Removing some old code which looked like it had to do with NACK handling but in reality did nothing.

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 114 matching lines @@
     rtp_rtcp_->SetREMBStatus(true);
     remb_->AddReceiveChannel(rtp_rtcp_.get());
   }
 
   for (size_t i = 0; i < config.rtp.extensions.size(); ++i) {
     EnableReceiveRtpHeaderExtension(config.rtp.extensions[i].name,
                                     config.rtp.extensions[i].id);
   }
 
   static const int kMaxPacketAgeToNack = 450;
-  NACKMethod nack_method =
-      config.rtp.nack.rtp_history_ms > 0 ? kNackRtcp : kNackOff;
-  const int max_reordering_threshold = (nack_method == kNackRtcp)
+  const int max_reordering_threshold = (config.rtp.nack.rtp_history_ms > 0)
       ? kMaxPacketAgeToNack : kDefaultMaxReorderingThreshold;
-  rtp_receiver_->SetNACKStatus(nack_method);
   rtp_receive_statistics_->SetMaxReorderingThreshold(max_reordering_threshold);
 
   // TODO(pbos): Support multiple RTX, per video payload.
   for (const auto& kv : config_.rtp.rtx) {
     RTC_DCHECK(kv.second.ssrc != 0);
     RTC_DCHECK(kv.second.payload_type != 0);
 
     rtp_payload_registry_.SetRtxSsrc(kv.second.ssrc);
     rtp_payload_registry_.SetRtxPayloadType(kv.second.payload_type,
                                             kv.first);
@@ skipping 386 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