Hallo all, i still bothering me and you with the old performance problem. I know that we can't imitate correct audio source without knowing what caps audio codec will need. And we can't start the call without have working stream.
Haw about this workaround: - use "adder" (a gstreamer element) as switch between different audio sources. Start with fake source: "audiotestsrc wave=silence ! audioresample ! audioconvert ! ..." - after the call is initiated add new pad to adder - initiate new source: for example pulsesrc - block/remove pad with fakesource - remove fake source. I found one example in python: http://stackoverflow.com/questions/3899666/adding-and-removing-audio-sources-to-from-gstreamer-pipeline-on-the-go _______________________________________________ telepathy mailing list telepathy@lists.freedesktop.org http://lists.freedesktop.org/mailman/listinfo/telepathy
