Adding AddTrack/RemoveTrack to native PeerConnection API.

talk/app/webrtc/rtpsender.h

 /*
  * 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 60 matching lines @@
 class AudioRtpSender : public ObserverInterface,
                        public rtc::RefCountedObject<RtpSenderInterface> {
  public:
   // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
   // at the appropriate times.
   AudioRtpSender(AudioTrackInterface* track,
                  const std::string& stream_id,
                  AudioProviderInterface* provider,
                  StatsCollector* stats);
 
+  // Randomly generates stream_id.
+  AudioRtpSender(AudioTrackInterface* track,
+                 AudioProviderInterface* provider,
+                 StatsCollector* stats);
+
   // Randomly generates id and stream_id.
   AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats);
 
   virtual ~AudioRtpSender();
 
   // ObserverInterface implementation
   void OnChanged() override;
 
   // RtpSenderInterface implementation
   bool SetTrack(MediaStreamTrackInterface* track) override;
@@ skipping 38 matching lines @@
   rtc::scoped_ptr<LocalAudioSinkAdapter> sink_adapter_;
 };
 
 class VideoRtpSender : public ObserverInterface,
                        public rtc::RefCountedObject<RtpSenderInterface> {
  public:
   VideoRtpSender(VideoTrackInterface* track,
                  const std::string& stream_id,
                  VideoProviderInterface* provider);
 
+  // Randomly generates stream_id.
+  VideoRtpSender(VideoTrackInterface* track, VideoProviderInterface* provider);
+
   // Randomly generates id and stream_id.
   explicit VideoRtpSender(VideoProviderInterface* provider);
 
   virtual ~VideoRtpSender();
 
   // ObserverInterface implementation
   void OnChanged() override;
 
   // RtpSenderInterface implementation
   bool SetTrack(MediaStreamTrackInterface* track) override;
@@ skipping 29 matching lines @@
   VideoProviderInterface* provider_;
   rtc::scoped_refptr<VideoTrackInterface> track_;
   uint32_t ssrc_ = 0;
   bool cached_track_enabled_ = false;
   bool stopped_ = false;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_RTPSENDER_H_