Move talk/app/webrtc to webrtc/api

talk/app/webrtc/mediastreamobserver.h

-/*
- * libjingle
- * 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_MEDIASTREAMOBSERVER_H_
-#define TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_
-
-#include "talk/app/webrtc/mediastreaminterface.h"
-#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/base/sigslot.h"
-
-namespace webrtc {
-
-// Helper class which will listen for changes to a stream and emit the
-// corresponding signals.
-class MediaStreamObserver : public ObserverInterface {
- public:
-  explicit MediaStreamObserver(MediaStreamInterface* stream);
-  ~MediaStreamObserver();
-
-  const MediaStreamInterface* stream() const { return stream_; }
-
-  void OnChanged() override;
-
-  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_;
-};
-
-}  // namespace webrtc
-
-#endif  // TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_