* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
-#include <sound/driver.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/slab.h>
ALC262_HP_BPC_D7000_WL,
ALC262_HP_BPC_D7000_WF,
ALC262_HP_TC_T5735,
+ ALC262_HP_RP5700,
ALC262_BENQ_ED8,
ALC262_SONY_ASSAMD,
ALC262_BENQ_T31,
ALC268_3ST,
ALC268_TOSHIBA,
ALC268_ACER,
+ ALC268_DELL,
+ ALC268_ZEPTO,
#ifdef CONFIG_SND_DEBUG
ALC268_TEST,
#endif
ALC662_5ST_DIG,
ALC662_LENOVO_101E,
ALC662_ASUS_EEEPC_P701,
+ ALC662_ASUS_EEEPC_EP20,
ALC662_AUTO,
ALC662_MODEL_LAST,
};
ALC883_HAIER_W66,
ALC888_6ST_HP,
ALC888_3ST_HP,
+ ALC888_6ST_DELL,
ALC883_MITAC,
ALC883_AUTO,
ALC883_MODEL_LAST,
char *stream_name_analog; /* 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; /* digital PCM stream */
struct hda_pcm_stream *stream_digital_playback;
* max_channels, dacs must be set
* dig_out_nid and hp_nid are optional
*/
+ hda_nid_t alt_dac_nid;
/* capture */
unsigned int num_adc_nids;
/* for pin sensing */
unsigned int sense_updated: 1;
unsigned int jack_present: 1;
+ unsigned int master_sw: 1;
+ /* for virtual master */
+ hda_nid_t vmaster_nid;
+ u32 vmaster_tlv[4];
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_loopback_check loopback;
#endif
/* check sum */
tmp = 0;
for (i = 1; i < 16; i++) {
- if ((ass >> i) && 1)
+ if ((ass >> i) & 1)
tmp++;
}
if (((ass >> 16) & 0xf) != tmp)
break;
}
- /* is laptop and enable the function "Mute internal speaker
+ /* is laptop or Desktop and enable the function "Mute internal speaker
* when the external headphone out jack is plugged"
*/
- if (!(ass & 0x4) || !(ass & 0x8000))
+ if (!(ass & 0x8000))
return;
/*
* 10~8 : Jack location
* when the external headphone out jack is plugged"
*/
if (!spec->autocfg.speaker_pins[0]) {
- if (spec->multiout.dac_nids[0])
+ if (spec->autocfg.line_out_pins[0])
spec->autocfg.speaker_pins[0] =
- spec->multiout.dac_nids[0];
+ spec->autocfg.line_out_pins[0];
else
return;
}
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
- * FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
- * FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
};
-/* FIXME! */
/*
* ALC880 F1734 model
*
static struct snd_kcontrol_new alc880_f1734_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Internal Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, 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),
};
-/* FIXME! */
/*
* ALC880 ASUS model
*
{ } /* end */
};
-/* FIXME! */
/*
* ALC880 ASUS W1V model
*
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
- * FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
/* Uniwill */
static struct snd_kcontrol_new alc880_uniwill_mixer[] = {
- HDA_CODEC_VOLUME("HPhone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("HPhone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("iSpeaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
};
static struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = {
- HDA_CODEC_VOLUME("HPhone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("HPhone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("iSpeaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
+/*
+ * virtual master controls
+ */
+
+/*
+ * slave controls for virtual master
+ */
+static const char *alc_slave_vols[] = {
+ "Front Playback Volume",
+ "Surround Playback Volume",
+ "Center Playback Volume",
+ "LFE Playback Volume",
