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

@ -14,7 +14,6 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
namespace webrtc {
@ -284,10 +283,6 @@ int32_t ReceiveStatisticsImpl::TimeUntilNextProcess() {
int32_t ReceiveStatisticsImpl::Process() {
incoming_bitrate_.Process();
TRACE_COUNTER_ID1("webrtc_rtp", "RTPReceiverBitrate", ssrc_,
incoming_bitrate_.BitrateLast());
TRACE_COUNTER_ID1("webrtc_rtp", "RTPReceiverPacketRate", ssrc_,
incoming_bitrate_.PacketRate());
return 0;
}