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

webrtc/modules/audio_coding/neteq/packet_buffer.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.
  */
 
 // This is the implementation of the PacketBuffer class. It is mostly based on
 // an STL list. The list is kept sorted at all times so that the next packet to
 // decode is at the beginning of the list.
 
 #include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
 
 #include <algorithm>  // find_if()
 
 #include "webrtc/api/audio_codecs/audio_decoder.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/tick_timer.h"
-#include "webrtc/rtc_base/logging.h"
 
 namespace webrtc {
 namespace {
 // Predicate used when inserting packets in the buffer list.
 // Operator() returns true when |packet| goes before |new_packet|.
 class NewTimestampIsLarger {
  public:
   explicit NewTimestampIsLarger(const Packet& new_packet)
       : new_packet_(new_packet) {
   }
@@ skipping 233 matching lines @@
   }
   return num_samples;
 }
 
 void PacketBuffer::BufferStat(int* num_packets, int* max_num_packets) const {
   *num_packets = static_cast<int>(buffer_.size());
   *max_num_packets = static_cast<int>(max_number_of_packets_);
 }
 
 }  // namespace webrtc