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..a4a3fb2496127af8ed48a532a5a3c4eef775953e 100644 |
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
@@ -19,6 +19,7 @@ |
#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 { |
@@ -78,7 +79,7 @@ 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( |
@@ -248,7 +249,7 @@ void AudioProcessingSimulator::SetupOutput() { |
void AudioProcessingSimulator::DestroyAudioProcessor() { |
if (settings_.aec_dump_output_filename) { |
- RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->StopDebugRecording()); |
+ ap_->StopDebugRecording(); |
} |
} |
@@ -385,11 +386,8 @@ 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)); |
+ ap_->StartDebugRecording(AecDumpFactory::Create( |
+ *settings_.aec_dump_output_filename, -1, &worker_queue_)); |
} |
} |