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

webrtc/modules/audio_processing/test/audio_processing_simulator.cc

 /*
  *  Copyright (c) 2016 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_processing/test/audio_processing_simulator.h"
 
 #include <algorithm>
 #include <iostream>
 #include <sstream>
 #include <string>
 #include <vector>
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/stringutils.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/stringutils.h"
 
 namespace webrtc {
 namespace test {
 namespace {
 
 void CopyFromAudioFrame(const AudioFrame& src, ChannelBuffer<float>* dest) {
   RTC_CHECK_EQ(src.num_channels_, dest->num_channels());
   RTC_CHECK_EQ(src.samples_per_channel_, dest->num_frames());
   // Copy the data from the input buffer.
   std::vector<float> tmp(src.samples_per_channel_ * src.num_channels_);
@@ skipping 356 matching lines @@
   }
 
   if (settings_.aec_dump_output_filename) {
     ap_->AttachAecDump(AecDumpFactory::Create(
         *settings_.aec_dump_output_filename, -1, &worker_queue_));
   }
 }
 
 }  // namespace test
 }  // namespace webrtc