Hi, I'm trying to stream an OPUS encoded audio signal, but I could not find out yet how to implement this.
I'm using a H264 encoder and a class derived from FramedSource - that one works perfectly and I can see the stream in VLC. So I wrote my own OPUS encoder along the same lines with "frames" of about 20ms each, but how do I set up an appropriate RTPAudioSink? m_audioSink = MPEG1or2AudioRTPSink::createNew(*m_env, m_rtpGroupsockAud); works perfectly if I use an MP3FileSource as in the demos, but does not work here, so I tried to go with SimpleRTPSink - however, what parameters do I need to set such that the packets really "leave" and are correctly interpreted by a receiving application like VLC? Any hints are highly appreciated. Best... _______________________________________________ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel
