Update includes for webrtc/{base => rtc_base} rename (1/3)

webrtc/modules/rtp_rtcp/source/packet_loss_stats.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
 
 #include <vector>
 
-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 
 // After this many packets are added, adding additional packets will cause the
 // oldest packets to be pruned from the buffer.
 static const int kBufferSize = 100;
 
 namespace webrtc {
 
 PacketLossStats::PacketLossStats()
     : single_loss_historic_count_(0),
       multiple_loss_historic_event_count_(0),
@@ skipping 102 matching lines @@
   }
   // Continue pruning if the wrapped buffer is beyond a threshold and there are
   // things left in the pre-wrapped buffer.
   if (!lost_packets_wrapped_buffer_.empty() &&
       *(lost_packets_wrapped_buffer_.rbegin()) > 0x4000) {
     PruneBuffer();
   }
 }
 
 }  // namespace webrtc