIO_MASK, 0}, - /* Internal speaker is connected to headphone pin */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Headphone/Line-out jack connects to Line1 pin; make it an output */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Mic/Line-in jack is connected to mic1 pin, so make it an input */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Ensure all other unused pins are disabled and muted. */ - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - - /* Disable digital (SPDIF) pins */ - {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, - {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Ensure Line1 pin widget takes its input from the OUT1 sum bus - * when acting as an output. - */ - {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, - - /* Start with output sum widgets muted and their output gains at min */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute Line1 pin widget output buffer since it starts as an output. - * If the pin mode is changed by the user the pin mode control will - * take care of enabling the pin's input/output buffers as needed. - * Therefore there's no need to enable the input buffer at this - * stage. - */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute input buffer of pin widget used for Line-in (no equiv - * mixer ctrl) - */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute capture amp left and right */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Set ADC connection select to match default mixer setting - line - * in (on mic1 pin) - */ - {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Do the same for the second ADC: mute capture input amp and - * set ADC connection to line in (on mic1 pin) - */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ - - { } -}; - static const struct hda_verb alc260_hp_dc7600_verbs[] = { {0x05, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -515,7 +387,6 @@ static const struct hda_verb alc260_test_init_verbs[] = { */ static const char * const alc260_models[ALC260_MODEL_LAST] = { [ALC260_BASIC] = "basic", - [ALC260_FUJITSU_S702X] = "fujitsu", #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = "test", #endif @@ -526,7 +397,6 @@ static const struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC), SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC), SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC), - SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X), SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC), {} }; @@ -544,18 +414,6 @@ static const struct alc_config_preset alc260_presets[] = { .channel_mode = alc260_modes, .input_mux = &alc260_capture_source, }, - [ALC260_FUJITSU_S702X] = { - .mixers = { alc260_fujitsu_mixer }, - .init_verbs = { alc260_fujitsu_init_verbs }, - .num_dacs = ARRAY_SIZE(alc260_dac_nids), - .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), - .adc_nids = alc260_dual_adc_nids, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources), - .input_mux = alc260_fujitsu_capture_sources, - }, #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = { .mixers = { alc260_test_mixer }, -- cgit v1.2.3-70-g09d2 From c3c2c9e7ff3e38bd9ff5b721b6ae8634fce42802 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 12:59:55 +0100 Subject: ALSA: hda/realtek - Remove leftover static quirks for ALC260 Now we can clean up all static quirks for ALC260. Also many codes in alc_quirks.c can be ripped off since they have been used only by ALC260 static quirks. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 6 +- sound/pci/hda/alc260_quirks.c | 432 --------------------------- sound/pci/hda/alc_quirks.c | 301 ------------------- sound/pci/hda/patch_realtek.c | 48 +-- 4 files changed, 9 insertions(+), 778 deletions(-) delete mode 100644 sound/pci/hda/alc260_quirks.c (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 53703392053..2d34be30465 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -29,11 +29,7 @@ ALC880 ALC260 ====== - basic fixed pin assignment (old default model) - test for testing/debugging purpose, almost all controls can - adjusted. Appearing only when compiled with - $CONFIG_SND_DEBUG=y - auto auto-config reading BIOS (default) + N/A ALC262 ====== diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c deleted file mode 100644 index 305341f892c..00000000000 --- a/sound/pci/hda/alc260_quirks.c +++ /dev/null @@ -1,432 +0,0 @@ -/* - * ALC260 quirk models - * included by patch_realtek.c - */ - -/* ALC260 models */ -enum { - ALC260_AUTO, - ALC260_BASIC, -#ifdef CONFIG_SND_DEBUG - ALC260_TEST, -#endif - ALC260_MODEL_LAST /* last tag */ -}; - -static const hda_nid_t alc260_dac_nids[1] = { - /* front */ - 0x02, -}; - -static const hda_nid_t alc260_adc_nids[1] = { - /* ADC0 */ - 0x04, -}; - -static const hda_nid_t alc260_adc_nids_alt[1] = { - /* ADC1 */ - 0x05, -}; - -/* NIDs used when simultaneous access to both ADCs makes sense. Note that - * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC. - */ -static const hda_nid_t alc260_dual_adc_nids[2] = { - /* ADC0, ADC1 */ - 0x04, 0x05 -}; - -#define ALC260_DIGOUT_NID 0x03 -#define ALC260_DIGIN_NID 0x06 - -static const struct hda_input_mux alc260_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to - * the Fujitsu S702x, but jacks are marked differently. - */ -static const struct hda_input_mux alc260_acer_capture_sources[2] = { - { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - { "Headphone", 0x5 }, - }, - }, - { - .num_items = 5, - .items = { - { "Mic", 0x0 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - { "Headphone", 0x6 }, - { "Mixer", 0x5 }, - }, - }, -}; - -/* - * This is just place-holder, so there's something for alc_build_pcms to look - * at when it calculates the maximum number of channels. ALC260 has no mixer - * element which allows changing the channel mode, so the verb list is - * never used. - */ -static const struct hda_channel_mode alc260_modes[1] = { - { 2, NULL }, -}; - - -/* Mixer combinations - * - * basic: base_output + input + pc_beep + capture - * fujitsu: fujitsu + capture - * acer: acer + capture - */ - -static const struct snd_kcontrol_new alc260_base_output_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc260_input_mixer[] = { - HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT), - { } /* end */ -}; - -/* - * initialization verbs - */ -static const struct hda_verb alc260_init_verbs[] = { - /* Line In pin widget for input */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* CD pin widget for input */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Mic1 (rear panel) pin widget for input and vref at 80% */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - /* Mic2 (front panel) pin widget for input and vref at 80% */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - /* LINE-2 is used for line-out in rear */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* select line-out */ - {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* LINE-OUT pin */ - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* enable HP */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* enable Mono */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* mute capture amp left and right */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* set connection select to line in (default select for this ADC) */ - {0x04, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* mute capture amp left and right */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* set connection select to line in (default select for this ADC) */ - {0x05, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* set vol=0 Line-Out mixer amp left and right */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* unmute pin widget amp left and right (no gain on this amp) */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* set vol=0 HP mixer amp left and right */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* unmute pin widget amp left and right (no gain on this amp) */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* set vol=0 Mono mixer amp left and right */ - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* unmute pin widget amp left and right (no gain on this amp) */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* unmute LINE-2 out pin */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & - * Line In 2 = 0x03 - */ - /* mute analog inputs */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */ - /* mute Front out path */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* mute Headphone out path */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* mute Mono out path */ - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - { } -}; - -static const struct hda_verb alc260_hp_dc7600_verbs[] = { - {0x05, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -/* Test configuration for debugging, modelled after the ALC880 test - * configuration. - */ -#ifdef CONFIG_SND_DEBUG -static const hda_nid_t alc260_test_dac_nids[1] = { - 0x02, -}; -static const hda_nid_t alc260_test_adc_nids[2] = { - 0x04, 0x05, -}; -/* For testing the ALC260, each input MUX needs its own definition since - * the signal assignments are different. This assumes that the first ADC - * is NID 0x04. - */ -static const struct hda_input_mux alc260_test_capture_sources[2] = { - { - .num_items = 7, - .items = { - { "MIC1 pin", 0x0 }, - { "MIC2 pin", 0x1 }, - { "LINE1 pin", 0x2 }, - { "LINE2 pin", 0x3 }, - { "CD pin", 0x4 }, - { "LINE-OUT pin", 0x5 }, - { "HP-OUT pin", 0x6 }, - }, - }, - { - .num_items = 8, - .items = { - { "MIC1 pin", 0x0 }, - { "MIC2 pin", 0x1 }, - { "LINE1 pin", 0x2 }, - { "LINE2 pin", 0x3 }, - { "CD pin", 0x4 }, - { "Mixer", 0x5 }, - { "LINE-OUT pin", 0x6 }, - { "HP-OUT pin", 0x7 }, - }, - }, -}; -static const struct snd_kcontrol_new alc260_test_mixer[] = { - /* Output driver widgets */ - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT), - HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT), - HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT), - - /* Modes for retasking pin widgets - * Note: the ALC260 doesn't seem to act on requests to enable mic - * bias from NIDs 0x0f and 0x10. The ALC260 datasheet doesn't - * mention this restriction. At this stage it's not clear whether - * this behaviour is intentional or is a hardware bug in chip - * revisions available at least up until early 2006. Therefore for - * now allow the "HP-OUT" and "LINE-OUT" Mode controls to span all - * choices, but if it turns out that the lack of mic bias for these - * NIDs is intentional we could change their modes from - * ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS. - */ - ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT), - ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT), - ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT), - ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT), - ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT), - ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT), - - /* Loopback mixer controls */ - HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT), - HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT), - HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT), - - /* Controls for GPIO pins, assuming they are configured as outputs */ - ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01), - ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02), - ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04), - ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08), - - /* Switches to allow the digital IO pins to be enabled. The datasheet - * is ambigious as to which NID is which; testing on laptops which - * make this output available should provide clarification. - */ - ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01), - ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01), - - /* A switch allowing EAPD to be enabled. Some laptops seem to use - * this output to turn on an external amplifier. - */ - ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02), - ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02), - - { } /* end */ -}; -static const struct hda_verb alc260_test_init_verbs[] = { - /* Enable all GPIOs as outputs with an initial value of 0 */ - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, - {0x01, AC_VERB_SET_GPIO_MASK, 0x0f}, - - /* Enable retasking pins as output, initially without power amp */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* Disable digital (SPDIF) pins initially, but users can enable - * them via a mixer switch. In the case of SPDIF-out, this initverb - * payload also sets the generation to 0, output to be in "consumer" - * PCM format, copyright asserted, no pre-emphasis and no validity - * control. - */ - {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, - {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Ensure mic1, mic2, line1 and line2 pin widgets take input from the - * OUT1 sum bus when acting as an output. - */ - {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, - {0x0c, AC_VERB_SET_CONNECT_SEL, 0}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, - {0x0e, AC_VERB_SET_CONNECT_SEL, 0}, - - /* Start with output sum widgets muted and their output gains at min */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* Unmute retasking pin widget output buffers since the default - * state appears to be output. As the pin mode is changed by the - * user the pin mode control will take care of enabling the pin's - * input/output buffers as needed. - */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Also unmute the mono-out pin widget */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Mute capture amp left and right */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Set ADC connection select to match default mixer setting (mic1 - * pin) - */ - {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Do the same for the second ADC: mute capture input amp and - * set ADC connection to mic1 pin - */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ - - { } -}; -#endif - -/* - * ALC260 configurations - */ -static const char * const alc260_models[ALC260_MODEL_LAST] = { - [ALC260_BASIC] = "basic", -#ifdef CONFIG_SND_DEBUG - [ALC260_TEST] = "test", -#endif - [ALC260_AUTO] = "auto", -}; - -static const struct snd_pci_quirk alc260_cfg_tbl[] = { - SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC), - SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC), - SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC), - SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC), - {} -}; - -static const struct alc_config_preset alc260_presets[] = { - [ALC260_BASIC] = { - .mixers = { alc260_base_output_mixer, - alc260_input_mixer }, - .init_verbs = { alc260_init_verbs }, - .num_dacs = ARRAY_SIZE(alc260_dac_nids), - .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), - .adc_nids = alc260_dual_adc_nids, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .input_mux = &alc260_capture_source, - }, -#ifdef CONFIG_SND_DEBUG - [ALC260_TEST] = { - .mixers = { alc260_test_mixer }, - .init_verbs = { alc260_test_init_verbs }, - .num_dacs = ARRAY_SIZE(alc260_test_dac_nids), - .dac_nids = alc260_test_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids), - .adc_nids = alc260_test_adc_nids, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .num_mux_defs = ARRAY_SIZE(alc260_test_capture_sources), - .input_mux = alc260_test_capture_sources, - }, -#endif -}; - diff --git a/sound/pci/hda/alc_quirks.c b/sound/pci/hda/alc_quirks.c index a18952ed431..b344603ac06 100644 --- a/sound/pci/hda/alc_quirks.c +++ b/sound/pci/hda/alc_quirks.c @@ -74,307 +74,6 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, return err; } -/* - * Control the mode of pin widget settings via the mixer. "pc" is used - * instead of "%" to avoid consequences of accidentally treating the % as - * being part of a format specifier. Maximum allowed length of a value is - * 63 characters plus NULL terminator. - * - * Note: some retasking pin complexes seem to ignore requests for input - * states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these - * are requested. Therefore order this list so that this behaviour will not - * cause problems when mixer clients move through the enum sequentially. - * NIDs 0x0f and 0x10 have been observed to have this behaviour as of - * March 2006. - */ -static const char * const alc_pin_mode_names[] = { - "Mic 50pc bias", "Mic 80pc bias", - "Line in", "Line out", "Headphone out", -}; -static const unsigned char alc_pin_mode_values[] = { - PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP, -}; -/* The control can present all 5 options, or it can limit the options based - * in the pin being assumed to be exclusively an input or an output pin. In - * addition, "input" pins may or may not process the mic bias option - * depending on actual widget capability (NIDs 0x0f and 0x10 don't seem to - * accept requests for bias as of chip versions up to March 2006) and/or - * wiring in the computer. - */ -#define ALC_PIN_DIR_IN 0x00 -#define ALC_PIN_DIR_OUT 0x01 -#define ALC_PIN_DIR_INOUT 0x02 -#define ALC_PIN_DIR_IN_NOMICBIAS 0x03 -#define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04 - -/* Info about the pin modes supported by the different pin direction modes. - * For each direction the minimum and maximum values are given. - */ -static const signed char alc_pin_mode_dir_info[5][2] = { - { 0, 2 }, /* ALC_PIN_DIR_IN */ - { 3, 4 }, /* ALC_PIN_DIR_OUT */ - { 0, 4 }, /* ALC_PIN_DIR_INOUT */ - { 2, 2 }, /* ALC_PIN_DIR_IN_NOMICBIAS */ - { 2, 4 }, /* ALC_PIN_DIR_INOUT_NOMICBIAS */ -}; -#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0]) -#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1]) -#define alc_pin_mode_n_items(_dir) \ - (alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1) - -static int alc_pin_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - unsigned int item_num = uinfo->value.enumerated.item; - unsigned char dir = (kcontrol->private_value >> 16) & 0xff; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = alc_pin_mode_n_items(dir); - - if (item_numalc_pin_mode_max(dir)) - item_num = alc_pin_mode_min(dir); - strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]); - return 0; -} - -static int alc_pin_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - unsigned int i; - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char dir = (kcontrol->private_value >> 16) & 0xff; - long *valp = ucontrol->value.integer.value; - unsigned int pinctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, - 0x00); - - /* Find enumerated value for current pinctl setting */ - i = alc_pin_mode_min(dir); - while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl) - i++; - *valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir); - return 0; -} - -static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - signed int change; - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char dir = (kcontrol->private_value >> 16) & 0xff; - long val = *ucontrol->value.integer.value; - unsigned int pinctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, - 0x00); - - if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir)) - val = alc_pin_mode_min(dir); - - change = pinctl != alc_pin_mode_values[val]; - if (change) { - /* Set pin mode to that requested */ - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - alc_pin_mode_values[val]); - - /* Also enable the retasking pin's input/output as required - * for the requested pin mode. Enum values of 2 or less are - * input modes. - * - * Dynamically switching the input/output buffers probably - * reduces noise slightly (particularly on input) so we'll - * do it. However, having both input and output buffers - * enabled simultaneously doesn't seem to be problematic if - * this turns out to be necessary in the future. - */ - if (val <= 2) { - snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0, - HDA_AMP_MUTE, 0); - } else { - snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, 0); - } - } - return change; -} - -#define ALC_PIN_MODE(xname, nid, dir) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ - .info = alc_pin_mode_info, \ - .get = alc_pin_mode_get, \ - .put = alc_pin_mode_put, \ - .private_value = nid | (dir<<16) } - -/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged - * together using a mask with more than one bit set. This control is - * currently used only by the ALC260 test model. At this stage they are not - * needed for any "production" models. - */ -#ifdef CONFIG_SND_DEBUG -#define alc_gpio_data_info snd_ctl_boolean_mono_info - -static int alc_gpio_data_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char mask = (kcontrol->private_value >> 16) & 0xff; - long *valp = ucontrol->value.integer.value; - unsigned int val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_GPIO_DATA, 0x00); - - *valp = (val & mask) != 0; - return 0; -} -static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - signed int change; - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char mask = (kcontrol->private_value >> 16) & 0xff; - long val = *ucontrol->value.integer.value; - unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_GPIO_DATA, - 0x00); - - /* Set/unset the masked GPIO bit(s) as needed */ - change = (val == 0 ? 0 : mask) != (gpio_data & mask); - if (val == 0) - gpio_data &= ~mask; - else - gpio_data |= mask; - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_GPIO_DATA, gpio_data); - - return change; -} -#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ - .info = alc_gpio_data_info, \ - .get = alc_gpio_data_get, \ - .put = alc_gpio_data_put, \ - .private_value = nid | (mask<<16) } -#endif /* CONFIG_SND_DEBUG */ - -/* A switch control to allow the enabling of the digital IO pins on the - * ALC260. This is incredibly simplistic; the intention of this control is - * to provide something in the test model allowing digital outputs to be - * identified if present. If models are found which can utilise these - * outputs a more complete mixer control can be devised for those models if - * necessary. - */ -#ifdef CONFIG_SND_DEBUG -#define alc_spdif_ctrl_info snd_ctl_boolean_mono_info - -static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char mask = (kcontrol->private_value >> 16) & 0xff; - long *valp = ucontrol->value.integer.value; - unsigned int val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT_1, 0x00); - - *valp = (val & mask) != 0; - return 0; -} -static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - signed int change; - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char mask = (kcontrol->private_value >> 16) & 0xff; - long val = *ucontrol->value.integer.value; - unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT_1, - 0x00); - - /* Set/unset the masked control bit(s) as needed */ - change = (val == 0 ? 0 : mask) != (ctrl_data & mask); - if (val==0) - ctrl_data &= ~mask; - else - ctrl_data |= mask; - snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, - ctrl_data); - - return change; -} -#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ - .info = alc_spdif_ctrl_info, \ - .get = alc_spdif_ctrl_get, \ - .put = alc_spdif_ctrl_put, \ - .private_value = nid | (mask<<16) } -#endif /* CONFIG_SND_DEBUG */ - -/* A switch control to allow the enabling EAPD digital outputs on the ALC26x. - * Again, this is only used in the ALC26x test models to help identify when - * the EAPD line must be asserted for features to work. - */ -#ifdef CONFIG_SND_DEBUG -#define alc_eapd_ctrl_info snd_ctl_boolean_mono_info - -static int alc_eapd_ctrl_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char mask = (kcontrol->private_value >> 16) & 0xff; - long *valp = ucontrol->value.integer.value; - unsigned int val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_EAPD_BTLENABLE, 0x00); - - *valp = (val & mask) != 0; - return 0; -} - -static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int change; - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char mask = (kcontrol->private_value >> 16) & 0xff; - long val = *ucontrol->value.integer.value; - unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_EAPD_BTLENABLE, - 0x00); - - /* Set/unset the masked control bit(s) as needed */ - change = (!val ? 