Add NACK rate throttling for audio channels.

webrtc/video/video_receive_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.
  */
 
 #include "webrtc/video/video_receive_stream.h"
 
 #include <stdlib.h>
 
 #include <set>
 #include <string>
+#include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/modules/video_coding/utility/ivf_file_writer.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video/call_stats.h"
@@ skipping 128 matching lines @@
     VieRemb* remb)
     : transport_adapter_(config.rtcp_send_transport),
       config_(std::move(config)),
       process_thread_(process_thread),
       clock_(Clock::GetRealTimeClock()),
       decode_thread_(DecodeThreadFunction, this, "DecodingThread"),
       congestion_controller_(congestion_controller),
       call_stats_(call_stats),
       video_receiver_(clock_, nullptr, this, this, this),
       stats_proxy_(&config_, clock_),
-      rtp_stream_receiver_(&video_receiver_,
-                           congestion_controller_->GetRemoteBitrateEstimator(
-                               UseSendSideBwe(config_)),
-                           &transport_adapter_,
-                           call_stats_->rtcp_rtt_stats(),
-                           congestion_controller_->pacer(),
-                           congestion_controller_->packet_router(),
-                           remb,
-                           &config_,
-                           &stats_proxy_,
-                           process_thread_),
+      rtp_stream_receiver_(
+          &video_receiver_,
+          congestion_controller_->GetRemoteBitrateEstimator(
+              UseSendSideBwe(config_)),
+          &transport_adapter_,
+          call_stats_->rtcp_rtt_stats(),
+          congestion_controller_->pacer(),
+          congestion_controller_->packet_router(),
+          remb,
+          &config_,
+          &stats_proxy_,
+          process_thread_,
+          congestion_controller_->GetRetransmissionRateLimiter()),
       vie_sync_(&video_receiver_) {
   LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
 
   RTC_DCHECK(process_thread_);
   RTC_DCHECK(congestion_controller_);
   RTC_DCHECK(call_stats_);
 
   RTC_DCHECK(!config_.decoders.empty());
   std::set<int> decoder_payload_types;
   for (const Decoder& decoder : config_.decoders) {
@@ skipping 176 matching lines @@
     const std::vector<uint16_t>& sequence_numbers) {
   rtp_stream_receiver_.RequestPacketRetransmit(sequence_numbers);
 }
 
 void VideoReceiveStream::RequestKeyFrame() {
   rtp_stream_receiver_.RequestKeyFrame();
 }
 
 }  // namespace internal
 }  // namespace webrtc