From ea1fb29ac95dea6b3063d6bce512faae9fec6a89 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 26 Aug 2008 12:58:38 +0200 Subject: [PATCH] ALSA: hda - fix spaces in patch_realtek.c Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 176 +++++++++++++++++----------------- 1 file changed, 88 insertions(+), 88 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f2049365e23c..3e594b2e1930 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -201,12 +201,12 @@ enum { ALC883_ACER, ALC883_ACER_ASPIRE, ALC883_MEDION, - ALC883_MEDION_MD2, + ALC883_MEDION_MD2, ALC883_LAPTOP_EAPD, ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, ALC888_LENOVO_MS7195_DIG, - ALC883_HAIER_W66, + ALC883_HAIER_W66, ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, @@ -399,7 +399,7 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, /* * Control the mode of pin widget settings via the mixer. "pc" is used - * instead of "%" to avoid consequences of accidently treating the % as + * instead of "%" to avoid consequences of accidently treating the % as * being part of a format specifier. Maximum allowed length of a value is * 63 characters plus NULL terminator. * @@ -430,7 +430,7 @@ static unsigned char alc_pin_mode_values[] = { #define ALC_PIN_DIR_IN_NOMICBIAS 0x03 #define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04 -/* Info about the pin modes supported by the different pin direction modes. +/* Info about the pin modes supported by the different pin direction modes. * For each direction the minimum and maximum values are given. */ static signed char alc_pin_mode_dir_info[5][2] = { @@ -503,7 +503,7 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, AC_VERB_SET_PIN_WIDGET_CONTROL, alc_pin_mode_values[val]); - /* Also enable the retasking pin's input/output as required + /* Also enable the retasking pin's input/output as required * for the requested pin mode. Enum values of 2 or less are * input modes. * @@ -708,7 +708,7 @@ static void setup_preset(struct alc_spec *spec, i++) spec->init_verbs[spec->num_init_verbs++] = preset->init_verbs[i]; - + spec->channel_mode = preset->channel_mode; spec->num_channel_mode = preset->num_channel_mode; spec->need_dac_fix = preset->need_dac_fix; @@ -719,7 +719,7 @@ static void setup_preset(struct alc_spec *spec, spec->multiout.dac_nids = preset->dac_nids; spec->multiout.dig_out_nid = preset->dig_out_nid; spec->multiout.hp_nid = preset->hp_nid; - + spec->num_mux_defs = preset->num_mux_defs; if (!spec->num_mux_defs) spec->num_mux_defs = 1; @@ -856,7 +856,7 @@ static void alc_subsystem_id(struct hda_codec *codec, if ((ass != codec->bus->pci->subsystem_device) && (ass & 1)) goto do_sku; - /* + /* * 31~30 : port conetcivity * 29~21 : reserve * 20 : PCBEEP input @@ -947,7 +947,7 @@ static void alc_subsystem_id(struct hda_codec *codec, tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0); snd_hda_codec_write(codec, 0x20, 0, - AC_VERB_SET_COEF_INDEX, 7); + AC_VERB_SET_COEF_INDEX, 7); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp | 0x2010); @@ -962,7 +962,7 @@ static void alc_subsystem_id(struct hda_codec *codec, tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0); snd_hda_codec_write(codec, 0x20, 0, - AC_VERB_SET_COEF_INDEX, 7); + AC_VERB_SET_COEF_INDEX, 7); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp | 0x3000); @@ -971,7 +971,7 @@ static void alc_subsystem_id(struct hda_codec *codec, default: break; } - + /* is laptop or Desktop and enable the function "Mute internal speaker * when the external headphone out jack is plugged" */ @@ -1007,6 +1007,7 @@ static void alc_subsystem_id(struct hda_codec *codec, snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT); + spec->unsol_event = alc_sku_unsol_event; } @@ -1297,7 +1298,7 @@ static struct snd_kcontrol_new alc880_six_stack_mixer[] = { * * The system also has a pair of internal speakers, and a headphone jack. * These are both connected to Line2 on the codec, hence to DAC 02. - * + * * There is a variable resistor to control the speaker or headphone * volume. This is a hardware-only device without a software API. * @@ -1825,7 +1826,7 @@ static struct hda_verb alc880_pin_6stack_init_verbs[] = { {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - + { } }; @@ -1870,7 +1871,7 @@ static struct hda_verb alc880_uniwill_init_verbs[] = { /* * Uniwill P53 -* HP = 0x14, InternalSpeaker = 0x15, mic = 0x19, +* HP = 0x14, InternalSpeaker = 0x15, mic = 0x19, */ static struct hda_verb alc880_uniwill_p53_init_verbs[] = { {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ @@ -1969,7 +1970,7 @@ static void alc880_uniwill_p53_hp_automute(struct hda_codec *codec) static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec) { unsigned int present; - + present = snd_hda_codec_read(codec, 0x21, 