speech_level_estimator.h

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/*

 *  Copyright (c) 2018 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_AGC2_SPEECH_LEVEL_ESTIMATOR_H_

#define MODULES_AUDIO_PROCESSING_AGC2_SPEECH_LEVEL_ESTIMATOR_H_

#include <stddef.h>

#include <type_traits>

#include "api/audio/audio_processing.h"

#include "modules/audio_processing/agc2/agc2_common.h"

namespace webrtc {

class ApmDataDumper;

// Active speech level estimator based on the analysis of the following

// framewise properties: RMS level (dBFS), peak level (dBFS), speech

// probability.

class SpeechLevelEstimator {

 public:

  SpeechLevelEstimator(

      ApmDataDumper* apm_data_dumper,

      const AudioProcessing::Config::GainController2::AdaptiveDigital& config,

      int adjacent_speech_frames_threshold);

  SpeechLevelEstimator(const SpeechLevelEstimator&) = delete;

  SpeechLevelEstimator& operator=(const SpeechLevelEstimator&) = delete;

  // Updates the level estimation.

  void Update(float rms_dbfs, float peak_dbfs, float speech_probability);

  // Returns the estimated speech plus noise level.

  float level_dbfs() const { return level_dbfs_; }

  // Returns true if the estimator is confident on its current estimate.

  bool is_confident() const { return is_confident_; }

  void Reset();

 private:

  // Part of the level estimator state used for check-pointing and restore ops.

  struct LevelEstimatorState {

    bool operator==(const LevelEstimatorState& s) const;

    inline bool operator!=(const LevelEstimatorState& s) const {

      return !(*this == s);

    // TODO(bugs.webrtc.org/7494): Remove `time_to_confidence_ms` if redundant.

    int time_to_confidence_ms;

    struct Ratio {

      float numerator;

      float denominator;

      float GetRatio() const;

    } level_dbfs;

};

  static_assert(std::is_trivially_copyable<LevelEstimatorState>::value, "");

  void UpdateIsConfident();

  void ResetLevelEstimatorState(LevelEstimatorState& state) const;

  void DumpDebugData() const;

  ApmDataDumper* const apm_data_dumper_;

  const float initial_speech_level_dbfs_;

  const int adjacent_speech_frames_threshold_;

  LevelEstimatorState preliminary_state_;

  LevelEstimatorState reliable_state_;

  float level_dbfs_;

  bool is_confident_;

  int num_adjacent_speech_frames_;

};

}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AGC2_SPEECH_LEVEL_ESTIMATOR_H_