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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
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
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@ -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
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@ -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);

2
webrtc/modules/video_coding/main/source/jitter_buffer.cc
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@ -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
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@ -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.

1
webrtc/video_engine/vie_encoder.cc
View File

@ -926,6 +926,7 @@ void ViEEncoder::OnReceivedIntraFrameRequest(uint32_t ssrc) {
// Key frame request from remote side, signal to VCM.
WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideo,
ViEId(engine_id_, channel_id_), "%s", __FUNCTION__);
TRACE_EVENT0("webrtc", "OnKeyFrameRequest");
int idx = 0;
{