I have here a .asoundrc for skype. I have no idea anymore why I created it, but it had to be:
Code:
pcm.skype {
type asym
playback.pcm "skypeout"
capture.pcm "skypein"
}
pcm.skypein {
# Convert from 8-bit unsigned mono (default format set by aoss when
# /dev/dsp is opened) to 16-bit signed stereo (expected by dsnoop)
#
# We can't just use a "plug" plugin because although the open will
# succeed, the buffer sizes will be wrong and we'll hear no sound at
# all.
type route
slave {
pcm "skypedsnoop"
format S16_LE
}
ttable {
0 {0 0.5}
1 {0 0.5}
}
}
pcm.skypeout {
# Just pass this on to the system dmix
type plug
slave {
pcm "dmix"
}
}
pcm.skypedsnoop {
type dsnoop
ipc_key 1133
slave {
# "Magic" buffer values to get skype audio to work
# If these are not set, opening /dev/dsp succeeds but no sound
# will be heard. According to the alsa developers this is due
# to skype abusing the OSS API.
pcm "hw:0,0"
period_size 256
periods 16
buffer_size 16384
}
bindings {
0 0
}
}
hubi |