commit dabc9449b79f2b17b6601ea34b22aef330054942
author Peter Boström <pbos@webrtc.org> Mon Apr 11 11:45:14 2016 +0200
committer Peter Boström <pbos@webrtc.org> Mon Apr 11 09:45:43 2016 +0000
tree 21ffcf4a12599fdba490de1b8917461143da2822
parent 18d3d1e466518acf8bf06b9ef935c5cb7a67f750

Add missing tracing to RtpSender objects.

BUG=
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1873793002 .

Cr-Commit-Position: refs/heads/master@{#12311}

webrtc/api/rtpsender.cc

diff --git a/webrtc/api/rtpsender.cc b/webrtc/api/rtpsender.cc
index 214b4a3..58cb18c 100644
--- a/webrtc/api/rtpsender.cc
+++ b/webrtc/api/rtpsender.cc
@@ -13,6 +13,7 @@
 #include "webrtc/api/localaudiosource.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/base/helpers.h"
+#include "webrtc/base/trace_event.h"
 
 namespace webrtc {
 
@@ -86,6 +87,7 @@
 }
 
 void AudioRtpSender::OnChanged() {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::OnChanged");
   RTC_DCHECK(!stopped_);
   if (cached_track_enabled_ != track_->enabled()) {
     cached_track_enabled_ = track_->enabled();
@@ -96,6 +98,7 @@
 }
 
 bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::SetTrack");
   if (stopped_) {
     LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
     return false;
@@ -140,6 +143,7 @@
 }
 
 void AudioRtpSender::SetSsrc(uint32_t ssrc) {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::SetSsrc");
   if (stopped_ || ssrc == ssrc_) {
     return;
   }
@@ -161,6 +165,7 @@
 }
 
 void AudioRtpSender::Stop() {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::Stop");
   // TODO(deadbeef): Need to do more here to fully stop sending packets.
   if (stopped_) {
     return;
@@ -204,6 +209,7 @@
 }
 
 bool AudioRtpSender::SetParameters(const RtpParameters& parameters) {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::SetParameters");
   return provider_->SetAudioRtpParameters(ssrc_, parameters);
 }
 
@@ -240,6 +246,7 @@
 }
 
 void VideoRtpSender::OnChanged() {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::OnChanged");
   RTC_DCHECK(!stopped_);
   if (cached_track_enabled_ != track_->enabled()) {
     cached_track_enabled_ = track_->enabled();
@@ -250,6 +257,7 @@
 }
 
 bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::SetTrack");
   if (stopped_) {
     LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
     return false;
@@ -292,6 +300,7 @@
 }
 
 void VideoRtpSender::SetSsrc(uint32_t ssrc) {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::SetSsrc");
   if (stopped_ || ssrc == ssrc_) {
     return;
   }
@@ -308,6 +317,7 @@
 }
 
 void VideoRtpSender::Stop() {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::Stop");
   // TODO(deadbeef): Need to do more here to fully stop sending packets.
   if (stopped_) {
     return;
@@ -338,6 +348,7 @@
 }
 
 bool VideoRtpSender::SetParameters(const RtpParameters& parameters) {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters");
   return provider_->SetVideoRtpParameters(ssrc_, parameters);
 }