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diff --git a/third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.h b/third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.h
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+++ b/third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.h
@@ -0,0 +1,66 @@
+/*
+ *  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 MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_
+#define MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_
+
+#include <memory>
+#include <vector>
+
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "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,
+                   size_t num_capture_input_channels,
+                   size_t num_render_output_channels,
+                   size_t num_capture_output_channels);
+  ~SimulatorBuffers();
+
+  void CreateConfigAndBuffer(int sample_rate_hz,
+                             size_t num_channels,
+                             Random* rand_gen,
+                             std::unique_ptr<AudioBuffer>* buffer,
+                             StreamConfig* config,
+                             std::vector<float*>* buffer_data,
+                             std::vector<float>* buffer_data_samples);
+
+  void UpdateInputBuffers();
+
+  std::unique_ptr<AudioBuffer> render_input_buffer;
+  std::unique_ptr<AudioBuffer> capture_input_buffer;
+  std::unique_ptr<AudioBuffer> render_output_buffer;
+  std::unique_ptr<AudioBuffer> capture_output_buffer;
+  StreamConfig render_input_config;
+  StreamConfig capture_input_config;
+  StreamConfig render_output_config;
+  StreamConfig capture_output_config;
+  std::vector<float*> render_input;
+  std::vector<float> render_input_samples;
+  std::vector<float*> capture_input;
+  std::vector<float> capture_input_samples;
+  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  // MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_