[ALSA] hda-codec - Update realtek codec support

1. Support  Acer Aspire 9810
2. Support  TOSHIBA A205
3. Support  HP TX1000

Signed-off-by: Kailang Yang <kailang@realtek.com.tw>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 171f641..778853c 100644 (file)
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -102,6 +102,7 @@ enum {
 /* ALC268 models */
 enum {
        ALC268_3ST,
+       ALC268_TOSHIBA,
        ALC268_AUTO,
        ALC268_MODEL_LAST /* last tag */
 };
@@ -129,6 +130,7 @@ enum {
        ALC861VD_6ST_DIG,
        ALC861VD_LENOVO,
        ALC861VD_DALLAS,
+       ALC861VD_HP,
        ALC861VD_AUTO,
        ALC861VD_MODEL_LAST,
 };
@@ -6253,16 +6255,14 @@ static struct snd_kcontrol_new alc888_3st_hp_mixer[] = {
 };
 
 static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
-       HDA_CODEC_VOLUME("Front Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
-       HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-       HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+       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("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("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", 0x18, 0, HDA_INPUT),
+       HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
        HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
        HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
        HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
@@ -6277,7 +6277,7 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
                .put = alc883_mux_enum_put,
        },
        { } /* end */
-};     
+};
 
 static struct snd_kcontrol_new alc883_chmode_mixer[] = {
        {
@@ -6490,18 +6490,6 @@ static struct hda_verb alc883_medion_md2_verbs[] = {
        { } /* end */
 };
 
-static struct hda_verb alc883_acer_aspire_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)},
-       {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
-       {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-
-       {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-       { } /* end */
-};
-
 /* toggle speaker-output according to the hp-jack state */
 static void alc883_medion_md2_automute(struct hda_codec *codec)
 {
@@ -6576,6 +6564,21 @@ static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec,
                alc883_lenovo_101e_ispeaker_automute(codec);
 }
 
+static 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) */
+       {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, 0x00},
+        /* eanable EAPD on medion laptop */
+       {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+       {0x20, AC_VERB_SET_PROC_COEF, 0x3050},
+       { }
+};
+
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
@@ -6718,6 +6721,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
        SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG),
        SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG),
        SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
+       SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
        SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
        SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER),
        SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
@@ -6834,8 +6838,13 @@ static struct alc_config_preset alc883_presets[] = {
                .input_mux = &alc883_capture_source,
        },
        [ALC883_ACER_ASPIRE] = {
-               .mixers = { alc883_acer_aspire_mixer},
-               .init_verbs = { alc883_init_verbs, alc883_acer_aspire_verbs},
+               .mixers = { alc883_acer_aspire_mixer, alc883_chmode_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, alc883_acer_eapd_verbs },
                .num_dacs = ARRAY_SIZE(alc883_dac_nids),
                .dac_nids = alc883_dac_nids,
                .dig_out_nid = ALC883_DIGOUT_NID,
@@ -6844,9 +6853,7 @@ 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_medion_md2_unsol_event,
-               .init_hook = alc883_medion_md2_automute,
-       },      
+       },
        [ALC883_MEDION] = {
                .mixers = { alc883_fivestack_mixer,
                            alc883_chmode_mixer },
@@ -8221,6 +8228,12 @@ static struct snd_kcontrol_new alc268_base_mixer[] = {
        { }
 };
 
+static struct hda_verb alc268_eapd_verbs[] = {
+       {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+       {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+       { }
+};
+
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
@@ -8610,6 +8623,7 @@ static const char *alc268_models[ALC268_MODEL_LAST] = {
 
 static struct snd_pci_quirk alc268_cfg_tbl[] = {
        SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
+       SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
        {}
 };
 
@@ -8627,6 +8641,18 @@ static struct alc_config_preset alc268_presets[] = {
                .channel_mode = alc268_modes,
                .input_mux = &alc268_capture_source,
        },
+       [ALC268_TOSHIBA] = {
+               .mixers = { alc268_base_mixer, alc268_capture_alt_mixer },
+               .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs },
+               .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+               .dac_nids = alc268_dac_nids,
+               .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+               .adc_nids = alc268_adc_nids_alt,
+               .hp_nid = 0x03,
+               .num_channel_mode = ARRAY_SIZE(alc268_modes),
+               .channel_mode = alc268_modes,
+               .input_mux = &alc268_capture_source,
+       },
 };
 
 static int patch_alc268(struct hda_codec *codec)
@@ -9916,6 +9942,14 @@ static struct hda_input_mux alc861vd_dallas_capture_source = {
        },
 };
 
+static struct hda_input_mux alc861vd_hp_capture_source = {
+       .num_items = 2,
+       .items = {
+               { "Front Mic", 0x0 },
+               { "ATAPI Mic", 0x1 },
+       },
+};
+
 #define alc861vd_mux_enum_info alc_mux_enum_info
 #define alc861vd_mux_enum_get alc_mux_enum_get
 
@@ -10117,6 +10151,22 @@ static struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
        { } /* end */
 };
 
+/* Pin assignment: Speaker=0x14, Line-out = 0x15,
+ *                 Front Mic=0x18, ATAPI Mic = 0x19,
+ */
+static struct snd_kcontrol_new alc861vd_hp_mixer[] = {
+       HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+       HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+       HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+       HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
+       HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+       HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+       HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+       HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+       
+       { } /* end */
+};
+
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
@@ -10399,6 +10449,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
        SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO),
        SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
        SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG),
+       SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
        {}
 };
 
@@ -10488,7 +10539,21 @@ static struct alc_config_preset alc861vd_presets[] = {
                .input_mux = &alc861vd_dallas_capture_source,
                .unsol_event = alc861vd_dallas_unsol_event,
                .init_hook = alc861vd_dallas_automute,
-       },      
+       },
+       [ALC861VD_HP] = {
+               .mixers = { alc861vd_hp_mixer },
+               .init_verbs = { alc861vd_dallas_verbs, alc861vd_eapd_verbs },
+               .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+               .dac_nids = alc861vd_dac_nids,
+               .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids),
+               .dig_out_nid = ALC861VD_DIGOUT_NID,
+               .adc_nids = alc861vd_adc_nids,
+               .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+               .channel_mode = alc861vd_3stack_2ch_modes,
+               .input_mux = &alc861vd_hp_capture_source,
+               .unsol_event = alc861vd_dallas_unsol_event,
+               .init_hook = alc861vd_dallas_automute,
+       },              
 };
 
 /*
@@ -10849,7 +10914,7 @@ static int alc662_mux_enum_put(struct snd_kcontrol *kcontrol,
        struct alc_spec *spec = codec->spec;
        const struct hda_input_mux *imux = spec->input_mux;
        unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-       static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 };
+       static hda_nid_t capture_mixers[2] = { 0x23, 0x22 };
        hda_nid_t nid = capture_mixers[adc_idx];
        unsigned int *cur_val = &spec->cur_mux[adc_idx];
        unsigned int i, idx;
@@ -11173,11 +11238,7 @@ static struct hda_verb alc662_auto_init_verbs[] = {
        /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
        /* Input mixer */
        {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-       {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-       {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-       /*{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},*/
-       {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-
+       {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
        { }
 };