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

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.
  */
 
 #include "webrtc/modules/audio_processing/test/audio_processing_simulator.h"
 
 #include <algorithm>
 #include <iostream>
 #include <sstream>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/common_audio/include/audio_util.h"
+#include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 
 namespace webrtc {
 namespace test {
 namespace {
 
 void CopyFromAudioFrame(const AudioFrame& src, ChannelBuffer<float>* dest) {
   RTC_CHECK_EQ(src.num_channels_, dest->num_channels());
   RTC_CHECK_EQ(src.samples_per_channel_, dest->num_frames());
   // Copy the data from the input buffer.
@@ skipping 40 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 149 matching lines @@
                       static_cast<size_t>(reverse_out_config_.num_channels())));
     reverse_buffer_writer_.reset(
         new ChannelBufferWavWriter(std::move(reverse_out_file)));
   }
 
   ++output_reset_counter_;
 }
 
 void AudioProcessingSimulator::DestroyAudioProcessor() {
   if (settings_.aec_dump_output_filename) {
-    RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->StopDebugRecording());
+    ap_->DetachAecDump();
   }
 }
 
 void AudioProcessingSimulator::CreateAudioProcessor() {
   Config config;
   AudioProcessing::Config apm_config;
   if (settings_.use_bf && *settings_.use_bf) {
     config.Set<Beamforming>(new Beamforming(
         true, ParseArrayGeometry(*settings_.microphone_positions),
         SphericalPointf(DegreesToRadians(settings_.target_angle_degrees), 0.f,
@@ skipping 119 matching lines @@
         AudioProcessing::kNoError,
         ap_->noise_suppression()->set_level(
             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));
+    ap_->AttachAecDump(AecDumpFactory::Create(
+        *settings_.aec_dump_output_filename, -1, &worker_queue_));
   }
 }
 
 }  // namespace test
 }  // namespace webrtc