Adding AddTrack/RemoveTrack to native PeerConnection API.

talk/app/webrtc/peerconnectioninterface.h

 /*
  * libjingle
  * Copyright 2012 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 313 matching lines @@
   // Add a new MediaStream to be sent on this PeerConnection.
   // Note that a SessionDescription negotiation is needed before the
   // remote peer can receive the stream.
   virtual bool AddStream(MediaStreamInterface* stream) = 0;
 
   // Remove a MediaStream from this PeerConnection.
   // Note that a SessionDescription negotiation is need before the
   // remote peer is notified.
   virtual void RemoveStream(MediaStreamInterface* stream) = 0;
 
+  // TODO(deadbeef): Make the following two methods pure virtual once
+  // implemented by all subclasses of PeerConnectionInterface.
+  // Add a new MediaStreamTrack to be sent on this PeerConnection.
+  // |streams| indicates which stream labels the track should be associated
+  // with.
+  virtual rtc::scoped_refptr<RtpSenderInterface> AddTrack(
+      MediaStreamTrackInterface* track,
+      std::vector<MediaStreamInterface*> streams) {
+    return nullptr;
+  }
+
+  // Remove an RtpSender from this PeerConnection.
+  // Returns true on success.
+  virtual bool RemoveTrack(RtpSenderInterface* sender) {
+    return false;
+  }
+
   // Returns pointer to the created DtmfSender on success.
   // Otherwise returns NULL.
   virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
       AudioTrackInterface* track) = 0;
 
   // TODO(deadbeef): Make these pure virtual once all subclasses implement them.
   // |kind| must be "audio" or "video".
   // |stream_id| is used to populate the msid attribute; if empty, one will
   // be generated automatically.
   virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
@@ skipping 337 matching lines @@
 CreatePeerConnectionFactory(
     rtc::Thread* worker_thread,
     rtc::Thread* signaling_thread,
     AudioDeviceModule* default_adm,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory);
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_