Index: webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
diff --git a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
index d2c274f460010af7976fc2a2f1032c8bb72ecc8a..59a009fb06326167c3a99503ef746d041e7c75ca 100644 |
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
@@ -78,7 +78,7 @@ void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) { |
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( |
@@ -387,9 +387,10 @@ void AudioProcessingSimulator::CreateAudioProcessor() { |
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_)); |
} |
} |