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..7c0213ba4882ca7c612dc4a2680da4db5249dfd7 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.h
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.h
@@ -17,11 +17,12 @@
 #include <memory>
 #include <string>
 
-#include "webrtc/base/timeutils.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/fake_recording_device.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 
 namespace webrtc {
@@ -74,6 +75,9 @@ struct SimulationSettings {
   rtc::Optional<int> vad_likelihood;
   rtc::Optional<int> ns_level;
   rtc::Optional<bool> use_refined_adaptive_filter;
+  int initial_mic_gain;
+  bool simulate_mic_gain = false;
+  rtc::Optional<int> simulated_mic_kind;
   bool report_performance = false;
   bool report_bitexactness = false;
   bool use_verbose_logging = false;
@@ -164,6 +168,8 @@ class AudioProcessingSimulator {
   AudioFrame rev_frame_;
   AudioFrame fwd_frame_;
   bool bitexact_output_ = true;
+  int new_analog_level_;
+  rtc::Optional<int> real_recording_device_level_;
 
  private:
   void SetupOutput();
@@ -175,6 +181,7 @@ class AudioProcessingSimulator {
   std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_;
   TickIntervalStats proc_time_;
   std::ofstream residual_echo_likelihood_graph_writer_;
+  FakeRecordingDevice fake_recording_device_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
 };