From ea4e7af1221237e7173ede198a817097d99e084b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Nov 2011 12:23:55 +0100 Subject: ALSA: hda/realtek - Convert alc262 model=tyan to a fixup-list Use the auto-parser for ALC262 model=tyan with a pin-config fix-up and drop the static configuration. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc262_quirks.c | 59 ------------------------------------------- sound/pci/hda/patch_realtek.c | 21 ++++++++++----- 2 files changed, 15 insertions(+), 65 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c index 7894b2b..e53f490 100644 --- a/sound/pci/hda/alc262_quirks.c +++ b/sound/pci/hda/alc262_quirks.c @@ -17,7 +17,6 @@ enum { ALC262_NEC, ALC262_TOSHIBA_S06, ALC262_TOSHIBA_RX1, - ALC262_TYAN, ALC262_MODEL_LAST /* last tag */ }; @@ -177,48 +176,6 @@ static const struct snd_kcontrol_new alc262_benq_t31_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc262_tyan_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Aux Playback Volume", 0x0b, 0x06, HDA_INPUT), - HDA_CODEC_MUTE("Aux Playback Switch", 0x0b, 0x06, 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("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_verb alc262_tyan_verbs[] = { - /* Headphone automute */ - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* P11 AUX_IN, white 4-pin connector */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, 0xe1}, - {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, 0x93}, - {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0x19}, - - {} -}; - -/* unsolicited event for HP jack sensing */ -static void alc262_tyan_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.speaker_pins[0] = 0x15; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - - #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -686,7 +643,6 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = { [ALC262_ULTRA] = "ultra", [ALC262_LENOVO_3000] = "lenovo-3000", [ALC262_NEC] = "nec", - [ALC262_TYAN] = "tyan", [ALC262_AUTO] = "auto", }; @@ -698,7 +654,6 @@ static const struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), - SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN), SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), @@ -857,19 +812,5 @@ static const struct alc_config_preset alc262_presets[] = { .setup = alc262_hippo_setup, .init_hook = alc_inithook, }, - [ALC262_TYAN] = { - .mixers = { alc262_tyan_mixer }, - .init_verbs = { alc262_init_verbs, alc262_tyan_verbs}, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .hp_nid = 0x02, - .dig_out_nid = ALC262_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc262_tyan_setup, - .init_hook = alc_hp_automute, - }, }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 308bb57..013a760 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4339,12 +4339,13 @@ static int alc262_parse_auto_config(struct hda_codec *codec) * Pin config fixes */ enum { - PINFIX_FSC_H270, - PINFIX_HP_Z200, + ALC262_FIXUP_FSC_H270, + ALC262_FIXUP_HP_Z200, + ALC262_FIXUP_TYAN, }; static const struct alc_fixup alc262_fixups[] = { - [PINFIX_FSC_H270] = { + [ALC262_FIXUP_FSC_H270] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { { 0x14, 0x99130110 }, /* speaker */ @@ -4353,18 +4354,26 @@ static const struct alc_fixup alc262_fixups[] = { { } } }, - [PINFIX_HP_Z200] = { + [ALC262_FIXUP_HP_Z200] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { { 0x16, 0x99130120 }, /* internal speaker */ { } } }, + [ALC262_FIXUP_TYAN] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x1993e1f0 }, /* int AUX */ + { } + } + }, }; static const struct snd_pci_quirk alc262_fixup_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", PINFIX_HP_Z200), - SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", PINFIX_FSC_H270), + SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", ALC262_FIXUP_HP_Z200), + SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_FIXUP_TYAN), + SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", ALC262_FIXUP_FSC_H270), {} }; -- cgit v1.1 From 12837c983dc5ec56155b1e95a6fa9a74e4da381f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Nov 2011 12:45:24 +0100 Subject: ALSA: hda/realtek - Convert ALC262 model=toshiba-rx1 to a fixup-list Use the auto-parser for ALC262 model=toshiba-rx1 with the fixed pin- configs. The BIOS table seems incorrect, so many pin entries are overwritten to match with the former quirk. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc262_quirks.c | 46 ------------------------------------------- sound/pci/hda/patch_realtek.c | 13 ++++++++++++ 2 files changed, 13 insertions(+), 46 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c index e53f490..ff7dc7e 100644 --- a/sound/pci/hda/alc262_quirks.c +++ b/sound/pci/hda/alc262_quirks.c @@ -16,7 +16,6 @@ enum { ALC262_LENOVO_3000, ALC262_NEC, ALC262_TOSHIBA_S06, - ALC262_TOSHIBA_RX1, ALC262_MODEL_LAST /* last tag */ }; @@ -465,18 +464,6 @@ static const struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = { - HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol), - ALC262_HIPPO_MASTER_SWITCH, - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), - { } /* end */ -}; - /* additional init verbs for Benq laptops */ static const struct hda_verb alc262_EAPD_verbs[] = { {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -611,23 +598,6 @@ static const struct snd_kcontrol_new alc262_ultra_capture_mixer[] = { { } /* end */ }; -static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = { - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Front Speaker */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x01}, - - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* MIC jack */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Front MIC */ - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) }, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) }, - - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP jack */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - /* * configuration and preset */ @@ -639,7 +609,6 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = { [ALC262_BENQ_ED8] = "benq", [ALC262_BENQ_T31] = "benq-t31", [ALC262_TOSHIBA_S06] = "toshiba-s06", - [ALC262_TOSHIBA_RX1] = "toshiba-rx1", [ALC262_ULTRA] = "ultra", [ALC262_LENOVO_3000] = "lenovo-3000", [ALC262_NEC] = "nec", @@ -649,8 +618,6 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = { static const struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC), - SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", - ALC262_TOSHIBA_RX1), SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), @@ -799,18 +766,5 @@ static const struct alc_config_preset alc262_presets[] = { .setup = alc262_toshiba_s06_setup, .init_hook = alc_inithook, }, - [ALC262_TOSHIBA_RX1] = { - .mixers = { alc262_toshiba_rx1_mixer }, - .init_verbs = { alc262_init_verbs, alc262_toshiba_rx1_unsol_verbs }, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc262_hippo_setup, - .init_hook = alc_inithook, - }, }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 013a760..6f344c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4342,6 +4342,7 @@ enum { ALC262_FIXUP_FSC_H270, ALC262_FIXUP_HP_Z200, ALC262_FIXUP_TYAN, + ALC262_FIXUP_TOSHIBA_RX1, }; static const struct alc_fixup alc262_fixups[] = { @@ -4368,11 +4369,23 @@ static const struct alc_fixup alc262_fixups[] = { { } } }, + [ALC262_FIXUP_TOSHIBA_RX1] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x90170110 }, /* speaker */ + { 0x15, 0x0421101f }, /* HP */ + { 0x1a, 0x40f000f0 }, /* N/A */ + { 0x1b, 0x40f000f0 }, /* N/A */ + { 0x1e, 0x40f000f0 }, /* N/A */ + } + }, }; static const struct snd_pci_quirk alc262_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", ALC262_FIXUP_HP_Z200), SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_FIXUP_TYAN), + SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", + ALC262_FIXUP_TOSHIBA_RX1), SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", ALC262_FIXUP_FSC_H270), {} }; -- cgit v1.1 From ee0eb25119c2c948b5d30da1a62f6d44020cb636 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Nov 2011 12:50:01 +0100 Subject: ALSA: hda/realtek - Drop ALC262 model=toshiba-s06 This laptop works fine with the current auto-parser and the BIOS setup, so let's drop the static configuration. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc262_quirks.c | 52 ------------------------------------------- 1 file changed, 52 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c index ff7dc7e..ace9d48 100644 --- a/sound/pci/hda/alc262_quirks.c +++ b/sound/pci/hda/alc262_quirks.c @@ -15,7 +15,6 @@ enum { ALC262_ULTRA, ALC262_LENOVO_3000, ALC262_NEC, - ALC262_TOSHIBA_S06, ALC262_MODEL_LAST /* last tag */ }; @@ -283,39 +282,6 @@ static const struct hda_verb alc262_sony_unsol_verbs[] = { {} }; -static const struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_verb alc262_toshiba_s06_verbs[] = { - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x22, AC_VERB_SET_CONNECT_SEL, 0x09}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -static void alc262_toshiba_s06_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x12; - spec->auto_mic = 1; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_PIN); -} - /* * nec model * 0x15 = headphone @@ -608,7 +574,6 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = { [ALC262_FUJITSU] = "fujitsu", [ALC262_BENQ_ED8] = "benq", [ALC262_BENQ_T31] = "benq-t31", - [ALC262_TOSHIBA_S06] = "toshiba-s06", [ALC262_ULTRA] = "ultra", [ALC262_LENOVO_3000] = "lenovo-3000", [ALC262_NEC] = "nec", @@ -618,7 +583,6 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = { static const struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC), - SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1", @@ -750,21 +714,5 @@ static const struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, }, - [ALC262_TOSHIBA_S06] = { - .mixers = { alc262_toshiba_s06_mixer }, - .init_verbs = { alc262_init_verbs, alc262_toshiba_s06_verbs, - alc262_eapd_verbs }, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .capsrc_nids = alc262_dmic_capsrc_nids, - .dac_nids = alc262_dac_nids, - .adc_nids = alc262_dmic_adc_nids, /* ADC0 */ - .num_adc_nids = 1, /* single ADC */ - .dig_out_nid = ALC262_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .unsol_event = alc_sku_unsol_event, - .setup = alc262_toshiba_s06_setup, - .init_hook = alc_inithook, - }, }; -- cgit v1.1 From 80f6b7736615cc7b09fa2a50f40b9773d0f15c6a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Nov 2011 12:57:16 +0100 Subject: ALSA: hda/realtek - Drop ALC262 model=nec quirk This laptop works also fine with the auto-parser and the BIOS setup. A good boy. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc262_quirks.c | 48 ------------------------------------------- 1 file changed, 48 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c index ace9d48..3dc843e 100644 --- a/sound/pci/hda/alc262_quirks.c +++ b/sound/pci/hda/alc262_quirks.c @@ -14,7 +14,6 @@ enum { ALC262_BENQ_T31, ALC262_ULTRA, ALC262_LENOVO_3000, - ALC262_NEC, ALC262_MODEL_LAST /* last tag */ }; @@ -283,41 +282,6 @@ static const struct hda_verb alc262_sony_unsol_verbs[] = { }; /* - * nec model - * 0x15 = headphone - * 0x16 = internal speaker - * 0x18 = external mic - */ - -static const struct snd_kcontrol_new alc262_nec_mixer[] = { - HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 0, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct hda_verb alc262_nec_verbs[] = { - /* Unmute Speaker */ - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Headphone */ - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* External mic to headphone */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* External mic to speaker */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {} -}; - -/* * fujitsu model * 0x14 = headphone/spdif-out, 0x15 = internal speaker, * 0x1b = port replicator headphone out @@ -576,13 +540,11 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = { [ALC262_BENQ_T31] = "benq-t31", [ALC262_ULTRA] = "ultra", [ALC262_LENOVO_3000] = "lenovo-3000", - [ALC262_NEC] = "nec", [ALC262_AUTO] = "auto", }; static const struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), - SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1", @@ -704,15 +666,5 @@ static const struct alc_config_preset alc262_presets[] = { .setup = alc262_lenovo_3000_setup, .init_hook = alc_inithook, }, - [ALC262_NEC] = { - .mixers = { alc262_nec_mixer }, - .init_verbs = { alc262_nec_verbs }, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_capture_source, - }, }; -- cgit v1.1 From c470150c537b76d9776a5434f6f8448db8188bd3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Nov 2011 14:20:07 +0100 Subject: ALSA: hda/realtek - Convert ALC262 lenovo-3000 quirks to fixup-list The static quirks for ALC262 Lenovo 3000 can be covered by the auto- parser with a fixup of the mic-pin to VREF50 and the additional COEF verb. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc262_quirks.c | 63 ------------------------------------------- sound/pci/hda/patch_realtek.c | 12 +++++++++ 2 files changed, 12 insertions(+), 63 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c index 3dc843e..dc4d1a1 100644 --- a/sound/pci/hda/alc262_quirks.c +++ b/sound/pci/hda/alc262_quirks.c @@ -13,7 +13,6 @@ enum { ALC262_BENQ_ED8, ALC262_BENQ_T31, ALC262_ULTRA, - ALC262_LENOVO_3000, ALC262_MODEL_LAST /* last tag */ }; @@ -295,19 +294,6 @@ static const struct hda_verb alc262_fujitsu_unsol_verbs[] = { {} }; -static const struct hda_verb alc262_lenovo_3000_unsol_verbs[] = { - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {} -}; - -static const struct hda_verb alc262_lenovo_3000_init_verbs[] = { - /* Front Mic pin: input vref at 50% */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {} -}; - static const struct hda_input_mux alc262_fujitsu_capture_source = { .num_items = 3, .items = { @@ -363,37 +349,6 @@ static const struct snd_kcontrol_new alc262_fujitsu_mixer[] = { { } /* end */ }; -static void alc262_lenovo_3000_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[1] = 0x16; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static const struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { - HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, - .info = snd_ctl_boolean_mono_info, - .get = alc262_hp_master_sw_get, - .put = alc262_hp_master_sw_put, - }, - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, 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 Boost Volume", 0x19, 0, 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 */ -}; - /* additional init verbs for Benq laptops */ static const struct hda_verb alc262_EAPD_verbs[] = { {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -539,7 +494,6 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = { [ALC262_BENQ_ED8] = "benq", [ALC262_BENQ_T31] = "benq-t31", [ALC262_ULTRA] = "ultra", - [ALC262_LENOVO_3000] = "lenovo-3000", [ALC262_AUTO] = "auto", }; @@ -550,7 +504,6 @@ static const struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), - SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1), @@ -650,21 +603,5 @@ static const struct alc_config_preset alc262_presets[] = { .unsol_event = alc262_ultra_unsol_event, .init_hook = alc262_ultra_automute, }, - [ALC262_LENOVO_3000] = { - .mixers = { alc262_lenovo_3000_mixer }, - .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs, - alc262_lenovo_3000_unsol_verbs, - alc262_lenovo_3000_init_verbs }, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .hp_nid = 0x03, - .dig_out_nid = ALC262_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_fujitsu_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc262_lenovo_3000_setup, - .init_hook = alc_inithook, - }, }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6f344c9..ee267be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4343,6 +4343,7 @@ enum { ALC262_FIXUP_HP_Z200, ALC262_FIXUP_TYAN, ALC262_FIXUP_TOSHIBA_RX1, + ALC262_FIXUP_LENOVO_3000, }; static const struct alc_fixup alc262_fixups[] = { @@ -4379,6 +4380,16 @@ static const struct alc_fixup alc262_fixups[] = { { 0x1e, 0x40f000f0 }, /* N/A */ } }, + [ALC262_FIXUP_LENOVO_3000] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x3070 }, + {} + } + }, + }; static const struct snd_pci_quirk alc262_fixup_tbl[] = { @@ -4387,6 +4398,7 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = { SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", ALC262_FIXUP_TOSHIBA_RX1), SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", ALC262_FIXUP_FSC_H270), + SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000", ALC262_FIXUP_LENOVO_3000), {} }; -- cgit v1.1 From b42590b865cb4358a0c37b2128e303a18d897bf6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Nov 2011 14:41:01 +0100 Subject: ALSA: hda/realtek - Convert ALC262 benq and benq-t31 to fixup-lists The conversion from ALC262 model=benq and model=benq-t31 static configs to auto-parser requires the manual COEF setups for corresponding models. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc262_quirks.c | 50 ------------------------------------------- sound/pci/hda/patch_realtek.c | 21 +++++++++++++++++- 2 files changed, 20 insertions(+), 51 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c index dc4d1a1..bf573bd 100644 --- a/sound/pci/hda/alc262_quirks.c +++ b/sound/pci/hda/alc262_quirks.c @@ -10,8 +10,6 @@ enum { ALC262_HIPPO, ALC262_HIPPO_1, ALC262_FUJITSU, - ALC262_BENQ_ED8, - ALC262_BENQ_T31, ALC262_ULTRA, ALC262_MODEL_LAST /* last tag */ }; @@ -161,17 +159,6 @@ static const struct snd_kcontrol_new alc262_sony_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc262_benq_t31_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - ALC262_HIPPO_MASTER_SWITCH, - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - { } /* end */ -}; - #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -356,15 +343,6 @@ static const struct hda_verb alc262_EAPD_verbs[] = { {} }; -static const struct hda_verb alc262_benq_t31_EAPD_verbs[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - - {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, - {0x20, AC_VERB_SET_PROC_COEF, 0x3050}, - {} -}; - /* Samsung Q1 Ultra Vista model setup */ static const struct snd_kcontrol_new alc262_ultra_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -491,8 +469,6 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = { [ALC262_HIPPO] = "hippo", [ALC262_HIPPO_1] = "hippo_1", [ALC262_FUJITSU] = "fujitsu", - [ALC262_BENQ_ED8] = "benq", - [ALC262_BENQ_T31] = "benq-t31", [ALC262_ULTRA] = "ultra", [ALC262_AUTO] = "auto", }; @@ -504,8 +480,6 @@ static const struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), - SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), - SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1), {} }; @@ -564,30 +538,6 @@ static const struct alc_config_preset alc262_presets[] = { .setup = alc262_fujitsu_setup, .init_hook = alc_inithook, }, - [ALC262_BENQ_ED8] = { - .mixers = { alc262_base_mixer }, - .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs }, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_capture_source, - }, - [ALC262_BENQ_T31] = { - .mixers = { alc262_benq_t31_mixer }, - .init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs, - alc_hp15_unsol_verbs }, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc262_hippo_setup, - .init_hook = alc_inithook, - }, [ALC262_ULTRA] = { .mixers = { alc262_ultra_mixer }, .cap_mixer = alc262_ultra_capture_mixer, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ee267be..55bdb73 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4344,6 +4344,8 @@ enum { ALC262_FIXUP_TYAN, ALC262_FIXUP_TOSHIBA_RX1, ALC262_FIXUP_LENOVO_3000, + ALC262_FIXUP_BENQ, + ALC262_FIXUP_BENQ_T31, }; static const struct alc_fixup alc262_fixups[] = { @@ -4384,12 +4386,27 @@ static const struct alc_fixup alc262_fixups[] = { .type = ALC_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 }, + {} + }, + .chained = true, + .chain_id = ALC262_FIXUP_BENQ, + }, + [ALC262_FIXUP_BENQ] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x3070 }, {} } }, - + [ALC262_FIXUP_BENQ_T31] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x3050 }, + {} + } + }, }; static const struct snd_pci_quirk alc262_fixup_tbl[] = { @@ -4399,6 +4416,8 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = { ALC262_FIXUP_TOSHIBA_RX1), SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", ALC262_FIXUP_FSC_H270), SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000", ALC262_FIXUP_LENOVO_3000), + SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_FIXUP_BENQ), + SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_FIXUP_BENQ_T31), {} }; -- cgit v1.1 From 3dcd3be33046c9828a6057ee0f14de6e5d3b48a2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Nov 2011 14:59:40 +0100 Subject: ALSA: hda/realtek - Convert ALC262 model=fujitsu to auto-parser It works well with the auto-parse and the default BIOS setup when an additional COEF setup (for benq) is used. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc262_quirks.c | 95 ------------------------------------------- sound/pci/hda/patch_realtek.c | 2 + 2 files changed, 2 insertions(+), 95 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c index bf573bd..ae10d00 100644 --- a/sound/pci/hda/alc262_quirks.c +++ b/sound/pci/hda/alc262_quirks.c @@ -9,7 +9,6 @@ enum { ALC262_BASIC, ALC262_HIPPO, ALC262_HIPPO_1, - ALC262_FUJITSU, ALC262_ULTRA, ALC262_MODEL_LAST /* last tag */ }; @@ -267,82 +266,6 @@ static const struct hda_verb alc262_sony_unsol_verbs[] = { {} }; -/* - * fujitsu model - * 0x14 = headphone/spdif-out, 0x15 = internal speaker, - * 0x1b = port replicator headphone out - */ - -static const struct hda_verb alc262_fujitsu_unsol_verbs[] = { - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {} -}; - -static const struct hda_input_mux alc262_fujitsu_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Internal Mic", 0x1 }, - { "CD", 0x4 }, - }, -}; - -static void alc262_fujitsu_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.hp_pins[1] = 0x1b; - spec->autocfg.speaker_pins[0] = 0x15; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -/* bind volumes of both NID 0x0c and 0x0d */ -static const struct hda_bind_ctls alc262_fujitsu_bind_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x0d, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static const struct snd_kcontrol_new alc262_fujitsu_mixer[] = { - HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, - .info = snd_ctl_boolean_mono_info, - .get = alc262_hp_master_sw_get, - .put = alc262_hp_master_sw_put, - }, - { - .iface = NID_MAPPING, - .name = "Master Playback Switch", - .private_value = 0x1b, - }, - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, 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 Boost Volume", 0x19, 0, 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 */ -}; - -/* additional init verbs for Benq laptops */ -static const struct hda_verb alc262_EAPD_verbs[] = { - {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, - {0x20, AC_VERB_SET_PROC_COEF, 0x3070}, - {} -}; - /* Samsung Q1 Ultra Vista model setup */ static const struct snd_kcontrol_new alc262_ultra_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -468,15 +391,12 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = { [ALC262_BASIC] = "basic", [ALC262_HIPPO] = "hippo", [ALC262_HIPPO_1] = "hippo_1", - [ALC262_FUJITSU] = "fujitsu", [ALC262_ULTRA] = "ultra", [ALC262_AUTO] = "auto", }; static const struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), - SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), - SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), @@ -523,21 +443,6 @@ static const struct alc_config_preset alc262_presets[] = { .setup = alc262_hippo1_setup, .init_hook = alc_inithook, }, - [ALC262_FUJITSU] = { - .mixers = { alc262_fujitsu_mixer }, - .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs, - alc262_fujitsu_unsol_verbs }, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .hp_nid = 0x03, - .dig_out_nid = ALC262_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_fujitsu_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc262_fujitsu_setup, - .init_hook = alc_inithook, - }, [ALC262_ULTRA] = { .mixers = { alc262_ultra_mixer }, .cap_mixer = alc262_ultra_capture_mixer, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 55bdb73..5082070 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4411,6 +4411,8 @@ static const struct alc_fixup alc262_fixups[] = { static const struct snd_pci_quirk alc262_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", ALC262_FIXUP_HP_Z200), + SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FIXUP_BENQ), + SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FIXUP_BENQ), SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_FIXUP_TYAN), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", ALC262_FIXUP_TOSHIBA_RX1), -- cgit v1.1 From 6fb9c82c47164ac5daa80e9f7979ccfa252ed2b5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Nov 2011 15:04:23 +0100 Subject: ALSA: hda/realtek - Drop ALC262 model=hippo static quirks This model (actually BenQ Joybook) works fine with the default auto-parser and the BIOS setup. Just drop the static quirks. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc262_quirks.c | 63 ------------------------------------------- 1 file changed, 63 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c index ae10d00..38ff995 100644 --- a/sound/pci/hda/alc262_quirks.c +++ b/sound/pci/hda/alc262_quirks.c @@ -8,7 +8,6 @@ enum { ALC262_AUTO, ALC262_BASIC, ALC262_HIPPO, - ALC262_HIPPO_1, ALC262_ULTRA, ALC262_MODEL_LAST /* last tag */ }; @@ -112,22 +111,6 @@ static const struct snd_kcontrol_new alc262_hippo_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc262_hippo1_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - ALC262_HIPPO_MASTER_SWITCH, - 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("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), - { } /* end */ -}; - /* mute/unmute internal speaker according to the hp jack and mute state */ static void alc262_hippo_setup(struct hda_codec *codec) { @@ -138,16 +121,6 @@ static void alc262_hippo_setup(struct hda_codec *codec) alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } -static void alc262_hippo1_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.speaker_pins[0] = 0x14; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - - static const struct snd_kcontrol_new alc262_sony_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), ALC262_HIPPO_MASTER_SWITCH, @@ -246,26 +219,6 @@ static const struct hda_verb alc262_eapd_verbs[] = { { } }; -static const struct hda_verb alc262_hippo1_unsol_verbs[] = { - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, - - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {} -}; - -static const struct hda_verb alc262_sony_unsol_verbs[] = { - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, // Front Mic - - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {} -}; - /* Samsung Q1 Ultra Vista model setup */ static const struct snd_kcontrol_new alc262_ultra_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -390,7 +343,6 @@ static const struct snd_kcontrol_new alc262_ultra_capture_mixer[] = { static const char * const alc262_models[ALC262_MODEL_LAST] = { [ALC262_BASIC] = "basic", [ALC262_HIPPO] = "hippo", - [ALC262_HIPPO_1] = "hippo_1", [ALC262_ULTRA] = "ultra", [ALC262_AUTO] = "auto", }; @@ -400,7 +352,6 @@ static const struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), - SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1), {} }; @@ -429,20 +380,6 @@ static const struct alc_config_preset alc262_presets[] = { .setup = alc262_hippo_setup, .init_hook = alc_inithook, }, - [ALC262_HIPPO_1] = { - .mixers = { alc262_hippo1_mixer }, - .init_verbs = { alc262_init_verbs, alc262_hippo1_unsol_verbs}, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .hp_nid = 0x02, - .dig_out_nid = ALC262_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc262_hippo1_setup, - .init_hook = alc_inithook, - }, [ALC262_ULTRA] = { .mixers = { alc262_ultra_mixer }, .cap_mixer = alc262_ultra_capture_mixer, -- cgit v1.1 From 46900b5c55ece43b0e6ad7147ffb58e37ff45c82 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Nov 2011 15:18:21 +0100 Subject: ALSA: hda/realtek - Drop ALC262 model=hippo static quirks Both entries for ALC262 model=hippo work well with the auto-parser and the default BIOS setup. No static configs are needed, so drop them. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc262_quirks.c | 104 ------------------------------------------ 1 file changed, 104 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c index 38ff995..fee7202 100644 --- a/sound/pci/hda/alc262_quirks.c +++ b/sound/pci/hda/alc262_quirks.c @@ -7,7 +7,6 @@ enum { ALC262_AUTO, ALC262_BASIC, - ALC262_HIPPO, ALC262_ULTRA, ALC262_MODEL_LAST /* last tag */ }; @@ -51,86 +50,6 @@ static const struct snd_kcontrol_new alc262_base_mixer[] = { { } /* end */ }; -/* bind hp and internal speaker mute (with plug check) as master switch */ - -static int alc262_hippo_master_sw_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; - *ucontrol->value.integer.value = !spec->master_mute; - return 0; -} - -static int alc262_hippo_master_sw_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 val = !*ucontrol->value.integer.value; - - if (val == spec->master_mute) - return 0; - spec->master_mute = val; - update_outputs(codec); - return 1; -} - -#define ALC262_HIPPO_MASTER_SWITCH \ - { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = "Master Playback Switch", \ - .info = snd_ctl_boolean_mono_info, \ - .get = alc262_hippo_master_sw_get, \ - .put = alc262_hippo_master_sw_put, \ - }, \ - { \ - .iface = NID_MAPPING, \ - .name = "Master Playback Switch", \ - .subdevice = SUBDEV_HP(0) | (SUBDEV_LINE(0) << 8) | \ - (SUBDEV_SPEAKER(0) << 16), \ - } - -#define alc262_hp_master_sw_get alc262_hippo_master_sw_get -#define alc262_hp_master_sw_put alc262_hippo_master_sw_put - -static const struct snd_kcontrol_new alc262_hippo_mixer[] = { - ALC262_HIPPO_MASTER_SWITCH, - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 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_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -/* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc262_hippo_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 const struct snd_kcontrol_new alc262_sony_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - ALC262_HIPPO_MASTER_SWITCH, - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - { } /* end */ -}; - #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -213,12 +132,6 @@ static const struct hda_verb alc262_init_verbs[] = { { } }; -static const struct hda_verb alc262_eapd_verbs[] = { - {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, - { } -}; - /* Samsung Q1 Ultra Vista model setup */ static const struct snd_kcontrol_new alc262_ultra_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -342,16 +255,13 @@ static const struct snd_kcontrol_new alc262_ultra_capture_mixer[] = { */ static const char * const alc262_models[ALC262_MODEL_LAST] = { [ALC262_BASIC] = "basic", - [ALC262_HIPPO] = "hippo", [ALC262_ULTRA] = "ultra", [ALC262_AUTO] = "auto", }; static const struct snd_pci_quirk alc262_cfg_tbl[] = { - SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1", ALC262_ULTRA), - SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), {} }; @@ -366,20 +276,6 @@ static const struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, }, - [ALC262_HIPPO] = { - .mixers = { alc262_hippo_mixer }, - .init_verbs = { alc262_init_verbs, alc_hp15_unsol_verbs}, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .hp_nid = 0x03, - .dig_out_nid = ALC262_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc262_hippo_setup, - .init_hook = alc_inithook, - }, [ALC262_ULTRA] = { .mixers = { alc262_ultra_mixer }, .cap_mixer = alc262_ultra_capture_mixer, -- cgit v1.1 From 82e14a4754599952f5504c1f696bbc77bdfd6009 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Nov 2011 15:24:17 +0100 Subject: ALSA: hda/realtek - Drop ALC262 model=basic static configs Now most of ALC262 stuff has been moved to the auto-parser, and no longer need for keeping model=basic. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc262_quirks.c | 118 ------------------------------------------ sound/pci/hda/patch_realtek.c | 8 --- 2 files changed, 126 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c index fee7202..813855a 100644 --- a/sound/pci/hda/alc262_quirks.c +++ b/sound/pci/hda/alc262_quirks.c @@ -6,7 +6,6 @@ /* ALC262 models */ enum { ALC262_AUTO, - ALC262_BASIC, ALC262_ULTRA, ALC262_MODEL_LAST /* last tag */ }; @@ -23,115 +22,9 @@ enum { #define alc262_modes alc260_modes #define alc262_capture_source alc882_capture_source -static const hda_nid_t alc262_dmic_adc_nids[1] = { - /* ADC0 */ - 0x09 -}; - -static const hda_nid_t alc262_dmic_capsrc_nids[1] = { 0x22 }; - -static const struct snd_kcontrol_new alc262_base_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front 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("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("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), - { } /* end */ -}; - #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer -/* - * generic initialization of ADC, input mixers and output mixers - */ -static const struct hda_verb alc262_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)}, - - /* Mute 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)}, - - /* - * Set up output mixers (0x0c - 0x0e) - */ - /* 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}, - /* set up input amps for analog loopback */ - /* Amp Indices: DAC = 0, mixer = 1 */ - {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)}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, - - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, 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, (0x7000 | (0x00 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, - /* Input mixer2 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, - /* Input mixer3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, - - { } -}; - /* Samsung Q1 Ultra Vista model setup */ static const struct snd_kcontrol_new alc262_ultra_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -254,7 +147,6 @@ static const struct snd_kcontrol_new alc262_ultra_capture_mixer[] = { * configuration and preset */ static const char * const alc262_models[ALC262_MODEL_LAST] = { - [ALC262_BASIC] = "basic", [ALC262_ULTRA] = "ultra", [ALC262_AUTO] = "auto", }; @@ -266,16 +158,6 @@ static const struct snd_pci_quirk alc262_cfg_tbl[] = { }; static const struct alc_config_preset alc262_presets[] = { - [ALC262_BASIC] = { - .mixers = { alc262_base_mixer }, - .init_verbs = { alc262_init_verbs }, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_capture_source, - }, [ALC262_ULTRA] = { .mixers = { alc262_ultra_mixer }, .cap_mixer = alc262_ultra_capture_mixer, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5082070..1f3d168 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4483,14 +4483,6 @@ static int patch_alc262(struct hda_codec *codec) err = alc262_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 = ALC262_BASIC; - } -#endif } if (board_config != ALC_MODEL_AUTO) -- cgit v1.1 From 24de183ed0369d73af89e6752fcc1ecbde59053d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Nov 2011 17:13:39 +0100 Subject: ALSA: hda/realtek - Add the support of shared HP/Mic A machine like Q1-ultra which has only a single HP but no mic-jack, we can re-task the headhpone as an external mic jack. This was done formerly in ALC262 model=ultra quirk, and now the auto-parser supports this mode. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 65 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 64 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1f3d168..ef040ec 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -183,6 +183,7 @@ struct alc_spec { unsigned int single_input_src:1; unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */ unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */ + unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */ /* auto-mute control */ int automute_mode; @@ -277,6 +278,8 @@ static bool alc_dyn_adc_pcm_resetup(struct hda_codec *codec, int cur) return false; } +static void call_update_outputs(struct hda_codec *codec); + /* select the given imux item; either unmute exclusively or select the route */ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, unsigned int idx, bool force) @@ -298,6 +301,19 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, return 0; spec->cur_mux[adc_idx] = idx; + /* for shared I/O, change the pin-control accordingly */ + if (spec->shared_mic_hp) { + /* NOTE: this assumes that there are only two inputs, the + * first is the real internal mic and the second is HP jack. + */ + snd_hda_codec_write(codec, spec->autocfg.inputs[1].pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->cur_mux[adc_idx] ? + PIN_VREF80 : PIN_HP); + spec->automute_speaker = !spec->cur_mux[adc_idx]; + call_update_outputs(codec); + } + if (spec->dyn_adc_switch) { alc_dyn_adc_pcm_resetup(codec, idx); adc_idx = spec->dyn_adc_idx[idx]; @@ -547,7 +563,8 @@ static void update_outputs(struct hda_codec *codec) * in general, HP pins/amps control should be enabled in all cases, * but currently set only for master_mute, just to be safe */ - do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), + if (!spec->shared_mic_hp) /* don't change HP-pin when shared with mic */ + do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), spec->autocfg.hp_pins, spec->master_mute, true); if (!spec->automute_speaker) @@ -1115,6 +1132,9 @@ static void alc_init_auto_mic(struct hda_codec *codec) hda_nid_t fixed, ext, dock; int i; + if (spec->shared_mic_hp) + return; /* no auto-mic for the shared I/O */ + spec->ext_mic_idx = spec->int_mic_idx = spec->dock_mic_idx = -1; fixed = ext = dock = 0; @@ -2667,6 +2687,9 @@ static int alc_auto_fill_adc_caps(struct hda_codec *codec) int max_nums = ARRAY_SIZE(spec->private_adc_nids); int i, nums = 0; + if (spec->shared_mic_hp) + max_nums = 1; /* no multi streams with the shared HP/mic */ + nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { hda_nid_t src; @@ -2729,6 +2752,8 @@ static int alc_auto_create_input_ctls(struct hda_codec *codec) continue; label = hda_get_autocfg_input_label(codec, cfg, i); + if (spec->shared_mic_hp && !strcmp(label, "Misc")) + label = "Headphone Mic"; if (prev_label && !strcmp(label, prev_label)) type_idx++; else @@ -2764,6 +2789,39 @@ static int alc_auto_create_input_ctls(struct hda_codec *codec) return 0; } +/* create a shared input with the headphone out */ +static int alc_auto_create_shared_input(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int defcfg; + hda_nid_t nid; + + /* only one internal input pin? */ + if (cfg->num_inputs != 1) + return 0; + defcfg = snd_hda_codec_get_pincfg(codec, cfg->inputs[0].pin); + if (snd_hda_get_input_pin_attr(defcfg) != INPUT_PIN_ATTR_INT) + return 0; + + if (cfg->hp_outs == 1 && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) + nid = cfg->hp_pins[0]; /* OK, we have a single HP-out */ + else if (cfg->line_outs == 1 && cfg->line_out_type == AUTO_PIN_HP_OUT) + nid = cfg->line_out_pins[0]; /* OK, we have a single line-out */ + else + return 0; /* both not available */ + + if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_IN)) + return 0; /* no input */ + + cfg->inputs[1].pin = nid; + cfg->inputs[1].type = AUTO_PIN_MIC; + cfg->num_inputs = 2; + spec->shared_mic_hp = 1; + snd_printdd("realtek: Enable shared I/O jack on NID 0x%x\n", nid); + return 0; +} + static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid, unsigned int pin_type) { @@ -3654,6 +3712,8 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec) char boost_label[32]; label = hda_get_autocfg_input_label(codec, cfg, i); + if (spec->shared_mic_hp && !strcmp(label, "Misc")) + label = "Headphone Mic"; if (prev_label && !strcmp(label, prev_label)) type_idx++; else @@ -3859,6 +3919,9 @@ static int alc_parse_auto_config(struct hda_codec *codec, err = alc_auto_create_speaker_out(codec); if (err < 0) return err; + err = alc_auto_create_shared_input(codec); + if (err < 0) + return err; err = alc_auto_create_input_ctls(codec); if (err < 0) return err; -- cgit v1.1 From 42399f7a71df817a0aab82dd88dc05521c88385b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Nov 2011 17:18:44 +0100 Subject: ALSA: hda/realtek - Remove all ALC262-quirk codes Now that model=ultra is supported well by the auto-parser, we can get rid of the whole alc262_quirks.c and its related codes. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc262_quirks.c | 177 ------------------------------------------ sound/pci/hda/patch_realtek.c | 36 ++------- 2 files changed, 7 insertions(+), 206 deletions(-) delete mode 100644 sound/pci/hda/alc262_quirks.c (limited to 'sound/pci') diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c deleted file mode 100644 index 813855a..0000000 --- a/sound/pci/hda/alc262_quirks.c +++ /dev/null @@ -1,177 +0,0 @@ -/* - * ALC262 quirk models - * included by patch_realtek.c - */ - -/* ALC262 models */ -enum { - ALC262_AUTO, - ALC262_ULTRA, - ALC262_MODEL_LAST /* last tag */ -}; - -#define ALC262_DIGOUT_NID ALC880_DIGOUT_NID -#define ALC262_DIGIN_NID ALC880_DIGIN_NID - -#define alc262_dac_nids alc260_dac_nids -#define alc262_adc_nids alc882_adc_nids -#define alc262_adc_nids_alt alc882_adc_nids_alt -#define alc262_capsrc_nids alc882_capsrc_nids -#define alc262_capsrc_nids_alt alc882_capsrc_nids_alt - -#define alc262_modes alc260_modes -#define alc262_capture_source alc882_capture_source - -#define alc262_capture_mixer alc882_capture_mixer -#define alc262_capture_alt_mixer alc882_capture_alt_mixer - -/* Samsung Q1 Ultra Vista model setup */ -static const struct snd_kcontrol_new alc262_ultra_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, 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("Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Mic Boost Volume", 0x15, 0, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_verb alc262_ultra_verbs[] = { - /* output mixer */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* speaker */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - /* internal mic */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* ADC, choose mic */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(8)}, - {} -}; - -/* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc262_ultra_automute(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - unsigned int mute; - - mute = 0; - /* auto-mute only when HP is used as HP */ - if (!spec->cur_mux[0]) { - spec->hp_jack_present = snd_hda_jack_detect(codec, 0x15); - if (spec->hp_jack_present) - mute = HDA_AMP_MUTE; - } - /* mute/unmute internal speaker */ - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - /* mute/unmute HP */ - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute ? 0 : HDA_AMP_MUTE); -} - -/* unsolicited event for HP jack sensing */ -static void alc262_ultra_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC_HP_EVENT) - return; - alc262_ultra_automute(codec); -} - -static const struct hda_input_mux alc262_ultra_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x1 }, - { "Headphone", 0x7 }, - }, -}; - -static int alc262_ultra_mux_enum_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 ret; - - ret = alc_mux_enum_put(kcontrol, ucontrol); - if (!ret) - return 0; - /* reprogram the HP pin as mic or HP according to the input source */ - snd_hda_codec_write_cache(codec, 0x15, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - spec->cur_mux[0] ? PIN_VREF80 : PIN_HP); - alc262_ultra_automute(codec); /* mute/unmute HP */ - return ret; -} - -static const struct snd_kcontrol_new alc262_ultra_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc262_ultra_mux_enum_put, - }, - { - .iface = NID_MAPPING, - .name = "Capture Source", - .private_value = 0x15, - }, - { } /* end */ -}; - -/* - * configuration and preset - */ -static const char * const alc262_models[ALC262_MODEL_LAST] = { - [ALC262_ULTRA] = "ultra", - [ALC262_AUTO] = "auto", -}; - -static const struct snd_pci_quirk alc262_cfg_tbl[] = { - SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1", - ALC262_ULTRA), - {} -}; - -static const struct alc_config_preset alc262_presets[] = { - [ALC262_ULTRA] = { - .mixers = { alc262_ultra_mixer }, - .cap_mixer = alc262_ultra_capture_mixer, - .init_verbs = { alc262_ultra_verbs }, - .num_dacs = ARRAY_SIZE(alc262_dac_nids), - .dac_nids = alc262_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc262_modes), - .channel_mode = alc262_modes, - .input_mux = &alc262_ultra_capture_source, - .adc_nids = alc262_adc_nids, /* ADC0 */ - .capsrc_nids = alc262_capsrc_nids, - .num_adc_nids = 1, /* single ADC */ - .unsol_event = alc262_ultra_unsol_event, - .init_hook = alc262_ultra_automute, - }, -}; - diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ef040ec..afdecd8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4493,14 +4493,9 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = { /* */ -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS -#include "alc262_quirks.c" -#endif - static int patch_alc262(struct hda_codec *codec) { struct alc_spec *spec; - int board_config; int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -4527,29 +4522,13 @@ static int patch_alc262(struct hda_codec *codec) alc_fix_pll_init(codec, 0x20, 0x0a, 10); - board_config = alc_board_config(codec, ALC262_MODEL_LAST, - alc262_models, alc262_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, alc262_fixup_tbl, alc262_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - } - - if (board_config == ALC_MODEL_AUTO) { - /* automatic parse from the BIOS config */ - err = alc262_parse_auto_config(codec); - if (err < 0) - goto error; - } + alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - if (board_config != ALC_MODEL_AUTO) - setup_preset(codec, &alc262_presets[board_config]); + /* automatic parse from the BIOS config */ + err = alc262_parse_auto_config(codec); + if (err < 0) + goto error; if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4572,8 +4551,7 @@ static int patch_alc262(struct hda_codec *codec) spec->vmaster_nid = 0x0c; codec->patch_ops = alc_patch_ops; - if (board_config == ALC_MODEL_AUTO) - spec->init_hook = alc_auto_init_std; + spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; alc_init_jacks(codec); -- cgit v1.1 From 5c0ebfbe56795cce558736e0023ebb85b9f753c6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Nov 2011 17:59:13 +0100 Subject: ALSA: hda/realtek - Rewrite ALC882 model=vaio-tt with auto-parser Providing a pincfg fix for VAIO-TT with ALC889 codec to work with the auto-parser, and drop the static configuration. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 46 ------------------------------------------- sound/pci/hda/patch_realtek.c | 39 ++++++++++++++++++++++-------------- 2 files changed, 24 insertions(+), 61 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index e251514..59c556d 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -55,7 +55,6 @@ enum { ALC888_ASUS_EEE1601, ALC889A_MB31, ALC1200_ASUS_P5Q, - ALC883_SONY_VAIO_TT, ALC882_MODEL_LAST, }; @@ -2255,16 +2254,6 @@ static const struct snd_kcontrol_new alc889A_mb31_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc883_vaiott_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - static const struct hda_bind_ctls alc883_bind_cap_vol = { .ops = &snd_hda_bind_vol, .values = { @@ -2475,17 +2464,6 @@ static const struct hda_verb alc888_6st_dell_verbs[] = { { } }; -static const struct hda_verb alc883_vaiott_verbs[] = { - /* HP */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - - /* enable unsolicited event */ - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - - { } /* end */ -}; - static void alc888_3st_hp_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -2682,16 +2660,6 @@ static void alc888_lenovo_sky_setup(struct hda_codec *codec) alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } -static void alc883_vaiott_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[1] = 0x17; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - static const struct hda_verb alc888_asus_m90v_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -2832,7 +2800,6 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC889_INTEL] = "intel-x58", [ALC1200_ASUS_P5Q] = "asus-p5q", [ALC889A_MB31] = "mb31", - [ALC883_SONY_VAIO_TT] = "sony-vaio-tt", [ALC882_AUTO] = "auto", }; @@ -2887,7 +2854,6 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601), - SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT), SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC), @@ -3711,18 +3677,6 @@ static const struct alc_config_preset alc882_presets[] = { .unsol_event = alc889A_mb31_unsol_event, .init_hook = alc889A_mb31_automute, }, - [ALC883_SONY_VAIO_TT] = { - .mixers = { alc883_vaiott_mixer }, - .init_verbs = { alc883_init_verbs, alc883_vaiott_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .input_mux = &alc883_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc883_vaiott_setup, - .init_hook = alc_hp_automute, - }, }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index afdecd8..959bda3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4214,15 +4214,16 @@ static int patch_alc260(struct hda_codec *codec) * Pin config fixes */ enum { - PINFIX_ABIT_AW9D_MAX, - PINFIX_LENOVO_Y530, - PINFIX_PB_M5210, - PINFIX_ACER_ASPIRE_7736, - PINFIX_ASUS_W90V, + ALC882_FIXUP_ABIT_AW9D_MAX, + ALC882_FIXUP_LENOVO_Y530, + ALC882_FIXUP_PB_M5210, + ALC882_FIXUP_ACER_ASPIRE_7736, + ALC882_FIXUP_ASUS_W90V, + ALC889_FIXUP_VAIO_TT, }; static const struct alc_fixup alc882_fixups[] = { - [PINFIX_ABIT_AW9D_MAX] = { + [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { { 0x15, 0x01080104 }, /* side */ @@ -4231,7 +4232,7 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, - [PINFIX_LENOVO_Y530] = { + [ALC882_FIXUP_LENOVO_Y530] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { { 0x15, 0x99130112 }, /* rear int speakers */ @@ -4239,32 +4240,40 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, - [PINFIX_PB_M5210] = { + [ALC882_FIXUP_PB_M5210] = { .type = ALC_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 }, {} } }, - [PINFIX_ACER_ASPIRE_7736] = { + [ALC882_FIXUP_ACER_ASPIRE_7736] = { .type = ALC_FIXUP_SKU, .v.sku = ALC_FIXUP_SKU_IGNORE, }, - [PINFIX_ASUS_W90V] = { + [ALC882_FIXUP_ASUS_W90V] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { { 0x16, 0x99130110 }, /* fix sequence for CLFE */ { } } }, + [ALC889_FIXUP_VAIO_TT] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x17, 0x90170111 }, /* hidden surround speaker */ + { } + } + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { - SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210), - SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", PINFIX_ASUS_W90V), - SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530), - SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), - SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736), + SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), + SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), + SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), + SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), + SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), {} }; -- cgit v1.1 From 6fdfa361282a69fc9bfea4dd7d6303adc9e4d6df Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2011 12:32:50 +0100 Subject: ALSA: hda/realtek - Drop ALC882 model=asus-p5q static config It works well with the auto-parser and the default BIOS setup. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 15 --------------- 1 file changed, 15 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index 59c556d..7f2a3a7 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -54,7 +54,6 @@ enum { ALC888_ASUS_M90V, ALC888_ASUS_EEE1601, ALC889A_MB31, - ALC1200_ASUS_P5Q, ALC882_MODEL_LAST, }; @@ -2798,7 +2797,6 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", [ALC889A_INTEL] = "intel-alc889a", [ALC889_INTEL] = "intel-x58", - [ALC1200_ASUS_P5Q] = "asus-p5q", [ALC889A_MB31] = "mb31", [ALC882_AUTO] = "auto", }; @@ -2851,7 +2849,6 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601), SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), @@ -3649,18 +3646,6 @@ static const struct alc_config_preset alc882_presets[] = { .unsol_event = alc_sku_unsol_event, .init_hook = alc883_eee1601_inithook, }, - [ALC1200_ASUS_P5Q] = { - .mixers = { alc883_base_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC1200_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .slave_dig_outs = alc1200_slave_dig_outs, - .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), - .channel_mode = alc883_sixstack_modes, - .input_mux = &alc883_capture_source, - }, [ALC889A_MB31] = { .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer}, .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs, -- cgit v1.1 From 0e7cc2e745450daa0aec8f32d663f7811cfac0f3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2011 12:42:48 +0100 Subject: ALSA: hda/realtek - Move ALC888 ASUS EEE1601 config to auto-parser The ASUS EEE1601 works almost fine with the auto-parser but the static configuration has a certain specific COEF verb. Add this to the fix-up list so that we can drop the whole EEE1601 static config from alc882_quirks.c. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 74 ------------------------------------------- sound/pci/hda/patch_realtek.c | 10 ++++++ 2 files changed, 10 insertions(+), 74 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index 7f2a3a7..2be5e07 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -52,7 +52,6 @@ enum { ALC889A_INTEL, ALC889_INTEL, ALC888_ASUS_M90V, - ALC888_ASUS_EEE1601, ALC889A_MB31, ALC882_MODEL_LAST, }; @@ -625,14 +624,6 @@ static const struct hda_input_mux alc883_lenovo_sky_capture_source = { }, }; -static const struct hda_input_mux alc883_asus_eee1601_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x0 }, - { "Line", 0x2 }, - }, -}; - static const struct hda_input_mux alc889A_mb31_capture_source = { .num_items = 2, .items = { @@ -2271,33 +2262,6 @@ static const struct hda_bind_ctls alc883_bind_cap_switch = { }, }; -static const struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, 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, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = { - HDA_BIND_VOL("Capture Volume", &alc883_bind_cap_vol), - HDA_BIND_SW("Capture Switch", &alc883_bind_cap_switch), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, - { } /* end */ -}; - static const struct snd_kcontrol_new alc883_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -2683,28 +2647,6 @@ static void alc883_mode2_setup(struct hda_codec *codec) alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } -static const struct hda_verb alc888_asus_eee1601_verbs[] = { - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_COEF_INDEX, 0x0b}, - {0x20, AC_VERB_SET_PROC_COEF, 0x0838}, - /* enable unsolicited event */ - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - { } /* end */ -}; - -static void alc883_eee1601_inithook(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x1b; - alc_hp_automute(codec); -} - 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 */ @@ -2849,7 +2791,6 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601), SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), @@ -3631,21 +3572,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc883_mode2_setup, .init_hook = alc_inithook, }, - [ALC888_ASUS_EEE1601] = { - .mixers = { alc883_asus_eee1601_mixer }, - .cap_mixer = alc883_asus_eee1601_cap_mixer, - .init_verbs = { alc883_init_verbs, alc888_asus_eee1601_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .need_dac_fix = 1, - .input_mux = &alc883_asus_eee1601_capture_source, - .unsol_event = alc_sku_unsol_event, - .init_hook = alc883_eee1601_inithook, - }, [ALC889A_MB31] = { .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer}, .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 959bda3..75f739b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4220,6 +4220,7 @@ enum { ALC882_FIXUP_ACER_ASPIRE_7736, ALC882_FIXUP_ASUS_W90V, ALC889_FIXUP_VAIO_TT, + ALC888_FIXUP_EEE1601, }; static const struct alc_fixup alc882_fixups[] = { @@ -4265,11 +4266,20 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [ALC888_FIXUP_EEE1601] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x0b }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0838 }, + { } + } + } }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), + SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), -- cgit v1.1 From 596830ee1d2d9cf56e5efe0c020eb588beecae62 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2011 15:06:45 +0100 Subject: ALSA: hda/realtek - Look through codec SSID for fix-up lists Not only PCI SSIDs but also look through codec SSIDs for fix-up table entries. MacBook tend to give the same PCI SSID but unique codec SSIDs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++---- 1 file changed, 18 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 75f739b..640cf28 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1526,6 +1526,7 @@ static void alc_pick_fixup(struct hda_codec *codec, const struct alc_fixup *fixlist) { struct alc_spec *spec = codec->spec; + const struct snd_pci_quirk *q; int id = -1; const char *name = NULL; @@ -1540,12 +1541,25 @@ static void alc_pick_fixup(struct hda_codec *codec, } } if (id < 0) { - quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (quirk) { - id = quirk->value; + q = snd_pci_quirk_lookup(codec->bus->pci, quirk); + if (q) { + id = q->value; +#ifdef CONFIG_SND_DEBUG_VERBOSE + name = q->name; +#endif + } + } + if (id < 0) { + for (q = quirk; q->subvendor; q++) { + unsigned int vendorid = + q->subdevice | (q->subvendor << 16); + if (vendorid == codec->subsystem_id) { + id = q->value; #ifdef CONFIG_SND_DEBUG_VERBOSE - name = quirk->name; + name = q->name; #endif + break; + } } } -- cgit v1.1 From 177943a39aabec31cdbd529fd5f3060850f5aef9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2011 12:55:18 +0100 Subject: ALSA: hda/realtek - Drop ALC882 asus-a7j and asus-a7m models These models work fine with the auto-parser with the additional COEF setup. The iMac 7,1 (106b:3200) also uses the same quirk, so remove it too. