commit 7bfb3a322738fdf79a8d2498fd35c00bcc4617a7
author justinlin@chromium.org <justinlin@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon May 13 22:59:00 2013 +0000
committer justinlin@chromium.org <justinlin@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon May 13 22:59:00 2013 +0000
tree 055599c4a06c67b609e12358a1661428b933624a
parent 941fcc584104ec42c54d5db2ebc50800d629d0de

Add more tracing for key frames.

R=mallinath@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1428004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4015 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc

diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index 3b9659a..bfb5f50 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -918,6 +918,8 @@
                              RTCPPacketInformation& rtcpPacketInformation) {
   const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
   if (_SSRC == rtcpPacket.PLI.MediaSSRC) {
+    TRACE_EVENT_INSTANT0("webrtc_rtp", "PLI");
+
     // Received a signal that we need to send a new key frame.
     rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpPli;
   }