Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/rtpsenderinterface.h

Index: talk/app/webrtc/rtpsenderinterface.h
diff --git a/talk/app/webrtc/audiotrack.h b/talk/app/webrtc/rtpsenderinterface.h
similarity index 56%
copy from talk/app/webrtc/audiotrack.h
copy to talk/app/webrtc/rtpsenderinterface.h
index 750f272ba2b79c8dcd4ccc14de83b55ca43d4ec4..fca98f21db5823397d789922bd21cc79b1d433b0 100644
--- a/talk/app/webrtc/audiotrack.h
+++ b/talk/app/webrtc/rtpsenderinterface.h
@@ -1,6 +1,6 @@
 /*
  * 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:
@@ -25,45 +25,46 @@
  * 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; }
+  // 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;
 
-  // MediaStreamTrack implementation.
-  std::string kind() const override;
+  virtual void Stop() = 0;
 
  protected:
-  AudioTrack(const std::string& label, AudioSourceInterface* audio_source);
-
- private:
-  rtc::scoped_refptr<AudioSourceInterface> audio_source_;
+  virtual ~RtpSenderInterface() {}
 };
 
+// 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)
+PROXY_METHOD0(void, Stop)
+END_PROXY()
+
 }  // namespace webrtc
 
-#endif  // TALK_APP_WEBRTC_AUDIOTRACK_H_
+#endif  // TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_