diff options
Diffstat (limited to 'sound/pci/hda')
-rw-r--r-- | sound/pci/hda/Kconfig | 9 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.c | 2 | ||||
-rw-r--r-- | sound/pci/hda/patch_analog.c | 128 | ||||
-rw-r--r-- | sound/pci/hda/patch_conexant.c | 4 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 283 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 10 |
6 files changed, 279 insertions, 157 deletions
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index c710150d506..04438f1d682 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -2,7 +2,6 @@ menuconfig SND_HDA_INTEL tristate "Intel HD Audio" select SND_PCM select SND_VMASTER - select SND_JACK if INPUT=y || INPUT=SND help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. @@ -39,6 +38,14 @@ config SND_HDA_INPUT_BEEP Say Y here to build a digital beep interface for HD-audio driver. This interface is used to generate digital beeps. +config SND_HDA_INPUT_JACK + bool "Support jack plugging notification via input layer" + depends on INPUT=y || INPUT=SND_HDA_INTEL + select SND_JACK + help + Say Y here to enable the jack plugging notification via + input layer. + config SND_HDA_CODEC_REALTEK bool "Build Realtek HD-audio codec support" default y diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 562403a2348..462e2cedaa6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -972,8 +972,6 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_SUBSYSTEM_ID, 0); } - if (bus->modelname) - codec->modelname = kstrdup(bus->modelname, GFP_KERNEL); /* power-up all before initialization */ hda_set_power_state(codec, diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 84cc49ca914..1988582d1ab 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -72,6 +72,7 @@ struct ad198x_spec { hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; unsigned int jack_present :1; + unsigned int inv_jack_detect:1; #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; @@ -669,39 +670,13 @@ static struct hda_input_mux ad1986a_automic_capture_source = { }, }; -static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { +static struct snd_kcontrol_new ad1986a_laptop_master_mixers[] = { HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "External Amplifier", - .info = ad198x_eapd_info, - .get = ad198x_eapd_get, - .put = ad198x_eapd_put, - .private_value = 0x1b | (1 << 8), /* port-D, inversed */ - }, { } /* end */ }; -static struct snd_kcontrol_new ad1986a_samsung_mixers[] = { - HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), - HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw), +static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), @@ -727,6 +702,12 @@ static struct snd_kcontrol_new ad1986a_samsung_mixers[] = { { } /* end */ }; +static struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = { + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, HDA_OUTPUT), + { } /* end */ +}; + /* re-connect the mic boost input according to the jack sensing */ static void ad1986a_automic(struct hda_codec *codec) { @@ -776,8 +757,9 @@ static void ad1986a_hp_automute(struct hda_codec *codec) unsigned int present; present = snd_hda_codec_read(codec, 0x1a, 0, AC_VERB_GET_PIN_SENSE, 0); - /* Lenovo N100 seems to report the reversed bit for HP jack-sensing */ - spec->jack_present = !(present & 0x80000000); + spec->jack_present = !!(present & 0x80000000); + if (spec->inv_jack_detect) + spec->jack_present = !spec->jack_present; ad1986a_update_hp(codec); } @@ -816,7 +798,7 @@ static int ad1986a_hp_master_sw_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new ad1986a_laptop_automute_mixers[] = { +static struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -826,33 +808,10 @@ static struct snd_kcontrol_new ad1986a_laptop_automute_mixers[] = { .put = ad1986a_hp_master_sw_put, .