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diff --git a/talk/app/webrtc/rtpsender.h b/talk/app/webrtc/rtpsender.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.
+ */
+
+// This file contains classes that implement RtpSenderInterface.
+// An RtpSender associates a MediaStreamTrackInterface with an underlying
+// transport (provided by AudioProviderInterface/VideoProviderInterface)
+
+#ifndef TALK_APP_WEBRTC_RTPSENDER_H_
+#define TALK_APP_WEBRTC_RTPSENDER_H_
+
+#include <string>
+
+#include "talk/app/webrtc/mediastreamprovider.h"
+#include "talk/app/webrtc/rtpsenderinterface.h"
+#include "talk/media/base/audiorenderer.h"
+#include "webrtc/base/basictypes.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
+
+namespace webrtc {
+
+// LocalAudioSinkAdapter receives data callback as a sink to the local
+// AudioTrack, and passes the data to the sink of AudioRenderer.
+class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
+                              public cricket::AudioRenderer {
+ public:
+  LocalAudioSinkAdapter();
+  virtual ~LocalAudioSinkAdapter();
+
+ private:
+  // AudioSinkInterface implementation.
+  void OnData(const void* audio_data,
+              int bits_per_sample,
+              int sample_rate,
+              int number_of_channels,
+              size_t number_of_frames) override;
+
+  // cricket::AudioRenderer implementation.
+  void SetSink(cricket::AudioRenderer::Sink* sink) override;
+
+  cricket::AudioRenderer::Sink* sink_;
+  // Critical section protecting |sink_|.
+  rtc::CriticalSection lock_;
+};
+
+class AudioRtpSender : public ObserverInterface,
+                       public rtc::RefCountedObject<RtpSenderInterface> {
+ public:
+  AudioRtpSender(AudioTrackInterface* track,
+                 uint32_t ssrc,
+                 AudioProviderInterface* provider);
+
+  virtual ~AudioRtpSender();
+
+  // ObserverInterface implementation
+  void OnChanged() override;
+
+  // RtpSenderInterface implementation
+  bool SetTrack(MediaStreamTrackInterface* track) override;
+  rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
+    return track_.get();
+  }
+
+  std::string id() const override { return id_; }
+
+  void Stop() override;
+
+ private:
+  void Reconfigure();
+
+  std::string id_;
+  rtc::scoped_refptr<AudioTrackInterface> track_;
+  uint32_t ssrc_;
+  AudioProviderInterface* provider_;
+  bool cached_track_enabled_;
+
+  // Used to pass the data callback from the |track_| to the other end of
+  // cricket::AudioRenderer.
+  rtc::scoped_ptr<LocalAudioSinkAdapter> sink_adapter_;
+};
+
+class VideoRtpSender : public ObserverInterface,
+                       public rtc::RefCountedObject<RtpSenderInterface> {
+ public:
+  VideoRtpSender(VideoTrackInterface* track,
+                 uint32_t ssrc,
+                 VideoProviderInterface* provider);
+
+  virtual ~VideoRtpSender();
+
+  // ObserverInterface implementation
+  void OnChanged() override;
+
+  // RtpSenderInterface implementation
+  bool SetTrack(MediaStreamTrackInterface* track) override;
+  rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
+    return track_.get();
+  }
+
+  std::string id() const override { return id_; }
+
+  void Stop() override;
+
+ private:
+  void Reconfigure();
+
+  std::string id_;
+  rtc::scoped_refptr<VideoTrackInterface> track_;
+  uint32_t ssrc_;
+  VideoProviderInterface* provider_;
+  bool cached_track_enabled_;
+};
+
+}  // namespace webrtc
+
+#endif  // TALK_APP_WEBRTC_RTPSENDER_H_