Index: webrtc/modules/audio_processing/test/debug_dump_test.cc |
diff --git a/webrtc/modules/audio_processing/test/debug_dump_test.cc b/webrtc/modules/audio_processing/test/debug_dump_test.cc |
index d67a73e40c71e07e079b17a3fc8f2f44b909688b..431c4db6e47c291d91112ff0f21d71182d1c0c98 100644 |
--- a/webrtc/modules/audio_processing/test/debug_dump_test.cc |
+++ b/webrtc/modules/audio_processing/test/debug_dump_test.cc |
@@ -14,6 +14,7 @@ |
#include <string> |
#include <vector> |
+#include "webrtc/base/task_queue.h" |
#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h" |
#include "webrtc/modules/audio_processing/test/debug_dump_replayer.h" |
#include "webrtc/modules/audio_processing/test/test_utils.h" |
@@ -104,6 +105,7 @@ class DebugDumpGenerator { |
std::unique_ptr<ChannelBuffer<float>> reverse_; |
std::unique_ptr<ChannelBuffer<float>> output_; |
+ rtc::TaskQueue worker_queue_; |
std::unique_ptr<AudioProcessing> apm_; |
const std::string dump_file_name_; |
@@ -130,9 +132,9 @@ DebugDumpGenerator::DebugDumpGenerator(const std::string& input_file_name, |
reverse_config_.num_channels())), |
output_(new ChannelBuffer<float>(output_config_.num_frames(), |
output_config_.num_channels())), |
+ worker_queue_("debug_dump_generator_worker_queue"), |
apm_(AudioProcessing::Create(config)), |
- dump_file_name_(dump_file_name) { |
-} |
+ dump_file_name_(dump_file_name) {} |
DebugDumpGenerator::DebugDumpGenerator( |
const Config& config, |
@@ -187,7 +189,7 @@ void DebugDumpGenerator::SetOutputChannels(int channels) { |
} |
void DebugDumpGenerator::StartRecording() { |
- apm_->StartDebugRecording(dump_file_name_.c_str(), -1); |
+ apm_->StartDebugRecording(dump_file_name_.c_str(), -1, &worker_queue_); |
} |
void DebugDumpGenerator::Process(size_t num_blocks) { |