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), }, - HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "External Amplifier", - .info = ad198x_eapd_info, - .get = ad198x_eapd_get, - .put = ad198x_eapd_put, - .private_value = 0x1b | (1 << 8), /* port-D, inversed */ - }, { } /* end */ }; + /* * initialization verbs */ @@ -981,6 +940,27 @@ static struct hda_verb ad1986a_hp_init_verbs[] = { {} }; +static void ad1986a_samsung_p50_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case AD1986A_HP_EVENT: + ad1986a_hp_automute(codec); + break; + case AD1986A_MIC_EVENT: + ad1986a_automic(codec); + break; + } +} + +static int ad1986a_samsung_p50_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1986a_hp_automute(codec); + ad1986a_automic(codec); + return 0; +} + /* models */ enum { @@ -991,6 +971,7 @@ enum { AD1986A_LAPTOP_AUTOMUTE, AD1986A_ULTRA, AD1986A_SAMSUNG, + AD1986A_SAMSUNG_P50, AD1986A_MODELS }; @@ -1002,6 +983,7 @@ static const char *ad1986a_models[AD1986A_MODELS] = { [AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute", [AD1986A_ULTRA] = "ultra", [AD1986A_SAMSUNG] = "samsung", + [AD1986A_SAMSUNG_P50] = "samsung-p50", }; static struct snd_pci_quirk ad1986a_cfg_tbl[] = { @@ -1024,6 +1006,7 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), + SND_PCI_QUIRK(0x144d, 0xc024, "Samsung P50", AD1986A_SAMSUNG_P50), SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA), SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK), @@ -1111,7 +1094,10 @@ static int patch_ad1986a(struct hda_codec *codec) spec->multiout.dac_nids = ad1986a_laptop_dac_nids; break; case AD1986A_LAPTOP_EAPD: - spec->mixers[0] = ad1986a_laptop_eapd_mixers; + spec->num_mixers = 3; + spec->mixers[0] = ad1986a_laptop_master_mixers; + spec->mixers[1] = ad1986a_laptop_eapd_mixers; + spec->mixers[2] = ad1986a_laptop_intmic_mixers; spec->num_init_verbs = 2; spec->init_verbs[1] = ad1986a_eapd_init_verbs; spec->multiout.max_channels = 2; @@ -1122,7 +1108,9 @@ static int patch_ad1986a(struct hda_codec *codec) spec->input_mux = &ad1986a_laptop_eapd_capture_source; break; case AD1986A_SAMSUNG: - spec->mixers[0] = ad1986a_samsung_mixers; + spec->num_mixers = 2; + spec->mixers[0] = ad1986a_laptop_master_mixers; + spec->mixers[1] = ad1986a_laptop_eapd_mixers; spec->num_init_verbs = 3; spec->init_verbs[1] = ad1986a_eapd_init_verbs; spec->init_verbs[2] = ad1986a_automic_verbs; @@ -1135,8 +1123,28 @@ static int patch_ad1986a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1986a_automic_unsol_event; codec->patch_ops.init = ad1986a_automic_init; break; + case AD1986A_SAMSUNG_P50: + spec->num_mixers = 2; + spec->mixers[0] = ad1986a_automute_master_mixers; + spec->mixers[1] = ad1986a_laptop_eapd_mixers; + spec->num_init_verbs = 4; + spec->init_verbs[1] = ad1986a_eapd_init_verbs; + spec->init_verbs[2] = ad1986a_automic_verbs; + spec->init_verbs[3] = ad1986a_hp_init_verbs; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = ad1986a_laptop_dac_nids; + if (!is_jack_available(codec, 0x25)) + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1986a_automic_capture_source; + codec->patch_ops.unsol_event = ad1986a_samsung_p50_unsol_event; + codec->patch_ops.init = ad1986a_samsung_p50_init; + break; case AD1986A_LAPTOP_AUTOMUTE: - spec->mixers[0] = ad1986a_laptop_automute_mixers; + spec->num_mixers = 3; + spec->mixers[0] = ad1986a_automute_master_mixers; + spec->mixers[1] = ad1986a_laptop_eapd_mixers; + spec->mixers[2] = ad1986a_laptop_intmic_mixers; spec->num_init_verbs = 3; spec->init_verbs[1] = ad1986a_eapd_init_verbs; spec->init_verbs[2] = ad1986a_hp_init_verbs; @@ -1148,6 +1156,10 @@ static int patch_ad1986a(struct hda_codec *codec) spec->input_mux = &ad1986a_laptop_eapd_capture_source; codec->patch_ops.unsol_event = ad1986a_hp_unsol_event; codec->patch_ops.init = ad1986a_hp_init; + /* Lenovo N100 seems to report the reversed bit + * for HP jack-sensing + */ + spec->inv_jack_detect = 1; break; case AD1986A_ULTRA: spec->mixers[0] = ad1986a_laptop_eapd_mixers; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4fcbe21829a..ac868c59f9e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -349,7 +349,7 @@ static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol, &spec->cur_mux[adc_idx]); } -#ifdef