0 : mask) != (ctrl_data & mask); - if (!val) - ctrl_data &= ~mask; - else - ctrl_data |= mask; - snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, - ctrl_data); - - return change; -} - -#define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ - .info = alc_eapd_ctrl_info, \ - .get = alc_eapd_ctrl_get, \ - .put = alc_eapd_ctrl_put, \ - .private_value = nid | (mask<<16) } -#endif /* CONFIG_SND_DEBUG */ - static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0d81eeb563c..3ea42069b8e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4308,14 +4308,10 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { /* */ -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS -#include "alc260_quirks.c" -#endif - static int patch_alc260(struct hda_codec *codec) { struct alc_spec *spec; - int err, board_config; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4325,38 +4321,13 @@ static int patch_alc260(struct hda_codec *codec) spec->mixer_nid = 0x07; - board_config = alc_board_config(codec, ALC260_MODEL_LAST, - alc260_models, alc260_cfg_tbl); - if (board_config < 0) { - snd_printd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = ALC_MODEL_AUTO; - } - - if (board_config == ALC_MODEL_AUTO) { - alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - } - - if (board_config == ALC_MODEL_AUTO) { - /* automatic parse from the BIOS config */ - err = alc260_parse_auto_config(codec); - if (err < 0) - goto error; -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS - else if (!err) { - printk(KERN_INFO - "hda_codec: Cannot set up configuration " - "from BIOS. Using base mode...\n"); - board_config = ALC260_BASIC; - } -#endif - } + alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - if (board_config != ALC_MODEL_AUTO) { - setup_preset(codec, &alc260_presets[board_config]); - spec->vmaster_nid = 0x08; - } + /* automatic parse from the BIOS config */ + err = alc260_parse_auto_config(codec); + if (err < 0) + goto error; if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4377,10 +4348,7 @@ static int patch_alc260(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; - if (board_config == ALC_MODEL_AUTO) - spec->init_hook = alc_auto_init_std; - else - codec->patch_ops.build_controls = __alc_build_controls; + spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) -- cgit v1.2.3-70-g09d2 From dc6af52dea5ada1269095cad5ed2c04e92114399 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Feb 2012 16:18:59 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=lg with the auto-parser ALC880 model=lg could work fine with the auto-parser due to the recent rewrite, but it still needs the manual adjustment; namely, the BIOS leaves unused pins as some real active jacks. This confuses the parser. Thus we just cover these pins and override the pin-configs as a fix-up. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - sound/pci/hda/alc880_quirks.c | 158 --------------------------- sound/pci/hda/patch_realtek.c | 14 +++ 3 files changed, 14 insertions(+), 159 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 2d34be30465..6387b41f35a 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -17,7 +17,6 @@ ALC880 uniwill 3-jack fujitsu Fujitsu Laptops (Pi1536) F1734 2-jack - lg LG laptop (m1 express dual) lg-lw LG LW20/LW25 laptop tcl TCL S700 clevo Clevo laptops (m520G, m665n) diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 501501ef36a..3b88bc561e1 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -25,7 +25,6 @@ enum { ALC880_UNIWILL_P53, ALC880_CLEVO, ALC880_TCL_S700, - ALC880_LG, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -773,11 +772,6 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec, } } -static void alc880_unsol_event(struct hda_codec *codec, unsigned int res) -{ - alc_exec_unsol_event(codec, res >> 28); -} - static void alc880_uniwill_p53_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -936,136 +930,6 @@ static const struct hda_verb alc880_pin_tcl_S700_init_verbs[] = { { } }; -/* - * LG m1 express dual - * - * Pin assignment: - * Rear Line-In/Out (blue): 0x14 - * Build-in Mic-In: 0x15 - * Speaker-out: 0x17 - * HP-Out (green): 0x1b - * Mic-In/Out (red): 0x19 - * SPDIF-Out: 0x1e - */ - -/* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */ -static const hda_nid_t alc880_lg_dac_nids[3] = { - 0x05, 0x02, 0x03 -}; - -/* seems analog CD is not working */ -static const struct hda_input_mux alc880_lg_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x1 }, - { "Line", 0x5 }, - { "Internal Mic", 0x6 }, - }, -}; - -/* 2,4,6 channel modes */ -static const struct hda_verb alc880_lg_ch2_init[] = { - /* set line-in and mic-in to input */ - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { } -}; - -static const struct hda_verb alc880_lg_ch4_init[] = { - /* set line-in to out and mic-in to input */ - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { } -}; - -static const struct hda_verb alc880_lg_ch6_init[] = { - /* set line-in and mic-in to output */ - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - { } -}; - -static const struct hda_channel_mode alc880_lg_ch_modes[3] = { - { 2, alc880_lg_ch2_init }, - { 4, alc880_lg_ch4_init }, - { 6, alc880_lg_ch6_init }, -}; - -static const struct snd_kcontrol_new alc880_lg_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x06, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x07, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x07, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - -static const struct hda_verb alc880_lg_init_verbs[] = { - /* set capture source to mic-in */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* mute all amp mixer inputs */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, - /* line-in to input */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* built-in mic */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* speaker-out */ - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* mic-in to input */ - {0x11, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* HP-out */ - {0x13, AC_VERB_SET_CONNECT_SEL, 0x03}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* jack sense */ - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - { } -}; - -/* toggle speaker-output according to the hp-jack state */ -static void alc880_lg_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.speaker_pins[0] = 0x17; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -#ifdef CONFIG_SND_HDA_POWER_SAVE -static const struct hda_amp_list alc880_lg_loopbacks[] = { - { 0x0b, HDA_INPUT, 1 }, - { 0x0b, HDA_INPUT, 6 }, - { 0x0b, HDA_INPUT, 7 }, - { } /* end */ -}; -#endif - /* * Test configuration for debugging * @@ -1352,7 +1216,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_UNIWILL_P53] = "uniwill-p53", [ALC880_FUJITSU] = "fujitsu", [ALC880_F1734] = "F1734", - [ALC880_LG] = "lg", #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = "test", #endif @@ -1409,9 +1272,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734), SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), - SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG), - SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_LG), - SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_LG), SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700), SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG), @@ -1673,24 +1533,6 @@ static const struct alc_config_preset alc880_presets[] = { .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, - [ALC880_LG] = { - .mixers = { alc880_lg_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_lg_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_lg_dac_nids), - .dac_nids = alc880_lg_dac_nids, - .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_unsol_event, - .setup = alc880_lg_setup, - .init_hook = alc_hp_automute, -#ifdef CONFIG_SND_HDA_POWER_SAVE - .loopbacks = alc880_lg_loopbacks, -#endif - }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d0c71d5be83..a39146528c2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4404,6 +4404,7 @@ static const struct hda_amp_list alc880_loopbacks[] = { enum { ALC880_FIXUP_GPIO2, ALC880_FIXUP_MEDION_RIM, + ALC880_FIXUP_LG, }; static const struct alc_fixup alc880_fixups[] = { @@ -4421,10 +4422,23 @@ static const struct alc_fixup alc880_fixups[] = { .chained = true, .chain_id = ALC880_FIXUP_GPIO2, }, + [ALC880_FIXUP_LG] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + /* disable bogus unused pins */ + { 0x16, 0x411111f0 }, + { 0x18, 0x411111f0 }, + { 0x1a, 0x411111f0 }, + { } + } + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM), + SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), + SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), + SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG), {} }; -- cgit v1.2.3-70-g09d2 From f02aab5d7fd53da95a78bb27bfbacc972ed75c10 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Feb 2012 16:33:56 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=w810 with auto-parser The Medion W810 with ALC880 has a typical BIOS bug, copying the pin-defaults without disabling the unused pins. At least, the pin 0x17 must be disabled. Also, it requires GPIO-2 setup. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - sound/pci/hda/alc880_quirks.c | 84 +--------------------------- sound/pci/hda/patch_realtek.c | 13 +++++ 3 files changed, 14 insertions(+), 84 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 6387b41f35a..24df3ab4193 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -8,7 +8,6 @@ ALC880 5stack-digout 5-jack in back, 2-jack in front, a SPDIF out 6stack 6-jack in back, 2-jack in front 6stack-digout 6-jack with a SPDIF out - w810 3-jack z71v 3-jack (HP shared SPDIF) asus 3-jack (ASUS Mobo) asus-w1v ASUS W1V diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 3b88bc561e1..41aecda30a8 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -10,7 +10,6 @@ enum { ALC880_3ST_DIG, ALC880_5ST, ALC880_5ST_DIG, - ALC880_W810, ALC880_Z71V, ALC880_6ST, ALC880_6ST_DIG, @@ -226,56 +225,11 @@ static const struct snd_kcontrol_new alc880_six_stack_mixer[] = { }; -/* - * ALC880 W810 model - * - * W810 has rear IO for: - * Front (DAC 02) - * Surround (DAC 03) - * Center/LFE (DAC 04) - * Digital out (06) - * - * The system also has a pair of internal speakers, and a headphone jack. - * These are both connected to Line2 on the codec, hence to DAC 02. - * - * There is a variable resistor to control the speaker or headphone - * volume. This is a hardware-only device without a software API. - * - * Plugging headphones in will disable the internal speakers. This is - * implemented in hardware, not via the driver using jack sense. In - * a similar fashion, plugging into the rear socket marked "front" will - * disable both the speakers and headphones. - * - * For input, there's a microphone jack, and an "audio in" jack. - * These may not do anything useful with this driver yet, because I - * haven't setup any initialization verbs for these yet... - */ - static const hda_nid_t alc880_w810_dac_nids[3] = { /* front, rear/surround, clfe */ 0x02, 0x03, 0x04 }; -/* fixed 6 channels */ -static const struct hda_channel_mode alc880_w810_modes[1] = { - { 6, NULL } -}; - -/* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */ -static const struct snd_kcontrol_new alc880_w810_base_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - { } /* end */ -}; - - /* * Z710V model * @@ -593,27 +547,6 @@ static const struct hda_verb alc880_pin_5stack_init_verbs[] = { { } }; -/* - * W810 pin configuration: - * front = 0x14, surround = 0x15, clfe = 0x16, HP = 0x1b - */ -static const struct hda_verb alc880_pin_w810_init_verbs[] = { - /* hphone/speaker input selector: front DAC */ - {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - { } -}; - /* * Z71V pin configuration: * Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?) @@ -1204,7 +1137,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_CLEVO] = "clevo", [ALC880_5ST] = "5stack", [ALC880_5ST_DIG] = "5stack-digout", - [ALC880_W810] = "w810", [ALC880_Z71V] = "z71v", [ALC880_6ST] = "6stack", [ALC880_6ST_DIG] = "6stack-digout", @@ -1223,7 +1155,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { }; static const struct snd_pci_quirk alc880_cfg_tbl[] = { - SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_W810), SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_6ST), SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_3ST_DIG), @@ -1265,7 +1196,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_F1734), SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL), SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53), - SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810), SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), @@ -1377,18 +1307,6 @@ static const struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_sixstack_modes, .input_mux = &alc880_6stack_capture_source, }, - [ALC880_W810] = { - .mixers = { alc880_w810_base_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_w810_init_verbs, - alc880_gpio2_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_w810_dac_nids), - .dac_nids = alc880_w810_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_w810_modes), - .channel_mode = alc880_w810_modes, - .input_mux = &alc880_capture_source, - }, [ALC880_Z71V] = { .mixers = { alc880_z71v_mixer }, .init_verbs = { alc880_volume_init_verbs, @@ -1499,7 +1417,7 @@ static const struct alc_config_preset alc880_presets[] = { alc880_uniwill_p53_init_verbs }, .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), .dac_nids = alc880_asus_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc880_w810_modes), + .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), .channel_mode = alc880_threestack_modes, .input_mux = &alc880_capture_source, .unsol_event = alc880_uniwill_p53_unsol_event, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a39146528c2..1cad6748e33 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4405,6 +4405,7 @@ enum { ALC880_FIXUP_GPIO2, ALC880_FIXUP_MEDION_RIM, ALC880_FIXUP_LG, + ALC880_FIXUP_W810, }; static const struct alc_fixup alc880_fixups[] = { @@ -4432,9 +4433,21 @@ static const struct alc_fixup alc880_fixups[] = { { } } }, + [ALC880_FIXUP_W810] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + /* disable bogus unused pins */ + { 0x17, 0x411111f0 }, + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_GPIO2, + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { + SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), + SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_FIXUP_W810), SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM), SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), -- cgit v1.2.3-70-g09d2 From 27e917f82bfcf8c51a2c025ddfb69e0b5947f50b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Feb 2012 17:49:54 +0100 Subject: ALSA: hda/realtek - Drop ALC880 model=clevo Clevo machines with ALC880 are all well with proper BIOS setup. It seems still requiring the additional COEF setup for the EAPD. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - sound/pci/hda/alc880_quirks.c | 45 ---------------------------- sound/pci/hda/patch_realtek.c | 11 +++++++ 3 files changed, 11 insertions(+), 46 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 24df3ab4193..58e8aac40a9 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -18,7 +18,6 @@ ALC880 F1734 2-jack lg-lw LG LW20/LW25 laptop tcl TCL S700 - clevo Clevo laptops (m520G, m665n) medion Medion Rim 2150 test for testing/debugging purpose, almost all controls can be adjusted. Appearing only when compiled with diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 41aecda30a8..b64d2464a78 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -22,7 +22,6 @@ enum { ALC880_UNIWILL_DIG, ALC880_UNIWILL, ALC880_UNIWILL_P53, - ALC880_CLEVO, ALC880_TCL_S700, #ifdef CONFIG_SND_DEBUG ALC880_TEST, @@ -809,35 +808,6 @@ static const struct hda_verb alc880_pin_asus_init_verbs[] = { #define alc880_gpio2_init_verbs alc_gpio2_init_verbs #define alc880_gpio3_init_verbs alc_gpio3_init_verbs -/* Clevo m520g init */ -static const struct hda_verb alc880_pin_clevo_init_verbs[] = { - /* headphone output */ - {0x11, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* line-out */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Line-in */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* CD */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Mic1 (rear panel) */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Mic2 (front panel) */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* headphone */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* change to EAPD mode */ - {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, - {0x20, AC_VERB_SET_PROC_COEF, 0x3060}, - - { } -}; - static const struct hda_verb alc880_pin_tcl_S700_init_verbs[] = { /* change to EAPD mode */ {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -1134,7 +1104,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_3ST] = "3stack", [ALC880_TCL_S700] = "tcl", [ALC880_3ST_DIG] = "3stack-digout", - [ALC880_CLEVO] = "clevo", [ALC880_5ST] = "5stack", [ALC880_5ST_DIG] = "5stack-digout", [ALC880_Z71V] = "z71v", @@ -1188,8 +1157,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1558, 0x0520, "Clevo m520G", ALC880_CLEVO), - SND_PCI_QUIRK(0x1558, 0x0660, "Clevo m655n", ALC880_CLEVO), SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2), SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG), @@ -1439,18 +1406,6 @@ static const struct alc_config_preset alc880_presets[] = { .setup = alc880_uniwill_p53_setup, .init_hook = alc_hp_automute, }, - [ALC880_CLEVO] = { - .mixers = { alc880_three_stack_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_clevo_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_dac_nids), - .dac_nids = alc880_dac_nids, - .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, - }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a0df05d0386..4f8c3620799 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4406,6 +4406,7 @@ enum { ALC880_FIXUP_MEDION_RIM, ALC880_FIXUP_LG, ALC880_FIXUP_W810, + ALC880_FIXUP_EAPD_COEF, }; static const struct alc_fixup alc880_fixups[] = { @@ -4443,10 +4444,20 @@ static const struct alc_fixup alc880_fixups[] = { .chained = true, .chain_id = ALC880_FIXUP_GPIO2, }, + [ALC880_FIXUP_EAPD_COEF] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* change to EAPD mode */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x3060 }, + {} + }, + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), + SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_FIXUP_W810), SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM), SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), -- cgit v1.2.3-70-g09d2 From b9368f5c10b15f2b79a58666849827edc1f2f3d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Feb 2012 17:54:44 +0100 Subject: ALSA: hda/realtek - Replace ALC880 model=tcl with auto-parser It needs a few extra setups for EAPD, but others look fairly straightforward. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 3 -- sound/pci/hda/alc880_quirks.c | 56 ---------------------------- sound/pci/hda/patch_realtek.c | 13 +++++++ 3 files changed, 13 insertions(+), 59 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 58e8aac40a9..fd3b3b25bd7 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -16,9 +16,6 @@ ALC880 uniwill 3-jack fujitsu Fujitsu Laptops (Pi1536) F1734 2-jack - lg-lw LG LW20/LW25 laptop - tcl TCL S700 - medion Medion Rim 2150 test for testing/debugging purpose, almost all controls can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index b64d2464a78..56f8fa1e346 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -22,7 +22,6 @@ enum { ALC880_UNIWILL_DIG, ALC880_UNIWILL, ALC880_UNIWILL_P53, - ALC880_TCL_S700, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -344,20 +343,6 @@ static const struct snd_kcontrol_new alc880_asus_w1v_mixer[] = { { } /* end */ }; -/* TCL S700 */ -static const struct snd_kcontrol_new alc880_tcl_s700_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 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", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - { } /* end */ -}; - /* Uniwill */ static const struct snd_kcontrol_new alc880_uniwill_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -808,31 +793,6 @@ static const struct hda_verb alc880_pin_asus_init_verbs[] = { #define alc880_gpio2_init_verbs alc_gpio2_init_verbs #define alc880_gpio3_init_verbs alc_gpio3_init_verbs -static const 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*/ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Line In pin widget for input */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* CD pin widget for input */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Mic1 (rear panel) pin widget for input and vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - - /* change to EAPD mode */ - {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, - {0x20, AC_VERB_SET_PROC_COEF, 0x3070}, - - { } -}; - /* * Test configuration for debugging * @@ -1102,7 +1062,6 @@ static const struct hda_verb alc880_test_init_verbs[] = { static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_3ST] = "3stack", - [ALC880_TCL_S700] = "tcl", [ALC880_3ST_DIG] = "3stack-digout", [ALC880_5ST] = "5stack", [ALC880_5ST_DIG] = "5stack-digout", @@ -1169,7 +1128,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734), SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), - SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700), SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG), SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG), @@ -1216,20 +1174,6 @@ static const struct alc_config_preset alc880_presets[] = { .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, - [ALC880_TCL_S700] = { - .mixers = { alc880_tcl_s700_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_tcl_S700_init_verbs, - alc880_gpio2_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_dac_nids), - .dac_nids = alc880_dac_nids, - .adc_nids = alc880_adc_nids_alt, /* FIXME: correct? */ - .num_adc_nids = 1, /* single ADC */ - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), - .channel_mode = alc880_2_jack_modes, - .input_mux = &alc880_capture_source, - }, [ALC880_5ST] = { .mixers = { alc880_three_stack_mixer, alc880_five_stack_mixer}, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4f8c3620799..e6eec9a9ab4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4407,6 +4407,7 @@ enum { ALC880_FIXUP_LG, ALC880_FIXUP_W810, ALC880_FIXUP_EAPD_COEF, + ALC880_FIXUP_TCL_S700, }; static const struct alc_fixup alc880_fixups[] = { @@ -4453,6 +4454,17 @@ static const struct alc_fixup alc880_fixups[] = { {} }, }, + [ALC880_FIXUP_TCL_S700] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* change to EAPD mode */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x3070 }, + {} + }, + .chained = true, + .chain_id = ALC880_FIXUP_GPIO2, + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { @@ -4463,6 +4475,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG), + SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_FIXUP_TCL_S700), {} }; -- cgit v1.2.3-70-g09d2 From cf5a22793cfa54c056655d374722dc5dfd496eca Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 16:31:07 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=futjisu with auto-parser Now adding the support for the volume-knob widget, we can move the static quirk for ALC880 model=fujitsu to the auto-parser completely. