Reland of Use RtxReceiveStream.

webrtc/call/rtx_receive_stream.h

 /*
  *  Copyright (c) 2017 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.
  */
 
 #ifndef WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
 #define WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
 
 #include <map>
 
 #include "webrtc/call/rtp_packet_sink_interface.h"
 
 namespace webrtc {
 
+class ReceiveStatistics;
+
 // This class is responsible for RTX decapsulation. The resulting media packets
 // are passed on to a sink representing the associated media stream.
 class RtxReceiveStream : public RtpPacketSinkInterface {
  public:
   RtxReceiveStream(RtpPacketSinkInterface* media_sink,
                    std::map<int, int> associated_payload_types,
-                   uint32_t media_ssrc);
+                   uint32_t media_ssrc,
+                   // TODO(nisse): Delete this argument, and
+                   // corresponding member variable, by moving the
+                   // responsibility for rtcp feedback to
+                   // RtpStreamReceiverController.
+                   ReceiveStatistics* rtp_receive_statistics = nullptr);
   ~RtxReceiveStream() override;
   // RtpPacketSinkInterface.
   void OnRtpPacket(const RtpPacketReceived& packet) override;
 
  private:
   RtpPacketSinkInterface* const media_sink_;
   // Map from rtx payload type -> media payload type.
   const std::map<int, int> associated_payload_types_;
   // TODO(nisse): Ultimately, the media receive stream shouldn't care about the
   // ssrc, and we should delete this.
   const uint32_t media_ssrc_;
+  ReceiveStatistics* const rtp_receive_statistics_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_RTX_RECEIVE_STREAM_H_