Hello,
I try on many ways but still something freezing my audio, Its like one second
audio and two not. I want to be sure I am doing this properly:
I am sending audio as rtsp server, its PCMU/8000, I give it every second frame
sized: 8000 bytes and send it. In Wireshark I can see that every second I am
sending those packets so in sending looks ok. I am wondering about
fPresentationTime can it be a problem? If I think good fDurationTime should be
one second when packet is 8000 so I tried to set it to 1000000 because of
microseconds. fPresentationTime:
void AudioOutStreamSource::doGetNextFrame()
{
if( m_buffFrames.Size() > 0 ) {
deliverFrame();
}
else {
gettimeofday(&m_tCurrentTime, NULL);
}
}
in deliverFrame before copying buffer: fPresentationTime = m_tCurrentTime;
RtpSink is set as: SimpleRTPSink::createNew(envir(), rtpGroupsock, 0, 8000,
"audio", "PCMU", 1);
Do you see here any issues?
thanks in advance!_______________________________________________
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