CONFIG_SND_JACK +#ifdef CONFIG_SND_HDA_INPUT_JACK static void conexant_free_jack_priv(struct snd_jack *jack) { struct conexant_jack *jacks = jack->private_data; @@ -463,7 +463,7 @@ static int conexant_init(struct hda_codec *codec) static void conexant_free(struct hda_codec *codec) { -#ifdef CONFIG_SND_JACK +#ifdef CONFIG_SND_HDA_INPUT_JACK struct conexant_spec *spec = codec->spec; if (spec->jacks.list) { struct conexant_jack *jacks = spec->jacks.list; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d22b2606801..3a8e58c483d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -224,6 +224,7 @@ enum { ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, + ALC888_ACER_ASPIRE_6530G, ALC888_ACER_ASPIRE_8930G, ALC883_MEDION, ALC883_MEDION_MD2, @@ -249,13 +250,6 @@ enum { ALC883_MODEL_LAST, }; -/* styles of capture selection */ -enum { - CAPT_MUX = 0, /* only mux based */ - CAPT_MIX, /* only mixer based */ - CAPT_1MUX_MIX, /* first mux and other mixers */ -}; - /* for GPIO Poll */ #define GPIO_MASK 0x03 @@ -305,7 +299,6 @@ struct alc_spec { hda_nid_t *adc_nids; hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ - int capture_style; /* capture style (CAPT_*) */ /* capture source */ unsigned int num_mux_defs; @@ -419,12 +412,13 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, unsigned int mux_idx; hda_nid_t nid = spec->capsrc_nids ? spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx]; + unsigned int type; mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; imux = &spec->input_mux[mux_idx]; - if (spec->capture_style && - !(spec->capture_style == CAPT_1MUX_MIX && !adc_idx)) { + type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (type == AC_WID_AUD_MIX) { /* Matrix-mixer style (e.g. ALC882) */ unsigned int *cur_val = &spec->cur_mux[adc_idx]; unsigned int i, idx; @@ -951,12 +945,13 @@ static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid, static void alc_automute_pin(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int present; + unsigned int present, pincap; unsigned int nid = spec->autocfg.hp_pins[0]; int i; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); + 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); present = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; @@ -970,7 +965,7 @@ static void alc_automute_pin(struct hda_codec *codec) } } -#if 0 /* it's broken in some acses -- temporarily disabled */ +#if 0 /* it's broken in some cases -- temporarily disabled */ static void alc_mic_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -1170,7 +1165,7 @@ static int alc_subsystem_id(struct hda_codec *codec, /* invalid SSID, check the special NID pin defcfg instead */ /* - * 31~30 : port conetcivity + * 31~30 : port connectivity * 29~21 : reserve * 20 : PCBEEP input * 19~16 : Check sum (15:1) @@ -1398,7 +1393,7 @@ static struct hda_verb alc888_fujitsu_xa3530_verbs[] = { static void alc_automute_amp(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int val, mute; + unsigned int val, mute, pincap; hda_nid_t nid; int i; @@ -1407,6 +1402,10 @@ static void alc_automute_amp(struct hda_codec *codec) nid = spec->autocfg.hp_pins[i]; if (!nid) break; + 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); val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); if (val & AC_PINSENSE_PRESENCE) { @@ -1471,6 +1470,29 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { }; /* + * ALC888 Acer Aspire 6530G model + */ + +static struct hda_verb alc888_acer_aspire_6530g_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, ALC880_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}, +/* 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}, + { } +}; + +/* * ALC889 Acer Aspire 8930G model */ @@ -1544,6 +1566,29 @@ static struct hda_input_mux alc888_2_capture_sources[2] = { } }; +static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { + /* Interal mic only available on one ADC */ + { + .num_items = 5, + .items = { + { "Ext Mic", 0x0 }, + { "Line In", 0x2 }, + { "CD", 0x4 }, + { "Input Mix", 0xa }, + { "Int Mic", 0xb }, + }, + }, + { + .num_items = 4, + .items = { + { "Ext Mic", 0x0 }, + { "Line In", 0x2 }, + { "CD", 0x4 }, + { "Input Mix", 0xa }, + }, + } +}; + static struct hda_input_mux alc889_capture_sources[3] = { /* Digital mic only available on first "ADC" */ { @@ -1607,6 +1652,17 @@ static void alc888_acer_aspire_4930g_init_hook(struct hda_codec *codec) alc_automute_amp(codec); } +static void alc888_acer_aspire_6530g_init_hook(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_automute_amp(codec); +} + static void alc889_acer_aspire_8930g_init_hook(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -6347,7 +6403,7 @@ static struct hda_channel_mode alc882_sixstack_modes[2] = { }; /* - * macbook pro ALC885 can switch LineIn to LineOut without loosing Mic + * macbook pro ALC885 can switch LineIn to LineOut without losing Mic */ /* @@ -7047,7 +7103,7 @@ static struct hda_verb alc882_auto_init_verbs[] = { #define alc882_loopbacks alc880_loopbacks #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc882_pcm_analog_playback alc880_pcm_analog_playback #define alc882_pcm_analog_capture alc880_pcm_analog_capture #define alc882_pcm_digital_playback alc880_pcm_digital_playback @@ -7518,7 +7574,6 @@ static int patch_alc882(struct hda_codec *codec) spec->stream_digital_playback = &alc882_pcm_digital_playback; spec->stream_digital_capture = &alc882_pcm_digital_capture; - spec->capture_style = CAPT_MIX; /* matrix-style capture */ if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -8068,7 +8123,7 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_tagra_mixer[] = { +static struct snd_kcontrol_new alc883_targa_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -8088,7 +8143,7 @@ static struct snd_kcontrol_new alc883_tagra_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_tagra_2ch_mixer[] = { +static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -8153,6 +8208,21 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 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("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", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + static 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), @@ -8417,7 +8487,7 @@ static struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = { { } /* end */ }; -static struct hda_verb alc883_tagra_verbs[] = { +static 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)}, @@ -8626,8 +8696,8 @@ static void alc883_medion_md2_init_hook(struct hda_codec *codec) } /* toggle speaker-output according to the hp-jack state */ -#define alc883_tagra_init_hook alc882_targa_init_hook -#define alc883_tagra_unsol_event alc882_targa_unsol_event +#define alc883_targa_init_hook alc882_targa_init_hook +#define alc883_targa_unsol_event alc882_targa_unsol_event static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) { @@ -8957,7 +9027,7 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) #define alc883_loopbacks alc880_loopbacks #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc883_pcm_analog_playback alc880_pcm_analog_playback #define alc883_pcm_analog_capture alc880_pcm_analog_capture #define alc883_pcm_analog_alt_capture alc880_pcm_analog_alt_capture @@ -8978,6 +9048,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_ACER] = "acer", [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", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", @@ -9019,9 +9090,9 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", - ALC888_ACER_ASPIRE_4930G), + ALC888_ACER_ASPIRE_6530G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", - ALC888_ACER_ASPIRE_4930G), + ALC888_ACER_ASPIRE_6530G), /* default Acer -- disabled as it causes more problems. * model=auto should work fine now */ @@ -9069,6 +9140,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { 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, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), @@ -9165,8 +9237,8 @@ static struct alc_config_preset alc883_presets[] = { .input_mux = &alc883_capture_source, }, [ALC883_TARGA_DIG] = { - .mixers = { alc883_tagra_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc883_tagra_verbs}, + .mixers = { alc883_targa_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc883_targa_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, @@ -9174,12 +9246,12 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, - .unsol_event = alc883_tagra_unsol_event, - .init_hook = alc883_tagra_init_hook, + .unsol_event = alc883_targa_unsol_event, + .init_hook = alc883_targa_init_hook, }, [ALC883_TARGA_2ch_DIG] = { - .mixers = { alc883_tagra_2ch_mixer}, - .init_verbs = { alc883_init_verbs, alc883_tagra_verbs}, + .mixers = { alc883_targa_2ch_mixer}, + .init_verbs = { alc883_init_verbs, alc883_targa_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .adc_nids = alc883_adc_nids_alt, @@ -9188,13 +9260,13 @@ static struct alc_config_preset alc883_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc883_tagra_unsol_event, - .init_hook = alc883_tagra_init_hook, + .unsol_event = alc883_targa_unsol_event, + .init_hook = alc883_targa_init_hook, }, [ALC883_TARGA_8ch_DIG] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, - alc883_tagra_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), @@ -9206,8 +9278,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_4ST_8ch_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, - .unsol_event = alc883_tagra_unsol_event, - .init_hook = alc883_tagra_init_hook, + .unsol_event = alc883_targa_unsol_event, + .init_hook = alc883_targa_init_hook, }, [ALC883_ACER] = { .mixers = { alc883_base_mixer }, @@ -9255,6 +9327,24 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc888_acer_aspire_4930g_init_hook, }, + [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_automute_amp_unsol_event, + .init_hook = alc888_acer_aspire_6530g_init_hook, + }, [ALC888_ACER_ASPIRE_8930G] = { .mixers = { alc888_base_mixer, alc883_chmode_mixer }, @@ -9361,7 +9451,7 @@ static struct alc_config_preset alc883_presets[] = { .init_hook = alc888_lenovo_ms7195_front_automute, }, [ALC883_HAIER_W66] = { - .mixers = { alc883_tagra_2ch_mixer}, + .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, @@ -9709,7 +9799,6 @@ static int patch_alc883(struct hda_codec *codec) } if (!spec->capsrc_nids) spec->capsrc_nids = alc883_capsrc_nids; - spec->capture_style = CAPT_MIX; /* matrix-style capture */ spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */ break; case 0x10ec0889: @@ -9719,8 +9808,6 @@ static int patch_alc883(struct hda_codec *codec) } if (!spec->capsrc_nids) spec->capsrc_nids = alc889_capsrc_nids; - spec->capture_style = CAPT_1MUX_MIX; /* 1mux/Nmix-style - capture */ break; default: if (!spec->num_adc_nids) { @@ -9729,7 +9816,6 @@ static int patch_alc883(struct hda_codec *codec) } if (!spec->capsrc_nids) spec->capsrc_nids = alc883_capsrc_nids; - spec->capture_style = CAPT_MIX; /* matrix-style capture */ break; } @@ -10841,9 +10927,27 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, return 0; } -/* identical with ALC880 */ -#define alc262_auto_create_analog_input_ctls \ - alc880_auto_create_analog_input_ctls +static int alc262_auto_create_analog_input_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int err; + + err = alc880_auto_create_analog_input_ctls(spec, cfg); + if (err < 0) + return err; + /* digital-mic input pin is excluded in alc880_auto_create..() + * because it's under 0x18 + */ + if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || + cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { + struct hda_input_mux *imux = &spec->private_imux[0]; + imux->items[imux->num_items].label = "Int Mic"; + imux->items[imux->num_items].index = 0x09; + imux->num_items++; + } + return 0; +} + /* * generic initialization of ADC, input mixers and output mixers @@ -11131,7 +11235,7 @@ static struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = { #define alc262_loopbacks alc880_loopbacks #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc262_pcm_analog_playback alc880_pcm_analog_playback #define alc262_pcm_analog_capture alc880_pcm_analog_capture #define alc262_pcm_digital_playback alc880_pcm_digital_playback @@ -11260,6 +11364,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO), SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */ + SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06), SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", @@ -11467,6 +11572,7 @@ static struct alc_config_preset alc262_presets[] = { .