Adding AddTrack/RemoveTrack to native PeerConnection API.

talk/app/webrtc/rtpsender.cc

 /*
  * libjingle
  * Copyright 2015 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 57 matching lines @@
       provider_(provider),
       stats_(stats),
       track_(track),
       cached_track_enabled_(track->enabled()),
       sink_adapter_(new LocalAudioSinkAdapter()) {
   RTC_DCHECK(provider != nullptr);
   track_->RegisterObserver(this);
   track_->AddSink(sink_adapter_.get());
 }
 
+AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
+                               AudioProviderInterface* provider,
+                               StatsCollector* stats)
+    : id_(track->id()),
+      stream_id_(rtc::CreateRandomUuid()),
+      provider_(provider),
+      stats_(stats),
+      track_(track),
+      cached_track_enabled_(track->enabled()),
+      sink_adapter_(new LocalAudioSinkAdapter()) {
+  RTC_DCHECK(provider != nullptr);
+  track_->RegisterObserver(this);
+  track_->AddSink(sink_adapter_.get());
+}
+
 AudioRtpSender::AudioRtpSender(AudioProviderInterface* provider,
                                StatsCollector* stats)
     : id_(rtc::CreateRandomUuid()),
       stream_id_(rtc::CreateRandomUuid()),
       provider_(provider),
       stats_(stats),
       sink_adapter_(new LocalAudioSinkAdapter()) {}
 
 AudioRtpSender::~AudioRtpSender() {
   Stop();
@@ skipping 116 matching lines @@
                                VideoProviderInterface* provider)
     : id_(track->id()),
       stream_id_(stream_id),
       provider_(provider),
       track_(track),
       cached_track_enabled_(track->enabled()) {
   RTC_DCHECK(provider != nullptr);
   track_->RegisterObserver(this);
 }
 
+VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
+                               VideoProviderInterface* provider)
+    : id_(track->id()),
+      stream_id_(rtc::CreateRandomUuid()),
+      provider_(provider),
+      track_(track),
+      cached_track_enabled_(track->enabled()) {
+  RTC_DCHECK(provider != nullptr);
+  track_->RegisterObserver(this);
+}
+
 VideoRtpSender::VideoRtpSender(VideoProviderInterface* provider)
     : id_(rtc::CreateRandomUuid()),
       stream_id_(rtc::CreateRandomUuid()),
       provider_(provider) {}
 
 VideoRtpSender::~VideoRtpSender() {
   Stop();
 }
 
 void VideoRtpSender::OnChanged() {
@@ skipping 87 matching lines @@
   RTC_DCHECK(!stopped_ && can_send_track());
   const cricket::VideoOptions* options = nullptr;
   VideoSourceInterface* source = track_->GetSource();
   if (track_->enabled() && source) {
     options = source->options();
   }
   provider_->SetVideoSend(ssrc_, track_->enabled(), options);
 }
 
 }  // namespace webrtc