Add NACK rate throttling for audio channels.

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 21 matching lines @@
 
 namespace webrtc {
 
 std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
     ReceiveStatistics* receive_statistics,
     Transport* outgoing_transport,
     RtcpRttStats* rtt_stats,
     RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
     RemoteBitrateEstimator* remote_bitrate_estimator,
     RtpPacketSender* paced_sender,
-    TransportSequenceNumberAllocator* transport_sequence_number_allocator) {
+    TransportSequenceNumberAllocator* transport_sequence_number_allocator,
+    RateLimiter* retransmission_rate_limiter) {
   RtpRtcp::Configuration configuration;
   configuration.audio = false;
   configuration.receiver_only = true;
   configuration.receive_statistics = receive_statistics;
   configuration.outgoing_transport = outgoing_transport;
   configuration.intra_frame_callback = nullptr;
   configuration.rtt_stats = rtt_stats;
   configuration.rtcp_packet_type_counter_observer =
       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;
+  configuration.retransmission_rate_limiter = retransmission_rate_limiter;
 
   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;
 
 RtpStreamReceiver::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)
+    ProcessThread* process_thread,
+    RateLimiter* retransmission_rate_limiter)
     : clock_(Clock::GetRealTimeClock()),
       config_(*config),
       video_receiver_(video_receiver),
       remote_bitrate_estimator_(remote_bitrate_estimator),
       packet_router_(packet_router),
       remb_(remb),
       process_thread_(process_thread),
       ntp_estimator_(clock_),
       rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
       rtp_header_parser_(RtpHeaderParser::Create()),
       rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_,
                                                      this,
                                                      this,
                                                      &rtp_payload_registry_)),
       rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
       fec_receiver_(FecReceiver::Create(this)),
       receiving_(false),
       restored_packet_in_use_(false),
       last_packet_log_ms_(-1),
       rtp_rtcp_(CreateRtpRtcpModule(rtp_receive_statistics_.get(),
                                     transport,
                                     rtt_stats,
                                     receive_stats_proxy,
                                     remote_bitrate_estimator_,
                                     paced_sender,
-                                    packet_router)) {
+                                    packet_router,
+                                    retransmission_rate_limiter)) {
   packet_router_->AddRtpModule(rtp_rtcp_.get());
   rtp_receive_statistics_->RegisterRtpStatisticsCallback(receive_stats_proxy);
   rtp_receive_statistics_->RegisterRtcpStatisticsCallback(receive_stats_proxy);
 
   RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
       << "A stream should not be configured with RTCP disabled. This value is "
          "reserved for internal usage.";
   RTC_DCHECK(config_.rtp.remote_ssrc != 0);
   // TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
   RTC_DCHECK(config_.rtp.local_ssrc != 0);
@@ skipping 415 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