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, @@ -11568,21 +11674,36 @@ static int patch_alc262(struct hda_codec *codec) spec->stream_digital_playback = &alc262_pcm_digital_playback; spec->stream_digital_capture = &alc262_pcm_digital_capture; - spec->capture_style = CAPT_MIX; if (!spec->adc_nids && spec->input_mux) { - /* check whether NID 0x07 is valid */ - unsigned int wcap = get_wcaps(codec, 0x07); - - /* get type */ - wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wcap != AC_WID_AUD_IN) { - spec->adc_nids = alc262_adc_nids_alt; - spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids_alt); - spec->capsrc_nids = alc262_capsrc_nids_alt; + int i; + /* check whether the digital-mic has to be supported */ + for (i = 0; i < spec->input_mux->num_items; i++) { + if (spec->input_mux->items[i].index >= 9) + break; + } + if (i < spec->input_mux->num_items) { + /* use only ADC0 */ + spec->adc_nids = alc262_dmic_adc_nids; + spec->num_adc_nids = 1; + spec->capsrc_nids = alc262_dmic_capsrc_nids; } else { - spec->adc_nids = alc262_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids); - spec->capsrc_nids = alc262_capsrc_nids; + /* all analog inputs */ + /* check whether NID 0x07 is valid */ + unsigned int wcap = get_wcaps(codec, 0x07); + + /* get type */ + wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (wcap != AC_WID_AUD_IN) { + spec->adc_nids = alc262_adc_nids_alt; + spec->num_adc_nids = + ARRAY_SIZE(alc262_adc_nids_alt); + spec->capsrc_nids = alc262_capsrc_nids_alt; + } else { + spec->adc_nids = alc262_adc_nids; + spec->num_adc_nids = + ARRAY_SIZE(alc262_adc_nids); + spec->capsrc_nids = alc262_capsrc_nids; + } } } if (!spec->cap_mixer && !spec->no_analog) @@ -12286,7 +12407,7 @@ static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec) AC_VERB_SET_AMP_GAIN_MUTE, dac_vol2); } -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc268_pcm_analog_playback alc880_pcm_analog_playback #define alc268_pcm_analog_capture alc880_pcm_analog_capture #define alc268_pcm_analog_alt_capture alc880_pcm_analog_alt_capture @@ -12342,6 +12463,8 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + alc_ssid_check(codec, 0x15, 0x1b, 0x14); + return 1; } @@ -13172,32 +13295,14 @@ static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec, return 0; } -static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec, - const struct auto_pin_cfg *cfg) -{ - int err; - - err = alc880_auto_create_analog_input_ctls(spec, cfg); - if (err < 0) - return err; - /* digital-mic input pin is excluded in alc880_auto_create..() - * because it's under 0x18 - */ - if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || - cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { - struct hda_input_mux *imux = &spec->private_imux[0]; - imux->items[imux->num_items].label = "Int Mic"; - imux->items[imux->num_items].index = 0x05; - imux->num_items++; - } - return 0; -} +#define alc269_auto_create_analog_input_ctls \ + alc262_auto_create_analog_input_ctls #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc269_loopbacks alc880_loopbacks #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc269_pcm_analog_playback alc880_pcm_analog_playback #define alc269_pcm_analog_capture alc880_pcm_analog_capture #define alc269_pcm_digital_playback alc880_pcm_digital_playback @@ -13268,6 +13373,8 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); + alc_ssid_check(codec, 0x15, 0x1b, 0x14); + return 1; } @@ -14059,7 +14166,7 @@ static void