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - sound/pci/hda/alc880_quirks.c | 33 ------------ sound/pci/hda/patch_realtek.c | 76 +++++++++++++++++++++++++++- 3 files changed, 74 insertions(+), 36 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index fd3b3b25bd7..936eef31ff9 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -14,7 +14,6 @@ ALC880 asus-dig ASUS with SPDIF out asus-dig2 ASUS with SPDIF out (using GPIO2) uniwill 3-jack - fujitsu Fujitsu Laptops (Pi1536) F1734 2-jack test for testing/debugging purpose, almost all controls can be adjusted. Appearing only when compiled with diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 56f8fa1e346..f062eaae6b1 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -18,7 +18,6 @@ enum { ALC880_ASUS_DIG, ALC880_ASUS_W1V, ALC880_ASUS_DIG2, - ALC880_FUJITSU, ALC880_UNIWILL_DIG, ALC880_UNIWILL, ALC880_UNIWILL_P53, @@ -371,20 +370,6 @@ static const struct snd_kcontrol_new alc880_uniwill_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc880_fujitsu_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - static const struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), @@ -1074,7 +1059,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_ASUS_DIG2] = "asus-dig2", [ALC880_UNIWILL_DIG] = "uniwill", [ALC880_UNIWILL_P53] = "uniwill-p53", - [ALC880_FUJITSU] = "fujitsu", [ALC880_F1734] = "F1734", #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = "test", @@ -1125,9 +1109,7 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), - SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734), - SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG), SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG), @@ -1335,21 +1317,6 @@ static const struct alc_config_preset alc880_presets[] = { .setup = alc880_uniwill_p53_setup, .init_hook = alc_hp_automute, }, - [ALC880_FUJITSU] = { - .mixers = { alc880_fujitsu_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_uniwill_p53_init_verbs, - alc880_beep_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_dac_nids), - .dac_nids = alc880_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), - .channel_mode = alc880_2_jack_modes, - .input_mux = &alc880_capture_source, - .unsol_event = alc880_uniwill_p53_unsol_event, - .setup = alc880_uniwill_p53_setup, - .init_hook = alc_hp_automute, - }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 895113ee385..6a6436a54f0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -651,15 +651,51 @@ static void alc_exec_unsol_event(struct hda_codec *codec, int action) snd_hda_jack_report_sync(codec); } +/* update the master volume per volume-knob's unsol event */ +static void alc_update_knob_master(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int val; + struct snd_kcontrol *kctl; + struct snd_ctl_elem_value *uctl; + + kctl = snd_hda_find_mixer_ctl(codec, "Master Playback Volume"); + if (!kctl) + return; + uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); + if (!uctl) + return; + val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_VOLUME_KNOB_CONTROL, 0); + val &= HDA_AMP_VOLMASK; + uctl->value.integer.value[0] = val; + uctl->value.integer.value[1] = val; + kctl->put(kctl, uctl); + kfree(uctl); +} + /* unsolicited event for HP jack sensing */ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) { + int action; + if (codec->vendor_id == 0x10ec0880) res >>= 28; else res >>= 26; - res = snd_hda_jack_get_action(codec, res); - alc_exec_unsol_event(codec, res); + action = snd_hda_jack_get_action(codec, res); + if (res == ALC_DCVOL_EVENT) { + /* Execute the dc-vol event here as it requires the NID + * but we don't pass NID to alc_exec_unsol_event(). + * Once when we convert all static quirks to the auto-parser, + * this can be integerated into there. + */ + struct hda_jack_tbl *jack; + jack = snd_hda_jack_tbl_get_from_tag(codec, res); + if (jack) + alc_update_knob_master(codec, jack->nid); + return; + } + alc_exec_unsol_event(codec, action); } /* call init functions of standard auto-mute helpers */ @@ -4408,8 +4444,18 @@ enum { ALC880_FIXUP_W810, ALC880_FIXUP_EAPD_COEF, ALC880_FIXUP_TCL_S700, + ALC880_FIXUP_VOL_KNOB, + ALC880_FIXUP_FUJITSU, }; +/* enable the volume-knob widget support on NID 0x21 */ +static void alc880_fixup_vol_knob(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action == ALC_FIXUP_ACT_PROBE) + snd_hda_jack_detect_enable(codec, 0x21, ALC_DCVOL_EVENT); +} + static const struct alc_fixup alc880_fixups[] = { [ALC880_FIXUP_GPIO2] = { .type = ALC_FIXUP_VERBS, @@ -4465,6 +4511,30 @@ static const struct alc_fixup alc880_fixups[] = { .chained = true, .chain_id = ALC880_FIXUP_GPIO2, }, + [ALC880_FIXUP_VOL_KNOB] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc880_fixup_vol_knob, + }, + [ALC880_FIXUP_FUJITSU] = { + /* override all pins as BIOS on old Amilo is broken */ + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x0121411f }, /* HP */ + { 0x15, 0x99030120 }, /* speaker */ + { 0x16, 0x99030130 }, /* bass speaker */ + { 0x17, 0x411111f0 }, /* N/A */ + { 0x18, 0x411111f0 }, /* N/A */ + { 0x19, 0x01a19950 }, /* mic-in */ + { 0x1a, 0x411111f0 }, /* N/A */ + { 0x1b, 0x411111f0 }, /* N/A */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + { 0x1e, 0x01454140 }, /* SPDIF out */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_VOL_KNOB, + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { @@ -4472,6 +4542,8 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_FIXUP_W810), SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM), + SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FIXUP_FUJITSU), + SND_PCI_QUIRK(0x1734, 0x10b0, "FSC Amilo Pi1556", ALC880_FIXUP_FUJITSU), SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG), -- cgit v1.2.3-70-g09d2 From ba5338185dd522696f1c0d0957a724a1fdd1f39d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 16:36:52 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=F1734 with auto-parser Similar as the previous patch for model=fujitsu, we can now move the static quirk for F1734 to the auto-parser. The only difference is the default pin configurations: F1734 has less pins than Amilo's. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - sound/pci/hda/alc880_quirks.c | 85 ---------------------------- sound/pci/hda/patch_realtek.c | 24 ++++++++ 3 files changed, 24 insertions(+), 86 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 936eef31ff9..a57d7718e4f 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -14,7 +14,6 @@ ALC880 asus-dig ASUS with SPDIF out asus-dig2 ASUS with SPDIF out (using GPIO2) uniwill 3-jack - F1734 2-jack test for testing/debugging purpose, almost all controls can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index f062eaae6b1..2ab7c3b9bb9 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -13,7 +13,6 @@ enum { ALC880_Z71V, ALC880_6ST, ALC880_6ST_DIG, - ALC880_F1734, ALC880_ASUS, ALC880_ASUS_DIG, ALC880_ASUS_W1V, @@ -257,40 +256,6 @@ static const struct snd_kcontrol_new alc880_z71v_mixer[] = { { } /* end */ }; - -/* - * ALC880 F1734 model - * - * DAC: HP = 0x02 (0x0c), Front = 0x03 (0x0d) - * Pin assignment: HP = 0x14, Front = 0x15, Mic = 0x18 - */ - -static const hda_nid_t alc880_f1734_dac_nids[1] = { - 0x03 -}; -#define ALC880_F1734_HP_DAC 0x02 - -static const struct snd_kcontrol_new alc880_f1734_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_input_mux alc880_f1734_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x1 }, - { "CD", 0x4 }, - }, -}; - - /* * ALC880 ASUS model * @@ -709,38 +674,6 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec, alc_exec_unsol_event(codec, res); } -/* - * F1734 pin configuration: - * HP = 0x14, speaker-out = 0x15, mic = 0x18 - */ -static const struct hda_verb alc880_pin_f1734_init_verbs[] = { - {0x07, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, - {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_HP_EVENT}, - {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_DCVOL_EVENT}, - - { } -}; - /* * ASUS pin configuration: * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a @@ -1059,7 +992,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_ASUS_DIG2] = "asus-dig2", [ALC880_UNIWILL_DIG] = "uniwill", [ALC880_UNIWILL_P53] = "uniwill-p53", - [ALC880_F1734] = "F1734", #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = "test", #endif @@ -1103,13 +1035,10 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2), SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG), - SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_F1734), SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL), SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53), SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), - SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734), SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG), SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG), @@ -1212,20 +1141,6 @@ static const struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_2_jack_modes, .input_mux = &alc880_capture_source, }, - [ALC880_F1734] = { - .mixers = { alc880_f1734_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_f1734_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_f1734_dac_nids), - .dac_nids = alc880_f1734_dac_nids, - .hp_nid = 0x02, - .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), - .channel_mode = alc880_2_jack_modes, - .input_mux = &alc880_f1734_capture_source, - .unsol_event = alc880_uniwill_p53_unsol_event, - .setup = alc880_uniwill_p53_setup, - .init_hook = alc_hp_automute, - }, [ALC880_ASUS] = { .mixers = { alc880_asus_mixer }, .init_verbs = { alc880_volume_init_verbs, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6a6436a54f0..2d102f70f78 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4446,6 +4446,7 @@ enum { ALC880_FIXUP_TCL_S700, ALC880_FIXUP_VOL_KNOB, ALC880_FIXUP_FUJITSU, + ALC880_FIXUP_F1734, }; /* enable the volume-knob widget support on NID 0x21 */ @@ -4535,14 +4536,37 @@ static const struct alc_fixup alc880_fixups[] = { .chained = true, .chain_id = ALC880_FIXUP_VOL_KNOB, }, + [ALC880_FIXUP_F1734] = { + /* almost compatible with FUJITSU, but no bass and SPDIF */ + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x0121411f }, /* HP */ + { 0x15, 0x99030120 }, /* speaker */ + { 0x16, 0x411111f0 }, /* N/A */ + { 0x17, 0x411111f0 }, /* N/A */ + { 0x18, 0x411111f0 }, /* N/A */ + { 0x19, 0x01a19950 }, /* mic-in */ + { 0x1a, 0x411111f0 }, /* N/A */ + { 0x1b, 0x411111f0 }, /* N/A */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + { 0x1e, 0x411111f0 }, /* N/A */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_VOL_KNOB, + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), + SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_FIXUP_F1734), SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_FIXUP_W810), SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM), + SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_FIXUP_F1734), SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FIXUP_FUJITSU), + SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_FIXUP_F1734), SND_PCI_QUIRK(0x1734, 0x10b0, "FSC Amilo Pi1556", ALC880_FIXUP_FUJITSU), SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), -- cgit v1.2.3-70-g09d2 From 817de92f1b52358f28534bb0b0c373f75e4b4baa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 17:20:48 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=uniwill with auto-parser The model=uniwill would work almost as is, but a couple of adjustments are needed to make the mutli-io working correctly. The headphone and speaker pins have to be marked properly in pin configs. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - sound/pci/hda/alc880_quirks.c | 118 --------------------------- sound/pci/hda/alc_quirks.c | 12 --- sound/pci/hda/patch_realtek.c | 12 +++ 4 files changed, 12 insertions(+), 131 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index a57d7718e4f..1af6354ec54 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -13,7 +13,6 @@ ALC880 asus-w1v ASUS W1V asus-dig ASUS with SPDIF out asus-dig2 ASUS with SPDIF out (using GPIO2) - uniwill 3-jack test for testing/debugging purpose, almost all controls can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 2a00271e065..c40f2446fcc 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -18,7 +18,6 @@ enum { ALC880_ASUS_W1V, ALC880_ASUS_DIG2, ALC880_UNIWILL_DIG, - ALC880_UNIWILL, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -306,34 +305,6 @@ static const struct snd_kcontrol_new alc880_asus_w1v_mixer[] = { { } /* end */ }; -/* Uniwill */ -static const struct snd_kcontrol_new alc880_uniwill_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - /* * initialize the codec volumes, etc */ @@ -518,83 +489,11 @@ static const struct hda_verb alc880_pin_6stack_init_verbs[] = { { } }; -/* - * Uniwill pin configuration: - * HP = 0x14, InternalSpeaker = 0x15, mic = 0x18, internal mic = 0x19, - * line = 0x1a - */ -static const struct hda_verb alc880_uniwill_init_verbs[] = { - {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, */ - /* {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - - { } -}; - static const struct hda_verb alc880_beep_init_verbs[] = { { 0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) }, { } }; -static void alc880_uniwill_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x16; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static void alc880_uniwill_init_hook(struct hda_codec *codec) -{ - alc_hp_automute(codec); - alc88x_simple_mic_automute(codec); -} - -static void alc880_uniwill_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - /* Looks like the unsol event is incompatible with the standard - * definition. 4bit tag is placed at 28 bit! - */ - res >>= 28; - switch (res) { - case ALC_MIC_EVENT: - alc88x_simple_mic_automute(codec); - break; - default: - alc_exec_unsol_event(codec, res); - break; - } -} - /* * ASUS pin configuration: * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a @@ -911,7 +810,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_ASUS_W1V] = "asus-w1v", [ALC880_ASUS_DIG] = "asus-dig", [ALC880_ASUS_DIG2] = "asus-dig2", - [ALC880_UNIWILL_DIG] = "uniwill", #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = "test", #endif @@ -955,7 +853,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2), SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG), - SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL), SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ @@ -1123,21 +1020,6 @@ static const struct alc_config_preset alc880_presets[] = { .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, - [ALC880_UNIWILL] = { - .mixers = { alc880_uniwill_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_uniwill_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), - .dac_nids = alc880_asus_dac_nids, - .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, - .unsol_event = alc880_uniwill_unsol_event, - .setup = alc880_uniwill_setup, - .init_hook = alc880_uniwill_init_hook, - }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, diff --git a/sound/pci/hda/alc_quirks.c b/sound/pci/hda/alc_quirks.c index b344603ac06..a63a517780d 100644 --- a/sound/pci/hda/alc_quirks.c +++ b/sound/pci/hda/alc_quirks.c @@ -165,15 +165,3 @@ static void alc_simple_setup_automute(struct alc_spec *spec, int mode) spec->automute_lo = spec->automute_lo_possible = !!lo_pin; spec->automute_speaker = spec->automute_speaker_possible = !!spec->autocfg.speaker_pins[0]; } - -/* auto-toggle front mic */ -static void alc88x_simple_mic_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_jack_detect(codec, 0x18); - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); -} - diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3c0a46ed9ca..ff4410cf75a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4447,6 +4447,7 @@ enum { ALC880_FIXUP_VOL_KNOB, ALC880_FIXUP_FUJITSU, ALC880_FIXUP_F1734, + ALC880_FIXUP_UNIWILL, }; /* enable the volume-knob widget support on NID 0x21 */ @@ -4556,12 +4557,23 @@ static const struct alc_fixup alc880_fixups[] = { .chained = true, .chain_id = ALC880_FIXUP_VOL_KNOB, }, + [ALC880_FIXUP_UNIWILL] = { + /* need to fix HP and speaker pins to be parsed correctly */ + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x0121411f }, /* HP */ + { 0x15, 0x99030120 }, /* speaker */ + { 0x16, 0x99030130 }, /* bass speaker */ + { } + }, + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_FIXUP_F1734), + SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_FIXUP_UNIWILL), SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_FIXUP_VOL_KNOB), SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_FIXUP_W810), SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM), -- cgit v1.2.3-70-g09d2 From 96e225f6922ecf3afafb55fdb0e6e771b3f71e94 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 17:41:51 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=z71v with auto-parser ASUS Z71V has a totally broken BIOS setup (at least the info I got), thus we need to override the whole pin-config table to make the auto-parser working correctly. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - sound/pci/hda/alc880_quirks.c | 58 ---------------------------- sound/pci/hda/patch_realtek.c | 20 ++++++++++ 3 files changed, 20 insertions(+), 59 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 1af6354ec54..43f3c711395 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -8,7 +8,6 @@ ALC880 5stack-digout 5-jack in back, 2-jack in front, a SPDIF out 6stack 6-jack in back, 2-jack in front 6stack-digout 6-jack with a SPDIF out - z71v 3-jack (HP shared SPDIF) asus 3-jack (ASUS Mobo) asus-w1v ASUS W1V asus-dig ASUS with SPDIF out diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 59899f8b056..6caa2010a85 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -10,7 +10,6 @@ enum { ALC880_3ST_DIG, ALC880_5ST, ALC880_5ST_DIG, - ALC880_Z71V, ALC880_6ST, ALC880_6ST_DIG, ALC880_ASUS, @@ -223,36 +222,11 @@ static const hda_nid_t alc880_w810_dac_nids[3] = { 0x02, 0x03, 0x04 }; -/* - * Z710V model - * - * DAC: Front = 0x02 (0x0c), HP = 0x03 (0x0d) - * Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?), - * Line = 0x1a - */ - -static const hda_nid_t alc880_z71v_dac_nids[1] = { - 0x02 -}; -#define ALC880_Z71V_HP_DAC 0x03 - /* fixed 2 channels */ static const struct hda_channel_mode alc880_2_jack_modes[1] = { { 2, NULL } }; -static const struct snd_kcontrol_new alc880_z71v_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - /* * ALC880 ASUS model * @@ -440,24 +414,6 @@ static const struct hda_verb alc880_pin_5stack_init_verbs[] = { { } }; -/* - * Z71V pin configuration: - * Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?) - */ -static const struct hda_verb alc880_pin_z71v_init_verbs[] = { - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - { } -}; - /* * 6-stack pin configuration: * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18, @@ -802,7 +758,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_3ST_DIG] = "3stack-digout", [ALC880_5ST] = "5stack", [ALC880_5ST_DIG] = "5stack-digout", - [ALC880_Z71V] = "z71v", [ALC880_6ST] = "6stack", [ALC880_6ST_DIG] = "6stack-digout", [ALC880_ASUS] = "asus", @@ -831,7 +786,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1113, "ASUS", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x1123, "ASUS", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x1173, "ASUS", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_Z71V), /* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */ SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS", ALC880_ASUS_DIG), @@ -943,18 +897,6 @@ static const struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_sixstack_modes, .input_mux = &alc880_6stack_capture_source, }, - [ALC880_Z71V] = { - .mixers = { alc880_z71v_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_z71v_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_z71v_dac_nids), - .dac_nids = alc880_z71v_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), - .channel_mode = alc880_2_jack_modes, - .input_mux = &alc880_capture_source, - }, [ALC880_ASUS] = { .mixers = { alc880_asus_mixer }, .init_verbs = { alc880_volume_init_verbs, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e88c753743d..71acd9b9a88 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4449,6 +4449,7 @@ enum { ALC880_FIXUP_F1734, ALC880_FIXUP_UNIWILL, ALC880_FIXUP_UNIWILL_DIG, + ALC880_FIXUP_Z71V, }; /* enable the volume-knob widget support on NID 0x21 */ @@ -4579,10 +4580,29 @@ static const struct alc_fixup alc880_fixups[] = { { } } }, + [ALC880_FIXUP_Z71V] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + /* set up the whole pins as BIOS is utterly broken */ + { 0x14, 0x99030120 }, /* speaker */ + { 0x15, 0x0121411f }, /* HP */ + { 0x16, 0x411111f0 }, /* N/A */ + { 0x17, 0x411111f0 }, /* N/A */ + { 0x18, 0x01a19950 }, /* mic-in */ + { 0x19, 0x411111f0 }, /* N/A */ + { 0x1a, 0x01813031 }, /* line-in */ + { 0x1b, 0x411111f0 }, /* N/A */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + { 0x1e, 0x0144111e }, /* SPDIF */ + { } + } + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), + SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_FIXUP_Z71V), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_FIXUP_UNIWILL_DIG), SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_FIXUP_F1734), -- cgit v1.2.3-70-g09d2 From 411225a01e57189b4116d5c61c0d64bd4b76e602 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 17:48:19 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=asus-w1v with auto-parser ASUS W1V has a sane pin-config table set by BIOS. The only missing piece is the setup of GPIO1. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - sound/pci/hda/alc880_quirks.c | 31 ---------------------------- sound/pci/hda/patch_realtek.c | 6 ++++++ 3 files changed, 6 insertions(+), 32 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 43f3c711395..f1820679985 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -9,7 +9,6 @@ ALC880 6stack 6-jack in back, 2-jack in front 6stack-digout 6-jack with a SPDIF out asus 3-jack (ASUS Mobo) - asus-w1v ASUS W1V asus-dig ASUS with SPDIF out asus-dig2 ASUS with SPDIF out (using GPIO2) test for testing/debugging purpose, almost all controls can be diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 6caa2010a85..c8af01b7f85 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -14,7 +14,6 @@ enum { ALC880_6ST_DIG, ALC880_ASUS, ALC880_ASUS_DIG, - ALC880_ASUS_W1V, ALC880_ASUS_DIG2, #ifdef CONFIG_SND_DEBUG ALC880_TEST, @@ -263,21 +262,6 @@ static const struct snd_kcontrol_new alc880_asus_mixer[] = { { } /* end */ }; -/* - * ALC880 ASUS W1V model - * - * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e) - * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16, - * Mic = 0x18, Line = 0x1a, Line2 = 0x1b - */ - -/* additional mixers to alc880_asus_mixer */ -static const struct snd_kcontrol_new alc880_asus_w1v_mixer[] = { - HDA_CODEC_VOLUME("Line2 Playback Volume", 0x0b, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Line2 Playback Switch", 0x0b, 0x03, HDA_INPUT), - { } /* end */ -}; - /* * initialize the codec volumes, etc */ @@ -761,7 +745,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_6ST] = "6stack", [ALC880_6ST_DIG] = "6stack-digout", [ALC880_ASUS] = "asus", - [ALC880_ASUS_W1V] = "asus-w1v", [ALC880_ASUS_DIG] = "asus-dig", [ALC880_ASUS_DIG2] = "asus-dig2", #ifdef CONFIG_SND_DEBUG @@ -780,7 +763,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG), SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST), SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V), SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x1113, "ASUS", ALC880_ASUS_DIG), @@ -935,19 +917,6 @@ static const struct alc_config_preset alc880_presets[] = { .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, - [ALC880_ASUS_W1V] = { - .mixers = { alc880_asus_mixer, alc880_asus_w1v_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_asus_init_verbs, - alc880_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), - .dac_nids = alc880_asus_dac_nids, - .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, - }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 71acd9b9a88..510ca928b84 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4438,6 +4438,7 @@ static const struct hda_amp_list alc880_loopbacks[] = { * ALC880 fix-ups */ enum { + ALC880_FIXUP_GPIO1, ALC880_FIXUP_GPIO2, ALC880_FIXUP_MEDION_RIM, ALC880_FIXUP_LG, @@ -4461,6 +4462,10 @@ static void alc880_fixup_vol_knob(struct hda_codec *codec, } static const struct alc_fixup alc880_fixups[] = { + [ALC880_FIXUP_GPIO1] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = alc_gpio1_init_verbs, + }, [ALC880_FIXUP_GPIO2] = { .type = ALC_FIXUP_VERBS, .v.verbs = alc_gpio2_init_verbs, @@ -4602,6 +4607,7 @@ static const struct alc_fixup alc880_fixups[] = { static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), + SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_FIXUP_GPIO1), SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_FIXUP_Z71V), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_FIXUP_UNIWILL_DIG), -- cgit v1.2.3-70-g09d2 From 29e3fdcc84e5da04cb7e6a36fee0a772c91d3b28 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 17:56:57 +0100 Subject: ALSA: hda/realtek - Drop model=asus* from ALC880 It turned out that BIOS on most of ASUS mobo's set the pin-config tables reasonably well for the auto-parser. We'd need GPIO setups, but should work as is other than that. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 3 - sound/pci/hda/alc880_quirks.c | 127 +-------------------------- sound/pci/hda/patch_realtek.c | 3 +- 3 files changed, 3 insertions(+), 130 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index f1820679985..62a67896fca 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -8,9 +8,6 @@ ALC880 5stack-digout 5-jack in back, 2-jack in front, a SPDIF out 6stack 6-jack in back, 2-jack in front 6stack-digout 6-jack with a SPDIF out - asus 3-jack (ASUS Mobo) - asus-dig ASUS with SPDIF out - asus-dig2 ASUS with SPDIF out (using GPIO2) test for testing/debugging purpose, almost all controls can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index c8af01b7f85..6917d78d4dc 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -12,9 +12,6 @@ enum { ALC880_5ST_DIG, ALC880_6ST, ALC880_6ST_DIG, - ALC880_ASUS, - ALC880_ASUS_DIG, - ALC880_ASUS_DIG2, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -226,42 +223,6 @@ static const struct hda_channel_mode alc880_2_jack_modes[1] = { { 2, NULL } }; -/* - * ALC880 ASUS model - * - * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e) - * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16, - * Mic = 0x18, Line = 0x1a - */ - -#define alc880_asus_dac_nids alc880_w810_dac_nids /* identical with w810 */ -#define alc880_asus_modes alc880_threestack_modes /* 2/6 channel mode */ - -static const struct snd_kcontrol_new alc880_asus_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - /* * initialize the codec volumes, etc */ @@ -433,38 +394,6 @@ static const struct hda_verb alc880_beep_init_verbs[] = { { } }; -/* - * ASUS pin configuration: - * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a - */ -static const struct hda_verb alc880_pin_asus_init_verbs[] = { - {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, - {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - { } -}; - /* Enable GPIO mask and set output */ #define alc880_gpio1_init_verbs alc_gpio1_init_verbs #define alc880_gpio2_init_verbs alc_gpio2_init_verbs @@ -744,9 +673,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_5ST_DIG] = "5stack-digout", [ALC880_6ST] = "6stack", [ALC880_6ST_DIG] = "6stack-digout", - [ALC880_ASUS] = "asus", - [ALC880_ASUS_DIG] = "asus-dig", - [ALC880_ASUS_DIG2] = "asus-dig2", #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = "test", #endif @@ -763,19 +689,7 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG), SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST), SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x1113, "ASUS", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x1123, "ASUS", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x1173, "ASUS", ALC880_ASUS_DIG), - /* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */ - SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x814e, "ASUS P5GD1 w/SPDIF", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST), - SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST), - SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_ASUS), /* default ASUS */ + SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST), SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST), SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST), @@ -785,7 +699,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2), SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), @@ -879,44 +792,6 @@ static const struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_sixstack_modes, .input_mux = &alc880_6stack_capture_source, }, - [ALC880_ASUS] = { - .mixers = { alc880_asus_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_asus_init_verbs, - alc880_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), - .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] = { - .mixers = { alc880_asus_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_asus_init_verbs, - alc880_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), - .dac_nids = alc880_asus_dac_nids, - .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] = { - .mixers = { alc880_asus_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_asus_init_verbs, - alc880_gpio2_init_verbs }, /* use GPIO2 */ - .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), - .dac_nids = alc880_asus_dac_nids, - .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, - }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 510ca928b84..fce31b050f4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4607,8 +4607,9 @@ static const struct alc_fixup alc880_fixups[] = { static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), - SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_FIXUP_GPIO1), SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_FIXUP_Z71V), + SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_FIXUP_GPIO1), + SND_PCI_QUIRK(0x1558, 0x5401, "Clevo GPIO2", ALC880_FIXUP_GPIO2), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_FIXUP_UNIWILL_DIG), SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_FIXUP_F1734), -- cgit v1.2.3-70-g09d2 From 67b6ec3196da235317ff1b9474f17379b78f3294 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 18:20:42 +0100 Subject: ALSA: hda/realtek - Drop all ALC880 static quirks Finally the all static quirks for ALC880 are converted to the auto-parser. Since we are never sure whether the BIOS on so many old machines are really correct, the quirk table entries are copied as they are, but just providing the proper pin-config values accordingly. Since alc880_quirks.c is removed, alc882_quirks.c has to be adjusted slightly to be built again. There might be some compile warnings due to the remaining alc882 quirks, but these shall be killed sooner or later, I don't care it much at this point. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 4 - sound/pci/hda/alc880_quirks.c | 808 --------------------------- sound/pci/hda/alc882_quirks.c | 24 +- sound/pci/hda/patch_realtek.c | 219 ++++++-- 4 files changed, 195 insertions(+), 860 deletions(-) delete mode 100644 sound/pci/hda/alc880_quirks.c (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 62a67896fca..1f64fb81052 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -8,10 +8,6 @@ ALC880 5stack-digout 5-jack in back, 2-jack in front, a SPDIF out 6stack 6-jack in back, 2-jack in front 6stack-digout 6-jack with a SPDIF out - test for testing/debugging purpose, almost all controls can be - adjusted. Appearing only when compiled with - $CONFIG_SND_DEBUG=y - auto auto-config reading BIOS (default) ALC260 ====== diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c deleted file mode 100644 index 6917d78d4dc..00000000000 --- a/sound/pci/hda/alc880_quirks.c +++ /dev/null @@ -1,808 +0,0 @@ -/* - * ALC880 quirk models - * included by patch_realtek.c - */ - -/* ALC880 board config type */ -enum { - ALC880_AUTO, - ALC880_3ST, - ALC880_3ST_DIG, - ALC880_5ST, - ALC880_5ST_DIG, - ALC880_6ST, - ALC880_6ST_DIG, -#ifdef CONFIG_SND_DEBUG - ALC880_TEST, -#endif - ALC880_MODEL_LAST /* last tag */ -}; - -/* - * ALC880 3-stack model - * - * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e) - * Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18, - * F-Mic = 0x1b, HP = 0x19 - */ - -static const hda_nid_t alc880_dac_nids[4] = { - /* front, rear, clfe, rear_surr */ - 0x02, 0x05, 0x04, 0x03 -}; - -static const hda_nid_t alc880_adc_nids[3] = { - /* ADC0-2 */ - 0x07, 0x08, 0x09, -}; - -/* The datasheet says the node 0x07 is connected from inputs, - * but it shows zero connection in the real implementation on some devices. - * Note: this is a 915GAV bug, fixed on 915GLV - */ -static const hda_nid_t alc880_adc_nids_alt[2] = { - /* ADC1-2 */ - 0x08, 0x09, -}; - -#define ALC880_DIGOUT_NID 0x06 -#define ALC880_DIGIN_NID 0x0a -#define ALC880_PIN_CD_NID 0x1c - -static const struct hda_input_mux alc880_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x3 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -/* channel source setting (2/6 channel selection for 3-stack) */ -/* 2ch mode */ -static const struct hda_verb alc880_threestack_ch2_init[] = { - /* set line-in to input, mute it */ - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - /* set mic-in to input vref 80%, mute it */ - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; - -/* 6ch mode */ -static const struct hda_verb alc880_threestack_ch6_init[] = { - /* set line-in to output, unmute it */ - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - /* set mic-in to output, unmute it */ - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { } /* end */ -}; - -static const struct hda_channel_mode alc880_threestack_modes[2] = { - { 2, alc880_threestack_ch2_init }, - { 6, alc880_threestack_ch6_init }, -}; - -static const struct snd_kcontrol_new alc880_three_stack_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - -/* - * ALC880 5-stack model - * - * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d), - * Side = 0x02 (0xd) - * Pin assignment: Front = 0x14, Surr = 0x17, CLFE = 0x16 - * Line-In/Side = 0x1a, Mic = 0x18, F-Mic = 0x1b, HP = 0x19 - */ - -/* additional mixers to alc880_three_stack_mixer */ -static const struct snd_kcontrol_new alc880_five_stack_mixer[] = { - HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT), - { } /* end */ -}; - -/* channel source setting (6/8 channel selection for 5-stack) */ -/* 6ch mode */ -static const struct hda_verb alc880_fivestack_ch6_init[] = { - /* set line-in to input, mute it */ - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; - -/* 8ch mode */ -static const struct hda_verb alc880_fivestack_ch8_init[] = { - /* set line-in to output, unmute it */ - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { } /* end */ -}; - -static const struct hda_channel_mode alc880_fivestack_modes[2] = { - { 6, alc880_fivestack_ch6_init }, - { 8, alc880_fivestack_ch8_init }, -}; - - -/* - * ALC880 6-stack model - * - * DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e), - * Side = 0x05 (0x0f) - * Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, Side = 0x17, - * Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b - */ - -static const hda_nid_t alc880_6st_dac_nids[4] = { - /* front, rear, clfe, rear_surr */ - 0x02, 0x03, 0x04, 0x05 -}; - -static const struct hda_input_mux alc880_6stack_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -/* fixed 8-channels */ -static const struct hda_channel_mode alc880_sixstack_modes[1] = { - { 8, NULL }, -}; - -static const struct snd_kcontrol_new alc880_six_stack_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - - -static const hda_nid_t alc880_w810_dac_nids[3] = { - /* front, rear/surround, clfe */ - 0x02, 0x03, 0x04 -}; - -/* fixed 2 channels */ -static const struct hda_channel_mode alc880_2_jack_modes[1] = { - { 2, NULL } -}; - -/* - * initialize the codec volumes, etc - */ - -/* - * generic initialization of ADC, input mixers and output mixers - */ -static const struct hda_verb alc880_volume_init_verbs[] = { - /* - * Unmute ADC0-2 and set the default input to mic-in - */ - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback - * mixer widget - * Note: PASD motherboards uses the Line In 2 as the input for front - * panel mic (mic 2) - */ - /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, - - /* - * Set up output mixers (0x0c - 0x0f) - */ - /* set vol=0 to output mixers */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* set up input amps for analog loopback */ - /* Amp Indices: DAC = 0, mixer = 1 */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - - { } -}; - -/* - * 3-stack pin configuration: - * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b - */ -static const struct hda_verb alc880_pin_3stack_init_verbs[] = { - /* - * preset connection lists of input pins - * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround - */ - {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ - {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ - {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */ - - /* - * Set pin mode and muting - */ - /* set front pin widgets 0x14 for output */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Mic1 (rear panel) pin widget for input and vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Mic2 (as headphone out) for HP output */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Line In pin widget for input */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line2 (as front mic) pin widget for input and vref at 80% */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* CD pin widget for input */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - { } -}; - -/* - * 5-stack pin configuration: - * front = 0x14, surround = 0x17, clfe = 0x16, mic = 0x18, HP = 0x19, - * line-in/side = 0x1a, f-mic = 0x1b - */ -static const struct hda_verb alc880_pin_5stack_init_verbs[] = { - /* - * preset connection lists of input pins - * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround - */ - {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ - {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/side */ - - /* - * Set pin mode and muting - */ - /* set pin widgets 0x14-0x17 for output */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* unmute pins for output (no gain on this amp) */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Mic1 (rear panel) pin widget for input and vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Mic2 (as headphone out) for HP output */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Line In pin widget for input */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line2 (as front mic) pin widget for input and vref at 80% */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* CD pin widget for input */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - { } -}; - -/* - * 6-stack pin configuration: - * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18, - * f-mic = 0x19, line = 0x1a, HP = 0x1b - */ -static const struct hda_verb alc880_pin_6stack_init_verbs[] = { - {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - { } -}; - -static const struct hda_verb alc880_beep_init_verbs[] = { - { 0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) }, - { } -}; - -/* Enable GPIO mask and set output */ -#define alc880_gpio1_init_verbs alc_gpio1_init_verbs -#define alc880_gpio2_init_verbs alc_gpio2_init_verbs -#define alc880_gpio3_init_verbs alc_gpio3_init_verbs - -/* - * Test configuration for debugging - * - * Almost all inputs/outputs are enabled. I/O pins can be configured via - * enum controls. - */ -#ifdef CONFIG_SND_DEBUG -static const hda_nid_t alc880_test_dac_nids[4] = { - 0x02, 0x03, 0x04, 0x05 -}; - -static const struct hda_input_mux alc880_test_capture_source = { - .num_items = 7, - .items = { - { "In-1", 0x0 }, - { "In-2", 0x1 }, - { "In-3", 0x2 }, - { "In-4", 0x3 }, - { "CD", 0x4 }, - { "Front", 0x5 }, - { "Surround", 0x6 }, - }, -}; - -static const struct hda_channel_mode alc880_test_modes[4] = { - { 2, NULL }, - { 4, NULL }, - { 6, NULL }, - { 8, NULL }, -}; - -static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - static const char * const texts[] = { - "N/A", "Line Out", "HP Out", - "In Hi-Z", "In 50%", "In Grd", "In 80%", "In 100%" - }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 8; - if (uinfo->value.enumerated.item >= 8) - uinfo->value.enumerated.item = 7; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = (hda_nid_t)kcontrol->private_value; - unsigned int pin_ctl, item = 0; - - pin_ctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - if (pin_ctl & AC_PINCTL_OUT_EN) { - if (pin_ctl & AC_PINCTL_HP_EN) - item = 2; - else - item = 1; - } else if (pin_ctl & AC_PINCTL_IN_EN) { - switch (pin_ctl & AC_PINCTL_VREFEN) { - case AC_PINCTL_VREF_HIZ: item = 3; break; - case AC_PINCTL_VREF_50: item = 4; break; - case AC_PINCTL_VREF_GRD: item = 5; break; - case AC_PINCTL_VREF_80: item = 6; break; - case AC_PINCTL_VREF_100: item = 7; break; - } - } - ucontrol->value.enumerated.item[0] = item; - return 0; -} - -static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = (hda_nid_t)kcontrol->private_value; - static const unsigned int ctls[] = { - 0, AC_PINCTL_OUT_EN, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN, - AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ, - AC_PINCTL_IN_EN | AC_PINCTL_VREF_50, - AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD, - AC_PINCTL_IN_EN | AC_PINCTL_VREF_80, - AC_PINCTL_IN_EN | AC_PINCTL_VREF_100, - }; - unsigned int old_ctl, new_ctl; - - old_ctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - new_ctl = ctls[ucontrol->value.enumerated.item[0]]; - if (old_ctl != new_ctl) { - int val; - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - new_ctl); - val = ucontrol->value.enumerated.item[0] >= 3 ? - HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, val); - return 1; - } - return 0; -} - -static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - static const char * const texts[] = { - "Front", "Surround", "CLFE", "Side" - }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item >= 4) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = (hda_nid_t)kcontrol->private_value; - unsigned int sel; - - sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0); - ucontrol->value.enumerated.item[0] = sel & 3; - return 0; -} - -static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = (hda_nid_t)kcontrol->private_value; - unsigned int sel; - - sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3; - if (ucontrol->value.enumerated.item[0] != sel) { - sel = ucontrol->value.enumerated.item[0] & 3; - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_CONNECT_SEL, sel); - return 1; - } - return 0; -} - -#define PIN_CTL_TEST(xname,nid) { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ - .info = alc_test_pin_ctl_info, \ - .get = alc_test_pin_ctl_get, \ - .put = alc_test_pin_ctl_put, \ - .private_value = nid \ - } - -#define PIN_SRC_TEST(xname,nid) { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ - .info = alc_test_pin_src_info, \ - .get = alc_test_pin_src_get, \ - .put = alc_test_pin_src_put, \ - .private_value = nid \ - } - -static const struct snd_kcontrol_new alc880_test_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CLFE Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_BIND_MUTE("CLFE Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - PIN_CTL_TEST("Front Pin Mode", 0x14), - PIN_CTL_TEST("Surround Pin Mode", 0x15), - PIN_CTL_TEST("CLFE Pin Mode", 0x16), - PIN_CTL_TEST("Side Pin Mode", 0x17), - PIN_CTL_TEST("In-1 Pin Mode", 0x18), - PIN_CTL_TEST("In-2 Pin Mode", 0x19), - PIN_CTL_TEST("In-3 Pin Mode", 0x1a), - PIN_CTL_TEST("In-4 Pin Mode", 0x1b), - PIN_SRC_TEST("In-1 Pin Source", 0x18), - PIN_SRC_TEST("In-2 Pin Source", 0x19), - PIN_SRC_TEST("In-3 Pin Source", 0x1a), - PIN_SRC_TEST("In-4 Pin Source", 0x1b), - HDA_CODEC_VOLUME("In-1 Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("In-1 Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("In-2 Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("In-2 Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("In-3 Playback Volume", 0x0b, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("In-3 Playback Switch", 0x0b, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("In-4 Playback Volume", 0x0b, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("In-4 Playback Switch", 0x0b, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x4, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - -static const struct hda_verb alc880_test_init_verbs[] = { - /* Unmute inputs of 0x0c - 0x0f */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Vol output for 0x0c-0x0f */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Set output pins 0x14-0x17 */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Unmute output pins 0x14-0x17 */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Set input pins 0x18-0x1c */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Mute input pins 0x18-0x1b */ - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* ADC set up */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Analog input/passthru */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - { } -}; -#endif - -/* - */ - -static const char * const alc880_models[ALC880_MODEL_LAST] = { - [ALC880_3ST] = "3stack", - [ALC880_3ST_DIG] = "3stack-digout", - [ALC880_5ST] = "5stack", - [ALC880_5ST_DIG] = "5stack-digout", - [ALC880_6ST] = "6stack", - [ALC880_6ST_DIG] = "6stack-digout", -#ifdef CONFIG_SND_DEBUG - [ALC880_TEST] = "test", -#endif - [ALC880_AUTO] = "auto", -}; - -static const struct snd_pci_quirk alc880_cfg_tbl[] = { - SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_6ST), - SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_3ST_DIG), - SND_PCI_QUIRK(0x1025, 0x0077, "ULI", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1025, 0x0078, "ULI", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1025, 0x0087, "ULI", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG), - SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST), - SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG), - - SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST), - SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST), - SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST), - SND_PCI_QUIRK(0x107b, 0x3033, "Gateway", ALC880_5ST), - SND_PCI_QUIRK(0x107b, 0x4039, "Gateway", ALC880_5ST), - SND_PCI_QUIRK(0x1297, 0xc790, "Shuttle ST20G5", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ - SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG), - SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xd400, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xd401, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_3ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe224, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_3ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_3ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG), - /* default Intel */ - SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_3ST), - SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG), - SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG), - {} -}; - -/* - * ALC880 codec presets - */ -static const struct alc_config_preset alc880_presets[] = { - [ALC880_3ST] = { - .mixers = { alc880_three_stack_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_3stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_dac_nids), - .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] = { - .mixers = { alc880_three_stack_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_3stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_dac_nids), - .dac_nids = alc880_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), - .channel_mode = alc880_threestack_modes, - .need_dac_fix = 1, - .input_mux = &alc880_capture_source, - }, - [ALC880_5ST] = { - .mixers = { alc880_three_stack_mixer, - alc880_five_stack_mixer}, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_5stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_dac_nids), - .dac_nids = alc880_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes), - .channel_mode = alc880_fivestack_modes, - .input_mux = &alc880_capture_source, - }, - [ALC880_5ST_DIG] = { - .mixers = { alc880_three_stack_mixer, - alc880_five_stack_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_5stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_dac_nids), - .dac_nids = alc880_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes), - .channel_mode = alc880_fivestack_modes, - .input_mux = &alc880_capture_source, - }, - [ALC880_6ST] = { - .mixers = { alc880_six_stack_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_6stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids), - .dac_nids = alc880_6st_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes), - .channel_mode = alc880_sixstack_modes, - .input_mux = &alc880_6stack_capture_source, - }, - [ALC880_6ST_DIG] = { - .mixers = { alc880_six_stack_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_6stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids), - .dac_nids = alc880_6st_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes), - .channel_mode = alc880_sixstack_modes, - .input_mux = &alc880_6stack_capture_source, - }, -#ifdef CONFIG_SND_DEBUG - [ALC880_TEST] = { - .mixers = { alc880_test_mixer }, - .init_verbs = { alc880_test_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_test_dac_nids), - .dac_nids = alc880_test_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_test_modes), - .channel_mode = alc880_test_modes, - .input_mux = &alc880_test_capture_source, - }, -#endif -}; - diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index bb364a53f54..0f4292688e1 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -34,8 +34,16 @@ static const hda_nid_t alc882_dac_nids[4] = { #define alc883_dac_nids alc882_dac_nids /* ADCs */ -#define alc882_adc_nids alc880_adc_nids -#define alc882_adc_nids_alt alc880_adc_nids_alt +static const hda_nid_t alc882_adc_nids[3] = { + /* ADC0-2 */ + 0x07, 0x08, 0x09, +}; + +static const hda_nid_t alc882_adc_nids_alt[2] = { + /* ADC1-2 */ + 0x08, 0x09, +}; + #define alc883_adc_nids alc882_adc_nids_alt static const hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 }; @@ -774,7 +782,7 @@ static const struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { static const struct alc_config_preset alc882_presets[] = { [ALC885_MBA21] = { .mixers = { alc885_mba21_mixer }, - .init_verbs = { alc885_mba21_init_verbs, alc880_gpio1_init_verbs }, + .init_verbs = { alc885_mba21_init_verbs, alc_gpio1_init_verbs }, .num_dacs = 2, .dac_nids = alc882_dac_nids, .channel_mode = alc885_mba21_ch_modes, @@ -787,7 +795,7 @@ static const struct alc_config_preset alc882_presets[] = { [ALC885_MBP3] = { .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, .init_verbs = { alc885_mbp3_init_verbs, - alc880_gpio1_init_verbs }, + alc_gpio1_init_verbs }, .num_dacs = 2, .dac_nids = alc882_dac_nids, .hp_nid = 0x04, @@ -803,7 +811,7 @@ static const struct alc_config_preset alc882_presets[] = { [ALC885_MB5] = { .mixers = { alc885_mb5_mixer, alc882_chmode_mixer }, .init_verbs = { alc885_mb5_init_verbs, - alc880_gpio1_init_verbs }, + alc_gpio1_init_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .channel_mode = alc885_mb5_6ch_modes, @@ -818,7 +826,7 @@ static const struct alc_config_preset alc882_presets[] = { [ALC885_MACMINI3] = { .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer }, .init_verbs = { alc885_macmini3_init_verbs, - alc880_gpio1_init_verbs }, + alc_gpio1_init_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .channel_mode = alc885_macmini3_6ch_modes, @@ -833,7 +841,7 @@ static const struct alc_config_preset alc882_presets[] = { [ALC885_IMAC91] = { .mixers = {alc885_imac91_mixer}, .init_verbs = { alc885_imac91_init_verbs, - alc880_gpio1_init_verbs }, + alc_gpio1_init_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .channel_mode = alc885_mba21_ch_modes, @@ -848,7 +856,7 @@ static const struct alc_config_preset alc882_presets[] = { [ALC889A_MB31] = { .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer}, .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs, - alc880_gpio1_init_verbs }, + alc_gpio1_init_verbs }, .adc_nids = alc883_adc_nids, .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), .capsrc_nids = alc883_capsrc_nids, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fce31b050f4..4ac1e3830af 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4451,6 +4451,15 @@ enum { ALC880_FIXUP_UNIWILL, ALC880_FIXUP_UNIWILL_DIG, ALC880_FIXUP_Z71V, + ALC880_FIXUP_3ST_BASE, + ALC880_FIXUP_3ST, + ALC880_FIXUP_3ST_DIG, + ALC880_FIXUP_5ST_BASE, + ALC880_FIXUP_5ST, + ALC880_FIXUP_5ST_DIG, + ALC880_FIXUP_6ST_BASE, + ALC880_FIXUP_6ST, + ALC880_FIXUP_6ST_DIG, }; /* enable the volume-knob widget support on NID 0x21 */ @@ -4603,6 +4612,114 @@ static const struct alc_fixup alc880_fixups[] = { { } } }, + [ALC880_FIXUP_3ST_BASE] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x01014010 }, /* line-out */ + { 0x15, 0x411111f0 }, /* N/A */ + { 0x16, 0x411111f0 }, /* N/A */ + { 0x17, 0x411111f0 }, /* N/A */ + { 0x18, 0x01a19c30 }, /* mic-in */ + { 0x19, 0x0121411f }, /* HP */ + { 0x1a, 0x01813031 }, /* line-in */ + { 0x1b, 0x02a19c40 }, /* front-mic */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + /* 0x1e is filled in below */ + { 0x1f, 0x411111f0 }, /* N/A */ + { } + } + }, + [ALC880_FIXUP_3ST] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1e, 0x411111f0 }, /* N/A */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_3ST_BASE, + }, + [ALC880_FIXUP_3ST_DIG] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1e, 0x0144111e }, /* SPDIF */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_3ST_BASE, + }, + [ALC880_FIXUP_5ST_BASE] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x01014010 }, /* front */ + { 0x15, 0x411111f0 }, /* N/A */ + { 0x16, 0x01011411 }, /* CLFE */ + { 0x17, 0x01016412 }, /* surr */ + { 0x18, 0x01a19c30 }, /* mic-in */ + { 0x19, 0x0121411f }, /* HP */ + { 0x1a, 0x01813031 }, /* line-in */ + { 0x1b, 0x02a19c40 }, /* front-mic */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + /* 0x1e is filled in below */ + { 0x1f, 0x411111f0 }, /* N/A */ + { } + } + }, + [ALC880_FIXUP_5ST] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1e, 0x411111f0 }, /* N/A */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_5ST_BASE, + }, + [ALC880_FIXUP_5ST_DIG] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1e, 0x0144111e }, /* SPDIF */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_5ST_BASE, + }, + [ALC880_FIXUP_6ST_BASE] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x01014010 }, /* front */ + { 0x15, 0x01016412 }, /* surr */ + { 0x16, 0x01011411 }, /* CLFE */ + { 0x17, 0x01012414 }, /* side */ + { 0x18, 0x01a19c30 }, /* mic-in */ + { 0x19, 0x02a19c40 }, /* front-mic */ + { 0x1a, 0x01813031 }, /* line-in */ + { 0x1b, 0x0121411f }, /* HP */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + /* 0x1e is filled in below */ + { 0x1f, 0x411111f0 }, /* N/A */ + { } + } + }, + [ALC880_FIXUP_6ST] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1e, 0x411111f0 }, /* N/A */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_6ST_BASE, + }, + [ALC880_FIXUP_6ST_DIG] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1e, 0x0144111e }, /* SPDIF */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_6ST_BASE, + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { @@ -4625,6 +4742,60 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_FIXUP_TCL_S700), + + /* Below is the copied entries from alc880_quirks.c. + * It's not quite sure whether BIOS sets the correct pin-config table + * on these machines, thus they are kept to be compatible with + * the old static quirks. Once when it's confirmed to work without + * these overrides, it'd be better to remove. + */ + SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_FIXUP_6ST), + SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_FIXUP_3ST_DIG), + SND_PCI_QUIRK(0x1025, 0x0077, "ULI", ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x1025, 0x0078, "ULI", ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x1025, 0x0087, "ULI", ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_FIXUP_3ST_DIG), + SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_FIXUP_3ST), + SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_FIXUP_3ST), + SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_FIXUP_3ST), + SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_FIXUP_5ST), + SND_PCI_QUIRK(0x107b, 0x3033, "Gateway", ALC880_FIXUP_5ST), + SND_PCI_QUIRK(0x107b, 0x4039, "Gateway", ALC880_FIXUP_5ST), + SND_PCI_QUIRK(0x1297, 0xc790, "Shuttle ST20G5", ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_FIXUP_6ST_DIG), /* broken BIOS */ + SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xd400, "Intel mobo", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xd401, "Intel mobo", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_FIXUP_3ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe224, "Intel mobo", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_FIXUP_3ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_FIXUP_3ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_FIXUP_5ST_DIG), + /* default Intel */ + SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_FIXUP_3ST), + SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_FIXUP_6ST_DIG), + {} +}; + +static const struct alc_model_fixup alc880_fixup_models[] = { + {.id = ALC880_FIXUP_3ST, .name = "3stack"}, + {.id = ALC880_FIXUP_3ST_DIG, .name = "3stack-digout"}, + {.id = ALC880_FIXUP_5ST, .name = "5stack"}, + {.id = ALC880_FIXUP_5ST_DIG, .name = "5stack-digout"}, + {.id = ALC880_FIXUP_6ST, .name = "6stack"}, + {.id = ALC880_FIXUP_6ST_DIG, .name = "6stack-digout"}, {} }; @@ -4647,14 +4818,9 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { /* * OK, here we have finally the patch for ALC880 */ -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS -#include "alc880_quirks.c" -#endif - static int patch_alc880(struct hda_codec *codec) { struct alc_spec *spec; - int board_config; int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -4666,38 +4832,14 @@ static int patch_alc880(struct hda_codec *codec) spec->mixer_nid = 0x0b; spec->need_dac_fix = 1; - board_config = alc_board_config(codec, ALC880_MODEL_LAST, - alc880_models, alc880_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = ALC_MODEL_AUTO; - } - - if (board_config == ALC_MODEL_AUTO) { - alc_pick_fixup(codec, NULL, alc880_fixup_tbl, alc880_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - } - - if (board_config == ALC_MODEL_AUTO) { - /* automatic parse from the BIOS config */ - err = alc880_parse_auto_config(codec); - if (err < 0) - goto error; -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS - else if (!err) { - printk(KERN_INFO - "hda_codec: Cannot set up configuration " - "from BIOS. Using 3-stack mode...\n"); - board_config = ALC880_3ST; - } -#endif - } + alc_pick_fixup(codec, alc880_fixup_models, alc880_fixup_tbl, + alc880_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - if (board_config != ALC_MODEL_AUTO) { - spec->vmaster_nid = 0x0c; - setup_preset(codec, &alc880_presets[board_config]); - } + /* automatic parse from the BIOS config */ + err = alc880_parse_auto_config(codec); + if (err < 0) + goto error; if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4716,10 +4858,7 @@ static int patch_alc880(struct hda_codec *codec) } codec->patch_ops = alc_patch_ops; - if (board_config == ALC_MODEL_AUTO) - spec->init_hook = alc_auto_init_std; - else - codec->patch_ops.build_controls = __alc_build_controls; + spec->init_hook = alc_auto_init_std; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc880_loopbacks; -- cgit v1.2.3-70-g09d2 From 1a97b7f22774b454531f013638b181803fba470f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Feb 2012 11:11:48 +0100 Subject: ALSA: hda/realtek - Remove the last static quirks for ALC882 Resitance is futile. The remaining static model quirks for Apple machines with ALC882-compatible codecs are converted to the auto-parser now. We can remove all alc*_quirks.c finally. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 50 +- sound/pci/hda/alc882_quirks.c | 874 --------------------------- sound/pci/hda/alc_quirks.c | 167 ----- sound/pci/hda/patch_realtek.c | 143 +++-- 4 files changed, 93 insertions(+), 1141 deletions(-) delete mode 100644 sound/pci/hda/alc882_quirks.c delete mode 100644 sound/pci/hda/alc_quirks.c (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 1f64fb81052..d97d992ced1 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -43,55 +43,7 @@ ALC680 ALC882/883/885/888/889 ====================== - 3stack-dig 3-jack with SPDIF I/O - 6stack-dig 6-jack digital with SPDIF I/O - arima Arima W820Di1 - targa Targa T8, MSI-1049 T8 - asus-a7j ASUS A7J - asus-a7m ASUS A7M - macpro MacPro support - mb5 Macbook 5,1 - macmini3 Macmini 3,1 - mba21 Macbook Air 2,1 - mbp3 Macbook Pro rev3 - imac24 iMac 24'' with jack detection - imac91 iMac 9,1 - w2jc ASUS W2JC - 3stack-2ch-dig 3-jack with SPDIF I/O (ALC883) - alc883-6stack-dig 6-jack digital with SPDIF I/O (ALC883) - 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) - acer-aspire Acer Aspire 9810 - acer-aspire-4930g Acer Aspire 4930G - acer-aspire-6530g Acer Aspire 6530G - acer-aspire-7730g Acer Aspire 7730G - acer-aspire-8930g Acer Aspire 8930G - medion Medion Laptops - targa-dig Targa/MSI - targa-2ch-dig Targa/MSI with 2-channel - targa-8ch-dig Targa/MSI with 8-channel (MSI GX620) - laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) - lenovo-101e Lenovo 101E - lenovo-nb0763 Lenovo NB0763 - lenovo-ms7195-dig Lenovo MS7195 - lenovo-sky Lenovo Sky - haier-w66 Haier W66 - 3stack-hp HP machines with 3stack (Lucknow, Samba boards) - 6stack-dell Dell machines with 6stack (Inspiron 530) - mitac Mitac 8252D - clevo-m540r Clevo M540R (6ch + digital) - clevo-m720 Clevo M720 laptop series - fujitsu-pi2515 Fujitsu AMILO Pi2515 - fujitsu-xa3530 Fujitsu AMILO XA3530 - 3stack-6ch-intel Intel DG33* boards - intel-alc889a Intel IbexPeak with ALC889A - intel-x58 Intel DX58 with ALC889 - asus-p5q ASUS P5Q-EM boards - mb31 MacBook 3,1 - sony-vaio-tt Sony VAIO TT - auto auto-config reading BIOS (default) + N/A ALC861/660 ========== diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c deleted file mode 100644 index 0f4292688e1..00000000000 --- a/sound/pci/hda/alc882_quirks.c +++ /dev/null @@ -1,874 +0,0 @@ -/* - * ALC882/ALC883/ALC888/ALC889 quirk models - * included by patch_realtek.c - */ - -/* ALC882 models */ -enum { - ALC882_AUTO, - ALC885_MBA21, - ALC885_MBP3, - ALC885_MB5, - ALC885_MACMINI3, - ALC885_IMAC91, - ALC889A_MB31, - ALC882_MODEL_LAST, -}; - -#define ALC882_DIGOUT_NID 0x06 -#define ALC882_DIGIN_NID 0x0a -#define ALC883_DIGOUT_NID ALC882_DIGOUT_NID -#define ALC883_DIGIN_NID ALC882_DIGIN_NID -#define ALC1200_DIGOUT_NID 0x10 - - -static const struct hda_channel_mode alc882_ch_modes[1] = { - { 8, NULL } -}; - -/* DACs */ -static const hda_nid_t alc882_dac_nids[4] = { - /* front, rear, clfe, rear_surr */ - 0x02, 0x03, 0x04, 0x05 -}; -#define alc883_dac_nids alc882_dac_nids - -/* ADCs */ -static const hda_nid_t alc882_adc_nids[3] = { - /* ADC0-2 */ - 0x07, 0x08, 0x09, -}; - -static const hda_nid_t alc882_adc_nids_alt[2] = { - /* ADC1-2 */ - 0x08, 0x09, -}; - -#define alc883_adc_nids alc882_adc_nids_alt - -static const hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 }; -#define alc883_capsrc_nids alc882_capsrc_nids_alt - -/* input MUX */ -/* FIXME: should be a matrix-type input source selection */ - -static const struct hda_input_mux alc882_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -#define alc883_capture_source alc882_capture_source - -static const struct hda_input_mux mb5_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x1 }, - { "Line", 0x7 }, - { "CD", 0x4 }, - }, -}; - -static const struct hda_input_mux macmini3_capture_source = { - .num_items = 2, - .items = { - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -static const struct hda_input_mux alc883_3stack_6ch_intel = { - .num_items = 4, - .items = { - { "Mic", 0x1 }, - { "Front Mic", 0x0 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -static const struct hda_input_mux alc889A_mb31_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x0 }, - /* Front Mic (0x01) unused */ - { "Line", 0x2 }, - /* Line 2 (0x03) unused */ - /* CD (0x04) unused? */ - }, -}; - -static const struct hda_input_mux alc889A_imac91_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x01 }, - { "Line", 0x2 }, /* Not sure! */ - }, -}; - -/* Macbook Air 2,1 */ - -static const struct hda_channel_mode alc885_mba21_ch_modes[1] = { - { 2, NULL }, -}; - -/* - * macbook pro ALC885 can switch LineIn to LineOut without losing Mic - */ - -/* - * 2ch mode - */ -static const struct hda_verb alc885_mbp_ch2_init[] = { - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - { } /* end */ -}; - -/* - * 4ch mode - */ -static const struct hda_verb alc885_mbp_ch4_init[] = { - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - { } /* end */ -}; - -static const struct hda_channel_mode alc885_mbp_4ch_modes[2] = { - { 2, alc885_mbp_ch2_init }, - { 4, alc885_mbp_ch4_init }, -}; - -/* - * 2ch - * Speakers/Woofer/HP = Front - * LineIn = Input - */ -static const struct hda_verb alc885_mb5_ch2_init[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - { } /* end */ -}; - -/* - * 6ch mode - * Speakers/HP = Front - * Woofer = LFE - * LineIn = Surround - */ -static const struct hda_verb alc885_mb5_ch6_init[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - { } /* end */ -}; - -static const struct hda_channel_mode alc885_mb5_6ch_modes[2] = { - { 2, alc885_mb5_ch2_init }, - { 6, alc885_mb5_ch6_init }, -}; - -#define alc885_macmini3_6ch_modes alc885_mb5_6ch_modes - -/* Macbook Air 2,1 same control for HP and internal Speaker */ - -static const struct snd_kcontrol_new alc885_mba21_mixer[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_OUTPUT), - { } -}; - - -static const struct snd_kcontrol_new alc885_mbp3_mixer[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0e, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Headphone Playback Switch", 0x0e, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), - HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc885_mb5_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT), - HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0x00, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc885_macmini3_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT), - HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT), - HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc885_imac91_mixer[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), - { } /* end */ -}; - - -static const struct snd_kcontrol_new alc882_chmode_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - -static const struct hda_verb alc882_base_init_verbs[] = { - /* Front mixer: unmute input/output amp left and right (volume = 0) */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Rear mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* CLFE mixer */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Side mixer */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - - /* Front Pin: output 0 (0x0c) */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Rear Pin: output 1 (0x0d) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* CLFE Pin: output 2 (0x0e) */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* Side Pin: output 3 (0x0f) */ - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, - /* Mic (rear) pin: input vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Front Mic pin: input vref at 80% */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line In pin: input */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line-2 In: Headphone output (output 0 - 0x0c) */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* CD pin widget for input */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer2 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Input mixer3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* ADC2: mute amp left and right */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* ADC3: mute amp left and right */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - - { } -}; - -#define alc883_init_verbs alc882_base_init_verbs - -/* Macbook 5,1 */ -static const struct hda_verb alc885_mb5_init_verbs[] = { - /* DACs */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {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}, - /* Front mixer */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Surround mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* LFE mixer */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* HP mixer */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Front Pin (0x0c) */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* LFE Pin (0x0e) */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* HP Pin (0x0f) */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - /* Front Mic pin: input vref at 80% */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line In pin */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)}, - { } -}; - -/* Macmini 3,1 */ -static const struct hda_verb alc885_macmini3_init_verbs[] = { - /* DACs */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {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}, - /* Front mixer */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Surround mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* LFE mixer */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* HP mixer */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Front Pin (0x0c) */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* LFE Pin (0x0e) */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* HP Pin (0x0f) */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - /* Line In pin */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - { } -}; - - -static const struct hda_verb alc885_mba21_init_verbs[] = { - /*Internal and HP Speaker Mixer*/ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /*Internal Speaker Pin (0x0c)*/ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP Pin: output 0 (0x0e) */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)}, - /* Line in (is hp when jack connected)*/ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - { } - }; - - -/* Macbook Pro rev3 */ -static const struct hda_verb alc885_mbp3_init_verbs[] = { - /* Front mixer: unmute input/output amp left and right (volume = 0) */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Rear mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* HP mixer */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Front Pin: output 0 (0x0c) */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP Pin: output 0 (0x0e) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - /* Mic (rear) pin: input vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Front Mic pin: input vref at 80% */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line In pin: use output 1 when in LineOut mode */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, - - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Input mixer2 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Input mixer3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* ADC1: mute amp left and right */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* ADC2: mute amp left and right */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* ADC3: mute amp left and right */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - - { } -}; - -/* iMac 9,1 */ -static const struct hda_verb alc885_imac91_init_verbs[] = { - /* Internal Speaker Pin (0x0c) */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP Pin: Rear */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)}, - /* Line in Rear */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Front Mic pin: input vref at 80% */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Rear mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Line-Out mixer: unmute input/output amp left and right (volume = 0) */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - { } -}; - -/* Toggle speaker-output according to the hp-jack state */ -static void alc885_imac24_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x18; - spec->autocfg.speaker_pins[1] = 0x1a; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -#define alc885_mb5_setup alc885_imac24_setup -#define alc885_macmini3_setup alc885_imac24_setup - -/* Macbook Air 2,1 */ -static void alc885_mba21_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x18; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - - - -static void alc885_mbp3_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static void alc885_imac91_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x18; - spec->autocfg.speaker_pins[1] = 0x1a; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -/* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */ -static const struct hda_verb