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; |