Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/rtpreceiverinterface.h

 /*
  * libjingle
- * Copyright 2011 Google Inc.
+ * 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,
  *     this list of conditions and the following disclaimer in the documentation
  *     and/or other materials provided with the distribution.
  *  3. The name of the author may not be used to endorse or promote products
  *     derived from this software without specific prior written permission.
  *
  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#ifndef TALK_APP_WEBRTC_MEDIASTREAMPROXY_H_
-#define TALK_APP_WEBRTC_MEDIASTREAMPROXY_H_
+// This file contains interfaces for RtpReceivers
+// http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
 
+#ifndef TALK_APP_WEBRTC_RTPRECEIVERINTERFACE_H_
+#define TALK_APP_WEBRTC_RTPRECEIVERINTERFACE_H_
+
+#include <string>
+
+#include "talk/app/webrtc/proxy.h"
 #include "talk/app/webrtc/mediastreaminterface.h"
-#include "talk/app/webrtc/proxy.h"
+#include "webrtc/base/refcount.h"
+#include "webrtc/base/scoped_ref_ptr.h"
 
 namespace webrtc {
 
-BEGIN_PROXY_MAP(MediaStream)
-  PROXY_CONSTMETHOD0(std::string, label)
-  PROXY_METHOD0(AudioTrackVector, GetAudioTracks)
-  PROXY_METHOD0(VideoTrackVector, GetVideoTracks)
-  PROXY_METHOD1(rtc::scoped_refptr<AudioTrackInterface>,
-                FindAudioTrack, const std::string&)
-  PROXY_METHOD1(rtc::scoped_refptr<VideoTrackInterface>,
-                FindVideoTrack, const std::string&)
-  PROXY_METHOD1(bool, AddTrack, AudioTrackInterface*)
-  PROXY_METHOD1(bool, AddTrack, VideoTrackInterface*)
-  PROXY_METHOD1(bool, RemoveTrack, AudioTrackInterface*)
-  PROXY_METHOD1(bool, RemoveTrack, VideoTrackInterface*)
-  PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
-  PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)
+class RtpReceiverInterface : public rtc::RefCountInterface {
+ public:
+  virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
+
+  // Not to be confused with "mid", this is a field we can temporarily use
+  // to uniquely identify a receiver until we implement Unified Plan SDP.
+  virtual std::string id() const = 0;
+
+ protected:
+  virtual ~RtpReceiverInterface() {}
+};
+
+// Define proxy for RtpReceiverInterface.
+BEGIN_PROXY_MAP(RtpReceiver)
+PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
+PROXY_CONSTMETHOD0(std::string, id)
 END_PROXY()
 
 }  // namespace webrtc
 
-#endif  // TALK_APP_WEBRTC_MEDIASTREAMPROXY_H_
+#endif  // TALK_APP_WEBRTC_RTPRECEIVERINTERFACE_H_