Cleanup WebRTC tracing

The goal of this change is to:
1. Remove unused tracing events.
2. Organize tracing events to facilitate measurement of end to end latency.

The major change in this CL is to use ASYNC_STEP such that operation
flow can be traced for the same frame.

R=marpan@webrtc.org, pwestin@webrtc.org, turaj@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1761004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4308 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
hclam@chromium.org
2013-07-08 21:31:18 +00:00
parent e80a934b36
commit 1a7b9b94be
15 changed files with 56 additions and 119 deletions

View File

@ -489,10 +489,6 @@ RTCPReceiver::HandleReportBlock(const RTCPUtility::RTCPPacket& rtcpPacket,
_lastReceivedRrMs = _clock->TimeInMilliseconds();
const RTCPPacketReportBlockItem& rb = rtcpPacket.ReportBlockItem;
TRACE_COUNTER_ID1("webrtc_rtp", "RRFractionLost", rb.SSRC, rb.FractionLost);
TRACE_COUNTER_ID1("webrtc_rtp", "RRCumulativeNumOfPacketLost",
rb.SSRC, rb.CumulativeNumOfPacketsLost);
TRACE_COUNTER_ID1("webrtc_rtp", "RRJitter", rb.SSRC, rb.Jitter);
reportBlock->remoteReceiveBlock.remoteSSRC = remoteSSRC;
reportBlock->remoteReceiveBlock.sourceSSRC = rb.SSRC;
reportBlock->remoteReceiveBlock.fractionLost = rb.FractionLost;