From: Takashi Iwai Date: Thu, 26 Jul 2007 09:49:22 +0000 (+0200) Subject: [ALSA] hda-codec - Fix the initial mixer state of ALC262 sony-assamd model X-Git-Url: https://git.stricted.de/?a=commitdiff_plain;h=5b31954e4b364f811450311e3b31d3512e575f63;p=GitHub%2Fexynos8895%2Fandroid_kernel_samsung_universal8895.git [ALSA] hda-codec - Fix the initial mixer state of ALC262 sony-assamd model Many of ALC262 codes don't call the automute function at the beginning, which may keep the silence until the HP jack is replugged. Now proper init_hook is added. Also, sony-assamd model doesn't handle the widget 0x14 properly, thus calling automute isn't enough. Now Front switch handles both widgets. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7e6cc01b521a..d839d567f8e4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7140,9 +7140,28 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = { { } /* end */ }; +static int alc262_sony_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned long private_save = kcontrol->private_value; + int change; + kcontrol->private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT); + change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); + kcontrol->private_value = private_save; + change |= snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); + return change; +} + static struct snd_kcontrol_new alc262_sony_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Front Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc262_sony_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x15, 3, 0, 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), @@ -7269,20 +7288,17 @@ static struct hda_verb alc262_sony_unsol_verbs[] = { }; /* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc262_hippo_automute(struct hda_codec *codec, int force) +static void alc262_hippo_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; unsigned int mute; + unsigned int present; - if (force || !spec->sense_updated) { - unsigned int present; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; - spec->sense_updated = 1; - } + /* need to execute and sync at first */ + snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & 0x80000000) != 0; if (spec->jack_present) { /* mute internal speaker */ snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, @@ -7306,24 +7322,19 @@ static void alc262_hippo_unsol_event(struct hda_codec *codec, { if ((res >> 26) != ALC880_HP_EVENT) return; - alc262_hippo_automute(codec, 1); + alc262_hippo_automute(codec); } -static void alc262_hippo1_automute(struct hda_codec *codec, int force) +static void alc262_hippo1_automute(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; unsigned int mute; + unsigned int present; - if (force || !spec->sense_updated) { - unsigned int present; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; - spec->sense_updated = 1; - } - if (spec->jack_present) { + snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0); + present = (present & 0x80000000) != 0; + if (present) { /* mute internal speaker */ snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, 0x80, 0x80); @@ -7346,7 +7357,7 @@ static void alc262_hippo1_unsol_event(struct hda_codec *codec, { if ((res >> 26) != ALC880_HP_EVENT) return; - alc262_hippo1_automute(codec, 1); + alc262_hippo1_automute(codec); } /* @@ -7923,6 +7934,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), + SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), @@ -7951,6 +7963,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, + .init_hook = alc262_hippo_automute, }, [ALC262_HIPPO_1] = { .mixers = { alc262_hippo1_mixer }, @@ -7963,6 +7976,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo1_unsol_event, + .init_hook = alc262_hippo1_automute, }, [ALC262_FUJITSU] = { .mixers = { alc262_fujitsu_mixer }, @@ -8027,6 +8041,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, + .init_hook = alc262_hippo_automute, }, [ALC262_BENQ_T31] = { .mixers = { alc262_benq_t31_mixer }, @@ -8038,6 +8053,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, + .init_hook = alc262_hippo_automute, }, };