Add NACK rate throttling for audio channels.

webrtc/modules/rtp_rtcp/source/rtp_sender.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 109 matching lines @@
       sequence_number_forced_(false),
       ssrc_forced_(false),
       timestamp_(0),
       capture_time_ms_(0),
       last_timestamp_time_ms_(0),
       media_has_been_sent_(false),
       last_packet_marker_bit_(false),
       csrcs_(),
       rtx_(kRtxOff),
       retransmission_rate_limiter_(retransmission_rate_limiter) {
+  RTC_DCHECK(retransmission_rate_limiter_ != nullptr);
+
   // We need to seed the random generator for BuildPaddingPacket() below.
   // TODO(holmer,tommi): Note that TimeInMilliseconds might return 0 on Mac
   // early on in the process.
   srand(static_cast<uint32_t>(clock_->TimeInMilliseconds()));
   ssrc_ = ssrc_db_->CreateSSRC();
   RTC_DCHECK(ssrc_ != 0);
   ssrc_rtx_ = ssrc_db_->CreateSSRC();
   RTC_DCHECK(ssrc_rtx_ != 0);
 
   // Random start, 16 bits. Can't be 0.
@@ skipping 1747 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc