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Add more tracing for key frames.

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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
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2
webrtc/modules/video_coding/main/source/jitter_buffer.cc
View File

@ -981,6 +981,8 @@ bool VCMJitterBuffer::HandleTooOldPackets(uint16_t latest_sequence_number) {
void VCMJitterBuffer::DropPacketsFromNackList(
uint16_t last_decoded_sequence_number) {
TRACE_EVENT_INSTANT1("webrtc", "JB::DropPacketsFromNackList",
"seqnum", last_decoded_sequence_number);
// Erase all sequence numbers from the NACK list which we won't need any
// longer.
missing_sequence_numbers_.erase(missing_sequence_numbers_.begin(),

11
webrtc/modules/video_coding/main/source/video_coding_impl.cc
View File

@ -1248,9 +1248,14 @@ VideoCodingModuleImpl::IncomingPacket(const uint8_t* incomingPayload,
uint32_t payloadLength,
const WebRtcRTPHeader& rtpInfo)
{
TRACE_EVENT2("webrtc", "VCM::Packet",
"seqnum", rtpInfo.header.sequenceNumber,
"type", rtpInfo.frameType);
if (rtpInfo.frameType == kVideoFrameKey) {
TRACE_EVENT1("webrtc", "VCM::PacketKeyFrame",
"seqnum", rtpInfo.header.sequenceNumber);
} else {
TRACE_EVENT2("webrtc", "VCM::Packet",
"seqnum", rtpInfo.header.sequenceNumber,
"type", rtpInfo.frameType);
}
if (incomingPayload == NULL) {
// The jitter buffer doesn't handle non-zero payload lengths for packets
// without payload.