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Add more tracing for key frames.

Browse Source 
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
This commit is contained in:
justinlin@chromium.org
2013-05-13 22:59:00 +00:00
parent 941fcc5841
commit 7bfb3a3227
5 changed files with 22 additions and 6 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
View File

@ -918,6 +918,8 @@ void RTCPReceiver::HandlePLI(RTCPUtility::RTCPParserV2& rtcpParser,
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;
}

12
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -330,9 +330,6 @@ int32_t RTPSender::SendOutgoingData(
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 @@ int32_t RTPSender::SendOutgoingData(
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);