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 106 ------------------------------------------ sound/pci/hda/patch_realtek.c | 14 +++++- 2 files changed, 13 insertions(+), 107 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index 2be5e07..c3438da 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -11,8 +11,6 @@ enum { ALC882_ARIMA, ALC882_W2JC, ALC882_TARGA, - ALC882_ASUS_A7J, - ALC882_ASUS_A7M, ALC885_MACPRO, ALC885_MBA21, ALC885_MBP3, @@ -1140,40 +1138,6 @@ static const struct snd_kcontrol_new alc882_targa_mixer[] = { { } /* end */ }; -/* Pin assignment: Front=0x14, HP = 0x15, Front = 0x16, ??? - * Front Mic=0x18, Line In = 0x1a, Line In = 0x1b, CD = 0x1c - */ -static const struct snd_kcontrol_new alc882_asus_a7j_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mobile Front Playback Switch", 0x16, 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("Mobile Line Playback Volume", 0x0b, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Mobile Line Playback Switch", 0x0b, 0x03, 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("Mic Boost Volume", 0x18, 0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc882_asus_a7m_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("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), - { } /* end */ -}; - static const struct snd_kcontrol_new alc882_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1770,42 +1734,6 @@ static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res) alc882_targa_automute(codec); } -static const struct hda_verb alc882_asus_a7j_verbs[] = { - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */ - - {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */ - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ - { } /* end */ -}; - -static const struct hda_verb alc882_asus_a7m_verbs[] = { - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */ - - {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */ - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ - { } /* end */ -}; - static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted) { unsigned int gpiostate, gpiomask, gpiodir; @@ -2699,8 +2627,6 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC882_ARIMA] = "arima", [ALC882_W2JC] = "w2jc", [ALC882_TARGA] = "targa", - [ALC882_ASUS_A7J] = "asus-a7j", - [ALC882_ASUS_A7M] = "asus-a7m", [ALC885_MACPRO] = "macpro", [ALC885_MB5] = "mb5", [ALC885_MACMINI3] = "macmini3", @@ -2782,9 +2708,6 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP), - SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J), - SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J), - SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M), SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC), SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), @@ -2876,7 +2799,6 @@ static const struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_IMAC24), SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3), SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889A_MB31), - SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_ASUS_A7M), 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), @@ -3059,34 +2981,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc882_targa_setup, .init_hook = alc882_targa_automute, }, - [ALC882_ASUS_A7J] = { - .mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, - alc882_asus_a7j_verbs}, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), - .adc_nids = alc882_adc_nids, - .capsrc_nids = alc882_capsrc_nids, - .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes), - .channel_mode = alc882_3ST_6ch_modes, - .need_dac_fix = 1, - .input_mux = &alc882_capture_source, - }, - [ALC882_ASUS_A7M] = { - .mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, - alc882_eapd_verbs, alc880_gpio1_init_verbs, - alc882_asus_a7m_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), - .channel_mode = alc880_threestack_modes, - .need_dac_fix = 1, - .input_mux = &alc882_capture_source, - }, [ALC883_3ST_2ch_DIG] = { .mixers = { alc883_3ST_2ch_mixer }, .init_verbs = { alc883_init_verbs }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 640cf28..43c7aea 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4235,6 +4235,7 @@ enum { ALC882_FIXUP_ASUS_W90V, ALC889_FIXUP_VAIO_TT, ALC888_FIXUP_EEE1601, + ALC882_FIXUP_EAPD, }; static const struct alc_fixup alc882_fixups[] = { @@ -4287,14 +4288,25 @@ static const struct alc_fixup alc882_fixups[] = { { 0x20, AC_VERB_SET_PROC_COEF, 0x0838 }, { } } - } + }, + [ALC882_FIXUP_EAPD] = { + .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 alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), + SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), + SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD), /* codec SSID */ SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), -- cgit v1.1 From 8af2d066d145f6fa97eb16145a4a918c5edfddc6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2011 13:06:37 +0100 Subject: ALSA: hda/realtek - Drop lenovo-sky, asus-m90v, fujitsu-pi2515 quirks These machines are working well with the auto-parser without static configurations. More diet. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 181 ------------------------------------------ 1 file changed, 181 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index c3438da..88f93fc 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -37,19 +37,16 @@ enum { ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, ALC888_LENOVO_MS7195_DIG, - ALC888_LENOVO_SKY, ALC883_HAIER_W66, ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, ALC883_CLEVO_M540R, ALC883_CLEVO_M720, - ALC883_FUJITSU_PI2515, ALC888_FUJITSU_XA3530, ALC883_3ST_6ch_INTEL, ALC889A_INTEL, ALC889_INTEL, - ALC888_ASUS_M90V, ALC889A_MB31, ALC882_MODEL_LAST, }; @@ -605,23 +602,6 @@ static const struct hda_input_mux alc883_lenovo_nb0763_capture_source = { }, }; -static const struct hda_input_mux alc883_fujitsu_pi2515_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x0 }, - { "Internal Mic", 0x1 }, - }, -}; - -static const struct hda_input_mux alc883_lenovo_sky_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x4 }, - }, -}; - static const struct hda_input_mux alc889A_mb31_capture_source = { .num_items = 2, .items = { @@ -1863,20 +1843,6 @@ static const struct snd_kcontrol_new alc883_clevo_m720_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_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("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("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - static const struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -2121,31 +2087,6 @@ static const struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 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("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_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), @@ -2258,20 +2199,6 @@ static const struct hda_verb alc883_clevo_m720_verbs[] = { { } /* end */ }; -static const struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = { - /* HP */ - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Subwoofer */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* enable unsolicited event */ - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - - { } /* end */ -}; - static const struct hda_verb alc883_targa_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -2338,18 +2265,6 @@ static const struct hda_verb alc883_haier_w66_verbs[] = { { } /* end */ }; -static const struct hda_verb alc888_lenovo_sky_verbs[] = { - {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)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - { } /* end */ -}; - static const struct hda_verb alc888_6st_dell_verbs[] = { {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, { } @@ -2468,16 +2383,6 @@ static void alc883_clevo_m720_unsol_event(struct hda_codec *codec, } } -/* toggle speaker-output according to the hp-jack state */ -static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x15; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - static void alc883_haier_w66_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -2538,43 +2443,6 @@ static void alc888_6st_dell_setup(struct hda_codec *codec) alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } -static void alc888_lenovo_sky_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[1] = 0x15; - spec->autocfg.speaker_pins[2] = 0x16; - spec->autocfg.speaker_pins[3] = 0x17; - spec->autocfg.speaker_pins[4] = 0x1a; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static const struct hda_verb alc888_asus_m90v_verbs[] = { - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* enable unsolicited event */ - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN}, - { } /* end */ -}; - -static void alc883_mode2_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[1] = 0x15; - spec->autocfg.speaker_pins[2] = 0x16; - spec->ext_mic_pin = 0x18; - spec->int_mic_pin = 0x19; - spec->auto_mic = 1; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - 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 */ @@ -2653,14 +2521,12 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC883_LENOVO_101E_2ch] = "lenovo-101e", [ALC883_LENOVO_NB0763] = "lenovo-nb0763", [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig", - [ALC888_LENOVO_SKY] = "lenovo-sky", [ALC883_HAIER_W66] = "haier-w66", [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", [ALC883_CLEVO_M540R] = "clevo-m540r", [ALC883_CLEVO_M720] = "clevo-m720", - [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515", [ALC888_FUJITSU_XA3530] = "fujitsu-xa3530", [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", [ALC889A_INTEL] = "intel-alc889a", @@ -2708,7 +2574,6 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP), - SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC), SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), @@ -2765,15 +2630,12 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), /* SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), */ SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), - SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1100, "FSC AMILO Xi/Pi25xx", - ALC883_FUJITSU_PI2515), SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1130, "Fujitsu AMILO Xa35xx", ALC888_FUJITSU_XA3530), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), - SND_PCI_QUIRK(0x17aa, 0x101d, "Lenovo Sky", ALC888_LENOVO_SKY), SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG), SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), @@ -3404,20 +3266,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc883_mitac_setup, .init_hook = alc_hp_automute, }, - [ALC883_FUJITSU_PI2515] = { - .mixers = { alc883_2ch_fujitsu_pi2515_mixer }, - .init_verbs = { alc883_init_verbs, - alc883_2ch_fujitsu_pi2515_verbs}, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .input_mux = &alc883_fujitsu_pi2515_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc883_2ch_fujitsu_pi2515_setup, - .init_hook = alc_hp_automute, - }, [ALC888_FUJITSU_XA3530] = { .mixers = { alc888_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, @@ -3437,35 +3285,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc888_fujitsu_xa3530_setup, .init_hook = alc_hp_automute, }, - [ALC888_LENOVO_SKY] = { - .mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc888_lenovo_sky_verbs}, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), - .channel_mode = alc883_sixstack_modes, - .need_dac_fix = 1, - .input_mux = &alc883_lenovo_sky_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc888_lenovo_sky_setup, - .init_hook = alc_hp_automute, - }, - [ALC888_ASUS_M90V] = { - .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc888_asus_m90v_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), - .channel_mode = alc883_3ST_6ch_modes, - .need_dac_fix = 1, - .input_mux = &alc883_fujitsu_pi2515_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc883_mode2_setup, - .init_hook = alc_inithook, - }, [ALC889A_MB31] = { .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer}, .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs, -- cgit v1.1 From f01f587bb779aabf220233b3aac52b437114ec84 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2011 14:32:38 +0100 Subject: ALSA: hda/realtek - Drop ALC882 lenovo and haier-w66 static configs Remove all ALC882 static configurations for all Lenovo and Haier-w66 quirks. They work fine with the auto-parser now. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 188 ------------------------------------------ 1 file changed, 188 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index 88f93fc..a71077d 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -34,10 +34,6 @@ enum { ALC883_MEDION, ALC883_MEDION_WIM2160, ALC883_LAPTOP_EAPD, - ALC883_LENOVO_101E_2ch, - ALC883_LENOVO_NB0763, - ALC888_LENOVO_MS7195_DIG, - ALC883_HAIER_W66, ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, @@ -584,24 +580,6 @@ static const struct hda_input_mux alc883_3stack_6ch_intel = { }, }; -static const struct hda_input_mux alc883_lenovo_101e_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x1 }, - { "Line", 0x2 }, - }, -}; - -static const struct hda_input_mux alc883_lenovo_nb0763_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Internal Mic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - static const struct hda_input_mux alc889A_mb31_capture_source = { .num_items = 2, .items = { @@ -2000,31 +1978,6 @@ static const struct snd_kcontrol_new alc883_targa_8ch_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front 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_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT), - 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("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 alc883_medion_wim2160_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -2228,43 +2181,6 @@ static const struct hda_verb alc883_targa_verbs[] = { { } /* end */ }; -static const struct hda_verb alc883_lenovo_101e_verbs[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_FRONT_EVENT|AC_USRSP_EN}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT|AC_USRSP_EN}, - { } /* end */ -}; - -static const struct hda_verb alc883_lenovo_nb0763_verbs[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - { } /* end */ -}; - -static const struct hda_verb alc888_lenovo_ms7195_verbs[] = { - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_FRONT_EVENT | AC_USRSP_EN}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - { } /* end */ -}; - -static const struct hda_verb alc883_haier_w66_verbs[] = { - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - { } /* end */ -}; - static const struct hda_verb alc888_6st_dell_verbs[] = { {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, { } @@ -2331,26 +2247,6 @@ static const struct hda_channel_mode alc888_3st_hp_modes[3] = { { 6, alc888_3st_hp_6ch_init }, }; -static void alc888_lenovo_ms7195_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.line_out_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x15; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -/* toggle speaker-output according to the hp-jack state */ -static void alc883_lenovo_nb0763_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x15; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - /* toggle speaker-output according to the hp-jack state */ #define alc883_targa_init_hook alc882_targa_init_hook #define alc883_targa_unsol_event alc882_targa_unsol_event @@ -2383,25 +2279,6 @@ static void alc883_clevo_m720_unsol_event(struct hda_codec *codec, } } -static void alc883_haier_w66_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.speaker_pins[0] = 0x14; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static void alc883_lenovo_101e_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.line_out_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x15; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - /* toggle speaker-output according to the hp-jack state */ static void alc883_acer_aspire_setup(struct hda_codec *codec) { @@ -2518,10 +2395,6 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC883_MEDION] = "medion", [ALC883_MEDION_WIM2160] = "medion-wim2160", [ALC883_LAPTOP_EAPD] = "laptop-eapd", - [ALC883_LENOVO_101E_2ch] = "lenovo-101e", - [ALC883_LENOVO_NB0763] = "lenovo-nb0763", - [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig", - [ALC883_HAIER_W66] = "haier-w66", [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", @@ -2632,13 +2505,7 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1130, "Fujitsu AMILO Xa35xx", ALC888_FUJITSU_XA3530), - SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), - SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), - SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763), - SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), - SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG), SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), @@ -3172,61 +3039,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc883_clevo_m720_setup, .init_hook = alc883_clevo_m720_init_hook, }, - [ALC883_LENOVO_101E_2ch] = { - .mixers = { alc883_lenovo_101e_2ch_mixer}, - .init_verbs = { alc883_init_verbs, alc883_lenovo_101e_verbs}, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .adc_nids = alc883_adc_nids_alt, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt), - .capsrc_nids = alc883_capsrc_nids, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .input_mux = &alc883_lenovo_101e_capture_source, - .setup = alc883_lenovo_101e_setup, - .unsol_event = alc_sku_unsol_event, - .init_hook = alc_inithook, - }, - [ALC883_LENOVO_NB0763] = { - .mixers = { alc883_lenovo_nb0763_mixer }, - .init_verbs = { alc883_init_verbs, alc883_lenovo_nb0763_verbs}, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .need_dac_fix = 1, - .input_mux = &alc883_lenovo_nb0763_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc883_lenovo_nb0763_setup, - .init_hook = alc_hp_automute, - }, - [ALC888_LENOVO_MS7195_DIG] = { - .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc888_lenovo_ms7195_verbs}, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), - .channel_mode = alc883_3ST_6ch_modes, - .need_dac_fix = 1, - .input_mux = &alc883_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc888_lenovo_ms7195_setup, - .init_hook = alc_inithook, - }, - [ALC883_HAIER_W66] = { - .mixers = { alc883_targa_2ch_mixer}, - .init_verbs = { alc883_init_verbs, alc883_haier_w66_verbs}, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .input_mux = &alc883_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc883_haier_w66_setup, - .init_hook = alc_hp_automute, - }, [ALC888_3ST_HP] = { .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs }, -- cgit v1.1 From ed63a88775c183dd3d6ee0bde0670960754e944a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2011 14:56:13 +0100 Subject: ALSA: hda/realtek - Drop ALC882 mitac and fujitsu-xa3530 static configs These are working well with the auto-parser although they have relatively complex setup. Let's go forward. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 131 ------------------------------------------ 1 file changed, 131 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index a71077d..2c5c181 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -36,10 +36,8 @@ enum { ALC883_LAPTOP_EAPD, ALC888_3ST_HP, ALC888_6ST_DELL, - ALC883_MITAC, ALC883_CLEVO_M540R, ALC883_CLEVO_M720, - ALC888_FUJITSU_XA3530, ALC883_3ST_6ch_INTEL, ALC889A_INTEL, ALC889_INTEL, @@ -114,43 +112,6 @@ static const struct hda_channel_mode alc888_4ST_8ch_intel_modes[4] = { { 8, alc888_4ST_ch8_intel_init }, }; -/* - * ALC888 Fujitsu Siemens Amillo xa3530 - */ - -static const struct hda_verb alc888_fujitsu_xa3530_verbs[] = { -/* Front Mic: set to PIN_IN (empty by default) */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, -/* Connect Internal HP to Front */ - {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}, -/* Connect Bass HP to Front */ - {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, 0x00}, -/* Connect Line-Out side jack (SPDIF) to Side */ - {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}, -/* Connect Mic jack to CLFE */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, -/* Connect Line-in jack to Surround */ - {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}, -/* Connect HP out jack to Front */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, -/* Enable unsolicited event for HP jack and Line-out jack */ - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - {} -}; - static void alc889_automute_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -170,17 +131,6 @@ static void alc889_intel_init_hook(struct hda_codec *codec) alc_hp_automute(codec); } -static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x17; /* line-out */ - spec->autocfg.hp_pins[1] = 0x1b; /* hp */ - spec->autocfg.speaker_pins[0] = 0x14; /* speaker */ - spec->autocfg.speaker_pins[1] = 0x15; /* bass */ - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - /* * ALC888 Acer Aspire 4930G model */ @@ -1790,23 +1740,6 @@ static const struct hda_verb alc883_medion_eapd_verbs[] = { #define alc883_base_mixer alc882_base_mixer -static const struct snd_kcontrol_new alc883_mitac_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 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", 0x15, 0x0, HDA_OUTPUT), - 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 alc883_clevo_m720_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), @@ -2095,32 +2028,6 @@ static const struct snd_kcontrol_new alc883_chmode_mixer[] = { { } /* end */ }; -/* toggle speaker-output according to the hp-jack state */ -static void alc883_mitac_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[1] = 0x17; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static const struct hda_verb alc883_mitac_verbs[] = { - /* HP */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Subwoofer */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* enable unsolicited event */ - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - /* {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN}, */ - - { } /* end */ -}; - static const struct hda_verb alc883_clevo_m540r_verbs[] = { /* HP */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -2397,10 +2304,8 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC883_LAPTOP_EAPD] = "laptop-eapd", [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", - [ALC883_MITAC] = "mitac", [ALC883_CLEVO_M540R] = "clevo-m540r", [ALC883_CLEVO_M720] = "clevo-m720", - [ALC888_FUJITSU_XA3530] = "fujitsu-xa3530", [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", [ALC889A_INTEL] = "intel-alc889a", [ALC889_INTEL] = "intel-x58", @@ -2455,8 +2360,6 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC), - SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL), SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), @@ -2503,13 +2406,10 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), /* SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), */ SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), - SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1130, "Fujitsu AMILO Xa35xx", - ALC888_FUJITSU_XA3530), SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG), SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), - SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC), SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC889_INTEL), SND_PCI_QUIRK(0x8086, 0x0021, "Intel IbexPeak", ALC889A_INTEL), SND_PCI_QUIRK(0x8086, 0x3b56, "Intel IbexPeak", ALC889A_INTEL), @@ -3066,37 +2966,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc888_6st_dell_setup, .init_hook = alc_hp_automute, }, - [ALC883_MITAC] = { - .mixers = { alc883_mitac_mixer }, - .init_verbs = { alc883_init_verbs, alc883_mitac_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .input_mux = &alc883_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc883_mitac_setup, - .init_hook = alc_hp_automute, - }, - [ALC888_FUJITSU_XA3530] = { - .mixers = { alc888_base_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, - alc888_fujitsu_xa3530_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), - .adc_nids = alc883_adc_nids_rev, - .capsrc_nids = alc883_capsrc_nids_rev, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc888_4ST_8ch_intel_modes), - .channel_mode = alc888_4ST_8ch_intel_modes, - .num_mux_defs = - ARRAY_SIZE(alc888_2_capture_sources), - .input_mux = alc888_2_capture_sources, - .unsol_event = alc_sku_unsol_event, - .setup = alc888_fujitsu_xa3530_setup, - .init_hook = alc_hp_automute, - }, [ALC889A_MB31] = { .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer}, .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs, -- cgit v1.1 From 7a6069bf64d22e1ca5413acf494dafb4200be44c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2011 15:22:01 +0100 Subject: ALSA: hda/realtek - Replace ALC882 arima, medion and laptop-eapd quirks Move these quitks to the auto-parser. They just need some EAPD setups in addition. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 104 ------------------------------------------ sound/pci/hda/patch_realtek.c | 13 ++++++ 2 files changed, 13 insertions(+), 104 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index 2c5c181..0f7f5e7 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -8,7 +8,6 @@ enum { ALC882_AUTO, ALC882_3ST_DIG, ALC882_6ST_DIG, - ALC882_ARIMA, ALC882_W2JC, ALC882_TARGA, ALC885_MACPRO, @@ -31,9 +30,6 @@ enum { ALC888_ACER_ASPIRE_6530G, ALC888_ACER_ASPIRE_8930G, ALC888_ACER_ASPIRE_7730G, - ALC883_MEDION, - ALC883_MEDION_WIM2160, - ALC883_LAPTOP_EAPD, ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_CLEVO_M540R, @@ -1731,13 +1727,6 @@ static const struct hda_channel_mode alc889A_mb31_6ch_modes[4] = { { 6, alc889A_mb31_ch6_init }, }; -static const struct hda_verb alc883_medion_eapd_verbs[] = { - /* eanable EAPD on medion laptop */ - {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, - {0x20, AC_VERB_SET_PROC_COEF, 0x3070}, - { } -}; - #define alc883_base_mixer alc882_base_mixer static const struct snd_kcontrol_new alc883_clevo_m720_mixer[] = { @@ -1911,43 +1900,6 @@ static const struct snd_kcontrol_new alc883_targa_8ch_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x08, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_verb alc883_medion_wim2160_verbs[] = { - /* Unmute front mixer */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - /* Set speaker pin to front mixer */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Init headphone pin */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - - { } /* end */ -}; - -/* toggle speaker-output according to the hp-jack state */ -static void alc883_medion_wim2160_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1a; - spec->autocfg.speaker_pins[0] = 0x15; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - static const struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -2276,7 +2228,6 @@ static const hda_nid_t alc1200_slave_dig_outs[] = { static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC882_3ST_DIG] = "3stack-dig", [ALC882_6ST_DIG] = "6stack-dig", - [ALC882_ARIMA] = "arima", [ALC882_W2JC] = "w2jc", [ALC882_TARGA] = "targa", [ALC885_MACPRO] = "macpro", @@ -2299,9 +2250,6 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g", [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", - [ALC883_MEDION] = "medion", - [ALC883_MEDION_WIM2160] = "medion-wim2160", - [ALC883_LAPTOP_EAPD] = "laptop-eapd", [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", [ALC883_CLEVO_M540R] = "clevo-m540r", @@ -2360,7 +2308,6 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL), SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), @@ -2402,10 +2349,7 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R), - SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), - /* SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), */ - SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG), SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), @@ -2471,16 +2415,6 @@ static const struct alc_config_preset alc882_presets[] = { .channel_mode = alc882_sixstack_modes, .input_mux = &alc882_capture_source, }, - [ALC882_ARIMA] = { - .mixers = { alc882_base_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, - alc882_eapd_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), - .channel_mode = alc882_sixstack_modes, - .input_mux = &alc882_capture_source, - }, [ALC882_W2JC] = { .mixers = { alc882_w2jc_mixer, alc882_chmode_mixer }, .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, @@ -2873,44 +2807,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc888_acer_aspire_7730g_setup, .init_hook = alc_hp_automute, }, - [ALC883_MEDION] = { - .mixers = { alc883_fivestack_mixer, - alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, - alc883_medion_eapd_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .adc_nids = alc883_adc_nids_alt, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt), - .capsrc_nids = alc883_capsrc_nids, - .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), - .channel_mode = alc883_sixstack_modes, - .input_mux = &alc883_capture_source, - }, - [ALC883_MEDION_WIM2160] = { - .mixers = { alc883_medion_wim2160_mixer }, - .init_verbs = { alc883_init_verbs, alc883_medion_wim2160_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .input_mux = &alc883_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc883_medion_wim2160_setup, - .init_hook = alc_hp_automute, - }, - [ALC883_LAPTOP_EAPD] = { - .mixers = { alc883_base_mixer }, - .init_verbs = { alc883_init_verbs, alc882_eapd_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .input_mux = &alc883_capture_source, - }, [ALC883_CLEVO_M540R] = { .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc883_clevo_m540r_verbs }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 43c7aea..422430d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4236,6 +4236,7 @@ enum { ALC889_FIXUP_VAIO_TT, ALC888_FIXUP_EEE1601, ALC882_FIXUP_EAPD, + ALC883_FIXUP_EAPD, }; static const struct alc_fixup alc882_fixups[] = { @@ -4298,6 +4299,15 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [ALC883_FIXUP_EAPD] = { + .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 }, + { } + } + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -4307,9 +4317,12 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD), /* codec SSID */ + SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), + SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), + SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), {} }; -- cgit v1.1 From 145fa008a208e824567d2d9b26133a4cd0e7fdbd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2011 17:28:27 +0100 Subject: ALSA: hda/realtek - Drop ALC882 3stack-hp, 6stack-dell and clevo-m540r models These static configs are no longer needed by replacement with the auto-parser. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 150 ------------------------------------------ 1 file changed, 150 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index 0f7f5e7..80321a1 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -30,9 +30,6 @@ enum { ALC888_ACER_ASPIRE_6530G, ALC888_ACER_ASPIRE_8930G, ALC888_ACER_ASPIRE_7730G, - ALC888_3ST_HP, - ALC888_6ST_DELL, - ALC883_CLEVO_M540R, ALC883_CLEVO_M720, ALC883_3ST_6ch_INTEL, ALC889A_INTEL, @@ -1980,22 +1977,6 @@ static const struct snd_kcontrol_new alc883_chmode_mixer[] = { { } /* end */ }; -static const struct hda_verb alc883_clevo_m540r_verbs[] = { - /* HP */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Int speaker */ - /*{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},*/ - - /* enable unsolicited event */ - /* - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN}, - */ - - { } /* end */ -}; - static const struct hda_verb alc883_clevo_m720_verbs[] = { /* HP */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -2040,72 +2021,6 @@ static const struct hda_verb alc883_targa_verbs[] = { { } /* end */ }; -static const struct hda_verb alc888_6st_dell_verbs[] = { - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - { } -}; - -static void alc888_3st_hp_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[1] = 0x16; - spec->autocfg.speaker_pins[2] = 0x18; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static const struct hda_verb alc888_3st_hp_verbs[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ - {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - { } /* end */ -}; - -/* - * 2ch mode - */ -static const struct hda_verb alc888_3st_hp_2ch_init[] = { - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; - -/* - * 4ch mode - */ -static const struct hda_verb alc888_3st_hp_4ch_init[] = { - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 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, 0x01 }, - { } /* end */ -}; - -/* - * 6ch mode - */ -static const struct hda_verb alc888_3st_hp_6ch_init[] = { - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 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, 0x01 }, - { } /* end */ -}; - -static const struct hda_channel_mode alc888_3st_hp_modes[3] = { - { 2, alc888_3st_hp_2ch_init }, - { 4, alc888_3st_hp_4ch_init }, - { 6, alc888_3st_hp_6ch_init }, -}; - /* toggle speaker-output according to the hp-jack state */ #define alc883_targa_init_hook alc882_targa_init_hook #define alc883_targa_unsol_event alc882_targa_unsol_event @@ -2167,18 +2082,6 @@ static const struct hda_verb alc883_acer_eapd_verbs[] = { { } }; -static void alc888_6st_dell_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[1] = 0x15; - spec->autocfg.speaker_pins[2] = 0x16; - spec->autocfg.speaker_pins[3] = 0x17; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - 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 */ @@ -2250,9 +2153,6 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g", [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", - [ALC888_3ST_HP] = "3stack-hp", - [ALC888_6ST_DELL] = "6stack-dell", - [ALC883_CLEVO_M540R] = "clevo-m540r", [ALC883_CLEVO_M720] = "clevo-m720", [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", [ALC889A_INTEL] = "intel-alc889a", @@ -2291,14 +2191,8 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { */ /* SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), */ - SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), - SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavilion", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), - SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP), - SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP), SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC), SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), @@ -2308,7 +2202,6 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL), SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), @@ -2348,7 +2241,6 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1558, 0x0571, "Clevo laptop M570U", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), - SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG), @@ -2807,21 +2699,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc888_acer_aspire_7730g_setup, .init_hook = alc_hp_automute, }, - [ALC883_CLEVO_M540R] = { - .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc883_clevo_m540r_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_clevo_modes), - .channel_mode = alc883_3ST_6ch_clevo_modes, - .need_dac_fix = 1, - .input_mux = &alc883_capture_source, - /* This machine has the hardware HP auto-muting, thus - * we need no software mute via unsol event - */ - }, [ALC883_CLEVO_M720] = { .mixers = { alc883_clevo_m720_mixer }, .init_verbs = { alc883_init_verbs, alc883_clevo_m720_verbs }, @@ -2835,33 +2712,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc883_clevo_m720_setup, .init_hook = alc883_clevo_m720_init_hook, }, - [ALC888_3ST_HP] = { - .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc888_3st_hp_modes), - .channel_mode = alc888_3st_hp_modes, - .need_dac_fix = 1, - .input_mux = &alc883_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc888_3st_hp_setup, - .init_hook = alc_hp_automute, - }, - [ALC888_6ST_DELL] = { - .mixers = { alc883_base_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc888_6st_dell_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), - .channel_mode = alc883_sixstack_modes, - .input_mux = &alc883_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc888_6st_dell_setup, - .init_hook = alc_hp_automute, - }, [ALC889A_MB31] = { .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer}, .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs, -- cgit v1.1 From b3ca3bf5e8e0fb445a81aef02092ae0b11fa4482 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2011 17:32:39 +0100 Subject: ALSA: hda/realtek - Drop ALC882 model=clevo-m720 quirk This works well without any special handling with the auto-parser. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 74 ------------------------------------------- 1 file changed, 74 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index 80321a1..370b5a7 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -30,7 +30,6 @@ enum { ALC888_ACER_ASPIRE_6530G, ALC888_ACER_ASPIRE_8930G, ALC888_ACER_ASPIRE_7730G, - ALC883_CLEVO_M720, ALC883_3ST_6ch_INTEL, ALC889A_INTEL, ALC889_INTEL, @@ -1726,20 +1725,6 @@ static const struct hda_channel_mode alc889A_mb31_6ch_modes[4] = { #define alc883_base_mixer alc882_base_mixer -static const struct snd_kcontrol_new alc883_clevo_m720_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("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("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - static const struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -1977,21 +1962,6 @@ static const struct snd_kcontrol_new alc883_chmode_mixer[] = { { } /* end */ }; -static const struct hda_verb alc883_clevo_m720_verbs[] = { - /* HP */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Int speaker */ - {0x14, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* enable unsolicited event */ - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN}, - - { } /* end */ -}; - static const struct hda_verb alc883_targa_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -2025,34 +1995,6 @@ static const struct hda_verb alc883_targa_verbs[] = { #define alc883_targa_init_hook alc882_targa_init_hook #define alc883_targa_unsol_event alc882_targa_unsol_event -static void alc883_clevo_m720_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 alc883_clevo_m720_init_hook(struct hda_codec *codec) -{ - alc_hp_automute(codec); - alc88x_simple_mic_automute(codec); -} - -static void alc883_clevo_m720_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC_MIC_EVENT: - alc88x_simple_mic_automute(codec); - break; - default: - alc_sku_unsol_event(codec, res); - break; - } -} - /* toggle speaker-output according to the hp-jack state */ static void alc883_acer_aspire_setup(struct hda_codec *codec) { @@ -2153,7 +2095,6 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g", [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", - [ALC883_CLEVO_M720] = "clevo-m720", [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", [ALC889A_INTEL] = "intel-alc889a", [ALC889_INTEL] = "intel-x58", @@ -2239,8 +2180,6 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x0571, "Clevo laptop M570U", ALC883_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), - SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG), @@ -2699,19 +2638,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc888_acer_aspire_7730g_setup, .init_hook = alc_hp_automute, }, - [ALC883_CLEVO_M720] = { - .mixers = { alc883_clevo_m720_mixer }, - .init_verbs = { alc883_init_verbs, alc883_clevo_m720_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .input_mux = &alc883_capture_source, - .unsol_event = alc883_clevo_m720_unsol_event, - .setup = alc883_clevo_m720_setup, - .init_hook = alc883_clevo_m720_init_hook, - }, [ALC889A_MB31] = { .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer}, .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs, -- cgit v1.1 From 25da1f86b4e7f5fef26cec1f65be3b120f2d36ac Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2011 17:33:27 +0100 Subject: ALSA: hda/realtek - Drop ALC882 model=acer quirk This quirk is anyway not used any more, so no problem to remove. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 20 -------------------- 1 file changed, 20 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index 370b5a7..29c9b8a 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -24,7 +24,6 @@ enum { ALC883_TARGA_DIG, ALC883_TARGA_2ch_DIG, ALC883_TARGA_8ch_DIG, - ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, ALC888_ACER_ASPIRE_6530G, @@ -2089,7 +2088,6 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC883_TARGA_DIG] = "targa-dig", [ALC883_TARGA_2ch_DIG] = "targa-2ch-dig", [ALC883_TARGA_8ch_DIG] = "targa-8ch-dig", - [ALC883_ACER] = "acer", [ALC883_ACER_ASPIRE] = "acer-aspire", [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", [ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g", @@ -2127,10 +2125,6 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { ALC888_ACER_ASPIRE_6530G), SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC888_ACER_ASPIRE_7730G), - /* default Acer -- disabled as it causes more problems. - * model=auto should work fine now - */ - /* SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), */ SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavilion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), @@ -2524,20 +2518,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc882_targa_setup, .init_hook = alc882_targa_automute, }, - [ALC883_ACER] = { - .mixers = { alc883_base_mixer }, - /* On TravelMate laptops, GPIO 0 enables the internal speaker - * and the headphone jack. Turn this on and rely on the - * standard mute methods whenever the user wants to turn - * these outputs off. - */ - .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .input_mux = &alc883_capture_source, - }, [ALC883_ACER_ASPIRE] = { .mixers = { alc883_acer_aspire_mixer }, .init_verbs = { alc883_init_verbs, alc883_acer_eapd_verbs }, -- cgit v1.1 From 8812c4f96178620ebaabbd6e32401411b244d26b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2011 17:39:15 +0100 Subject: ALSA: hda/realtek - Move ALC882 model=acer-aspire to auto-parser The ALC882 model=acer-aspire requires the additional COEF setup. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 62 ------------------------------------------- sound/pci/hda/patch_realtek.c | 16 +++++++++++ 2 files changed, 16 insertions(+), 62 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index 29c9b8a..707dd28 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -24,7 +24,6 @@ enum { ALC883_TARGA_DIG, ALC883_TARGA_2ch_DIG, ALC883_TARGA_8ch_DIG, - ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, ALC888_ACER_ASPIRE_6530G, ALC888_ACER_ASPIRE_8930G, @@ -1881,18 +1880,6 @@ static const struct snd_kcontrol_new alc883_targa_8ch_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, 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, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - static const struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), @@ -1994,35 +1981,6 @@ static const struct hda_verb alc883_targa_verbs[] = { #define alc883_targa_init_hook alc882_targa_init_hook #define alc883_targa_unsol_event alc882_targa_unsol_event -/* toggle speaker-output according to the hp-jack state */ -static void alc883_acer_aspire_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[1] = 0x16; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static const struct hda_verb alc883_acer_eapd_verbs[] = { - /* HP Pin: output 0 (0x0c) */ - {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}, - /* Front Pin: output 0 (0x0c) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* eanable EAPD on medion laptop */ - {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, - {0x20, AC_VERB_SET_PROC_COEF, 0x3050}, - /* enable unsolicited event */ - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - { } -}; - 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 */ @@ -2088,7 +2046,6 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC883_TARGA_DIG] = "targa-dig", [ALC883_TARGA_2ch_DIG] = "targa-2ch-dig", [ALC883_TARGA_8ch_DIG] = "targa-8ch-dig", - [ALC883_ACER_ASPIRE] = "acer-aspire", [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", [ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g", [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", @@ -2103,12 +2060,6 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE), - SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE), - SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_ACER_ASPIRE), - SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE), - SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE), - SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G", ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", @@ -2518,19 +2469,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc882_targa_setup, .init_hook = alc882_targa_automute, }, - [ALC883_ACER_ASPIRE] = { - .mixers = { alc883_acer_aspire_mixer }, - .init_verbs = { alc883_init_verbs, alc883_acer_eapd_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .input_mux = &alc883_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc883_acer_aspire_setup, - .init_hook = alc_hp_automute, - }, [ALC888_ACER_ASPIRE_4930G] = { .mixers = { alc888_acer_aspire_4930g_mixer, alc883_chmode_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 422430d..32663c7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4237,6 +4237,7 @@ enum { ALC888_FIXUP_EEE1601, ALC882_FIXUP_EAPD, ALC883_FIXUP_EAPD, + ALC883_FIXUP_ACER_EAPD, }; static const struct alc_fixup alc882_fixups[] = { @@ -4308,9 +4309,24 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [ALC883_FIXUP_ACER_EAPD] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* eanable EAPD on Acer laptops */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x3050 }, + { } + } + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_FIXUP_ACER_EAPD), + SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_FIXUP_ACER_EAPD), + SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_FIXUP_ACER_EAPD), + SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_FIXUP_ACER_EAPD), + SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_FIXUP_ACER_EAPD), + SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_FIXUP_ACER_EAPD), SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), -- cgit v1.1 From ac9b1cddf10a299fb3a4dd411e518d07ad17f89f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2011 17:45:55 +0100 Subject: ALSA: hda/realtek - Reorder alc882_fixup_tbl[] No, I'm not Mr. Monk, but can't resist... Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 32663c7..765780c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4328,17 +4328,17 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_FIXUP_ACER_EAPD), SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_FIXUP_ACER_EAPD), SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), + SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), - SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), + SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD), /* codec SSID */ SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), - SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), - SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), {} }; -- cgit v1.1 From eb844d51cccca0ce9fad316da803f1bbe53d323b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2011 18:03:07 +0100 Subject: ALSA: hda/realtek - Remove ALC882 targa-* models All ALC882 targa-* models can be replaced with the auto-parser just with the additional GPIO3 setup. And it's generically applied to all MSI boards unless other quirks are present. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 235 ------------------------------------------ sound/pci/hda/patch_realtek.c | 6 ++ 2 files changed, 6 insertions(+), 235 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index 707dd28..185ad65 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -9,7 +9,6 @@ enum { ALC882_3ST_DIG, ALC882_6ST_DIG, ALC882_W2JC, - ALC882_TARGA, ALC885_MACPRO, ALC885_MBA21, ALC885_MBP3, @@ -21,9 +20,6 @@ enum { ALC883_3ST_6ch_DIG, ALC883_3ST_6ch, ALC883_6ST_DIG, - ALC883_TARGA_DIG, - ALC883_TARGA_2ch_DIG, - ALC883_TARGA_8ch_DIG, ALC888_ACER_ASPIRE_4930G, ALC888_ACER_ASPIRE_6530G, ALC888_ACER_ASPIRE_8930G, @@ -1019,23 +1015,6 @@ static const struct snd_kcontrol_new alc882_w2jc_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc882_targa_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 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_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), - { } /* end */ -}; - static const struct snd_kcontrol_new alc882_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1593,45 +1572,7 @@ static void alc885_imac91_setup(struct hda_codec *codec) alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } -static const struct hda_verb alc882_targa_verbs[] = { - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */ - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ - - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - { } /* end */ -}; - /* toggle speaker-output according to the hp-jack state */ -static void alc882_targa_automute(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - alc_hp_automute(codec); - snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA, - spec->hp_jack_present ? 1 : 3); -} - -static void alc882_targa_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x1b; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res) -{ - if ((res >> 26) == ALC_HP_EVENT) - alc882_targa_automute(codec); -} - static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted) { unsigned int gpiostate, gpiomask, gpiodir; @@ -1834,52 +1775,6 @@ static const struct snd_kcontrol_new alc883_fivestack_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc883_targa_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - 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_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc883_targa_2ch_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker 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("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("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc883_targa_8ch_mixer[] = { - HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - static const struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), @@ -1948,39 +1843,6 @@ static const struct snd_kcontrol_new alc883_chmode_mixer[] = { { } /* end */ }; -static const struct hda_verb alc883_targa_verbs[] = { - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - -/* Connect Line-Out side jack (SPDIF) to Side */ - {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}, -/* Connect Mic jack to CLFE */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, -/* Connect Line-in jack to Surround */ - {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}, -/* Connect HP out jack to Front */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - - { } /* end */ -}; - -/* toggle speaker-output according to the hp-jack state */ -#define alc883_targa_init_hook alc882_targa_init_hook -#define alc883_targa_unsol_event alc882_targa_unsol_event - 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 */ @@ -2031,7 +1893,6 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC882_3ST_DIG] = "3stack-dig", [ALC882_6ST_DIG] = "6stack-dig", [ALC882_W2JC] = "w2jc", - [ALC882_TARGA] = "targa", [ALC885_MACPRO] = "macpro", [ALC885_MB5] = "mb5", [ALC885_MACMINI3] = "macmini3", @@ -2043,9 +1904,6 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC883_3ST_6ch_DIG] = "3stack-6ch-dig", [ALC883_3ST_6ch] = "3stack-6ch", [ALC883_6ST_DIG] = "alc883-6stack-dig", - [ALC883_TARGA_DIG] = "targa-dig", - [ALC883_TARGA_2ch_DIG] = "targa-2ch-dig", - [ALC883_TARGA_8ch_DIG] = "targa-8ch-dig", [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", [ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g", [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", @@ -2091,37 +1949,14 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), - SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), - SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), - SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ - SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC882_AUTO), SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), - SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG), - SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG), - SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG), - SND_PCI_QUIRK(0x1462, 0x3fc1, "MSI", ALC883_TARGA_DIG), - SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG), - SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG), - SND_PCI_QUIRK(0x1462, 0x3fdf, "MSI", ALC883_TARGA_DIG), - SND_PCI_QUIRK(0x1462, 0x42cd, "MSI", ALC883_TARGA_DIG), - SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG), - SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG), - SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG), - SND_PCI_QUIRK(0x1462, 0x4570, "MSI Wind Top AE2220", ALC883_TARGA_DIG), - SND_PCI_QUIRK(0x1462, 0x6510, "MSI GX620", ALC883_TARGA_8ch_DIG), SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x7260, "MSI 7260", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x7437, "MSI NetOn AP1900", ALC883_TARGA_DIG), - SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), - SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x0571, "Clevo laptop M570U", ALC883_3ST_6ch_DIG), @@ -2302,24 +2137,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc885_imac91_setup, .init_hook = alc_hp_automute, }, - [ALC882_TARGA] = { - .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, - alc880_gpio3_init_verbs, alc882_targa_verbs}, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), - .adc_nids = alc882_adc_nids, - .capsrc_nids = alc882_capsrc_nids, - .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes), - .channel_mode = alc882_3ST_6ch_modes, - .need_dac_fix = 1, - .input_mux = &alc882_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc882_targa_setup, - .init_hook = alc882_targa_automute, - }, [ALC883_3ST_2ch_DIG] = { .mixers = { alc883_3ST_2ch_mixer }, .init_verbs = { alc883_init_verbs }, @@ -2417,58 +2234,6 @@ static const struct alc_config_preset alc882_presets[] = { .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, }, - [ALC883_TARGA_DIG] = { - .mixers = { alc883_targa_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, - alc883_targa_verbs}, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), - .channel_mode = alc883_3ST_6ch_modes, - .need_dac_fix = 1, - .input_mux = &alc883_capture_source, - .unsol_event = alc883_targa_unsol_event, - .setup = alc882_targa_setup, - .init_hook = alc882_targa_automute, - }, - [ALC883_TARGA_2ch_DIG] = { - .mixers = { alc883_targa_2ch_mixer}, - .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, - alc883_targa_verbs}, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .adc_nids = alc883_adc_nids_alt, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt), - .capsrc_nids = alc883_capsrc_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .input_mux = &alc883_capture_source, - .unsol_event = alc883_targa_unsol_event, - .setup = alc882_targa_setup, - .init_hook = alc882_targa_automute, - }, - [ALC883_TARGA_8ch_DIG] = { - .mixers = { alc883_targa_mixer, alc883_targa_8ch_mixer, - alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, - alc883_targa_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), - .adc_nids = alc883_adc_nids_rev, - .capsrc_nids = alc883_capsrc_nids_rev, - .dig_out_nid = ALC883_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc883_4ST_8ch_modes), - .channel_mode = alc883_4ST_8ch_modes, - .need_dac_fix = 1, - .input_mux = &alc883_capture_source, - .unsol_event = alc883_targa_unsol_event, - .setup = alc882_targa_setup, - .init_hook = alc882_targa_automute, - }, [ALC888_ACER_ASPIRE_4930G] = { .mixers = { alc888_acer_aspire_4930g_mixer, alc883_chmode_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 765780c..5aa8deb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4238,6 +4238,7 @@ enum { ALC882_FIXUP_EAPD, ALC883_FIXUP_EAPD, ALC883_FIXUP_ACER_EAPD, + ALC882_FIXUP_GPIO3, }; static const struct alc_fixup alc882_fixups[] = { @@ -4318,6 +4319,10 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [ALC882_FIXUP_GPIO3] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = alc_gpio3_init_verbs, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -4335,6 +4340,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD), /* codec SSID */ SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), + SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), -- cgit v1.1 From 68ef0561efe494143516df38c03a16b837b8e79c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2011 18:24:44 +0100 Subject: ALSA: hda/realtek - Drop ALC882 desktop model quirks Now we're touching the desktop static configs for ALC88x codecs. These are mostly OK with the auto-parser, but some models need careful handling; ALC889 intel mobo requires the COEF setup, and W2JC needs GPIO1 and COEF. