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so far, everything has been working fine. all i had to do was turn the Capture sliders up in alsamixer and my microphone was working well enough to use with google talk in gmail and other flash sites. but, i've noticed that the mic does not work when applications try to use it in mono mode, which normally only looks at the left channel. when i record things in audacity using stereo, i see empty input on the left channel and see my voice on the right channel.
normally i wouldn't care, but the only downside is i can't make skype calls. i was wondering if anyone has seen this problem before. is there another driver i should be loading? or is there a way i can make alsa forge a device that copies the actual device's right channel to the fake device's left channel?
$ lspci | grep Audio
00:08.0 Audio device: nVidia Corporation MCP79 High Definition Audio (rev b1)
$ lsmod | grep snd
snd_hda_codec_cirrus 10870 1
snd_hda_intel 22122 0
snd_hda_codec 77927 2 snd_hda_codec_cirrus,snd_hda_intel
snd_hwdep 6325 1 snd_hda_codec
snd_pcm 73856 2 snd_hda_intel,snd_hda_codec
snd_timer 19416 1 snd_pcm
snd 57786 6 snd_hda_codec_cirrus,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer
soundcore 6146 1 snd
snd_page_alloc 7121 2 snd_hda_intel,snd_pcm
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: NVidia [HDA NVidia], device 0: Cirrus Analog [Cirrus Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: NVidia [HDA NVidia], device 1: Cirrus Digital [Cirrus Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0Offline
Hello, I have exactly the same problem. Do you have any new ideas?
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What I did to solve this issue for Skype is the following:
Create $HOME/.asoundrc or /etc/asound.conf file looking like this
pcm.skype {
type route
slave.pcm "hw:0,0"
slave.channels 2
ttable.0.1 4
ttable.1.0 0
}Set in Skype -> Ctrl+O -> Sound Devices -> Microphone -> skype (route)
Please confirm it works for you. If you use Google Talk, please confirm whether it works there as well.
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Thank you woo, I also had that issue and the solution you provided works perfectly.
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And now I have the same problem (I guess) with Linphone. I tried to add to the .asoundrc the same "paragraph" replacing skype with linphone, but it doesn't seem to work. Any idea?
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