audioproc_f with fake microphone.

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 db3c92f942a7bdda6d2903f40d164c69f33bd7da..dacc88383add35fa56b912222b1e724b187a43ad 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 {
@@ -79,6 +80,7 @@ struct SimulationSettings {
   rtc::Optional<int> ns_level;
   rtc::Optional<bool> use_refined_adaptive_filter;
   bool simulate_mic_gain = false;
+  rtc::Optional<int> simulated_mic_kind;
   bool report_performance = false;
   bool report_bitexactness = false;
   bool use_verbose_logging = false;
@@ -169,6 +171,7 @@ class AudioProcessingSimulator {
   AudioFrame rev_frame_;
   AudioFrame fwd_frame_;
   bool bitexact_output_ = true;
+  rtc::Optional<FakeRecordingDevice> fake_recording_device_;
 
  private:
   void SetupOutput();