Added calling of the stream_analog_level api in audioproc_f

webrtc/modules/audio_processing/test/aec_dump_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.
  */
 
@@ skipping 24 matching lines @@
   ~AecDumpBasedSimulator() override {}
 
   // Processes the messages in the aecdump file.
   void Process() override;
 
  private:
   void HandleMessage(const webrtc::audioproc::Init& msg);
   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);
+  void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg,
+                                bool* set_stream_analog_level_called);
   void PrepareReverseProcessStreamCall(
       const webrtc::audioproc::ReverseStream& msg);
   void VerifyProcessStreamBitExactness(const webrtc::audioproc::Stream& msg);
 
   enum InterfaceType {
     kFixedInterface,
     kFloatInterface,
     kNotSpecified,
   };
 
   FILE* dump_input_file_;
   InterfaceType interface_used_ = InterfaceType::kNotSpecified;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AecDumpBasedSimulator);
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_