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diff --git a/talk/app/webrtc/mediastreamobserver.h b/talk/app/webrtc/mediastreamobserver.h
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+/*
+ * 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_