Add missing tracing to RtpSender objects.

webrtc/api/rtpsender.cc

 /*
  *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/api/rtpsender.h"
 
 #include "webrtc/api/localaudiosource.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/base/helpers.h"
+#include "webrtc/base/trace_event.h"
 
 namespace webrtc {
 
 LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {}
 
 LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
   rtc::CritScope lock(&lock_);
   if (sink_)
     sink_->OnClose();
 }
@@ skipping 53 matching lines @@
       stream_id_(rtc::CreateRandomUuid()),
       provider_(provider),
       stats_(stats),
       sink_adapter_(new LocalAudioSinkAdapter()) {}
 
 AudioRtpSender::~AudioRtpSender() {
   Stop();
 }
 
 void AudioRtpSender::OnChanged() {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::OnChanged");
   RTC_DCHECK(!stopped_);
   if (cached_track_enabled_ != track_->enabled()) {
     cached_track_enabled_ = track_->enabled();
     if (can_send_track()) {
       SetAudioSend();
     }
   }
 }
 
 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;
   }
   if (track && track->kind() != MediaStreamTrackInterface::kAudioKind) {
     LOG(LS_ERROR) << "SetTrack called on audio RtpSender with " << track->kind()
                   << " track.";
     return false;
   }
   AudioTrackInterface* audio_track = static_cast<AudioTrackInterface*>(track);
@@ skipping 24 matching lines @@
       stats_->AddLocalAudioTrack(track_.get(), ssrc_);
     }
   } else if (prev_can_send_track) {
     cricket::AudioOptions options;
     provider_->SetAudioSend(ssrc_, false, options, nullptr);
   }
   return true;
 }
 
 void AudioRtpSender::SetSsrc(uint32_t ssrc) {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::SetSsrc");
   if (stopped_ || ssrc == ssrc_) {
     return;
   }
   // If we are already sending with a particular SSRC, stop sending.
   if (can_send_track()) {
     cricket::AudioOptions options;
     provider_->SetAudioSend(ssrc_, false, options, nullptr);
     if (stats_) {
       stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
     }
   }
   ssrc_ = ssrc;
   if (can_send_track()) {
     SetAudioSend();
     if (stats_) {
       stats_->AddLocalAudioTrack(track_.get(), ssrc_);
     }
   }
 }
 
 void AudioRtpSender::Stop() {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::Stop");
   // TODO(deadbeef): Need to do more here to fully stop sending packets.
   if (stopped_) {
     return;
   }
   if (track_) {
     track_->RemoveSink(sink_adapter_.get());
     track_->UnregisterObserver(this);
   }
   if (can_send_track()) {
     cricket::AudioOptions options;
@@ skipping 23 matching lines @@
   cricket::AudioSource* source = sink_adapter_.get();
   ASSERT(source != nullptr);
   provider_->SetAudioSend(ssrc_, track_->enabled(), options, source);
 }
 
 RtpParameters AudioRtpSender::GetParameters() const {
   return provider_->GetAudioRtpParameters(ssrc_);
 }
 
 bool AudioRtpSender::SetParameters(const RtpParameters& parameters) {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::SetParameters");
   return provider_->SetAudioRtpParameters(ssrc_, parameters);
 }
 
 VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
                                const std::string& stream_id,
                                VideoProviderInterface* provider)
     : id_(track->id()),
       stream_id_(stream_id),
       provider_(provider),
       track_(track),
@@ skipping 16 matching lines @@
 VideoRtpSender::VideoRtpSender(VideoProviderInterface* provider)
     : id_(rtc::CreateRandomUuid()),
       stream_id_(rtc::CreateRandomUuid()),
       provider_(provider) {}
 
 VideoRtpSender::~VideoRtpSender() {
   Stop();
 }
 
 void VideoRtpSender::OnChanged() {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::OnChanged");
   RTC_DCHECK(!stopped_);
   if (cached_track_enabled_ != track_->enabled()) {
     cached_track_enabled_ = track_->enabled();
     if (can_send_track()) {
       SetVideoSend();
     }
   }
 }
 
 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;
   }
   if (track && track->kind() != MediaStreamTrackInterface::kVideoKind) {
     LOG(LS_ERROR) << "SetTrack called on video RtpSender with " << track->kind()
                   << " track.";
     return false;
   }
   VideoTrackInterface* video_track = static_cast<VideoTrackInterface*>(track);
@@ skipping 22 matching lines @@
     provider_->SetSource(ssrc_, track_);
     SetVideoSend();
   } else if (prev_can_send_track) {
     provider_->SetSource(ssrc_, nullptr);
     provider_->SetVideoSend(ssrc_, false, nullptr);
   }
   return true;
 }
 
 void VideoRtpSender::SetSsrc(uint32_t ssrc) {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::SetSsrc");
   if (stopped_ || ssrc == ssrc_) {
     return;
   }
   // If we are already sending with a particular SSRC, stop sending.
   if (can_send_track()) {
     provider_->SetSource(ssrc_, nullptr);
     provider_->SetVideoSend(ssrc_, false, nullptr);
   }
   ssrc_ = ssrc;
   if (can_send_track()) {
     provider_->SetSource(ssrc_, track_);
     SetVideoSend();
   }
 }
 
 void VideoRtpSender::Stop() {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::Stop");
   // TODO(deadbeef): Need to do more here to fully stop sending packets.
   if (stopped_) {
     return;
   }
   if (track_) {
     track_->UnregisterObserver(this);
   }
   if (can_send_track()) {
     provider_->SetSource(ssrc_, nullptr);
     provider_->SetVideoSend(ssrc_, false, nullptr);
@@ skipping 10 matching lines @@
     options.video_noise_reduction = source->needs_denoising();
   }
   provider_->SetVideoSend(ssrc_, track_->enabled(), &options);
 }
 
 RtpParameters VideoRtpSender::GetParameters() const {
   return provider_->GetVideoRtpParameters(ssrc_);
 }
 
 bool VideoRtpSender::SetParameters(const RtpParameters& parameters) {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters");
   return provider_->SetVideoRtpParameters(ssrc_, parameters);
 }
 
 }  // namespace webrtc