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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_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
#include <stddef.h>
#include <type_traits>
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/agc2/vad_wrapper.h"
#include "modules/audio_processing/include/audio_processing.h"
namespace webrtc {
class ApmDataDumper;
// Level estimator for the digital adaptive gain controller.
class AdaptiveModeLevelEstimator {
public:
AdaptiveModeLevelEstimator(
ApmDataDumper* apm_data_dumper,
const AudioProcessing::Config::GainController2::AdaptiveDigital& config);
AdaptiveModeLevelEstimator(const AdaptiveModeLevelEstimator&) = delete;
AdaptiveModeLevelEstimator& operator=(const AdaptiveModeLevelEstimator&) =
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 IsConfident() const;
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 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_;
int num_adjacent_speech_frames_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_