Refactor NACK bitrate allocation

webrtc/video/video_send_stream.cc

 /*
  *  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 34 matching lines @@
     Transport* outgoing_transport,
     RtcpIntraFrameObserver* intra_frame_callback,
     RtcpBandwidthObserver* bandwidth_callback,
     TransportFeedbackObserver* transport_feedback_callback,
     RtcpRttStats* rtt_stats,
     RtpPacketSender* paced_sender,
     TransportSequenceNumberAllocator* transport_sequence_number_allocator,
     SendStatisticsProxy* stats_proxy,
     SendDelayStats* send_delay_stats,
     RtcEventLog* event_log,
+    NackRateLimiter* nack_rate_limiter,
     size_t num_modules) {
   RTC_DCHECK_GT(num_modules, 0u);
   RtpRtcp::Configuration configuration;
   ReceiveStatistics* null_receive_statistics = configuration.receive_statistics;
   configuration.audio = false;
   configuration.receiver_only = false;
   configuration.receive_statistics = null_receive_statistics;
   configuration.outgoing_transport = outgoing_transport;
   configuration.intra_frame_callback = intra_frame_callback;
   configuration.bandwidth_callback = bandwidth_callback;
   configuration.transport_feedback_callback = transport_feedback_callback;
   configuration.rtt_stats = rtt_stats;
   configuration.rtcp_packet_type_counter_observer = stats_proxy;
   configuration.paced_sender = paced_sender;
   configuration.transport_sequence_number_allocator =
       transport_sequence_number_allocator;
   configuration.send_bitrate_observer = stats_proxy;
   configuration.send_frame_count_observer = stats_proxy;
   configuration.send_side_delay_observer = stats_proxy;
   configuration.send_packet_observer = send_delay_stats;
   configuration.event_log = event_log;
+  configuration.nack_rate_limiter = nack_rate_limiter;
 
   std::vector<RtpRtcp*> modules;
   for (size_t i = 0; i < num_modules; ++i) {
     RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_rtcp->SetSendingStatus(false);
     rtp_rtcp->SetSendingMediaStatus(false);
     rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
     modules.push_back(rtp_rtcp);
   }
   return modules;
@@ skipping 333 matching lines @@
           config.send_transport,
           &encoder_feedback_,
           bandwidth_observer_.get(),
           congestion_controller_->GetTransportFeedbackObserver(),
           call_stats_->rtcp_rtt_stats(),
           congestion_controller_->pacer(),
           congestion_controller_->packet_router(),
           &stats_proxy_,
           send_delay_stats,
           event_log,
+          congestion_controller_->GetNackRateLimiter(),
           config_.rtp.ssrcs.size())),
       payload_router_(rtp_rtcp_modules_, config.encoder_settings.payload_type),
       input_(&encoder_wakeup_event_,
              config_.local_renderer,
              &stats_proxy_,
              &overuse_detector_) {
   LOG(LS_INFO) << "VideoSendStream: " << config_.ToString();
 
   RTC_DCHECK(!config_.rtp.ssrcs.empty());
   RTC_DCHECK(module_process_thread_);
@@ skipping 456 matching lines @@
                           &module_nack_rate);
     *sent_video_rate_bps += module_video_rate;
     *sent_nack_rate_bps += module_nack_rate;
     *sent_fec_rate_bps += module_fec_rate;
   }
   return 0;
 }
 
 }  // namespace internal
 }  // namespace webrtc