+ "Side Playback Volume",
+ "Headphone Playback Volume",
+ "Speaker Playback Volume",
+ "Mono Playback Volume",
+ "Line-Out Playback Volume",
+ NULL,
+};
+
+static const char *alc_slave_sws[] = {
+ "Front Playback Switch",
+ "Surround Playback Switch",
+ "Center Playback Switch",
+ "LFE Playback Switch",
+ "Side Playback Switch",
+ "Headphone Playback Switch",
+ "Speaker Playback Switch",
+ "Mono Playback Switch",
+ "IEC958 Playback Switch",
+ NULL,
+};
+
/*
* build control elements
*/
if (err < 0)
return err;
}
+
+ /* if we have no master control, let's create it */
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
+ snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
+ HDA_OUTPUT, spec->vmaster_tlv);
+ err = snd_hda_add_vmaster(codec, "Master Playback Volume",
+ spec->vmaster_tlv, alc_slave_vols);
+ if (err < 0)
+ return err;
+ }
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
+ err = snd_hda_add_vmaster(codec, "Master Playback Switch",
+ NULL, alc_slave_sws);
+ if (err < 0)
+ return err;
+ }
+
return 0;
}
alc880_uniwill_p53_dcvol_automute(codec);
}
-/* FIXME! */
/*
* F1734 pin configuration:
* HP = 0x14, speaker-out = 0x15, mic = 0x18
{ }
};
-/* FIXME! */
/*
* ASUS pin configuration:
* HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a
};
static struct snd_kcontrol_new alc880_lg_mixer[] = {
- /* FIXME: it's not really "master" but front channels */
- HDA_CODEC_VOLUME("Master Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
/*
* Analog capture
*/
-static int alc880_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+static int alc880_alt_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
{
struct alc_spec *spec = codec->spec;
- snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
+ snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number + 1],
stream_tag, 0, format);
return 0;
}
-static int alc880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+static int alc880_alt_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
- snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
+ snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number + 1],
0, 0, 0);
return 0;
}
};
static struct hda_pcm_stream alc880_pcm_analog_capture = {
- .substreams = 2,
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in alc_build_pcms */
+};
+
+static struct hda_pcm_stream alc880_pcm_analog_alt_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in alc_build_pcms */
+};
+
+static struct hda_pcm_stream alc880_pcm_analog_alt_capture = {
+ .substreams = 2, /* can be overridden */
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
.ops = {
- .prepare = alc880_capture_pcm_prepare,
- .cleanup = alc880_capture_pcm_cleanup
+ .prepare = alc880_alt_capture_pcm_prepare,
+ .cleanup = alc880_alt_capture_pcm_cleanup
},
};
};
/* Used by alc_build_pcms to flag that a PCM has no playback stream */
-static struct hda_pcm_stream alc_pcm_null_playback = {
+static struct hda_pcm_stream alc_pcm_null_stream = {
.substreams = 0,
.channels_min = 0,
.channels_max = 0,
codec->num_pcms = 2;
info = spec->pcm_rec + 1;
info->name = spec->stream_name_digital;
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
if (spec->multiout.dig_out_nid &&
spec->stream_digital_playback) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback);
* model, configure a second analog capture-only PCM.
