audioproc_f with simulated mic analog gain

webrtc/modules/audio_processing/test/aec_dump_based_simulator.h

Index: webrtc/modules/audio_processing/test/aec_dump_based_simulator.h
diff --git a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h
index 5f7d138aa41eff5b2bf5a3f9374ba99836939fb5..188850bf7278934af4a5672fb5d25fac62cde805 100644
--- a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h
+++ b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_
 
+#include <memory>
+
 #include "webrtc/modules/audio_processing/test/audio_processing_simulator.h"
 
 #include "webrtc/base/constructormagic.h"
@@ -41,8 +43,7 @@ class AecDumpBasedSimulator final : public AudioProcessingSimulator {
   void HandleMessage(const webrtc::audioproc::Stream& msg);
   void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
   void HandleMessage(const webrtc::audioproc::Config& msg);
-  void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg,
-                                bool* set_stream_analog_level_called);
+  void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg);
   void PrepareReverseProcessStreamCall(
       const webrtc::audioproc::ReverseStream& msg);
   void VerifyProcessStreamBitExactness(const webrtc::audioproc::Stream& msg);
@@ -59,6 +60,10 @@ class AecDumpBasedSimulator final : public AudioProcessingSimulator {
   bool artificial_nearend_eof_reported_ = false;
   InterfaceType interface_used_ = InterfaceType::kNotSpecified;
 
+  // TODO(aleloi): Remove once a FakeRecordingDevice is addded as protected
+  // member of AudioProcessingSimulator.
+  int last_specified_microphone_level_ = kInitialMicrophoneGainLevel;
+
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AecDumpBasedSimulator);
 };