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Only report the first rtp packet because it indicates the media has started flowing.

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BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37829004

Cr-Commit-Position: refs/heads/master@{#8189}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8189 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
honghaiz@google.com 2015-01-28 19:48:33 +00:00
parent a094cac11f
commit a67ca1a3bb
1 changed files with 3 additions and 1 deletions
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4
talk/session/media/channel.cc
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@ -554,7 +554,9 @@ void BaseChannel::HandlePacket(bool rtcp, rtc::Buffer* packet,
return; return;
} }
if (!has_received_packet_) { // We are only interested in the first rtp packet because that
// indicates the media has started flowing.
if (!has_received_packet_ && !rtcp) {
has_received_packet_ = true; has_received_packet_ = true;
signaling_thread()->Post(this, MSG_FIRSTPACKETRECEIVED); signaling_thread()->Post(this, MSG_FIRSTPACKETRECEIVED);
} }