| 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..680041e8cac5d9591f585d18574ef116a4215aad 100644
|
| --- a/webrtc/modules/audio_processing/test/debug_dump_test.cc
|
| +++ b/webrtc/modules/audio_processing/test/debug_dump_test.cc
|
| @@ -14,7 +14,9 @@
|
| #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/aec_dump/aec_dump_factory.h"
|
| #include "webrtc/modules/audio_processing/test/debug_dump_replayer.h"
|
| #include "webrtc/modules/audio_processing/test/test_utils.h"
|
| #include "webrtc/test/gtest.h"
|
| @@ -104,6 +106,7 @@ 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 +133,9 @@ 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 +190,8 @@ void DebugDumpGenerator::SetOutputChannels(int channels) {
|
| }
|
|
|
| void DebugDumpGenerator::StartRecording() {
|
| - apm_->StartDebugRecording(dump_file_name_.c_str(), -1);
|
| + apm_->StartDebugRecording(
|
| + AecDumpFactory::Create(dump_file_name_.c_str(), -1, &worker_queue_));
|
| }
|
|
|
| void DebugDumpGenerator::Process(size_t num_blocks) {
|
|
|