Drop pacer and retransmission_rate_limiter from RtpStreamReceiver constructor.

webrtc/video/rtp_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 51 matching lines @@
                           public VCMPacketRequestCallback,
                           public video_coding::OnReceivedFrameCallback,
                           public video_coding::OnCompleteFrameCallback,
                           public CallStatsObserver {
  public:
   RtpStreamReceiver(
       vcm::VideoReceiver* video_receiver,
       RemoteBitrateEstimator* remote_bitrate_estimator,
       Transport* transport,
       RtcpRttStats* rtt_stats,
-      PacedSender* paced_sender,
       PacketRouter* packet_router,
       VieRemb* remb,
       const VideoReceiveStream::Config* config,
       ReceiveStatisticsProxy* receive_stats_proxy,
       ProcessThread* process_thread,
-      RateLimiter* retransmission_rate_limiter,
       NackSender* nack_sender,
       KeyFrameRequestSender* keyframe_request_sender,
       video_coding::OnCompleteFrameCallback* complete_frame_callback,
       VCMTiming* timing);
   ~RtpStreamReceiver();
 
   bool AddReceiveCodec(const VideoCodec& video_codec,
                        const std::map<std::string, std::string>& codec_params);
 
   bool AddReceiveCodec(const VideoCodec& video_codec);
@@ skipping 114 matching lines @@
   // TODO(johan): Remove pt_codec_params_ once
   // https://bugs.chromium.org/p/webrtc/issues/detail?id=6883 is resolved.
   // Maps a payload type to a map of out-of-band supplied codec parameters.
   std::map<uint8_t, std::map<std::string, std::string>> pt_codec_params_;
   int16_t last_payload_type_ = -1;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_