alc889A_mb31_ch2_init[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */ - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */ - { } /* end */ -}; - -/* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */ -static const struct hda_verb alc889A_mb31_ch4_init[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */ - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */ - { } /* end */ -}; - -/* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */ -static const struct hda_verb alc889A_mb31_ch5_init[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as rear */ - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */ - { } /* end */ -}; - -/* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */ -static const struct hda_verb alc889A_mb31_ch6_init[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as front */ - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Subwoofer off */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */ - { } /* end */ -}; - -static const struct hda_channel_mode alc889A_mb31_6ch_modes[4] = { - { 2, alc889A_mb31_ch2_init }, - { 4, alc889A_mb31_ch4_init }, - { 5, alc889A_mb31_ch5_init }, - { 6, alc889A_mb31_ch6_init }, -}; - -static const struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc889A_mb31_mixer[] = { - /* Output mixers */ - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x00, - HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x02, HDA_INPUT), - /* Output switches */ - HDA_CODEC_MUTE("Enable Speaker", 0x14, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("Enable Headphones", 0x15, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Enable LFE", 0x16, 2, 0x00, HDA_OUTPUT), - /* Boost mixers */ - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT), - /* Input mixers */ - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc883_chmode_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - -static const struct hda_verb alc889A_mb31_verbs[] = { - /* Init rear pin (used as headphone output) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, /* Apple Headphones */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Connect to front */ - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - /* Init line pin (used as output in 4ch and 6ch mode) */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect to CLFE */ - /* Init line 2 pin (used as headphone out by default) */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Use as input */ - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Mute output */ - { } /* end */ -}; - -/* Mute speakers according to the headphone jack state */ -static void alc889A_mb31_automute(struct hda_codec *codec) -{ - unsigned int present; - - /* Mute only in 2ch or 4ch mode */ - if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0) - == 0x00) { - present = snd_hda_jack_detect(codec, 0x15); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - } -} - -static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) -{ - if ((res >> 26) == ALC_HP_EVENT) - alc889A_mb31_automute(codec); -} - -static void alc882_unsol_event(struct hda_codec *codec, unsigned int res) -{ - alc_exec_unsol_event(codec, res >> 26); -} - -/* - * configuration and preset - */ -static const char * const alc882_models[ALC882_MODEL_LAST] = { - [ALC885_MB5] = "mb5", - [ALC885_MACMINI3] = "macmini3", - [ALC885_MBA21] = "mba21", - [ALC885_MBP3] = "mbp3", - [ALC885_IMAC91] = "imac91", - [ALC889A_MB31] = "mb31", - [ALC882_AUTO] = "auto", -}; - -/* codec SSID table for Intel Mac */ -static const struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { - SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC885_MBP3), - SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC885_MBP3), - SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC885_MBP3), - SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3), - SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889A_MB31), - SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC885_MBP3), - SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC885_MBA21), - SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31), - SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3), - SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC885_IMAC91), - SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5), - SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC885_MB5), - /* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2, - * so apparently no perfect solution yet - */ - SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), - SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), - SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC885_MACMINI3), - {} /* terminator */ -}; - -static const struct alc_config_preset alc882_presets[] = { - [ALC885_MBA21] = { - .mixers = { alc885_mba21_mixer }, - .init_verbs = { alc885_mba21_init_verbs, alc_gpio1_init_verbs }, - .num_dacs = 2, - .dac_nids = alc882_dac_nids, - .channel_mode = alc885_mba21_ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), - .input_mux = &alc882_capture_source, - .unsol_event = alc882_unsol_event, - .setup = alc885_mba21_setup, - .init_hook = alc_hp_automute, - }, - [ALC885_MBP3] = { - .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, - .init_verbs = { alc885_mbp3_init_verbs, - alc_gpio1_init_verbs }, - .num_dacs = 2, - .dac_nids = alc882_dac_nids, - .hp_nid = 0x04, - .channel_mode = alc885_mbp_4ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes), - .input_mux = &alc882_capture_source, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc882_unsol_event, - .setup = alc885_mbp3_setup, - .init_hook = alc_hp_automute, - }, - [ALC885_MB5] = { - .mixers = { alc885_mb5_mixer, alc882_chmode_mixer }, - .init_verbs = { alc885_mb5_init_verbs, - alc_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .channel_mode = alc885_mb5_6ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes), - .input_mux = &mb5_capture_source, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc882_unsol_event, - .setup = alc885_mb5_setup, - .init_hook = alc_hp_automute, - }, - [ALC885_MACMINI3] = { - .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer }, - .init_verbs = { alc885_macmini3_init_verbs, - alc_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .channel_mode = alc885_macmini3_6ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_macmini3_6ch_modes), - .input_mux = &macmini3_capture_source, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc882_unsol_event, - .setup = alc885_macmini3_setup, - .init_hook = alc_hp_automute, - }, - [ALC885_IMAC91] = { - .mixers = {alc885_imac91_mixer}, - .init_verbs = { alc885_imac91_init_verbs, - alc_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .channel_mode = alc885_mba21_ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), - .input_mux = &alc889A_imac91_capture_source, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc882_unsol_event, - .setup = alc885_imac91_setup, - .init_hook = alc_hp_automute, - }, - [ALC889A_MB31] = { - .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer}, - .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs, - alc_gpio1_init_verbs }, - .adc_nids = alc883_adc_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .capsrc_nids = alc883_capsrc_nids, - .dac_nids = alc883_dac_nids, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .channel_mode = alc889A_mb31_6ch_modes, - .num_channel_mode = ARRAY_SIZE(alc889A_mb31_6ch_modes), - .input_mux = &alc889A_mb31_capture_source, - .dig_out_nid = ALC883_DIGOUT_NID, - .unsol_event = alc889A_mb31_unsol_event, - .init_hook = alc889A_mb31_automute, - }, -}; - - diff --git a/sound/pci/hda/alc_quirks.c b/sound/pci/hda/alc_quirks.c deleted file mode 100644 index a63a517780d..00000000000 --- a/sound/pci/hda/alc_quirks.c +++ /dev/null @@ -1,167 +0,0 @@ -/* - * Common codes for Realtek codec quirks - * included by patch_realtek.c - */ - -/* - * configuration template - to be copied to the spec instance - */ -struct alc_config_preset { - const struct snd_kcontrol_new *mixers[5]; /* should be identical size - * with spec - */ - const struct snd_kcontrol_new *cap_mixer; /* capture mixer */ - const struct hda_verb *init_verbs[5]; - unsigned int num_dacs; - const hda_nid_t *dac_nids; - hda_nid_t dig_out_nid; /* optional */ - hda_nid_t hp_nid; /* optional */ - const hda_nid_t *slave_dig_outs; - unsigned int num_adc_nids; - const hda_nid_t *adc_nids; - const hda_nid_t *capsrc_nids; - hda_nid_t dig_in_nid; - unsigned int num_channel_mode; - const struct hda_channel_mode *channel_mode; - int need_dac_fix; - int const_channel_count; - unsigned int num_mux_defs; - const struct hda_input_mux *input_mux; - void (*unsol_event)(struct hda_codec *, unsigned int); - void (*setup)(struct hda_codec *); - void (*init_hook)(struct hda_codec *); -#ifdef CONFIG_SND_HDA_POWER_SAVE - const struct hda_amp_list *loopbacks; - void (*power_hook)(struct hda_codec *codec); -#endif -}; - -/* - * channel mode setting - */ -static int alc_ch_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode, - spec->num_channel_mode); -} - -static int alc_ch_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - spec->ext_channel_count); -} - -static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - &spec->ext_channel_count); - if (err >= 0 && !spec->const_channel_count) { - spec->multiout.max_channels = spec->ext_channel_count; - if (spec->need_dac_fix) - spec->multiout.num_dacs = spec->multiout.max_channels / 2; - } - return err; -} - -static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - - if (!cfg->line_outs) { - while (cfg->line_outs < AUTO_CFG_MAX_OUTS && - cfg->line_out_pins[cfg->line_outs]) - cfg->line_outs++; - } - if (!cfg->speaker_outs) { - while (cfg->speaker_outs < AUTO_CFG_MAX_OUTS && - cfg->speaker_pins[cfg->speaker_outs]) - cfg->speaker_outs++; - } - if (!cfg->hp_outs) { - while (cfg->hp_outs < AUTO_CFG_MAX_OUTS && - cfg->hp_pins[cfg->hp_outs]) - cfg->hp_outs++; - } -} - -/* - * set up from the preset table - */ -static void setup_preset(struct hda_codec *codec, - const struct alc_config_preset *preset) -{ - struct alc_spec *spec = codec->spec; - int i; - - for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++) - add_mixer(spec, preset->mixers[i]); - spec->cap_mixer = preset->cap_mixer; - for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i]; - i++) - add_verb(spec, preset->init_verbs[i]); - - spec->channel_mode = preset->channel_mode; - spec->num_channel_mode = preset->num_channel_mode; - spec->need_dac_fix = preset->need_dac_fix; - spec->const_channel_count = preset->const_channel_count; - - if (preset->const_channel_count) - spec->multiout.max_channels = preset->const_channel_count; - else - spec->multiout.max_channels = spec->channel_mode[0].channels; - spec->ext_channel_count = spec->channel_mode[0].channels; - - spec->multiout.num_dacs = preset->num_dacs; - spec->multiout.dac_nids = preset->dac_nids; - spec->multiout.dig_out_nid = preset->dig_out_nid; - spec->multiout.slave_dig_outs = preset->slave_dig_outs; - spec->multiout.hp_nid = preset->hp_nid; - - spec->num_mux_defs = preset->num_mux_defs; - if (!spec->num_mux_defs) - spec->num_mux_defs = 1; - spec->input_mux = preset->input_mux; - - spec->num_adc_nids = preset->num_adc_nids; - spec->adc_nids = preset->adc_nids; - spec->capsrc_nids = preset->capsrc_nids; - spec->dig_in_nid = preset->dig_in_nid; - - spec->unsol_event = preset->unsol_event; - spec->init_hook = preset->init_hook; -#ifdef CONFIG_SND_HDA_POWER_SAVE - spec->power_hook = preset->power_hook; - spec->loopback.amplist = preset->loopbacks; -#endif - - if (preset->setup) - preset->setup(codec); - - alc_fixup_autocfg_pin_nums(codec); -} - -static void alc_simple_setup_automute(struct alc_spec *spec, int mode) -{ - int lo_pin = spec->autocfg.line_out_pins[0]; - - if (lo_pin == spec->autocfg.speaker_pins[0] || - lo_pin == spec->autocfg.hp_pins[0]) - lo_pin = 0; - spec->automute_mode = mode; - spec->detect_hp = !!spec->autocfg.hp_pins[0]; - spec->detect_lo = !!lo_pin; - spec->automute_lo = spec->automute_lo_possible = !!lo_pin; - spec->automute_speaker = spec->automute_speaker_possible = !!spec->autocfg.speaker_pins[0]; -} diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3c6f5b5161f..c5216b58d21 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4807,21 +4807,6 @@ static const struct alc_model_fixup alc880_fixup_models[] = { }; -/* - * board setups - */ -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS -#define alc_board_config \ - snd_hda_check_board_config -#define alc_board_codec_sid_config \ - snd_hda_check_board_codec_sid_config -#include "alc_quirks.c" -#else -#define alc_board_config(codec, nums, models, tbl) -1 -#define alc_board_codec_sid_config(codec, nums, models, tbl) -1 -#define setup_preset(codec, x) /* NOP */ -#endif - /* * OK, here we have finally the patch for ALC880 */ @@ -5091,6 +5076,8 @@ enum { ALC882_FIXUP_EAPD, ALC883_FIXUP_EAPD, ALC883_FIXUP_ACER_EAPD, + ALC882_FIXUP_GPIO1, + ALC882_FIXUP_GPIO2, ALC882_FIXUP_GPIO3, ALC889_FIXUP_COEF, ALC882_FIXUP_ASUS_W2JC, @@ -5099,6 +5086,8 @@ enum { ALC882_FIXUP_ASPIRE_8930G_VERBS, ALC885_FIXUP_MACPRO_GPIO, ALC889_FIXUP_DAC_ROUTE, + ALC889_FIXUP_MBP_VREF, + ALC889_FIXUP_IMAC91_VREF, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -5169,6 +5158,51 @@ static void alc889_fixup_dac_route(struct hda_codec *codec, } } +/* Set VREF on HP pin */ +static void alc889_fixup_mbp_vref(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static hda_nid_t nids[2] = { 0x14, 0x15 }; + int i; + + if (action != ALC_FIXUP_ACT_INIT) + return; + for (i = 0; i < ARRAY_SIZE(nids); i++) { + unsigned int val = snd_hda_codec_get_pincfg(codec, nids[i]); + if (get_defcfg_device(val) != AC_JACK_HP_OUT) + continue; + val = snd_hda_codec_read(codec, nids[i], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + val |= AC_PINCTL_VREF_80; + snd_hda_codec_write(codec, nids[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, val); + spec->keep_vref_in_automute = 1; + break; + } +} + +/* Set VREF on speaker pins on imac91 */ +static void alc889_fixup_imac91_vref(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static hda_nid_t nids[2] = { 0x18, 0x1a }; + int i; + + if (action != ALC_FIXUP_ACT_INIT) + return; + for (i = 0; i < ARRAY_SIZE(nids); i++) { + unsigned int val; + val = snd_hda_codec_read(codec, nids[i], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + val |= AC_PINCTL_VREF_50; + snd_hda_codec_write(codec, nids[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, val); + } + spec->keep_vref_in_automute = 1; +} + static const struct alc_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = ALC_FIXUP_PINS, @@ -5247,6 +5281,14 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [ALC882_FIXUP_GPIO1] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = alc_gpio1_init_verbs, + }, + [ALC882_FIXUP_GPIO2] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = alc_gpio2_init_verbs, + }, [ALC882_FIXUP_GPIO3] = { .type = ALC_FIXUP_VERBS, .v.verbs = alc_gpio3_init_verbs, @@ -5320,6 +5362,18 @@ static const struct alc_fixup alc882_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc889_fixup_dac_route, }, + [ALC889_FIXUP_MBP_VREF] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc889_fixup_mbp_vref, + .chained = true, + .chain_id = ALC882_FIXUP_GPIO1, + }, + [ALC889_FIXUP_IMAC91_VREF] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc889_fixup_imac91_vref, + .chained = true, + .chain_id = ALC882_FIXUP_GPIO1, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -5353,11 +5407,26 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), /* All Apple entries are in codec SSIDs */ + SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD), + SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), @@ -5382,14 +5451,10 @@ static int alc882_parse_auto_config(struct hda_codec *codec) /* */ -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS -#include "alc882_quirks.c" -#endif - static int patch_alc882(struct hda_codec *codec) { struct alc_spec *spec; - int err, board_config; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -5413,36 +5478,15 @@ static int patch_alc882(struct hda_codec *codec) if (err < 0) goto error; - board_config = alc_board_config(codec, ALC882_MODEL_LAST, - alc882_models, NULL); - if (board_config < 0) - board_config = alc_board_codec_sid_config(codec, - ALC882_MODEL_LAST, alc882_models, alc882_ssid_cfg_tbl); - - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = ALC_MODEL_AUTO; - } - - if (board_config == ALC_MODEL_AUTO) { - alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - } + alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); - if (board_config == ALC_MODEL_AUTO) { - /* automatic parse from the BIOS config */ - err = alc882_parse_auto_config(codec); - if (err < 0) - goto error; - } - - if (board_config != ALC_MODEL_AUTO) { - setup_preset(codec, &alc882_presets[board_config]); - spec->vmaster_nid = 0x0c; - } + /* automatic parse from the BIOS config */ + err = alc882_parse_auto_config(codec); + if (err < 0) + goto error; if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -5461,10 +5505,7 @@ static int patch_alc882(struct hda_codec *codec) } codec->patch_ops = alc_patch_ops; - if (board_config == ALC_MODEL_AUTO) - spec->init_hook = alc_auto_init_std; - else - codec->patch_ops.build_controls = __alc_build_controls; + spec->init_hook = alc_auto_init_std; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) -- cgit v1.2.3-70-g09d2 From a6f2fd557f993aecc93d51afd9e339524107937f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Feb 2012 11:58:40 +0100 Subject: ALSA: hda - Add position_fix=4 (COMBO) option This patch adds a new position_fix option value, 4, as a combo mode to use LPIB for playbacks and POSBUF for captures. It's the way recommended by Intel hardware guys. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 8 +++++++- Documentation/sound/alsa/HD-Audio.txt | 7 ++++++- sound/pci/hda/hda_intel.c | 10 +++++++++- 3 files changed, 22 insertions(+), 3 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 936699e4f04..9af64c508ab 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -860,7 +860,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. [Multiple options for each card instance] model - force the model name - position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF) + position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF, + 3 = VIACOMBO, 4 = COMBO) probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) When the bit 8 (0x100) is set, the lower 8 bits are used as the "fixed" codec slots; i.e. the driver probes the @@ -925,6 +926,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. (Usually SD_LPIB register is more accurate than the position buffer.) + position_fix=3 is specific to VIA devices. The position + of the capture stream is checked from both LPIB and POSBUF + values. position_fix=4 is a combination mode, using LPIB + for playback and POSBUF for capture. + NB: If you get many "azx_get_response timeout" messages at loading, it's likely a problem of interrupts (e.g. ACPI irq routing). Try to boot with options like "pci=noacpi". Also, you diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 91fee3b45fb..7813c06a5c7 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -59,7 +59,12 @@ a case, you can change the default method via `position_fix` option. `position_fix=1` means to use LPIB method explicitly. `position_fix=2` means to use the position-buffer. `position_fix=3` means to use a combination of both methods, needed -for some VIA and ATI controllers. 0 is the default value for all other +for some VIA controllers. The capture stream position is corrected +by comparing both LPIB and position-buffer values. +`position_fix=4` is another combination available for all controllers, +and uses LPIB for the playback and the position-buffer for the capture +streams. +0 is the default value for all other controllers, the automatic check and fallback to LPIB as described in the above. If you get a problem of repeated sounds, this option might help. diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e354c161654..6e958bf9419 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -84,7 +84,7 @@ module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "DMA pointer read method." - "(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO)."); + "(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO)."); module_param_array(bdl_pos_adj, int, NULL, 0644); MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); @@ -330,6 +330,7 @@ enum { POS_FIX_LPIB, POS_FIX_POSBUF, POS_FIX_VIACOMBO, + POS_FIX_COMBO, }; /* Defines for ATI HD Audio support in SB450 south bridge */ @@ -2520,6 +2521,7 @@ static int __devinit check_position_fix(struct azx *chip, int fix) case POS_FIX_LPIB: case POS_FIX_POSBUF: case POS_FIX_VIACOMBO: + case POS_FIX_COMBO: return fix; } @@ -2699,6 +2701,12 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->position_fix[0] = chip->position_fix[1] = check_position_fix(chip, position_fix[dev]); + /* combo mode uses LPIB for playback */ + if (chip->position_fix[0] == POS_FIX_COMBO) { + chip->position_fix[0] = POS_FIX_LPIB; + chip->position_fix[1] = POS_FIX_AUTO; + } + check_probe_mask(chip, dev); chip->single_cmd = single_cmd; -- cgit v1.2.3-70-g09d2 From 970e2486492aa1eb47a436a5a4c81e92558986a9 Mon Sep 17 00:00:00 2001 From: Lucas De Marchi Date: Fri, 30 Mar 2012 13:37:16 -0700 Subject: Documentation: remove references to /etc/modprobe.conf Usage of /etc/modprobe.conf file was deprecated by module-init-tools and is no longer parsed by new kmod tool. References to this file are replaced in Documentation, comments and Kconfig according to the context. There are also some references to the old /etc/modules.conf from 2.4 kernels that are being removed. Signed-off-by: Lucas De Marchi Acked-by: Takashi Iwai Acked-by: Mauro Carvalho Chehab Signed-off-by: Randy Dunlap Signed-off-by: Linus Torvalds --- Documentation/aoe/aoe.txt | 2 +- Documentation/aoe/autoload.sh | 4 +-- Documentation/blockdev/floppy.txt | 2 +- Documentation/fb/intel810.txt | 2 +- Documentation/fb/intelfb.txt | 2 +- Documentation/i2c/busses/scx200_acb | 2 +- Documentation/ide/ide.txt | 2 +- Documentation/isdn/README.gigaset | 16 ++++----- Documentation/laptops/sonypi.txt | 2 +- Documentation/mono.txt | 8 ++--- Documentation/networking/baycom.txt | 2 +- Documentation/networking/bonding.txt | 43 +++++++++++-------------- Documentation/networking/dl2k.txt | 11 ++++--- Documentation/networking/e100.txt | 6 ++-- Documentation/networking/ipv6.txt | 6 ++-- Documentation/networking/ixgb.txt | 6 ++-- Documentation/networking/ltpc.txt | 2 +- Documentation/networking/vortex.txt | 6 ++-- Documentation/parport.txt | 13 ++++---- Documentation/s390/3270.txt | 21 ++++++------ Documentation/scsi/aic79xx.txt | 2 +- Documentation/scsi/aic7xxx.txt | 2 +- Documentation/scsi/osst.txt | 2 +- Documentation/serial/computone.txt | 8 ++--- Documentation/serial/rocket.txt | 2 +- Documentation/serial/stallion.txt | 4 +-- Documentation/sound/alsa/ALSA-Configuration.txt | 10 +++--- Documentation/sound/alsa/Audiophile-Usb.txt | 4 +-- Documentation/sound/alsa/MIXART.txt | 6 ++-- Documentation/sound/alsa/OSS-Emulation.txt | 2 +- Documentation/sound/oss/AudioExcelDSP16 | 6 ++-- Documentation/sound/oss/CMI8330 | 5 ++- Documentation/sound/oss/Introduction | 10 +++--- Documentation/sound/oss/Opti | 8 ++--- Documentation/sound/oss/PAS16 | 4 +-- Documentation/sound/oss/README.modules | 8 ++--- Documentation/usb/power-management.txt | 3 +- Documentation/video4linux/CQcam.txt | 14 ++------ Documentation/video4linux/Zoran | 2 +- Documentation/video4linux/bttv/Modules.conf | 2 +- Documentation/video4linux/meye.txt | 2 +- drivers/net/wan/Kconfig | 4 +-- drivers/scsi/aic7xxx/aic79xx_osm.c | 8 ++--- drivers/scsi/aic7xxx/aic7xxx_osm.c | 8 ++--- drivers/staging/asus_oled/README | 2 +- drivers/tty/isicom.c | 2 +- drivers/usb/serial/ftdi_sio.c | 3 +- drivers/usb/storage/Kconfig | 2 +- sound/core/seq/seq_dummy.c | 2 +- sound/drivers/Kconfig | 3 +- 50 files changed, 138 insertions(+), 160 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/aoe/aoe.txt b/Documentation/aoe/aoe.txt index b5aada9f20c..5f5aa16047f 100644 --- a/Documentation/aoe/aoe.txt +++ b/Documentation/aoe/aoe.txt @@ -35,7 +35,7 @@ CREATING DEVICE NODES sh Documentation/aoe/mkshelf.sh /dev/etherd 0 There is also an autoload script that shows how to edit - /etc/modprobe.conf to ensure that the aoe module is loaded when + /etc/modprobe.d/aoe.conf to ensure that the aoe module is loaded when necessary. USING DEVICE NODES diff --git a/Documentation/aoe/autoload.sh b/Documentation/aoe/autoload.sh index 78dad1334c6..815dff4691c 100644 --- a/Documentation/aoe/autoload.sh +++ b/Documentation/aoe/autoload.sh @@ -1,8 +1,8 @@ #!