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authorLinus Torvalds <torvalds@linux-foundation.org>2010-06-27 07:39:57 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2010-06-27 07:39:57 -0700
commit29ccb201a28f20885c90954152db8421a2efe779 (patch)
tree21e29b86d002cc9635e8929882f4b31435266bd1 /sound/pci/hda/patch_realtek.c
parentd94b20497b419e8394654f995f94742bd6b06640 (diff)
parentd69f309f0477fc13418f7526639f9ed527ff01e5 (diff)
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: ALSA: usb/endpoint, fix dangling pointer use ALSA: asihpi - Get rid of incorrect "long" types and casts. ASoC: DaVinci: Fix McASP hardware FIFO configuration ALSA: hda - Fix line-in for mb5 model MacBook (Pro) 5,1 / 5,2 ALSA: usb-audio: fix UAC2 control value queries ALSA: usb-audio: parse UAC2 sample rate ranges correctly ALSA: usb-audio: fix control messages for USB_RECIP_INTERFACE ALSA: usb-audio: add check for faulty clock in parse_audio_format_rates_v2() ALSA: hda - Don't check capture source mixer if no ADC is available
Diffstat (limited to 'sound/pci/hda/patch_realtek.c')
-rw-r--r--sound/pci/hda/patch_realtek.c35
1 files changed, 18 insertions, 17 deletions
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index fc767b6b478..f1ce7d7f5aa 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2619,16 +2619,18 @@ static int alc_build_controls(struct hda_codec *codec)
}
/* assign Capture Source enums to NID */
- kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
- if (!kctl)
- kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
- for (i = 0; kctl && i < kctl->count; i++) {
- hda_nid_t *nids = spec->capsrc_nids;
- if (!nids)
- nids = spec->adc_nids;
- err = snd_hda_add_nid(codec, kctl, i, nids[i]);
- if (err < 0)
- return err;
+ if (spec->capsrc_nids || spec->adc_nids) {
+ kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
+ if (!kctl)
+ kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
+ for (i = 0; kctl && i < kctl->count; i++) {
+ hda_nid_t *nids = spec->capsrc_nids;
+ if (!nids)
+ nids = spec->adc_nids;
+ err = snd_hda_add_nid(codec, kctl, i, nids[i]);
+ if (err < 0)
+ return err;
+ }
}
if (spec->cap_mixer) {
const char *kname = kctl ? kctl->id.name : NULL;
@@ -6948,7 +6950,7 @@ static struct hda_input_mux mb5_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x1 },
- { "Line", 0x2 },
+ { "Line", 0x7 },
{ "CD", 0x4 },
},
};
@@ -7469,8 +7471,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = {
HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, 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("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
+ HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT),
@@ -7853,10 +7855,9 @@ static struct hda_verb alc885_mb5_init_verbs[] = {
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)},
{ }
};