Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/rtpsenderinterface.h

Index: talk/app/webrtc/rtpsenderinterface.h
diff --git a/talk/app/webrtc/mediacontroller.h b/talk/app/webrtc/rtpsenderinterface.h
similarity index 68%
copy from talk/app/webrtc/mediacontroller.h
copy to talk/app/webrtc/rtpsenderinterface.h
index 68798515d00984c586a88fbf8346f75ca96f309b..76345b77ad24fbfe01992506513d8e7917ac20ae 100644
--- a/talk/app/webrtc/mediacontroller.h
+++ b/talk/app/webrtc/rtpsenderinterface.h
@@ -25,25 +25,31 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#ifndef TALK_APP_WEBRTC_MEDIACONTROLLER_H_
-#define TALK_APP_WEBRTC_MEDIACONTROLLER_H_
+// This file contains interfaces for RtpSenders
+// http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
 
-#include "webrtc/base/thread.h"
+#ifndef TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_
+#define TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_
+
+#include <string>
+
+#include "talk/app/webrtc/mediastreaminterface.h"
+#include "webrtc/base/refcount.h"
+#include "webrtc/base/scoped_ref_ptr.h"
 
 namespace webrtc {
-class Call;
-class VoiceEngine;
 
-// The MediaController currently owns shared state between media channels, but
-// in the future will create and own RtpSenders and RtpReceivers.
-class MediaControllerInterface {
+class RtpSenderInterface : public rtc::RefCountInterface {
  public:
-  static MediaControllerInterface* Create(rtc::Thread* worker_thread,
-                                          webrtc::VoiceEngine* voice_engine);
+  virtual void SetTrack(MediaStreamTrackInterface* track) = 0;
+  virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
+  virtual std::string mid() const = 0;

[pthatcher1, 2015/09/17 04:25:46]

I think there is a whole lot of code added to pass around the MID, but I don't
think we actually need it on the object.  I'd say we just remove it all if we
can.

[Taylor Brandstetter, 2015/09/23 00:10:45]

Done.

+  virtual void Stop() = 0;
 
-  virtual ~MediaControllerInterface() {}
-  virtual webrtc::Call* call_w() = 0;
+ protected:
+  virtual ~RtpSenderInterface() {}
 };
+
 }  // namespace webrtc
 
-#endif  // TALK_APP_WEBRTC_MEDIACONTROLLER_H_
+#endif  // TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_