/bin/sh # set aoe to autoload by installing the -# aliases in /etc/modprobe.conf +# aliases in /etc/modprobe.d/ -f=/etc/modprobe.conf +f=/etc/modprobe.d/aoe.conf if test ! -r $f || test ! -w $f; then echo "cannot configure $f for module autoloading" 1>&2 diff --git a/Documentation/blockdev/floppy.txt b/Documentation/blockdev/floppy.txt index 6ccab88705c..470fe4b5e37 100644 --- a/Documentation/blockdev/floppy.txt +++ b/Documentation/blockdev/floppy.txt @@ -49,7 +49,7 @@ you can put: options floppy omnibook messages -in /etc/modprobe.conf. +in a configuration file in /etc/modprobe.d/. The floppy driver related options are: diff --git a/Documentation/fb/intel810.txt b/Documentation/fb/intel810.txt index be3e7836abe..a8e9f5bca6f 100644 --- a/Documentation/fb/intel810.txt +++ b/Documentation/fb/intel810.txt @@ -211,7 +211,7 @@ Using the same setup as described above, load the module like this: modprobe i810fb vram=2 xres=1024 bpp=8 hsync1=30 hsync2=55 vsync1=50 \ vsync2=85 accel=1 mtrr=1 -Or just add the following to /etc/modprobe.conf +Or just add the following to a configuration file in /etc/modprobe.d/ options i810fb vram=2 xres=1024 bpp=16 hsync1=30 hsync2=55 vsync1=50 \ vsync2=85 accel=1 mtrr=1 diff --git a/Documentation/fb/intelfb.txt b/Documentation/fb/intelfb.txt index dd9e944ea62..feac4e4d696 100644 --- a/Documentation/fb/intelfb.txt +++ b/Documentation/fb/intelfb.txt @@ -120,7 +120,7 @@ Using the same setup as described above, load the module like this: modprobe intelfb mode=800x600-32@75 vram=8 accel=1 hwcursor=1 -Or just add the following to /etc/modprobe.conf +Or just add the following to a configuration file in /etc/modprobe.d/ options intelfb mode=800x600-32@75 vram=8 accel=1 hwcursor=1 diff --git a/Documentation/i2c/busses/scx200_acb b/Documentation/i2c/busses/scx200_acb index 7c07883d4df..ce83c871fe9 100644 --- a/Documentation/i2c/busses/scx200_acb +++ b/Documentation/i2c/busses/scx200_acb @@ -28,5 +28,5 @@ If the scx200_acb driver is built into the kernel, add the following parameter to your boot command line: scx200_acb.base=0x810,0x820 If the scx200_acb driver is built as a module, add the following line to -the file /etc/modprobe.conf instead: +a configuration file in /etc/modprobe.d/ instead: options scx200_acb base=0x810,0x820 diff --git a/Documentation/ide/ide.txt b/Documentation/ide/ide.txt index e77bebfa7b0..7aca987c23d 100644 --- a/Documentation/ide/ide.txt +++ b/Documentation/ide/ide.txt @@ -169,7 +169,7 @@ When using ide.c as a module in combination with kmod, add: alias block-major-3 ide-probe -to /etc/modprobe.conf. +to a configuration file in /etc/modprobe.d/. When ide.c is used as a module, you can pass command line parameters to the driver using the "options=" keyword to insmod, while replacing any ',' with diff --git a/Documentation/isdn/README.gigaset b/Documentation/isdn/README.gigaset index ef3343eaa00..7534c6039ad 100644 --- a/Documentation/isdn/README.gigaset +++ b/Documentation/isdn/README.gigaset @@ -97,8 +97,7 @@ GigaSet 307x Device Driver 2.5.): 1=on (default), 0=off Depending on your distribution you may want to create a separate module - configuration file /etc/modprobe.d/gigaset for these, or add them to a - custom file like /etc/modprobe.conf.local. + configuration file like /etc/modprobe.d/gigaset.conf for these. 2.2. Device nodes for user space programs ------------------------------------ @@ -212,8 +211,8 @@ GigaSet 307x Device Driver options ppp_async flag_time=0 - to an appropriate module configuration file, like /etc/modprobe.d/gigaset - or /etc/modprobe.conf.local. + to an appropriate module configuration file, like + /etc/modprobe.d/gigaset.conf. Unimodem mode is needed for making some devices [e.g. SX100] work which do not support the regular Gigaset command set. If debug output (see @@ -237,8 +236,8 @@ GigaSet 307x Device Driver modprobe usb_gigaset startmode=0 or by adding a line like options usb_gigaset startmode=0 - to an appropriate module configuration file, like /etc/modprobe.d/gigaset - or /etc/modprobe.conf.local. + to an appropriate module configuration file, like + /etc/modprobe.d/gigaset.conf 2.6. Call-ID (CID) mode ------------------ @@ -310,7 +309,7 @@ GigaSet 307x Device Driver options isdn dialtimeout=15 - to /etc/modprobe.d/gigaset, /etc/modprobe.conf.local or a similar file. + to /etc/modprobe.d/gigaset.conf or a similar file. Problem: The isdnlog program emits error messages or just doesn't work. @@ -350,8 +349,7 @@ GigaSet 307x Device Driver The initial value can be set using the debug parameter when loading the module "gigaset", e.g. by adding a line options gigaset debug=0 - to your module configuration file, eg. /etc/modprobe.d/gigaset or - /etc/modprobe.conf.local. + to your module configuration file, eg. /etc/modprobe.d/gigaset.conf Generated debugging information can be found - as output of the command diff --git a/Documentation/laptops/sonypi.txt b/Documentation/laptops/sonypi.txt index 4857acfc50f..606bdb9ce03 100644 --- a/Documentation/laptops/sonypi.txt +++ b/Documentation/laptops/sonypi.txt @@ -110,7 +110,7 @@ Module use: ----------- In order to automatically load the sonypi module on use, you can put those -lines in your /etc/modprobe.conf file: +lines a configuration file in /etc/modprobe.d/: alias char-major-10-250 sonypi options sonypi minor=250 diff --git a/Documentation/mono.txt b/Documentation/mono.txt index e8e1758e87d..d01ac605219 100644 --- a/Documentation/mono.txt +++ b/Documentation/mono.txt @@ -38,11 +38,11 @@ if [ ! -e /proc/sys/fs/binfmt_misc/register ]; then /sbin/modprobe binfmt_misc # Some distributions, like Fedora Core, perform # the following command automatically when the - # binfmt_misc module is loaded into the kernel. + # binfmt_misc module is loaded into the kernel + # or during normal boot up (systemd-based systems). # Thus, it is possible that the following line - # is not needed at all. Look at /etc/modprobe.conf - # to check whether this is applicable or not. - mount -t binfmt_misc none /proc/sys/fs/binfmt_misc + # is not needed at all. + mount -t binfmt_misc none /proc/sys/fs/binfmt_misc fi # Register support for .NET CLR binaries diff --git a/Documentation/networking/baycom.txt b/Documentation/networking/baycom.txt index 4e68849d563..688f18fd446 100644 --- a/Documentation/networking/baycom.txt +++ b/Documentation/networking/baycom.txt @@ -93,7 +93,7 @@ Every time a driver is inserted into the kernel, it has to know which modems it should access at which ports. This can be done with the setbaycom utility. If you are only using one modem, you can also configure the driver from the insmod command line (or by means of an option line in -/etc/modprobe.conf). +/etc/modprobe.d/*.conf). Examples: modprobe baycom_ser_fdx mode="ser12*" iobase=0x3f8 irq=4 diff --git a/Documentation/networking/bonding.txt b/Documentation/networking/bonding.txt index 080ad26690a..d5e86981404 100644 --- a/Documentation/networking/bonding.txt +++ b/Documentation/networking/bonding.txt @@ -173,9 +173,8 @@ bonding module at load time, or are specified via sysfs. Module options may be given as command line arguments to the insmod or modprobe command, but are usually specified in either the -/etc/modules.conf or /etc/modprobe.conf configuration file, or in a -distro-specific configuration file (some of which are detailed in the next -section). +/etc/modrobe.d/*.conf configuration files, or in a distro-specific +configuration file (some of which are detailed in the next section). Details on bonding support for sysfs is provided in the "Configuring Bonding Manually via Sysfs" section, below. @@ -1021,7 +1020,7 @@ ifcfg-bondX files. Because the sysconfig scripts supply the bonding module options in the ifcfg-bondX file, it is not necessary to add them to -the system /etc/modules.conf or /etc/modprobe.conf configuration file. +the system /etc/modules.d/*.conf configuration files. 3.2 Configuration with Initscripts Support ------------------------------------------ @@ -1098,15 +1097,13 @@ queried targets, e.g., arp_ip_target=+192.168.1.1 arp_ip_target=+192.168.1.2 is the proper syntax to specify multiple targets. When specifying -options via BONDING_OPTS, it is not necessary to edit /etc/modules.conf or -/etc/modprobe.conf. +options via BONDING_OPTS, it is not necessary to edit /etc/modprobe.d/*.conf. For even older versions of initscripts that do not support -BONDING_OPTS, it is necessary to edit /etc/modules.conf (or -/etc/modprobe.conf, depending upon your distro) to load the bonding module -with your desired options when the bond0 interface is brought up. The -following lines in /etc/modules.conf (or modprobe.conf) will load the -bonding module, and select its options: +BONDING_OPTS, it is necessary to edit /etc/modprobe.d/*.conf, depending upon +your distro) to load the bonding module with your desired options when the +bond0 interface is brought up. The following lines in /etc/modprobe.d/*.conf +will load the bonding module, and select its options: alias bond0 bonding options bond0 mode=balance-alb miimon=100 @@ -1152,7 +1149,7 @@ knowledge of bonding. One such distro is SuSE Linux Enterprise Server version 8. The general method for these systems is to place the bonding -module parameters into /etc/modules.conf or /etc/modprobe.conf (as +module parameters into a config file in /etc/modprobe.d/ (as appropriate for the installed distro), then add modprobe and/or ifenslave commands to the system's global init script. The name of the global init script differs; for sysconfig, it is @@ -1228,7 +1225,7 @@ network initialization scripts. specify a different name for each instance (the module loading system requires that every loaded module, even multiple instances of the same module, have a unique name). This is accomplished by supplying multiple -sets of bonding options in /etc/modprobe.conf, for example: +sets of bonding options in /etc/modprobe.d/*.conf, for example: alias bond0 bonding options bond0 -o bond0 mode=balance-rr miimon=100 @@ -1793,8 +1790,8 @@ route additions may cause trouble. On systems with network configuration scripts that do not associate physical devices directly with network interface names (so that the same physical device always has the same "ethX" name), it may -be necessary to add some special logic to either /etc/modules.conf or -/etc/modprobe.conf (depending upon which is installed on the system). +be necessary to add some special logic to config files in +/etc/modprobe.d/. For example, given a modules.conf containing the following: @@ -1821,20 +1818,16 @@ add above bonding e1000 tg3 bonding is loaded. This command is fully documented in the modules.conf manual page. - On systems utilizing modprobe.conf (or modprobe.conf.local), -an equivalent problem can occur. In this case, the following can be -added to modprobe.conf (or modprobe.conf.local, as appropriate), as -follows (all on one line; it has been split here for clarity): + On systems utilizing modprobe an equivalent problem can occur. +In this case, the following can be added to config files in +/etc/modprobe.d/ as: install bonding /sbin/modprobe tg3; /sbin/modprobe e1000; /sbin/modprobe --ignore-install bonding - This will, when loading the bonding module, rather than -performing the normal action, instead execute the provided command. -This command loads the device drivers in the order needed, then calls -modprobe with --ignore-install to cause the normal action to then take -place. Full documentation on this can be found in the modprobe.conf -and modprobe manual pages. + This will load tg3 and e1000 modules before loading the bonding one. +Full documentation on this can be found in the modprobe.d and modprobe +manual pages. 8.3. Painfully Slow Or No Failed Link Detection By Miimon --------------------------------------------------------- diff --git a/Documentation/networking/dl2k.txt b/Documentation/networking/dl2k.txt index 10e8490fa40..cba74f7a3ab 100644 --- a/Documentation/networking/dl2k.txt +++ b/Documentation/networking/dl2k.txt @@ -45,12 +45,13 @@ Now eth0 should active, you can test it by "ping" or get more information by "ifconfig". If tested ok, continue the next step. 4. cp dl2k.ko /lib/modules/`uname -r`/kernel/drivers/net -5. Add the following line to /etc/modprobe.conf: +5. Add the following line to /etc/modprobe.d/dl2k.conf: alias eth0 dl2k -6. Run "netconfig" or "netconf" to create configuration script ifcfg-eth0 +6. Run depmod to updated module indexes. +7. Run "netconfig" or "netconf" to create configuration script ifcfg-eth0 located at /etc/sysconfig/network-scripts or create it manually. [see - Configuration Script Sample] -7. Driver will automatically load and configure at next boot time. +8. Driver will automatically load and configure at next boot time. Compiling the Driver ==================== @@ -154,8 +155,8 @@ Installing the Driver ----------------- 1. Copy dl2k.o to the network modules directory, typically /lib/modules/2.x.x-xx/net or /lib/modules/2.x.x/kernel/drivers/net. - 2. Locate the boot module configuration file, most commonly modprobe.conf - or modules.conf (for 2.4) in the /etc directory. Add the following lines: + 2. Locate the boot module configuration file, most commonly in the + /etc/modprobe.d/ directory. Add the following lines: alias ethx dl2k options dl2k diff --git a/Documentation/networking/e100.txt b/Documentation/networking/e100.txt index 162f323a7a1..fcb6c71cdb6 100644 --- a/Documentation/networking/e100.txt +++ b/Documentation/networking/e100.txt @@ -94,8 +94,8 @@ Additional Configurations Configuring a network driver to load properly when the system is started is distribution dependent. Typically, the configuration process involves adding - an alias line to /etc/modules.conf or /etc/modprobe.conf as well as editing - other system startup scripts and/or configuration files. Many popular Linux + an alias line to /etc/modprobe.d/*.conf as well as editing other system + startup scripts and/or configuration files. Many popular Linux distributions ship with tools to make these changes for you. To learn the proper way to configure a network device for your system, refer to your distribution documentation. If during this process you are asked for the @@ -103,7 +103,7 @@ Additional Configurations PRO/100 Family of Adapters is e100. As an example, if you install the e100 driver for two PRO/100 adapters - (eth0 and eth1), add the following to modules.conf or modprobe.conf: + (eth0 and eth1), add the following to a configuraton file in /etc/modprobe.d/ alias eth0 e100 alias eth1 e100 diff --git a/Documentation/networking/ipv6.txt b/Documentation/networking/ipv6.txt index 9fd7e21296c..6cd74fa5535 100644 --- a/Documentation/networking/ipv6.txt +++ b/Documentation/networking/ipv6.txt @@ -2,9 +2,9 @@ Options for the ipv6 module are supplied as parameters at load time. Module options may be given as command line arguments to the insmod -or modprobe command, but are usually specified in either the -/etc/modules.conf or /etc/modprobe.conf configuration file, or in a -distro-specific configuration file. +or modprobe command, but are usually specified in either +/etc/modules.d/*.conf configuration files, or in a distro-specific +configuration file. The available ipv6 module parameters are listed below. If a parameter is not specified the default value is used. diff --git a/Documentation/networking/ixgb.txt b/Documentation/networking/ixgb.txt index e196f16df31..d75a1f9565b 100644 --- a/Documentation/networking/ixgb.txt +++ b/Documentation/networking/ixgb.txt @@ -274,9 +274,9 @@ Additional Configurations ------------------------------------------------- Configuring a network driver to load properly when the system is started is distribution dependent. Typically, the configuration process involves adding - an alias line to /etc/modprobe.conf as well as editing other system startup - scripts and/or configuration files. Many popular Linux distributions ship - with tools to make these changes for you. To learn the proper way to + an alias line to files in /etc/modprobe.d/ as well as editing other system + startup scripts and/or configuration files. Many popular Linux distributions + ship with tools to make these changes for you. To learn the proper way to configure a network device for your system, refer to your distribution documentation. If during this process you are asked for the driver or module name, the name for the Linux Base Driver for the Intel 10GbE Family of diff --git a/Documentation/networking/ltpc.txt b/Documentation/networking/ltpc.txt index fe2a9129d95..0bf3220c715 100644 --- a/Documentation/networking/ltpc.txt +++ b/Documentation/networking/ltpc.txt @@ -25,7 +25,7 @@ the driver will try to determine them itself. If you load the driver as a module, you can pass the parameters "io=", "irq=", and "dma=" on the command line with insmod or modprobe, or add -them as options in /etc/modprobe.conf: +them as options in a configuration file in /etc/modprobe.d/ directory: alias lt0 ltpc # autoload the module when the interface is configured options ltpc io=0x240 irq=9 dma=1 diff --git a/Documentation/networking/vortex.txt b/Documentation/networking/vortex.txt index bd70976b816..b4038ffb3bc 100644 --- a/Documentation/networking/vortex.txt +++ b/Documentation/networking/vortex.txt @@ -67,8 +67,8 @@ Module parameters ================= There are several parameters which may be provided to the driver when -its module is loaded. These are usually placed in /etc/modprobe.conf -(/etc/modules.conf in 2.4). Example: +its module is loaded. These are usually placed in /etc/modprobe.d/*.conf +configuretion files. Example: options 3c59x debug=3 rx_copybreak=300 @@ -425,7 +425,7 @@ steps you should take: 1) Increase the debug level. Usually this is done via: a) modprobe driver debug=7 - b) In /etc/modprobe.conf (or /etc/modules.conf for 2.4): + b) In /etc/modprobe.d/driver.conf: options driver debug=7 2) Recreate the problem with the higher debug level, diff --git a/Documentation/parport.txt b/Documentation/parport.txt index 93a7ceef398..c208e4366c0 100644 --- a/Documentation/parport.txt +++ b/Documentation/parport.txt @@ -36,18 +36,17 @@ addresses should not be specified for supported PCI cards since they are automatically detected. -KMod ----- +modprobe +-------- -If you use kmod, you will find it useful to edit /etc/modprobe.conf. -Here is an example of the lines that need to be added: +If you use modprobe , you will find it useful to add lines as below to a +configuration file in /etc/modprobe.d/ directory:. alias parport_lowlevel parport_pc options parport_pc io=0x378,0x278 irq=7,auto -KMod will then automatically load parport_pc (with the options -"io=0x378,0x278 irq=7,auto") whenever a parallel port device driver -(such as lp) is loaded. +modprobe will load parport_pc (with the options "io=0x378,0x278 irq=7,auto") +whenever a parallel port device driver (such as lp) is loaded. Note that these are example lines only! You shouldn't in general need to specify any options to parport_pc in order to be able to use a diff --git a/Documentation/s390/3270.txt b/Documentation/s390/3270.txt index 7a5c73a7ed7..7c715de9977 100644 --- a/Documentation/s390/3270.txt +++ b/Documentation/s390/3270.txt @@ -47,9 +47,9 @@ including the console 3270, changes subchannel identifier relative to one another. ReIPL as soon as possible after running the configuration script and the resulting /tmp/mkdev3270. -If you have chosen to make tub3270 a module, you add a line to -/etc/modprobe.conf. If you are working on a VM virtual machine, you -can use DEF GRAF to define virtual 3270 devices. +If you have chosen to make tub3270 a module, you add a line to a +configuration file under /etc/modprobe.d/. If you are working on a VM +virtual machine, you can use DEF GRAF to define virtual 3270 devices. You may generate both 3270 and 3215 console support, or one or the other, or neither. If you generate both, the console type under VM is @@ -60,7 +60,7 @@ at boot time to a 3270 if it is a 3215. In brief, these are the steps: 1. Install the tub3270 patch - 2. (If a module) add a line to /etc/modprobe.conf + 2. (If a module) add a line to a file in /etc/modprobe.d/*.conf 3. (If VM) define devices with DEF GRAF 4. Reboot 5. Configure @@ -84,13 +84,12 @@ Here are the installation steps in detail: make modules_install 2. (Perform this step only if you have configured tub3270 as a - module.) Add a line to /etc/modprobe.conf to automatically - load the driver when it's needed. With this line added, - you will see login prompts appear on your 3270s as soon as - boot is complete (or with emulated 3270s, as soon as you dial - into your vm guest using the command "DIAL "). - Since the line-mode major number is 227, the line to add to - /etc/modprobe.conf should be: + module.) Add a line to a file /etc/modprobe.d/*.conf to automatically + load the driver when it's needed. With this line added, you will see + login prompts appear on your 3270s as soon as boot is complete (or + with emulated 3270s, as soon as you dial into your vm guest using the + command "DIAL "). Since the line-mode major number is + 227, the line to add should be: alias char-major-227 tub3270 3. Define graphic devices to your vm guest machine, if you diff --git a/Documentation/scsi/aic79xx.txt b/Documentation/scsi/aic79xx.txt index 64ac7093c87..e2d3273000d 100644 --- a/Documentation/scsi/aic79xx.txt +++ b/Documentation/scsi/aic79xx.txt @@ -215,7 +215,7 @@ The following information is available in this file: INCORRECTLY CAN RENDER YOUR SYSTEM INOPERABLE. USE THEM WITH CAUTION. - Edit the file "modprobe.conf" in the directory /etc and add/edit a + Put a .conf file in the /etc/modprobe.d/ directory and add/edit a line containing 'options aic79xx aic79xx=[command[,command...]]' where 'command' is one or more of the following: ----------------------------------------------------------------- diff --git a/Documentation/scsi/aic7xxx.txt b/Documentation/scsi/aic7xxx.txt index 18f8d1905e6..7c5d0223d44 100644 --- a/Documentation/scsi/aic7xxx.txt +++ b/Documentation/scsi/aic7xxx.txt @@ -190,7 +190,7 @@ The following information is available in this file: INCORRECTLY CAN RENDER YOUR SYSTEM INOPERABLE. USE THEM WITH CAUTION. - Edit the file "modprobe.conf" in the directory /etc and add/edit a + Put a .conf file in the /etc/modprobe.d directory and add/edit a line containing 'options aic7xxx aic7xxx=[command[,command...]]' where 'command' is one or more of the following: ----------------------------------------------------------------- diff --git a/Documentation/scsi/osst.txt b/Documentation/scsi/osst.txt index ad86c6d1e89..00c8ebb2fd1 100644 --- a/Documentation/scsi/osst.txt +++ b/Documentation/scsi/osst.txt @@ -66,7 +66,7 @@ recognized. If you want to have the module autoloaded on access to /dev/osst, you may add something like alias char-major-206 osst -to your /etc/modprobe.conf (before 2.6: modules.conf). +to a file under /etc/modprobe.d/ directory. You may find it convenient to create a symbolic link ln -s nosst0 /dev/tape diff --git a/Documentation/serial/computone.txt b/Documentation/serial/computone.txt index 39ddcdbeeb8..a6a1158ea2b 100644 --- a/Documentation/serial/computone.txt +++ b/Documentation/serial/computone.txt @@ -49,7 +49,7 @@ Hardware - If you have an ISA card, find a free interrupt and io port. Note the hardware address from the Computone ISA cards installed into the system. These are required for editing ip2.c or editing - /etc/modprobe.conf, or for specification on the modprobe + /etc/modprobe.d/*.conf, or for specification on the modprobe command line. Note that the /etc/modules.conf should be used for older (pre-2.6) @@ -66,7 +66,7 @@ b) Run "make config" or "make menuconfig" or "make xconfig" c) Set address on ISA cards then: edit /usr/src/linux/drivers/char/ip2.c if needed or - edit /etc/modprobe.conf if needed (module). + edit config file in /etc/modprobe.d/ if needed (module). or both to match this setting. d) Run "make modules" e) Run "make modules_install" @@ -153,11 +153,11 @@ the irqs are not specified the driver uses the default in ip2.c (which selects polled mode). If no base addresses are specified the defaults in ip2.c are used. If you are autoloading the driver module with kerneld or kmod the base addresses and interrupt number must also be set in ip2.c -and recompile or just insert and options line in /etc/modprobe.conf or both. +and recompile or just insert and options line in /etc/modprobe.d/*.conf or both. The options line is equivalent to the command line and takes precedence over what is in ip2.c. -/etc/modprobe.conf sample: +config sample to put /etc/modprobe.d/*.conf: options ip2 io=1,0x328 irq=1,10 alias char-major-71 ip2 alias char-major-72 ip2 diff --git a/Documentation/serial/rocket.txt b/Documentation/serial/rocket.txt index 1d858299043..60b03989105 100644 --- a/Documentation/serial/rocket.txt +++ b/Documentation/serial/rocket.txt @@ -62,7 +62,7 @@ in the system log at /var/log/messages. If installed as a module, the module must be loaded. This can be done manually by entering "modprobe rocket". To have the module loaded automatically -upon system boot, edit the /etc/modprobe.conf file and add the line +upon system boot, edit a /etc/modprobe.d/*.conf file and add the line "alias char-major-46 rocket". In order to use the ports, their device names (nodes) must be created with mknod. diff --git a/Documentation/serial/stallion.txt b/Documentation/serial/stallion.txt index 5c4902d9a5b..55090914a9c 100644 --- a/Documentation/serial/stallion.txt +++ b/Documentation/serial/stallion.txt @@ -139,8 +139,8 @@ secondary address 0x280 and IRQ 10. You will probably want to enter this module load and configuration information into your system startup scripts so that the drivers are loaded and configured -on each system boot. Typically the start up script would be something like -/etc/modprobe.conf. +on each system boot. Typically configuration files are put in the +/etc/modprobe.d/ directory. 