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 780 ------------------------------------------ sound/pci/hda/patch_realtek.c | 30 +- 2 files changed, 22 insertions(+), 788 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index 185ad65..ccd20d1 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -6,9 +6,6 @@ /* ALC882 models */ enum { ALC882_AUTO, - ALC882_3ST_DIG, - ALC882_6ST_DIG, - ALC882_W2JC, ALC885_MACPRO, ALC885_MBA21, ALC885_MBP3, @@ -16,108 +13,15 @@ enum { ALC885_MACMINI3, ALC885_IMAC24, ALC885_IMAC91, - ALC883_3ST_2ch_DIG, - ALC883_3ST_6ch_DIG, - ALC883_3ST_6ch, - ALC883_6ST_DIG, ALC888_ACER_ASPIRE_4930G, ALC888_ACER_ASPIRE_6530G, ALC888_ACER_ASPIRE_8930G, ALC888_ACER_ASPIRE_7730G, - ALC883_3ST_6ch_INTEL, - ALC889A_INTEL, - ALC889_INTEL, ALC889A_MB31, ALC882_MODEL_LAST, }; /* - * 2ch mode - */ -static const struct hda_verb alc888_4ST_ch2_intel_init[] = { -/* Mic-in jack as mic in */ - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, -/* Line-in jack as Line in */ - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, -/* Line-Out as Front */ - { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00}, - { } /* end */ -}; - -/* - * 4ch mode - */ -static const struct hda_verb alc888_4ST_ch4_intel_init[] = { -/* Mic-in jack as mic in */ - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, -/* Line-in jack as Surround */ - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, -/* Line-Out as Front */ - { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00}, - { } /* end */ -}; - -/* - * 6ch mode - */ -static const struct hda_verb alc888_4ST_ch6_intel_init[] = { -/* Mic-in jack as CLFE */ - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, -/* Line-in jack as Surround */ - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, -/* Line-Out as CLFE (workaround because Mic-in is not loud enough) */ - { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, - { } /* end */ -}; - -/* - * 8ch mode - */ -static const struct hda_verb alc888_4ST_ch8_intel_init[] = { -/* Mic-in jack as CLFE */ - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, -/* Line-in jack as Surround */ - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, -/* Line-Out as Side */ - { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, - { } /* end */ -}; - -static const struct hda_channel_mode alc888_4ST_8ch_intel_modes[4] = { - { 2, alc888_4ST_ch2_intel_init }, - { 4, alc888_4ST_ch4_intel_init }, - { 6, alc888_4ST_ch6_intel_init }, - { 8, alc888_4ST_ch8_intel_init }, -}; - -static void alc889_automute_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[1] = 0x16; - spec->autocfg.speaker_pins[2] = 0x17; - spec->autocfg.speaker_pins[3] = 0x19; - spec->autocfg.speaker_pins[4] = 0x1a; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static void alc889_intel_init_hook(struct hda_codec *codec) -{ - alc889_coef_init(codec); - alc_hp_automute(codec); -} - -/* * ALC888 Acer Aspire 4930G model */ @@ -586,79 +490,6 @@ static const struct hda_channel_mode alc882_3ST_6ch_modes[3] = { #define alc883_3ST_6ch_modes alc882_3ST_6ch_modes -/* - * 2ch mode - */ -static const struct hda_verb alc883_3ST_ch2_clevo_init[] = { - { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, 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 }, - { } /* end */ -}; - -/* - * 4ch mode - */ -static const struct hda_verb alc883_3ST_ch4_clevo_init[] = { - { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, 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 }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -/* - * 6ch mode - */ -static const struct hda_verb alc883_3ST_ch6_clevo_init[] = { - { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 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 }, - { } /* end */ -}; - -static const struct hda_channel_mode alc883_3ST_6ch_clevo_modes[3] = { - { 2, alc883_3ST_ch2_clevo_init }, - { 4, alc883_3ST_ch4_clevo_init }, - { 6, alc883_3ST_ch6_clevo_init }, -}; - - -/* - * 6ch mode - */ -static const struct hda_verb alc882_sixstack_ch6_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, - { 0x16, 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 }, - { } /* end */ -}; - -/* - * 8ch mode - */ -static const struct hda_verb alc882_sixstack_ch8_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x16, 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 }, - { } /* end */ -}; - -static const struct hda_channel_mode alc882_sixstack_modes[2] = { - { 6, alc882_sixstack_ch6_init }, - { 8, alc882_sixstack_ch8_init }, -}; - /* Macbook Air 2,1 */ @@ -728,216 +559,6 @@ static const struct hda_channel_mode alc885_mb5_6ch_modes[2] = { #define alc885_macmini3_6ch_modes alc885_mb5_6ch_modes -/* - * 2ch mode - */ -static const struct hda_verb alc883_4ST_ch2_init[] = { - { 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 }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; - -/* - * 4ch mode - */ -static const struct hda_verb alc883_4ST_ch4_init[] = { - { 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 }, - { 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 }, - { } /* end */ -}; - -/* - * 6ch mode - */ -static const struct hda_verb alc883_4ST_ch6_init[] = { - { 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_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 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 }, - { } /* end */ -}; - -/* - * 8ch mode - */ -static const struct hda_verb alc883_4ST_ch8_init[] = { - { 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 }, - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 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 }, - { } /* end */ -}; - -static const struct hda_channel_mode alc883_4ST_8ch_modes[4] = { - { 2, alc883_4ST_ch2_init }, - { 4, alc883_4ST_ch4_init }, - { 6, alc883_4ST_ch6_init }, - { 8, alc883_4ST_ch8_init }, -}; - - -/* - * 2ch mode - */ -static const struct hda_verb alc883_3ST_ch2_intel_init[] = { - { 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 }, - { } /* end */ -}; - -/* - * 4ch mode - */ -static const struct hda_verb alc883_3ST_ch4_intel_init[] = { - { 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_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -/* - * 6ch mode - */ -static const struct hda_verb alc883_3ST_ch6_intel_init[] = { - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 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 }, - { } /* end */ -}; - -static const struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = { - { 2, alc883_3ST_ch2_intel_init }, - { 4, alc883_3ST_ch4_intel_init }, - { 6, alc883_3ST_ch6_intel_init }, -}; - -/* - * 2ch mode - */ -static const struct hda_verb alc889_ch2_intel_init[] = { - { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, - { 0x19, AC_VERB_SET_CONNECT_SEL, 0x00 }, - { 0x16, AC_VERB_SET_CONNECT_SEL, 0x00 }, - { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00 }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; - -/* - * 6ch mode - */ -static const struct hda_verb alc889_ch6_intel_init[] = { - { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, - { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, - { 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 alc889_ch8_intel_init[] = { - { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, - { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 }, - { 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 alc889_8ch_intel_modes[3] = { - { 2, alc889_ch2_intel_init }, - { 6, alc889_ch6_intel_init }, - { 8, alc889_ch8_intel_init }, -}; - -/* - * 6ch mode - */ -static const struct hda_verb alc883_sixstack_ch6_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, - { 0x16, 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 }, - { } /* end */ -}; - -/* - * 8ch mode - */ -static const struct hda_verb alc883_sixstack_ch8_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x16, 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 }, - { } /* end */ -}; - -static const struct hda_channel_mode alc883_sixstack_modes[2] = { - { 6, alc883_sixstack_ch6_init }, - { 8, alc883_sixstack_ch8_init }, -}; - - -/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 - * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b - */ -static const struct snd_kcontrol_new alc882_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_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 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 */ -}; - /* Macbook Air 2,1 same control for HP and internal Speaker */ static const struct snd_kcontrol_new alc885_mba21_mixer[] = { @@ -1002,19 +623,6 @@ static const struct snd_kcontrol_new alc885_imac91_mixer[] = { }; -static const struct snd_kcontrol_new alc882_w2jc_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 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_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - static const struct snd_kcontrol_new alc882_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1100,100 +708,12 @@ static const struct hda_verb alc882_adc1_init_verbs[] = { { } }; -static const struct hda_verb alc882_eapd_verbs[] = { - /* change to EAPD mode */ - {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, - {0x20, AC_VERB_SET_PROC_COEF, 0x3060}, - { } -}; - static const struct hda_verb alc889_eapd_verbs[] = { {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } }; -static const struct hda_verb alc_hp15_unsol_verbs[] = { - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {} -}; - -static const struct hda_verb alc885_init_verbs[] = { - /* Front mixer: unmute input/output amp left and right (volume = 0) */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Rear mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* CLFE mixer */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Side mixer */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - /* Front HP Pin: output 0 (0x0c) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* 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) */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x19, 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% */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Front Mic pin: input vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, 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}, - - /* Mixer elements: 0x18, , 0x1a, 0x1b */ - /* Input mixer1 */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* 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)}, - /* ADC3: mute amp left and right */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - - { } -}; - -static const struct hda_verb alc885_init_input_verbs[] = { - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, - { } -}; - - -/* Unmute Selector 24h and set the default input to front mic */ -static const struct hda_verb alc889_init_input_verbs[] = { - {0x24, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - { } -}; - - #define alc883_init_verbs alc882_base_init_verbs /* Mac Pro test */ @@ -1662,25 +1182,6 @@ static const struct hda_channel_mode alc889A_mb31_6ch_modes[4] = { { 6, alc889A_mb31_ch6_init }, }; -#define alc883_base_mixer alc882_base_mixer - -static const struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 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 alc883_3ST_6ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -1704,77 +1205,6 @@ static const struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc883_3ST_6ch_intel_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", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc885_8ch_intel_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("Speaker Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, 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, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x1b, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc883_fivestack_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 alc888_acer_aspire_6530_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), @@ -1814,24 +1244,6 @@ static const struct snd_kcontrol_new alc889A_mb31_mixer[] = { { } /* end */ }; -static const struct hda_bind_ctls alc883_bind_cap_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), - 0 - }, -}; - -static const struct hda_bind_ctls alc883_bind_cap_switch = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), - 0 - }, -}; - static const struct snd_kcontrol_new alc883_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1878,21 +1290,10 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) alc889A_mb31_automute(codec); } -static const hda_nid_t alc883_slave_dig_outs[] = { - ALC1200_DIGOUT_NID, 0, -}; - -static const hda_nid_t alc1200_slave_dig_outs[] = { - ALC883_DIGOUT_NID, 0, -}; - /* * configuration and preset */ static const char * const alc882_models[ALC882_MODEL_LAST] = { - [ALC882_3ST_DIG] = "3stack-dig", - [ALC882_6ST_DIG] = "6stack-dig", - [ALC882_W2JC] = "w2jc", [ALC885_MACPRO] = "macpro", [ALC885_MB5] = "mb5", [ALC885_MACMINI3] = "macmini3", @@ -1900,24 +1301,15 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC885_MBP3] = "mbp3", [ALC885_IMAC24] = "imac24", [ALC885_IMAC91] = "imac91", - [ALC883_3ST_2ch_DIG] = "3stack-2ch-dig", - [ALC883_3ST_6ch_DIG] = "3stack-6ch-dig", - [ALC883_3ST_6ch] = "3stack-6ch", - [ALC883_6ST_DIG] = "alc883-6stack-dig", [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", [ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g", [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", - [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", - [ALC889A_INTEL] = "intel-alc889a", - [ALC889_INTEL] = "intel-x58", [ALC889A_MB31] = "mb31", [ALC882_AUTO] = "auto", }; static const struct snd_pci_quirk alc882_cfg_tbl[] = { - SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G", ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", @@ -1926,50 +1318,12 @@ static const struct snd_pci_quirk alc882_cfg_tbl[] = { ALC888_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G", ALC888_ACER_ASPIRE_8930G), - SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC882_AUTO), - SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC882_AUTO), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_6530G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", ALC888_ACER_ASPIRE_6530G), SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC888_ACER_ASPIRE_7730G), - - SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavilion", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), - - SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC), - SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG), - - SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), - SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), - - SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG), - - SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x1558, 0x0571, "Clevo laptop M570U", ALC883_3ST_6ch_DIG), - SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), - SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG), - - SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), - SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), - SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC889_INTEL), - SND_PCI_QUIRK(0x8086, 0x0021, "Intel IbexPeak", ALC889A_INTEL), - SND_PCI_QUIRK(0x8086, 0x3b56, "Intel IbexPeak", ALC889A_INTEL), - SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC882_6ST_DIG), - {} }; @@ -2001,43 +1355,6 @@ static const struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { }; static const struct alc_config_preset alc882_presets[] = { - [ALC882_3ST_DIG] = { - .mixers = { alc882_base_mixer }, - .init_verbs = { alc882_base_init_verbs, - alc882_adc1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), - .channel_mode = alc882_ch_modes, - .need_dac_fix = 1, - .input_mux = &alc882_capture_source, - }, - [ALC882_6ST_DIG] = { - .mixers = { alc882_base_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_base_init_verbs, - alc882_adc1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), - .channel_mode = alc882_sixstack_modes, - .input_mux = &alc882_capture_source, - }, - [ALC882_W2JC] = { - .mixers = { alc882_w2jc_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, - alc882_eapd_verbs, alc880_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), - .channel_mode = alc880_threestack_modes, - .need_dac_fix = 1, - .input_mux = &alc882_capture_source, - .dig_out_nid = ALC882_DIGOUT_NID, - }, [ALC885_MBA21] = { .mixers = { alc885_mba21_mixer }, .init_verbs = { alc885_mba21_init_verbs, alc880_gpio1_init_verbs }, @@ -2137,103 +1454,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc885_imac91_setup, .init_hook = alc_hp_automute, }, - [ALC883_3ST_2ch_DIG] = { - .mixers = { alc883_3ST_2ch_mixer }, - .init_verbs = { alc883_init_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .input_mux = &alc883_capture_source, - }, - [ALC883_3ST_6ch_DIG] = { - .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), - .channel_mode = alc883_3ST_6ch_modes, - .need_dac_fix = 1, - .input_mux = &alc883_capture_source, - }, - [ALC883_3ST_6ch] = { - .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), - .channel_mode = alc883_3ST_6ch_modes, - .need_dac_fix = 1, - .input_mux = &alc883_capture_source, - }, - [ALC883_3ST_6ch_INTEL] = { - .mixers = { alc883_3ST_6ch_intel_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .slave_dig_outs = alc883_slave_dig_outs, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes), - .channel_mode = alc883_3ST_6ch_intel_modes, - .need_dac_fix = 1, - .input_mux = &alc883_3stack_6ch_intel, - }, - [ALC889A_INTEL] = { - .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer }, - .init_verbs = { alc885_init_verbs, alc885_init_input_verbs, - alc_hp15_unsol_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), - .adc_nids = alc889_adc_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .slave_dig_outs = alc883_slave_dig_outs, - .num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes), - .channel_mode = alc889_8ch_intel_modes, - .capsrc_nids = alc889_capsrc_nids, - .input_mux = &alc889_capture_source, - .setup = alc889_automute_setup, - .init_hook = alc_hp_automute, - .unsol_event = alc_sku_unsol_event, - .need_dac_fix = 1, - }, - [ALC889_INTEL] = { - .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer }, - .init_verbs = { alc885_init_verbs, alc889_init_input_verbs, - alc889_eapd_verbs, alc_hp15_unsol_verbs}, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), - .adc_nids = alc889_adc_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .slave_dig_outs = alc883_slave_dig_outs, - .num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes), - .channel_mode = alc889_8ch_intel_modes, - .capsrc_nids = alc889_capsrc_nids, - .input_mux = &alc889_capture_source, - .setup = alc889_automute_setup, - .init_hook = alc889_intel_init_hook, - .unsol_event = alc_sku_unsol_event, - .need_dac_fix = 1, - }, - [ALC883_6ST_DIG] = { - .mixers = { alc883_base_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), - .channel_mode = alc883_sixstack_modes, - .input_mux = &alc883_capture_source, - }, [ALC888_ACER_ASPIRE_4930G] = { .mixers = { alc888_acer_aspire_4930g_mixer, alc883_chmode_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5aa8deb..c1fa4c3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4239,8 +4239,18 @@ enum { ALC883_FIXUP_EAPD, ALC883_FIXUP_ACER_EAPD, ALC882_FIXUP_GPIO3, + ALC889_FIXUP_COEF, + ALC882_FIXUP_ASUS_W2JC, }; +static void alc889_fixup_coef(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action != ALC_FIXUP_ACT_INIT) + return; + alc889_coef_init(codec); +} + static const struct alc_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = ALC_FIXUP_PINS, @@ -4323,6 +4333,16 @@ static const struct alc_fixup alc882_fixups[] = { .type = ALC_FIXUP_VERBS, .v.verbs = alc_gpio3_init_verbs, }, + [ALC882_FIXUP_ASUS_W2JC] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = alc_gpio1_init_verbs, + .chained = true, + .chain_id = ALC882_FIXUP_EAPD, + }, + [ALC889_FIXUP_COEF] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc889_fixup_coef, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -4336,6 +4356,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), + SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD), /* codec SSID */ @@ -4345,6 +4366,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), + SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC889_FIXUP_COEF), {} }; @@ -4417,14 +4439,6 @@ static int patch_alc882(struct hda_codec *codec) err = alc882_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 = ALC882_3ST_DIG; - } -#endif } if (board_config != ALC_MODEL_AUTO) -- cgit v1.1 From 07b18f69a766375736a5313c29d808e59b1e13e9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Nov 2011 15:42:54 +0100 Subject: ALSA: hda/realtek - Create multi-io jacks more aggresively So far the driver creates the multi-io jacks only when a single output jack, i.e. no multiple speakers are assigned. This patch adds the similar multi-io detection even with multiple speakers are assigned primarily, so that 5.1-speakers + HP/mic/LI combination can work. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 63 ++++++++++++++++++++++++++++++++++--------- 1 file changed, 51 insertions(+), 12 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c1fa4c3..8194c07 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2974,6 +2974,23 @@ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin) return 0; } +/* check whether the DAC is reachable from the pin */ +static bool alc_auto_is_dac_reachable(struct hda_codec *codec, + hda_nid_t pin, hda_nid_t dac) +{ + hda_nid_t srcs[5]; + int i, num; + + pin = alc_go_down_to_selector(codec, pin); + num = snd_hda_get_connections(codec, pin, srcs, ARRAY_SIZE(srcs)); + for (i = 0; i < num; i++) { + hda_nid_t nid = alc_auto_mix_to_dac(codec, srcs[i]); + if (nid == dac) + return true; + } + return false; +} + static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin) { hda_nid_t sel = alc_go_down_to_selector(codec, pin); @@ -3003,13 +3020,15 @@ static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs, } static int alc_auto_fill_multi_ios(struct hda_codec *codec, - unsigned int location); + unsigned int location, int offset); /* fill in the dac_nids table from the parsed pin configuration */ static int alc_auto_fill_dac_nids(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; const struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int location, defcfg; + int num_pins; bool redone = false; int i; @@ -3061,13 +3080,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { /* try to fill multi-io first */ - unsigned int location, defcfg; - int num_pins; - defcfg = snd_hda_codec_get_pincfg(codec, cfg->line_out_pins[0]); location = get_defcfg_location(defcfg); - num_pins = alc_auto_fill_multi_ios(codec, location); + num_pins = alc_auto_fill_multi_ios(codec, location, 0); if (num_pins > 0) { spec->multi_ios = num_pins; spec->ext_channel_count = 2; @@ -3082,6 +3098,21 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) alc_auto_fill_extra_dacs(codec, cfg->speaker_outs, cfg->speaker_pins, spec->multiout.extra_out_nid); + if (!spec->multi_ios && + cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && + cfg->hp_outs) { + /* try multi-ios with HP + inputs */ + defcfg = snd_hda_codec_get_pincfg(codec, cfg->hp_pins[0]); + location = get_defcfg_location(defcfg); + + num_pins = alc_auto_fill_multi_ios(codec, location, 1); + if (num_pins > 0) { + spec->multi_ios = num_pins; + spec->ext_channel_count = 2; + spec->multiout.num_dacs = num_pins + 1; + } + } + return 0; } @@ -3467,17 +3498,19 @@ static void alc_auto_init_extra_out(struct hda_codec *codec) * multi-io helper */ static int alc_auto_fill_multi_ios(struct hda_codec *codec, - unsigned int location) + unsigned int location, + int offset) { struct alc_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t prime_dac = spec->private_dac_nids[0]; - int type, i, num_pins = 0; + int type, i, dacs, num_pins = 0; + dacs = spec->multiout.num_dacs; for (type = AUTO_PIN_LINE_IN; type >= AUTO_PIN_MIC; type--) { for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; - hda_nid_t dac; + hda_nid_t dac = 0; unsigned int defcfg, caps; if (cfg->inputs[i].type != type) continue; @@ -3489,7 +3522,13 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec, caps = snd_hda_query_pin_caps(codec, nid); if (!(caps & AC_PINCAP_OUT)) continue; - dac = alc_auto_look_for_dac(codec, nid); + if (offset && offset + num_pins < dacs) { + dac = spec->private_dac_nids[offset + num_pins]; + if (!alc_auto_is_dac_reachable(codec, nid, dac)) + dac = 0; + } + if (!dac) + dac = alc_auto_look_for_dac(codec, nid); if (!dac) continue; spec->multi_io[num_pins].pin = nid; @@ -3498,11 +3537,11 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec, spec->private_dac_nids[spec->multiout.num_dacs++] = dac; } } - spec->multiout.num_dacs = 1; + spec->multiout.num_dacs = dacs; if (num_pins < 2) { /* clear up again */ - memset(spec->private_dac_nids, 0, - sizeof(spec->private_dac_nids)); + memset(spec->private_dac_nids + dacs, 0, + sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - dacs)); spec->private_dac_nids[0] = prime_dac; return 0; } -- cgit v1.1 From c3e837bbcc03e39991da5dd1f791c06cde9f7a45 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Nov 2011 16:01:47 +0100 Subject: ALSA: hda/realtek - Rewrite ALC882 acer-aspire-* models with the auto-parser Now we can move the big acer-aspire-* static quirks to the auto-paresr with some additional pin-configs and verbs. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 425 ------------------------------------------ sound/pci/hda/patch_realtek.c | 74 +++++++- 2 files changed, 69 insertions(+), 430 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index ccd20d1..c1f6ac6 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -21,322 +21,6 @@ enum { ALC882_MODEL_LAST, }; -/* - * ALC888 Acer Aspire 4930G model - */ - -static const struct hda_verb alc888_acer_aspire_4930g_verbs[] = { -/* Front Mic: set to PIN_IN (empty by default) */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, -/* Unselect Front Mic by default in input mixer 3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, -/* Enable unsolicited event for HP jack */ - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, -/* Connect Internal HP to front */ - {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}, -/* Connect HP out to front */ - {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, 0x00}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, - { } -}; - -/* - * ALC888 Acer Aspire 6530G model - */ - -static const struct hda_verb alc888_acer_aspire_6530g_verbs[] = { -/* Route to built-in subwoofer as well as speakers */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, 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)}, -/* Bias voltage on for external mic port */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80}, -/* Front Mic: set to PIN_IN (empty by default) */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, -/* Unselect Front Mic by default in input mixer 3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, -/* Enable unsolicited event for HP jack */ - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, -/* Enable speaker output */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, -/* Enable headphone output */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, - { } -}; - -/* - *ALC888 Acer Aspire 7730G model - */ - -static const struct hda_verb alc888_acer_aspire_7730G_verbs[] = { -/* Bias voltage on for external mic port */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80}, -/* Front Mic: set to PIN_IN (empty by default) */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, -/* Unselect Front Mic by default in input mixer 3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, -/* Enable unsolicited event for HP jack */ - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, -/* Enable speaker output */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, -/* Enable headphone output */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, -/*Enable internal subwoofer */ - {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, 0x02}, - {0x17, AC_VERB_SET_EAPD_BTLENABLE, 2}, - { } -}; - -/* - * ALC889 Acer Aspire 8930G model - */ - -static const struct hda_verb alc889_acer_aspire_8930g_verbs[] = { -/* Front Mic: set to PIN_IN (empty by default) */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, -/* Unselect Front Mic by default in input mixer 3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, -/* Enable unsolicited event for HP jack */ - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, -/* Connect Internal Front to Front */ - {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}, -/* Connect Internal Rear to Rear */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, -/* Connect Internal CLFE to CLFE */ - {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}, -/* Connect HP out to Front */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, -/* Enable all DACs */ -/* DAC DISABLE/MUTE 1? */ -/* setting bits 1-5 disables DAC nids 0x02-0x06 apparently. Init=0x38 */ - {0x20, AC_VERB_SET_COEF_INDEX, 0x03}, - {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, -/* DAC DISABLE/MUTE 2? */ -/* some bit here disables the other DACs. Init=0x4900 */ - {0x20, AC_VERB_SET_COEF_INDEX, 0x08}, - {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, -/* DMIC fix - * This laptop has a stereo digital microphone. The mics are only 1cm apart - * which makes the stereo useless. However, either the mic or the ALC889 - * makes the signal become a difference/sum signal instead of standard - * stereo, which is annoying. So instead we flip this bit which makes the - * codec replicate the sum signal to both channels, turning it into a - * normal mono mic. - */ -/* DMIC_CONTROL? Init value = 0x0001 */ - {0x20, AC_VERB_SET_COEF_INDEX, 0x0b}, - {0x20, AC_VERB_SET_PROC_COEF, 0x0003}, - { } -}; - -static const struct hda_input_mux alc888_2_capture_sources[2] = { - /* Front mic only available on one ADC */ - { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - { "Front Mic", 0xb }, - }, - }, - { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, - } -}; - -static const struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { - /* Interal mic only available on one ADC */ - { - .num_items = 5, - .items = { - { "Mic", 0x0 }, - { "Line In", 0x2 }, - { "CD", 0x4 }, - { "Input Mix", 0xa }, - { "Internal Mic", 0xb }, - }, - }, - { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Line In", 0x2 }, - { "CD", 0x4 }, - { "Input Mix", 0xa }, - }, - } -}; - -static const struct hda_input_mux alc889_capture_sources[3] = { - /* Digital mic only available on first "ADC" */ - { - .num_items = 5, - .items = { - { "Mic", 0x0 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - { "Front Mic", 0xb }, - { "Input Mix", 0xa }, - }, - }, - { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - { "Input Mix", 0xa }, - }, - }, - { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - { "Input Mix", 0xa }, - }, - } -}; - -static const struct snd_kcontrol_new alc888_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_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_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc888_acer_aspire_4930g_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_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Internal LFE Playback Volume", 0x0f, 1, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Internal LFE Playback Switch", 0x0f, 1, 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_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc889_acer_aspire_8930g_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("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), - { } /* end */ -}; - - -static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[1] = 0x16; - spec->autocfg.speaker_pins[2] = 0x17; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[1] = 0x16; - spec->autocfg.speaker_pins[2] = 0x17; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[1] = 0x16; - spec->autocfg.speaker_pins[2] = 0x17; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->autocfg.speaker_pins[1] = 0x16; - spec->autocfg.speaker_pins[2] = 0x1b; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - #define ALC882_DIGOUT_NID 0x06 #define ALC882_DIGIN_NID 0x0a #define ALC883_DIGOUT_NID ALC882_DIGOUT_NID @@ -1301,32 +985,10 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC885_MBP3] = "mbp3", [ALC885_IMAC24] = "imac24", [ALC885_IMAC91] = "imac91", - [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", - [ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g", - [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", - [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", [ALC889A_MB31] = "mb31", [ALC882_AUTO] = "auto", }; -static const struct snd_pci_quirk alc882_cfg_tbl[] = { - SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G", - ALC888_ACER_ASPIRE_4930G), - SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", - ALC888_ACER_ASPIRE_4930G), - SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G", - ALC888_ACER_ASPIRE_8930G), - SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G", - ALC888_ACER_ASPIRE_8930G), - SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", - ALC888_ACER_ASPIRE_6530G), - SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", - ALC888_ACER_ASPIRE_6530G), - SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", - ALC888_ACER_ASPIRE_7730G), - {} -}; - /* 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), @@ -1454,93 +1116,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc885_imac91_setup, .init_hook = alc_hp_automute, }, - [ALC888_ACER_ASPIRE_4930G] = { - .mixers = { alc888_acer_aspire_4930g_mixer, - alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, - alc888_acer_aspire_4930g_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), - .adc_nids = alc883_adc_nids_rev, - .capsrc_nids = alc883_capsrc_nids_rev, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), - .channel_mode = alc883_3ST_6ch_modes, - .need_dac_fix = 1, - .const_channel_count = 6, - .num_mux_defs = - ARRAY_SIZE(alc888_2_capture_sources), - .input_mux = alc888_2_capture_sources, - .unsol_event = alc_sku_unsol_event, - .setup = alc888_acer_aspire_4930g_setup, - .init_hook = alc_hp_automute, - }, - [ALC888_ACER_ASPIRE_6530G] = { - .mixers = { alc888_acer_aspire_6530_mixer }, - .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, - alc888_acer_aspire_6530g_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), - .adc_nids = alc883_adc_nids_rev, - .capsrc_nids = alc883_capsrc_nids_rev, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .num_mux_defs = - ARRAY_SIZE(alc888_2_capture_sources), - .input_mux = alc888_acer_aspire_6530_sources, - .unsol_event = alc_sku_unsol_event, - .setup = alc888_acer_aspire_6530g_setup, - .init_hook = alc_hp_automute, - }, - [ALC888_ACER_ASPIRE_8930G] = { - .mixers = { alc889_acer_aspire_8930g_mixer, - alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, - alc889_acer_aspire_8930g_verbs, - alc889_eapd_verbs}, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), - .adc_nids = alc889_adc_nids, - .capsrc_nids = alc889_capsrc_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), - .channel_mode = alc883_3ST_6ch_modes, - .need_dac_fix = 1, - .const_channel_count = 6, - .num_mux_defs = - ARRAY_SIZE(alc889_capture_sources), - .input_mux = alc889_capture_sources, - .unsol_event = alc_sku_unsol_event, - .setup = alc889_acer_aspire_8930g_setup, - .init_hook = alc_hp_automute, -#ifdef CONFIG_SND_HDA_POWER_SAVE - .power_hook = alc_power_eapd, -#endif - }, - [ALC888_ACER_ASPIRE_7730G] = { - .mixers = { alc883_3ST_6ch_mixer, - alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, - alc888_acer_aspire_7730G_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), - .adc_nids = alc883_adc_nids_rev, - .capsrc_nids = alc883_capsrc_nids_rev, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), - .channel_mode = alc883_3ST_6ch_modes, - .need_dac_fix = 1, - .const_channel_count = 6, - .input_mux = &alc883_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc888_acer_aspire_7730g_setup, - .init_hook = alc_hp_automute, - }, [ALC889A_MB31] = { .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer}, .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8194c07..9fc2ba0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4280,6 +4280,9 @@ enum { ALC882_FIXUP_GPIO3, ALC889_FIXUP_COEF, ALC882_FIXUP_ASUS_W2JC, + ALC882_FIXUP_ACER_ASPIRE_4930G, + ALC882_FIXUP_ACER_ASPIRE_8930G, + ALC882_FIXUP_ASPIRE_8930G_VERBS, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -4382,6 +4385,57 @@ static const struct alc_fixup alc882_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc889_fixup_coef, }, + [ALC882_FIXUP_ACER_ASPIRE_4930G] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x16, 0x99130111 }, /* CLFE speaker */ + { 0x17, 0x99130112 }, /* surround speaker */ + { } + } + }, + [ALC882_FIXUP_ACER_ASPIRE_8930G] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x16, 0x99130111 }, /* CLFE speaker */ + { 0x1b, 0x99130112 }, /* surround speaker */ + { } + }, + .chained = true, + .chain_id = ALC882_FIXUP_ASPIRE_8930G_VERBS, + }, + [ALC882_FIXUP_ASPIRE_8930G_VERBS] = { + /* additional init verbs for Acer Aspire 8930G */ + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Enable all DACs */ + /* DAC DISABLE/MUTE 1? */ + /* setting bits 1-5 disables DAC nids 0x02-0x06 + * apparently. Init=0x38 */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x03 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, + /* DAC DISABLE/MUTE 2? */ + /* some bit here disables the other DACs. + * Init=0x4900 */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x08 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, + /* DMIC fix + * This laptop has a stereo digital microphone. + * The mics are only 1cm apart which makes the stereo + * useless. However, either the mic or the ALC889 + * makes the signal become a difference/sum signal + * instead of standard stereo, which is annoying. + * So instead we flip this bit which makes the + * codec replicate the sum signal to both channels, + * turning it into a normal mono mic. + */ + /* DMIC_CONTROL? Init value = 0x0001 */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x0b }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x3050 }, + { } + } + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -4391,6 +4445,20 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_FIXUP_ACER_EAPD), SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_FIXUP_ACER_EAPD), SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_FIXUP_ACER_EAPD), + SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G", + ALC882_FIXUP_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", + ALC882_FIXUP_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G", + ALC882_FIXUP_ACER_ASPIRE_8930G), + SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G", + ALC882_FIXUP_ACER_ASPIRE_8930G), + SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", + ALC882_FIXUP_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", + ALC882_FIXUP_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", + ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD), @@ -4453,11 +4521,7 @@ static int patch_alc882(struct hda_codec *codec) if (err < 0) goto error; - board_config = alc_board_config(codec, ALC882_MODEL_LAST, - alc882_models, alc882_cfg_tbl); - - if (board_config < 0) - board_config = alc_board_codec_sid_config(codec, + board_config = alc_board_codec_sid_config(codec, ALC882_MODEL_LAST, alc882_models, alc882_ssid_cfg_tbl); if (board_config < 0) { -- cgit v1.1 From e29d377814b83af816fb8c1857605b9c4196477b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 14 Nov 2011 17:13:23 +0100 Subject: ALSA: hda/realtek - Create mono volume controls for mono-outputs When the pin or the DAC doesn't support the stereo, create a mono control instead of creating a stereo control blindly. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9fc2ba0..6f0f2f6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3144,8 +3144,15 @@ static int alc_auto_add_vol_ctl(struct hda_codec *codec, val); } -#define alc_auto_add_stereo_vol(codec, pfx, cidx, nid) \ - alc_auto_add_vol_ctl(codec, pfx, cidx, nid, 3) +static int alc_auto_add_stereo_vol(struct hda_codec *codec, + const char *pfx, int cidx, + hda_nid_t nid) +{ + int chs = 1; + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + chs = 3; + return alc_auto_add_vol_ctl(codec, pfx, cidx, nid, chs); +} /* create a mute-switch for the given mixer widget; * if it has multiple sources (e.g. DAC and loopback), create a bind-mute @@ -3177,8 +3184,14 @@ static int alc_auto_add_sw_ctl(struct hda_codec *codec, return __add_pb_sw_ctrl(codec->spec, type, pfx, cidx, val); } -#define alc_auto_add_stereo_sw(codec, pfx, cidx, nid) \ - alc_auto_add_sw_ctl(codec, pfx, cidx, nid, 3) +static int alc_auto_add_stereo_sw(struct hda_codec *codec, const char *pfx, + int cidx, hda_nid_t nid) +{ + int chs = 1; + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + chs = 3; + return alc_auto_add_sw_ctl(codec, pfx, cidx, nid, chs); +} static hda_nid_t alc_look_for_out_mute_nid(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) -- cgit v1.1 From 3ccbf1c3763510d5c443133f9aa40d8bf4ae8db9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 14 Nov 2011 17:20:49 +0100 Subject: ALSA: hda/realtek - Remove left-over chunks in alc882_quirks.c Remove unused variables. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 102 ------------------------------------------ 1 file changed, 102 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index c1f6ac6..403e30a 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -13,10 +13,6 @@ enum { ALC885_MACMINI3, ALC885_IMAC24, ALC885_IMAC91, - ALC888_ACER_ASPIRE_4930G, - ALC888_ACER_ASPIRE_6530G, - ALC888_ACER_ASPIRE_8930G, - ALC888_ACER_ASPIRE_7730G, ALC889A_MB31, ALC882_MODEL_LAST, }; @@ -43,15 +39,9 @@ static const hda_nid_t alc882_dac_nids[4] = { #define alc882_adc_nids alc880_adc_nids #define alc882_adc_nids_alt alc880_adc_nids_alt #define alc883_adc_nids alc882_adc_nids_alt -static const hda_nid_t alc883_adc_nids_alt[1] = { 0x08 }; -static const hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 }; -#define alc889_adc_nids alc880_adc_nids -static const hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 }; static const hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 }; #define alc883_capsrc_nids alc882_capsrc_nids_alt -static const hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 }; -#define alc889_capsrc_nids alc882_capsrc_nids /* input MUX */ /* FIXME: should be a matrix-type input source selection */ @@ -68,15 +58,6 @@ static const struct hda_input_mux alc882_capture_source = { #define alc883_capture_source alc882_capture_source -static const struct hda_input_mux alc889_capture_source = { - .num_items = 3, - .items = { - { "Front Mic", 0x0 }, - { "Mic", 0x3 }, - { "Line", 0x2 }, - }, -}; - static const struct hda_input_mux mb5_capture_source = { .num_items = 3, .items = { @@ -123,58 +104,6 @@ static const struct hda_input_mux alc889A_imac91_capture_source = { }, }; -/* - * 2ch mode - */ -static const struct hda_channel_mode alc883_3ST_2ch_modes[1] = { - { 2, NULL } -}; - -/* - * 2ch mode - */ -static const struct hda_verb alc882_3ST_ch2_init[] = { - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, 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 }, - { } /* end */ -}; - -/* - * 4ch mode - */ -static const struct hda_verb alc882_3ST_ch4_init[] = { - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, 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 }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -/* - * 6ch mode - */ -static const struct hda_verb alc882_3ST_ch6_init[] = { - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 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 }, - { } /* end */ -}; - -static const struct hda_channel_mode alc882_3ST_6ch_modes[3] = { - { 2, alc882_3ST_ch2_init }, - { 4, alc882_3ST_ch4_init }, - { 6, alc882_3ST_ch6_init }, -}; - -#define alc883_3ST_6ch_modes alc882_3ST_6ch_modes - - /* Macbook Air 2,1 */ static const struct hda_channel_mode alc885_mba21_ch_modes[1] = { @@ -380,24 +309,6 @@ static const struct hda_verb alc882_base_init_verbs[] = { { } }; -static const struct hda_verb alc882_adc1_init_verbs[] = { - /* 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)}, - /* ADC1: mute amp left and right */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - { } -}; - -static const struct hda_verb alc889_eapd_verbs[] = { - {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, - { } -}; - #define alc883_init_verbs alc882_base_init_verbs /* Mac Pro test */ @@ -889,19 +800,6 @@ static const struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, 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("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - static const struct snd_kcontrol_new alc889A_mb31_mixer[] = { /* Output mixers */ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), -- cgit v1.1 From b25396994b90f69c5fc6d7cd448174d7eea69f14 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 14 Nov 2011 17:32:17 +0100 Subject: ALSA: hda/realtek - Re-add the model string selection for ALC88x In the commit [c3e837bb: ALSA: hda/realtek - Rewrite ALC882 acer-aspire-* models with the auto-parser], the check of the model option got removed mistakenly. Re-added the board_config check again. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6f0f2f6..41590e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4534,7 +4534,10 @@ static int patch_alc882(struct hda_codec *codec) if (err < 0) goto error; - board_config = alc_board_codec_sid_config(codec, + 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) { -- cgit v1.1 From 5671087ffa80ea7fcc254c08de9697551fecedcf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 14 Nov 2011 17:42:11 +0100 Subject: ALSA: hda/realtek - Move ALC885 macpro and imac24 models to auto-parser The ALC882 macpro and imac24 static configs can be transferred to the auto-parser with the additional GPIO setup. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 174 ------------------------------------------ sound/pci/hda/patch_realtek.c | 57 +++++++++++++- 2 files changed, 56 insertions(+), 175 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index 403e30a..bdf0ed4 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -6,12 +6,10 @@ /* ALC882 models */ enum { ALC882_AUTO, - ALC885_MACPRO, ALC885_MBA21, ALC885_MBP3, ALC885_MB5, ALC885_MACMINI3, - ALC885_IMAC24, ALC885_IMAC91, ALC889A_MB31, ALC882_MODEL_LAST, @@ -311,71 +309,6 @@ static const struct hda_verb alc882_base_init_verbs[] = { #define alc883_init_verbs alc882_base_init_verbs -/* Mac Pro test */ -static const struct snd_kcontrol_new alc882_macpro_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), - /* FIXME: this looks suspicious... - HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x02, HDA_INPUT), - */ - { } /* end */ -}; - -static const struct hda_verb alc882_macpro_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)}, - /* Front Pin: output 0 (0x0c) */ - {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, 0x00}, - /* 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}, - /* Speaker: output */ - {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, 0x04}, - /* Headphone output (output 0 - 0x0c) */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* 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}, - - { } -}; - /* Macbook 5,1 */ static const struct hda_verb alc885_mb5_init_verbs[] = { /* DACs */ @@ -614,34 +547,6 @@ static const struct hda_verb alc885_imac91_init_verbs[] = { { } }; -/* iMac 24 mixer. */ -static const struct snd_kcontrol_new alc885_imac24_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x00, HDA_INPUT), - { } /* end */ -}; - -/* iMac 24 init verbs. */ -static const struct hda_verb alc885_imac24_init_verbs[] = { - /* Internal speakers: output 0 (0x0c) */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Internal speakers: output 0 (0x0c) */ - {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, 0x00}, - /* Headphone: output 0 (0x0c) */ - {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}, - /* Front Mic: input vref at 80% */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - { } -}; - /* Toggle speaker-output according to the hp-jack state */ static void alc885_imac24_setup(struct hda_codec *codec) { @@ -687,53 +592,6 @@ static void alc885_imac91_setup(struct hda_codec *codec) alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } -/* toggle speaker-output according to the hp-jack state */ -static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted) -{ - unsigned int gpiostate, gpiomask, gpiodir; - - gpiostate = snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_DATA, 0); - - if (!muted) - gpiostate |= (1 << pin); - else - gpiostate &= ~(1 << pin); - - gpiomask = snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_MASK, 0); - gpiomask |= (1 << pin); - - gpiodir = snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_DIRECTION, 0); - gpiodir |= (1 << pin); - - - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_MASK, gpiomask); - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DIRECTION, gpiodir); - - msleep(1); - - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DATA, gpiostate); -} - -/* set up GPIO at initialization */ -static void alc885_macpro_init_hook(struct hda_codec *codec) -{ - alc882_gpio_mute(codec, 0, 0); - alc882_gpio_mute(codec, 1, 0); -} - -/* set up GPIO and update auto-muting at initialization */ -static void alc885_imac24_init_hook(struct hda_codec *codec) -{ - alc885_macpro_init_hook(codec); - alc_hp_automute(codec); -} - /* 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 */ @@ -876,12 +734,10 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) * configuration and preset */ static const char * const alc882_models[ALC882_MODEL_LAST] = { - [ALC885_MACPRO] = "macpro", [ALC885_MB5] = "mb5", [ALC885_MACMINI3] = "macmini3", [ALC885_MBA21] = "mba21", [ALC885_MBP3] = "mbp3", - [ALC885_IMAC24] = "imac24", [ALC885_IMAC91] = "imac91", [ALC889A_MB31] = "mb31", [ALC882_AUTO] = "auto", @@ -892,16 +748,12 @@ 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, 0x0c00, "Mac Pro", ALC885_MACPRO), - SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_IMAC24), - SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_IMAC24), 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, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24), 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), @@ -973,32 +825,6 @@ static const struct alc_config_preset alc882_presets[] = { .setup = alc885_macmini3_setup, .init_hook = alc_hp_automute, }, - [ALC885_MACPRO] = { - .mixers = { alc882_macpro_mixer }, - .init_verbs = { alc882_macpro_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), - .channel_mode = alc882_ch_modes, - .input_mux = &alc882_capture_source, - .init_hook = alc885_macpro_init_hook, - }, - [ALC885_IMAC24] = { - .mixers = { alc885_imac24_mixer }, - .init_verbs = { alc885_imac24_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), - .channel_mode = alc882_ch_modes, - .input_mux = &alc882_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc885_imac24_setup, - .init_hook = alc885_imac24_init_hook, - }, [ALC885_IMAC91] = { .mixers = {alc885_imac91_mixer}, .init_verbs = { alc885_imac91_init_verbs, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 41590e3..fb48085 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4296,6 +4296,7 @@ enum { ALC882_FIXUP_ACER_ASPIRE_4930G, ALC882_FIXUP_ACER_ASPIRE_8930G, ALC882_FIXUP_ASPIRE_8930G_VERBS, + ALC885_FIXUP_MACPRO_GPIO, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -4306,6 +4307,49 @@ static void alc889_fixup_coef(struct hda_codec *codec, alc889_coef_init(codec); } +/* toggle speaker-output according to the hp-jack state */ +static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted) +{ + unsigned int gpiostate, gpiomask, gpiodir; + + gpiostate = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DATA, 0); + + if (!muted) + gpiostate |= (1 << pin); + else + gpiostate &= ~(1 << pin); + + gpiomask = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_MASK, 0); + gpiomask |= (1 << pin); + + gpiodir = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DIRECTION, 0); + gpiodir |= (1 << pin); + + + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_MASK, gpiomask); + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DIRECTION, gpiodir); + + msleep(1); + + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DATA, gpiostate); +} + +/* set up GPIO at initialization */ +static void alc885_fixup_macpro_gpio(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action != ALC_FIXUP_ACT_INIT) + return; + alc882_gpio_mute(codec, 0, 0); + alc882_gpio_mute(codec, 1, 0); +} + static const struct alc_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = ALC_FIXUP_PINS, @@ -4449,6 +4493,10 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [ALC885_FIXUP_MACPRO_GPIO] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc885_fixup_macpro_gpio, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -4479,7 +4527,14 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), - SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD), /* codec SSID */ + + /* All Apple entries are in codec SSIDs */ + 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, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD), + SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), -- cgit v1.1 From ee3b29693cb56b36fa4913b9a573d6c233bb8f03 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 15 Nov 2011 14:26:54 +0100 Subject: ALSA: hda/realtek - Move ALC880 model=medion-rim to auto-parser Translate ALC880 medion-rim static configs to the auto-parser with the additional GPIO2 verb and COEF setup. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 90 ------------------------------------------- sound/pci/hda/patch_realtek.c | 38 ++++++++++++++++++ 2 files changed, 38 insertions(+), 90 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index bea22ed..dffa7a9 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -27,7 +27,6 @@ enum { ALC880_TCL_S700, ALC880_LG, ALC880_LG_LW, - ALC880_MEDION_RIM, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -1137,78 +1136,6 @@ static void alc880_lg_lw_setup(struct hda_codec *codec) alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } -static const struct snd_kcontrol_new alc880_medion_rim_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, 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 Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_input_mux alc880_medion_rim_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x0 }, - { "Internal Mic", 0x1 }, - }, -}; - -static const struct hda_verb alc880_medion_rim_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}, - - /* 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_MUTE}, - /* Internal Speaker */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, - {0x20, AC_VERB_SET_PROC_COEF, 0x3060}, - - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - { } -}; - -/* toggle speaker-output according to the hp-jack state */ -static void alc880_medion_rim_automute(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - alc_hp_automute(codec); - /* toggle EAPD */ - if (spec->hp_jack_present) - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0); - else - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2); -} - -static void alc880_medion_rim_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! - */ - if ((res >> 28) == ALC_HP_EVENT) - alc880_medion_rim_automute(codec); -} - -static void alc880_medion_rim_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x1b; - 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 }, @@ -1506,7 +1433,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_F1734] = "F1734", [ALC880_LG] = "lg", [ALC880_LG_LW] = "lg-lw", - [ALC880_MEDION_RIM] = "medion", #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = "test", #endif @@ -1557,7 +1483,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { 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(0x161f, 0x205d, "Medion Rim 2150", ALC880_MEDION_RIM), 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), @@ -1862,21 +1787,6 @@ static const struct alc_config_preset alc880_presets[] = { .setup = alc880_lg_lw_setup, .init_hook = alc_hp_automute, }, - [ALC880_MEDION_RIM] = { - .mixers = { alc880_medion_rim_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_medion_rim_init_verbs, - alc_gpio2_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_medion_rim_capture_source, - .unsol_event = alc880_medion_rim_unsol_event, - .setup = alc880_medion_rim_setup, - .init_hook = alc880_medion_rim_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 fb48085..8d1b27b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4035,6 +4035,37 @@ static const struct hda_amp_list alc880_loopbacks[] = { #endif /* + * ALC880 fix-ups + */ +enum { + ALC880_FIXUP_GPIO2, + ALC880_FIXUP_MEDION_RIM, +}; + +static const struct alc_fixup alc880_fixups[] = { + [ALC880_FIXUP_GPIO2] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = alc_gpio2_init_verbs, + }, + [ALC880_FIXUP_MEDION_RIM] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x3060 }, + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_GPIO2, + }, +}; + +static const struct snd_pci_quirk alc880_fixup_tbl[] = { + SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM), + {} +}; + + +/* * board setups */ #ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS @@ -4080,6 +4111,11 @@ static int patch_alc880(struct hda_codec *codec) } 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) @@ -4113,6 +4149,8 @@ static int patch_alc880(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + spec->vmaster_nid = 0x0c; codec->patch_ops = alc_patch_ops; -- cgit v1.1 From 19723079db3ef1769803e05293314461ba81dede Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 15 Nov 2011 14:46:25 +0100 Subject: ALSA: hda/realtek - Move ALC880 model=lg-lw to auto-parser ALC880 model=lg-lw works fine with the auto-parser as is. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 103 ------------------------------------------ 1 file changed, 103 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index dffa7a9..5b68435 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -26,7 +26,6 @@ enum { ALC880_CLEVO, ALC880_TCL_S700, ALC880_LG, - ALC880_LG_LW, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -1051,91 +1050,6 @@ static void alc880_lg_setup(struct hda_codec *codec) alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } -/* - * LG LW20 - * - * Pin assignment: - * Speaker-out: 0x14 - * Mic-In: 0x18 - * Built-in Mic-In: 0x19 - * Line-In: 0x1b - * HP-Out: 0x1a - * SPDIF-Out: 0x1e - */ - -static const struct hda_input_mux alc880_lg_lw_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Internal Mic", 0x1 }, - { "Line In", 0x2 }, - }, -}; - -#define alc880_lg_lw_modes alc880_threestack_modes - -static const struct snd_kcontrol_new alc880_lg_lw_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("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("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, 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_lw_init_verbs[] = { - {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ - {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ - {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */ - - /* set capture source to mic-in */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, - /* speaker-out */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* HP-out */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* mic-in to input */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* built-in mic */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, 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_lw_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.speaker_pins[0] = 0x14; - 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 }, @@ -1432,7 +1346,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_FUJITSU] = "fujitsu", [ALC880_F1734] = "F1734", [ALC880_LG] = "lg", - [ALC880_LG_LW] = "lg-lw", #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = "test", #endif @@ -1489,11 +1402,9 @@ 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, 0x0018, "LG LW20", ALC880_LG_LW), 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(0x1854, 0x0077, "LG LW25", ALC880_LG_LW), 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), @@ -1773,20 +1684,6 @@ static const struct alc_config_preset alc880_presets[] = { .loopbacks = alc880_lg_loopbacks, #endif }, - [ALC880_LG_LW] = { - .mixers = { alc880_lg_lw_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_lg_lw_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_lg_lw_modes), - .channel_mode = alc880_lg_lw_modes, - .input_mux = &alc880_lg_lw_capture_source, - .unsol_event = alc_sku_unsol_event, - .setup = alc880_lg_lw_setup, - .init_hook = alc_hp_automute, - }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, -- cgit v1.1 From 04f5ade6afc4326dc6cd10d235500972fba548eb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 27 Oct 2011 20:47:07 +0200 Subject: ALSA: hda - Introduce snd_hda_get_pin_label() Create a new helper function snd_hda_get_pin_label() for getting a label string for both input and output pins. hda_get_input_pin_label() is obsoleted by this function, and the callers are replaced appropriately now by this patch. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 63 ++++++++++++++++++++++++++++++++++++++++-- sound/pci/hda/hda_local.h | 4 +-- sound/pci/hda/patch_ca0110.c | 2 +- sound/pci/hda/patch_cirrus.c | 2 +- sound/pci/hda/patch_sigmatel.c | 4 +-- 5 files changed, 66 insertions(+), 9 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e44b107..c6ff9b9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5004,8 +5004,8 @@ EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr); * "Rear", "Internal". */ -const char *hda_get_input_pin_label(struct hda_codec *codec, hda_nid_t pin, - int check_location) +static const char *hda_get_input_pin_label(struct hda_codec *codec, + hda_nid_t pin, bool check_location) { unsigned int def_conf; static const char * const mic_names[] = { @@ -5044,7 +5044,6 @@ const char *hda_get_input_pin_label(struct hda_codec *codec, hda_nid_t pin, return "Misc"; } } -EXPORT_SYMBOL_HDA(hda_get_input_pin_label); /* Check whether the location prefix needs to be added to the label. * If all mic-jacks are in the same location (e.g. rear panel), we don't @@ -5102,6 +5101,64 @@ const char *hda_get_autocfg_input_label(struct hda_codec *codec, EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label); /** + * snd_hda_get_pin_label - Get a label for the given I/O pin + * + * Get a label for the given pin. This function works for both input and + * output pins. When @cfg is given as non-NULL, the function tries to get + * an optimized label using hda_get_autocfg_input_label(). + */ +const char *snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, + const struct auto_pin_cfg *cfg) +{ + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); + int attr; + int i; + + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) + return NULL; + + attr = snd_hda_get_input_pin_attr(def_conf); + switch (get_defcfg_device(def_conf)) { + case AC_JACK_LINE_OUT: + switch (attr) { + case INPUT_PIN_ATTR_INT: + return "Speaker"; + case INPUT_PIN_ATTR_DOCK: + return "Dock Line-Out"; + case INPUT_PIN_ATTR_FRONT: + return "Front Line-Out"; + default: + return "Line-Out"; + } + case AC_JACK_SPEAKER: + return "Speaker"; + case AC_JACK_HP_OUT: + switch (attr) { + case INPUT_PIN_ATTR_DOCK: + return "Dock Headphone"; + case INPUT_PIN_ATTR_FRONT: + return "Front Headphone"; + default: + return "Headphone"; + } + case AC_JACK_SPDIF_OUT: + case AC_JACK_DIG_OTHER_OUT: + if (get_defcfg_location(def_conf) == AC_JACK_LOC_HDMI) + return "HDMI"; + else + return "SPDIF"; + } + + if (cfg) { + for (i = 0; i < cfg->num_inputs; i++) + if (cfg->inputs[i].pin == nid) + return hda_get_autocfg_input_label(codec, cfg, i); + } + return hda_get_input_pin_label(codec, nid, true); +} +EXPORT_SYMBOL_HDA(snd_hda_get_pin_label); + +/** * snd_hda_add_imux_item - Add an item to input_mux * * When the same label is used already in the existing items, the number diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 618ddad..7a10fe9 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -394,11 +394,11 @@ struct auto_pin_cfg_item { }; struct auto_pin_cfg; -const char *hda_get_input_pin_label(struct hda_codec *codec, hda_nid_t pin, - int check_location); const char *hda_get_autocfg_input_label(struct hda_codec *codec, const struct auto_pin_cfg *cfg, int input); +const char *snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, + const struct auto_pin_cfg *cfg); int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label, int index, int *type_index_ret); diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 993757b..6bd602b 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -476,7 +476,7 @@ static void parse_input(struct hda_codec *codec) if (j >= cfg->num_inputs) continue; spec->input_pins[n] = pin; - spec->input_labels[n] = hda_get_input_pin_label(codec, pin, 1); + spec->input_labels[n] = snd_hda_get_pin_label(codec, pin, NULL); spec->adcs[n] = nid; n++; } diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 2a2d864..34a460b 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -711,7 +711,7 @@ static int cs_capture_source_info(struct snd_kcontrol *kcontrol, uinfo->value.enumerated.item = spec->num_inputs - 1; idx = spec->input_idx[uinfo->value.enumerated.item]; strcpy(uinfo->value.enumerated.name, - hda_get_input_pin_label(codec, cfg->inputs[idx].pin, 1)); + snd_hda_get_pin_label(codec, cfg->inputs[idx].pin, NULL)); return 0; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 470f6f2..3b4ef0c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2872,7 +2872,7 @@ static inline int stac92xx_add_jack_mode_control(struct hda_codec *codec, } if (control) { - strcpy(name, hda_get_input_pin_label(codec, nid, 1)); + strcpy(name, snd_hda_get_pin_label(codec, nid, NULL)); return stac92xx_add_control(codec->spec, control, strcat(name, " Jack Mode"), nid); } @@ -3563,7 +3563,7 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, if (index < 0) continue; - label = hda_get_input_pin_label(codec, nid, 1); + label = snd_hda_get_pin_label(codec, nid, NULL); snd_hda_add_imux_item(dimux, label, index, &type_idx); if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) snd_hda_add_imux_item(imux, label, index, &type_idx); -- cgit v1.1 From 1835a0f9a2121ce3198dab67507d4d3e960cc09e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 27 Oct 2011 22:12:46 +0200 Subject: ALSA: hda - Cache the jack-detection value Introduce a table containing the pins and their jack-detection states for avoiding the unnecessary verbs to check the pin status at each time. When the unsol event is enabled via snd_hda_jack_detect_enable(), it automatically adds the given NID to the table. Then the driver supposes that the codec driver will set the dirty flag appropariately when an unsolicited event is invoked for that pin. The behavior for reading other pins that aren't registered in the table doesn't change. Only the pins assigned to the table are cached, so far. In near futre, this table can be extended to use the central place for the unsolicited events of all pins, etc, and eventually include the jack-detect kcontrols that replace the current input-jack stuff. Signed-off-by: Takashi Iwai --- sound/pci/hda/Makefile | 2 +- sound/pci/hda/hda_codec.c | 40 +---------- sound/pci/hda/hda_codec.h | 3 + sound/pci/hda/hda_jack.c | 148 +++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_jack.h | 63 ++++++++++++++++++ sound/pci/hda/hda_local.h | 15 ----- sound/pci/hda/patch_analog.c | 1 + sound/pci/hda/patch_cirrus.c | 18 +++-- sound/pci/hda/patch_conexant.c | 20 +++--- sound/pci/hda/patch_hdmi.c | 6 +- sound/pci/hda/patch_realtek.c | 20 +++--- sound/pci/hda/patch_sigmatel.c | 46 +++++++------ sound/pci/hda/patch_via.c | 19 +++--- 13 files changed, 284 insertions(+), 117 deletions(-) create mode 100644 sound/pci/hda/hda_jack.c create mode 100644 sound/pci/hda/hda_jack.h (limited to 'sound/pci') diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index f928d66..ace157c 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,6 +1,6 @@ snd-hda-intel-objs := hda_intel.o -snd-hda-codec-y := hda_codec.o +snd-hda-codec-y := hda_codec.o hda_jack.o snd-hda-codec-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c6ff9b9..8217ff7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -33,6 +33,7 @@ #include #include "hda_local.h" #include "hda_beep.h" +#include "hda_jack.h" #include #define CREATE_TRACE_POINTS @@ -1723,43 +1724,6 @@ int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_override_pin_caps); -/** - * snd_hda_pin_sense - execute pin sense measurement - * @codec: the CODEC to sense - * @nid: the pin NID to sense - * - * Execute necessary pin sense measurement and return its Presence Detect, - * Impedance, ELD Valid etc. status bits. - */ -u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) -{ - u32 pincap; - - if (!codec->no_trigger_sense) { - pincap = snd_hda_query_pin_caps(codec, nid); - if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ - snd_hda_codec_read(codec, nid, 0, - AC_VERB_SET_PIN_SENSE, 0); - } - return snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_SENSE, 0); -} -EXPORT_SYMBOL_HDA(snd_hda_pin_sense); - -/** - * snd_hda_jack_detect - query pin Presence Detect status - * @codec: the CODEC to sense - * @nid: the pin NID to sense - * - * Query and return the pin's Presence Detect status. - */ -int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) -{ - u32 sense = snd_hda_pin_sense(codec, nid); - return !!(sense & AC_PINSENSE_PRESENCE); -} -EXPORT_SYMBOL_HDA(snd_hda_jack_detect); - /* * read the current volume to info * if the cache exists, read the cache value. @@ -2308,6 +2272,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) } if (codec->patch_ops.free) codec->patch_ops.free(codec); + snd_hda_jack_tbl_clear(codec); codec->proc_widget_hook = NULL; codec->spec = NULL; free_hda_cache(&codec->amp_cache); @@ -3364,6 +3329,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) restore_pincfgs(codec); /* restore all current pin configs */ restore_shutup_pins(codec); hda_exec_init_verbs(codec); + snd_hda_jack_set_dirty_all(codec); if (codec->patch_ops.resume) codec->patch_ops.resume(codec); else { diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5644711..97aa65d 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -869,6 +869,9 @@ struct hda_codec { void (*proc_widget_hook)(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid); + /* jack detection */ + struct snd_array jacktbl; + #ifdef CONFIG_SND_HDA_INPUT_JACK /* jack detection */ struct snd_array jacks; diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c new file mode 100644 index 0000000..64b78a2 --- /dev/null +++ b/sound/pci/hda/hda_jack.c @@ -0,0 +1,148 @@ +/* + * Jack-detection handling for HD-audio + * + * Copyright (c) 2011 Takashi Iwai + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include +#include +#include +#include "hda_codec.h" +#include "hda_local.h" +#include "hda_jack.h" + +/* execute pin sense measurement */ +static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid) +{ + u32 pincap; + + if (!codec->no_trigger_sense) { + pincap = snd_hda_query_pin_caps(codec, nid); + if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ + snd_hda_codec_read(codec, nid, 0, + AC_VERB_SET_PIN_SENSE, 0); + } + return snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_SENSE, 0); +} + +/** + * snd_hda_jack_tbl_get - query the jack-table entry for the given NID + */ +struct hda_jack_tbl * +snd_hda_jack_tbl_get(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_jack_tbl *jack = codec->jacktbl.list; + int i; + + if (!nid || !jack) + return NULL; + for (i = 0; i < codec->jacktbl.used; i++, jack++) + if (jack->nid == nid) + return jack; + return NULL; +} +EXPORT_SYMBOL_HDA(snd_hda_jack_tbl_get); + +/** + * snd_hda_jack_tbl_new - create a jack-table entry for the given NID + */ +struct hda_jack_tbl * +snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); + if (jack) + return jack; + snd_array_init(&codec->jacktbl, sizeof(*jack), 16); + jack = snd_array_new(&codec->jacktbl); + if (!jack) + return NULL; + jack->nid = nid; + jack->jack_dirty = 1; + return jack; +} + +void snd_hda_jack_tbl_clear(struct hda_codec *codec) +{ + snd_array_free(&codec->jacktbl); +} + +/* update the cached value and notification flag if needed */ +static void jack_detect_update(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + if (jack->jack_dirty) { + jack->pin_sense = read_pin_sense(codec, jack->nid); + jack->jack_dirty = 0; + } +} + +/** + * snd_hda_set_dirty_all - Mark all the cached as dirty + * + * This function sets the dirty flag to all entries of jack table. + * It's called from the resume path in hda_codec.c. + */ +void snd_hda_jack_set_dirty_all(struct hda_codec *codec) +{ + struct hda_jack_tbl *jack = codec->jacktbl.list; + int i; + + for (i = 0; i < codec->jacktbl.used; i++, jack++) + if (jack->nid) + jack->jack_dirty = 1; +} +EXPORT_SYMBOL_HDA(snd_hda_jack_set_dirty_all); + +/** + * snd_hda_pin_sense - execute pin sense measurement + * @codec: the CODEC to sense + * @nid: the pin NID to sense + * + * Execute necessary pin sense measurement and return its Presence Detect, + * Impedance, ELD Valid etc. status bits. + */ +u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); + if (jack) { + jack_detect_update(codec, jack); + return jack->pin_sense; + } + return read_pin_sense(codec, nid); +} +EXPORT_SYMBOL_HDA(snd_hda_pin_sense); + +/** + * snd_hda_jack_detect - query pin Presence Detect status + * @codec: the CODEC to sense + * @nid: the pin NID to sense + * + * Query and return the pin's Presence Detect status. + */ +int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) +{ + u32 sense = snd_hda_pin_sense(codec, nid); + return !!(sense & AC_PINSENSE_PRESENCE); +} +EXPORT_SYMBOL_HDA(snd_hda_jack_detect); + +/** + * snd_hda_jack_detect_enable - enable the jack-detection + */ +int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, + unsigned int tag) +{ + struct hda_jack_tbl *jack = snd_hda_jack_tbl_new(codec, nid); + if (!jack) + return -ENOMEM; + return snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | tag); +} +EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable); diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h new file mode 100644 index 0000000..5c1bcb8 --- /dev/null +++ b/sound/pci/hda/hda_jack.h @@ -0,0 +1,63 @@ +/* + * Jack-detection handling for HD-audio + * + * Copyright (c) 2011 Takashi Iwai + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#ifndef __SOUND_HDA_JACK_H +#define __SOUND_HDA_JACK_H + +struct hda_jack_tbl { + hda_nid_t nid; + unsigned int pin_sense; /* cached pin-sense value */ + unsigned int jack_dirty:1; /* needs to update? */ +}; + +struct hda_jack_tbl * +snd_hda_jack_tbl_get(struct hda_codec *codec, hda_nid_t nid); + +struct hda_jack_tbl * +snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid); +void snd_hda_jack_tbl_clear(struct hda_codec *codec); + +/** + * snd_hda_jack_set_dirty - set the dirty flag for the given jack-entry + * + * Call this function when a pin-state may change, e.g. when the hardware + * notifies via an unsolicited event. + */ +static inline void snd_hda_jack_set_dirty(struct hda_codec *codec, + hda_nid_t nid) +{ + struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); + if (jack) + jack->jack_dirty = 1; +} + +void snd_hda_jack_set_dirty_all(struct hda_codec *codec); + +int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, + unsigned int tag); + +u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); +int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); + +static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) +{ + if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT)) + return false; + if (!codec->ignore_misc_bit && + (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & + AC_DEFCFG_MISC_NO_PRESENCE)) + return false; + if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)) + return false; + return true; +} + +#endif /* __SOUND_HDA_JACK_H */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 7a10fe9..08e88b82 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -505,21 +505,6 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid); int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid, unsigned int caps); -u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); -int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); - -static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) -{ - if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT)) - return false; - if (!codec->ignore_misc_bit && - (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & - AC_DEFCFG_MISC_NO_PRESENCE)) - return false; - if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)) - return false; - return true; -} /* flags for hda_nid_item */ #define HDA_NID_ITEM_AMP (1<<0) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index bcb3310..9cb14b4 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -29,6 +29,7 @@ #include "hda_codec.h" #include "hda_local.h" #include "hda_beep.h" +#include "hda_jack.h" struct ad198x_spec { const struct snd_kcontrol_new *mixers[6]; diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 34a460b..6f15877 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -26,6 +26,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_jack.h" #include /* @@ -1020,9 +1021,7 @@ static void init_output(struct hda_codec *codec) if (!cfg->speaker_outs) continue; if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | HP_EVENT); + snd_hda_jack_detect_enable(codec, nid, HP_EVENT); spec->hp_detect = 1; } } @@ -1063,9 +1062,7 @@ static void init_input(struct hda_codec *codec) AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(spec->adc_idx[i])); if (spec->mic_detect && spec->automic_idx == i) - snd_hda_codec_write(codec, pin, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | MIC_EVENT); + snd_hda_jack_detect_enable(codec, pin, MIC_EVENT); } /* specific to CS421x */ if (spec->vendor_nid == CS421X_VENDOR_NID) { @@ -1227,6 +1224,8 @@ static void cs_free(struct hda_codec *codec) static void cs_unsol_event(struct hda_codec *codec, unsigned int res) { + snd_hda_jack_set_dirty_all(codec); /* FIXME: to be more fine-grained */ + switch ((res >> 26) & 0x7f) { case HP_EVENT: cs_automute(codec); @@ -1585,10 +1584,7 @@ static void init_cs421x_digital(struct hda_codec *codec) if (!cfg->speaker_outs) continue; if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { - - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | SPDIF_EVENT); + snd_hda_jack_detect_enable(codec, nid, SPDIF_EVENT); spec->spdif_detect = 1; } } @@ -1806,6 +1802,8 @@ static int cs421x_build_controls(struct hda_codec *codec) static void cs421x_unsol_event(struct hda_codec *codec, unsigned int res) { + snd_hda_jack_set_dirty_all(codec); /* FIXME: to be more fine-grained */ + switch ((res >> 26) & 0x3f) { case HP_EVENT: case SPDIF_EVENT: diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 0de2119..220e567 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -31,6 +31,7 @@ #include "hda_codec.h" #include "hda_local.h" #include "hda_beep.h" +#include "hda_jack.h" #define CXT_PIN_DIR_IN 0x00 #define CXT_PIN_DIR_OUT 0x01 @@ -3756,6 +3757,7 @@ static void cx_auto_automic(struct hda_codec *codec) static void cx_auto_unsol_event(struct hda_codec *codec, unsigned int res) { int nid = (res & AC_UNSOL_RES_SUBTAG) >> 20; + snd_hda_jack_set_dirty(codec, nid); switch (res >> 26) { case CONEXANT_HP_EVENT: cx_auto_hp_automute(codec); @@ -3983,9 +3985,7 @@ static void enable_unsol_pins(struct hda_codec *codec, int num_pins, { int i; for (i = 0; i < num_pins; i++) - snd_hda_codec_write(codec, pins[i], 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | tag); + snd_hda_jack_detect_enable(codec, pins[i], tag); } static void cx_auto_init_output(struct hda_codec *codec) @@ -4060,16 +4060,14 @@ static void cx_auto_init_input(struct hda_codec *codec) if (spec->auto_mic) { if (spec->auto_mic_ext >= 0) { - snd_hda_codec_write(codec, - cfg->inputs[spec->auto_mic_ext].pin, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | CONEXANT_MIC_EVENT); + snd_hda_jack_detect_enable(codec, + cfg->inputs[spec->auto_mic_ext].pin, + CONEXANT_MIC_EVENT); } if (spec->auto_mic_dock >= 0) { - snd_hda_codec_write(codec, - cfg->inputs[spec->auto_mic_dock].pin, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | CONEXANT_MIC_EVENT); + snd_hda_jack_detect_enable(codec, + cfg->inputs[spec->auto_mic_dock].pin, + CONEXANT_MIC_EVENT); } cx_auto_automic(codec); } else { diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9850c5b..ea6d85d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -36,6 +36,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_jack.h" static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); @@ -766,6 +767,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; + snd_hda_jack_set_dirty(codec, pin_nid); hdmi_present_sense(&spec->pins[pin_idx], true); } @@ -1282,9 +1284,7 @@ static int generic_hdmi_init(struct hda_codec *codec) struct hdmi_eld *eld = &per_pin->sink_eld; hdmi_init_pin(codec, pin_nid); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | pin_nid); + snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); per_pin->codec = codec; INIT_DELAYED_WORK(&per_pin->work, hdmi_repoll_eld); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 14feecf..da9d227 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -33,6 +33,7 @@ #include "hda_codec.h" #include "hda_local.h" #include "hda_beep.h" +#include "hda_jack.h" /* unsol event tags */ #define ALC_FRONT_EVENT 0x01 @@ -664,6 +665,7 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) res >>= 28; else res >>= 26; + snd_hda_jack_set_dirty_all(codec); /* FIXME: to be more fine-grained */ switch (res) { case ALC_HP_EVENT: alc_hp_automute(codec); @@ -964,9 +966,7 @@ static void alc_init_automute(struct hda_codec *codec) continue; snd_printdd("realtek: Enable HP auto-muting on NID 0x%x\n", nid); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ALC_HP_EVENT); + snd_hda_jack_detect_enable(codec, nid, ALC_HP_EVENT); spec->detect_hp = 1; } @@ -978,9 +978,8 @@ static void alc_init_automute(struct hda_codec *codec) continue; snd_printdd("realtek: Enable Line-Out " "auto-muting on NID 0x%x\n", nid); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ALC_FRONT_EVENT); + snd_hda_jack_detect_enable(codec, nid, + ALC_FRONT_EVENT); spec->detect_lo = 1; } spec->automute_lo_possible = spec->detect_hp; @@ -1108,13 +1107,10 @@ static bool alc_auto_mic_check_imux(struct hda_codec *codec) return false; /* no corresponding imux */ } - snd_hda_codec_write_cache(codec, spec->ext_mic_pin, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ALC_MIC_EVENT); + snd_hda_jack_detect_enable(codec, spec->ext_mic_pin, ALC_MIC_EVENT); if (spec->dock_mic_pin) - snd_hda_codec_write_cache(codec, spec->dock_mic_pin, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ALC_MIC_EVENT); + snd_hda_jack_detect_enable(codec, spec->dock_mic_pin, + ALC_MIC_EVENT); spec->auto_mic_valid_imux = 1; spec->auto_mic = 1; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3b4ef0c..97c6df9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -37,6 +37,7 @@ #include "hda_codec.h" #include "hda_local.h" #include "hda_beep.h" +#include "hda_jack.h" enum { STAC_VREF_EVENT = 1, @@ -4244,9 +4245,7 @@ static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, if (tag < 0) return 0; } - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | tag); + snd_hda_jack_detect_enable(codec, nid, tag); return 1; } @@ -4795,24 +4794,11 @@ static void stac92xx_mic_detect(struct hda_codec *codec) mic->mux_idx); } -static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid) -{ - struct sigmatel_event *event = stac_get_event(codec, nid); - if (!event) - return; - codec->patch_ops.unsol_event(codec, (unsigned)event->tag << 26); -} - -static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) +static void handle_unsol_event(struct hda_codec *codec, + struct sigmatel_event *event) { struct sigmatel_spec *spec = codec->spec; - struct sigmatel_event *event; - int tag, data; - - tag = (res >> 26) & 0x7f; - event = stac_get_event_from_tag(codec, tag); - if (!event) - return; + int data; switch (event->type) { case STAC_HP_EVENT: @@ -4862,6 +4848,28 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) } } +static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_event *event = stac_get_event(codec, nid); + if (!event) + return; + handle_unsol_event(codec, event); +} + +static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) +{ + struct sigmatel_spec *spec = codec->spec; + struct sigmatel_event *event; + int tag; + + tag = (res >> 26) & 0x7f; + event = stac_get_event_from_tag(codec, tag); + if (!event) + return; + snd_hda_jack_set_dirty(codec, event->nid); + handle_unsol_event(codec, event); +} + static int hp_blike_system(u32 subsystem_id); static void set_hp_led_gpio(struct hda_codec *codec) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 431c0d4..3467d0c 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -54,6 +54,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_jack.h" /* Pin Widget NID */ #define VT1708_HP_PIN_NID 0x20 @@ -1708,6 +1709,8 @@ static void via_gpio_control(struct hda_codec *codec) static void via_unsol_event(struct hda_codec *codec, unsigned int res) { + snd_hda_jack_set_dirty_all(codec); /* FIXME: to be more fine-grained */ + res >>= 26; if (res & VIA_JACK_EVENT) @@ -2729,9 +2732,8 @@ static void via_auto_init_unsol_event(struct hda_codec *codec) int i; if (cfg->hp_pins[0] && is_jack_detectable(codec, cfg->hp_pins[0])) - snd_hda_codec_write(codec, cfg->hp_pins[0], 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT); + snd_hda_jack_detect_enable(codec, cfg->hp_pins[0], + VIA_HP_EVENT | VIA_JACK_EVENT); if (cfg->speaker_pins[0]) ev = VIA_LINE_EVENT; @@ -2740,16 +2742,14 @@ static void via_auto_init_unsol_event(struct hda_codec *codec) for (i = 0; i < cfg->line_outs; i++) { if (cfg->line_out_pins[i] && is_jack_detectable(codec, cfg->line_out_pins[i])) - snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ev | VIA_JACK_EVENT); + snd_hda_jack_detect_enable(codec, cfg->line_out_pins[i], + ev | VIA_JACK_EVENT); } for (i = 0; i < cfg->num_inputs; i++) { if (is_jack_detectable(codec, cfg->inputs[i].pin)) - snd_hda_codec_write(codec, cfg->inputs[i].pin, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | VIA_JACK_EVENT); + snd_hda_jack_detect_enable(codec, cfg->inputs[i].pin, + VIA_JACK_EVENT); } } @@ -2781,6 +2781,7 @@ static void vt1708_update_hp_jack_state(struct work_struct *work) vt1708_hp_work.work); if (spec->codec_type != VT1708) return; + snd_hda_jack_set_dirty_all(spec->codec); /* if jack state toggled */ if (spec->vt1708_hp_present != snd_hda_jack_detect(spec->codec, spec->autocfg.hp_pins[0])) { -- cgit v1.1 From 01a61e12b4602c82bde9797d0e153f3e53c95b04 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Oct 2011 00:03:22 +0200 Subject: ALSA: hda - Create jack-detection kcontrols Create kcontrols for pin jack-detections, which work similarly like jack-input layer. Each control will notify when the jack is plugged or unplugged, and also user can read the value at any time via the normal control API. The control elements are created with iface=CARD, so that they won't appear in the mixer apps. So far, only the pins that enabled the jack-detection are registered. For covering all pins, the transition of the common unsol-tag handling would be needed. Stay tuned. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_jack.c | 168 ++++++++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_jack.h | 12 +++ sound/pci/hda/patch_cirrus.c | 27 ++++++- sound/pci/hda/patch_conexant.c | 2 + sound/pci/hda/patch_hdmi.c | 6 ++ sound/pci/hda/patch_realtek.c | 7 ++ sound/pci/hda/patch_sigmatel.c | 7 ++ sound/pci/hda/patch_via.c | 7 ++ 8 files changed, 232 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 64b78a2..cee6a00 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -12,6 +12,7 @@ #include #include #include +#include #include "hda_codec.h" #include "hda_local.h" #include "hda_jack.h" @@ -76,9 +77,13 @@ void snd_hda_jack_tbl_clear(struct hda_codec *codec) static void jack_detect_update(struct hda_codec *codec, struct hda_jack_tbl *jack) { - if (jack->jack_dirty) { - jack->pin_sense = read_pin_sense(codec, jack->nid); + if (jack->jack_dirty || !jack->jack_cachable) { + unsigned int val = read_pin_sense(codec, jack->nid); jack->jack_dirty = 0; + if (val != jack->pin_sense) { + jack->need_notify = 1; + jack->pin_sense = val; + } } } @@ -141,8 +146,167 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, struct hda_jack_tbl *jack = snd_hda_jack_tbl_new(codec, nid); if (!jack) return -ENOMEM; + if (jack->jack_cachable) + return 0; /* already registered */ + jack->jack_cachable = 1; return snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | tag); } EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable); + +/* queue the notification when needed */ +static void jack_detect_report(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + jack_detect_update(codec, jack); + if (jack->need_notify) { + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, + &jack->kctl->id); + jack->need_notify = 0; + } +} + +/** + * snd_hda_jack_report - notify kctl when the jack state was changed + */ +void snd_hda_jack_report(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); + + if (jack) + jack_detect_report(codec, jack); +} +EXPORT_SYMBOL_HDA(snd_hda_jack_report); + +/** + * snd_hda_jack_report_sync - sync the states of all jacks and report if changed + */ +void snd_hda_jack_report_sync(struct hda_codec *codec) +{ + struct hda_jack_tbl *jack = codec->jacktbl.list; + int i; + + for (i = 0; i < codec->jacktbl.used; i++, jack++) + if (jack->nid) { + jack_detect_update(codec, jack); + jack_detect_report(codec, jack); + } +} +EXPORT_SYMBOL_HDA(snd_hda_jack_report_sync); + +/* + * jack-detection kcontrols + */ + +#define jack_detect_kctl_info snd_ctl_boolean_mono_info + +static int jack_detect_kctl_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; + + ucontrol->value.integer.value[0] = snd_hda_jack_detect(codec, nid); + return 0; +} + +static struct snd_kcontrol_new jack_detect_kctl = { + /* name is filled later */ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .info = jack_detect_kctl_info, + .get = jack_detect_kctl_get, +}; + +/** + * snd_hda_jack_add_kctl - Add a kctl for the given pin + * + * This assigns a jack-detection kctl to the given pin. The kcontrol + * will have the given name and index. + */ +int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, + const char *name, int idx) +{ + struct hda_jack_tbl *jack; + struct snd_kcontrol *kctl; + + jack = snd_hda_jack_tbl_get(codec, nid); + if (!jack) + return 0; + if (jack->kctl) + return 0; /* already created */ + kctl = snd_ctl_new1(&jack_detect_kctl, codec); + if (!kctl) + return -ENOMEM; + snprintf(kctl->id.name, sizeof(kctl->id.name), "%s Jack", name); + kctl->id.index = idx; + kctl->private_value = nid; + if (snd_hda_ctl_add(codec, nid, kctl) < 0) + return -ENOMEM; + jack->kctl = kctl; + return 0; +} + +static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, int idx, + const struct auto_pin_cfg *cfg) +{ + if (!nid) + return 0; + if (!is_jack_detectable(codec, nid)) + return 0; + return snd_hda_jack_add_kctl(codec, nid, + snd_hda_get_pin_label(codec, nid, cfg), + idx); +} + +/** + * snd_hda_jack_add_kctls - Add kctls for all pins included in the given pincfg + * + * As of now, it assigns only to the pins that enabled the detection. + * Usually this is called at the end of build_controls callback. + */ +int snd_hda_jack_add_kctls(struct hda_codec *codec, + const struct auto_pin_cfg *cfg) +{ + const hda_nid_t *p; + int i, err; + + for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) { + err = add_jack_kctl(codec, *p, i, cfg); + if (err < 0) + return err; + } + for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) { + if (*p == *cfg->line_out_pins) /* might be duplicated */ + break; + err = add_jack_kctl(codec, *p, i, cfg); + if (err < 0) + return err; + } + for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) { + if (*p == *cfg->line_out_pins) /* might be duplicated */ + break; + err = add_jack_kctl(codec, *p, i, cfg); + if (err < 0) + return err; + } + for (i = 0; i < cfg->num_inputs; i++) { + err = add_jack_kctl(codec, cfg->inputs[i].pin, 0, cfg); + if (err < 0) + return err; + } + for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) { + err = add_jack_kctl(codec, *p, i, cfg); + if (err < 0) + return err; + } + err = add_jack_kctl(codec, cfg->dig_in_pin, 0, cfg); + if (err < 0) + return err; + err = add_jack_kctl(codec, cfg->mono_out_pin, 0, cfg); + if (err < 0) + return err; + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctls); diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 5c1bcb8..b5983ea 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -15,7 +15,10 @@ struct hda_jack_tbl { hda_nid_t nid; unsigned int pin_sense; /* cached pin-sense value */ + unsigned int jack_cachable:1; /* can be updated via unsol events */ unsigned int jack_dirty:1; /* needs to update? */ + unsigned int need_notify:1; /* to be notified? */ + struct snd_kcontrol *kctl; /* assigned kctl for jack-detection */ }; struct hda_jack_tbl * @@ -60,4 +63,13 @@ static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) return true; } +int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, + const char *name, int idx); +int snd_hda_jack_add_kctls(struct hda_codec *codec, + const struct auto_pin_cfg *cfg); + +void snd_hda_jack_report(struct hda_codec *codec, hda_nid_t nid); +void snd_hda_jack_report_sync(struct hda_codec *codec); + + #endif /* __SOUND_HDA_JACK_H */ diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 6f15877..135fd49 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1192,11 +1192,14 @@ static int cs_init(struct hda_codec *codec) init_output(codec); init_input(codec); init_digital(codec); + snd_hda_jack_report_sync(codec); + return 0; } static int cs_build_controls(struct hda_codec *codec) { + struct cs_spec *spec = codec->spec; int err; err = build_output(codec); @@ -1211,7 +1214,15 @@ static int cs_build_controls(struct hda_codec *codec) err = build_digital_input(codec); if (err < 0) return err; - return cs_init(codec); + err = cs_init(codec); + if (err < 0) + return err; + + err = snd_hda_jack_add_kctls(codec, &spec->autocfg); + if (err < 0) + return err; + + return 0; } static void cs_free(struct hda_codec *codec) @@ -1234,6 +1245,7 @@ static void cs_unsol_event(struct hda_codec *codec, unsigned int res) cs_automic(codec); break; } + snd_hda_jack_report_sync(codec); } static const struct hda_codec_ops cs_patch_ops = { @@ -1611,6 +1623,7 @@ static int cs421x_init(struct hda_codec *codec) init_output(codec); init_input(codec); init_cs421x_digital(codec); + snd_hda_jack_report_sync(codec); return 0; } @@ -1786,6 +1799,7 @@ static int build_cs421x_output(struct hda_codec *codec) static int cs421x_build_controls(struct hda_codec *codec) { + struct cs_spec *spec = codec->spec; int err; err = build_cs421x_output(codec); @@ -1797,7 +1811,15 @@ static int cs421x_build_controls(struct hda_codec *codec) err = build_digital_output(codec); if (err < 0) return err; - return cs421x_init(codec); + err = cs421x_init(codec); + if (err < 0) + return err; + + err = snd_hda_jack_add_kctls(codec, &spec->autocfg); + if (err < 0) + return err; + + return 0; } static void cs421x_unsol_event(struct hda_codec *codec, unsigned int res) @@ -1814,6 +1836,7 @@ static void cs421x_unsol_event(struct hda_codec *codec, unsigned int res) cs_automic(codec); break; } + snd_hda_jack_report_sync(codec); } static int parse_cs421x_input(struct hda_codec *codec) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 220e567..25fdd1e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3770,6 +3770,7 @@ static void cx_auto_unsol_event(struct hda_codec *codec, unsigned int res) snd_hda_input_jack_report(codec, nid); break; } + snd_hda_jack_report_sync(codec); } /* check whether the pin config is suitable for auto-mic switching; @@ -4095,6 +4096,7 @@ static int cx_auto_init(struct hda_codec *codec) cx_auto_init_output(codec); cx_auto_init_input(codec); cx_auto_init_digital(codec); + snd_hda_jack_report_sync(codec); return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ea6d85d..f01c5ef 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -769,6 +769,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) snd_hda_jack_set_dirty(codec, pin_nid); hdmi_present_sense(&spec->pins[pin_idx], true); + snd_hda_jack_report_sync(codec); } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) @@ -1268,6 +1269,10 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) if (err < 0) return err; + err = snd_hda_jack_add_kctl(codec, per_pin->pin_nid, + "HDMI", pin_idx); + if (err < 0) + return err; } return 0; @@ -1290,6 +1295,7 @@ static int generic_hdmi_init(struct hda_codec *codec) INIT_DELAYED_WORK(&per_pin->work, hdmi_repoll_eld); snd_hda_eld_proc_new(codec, eld, pin_idx); } + snd_hda_jack_report_sync(codec); return 0; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index da9d227..04beae0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -677,6 +677,7 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) alc_mic_automute(codec); break; } + snd_hda_jack_report_sync(codec); } /* call init functions of standard auto-mute helpers */ @@ -2054,6 +2055,10 @@ static int alc_build_controls(struct hda_codec *codec) alc_free_kctls(codec); /* no longer needed */ + err = snd_hda_jack_add_kctls(codec, &spec->autocfg); + if (err < 0) + return err; + return 0; } @@ -2081,6 +2086,8 @@ static int alc_init(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_INIT); + snd_hda_jack_report_sync(codec); + hda_call_check_power_status(codec, 0x01); return 0; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 97c6df9..90954b8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1212,6 +1212,10 @@ static int stac92xx_build_controls(struct hda_codec *codec) return err; } + err = snd_hda_jack_add_kctls(codec, &spec->autocfg); + if (err < 0) + return err; + return 0; } @@ -4473,6 +4477,8 @@ static int stac92xx_init(struct hda_codec *codec) stac_toggle_power_map(codec, nid, 0); } + snd_hda_jack_report_sync(codec); + /* sync mute LED */ if (spec->gpio_led) hda_call_check_power_status(codec, 0x01); @@ -4868,6 +4874,7 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) return; snd_hda_jack_set_dirty(codec, event->nid); handle_unsol_event(codec, event); + snd_hda_jack_report_sync(codec); } static int hp_blike_system(u32 subsystem_id); diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 3467d0c..8529396 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1500,6 +1500,11 @@ static int via_build_controls(struct hda_codec *codec) analog_low_current_mode(codec); via_free_kctls(codec); /* no longer needed */ + + err = snd_hda_jack_add_kctls(codec, &spec->autocfg); + if (err < 0) + return err; + return 0; } @@ -1722,6 +1727,7 @@ static void via_unsol_event(struct hda_codec *codec, via_hp_automute(codec); else if (res == VIA_GPIO_EVENT) via_gpio_control(codec); + snd_hda_jack_report_sync(codec); } #ifdef CONFIG_PM @@ -2771,6 +2777,7 @@ static int via_init(struct hda_codec *codec) via_auto_init_unsol_event(codec); via_hp_automute(codec); + snd_hda_jack_report_sync(codec); return 0; } -- cgit v1.1 From 3a93897ea37cbb8277f8a4232c12c0c18168a7db Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Oct 2011 01:16:55 +0200 Subject: ALSA: hda - Manage unsol tags in hda_jack.c Manage the tags assigned for unsolicited events dynamically together with the jack-detection routines. Basically this is almost same as what we've done in patch_sigmatel.c. Assign the new tag number for each new unsol event, associate with the given NID and the action type, etc. With this change, now all pins looked over in snd_hda_jack_add_kctls() are actually enabled for detection now even if the pins aren't used for jack-retasking by the driver. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_jack.c | 61 +++++++++++++++--------- sound/pci/hda/hda_jack.h | 28 +++++++---- sound/pci/hda/patch_cirrus.c | 8 +--- sound/pci/hda/patch_conexant.c | 8 ++-- sound/pci/hda/patch_hdmi.c | 13 ++++-- sound/pci/hda/patch_realtek.c | 6 ++- sound/pci/hda/patch_sigmatel.c | 104 +++++++++++------------------------------ sound/pci/hda/patch_via.c | 3 +- 8 files changed, 105 insertions(+), 126 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index cee6a00..8829d5c 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -51,6 +51,24 @@ snd_hda_jack_tbl_get(struct hda_codec *codec, hda_nid_t nid) EXPORT_SYMBOL_HDA(snd_hda_jack_tbl_get); /** + * snd_hda_jack_tbl_get_from_tag - query the jack-table entry for the given tag + */ +struct hda_jack_tbl * +snd_hda_jack_tbl_get_from_tag(struct hda_codec *codec, unsigned char tag) +{ + struct hda_jack_tbl *jack = codec->jacktbl.list; + int i; + + if (!tag || !jack) + return NULL; + for (i = 0; i < codec->jacktbl.used; i++, jack++) + if (jack->tag == tag) + return jack; + return NULL; +} +EXPORT_SYMBOL_HDA(snd_hda_jack_tbl_get_from_tag); + +/** * snd_hda_jack_tbl_new - create a jack-table entry for the given NID */ struct hda_jack_tbl * @@ -65,6 +83,7 @@ snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid) return NULL; jack->nid = nid; jack->jack_dirty = 1; + jack->tag = codec->jacktbl.used; return jack; } @@ -77,7 +96,7 @@ void snd_hda_jack_tbl_clear(struct hda_codec *codec) static void jack_detect_update(struct hda_codec *codec, struct hda_jack_tbl *jack) { - if (jack->jack_dirty || !jack->jack_cachable) { + if (jack->jack_dirty || !jack->jack_detect) { unsigned int val = read_pin_sense(codec, jack->nid); jack->jack_dirty = 0; if (val != jack->pin_sense) { @@ -141,17 +160,19 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_detect); * snd_hda_jack_detect_enable - enable the jack-detection */ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, - unsigned int tag) + unsigned char action) { struct hda_jack_tbl *jack = snd_hda_jack_tbl_new(codec, nid); if (!jack) return -ENOMEM; - if (jack->jack_cachable) + if (jack->jack_detect) return 0; /* already registered */ - jack->jack_cachable = 1; + jack->jack_detect = 1; + if (action) + jack->action = action; return snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | tag); + AC_USRSP_EN | jack->tag); } EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable); @@ -168,18 +189,6 @@ static void jack_detect_report(struct hda_codec *codec, } /** - * snd_hda_jack_report - notify kctl when the jack state was changed - */ -void snd_hda_jack_report(struct hda_codec *codec, hda_nid_t nid) -{ - struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); - - if (jack) - jack_detect_report(codec, jack); -} -EXPORT_SYMBOL_HDA(snd_hda_jack_report); - -/** * snd_hda_jack_report_sync - sync the states of all jacks and report if changed */ void snd_hda_jack_report_sync(struct hda_codec *codec) @@ -231,7 +240,7 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, struct hda_jack_tbl *jack; struct snd_kcontrol *kctl; - jack = snd_hda_jack_tbl_get(codec, nid); + jack = snd_hda_jack_tbl_new(codec, nid); if (!jack) return 0; if (jack->kctl) @@ -251,20 +260,28 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, int idx, const struct auto_pin_cfg *cfg) { + unsigned int def_conf, conn; + int err; + if (!nid) return 0; if (!is_jack_detectable(codec, nid)) return 0; - return snd_hda_jack_add_kctl(codec, nid, + def_conf = snd_hda_codec_get_pincfg(codec, nid); + conn = get_defcfg_connect(def_conf); + if (conn != AC_JACK_PORT_COMPLEX) + return 0; + + err = snd_hda_jack_add_kctl(codec, nid, snd_hda_get_pin_label(codec, nid, cfg), idx); + if (err < 0) + return err; + return snd_hda_jack_detect_enable(codec, nid, 0); } /** * snd_hda_jack_add_kctls - Add kctls for all pins included in the given pincfg - * - * As of now, it assigns only to the pins that enabled the detection. - * Usually this is called at the end of build_controls callback. */ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index b5983ea..69a67f8 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -14,8 +14,12 @@ struct hda_jack_tbl { hda_nid_t nid; + unsigned char action; /* event action (0 = none) */ + unsigned char tag; /* unsol event tag */ + unsigned int private_data; /* arbitrary data */ + /* jack-detection stuff */ unsigned int pin_sense; /* cached pin-sense value */ - unsigned int jack_cachable:1; /* can be updated via unsol events */ + unsigned int jack_detect:1; /* capable of jack-detection? */ unsigned int jack_dirty:1; /* needs to update? */ unsigned int need_notify:1; /* to be notified? */ struct snd_kcontrol *kctl; /* assigned kctl for jack-detection */ @@ -23,29 +27,34 @@ struct hda_jack_tbl { struct hda_jack_tbl * snd_hda_jack_tbl_get(struct hda_codec *codec, hda_nid_t nid); +struct hda_jack_tbl * +snd_hda_jack_tbl_get_from_tag(struct hda_codec *codec, unsigned char tag); struct hda_jack_tbl * snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid); void snd_hda_jack_tbl_clear(struct hda_codec *codec); /** - * snd_hda_jack_set_dirty - set the dirty flag for the given jack-entry + * snd_hda_jack_get_action - get jack-tbl entry for the tag * - * Call this function when a pin-state may change, e.g. when the hardware - * notifies via an unsolicited event. + * Call this from the unsol event handler to get the assigned action for the + * event. This will mark the dirty flag for the later reporting, too. */ -static inline void snd_hda_jack_set_dirty(struct hda_codec *codec, - hda_nid_t nid) +static inline unsigned char +snd_hda_jack_get_action(struct hda_codec *codec, unsigned int tag) { - struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); - if (jack) + struct hda_jack_tbl *jack = snd_hda_jack_tbl_get_from_tag(codec, tag); + if (jack) { jack->jack_dirty = 1; + return jack->action; + } + return 0; } void snd_hda_jack_set_dirty_all(struct hda_codec *codec); int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, - unsigned int tag); + unsigned char action); u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); @@ -68,7 +77,6 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg); -void snd_hda_jack_report(struct hda_codec *codec, hda_nid_t nid); void snd_hda_jack_report_sync(struct hda_codec *codec); diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 135fd49..0e34554 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1235,9 +1235,7 @@ static void cs_free(struct hda_codec *codec) static void cs_unsol_event(struct hda_codec *codec, unsigned int res) { - snd_hda_jack_set_dirty_all(codec); /* FIXME: to be more fine-grained */ - - switch ((res >> 26) & 0x7f) { + switch (snd_hda_jack_get_action(codec, res >> 26)) { case HP_EVENT: cs_automute(codec); break; @@ -1824,9 +1822,7 @@ static int cs421x_build_controls(struct hda_codec *codec) static void cs421x_unsol_event(struct hda_codec *codec, unsigned int res) { - snd_hda_jack_set_dirty_all(codec); /* FIXME: to be more fine-grained */ - - switch ((res >> 26) & 0x3f) { + switch (snd_hda_jack_get_action(codec, res >> 26)) { case HP_EVENT: case SPDIF_EVENT: cs_automute(codec); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 25fdd1e..40bd75b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3757,8 +3757,8 @@ static void cx_auto_automic(struct hda_codec *codec) static void cx_auto_unsol_event(struct hda_codec *codec, unsigned int res) { int nid = (res & AC_UNSOL_RES_SUBTAG) >> 20; - snd_hda_jack_set_dirty(codec, nid); - switch (res >> 26) { + + switch (snd_hda_jack_get_action(codec, res >> 26)) { case CONEXANT_HP_EVENT: cx_auto_hp_automute(codec); break; @@ -3982,11 +3982,11 @@ static void mute_outputs(struct hda_codec *codec, int num_nids, } static void enable_unsol_pins(struct hda_codec *codec, int num_pins, - hda_nid_t *pins, unsigned int tag) + hda_nid_t *pins, unsigned int action) { int i; for (i = 0; i < num_pins; i++) - snd_hda_jack_detect_enable(codec, pins[i], tag); + snd_hda_jack_detect_enable(codec, pins[i], action); } static void cx_auto_init_output(struct hda_codec *codec) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f01c5ef..ea30bf4 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -754,10 +754,18 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, bool retry); static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) { struct hdmi_spec *spec = codec->spec; - int pin_nid = res >> AC_UNSOL_RES_TAG_SHIFT; + int tag = res >> AC_UNSOL_RES_TAG_SHIFT; + int pin_nid; int pd = !!(res & AC_UNSOL_RES_PD); int eldv = !!(res & AC_UNSOL_RES_ELDV); int pin_idx; + struct hda_jack_tbl *jack; + + jack = snd_hda_jack_tbl_get_from_tag(codec, tag); + if (!jack) + return; + pin_nid = jack->nid; + jack->jack_dirty = 1; printk(KERN_INFO "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", @@ -767,7 +775,6 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; - snd_hda_jack_set_dirty(codec, pin_nid); hdmi_present_sense(&spec->pins[pin_idx], true); snd_hda_jack_report_sync(codec); } @@ -801,7 +808,7 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) int tag = res >> AC_UNSOL_RES_TAG_SHIFT; int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; - if (pin_nid_to_pin_index(spec, tag) < 0) { + if (!snd_hda_jack_tbl_get_from_tag(codec, tag)) { snd_printd(KERN_INFO "Unexpected HDMI event tag 0x%x\n", tag); return; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 04beae0..9a90cda 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -185,6 +185,7 @@ struct alc_spec { unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */ unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */ unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */ + unsigned int use_jack_tbl:1; /* 1 for model=auto */ /* auto-mute control */ int automute_mode; @@ -661,11 +662,13 @@ static void alc_mic_automute(struct hda_codec *codec) /* unsolicited event for HP jack sensing */ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) { + struct alc_spec *spec = codec->spec; if (codec->vendor_id == 0x10ec0880) res >>= 28; else res >>= 26; - snd_hda_jack_set_dirty_all(codec); /* FIXME: to be more fine-grained */ + if (spec->use_jack_tbl) + res = snd_hda_jack_get_action(codec, res); switch (res) { case ALC_HP_EVENT: alc_hp_automute(codec); @@ -3896,6 +3899,7 @@ static void set_capture_mixer(struct hda_codec *codec) static void alc_auto_init_std(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + spec->use_jack_tbl = 1; alc_auto_init_multi_out(codec); alc_auto_init_extra_out(codec); alc_auto_init_analog_input(codec); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 90954b8..dd6569f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -177,13 +177,6 @@ enum { STAC_9872_MODELS }; -struct sigmatel_event { - hda_nid_t nid; - unsigned char type; - unsigned char tag; - int data; -}; - struct sigmatel_mic_route { hda_nid_t pin; signed char mux_idx; @@ -231,9 +224,6 @@ struct sigmatel_spec { const hda_nid_t *pwr_nids; const hda_nid_t *dac_list; - /* events */ - struct snd_array events; - /* playback */ struct hda_input_mux *mono_mux; unsigned int cur_mmux; @@ -4182,49 +4172,18 @@ static int stac92xx_add_jack(struct hda_codec *codec, #endif /* CONFIG_SND_HDA_INPUT_JACK */ } -static int stac_add_event(struct sigmatel_spec *spec, hda_nid_t nid, +static int stac_add_event(struct hda_codec *codec, hda_nid_t nid, unsigned char type, int data) { - struct sigmatel_event *event; + struct hda_jack_tbl *event; - snd_array_init(&spec->events, sizeof(*event), 32); - event = snd_array_new(&spec->events); + event = snd_hda_jack_tbl_new(codec, nid); if (!event) return -ENOMEM; - event->nid = nid; - event->type = type; - event->tag = spec->events.used; - event->data = data; - - return event->tag; -} - -static struct sigmatel_event *stac_get_event(struct hda_codec *codec, - hda_nid_t nid) -{ - struct sigmatel_spec *spec = codec->spec; - struct sigmatel_event *event = spec->events.list; - int i; - - for (i = 0; i < spec->events.used; i++, event++) { - if (event->nid == nid) - return event; - } - return NULL; -} + event->action = type; + event->private_data = data; -static struct sigmatel_event *stac_get_event_from_tag(struct hda_codec *codec, - unsigned char tag) -{ - struct sigmatel_spec *spec = codec->spec; - struct sigmatel_event *event = spec->events.list; - int i; - - for (i = 0; i < spec->events.used; i++, event++) { - if (event->tag == tag) - return event; - } - return NULL; + return 0; } /* check if given nid is a valid pin and no other events are assigned @@ -4234,22 +4193,17 @@ static struct sigmatel_event *stac_get_event_from_tag(struct hda_codec *codec, static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, unsigned int type) { - struct sigmatel_event *event; - int tag; + struct hda_jack_tbl *event; if (!is_jack_detectable(codec, nid)) return 0; - event = stac_get_event(codec, nid); - if (event) { - if (event->type != type) - return 0; - tag = event->tag; - } else { - tag = stac_add_event(codec->spec, nid, type, 0); - if (tag < 0) - return 0; - } - snd_hda_jack_detect_enable(codec, nid, tag); + event = snd_hda_jack_tbl_new(codec, nid); + if (!event) + return -ENOMEM; + if (event->action && event->action != type) + return 0; + event->action = type; + snd_hda_jack_detect_enable(codec, nid, 0); return 1; } @@ -4536,7 +4490,6 @@ static void stac92xx_free(struct hda_codec *codec) stac92xx_shutup(codec); snd_hda_input_jack_free(codec); - snd_array_free(&spec->events); kfree(spec); snd_hda_detach_beep_device(codec); @@ -4801,12 +4754,12 @@ static void stac92xx_mic_detect(struct hda_codec *codec) } static void handle_unsol_event(struct hda_codec *codec, - struct sigmatel_event *event) + struct hda_jack_tbl *event) { struct sigmatel_spec *spec = codec->spec; int data; - switch (event->type) { + switch (event->action) { case STAC_HP_EVENT: case STAC_LO_EVENT: stac92xx_hp_detect(codec); @@ -4816,7 +4769,7 @@ static void handle_unsol_event(struct hda_codec *codec, break; } - switch (event->type) { + switch (event->action) { case STAC_HP_EVENT: case STAC_LO_EVENT: case STAC_MIC_EVENT: @@ -4849,14 +4802,14 @@ static void handle_unsol_event(struct hda_codec *codec, AC_VERB_GET_GPIO_DATA, 0); /* toggle VREF state based on GPIOx status */ snd_hda_codec_write(codec, codec->afg, 0, 0x7e0, - !!(data & (1 << event->data))); + !!