alc861_toshiba_unsol_event(struct hda_codec *codec, alc861_toshiba_automute(codec); } -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc861_pcm_analog_playback alc880_pcm_analog_playback #define alc861_pcm_analog_capture alc880_pcm_analog_capture #define alc861_pcm_digital_playback alc880_pcm_digital_playback @@ -14582,7 +14689,7 @@ static hda_nid_t alc861vd_dac_nids[4] = { /* dac_nids for ALC660vd are in a different order - according to * Realtek's driver. - * This should probably tesult in a different mixer for 6stack models + * This should probably result in a different mixer for 6stack models * of ALC660vd codecs, but for now there is only 3stack mixer * - and it is the same as in 861vd. * adc_nids in ALC660vd are (is) the same as in 861vd @@ -15027,7 +15134,7 @@ static void alc861vd_dallas_init_hook(struct hda_codec *codec) #define alc861vd_loopbacks alc880_loopbacks #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc861vd_pcm_analog_playback alc880_pcm_analog_playback #define alc861vd_pcm_analog_capture alc880_pcm_analog_capture #define alc861vd_pcm_digital_playback alc880_pcm_digital_playback @@ -15206,7 +15313,7 @@ static void alc861vd_auto_init_hp_out(struct hda_codec *codec) hda_nid_t pin; pin = spec->autocfg.hp_pins[0]; - if (pin) /* connect to front and use dac 0 */ + if (pin) /* connect to front and use dac 0 */ alc861vd_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); pin = spec->autocfg.speaker_pins[0]; if (pin) @@ -15482,7 +15589,6 @@ static int patch_alc861vd(struct hda_codec *codec) spec->adc_nids = alc861vd_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids); spec->capsrc_nids = alc861vd_capsrc_nids; - spec->capture_style = CAPT_MIX; set_capture_mixer(spec); set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); @@ -16669,7 +16775,7 @@ static struct snd_kcontrol_new alc272_nc10_mixer[] = { #endif -/* pcm configuration: identiacal with ALC880 */ +/* pcm configuration: identical with ALC880 */ #define alc662_pcm_analog_playback alc880_pcm_analog_playback #define alc662_pcm_analog_capture alc880_pcm_analog_capture #define alc662_pcm_digital_playback alc880_pcm_digital_playback @@ -17402,7 +17508,6 @@ static int patch_alc662(struct hda_codec *codec) spec->adc_nids = alc662_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids); spec->capsrc_nids = alc662_capsrc_nids; - spec->capture_style = CAPT_MIX; if (!spec->cap_mixer) set_capture_mixer(spec); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 93e47c96a38..14f3c3e0f62 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -639,7 +639,7 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, static unsigned int stac92xx_vref_set(struct hda_codec *codec, hda_nid_t nid, unsigned int new_vref) { - unsigned int error; + int error; unsigned int pincfg; pincfg = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); @@ -2703,7 +2703,7 @@ static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int new_vref = 0; - unsigned int error; + int error; hda_nid_t nid = kcontrol->private_value; if (ucontrol->value.enumerated.item[0] == 0) @@ -4035,7 +4035,7 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */ } -#ifdef CONFIG_SND_JACK +#ifdef CONFIG_SND_HDA_INPUT_JACK static void stac92xx_free_jack_priv(struct snd_jack *jack) { struct sigmatel_jack *jacks = jack->private_data; @@ -4047,7 +4047,7 @@ static void stac92xx_free_jack_priv(struct snd_jack *jack) static int stac92xx_add_jack(struct hda_codec *codec, hda_nid_t nid, int type) { -#ifdef CONFIG_SND_JACK +#ifdef CONFIG_SND_HDA_INPUT_JACK struct sigmatel_spec *spec = codec->spec; struct sigmatel_jack *jack; int def_conf = snd_hda_codec_get_pincfg(codec, nid); @@ -4336,7 +4336,7 @@ static int stac92xx_init(struct hda_codec *codec) static void stac92xx_free_jacks(struct hda_codec *codec) { -#ifdef CONFIG_SND_JACK +#ifdef CONFIG_SND_HDA_INPUT_JACK /* free jack instances manually when clearing/reconfiguring */ struct sigmatel_spec *spec = codec->spec; if (!codec->bus->shutdown && spec->jacks.list) { |