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

webrtc/modules/video_coding/test/rtp_player.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 210 matching lines @@
     RtpRtcp::Configuration configuration;
     configuration.clock = clock;
     configuration.audio = false;
     handler->rtp_module_.reset(RtpReceiver::CreateVideoReceiver(
         configuration.clock, handler->payload_sink_.get(), NULL,
         handler->rtp_payload_registry_.get()));
     if (handler->rtp_module_.get() == NULL) {
       return -1;
     }
 
-    handler->rtp_module_->SetNACKStatus(kNackOff);
     handler->rtp_header_parser_->RegisterRtpHeaderExtension(
         kRtpExtensionTransmissionTimeOffset,
         kDefaultTransmissionTimeOffsetExtensionId);
 
     for (PayloadTypesIterator it = payload_types_.begin();
          it != payload_types_.end(); ++it) {
       VideoCodec codec;
       memset(&codec, 0, sizeof(codec));
       strncpy(codec.plName, it->name().c_str(), sizeof(codec.plName) - 1);
       codec.plType = it->payload_type();
@@ skipping 242 matching lines @@
     }
   }
 
   std::unique_ptr<RtpPlayerImpl> impl(
       new RtpPlayerImpl(payload_sink_factory, payload_types, clock,
                         &packet_source, loss_rate, rtt_ms, reordering));
   return impl.release();
 }
 }  // namespace rtpplayer
 }  // namespace webrtc