From dabc9449b79f2b17b6601ea34b22aef330054942 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Peter=20Bostr=C3=B6m?= <pbos@webrtc.org>
Date: Mon, 11 Apr 2016 11:45:14 +0200
Subject: [PATCH] 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                   | 11 +++++++++++
 webrtc/media/engine/webrtcvideoengine2.cc |  2 ++
 webrtc/video/video_send_stream.cc         |  2 ++
 3 files changed, 15 insertions(+)

diff --git a/webrtc/api/rtpsender.cc b/webrtc/api/rtpsender.cc
index 214b4a39ec..58cb18c6cd 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 @@ AudioRtpSender::~AudioRtpSender() {
 }
 
 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 @@ void AudioRtpSender::OnChanged() {
 }
 
 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 @@ bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
 }
 
 void AudioRtpSender::SetSsrc(uint32_t ssrc) {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::SetSsrc");
   if (stopped_ || ssrc == ssrc_) {
     return;
   }
@@ -161,6 +165,7 @@ void AudioRtpSender::SetSsrc(uint32_t ssrc) {
 }
 
 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 @@ RtpParameters AudioRtpSender::GetParameters() const {
 }
 
 bool AudioRtpSender::SetParameters(const RtpParameters& parameters) {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::SetParameters");
   return provider_->SetAudioRtpParameters(ssrc_, parameters);
 }
 
@@ -240,6 +246,7 @@ VideoRtpSender::~VideoRtpSender() {
 }
 
 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 @@ void VideoRtpSender::OnChanged() {
 }
 
 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 @@ bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
 }
 
 void VideoRtpSender::SetSsrc(uint32_t ssrc) {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::SetSsrc");
   if (stopped_ || ssrc == ssrc_) {
     return;
   }
@@ -308,6 +317,7 @@ void VideoRtpSender::SetSsrc(uint32_t ssrc) {
 }
 
 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 @@ RtpParameters VideoRtpSender::GetParameters() const {
 }
 
 bool VideoRtpSender::SetParameters(const RtpParameters& parameters) {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters");
   return provider_->SetVideoRtpParameters(ssrc_, parameters);
 }
 
diff --git a/webrtc/media/engine/webrtcvideoengine2.cc b/webrtc/media/engine/webrtcvideoengine2.cc
index fc87607d20..ec6b033d2f 100644
--- a/webrtc/media/engine/webrtcvideoengine2.cc
+++ b/webrtc/media/engine/webrtcvideoengine2.cc
@@ -883,6 +883,7 @@ webrtc::RtpParameters WebRtcVideoChannel2::GetRtpParameters(
 bool WebRtcVideoChannel2::SetRtpParameters(
     uint32_t ssrc,
     const webrtc::RtpParameters& parameters) {
+  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRtpParameters");
   rtc::CritScope stream_lock(&stream_crit_);
   auto it = send_streams_.find(ssrc);
   if (it == send_streams_.end()) {
@@ -985,6 +986,7 @@ bool WebRtcVideoChannel2::GetSendCodec(VideoCodec* codec) {
 }
 
 bool WebRtcVideoChannel2::SetSend(bool send) {
+  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSend");
   LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false");
   if (send && !send_codec_) {
     LOG(LS_ERROR) << "SetSend(true) called before setting codec.";
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index 6b6b1af346..1c6638ebf4 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -355,6 +355,7 @@ VideoCaptureInput* VideoSendStream::Input() {
 void VideoSendStream::Start() {
   if (payload_router_.active())
     return;
+  TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start");
   vie_encoder_.Pause();
   payload_router_.set_active(true);
   // Was not already started, trigger a keyframe.
@@ -366,6 +367,7 @@ void VideoSendStream::Start() {
 void VideoSendStream::Stop() {
   if (!payload_router_.active())
     return;
+  TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop");
   // TODO(pbos): Make sure the encoder stops here.
   payload_router_.set_active(false);
   vie_receiver_->StopReceive();