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Diffstat (limited to 'sound/pci/hda/patch_realtek.c')
-rw-r--r--sound/pci/hda/patch_realtek.c103
1 files changed, 51 insertions, 52 deletions
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3a8e58c483df..fea976793ae5 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -275,13 +275,13 @@ struct alc_spec {
*/
unsigned int num_init_verbs;
- char stream_name_analog[16]; /* analog PCM stream */
+ char stream_name_analog[32]; /* analog PCM stream */
struct hda_pcm_stream *stream_analog_playback;
struct hda_pcm_stream *stream_analog_capture;
struct hda_pcm_stream *stream_analog_alt_playback;
struct hda_pcm_stream *stream_analog_alt_capture;
- char stream_name_digital[16]; /* digital PCM stream */
+ char stream_name_digital[32]; /* digital PCM stream */
struct hda_pcm_stream *stream_digital_playback;
struct hda_pcm_stream *stream_digital_capture;
@@ -559,7 +559,7 @@ static int alc_pin_mode_get(struct snd_kcontrol *kcontrol,
/* Find enumerated value for current pinctl setting */
i = alc_pin_mode_min(dir);
- while (alc_pin_mode_values[i] != pinctl && i <= alc_pin_mode_max(dir))
+ while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl)
i++;
*valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
return 0;
@@ -4505,6 +4505,12 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
&dig_nid, 1);
if (err < 0)
continue;
+ if (dig_nid > 0x7f) {
+ printk(KERN_ERR "alc880_auto: invalid dig_nid "
+ "connection 0x%x for NID 0x%x\n", dig_nid,
+ spec->autocfg.dig_out_pins[i]);
+ continue;
+ }
if (!i)
spec->multiout.dig_out_nid = dig_nid;
else {
@@ -6919,9 +6925,6 @@ static struct hda_verb alc882_targa_verbs[] = {
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x01, AC_VERB_SET_GPIO_MASK, 0x03},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x03},
{ } /* end */
};
@@ -7241,7 +7244,8 @@ static struct alc_config_preset alc882_presets[] = {
},
[ALC882_TARGA] = {
.mixers = { alc882_targa_mixer, alc882_chmode_mixer },
- .init_verbs = { alc882_init_verbs, alc882_targa_verbs},
+ .init_verbs = { alc882_init_verbs, alc880_gpio3_init_verbs,
+ alc882_targa_verbs},
.num_dacs = ARRAY_SIZE(alc882_dac_nids),
.dac_nids = alc882_dac_nids,
.dig_out_nid = ALC882_DIGOUT_NID,
@@ -9238,7 +9242,8 @@ static struct alc_config_preset alc883_presets[] = {
},
[ALC883_TARGA_DIG] = {
.mixers = { alc883_targa_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs, alc883_targa_verbs},
+ .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
+ alc883_targa_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
@@ -9251,7 +9256,8 @@ static struct alc_config_preset alc883_presets[] = {
},
[ALC883_TARGA_2ch_DIG] = {
.mixers = { alc883_targa_2ch_mixer},
- .init_verbs = { alc883_init_verbs, alc883_targa_verbs},
+ .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
+ alc883_targa_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.adc_nids = alc883_adc_nids_alt,
@@ -10625,6 +10631,18 @@ static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec,
alc262_lenovo_3000_automute(codec, 1);
}
+static int amp_stereo_mute_update(struct hda_codec *codec, hda_nid_t nid,
+ int dir, int idx, long *valp)
+{
+ int i, change = 0;
+
+ for (i = 0; i < 2; i++, valp++)
+ change |= snd_hda_codec_amp_update(codec, nid, i, dir, idx,
+ HDA_AMP_MUTE,
+ *valp ? 0 : HDA_AMP_MUTE);
+ return change;
+}
+
/* bind hp and internal speaker mute (with plug check) */
static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -10633,13 +10651,8 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change;
- change = snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp ? 0 : HDA_AMP_MUTE);
- change |= snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp ? 0 : HDA_AMP_MUTE);
-
+ change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
+ change |= amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
if (change)
alc262_fujitsu_automute(codec, 0);
return change;
@@ -10674,10 +10687,7 @@ static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change;
- change = snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp ? 0 : HDA_AMP_MUTE);
-
+ change = amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
if (change)
alc262_lenovo_3000_automute(codec, 0);
return change;
@@ -11848,12 +11858,7 @@ static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change;
- change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp[0] ? 0 : HDA_AMP_MUTE);
- change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp[1] ? 0 : HDA_AMP_MUTE);
+ change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
if (change)
alc268_acer_automute(codec, 0);
return change;
@@ -12876,20 +12881,11 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = {
{ }
};
-/* bind volumes of both NID 0x0c and 0x0d */
-static struct hda_bind_ctls alc269_epc_bind_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
static struct snd_kcontrol_new alc269_eeepc_mixer[] = {
- HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_BIND_VOL("LineOut Playback Volume", &alc269_epc_bind_vol),
- HDA_CODEC_MUTE("LineOut Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
{ } /* end */
};
@@ -12902,12 +12898,7 @@ static struct snd_kcontrol_new alc269_epc_capture_mixer[] = {
};
/* FSC amilo */
-static struct snd_kcontrol_new alc269_fujitsu_mixer[] = {
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_BIND_VOL("PCM Playback Volume", &alc269_epc_bind_vol),
- { } /* end */
-};
+#define alc269_fujitsu_mixer alc269_eeepc_mixer
static struct hda_verb alc269_quanta_fl1_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
@@ -13572,6 +13563,8 @@ static int patch_alc269(struct hda_codec *codec)
set_capture_mixer(spec);
set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
+ spec->vmaster_nid = 0x02;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC269_AUTO)
spec->init_hook = alc269_auto_init;
@@ -15166,7 +15159,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
/*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/
- SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS),
+ SND_PCI_QUIRK(0x1179, 0xff01, "Toshiba A135", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS),
SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
@@ -15586,9 +15579,12 @@ static int patch_alc861vd(struct hda_codec *codec)
spec->stream_digital_playback = &alc861vd_pcm_digital_playback;
spec->stream_digital_capture = &alc861vd_pcm_digital_capture;
- spec->adc_nids = alc861vd_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids);
- spec->capsrc_nids = alc861vd_capsrc_nids;
+ if (!spec->adc_nids) {
+ spec->adc_nids = alc861vd_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids);
+ }
+ if (!spec->capsrc_nids)
+ spec->capsrc_nids = alc861vd_capsrc_nids;
set_capture_mixer(spec);
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
@@ -17505,9 +17501,12 @@ static int patch_alc662(struct hda_codec *codec)
spec->stream_digital_playback = &alc662_pcm_digital_playback;
spec->stream_digital_capture = &alc662_pcm_digital_capture;
- spec->adc_nids = alc662_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
- spec->capsrc_nids = alc662_capsrc_nids;
+ if (!spec->adc_nids) {
+ spec->adc_nids = alc662_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
+ }
+ if (!spec->capsrc_nids)
+ spec->capsrc_nids = alc662_capsrc_nids;
if (!spec->cap_mixer)
set_capture_mixer(spec);
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-24 18:52:02 +0100