Refactor NACK bitrate allocation

webrtc/video/rtp_stream_receiver.cc

 /*
  *  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 40 matching lines @@
       rtcp_packet_type_counter_observer;
   configuration.paced_sender = paced_sender;
   configuration.transport_sequence_number_allocator =
       transport_sequence_number_allocator;
   configuration.send_bitrate_observer = nullptr;
   configuration.send_frame_count_observer = nullptr;
   configuration.send_side_delay_observer = nullptr;
   configuration.send_packet_observer = nullptr;
   configuration.bandwidth_callback = nullptr;
   configuration.transport_feedback_callback = nullptr;
+  configuration.retransmission_rate_limiter = nullptr;
 
   std::unique_ptr<RtpRtcp> rtp_rtcp(RtpRtcp::CreateRtpRtcp(configuration));
   rtp_rtcp->SetSendingStatus(false);
   rtp_rtcp->SetSendingMediaStatus(false);
   rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
 
   return rtp_rtcp;
 }
 
 static const int kPacketLogIntervalMs = 10000;
@@ skipping 107 matching lines @@
                                    config_.rtp.fec.red_payload_type,
                                    config_.rtp.fec.ulpfec_payload_type);
   }
 
   if (config_.rtp.rtcp_xr.receiver_reference_time_report)
     rtp_rtcp_->SetRtcpXrRrtrStatus(true);
 
   // Stats callback for CNAME changes.
   rtp_rtcp_->RegisterRtcpStatisticsCallback(receive_stats_proxy);
 
-  process_thread_->RegisterModule(rtp_receive_statistics_.get());
   process_thread_->RegisterModule(rtp_rtcp_.get());
 }
 
 RtpStreamReceiver::~RtpStreamReceiver() {
-  process_thread_->DeRegisterModule(rtp_receive_statistics_.get());
   process_thread_->DeRegisterModule(rtp_rtcp_.get());
 
   packet_router_->RemoveRtpModule(rtp_rtcp_.get());
   rtp_rtcp_->SetREMBStatus(false);
   remb_->RemoveReceiveChannel(rtp_rtcp_.get());
   UpdateHistograms();
 }
 
 bool RtpStreamReceiver::SetReceiveCodec(const VideoCodec& video_codec) {
   int8_t old_pltype = -1;
@@ skipping 332 matching lines @@
     const std::string& extension, int id) {
   // One-byte-extension local identifiers are in the range 1-14 inclusive.
   RTC_DCHECK_GE(id, 1);
   RTC_DCHECK_LE(id, 14);
   RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
   RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
       StringToRtpExtensionType(extension), id));
 }
 
 }  // namespace webrtc