0, AC_VERB_GET_VOLUME_KNOB_CONTROL, 0); present &= HDA_AMP_VOLMASK; @@ -2051,7 +2052,7 @@ static struct hda_verb alc880_pin_asus_init_verbs[] = { {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - + { } }; @@ -3688,7 +3689,7 @@ static void alc880_auto_init_multi_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int i; - + alc_subsystem_id(codec, 0x15, 0x1b, 0x14); for (i = 0; i < spec->autocfg.line_outs; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; @@ -4483,7 +4484,7 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - /* Ensure Line1 pin widget takes its input from the OUT1 sum bus + /* Ensure Line1 pin widget takes its input from the OUT1 sum bus * when acting as an output. */ {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, @@ -4508,14 +4509,14 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { * stage. */ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute input buffer of pin widget used for Line-in (no equiv + /* Unmute input buffer of pin widget used for Line-in (no equiv * mixer ctrl) */ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Mute capture amp left and right */ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Set ADC connection select to match default mixer setting - line + /* Set ADC connection select to match default mixer setting - line * in (on mic1 pin) */ {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -4569,7 +4570,7 @@ static struct hda_verb alc260_acer_init_verbs[] = { {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum + /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum * bus when acting as outputs. */ {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, @@ -4691,7 +4692,7 @@ static hda_nid_t alc260_test_adc_nids[2] = { 0x04, 0x05, }; /* For testing the ALC260, each input MUX needs its own definition since - * the signal assignments are different. This assumes that the first ADC + * the signal assignments are different. This assumes that the first ADC * is NID 0x04. */ static struct hda_input_mux alc260_test_capture_sources[2] = { @@ -4774,7 +4775,7 @@ static struct snd_kcontrol_new alc260_test_mixer[] = { /* Switches to allow the digital IO pins to be enabled. The datasheet * is ambigious as to which NID is which; testing on laptops which - * make this output available should provide clarification. + * make this output available should provide clarification. */ ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01), ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01), @@ -4810,7 +4811,7 @@ static struct hda_verb alc260_test_init_verbs[] = { {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - /* Ensure mic1, mic2, line1 and line2 pin widgets take input from the + /* Ensure mic1, mic2, line1 and line2 pin widgets take input from the * OUT1 sum bus when acting as an output. */ {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, @@ -4902,7 +4903,7 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid, sw_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); } else return 0; /* N/A */ - + snprintf(name, sizeof(name), "%s Playback Volume", pfx); err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val); if (err < 0) @@ -5008,7 +5009,7 @@ static void alc260_auto_init_multi_out(struct hda_codec *codec) int pin_type = get_pin_type(spec->autocfg.line_out_type); alc260_auto_set_output_and_unmute(codec, nid, pin_type, 0); } - + nid = spec->autocfg.speaker_pins[0]; if (nid) alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0); @@ -5050,7 +5051,7 @@ static struct hda_verb alc260_volume_init_verbs[] = { {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback * mixer widget * Note: PASD motherboards uses the Line In 2 as the input for @@ -5079,7 +5080,7 @@ static struct hda_verb alc260_volume_init_verbs[] = { {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - + { } }; @@ -5938,7 +5939,7 @@ static struct hda_verb alc882_targa_verbs[] = { {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 */ @@ -5954,7 +5955,7 @@ static struct hda_verb alc882_targa_verbs[] = { static void alc882_targa_automute(struct hda_codec *codec) { unsigned int present; - + present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, @@ -5980,7 +5981,7 @@ static struct hda_verb alc882_asus_a7j_verbs[] = { {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 */ @@ -5998,7 +5999,7 @@ static struct hda_verb alc882_asus_a7m_verbs[] = { {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 */ @@ -6324,7 +6325,7 @@ static struct alc_config_preset alc882_presets[] = { .