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Review URL: http://webrtc-codereview.appspot.com/353002

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@1432 4adac7df-926f-26a2-2b94-8c16560cd09d
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asapersson@webrtc.org 2012-01-16 11:58:36 +00:00
parent 0b3c35a258
commit 869ce2d441
1 changed files with 7 additions and 3 deletions
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10
src/modules/rtp_rtcp/source/rtp_sender.cc
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@ -1074,15 +1074,19 @@ void RTPSender::ProcessSendToNetwork() {
WebRtc_UWord16 length = IP_PACKET_SIZE;
WebRtc_UWord32 stored_time_ms;
StorageType type;
assert(_packetHistory->GetRTPPacket(seq_num, 0, data_buffer, &length,
&stored_time_ms, &type));
bool found = _packetHistory->GetRTPPacket(seq_num, 0, data_buffer, &length,
&stored_time_ms, &type);
if (!found) {
assert(false);
return;
}
assert(length > 0);
WebRtc_UWord32 diff_ms = _clock.GetTimeInMS() - stored_time_ms;
ModuleRTPUtility::RTPHeaderParser rtpParser(data_buffer, length);
WebRtcRTPHeader rtp_header;
assert(rtpParser.Parse(rtp_header));
rtpParser.Parse(rtp_header);
UpdateTransmissionTimeOffset(data_buffer, length, rtp_header, diff_ms);