*/
/* Additional Analaog capture for index #2 */
- if (spec->num_adc_nids > 1 && spec->stream_analog_capture &&
- spec->adc_nids) {
+ if ((spec->alt_dac_nid && spec->stream_analog_alt_playback) ||
+ (spec->num_adc_nids > 1 && spec->stream_analog_alt_capture)) {
codec->num_pcms = 3;
info = spec->pcm_rec + 2;
info->name = spec->stream_name_analog;
- /* No playback stream for second PCM */
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = alc_pcm_null_playback;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0;
- if (spec->stream_analog_capture) {
- info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
- info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[1];
+ if (spec->alt_dac_nid) {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+ *spec->stream_analog_alt_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+ spec->alt_dac_nid;
+ } else {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+ alc_pcm_null_stream;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0;
+ }
+ if (spec->num_adc_nids > 1) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ *spec->stream_analog_alt_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
+ spec->adc_nids[1];
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams =
+ spec->num_adc_nids - 1;
+ } else {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ alc_pcm_null_stream;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = 0;
}
}
spec->stream_name_analog = "ALC880 Analog";
spec->stream_analog_playback = &alc880_pcm_analog_playback;
spec->stream_analog_capture = &alc880_pcm_analog_capture;
+ spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture;
spec->stream_name_digital = "ALC880 Digital";
spec->stream_digital_playback = &alc880_pcm_digital_playback;
}
}
+ spec->vmaster_nid = 0x0c;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC880_AUTO)
spec->init_hook = alc880_auto_init;
{ } /* end */
};
+/* update HP, line and mono out pins according to the master switch */
+static void alc260_hp_master_update(struct hda_codec *codec,
+ hda_nid_t hp, hda_nid_t line,
+ hda_nid_t mono)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int val = spec->master_sw ? PIN_HP : 0;
+ /* change HP and line-out pins */
+ snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ val);
+ snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ val);
+ /* mono (speaker) depending on the HP jack sense */
+ val = (val && !spec->jack_present) ? PIN_OUT : 0;
+ snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ val);
+}
+
+static int alc260_hp_master_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ *ucontrol->value.integer.value = spec->master_sw;
+ return 0;
+}
+
+static int alc260_hp_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ int val = !!*ucontrol->value.integer.value;
+ hda_nid_t hp, line, mono;
+
+ if (val == spec->master_sw)
+ return 0;
+ spec->master_sw = val;
+ hp = (kcontrol->private_value >> 16) & 0xff;
+ line = (kcontrol->private_value >> 8) & 0xff;
+ mono = kcontrol->private_value & 0xff;
+ alc260_hp_master_update(codec, hp, line, mono);
+ return 1;
+}
+
+static struct snd_kcontrol_new alc260_hp_output_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc260_hp_master_sw_get,
+ .put = alc260_hp_master_sw_put,
+ .private_value = (0x0f << 16) | (0x10 << 8) | 0x11
+ },
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_verb alc260_hp_unsol_verbs[] = {
+ {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {},
+};
+
+static void alc260_hp_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x10, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
+ alc260_hp_master_update(codec, 0x0f, 0x10, 0x11);
+}
+
+static void alc260_hp_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc260_hp_automute(codec);
+}
+
static struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc260_hp_master_sw_get,
+ .put = alc260_hp_master_sw_put,
+ .private_value = (0x10 << 16) | (0x15 << 8) | 0x11
+ },
HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Aux-In Playback Volume", 0x07, 0x06, HDA_INPUT),
HDA_CODEC_MUTE("Aux-In Playback Switch", 0x07, 0x06, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("iSpeaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("iSpeaker Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT),
{ } /* end */
};
+static struct hda_verb alc260_hp_3013_unsol_verbs[] = {
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {},
+};
+
+static void alc260_hp_3013_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
+ alc260_hp_master_update(codec, 0x10, 0x15, 0x11);
+}
+
+static void alc260_hp_3013_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc260_hp_3013_automute(codec);
+}
+
/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12,
* HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10.
