[Live-devel] Send and receive wav (streaming)

anaino-9 at student.ltu.se anaino-9 at student.ltu.se
Mon Apr 5 16:50:25 PDT 2004

Hi there!

We are currently working on a project where we are supposed to make this 
half-duplex voice communication program. We have managed to send a WAV file from 
one computer to another, but on its way there it gets corrupt somehow. When we 
do this with a MP3 file (according to the test program) it works just fine.

We are using the testMP3Streamer.cpp as a model and we are using this source and 
sink. In the case of SimpleRTPSink, what do we put where it says xxx? We have no 
requirements on audio compression in this stage, so what is the simplest thing 
here? We may want to add an AMR codec later on...

rtpSource = SimpleRTPSource::createNew(*env, &rtpGroupsock, 0, 8000, 

sessionState.sink = SimpleRTPSink::createNew(*env, sessionState.rtpGroupsock, 0, 
8000, "audio/raw", xxx);

Is there any kind of list of arguments for the different kinds of sources and 
sinks? I.e. the SimpleRTPSource takes a bunch of arguments that we don't really 
know what they are.

We'd really appreciate some help with this since the documentation of the 
liveMedia stack is quite non-existent :) 


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