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

webrtc/modules/audio_processing/test/simulator_buffers.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/simulator_buffers.h"
 
-#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
+#include "webrtc/rtc_base/checks.h"
 
 namespace webrtc {
 namespace test {
 
 SimulatorBuffers::SimulatorBuffers(int render_input_sample_rate_hz,
                                    int capture_input_sample_rate_hz,
                                    int render_output_sample_rate_hz,
                                    int capture_output_sample_rate_hz,
                                    size_t num_render_input_channels,
                                    size_t num_capture_input_channels,
@@ skipping 51 matching lines @@
 
 void SimulatorBuffers::UpdateInputBuffers() {
   test::CopyVectorToAudioBuffer(capture_input_config, capture_input_samples,
                                 capture_input_buffer.get());
   test::CopyVectorToAudioBuffer(render_input_config, render_input_samples,
                                 render_input_buffer.get());
 }
 
 }  // namespace test
 }  // namespace webrtc