(data & (1 << event->private_data))); break; } } static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid) { - struct sigmatel_event *event = stac_get_event(codec, nid); + struct hda_jack_tbl *event = snd_hda_jack_tbl_get(codec, nid); if (!event) return; handle_unsol_event(codec, event); @@ -4864,15 +4817,14 @@ static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid) static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) { - struct sigmatel_spec *spec = codec->spec; - struct sigmatel_event *event; + struct hda_jack_tbl *event; int tag; tag = (res >> 26) & 0x7f; - event = stac_get_event_from_tag(codec, tag); + event = snd_hda_jack_tbl_get_from_tag(codec, tag); if (!event) return; - snd_hda_jack_set_dirty(codec, event->nid); + event->jack_dirty = 1; handle_unsol_event(codec, event); snd_hda_jack_report_sync(codec); } @@ -5857,15 +5809,13 @@ again: switch (spec->board_config) { case STAC_HP_M4: /* Enable VREF power saving on GPIO1 detect */ - err = stac_add_event(spec, codec->afg, + err = stac_add_event(codec, codec->afg, STAC_VREF_EVENT, 0x02); if (err < 0) return err; snd_hda_codec_write_cache(codec, codec->afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02); - snd_hda_codec_write_cache(codec, codec->afg, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | err); + snd_hda_jack_detect_enable(codec, codec->afg, 0); spec->gpio_mask |= 0x02; break; } @@ -6338,14 +6288,12 @@ static int patch_stac9205(struct hda_codec *codec) snd_hda_codec_set_pincfg(codec, 0x20, 0x1c410030); /* Enable unsol response for GPIO4/Dock HP connection */ - err = stac_add_event(spec, codec->afg, STAC_VREF_EVENT, 0x01); + err = stac_add_event(codec, codec->afg, STAC_VREF_EVENT, 0x01); if (err < 0) return err; snd_hda_codec_write_cache(codec, codec->afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x10); - snd_hda_codec_write_cache(codec, codec->afg, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | err); + snd_hda_jack_detect_enable(codec, codec->afg, 0); spec->gpio_dir = 0x0b; spec->eapd_mask = 0x01; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 8529396..f73c986 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1714,9 +1714,8 @@ static void via_gpio_control(struct hda_codec *codec) static void via_unsol_event(struct hda_codec *codec, unsigned int res) { - snd_hda_jack_set_dirty_all(codec); /* FIXME: to be more fine-grained */ - res >>= 26; + res = snd_hda_jack_get_action(codec, res); if (res & VIA_JACK_EVENT) set_widgets_power_state(codec); -- cgit v1.1 From 35be544af367170a9c6bf63adcf9d0cb2d569dbb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 2 Nov 2011 08:36:06 +0100 Subject: ALSA: Introduce common helper functions for jack-detection control Now move the helper function for creating and reporting the jack-detection to the common place. The driver that needs this functionality should select CONFIG_SND_KCTL_JACK kconfig. Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 1 + sound/pci/hda/hda_jack.c | 56 +++++++----------------------------------------- sound/pci/hda/hda_jack.h | 1 - 3 files changed, 9 insertions(+), 49 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bb7e102..163b6b5 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -2,6 +2,7 @@ menuconfig SND_HDA_INTEL tristate "Intel HD Audio" select SND_PCM select SND_VMASTER + select SND_KCTL_JACK help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 8829d5c..a2ab52b 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -97,12 +97,8 @@ static void jack_detect_update(struct hda_codec *codec, struct hda_jack_tbl *jack) { if (jack->jack_dirty || !jack->jack_detect) { - unsigned int val = read_pin_sense(codec, jack->nid); + jack->pin_sense = read_pin_sense(codec, jack->nid); jack->jack_dirty = 0; - if (val != jack->pin_sense) { - jack->need_notify = 1; - jack->pin_sense = val; - } } } @@ -142,6 +138,8 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) } EXPORT_SYMBOL_HDA(snd_hda_pin_sense); +#define get_jack_plug_state(sense) !!(sense & AC_PINSENSE_PRESENCE) + /** * snd_hda_jack_detect - query pin Presence Detect status * @codec: the CODEC to sense @@ -152,7 +150,7 @@ EXPORT_SYMBOL_HDA(snd_hda_pin_sense); int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) { u32 sense = snd_hda_pin_sense(codec, nid); - return !!(sense & AC_PINSENSE_PRESENCE); + return get_jack_plug_state(sense); } EXPORT_SYMBOL_HDA(snd_hda_jack_detect); @@ -176,58 +174,23 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable); -/* queue the notification when needed */ -static void jack_detect_report(struct hda_codec *codec, - struct hda_jack_tbl *jack) -{ - jack_detect_update(codec, jack); - if (jack->need_notify) { - snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, - &jack->kctl->id); - jack->need_notify = 0; - } -} - /** * snd_hda_jack_report_sync - sync the states of all jacks and report if changed */ void snd_hda_jack_report_sync(struct hda_codec *codec) { struct hda_jack_tbl *jack = codec->jacktbl.list; - int i; + int i, state; for (i = 0; i < codec->jacktbl.used; i++, jack++) if (jack->nid) { jack_detect_update(codec, jack); - jack_detect_report(codec, jack); + state = get_jack_plug_state(jack->pin_sense); + snd_kctl_jack_notify(codec->bus->card, jack->kctl, state); } } EXPORT_SYMBOL_HDA(snd_hda_jack_report_sync); -/* - * jack-detection kcontrols - */ - -#define jack_detect_kctl_info snd_ctl_boolean_mono_info - -static int jack_detect_kctl_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; - - ucontrol->value.integer.value[0] = snd_hda_jack_detect(codec, nid); - return 0; -} - -static struct snd_kcontrol_new jack_detect_kctl = { - /* name is filled later */ - .iface = SNDRV_CTL_ELEM_IFACE_CARD, - .access = SNDRV_CTL_ELEM_ACCESS_READ, - .info = jack_detect_kctl_info, - .get = jack_detect_kctl_get, -}; - /** * snd_hda_jack_add_kctl - Add a kctl for the given pin * @@ -245,12 +208,9 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, return 0; if (jack->kctl) return 0; /* already created */ - kctl = snd_ctl_new1(&jack_detect_kctl, codec); + kctl = snd_kctl_jack_new(name, idx, codec); if (!kctl) return -ENOMEM; - snprintf(kctl->id.name, sizeof(kctl->id.name), "%s Jack", name); - kctl->id.index = idx; - kctl->private_value = nid; if (snd_hda_ctl_add(codec, nid, kctl) < 0) return -ENOMEM; jack->kctl = kctl; diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 69a67f8..4bb75ee 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -21,7 +21,6 @@ struct hda_jack_tbl { unsigned int pin_sense; /* cached pin-sense value */ unsigned int jack_detect:1; /* capable of jack-detection? */ unsigned int jack_dirty:1; /* needs to update? */ - unsigned int need_notify:1; /* to be notified? */ struct snd_kcontrol *kctl; /* assigned kctl for jack-detection */ }; -- cgit v1.1 From aad37dbd563010252e1bedb6dad6cddb867b9235 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 2 Nov 2011 08:54:51 +0100 Subject: ALSA: hda - Merge input-jack helpers to hda_jack.c We can use the very same table in hda_jack.c for managing the list for input-jack elements, too. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 108 ----------------------------------------- sound/pci/hda/hda_jack.c | 97 +++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_jack.h | 4 ++ sound/pci/hda/hda_local.h | 4 -- sound/pci/hda/patch_conexant.c | 1 - sound/pci/hda/patch_hdmi.c | 1 - sound/pci/hda/patch_realtek.c | 1 - sound/pci/hda/patch_sigmatel.c | 1 - 8 files changed, 100 insertions(+), 117 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8217ff7..e57698f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5275,113 +5275,5 @@ void snd_print_pcm_bits(int pcm, char *buf, int buflen) } EXPORT_SYMBOL_HDA(snd_print_pcm_bits); -#ifdef CONFIG_SND_HDA_INPUT_JACK -/* - * Input-jack notification support - */ -struct hda_jack_item { - hda_nid_t nid; - int type; - struct snd_jack *jack; -}; - -static const char *get_jack_default_name(struct hda_codec *codec, hda_nid_t nid, - int type) -{ - switch (type) { - case SND_JACK_HEADPHONE: - return "Headphone"; - case SND_JACK_MICROPHONE: - return "Mic"; - case SND_JACK_LINEOUT: - return "Line-out"; - case SND_JACK_LINEIN: - return "Line-in"; - case SND_JACK_HEADSET: - return "Headset"; - case SND_JACK_VIDEOOUT: - return "HDMI/DP"; - default: - return "Misc"; - } -} - -static void hda_free_jack_priv(struct snd_jack *jack) -{ - struct hda_jack_item *jacks = jack->private_data; - jacks->nid = 0; - jacks->jack = NULL; -} - -int snd_hda_input_jack_add(struct hda_codec *codec, hda_nid_t nid, int type, - const char *name) -{ - struct hda_jack_item *jack; - int err; - - snd_array_init(&codec->jacks, sizeof(*jack), 32); - jack = snd_array_new(&codec->jacks); - if (!jack) - return -ENOMEM; - - jack->nid = nid; - jack->type = type; - if (!name) - name = get_jack_default_name(codec, nid, type); - err = snd_jack_new(codec->bus->card, name, type, &jack->jack); - if (err < 0) { - jack->nid = 0; - return err; - } - jack->jack->private_data = jack; - jack->jack->private_free = hda_free_jack_priv; - return 0; -} -EXPORT_SYMBOL_HDA(snd_hda_input_jack_add); - -void snd_hda_input_jack_report(struct hda_codec *codec, hda_nid_t nid) -{ - struct hda_jack_item *jacks = codec->jacks.list; - int i; - - if (!jacks) - return; - - for (i = 0; i < codec->jacks.used; i++, jacks++) { - unsigned int pin_ctl; - unsigned int present; - int type; - - if (jacks->nid != nid) - continue; - present = snd_hda_jack_detect(codec, nid); - type = jacks->type; - if (type == (SND_JACK_HEADPHONE | SND_JACK_LINEOUT)) { - pin_ctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - type = (pin_ctl & AC_PINCTL_HP_EN) ? - SND_JACK_HEADPHONE : SND_JACK_LINEOUT; - } - snd_jack_report(jacks->jack, present ? type : 0); - } -} -EXPORT_SYMBOL_HDA(snd_hda_input_jack_report); - -/* free jack instances manually when clearing/reconfiguring */ -void snd_hda_input_jack_free(struct hda_codec *codec) -{ - if (!codec->bus->shutdown && codec->jacks.list) { - struct hda_jack_item *jacks = codec->jacks.list; - int i; - for (i = 0; i < codec->jacks.used; i++, jacks++) { - if (jacks->jack) - snd_device_free(codec->bus->card, jacks->jack); - } - } - snd_array_free(&codec->jacks); -} -EXPORT_SYMBOL_HDA(snd_hda_input_jack_free); -#endif /* CONFIG_SND_HDA_INPUT_JACK */ - MODULE_DESCRIPTION("HDA codec core"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index a2ab52b..1389958 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -13,6 +13,7 @@ #include #include #include +#include #include "hda_codec.h" #include "hda_local.h" #include "hda_jack.h" @@ -87,8 +88,15 @@ snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid) return jack; } +#ifdef CONFIG_SND_HDA_INPUT_JACK +static void snd_hda_input_jack_free(struct hda_codec *codec); +#else +#define snd_hda_input_jack_free(codec) +#endif + void snd_hda_jack_tbl_clear(struct hda_codec *codec) { + snd_hda_input_jack_free(codec); snd_array_free(&codec->jacktbl); } @@ -186,7 +194,7 @@ void snd_hda_jack_report_sync(struct hda_codec *codec) if (jack->nid) { jack_detect_update(codec, jack); state = get_jack_plug_state(jack->pin_sense); - snd_kctl_jack_notify(codec->bus->card, jack->kctl, state); + snd_kctl_jack_report(codec->bus->card, jack->kctl, state); } } EXPORT_SYMBOL_HDA(snd_hda_jack_report_sync); @@ -287,3 +295,90 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, return 0; } EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctls); + +#ifdef CONFIG_SND_HDA_INPUT_JACK +/* + * Input-jack notification support + */ +static const char *get_jack_default_name(struct hda_codec *codec, hda_nid_t nid, + int type) +{ + switch (type) { + case SND_JACK_HEADPHONE: + return "Headphone"; + case SND_JACK_MICROPHONE: + return "Mic"; + case SND_JACK_LINEOUT: + return "Line-out"; + case SND_JACK_LINEIN: + return "Line-in"; + case SND_JACK_HEADSET: + return "Headset"; + case SND_JACK_VIDEOOUT: + return "HDMI/DP"; + default: + return "Misc"; + } +} + +static void hda_free_jack_priv(struct snd_jack *jack) +{ + struct hda_jack_tbl *jacks = jack->private_data; + jacks->nid = 0; + jacks->jack = NULL; +} + +int snd_hda_input_jack_add(struct hda_codec *codec, hda_nid_t nid, int type, + const char *name) +{ + struct hda_jack_tbl *jack = snd_hda_jack_tbl_new(codec, nid); + int err; + + if (!jack) + return -ENOMEM; + if (!name) + name = get_jack_default_name(codec, nid, type); + err = snd_jack_new(codec->bus->card, name, type, &jack->jack); + if (err < 0) + return err; + jack->jack->private_data = jack; + jack->jack->private_free = hda_free_jack_priv; + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_input_jack_add); + +void snd_hda_input_jack_report(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); + unsigned int pin_ctl; + unsigned int present; + int type; + + if (!jack) + return; + + present = snd_hda_jack_detect(codec, nid); + type = jack->type; + if (type == (SND_JACK_HEADPHONE | SND_JACK_LINEOUT)) { + pin_ctl = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + type = (pin_ctl & AC_PINCTL_HP_EN) ? + SND_JACK_HEADPHONE : SND_JACK_LINEOUT; + } + snd_jack_report(jack->jack, present ? type : 0); +} +EXPORT_SYMBOL_HDA(snd_hda_input_jack_report); + +/* free jack instances manually when clearing/reconfiguring */ +static void snd_hda_input_jack_free(struct hda_codec *codec) +{ + if (!codec->bus->shutdown && codec->jacktbl.list) { + struct hda_jack_tbl *jack = codec->jacktbl.list; + int i; + for (i = 0; i < codec->jacktbl.used; i++, jack++) { + if (jack->jack) + snd_device_free(codec->bus->card, jack->jack); + } + } +} +#endif /* CONFIG_SND_HDA_INPUT_JACK */ diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 4bb75ee..f8f97c7 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -22,6 +22,10 @@ struct hda_jack_tbl { unsigned int jack_detect:1; /* capable of jack-detection? */ unsigned int jack_dirty:1; /* needs to update? */ struct snd_kcontrol *kctl; /* assigned kctl for jack-detection */ +#ifdef CONFIG_SND_HDA_INPUT_JACK + int type; + struct snd_jack *jack; +#endif }; struct hda_jack_tbl * diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 08e88b82..13f6814 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -680,7 +680,6 @@ void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen); int snd_hda_input_jack_add(struct hda_codec *codec, hda_nid_t nid, int type, const char *name); void snd_hda_input_jack_report(struct hda_codec *codec, hda_nid_t nid); -void snd_hda_input_jack_free(struct hda_codec *codec); #else /* CONFIG_SND_HDA_INPUT_JACK */ static inline int snd_hda_input_jack_add(struct hda_codec *codec, hda_nid_t nid, int type, @@ -692,9 +691,6 @@ static inline void snd_hda_input_jack_report(struct hda_codec *codec, hda_nid_t nid) { } -static inline void snd_hda_input_jack_free(struct hda_codec *codec) -{ -} #endif /* CONFIG_SND_HDA_INPUT_JACK */ #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 40bd75b..ae9c028 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -475,7 +475,6 @@ static int conexant_init(struct hda_codec *codec) static void conexant_free(struct hda_codec *codec) { - snd_hda_input_jack_free(codec); snd_hda_detach_beep_device(codec); kfree(codec->spec); } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ea30bf4..bb8cfc6 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1318,7 +1318,6 @@ static void generic_hdmi_free(struct hda_codec *codec) cancel_delayed_work(&per_pin->work); snd_hda_eld_proc_free(codec, eld); } - snd_hda_input_jack_free(codec); flush_workqueue(codec->bus->workq); kfree(spec); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9a90cda..933c8cf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2474,7 +2474,6 @@ static void alc_free(struct hda_codec *codec) return; alc_shutup(codec); - snd_hda_input_jack_free(codec); alc_free_kctls(codec); alc_free_bind_ctls(codec); kfree(spec); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dd6569f..73bf7cd0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4489,7 +4489,6 @@ static void stac92xx_free(struct hda_codec *codec) return; stac92xx_shutup(codec); - snd_hda_input_jack_free(codec); kfree(spec); snd_hda_detach_beep_device(codec); -- cgit v1.1 From d1cb620081f51c78cf95224efb593a886875078f Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 11 Nov 2011 17:13:15 +0100 Subject: ALSA: HDA: Jack: Export required functions from hda_jack.c These two functions are being used by the codec-idt and codec-hdmi modules, so they need to be exported properly. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_jack.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 1389958..eac002d 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -87,6 +87,7 @@ snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid) jack->tag = codec->jacktbl.used; return jack; } +EXPORT_SYMBOL_HDA(snd_hda_jack_tbl_new); #ifdef CONFIG_SND_HDA_INPUT_JACK static void snd_hda_input_jack_free(struct hda_codec *codec); @@ -224,6 +225,7 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, jack->kctl = kctl; return 0; } +EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl); static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, int idx, const struct auto_pin_cfg *cfg) -- cgit v1.1 From cfc7c9d307b6a3557e333f960218d344d3a70ce7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 11 Nov 2011 17:53:03 +0100 Subject: ALSA: hda/jack - Fix NULL-dereference at probing At probing time, the elements that aren't assigned to kctl or jack may be called. Need proper NULL-checks. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_jack.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index eac002d..ef36cbb9 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -194,6 +194,8 @@ void snd_hda_jack_report_sync(struct hda_codec *codec) for (i = 0; i < codec->jacktbl.used; i++, jack++) if (jack->nid) { jack_detect_update(codec, jack); + if (!jack->kctl) + continue; state = get_jack_plug_state(jack->pin_sense); snd_kctl_jack_report(codec->bus->card, jack->kctl, state); } @@ -356,7 +358,7 @@ void snd_hda_input_jack_report(struct hda_codec *codec, hda_nid_t nid) unsigned int present; int type; - if (!jack) + if (!jack || !jack->jack) return; present = snd_hda_jack_detect(codec, nid); -- cgit v1.1 From 344b01aecdc1e1173b5aa86208ef583489de7710 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 11 Nov 2011 17:54:19 +0100 Subject: ALSA: hda/jack - Fix the assignment of input jack-type The type field was lost during the transition. Restored. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_jack.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index ef36cbb9..3bcf623 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -345,6 +345,7 @@ int snd_hda_input_jack_add(struct hda_codec *codec, hda_nid_t nid, int type, err = snd_jack_new(codec->bus->card, name, type, &jack->jack); if (err < 0) return err; + jack->type = type; jack->jack->private_data = jack; jack->jack->private_free = hda_free_jack_priv; return 0; -- cgit v1.1 From fc5b15f13886afad43fc6c0040af7cb5172a0bd8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 14 Nov 2011 10:32:21 +0100 Subject: ALSA: hda - Add missing initialization of kctl jack status Otherwise the jack kctls will report invalid values until the jack is re-plugged. Reported-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_jack.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 3bcf623..e014562 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -225,6 +225,8 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, if (snd_hda_ctl_add(codec, nid, kctl) < 0) return -ENOMEM; jack->kctl = kctl; + snd_kctl_jack_report(codec->bus->card, kctl, + snd_hda_jack_detect(codec, nid)); return 0; } EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl); -- cgit v1.1 From bf815bf0a3c3b8ad6cd97cda6bc29cc3708fe749 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 16 Nov 2011 14:28:33 +0100 Subject: ALSA: hda - Add missing inclusion of linux/export.h This is needed newly since 3.2... Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_jack.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index e014562..25f7565 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -11,6 +11,7 @@ #include #include +#include #include #include #include -- cgit v1.1 From 201e06ffa9ef9b5265e636617f4fa20cd1490343 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 16 Nov 2011 15:33:26 +0100 Subject: ALSA: hda - Give more unique names by snd_hda_get_pin_label() The function now gives more unique names for the output pins by adding some prefix and suffix for the location and the channels. Otherwise, it can pass the index number. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 141 +++++++++++++++++++++++++++++++---------- sound/pci/hda/hda_jack.c | 24 +++---- sound/pci/hda/hda_local.h | 5 +- sound/pci/hda/patch_ca0110.c | 6 +- sound/pci/hda/patch_cirrus.c | 5 +- sound/pci/hda/patch_sigmatel.c | 8 ++- 6 files changed, 135 insertions(+), 54 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e57698f..e050f89 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5066,61 +5066,136 @@ const char *hda_get_autocfg_input_label(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label); +/* get a unique suffix or an index number */ +static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins, + int num_pins, int *indexp) +{ + static const char * const channel_sfx[] = { + " Front", " Surrount", " CLFE", " Side" + }; + int i; + + for (i = 0; i < num_pins; i++) { + if (pins[i] == nid) { + if (num_pins == 1) + return ""; + if (num_pins > ARRAY_SIZE(channel_sfx)) { + if (indexp) + *indexp = i; + return ""; + } + return channel_sfx[i]; + } + } + return NULL; +} + +static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, + const struct auto_pin_cfg *cfg, + const char *name, char *label, int maxlen, + int *indexp) +{ + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); + int attr = snd_hda_get_input_pin_attr(def_conf); + const char *pfx = "", *sfx = ""; + + /* handle as a speaker if it's a fixed line-out */ + if (!strcmp(name, "Line-Out") && attr == INPUT_PIN_ATTR_INT) + name = "Speaker"; + /* check the location */ + switch (attr) { + case INPUT_PIN_ATTR_DOCK: + pfx = "Dock "; + break; + case INPUT_PIN_ATTR_FRONT: + pfx = "Front "; + break; + } + if (cfg) { + /* try to give a unique suffix if needed */ + sfx = check_output_sfx(nid, cfg->line_out_pins, cfg->line_outs, + indexp); + if (!sfx) + sfx = check_output_sfx(nid, cfg->hp_pins, cfg->hp_outs, + indexp); + if (!sfx) + sfx = check_output_sfx(nid, cfg->speaker_pins, cfg->speaker_outs, + indexp); + if (!sfx) + sfx = ""; + } + snprintf(label, maxlen, "%s%s%s", pfx, name, sfx); + return 1; +} + /** * snd_hda_get_pin_label - Get a label for the given I/O pin * * Get a label for the given pin. This function works for both input and * output pins. When @cfg is given as non-NULL, the function tries to get * an optimized label using hda_get_autocfg_input_label(). + * + * This function tries to give a unique label string for the pin as much as + * possible. For example, when the multiple line-outs are present, it adds + * the channel suffix like "Front", "Surround", etc (only when @cfg is given). + * If no unique name with a suffix is available and @indexp is non-NULL, the + * index number is stored in the pointer. */ -const char *snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg) +int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, + const struct auto_pin_cfg *cfg, + char *label, int maxlen, int *indexp) { unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); - int attr; + const char *name = NULL; int i; + if (indexp) + *indexp = 0; if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) - return NULL; + return 0; - attr = snd_hda_get_input_pin_attr(def_conf); switch (get_defcfg_device(def_conf)) { case AC_JACK_LINE_OUT: - switch (attr) { - case INPUT_PIN_ATTR_INT: - return "Speaker"; - case INPUT_PIN_ATTR_DOCK: - return "Dock Line-Out"; - case INPUT_PIN_ATTR_FRONT: - return "Front Line-Out"; - default: - return "Line-Out"; - } + return fill_audio_out_name(codec, nid, cfg, "Line-Out", + label, maxlen, indexp); case AC_JACK_SPEAKER: - return "Speaker"; + return fill_audio_out_name(codec, nid, cfg, "Speaker", + label, maxlen, indexp); case AC_JACK_HP_OUT: - switch (attr) { - case INPUT_PIN_ATTR_DOCK: - return "Dock Headphone"; - case INPUT_PIN_ATTR_FRONT: - return "Front Headphone"; - default: - return "Headphone"; - } + return fill_audio_out_name(codec, nid, cfg, "Headphone", + label, maxlen, indexp); case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: if (get_defcfg_location(def_conf) == AC_JACK_LOC_HDMI) - return "HDMI"; + name = "HDMI"; else - return "SPDIF"; - } - - if (cfg) { - for (i = 0; i < cfg->num_inputs; i++) - if (cfg->inputs[i].pin == nid) - return hda_get_autocfg_input_label(codec, cfg, i); + name = "SPDIF"; + if (cfg && indexp) { + for (i = 0; i < cfg->dig_outs; i++) + if (cfg->dig_out_pins[i] == nid) { + *indexp = i; + break; + } + } + break; + default: + if (cfg) { + for (i = 0; i < cfg->num_inputs; i++) { + if (cfg->inputs[i].pin != nid) + continue; + name = hda_get_autocfg_input_label(codec, cfg, i); + if (name) + break; + } + } + if (!name) + name = hda_get_input_pin_label(codec, nid, true); + break; } - return hda_get_input_pin_label(codec, nid, true); + if (!name) + return 0; + strlcpy(label, name, maxlen); + return 1; } EXPORT_SYMBOL_HDA(snd_hda_get_pin_label); diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 25f7565..39490151 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -232,11 +232,12 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl); -static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, int idx, +static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, const struct auto_pin_cfg *cfg) { unsigned int def_conf, conn; - int err; + char name[44]; + int idx, err; if (!nid) return 0; @@ -247,9 +248,8 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, int idx, if (conn != AC_JACK_PORT_COMPLEX) return 0; - err = snd_hda_jack_add_kctl(codec, nid, - snd_hda_get_pin_label(codec, nid, cfg), - idx); + snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx); + err = snd_hda_jack_add_kctl(codec, nid, name, idx); if (err < 0) return err; return snd_hda_jack_detect_enable(codec, nid, 0); @@ -265,38 +265,38 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, int i, err; for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) { - err = add_jack_kctl(codec, *p, i, cfg); + err = add_jack_kctl(codec, *p, cfg); if (err < 0) return err; } for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, i, cfg); + err = add_jack_kctl(codec, *p, cfg); if (err < 0) return err; } for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, i, cfg); + err = add_jack_kctl(codec, *p, cfg); if (err < 0) return err; } for (i = 0; i < cfg->num_inputs; i++) { - err = add_jack_kctl(codec, cfg->inputs[i].pin, 0, cfg); + err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg); if (err < 0) return err; } for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) { - err = add_jack_kctl(codec, *p, i, cfg); + err = add_jack_kctl(codec, *p, cfg); if (err < 0) return err; } - err = add_jack_kctl(codec, cfg->dig_in_pin, 0, cfg); + err = add_jack_kctl(codec, cfg->dig_in_pin, cfg); if (err < 0) return err; - err = add_jack_kctl(codec, cfg->mono_out_pin, 0, cfg); + err = add_jack_kctl(codec, cfg->mono_out_pin, cfg); if (err < 0) return err; return 0; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 13f6814..ef09716 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -397,8 +397,9 @@ struct auto_pin_cfg; const char *hda_get_autocfg_input_label(struct hda_codec *codec, const struct auto_pin_cfg *cfg, int input); -const char *snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg); +int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, + const struct auto_pin_cfg *cfg, + char *label, int maxlen, int *indexp); int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label, int index, int *type_index_ret); diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 6bd602b..09ccfab 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -41,7 +41,7 @@ struct ca0110_spec { hda_nid_t dig_out; hda_nid_t dig_in; unsigned int num_inputs; - const char *input_labels[AUTO_PIN_LAST]; + char input_labels[AUTO_PIN_LAST][32]; struct hda_pcm pcm_rec[2]; /* PCM information */ }; @@ -476,7 +476,9 @@ static void parse_input(struct hda_codec *codec) if (j >= cfg->num_inputs) continue; spec->input_pins[n] = pin; - spec->input_labels[n] = snd_hda_get_pin_label(codec, pin, NULL); + snd_hda_get_pin_label(codec, pin, cfg, + spec->input_labels[n], + sizeof(spec->input_labels[n]), NULL); spec->adcs[n] = nid; n++; } diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 0e34554..0ba0387 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -711,8 +711,9 @@ static int cs_capture_source_info(struct snd_kcontrol *kcontrol, if (uinfo->value.enumerated.item >= spec->num_inputs) uinfo->value.enumerated.item = spec->num_inputs - 1; idx = spec->input_idx[uinfo->value.enumerated.item]; - strcpy(uinfo->value.enumerated.name, - snd_hda_get_pin_label(codec, cfg->inputs[idx].pin, NULL)); + snd_hda_get_pin_label(codec, cfg->inputs[idx].pin, cfg, + uinfo->value.enumerated.name, + sizeof(uinfo->value.enumerated.name), NULL); return 0; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 73bf7cd0..0988dc4 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2867,7 +2867,8 @@ static inline int stac92xx_add_jack_mode_control(struct hda_codec *codec, } if (control) { - strcpy(name, snd_hda_get_pin_label(codec, nid, NULL)); + snd_hda_get_pin_label(codec, nid, &spec->autocfg, + name, sizeof(name), NULL); return stac92xx_add_control(codec->spec, control, strcat(name, " Jack Mode"), nid); } @@ -3545,7 +3546,7 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, for (i = 0; i < spec->num_dmics; i++) { hda_nid_t nid; int index, type_idx; - const char *label; + char label[32]; nid = spec->dmic_nids[i]; if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN) @@ -3558,7 +3559,8 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, if (index < 0) continue; - label = snd_hda_get_pin_label(codec, nid, NULL); + snd_hda_get_pin_label(codec, nid, &spec->autocfg, + label, sizeof(label), NULL); snd_hda_add_imux_item(dimux, label, index, &type_idx); if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) snd_hda_add_imux_item(imux, label, index, &type_idx); -- cgit v1.1 From d6018bb566f6eef277184278b105e04705e8aeb6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Nov 2011 14:17:16 +0100 Subject: ALSA: hda - Fix a typo Reported-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e050f89..b703e25 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5071,7 +5071,7 @@ static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins, int num_pins, int *indexp) { static const char * const channel_sfx[] = { - " Front", " Surrount", " CLFE", " Side" + " Front", " Surround", " CLFE", " Side" }; int i; -- cgit v1.1 From 358b6e62b86f6313d114e0f6b7d8f8adaf85ed9c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Nov 2011 14:34:20 +0100 Subject: ALSA: hda - Don't add channel suffix for headphone pin labels The multiple headphone pins are usually handled as copied from the same source, not as individual channels like front and surround. Thus it'd be more correct to avoid the channel suffix for "Headphone" pin labels in snd_hda_get_pin_label() but give an index number instead. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 52 ++++++++++++++++++++++++++++------------------- 1 file changed, 31 insertions(+), 21 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b703e25..de3325e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5066,6 +5066,16 @@ const char *hda_get_autocfg_input_label(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label); +/* return the position of NID in the list, or -1 if not found */ +static int find_idx_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums) +{ + int i; + for (i = 0; i < nums; i++) + if (list[i] == nid) + return i; + return -1; +} + /* get a unique suffix or an index number */ static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins, int num_pins, int *indexp) @@ -5075,19 +5085,17 @@ static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins, }; int i; - for (i = 0; i < num_pins; i++) { - if (pins[i] == nid) { - if (num_pins == 1) - return ""; - if (num_pins > ARRAY_SIZE(channel_sfx)) { - if (indexp) - *indexp = i; - return ""; - } - return channel_sfx[i]; - } + i = find_idx_in_nid_list(nid, pins, num_pins); + if (i < 0) + return NULL; + if (num_pins == 1) + return ""; + if (num_pins > ARRAY_SIZE(channel_sfx)) { + if (indexp) + *indexp = i; + return ""; } - return NULL; + return channel_sfx[i]; } static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, @@ -5116,13 +5124,16 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, sfx = check_output_sfx(nid, cfg->line_out_pins, cfg->line_outs, indexp); if (!sfx) - sfx = check_output_sfx(nid, cfg->hp_pins, cfg->hp_outs, - indexp); - if (!sfx) sfx = check_output_sfx(nid, cfg->speaker_pins, cfg->speaker_outs, indexp); - if (!sfx) + if (!sfx) { + /* don't add channel suffix for Headphone controls */ + int idx = find_idx_in_nid_list(nid, cfg->hp_pins, + cfg->hp_outs); + if (idx >= 0) + *indexp = idx; sfx = ""; + } } snprintf(label, maxlen, "%s%s%s", pfx, name, sfx); return 1; @@ -5171,11 +5182,10 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, else name = "SPDIF"; if (cfg && indexp) { - for (i = 0; i < cfg->dig_outs; i++) - if (cfg->dig_out_pins[i] == nid) { - *indexp = i; - break; - } + i = find_idx_in_nid_list(nid, cfg->dig_out_pins, + cfg->dig_outs); + if (i >= 0) + *indexp = i; } break; default: -- cgit v1.1 From 01b65bfb4f8cd45b0d44547c961ef59a0bcf74be Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 24 Nov 2011 14:31:46 +0100 Subject: ALSA: hda - Supports more audio streams So far, the driver supports up to 10 streams. This is a restriction in hda_intel.c and hda_codec.c: in the former, the fixed array size limits the amount, and in the latter, the fixed device-number assignment table (in get_empty_pcm_device()) limits the possibility. This patch reduces the restriction by - using linked list for managing PCM instances in hda_intel.c, and - assigning non-fixed device numbers for the extra devices Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 6 ++++++ sound/pci/hda/hda_codec.h | 3 --- sound/pci/hda/hda_intel.c | 48 +++++++++++++++++++++++------------------------ 3 files changed, 30 insertions(+), 27 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4562e9d..4463f9a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3850,6 +3850,12 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) if (!test_and_set_bit(audio_idx[type][i], bus->pcm_dev_bits)) return audio_idx[type][i]; + /* non-fixed slots starting from 10 */ + for (i = 10; i < 32; i++) { + if (!test_and_set_bit(i, bus->pcm_dev_bits)) + return i; + } + snd_printk(KERN_WARNING "Too many %s devices\n", snd_hda_pcm_type_name[type]); return -EAGAIN; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5644711..17cee4e 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -547,9 +547,6 @@ enum { /* max. codec address */ #define HDA_MAX_CODEC_ADDRESS 0x0f -/* max number of PCM devics per card */ -#define HDA_MAX_PCMS 10 - /* * generic arrays */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 096507d..ddd7f3b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -407,6 +407,14 @@ struct azx_rb { u32 res[AZX_MAX_CODECS]; /* last read value */ }; +struct azx_pcm { + struct azx *chip; + struct snd_pcm *pcm; + struct hda_codec *codec; + struct hda_pcm_stream *hinfo[2]; + struct list_head list; +}; + struct azx { struct snd_card *card; struct pci_dev *pci; @@ -434,7 +442,7 @@ struct azx { struct azx_dev *azx_dev; /* PCM */ - struct snd_pcm *pcm[HDA_MAX_PCMS]; + struct list_head pcm_list; /* azx_pcm list */ /* HD codec */ unsigned short codec_mask; @@ -1486,10 +1494,9 @@ static void azx_bus_reset(struct hda_bus *bus) azx_init_chip(chip, 1); #ifdef CONFIG_PM if (chip->initialized) { - int i; - - for (i = 0; i < HDA_MAX_PCMS; i++) - snd_pcm_suspend_all(chip->pcm[i]); + struct azx_pcm *p; + list_for_each_entry(p, &chip->pcm_list, list) + snd_pcm_suspend_all(p->pcm); snd_hda_suspend(chip->bus); snd_hda_resume(chip->bus); } @@ -1667,12 +1674,6 @@ static struct snd_pcm_hardware azx_pcm_hw = { .fifo_size = 0, }; -struct azx_pcm { - struct azx *chip; - struct hda_codec *codec; - struct hda_pcm_stream *hinfo[2]; -}; - static int azx_pcm_open(struct snd_pcm_substream *substream) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); @@ -2197,7 +2198,7 @@ static void azx_pcm_free(struct snd_pcm *pcm) { struct azx_pcm *apcm = pcm->private_data; if (apcm) { - apcm->chip->pcm[pcm->device] = NULL; + list_del(&apcm->list); kfree(apcm); } } @@ -2215,14 +2216,11 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, unsigned int size; int s, err; - if (pcm_dev >= HDA_MAX_PCMS) { - snd_printk(KERN_ERR SFX "Invalid PCM device number %d\n", - pcm_dev); - return -EINVAL; - } - if (chip->pcm[pcm_dev]) { - snd_printk(KERN_ERR SFX "PCM %d already exists\n", pcm_dev); - return -EBUSY; + list_for_each_entry(apcm, &chip->pcm_list, list) { + if (apcm->pcm->device == pcm_dev) { + snd_printk(KERN_ERR SFX "PCM %d already exists\n", pcm_dev); + return -EBUSY; + } } err = snd_pcm_new(chip->card, cpcm->name, pcm_dev, cpcm->stream[SNDRV_PCM_STREAM_PLAYBACK].substreams, @@ -2235,12 +2233,13 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, if (apcm == NULL) return -ENOMEM; apcm->chip = chip; + apcm->pcm = pcm; apcm->codec = codec; pcm->private_data = apcm; pcm->private_free = azx_pcm_free; if (cpcm->pcm_type == HDA_PCM_TYPE_MODEM) pcm->dev_class = SNDRV_PCM_CLASS_MODEM; - chip->pcm[pcm_dev] = pcm; + list_add_tail(&apcm->list, &chip->pcm_list); cpcm->pcm = pcm; for (s = 0; s < 2; s++) { apcm->hinfo[s] = &cpcm->stream[s]; @@ -2370,12 +2369,12 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); struct azx *chip = card->private_data; - int i; + struct azx_pcm *p; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); azx_clear_irq_pending(chip); - for (i = 0; i < HDA_MAX_PCMS; i++) - snd_pcm_suspend_all(chip->pcm[i]); + list_for_each_entry(p, &chip->pcm_list, list) + snd_pcm_suspend_all(p->pcm); if (chip->initialized) snd_hda_suspend(chip->bus); azx_stop_chip(chip); @@ -2672,6 +2671,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, check_msi(chip); chip->dev_index = dev; INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work); + INIT_LIST_HEAD(&chip->pcm_list); chip->position_fix[0] = chip->position_fix[1] = check_position_fix(chip, position_fix[dev]); -- cgit v1.1 From a4567cb389301262f7ff392074eb3b0864498737 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 24 Nov 2011 14:44:19 +0100 Subject: ALSA: hda - Increase the max number of coverters/pins in patch_hdmi.c The new hardware tends to have more and more. As a temporary fix, just increase the number for now. For a long-term solution, we should assign the cvts/pins dynamically. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c505fd5..6e0756f 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -48,8 +48,8 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); * * The HDA correspondence of pipes/ports are converter/pin nodes. */ -#define MAX_HDMI_CVTS 4 -#define MAX_HDMI_PINS 4 +#define MAX_HDMI_CVTS 8 +#define MAX_HDMI_PINS 8 struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; @@ -1126,12 +1126,12 @@ static int hdmi_parse_codec(struct hda_codec *codec) /* */ -static char *generic_hdmi_pcm_names[MAX_HDMI_PINS] = { - "HDMI 0", - "HDMI 1", - "HDMI 2", - "HDMI 3", -}; +static char *get_hdmi_pcm_name(int idx) +{ + static char names[MAX_HDMI_PINS][8]; + sprintf(&names[idx][0], "HDMI %d", idx); + return &names[idx][0]; +} /* * HDMI callbacks @@ -1209,7 +1209,7 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) struct hda_pcm_stream *pstr; info = &spec->pcm_rec[pin_idx]; - info->name = generic_hdmi_pcm_names[pin_idx]; + info->name = get_hdmi_pcm_name(pin_idx); info->pcm_type = HDA_PCM_TYPE_HDMI; pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; @@ -1364,7 +1364,7 @@ static int simple_playback_build_pcms(struct hda_codec *codec) chans = get_wcaps(codec, spec->cvts[i].cvt_nid); chans = get_wcaps_channels(chans); - info->name = generic_hdmi_pcm_names[i]; + info->name = get_hdmi_pcm_name(i); info->pcm_type = HDA_PCM_TYPE_HDMI; pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; snd_BUG_ON(!spec->pcm_playback); -- cgit v1.1 From 31ef22579302ac42054bebecb528710f46580925 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Dec 2011 17:41:36 +0100 Subject: ALSA: hda - Integrate input-jack stuff into kctl-jack Instead of managing input-jack stuff separately, call all stuff inside the kctl-jack creation, deletion and report. The caller no longer needs to care about input-jack. The better integration between input-jack and kctl-jack should be done in the upper layer in near future, but for now, it's implemented locally for more tests. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_jack.c | 159 ++++++++++++++++------------------------- sound/pci/hda/hda_local.h | 20 ------ sound/pci/hda/patch_conexant.c | 52 +++----------- sound/pci/hda/patch_hdmi.c | 22 ++---- sound/pci/hda/patch_realtek.c | 52 -------------- sound/pci/hda/patch_sigmatel.c | 46 ------------ 6 files changed, 74 insertions(+), 277 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 39490151..d8a35da 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -90,15 +90,19 @@ snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid) } EXPORT_SYMBOL_HDA(snd_hda_jack_tbl_new); -#ifdef CONFIG_SND_HDA_INPUT_JACK -static void snd_hda_input_jack_free(struct hda_codec *codec); -#else -#define snd_hda_input_jack_free(codec) -#endif - void snd_hda_jack_tbl_clear(struct hda_codec *codec) { - snd_hda_input_jack_free(codec); +#ifdef CONFIG_SND_HDA_INPUT_JACK + /* free jack instances manually when clearing/reconfiguring */ + if (!codec->bus->shutdown && codec->jacktbl.list) { + struct hda_jack_tbl *jack = codec->jacktbl.list; + int i; + for (i = 0; i < codec->jacktbl.used; i++, jack++) { + if (jack->jack) + snd_device_free(codec->bus->card, jack->jack); + } + } +#endif snd_array_free(&codec->jacktbl); } @@ -199,10 +203,44 @@ void snd_hda_jack_report_sync(struct hda_codec *codec) continue; state = get_jack_plug_state(jack->pin_sense); snd_kctl_jack_report(codec->bus->card, jack->kctl, state); +#ifdef CONFIG_SND_HDA_INPUT_JACK + if (jack->jack) + snd_jack_report(jack->jack, + state ? jack->type : 0); +#endif } } EXPORT_SYMBOL_HDA(snd_hda_jack_report_sync); +#ifdef CONFIG_SND_HDA_INPUT_JACK +/* guess the jack type from the pin-config */ +static int get_input_jack_type(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); + switch (get_defcfg_device(def_conf)) { + case AC_JACK_LINE_OUT: + case AC_JACK_SPEAKER: + return SND_JACK_LINEOUT; + case AC_JACK_HP_OUT: + return SND_JACK_HEADPHONE; + case AC_JACK_SPDIF_OUT: + case AC_JACK_DIG_OTHER_OUT: + return SND_JACK_AVOUT; + case AC_JACK_MIC_IN: + return SND_JACK_MICROPHONE; + default: + return SND_JACK_LINEIN; + } +} + +static void hda_free_jack_priv(struct snd_jack *jack) +{ + struct hda_jack_tbl *jacks = jack->private_data; + jacks->nid = 0; + jacks->jack = NULL; +} +#endif + /** * snd_hda_jack_add_kctl - Add a kctl for the given pin * @@ -214,6 +252,7 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, { struct hda_jack_tbl *jack; struct snd_kcontrol *kctl; + int err, state; jack = snd_hda_jack_tbl_new(codec, nid); if (!jack) @@ -223,11 +262,21 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, kctl = snd_kctl_jack_new(name, idx, codec); if (!kctl) return -ENOMEM; - if (snd_hda_ctl_add(codec, nid, kctl) < 0) - return -ENOMEM; + err = snd_hda_ctl_add(codec, nid, kctl); + if (err < 0) + return err; jack->kctl = kctl; - snd_kctl_jack_report(codec->bus->card, kctl, - snd_hda_jack_detect(codec, nid)); + state = snd_hda_jack_detect(codec, nid); + snd_kctl_jack_report(codec->bus->card, kctl, state); +#ifdef CONFIG_SND_HDA_INPUT_JACK + jack->type = get_input_jack_type(codec, nid); + err = snd_jack_new(codec->bus->card, name, jack->type, &jack->jack); + if (err < 0) + return err; + jack->jack->private_data = jack; + jack->jack->private_free = hda_free_jack_priv; + snd_jack_report(jack->jack, state ? jack->type : 0); +#endif return 0; } EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl); @@ -302,91 +351,3 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, return 0; } EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctls); - -#ifdef CONFIG_SND_HDA_INPUT_JACK -/* - * Input-jack notification support - */ -static const char *get_jack_default_name(struct hda_codec *codec, hda_nid_t nid, - int type) -{ - switch (type) { - case SND_JACK_HEADPHONE: - return "Headphone"; - case SND_JACK_MICROPHONE: - return "Mic"; - case SND_JACK_LINEOUT: - return "Line-out"; - case SND_JACK_LINEIN: - return "Line-in"; - case SND_JACK_HEADSET: - return "Headset"; - case SND_JACK_VIDEOOUT: - return "HDMI/DP"; - default: - return "Misc"; - } -} - -static void hda_free_jack_priv(struct snd_jack *jack) -{ - struct hda_jack_tbl *jacks = jack->private_data; - jacks->nid = 0; - jacks->jack = NULL; -} - -int snd_hda_input_jack_add(struct hda_codec *codec, hda_nid_t nid, int type, - const char *name) -{ - struct hda_jack_tbl *jack = snd_hda_jack_tbl_new(codec, nid); - int err; - - if (!jack) - return -ENOMEM; - if (!name) - name = get_jack_default_name(codec, nid, type); - err = snd_jack_new(codec->bus->card, name, type, &jack->jack); - if (err < 0) - return err; - jack->type = type; - jack->jack->private_data = jack; - jack->jack->private_free = hda_free_jack_priv; - return 0; -} -EXPORT_SYMBOL_HDA(snd_hda_input_jack_add); - -void snd_hda_input_jack_report(struct hda_codec *codec, hda_nid_t nid) -{ - struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); - unsigned int pin_ctl; - unsigned int present; - int type; - - if (!jack || !jack->jack) - return; - - present = snd_hda_jack_detect(codec, nid); - type = jack->type; - if (type == (SND_JACK_HEADPHONE | SND_JACK_LINEOUT)) { - pin_ctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - type = (pin_ctl & AC_PINCTL_HP_EN) ? - SND_JACK_HEADPHONE : SND_JACK_LINEOUT; - } - snd_jack_report(jack->jack, present ? type : 0); -} -EXPORT_SYMBOL_HDA(snd_hda_input_jack_report); - -/* free jack instances manually when clearing/reconfiguring */ -static void snd_hda_input_jack_free(struct hda_codec *codec) -{ - if (!codec->bus->shutdown && codec->jacktbl.list) { - struct hda_jack_tbl *jack = codec->jacktbl.list; - int i; - for (i = 0; i < codec->jacktbl.used; i++, jack++) { - if (jack->jack) - snd_device_free(codec->bus->card, jack->jack); - } - } -} -#endif /* CONFIG_SND_HDA_INPUT_JACK */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index ef09716..e1abc07 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -674,24 +674,4 @@ static inline void snd_hda_eld_proc_free(struct hda_codec *codec, #define SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE 80 void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen); -/* - * Input-jack notification support - */ -#ifdef CONFIG_SND_HDA_INPUT_JACK -int snd_hda_input_jack_add(struct hda_codec *codec, hda_nid_t nid, int type, - const char *name); -void snd_hda_input_jack_report(struct hda_codec *codec, hda_nid_t nid); -#else /* CONFIG_SND_HDA_INPUT_JACK */ -static inline int snd_hda_input_jack_add(struct hda_codec *codec, - hda_nid_t nid, int type, - const char *name) -{ - return 0; -} -static inline void snd_hda_input_jack_report(struct hda_codec *codec, - hda_nid_t nid) -{ -} -#endif /* CONFIG_SND_HDA_INPUT_JACK */ - #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ae9c028..bf14a0a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -416,40 +416,6 @@ static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol, &spec->cur_mux[adc_idx]); } -static int conexant_init_jacks(struct hda_codec *codec) -{ -#ifdef CONFIG_SND_HDA_INPUT_JACK - struct conexant_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_init_verbs; i++) { - const struct hda_verb *hv; - - hv = spec->init_verbs[i]; - while (hv->nid) { - int err = 0; - switch (hv->param ^ AC_USRSP_EN) { - case CONEXANT_HP_EVENT: - err = snd_hda_input_jack_add(codec, hv->nid, - SND_JACK_HEADPHONE, NULL); - snd_hda_input_jack_report(codec, hv->nid); - break; - case CXT5051_PORTC_EVENT: - case CONEXANT_MIC_EVENT: - err = snd_hda_input_jack_add(codec, hv->nid, - SND_JACK_MICROPHONE, NULL); - snd_hda_input_jack_report(codec, hv->nid); - break; - } - if (err < 0) - return err; - ++hv; - } - } -#endif /* CONFIG_SND_HDA_INPUT_JACK */ - return 0; -} - static void conexant_set_power(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state) { @@ -1750,7 +1716,6 @@ static void cxt5051_hp_automute(struct hda_codec *codec) static void cxt5051_hp_unsol_event(struct hda_codec *codec, unsigned int res) { - int nid = (res & AC_UNSOL_RES_SUBTAG) >> 20; switch (res >> 26) { case CONEXANT_HP_EVENT: cxt5051_hp_automute(codec); @@ -1762,7 +1727,6 @@ static void cxt5051_hp_unsol_event(struct hda_codec *codec, cxt5051_portc_automic(codec); break; } - snd_hda_input_jack_report(codec, nid); } static const struct snd_kcontrol_new cxt5051_playback_mixers[] = { @@ -1901,8 +1865,6 @@ static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | event); - snd_hda_input_jack_add(codec, nid, SND_JACK_MICROPHONE, NULL); - snd_hda_input_jack_report(codec, nid); } static const struct hda_verb cxt5051_ideapad_init_verbs[] = { @@ -1918,7 +1880,6 @@ static int cxt5051_init(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; conexant_init(codec); - conexant_init_jacks(codec); if (spec->auto_mic & AUTO_MIC_PORTB) cxt5051_init_mic_port(codec, 0x17, CXT5051_PORTB_EVENT); @@ -3450,7 +3411,6 @@ static int detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) hda_nid_t nid = pins[i]; if (!nid || !is_jack_detectable(codec, nid)) break; - snd_hda_input_jack_report(codec, nid); present |= snd_hda_jack_detect(codec, nid); } return present; @@ -3755,8 +3715,6 @@ static void cx_auto_automic(struct hda_codec *codec) static void cx_auto_unsol_event(struct hda_codec *codec, unsigned int res) { - int nid = (res & AC_UNSOL_RES_SUBTAG) >> 20; - switch (snd_hda_jack_get_action(codec, res >> 26)) { case CONEXANT_HP_EVENT: cx_auto_hp_automute(codec); @@ -3766,7 +3724,6 @@ static void cx_auto_unsol_event(struct hda_codec *codec, unsigned int res) break; case CONEXANT_MIC_EVENT: cx_auto_automic(codec); - snd_hda_input_jack_report(codec, nid); break; } snd_hda_jack_report_sync(codec); @@ -4325,6 +4282,7 @@ static int cx_auto_build_input_controls(struct hda_codec *codec) static int cx_auto_build_controls(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; int err; err = cx_auto_build_output_controls(codec); @@ -4333,7 +4291,13 @@ static int cx_auto_build_controls(struct hda_codec *codec) err = cx_auto_build_input_controls(codec); if (err < 0) return err; - return conexant_build_controls(codec); + err = conexant_build_controls(codec); + if (err < 0) + return err; + err = snd_hda_jack_add_kctls(codec, &spec->autocfg); + if (err < 0) + return err; + return 0; } static int cx_auto_search_adcs(struct hda_codec *codec) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bb8cfc6..66dc7e6 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1005,8 +1005,6 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, bool retry) msecs_to_jiffies(300)); } } - - snd_hda_input_jack_report(codec, pin_nid); } static void hdmi_repoll_eld(struct work_struct *work) @@ -1232,21 +1230,15 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx) { - int err; - char hdmi_str[32]; + char hdmi_str[32] = "HDMI/DP"; struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; int pcmdev = spec->pcm_rec[pin_idx].device; - snprintf(hdmi_str, sizeof(hdmi_str), "HDMI/DP,pcm=%d", pcmdev); - - err = snd_hda_input_jack_add(codec, per_pin->pin_nid, - SND_JACK_VIDEOOUT, pcmdev > 0 ? hdmi_str : NULL); - if (err < 0) - return err; + if (pcmdev > 0) + sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev); - hdmi_present_sense(per_pin, false); - return 0; + return snd_hda_jack_add_kctl(codec, per_pin->pin_nid, hdmi_str, 0); } static int generic_hdmi_build_controls(struct hda_codec *codec) @@ -1276,10 +1268,8 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) if (err < 0) return err; - err = snd_hda_jack_add_kctl(codec, per_pin->pin_nid, - "HDMI", pin_idx); - if (err < 0) - return err; + + hdmi_present_sense(per_pin, false); } return 0; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 933c8cf..641c829 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -461,46 +461,6 @@ static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid, } /* - * Jack-reporting via input-jack layer - */ - -/* initialization of jacks; currently checks only a few known pins */ -static int alc_init_jacks(struct hda_codec *codec) -{ -#ifdef CONFIG_SND_HDA_INPUT_JACK - struct alc_spec *spec = codec->spec; - int err; - unsigned int hp_nid = spec->autocfg.hp_pins[0]; - unsigned int mic_nid = spec->ext_mic_pin; - unsigned int dock_nid = spec->dock_mic_pin; - - if (hp_nid) { - err = snd_hda_input_jack_add(codec, hp_nid, - SND_JACK_HEADPHONE, NULL); - if (err < 0) - return err; - snd_hda_input_jack_report(codec, hp_nid); - } - - if (mic_nid) { - err = snd_hda_input_jack_add(codec, mic_nid, - SND_JACK_MICROPHONE, NULL); - if (err < 0) - return err; - snd_hda_input_jack_report(codec, mic_nid); - } - if (dock_nid) { - err = snd_hda_input_jack_add(codec, dock_nid, - SND_JACK_MICROPHONE, NULL); - if (err < 0) - return err; - snd_hda_input_jack_report(codec, dock_nid); - } -#endif /* CONFIG_SND_HDA_INPUT_JACK */ - return 0; -} - -/* * Jack detections for HP auto-mute and mic-switch */ @@ -513,7 +473,6 @@ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) hda_nid_t nid = pins[i]; if (!nid) break; - snd_hda_input_jack_report(codec, nid); present |= snd_hda_jack_detect(codec, nid); } return present; @@ -653,10 +612,6 @@ static void alc_mic_automute(struct hda_codec *codec) alc_mux_select(codec, 0, spec->dock_mic_idx, false); else alc_mux_select(codec, 0, spec->int_mic_idx, false); - - snd_hda_input_jack_report(codec, pins[spec->ext_mic_idx]); - if (spec->dock_mic_idx >= 0) - snd_hda_input_jack_report(codec, pins[spec->dock_mic_idx]); } /* unsolicited event for HP jack sensing */ @@ -4686,7 +4641,6 @@ static int patch_alc882(struct hda_codec *codec) if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; - alc_init_jacks(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc882_loopbacks; @@ -4866,7 +4820,6 @@ static int patch_alc262(struct hda_codec *codec) spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; - alc_init_jacks(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc262_loopbacks; @@ -4979,8 +4932,6 @@ static int patch_alc268(struct hda_codec *codec) spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; - alc_init_jacks(codec); - return 0; error: @@ -5538,7 +5489,6 @@ static int patch_alc269(struct hda_codec *codec) spec->init_hook = alc_auto_init_std; spec->shutup = alc269_shutup; - alc_init_jacks(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc269_loopbacks; @@ -6153,8 +6103,6 @@ static int patch_alc662(struct hda_codec *codec) spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; - alc_init_jacks(codec); - #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc662_loopbacks; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0988dc4..2d4156c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1084,13 +1084,10 @@ static const char * const slave_sws[] = { }; static void stac92xx_free_kctls(struct hda_codec *codec); -static int stac92xx_add_jack(struct hda_codec *codec, hda_nid_t nid, int type); static int stac92xx_build_controls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - hda_nid_t nid; int err; int i; @@ -1176,32 +1173,6 @@ static int stac92xx_build_controls(struct hda_codec *codec) stac92xx_free_kctls(codec); /* no longer needed */ - /* create jack input elements */ - if (spec->hp_detect) { - for (i = 0; i < cfg->hp_outs; i++) { - int type = SND_JACK_HEADPHONE; - nid = cfg->hp_pins[i]; - /* jack detection */ - if (cfg->hp_outs == i) - type |= SND_JACK_LINEOUT; - err = stac92xx_add_jack(codec, nid, type); - if (err < 0) - return err; - } - } - for (i = 0; i < cfg->line_outs; i++) { - err = stac92xx_add_jack(codec, cfg->line_out_pins[i], - SND_JACK_LINEOUT); - if (err < 0) - return err; - } - for (i = 0; i < cfg->num_inputs; i++) { - nid = cfg->inputs[i].pin; - err = stac92xx_add_jack(codec, nid, SND_JACK_MICROPHONE); - if (err < 0) - return err; - } - err = snd_hda_jack_add_kctls(codec, &spec->autocfg); if (err < 0) return err; @@ -4158,22 +4129,6 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */ } -static int stac92xx_add_jack(struct hda_codec *codec, - hda_nid_t nid, int type) -{ -#ifdef CONFIG_SND_HDA_INPUT_JACK - int def_conf = snd_hda_codec_get_pincfg(codec, nid); - int connectivity = get_defcfg_connect(def_conf); - - if (connectivity && connectivity != AC_JACK_PORT_FIXED) - return 0; - - return snd_hda_input_jack_add(codec, nid, type, NULL); -#else - return 0; -#endif /* CONFIG_SND_HDA_INPUT_JACK */ -} - static int stac_add_event(struct hda_codec *codec, hda_nid_t nid, unsigned char type, int data) { @@ -4778,7 +4733,6 @@ static void handle_unsol_event(struct hda_codec *codec, case STAC_PWR_EVENT: if (spec->num_pwrs > 0) stac92xx_pin_sense(codec, event->nid); - snd_hda_input_jack_report(codec, event->nid); switch (codec->subsystem_id) { case 0x103c308f: -- cgit v1.1 From 705978516fe4a5e0b5726e2ea860c1bfc6909472 Mon Sep 17 00:00:00 2001 From: David Dillow Date: Thu, 1 Dec 2011 23:26:57 -0500 Subject: ALSA: sis7019 - convert to dev_*() logging Signed-off-by: David Dillow Signed-off-by: Takashi Iwai --- sound/pci/sis7019.c | 29 ++++++++++++++--------------- 1 file changed, 14 insertions(+), 15 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 28dfafb..7331b2d 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -983,7 +983,7 @@ timeout: mutex_unlock(&sis->ac97_mutex); if (!count) { - printk(KERN_ERR "sis7019: ac97 codec %d timeout cmd 0x%08x\n", + dev_err(&sis->pci->dev, "ac97 codec %d timeout cmd 0x%08x\n", codec, cmd); } @@ -1142,13 +1142,13 @@ static int sis_chip_init(struct sis7019 *sis) /* All done, check for errors. */ if (!sis->codecs_present) { - printk(KERN_ERR "sis7019: could not find any codecs\n"); + dev_err(&sis->pci->dev, "could not find any codecs\n"); return -EIO; } if (sis->codecs_present != codecs) { - printk(KERN_WARNING "sis7019: missing codecs, found %0x, expected %0x\n", - sis->codecs_present, codecs); + dev_warn(&sis->pci->dev, "missing codecs, found %0x, expected %0x\n", + sis->codecs_present, codecs); } /* Let the hardware know that the audio driver is alive, @@ -1256,18 +1256,18 @@ static int sis_resume(struct pci_dev *pci) pci_restore_state(pci); if (pci_enable_device(pci) < 0) { - printk(KERN_ERR "sis7019: unable to re-enable device\n"); + dev_err(&pci->dev, "unable to re-enable device\n"); goto error; } if (sis_chip_init(sis)) { - printk(KERN_ERR "sis7019: unable to re-init controller\n"); + dev_err(&pci->dev, "unable to re-init controller\n"); goto error; } if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, sis)) { - printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq); + dev_err(&pci->dev, "unable to regain IRQ %d\n", pci->irq); goto error; } @@ -1335,8 +1335,7 @@ static int __devinit sis_chip_create(struct snd_card *card, goto error_out; if (pci_set_dma_mask(pci, DMA_BIT_MASK(30)) < 0) { - printk(KERN_ERR "sis7019: architecture does not support " - "30-bit PCI busmaster DMA"); + dev_err(&pci->dev, "architecture does not support 30-bit PCI busmaster DMA"); goto error_out_enabled; } @@ -1350,20 +1349,20 @@ static int __devinit sis_chip_create(struct snd_card *card, rc = pci_request_regions(pci, "SiS7019"); if (rc) { - printk(KERN_ERR "sis7019: unable request regions\n"); + dev_err(&pci->dev, "unable request regions\n"); goto error_out_enabled; } rc = -EIO; sis->ioaddr = ioremap_nocache(pci_resource_start(pci, 1), 0x4000); if (!sis->ioaddr) { - printk(KERN_ERR "sis7019: unable to remap MMIO, aborting\n"); + dev_err(&pci->dev, "unable to remap MMIO, aborting\n"); goto error_out_cleanup; } rc = sis_alloc_suspend(sis); if (rc < 0) { - printk(KERN_ERR "sis7019: unable to allocate state storage\n"); + dev_err(&pci->dev, "unable to allocate state storage\n"); goto error_out_cleanup; } @@ -1371,9 +1370,9 @@ static int __devinit sis_chip_create(struct snd_card *card, if (rc) goto error_out_cleanup; - if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, - KBUILD_MODNAME, sis)) { - printk(KERN_ERR "unable to allocate irq %d\n", sis->irq); + if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, + sis)) { + dev_err(&pci->dev, "unable to allocate irq %d\n", sis->irq); goto error_out_cleanup; } -- cgit v1.1 From 1d5d37f408e530ce1eab1deb66d2331535665ec7 Mon Sep 17 00:00:00 2001 From: Thomas Meyer Date: Tue, 29 Nov 2011 22:08:00 +0100 Subject: ALSA: ctxf: Use kcalloc instead of kzalloc to allocate array The advantage of kcalloc is, that will prevent integer overflows which could result from the multiplication of number of elements and size and it is also a bit nicer to read. The semantic patch that makes this change is available in https://lkml.org/lkml/2011/11/25/107 Signed-off-by: Thomas Meyer Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctsrc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index e134b3a..6e77e86 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -437,7 +437,7 @@ get_src_rsc(struct src_mgr *mgr, const struct src_desc *desc, struct src **rsrc) /* Allocate mem for master src resource */ if (MEMRD == desc->mode) - src = kzalloc(sizeof(*src)*desc->multi, GFP_KERNEL); + src = kcalloc(desc->multi, sizeof(*src), GFP_KERNEL); else src = kzalloc(sizeof(*src), GFP_KERNEL); -- cgit v1.1 From 6d2d4313690f2f81a9a54c6a0c8ae645c4598063 Mon Sep 17 00:00:00 2001 From: Thomas Meyer Date: Tue, 29 Nov 2011 22:08:00 +0100 Subject: ALSA: asihp: Use kcalloc instead of kzalloc to allocate array The advantage of kcalloc is, that will prevent integer overflows which could result from the multiplication of number of elements and size and it is also a bit nicer to read. The semantic patch that makes this change is available in https://lkml.org/lkml/2011/11/25/107 Signed-off-by: Thomas Meyer Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpicmn.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c index bd47521..44c7eb4 100644 --- a/sound/pci/asihpi/hpicmn.c +++ b/sound/pci/asihpi/hpicmn.c @@ -631,7 +631,7 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32 control_count, if (!p_cache) return NULL; - p_cache->p_info = kzalloc(sizeof(*p_cache->p_info) * control_count, + p_cache->p_info = kcalloc(control_count, sizeof(*p_cache->p_info), GFP_KERNEL); if (!p_cache->p_info) { kfree(p_cache); -- cgit v1.1 From fde48a1f808e2bb6aaad5709d2470d814a157c86 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 9 Dec 2011 18:27:42 +0800 Subject: ALSA: HDA: Realtek: Take vmaster dac from multiout dac list With the auto-parser we can choose the dac nid for vmaster from the DACs we already know, instead of hard-coding it. This is more future-proof and was actually wrong on one machine. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 37 +++++++++++++++---------------------- 1 file changed, 15 insertions(+), 22 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8a74c1e..690f2a2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3039,6 +3039,8 @@ static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs, static int alc_auto_fill_multi_ios(struct hda_codec *codec, unsigned int location, int offset); +static hda_nid_t alc_look_for_out_vol_nid(struct hda_codec *codec, + hda_nid_t pin, hda_nid_t dac); /* fill in the dac_nids table from the parsed pin configuration */ static int alc_auto_fill_dac_nids(struct hda_codec *codec) @@ -3153,6 +3155,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) } } + if (cfg->line_out_pins[0]) + spec->vmaster_nid = + alc_look_for_out_vol_nid(codec, cfg->line_out_pins[0], + spec->multiout.dac_nids[0]); return 0; } @@ -4175,8 +4181,10 @@ static int patch_alc880(struct hda_codec *codec) #endif } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { + spec->vmaster_nid = 0x0c; setup_preset(codec, &alc880_presets[board_config]); + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4196,8 +4204,6 @@ static int patch_alc880(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4304,8 +4310,10 @@ static int patch_alc260(struct hda_codec *codec) #endif } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { setup_preset(codec, &alc260_presets[board_config]); + spec->vmaster_nid = 0x08; + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4325,8 +4333,6 @@ static int patch_alc260(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x08; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4698,8 +4704,10 @@ static int patch_alc882(struct hda_codec *codec) goto error; } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { setup_preset(codec, &alc882_presets[board_config]); + spec->vmaster_nid = 0x0c; + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4719,8 +4727,6 @@ static int patch_alc882(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4899,8 +4905,6 @@ static int patch_alc262(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -5012,8 +5016,6 @@ static int patch_alc268(struct hda_codec *codec) if (!spec->no_analog && !spec->cap_mixer) set_capture_mixer(codec); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -5568,8 +5570,6 @@ static int patch_alc269(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; #ifdef CONFIG_PM codec->patch_ops.resume = alc269_resume; @@ -5674,8 +5674,6 @@ static int patch_alc861(struct hda_codec *codec) set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); } - spec->vmaster_nid = 0x03; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -5800,8 +5798,6 @@ static int patch_alc861vd(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } - spec->vmaster_nid = 0x02; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -6184,7 +6180,6 @@ static int patch_alc662(struct hda_codec *codec) break; } } - spec->vmaster_nid = 0x02; alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -6240,8 +6235,6 @@ static int patch_alc680(struct hda_codec *codec) if (!spec->no_analog && !spec->cap_mixer) set_capture_mixer(codec); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; -- cgit v1.1 From 6a557c94737a261f1b78767c7c41406206dd25c8 Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Mon, 12 Dec 2011 17:35:13 -0600 Subject: ALSA: hda - GPIO to control mute LED may be enabled on HP systems with no such HW This may lead to problems (like loss of sound) as GPIO pin may be used for different function (SPDIF OUT, EAPD etc) on those systems. This patch disables default mute LED GPIO configuration on all new codecs as all new HP systems are expected to provide explicit mute LED configuration in SMBIOS. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index eeb25d52..85740e0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4903,7 +4903,7 @@ static void set_hp_led_gpio(struct hda_codec *codec) * Need more information on whether it is true across the entire series. * -- kunal */ -static int find_mute_led_gpio(struct hda_codec *codec, int default_polarity) +static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) { struct sigmatel_spec *spec = codec->spec; const struct dmi_device *dev = NULL; @@ -4933,9 +4933,11 @@ static int find_mute_led_gpio(struct hda_codec *codec, int default_polarity) /* * Fallback case - if we don't find the DMI strings, - * we statically set the GPIO - if not a B-series system. + * we statically set the GPIO - if not a B-series system + * and default polarity is provided */ - if (!hp_blike_system(codec->subsystem_id)) { + if (!hp_blike_system(codec->subsystem_id) && + (default_polarity == 0 || default_polarity == 1)) { set_hp_led_gpio(codec); spec->gpio_led_polarity = default_polarity; return 1; @@ -5645,7 +5647,7 @@ again: codec->patch_ops = stac92xx_patch_ops; - if (find_mute_led_gpio(codec, 0)) + if (find_mute_led_cfg(codec, -1/*no default cfg*/)) snd_printd("mute LED gpio %d polarity %d\n", spec->gpio_led, spec->gpio_led_polarity); @@ -5958,7 +5960,7 @@ again: } } - if (find_mute_led_gpio(codec, 1)) + if (find_mute_led_cfg(codec, 1)) snd_printd("mute LED gpio %d polarity %d\n", spec->gpio_led, spec->gpio_led_polarity); -- cgit v1.1 From 97e287626a219e3754a54ac654691c608b78341d Mon Sep 17 00:00:00 2001 From: Gustavo Maciel Dias Vieira Date: Sun, 4 Dec 2011 15:14:10 -0200 Subject: ALSA: hda: remove unused quirk for inverted mute led Commit b99a776d0b17ae0f3a54e86009887a00ac4889d0 removed all effects of the STAC92HD83* model quirk "hp". However, it left the model selection and documentation behind, confusing users with inverted mute leds. Completely remove this quirk and its documentation. Signed-off-by: Gustavo Maciel Dias Vieira Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 85740e0..65f5179 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -96,7 +96,6 @@ enum { STAC_92HD83XXX_PWR_REF, STAC_DELL_S14, STAC_DELL_VOSTRO_3500, - STAC_92HD83XXX_HP, STAC_92HD83XXX_HP_cNB11_INTQUAD, STAC_HP_DV7_4000, STAC_92HD83XXX_MODELS @@ -1692,7 +1691,6 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_PWR_REF] = "mic-ref", [STAC_DELL_S14] = "dell-s14", [STAC_DELL_VOSTRO_3500] = "dell-vostro-3500", - [STAC_92HD83XXX_HP] = "hp", [STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad", [STAC_HP_DV7_4000] = "hp-dv7-4000", }; @@ -1707,8 +1705,6 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "unknown Dell", STAC_DELL_S14), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x1028, "Dell Vostro 3500", STAC_DELL_VOSTRO_3500), - SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x3600, - "HP", STAC_92HD83XXX_HP), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1656, "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1657, -- cgit v1.1 From 1815b34a626742733f846ddd266a89bd2fcea28f Mon Sep 17 00:00:00 2001 From: Andiry Xu Date: Wed, 14 Dec 2011 16:10:27 +0800 Subject: ALSA: HDA: Add support for new AMD products This patch adds HDMI audio support for new AMD products. As HW default disable snoop, force non-snoop mode in HD audio driver. Signed-off-by: Andiry Xu Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d1582dd..9f7c901a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -487,6 +487,7 @@ enum { AZX_DRIVER_SCH, AZX_DRIVER_ATI, AZX_DRIVER_ATIHDMI, + AZX_DRIVER_ATIHDMI_NS, AZX_DRIVER_VIA, AZX_DRIVER_SIS, AZX_DRIVER_ULI, @@ -533,6 +534,7 @@ static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_SCH] = "HDA Intel MID", [AZX_DRIVER_ATI] = "HDA ATI SB", [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", + [AZX_DRIVER_ATIHDMI_NS] = "HDA ATI HDMI", [AZX_DRIVER_VIA] = "HDA VIA VT82xx", [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", @@ -2678,6 +2680,8 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->single_cmd = single_cmd; chip->snoop = hda_snoop; + if (chip->driver_type == AZX_DRIVER_ATIHDMI_NS) + chip->snoop = 0; if (bdl_pos_adj[dev] < 0) { switch (chip->driver_type) { @@ -2776,6 +2780,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->capture_streams = ULI_NUM_CAPTURE; break; case AZX_DRIVER_ATIHDMI: + case AZX_DRIVER_ATIHDMI_NS: chip->playback_streams = ATIHDMI_NUM_PLAYBACK; chip->capture_streams = ATIHDMI_NUM_CAPTURE; break; @@ -3037,6 +3042,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0xaa48), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, + { PCI_DEVICE(0x1002, 0x9902), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, + { PCI_DEVICE(0x1002, 0xaaa0), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, + { PCI_DEVICE(0x1002, 0xaaa8), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, + { PCI_DEVICE(0x1002, 0xaab0), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA }, -- cgit v1.1 From a1585d769731323a792277f15b7a3ee2ae36b698 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 14 Dec 2011 09:27:04 +0100 Subject: ALSA: hda - Check non-snoop in a single place Merge the checks for VIA and ATI-HDMI into a single place for better code-flow management. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 42 ++++++++++++++++++++++++++++++------------ 1 file changed, 30 insertions(+), 12 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9f7c901a..8d17963 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1153,16 +1153,6 @@ static void update_pci_byte(struct pci_dev *pci, unsigned int reg, static void azx_init_pci(struct azx *chip) { - /* force to non-snoop mode for a new VIA controller when BIOS is set */ - if (chip->snoop && chip->driver_type == AZX_DRIVER_VIA) { - u8 snoop; - pci_read_config_byte(chip->pci, 0x42, &snoop); - if (!(snoop & 0x80) && chip->pci->revision == 0x30) { - chip->snoop = 0; - snd_printdd(SFX "Force to non-snoop mode\n"); - } - } - /* Clear bits 0-2 of PCI register TCSEL (at offset 0x44) * TCSEL == Traffic Class Select Register, which sets PCI express QOS * Ensuring these bits are 0 clears playback static on some HD Audio @@ -2634,6 +2624,35 @@ static void __devinit check_msi(struct azx *chip) } } +/* check the snoop mode availability */ +static void __devinit azx_check_snoop_available(struct azx *chip) +{ + bool snoop = chip->snoop; + + switch (chip->driver_type) { + case AZX_DRIVER_VIA: + /* force to non-snoop mode for a new VIA controller + * when BIOS is set + */ + if (snoop) { + u8 val; + pci_read_config_byte(chip->pci, 0x42, &val); + if (!(val & 0x80) && chip->pci->revision == 0x30) + snoop = false; + } + break; + case AZX_DRIVER_ATIHDMI_NS: + /* new ATI HDMI requires non-snoop */ + snoop = false; + break; + } + + if (snoop != chip->snoop) { + snd_printk(KERN_INFO SFX "Force to %s mode\n", + snoop ? "snoop" : "non-snoop"); + chip->snoop = snoop; + } +} /* * constructor @@ -2680,8 +2699,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->single_cmd = single_cmd; chip->snoop = hda_snoop; - if (chip->driver_type == AZX_DRIVER_ATIHDMI_NS) - chip->snoop = 0; + azx_check_snoop_available(chip); if (bdl_pos_adj[dev] < 0) { switch (chip->driver_type) { -- cgit v1.1 From a67ff6a54095e27093ea501fb143fefe51a536c2 Mon Sep 17 00:00:00 2001 From: Rusty Russell Date: Thu, 15 Dec 2011 13:49:36 +1030 Subject: ALSA: module_param: make bool parameters really bool module_param(bool) used to counter-intuitively take an int. In fddd5201 (mid-2009) we allowed bool or int/unsigned int using a messy trick. It's time to remove the int/unsigned int option. For this version it'll simply give a warning, but it'll break next kernel version. Signed-off-by: Rusty Russell Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 2 +- sound/pci/ad1889.c | 2 +- sound/pci/ali5451/ali5451.c | 4 ++-- sound/pci/als4000.c | 2 +- sound/pci/asihpi/asihpi.c | 4 ++-- sound/pci/atiixp.c | 4 ++-- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au88x0.c | 2 +- sound/pci/aw2/aw2-alsa.c | 2 +- sound/pci/azt3328.c | 2 +- sound/pci/bt87x.c | 4 ++-- sound/pci/ca0106/ca0106_main.c | 2 +- sound/pci/cmipci.c | 4 ++-- sound/pci/cs4281.c | 4 ++-- sound/pci/cs46xx/cs46xx.c | 8 ++++---- sound/pci/cs5535audio/cs5535audio.c | 2 +- sound/pci/ctxfi/cttimer.c | 4 ++-- sound/pci/ctxfi/xfi.c | 2 +- sound/pci/echoaudio/echoaudio.c | 2 +- sound/pci/emu10k1/emu10k1.c | 4 ++-- sound/pci/emu10k1/emu10k1x.c | 2 +- sound/pci/ens1370.c | 4 ++-- sound/pci/es1938.c | 2 +- sound/pci/es1968.c | 4 ++-- sound/pci/fm801.c | 2 +- sound/pci/hda/hda_intel.c | 8 ++++---- sound/pci/ice1712/ice1712.c | 4 ++-- sound/pci/ice1712/ice1724.c | 2 +- sound/pci/intel8x0.c | 6 +++--- sound/pci/intel8x0m.c | 2 +- sound/pci/korg1212/korg1212.c | 2 +- sound/pci/lola/lola.c | 2 +- sound/pci/lx6464es/lx6464es.c | 2 +- sound/pci/maestro3.c | 4 ++-- sound/pci/mixart/mixart.c | 2 +- sound/pci/nm256/nm256.c | 12 ++++++------ sound/pci/oxygen/oxygen.c | 2 +- sound/pci/oxygen/virtuoso.c | 2 +- sound/pci/pcxhr/pcxhr.c | 4 ++-- sound/pci/riptide/riptide.c | 2 +- sound/pci/rme32.c | 4 ++-- sound/pci/rme96.c | 2 +- sound/pci/rme9652/hdsp.c | 2 +- sound/pci/rme9652/hdspm.c | 2 +- sound/pci/rme9652/rme9652.c | 4 ++-- sound/pci/sis7019.c | 2 +- sound/pci/sonicvibes.c | 6 +++--- sound/pci/trident/trident.c | 2 +- sound/pci/via82xx.c | 4 ++-- sound/pci/via82xx_modem.c | 2 +- sound/pci/vx222/vx222.c | 4 ++-- sound/pci/ymfpci/ymfpci.c | 4 ++-- 52 files changed, 85 insertions(+), 85 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index fac51ee..9473fca 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -42,7 +42,7 @@ MODULE_AUTHOR("Jaroslav Kysela "); MODULE_DESCRIPTION("Universal interface for Audio Codec '97"); MODULE_LICENSE("GPL"); -static int enable_loopback; +static bool enable_loopback; module_param(enable_loopback, bool, 0444); MODULE_PARM_DESC(enable_loopback, "Enable AC97 ADC/DAC Loopback Control"); diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 6e31118..9d91d61 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -66,7 +66,7 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for the AD1889 soundcard."); -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable AD1889 soundcard."); diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index ef85ac5..bdd6164 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -48,7 +48,7 @@ MODULE_SUPPORTED_DEVICE("{{ALI,M5451,pci},{ALI,M5451}}"); static int index = SNDRV_DEFAULT_IDX1; /* Index */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int pcm_channels = 32; -static int spdif; +static bool spdif; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for ALI M5451 PCI Audio."); @@ -60,7 +60,7 @@ module_param(spdif, bool, 0444); MODULE_PARM_DESC(spdif, "Support SPDIF I/O"); /* just for backward compatibility */ -static int enable; +static bool enable; module_param(enable, bool, 0444); diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 28ef40e..3269b80 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -90,7 +90,7 @@ MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS4000}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ #ifdef SUPPORT_JOYSTICK static int joystick_port[SNDRV_CARDS]; #endif diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index f4b9e2b..e9de799 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -63,8 +63,8 @@ MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -static int enable_hpi_hwdep = 1; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable_hpi_hwdep = 1; module_param_array(index, int, NULL, S_IRUGO); MODULE_PARM_DESC(index, "ALSA index value for AudioScience soundcard."); diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 15e4e5e..590682f 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -43,7 +43,7 @@ static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int ac97_clock = 48000; static char *ac97_quirk; -static int spdif_aclink = 1; +static bool spdif_aclink = 1; static int ac97_codec = -1; module_param(index, int, 0444); @@ -60,7 +60,7 @@ module_param(spdif_aclink, bool, 0444); MODULE_PARM_DESC(spdif_aclink, "S/PDIF over AC-link."); /* just for backward compatibility */ -static int enable; +static bool enable; module_param(enable, bool, 0444); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 57bf8f4..524d35f 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -51,7 +51,7 @@ module_param(ac97_clock, int, 0444); MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (default 48000Hz)."); /* just for backward compatibility */ -static int enable; +static bool enable; module_param(enable, bool, 0444); diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index dc326be..762bb10 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -26,7 +26,7 @@ // module parameters (see "Module Parameters") static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static int pcifix[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 255 }; module_param_array(index, int, NULL, 0444); diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 7a58115..1c523193 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -153,7 +153,7 @@ static int snd_aw2_control_switch_capture_put(struct snd_kcontrol *kcontrol, ********************************/ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Audiowerk2 soundcard."); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index bc1e683..95ffa6a 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -301,7 +301,7 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for AZF3328 soundcard."); -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable AZF3328 soundcard."); diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index c1c2d0c..62d6163 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -42,9 +42,9 @@ MODULE_SUPPORTED_DEVICE("{{Brooktree,Bt878}," static int index[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -2}; /* Exclude the first card */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ static int digital_rate[SNDRV_CARDS]; /* digital input rate */ -static int load_all; /* allow to load the non-whitelisted cards */ +static bool load_all; /* allow to load the non-whitelisted cards */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Bt87x soundcard"); diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index fe99fde..08d6ebf 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -156,7 +156,7 @@ MODULE_SUPPORTED_DEVICE("{{Creative,SB CA0106 chip}}"); // module parameters (see "Module Parameters") static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */ module_param_array(index, int, NULL, 0444); diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 954c993..19b0626 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -54,10 +54,10 @@ MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8738}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ static long mpu_port[SNDRV_CARDS]; static long fm_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)]=1}; -static int soft_ac3[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)]=1}; +static bool soft_ac3[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)]=1}; #ifdef SUPPORT_JOYSTICK static int joystick_port[SNDRV_CARDS]; #endif diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index a6c6c5c..a9f368f 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -44,8 +44,8 @@ MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,CS4281}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ -static int dual_codec[SNDRV_CARDS]; /* dual codec */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ +static bool dual_codec[SNDRV_CARDS]; /* dual codec */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for CS4281 soundcard."); diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index a4ecb40..819d79d 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -46,10 +46,10 @@ MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,Sound Fusion (CS4280)}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int external_amp[SNDRV_CARDS]; -static int thinkpad[SNDRV_CARDS]; -static int mmap_valid[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool external_amp[SNDRV_CARDS]; +static bool thinkpad[SNDRV_CARDS]; +static bool mmap_valid[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the CS46xx soundcard."); diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index b8959d2..a2fb217 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -57,7 +57,7 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for " DRIVER_NAME); diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index 93b0aed..03fb909 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -15,8 +15,8 @@ #include "cthardware.h" #include "cttimer.h" -static int use_system_timer; -MODULE_PARM_DESC(use_system_timer, "Foce to use system-timer"); +static bool use_system_timer; +MODULE_PARM_DESC(use_system_timer, "Force to use system-timer"); module_param(use_system_timer, bool, S_IRUGO); struct ct_timer_ops { diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index 33931ef..15d95d2 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -32,7 +32,7 @@ module_param(multiple, uint, S_IRUGO); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static unsigned int subsystem[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 9fd694c..595c11f 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -26,7 +26,7 @@ MODULE_DEVICE_TABLE(pci, snd_echo_ids); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for " ECHOCARD_NAME " soundcard."); diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index eaa198e..790c65d 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -44,13 +44,13 @@ MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB Live!/PCI512/E-mu APS}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ static int extin[SNDRV_CARDS]; static int extout[SNDRV_CARDS]; static int seq_ports[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4}; static int max_synth_voices[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 64}; static int max_buffer_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 128}; -static int enable_ir[SNDRV_CARDS]; +static bool enable_ir[SNDRV_CARDS]; static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */ static uint delay_pcm_irq[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 2228be9..47a651c 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -50,7 +50,7 @@ MODULE_SUPPORTED_DEVICE("{{Dell Creative Labs,SB Live!}"); // module parameters (see "Module Parameters") static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the EMU10K1X soundcard."); diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index d085ad0..47a245e 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -83,12 +83,12 @@ MODULE_SUPPORTED_DEVICE("{{Ensoniq,AudioPCI ES1371/73}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ #ifdef SUPPORT_JOYSTICK #ifdef CHIP1371 static int joystick_port[SNDRV_CARDS]; #else -static int joystick[SNDRV_CARDS]; +static bool joystick[SNDRV_CARDS]; #endif #endif #ifdef CHIP1371 diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 04cc21f..53eb76b 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -79,7 +79,7 @@ MODULE_SUPPORTED_DEVICE("{{ESS,ES1938}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for ESS Solo-1 soundcard."); diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 297a151..cb557c6 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -132,7 +132,7 @@ MODULE_SUPPORTED_DEVICE("{{ESS,Maestro 2e}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 1-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ static int total_bufsize[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1024 }; static int pcm_substreams_p[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4 }; static int pcm_substreams_c[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1 }; @@ -140,7 +140,7 @@ static int clock[SNDRV_CARDS]; static int use_pm[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; static int enable_mpu[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; #ifdef SUPPORT_JOYSTICK -static int joystick[SNDRV_CARDS]; +static bool joystick[SNDRV_CARDS]; #endif module_param_array(index, int, NULL, 0444); diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index ec05ef5..9597ef1 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -48,7 +48,7 @@ MODULE_SUPPORTED_DEVICE("{{ForteMedia,FM801}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ /* * Enable TEA575x tuner * 1 = MediaForte 256-PCS diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7d98240..06fe2c5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -58,13 +58,13 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static char *model[SNDRV_CARDS]; static int position_fix[SNDRV_CARDS]; static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_only[SNDRV_CARDS]; -static int single_cmd; +static bool single_cmd; static int enable_msi = -1; #ifdef CONFIG_SND_HDA_PATCH_LOADER static char *patch[SNDRV_CARDS]; @@ -116,12 +116,12 @@ MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " * this may give more power-saving, but will take longer time to * wake up. */ -static int power_save_controller = 1; +static bool power_save_controller = 1; module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #endif -static int align_buffer_size = 1; +static bool align_buffer_size = 1; module_param(align_buffer_size, bool, 0644); MODULE_PARM_DESC(align_buffer_size, "Force buffer and period sizes to be multiple of 128 bytes."); diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 44446f2..132a86e 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -84,9 +84,9 @@ MODULE_SUPPORTED_DEVICE("{" static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ static char *model[SNDRV_CARDS]; -static int omni[SNDRV_CARDS]; /* Delta44 & 66 Omni I/O support */ +static bool omni[SNDRV_CARDS]; /* Delta44 & 66 Omni I/O support */ static int cs8427_timeout[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 500}; /* CS8427 S/PDIF transceiver reset timeout value in msec */ static int dxr_enable[SNDRV_CARDS]; /* DXR enable for DMX6FIRE */ diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 4353e76..4dc5124 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -80,7 +80,7 @@ MODULE_SUPPORTED_DEVICE("{" static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ static char *model[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 11718b49..40b181b 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -79,9 +79,9 @@ static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int ac97_clock; static char *ac97_quirk; -static int buggy_semaphore; +static bool buggy_semaphore; static int buggy_irq = -1; /* auto-check */ -static int xbox; +static bool xbox; static int spdif_aclink = -1; static int inside_vm = -1; @@ -105,7 +105,7 @@ module_param(inside_vm, bool, 0444); MODULE_PARM_DESC(inside_vm, "KVM/Parallels optimization."); /* just for backward compatibility */ -static int enable; +static bool enable; module_param(enable, bool, 0444); static int joystick; module_param(joystick, int, 0444); diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 0f7041e..d689913 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -68,7 +68,7 @@ module_param(ac97_clock, int, 0444); MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (0 = auto-detect)."); /* just for backward compatibility */ -static int enable; +static bool enable; module_param(enable, bool, 0444); /* diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 841864b..8fea45a 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -408,7 +408,7 @@ MODULE_FIRMWARE("korg/k1212.dsp"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Korg 1212 soundcard."); diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 924168e..3759827 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -35,7 +35,7 @@ /* Standard options */ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Digigram Lola driver."); diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 04ae84b2..d94c0c2 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -42,7 +42,7 @@ MODULE_SUPPORTED_DEVICE("{digigram lx6464es{}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Digigram LX6464ES interface."); diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 863c8bd..78229b0 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -64,8 +64,8 @@ MODULE_FIRMWARE("ess/maestro3_assp_minisrc.fw"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* all enabled */ -static int external_amp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* all enabled */ +static bool external_amp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; static int amp_gpio[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -1}; module_param_array(index, int, NULL, 0444); diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index a0bd1d9..487837c 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -49,7 +49,7 @@ MODULE_SUPPORTED_DEVICE("{{Digigram," CARD_NAME "}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Digigram " CARD_NAME " soundcard."); diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index c6c45d9..ade2c64 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -57,12 +57,12 @@ static int index = SNDRV_DEFAULT_IDX1; /* Index */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int playback_bufsize = 16; static int capture_bufsize = 16; -static int force_ac97; /* disabled as default */ +static bool force_ac97; /* disabled as default */ static int buffer_top; /* not specified */ -static int use_cache; /* disabled */ -static int vaio_hack; /* disabled */ -static int reset_workaround; -static int reset_workaround_2; +static bool use_cache; /* disabled */ +static bool vaio_hack; /* disabled */ +static bool reset_workaround; +static bool reset_workaround_2; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); @@ -86,7 +86,7 @@ module_param(reset_workaround_2, bool, 0444); MODULE_PARM_DESC(reset_workaround_2, "Enable extended AC97 RESET workaround for some other laptops."); /* just for backward compatibility */ -static int enable; +static bool enable; module_param(enable, bool, 0444); diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 5f3a13d..eab663e 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -74,7 +74,7 @@ MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8786}" static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "card index"); diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 4149a0c..3fdee49 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -32,7 +32,7 @@ MODULE_SUPPORTED_DEVICE("{{Asus,AV66},{Asus,AV100},{Asus,AV200}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "card index"); diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 56a5265..fd1809a 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -52,8 +52,8 @@ MODULE_SUPPORTED_DEVICE("{{Digigram," DRIVER_NAME "}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ -static int mono[SNDRV_CARDS]; /* capture mono only */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ +static bool mono[SNDRV_CARDS]; /* capture mono only */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Digigram " DRIVER_NAME " soundcard"); diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index dcbedd3..0481d94 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -122,7 +122,7 @@ MODULE_FIRMWARE("riptide.hex"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; #ifdef SUPPORT_JOYSTICK static int joystick_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x200 }; diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 21bcb47..b4819d5 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -89,8 +89,8 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int fullduplex[SNDRV_CARDS]; // = {[0 ... (SNDRV_CARDS - 1)] = 1}; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool fullduplex[SNDRV_CARDS]; // = {[0 ... (SNDRV_CARDS - 1)] = 1}; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for RME Digi32 soundcard."); diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 4585c97..ba89415 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -53,7 +53,7 @@ MODULE_SUPPORTED_DEVICE("{{RME,Digi96}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for RME Digi96 soundcard."); diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index f2a3758..0111203 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -45,7 +45,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for RME Hammerfall DSP interface."); diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 19ee220..d623451 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -61,7 +61,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for RME HDSPM interface."); diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 732c5e8..b737d16 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -38,8 +38,8 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int precise_ptr[SNDRV_CARDS]; /* Enable precise pointer */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool precise_ptr[SNDRV_CARDS]; /* Enable precise pointer */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for RME Digi9652 (Hammerfall) soundcard."); diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 7331b2d..ff500a8 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -40,7 +40,7 @@ MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}"); static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ -static int enable = 1; +static bool enable = 1; static int codecs = 1; module_param(index, int, 0444); diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 31b6ad3..54cc802 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -52,9 +52,9 @@ MODULE_SUPPORTED_DEVICE("{{S3,SonicVibes PCI}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int reverb[SNDRV_CARDS]; -static int mge[SNDRV_CARDS]; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool reverb[SNDRV_CARDS]; +static bool mge[SNDRV_CARDS]; static unsigned int dmaio = 0x7a00; /* DDMA i/o address */ module_param_array(index, int, NULL, 0444); diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index deb04b9..5f1def7 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -47,7 +47,7 @@ MODULE_SUPPORTED_DEVICE("{{Trident,4DWave DX}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ static int pcm_channels[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 32}; static int wavetable_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8192}; diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index ae98d56..7563040 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -80,7 +80,7 @@ static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static long mpu_port; #ifdef SUPPORT_JOYSTICK -static int joystick; +static bool joystick; #endif static int ac97_clock = 48000; static char *ac97_quirk; @@ -110,7 +110,7 @@ module_param(nodelay, int, 0444); MODULE_PARM_DESC(nodelay, "Disable 500ms init delay"); /* just for backward compatibility */ -static int enable; +static bool enable; module_param(enable, bool, 0444); diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 80a9c2b..5efcbca 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -66,7 +66,7 @@ module_param(ac97_clock, int, 0444); MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (default 48000Hz)."); /* just for backward compatibility */ -static int enable; +static bool enable; module_param(enable, bool, 0444); diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index 6765822..6a534bf 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -37,8 +37,8 @@ MODULE_SUPPORTED_DEVICE("{{Digigram," CARD_NAME "}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int mic[SNDRV_CARDS]; /* microphone */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool mic[SNDRV_CARDS]; /* microphone */ static int ibl[SNDRV_CARDS]; /* microphone */ module_param_array(index, int, NULL, 0444); diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index e97ddca..e57b89e8 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -41,13 +41,13 @@ MODULE_SUPPORTED_DEVICE("{{Yamaha,YMF724}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ static long fm_port[SNDRV_CARDS]; static long mpu_port[SNDRV_CARDS]; #ifdef SUPPORT_JOYSTICK static long joystick_port[SNDRV_CARDS]; #endif -static int rear_switch[SNDRV_CARDS]; +static bool rear_switch[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the Yamaha DS-1 PCI soundcard."); -- cgit v1.1 From cde944803d12450f70f0adc4d418afcd8e42db2e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 19 Dec 2011 10:32:48 +0100 Subject: ALSA: Add missing module parameters for als300 and cs5530 drivers These drviers defined only variables but didn't declare as module parameters. Also fix the enable variable to bool type. Signed-off-by: Takashi Iwai --- sound/pci/als300.c | 9 ++++++++- sound/pci/cs5530.c | 9 ++++++++- 2 files changed, 16 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 8dc77a0..8196e22 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -115,7 +115,14 @@ MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS300},{Avance Logic,ALS300+}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for ALS300 sound card."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for ALS300 sound card."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable ALS300 sound card."); struct snd_als300 { unsigned long port; diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 958f494..c47cabf 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -50,7 +50,14 @@ MODULE_LICENSE("GPL"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for CS5530 Audio driver."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for CS5530 Audio driver."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable CS5530 Audio driver."); struct snd_cs5530 { struct snd_card *card; -- cgit v1.1 From 82b1d73f1f22df2c8384cb7cea4aabd9db5273a9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Dec 2011 15:53:07 +0100 Subject: ALSA: hda - Fix left-over merge issues in patch_hdmi.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3f42cc9..1168ebd 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -805,7 +805,6 @@ static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) { - struct hdmi_spec *spec = codec->spec; int tag = res >> AC_UNSOL_RES_TAG_SHIFT; int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; @@ -1242,7 +1241,6 @@ static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx) if (pcmdev > 0) sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev); - hdmi_present_sense(per_pin, 0); return snd_hda_jack_add_kctl(codec, per_pin->pin_nid, hdmi_str, 0); } @@ -1274,7 +1272,7 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) if (err < 0) return err; - hdmi_present_sense(per_pin, false); + hdmi_present_sense(per_pin, 0); } return 0; -- cgit v1.1 From 47a74a5d1ed2af23c2dc1ccfdcc0176e40404345 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 11:54:02 +1300 Subject: ALSA: asihpi - fix pcm dma pointer tracking Elapsed counter should only count data committed to snd_pcm_period_elapsed, rather than all data available Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index e9de799..1ba50e3 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -888,8 +888,8 @@ static void snd_card_asihpi_timer_function(unsigned long data) pd, xfer2)); } } - ds->pcm_buf_host_rw_ofs = ds->pcm_buf_host_rw_ofs + xfercount; - ds->pcm_buf_elapsed_dma_ofs = pcm_buf_dma_ofs; + ds->pcm_buf_host_rw_ofs += xfercount; + ds->pcm_buf_elapsed_dma_ofs += xfercount; snd_pcm_period_elapsed(s); } } -- cgit v1.1 From f6baaec2af36494469aa37558db8c79186f2fa03 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:31 +1300 Subject: ALSA: asihpi - Split hpi version info into separate header file. and update HPI version to 4.10 Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi.h | 18 ------------------ sound/pci/asihpi/hpi_version.h | 32 ++++++++++++++++++++++++++++++++ sound/pci/asihpi/hpidspcd.c | 23 ++++++++++++----------- sound/pci/asihpi/hpidspcd.h | 4 ---- sound/pci/asihpi/hpimsgx.c | 3 ++- sound/pci/asihpi/hpioctl.c | 1 + 6 files changed, 47 insertions(+), 34 deletions(-) create mode 100644 sound/pci/asihpi/hpi_version.h (limited to 'sound/pci') diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index f207272..7714937 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -30,26 +30,8 @@ #ifndef _HPI_H_ #define _HPI_H_ -/* HPI Version -If HPI_VER_MINOR is odd then its a development release not intended for the -public. If HPI_VER_MINOR is even then is a release version -i.e 3.05.02 is a development version -*/ -#define HPI_VERSION_CONSTRUCTOR(maj, min, rel) \ - ((maj << 16) + (min << 8) + rel) - -#define HPI_VER_MAJOR(v) ((int)(v >> 16)) -#define HPI_VER_MINOR(v) ((int)((v >> 8) & 0xFF)) -#define HPI_VER_RELEASE(v) ((int)(v & 0xFF)) - -#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 8, 0) -#define HPI_VER_STRING "4.08.00" - -/* Library version as documented in hpi-api-versions.txt */ -#define HPI_LIB_VER HPI_VERSION_CONSTRUCTOR(10, 0, 0) #include -#define HPI_BUILD_EXCLUDE_DEPRECATED #define HPI_BUILD_KERNEL_MODE /******************************************************************************/ diff --git a/sound/pci/asihpi/hpi_version.h b/sound/pci/asihpi/hpi_version.h new file mode 100644 index 0000000..e9146e5 --- /dev/null +++ b/sound/pci/asihpi/hpi_version.h @@ -0,0 +1,32 @@ +/** HPI Version Definitions +Development releases have odd minor version. +Production releases have even minor version. + +\file hpi_version.h +*/ + +#ifndef _HPI_VERSION_H +#define _HPI_VERSION_H + +/* Use single digits for versions less that 10 to avoid octal. */ +/* *** HPI_VER is the only edit required to update version *** */ +/** HPI version */ +#define HPI_VER HPI_VERSION_CONSTRUCTOR(4, 10, 1) + +/** HPI version string in dotted decimal format */ +#define HPI_VER_STRING "4.10.01" + +/** Library version as documented in hpi-api-versions.txt */ +#define HPI_LIB_VER HPI_VERSION_CONSTRUCTOR(10, 2, 0) + +/** Construct hpi version number from major, minor, release numbers */ +#define HPI_VERSION_CONSTRUCTOR(maj, min, r) ((maj << 16) + (min << 8) + r) + +/** Extract major version from hpi version number */ +#define HPI_VER_MAJOR(v) ((int)(v >> 16)) +/** Extract minor version from hpi version number */ +#define HPI_VER_MINOR(v) ((int)((v >> 8) & 0xFF)) +/** Extract release from hpi version number */ +#define HPI_VER_RELEASE(v) ((int)(v & 0xFF)) + +#endif diff --git a/sound/pci/asihpi/hpidspcd.c b/sound/pci/asihpi/hpidspcd.c index 71d32c8..21cdb9e 100644 --- a/sound/pci/asihpi/hpidspcd.c +++ b/sound/pci/asihpi/hpidspcd.c @@ -25,6 +25,7 @@ hotplug firmware loader from individual dsp code files #define SOURCEFILE_NAME "hpidspcd.c" #include "hpidspcd.h" #include "hpidebug.h" +#include "hpi_version.h" struct dsp_code_private { /** Firmware descriptor */ @@ -32,9 +33,6 @@ struct dsp_code_private { struct pci_dev *dev; }; -#define HPI_VER_DECIMAL ((int)(HPI_VER_MAJOR(HPI_VER) * 10000 + \ - HPI_VER_MINOR(HPI_VER) * 100 + HPI_VER_RELEASE(HPI_VER))) - /*-------------------------------------------------------------------*/ short hpi_dsp_code_open(u32 adapter, void *os_data, struct dsp_code *dsp_code, u32 *os_error_code) @@ -66,22 +64,25 @@ short hpi_dsp_code_open(u32 adapter, void *os_data, struct dsp_code *dsp_code, if ((header.type != 0x45444F43) || /* "CODE" */ (header.adapter != adapter) || (header.size != firmware->size)) { - dev_printk(KERN_ERR, &dev->dev, "Invalid firmware file\n"); + dev_printk(KERN_ERR, &dev->dev, + "Invalid firmware header size %d != file %zd\n", + header.size, firmware->size); goto error2; } - if ((header.version / 100 & ~1) != (HPI_VER_DECIMAL / 100 & ~1)) { + if ((header.version >> 9) != (HPI_VER >> 9)) { + /* Consider even and subsequent odd minor versions to be compatible */ dev_printk(KERN_ERR, &dev->dev, "Incompatible firmware version " - "DSP image %d != Driver %d\n", header.version, - HPI_VER_DECIMAL); + "DSP image %X != Driver %X\n", header.version, + HPI_VER); goto error2; } - if (header.version != HPI_VER_DECIMAL) { - dev_printk(KERN_WARNING, &dev->dev, - "Firmware: release version mismatch DSP image %d != Driver %d\n", - header.version, HPI_VER_DECIMAL); + if (header.version != HPI_VER) { + dev_printk(KERN_INFO, &dev->dev, + "Firmware: release version mismatch DSP image %X != Driver %X\n", + header.version, HPI_VER); } HPI_DEBUG_LOG(DEBUG, "dsp code %s opened\n", fw_name); diff --git a/sound/pci/asihpi/hpidspcd.h b/sound/pci/asihpi/hpidspcd.h index b228811..659d19c 100644 --- a/sound/pci/asihpi/hpidspcd.h +++ b/sound/pci/asihpi/hpidspcd.h @@ -27,10 +27,6 @@ Functions for reading DSP code to load into DSP #include "hpi_internal.h" -/** Code header version is decimal encoded e.g. 4.06.10 is 40601 */ -#define HPI_VER_DECIMAL ((int)(HPI_VER_MAJOR(HPI_VER) * 10000 + \ -HPI_VER_MINOR(HPI_VER) * 100 + HPI_VER_RELEASE(HPI_VER))) - /** Header structure for dsp firmware file This structure must match that used in s2bin.c for generation of asidsp.bin */ diff --git a/sound/pci/asihpi/hpimsgx.c b/sound/pci/asihpi/hpimsgx.c index 2e77942..d4790dd 100644 --- a/sound/pci/asihpi/hpimsgx.c +++ b/sound/pci/asihpi/hpimsgx.c @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2010 AudioScience Inc. + Copyright (C) 1997-2011 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -22,6 +22,7 @@ Extended Message Function With Response Caching *****************************************************************************/ #define SOURCEFILE_NAME "hpimsgx.c" #include "hpi_internal.h" +#include "hpi_version.h" #include "hpimsginit.h" #include "hpicmn.h" #include "hpimsgx.h" diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index f6b9517..75f7a2d 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -21,6 +21,7 @@ Common Linux HPI ioctl and module probe/remove functions #define SOURCEFILE_NAME "hpioctl.c" #include "hpi_internal.h" +#include "hpi_version.h" #include "hpimsginit.h" #include "hpidebug.h" #include "hpimsgx.h" -- cgit v1.1 From 40818b6242513676c8adf30811fb7877b02005fb Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:32 +1300 Subject: ALSA: asihpi - Update copyright to 2011 Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi6000.c | 2 +- sound/pci/asihpi/hpi6000.h | 2 +- sound/pci/asihpi/hpi6205.c | 2 +- sound/pci/asihpi/hpicmn.c | 2 +- sound/pci/asihpi/hpicmn.h | 2 +- sound/pci/asihpi/hpidebug.c | 2 +- sound/pci/asihpi/hpidebug.h | 2 +- sound/pci/asihpi/hpimsginit.c | 2 +- sound/pci/asihpi/hpimsginit.h | 2 +- sound/pci/asihpi/hpimsgx.h | 2 +- sound/pci/asihpi/hpioctl.h | 2 +- sound/pci/asihpi/hpios.c | 2 +- sound/pci/asihpi/hpios.h | 2 +- sound/pci/asihpi/hpipcida.h | 2 +- 14 files changed, 14 insertions(+), 14 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c index 3cc6f11..278bec8 100644 --- a/sound/pci/asihpi/hpi6000.c +++ b/sound/pci/asihpi/hpi6000.c @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2010 AudioScience Inc. + Copyright (C) 1997-2011 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as diff --git a/sound/pci/asihpi/hpi6000.h b/sound/pci/asihpi/hpi6000.h index 4c7d507..7e0deef 100644 --- a/sound/pci/asihpi/hpi6000.h +++ b/sound/pci/asihpi/hpi6000.h @@ -1,7 +1,7 @@ /***************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2010 AudioScience Inc. + Copyright (C) 1997-2011 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index e041a6a..7f65602 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2010 AudioScience Inc. + Copyright (C) 1997-2011 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c index 44c7eb4..358853a 100644 --- a/sound/pci/asihpi/hpicmn.c +++ b/sound/pci/asihpi/hpicmn.c @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2010 AudioScience Inc. + Copyright (C) 1997-2011 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as diff --git a/sound/pci/asihpi/hpicmn.h b/sound/pci/asihpi/hpicmn.h index d53cdf6..67cc1b0 100644 --- a/sound/pci/asihpi/hpicmn.h +++ b/sound/pci/asihpi/hpicmn.h @@ -1,7 +1,7 @@ /** AudioScience HPI driver - Copyright (C) 1997-2010 AudioScience Inc. + Copyright (C) 1997-2011 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as diff --git a/sound/pci/asihpi/hpidebug.c b/sound/pci/asihpi/hpidebug.c index b52baf6..ac86a1f 100644 --- a/sound/pci/asihpi/hpidebug.c +++ b/sound/pci/asihpi/hpidebug.c @@ -1,7 +1,7 @@ /************************************************************************ AudioScience HPI driver - Copyright (C) 1997-2010 AudioScience Inc. + Copyright (C) 1997-2011 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as diff --git a/sound/pci/asihpi/hpidebug.h b/sound/pci/asihpi/hpidebug.h index 940f54c..2c9af23 100644 --- a/sound/pci/asihpi/hpidebug.h +++ b/sound/pci/asihpi/hpidebug.h @@ -1,7 +1,7 @@ /***************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2010 AudioScience Inc. + Copyright (C) 1997-2011 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as diff --git a/sound/pci/asihpi/hpimsginit.c b/sound/pci/asihpi/hpimsginit.c index 52400a6..032d563 100644 --- a/sound/pci/asihpi/hpimsginit.c +++ b/sound/pci/asihpi/hpimsginit.c @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2010 AudioScience Inc. + Copyright (C) 1997-2011 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as diff --git a/sound/pci/asihpi/hpimsginit.h b/sound/pci/asihpi/hpimsginit.h index bfd330d..5b48708 100644 --- a/sound/pci/asihpi/hpimsginit.h +++ b/sound/pci/asihpi/hpimsginit.h @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2010 AudioScience Inc. + Copyright (C) 1997-2011 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as diff --git a/sound/pci/asihpi/hpimsgx.h b/sound/pci/asihpi/hpimsgx.h index fd49e75..37f3efd 100644 --- a/sound/pci/asihpi/hpimsgx.h +++ b/sound/pci/asihpi/hpimsgx.h @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2010 AudioScience Inc. + Copyright (C) 1997-2011 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as diff --git a/sound/pci/asihpi/hpioctl.h b/sound/pci/asihpi/hpioctl.h index 847f72f..2614aff 100644 --- a/sound/pci/asihpi/hpioctl.h +++ b/sound/pci/asihpi/hpioctl.h @@ -1,7 +1,7 @@ /******************************************************************************* AudioScience HPI driver - Copyright (C) 1997-2010 AudioScience Inc. + Copyright (C) 1997-2011 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c index ff2a19b..2d7d1c2 100644 --- a/sound/pci/asihpi/hpios.c +++ b/sound/pci/asihpi/hpios.c @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2010 AudioScience Inc. + Copyright (C) 1997-2011 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h index 2f605e3..d59a059 100644 --- a/sound/pci/asihpi/hpios.h +++ b/sound/pci/asihpi/hpios.h @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2010 AudioScience Inc. + Copyright (C) 1997-2011 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as diff --git a/sound/pci/asihpi/hpipcida.h b/sound/pci/asihpi/hpipcida.h index bb30868..db570dd 100644 --- a/sound/pci/asihpi/hpipcida.h +++ b/sound/pci/asihpi/hpipcida.h @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2010 AudioScience Inc. + Copyright (C) 1997-2011 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as -- cgit v1.1 From c382a5da5cda3e0d8a8f2e8809460285d0a7c1cb Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:33 +1300 Subject: ALSA: asihpi - Low latency mode stream has fixed channel count. Unlike other streams which support 1..max channels, Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 1ba50e3..44e6ef3 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -135,6 +135,8 @@ struct snd_card_asihpi { u16 update_interval_frames; u16 in_max_chans; u16 out_max_chans; + u16 in_min_chans; + u16 out_min_chans; }; /* Per stream data */ @@ -968,8 +970,6 @@ static void snd_card_asihpi_playback_format(struct snd_card_asihpi *asihpi, } static struct snd_pcm_hardware snd_card_asihpi_playback = { - .channels_min = 1, - .channels_max = 2, .buffer_bytes_max = BUFFER_BYTES_MAX, .period_bytes_min = PERIOD_BYTES_MIN, .period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN, @@ -1013,6 +1013,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) runtime->private_free = snd_card_asihpi_runtime_free; snd_card_asihpi_playback.channels_max = card->out_max_chans; + snd_card_asihpi_playback.channels_min = card->out_min_chans; /*?snd_card_asihpi_playback.period_bytes_min = card->out_max_chans * 4096; */ @@ -1150,8 +1151,6 @@ static void snd_card_asihpi_capture_format(struct snd_card_asihpi *asihpi, static struct snd_pcm_hardware snd_card_asihpi_capture = { - .channels_min = 1, - .channels_max = 2, .buffer_bytes_max = BUFFER_BYTES_MAX, .period_bytes_min = PERIOD_BYTES_MIN, .period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN, @@ -1193,6 +1192,7 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) runtime->private_free = snd_card_asihpi_runtime_free; snd_card_asihpi_capture.channels_max = card->in_max_chans; + snd_card_asihpi_capture.channels_min = card->in_min_chans; snd_card_asihpi_capture_format(card, dpcm->h_stream, &snd_card_asihpi_capture); snd_card_asihpi_pcm_samplerates(card, &snd_card_asihpi_capture); @@ -2883,6 +2883,15 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, asihpi->out_max_chans = 2; } + if (asihpi->out_max_chans > 2) { /* assume LL mode */ + asihpi->out_min_chans = asihpi->out_max_chans; + asihpi->in_min_chans = asihpi->in_max_chans; + asihpi->support_grouping = 0; + } else { + asihpi->out_min_chans = 1; + asihpi->in_min_chans = 1; + } + snd_printk(KERN_INFO "has dma:%d, grouping:%d, mrx:%d\n", asihpi->can_dma, asihpi->support_grouping, -- cgit v1.1 From d4b06d23ab1c5aefbb1377fb01939b555236f57f Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:34 +1300 Subject: ALSA: asihpi - Volumes and meters may have 1 or 2 channels. The channel count can be queried to determine which. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 44e6ef3..6e89e5b 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1410,6 +1410,7 @@ static int snd_asihpi_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { u32 h_control = kcontrol->private_value; + u32 count; u16 err; /* native gains are in millibels */ short min_gain_mB; @@ -1424,8 +1425,12 @@ static int snd_asihpi_volume_info(struct snd_kcontrol *kcontrol, step_gain_mB = VOL_STEP_mB; } + err = hpi_meter_query_channels(h_control, &count); + if (err) + count = HPI_MAX_CHANNELS; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; + uinfo->count = count; uinfo->value.integer.min = min_gain_mB / VOL_STEP_mB; uinfo->value.integer.max = max_gain_mB / VOL_STEP_mB; uinfo->value.integer.step = step_gain_mB / VOL_STEP_mB; @@ -2033,8 +2038,15 @@ static int __devinit snd_asihpi_tuner_add(struct snd_card_asihpi *asihpi, static int snd_asihpi_meter_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { + u32 h_control = kcontrol->private_value; + u32 count; + u16 err; + err = hpi_meter_query_channels(h_control, &count); + if (err) + count = HPI_MAX_CHANNELS; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = HPI_MAX_CHANNELS; + uinfo->count = count; uinfo->value.integer.min = 0; uinfo->value.integer.max = 0x7FFFFFFF; return 0; -- cgit v1.1 From cbd757daf5ed29618214b4ec4e298c79117baa8e Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:35 +1300 Subject: ALSA: asihpi - Use snd_pcm_debug_name to get substream name. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 6e89e5b..96a6eb0 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -904,7 +904,9 @@ static void snd_card_asihpi_timer_function(unsigned long data) static int snd_card_asihpi_playback_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) { - snd_printddd(KERN_INFO "P%d ioctl %d\n", substream->number, cmd); + char name[16]; + snd_pcm_debug_name(substream, name, sizeof(name)); + snd_printddd(KERN_INFO "%s ioctl %d\n", name, cmd); return snd_pcm_lib_ioctl(substream, cmd, arg); } @@ -929,9 +931,11 @@ snd_card_asihpi_playback_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_card_asihpi_pcm *dpcm = runtime->private_data; snd_pcm_uframes_t ptr; + char name[16]; + snd_pcm_debug_name(substream, name, sizeof(name)); ptr = bytes_to_frames(runtime, dpcm->pcm_buf_dma_ofs % dpcm->buffer_bytes); - snd_printddd("P%d pointer = 0x%04lx\n", substream->number, (unsigned long)ptr); + snd_printddd("%s pointer = 0x%04lx\n", name, (unsigned long)ptr); return ptr; } -- cgit v1.1 From 8e0874ea72d9dde8d300a5d4d6dfa87238ff7a81 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:36 +1300 Subject: ALSA: asihpi - Correct stray capital letters in identifier. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi6205.c | 12 ++++++------ sound/pci/asihpi/hpi_internal.h | 4 ++-- 2 files changed, 8 insertions(+), 8 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 7f65602..95d1cd5 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -803,8 +803,8 @@ static void outstream_host_buffer_allocate(struct hpi_adapter_obj *pao, obj_index]; status->samples_processed = 0; status->stream_state = HPI_STATE_STOPPED; - status->dSP_index = 0; - status->host_index = status->dSP_index; + status->dsp_index = 0; + status->host_index = status->dsp_index; status->size_in_bytes = phm->u.d.u.buffer.buffer_size; status->auxiliary_data_available = 0; @@ -878,7 +878,7 @@ static void outstream_host_buffer_free(struct hpi_adapter_obj *pao, static u32 outstream_get_space_available(struct hpi_hostbuffer_status *status) { return status->size_in_bytes - (status->host_index - - status->dSP_index); + status->dsp_index); } static void outstream_write(struct hpi_adapter_obj *pao, @@ -1080,8 +1080,8 @@ static void instream_host_buffer_allocate(struct hpi_adapter_obj *pao, obj_index]; status->samples_processed = 0; status->stream_state = HPI_STATE_STOPPED; - status->dSP_index = 0; - status->host_index = status->dSP_index; + status->dsp_index = 0; + status->host_index = status->dsp_index; status->size_in_bytes = phm->u.d.u.buffer.buffer_size; status->auxiliary_data_available = 0; @@ -1162,7 +1162,7 @@ static void instream_start(struct hpi_adapter_obj *pao, static u32 instream_get_bytes_available(struct hpi_hostbuffer_status *status) { - return status->dSP_index - status->host_index; + return status->dsp_index - status->host_index; } static void instream_read(struct hpi_adapter_obj *pao, diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index d497030..e5e9e33 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -618,7 +618,7 @@ struct hpi_hostbuffer_status { u32 auxiliary_data_available; u32 stream_state; /* DSP index in to the host bus master buffer. */ - u32 dSP_index; + u32 dsp_index; /* Host index in to the host bus master buffer. */ u32 host_index; u32 size_in_bytes; @@ -1461,7 +1461,7 @@ struct hpi_control_cache_pad { /* 2^N sized FIFO buffer (internal to HPI<->DSP interaction) */ struct hpi_fifo_buffer { u32 size; - u32 dSP_index; + u32 dsp_index; u32 host_index; }; -- cgit v1.1 From 0be55c453f0b46de3d4e9025749dbba571cd7b84 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:37 +1300 Subject: ALSA: asihpi - Relax drained check for more reliable playback startup. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 96a6eb0..d3cab35 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -759,8 +759,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { pcm_buf_dma_ofs = ds->pcm_buf_host_rw_ofs - bytes_avail; if (state == HPI_STATE_STOPPED) { - if ((bytes_avail == 0) && - (on_card_bytes < ds->pcm_buf_host_rw_ofs)) { + if (bytes_avail == 0) { hpi_handle_error(hpi_stream_start(ds->h_stream)); snd_printdd("P%d start\n", s->number); ds->drained_count = 0; @@ -769,7 +768,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) snd_printd(KERN_WARNING "P%d drained\n", s->number); ds->drained_count++; - if (ds->drained_count > 2) { + if (ds->drained_count > 20) { snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN); continue; } -- cgit v1.1 From 09c728aced2bb212ce060a91c28d2ee40a6bf33c Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:38 +1300 Subject: ALSA: asihpi - Only set sync if card supports hardware stream grouping. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index d3cab35..62f094c 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1033,8 +1033,10 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID; - if (card->support_grouping) + if (card->support_grouping) { snd_card_asihpi_playback.info |= SNDRV_PCM_INFO_SYNC_START; + snd_pcm_set_sync(substream); + } /* struct is copied, so can create initializer dynamically */ runtime->hw = snd_card_asihpi_playback; @@ -1051,8 +1053,6 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, card->update_interval_frames * 2, UINT_MAX); - snd_pcm_set_sync(substream); - snd_printdd("playback open\n"); return 0; -- cgit v1.1 From 502f271ae3d61cbe8ae75a11f28f17174d7bd380 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:39 +1300 Subject: ALSA: asihpi - Update node types. Add "Internal" node type. Remove GPI and GPO node types. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 3 +-- sound/pci/asihpi/hpi.h | 5 ++--- 2 files changed, 3 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 62f094c..f56a1b8 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1317,7 +1317,7 @@ static const char * const asihpi_src_names[] = { "Analog", "Adapter", "RTP", - "GPI", + "Internal" }; compile_time_assert( @@ -1335,7 +1335,6 @@ static const char * const asihpi_dst_names[] = { "Net", "Analog", "RTP", - "GPO", }; compile_time_assert( diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index 7714937..867c144 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -195,7 +195,7 @@ enum HPI_SOURCENODES { /** RTP stream input node - This node is a destination for packets of RTP audio samples from other devices. */ HPI_SOURCENODE_RTP_DESTINATION = 112, - HPI_SOURCENODE_GP_IN = 113, /**< general purpose input. */ + HPI_SOURCENODE_INTERNAL = 113, /**< node internal to the device. */ /* !!!Update this AND hpidebug.h if you add a new sourcenode type!!! */ HPI_SOURCENODE_LAST_INDEX = 113 /**< largest ID */ /* AX6 max sourcenode types = 15 */ @@ -224,9 +224,8 @@ enum HPI_DESTNODES { /** RTP stream output node - This node is a source for packets of RTP audio samples that are sent to other devices. */ HPI_DESTNODE_RTP_SOURCE = 208, - HPI_DESTNODE_GP_OUT = 209, /**< general purpose output node. */ /* !!!Update this AND hpidebug.h if you add a new destnode type!!! */ - HPI_DESTNODE_LAST_INDEX = 209 /**< largest ID */ + HPI_DESTNODE_LAST_INDEX = 208 /**< largest ID */ /* AX6 max destnode types = 15 */ }; -- cgit v1.1 From d8aefaef1b10f1063f295342b8967543e0d11d72 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:40 +1300 Subject: ALSA: asihpi - Remove unused structs and defs Structs related to network flash update are not required in kernel. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi_internal.h | 96 +---------------------------------------- 1 file changed, 2 insertions(+), 94 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index e5e9e33..e06c5e0 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -25,6 +25,7 @@ HPI internal definitions #define _HPI_INTERNAL_H_ #include "hpi.h" + /** maximum number of memory regions mapped to an adapter */ #define HPI_MAX_ADAPTER_MEM_SPACES (2) @@ -220,8 +221,6 @@ enum HPI_CONTROL_ATTRIBUTES { HPI_COBRANET_SET = HPI_CTL_ATTR(COBRANET, 1), HPI_COBRANET_GET = HPI_CTL_ATTR(COBRANET, 2), - /*HPI_COBRANET_SET_DATA = HPI_CTL_ATTR(COBRANET, 3), */ - /*HPI_COBRANET_GET_DATA = HPI_CTL_ATTR(COBRANET, 4), */ HPI_COBRANET_GET_STATUS = HPI_CTL_ATTR(COBRANET, 5), HPI_COBRANET_SEND_PACKET = HPI_CTL_ATTR(COBRANET, 6), HPI_COBRANET_GET_PACKET = HPI_CTL_ATTR(COBRANET, 7), @@ -393,14 +392,10 @@ enum HPI_FUNCTION_IDS { HPI_SUBSYS_OPEN = HPI_FUNC_ID(SUBSYSTEM, 1), HPI_SUBSYS_GET_VERSION = HPI_FUNC_ID(SUBSYSTEM, 2), HPI_SUBSYS_GET_INFO = HPI_FUNC_ID(SUBSYSTEM, 3), - /* HPI_SUBSYS_FIND_ADAPTERS = HPI_FUNC_ID(SUBSYSTEM, 4), */ HPI_SUBSYS_CREATE_ADAPTER = HPI_FUNC_ID(SUBSYSTEM, 5), HPI_SUBSYS_CLOSE = HPI_FUNC_ID(SUBSYSTEM, 6), - /* HPI_SUBSYS_DELETE_ADAPTER = HPI_FUNC_ID(SUBSYSTEM, 7), */ HPI_SUBSYS_DRIVER_LOAD = HPI_FUNC_ID(SUBSYSTEM, 8), HPI_SUBSYS_DRIVER_UNLOAD = HPI_FUNC_ID(SUBSYSTEM, 9), - /* HPI_SUBSYS_READ_PORT_8 = HPI_FUNC_ID(SUBSYSTEM, 10), */ - /* HPI_SUBSYS_WRITE_PORT_8 = HPI_FUNC_ID(SUBSYSTEM, 11), */ HPI_SUBSYS_GET_NUM_ADAPTERS = HPI_FUNC_ID(SUBSYSTEM, 12), HPI_SUBSYS_GET_ADAPTER = HPI_FUNC_ID(SUBSYSTEM, 13), HPI_SUBSYS_SET_NETWORK_INTERFACE = HPI_FUNC_ID(SUBSYSTEM, 14), @@ -661,13 +656,6 @@ union hpi_adapterx_msg { u16 index; } module_info; struct { - u32 checksum; - u16 sequence; - u16 length; - u16 offset; /**< offset from start of msg to data */ - u16 unused; - } program_flash; - struct { u16 index; u16 what; u16 property_index; @@ -678,25 +666,18 @@ union hpi_adapterx_msg { u16 parameter2; } property_set; struct { - u32 offset; - } query_flash; - struct { u32 pad32; u16 key1; u16 key2; } restart; struct { - u32 offset; - u32 length; - u32 key; - } start_flash; - struct { u32 pad32; u16 value; } test_assert; struct { u32 yes; } irq_query; + u32 pad[3]; }; struct hpi_adapter_res { @@ -724,18 +705,10 @@ union hpi_adapterx_res { u32 adapter_mode; } mode; struct { - u16 sequence; - } program_flash; - struct { u16 parameter1; u16 parameter2; } property_get; struct { - u32 checksum; - u32 length; - u32 version; - } query_flash; - struct { u32 yes; } irq_query; }; @@ -1150,71 +1123,6 @@ struct hpi_res_adapter_get_info { struct hpi_adapter_res p; }; -/* padding is so these are same size as v0 hpi_message */ -struct hpi_msg_adapter_query_flash { - struct hpi_message_header h; - u32 offset; - u8 pad_to_version0_size[sizeof(struct hpi_message) - /* V0 res */ - sizeof(struct hpi_message_header) - 1 * sizeof(u32)]; -}; - -/* padding is so these are same size as v0 hpi_response */ -struct hpi_res_adapter_query_flash { - struct hpi_response_header h; - u32 checksum; - u32 length; - u32 version; - u8 pad_to_version0_size[sizeof(struct hpi_response) - /* V0 res */ - sizeof(struct hpi_response_header) - 3 * sizeof(u32)]; -}; - -struct hpi_msg_adapter_start_flash { - struct hpi_message_header h; - u32 offset; - u32 length; - u32 key; - u8 pad_to_version0_size[sizeof(struct hpi_message) - /* V0 res */ - sizeof(struct hpi_message_header) - 3 * sizeof(u32)]; -}; - -struct hpi_res_adapter_start_flash { - struct hpi_response_header h; - u8 pad_to_version0_size[sizeof(struct hpi_response) - /* V0 res */ - sizeof(struct hpi_response_header)]; -}; - -struct hpi_msg_adapter_program_flash_payload { - u32 checksum; - u16 sequence; - u16 length; - u16 offset; /**< offset from start of msg to data */ - u16 unused; - /* ensure sizeof(header + payload) == sizeof(hpi_message_V0) - because old firmware expects data after message of this size */ - u8 pad_to_version0_size[sizeof(struct hpi_message) - /* V0 message */ - sizeof(struct hpi_message_header) - sizeof(u32) - - 4 * sizeof(u16)]; -}; - -struct hpi_msg_adapter_program_flash { - struct hpi_message_header h; - struct hpi_msg_adapter_program_flash_payload p; - u32 data[256]; -}; - -struct hpi_res_adapter_program_flash { - struct hpi_response_header h; - u16 sequence; - u8 pad_to_version0_size[sizeof(struct hpi_response) - /* V0 res */ - sizeof(struct hpi_response_header) - sizeof(u16)]; -}; - -struct hpi_msg_adapter_debug_read { - struct hpi_message_header h; - u32 dsp_address; - u32 count_bytes; -}; - struct hpi_res_adapter_debug_read { struct hpi_response_header h; u8 bytes[256]; -- cgit v1.1 From 72868339e45cbd01ec20c2d3bb8c43e8d6911331 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:41 +1300 Subject: ALSA: asihpi - Add new function codes. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi_internal.h | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index e06c5e0..2a63308 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -240,7 +240,9 @@ enum HPI_CONTROL_ATTRIBUTES { HPI_PAD_PROGRAM_TYPE = HPI_CTL_ATTR(PAD, 5), HPI_PAD_PROGRAM_ID = HPI_CTL_ATTR(PAD, 6), HPI_PAD_TA_SUPPORT = HPI_CTL_ATTR(PAD, 7), - HPI_PAD_TA_ACTIVE = HPI_CTL_ATTR(PAD, 8) + HPI_PAD_TA_ACTIVE = HPI_CTL_ATTR(PAD, 8), + + HPI_UNIVERSAL_ENTITY = HPI_CTL_ATTR(UNIVERSAL, 1) }; #define HPI_POLARITY_POSITIVE 0 @@ -425,7 +427,10 @@ enum HPI_FUNCTION_IDS { HPI_ADAPTER_IRQ_QUERY_AND_CLEAR = HPI_FUNC_ID(ADAPTER, 19), HPI_ADAPTER_IRQ_CALLBACK = HPI_FUNC_ID(ADAPTER, 20), HPI_ADAPTER_DELETE = HPI_FUNC_ID(ADAPTER, 21), -#define HPI_ADAPTER_FUNCTION_COUNT 21 + HPI_ADAPTER_READ_FLASH = HPI_FUNC_ID(ADAPTER, 22), + HPI_ADAPTER_END_FLASH = HPI_FUNC_ID(ADAPTER, 23), + HPI_ADAPTER_FILESTORE_DELETE_ALL = HPI_FUNC_ID(ADAPTER, 24), +#define HPI_ADAPTER_FUNCTION_COUNT 24 HPI_OSTREAM_OPEN = HPI_FUNC_ID(OSTREAM, 1), HPI_OSTREAM_CLOSE = HPI_FUNC_ID(OSTREAM, 2), @@ -490,7 +495,9 @@ enum HPI_FUNCTION_IDS { HPI_MIXER_GET_CONTROL_MULTIPLE_VALUES = HPI_FUNC_ID(MIXER, 10), HPI_MIXER_STORE = HPI_FUNC_ID(MIXER, 11), HPI_MIXER_GET_CACHE_INFO = HPI_FUNC_ID(MIXER, 12), -#define HPI_MIXER_FUNCTION_COUNT 12 + HPI_MIXER_GET_BLOCK_HANDLE = HPI_FUNC_ID(MIXER, 13), + HPI_MIXER_GET_PARAMETER_HANDLE = HPI_FUNC_ID(MIXER, 14), +#define HPI_MIXER_FUNCTION_COUNT 14 HPI_CONTROL_GET_INFO = HPI_FUNC_ID(CONTROL, 1), HPI_CONTROL_GET_STATE = HPI_FUNC_ID(CONTROL, 2), -- cgit v1.1 From 3dad06ac89f4b63fcce5020abbe1b3e5754e26dd Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:42 +1300 Subject: ALSA: asihpi - Increase debug response buffer size. Enables retrieving more debug info in fewer transactions. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi_internal.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index 2a63308..4cc315d 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -1132,7 +1132,7 @@ struct hpi_res_adapter_get_info { struct hpi_res_adapter_debug_read { struct hpi_response_header h; - u8 bytes[256]; + u8 bytes[1024]; }; struct hpi_msg_cobranet_hmi { -- cgit v1.1 From 7036b92d303a01477e27a5a9b2d582a5df3cc8ef Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:43 +1300 Subject: ALSA: asihpi - Remove redundant struct members. Structs hpi_adapter and snd_card_asihpi had members that duplicate those in underlying hpi_adapter_obj or whose info can be retrieved using hpi_adapter_get_info(). Print less info in probe function, it can be retrieved from /proc. Avoid name redundancy: hpi_adapter_obj.adapter_type renamed to .type Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 143 +++++++++++++++++++++------------------------ sound/pci/asihpi/hpi6000.c | 59 +++++++++---------- sound/pci/asihpi/hpi6205.c | 8 +-- sound/pci/asihpi/hpicmn.c | 20 +++---- sound/pci/asihpi/hpicmn.h | 11 ++-- sound/pci/asihpi/hpioctl.c | 62 +++++++++----------- sound/pci/asihpi/hpios.h | 14 ++--- 7 files changed, 147 insertions(+), 170 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index f56a1b8..2402801 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -25,6 +25,8 @@ #include "hpi_internal.h" #include "hpimsginit.h" #include "hpioctl.h" +#include "hpicmn.h" + #include #include @@ -119,12 +121,7 @@ struct clk_cache { struct snd_card_asihpi { struct snd_card *card; struct pci_dev *pci; - u16 adapter_index; - u32 serial_number; - u16 type; - u16 version; - u16 num_outstreams; - u16 num_instreams; + struct hpi_adapter *hpi; u32 h_mixer; struct clk_cache cc; @@ -497,6 +494,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, snd_printdd("stream_host_buffer_attach status 0x%x\n", dpcm->hpi_buffer_attached); + } bytes_per_sec = params_rate(params) * params_channels(params); width = snd_pcm_format_width(params_format(params)); @@ -993,7 +991,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) return -ENOMEM; err = - hpi_outstream_open(card->adapter_index, + hpi_outstream_open(card->hpi->adapter->index, substream->number, &dpcm->h_stream); hpi_handle_error(err); if (err) @@ -1174,10 +1172,10 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) return -ENOMEM; snd_printdd("capture open adapter %d stream %d\n", - card->adapter_index, substream->number); + card->hpi->adapter->index, substream->number); err = hpi_handle_error( - hpi_instream_open(card->adapter_index, + hpi_instream_open(card->hpi->adapter->index, substream->number, &dpcm->h_stream)); if (err) kfree(dpcm); @@ -1186,7 +1184,6 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) if (err) return -EIO; - init_timer(&dpcm->timer); dpcm->timer.data = (unsigned long) dpcm; dpcm->timer.function = snd_card_asihpi_timer_function; @@ -1243,15 +1240,20 @@ static struct snd_pcm_ops snd_card_asihpi_capture_mmap_ops = { .pointer = snd_card_asihpi_capture_pointer, }; -static int __devinit snd_card_asihpi_pcm_new(struct snd_card_asihpi *asihpi, - int device, int substreams) +static int __devinit snd_card_asihpi_pcm_new( + struct snd_card_asihpi *asihpi, int device) { struct snd_pcm *pcm; int err; + u16 num_instreams, num_outstreams, x16; + u32 x32; + + err = hpi_adapter_get_info(asihpi->hpi->adapter->index, + &num_outstreams, &num_instreams, + &x16, &x32, &x16); err = snd_pcm_new(asihpi->card, "Asihpi PCM", device, - asihpi->num_outstreams, asihpi->num_instreams, - &pcm); + num_outstreams, num_instreams, &pcm); if (err < 0) return err; /* pointer to ops struct is stored, dont change ops afterwards! */ @@ -2561,7 +2563,7 @@ static int __devinit snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi) strcpy(card->mixername, "Asihpi Mixer"); err = - hpi_mixer_open(asihpi->adapter_index, + hpi_mixer_open(asihpi->hpi->adapter->index, &asihpi->h_mixer); hpi_handle_error(err); if (err) @@ -2679,24 +2681,33 @@ snd_asihpi_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_card_asihpi *asihpi = entry->private_data; - u16 version; u32 h_control; u32 rate = 0; u16 source = 0; + + u16 num_outstreams; + u16 num_instreams; + u16 version; + u32 serial_number; + u16 type; + int err; snd_iprintf(buffer, "ASIHPI driver proc file\n"); + + hpi_handle_error(hpi_adapter_get_info(asihpi->hpi->adapter->index, + &num_outstreams, &num_instreams, + &version, &serial_number, &type)); + snd_iprintf(buffer, - "adapter ID=%4X\n_index=%d\n" - "num_outstreams=%d\n_num_instreams=%d\n", - asihpi->type, asihpi->adapter_index, - asihpi->num_outstreams, asihpi->num_instreams); + "Adapter type ASI%4X\nHardware Index %d\n" + "%d outstreams\n%d instreams\n", + type, asihpi->hpi->adapter->index, + num_outstreams, num_instreams); - version = asihpi->version; snd_iprintf(buffer, - "serial#=%d\n_hw version %c%d\nDSP code version %03d\n", - asihpi->serial_number, ((version >> 3) & 0xf) + 'A', - version & 0x7, + "Serial#%d\nHardware version %c%d\nDSP code version %03d\n", + serial_number, ((version >> 3) & 0xf) + 'A', version & 0x7, ((version >> 13) * 100) + ((version >> 7) & 0x3f)); err = hpi_mixer_get_control(asihpi->h_mixer, @@ -2704,18 +2715,15 @@ snd_asihpi_proc_read(struct snd_info_entry *entry, HPI_CONTROL_SAMPLECLOCK, &h_control); if (!err) { - err = hpi_sample_clock_get_sample_rate( - h_control, &rate); + err = hpi_sample_clock_get_sample_rate(h_control, &rate); err += hpi_sample_clock_get_source(h_control, &source); if (!err) - snd_iprintf(buffer, "sample_clock=%d_hz, source %s\n", + snd_iprintf(buffer, "Sample Clock %dHz, source %s\n", rate, sampleclock_sources[source]); } - } - static void __devinit snd_asihpi_proc_init(struct snd_card_asihpi *asihpi) { struct snd_info_entry *entry; @@ -2787,35 +2795,34 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, const struct pci_device_id *pci_id) { int err; - - u16 version; - int pcm_substreams; - - struct hpi_adapter *hpi_card; + struct hpi_adapter *hpi; struct snd_card *card; struct snd_card_asihpi *asihpi; u32 h_control; u32 h_stream; + u32 adapter_index; static int dev; if (dev >= SNDRV_CARDS) return -ENODEV; - /* Should this be enable[hpi_card->index] ? */ + /* Should this be enable[hpi->index] ? */ if (!enable[dev]) { dev++; return -ENOENT; } + /* Initialise low-level HPI driver */ err = asihpi_adapter_probe(pci_dev, pci_id); if (err < 0) return err; - hpi_card = pci_get_drvdata(pci_dev); + hpi = pci_get_drvdata(pci_dev); + adapter_index = hpi->adapter->index; /* first try to give the card the same index as its hardware index */ - err = snd_card_create(hpi_card->index, - id[hpi_card->index], THIS_MODULE, + err = snd_card_create(adapter_index, + id[adapter_index], THIS_MODULE, sizeof(struct snd_card_asihpi), &card); if (err < 0) { @@ -2829,50 +2836,32 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, return err; snd_printk(KERN_WARNING "**** WARNING **** Adapter index %d->ALSA index %d\n", - hpi_card->index, card->number); + adapter_index, card->number); } snd_card_set_dev(card, &pci_dev->dev); - asihpi = (struct snd_card_asihpi *) card->private_data; + asihpi = card->private_data; asihpi->card = card; asihpi->pci = pci_dev; - asihpi->adapter_index = hpi_card->index; - hpi_handle_error(hpi_adapter_get_info( - asihpi->adapter_index, - &asihpi->num_outstreams, - &asihpi->num_instreams, - &asihpi->version, - &asihpi->serial_number, &asihpi->type)); - - version = asihpi->version; - snd_printk(KERN_INFO "adapter ID=%4X index=%d num_outstreams=%d " - "num_instreams=%d S/N=%d\n" - "Hw Version %c%d DSP code version %03d\n", - asihpi->type, asihpi->adapter_index, - asihpi->num_outstreams, - asihpi->num_instreams, asihpi->serial_number, - ((version >> 3) & 0xf) + 'A', - version & 0x7, - ((version >> 13) * 100) + ((version >> 7) & 0x3f)); - - pcm_substreams = asihpi->num_outstreams; - if (pcm_substreams < asihpi->num_instreams) - pcm_substreams = asihpi->num_instreams; - - err = hpi_adapter_get_property(asihpi->adapter_index, + asihpi->hpi = hpi; + + snd_printk(KERN_INFO "adapter ID=%4X index=%d\n", + asihpi->hpi->adapter->type, adapter_index); + + err = hpi_adapter_get_property(adapter_index, HPI_ADAPTER_PROPERTY_CAPS1, NULL, &asihpi->support_grouping); if (err) asihpi->support_grouping = 0; - err = hpi_adapter_get_property(asihpi->adapter_index, + err = hpi_adapter_get_property(adapter_index, HPI_ADAPTER_PROPERTY_CAPS2, &asihpi->support_mrx, NULL); if (err) asihpi->support_mrx = 0; - err = hpi_adapter_get_property(asihpi->adapter_index, + err = hpi_adapter_get_property(adapter_index, HPI_ADAPTER_PROPERTY_INTERVAL, NULL, &asihpi->update_interval_frames); if (err) @@ -2881,7 +2870,7 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, if (!asihpi->can_dma) asihpi->update_interval_frames *= 2; - hpi_handle_error(hpi_instream_open(asihpi->adapter_index, + hpi_handle_error(hpi_instream_open(adapter_index, 0, &h_stream)); err = hpi_instream_host_buffer_free(h_stream); @@ -2889,7 +2878,7 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, hpi_handle_error(hpi_instream_close(h_stream)); - err = hpi_adapter_get_property(asihpi->adapter_index, + err = hpi_adapter_get_property(adapter_index, HPI_ADAPTER_PROPERTY_CURCHANNELS, &asihpi->in_max_chans, &asihpi->out_max_chans); if (err) { @@ -2906,13 +2895,13 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, asihpi->in_min_chans = 1; } - snd_printk(KERN_INFO "has dma:%d, grouping:%d, mrx:%d\n", + snd_printk(KERN_INFO "Has dma:%d, grouping:%d, mrx:%d\n", asihpi->can_dma, asihpi->support_grouping, asihpi->support_mrx ); - err = snd_card_asihpi_pcm_new(asihpi, 0, pcm_substreams); + err = snd_card_asihpi_pcm_new(asihpi, 0); if (err < 0) { snd_printk(KERN_ERR "pcm_new failed\n"); goto __nodev; @@ -2939,13 +2928,14 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, strcpy(card->driver, "ASIHPI"); - sprintf(card->shortname, "AudioScience ASI%4X", asihpi->type); + sprintf(card->shortname, "AudioScience ASI%4X", + asihpi->hpi->adapter->type); sprintf(card->longname, "%s %i", - card->shortname, asihpi->adapter_index); + card->shortname, adapter_index); err = snd_card_register(card); if (!err) { - hpi_card->snd_card_asihpi = card; + hpi->snd_card = card; dev++; return 0; } @@ -2958,10 +2948,9 @@ __nodev: static void __devexit snd_asihpi_remove(struct pci_dev *pci_dev) { - struct hpi_adapter *hpi_card = pci_get_drvdata(pci_dev); - - snd_card_free(hpi_card->snd_card_asihpi); - hpi_card->snd_card_asihpi = NULL; + struct hpi_adapter *hpi = pci_get_drvdata(pci_dev); + snd_card_free(hpi->snd_card); + hpi->snd_card = NULL; asihpi_adapter_remove(pci_dev); } diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c index 278bec8..2414d7a 100644 --- a/sound/pci/asihpi/hpi6000.c +++ b/sound/pci/asihpi/hpi6000.c @@ -231,6 +231,8 @@ static void subsys_message(struct hpi_message *phm, struct hpi_response *phr) static void control_message(struct hpi_adapter_obj *pao, struct hpi_message *phm, struct hpi_response *phr) { + struct hpi_hw_obj *phw = pao->priv; + switch (phm->function) { case HPI_CONTROL_GET_STATE: if (pao->has_control_cache) { @@ -248,17 +250,14 @@ static void control_message(struct hpi_adapter_obj *pao, break; } - if (hpi_check_control_cache(((struct hpi_hw_obj *) - pao->priv)->p_cache, phm, - phr)) + if (hpi_check_control_cache(phw->p_cache, phm, phr)) break; } hw_message(pao, phm, phr); break; case HPI_CONTROL_SET_STATE: hw_message(pao, phm, phr); - hpi_cmn_control_cache_sync_to_msg(((struct hpi_hw_obj *)pao-> - priv)->p_cache, phm, phr); + hpi_cmn_control_cache_sync_to_msg(phw->p_cache, phm, phr); break; case HPI_CONTROL_GET_INFO: @@ -451,11 +450,11 @@ static void subsys_create_adapter(struct hpi_message *phm, } for (dsp_index = 0; dsp_index < MAX_DSPS; dsp_index++) { - struct hpi_hw_obj *phw = (struct hpi_hw_obj *)pao->priv; + struct hpi_hw_obj *phw = pao->priv; phw->ado[dsp_index].pa_parent_adapter = pao; } - phr->u.s.adapter_type = ao.adapter_type; + phr->u.s.adapter_type = ao.type; phr->u.s.adapter_index = ao.index; phr->error = 0; } @@ -476,7 +475,7 @@ static short create_adapter_obj(struct hpi_adapter_obj *pao, u32 dsp_index = 0; u32 control_cache_size = 0; u32 control_cache_count = 0; - struct hpi_hw_obj *phw = (struct hpi_hw_obj *)pao->priv; + struct hpi_hw_obj *phw = pao->priv; /* The PCI2040 has the following address map */ /* BAR0 - 4K = HPI control and status registers on PCI2040 (HPI CSR) */ @@ -559,7 +558,7 @@ static short create_adapter_obj(struct hpi_adapter_obj *pao, if (error) return error; } - pao->adapter_type = hr0.u.ax.info.adapter_type; + pao->type = hr0.u.ax.info.adapter_type; pao->index = hr0.u.ax.info.adapter_index; } @@ -584,9 +583,8 @@ static short create_adapter_obj(struct hpi_adapter_obj *pao, pao->has_control_cache = 1; } - HPI_DEBUG_LOG(DEBUG, "get adapter info ASI%04X index %d\n", - pao->adapter_type, pao->index); - pao->open = 0; /* upon creation the adapter is closed */ + HPI_DEBUG_LOG(DEBUG, "get adapter info ASI%04X index %d\n", pao->type, + pao->index); if (phw->p_cache) phw->p_cache->adap_idx = pao->index; @@ -596,7 +594,7 @@ static short create_adapter_obj(struct hpi_adapter_obj *pao, static void delete_adapter_obj(struct hpi_adapter_obj *pao) { - struct hpi_hw_obj *phw = (struct hpi_hw_obj *)pao->priv; + struct hpi_hw_obj *phw = pao->priv; if (pao->has_control_cache) hpi_free_control_cache(phw->p_cache); @@ -639,7 +637,7 @@ static void adapter_get_asserts(struct hpi_adapter_obj *pao, static short hpi6000_adapter_boot_load_dsp(struct hpi_adapter_obj *pao, u32 *pos_error_code) { - struct hpi_hw_obj *phw = (struct hpi_hw_obj *)pao->priv; + struct hpi_hw_obj *phw = pao->priv; short error; u32 timeout; u32 read = 0; @@ -1220,8 +1218,8 @@ static void hpi_read_block(struct dsp_obj *pdo, u32 address, u32 *pdata, static u16 hpi6000_dsp_block_write32(struct hpi_adapter_obj *pao, u16 dsp_index, u32 hpi_address, u32 *source, u32 count) { - struct dsp_obj *pdo = - &(*(struct hpi_hw_obj *)pao->priv).ado[dsp_index]; + struct hpi_hw_obj *phw = pao->priv; + struct dsp_obj *pdo = &phw->ado[dsp_index]; u32 time_out = PCI_TIMEOUT; int c6711_burst_size = 128; u32 local_hpi_address = hpi_address; @@ -1258,8 +1256,8 @@ static u16 hpi6000_dsp_block_write32(struct hpi_adapter_obj *pao, static u16 hpi6000_dsp_block_read32(struct hpi_adapter_obj *pao, u16 dsp_index, u32 hpi_address, u32 *dest, u32 count) { - struct dsp_obj *pdo = - &(*(struct hpi_hw_obj *)pao->priv).ado[dsp_index]; + struct hpi_hw_obj *phw = pao->priv; + struct dsp_obj *pdo = &phw->ado[dsp_index]; u32 time_out = PCI_TIMEOUT; int c6711_burst_size = 16; u32 local_hpi_address = hpi_address; @@ -1298,7 +1296,7 @@ static u16 hpi6000_dsp_block_read32(struct hpi_adapter_obj *pao, static short hpi6000_message_response_sequence(struct hpi_adapter_obj *pao, u16 dsp_index, struct hpi_message *phm, struct hpi_response *phr) { - struct hpi_hw_obj *phw = (struct hpi_hw_obj *)pao->priv; + struct hpi_hw_obj *phw = pao->priv; struct dsp_obj *pdo = &phw->ado[dsp_index]; u32 timeout; u16 ack; @@ -1414,8 +1412,8 @@ static short hpi6000_send_data_check_adr(u32 address, u32 length_in_dwords) static short hpi6000_send_data(struct hpi_adapter_obj *pao, u16 dsp_index, struct hpi_message *phm, struct hpi_response *phr) { - struct dsp_obj *pdo = - &(*(struct hpi_hw_obj *)pao->priv).ado[dsp_index]; + struct hpi_hw_obj *phw = pao->priv; + struct dsp_obj *pdo = &phw->ado[dsp_index]; u32 data_sent = 0; u16 ack; u32 length, address; @@ -1487,8 +1485,8 @@ static short hpi6000_send_data(struct hpi_adapter_obj *pao, u16 dsp_index, static short hpi6000_get_data(struct hpi_adapter_obj *pao, u16 dsp_index, struct hpi_message *phm, struct hpi_response *phr) { - struct dsp_obj *pdo = - &(*(struct hpi_hw_obj *)pao->priv).ado[dsp_index]; + struct hpi_hw_obj *phw = pao->priv; + struct dsp_obj *pdo = &phw->ado[dsp_index]; u32 data_got = 0; u16 ack; u32 length, address; @@ -1551,8 +1549,8 @@ static void hpi6000_send_dsp_interrupt(struct dsp_obj *pdo) static short hpi6000_send_host_command(struct hpi_adapter_obj *pao, u16 dsp_index, u32 host_cmd) { - struct dsp_obj *pdo = - &(*(struct hpi_hw_obj *)pao->priv).ado[dsp_index]; + struct hpi_hw_obj *phw = pao->priv; + struct dsp_obj *pdo = &phw->ado[dsp_index]; u32 timeout = TIMEOUT; /* set command */ @@ -1577,7 +1575,7 @@ static short hpi6000_check_PCI2040_error_flag(struct hpi_adapter_obj *pao, { u32 hPI_error; - struct hpi_hw_obj *phw = (struct hpi_hw_obj *)pao->priv; + struct hpi_hw_obj *phw = pao->priv; /* read the error bits from the PCI2040 */ hPI_error = ioread32(phw->dw2040_HPICSR + HPI_ERROR_REPORT); @@ -1597,8 +1595,8 @@ static short hpi6000_check_PCI2040_error_flag(struct hpi_adapter_obj *pao, static short hpi6000_wait_dsp_ack(struct hpi_adapter_obj *pao, u16 dsp_index, u32 ack_value) { - struct dsp_obj *pdo = - &(*(struct hpi_hw_obj *)pao->priv).ado[dsp_index]; + struct hpi_hw_obj *phw = pao->priv; + struct dsp_obj *pdo = &phw->ado[dsp_index]; u32 ack = 0L; u32 timeout; u32 hPIC = 0L; @@ -1640,7 +1638,7 @@ static short hpi6000_update_control_cache(struct hpi_adapter_obj *pao, struct hpi_message *phm) { const u16 dsp_index = 0; - struct hpi_hw_obj *phw = (struct hpi_hw_obj *)pao->priv; + struct hpi_hw_obj *phw = pao->priv; struct dsp_obj *pdo = &phw->ado[dsp_index]; u32 timeout; u32 cache_dirty_flag; @@ -1740,7 +1738,8 @@ static void hw_message(struct hpi_adapter_obj *pao, struct hpi_message *phm, { u16 error = 0; u16 dsp_index = 0; - u16 num_dsp = ((struct hpi_hw_obj *)pao->priv)->num_dsp; + struct hpi_hw_obj *phw = pao->priv; + u16 num_dsp = phw->num_dsp; if (num_dsp < 2) dsp_index = 0; diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 95d1cd5..e3d0f55 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -488,7 +488,7 @@ static void subsys_create_adapter(struct hpi_message *phm, return; } - phr->u.s.adapter_type = ao.adapter_type; + phr->u.s.adapter_type = ao.type; phr->u.s.adapter_index = ao.index; phr->error = 0; } @@ -503,7 +503,7 @@ static void adapter_delete(struct hpi_adapter_obj *pao, phr->error = HPI_ERROR_INVALID_OBJ_INDEX; return; } - phw = (struct hpi_hw_obj *)pao->priv; + phw = pao->priv; /* reset adapter h/w */ /* Reset C6713 #1 */ boot_loader_write_mem32(pao, 0, C6205_BAR0_TIMER1_CTL, 0); @@ -652,7 +652,7 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao, if (hr.error) return hr.error; - pao->adapter_type = hr.u.ax.info.adapter_type; + pao->type = hr.u.ax.info.adapter_type; pao->index = hr.u.ax.info.adapter_index; max_streams = @@ -665,8 +665,6 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao, hr.u.ax.info.serial_number); } - pao->open = 0; /* upon creation the adapter is closed */ - if (phw->p_cache) phw->p_cache->adap_idx = pao->index; diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c index 358853a..c54a49f 100644 --- a/sound/pci/asihpi/hpicmn.c +++ b/sound/pci/asihpi/hpicmn.c @@ -68,7 +68,7 @@ u16 hpi_validate_response(struct hpi_message *phm, struct hpi_response *phr) u16 hpi_add_adapter(struct hpi_adapter_obj *pao) { u16 retval = 0; - /*HPI_ASSERT(pao->wAdapterType); */ + /*HPI_ASSERT(pao->type); */ hpios_alistlock_lock(&adapters); @@ -77,13 +77,13 @@ u16 hpi_add_adapter(struct hpi_adapter_obj *pao) goto unlock; } - if (adapters.adapter[pao->index].adapter_type) { + if (adapters.adapter[pao->index].type) { int a; for (a = HPI_MAX_ADAPTERS - 1; a >= 0; a--) { - if (!adapters.adapter[a].adapter_type) { + if (!adapters.adapter[a].type) { HPI_DEBUG_LOG(WARNING, "ASI%X duplicate index %d moved to %d\n", - pao->adapter_type, pao->index, a); + pao->type, pao->index, a); pao->index = a; break; } @@ -104,13 +104,13 @@ unlock: void hpi_delete_adapter(struct hpi_adapter_obj *pao) { - if (!pao->adapter_type) { + if (!pao->type) { HPI_DEBUG_LOG(ERROR, "removing null adapter?\n"); return; } hpios_alistlock_lock(&adapters); - if (adapters.adapter[pao->index].adapter_type) + if (adapters.adapter[pao->index].type) adapters.gw_num_adapters--; memset(&adapters.adapter[pao->index], 0, sizeof(adapters.adapter[0])); hpios_alistlock_unlock(&adapters); @@ -132,7 +132,7 @@ struct hpi_adapter_obj *hpi_find_adapter(u16 adapter_index) } pao = &adapters.adapter[adapter_index]; - if (pao->adapter_type != 0) { + if (pao->type != 0) { /* HPI_DEBUG_LOG(VERBOSE, "Found adapter index %d\n", wAdapterIndex); @@ -165,7 +165,7 @@ static void subsys_get_adapter(struct hpi_message *phm, /* find the nCount'th nonzero adapter in array */ for (index = 0; index < HPI_MAX_ADAPTERS; index++) { - if (adapters.adapter[index].adapter_type) { + if (adapters.adapter[index].type) { if (!count) break; count--; @@ -174,11 +174,11 @@ static void subsys_get_adapter(struct hpi_message *phm, if (index < HPI_MAX_ADAPTERS) { phr->u.s.adapter_index = adapters.adapter[index].index; - phr->u.s.adapter_type = adapters.adapter[index].adapter_type; + phr->u.s.adapter_type = adapters.adapter[index].type; } else { phr->u.s.adapter_index = 0; phr->u.s.adapter_type = 0; - phr->error = HPI_ERROR_BAD_ADAPTER_NUMBER; + phr->error = HPI_ERROR_INVALID_OBJ_INDEX; } } diff --git a/sound/pci/asihpi/hpicmn.h b/sound/pci/asihpi/hpicmn.h index 67cc1b0..e441212 100644 --- a/sound/pci/asihpi/hpicmn.h +++ b/sound/pci/asihpi/hpicmn.h @@ -18,12 +18,15 @@ */ +struct hpi_adapter_obj; + +/* a function that takes an adapter obj and returns an int */ +typedef int adapter_int_func(struct hpi_adapter_obj *pao); + struct hpi_adapter_obj { struct hpi_pci pci; /* PCI info - bus#,dev#,address etc */ - u16 adapter_type; /* ASI6701 etc */ - u16 index; /* */ - u16 open; /* =1 when adapter open */ - u16 mixer_open; + u16 type; /* 0x6644 == ASI6644 etc */ + u16 index; struct hpios_spinlock dsp_lock; diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 75f7a2d..6091562 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -66,9 +66,7 @@ static struct hpi_adapter adapters[HPI_MAX_ADAPTERS]; static void hpi_send_recv_f(struct hpi_message *phm, struct hpi_response *phr, struct file *file) { - int adapter = phm->adapter_index; - - if ((adapter >= HPI_MAX_ADAPTERS || adapter < 0) + if ((phm->adapter_index >= HPI_MAX_ADAPTERS) && (phm->object != HPI_OBJ_SUBSYSTEM)) phr->error = HPI_ERROR_INVALID_OBJ_INDEX; else @@ -179,19 +177,14 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) } else { u16 __user *ptr = NULL; u32 size = 0; - u32 adapter_present; /* -1=no data 0=read from user mem, 1=write to user mem */ int wrflag = -1; - struct hpi_adapter *pa; + struct hpi_adapter *pa = NULL; - if (hm->h.adapter_index < HPI_MAX_ADAPTERS) { + if (hm->h.adapter_index < ARRAY_SIZE(adapters)) pa = &adapters[hm->h.adapter_index]; - adapter_present = pa->type; - } else { - adapter_present = 0; - } - if (!adapter_present) { + if (!pa || !pa->adapter || !pa->adapter->type) { hpi_init_response(&hr->r0, hm->h.object, hm->h.function, HPI_ERROR_BAD_ADAPTER_NUMBER); @@ -318,6 +311,7 @@ int __devinit asihpi_adapter_probe(struct pci_dev *pci_dev, const struct pci_device_id *pci_id) { int idx, nm; + int adapter_index; unsigned int memlen; struct hpi_message hm; struct hpi_response hr; @@ -346,8 +340,6 @@ int __devinit asihpi_adapter_probe(struct pci_dev *pci_dev, hm.adapter_index = HPI_ADAPTER_INDEX_INVALID; - adapter.pci = pci_dev; - nm = HPI_MAX_ADAPTER_MEM_SPACES; for (idx = 0; idx < nm; idx++) { @@ -356,18 +348,16 @@ int __devinit asihpi_adapter_probe(struct pci_dev *pci_dev, if (pci_resource_flags(pci_dev, idx) & IORESOURCE_MEM) { memlen = pci_resource_len(pci_dev, idx); - adapter.ap_remapped_mem_base[idx] = + pci.ap_mem_base[idx] = ioremap(pci_resource_start(pci_dev, idx), memlen); - if (!adapter.ap_remapped_mem_base[idx]) { + if (!pci.ap_mem_base[idx]) { HPI_DEBUG_LOG(ERROR, "ioremap failed, aborting\n"); /* unmap previously mapped pci mem space */ goto err; } } - - pci.ap_mem_base[idx] = adapter.ap_remapped_mem_base[idx]; } pci.pci_dev = pci_dev; @@ -379,6 +369,9 @@ int __devinit asihpi_adapter_probe(struct pci_dev *pci_dev, if (hr.error) goto err; + adapter_index = hr.u.s.adapter_index; + adapter.adapter = hpi_find_adapter(adapter_index); + if (prealloc_stream_buf) { adapter.p_buffer = vmalloc(prealloc_stream_buf); if (!adapter.p_buffer) { @@ -390,36 +383,32 @@ int __devinit asihpi_adapter_probe(struct pci_dev *pci_dev, } } - adapter.index = hr.u.s.adapter_index; - adapter.type = hr.u.s.adapter_type; - hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, HPI_ADAPTER_OPEN); - hm.adapter_index = adapter.index; + hm.adapter_index = adapter.adapter->index; hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL); if (hr.error) goto err; - adapter.snd_card_asihpi = NULL; /* WARNING can't init mutex in 'adapter' * and then copy it to adapters[] ?!?! */ - adapters[adapter.index] = adapter; - mutex_init(&adapters[adapter.index].mutex); - pci_set_drvdata(pci_dev, &adapters[adapter.index]); + adapters[adapter_index] = adapter; + mutex_init(&adapters[adapter_index].mutex); + pci_set_drvdata(pci_dev, &adapters[adapter_index]); dev_printk(KERN_INFO, &pci_dev->dev, - "probe succeeded for ASI%04X HPI index %d\n", adapter.type, - adapter.index); + "probe succeeded for ASI%04X HPI index %d\n", + adapter.adapter->type, adapter_index); return 0; err: for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) { - if (adapter.ap_remapped_mem_base[idx]) { - iounmap(adapter.ap_remapped_mem_base[idx]); - adapter.ap_remapped_mem_base[idx] = NULL; + if (pci.ap_mem_base[idx]) { + iounmap(pci.ap_mem_base[idx]); + pci.ap_mem_base[idx] = NULL; } } @@ -438,19 +427,20 @@ void __devexit asihpi_adapter_remove(struct pci_dev *pci_dev) struct hpi_message hm; struct hpi_response hr; struct hpi_adapter *pa; + struct hpi_pci pci; + pa = pci_get_drvdata(pci_dev); + pci = pa->adapter->pci; hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, HPI_ADAPTER_DELETE); - hm.adapter_index = pa->index; + hm.adapter_index = pa->adapter->index; hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL); /* unmap PCI memory space, mapped during device init. */ for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) { - if (pa->ap_remapped_mem_base[idx]) { - iounmap(pa->ap_remapped_mem_base[idx]); - pa->ap_remapped_mem_base[idx] = NULL; - } + if (pci.ap_mem_base[idx]) + iounmap(pci.ap_mem_base[idx]); } if (pa->p_buffer) @@ -462,7 +452,7 @@ void __devexit asihpi_adapter_remove(struct pci_dev *pci_dev) "remove %04x:%04x,%04x:%04x,%04x," " HPI index %d.\n", pci_dev->vendor, pci_dev->device, pci_dev->subsystem_vendor, pci_dev->subsystem_device, - pci_dev->devfn, pa->index); + pci_dev->devfn, pa->adapter->index); memset(pa, 0, sizeof(*pa)); } diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h index d59a059..c5cef11 100644 --- a/sound/pci/asihpi/hpios.h +++ b/sound/pci/asihpi/hpios.h @@ -149,20 +149,18 @@ static inline void cond_unlock(struct hpios_spinlock *l) #define hpios_alistlock_lock(obj) spin_lock(&((obj)->list_lock.lock)) #define hpios_alistlock_unlock(obj) spin_unlock(&((obj)->list_lock.lock)) +struct snd_card; + +/** pci drvdata points to an instance of this struct */ struct hpi_adapter { + struct hpi_adapter_obj *adapter; + struct snd_card *snd_card; + /* mutex prevents contention for one card between multiple user programs (via ioctl) */ struct mutex mutex; - u16 index; - u16 type; - - /* ALSA card structure */ - void *snd_card_asihpi; - char *p_buffer; size_t buffer_size; - struct pci_dev *pci; - void __iomem *ap_remapped_mem_base[HPI_MAX_ADAPTER_MEM_SPACES]; }; #endif -- cgit v1.1 From 50d5f773ecc42fec029530f0e2e22686ccdf0ac7 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:44 +1300 Subject: ALSA: asihpi - Simplify dsp code close. dsp_code struct is not created if firmware is invalid, so check and zero of firmware pointer is not necessary Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpidspcd.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/hpidspcd.c b/sound/pci/asihpi/hpidspcd.c index 21cdb9e..456a758 100644 --- a/sound/pci/asihpi/hpidspcd.c +++ b/sound/pci/asihpi/hpidspcd.c @@ -109,11 +109,8 @@ error1: /*-------------------------------------------------------------------*/ void hpi_dsp_code_close(struct dsp_code *dsp_code) { - if (dsp_code->pvt->firmware) { - HPI_DEBUG_LOG(DEBUG, "dsp code closed\n"); - release_firmware(dsp_code->pvt->firmware); - dsp_code->pvt->firmware = NULL; - } + HPI_DEBUG_LOG(DEBUG, "dsp code closed\n"); + release_firmware(dsp_code->pvt->firmware); kfree(dsp_code->pvt); } -- cgit v1.1 From 862e14185b46e1d65d560c7dbaa0f2c842b71d20 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:45 +1300 Subject: ALSA: asihpi - Add autofade query. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi.h | 3 +++ sound/pci/asihpi/hpifunc.c | 10 ++++++++++ 2 files changed, 13 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index 867c144..f2f1d98 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -1347,6 +1347,9 @@ u16 hpi_volume_auto_fade_profile(u32 h_control, short an_stop_gain0_01dB[HPI_MAX_CHANNELS], u32 duration_ms, u16 profile); +u16 hpi_volume_query_auto_fade_profile(const u32 h_control, const u32 i, + u16 *profile); + /*****************/ /* Level control */ /*****************/ diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index ebb568d..510e56c 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -2826,6 +2826,16 @@ u16 hpi_volume_auto_fade(u32 h_control, duration_ms, HPI_VOLUME_AUTOFADE_LOG); } +u16 hpi_volume_query_auto_fade_profile(const u32 h_volume, const u32 i, + u16 *profile) +{ + u16 e; + u32 u; + e = hpi_control_query(h_volume, HPI_VOLUME_AUTOFADE, i, 0, &u); + *profile = (u16)u; + return e; +} + u16 hpi_vox_set_threshold(u32 h_control, short an_gain0_01dB) { struct hpi_message hm; -- cgit v1.1 From 812550e9efd149062fdc6cc8bf00b5f15d0734b7 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:46 +1300 Subject: ALSA: asihpi - New defs and comments. Add new error codes, and adapter properties. Clean up some comments. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi.h | 48 ++++++++++++++++++++++++++++++++++-------------- 1 file changed, 34 insertions(+), 14 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index f2f1d98..2088724 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -431,7 +431,19 @@ Indicates that the adapter in it's current mode supports interrupts across the host bus. Note, this does not imply that interrupts are enabled. Instead it indicates that they can be enabled. */ - HPI_ADAPTER_PROPERTY_SUPPORTS_IRQ = 272 + HPI_ADAPTER_PROPERTY_SUPPORTS_IRQ = 272, +/** Readonly supports firmware updating. +Indicates that the adapter implements an interface to update firmware +on the adapter. +*/ + HPI_ADAPTER_PROPERTY_SUPPORTS_FW_UPDATE = 273, +/** Readonly Firmware IDs +Identifiy firmware independent of individual adapter type. +May be used as a filter for firmware update images. +Property 1 = Bootloader ID +Property 2 = Main program ID +*/ + HPI_ADAPTER_PROPERTY_FIRMWARE_ID = 274 }; /** Adapter mode commands @@ -619,7 +631,7 @@ enum HPI_MIXER_STORE_COMMAND { HPI_MIXER_STORE_ENABLE = 4, /** Disable auto storage of some control settings. */ HPI_MIXER_STORE_DISABLE = 5, -/** Save the attributes of a single control. */ +/** Unimplemented - save the attributes of a single control. */ HPI_MIXER_STORE_SAVE_SINGLE = 6 }; @@ -922,7 +934,7 @@ enum HPI_ERROR_CODES { HPI_ERROR_BAD_ADAPTER_NUMBER = 202, /** 2 adapters with the same adapter number. */ HPI_ERROR_DUPLICATE_ADAPTER_NUMBER = 203, - /** DSP code failed to bootload. (unused?) */ + /** DSP code failed to bootload. Usually a DSP memory test failure. */ HPI_ERROR_DSP_BOOTLOAD = 204, /** Couldn't find or open the DSP code file. */ HPI_ERROR_DSP_FILE_NOT_FOUND = 206, @@ -959,6 +971,9 @@ enum HPI_ERROR_CODES { HPI_ERROR_FLASH_VERIFY = 225, HPI_ERROR_FLASH_TYPE = 226, HPI_ERROR_FLASH_START = 227, + HPI_ERROR_FLASH_READ = 228, + HPI_ERROR_FLASH_READ_NO_FILE = 229, + HPI_ERROR_FLASH_SIZE = 230, /** Reserved for OEMs. */ HPI_ERROR_RESERVED_1 = 290, @@ -1001,6 +1016,8 @@ enum HPI_ERROR_CODES { HPI_ERROR_NO_INTERDSP_GROUPS = 315, /** Stream wait cancelled before threshold reached. */ HPI_ERROR_WAIT_CANCELLED = 316, + /** A character string is invalid. */ + HPI_ERROR_INVALID_STRING = 317, /** Invalid mixer node for this adapter. */ HPI_ERROR_INVALID_NODE = 400, @@ -1027,11 +1044,15 @@ enum HPI_ERROR_CODES { /** I2C */ HPI_ERROR_I2C_BAD_ADR = 460, - /** Entity errors */ + /** Entity type did not match requested type */ HPI_ERROR_ENTITY_TYPE_MISMATCH = 470, + /** Entity item count did not match requested count */ HPI_ERROR_ENTITY_ITEM_COUNT = 471, + /** Entity type is not one of the valid types */ HPI_ERROR_ENTITY_TYPE_INVALID = 472, + /** Entity role is not one of the valid roles */ HPI_ERROR_ENTITY_ROLE_INVALID = 473, + /** Entity size doesn't match target size */ HPI_ERROR_ENTITY_SIZE_MISMATCH = 474, /* AES18 specific errors were 500..507 */ @@ -1059,8 +1080,7 @@ enum HPI_ERROR_CODES { /** \defgroup maximums HPI maximum values \{ */ -/** Maximum number of adapters per HPI sub-system - WARNING: modifying this value changes the response structure size.*/ +/** Maximum number of PCI HPI adapters */ #define HPI_MAX_ADAPTERS 20 /** Maximum number of in or out streams per adapter */ #define HPI_MAX_STREAMS 16 @@ -1071,6 +1091,9 @@ enum HPI_ERROR_CODES { #define HPI_MAX_ANC_BYTES_PER_FRAME (64) #define HPI_STRING_LEN 16 +/** Networked adapters have index >= 100 */ +#define HPI_MIN_NETWORK_ADAPTER_IDX 100 + /** Velocity units */ #define HPI_OSTREAM_VELOCITY_UNITS 4096 /** OutStream timescale units */ @@ -1092,14 +1115,14 @@ enum HPI_ERROR_CODES { struct hpi_format { u32 sample_rate; /**< 11025, 32000, 44100 ... */ - u32 bit_rate; /**< for MPEG */ + u32 bit_rate; /**< for MPEG */ u32 attributes; /**< Stereo/JointStereo/Mono */ u16 mode_legacy; /**< Legacy ancillary mode or idle bit */ - u16 unused; /**< Unused */ - u16 channels; /**< 1,2..., (or ancillary mode or idle bit */ - u16 format; /**< HPI_FORMAT_PCM16, _MPEG etc. see #HPI_FORMATS. */ + u16 unused; /**< Unused */ + u16 channels; /**< 1,2..., (or ancillary mode or idle bit */ + u16 format; /**< HPI_FORMAT_PCM16, _MPEG etc. see #HPI_FORMATS. */ }; struct hpi_anc_frame { @@ -1125,9 +1148,6 @@ struct hpi_async_event { } u; }; -/* skip host side function declarations for - DSP compile and documentation extraction */ - #ifndef DISABLE_PRAGMA_PACK1 #pragma pack(pop) #endif @@ -1338,7 +1358,7 @@ u16 hpi_volume_get_mute(u32 h_control, u32 *mute); u16 hpi_volume_query_range(u32 h_control, short *min_gain_01dB, short *max_gain_01dB, short *step_gain_01dB); -u16 hpi_volume_query_channels(const u32 h_volume, u32 *p_channels); +u16 hpi_volume_query_channels(const u32 h_control, u32 *p_channels); u16 hpi_volume_auto_fade(u32 h_control, short an_stop_gain0_01dB[HPI_MAX_CHANNELS], u32 duration_ms); -- cgit v1.1 From 4e225e2649660af1510aa06f20b32e1dcc45545e Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:47 +1300 Subject: ALSA: asihpi - Distinguish four different emif init errors. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi6205.c | 35 +++++++++++++++++++---------------- 1 file changed, 19 insertions(+), 16 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index e3d0f55..4f28738 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -45,18 +45,21 @@ #define HPI6205_ERROR_MSG_RESP_TIMEOUT 1016 /* initialization/bootload errors */ -#define HPI6205_ERROR_6205_NO_IRQ 1002 -#define HPI6205_ERROR_6205_INIT_FAILED 1003 -#define HPI6205_ERROR_6205_REG 1006 -#define HPI6205_ERROR_6205_DSPPAGE 1007 -#define HPI6205_ERROR_C6713_HPIC 1009 -#define HPI6205_ERROR_C6713_HPIA 1010 -#define HPI6205_ERROR_C6713_PLL 1011 -#define HPI6205_ERROR_DSP_INTMEM 1012 -#define HPI6205_ERROR_DSP_EXTMEM 1013 -#define HPI6205_ERROR_DSP_PLD 1014 -#define HPI6205_ERROR_6205_EEPROM 1017 -#define HPI6205_ERROR_DSP_EMIF 1018 +#define HPI6205_ERROR_6205_NO_IRQ 1002 +#define HPI6205_ERROR_6205_INIT_FAILED 1003 +#define HPI6205_ERROR_6205_REG 1006 +#define HPI6205_ERROR_6205_DSPPAGE 1007 +#define HPI6205_ERROR_C6713_HPIC 1009 +#define HPI6205_ERROR_C6713_HPIA 1010 +#define HPI6205_ERROR_C6713_PLL 1011 +#define HPI6205_ERROR_DSP_INTMEM 1012 +#define HPI6205_ERROR_DSP_EXTMEM 1013 +#define HPI6205_ERROR_DSP_PLD 1014 +#define HPI6205_ERROR_6205_EEPROM 1017 +#define HPI6205_ERROR_DSP_EMIF1 1018 +#define HPI6205_ERROR_DSP_EMIF2 1019 +#define HPI6205_ERROR_DSP_EMIF3 1020 +#define HPI6205_ERROR_DSP_EMIF4 1021 /*****************************************************************************/ /* for C6205 PCI i/f */ @@ -1612,7 +1615,7 @@ static u16 boot_loader_config_emif(struct hpi_adapter_obj *pao, int dsp_index) boot_loader_write_mem32(pao, dsp_index, 0x01800008, setting); if (setting != boot_loader_read_mem32(pao, dsp_index, 0x01800008)) - return HPI6205_ERROR_DSP_EMIF; + return HPI6205_ERROR_DSP_EMIF1; /* EMIF CE1 setup - 32 bit async. This is 6713 #1 HPI, */ /* which occupies D15..0. 6713 starts at 27MHz, so need */ @@ -1625,7 +1628,7 @@ static u16 boot_loader_config_emif(struct hpi_adapter_obj *pao, int dsp_index) boot_loader_write_mem32(pao, dsp_index, 0x01800004, setting); if (setting != boot_loader_read_mem32(pao, dsp_index, 0x01800004)) - return HPI6205_ERROR_DSP_EMIF; + return HPI6205_ERROR_DSP_EMIF2; /* EMIF CE2 setup - 32 bit async. This is 6713 #2 HPI, */ /* which occupies D15..0. 6713 starts at 27MHz, so need */ @@ -1637,7 +1640,7 @@ static u16 boot_loader_config_emif(struct hpi_adapter_obj *pao, int dsp_index) boot_loader_write_mem32(pao, dsp_index, 0x01800010, setting); if (setting != boot_loader_read_mem32(pao, dsp_index, 0x01800010)) - return HPI6205_ERROR_DSP_EMIF; + return HPI6205_ERROR_DSP_EMIF3; /* EMIF CE3 setup - 32 bit async. */ /* This is the PLD on the ASI5000 cards only */ @@ -1648,7 +1651,7 @@ static u16 boot_loader_config_emif(struct hpi_adapter_obj *pao, int dsp_index) boot_loader_write_mem32(pao, dsp_index, 0x01800014, setting); if (setting != boot_loader_read_mem32(pao, dsp_index, 0x01800014)) - return HPI6205_ERROR_DSP_EMIF; + return HPI6205_ERROR_DSP_EMIF4; /* set EMIF SDRAM control for 2Mx32 SDRAM (512x32x4 bank) */ /* need to use this else DSP code crashes? */ -- cgit v1.1 From f50efa2d9b10e32bf9ccd1a4692df3096512dcfc Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:48 +1300 Subject: ALSA: asihpi - Add HPI version to module description. It is useful to know the HPI version without having to load the module, in order to determine the matching firmware version. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 2402801..c94d5d5 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -23,6 +23,7 @@ */ #include "hpi_internal.h" +#include "hpi_version.h" #include "hpimsginit.h" #include "hpioctl.h" #include "hpicmn.h" @@ -46,7 +47,8 @@ MODULE_LICENSE("GPL"); MODULE_AUTHOR("AudioScience inc. "); -MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx"); +MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx " + HPI_VER_STRING); #if defined CONFIG_SND_DEBUG_VERBOSE /** -- cgit v1.1 From 8637bc94f6a36c138229ac1ea09faca343f48bd7 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:49 +1300 Subject: ALSA: asihpi - Correct headers in cached control responses. Previously, only payload and size were correct, sufficient for reading, but other fields produced spurious debug output. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpicmn.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c index c54a49f..7ed5c26 100644 --- a/sound/pci/asihpi/hpicmn.c +++ b/sound/pci/asihpi/hpicmn.c @@ -324,6 +324,8 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache, } phr->error = 0; + phr->specific_error = 0; + phr->version = 0; /* set the default response size */ response_size = @@ -531,8 +533,12 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache, found ? "Cached" : "Uncached", phm->adapter_index, pI->control_index, pI->control_type, phm->u.c.attribute); - if (found) + if (found) { phr->size = (u16)response_size; + phr->type = HPI_TYPE_RESPONSE; + phr->object = phm->object; + phr->function = phm->function; + } return found; } -- cgit v1.1 From c1d70dd9c44d7554b97f38b5ce8001d3cbe10f61 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:50 +1300 Subject: ALSA: asihpi - Use valid channel count in format enumeration. Since introduction of mono and low latency modes, fixed channel count of 2 is not always valid. Use reported max_channels instead. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 20 ++++++++------------ 1 file changed, 8 insertions(+), 12 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index c94d5d5..fdec4aa 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -961,14 +961,12 @@ static void snd_card_asihpi_playback_format(struct snd_card_asihpi *asihpi, for (format = HPI_FORMAT_PCM8_UNSIGNED; format <= HPI_FORMAT_PCM24_SIGNED; format++) { - err = hpi_format_create(&hpi_format, - 2, format, sample_rate, 128000, 0); + err = hpi_format_create(&hpi_format, asihpi->out_max_chans, + format, sample_rate, 128000, 0); if (!err) - err = hpi_outstream_query_format(h_stream, - &hpi_format); + err = hpi_outstream_query_format(h_stream, &hpi_format); if (!err && (hpi_to_alsa_formats[format] != -1)) - pcmhw->formats |= - (1ULL << hpi_to_alsa_formats[format]); + pcmhw->formats |= (1ULL << hpi_to_alsa_formats[format]); } } @@ -1141,14 +1139,12 @@ static void snd_card_asihpi_capture_format(struct snd_card_asihpi *asihpi, for (format = HPI_FORMAT_PCM8_UNSIGNED; format <= HPI_FORMAT_PCM24_SIGNED; format++) { - err = hpi_format_create(&hpi_format, 2, format, - sample_rate, 128000, 0); + err = hpi_format_create(&hpi_format, asihpi->in_max_chans, + format, sample_rate, 128000, 0); if (!err) - err = hpi_instream_query_format(h_stream, - &hpi_format); + err = hpi_instream_query_format(h_stream, &hpi_format); if (!err) - pcmhw->formats |= - (1ULL << hpi_to_alsa_formats[format]); + pcmhw->formats |= (1ULL << hpi_to_alsa_formats[format]); } } -- cgit v1.1 From 68d533932217c6b3da4ab9abb15ab79d3a79474c Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 22 Dec 2011 13:38:51 +1300 Subject: ALSA: asihpi - Fix format validity check. Sharing and not reinitialising static pcm_hardware struct resulted in stream format validity flags being incorrectly shared between cards. Fix and clarify by declaring locally and initialising in the open functions. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 69 +++++++++++++++++++++++------------------------ 1 file changed, 33 insertions(+), 36 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index fdec4aa..fd3926f 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -938,15 +938,15 @@ snd_card_asihpi_playback_pointer(struct snd_pcm_substream *substream) return ptr; } -static void snd_card_asihpi_playback_format(struct snd_card_asihpi *asihpi, - u32 h_stream, - struct snd_pcm_hardware *pcmhw) +static u64 snd_card_asihpi_playback_formats(struct snd_card_asihpi *asihpi, + u32 h_stream) { struct hpi_format hpi_format; u16 format; u16 err; u32 h_control; u32 sample_rate = 48000; + u64 formats = 0; /* on cards without SRC, must query at valid rate, * maybe set by external sync @@ -966,32 +966,24 @@ static void snd_card_asihpi_playback_format(struct snd_card_asihpi *asihpi, if (!err) err = hpi_outstream_query_format(h_stream, &hpi_format); if (!err && (hpi_to_alsa_formats[format] != -1)) - pcmhw->formats |= (1ULL << hpi_to_alsa_formats[format]); + formats |= (1ULL << hpi_to_alsa_formats[format]); } + return formats; } -static struct snd_pcm_hardware snd_card_asihpi_playback = { - .buffer_bytes_max = BUFFER_BYTES_MAX, - .period_bytes_min = PERIOD_BYTES_MIN, - .period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN, - .periods_min = PERIODS_MIN, - .periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN, - .fifo_size = 0, -}; - static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_card_asihpi_pcm *dpcm; struct snd_card_asihpi *card = snd_pcm_substream_chip(substream); + struct snd_pcm_hardware snd_card_asihpi_playback; int err; dpcm = kzalloc(sizeof(*dpcm), GFP_KERNEL); if (dpcm == NULL) return -ENOMEM; - err = - hpi_outstream_open(card->hpi->adapter->index, + err = hpi_outstream_open(card->hpi->adapter->index, substream->number, &dpcm->h_stream); hpi_handle_error(err); if (err) @@ -1013,13 +1005,19 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) runtime->private_data = dpcm; runtime->private_free = snd_card_asihpi_runtime_free; - snd_card_asihpi_playback.channels_max = card->out_max_chans; - snd_card_asihpi_playback.channels_min = card->out_min_chans; + memset(&snd_card_asihpi_playback, 0, sizeof(snd_card_asihpi_playback)); + snd_card_asihpi_playback.buffer_bytes_max = BUFFER_BYTES_MAX; + snd_card_asihpi_playback.period_bytes_min = PERIOD_BYTES_MIN; /*?snd_card_asihpi_playback.period_bytes_min = card->out_max_chans * 4096; */ - - snd_card_asihpi_playback_format(card, dpcm->h_stream, - &snd_card_asihpi_playback); + snd_card_asihpi_playback.period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN; + snd_card_asihpi_playback.periods_min = PERIODS_MIN; + snd_card_asihpi_playback.periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN; + /* snd_card_asihpi_playback.fifo_size = 0; */ + snd_card_asihpi_playback.channels_max = card->out_max_chans; + snd_card_asihpi_playback.channels_min = card->out_min_chans; + snd_card_asihpi_playback.formats = + snd_card_asihpi_playback_formats(card, dpcm->h_stream); snd_card_asihpi_pcm_samplerates(card, &snd_card_asihpi_playback); @@ -1116,15 +1114,15 @@ static int snd_card_asihpi_capture_prepare(struct snd_pcm_substream *substream) -static void snd_card_asihpi_capture_format(struct snd_card_asihpi *asihpi, - u32 h_stream, - struct snd_pcm_hardware *pcmhw) +static u64 snd_card_asihpi_capture_formats(struct snd_card_asihpi *asihpi, + u32 h_stream) { struct hpi_format hpi_format; u16 format; u16 err; u32 h_control; u32 sample_rate = 48000; + u64 formats = 0; /* on cards without SRC, must query at valid rate, maybe set by external sync */ @@ -1144,25 +1142,17 @@ static void snd_card_asihpi_capture_format(struct snd_card_asihpi *asihpi, if (!err) err = hpi_instream_query_format(h_stream, &hpi_format); if (!err) - pcmhw->formats |= (1ULL << hpi_to_alsa_formats[format]); + formats |= (1ULL << hpi_to_alsa_formats[format]); } + return formats; } - -static struct snd_pcm_hardware snd_card_asihpi_capture = { - .buffer_bytes_max = BUFFER_BYTES_MAX, - .period_bytes_min = PERIOD_BYTES_MIN, - .period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN, - .periods_min = PERIODS_MIN, - .periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN, - .fifo_size = 0, -}; - static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_card_asihpi *card = snd_pcm_substream_chip(substream); struct snd_card_asihpi_pcm *dpcm; + struct snd_pcm_hardware snd_card_asihpi_capture; int err; dpcm = kzalloc(sizeof(*dpcm), GFP_KERNEL); @@ -1189,10 +1179,17 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) runtime->private_data = dpcm; runtime->private_free = snd_card_asihpi_runtime_free; + memset(&snd_card_asihpi_capture, 0, sizeof(snd_card_asihpi_capture)); + snd_card_asihpi_capture.buffer_bytes_max = BUFFER_BYTES_MAX; + snd_card_asihpi_capture.period_bytes_min = PERIOD_BYTES_MIN; + snd_card_asihpi_capture.period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN; + snd_card_asihpi_capture.periods_min = PERIODS_MIN; + snd_card_asihpi_capture.periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN; + /* snd_card_asihpi_capture.fifo_size = 0; */ snd_card_asihpi_capture.channels_max = card->in_max_chans; snd_card_asihpi_capture.channels_min = card->in_min_chans; - snd_card_asihpi_capture_format(card, dpcm->h_stream, - &snd_card_asihpi_capture); + snd_card_asihpi_capture.formats = + snd_card_asihpi_capture_formats(card, dpcm->h_stream); snd_card_asihpi_pcm_samplerates(card, &snd_card_asihpi_capture); snd_card_asihpi_capture.info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | -- cgit v1.1 From 09904b9506e56579282a16ac38d313f2dd08d0f1 Mon Sep 17 00:00:00 2001 From: Li Peng Date: Wed, 28 Dec 2011 15:17:26 +0000 Subject: ALSA: hda_intel: Add Oaktrail identifiers Oaktrail has 0x8086, 0x080a - AZX_DRIVER_SCH Taken from the Meego patches for Oaktrail Signed-off-by: Li Peng Signed-off-by: Alan Cox Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4045b0c..c6c896b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2995,6 +2995,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_LPIB }, /* Poulsbo */ + { PCI_DEVICE(0x8086, 0x080a), + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE}, /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | -- cgit v1.1 From dcc2cf7507861476c874cc7974e5d4557a32475f Mon Sep 17 00:00:00 2001 From: Tim Yamin Date: Thu, 29 Dec 2011 18:50:56 +0000 Subject: ALSA: emu10k1 - add another Audigy 2 ZS ID 0x20051102 is an Audigy 2 ZS. Signed-off-by: Tim Yamin Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 6a3e567..7549240 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1480,6 +1480,18 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spdif_bug = 1, .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , + /* 0x20051102 also has SB0350 written on it, treated as Audigy 2 ZS by + Creative's Windows driver */ + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20051102, + .driver = "Audigy2", .name = "SB Audigy 2 ZS [SB0350a]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .ca0151_chip = 1, + .spk71 = 1, + .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ + .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20021102, .driver = "Audigy2", .name = "SB Audigy 2 ZS [SB0350]", .id = "Audigy2", -- cgit v1.1 From 716e5db48861be408f9bbb5b49c72818ba85e4d2 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 4 Jan 2012 10:12:54 +0100 Subject: ALSA: HDA: Use LPIB position fix for Oaktrail According to the thread on alsa-devel, the LPIB method is to prefer for Oaktrail controller chip. Reference: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-January/047800.html Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c6c896b..bdc4336 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2997,7 +2997,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_LPIB }, /* Poulsbo */ { PCI_DEVICE(0x8086, 0x080a), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE}, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_LPIB }, /* Oaktrail */ /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | -- cgit v1.1 From 40d03e63e91af8ddccdfd5a536cc2a6e51433e1d Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 2 Jan 2012 12:40:15 +0100 Subject: ALSA: HDA: Fix master control for Cirrus Logic 421X The control name "HP/Speakers" is non-standard, and since there is only one DAC on this chip there is no need for a virtual master anyway. Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index acfb645..9139558 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1779,30 +1779,19 @@ static int build_cs421x_output(struct hda_codec *codec) struct auto_pin_cfg *cfg = &spec->autocfg; struct snd_kcontrol *kctl; int err; - char *name = "HP/Speakers"; + char *name = "Master"; fix_volume_caps(codec, dac); - if (!spec->vmaster_sw) { - err = add_vmaster(codec, dac); - if (err < 0) - return err; - } err = add_mute(codec, name, 0, HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); if (err < 0) return err; - err = snd_ctl_add_slave(spec->vmaster_sw, kctl); - if (err < 0) - return err; err = add_volume(codec, name, 0, HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); if (err < 0) return err; - err = snd_ctl_add_slave(spec->vmaster_vol, kctl); - if (err < 0) - return err; if (cfg->speaker_outs) { err = snd_hda_ctl_add(codec, 0, -- cgit v1.1 From 78e2a928e377d5124932d4399c6c581908b027a0 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 2 Jan 2012 12:40:16 +0100 Subject: ALSA: HDA: Fix automute for Cirrus Logic 421x There was a bug in the automute logic causing speakers not to mute when headphones were plugged in. Cc: stable@kernel.org Tested-by: Hsin-Yi Chen Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 9139558..ea66042 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -922,16 +922,14 @@ static void cs_automute(struct hda_codec *codec) /* mute speakers if spdif or hp jack is plugged in */ for (i = 0; i < cfg->speaker_outs; i++) { + int pin_ctl = hp_present ? 0 : PIN_OUT; + /* detect on spdif is specific to CS421x */ + if (spdif_present && (spec->vendor_nid == CS421X_VENDOR_NID)) + pin_ctl = 0; + nid = cfg->speaker_pins[i]; snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - hp_present ? 0 : PIN_OUT); - /* detect on spdif is specific to CS421x */ - if (spec->vendor_nid == CS421X_VENDOR_NID) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - spdif_present ? 0 : PIN_OUT); - } + AC_VERB_SET_PIN_WIDGET_CONTROL, pin_ctl); } if (spec->gpio_eapd_hp) { unsigned int gpio = hp_present ? -- cgit v1.1 From 5660ffd06935e564404412997a703279e325fa64 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 2 Jan 2012 12:40:17 +0100 Subject: ALSA: HDA: Add support for Cirrus Logic 4213 The CS4213 chip is similar to the CS4210, but it does not have SPDIF capabilities. Also, it has fewer pins, and the vendor specific nid is different. With this patch, we have working inputs and outputs (and automute/autoswitch). However, we don't know anything about the vendor specific processing coefficients, so we don't read or write to that node in this patch. BugLink: https://bugs.launchpad.net/bugs/910792 Tested-by: Hsin-Yi Chen Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 105 ++++++++++++++++++++++++++++--------------- 1 file changed, 70 insertions(+), 35 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index ea66042..036056c4 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -138,7 +138,7 @@ enum { */ #define CS4210_DAC_NID 0x02 #define CS4210_ADC_NID 0x03 -#define CS421X_VENDOR_NID 0x0B +#define CS4210_VENDOR_NID 0x0B #define CS421X_DMIC_PIN_NID 0x09 /* Port E */ #define CS421X_SPDIF_PIN_NID 0x0A /* Port H */ @@ -149,6 +149,10 @@ enum { #define SPDIF_EVENT 0x04 +/* Cirrus Logic CS4213 is like CS4210 but does not have SPDIF input/output */ +#define CS4213_VENDOR_NID 0x09 + + static inline int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx) { struct cs_spec *spec = codec->spec; @@ -923,8 +927,8 @@ static void cs_automute(struct hda_codec *codec) /* mute speakers if spdif or hp jack is plugged in */ for (i = 0; i < cfg->speaker_outs; i++) { int pin_ctl = hp_present ? 0 : PIN_OUT; - /* detect on spdif is specific to CS421x */ - if (spdif_present && (spec->vendor_nid == CS421X_VENDOR_NID)) + /* detect on spdif is specific to CS4210 */ + if (spdif_present && (spec->vendor_nid == CS4210_VENDOR_NID)) pin_ctl = 0; nid = cfg->speaker_pins[i]; @@ -938,8 +942,8 @@ static void cs_automute(struct hda_codec *codec) AC_VERB_SET_GPIO_DATA, gpio); } - /* specific to CS421x */ - if (spec->vendor_nid == CS421X_VENDOR_NID) { + /* specific to CS4210 */ + if (spec->vendor_nid == CS4210_VENDOR_NID) { /* mute HPs if spdif jack (SENSE_B) is present */ for (i = 0; i < cfg->hp_outs; i++) { nid = cfg->hp_pins[i]; @@ -976,7 +980,12 @@ static void cs_automic(struct hda_codec *codec) present = snd_hda_jack_detect(codec, nid); /* specific to CS421x, single ADC */ - if (spec->vendor_nid == CS421X_VENDOR_NID) { + if (spec->vendor_nid == CS420X_VENDOR_NID) { + if (present) + change_cur_input(codec, spec->automic_idx, 0); + else + change_cur_input(codec, !spec->automic_idx, 0); + } else { if (present) { spec->last_input = spec->cur_input; spec->cur_input = spec->automic_idx; @@ -984,11 +993,6 @@ static void cs_automic(struct hda_codec *codec) spec->cur_input = spec->last_input; } cs_update_input_select(codec); - } else { - if (present) - change_cur_input(codec, spec->automic_idx, 0); - else - change_cur_input(codec, !spec->automic_idx, 0); } } @@ -1070,15 +1074,8 @@ static void init_input(struct hda_codec *codec) if (spec->mic_detect && spec->automic_idx == i) snd_hda_jack_detect_enable(codec, pin, MIC_EVENT); } - /* specific to CS421x */ - if (spec->vendor_nid == CS421X_VENDOR_NID) { - if (spec->mic_detect) - cs_automic(codec); - else { - spec->cur_adc = spec->adc_nid[spec->cur_input]; - cs_update_input_select(codec); - } - } else { + /* CS420x has multiple ADC, CS421x has single ADC */ + if (spec->vendor_nid == CS420X_VENDOR_NID) { change_cur_input(codec, spec->cur_input, 1); if (spec->mic_detect) cs_automic(codec); @@ -1092,6 +1089,13 @@ static void init_input(struct hda_codec *codec) * selected in IDX_SPDIF_CTL. */ cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + } else { + if (spec->mic_detect) + cs_automic(codec); + else { + spec->cur_adc = spec->adc_nid[spec->cur_input]; + cs_update_input_select(codec); + } } } @@ -1565,7 +1569,7 @@ static const struct snd_kcontrol_new cs421x_speaker_bost_ctl = { .tlv = { .p = cs421x_speaker_boost_db_scale }, }; -static void cs421x_pinmux_init(struct hda_codec *codec) +static void cs4210_pinmux_init(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; unsigned int def_conf, coef; @@ -1620,10 +1624,11 @@ static int cs421x_init(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; - snd_hda_sequence_write(codec, cs421x_coef_init_verbs); - snd_hda_sequence_write(codec, cs421x_coef_init_verbs_A1_silicon_fixes); - - cs421x_pinmux_init(codec); + if (spec->vendor_nid == CS4210_VENDOR_NID) { + snd_hda_sequence_write(codec, cs421x_coef_init_verbs); + snd_hda_sequence_write(codec, cs421x_coef_init_verbs_A1_silicon_fixes); + cs4210_pinmux_init(codec); + } if (spec->gpio_mask) { snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, @@ -1791,7 +1796,7 @@ static int build_cs421x_output(struct hda_codec *codec) if (err < 0) return err; - if (cfg->speaker_outs) { + if (cfg->speaker_outs && (spec->vendor_nid == CS4210_VENDOR_NID)) { err = snd_hda_ctl_add(codec, 0, snd_ctl_new1(&cs421x_speaker_bost_ctl, codec)); if (err < 0) @@ -1888,6 +1893,7 @@ static int cs421x_parse_auto_config(struct hda_codec *codec) */ static int cs421x_suspend(struct hda_codec *codec, pm_message_t state) { + struct cs_spec *spec = codec->spec; unsigned int coef; snd_hda_shutup_pins(codec); @@ -1897,15 +1903,17 @@ static int cs421x_suspend(struct hda_codec *codec, pm_message_t state) snd_hda_codec_write(codec, CS4210_ADC_NID, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); - coef |= 0x0004; /* PDREF */ - cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + if (spec->vendor_nid == CS4210_VENDOR_NID) { + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + coef |= 0x0004; /* PDREF */ + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + } return 0; } #endif -static struct hda_codec_ops cs4210_patch_ops = { +static struct hda_codec_ops cs421x_patch_ops = { .build_controls = cs421x_build_controls, .build_pcms = cs_build_pcms, .init = cs421x_init, @@ -1916,7 +1924,7 @@ static struct hda_codec_ops cs4210_patch_ops = { #endif }; -static int patch_cs421x(struct hda_codec *codec) +static int patch_cs4210(struct hda_codec *codec) { struct cs_spec *spec; int err; @@ -1926,7 +1934,7 @@ static int patch_cs421x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - spec->vendor_nid = CS421X_VENDOR_NID; + spec->vendor_nid = CS4210_VENDOR_NID; spec->board_config = snd_hda_check_board_config(codec, CS421X_MODELS, @@ -1954,14 +1962,39 @@ static int patch_cs421x(struct hda_codec *codec) is auto-parsed. If GPIO or SENSE_B is forced, DMIC input is disabled. */ - cs421x_pinmux_init(codec); + cs4210_pinmux_init(codec); err = cs421x_parse_auto_config(codec); if (err < 0) goto error; - codec->patch_ops = cs4210_patch_ops; + codec->patch_ops = cs421x_patch_ops; + + return 0; + + error: + kfree(codec->spec); + codec->spec = NULL; + return err; +} + +static int patch_cs4213(struct hda_codec *codec) +{ + struct cs_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + spec->vendor_nid = CS4213_VENDOR_NID; + + err = cs421x_parse_auto_config(codec); + if (err < 0) + goto error; + codec->patch_ops = cs421x_patch_ops; return 0; error: @@ -1977,13 +2010,15 @@ static int patch_cs421x(struct hda_codec *codec) static const struct hda_codec_preset snd_hda_preset_cirrus[] = { { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x }, { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x }, - { .id = 0x10134210, .name = "CS4210", .patch = patch_cs421x }, + { .id = 0x10134210, .name = "CS4210", .patch = patch_cs4210 }, + { .id = 0x10134213, .name = "CS4213", .patch = patch_cs4213 }, {} /* terminator */ }; MODULE_ALIAS("snd-hda-codec-id:10134206"); MODULE_ALIAS("snd-hda-codec-id:10134207"); MODULE_ALIAS("snd-hda-codec-id:10134210"); +MODULE_ALIAS("snd-hda-codec-id:10134213"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Cirrus Logic HD-audio codec"); -- cgit v1.1 From 2267ea9762c7b0080d5747726f95cdd32d521361 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 3 Jan 2012 08:45:56 +0100 Subject: ALSA: HDA: Fix typo for ALC269VB_FIXUP_DMIC This fixup is not actually used, so in practice this is just a cosmetic fix. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d24adbd..5e82acf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5320,7 +5320,7 @@ static const struct alc_fixup alc269_fixups[] = { { } }, }, - [ALC269_FIXUP_DMIC] = { + [ALC269VB_FIXUP_DMIC] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { { 0x12, 0x99a3092f }, /* int-mic */ -- cgit v1.1 From f16c2cc3c40ea8b7860f3abb9c7bb887a1bdd703 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 4 Jan 2012 10:48:27 +0100 Subject: ALSA: HDA: Remove Poulsbo position fix quirks Now that we have changed the poulsbo chip to use LPIB position fix, we can remove the individual machine quirks that do the same thing. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bdc4336..8a46450 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2493,12 +2493,10 @@ static int azx_dev_free(struct snd_device *device) static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1028, 0x02c6, "Dell Inspiron 1010", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS 1101HA", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), -- cgit v1.1 From e7848163aa2a649d9065f230fadff80dc3519775 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Thu, 5 Jan 2012 23:05:18 +0100 Subject: ALSA: ice1724 - Check for ac97 to avoid kernel oops Cards with identical PCI ids but no AC97 config in EEPROM do not have the ac97 field initialized. We must check for this case to avoid kernel oops. Signed-off-by: Pavel Hofman Cc: Signed-off-by: Takashi Iwai --- sound/pci/ice1712/amp.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c index e328cfb..e525da2 100644 --- a/sound/pci/ice1712/amp.c +++ b/sound/pci/ice1712/amp.c @@ -68,8 +68,11 @@ static int __devinit snd_vt1724_amp_init(struct snd_ice1712 *ice) static int __devinit snd_vt1724_amp_add_controls(struct snd_ice1712 *ice) { - /* we use pins 39 and 41 of the VT1616 for left and right read outputs */ - snd_ac97_write_cache(ice->ac97, 0x5a, snd_ac97_read(ice->ac97, 0x5a) & ~0x8000); + if (ice->ac97) + /* we use pins 39 and 41 of the VT1616 for left and right + read outputs */ + snd_ac97_write_cache(ice->ac97, 0x5a, + snd_ac97_read(ice->ac97, 0x5a) & ~0x8000); return 0; } -- cgit v1.1 From 219e2cd41b5014c5a6ed4d1748f65f55f74a862f Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Thu, 5 Jan 2012 22:01:56 +0100 Subject: ALSA: ice1724 - External clock item only for cards with SPDIF_IN Append the external clock item to the clock list only if the SPDIF_IN capability is defined in the SPDIF register. Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 4dc5124..9e18d3a 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1825,7 +1825,12 @@ static int snd_vt1724_pro_internal_clock_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = hw_rates_count + ice->ext_clock_count; + /* internal clocks */ + uinfo->value.enumerated.items = hw_rates_count; + /* external clocks */ + if (ice->force_rdma1 || + (ice->eeprom.data[ICE_EEP2_SPDIF] & VT1724_CFG_SPDIF_IN)) + uinfo->value.enumerated.items += ice->ext_clock_count; /* upper limit - keep at top */ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; -- cgit v1.1 From 9489f2c63f4b59e0cc1db6a5fd136df5cbb6fe0a Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Wed, 4 Jan 2012 15:42:44 +0800 Subject: ALSA: Au88x0 - Xtalk - fix write/read of eq and xt instates Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_xtalk.c | 64 ++++++++++++++++++++--------------------- 1 file changed, 32 insertions(+), 32 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/au88x0/au88x0_xtalk.c b/sound/pci/au88x0/au88x0_xtalk.c index b4151e2..e55312e 100644 --- a/sound/pci/au88x0/au88x0_xtalk.c +++ b/sound/pci/au88x0/au88x0_xtalk.c @@ -306,10 +306,10 @@ vortex_XtalkHw_SetLeftEQStates(vortex_t * vortex, hwwrite(vortex->mmio, 0x2421C + i * 0x24, coefs[i][2]); hwwrite(vortex->mmio, 0x24220 + i * 0x24, coefs[i][3]); } - hwwrite(vortex->mmio, 0x244F8 + i * 0x24, arg_0[0]); - hwwrite(vortex->mmio, 0x244FC + i * 0x24, arg_0[1]); - hwwrite(vortex->mmio, 0x24500 + i * 0x24, arg_0[2]); - hwwrite(vortex->mmio, 0x24504 + i * 0x24, arg_0[3]); + hwwrite(vortex->mmio, 0x244F8, arg_0[0]); + hwwrite(vortex->mmio, 0x244FC, arg_0[1]); + hwwrite(vortex->mmio, 0x24500, arg_0[2]); + hwwrite(vortex->mmio, 0x24504, arg_0[3]); } static void @@ -325,10 +325,10 @@ vortex_XtalkHw_SetRightEQStates(vortex_t * vortex, hwwrite(vortex->mmio, 0x242D0 + i * 0x24, coefs[i][2]); hwwrite(vortex->mmio, 0x244D4 + i * 0x24, coefs[i][3]); } - hwwrite(vortex->mmio, 0x24508 + i * 0x24, arg_0[0]); - hwwrite(vortex->mmio, 0x2450C + i * 0x24, arg_0[1]); - hwwrite(vortex->mmio, 0x24510 + i * 0x24, arg_0[2]); - hwwrite(vortex->mmio, 0x24514 + i * 0x24, arg_0[3]); + hwwrite(vortex->mmio, 0x24508, arg_0[0]); + hwwrite(vortex->mmio, 0x2450C, arg_0[1]); + hwwrite(vortex->mmio, 0x24510, arg_0[2]); + hwwrite(vortex->mmio, 0x24514, arg_0[3]); } static void @@ -344,10 +344,10 @@ vortex_XtalkHw_SetLeftXTStates(vortex_t * vortex, hwwrite(vortex->mmio, 0x24384 + i * 0x24, coefs[i][2]); hwwrite(vortex->mmio, 0x24388 + i * 0x24, coefs[i][3]); } - hwwrite(vortex->mmio, 0x24518 + i * 0x24, arg_0[0]); - hwwrite(vortex->mmio, 0x2451C + i * 0x24, arg_0[1]); - hwwrite(vortex->mmio, 0x24520 + i * 0x24, arg_0[2]); - hwwrite(vortex->mmio, 0x24524 + i * 0x24, arg_0[3]); + hwwrite(vortex->mmio, 0x24518, arg_0[0]); + hwwrite(vortex->mmio, 0x2451C, arg_0[1]); + hwwrite(vortex->mmio, 0x24520, arg_0[2]); + hwwrite(vortex->mmio, 0x24524, arg_0[3]); } static void @@ -363,10 +363,10 @@ vortex_XtalkHw_SetRightXTStates(vortex_t * vortex, hwwrite(vortex->mmio, 0x24438 + i * 0x24, coefs[i][2]); hwwrite(vortex->mmio, 0x2443C + i * 0x24, coefs[i][3]); } - hwwrite(vortex->mmio, 0x24528 + i * 0x24, arg_0[0]); - hwwrite(vortex->mmio, 0x2452C + i * 0x24, arg_0[1]); - hwwrite(vortex->mmio, 0x24530 + i * 0x24, arg_0[2]); - hwwrite(vortex->mmio, 0x24534 + i * 0x24, arg_0[3]); + hwwrite(vortex->mmio, 0x24528, arg_0[0]); + hwwrite(vortex->mmio, 0x2452C, arg_0[1]); + hwwrite(vortex->mmio, 0x24530, arg_0[2]); + hwwrite(vortex->mmio, 0x24534, arg_0[3]); } #if 0 @@ -450,10 +450,10 @@ vortex_XtalkHw_GetLeftEQStates(vortex_t * vortex, xtalk_instate_t arg_0, coefs[i][2] = hwread(vortex->mmio, 0x2421C + i * 0x24); coefs[i][3] = hwread(vortex->mmio, 0x24220 + i * 0x24); } - arg_0[0] = hwread(vortex->mmio, 0x244F8 + i * 0x24); - arg_0[1] = hwread(vortex->mmio, 0x244FC + i * 0x24); - arg_0[2] = hwread(vortex->mmio, 0x24500 + i * 0x24); - arg_0[3] = hwread(vortex->mmio, 0x24504 + i * 0x24); + arg_0[0] = hwread(vortex->mmio, 0x244F8); + arg_0[1] = hwread(vortex->mmio, 0x244FC); + arg_0[2] = hwread(vortex->mmio, 0x24500); + arg_0[3] = hwread(vortex->mmio, 0x24504); } static void @@ -468,10 +468,10 @@ vortex_XtalkHw_GetRightEQStates(vortex_t * vortex, xtalk_instate_t arg_0, coefs[i][2] = hwread(vortex->mmio, 0x242D0 + i * 0x24); coefs[i][3] = hwread(vortex->mmio, 0x242D4 + i * 0x24); } - arg_0[0] = hwread(vortex->mmio, 0x24508 + i * 0x24); - arg_0[1] = hwread(vortex->mmio, 0x2450C + i * 0x24); - arg_0[2] = hwread(vortex->mmio, 0x24510 + i * 0x24); - arg_0[3] = hwread(vortex->mmio, 0x24514 + i * 0x24); + arg_0[0] = hwread(vortex->mmio, 0x24508); + arg_0[1] = hwread(vortex->mmio, 0x2450C); + arg_0[2] = hwread(vortex->mmio, 0x24510); + arg_0[3] = hwread(vortex->mmio, 0x24514); } static void @@ -486,10 +486,10 @@ vortex_XtalkHw_GetLeftXTStates(vortex_t * vortex, xtalk_instate_t arg_0, coefs[i][2] = hwread(vortex->mmio, 0x24384 + i * 0x24); coefs[i][3] = hwread(vortex->mmio, 0x24388 + i * 0x24); } - arg_0[0] = hwread(vortex->mmio, 0x24518 + i * 0x24); - arg_0[1] = hwread(vortex->mmio, 0x2451C + i * 0x24); - arg_0[2] = hwread(vortex->mmio, 0x24520 + i * 0x24); - arg_0[3] = hwread(vortex->mmio, 0x24524 + i * 0x24); + arg_0[0] = hwread(vortex->mmio, 0x24518); + arg_0[1] = hwread(vortex->mmio, 0x2451C); + arg_0[2] = hwread(vortex->mmio, 0x24520); + arg_0[3] = hwread(vortex->mmio, 0x24524); } static void @@ -504,10 +504,10 @@ vortex_XtalkHw_GetRightXTStates(vortex_t * vortex, xtalk_instate_t arg_0, coefs[i][2] = hwread(vortex->mmio, 0x24438 + i * 0x24); coefs[i][3] = hwread(vortex->mmio, 0x2443C + i * 0x24); } - arg_0[0] = hwread(vortex->mmio, 0x24528 + i * 0x24); - arg_0[1] = hwread(vortex->mmio, 0x2452C + i * 0x24); - arg_0[2] = hwread(vortex->mmio, 0x24530 + i * 0x24); - arg_0[3] = hwread(vortex->mmio, 0x24534 + i * 0x24); + arg_0[0] = hwread(vortex->mmio, 0x24528); + arg_0[1] = hwread(vortex->mmio, 0x2452C); + arg_0[2] = hwread(vortex->mmio, 0x24530); + arg_0[3] = hwread(vortex->mmio, 0x24534); } #endif -- cgit v1.1 From 76474da05ff7b1729b36cd2b6d7aa6b1865d68f1 Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Wed, 4 Jan 2012 16:16:26 +0800 Subject: ALSA: Au88x0 - Fix Xtalk's constants - Fix XtalkGainsDefault, XtalkGains1Chn - Fix XtalkWideCoefsLeftEQ, XtalkWideCoefsRightEQ - Fix XtlakWideCoefsLeftXT, XtalkWideCoefsRightXT Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_xtalk.c | 87 ++++++++++++++++++++++++++--------------- 1 file changed, 56 insertions(+), 31 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/au88x0/au88x0_xtalk.c b/sound/pci/au88x0/au88x0_xtalk.c index e55312e..b278e28 100644 --- a/sound/pci/au88x0/au88x0_xtalk.c +++ b/sound/pci/au88x0/au88x0_xtalk.c @@ -48,43 +48,61 @@ static unsigned short const wXtalkNarrowLeftDelay = 0x7; static unsigned short const wXtalkNarrowRightDelay = 0x7; static xtalk_gains_t const asXtalkGainsDefault = { - 0x4000, 0x4000, 4000, 0x4000, 4000, 0x4000, 4000, 0x4000, 4000, - 0x4000 + 0x4000, 0x4000, 0x4000, 0x4000, 0x4000, + 0x4000, 0x4000, 0x4000, 0x4000, 0x4000 }; static xtalk_gains_t const asXtalkGainsTest = { - 0x8000, 0x7FFF, 0, 0xFFFF, 0x0001, 0xC000, 0x4000, 0xFFFE, 0x0002, - 0 + 0x7fff, 0x8000, 0x0000, 0x0000, 0x0001, + 0xffff, 0x4000, 0xc000, 0x0002, 0xfffe }; + static xtalk_gains_t const asXtalkGains1Chan = { - 0x7FFF, 0, 0, 0, 0x7FFF, 0, 0, 0, 0, 0 + 0x7FFF, 0, 0, 0, 0, + 0x7FFF, 0, 0, 0, 0, }; // Input gain for 4 A3D slices. One possible input pair is left zero. static xtalk_gains_t const asXtalkGainsAllChan = { - 0x7FFF, 0x7FFF, 0x7FFF, 0x7FFF, 0, 0x7FFF, 0x7FFF, 0x7FFF, 0x7FFF, - 0 - //0x7FFF,0x7FFF,0x7FFF,0x7FFF,0x7fff,0x7FFF,0x7FFF,0x7FFF,0x7FFF,0x7fff + 0x7FFF, 0x7FFF, 0x7FFF, 0x7FFF, 0, + 0x7FFF, 0x7FFF, 0x7FFF, 0x7FFF, 0 +}; + +static xtalk_gains_t const asXtalkGainsZeros = { + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }; -static xtalk_gains_t const asXtalkGainsZeros; -static xtalk_dline_t const alXtalkDlineZeros; +static xtalk_dline_t const alXtalkDlineZeros = { + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 +}; static xtalk_dline_t const alXtalkDlineTest = { - 0xFC18, 0x03E8FFFF, 0x186A0, 0x7960FFFE, 1, 0xFFFFFFFF, - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0x0000fc18, 0xfff03e8, 0x000186a0, 0xfffe7960, 1, 0xffffffff, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0 +}; + +static xtalk_instate_t const asXtalkInStateZeros = { 0, 0, 0, 0 }; -static xtalk_instate_t const asXtalkInStateZeros; -static xtalk_instate_t const asXtalkInStateTest = - { 0xFF80, 0x0080, 0xFFFF, 0x0001 }; -static xtalk_state_t const asXtalkOutStateZeros; +static xtalk_instate_t const asXtalkInStateTest = { + 0x0080, 0xff80, 0x0001, 0xffff +}; + +static xtalk_state_t const asXtalkOutStateZeros = { + {0, 0, 0, 0}, + {0, 0, 0, 0}, + {0, 0, 0, 0}, + {0, 0, 0, 0}, + {0, 0, 0, 0} +}; static short const sDiamondKLeftEq = 0x401d; static short const sDiamondKRightEq = 0x401d; static short const sDiamondKLeftXt = 0xF90E; static short const sDiamondKRightXt = 0xF90E; -static short const sDiamondShiftLeftEq = 1; /* 0xF90E Is this a bug ??? */ +static short const sDiamondShiftLeftEq = 1; static short const sDiamondShiftRightEq = 1; static short const sDiamondShiftLeftXt = 0; static short const sDiamondShiftRightXt = 0; @@ -94,29 +112,29 @@ static unsigned short const wDiamondRightDelay = 0xb; static xtalk_coefs_t const asXtalkWideCoefsLeftEq = { {0xEC4C, 0xDCE9, 0xFDC2, 0xFEEC, 0}, {0x5F60, 0xCBCB, 0xFC26, 0x0305, 0}, - {0x340B, 0xf504, 0x6CE8, 0x0D23, 0x00E4}, - {0xD500, 0x8D76, 0xACC7, 0x5B05, 0x00FA}, + {0x340B, 0xe8f5, 0x236c, 0xe40d, 0}, + {0x76d5, 0xc78d, 0x05ac, 0xfa5b, 0}, {0x7F04, 0xC0FA, 0x0263, 0xFDA2, 0} }; static xtalk_coefs_t const asXtalkWideCoefsRightEq = { {0xEC4C, 0xDCE9, 0xFDC2, 0xFEEC, 0}, {0x5F60, 0xCBCB, 0xFC26, 0x0305, 0}, - {0x340B, 0xF504, 0x6CE8, 0x0D23, 0x00E4}, - {0xD500, 0x8D76, 0xACC7, 0x5B05, 0x00FA}, + {0x340B, 0xe8f5, 0x236c, 0xe40d, 0}, + {0x76d5, 0xc78d, 0x05ac, 0xfa5b, 0}, {0x7F04, 0xC0FA, 0x0263, 0xFDA2, 0} }; static xtalk_coefs_t const asXtalkWideCoefsLeftXt = { - {0x86C3, 0x7B55, 0x89C3, 0x005B, 0x0047}, - {0x6000, 0x206A, 0xC6CA, 0x40FF, 0}, - {0x1100, 0x1164, 0xA1D7, 0x90FC, 0x0001}, - {0xDC00, 0x9E77, 0xB8C7, 0x0AFF, 0}, + {0x55c6, 0xc97b, 0x005b, 0x0047, 0}, + {0x6a60, 0xca20, 0xffc6, 0x0040, 0}, + {0x6411, 0xd711, 0xfca1, 0x0190, 0}, + {0x77dc, 0xc79e, 0xffb8, 0x000a, 0}, {0, 0, 0, 0, 0} }; static xtalk_coefs_t const asXtalkWideCoefsRightXt = { - {0x86C3, 0x7B55, 0x89C3, 0x005B, 0x0047}, - {0x6000, 0x206A, 0xC6CA, 0x40FF, 0}, - {0x1100, 0x1164, 0xA1D7, 0x90FC, 0x0001}, - {0xDC00, 0x9E77, 0xB8C7, 0x0AFF, 0}, + {0x55c6, 0xc97b, 0x005b, 0x0047, 0}, + {0x6a60, 0xca20, 0xffc6, 0x0040, 0}, + {0x6411, 0xd711, 0xfca1, 0x0190, 0}, + {0x77dc, 0xc79e, 0xffb8, 0x000a, 0}, {0, 0, 0, 0, 0} }; static xtalk_coefs_t const asXtalkNarrowCoefsLeftEq = { @@ -151,7 +169,14 @@ static xtalk_coefs_t const asXtalkNarrowCoefsRightXt = { {0, 0, 0, 0, 0} }; -static xtalk_coefs_t const asXtalkCoefsZeros; +static xtalk_coefs_t const asXtalkCoefsZeros = { + {0, 0, 0, 0, 0}, + {0, 0, 0, 0, 0}, + {0, 0, 0, 0, 0}, + {0, 0, 0, 0, 0}, + {0, 0, 0, 0, 0} +}; + static xtalk_coefs_t const asXtalkCoefsPipe = { {0, 0, 0x0FA0, 0, 0}, {0, 0, 0x0FA0, 0, 0}, @@ -186,7 +211,7 @@ static xtalk_coefs_t const asXtalkCoefsDenTest = { static xtalk_state_t const asXtalkOutStateTest = { {0x7FFF, 0x0004, 0xFFFC, 0}, {0xFE00, 0x0008, 0xFFF8, 0x4000}, - {0x200, 0x0010, 0xFFF0, 0xC000}, + {0x0200, 0x0010, 0xFFF0, 0xC000}, {0x8000, 0x0020, 0xFFE0, 0}, {0, 0, 0, 0} }; -- cgit v1.1 From 3ae4e1f7a0dab95f7e6049272cdb59c7bdc34365 Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Fri, 6 Jan 2012 09:19:29 +0800 Subject: ALSA: Au88x0 - Fix IRQ fifo error and channels swap of 4 channels playback Fix IRQ fifo error when playing stereo by set stereo flag of fifo control. This also fix the swap of front and rear channels on au8830. Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_core.c | 8 ++++---- sound/pci/au88x0/au88x0_pcm.c | 4 ++-- 2 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 4891503..d49e2c5 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -805,7 +805,7 @@ static void vortex_fifo_setadbvalid(vortex_t * vortex, int fifo, int en) } static void -vortex_fifo_setadbctrl(vortex_t * vortex, int fifo, int b, int priority, +vortex_fifo_setadbctrl(vortex_t * vortex, int fifo, int stereo, int priority, int empty, int valid, int f) { int temp, lifeboat = 0; @@ -837,7 +837,7 @@ vortex_fifo_setadbctrl(vortex_t * vortex, int fifo, int b, int priority, #else temp = (this_4 & 0x3f) << 0xc; #endif - temp = (temp & 0xfffffffd) | ((b & 1) << 1); + temp = (temp & 0xfffffffd) | ((stereo & 1) << 1); temp = (temp & 0xfffffff3) | ((priority & 3) << 2); temp = (temp & 0xffffffef) | ((valid & 1) << 4); temp |= FIFO_U1; @@ -1148,11 +1148,11 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma, static void vortex_adbdma_setmode(vortex_t * vortex, int adbdma, int ie, int dir, - int fmt, int d, u32 offset) + int fmt, int stereo, u32 offset) { stream_t *dma = &vortex->dma_adb[adbdma]; - dma->dma_unknown = d; + dma->dma_unknown = stereo; dma->dma_ctrl = ((offset & OFFSET_MASK) | (dma->dma_ctrl & ~OFFSET_MASK)); /* Enable PCMOUT interrupts. */ diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index c5f7ae4..5099690 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -307,8 +307,8 @@ static int snd_vortex_pcm_prepare(struct snd_pcm_substream *substream) fmt = vortex_alsafmt_aspfmt(runtime->format); spin_lock_irq(&chip->lock); if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) { - vortex_adbdma_setmode(chip, dma, 1, dir, fmt, 0 /*? */ , - 0); + vortex_adbdma_setmode(chip, dma, 1, dir, fmt, + runtime->channels == 1 ? 0 : 1, 0); vortex_adbdma_setstartbuffer(chip, dma, 0); if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_SPDIF) vortex_adb_setsrc(chip, dma, runtime->rate, dir); -- cgit v1.1 From fb65c2dfe60d38be6b9193d0b85e66e780cd4373 Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Sat, 7 Jan 2012 14:35:17 +0800 Subject: ALSA: Au88x0 - Fix channels swapping of 4 channels playback Fix channels swapping of 4 channels playback by using vortex_adbdma_stopfifo instead of vortex_adbdma_pausefifo for SNDRV_PCM_TRIGGER_STOP event Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_core.c | 2 -- sound/pci/au88x0/au88x0_pcm.c | 3 +-- 2 files changed, 1 insertion(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index d49e2c5..6933a27 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1336,7 +1336,6 @@ static void vortex_adbdma_pausefifo(vortex_t * vortex, int adbdma) dma->fifo_status = FIFO_PAUSE; } -#if 0 // Using pause instead static void vortex_adbdma_stopfifo(vortex_t * vortex, int adbdma) { stream_t *dma = &vortex->dma_adb[adbdma]; @@ -1351,7 +1350,6 @@ static void vortex_adbdma_stopfifo(vortex_t * vortex, int adbdma) dma->fifo_enabled = 0; } -#endif /* WTDMA */ #ifndef CHIP_AU8810 diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 5099690..0488633 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -353,8 +353,7 @@ static int snd_vortex_pcm_trigger(struct snd_pcm_substream *substream, int cmd) //printk(KERN_INFO "vortex: stop %d\n", dma); stream->fifo_enabled = 0; if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) - vortex_adbdma_pausefifo(chip, dma); - //vortex_adbdma_stopfifo(chip, dma); + vortex_adbdma_stopfifo(chip, dma); #ifndef CHIP_AU8810 else { printk(KERN_INFO "vortex: wt stop %d\n", dma); -- cgit v1.1 From 7d53a631ed92abd19d3c948a5daa535e53bd2bff Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Wed, 4 Jan 2012 14:31:16 +0100 Subject: ALSA: hdspm - Refactor serial number to avoid code duplication The serial number is used multiple times in hdspm.c. Since it belongs to the card, let's store it in struct hdspm and refer to it whenever necessary. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index d623451..1609253 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -941,6 +941,8 @@ struct hdspm { cycles_t last_interrupt; + unsigned int serial; + struct hdspm_peak_rms peak_rms; }; @@ -4694,7 +4696,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n", (hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF, - (hdspm_read(hdspm, HDSPM_midiStatusIn0)>>8) & 0xFFFFFF); + hdspm->serial); snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n", hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase); @@ -6266,8 +6268,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, hdspm_version.card_type = hdspm->io_type; strncpy(hdspm_version.cardname, hdspm->card_name, sizeof(hdspm_version.cardname)); - hdspm_version.serial = (hdspm_read(hdspm, - HDSPM_midiStatusIn0)>>8) & 0xFFFFFF; + hdspm_version.serial = hdspm->serial; hdspm_version.firmware_rev = hdspm->firmware_rev; hdspm_version.addons = 0; if (hdspm->tco) @@ -6866,12 +6867,14 @@ static int __devinit snd_hdspm_probe(struct pci_dev *pci, } if (hdspm->io_type != MADIface) { + hdspm->serial = (hdspm_read(hdspm, + HDSPM_midiStatusIn0)>>8) & 0xFFFFFF; sprintf(card->shortname, "%s_%x", hdspm->card_name, - (hdspm_read(hdspm, HDSPM_midiStatusIn0)>>8) & 0xFFFFFF); + hdspm->serial); sprintf(card->longname, "%s S/N 0x%x at 0x%lx, irq %d", hdspm->card_name, - (hdspm_read(hdspm, HDSPM_midiStatusIn0)>>8) & 0xFFFFFF, + hdspm->serial, hdspm->port, hdspm->irq); } else { sprintf(card->shortname, "%s", hdspm->card_name); -- cgit v1.1 From 7e5bea19aed376855eb2928c6d3c9ab0b35b5af7 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?J=C3=A9r=C3=A9my=20Lal?= Date: Mon, 9 Jan 2012 17:19:45 +0100 Subject: ALSA: hda/cirrus - support for iMac12,2 model MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This early 2011 model just need to have headphones on GPI02 instead of GPI01, and use BIOS pincfgs. It is detected by codec SSID. The iMac12,1 model is known to work the same way, although maybe not with the same codec SSID. Signed-off-by: Jérémy Lal Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 036056c4..0e99357 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -79,6 +79,7 @@ enum { CS420X_MBP53, CS420X_MBP55, CS420X_IMAC27, + CS420X_IMAC27_122, CS420X_APPLE, CS420X_AUTO, CS420X_MODELS @@ -1290,6 +1291,7 @@ static const char * const cs420x_models[CS420X_MODELS] = { [CS420X_MBP53] = "mbp53", [CS420X_MBP55] = "mbp55", [CS420X_IMAC27] = "imac27", + [CS420X_IMAC27_122] = "imac27_122", [CS420X_APPLE] = "apple", [CS420X_AUTO] = "auto", }; @@ -1306,6 +1308,7 @@ static const struct snd_pci_quirk cs420x_cfg_tbl[] = { }; static const struct snd_pci_quirk cs420x_codec_cfg_tbl[] = { + SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE), {} /* terminator */ }; @@ -1405,6 +1408,12 @@ static int patch_cs420x(struct hda_codec *codec) spec->gpio_mask = spec->gpio_dir = spec->gpio_eapd_hp | spec->gpio_eapd_speaker; break; + case CS420X_IMAC27_122: + spec->gpio_eapd_hp = 4; /* GPIO2 = headphones */ + spec->gpio_eapd_speaker = 8; /* GPIO3 = speakers */ + spec->gpio_mask = spec->gpio_dir = + spec->gpio_eapd_hp | spec->gpio_eapd_speaker; + break; } err = cs_parse_auto_config(codec); -- cgit v1.1 From 9badda0a0afffebbe1cb30565800896534a6c5bd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Jan 2012 18:22:35 +0100 Subject: ALSA: hdsp - Fix potential Oops in snd_hdsp_info_pref_sync_ref() Dan Carpenter reported that setting 0 to uinfo->value.enumerated.items in snd_hdsp_info_pref_sync_ref() may lead to Oops. This function should return an error immediately in such a case instead. Cc: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 0111203..b68cdec 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -2640,8 +2640,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd uinfo->value.enumerated.items = 3; break; default: - uinfo->value.enumerated.items = 0; - break; + return -EINVAL; } if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) -- cgit v1.1 From 74eeb141d3bdf5a9a65c84dd637c41f12c40f41c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Jan 2012 18:26:05 +0100 Subject: ALSA: asihpi - Fix potential Oops in snd_asihpi_cmode_info() Dan Carpenter reported that setting 0 to uinfo->value.enumerated.items in snd_asihpi_cmode_info() may lead to Oops. This function should return an error immediately in such a case instead. Cc: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index fd3926f..e8de831 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -2259,6 +2259,9 @@ static int snd_asihpi_cmode_info(struct snd_kcontrol *kcontrol, valid_modes++; } + if (!valid_modes) + return -EINVAL; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = valid_modes; -- cgit v1.1 From de4da59e480cdf1075b33dbaf8078fc87bc52241 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jan 2012 08:59:56 +0100 Subject: ALSA: hda - Use auto-parser for HP laptops with cx20459 codec These laptops can work well with the auto-parser and their BIOS setups, and in addition, the auto-parser fixes the problem with S3/S4 where the unsol event handling is killed after resume due to fallback to the single-cmd mode. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=740115 Cc: [v3.1+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index bf14a0a..8a32a69 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1086,8 +1086,6 @@ static const char * const cxt5045_models[CXT5045_MODELS] = { static const struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530), - SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series", - CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P105", CXT5045_LAPTOP_MICSENSE), SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), -- cgit v1.1 From 3a90274de3548ebb2aabfbf488cea8e275a73dc6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jan 2012 12:41:22 +0100 Subject: ALSA: hda - Return the error from get_wcaps_type() for invalid NIDs When an invalid NID is given, get_wcaps() returns zero as the error, but get_wcaps_type() takes it as the normal value and returns a bogus AC_WID_AUD_OUT value. This confuses the parser. With this patch, get_wcaps_type() returns -1 when value 0 is given, i.e. an invalid NID is passed to get_wcaps(). Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=740118 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_local.h | 7 ++++++- sound/pci/hda/hda_proc.c | 2 ++ 2 files changed, 8 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index e1abc07..aca8d31 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -488,7 +488,12 @@ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid) } /* get the widget type from widget capability bits */ -#define get_wcaps_type(wcaps) (((wcaps) & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT) +static inline int get_wcaps_type(unsigned int wcaps) +{ + if (!wcaps) + return -1; /* invalid type */ + return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; +} static inline unsigned int get_wcaps_channels(u32 wcaps) { diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 2c981b5..254ab52 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -54,6 +54,8 @@ static const char *get_wid_type_name(unsigned int wid_value) [AC_WID_BEEP] = "Beep Generator Widget", [AC_WID_VENDOR] = "Vendor Defined Widget", }; + if (wid_value == -1) + return "UNKNOWN Widget"; wid_value &= 0xf; if (names[wid_value]) return names[wid_value]; -- cgit v1.1 From 4808d12d1dddb046ec86425e5f6766f02e950292 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jan 2012 15:16:02 +0100 Subject: ALSA: hda - Fix the detection of "Loopback Mixing" control for VIA codecs Currently the driver checks only the out_mix_path[] for the primary output route for judging whether to create the loopback-mixing control or not. But, there are cases where aamix-routing is available only on headphone or speaker paths but not on the primary output path. So, the driver ignores such cases inappropriately. This patch fixes the check of the loopback-mixing control by testing all mix-routing paths. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index ab56866..03e63fe 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -2208,7 +2208,10 @@ static int via_auto_create_loopback_switch(struct hda_codec *codec) { struct via_spec *spec = codec->spec; - if (!spec->aa_mix_nid || !spec->out_mix_path.depth) + if (!spec->aa_mix_nid) + return 0; /* no loopback switching available */ + if (!(spec->out_mix_path.depth || spec->hp_mix_path.depth || + spec->speaker_path.depth)) return 0; /* no loopback switching available */ if (!via_clone_control(spec, &via_aamix_ctl_enum)) return -ENOMEM; -- cgit v1.1 From f7de8ba3fcf19487d2f0af9aee0c510fc79efa15 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Tue, 10 Jan 2012 20:58:40 +0100 Subject: ALSA: hdspm - Provide unique driver id based on card serial Before, /proc/asound looked like this: 2 [Default ]: HDSPM - RME RayDAT_f1cd85 RME RayDAT S/N 0xf1cd85 at 0xf7300000, irq 18 In case of a second HDSPM card, its name would be Default_1. This is cumbersome, because the order of the cards isn't stable across reboots. To help userspace tools referring to the correct card, this commit provides a unique id for each card: 2 [HDSPMxf1cd85 ]: HDSPM - RME RayDAT_f1cd85 RME RayDAT S/N 0xf1cd85 at 0xf7300000, irq 18 In this example, userspace (configuration files) would then use hw:HDSPMxf1cd85 to choose the right card. The serial is masked to 24bits, so this string is always shorter than sixteen chars. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 21 +++++++++++++++++++-- 1 file changed, 19 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 1609253..cc9f6c8 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6783,6 +6783,25 @@ static int __devinit snd_hdspm_create(struct snd_card *card, tasklet_init(&hdspm->midi_tasklet, hdspm_midi_tasklet, (unsigned long) hdspm); + + if (hdspm->io_type != MADIface) { + hdspm->serial = (hdspm_read(hdspm, + HDSPM_midiStatusIn0)>>8) & 0xFFFFFF; + /* id contains either a user-provided value or the default + * NULL. If it's the default, we're safe to + * fill card->id with the serial number. + * + * If the serial number is 0xFFFFFF, then we're dealing with + * an old PCI revision that comes without a sane number. In + * this case, we don't set card->id to avoid collisions + * when running with multiple cards. + */ + if (NULL == id[hdspm->dev] && hdspm->serial != 0xFFFFFF) { + sprintf(card->id, "HDSPMx%06x", hdspm->serial); + snd_card_set_id(card, card->id); + } + } + snd_printdd("create alsa devices.\n"); err = snd_hdspm_create_alsa_devices(card, hdspm); if (err < 0) @@ -6867,8 +6886,6 @@ static int __devinit snd_hdspm_probe(struct pci_dev *pci, } if (hdspm->io_type != MADIface) { - hdspm->serial = (hdspm_read(hdspm, - HDSPM_midiStatusIn0)>>8) & 0xFFFFFF; sprintf(card->shortname, "%s_%x", hdspm->card_name, hdspm->serial); -- cgit v1.1 From ffd364ddd3090e2ef0d4882970c1e342db8b482f Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Tue, 10 Jan 2012 20:45:28 +0100 Subject: ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations Add the capture pcm only if there is at least one ADC configured in the SYSCONF register. Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- sound/pci/ice1712/envy24ht.h | 1 + sound/pci/ice1712/ice1724.c | 13 ++++++++++--- 2 files changed, 11 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ice1712/envy24ht.h b/sound/pci/ice1712/envy24ht.h index a0c5e00..4ca33a8 100644 --- a/sound/pci/ice1712/envy24ht.h +++ b/sound/pci/ice1712/envy24ht.h @@ -66,6 +66,7 @@ enum { #define VT1724_CFG_CLOCK384 0x40 /* 16.9344Mhz, 44.1kHz*384 */ #define VT1724_CFG_MPU401 0x20 /* MPU401 UARTs */ #define VT1724_CFG_ADC_MASK 0x0c /* one, two or one and S/PDIF, stereo ADCs */ +#define VT1724_CFG_ADC_NONE 0x0c /* no ADCs */ #define VT1724_CFG_DAC_MASK 0x03 /* one, two, three, four stereo DACs */ #define VT1724_REG_AC97_CFG 0x05 /* byte */ diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 9e18d3a..e797823 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1117,14 +1117,21 @@ static struct snd_pcm_ops snd_vt1724_capture_pro_ops = { static int __devinit snd_vt1724_pcm_profi(struct snd_ice1712 *ice, int device) { struct snd_pcm *pcm; - int err; + int capt, err; - err = snd_pcm_new(ice->card, "ICE1724", device, 1, 1, &pcm); + if ((ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_ADC_MASK) == + VT1724_CFG_ADC_NONE) + capt = 0; + else + capt = 1; + err = snd_pcm_new(ice->card, "ICE1724", device, 1, capt, &pcm); if (err < 0) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_vt1724_playback_pro_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_vt1724_capture_pro_ops); + if (capt) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_vt1724_capture_pro_ops); pcm->private_data = ice; pcm->info_flags = 0; -- cgit v1.1 From 2b151ef734b1be749e355f32f94f649acfde0f48 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Tue, 10 Jan 2012 20:28:47 +0100 Subject: ALSA: ice1724 - Allow card info based on model only When two different cards share the same PCI vendor/subvendor identification, allow card info based on model only. Do not require subvendor ID. Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index e797823..352f3ff 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2282,7 +2282,7 @@ static int __devinit snd_vt1724_read_eeprom(struct snd_ice1712 *ice, } } for (tbl = card_tables; *tbl; tbl++) { - for (c = *tbl; c->subvendor; c++) { + for (c = *tbl; c->name; c++) { if (modelname && c->model && !strcmp(modelname, c->model)) { printk(KERN_INFO "ice1724: Using board model %s\n", @@ -2591,8 +2591,10 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, ice->ext_clock_count = 0; for (tbl = card_tables; *tbl; tbl++) { - for (c = *tbl; c->subvendor; c++) { - if (c->subvendor == ice->eeprom.subvendor) { + for (c = *tbl; c->name; c++) { + if ((model[dev] && c->model && + !strcmp(model[dev], c->model)) || + (c->subvendor == ice->eeprom.subvendor)) { strcpy(card->shortname, c->name); if (c->driver) /* specific driver? */ strcpy(card->driver, c->driver); -- cgit v1.1 From 52cd0a76fd7e7b47f0b0ad594ad5fd3b69949f76 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Tue, 10 Jan 2012 20:28:48 +0100 Subject: ALSA: ice1724 - Support for ooAoo SQ210a This card shares PCI ids with Chaintec AV710. Therefore, it will not be detected automatically, it can only be activated by the module parameter model=sq210a. Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 35 +++++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 352f3ff..9236297 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2185,6 +2185,40 @@ static struct snd_kcontrol_new snd_vt1724_mixer_pro_peak __devinitdata = { static struct snd_ice1712_card_info no_matched __devinitdata; + +/* + ooAoo cards with no controls +*/ +static unsigned char ooaoo_sq210_eeprom[] __devinitdata = { + [ICE_EEP2_SYSCONF] = 0x4c, /* 49MHz crystal, no mpu401, no ADC, + 1xDACs */ + [ICE_EEP2_ACLINK] = 0x80, /* I2S */ + [ICE_EEP2_I2S] = 0x78, /* no volume, 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc1, /* out-en, out-int, out-ext */ + [ICE_EEP2_GPIO_DIR] = 0x00, /* no GPIOs are used */ + [ICE_EEP2_GPIO_DIR1] = 0x00, + [ICE_EEP2_GPIO_DIR2] = 0x00, + [ICE_EEP2_GPIO_MASK] = 0xff, + [ICE_EEP2_GPIO_MASK1] = 0xff, + [ICE_EEP2_GPIO_MASK2] = 0xff, + + [ICE_EEP2_GPIO_STATE] = 0x00, /* inputs */ + [ICE_EEP2_GPIO_STATE1] = 0x00, /* all 1, but GPIO_CPLD_RW + and GPIO15 always zero */ + [ICE_EEP2_GPIO_STATE2] = 0x00, /* inputs */ +}; + + +struct snd_ice1712_card_info snd_vt1724_ooaoo_cards[] __devinitdata = { + { + .name = "ooAoo SQ210a", + .model = "sq210a", + .eeprom_size = sizeof(ooaoo_sq210_eeprom), + .eeprom_data = ooaoo_sq210_eeprom, + }, + { } /* terminator */ +}; + static struct snd_ice1712_card_info *card_tables[] __devinitdata = { snd_vt1724_revo_cards, snd_vt1724_amp_cards, @@ -2199,6 +2233,7 @@ static struct snd_ice1712_card_info *card_tables[] __devinitdata = { snd_vt1724_wtm_cards, snd_vt1724_se_cards, snd_vt1724_qtet_cards, + snd_vt1724_ooaoo_cards, NULL, }; -- cgit v1.1 From 56225e4cc88a24d3e1632bdfb901a3c38615fc42 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 6 Dec 2011 10:07:43 +0100 Subject: ALSA: virtuoso: add S/PDIF input support for all Xonars All Xonar cards support S/PDIF input, but the cards without optical or coaxial plugs have only undocumented pin connectors. Support for the ST/STX was already added in a previous patch; this adds support for the D1/DX (JP2), DG (J5), DS (J5), and HDAV Slim (J12). Many thanks to Zoltan Miklos for testing the DS and DX. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_cs43xx.c | 1 + sound/pci/oxygen/xonar_dg.c | 3 ++- sound/pci/oxygen/xonar_wm87x6.c | 6 ++++-- 3 files changed, 7 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index 2527191..c8febf4 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -418,6 +418,7 @@ static const struct oxygen_model model_xonar_d1 = { .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF | AC97_FMIC_SWITCH, .dac_channels_pcm = 8, .dac_channels_mixer = 8, diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index bc6eb58..793bdf0 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -597,7 +597,8 @@ struct oxygen_model model_xonar_dg = { .model_data_size = sizeof(struct dg), .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2, + CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF, .dac_channels_pcm = 6, .dac_channels_mixer = 0, .function_flags = OXYGEN_FUNCTION_SPI, diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 42d1ab1..478303e 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -1274,7 +1274,8 @@ static const struct oxygen_model model_xonar_ds = { .model_data_size = sizeof(struct xonar_wm87x6), .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_1, + CAPTURE_0_FROM_I2S_1 | + CAPTURE_1_FROM_SPDIF, .dac_channels_pcm = 8, .dac_channels_mixer = 8, .dac_volume_min = 255 - 2*60, @@ -1306,7 +1307,8 @@ static const struct oxygen_model model_xonar_hdav_slim = { .model_data_size = sizeof(struct xonar_wm87x6), .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_1, + CAPTURE_0_FROM_I2S_1 | + CAPTURE_1_FROM_SPDIF, .dac_channels_pcm = 8, .dac_channels_mixer = 2, .dac_volume_min = 255 - 2*60, -- cgit v1.1 From f2cbba7602383cd9cdd21f0a5d0b8bd1aad47b33 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Jan 2012 12:34:11 +0100 Subject: ALSA: hda - Fix the lost power-setup of seconary pins after PM resume When multiple headphone or other detectable output pins are present, the power-map has to be updated after resume appropriately, but the current driver doesn't check all pins but only the first pin (since it's enough to check it for the mute-behavior). This resulted in the silent output from the secondary outputs after PM resume. This patch fixes the problem by checking all pins at (re-)init time. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=740347 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 03145ae..87e684f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4236,6 +4236,27 @@ static void stac_store_hints(struct hda_codec *codec) } } +static void stac_issue_unsol_events(struct hda_codec *codec, int num_pins, + const hda_nid_t *pins) +{ + while (num_pins--) + stac_issue_unsol_event(codec, *pins++); +} + +/* fake event to set up pins */ +static void stac_fake_hp_events(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + + if (spec->autocfg.hp_outs) + stac_issue_unsol_events(codec, spec->autocfg.hp_outs, + spec->autocfg.hp_pins); + if (spec->autocfg.line_outs && + spec->autocfg.line_out_pins[0] != spec->autocfg.hp_pins[0]) + stac_issue_unsol_events(codec, spec->autocfg.line_outs, + spec->autocfg.line_out_pins); +} + static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -4286,10 +4307,7 @@ static int stac92xx_init(struct hda_codec *codec) stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0], AC_PINCTL_OUT_EN); /* fake event to set up pins */ - if (cfg->hp_pins[0]) - stac_issue_unsol_event(codec, cfg->hp_pins[0]); - else if (cfg->line_out_pins[0]) - stac_issue_unsol_event(codec, cfg->line_out_pins[0]); + stac_fake_hp_events(codec); } else { stac92xx_auto_init_multi_out(codec); stac92xx_auto_init_hp_out(codec); @@ -4948,19 +4966,11 @@ static void stac927x_proc_hook(struct snd_info_buffer *buffer, #ifdef CONFIG_PM static int stac92xx_resume(struct hda_codec *codec) { - struct sigmatel_spec *spec = codec->spec; - stac92xx_init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); /* fake event to set up pins again to override cached values */ - if (spec->hp_detect) { - if (spec->autocfg.hp_pins[0]) - stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]); - else if (spec->autocfg.line_out_pins[0]) - stac_issue_unsol_event(codec, - spec->autocfg.line_out_pins[0]); - } + stac_fake_hp_events(codec); return 0; } -- cgit v1.1