audioproc_f with simulated mic analog gain

webrtc/modules/audio_processing/test/audio_processing_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 56 matching lines @@
   rtc::Optional<bool> use_experimental_agc;
   rtc::Optional<int> aecm_routing_mode;
   rtc::Optional<bool> use_aecm_comfort_noise;
   rtc::Optional<int> agc_mode;
   rtc::Optional<int> agc_target_level;
   rtc::Optional<bool> use_agc_limiter;
   rtc::Optional<int> agc_compression_gain;
   rtc::Optional<int> vad_likelihood;
   rtc::Optional<int> ns_level;
   rtc::Optional<bool> use_refined_adaptive_filter;
+  bool simulate_mic_gain = false;
   bool report_performance = false;
   bool report_bitexactness = false;
   bool use_verbose_logging = false;
   bool discard_all_settings_in_aecdump = true;
   rtc::Optional<std::string> aec_dump_input_filename;
   rtc::Optional<std::string> aec_dump_output_filename;
   bool fixed_interface = false;
   bool store_intermediate_output = false;
   rtc::Optional<std::string> custom_call_order_filename;
 };
@@ skipping 41 matching lines @@
         : proc_time_(proc_time), start_time_(rtc::TimeNanos()) {}
 
     ~ScopedTimer();
 
    private:
     TickIntervalStats* const proc_time_;
     int64_t start_time_;
   };
 
   TickIntervalStats* mutable_proc_time() { return &proc_time_; }
-  void ProcessStream(bool fixed_interface);
+  void ProcessStream(bool fixed_interface,
+                     bool skip_analog_level_update = false);

[aleloi, 2017/04/21 11:46:42]

Suggest rename into do_analog_level_update instead.

[AleBzk, 2017/04/24 09:40:26]

Done.

   void ProcessReverseStream(bool fixed_interface);
   void CreateAudioProcessor();
   void DestroyAudioProcessor();
   void SetupBuffersConfigsOutputs(int input_sample_rate_hz,
                                   int output_sample_rate_hz,
                                   int reverse_input_sample_rate_hz,
                                   int reverse_output_sample_rate_hz,
                                   int input_num_channels,
                                   int output_num_channels,
                                   int reverse_input_num_channels,
                                   int reverse_output_num_channels);
 
   const SimulationSettings settings_;
   std::unique_ptr<AudioProcessing> ap_;
 
   std::unique_ptr<ChannelBuffer<float>> in_buf_;
   std::unique_ptr<ChannelBuffer<float>> out_buf_;
   std::unique_ptr<ChannelBuffer<float>> reverse_in_buf_;
   std::unique_ptr<ChannelBuffer<float>> reverse_out_buf_;
   StreamConfig in_config_;
   StreamConfig out_config_;
   StreamConfig reverse_in_config_;
   StreamConfig reverse_out_config_;
   std::unique_ptr<ChannelBufferWavReader> buffer_reader_;
   std::unique_ptr<ChannelBufferWavReader> reverse_buffer_reader_;
   AudioFrame rev_frame_;
   AudioFrame fwd_frame_;
   bool bitexact_output_ = true;
+  // TODO(alessiob): Check what initial value makes sense, 100 comes from
+  // WavBasedSimulator::last_specified_microphone_level_.
+  int last_specified_microphone_level_ = 100;
 
  private:
   void SetupOutput();
 
   size_t num_process_stream_calls_ = 0;
   size_t num_reverse_process_stream_calls_ = 0;
   size_t output_reset_counter_ = 0;
   std::unique_ptr<ChannelBufferWavWriter> buffer_writer_;
   std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_;
   TickIntervalStats proc_time_;
   std::ofstream residual_echo_likelihood_graph_writer_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_