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..8d5d7c809dd6640afbfbde104cda3b8ed7e35325 100644 |
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.h |
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.h |
@@ -74,6 +74,7 @@ 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; |
@@ -135,7 +136,8 @@ class AudioProcessingSimulator { |
}; |
TickIntervalStats* mutable_proc_time() { return &proc_time_; } |
- void ProcessStream(bool fixed_interface); |
+ void ProcessStream(bool fixed_interface, |
+ bool skip_analog_level_update = false); |
aleloi
2017/04/21 11:46:42
Suggest rename into do_analog_level_update instead
AleBzk
2017/04/24 09:40:26
Done.
|
void ProcessReverseStream(bool fixed_interface); |
void CreateAudioProcessor(); |
void DestroyAudioProcessor(); |
@@ -164,6 +166,9 @@ class AudioProcessingSimulator { |
AudioFrame rev_frame_; |
AudioFrame fwd_frame_; |
bool bitexact_output_ = true; |
+ // TODO(alessiob): Check what initial value makes sense, 100 comes from |
+ // WavBasedSimulator::last_specified_microphone_level_. |
+ int last_specified_microphone_level_ = 100; |
private: |
void SetupOutput(); |