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<p class="MsoNormal">We are building a server to transcode live JPEG video and AAC audio streams to clients from IP cameras. We have used the testRTSPClient program as a guide to get the JPEG and AAC audio from the cameras and copied the MediaSink buffer into
our own shared buffer in the attempt to encode the JPEG video in the near future. We then pull from the shared buffer using tbb::tasks to stream the JPEG and Audio. The video works fine; however we are having issues getting VLC to play the audio regardless
of if we stream it or save it to a file. Can you please provide some insight as to the frames generated by MPEG4GenericRTPSource so that we can have an idea of how to restream the AAC audio. Otherwise, can you give further insight as to how to format the
raw AAC frames for live555.<o:p></o:p></p>
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