Introduce RtpStreamReceiver and RtpStreamReceiverControllerInterface.

webrtc/video/video_receive_stream.h

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 18 matching lines @@
 #include "webrtc/video/transport_adapter.h"
 #include "webrtc/video/video_stream_decoder.h"
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {
 
 class CallStats;
 class IvfFileWriter;
 class ProcessThread;
 class RTPFragmentationHeader;
+class RtpStreamReceiverInterface;
+class RtpStreamReceiverControllerInterface;
 class VCMTiming;
 class VCMJitterEstimator;
 
 namespace internal {
 
 class VideoReceiveStream : public webrtc::VideoReceiveStream,
                            public rtc::VideoSinkInterface<VideoFrame>,
                            public EncodedImageCallback,
                            public NackSender,
                            public KeyFrameRequestSender,
                            public video_coding::OnCompleteFrameCallback,
-                           public Syncable,
-                           public RtpPacketSinkInterface {
+                           public Syncable {
  public:
-  VideoReceiveStream(int num_cpu_cores,
+  VideoReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller,
+                     int num_cpu_cores,
                      PacketRouter* packet_router,
                      VideoReceiveStream::Config config,
                      ProcessThread* process_thread,
                      CallStats* call_stats);
   ~VideoReceiveStream() override;
 
   const Config& config() const { return config_; }
 
   void SignalNetworkState(NetworkState state);
   bool DeliverRtcp(const uint8_t* packet, size_t length);
 
   void SetSync(Syncable* audio_syncable);
 
   // Implements webrtc::VideoReceiveStream.
   void Start() override;
   void Stop() override;
 
   webrtc::VideoReceiveStream::Stats GetStats() const override;
 
   // Takes ownership of the file, is responsible for closing it later.
   // Calling this method will close and finalize any current log.
   // Giving rtc::kInvalidPlatformFileValue disables logging.
   // If a frame to be written would make the log too large the write fails and
   // the log is closed and finalized. A |byte_limit| of 0 means no limit.
   void EnableEncodedFrameRecording(rtc::PlatformFile file,
                                    size_t byte_limit) override;
 
-  // RtpPacketSinkInterface.
-  void OnRtpPacket(const RtpPacketReceived& packet) override;
-
   // Implements rtc::VideoSinkInterface<VideoFrame>.
   void OnFrame(const VideoFrame& video_frame) override;
 
   // Implements EncodedImageCallback.
   EncodedImageCallback::Result OnEncodedImage(
       const EncodedImage& encoded_image,
       const CodecSpecificInfo* codec_specific_info,
       const RTPFragmentationHeader* fragmentation) override;
 
   // Implements NackSender.
@@ skipping 36 matching lines @@
   RtpVideoStreamReceiver rtp_video_stream_receiver_;
   std::unique_ptr<VideoStreamDecoder> video_stream_decoder_;
   RtpStreamsSynchronizer rtp_stream_sync_;
 
   rtc::CriticalSection ivf_writer_lock_;
   std::unique_ptr<IvfFileWriter> ivf_writer_ GUARDED_BY(ivf_writer_lock_);
 
   // Members for the new jitter buffer experiment.
   std::unique_ptr<VCMJitterEstimator> jitter_estimator_;
   std::unique_ptr<video_coding::FrameBuffer> frame_buffer_;
+
+  std::unique_ptr<RtpStreamReceiverInterface> media_receiver_;
+  std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_;
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_