[Live-devel] How to set Presentation Time correctly for audio frame in rtsp server?

[Karol Zdancewicz]

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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