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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
 
 #include <algorithm>
 #include <fstream>
 #include <limits>
 #include <memory>
 #include <string>
 
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/fake_recording_device.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/task_queue.h"
 #include "webrtc/rtc_base/timeutils.h"
 
 namespace webrtc {
 namespace test {
 
 // Holds all the parameters available for controlling the simulation.
@@ skipping 37 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;
+  int initial_mic_level;
+  bool simulate_mic_gain = false;
+  rtc::Optional<int> simulated_mic_kind;
   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 70 matching lines @@
   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;
+  rtc::Optional<int> aec_dump_mic_level_;
 
  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_;
 

[peah-webrtc, 2017/09/15 09:36:20]

Please remove the empty line on 185

[AleBzk, 2017/09/22 12:33:55]

Done.

+  int analog_mic_level_;
+  FakeRecordingDevice fake_recording_device_;
+
   rtc::TaskQueue worker_queue_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_