| Index: webrtc/modules/audio_processing/test/audio_processing_simulator.h
|
| diff --git a/webrtc/modules/audio_processing/test/audio_processing_simulator.h b/webrtc/modules/audio_processing/test/audio_processing_simulator.h
|
| index d1def43b352248432f0bd89cf6e8c5993069c8a8..832ab10338ce296c51fcad2753f1691683eb271e 100644
|
| --- a/webrtc/modules/audio_processing/test/audio_processing_simulator.h
|
| +++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.h
|
| @@ -12,6 +12,7 @@
|
| #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
|
|
|
| #include <algorithm>
|
| +#include <fstream>
|
| #include <limits>
|
| #include <memory>
|
| #include <string>
|
| @@ -45,6 +46,8 @@ struct SimulationSettings {
|
| rtc::Optional<std::string> reverse_input_filename;
|
| rtc::Optional<bool> use_aec;
|
| rtc::Optional<bool> use_aecm;
|
| + rtc::Optional<bool> use_red; // Residual Echo Detector.
|
| + rtc::Optional<std::string> red_graph_output_filename;
|
| rtc::Optional<bool> use_agc;
|
| rtc::Optional<bool> use_hpf;
|
| rtc::Optional<bool> use_ns;
|
| @@ -168,6 +171,7 @@ class AudioProcessingSimulator {
|
| std::unique_ptr<ChannelBufferWavWriter> buffer_writer_;
|
| std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_;
|
| TickIntervalStats proc_time_;
|
| + std::ofstream residual_echo_likelihood_graph_writer_;
|
|
|
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
|
| };
|
|
|