Test submission of complete AEC-dump refactoring.

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.
  */
 
@@ skipping 60 matching lines @@
   for (size_t ch = 0; ch < dest->num_channels_; ++ch) {
     for (size_t sample = 0; sample < dest->samples_per_channel_; ++sample) {
       dest->data_[sample * dest->num_channels_ + ch] =
           src.channels()[ch][sample] * 32767;
     }
   }
 }
 
 AudioProcessingSimulator::AudioProcessingSimulator(
     const SimulationSettings& settings)
-    : settings_(settings) {
+    : settings_(settings), worker_queue_("file_writer_task_queue") {
   if (settings_.ed_graph_output_filename &&
       settings_.ed_graph_output_filename->size() > 0) {
     residual_echo_likelihood_graph_writer_.open(
         *settings_.ed_graph_output_filename);
     RTC_CHECK(residual_echo_likelihood_graph_writer_.is_open());
     WriteEchoLikelihoodGraphFileHeader(&residual_echo_likelihood_graph_writer_);
   }
 }
 
 AudioProcessingSimulator::~AudioProcessingSimulator() {
@@ skipping 288 matching lines @@
             static_cast<NoiseSuppression::Level>(*settings_.ns_level)));
   }
 
   if (settings_.use_ts) {
     ap_->set_stream_key_pressed(*settings_.use_ts);
   }
 
   if (settings_.aec_dump_output_filename) {
     size_t kMaxFilenameSize = AudioProcessing::kMaxFilenameSize;
     RTC_CHECK_LE(settings_.aec_dump_output_filename->size(), kMaxFilenameSize);
-    RTC_CHECK_EQ(AudioProcessing::kNoError,
-                 ap_->StartDebugRecording(
-                     settings_.aec_dump_output_filename->c_str(), -1));
+    RTC_CHECK_EQ(
+        AudioProcessing::kNoError,
+        ap_->StartDebugRecording(settings_.aec_dump_output_filename->c_str(),
+                                 -1, &worker_queue_));
   }
 }
 
 }  // namespace test
 }  // namespace webrtc