Use RtxReceiveStream.

webrtc/video/rtp_video_stream_receiver.h

 /*
  *  Copyright (c) 2012 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 167 matching lines @@
   RemoteNtpTimeEstimator ntp_estimator_;
   RTPPayloadRegistry rtp_payload_registry_;
 
   const std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
   const std::unique_ptr<RtpReceiver> rtp_receiver_;
   const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
   std::unique_ptr<UlpfecReceiver> ulpfec_receiver_;
 
   rtc::SequencedTaskChecker worker_task_checker_;
   bool receiving_ GUARDED_BY(worker_task_checker_);
-  uint8_t restored_packet_[IP_PACKET_SIZE] GUARDED_BY(worker_task_checker_);
-  bool restored_packet_in_use_ GUARDED_BY(worker_task_checker_);
   int64_t last_packet_log_ms_ GUARDED_BY(worker_task_checker_);
 
   const std::unique_ptr<RtpRtcp> rtp_rtcp_;
 
   // Members for the new jitter buffer experiment.
   video_coding::OnCompleteFrameCallback* complete_frame_callback_;
   KeyFrameRequestSender* keyframe_request_sender_;
   VCMTiming* timing_;
   std::unique_ptr<NackModule> nack_module_;
   rtc::scoped_refptr<video_coding::PacketBuffer> packet_buffer_;
@@ skipping 10 matching lines @@
 
   bool has_received_frame_;
 
   std::vector<RtpPacketSinkInterface*> secondary_sinks_
       GUARDED_BY(worker_task_checker_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_