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Commit ea1fb29a authored by Kailang Yang's avatar Kailang Yang Committed by Jaroslav Kysela
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ALSA: hda - fix spaces in patch_realtek.c 

Signed-off-by: default avatarKailang Yang <kailang@realtek.com>
Signed-off-by: default avatarTakashi Iwai <tiwai@suse.de>
Signed-off-by: default avatarJaroslav Kysela <perex@perex.cz>
parent f1c0a02f
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Showing with 88 additions and 88 deletions
sound/pci/hda/patch_realtek.c
View file @ ea1fb29a
......@@ -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++] =
......
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