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

webrtc/modules/audio_processing/test/simulator_buffers.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_
 
 #include <memory>
 #include <vector>
 
-#include "webrtc/base/random.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/rtc_base/random.h"
 
 namespace webrtc {
 namespace test {
 
 struct 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,
@@ skipping 27 matching lines @@
   std::vector<float*> render_output;
   std::vector<float> render_output_samples;
   std::vector<float*> capture_output;
   std::vector<float> capture_output_samples;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_