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 WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
  #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
  
  #include <algorithm>
  #include <limits>
  #include <memory>
  #include <string>
  
  #include "webrtc/base/timeutils.h"
  #include "webrtc/base/constructormagic.h"
  #include "webrtc/base/optional.h"
  #include "webrtc/common_audio/channel_buffer.h"
  #include "webrtc/modules/audio_processing/include/audio_processing.h"
  #include "webrtc/modules/audio_processing/test/test_utils.h"
  
  namespace webrtc {
  namespace test {
  
  // Holds all the parameters available for controlling the simulation.
  struct SimulationSettings {
    SimulationSettings();
    SimulationSettings(const SimulationSettings&);
    ~SimulationSettings();
    rtc::Optional<int> stream_delay;
    rtc::Optional<int> stream_drift_samples;
    rtc::Optional<int> output_sample_rate_hz;
    rtc::Optional<int> output_num_channels;
    rtc::Optional<int> reverse_output_sample_rate_hz;
    rtc::Optional<int> reverse_output_num_channels;
    rtc::Optional<std::string> microphone_positions;
    int target_angle_degrees = 90;
    rtc::Optional<std::string> output_filename;
    rtc::Optional<std::string> reverse_output_filename;
    rtc::Optional<std::string> input_filename;
    rtc::Optional<std::string> reverse_input_filename;
    rtc::Optional<bool> use_aec;
    rtc::Optional<bool> use_aecm;
    rtc::Optional<bool> use_agc;
    rtc::Optional<bool> use_hpf;
    rtc::Optional<bool> use_ns;
    rtc::Optional<bool> use_ts;
    rtc::Optional<bool> use_bf;
    rtc::Optional<bool> use_ie;
    rtc::Optional<bool> use_vad;
    rtc::Optional<bool> use_le;
    rtc::Optional<bool> use_all;
    rtc::Optional<int> aec_suppression_level;
    rtc::Optional<bool> use_delay_agnostic;
    rtc::Optional<bool> use_extended_filter;
    rtc::Optional<bool> use_drift_compensation;
    rtc::Optional<bool> use_aec3;
    rtc::Optional<bool> use_lc;
    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 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;
  };
  
  // Holds a few statistics about a series of TickIntervals.
  struct TickIntervalStats {
    TickIntervalStats() : min(std::numeric_limits<int64_t>::max()) {}
    int64_t sum;
    int64_t max;
    int64_t min;
  };
  
  // 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;
  
    explicit AudioProcessingSimulator(const SimulationSettings& settings);
    virtual ~AudioProcessingSimulator();
  
    // Processes the data in the input.
    virtual void Process() = 0;
  
    // Returns the execution time of all AudioProcessing calls.
    const TickIntervalStats& proc_time() const { return proc_time_; }
  
    // 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:
    // RAII class for execution time measurement. Updates the provided
    // TickIntervalStats based on the time between ScopedTimer creation and
    // leaving the enclosing scope.
    class ScopedTimer {
     public:
      explicit ScopedTimer(TickIntervalStats* proc_time)
          : 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 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;
  
   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_;
  
    RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
  };
  
  }  // namespace test
  }  // namespace webrtc
  
  #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
  