2.2 STATIC DRIVER CONFIGURATION: diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 6f75ba3b8a3..8c16d50f6cb 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -2044,7 +2044,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Install the necessary firmware files in alsa-firmware package. When no hotplug fw loader is available, you need to load the firmware via vxloader utility in alsa-tools package. To invoke - vxloader automatically, add the following to /etc/modprobe.conf + vxloader automatically, add the following to /etc/modprobe.d/alsa.conf install snd-vx222 /sbin/modprobe --first-time -i snd-vx222 && /usr/bin/vxloader @@ -2168,10 +2168,10 @@ corresponds to the card index of ALSA. Usually, define this as the same card module. An example configuration for a single emu10k1 card is like below: ------ /etc/modprobe.conf +----- /etc/modprobe.d/alsa.conf alias snd-card-0 snd-emu10k1 alias sound-slot-0 snd-emu10k1 ------ /etc/modprobe.conf +----- /etc/modprobe.d/alsa.conf The available number of auto-loaded sound cards depends on the module option "cards_limit" of snd module. As default it's set to 1. @@ -2184,7 +2184,7 @@ cards is kept consistent. An example configuration for two sound cards is like below: ------ /etc/modprobe.conf +----- /etc/modprobe.d/alsa.conf # ALSA portion options snd cards_limit=2 alias snd-card-0 snd-interwave @@ -2194,7 +2194,7 @@ options snd-ens1371 index=1 # OSS/Free portion alias sound-slot-0 snd-interwave alias sound-slot-1 snd-ens1371 ------ /etc/modprobe.conf +----- /etc/modprobe.d/alsa.conf In this example, the interwave card is always loaded as the first card (index 0) and ens1371 as the second (index 1). diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt index a4c53d8961e..654dd3b694a 100644 --- a/Documentation/sound/alsa/Audiophile-Usb.txt +++ b/Documentation/sound/alsa/Audiophile-Usb.txt @@ -232,7 +232,7 @@ The parameter can be given: # modprobe snd-usb-audio index=1 device_setup=0x09 * Or while configuring the modules options in your modules configuration file - - For Fedora distributions, edit the /etc/modprobe.conf file: + (tipically a .conf file in /etc/modprobe.d/ directory: alias snd-card-1 snd-usb-audio options snd-usb-audio index=1 device_setup=0x09 @@ -253,7 +253,7 @@ CAUTION when initializing the device - first turn off the device - de-register the snd-usb-audio module (modprobe -r) - change the device_setup parameter by changing the device_setup - option in /etc/modprobe.conf + option in /etc/modprobe.d/*.conf - turn on the device * A workaround for this last issue has been applied to kernel 2.6.23, but it may not be enough to ensure the 'stability' of the device initialization. diff --git a/Documentation/sound/alsa/MIXART.txt b/Documentation/sound/alsa/MIXART.txt index ef42c44fa1f..4ee35b4fbe4 100644 --- a/Documentation/sound/alsa/MIXART.txt +++ b/Documentation/sound/alsa/MIXART.txt @@ -76,9 +76,9 @@ FIRMWARE when CONFIG_FW_LOADER is set. The mixartloader is necessary only for older versions or when you build the driver into kernel.] -For loading the firmware automatically after the module is loaded, use -the post-install command. For example, add the following entry to -/etc/modprobe.conf for miXart driver: +For loading the firmware automatically after the module is loaded, use a +install command. For example, add the following entry to +/etc/modprobe.d/mixart.conf for miXart driver: install snd-mixart /sbin/modprobe --first-time -i snd-mixart && \ /usr/bin/mixartloader diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt index 022aaeb0e9d..152ca2a3f1b 100644 --- a/Documentation/sound/alsa/OSS-Emulation.txt +++ b/Documentation/sound/alsa/OSS-Emulation.txt @@ -19,7 +19,7 @@ the card number and the minor unit number. Usually you don't have to define these aliases by yourself. Only necessary step for auto-loading of OSS modules is to define the -card alias in /etc/modprobe.conf, such as +card alias in /etc/modprobe.d/alsa.conf, such as alias sound-slot-0 snd-emu10k1 diff --git a/Documentation/sound/oss/AudioExcelDSP16 b/Documentation/sound/oss/AudioExcelDSP16 index e0dc0641b48..e863f9cd594 100644 --- a/Documentation/sound/oss/AudioExcelDSP16 +++ b/Documentation/sound/oss/AudioExcelDSP16 @@ -41,7 +41,7 @@ mpu_base I/O base address for activate MPU-401 mode (0x300, 0x310, 0x320 or 0x330) mpu_irq MPU-401 irq line (5, 7, 9, 10 or 0) -The /etc/modprobe.conf will have lines like this: +A configuration file in /etc/modprobe.d/ directory will have lines like this: options opl3 io=0x388 options ad1848 io=0x530 irq=11 dma=3 @@ -51,8 +51,8 @@ Where the aedsp16 options are the options for this driver while opl3 and ad1848 are the corresponding options for the MSS and OPL3 modules. Loading MSS and OPL3 needs to pre load the aedsp16 module to set up correctly -the sound card. Installation dependencies must be written in the modprobe.conf -file: +the sound card. Installation dependencies must be written in configuration +files under /etc/modprobe.d/ directory: install ad1848 /sbin/modprobe aedsp16 && /sbin/modprobe -i ad1848 install opl3 /sbin/modprobe aedsp16 && /sbin/modprobe -i opl3 diff --git a/Documentation/sound/oss/CMI8330 b/Documentation/sound/oss/CMI8330 index 9c439f1a6db..8a5fd1611c6 100644 --- a/Documentation/sound/oss/CMI8330 +++ b/Documentation/sound/oss/CMI8330 @@ -143,11 +143,10 @@ CONFIG_SOUND_MSS=m -Alma Chao suggests the following /etc/modprobe.conf: +Alma Chao suggests the following in +a /etc/modprobe.d/*conf file: alias sound ad1848 alias synth0 opl3 options ad1848 io=0x530 irq=7 dma=0 soundpro=1 options opl3 io=0x388 - - diff --git a/Documentation/sound/oss/Introduction b/Documentation/sound/oss/Introduction index 75d967ff926..42da2d8fa37 100644 --- a/Documentation/sound/oss/Introduction +++ b/Documentation/sound/oss/Introduction @@ -167,8 +167,8 @@ in a file such as /root/soundon.sh. MODPROBE: ========= -If loading via modprobe, these common files are automatically loaded -when requested by modprobe. For example, my /etc/modprobe.conf contains: +If loading via modprobe, these common files are automatically loaded when +requested by modprobe. For example, my /etc/modprobe.d/oss.conf contains: alias sound sb options sb io=0x240 irq=9 dma=3 dma16=5 mpu_io=0x300 @@ -228,7 +228,7 @@ http://www.opensound.com. Before loading the commercial sound driver, you should do the following: 1. remove sound modules (detailed above) -2. remove the sound modules from /etc/modprobe.conf +2. remove the sound modules from /etc/modprobe.d/*.conf 3. move the sound modules from /lib/modules//misc (for example, I make a /lib/modules//misc/tmp directory and copy the sound module files to that @@ -265,7 +265,7 @@ twice, you need to do the following: sb.o could be copied (or symlinked) to sb1.o for the second SoundBlaster. -2. Make a second entry in /etc/modprobe.conf, for example, +2. Make a second entry in /etc/modprobe.d/*conf, for example, sound1 or sb1. This second entry should refer to the new module names for example sb1, and should include the I/O, etc. for the second sound card. @@ -369,7 +369,7 @@ There are several ways of configuring your sound: 2) On the command line when using insmod or in a bash script using command line calls to load sound. -3) In /etc/modprobe.conf when using modprobe. +3) In /etc/modprobe.d/*conf when using modprobe. 4) Via Red Hat's GPL'd /usr/sbin/sndconfig program (text based). diff --git a/Documentation/sound/oss/Opti b/Documentation/sound/oss/Opti index c15af3c07d4..4cd5d9ab358 100644 --- a/Documentation/sound/oss/Opti +++ b/Documentation/sound/oss/Opti @@ -18,7 +18,7 @@ force the card into a mode in which it can be programmed. If you have another OS installed on your computer it is recommended that Linux and the other OS use the same resources. -Also, it is recommended that resources specified in /etc/modprobe.conf +Also, it is recommended that resources specified in /etc/modprobe.d/*.conf and resources specified in /etc/isapnp.conf agree. Compiling the sound driver @@ -67,11 +67,7 @@ address is hard-coded into the driver. Using kmod and autoloading the sound driver ------------------------------------------- -Comment: as of linux-2.1.90 kmod is replacing kerneld. -The config file '/etc/modprobe.conf' is used as before. - -This is the sound part of my /etc/modprobe.conf file. -Following that I will explain each line. +Config files in '/etc/modprobe.d/' are used as below: alias mixer0 mad16 alias audio0 mad16 diff --git a/Documentation/sound/oss/PAS16 b/Documentation/sound/oss/PAS16 index 3dca4b75988..5c27229eec8 100644 --- a/Documentation/sound/oss/PAS16 +++ b/Documentation/sound/oss/PAS16 @@ -128,7 +128,7 @@ CONFIG_SOUND_YM3812 You can then get OPL3 functionality by issuing the command: insmod opl3 In addition, you must either add the following line to - /etc/modprobe.conf: + /etc/modprobe.d/*.conf: options opl3 io=0x388 or else add the following line to /etc/lilo.conf: opl3=0x388 @@ -158,5 +158,5 @@ following line would be appropriate: append="pas2=0x388,10,3,-1,0,-1,-1,-1 opl3=0x388" If sound is built totally modular, the above options may be -specified in /etc/modprobe.conf for pas2, sb and opl3 +specified in /etc/modprobe.d/*.conf for pas2, sb and opl3 respectively. diff --git a/Documentation/sound/oss/README.modules b/Documentation/sound/oss/README.modules index e691d74e1e5..bf5142a7be7 100644 --- a/Documentation/sound/oss/README.modules +++ b/Documentation/sound/oss/README.modules @@ -26,7 +26,7 @@ Note that it is no longer necessary or possible to configure sound in the drivers/sound dir. Now one simply configures and makes one's kernel and modules in the usual way. - Then, add to your /etc/modprobe.conf something like: + Then, add to your /etc/modprobe.d/oss.conf something like: alias char-major-14-* sb install sb /sbin/modprobe -i sb && /sbin/modprobe adlib_card @@ -66,12 +66,12 @@ args are expected. Note that at present there is no way to configure the io, irq and other parameters for the modular drivers as one does for the wired drivers.. One needs to pass the modules the necessary parameters as arguments, either -with /etc/modprobe.conf or with command-line args to modprobe, e.g. +with /etc/modprobe.d/*.conf or with command-line args to modprobe, e.g. modprobe sb io=0x220 irq=7 dma=1 dma16=5 mpu_io=0x330 modprobe adlib_card io=0x388 - recommend using /etc/modprobe.conf. + recommend using /etc/modprobe.d/*.conf. Persistent DMA Buffers: @@ -89,7 +89,7 @@ wasteful of RAM, but it guarantees that sound always works. To make the sound driver use persistent DMA buffers we need to pass the sound.o module a "dmabuf=1" command-line argument. This is normally done -in /etc/modprobe.conf like so: +in /etc/modprobe.d/*.conf files like so: options sound dmabuf=1 diff --git a/Documentation/usb/power-management.txt b/Documentation/usb/power-management.txt index 817df299ea0..4204eb01fd3 100644 --- a/Documentation/usb/power-management.txt +++ b/Documentation/usb/power-management.txt @@ -179,7 +179,8 @@ do: modprobe usbcore autosuspend=5 -Equivalently, you could add to /etc/modprobe.conf a line saying: +Equivalently, you could add to a configuration file in /etc/modprobe.d +a line saying: options usbcore autosuspend=5 diff --git a/Documentation/video4linux/CQcam.txt b/Documentation/video4linux/CQcam.txt index 8977e7ce4da..6e680fec1e9 100644 --- a/Documentation/video4linux/CQcam.txt +++ b/Documentation/video4linux/CQcam.txt @@ -61,29 +61,19 @@ But that is my personal preference. 2.2 Configuration The configuration requires module configuration and device -configuration. I like kmod or kerneld process with the -/etc/modprobe.conf file so the modules can automatically load/unload as -they are used. The video devices could already exist, be generated -using MAKEDEV, or need to be created. The following sections detail -these procedures. +configuration. The following sections detail these procedures. 2.1 Module Configuration Using modules requires a bit of work to install and pass the -parameters. Understand that entries in /etc/modprobe.conf of: +parameters. Understand that entries in /etc/modprobe.d/*.conf of: alias parport_lowlevel parport_pc options parport_pc io=0x378 irq=none alias char-major-81 videodev alias char-major-81-0 c-qcam -will cause the kmod/modprobe to do certain things. If you are -using kmod, then a request for a 'char-major-81-0' will cause -the 'c-qcam' module to load. If you have other video sources with -modules, you might want to assign the different minor numbers to -different modules. - 2.2 Device Configuration At this point, we need to ensure that the device files exist. diff --git a/Documentation/video4linux/Zoran b/Documentation/video4linux/Zoran index 9ed629d4874..b5a911fd060 100644 --- a/Documentation/video4linux/Zoran +++ b/Documentation/video4linux/Zoran @@ -255,7 +255,7 @@ Load zr36067.o. If it can't autodetect your card, use the card=X insmod option with X being the card number as given in the previous section. To have more than one card, use card=X1[,X2[,X3,[X4[..]]]] -To automate this, add the following to your /etc/modprobe.conf: +To automate this, add the following to your /etc/modprobe.d/zoran.conf: options zr36067 card=X1[,X2[,X3[,X4[..]]]] alias char-major-81-0 zr36067 diff --git a/Documentation/video4linux/bttv/Modules.conf b/Documentation/video4linux/bttv/Modules.conf index 753f15956eb..8f258faf18f 100644 --- a/Documentation/video4linux/bttv/Modules.conf +++ b/Documentation/video4linux/bttv/Modules.conf @@ -1,4 +1,4 @@ -# For modern kernels (2.6 or above), this belongs in /etc/modprobe.conf +# For modern kernels (2.6 or above), this belongs in /etc/modprobe.d/*.conf # For for 2.4 kernels or earlier, this belongs in /etc/modules.conf. # i2c diff --git a/Documentation/video4linux/meye.txt b/Documentation/video4linux/meye.txt index 34e2842c70a..a051152ea99 100644 --- a/Documentation/video4linux/meye.txt +++ b/Documentation/video4linux/meye.txt @@ -55,7 +55,7 @@ Module use: ----------- In order to automatically load the meye module on use, you can put those lines -in your /etc/modprobe.conf file: +in your /etc/modprobe.d/meye.conf file: alias char-major-81 videodev alias char-major-81-0 meye diff --git a/drivers/net/wan/Kconfig b/drivers/net/wan/Kconfig index 423eb26386c..d70ede7a7f9 100644 --- a/drivers/net/wan/Kconfig +++ b/drivers/net/wan/Kconfig @@ -290,8 +290,8 @@ config FARSYNC Frame Relay or X.25/LAPB. If you want the module to be automatically loaded when the interface - is referenced then you should add "alias hdlcX farsync" to - /etc/modprobe.conf for each interface, where X is 0, 1, 2, ..., or + is referenced then you should add "alias hdlcX farsync" to a file + in /etc/modprobe.d/ for each interface, where X is 0, 1, 2, ..., or simply use "alias hdlc* farsync" to indicate all of them. To compile this driver as a module, choose M here: the diff --git a/drivers/scsi/aic7xxx/aic79xx_osm.c b/drivers/scsi/aic7xxx/aic79xx_osm.c index 7d48700257a..9328121804b 100644 --- a/drivers/scsi/aic7xxx/aic79xx_osm.c +++ b/drivers/scsi/aic7xxx/aic79xx_osm.c @@ -341,10 +341,10 @@ MODULE_PARM_DESC(aic79xx, " (0/256ms,1/128ms,2/64ms,3/32ms)\n" " slowcrc Turn on the SLOWCRC bit (Rev B only)\n" "\n" -" Sample /etc/modprobe.conf line:\n" -" Enable verbose logging\n" -" Set tag depth on Controller 2/Target 2 to 10 tags\n" -" Shorten the selection timeout to 128ms\n" +" Sample modprobe configuration file:\n" +" # Enable verbose logging\n" +" # Set tag depth on Controller 2/Target 2 to 10 tags\n" +" # Shorten the selection timeout to 128ms\n" "\n" " options aic79xx 'aic79xx=verbose.tag_info:{{}.{}.{..10}}.seltime:1'\n" ); diff --git a/drivers/scsi/aic7xxx/aic7xxx_osm.c b/drivers/scsi/aic7xxx/aic7xxx_osm.c index c6251bb4f43..5a477cdc780 100644 --- a/drivers/scsi/aic7xxx/aic7xxx_osm.c +++ b/drivers/scsi/aic7xxx/aic7xxx_osm.c @@ -360,10 +360,10 @@ MODULE_PARM_DESC(aic7xxx, " seltime: Selection Timeout\n" " (0/256ms,1/128ms,2/64ms,3/32ms)\n" "\n" -" Sample /etc/modprobe.conf line:\n" -" Toggle EISA/VLB probing\n" -" Set tag depth on Controller 1/Target 1 to 10 tags\n" -" Shorten the selection timeout to 128ms\n" +" Sample modprobe configuration file:\n" +" # Toggle EISA/VLB probing\n" +" # Set tag depth on Controller 1/Target 1 to 10 tags\n" +" # Shorten the selection timeout to 128ms\n" "\n" " options aic7xxx 'aic7xxx=probe_eisa_vl.tag_info:{{}.{.10}}.seltime:1'\n" ); diff --git a/drivers/staging/asus_oled/README b/drivers/staging/asus_oled/README index 0d82a6d5fa5..2d721232467 100644 --- a/drivers/staging/asus_oled/README +++ b/drivers/staging/asus_oled/README @@ -52,7 +52,7 @@ Configuration There is only one option: start_off. You can use it by: 'modprobe asus_oled start_off=1', or by adding this - line to /etc/modprobe.conf: + line to /etc/modprobe.d/asus_oled.conf: options asus_oled start_off=1 With this option provided, asus_oled driver will switch off the display diff --git a/drivers/tty/isicom.c b/drivers/tty/isicom.c index 794ecb40017..e1235accab7 100644 --- a/drivers/tty/isicom.c +++ b/drivers/tty/isicom.c @@ -102,7 +102,7 @@ * You can find the original tools for this direct from Multitech * ftp://ftp.multitech.com/ISI-Cards/ * - * Having installed the cards the module options (/etc/modprobe.conf) + * Having installed the cards the module options (/etc/modprobe.d/) * * options isicom io=card1,card2,card3,card4 irq=card1,card2,card3,card4 * diff --git a/drivers/usb/serial/ftdi_sio.c b/drivers/usb/serial/ftdi_sio.c index 7c229d30468..ff8605b4b4b 100644 --- a/drivers/usb/serial/ftdi_sio.c +++ b/drivers/usb/serial/ftdi_sio.c @@ -1724,7 +1724,8 @@ static void ftdi_HE_TIRA1_setup(struct ftdi_private *priv) /* * Module parameter to control latency timer for NDI FTDI-based USB devices. - * If this value is not set in modprobe.conf.local its value will be set to 1ms. + * If this value is not set in /etc/modprobe.d/ its value will be set + * to 1ms. */ static int ndi_latency_timer = 1; diff --git a/drivers/usb/storage/Kconfig b/drivers/usb/storage/Kconfig index fe2d803a634..7691c866637 100644 --- a/drivers/usb/storage/Kconfig +++ b/drivers/usb/storage/Kconfig @@ -222,7 +222,7 @@ config USB_LIBUSUAL for usb-storage and ub drivers, and allows to switch binding of these devices without rebuilding modules. - Typical syntax of /etc/modprobe.conf is: + Typical syntax of /etc/modprobe.d/*conf is: options libusual bias="ub" diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index bbe32d2177d..dbc55071679 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -46,7 +46,7 @@ The number of ports to be created can be specified via the module parameter "ports". For example, to create four ports, add the - following option in /etc/modprobe.conf: + following option in a configuration file under /etc/modprobe.d/: option snd-seq-dummy ports=4 diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index c8961165277..fe5ae09ffcc 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -50,7 +50,8 @@ config SND_PCSP before the other sound driver of yours, making the pc-speaker a default sound device. Which is likely not what you want. To make this driver play nicely with other - sound driver, you can add this into your /etc/modprobe.conf: + sound driver, you can add this in a configuration file under + /etc/modprobe.d/ directory: options snd-pcsp index=2 You don't need this driver if you only want your pc-speaker to beep. -- cgit v1.2.3-70-g09d2 From 78286cdf054212c6d2fe6524fbf673fb9ead1abe Mon Sep 17 00:00:00 2001 From: Lucas De Marchi Date: Fri, 30 Mar 2012 13:37:20 -0700 Subject: Documentation: replace install commands with softdeps Install commands should not be used to specify soft dependencies among modules. When loading modules it's much better to have a softdep that modprobe knows what's being done than having to fork/exec another instance of modprobe to load the other module. By using a softdep user has also an option to remove the dependencies when removing the module (and if its refcount dropped to 0) Signed-off-by: Lucas De Marchi Signed-off-by: Linus Torvalds --- Documentation/networking/bonding.txt | 3 +-- Documentation/sound/oss/AudioExcelDSP16 | 4 ++-- Documentation/sound/oss/README.modules | 2 +- 3 files changed, 4 insertions(+), 5 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/networking/bonding.txt b/Documentation/networking/bonding.txt index d5e86981404..bfea8a33890 100644 --- a/Documentation/networking/bonding.txt +++ b/Documentation/networking/bonding.txt @@ -1822,8 +1822,7 @@ modules.conf manual page. In this case, the following can be added to config files in /etc/modprobe.d/ as: -install bonding /sbin/modprobe tg3; /sbin/modprobe e1000; - /sbin/modprobe --ignore-install bonding +softdep bonding pre: tg3 e1000 This will load tg3 and e1000 modules before loading the bonding one. Full documentation on this can be found in the modprobe.d and modprobe diff --git a/Documentation/sound/oss/AudioExcelDSP16 b/Documentation/sound/oss/AudioExcelDSP16 index e863f9cd594..ea8549faede 100644 --- a/Documentation/sound/oss/AudioExcelDSP16 +++ b/Documentation/sound/oss/AudioExcelDSP16 @@ -54,8 +54,8 @@ Loading MSS and OPL3 needs to pre load the aedsp16 module to set up correctly the sound card. Installation dependencies must be written in configuration files under /etc/modprobe.d/ directory: -install ad1848 /sbin/modprobe aedsp16 && /sbin/modprobe -i ad1848 -install opl3 /sbin/modprobe aedsp16 && /sbin/modprobe -i opl3 +softdep ad1848 pre: aedsp16 +softdep opl3 pre: aedsp16 Then you must load the sound modules stack in this order: sound -> aedsp16 -> [ ad1848, opl3 ] diff --git a/Documentation/sound/oss/README.modules b/Documentation/sound/oss/README.modules index bf5142a7be7..cdc039421a4 100644 --- a/Documentation/sound/oss/README.modules +++ b/Documentation/sound/oss/README.modules @@ -36,7 +36,7 @@ options adlib_card io=0x388 # FM synthesizer Alternatively, if you have compiled in kernel level ISAPnP support: alias char-major-14 sb -post-install sb /sbin/modprobe "-k" "adlib_card" +softdep sb post: adlib_card options adlib_card io=0x388 The effect of this is that the sound driver and all necessary bits and -- cgit v1.2.3-70-g09d2