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_init_verbs, alc882_eapd_verbs, @@ -6337,14 +6338,14 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc880_threestack_modes, .need_dac_fix = 1, .input_mux = &alc882_capture_source, - }, + }, }; /* * Pin config fixes */ -enum { +enum { PINFIX_ABIT_AW9D_MAX }; @@ -7261,7 +7262,7 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { .put = alc883_mux_enum_put, }, { } /* end */ -}; +}; static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -7473,7 +7474,7 @@ static struct hda_verb alc883_tagra_verbs[] = { {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 */ @@ -7560,7 +7561,7 @@ static struct hda_channel_mode alc888_3st_hp_modes[2] = { static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec) { unsigned int present; - + present = snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, @@ -7573,7 +7574,7 @@ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec) static void alc888_lenovo_ms7195_rca_automute(struct hda_codec *codec) { unsigned int present; - + present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, @@ -7603,7 +7604,7 @@ static struct hda_verb alc883_medion_md2_verbs[] = { static void alc883_medion_md2_automute(struct hda_codec *codec) { unsigned int present; - + present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, @@ -7758,7 +7759,7 @@ static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec, static void alc883_acer_aspire_automute(struct hda_codec *codec) { unsigned int present; - + present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, @@ -7795,7 +7796,7 @@ static struct hda_verb alc883_acer_eapd_verbs[] = { static void alc888_6st_dell_front_automute(struct hda_codec *codec) { unsigned int present; - + present = snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, @@ -8133,7 +8134,7 @@ static struct alc_config_preset alc883_presets[] = { .input_mux = &alc883_capture_source, .unsol_event = alc883_medion_md2_unsol_event, .init_hook = alc883_medion_md2_automute, - }, + }, [ALC883_LAPTOP_EAPD] = { .mixers = { alc883_base_mixer }, .init_verbs = { alc883_init_verbs, alc882_eapd_verbs }, @@ -8838,10 +8839,10 @@ static struct hda_verb alc262_init_verbs[] = { {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 */ @@ -9467,7 +9468,7 @@ static struct hda_verb alc262_volume_init_verbs[] = { {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)}, @@ -9522,7 +9523,7 @@ static struct hda_verb alc262_HP_BPC_init_verbs[] = { {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - + /* * Set up output mixers (0x0c - 0x0e) */ @@ -9960,7 +9961,7 @@ static struct alc_config_preset alc262_presets[] = { .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, .init_hook = alc262_hippo_automute, - }, + }, [ALC262_ULTRA] = { .mixers = { alc262_ultra_mixer, alc262_ultra_capture_mixer }, .init_verbs = { alc262_ultra_verbs }, @@ -10056,7 +10057,7 @@ static int patch_alc262(struct hda_codec *codec) spec->stream_name_analog = "ALC262 Analog"; spec->stream_analog_playback = &alc262_pcm_analog_playback; spec->stream_analog_capture = &alc262_pcm_analog_capture; - + spec->stream_name_digital = "ALC262 Digital"; spec->stream_digital_playback = &alc262_pcm_digital_playback; spec->stream_digital_capture = &alc262_pcm_digital_capture; @@ -10092,7 +10093,7 @@ static int patch_alc262(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc262_loopbacks; #endif - + return 0; } @@ -10101,7 +10102,7 @@ static int patch_alc262(struct hda_codec *codec) */ #define ALC268_DIGOUT_NID ALC880_DIGOUT_NID #define alc268_modes alc260_modes - + static hda_nid_t alc268_dac_nids[2] = { /* front, hp */ 0x02, 0x03 @@ -10237,7 +10238,6 @@ static struct hda_verb alc268_acer_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, { } }; @@ -10412,7 +10412,7 @@ static struct hda_verb alc268_base_init_verbs[] = { {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Unmute Selector 23h,24h and set the default input to mic-in */ - + {0x23, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x24, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -10611,7 +10611,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, nid = cfg->line_out_pins[0]; if (nid) - alc268_new_analog_output(spec, nid, "Front", 0); + alc268_new_analog_output(spec, nid, "Front", 0); nid = cfg->speaker_pins[0]; if (nid == 0x1d) { @@ -10633,7 +10633,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, if (err < 0) return err; } - return 0; + return 0; } /* create playback/capture controls for input pins */ @@ -10654,7 +10654,7 @@ static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, case 0x1a: idx1 = 2; /* Line In */ break; - case 0x1c: + case 0x1c: idx1 = 3; /* CD */ break; case 0x12: @@ -10666,7 +10666,7 @@ static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, } imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; imux->items[imux->num_items].index = idx1; - imux->num_items++; + imux->num_items++; } return 0; } @@ -10696,11 +10696,11 @@ static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec) } dac_vol1 = dac_vol2 = 0xb000 | 0x40; /* set max volume */ - if (line_nid == 0x14) + if (line_nid == 0x14) dac_vol2 = AMP_OUT_ZERO; else if (line_nid == 0x15) dac_vol1 = AMP_OUT_ZERO; - if (hp_nid == 0x14) + if (hp_nid == 0x14) dac_vol2 = AMP_OUT_ZERO; else if (hp_nid == 0x15) dac_vol1 = AMP_OUT_ZERO; @@ -11026,7 +11026,7 @@ static int patch_alc268(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC268_AUTO) spec->init_hook = alc268_auto_init; - + return 0; } @@ -11284,7 +11284,7 @@ static void alc269_eeepc_dmic_inithook(struct hda_codec *codec) /* unsolicited event for HP jack sensing */ static void alc269_eeepc_amic_unsol_event(struct hda_codec *codec, - unsigned int res) + unsigned int res) { if ((res >> 26) == ALC880_HP_EVENT) alc269_speaker_automute(codec); @@ -11770,7 +11770,7 @@ static struct snd_kcontrol_new alc861_toshiba_mixer[] = { HDA_CODEC_MUTE("Master Playback Switch", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT), - + /*Capture mixer control */ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), @@ -11913,20 +11913,20 @@ static struct hda_verb alc861_base_init_verbs[] = { /* route front mic to ADC1*/ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - + /* Unmute DAC0~3 & spdif out*/ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - + /* Unmute Mixer 14 (mic) 1c (Line in)*/ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - + /* Unmute Stereo Mixer 15 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -11982,13 +11982,13 @@ static struct hda_verb alc861_threestack_init_verbs[] = { {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - + /* Unmute Mixer 14 (mic) 1c (Line in)*/ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - + /* Unmute Stereo Mixer 15 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -12044,13 +12044,13 @@ static struct hda_verb alc861_uniwill_m31_init_verbs[] = { {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - + /* Unmute Mixer 14 (mic) 1c (Line in)*/ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - + /* Unmute Stereo Mixer 15 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -12115,7 +12115,7 @@ static struct hda_verb alc861_asus_init_verbs[] = { {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - + /* Unmute Stereo Mixer 15 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -12152,20 +12152,20 @@ static struct hda_verb alc861_auto_init_verbs[] = { */ /* {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, */ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - + /* Unmute DAC0~3 & spdif out*/ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - + /* Unmute Mixer 14 (mic) 1c (Line in)*/ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - + /* Unmute Stereo Mixer 15 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -12740,7 +12740,7 @@ static int patch_alc861(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc861_loopbacks; #endif - + return 0; } @@ -12994,7 +12994,7 @@ static struct snd_kcontrol_new alc861vd_hp_mixer[] = { 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 */ }; @@ -13139,7 +13139,7 @@ static struct hda_verb alc861vd_lenovo_unsol_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, {} }; @@ -13201,7 +13201,7 @@ static struct hda_verb alc861vd_dallas_verbs[] = { {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -13226,7 +13226,7 @@ static struct hda_verb alc861vd_dallas_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, { } /* end */ @@ -13385,7 +13385,7 @@ static struct alc_config_preset alc861vd_presets[] = { .input_mux = &alc861vd_hp_capture_source, .unsol_event = alc861vd_dallas_unsol_event, .init_hook = alc861vd_dallas_automute, - }, + }, }; /* @@ -14290,12 +14290,12 @@ static void alc662_eeepc_ep20_automute(struct hda_codec *codec) if (present) { /* mute internal speaker */ snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); + HDA_AMP_MUTE, HDA_AMP_MUTE); } else { /* unmute internal speaker if necessary */ mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0); snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); + HDA_AMP_MUTE, mute); } } @@ -14330,16 +14330,16 @@ static void alc663_m51va_mic_automute(struct hda_codec *codec) unsigned int present; present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); + 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); + 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); + 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); + 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); } static void alc663_m51va_unsol_event(struct hda_codec *codec, @@ -14858,7 +14858,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; - + spec->init_verbs[spec->num_init_verbs++] = alc662_auto_init_verbs; if (codec->vendor_id == 0x10ec0663) spec->init_verbs[spec->num_init_verbs++] = -- GitLab