| 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 f597fa101a76e7a1a705464957458308fb1b0f81..db3c92f942a7bdda6d2903f40d164c69f33bd7da 100644
|
| --- a/webrtc/modules/audio_processing/test/audio_processing_simulator.h
|
| +++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.h
|
| @@ -27,6 +27,10 @@
|
| namespace webrtc {
|
| namespace test {
|
|
|
| +// TODO(alessiob): Check what initial value makes sense, 100 was used in
|
| +// WavBasedSimulator::last_specified_microphone_level_.
|
| +constexpr int kInitialMicrophoneGainLevel = 100;
|
| +
|
| // Holds all the parameters available for controlling the simulation.
|
| struct SimulationSettings {
|
| SimulationSettings();
|
| @@ -74,6 +78,7 @@ struct SimulationSettings {
|
| rtc::Optional<int> vad_likelihood;
|
| rtc::Optional<int> ns_level;
|
| rtc::Optional<bool> use_refined_adaptive_filter;
|
| + bool simulate_mic_gain = false;
|
| bool report_performance = false;
|
| bool report_bitexactness = false;
|
| bool use_verbose_logging = false;
|
|
|