audioproc_f with simulated mic analog gain

webrtc/modules/audio_processing/test/wav_based_simulator.h

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_WAV_BASED_SIMULATOR_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_WAV_BASED_SIMULATOR_H_
 
+#include <string>
 #include <vector>
 
 #include "webrtc/modules/audio_processing/test/audio_processing_simulator.h"
 
 #include "webrtc/base/constructormagic.h"
 
 namespace webrtc {
 namespace test {
 
 // Used to perform an audio processing simulation from wav files.
@@ skipping 14 matching lines @@
   void Initialize();
   bool HandleProcessStreamCall();
   bool HandleProcessReverseStreamCall();
   void PrepareProcessStreamCall();
   void PrepareReverseProcessStreamCall();
   static std::vector<SimulationEventType> GetDefaultEventChain();
   static std::vector<SimulationEventType> GetCustomEventChain(
       const std::string& filename);
 
   std::vector<SimulationEventType> call_chain_;
-  int last_specified_microphone_level_ = 100;

[aleloi, 2017/04/21 11:46:43]

Since the initial value is used in several places, I think it should be defined
as 'constexpr int kInitialMicrophoneLevel = 100' somewhere.

 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WavBasedSimulator);
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_WAV_BASED_SIMULATOR_H_