[streaming] send in rtp/rtcp modules

[Axel Voss]

Hi,

I transcode video mpeg 2 ts to h264 and audio ac3 to aac. The output is a rtp stream without any muxer (nal in rtp). With the suitable sdp file the software receiver like vlc or quicktime can decode and play it fine. But now a have a hardware like a set top box. The result here is an unsynchronized audio/video with a stottering video. This decoder hardware works fine with other encoders. Wireshark shows me the differences. The other encoders send the rtcp packets for audio and video as a pair at the same time and near the same distance. I see that in the rtp module the ThreadSend function calls the SendRTCP function. I assume that the audio und video threads works relatively seperate (without any synchronize). I would like test a send out with paired audio/video rtcp packets. I understand the code and processes, but ones is a black hole: the send () function. Where is the definition/implementation of this function?
 
Thanks
 
Axel


      
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