Hi,
I made new experiments with gstreamer, and with sound output to be more
precise.
Sound events work great ; the buffer size is 1764 and the pipeline looks
like this :
appsrc is-live=true name=ekiga_src !
audio/x-raw-int,rate=44100,channels=2,width=16,depth=16,signed=true,endianness=1234
! audiorate ! volume name=ekiga_volume ! pulsesink
device=alsa_output.pci-0000_00_05.0.analog-stereo
Sound in a call doesn't seem to work ; the buffer size is 160 and the
pipeline looks like this :
appsrc is-live=true name=ekiga_src !
audio/x-raw-int,rate=8000,channels=1,width=16,depth=16,signed=true,endianness=1234
! audiorate ! volume name=ekiga_volume ! pulsesink
device=alsa_output.pci-0000_00_05.0.analog-stereo
As you see differences lie in the buffer size, rate and channels settings.
I wrote "doesn't seem to work" because when the call starts, there's a
small noise ; after that it seems I get silence.
Does the above data look sane?
Snark
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