Restoring behavior where PeerConnection tracks changes to MediaStreams.

talk/app/webrtc/mediastreamobserver.h

 /*
  * libjingle
  * Copyright 2015 Google Inc.
  *
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  * 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,
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-// 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;
+
+ private:
+  rtc::scoped_refptr<MediaStreamInterface> stream_;
+  AudioTrackVector cached_audio_tracks_;
+  VideoTrackVector cached_video_tracks_;
 };
 
-// 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()
-
 }  // namespace webrtc
 
-#endif  // TALK_APP_WEBRTC_RTPRECEIVERINTERFACE_H_
+#endif  // APP_WEBRTC_MEDIASTREAMOBSERVER_H_