Restoring behavior where PeerConnection tracks changes to MediaStreams.

talk/app/webrtc/mediastreamobserver.h

Index: talk/app/webrtc/mediastreamobserver.h
diff --git a/talk/app/webrtc/rtpreceiverinterface.h b/talk/app/webrtc/mediastreamobserver.h
similarity index 61%
copy from talk/app/webrtc/rtpreceiverinterface.h
copy to talk/app/webrtc/mediastreamobserver.h
index 099699efc4be7a6d729558144131a4163eeed865..5aa7c12550154789811a1581ecebae0c6abf20c6 100644
--- a/talk/app/webrtc/rtpreceiverinterface.h
+++ b/talk/app/webrtc/mediastreamobserver.h
@@ -25,42 +25,41 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-// This file contains interfaces for RtpReceivers
-// http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
+#ifndef APP_WEBRTC_MEDIASTREAMOBSERVER_H_
+#define APP_WEBRTC_MEDIASTREAMOBSERVER_H_
 
-#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 "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/sigslot.h"
 
 namespace webrtc {
 
-class RtpReceiverInterface : public rtc::RefCountInterface {
+// Helper class which will listen for changes to a stream and emit the
+// corresponding signals.
+class MediaStreamObserver : public ObserverInterface {
  public:
-  virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
+  explicit MediaStreamObserver(MediaStreamInterface* stream);
+  ~MediaStreamObserver();
 
-  // 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;
+  const MediaStreamInterface* stream() const { return stream_; }
 
-  virtual void Stop() = 0;
+  void OnChanged() override;
 
- protected:
-  virtual ~RtpReceiverInterface() {}
-};
+  sigslot::signal2<AudioTrackInterface*, MediaStreamInterface*>
+      SignalAudioTrackAdded;
+  sigslot::signal2<AudioTrackInterface*, MediaStreamInterface*>
+      SignalAudioTrackRemoved;
+  sigslot::signal2<VideoTrackInterface*, MediaStreamInterface*>
+      SignalVideoTrackAdded;
+  sigslot::signal2<VideoTrackInterface*, MediaStreamInterface*>
+      SignalVideoTrackRemoved;
 
-// Define proxy for RtpReceiverInterface.
-BEGIN_PROXY_MAP(RtpReceiver)
-PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
-PROXY_CONSTMETHOD0(std::string, id)
-PROXY_METHOD0(void, Stop)
-END_PROXY()
+ private:
+  rtc::scoped_refptr<MediaStreamInterface> stream_;
+  AudioTrackVector cached_audio_tracks_;
+  VideoTrackVector cached_video_tracks_;
+};
 
 }  // namespace webrtc
 
-#endif  // TALK_APP_WEBRTC_RTPRECEIVERINTERFACE_H_
+#endif  // APP_WEBRTC_MEDIASTREAMOBSERVER_H_