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

webrtc/modules/audio_device/fine_audio_buffer.cc

 /*
  *  Copyright (c) 2013 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/audio_device/fine_audio_buffer.h"
 
 #include <memory.h>
 #include <stdio.h>
 #include <algorithm>
 
-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_device/audio_device_buffer.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
 
 namespace webrtc {
 
 FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* device_buffer,
                                  int sample_rate,
                                  size_t capacity)
     : device_buffer_(device_buffer),
       sample_rate_(sample_rate),
       samples_per_10_ms_(static_cast<size_t>(sample_rate_ * 10 / 1000)),
       bytes_per_10_ms_(samples_per_10_ms_ * sizeof(int16_t)),
@@ skipping 53 matching lines @@
                                       samples_per_10_ms_);
     device_buffer_->SetVQEData(playout_delay_ms, record_delay_ms, 0);
     device_buffer_->DeliverRecordedData();
     memmove(record_buffer_.data(), record_buffer_.data() + bytes_per_10_ms_,
             record_buffer_.size() - bytes_per_10_ms_);
     record_buffer_.SetSize(record_buffer_.size() - bytes_per_10_ms_);
   }
 }
 
 }  // namespace webrtc