Change existing aec dump tests to use webrtc::AecDump.

webrtc/modules/audio_processing/test/audio_processing_simulator.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_AUDIO_PROCESSING_SIMULATOR_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
 
 #include <algorithm>
 #include <fstream>
 #include <limits>
 #include <memory>
 #include <string>
 
-#include "webrtc/base/timeutils.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
+#include "webrtc/base/task_queue.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 
 namespace webrtc {
 namespace test {
 
 // Holds all the parameters available for controlling the simulation.
 struct SimulationSettings {
   SimulationSettings();
@@ skipping 137 matching lines @@
   void SetupOutput();
 
   size_t num_process_stream_calls_ = 0;
   size_t num_reverse_process_stream_calls_ = 0;
   size_t output_reset_counter_ = 0;
   std::unique_ptr<ChannelBufferWavWriter> buffer_writer_;
   std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_;
   TickIntervalStats proc_time_;
   std::ofstream residual_echo_likelihood_graph_writer_;
 
+  rtc::TaskQueue worker_queue_;
+
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_