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Cleanup WebRTC tracing

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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
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2
webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
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@ -19,7 +19,6 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
namespace webrtc {
@ -211,7 +210,6 @@ bool RtpReceiverImpl::IncomingRtpPacket(
int packet_length,
PayloadUnion payload_specific,
bool in_order) {
TRACE_EVENT0("webrtc_rtp", "RTPRecv::Packet");
// The rtp_header argument contains the parsed RTP header.
int length = packet_length - rtp_header->paddingLength;