*/
ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN),
HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Internal Speaker Playback Switch", 0x09, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT),
{ } /* end */
};
HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
- HDA_CODEC_VOLUME_MONO("Mono Speaker Playback Volume", 0x0a, 1, 0x0,
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Mono Speaker Playback Switch", 0x0a, 1, 2,
+ HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2,
HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
- * FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
- * FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
};
#endif
-static struct hda_pcm_stream alc260_pcm_analog_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
-};
-
-static struct hda_pcm_stream alc260_pcm_analog_capture = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
-};
+#define alc260_pcm_analog_playback alc880_pcm_analog_alt_playback
+#define alc260_pcm_analog_capture alc880_pcm_analog_capture
#define alc260_pcm_digital_playback alc880_pcm_digital_playback
#define alc260_pcm_digital_capture alc880_pcm_digital_capture
.input_mux = &alc260_capture_source,
},
[ALC260_HP] = {
- .mixers = { alc260_base_output_mixer,
+ .mixers = { alc260_hp_output_mixer,
alc260_input_mixer,
alc260_capture_alt_mixer },
- .init_verbs = { alc260_init_verbs },
+ .init_verbs = { alc260_init_verbs,
+ alc260_hp_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids),
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
+ .unsol_event = alc260_hp_unsol_event,
+ .init_hook = alc260_hp_automute,
},
[ALC260_HP_3013] = {
.mixers = { alc260_hp_3013_mixer,
alc260_input_mixer,
alc260_capture_alt_mixer },
- .init_verbs = { alc260_hp_3013_init_verbs },
+ .init_verbs = { alc260_hp_3013_init_verbs,
+ alc260_hp_3013_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc260_dac_nids),
.dac_nids = alc260_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids),
.num_channel_mode = ARRAY_SIZE(alc260_modes),
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
+ .unsol_event = alc260_hp_3013_unsol_event,
+ .init_hook = alc260_hp_3013_automute,
},
[ALC260_FUJITSU_S702X] = {
.mixers = { alc260_fujitsu_mixer,
spec->stream_digital_playback = &alc260_pcm_digital_playback;
spec->stream_digital_capture = &alc260_pcm_digital_capture;
+ spec->vmaster_nid = 0x08;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC260_AUTO)
spec->init_hook = alc260_auto_init;
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 };
- hda_nid_t nid = capture_mixers[adc_idx];
+ hda_nid_t nid;
unsigned int *cur_val = &spec->cur_mux[adc_idx];
unsigned int i, idx;
+ if (spec->num_adc_nids < 3)
+ nid = capture_mixers[adc_idx + 1];
+ else
+ nid = capture_mixers[adc_idx];
idx = ucontrol->value.enumerated.item[0];
if (idx >= imux->num_items)
idx = imux->num_items - 1;
};
static struct snd_kcontrol_new alc885_mbp3_mixer[] = {
- HDA_CODEC_VOLUME("Master Volume", 0x0c, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Master Switch", 0x0c, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE ("Speaker Switch", 0x14, 0x00, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Out Volume", 0x0d,0x00, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line In Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE ("Line In Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE ("Speaker Playback Switch", 0x14, 0x00, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
+ 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, 0x00, HDA_INPUT),
HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Boost", 0x1a, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT),
{ } /* end */
};
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
- * FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
- * FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG),
+ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */
SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA),
hda_nid_t nid;
nid = spec->autocfg.input_pins[AUTO_PIN_MIC];
- if (nid) {
+ if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) {
err = add_control(spec, ALC_CTL_WIDGET_VOL,
"Mic Boost",
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
return err;
}
nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC];
- if (nid) {
+ if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) {
err = add_control(spec, ALC_CTL_WIDGET_VOL,
"Front Mic Boost",
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
case 0x106b1000: /* iMac 24 */
board_config = ALC885_IMAC24;
break;
+ case 0x106b00a1: /* Macbook */
case 0x106b2c00: /* Macbook Pro rev3 */
board_config = ALC885_MBP3;
break;
spec->stream_name_analog = "ALC882 Analog";
spec->stream_analog_playback = &alc882_pcm_analog_playback;
spec->stream_analog_capture = &alc882_pcm_analog_capture;
+ /* FIXME: setup DAC5 */
+ /*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/
+ spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture;
spec->stream_name_digital = "ALC882 Digital";
spec->stream_digital_playback = &alc882_pcm_digital_playback;
}
}
+ spec->vmaster_nid = 0x0c;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC882_AUTO)
spec->init_hook = alc882_auto_init;
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;
static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("iSpeaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
{ } /* end */
};
+static struct snd_kcontrol_new alc888_6st_dell_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 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("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("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, 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),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .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),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
{ }
};
+static struct hda_verb alc888_6st_dell_verbs[] = {
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 1 (0x0e) */
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 2 (0x0d) */
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { }
+};
+
static struct hda_verb alc888_3st_hp_2ch_init[] = {
{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
{ }
};
+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,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+static void alc888_6st_dell_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ printk("hp_event\n");
+ alc888_6st_dell_front_automute(codec);
+ break;
+ }
+}
+
/*
* generic initialization of ADC, input mixers and output mixers
*/
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
- * FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
/* pcm configuration: identiacal 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
#define alc883_pcm_digital_playback alc880_pcm_digital_playback
#define alc883_pcm_digital_capture alc880_pcm_digital_capture
[ALC883_HAIER_W66] = "haier-w66",
[ALC888_6ST_HP] = "6stack-hp",
[ALC888_3ST_HP] = "3stack-hp",
+ [ALC888_6ST_DELL] = "6stack-dell",
[ALC883_MITAC] = "mitac",
[ALC883_AUTO] = "auto",
};
SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */
+ SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),
SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
+ SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch),
{}
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
},
+ [ALC888_6ST_DELL] = {
+ .mixers = { alc888_6st_dell_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc888_6st_dell_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+ .adc_nids = alc883_adc_nids,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+ .channel_mode = alc883_sixstack_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc888_6st_dell_unsol_event,
+ .init_hook = alc888_6st_dell_front_automute,
+ },
[ALC883_MITAC] = {
.mixers = { alc883_mitac_mixer },
.init_verbs = { alc883_init_verbs, alc883_mitac_verbs },
spec->stream_name_analog = "ALC883 Analog";
spec->stream_analog_playback = &alc883_pcm_analog_playback;
spec->stream_analog_capture = &alc883_pcm_analog_capture;
+ spec->stream_analog_alt_capture = &alc883_pcm_analog_alt_capture;
spec->stream_name_digital = "ALC883 Digital";
spec->stream_digital_playback = &alc883_pcm_digital_playback;
spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
}
+ spec->vmaster_nid = 0x0c;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC883_AUTO)
spec->init_hook = alc883_auto_init;
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
/* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
- HDA_CODEC_MUTE("PC Beelp Playback Switch", 0x0b, 0x05, HDA_INPUT), */
+ HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
/* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
- HDA_CODEC_MUTE("PC Beelp Playback Switch", 0x0b, 0x05, HDA_INPUT), */
+ HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */
/*HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),*/
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
{ } /* end */
};
+/* update HP, line and mono-out pins according to the master switch */
+static void alc262_hp_master_update(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int val = spec->master_sw;
+
+ /* HP & line-out */
+ snd_hda_codec_write_cache(codec, 0x1b, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ val ? PIN_HP : 0);
+ snd_hda_codec_write_cache(codec, 0x15, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ val ? PIN_HP : 0);
+ /* mono (speaker) depending on the HP jack sense */
+ val = val && !spec->jack_present;
+ snd_hda_codec_write_cache(codec, 0x16, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ val ? PIN_OUT : 0);
+}
+
+static void alc262_hp_bpc_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int presence;
