Extension and refactoring of the audioproc_f tool to be a fully fledged tool for audio processing m…

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 <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 {
+  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<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)
+      : settings_(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_