Add NACK rate throttling for audio channels.

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 44 matching lines @@
  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);
+                    ProcessThread* process_thread,
+                    RateLimiter* retransmission_rate_limiter);
   ~RtpStreamReceiver();
 
   bool SetReceiveCodec(const VideoCodec& video_codec);
 
   uint32_t GetRemoteSsrc() const;
   int GetCsrcs(uint32_t* csrcs) const;
 
   RtpReceiver* GetRtpReceiver() const;
   RtpRtcp* rtp_rtcp() const { return rtp_rtcp_.get(); }
 
@@ skipping 73 matching lines @@
   uint8_t restored_packet_[IP_PACKET_SIZE] GUARDED_BY(receive_cs_);
   bool restored_packet_in_use_ GUARDED_BY(receive_cs_);
   int64_t last_packet_log_ms_ GUARDED_BY(receive_cs_);
 
   const std::unique_ptr<RtpRtcp> rtp_rtcp_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_