+ presence = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = !!(presence & AC_PINSENSE_PRESENCE);
+ alc262_hp_master_update(codec);
+}
+
+static void alc262_hp_bpc_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if ((res >> 26) != ALC880_HP_EVENT)
+ return;
+ alc262_hp_bpc_automute(codec);
+}
+
+static void alc262_hp_wildwest_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int presence;
+ presence = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = !!(presence & AC_PINSENSE_PRESENCE);
+ alc262_hp_master_update(codec);
+}
+
+static void alc262_hp_wildwest_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC880_HP_EVENT)
+ return;
+ alc262_hp_wildwest_automute(codec);
+}
+
+static int alc262_hp_master_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ *ucontrol->value.integer.value = spec->master_sw;
+ return 0;
+}
+
+static int alc262_hp_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ int val = !!*ucontrol->value.integer.value;
+
+ if (val == spec->master_sw)
+ return 0;
+ spec->master_sw = val;
+ alc262_hp_master_update(codec);
+ return 1;
+}
+
static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc262_hp_master_sw_get,
+ .put = alc262_hp_master_sw_put,
+ },
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
-
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
+ HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
};
static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc262_hp_master_sw_get,
+ .put = alc262_hp_master_sw_put,
+ },
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
+ HDA_OUTPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost", 0x1a, 0, HDA_INPUT),
{ } /* end */
};
-static struct hda_bind_ctls alc262_hp_t5735_bind_front_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x0d, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-static struct hda_bind_ctls alc262_hp_t5735_bind_front_sw = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
/* mute/unmute internal speaker according to the hp jack and mute state */
static void alc262_hp_t5735_automute(struct hda_codec *codec, int force)
{
struct alc_spec *spec = codec->spec;
- unsigned int mute;
if (force || !spec->sense_updated) {
unsigned int present;
present = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0);
- spec->jack_present = (present & 0x80000000) != 0;
+ spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
spec->sense_updated = 1;
}
- if (spec->jack_present)
- mute = (0x7080 | ((0)<<8)); /* mute internal speaker */
- else /* unmute internal speaker if necessary */
- mute = (0x7000 | ((0)<<8));
- snd_hda_codec_write(codec, 0x0c, 0,
- AC_VERB_SET_AMP_GAIN_MUTE, mute );
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0, HDA_AMP_MUTE,
+ spec->jack_present ? HDA_AMP_MUTE : 0);
}
static void alc262_hp_t5735_unsol_event(struct hda_codec *codec,
}
static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
- HDA_BIND_VOL("PCM Playback Volume", &alc262_hp_t5735_bind_front_vol),
- HDA_BIND_SW("PCM Playback Switch",&alc262_hp_t5735_bind_front_sw),
- HDA_CODEC_VOLUME("LineOut Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("LineOut Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
{ }
};
+static struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_verb alc262_hp_rp5700_verbs[] = {
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
+ {}
+};
+
+static struct hda_input_mux alc262_hp_rp5700_capture_source = {
+ .num_items = 1,
+ .items = {
+ { "Line", 0x1 },
+ },
+};
+
/* bind hp and internal speaker mute (with plug check) */
static int alc262_sony_master_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+
{ }
};
/* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+
{ }
};
[ALC262_HP_BPC] = "hp-bpc",
[ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000",
[ALC262_HP_TC_T5735] = "hp-tc-t5735",
+ [ALC262_HP_RP5700] = "hp-rp5700",
[ALC262_BENQ_ED8] = "benq",
[ALC262_BENQ_T31] = "benq-t31",
[ALC262_SONY_ASSAMD] = "sony-assamd",
SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x302f, "HP Thin Client T5735",
ALC262_HP_TC_T5735),
+ SND_PCI_QUIRK(0x103c, 0x2817, "HP RP5700", ALC262_HP_RP5700),
SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO),
SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
+ SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU),
SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA),
SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8),
SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31),
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_HP_capture_source,
+ .unsol_event = alc262_hp_bpc_unsol_event,
+ .init_hook = alc262_hp_bpc_automute,
},
[ALC262_HP_BPC_D7000_WF] = {
.mixers = { alc262_HP_BPC_WildWest_mixer },
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_HP_D7000_capture_source,
+ .unsol_event = alc262_hp_wildwest_unsol_event,
+ .init_hook = alc262_hp_wildwest_automute,
},
[ALC262_HP_BPC_D7000_WL] = {
