Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/rtpreceiver.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.
  */
 
-// This file is currently stubbed so that Chromium's build files can be updated.
+// This file contains classes that implement RtpReceiverInterface.
+// An RtpReceiver associates a MediaStreamTrackInterface with an underlying
+// transport (provided by AudioProviderInterface/VideoProviderInterface)
+
+#ifndef TALK_APP_WEBRTC_RTPRECEIVER_H_
+#define TALK_APP_WEBRTC_RTPRECEIVER_H_
+
+#include <string>
+
+#include "talk/app/webrtc/mediastreamprovider.h"
+#include "talk/app/webrtc/rtpreceiverinterface.h"
+#include "webrtc/base/basictypes.h"
+
+namespace webrtc {
+
+class AudioRtpReceiver : public ObserverInterface,
+                         public AudioSourceInterface::AudioObserver,
+                         public rtc::RefCountedObject<RtpReceiverInterface> {
+ public:
+  AudioRtpReceiver(AudioTrackInterface* track,
+                   uint32 ssrc,
+                   AudioProviderInterface* provider);
+
+  virtual ~AudioRtpReceiver();
+
+  // ObserverInterface implementation
+  void OnChanged() override;
+
+  // AudioSourceInterface::AudioObserver implementation
+  void OnSetVolume(double volume) override;
+
+  // RtpReceiverInterface implementation
+  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 ssrc_;
+  AudioProviderInterface* provider_;
+  bool cached_track_enabled_;
+};
+
+class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInterface> {
+ public:
+  VideoRtpReceiver(VideoTrackInterface* track,
+                   uint32 ssrc,
+                   VideoProviderInterface* provider);
+
+  virtual ~VideoRtpReceiver();
+
+  // RtpReceiverInterface implementation
+  rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
+    return track_.get();
+  }
+
+  std::string id() const override { return id_; }
+
+  void Stop() override;
+
+ private:
+  std::string id_;
+  rtc::scoped_refptr<VideoTrackInterface> track_;
+  uint32 ssrc_;
+  VideoProviderInterface* provider_;
+};
+
+}  // namespace webrtc
+
+#endif  // TALK_APP_WEBRTC_RTPRECEIVER_H_