Added calling of the stream_analog_level api in audioproc_f

webrtc/modules/audio_processing/test/wav_based_simulator.cc

 /*
  *  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 82 matching lines @@
   }
 
   DestroyAudioProcessor();
 }
 
 bool WavBasedSimulator::HandleProcessStreamCall() {
   bool samples_left_to_process = buffer_reader_->Read(in_buf_.get());
   if (samples_left_to_process) {
     PrepareProcessStreamCall();
     ProcessStream(settings_.fixed_interface);
+    // Call stream analog level to ensure that any side-effects are triggered.
+    (void)ap_->gain_control()->stream_analog_level();
     last_specified_microphone_level_ =
         ap_->gain_control()->stream_analog_level();
   }
   return samples_left_to_process;
 }
 
 bool WavBasedSimulator::HandleProcessReverseStreamCall() {
   bool samples_left_to_process =
       reverse_buffer_reader_->Read(reverse_in_buf_.get());
   if (samples_left_to_process) {
@@ skipping 39 matching lines @@
   }
 
   SetupBuffersConfigsOutputs(
       input_sample_rate_hz, output_sample_rate_hz, reverse_sample_rate_hz,
       reverse_output_sample_rate_hz, input_num_channels, output_num_channels,
       reverse_num_channels, reverse_output_num_channels);
 }
 
 }  // namespace test
 }  // namespace webrtc