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
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;
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index b4cdf69..cfc7c66 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -330,9 +330,6 @@
const uint8_t *payload_data, const uint32_t payload_size,
const RTPFragmentationHeader *fragmentation,
VideoCodecInformation *codec_info, const RTPVideoTypeHeader *rtp_type_hdr) {
- TRACE_EVENT2("webrtc_rtp", "RTPSender::SendOutgoingData",
- "timestsamp", capture_timestamp,
- "frame_type", FrameTypeToString(frame_type));
{
// Drop this packet if we're not sending media packets.
CriticalSectionScoped cs(send_critsect_);
@@ -348,6 +345,15 @@
return -1;
}
+ if (frame_type == kVideoFrameKey) {
+ TRACE_EVENT_INSTANT1("webrtc_rtp", "SendKeyFrame",
+ "timestamp", capture_timestamp);
+ } else {
+ TRACE_EVENT_INSTANT2("webrtc_rtp", "SendFrame",
+ "timestsamp", capture_timestamp,
+ "frame_type", FrameTypeToString(frame_type));
+ }
+
if (audio_configured_) {
assert(frame_type == kAudioFrameSpeech || frame_type == kAudioFrameCN ||
frame_type == kFrameEmpty);