audioproc_f with simulated mic analog gain

webrtc/modules/audio_processing/test/audio_processing_simulator.h

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..671210a8cf2dd2bf249e3d8de22e7062ef490c99 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;
@@ -135,7 +140,8 @@ class AudioProcessingSimulator {
   };
 
   TickIntervalStats* mutable_proc_time() { return &proc_time_; }
-  void ProcessStream(bool fixed_interface);
+  void ProcessStream(bool fixed_interface,
+                     bool update_analog_level = true);
   void ProcessReverseStream(bool fixed_interface);
   void CreateAudioProcessor();
   void DestroyAudioProcessor();
@@ -164,6 +170,7 @@ class AudioProcessingSimulator {
   AudioFrame rev_frame_;
   AudioFrame fwd_frame_;
   bool bitexact_output_ = true;
+  int last_specified_microphone_level_ = kInitialMicrophoneGainLevel;
 
  private:
   void SetupOutput();