Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/rtpsender.h

Index: talk/app/webrtc/rtpsender.h
diff --git a/talk/app/webrtc/rtpsender.h b/talk/app/webrtc/rtpsender.h
index aee77e173c9c438fed345afcd68adc0e7331e955..a0eae5dd6add152d29e51164e6635846fb3d3448 100644
--- a/talk/app/webrtc/rtpsender.h
+++ b/talk/app/webrtc/rtpsender.h
@@ -25,4 +25,116 @@
  * 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 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 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 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 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 ssrc_;
+  VideoProviderInterface* provider_;
+  bool cached_track_enabled_;
+};
+
+}  // namespace webrtc
+
+#endif  // TALK_APP_WEBRTC_RTPSENDER_H_