.mixers = { alc262_HP_BPC_WildWest_mixer,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_HP_D7000_capture_source,
+ .unsol_event = alc262_hp_wildwest_unsol_event,
+ .init_hook = alc262_hp_wildwest_automute,
},
[ALC262_HP_TC_T5735] = {
.mixers = { alc262_hp_t5735_mixer },
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hp_t5735_unsol_event,
.init_hook = alc262_hp_t5735_init_hook,
+ },
+ [ALC262_HP_RP5700] = {
+ .mixers = { alc262_hp_rp5700_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_hp_rp5700_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_hp_rp5700_capture_source,
},
[ALC262_BENQ_ED8] = {
.mixers = { alc262_base_mixer },
}
}
+ spec->vmaster_nid = 0x0c;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC262_AUTO)
spec->init_hook = alc262_auto_init;
alc268_acer_automute(codec, 1);
}
+static struct snd_kcontrol_new alc268_dell_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT),
+ { }
+};
+
+static struct hda_verb alc268_dell_verbs[] = {
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { }
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc268_dell_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned int mute;
+
+ present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0);
+ if (present & 0x80000000)
+ mute = HDA_AMP_MUTE;
+ else
+ mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+}
+
+static void alc268_dell_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC880_HP_EVENT)
+ return;
+ alc268_dell_automute(codec);
+}
+
+#define alc268_dell_init_hook alc268_dell_automute
+
/*
* generic initialization of ADC, input mixers and output mixers
*/
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
- * FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
- * FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
HDA_CODEC_VOLUME("MIC2 Capture Volume", 0x19, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("LINE1 Capture Volume", 0x1a, 0x0, HDA_INPUT),
HDA_BIND_MUTE("LINE1 Capture Switch", 0x1a, 2, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCBEEP Playback Volume", 0x1d, 0x0, HDA_INPUT),
+ /* The below appears problematic on some hardwares */
+ /*HDA_CODEC_VOLUME("PCBEEP Playback Volume", 0x1d, 0x0, HDA_INPUT),*/
HDA_CODEC_VOLUME("PCM-IN1 Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("PCM-IN1 Capture Switch", 0x23, 2, HDA_OUTPUT),
HDA_CODEC_VOLUME("PCM-IN2 Capture Volume", 0x24, 0x0, HDA_OUTPUT),
/* pcm configuration: identiacal 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
#define alc268_pcm_digital_playback alc880_pcm_digital_playback
/*
[ALC268_3ST] = "3stack",
[ALC268_TOSHIBA] = "toshiba",
[ALC268_ACER] = "acer",
+ [ALC268_DELL] = "dell",
+ [ALC268_ZEPTO] = "zepto",
#ifdef CONFIG_SND_DEBUG
[ALC268_TEST] = "test",
#endif
static struct snd_pci_quirk alc268_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER),
+ SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER),
+ SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER),
+ SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
+ SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
{}
};
.unsol_event = alc268_acer_unsol_event,
.init_hook = alc268_acer_init_hook,
},
+ [ALC268_DELL] = {
+ .mixers = { alc268_dell_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_dell_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .hp_nid = 0x02,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .unsol_event = alc268_dell_unsol_event,
+ .init_hook = alc268_dell_init_hook,
+ .input_mux = &alc268_capture_source,
+ },
+ [ALC268_ZEPTO] = {
+ .mixers = { alc268_base_mixer, alc268_capture_alt_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_toshiba_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,
+ .dig_out_nid = ALC268_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_capture_source,
+ .unsol_event = alc268_toshiba_unsol_event,
+ .init_hook = alc268_toshiba_automute
+ },
#ifdef CONFIG_SND_DEBUG
[ALC268_TEST] = {
.mixers = { alc268_test_mixer, alc268_capture_mixer },
spec->stream_name_analog = "ALC268 Analog";
spec->stream_analog_playback = &alc268_pcm_analog_playback;
spec->stream_analog_capture = &alc268_pcm_analog_capture;
+ spec->stream_analog_alt_capture = &alc268_pcm_analog_alt_capture;
spec->stream_name_digital = "ALC268 Digital";
spec->stream_digital_playback = &alc268_pcm_digital_playback;
- if (board_config == ALC268_AUTO) {
- 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 = alc268_adc_nids_alt;
- spec->num_adc_nids =
- ARRAY_SIZE(alc268_adc_nids_alt);
- spec->mixers[spec->num_mixers] =
+ 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 = alc268_adc_nids_alt;
+ spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt);
+ spec->mixers[spec->num_mixers] =
alc268_capture_alt_mixer;
- spec->num_mixers++;
- } else {
- spec->adc_nids = alc268_adc_nids;
- spec->num_adc_nids =
- ARRAY_SIZE(alc268_adc_nids);
- spec->mixers[spec->num_mixers] =
- alc268_capture_mixer;
