Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/rtpsenderinterface.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_AUDIOTRACK_H_
-#define TALK_APP_WEBRTC_AUDIOTRACK_H_
+// This file contains interfaces for RtpSenders
+// http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
 
+#ifndef TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_
+#define TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_
+
+#include <string>
+
+#include "talk/app/webrtc/proxy.h"
 #include "talk/app/webrtc/mediastreaminterface.h"
-#include "talk/app/webrtc/mediastreamtrack.h"
-#include "talk/app/webrtc/notifier.h"
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 
 namespace webrtc {
 
-class AudioTrack : public MediaStreamTrack<AudioTrackInterface> {
+class RtpSenderInterface : public rtc::RefCountInterface {
  public:
-  static rtc::scoped_refptr<AudioTrack> Create(
-      const std::string& id, AudioSourceInterface* source);
+  // Returns true if successful in setting the track.
+  // Fails if an audio track is set on a video RtpSender, or vice-versa.
+  virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
+  virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
 
-  // AudioTrackInterface implementation.
-  AudioSourceInterface* GetSource() const override {
-    return audio_source_.get();
-  }
-  // TODO(xians): Implement these methods.
-  void AddSink(AudioTrackSinkInterface* sink) override {}
-  void RemoveSink(AudioTrackSinkInterface* sink) override {}
-  bool GetSignalLevel(int* level) override { return false; }
-  rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor() override {
-    return NULL;
-  }
-  cricket::AudioRenderer* GetRenderer() override { return NULL; }
-
-  // MediaStreamTrack implementation.
-  std::string kind() const override;
+  // 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:
-  AudioTrack(const std::string& label, AudioSourceInterface* audio_source);
+  virtual ~RtpSenderInterface() {}
+};
 
- private:
-  rtc::scoped_refptr<AudioSourceInterface> audio_source_;
-};
+// Define proxy for RtpSenderInterface.
+BEGIN_PROXY_MAP(RtpSender)
+PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
+PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
+PROXY_CONSTMETHOD0(std::string, id)
+END_PROXY()
 
 }  // namespace webrtc
 
-#endif  // TALK_APP_WEBRTC_AUDIOTRACK_H_
+#endif  // TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_