text: /*
   *  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 MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
  #define MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
  
  #include <algorithm>
  #include <fstream>
  #include <limits>
  #include <memory>
  #include <string>
  
  #include "absl/types/optional.h"
  #include "common_audio/channel_buffer.h"
  #include "modules/audio_processing/include/audio_processing.h"
  #include "modules/audio_processing/test/api_call_statistics.h"
  #include "modules/audio_processing/test/fake_recording_device.h"
  #include "modules/audio_processing/test/test_utils.h"
  #include "rtc_base/constructor_magic.h"
  #include "rtc_base/task_queue_for_test.h"
  #include "rtc_base/time_utils.h"
  
  namespace webrtc {
  namespace test {
  
  // Holds all the parameters available for controlling the simulation.
  struct SimulationSettings {
    SimulationSettings();
    SimulationSettings(const SimulationSettings&);
    ~SimulationSettings();
    absl::optional<int> stream_delay;
    absl::optional<bool> use_stream_delay;
    absl::optional<int> stream_drift_samples;
    absl::optional<int> output_sample_rate_hz;
    absl::optional<int> output_num_channels;
    absl::optional<int> reverse_output_sample_rate_hz;
    absl::optional<int> reverse_output_num_channels;
    absl::optional<std::string> output_filename;
    absl::optional<std::string> reverse_output_filename;
    absl::optional<std::string> input_filename;
    absl::optional<std::string> reverse_input_filename;
    absl::optional<std::string> artificial_nearend_filename;
    absl::optional<bool> use_aec;
    absl::optional<bool> use_aecm;
    absl::optional<bool> use_ed;  // Residual Echo Detector.
    absl::optional<std::string> ed_graph_output_filename;
    absl::optional<bool> use_agc;
    absl::optional<bool> use_agc2;
    absl::optional<bool> use_pre_amplifier;
    absl::optional<bool> use_hpf;
    absl::optional<bool> use_ns;
    absl::optional<bool> use_ts;
    absl::optional<bool> use_ie;
    absl::optional<bool> use_vad;
    absl::optional<bool> use_le;
    absl::optional<bool> use_all;
    absl::optional<int> aec_suppression_level;
    absl::optional<bool> use_delay_agnostic;
    absl::optional<bool> use_extended_filter;
    absl::optional<bool> use_drift_compensation;
    absl::optional<bool> use_legacy_aec;
    absl::optional<bool> use_experimental_agc;
    absl::optional<bool> use_experimental_agc_agc2_level_estimator;
    absl::optional<bool> experimental_agc_disable_digital_adaptive;
    absl::optional<bool> experimental_agc_analyze_before_aec;
    absl::optional<int> agc_mode;
    absl::optional<int> agc_target_level;
    absl::optional<bool> use_agc_limiter;
    absl::optional<int> agc_compression_gain;
    absl::optional<bool> agc2_use_adaptive_gain;
    absl::optional<float> agc2_fixed_gain_db;
    AudioProcessing::Config::GainController2::LevelEstimator
        agc2_adaptive_level_estimator;
    absl::optional<float> pre_amplifier_gain_factor;
    absl::optional<int> vad_likelihood;
    absl::optional<int> ns_level;
    absl::optional<bool> use_refined_adaptive_filter;
    int initial_mic_level;
    bool simulate_mic_gain = false;
    absl::optional<int> simulated_mic_kind;
    bool report_performance = false;
    absl::optional<std::string> performance_report_output_filename;
    bool report_bitexactness = false;
    bool use_verbose_logging = false;
    bool use_quiet_output = false;
    bool discard_all_settings_in_aecdump = true;
    absl::optional<std::string> aec_dump_input_filename;
    absl::optional<std::string> aec_dump_output_filename;
    bool fixed_interface = false;
    bool store_intermediate_output = false;
    bool print_aec_parameter_values = false;
    bool dump_internal_data = false;
    absl::optional<std::string> dump_internal_data_output_dir;
    absl::optional<std::string> call_order_input_filename;
    absl::optional<std::string> call_order_output_filename;
    absl::optional<std::string> aec_settings_filename;
  };
  
  // Copies samples present in a ChannelBuffer into an AudioFrame.
  void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest);
  
  // Provides common functionality for performing audioprocessing simulations.
  class AudioProcessingSimulator {
   public:
    static const int kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs;
  
    AudioProcessingSimulator(const SimulationSettings& settings,
                             std::unique_ptr<AudioProcessingBuilder> ap_builder);
    virtual ~AudioProcessingSimulator();
  
    // Processes the data in the input.
    virtual void Process() = 0;
  
    // Returns the execution times of all AudioProcessing calls.
    const ApiCallStatistics& GetApiCallStatistics() const {
      return api_call_statistics_;
    }
  
    // Reports whether the processed recording was bitexact.
    bool OutputWasBitexact() { return bitexact_output_; }
  
    size_t get_num_process_stream_calls() { return num_process_stream_calls_; }
    size_t get_num_reverse_process_stream_calls() {
      return num_reverse_process_stream_calls_;
    }
  
   protected:
    void ProcessStream(bool fixed_interface);
    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<AudioProcessingBuilder> ap_builder_;
  
    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;
    int aec_dump_mic_level_ = 0;
  
   protected:
    size_t output_reset_counter_ = 0;
  
   private:
    void SetupOutput();
  
    size_t num_process_stream_calls_ = 0;
    size_t num_reverse_process_stream_calls_ = 0;
    std::unique_ptr<ChannelBufferWavWriter> buffer_writer_;
    std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_;
    ApiCallStatistics api_call_statistics_;
    std::ofstream residual_echo_likelihood_graph_writer_;
    int analog_mic_level_;
    FakeRecordingDevice fake_recording_device_;
  
    TaskQueueForTest worker_queue_;
  
    RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
  };
  
  }  // namespace test
  }  // namespace webrtc
  
  #endif  // MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
  