- spec->num_mixers++;
- }
+ spec->num_mixers++;
+ } else {
+ spec->adc_nids = alc268_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids);
+ spec->mixers[spec->num_mixers] =
+ alc268_capture_mixer;
+ spec->num_mixers++;
}
}
+
+ spec->vmaster_nid = 0x02;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC268_AUTO)
spec->init_hook = alc268_auto_init;
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
- * FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
- *FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
spec->stream_digital_playback = &alc861_pcm_digital_playback;
spec->stream_digital_capture = &alc861_pcm_digital_capture;
+ spec->vmaster_nid = 0x03;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC861_AUTO)
spec->init_hook = alc861_auto_init;
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
- *FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
spec->mixers[spec->num_mixers] = alc861vd_capture_mixer;
spec->num_mixers++;
+ spec->vmaster_nid = 0x02;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC861VD_AUTO)
static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("iSpeaker Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("iSpeaker Playback Switch", 0x03, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x03, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
};
static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
- HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("LineOut Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("LineOut Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Line-Out Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("e-Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("e-Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
{ } /* end */
};
+static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = {
+ HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Line-Out Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x03, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("MuteCtrl Playback Switch", 0x0c, 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("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
static struct snd_kcontrol_new alc662_chmode_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
{}
};
+/* Set Unsolicited Event*/
+static struct hda_verb alc662_eeepc_ep20_sue_init_verbs[] = {
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
/*
* generic initialization of ADC, input mixers and output mixers
*/
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
* So call somewhat different..
- * FIXME: the controls appear in the "playback" view!
*/
/* .name = "Capture Source", */
.name = "Input Source",
alc662_eeepc_mic_automute(codec);
}
+static void alc662_eeepc_ep20_automute(struct hda_codec *codec)
+{
+ unsigned int mute;
+ unsigned int present;
+
+ snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ present = (present & 0x80000000) != 0;
+ if (present) {
+ /* mute internal speaker */
+ snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+ 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);
+ }
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc662_eeepc_ep20_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc662_eeepc_ep20_automute(codec);
+}
+
+static void alc662_eeepc_ep20_inithook(struct hda_codec *codec)
+{
+ alc662_eeepc_ep20_automute(codec);
+}
+
#ifdef CONFIG_SND_HDA_POWER_SAVE
#define alc662_loopbacks alc880_loopbacks
#endif
[ALC662_5ST_DIG] = "6stack-dig",
[ALC662_LENOVO_101E] = "lenovo-101e",
[ALC662_ASUS_EEEPC_P701] = "eeepc-p701",
+ [ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20",
[ALC662_AUTO] = "auto",
};
static struct snd_pci_quirk alc662_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701),
+ SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20),
SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
{}
};
.unsol_event = alc662_eeepc_unsol_event,
.init_hook = alc662_eeepc_inithook,
},
+ [ALC662_ASUS_EEEPC_EP20] = {
+ .mixers = { alc662_eeepc_ep20_mixer, alc662_capture_mixer,
+ alc662_chmode_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eeepc_ep20_sue_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc662_adc_nids),
+ .adc_nids = alc662_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
+ .channel_mode = alc662_3ST_6ch_modes,
+ .input_mux = &alc662_lenovo_101e_capture_source,
+ .unsol_event = alc662_eeepc_ep20_unsol_event,
+ .init_hook = alc662_eeepc_ep20_inithook,
+ },
};
struct alc_spec *spec = codec->spec;
int i;
+ alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
for (i = 0; i <= HDA_SIDE; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
int pin_type = get_pin_type(spec->autocfg.line_out_type);
spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
}
+ spec->vmaster_nid = 0x02;
+
codec->patch_ops = alc_patch_ops;
if (board_config == ALC662_AUTO)
spec->init_hook = alc662_auto_init;
projects / linux-2.6 / blobdiff