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/*
* Copyright (c) 2025 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_EXPERIMENTAL_IMPL_H_
#define MODULES_AUDIO_PROCESSING_AGC2_SPEECH_LEVEL_ESTIMATOR_EXPERIMENTAL_IMPL_H_
#include <type_traits>
#include "api/audio/audio_processing.h"
#include "modules/audio_processing/agc2/speech_level_estimator.h"
namespace webrtc {
class ApmDataDumper;
// Active speech level estimator based on the analysis of RMS level (dBFS), and
// speech probability.
class SpeechLevelEstimatorExperimentalImpl : public SpeechLevelEstimator {
public:
SpeechLevelEstimatorExperimentalImpl(
ApmDataDumper* apm_data_dumper,
const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
int adjacent_speech_frames_threshold);
SpeechLevelEstimatorExperimentalImpl(
const SpeechLevelEstimatorExperimentalImpl&) = delete;
SpeechLevelEstimatorExperimentalImpl& operator=(
const SpeechLevelEstimatorExperimentalImpl&) = delete;
// Updates the level estimation.
void Update(float rms_dbfs, float speech_probability) override;
// Returns the estimated speech plus noise level.
float GetLevelDbfs() const override { return level_dbfs_; }
// Returns true if the estimator is confident on its current estimate.
bool IsConfident() const override { return is_confident_; }
void Reset() override;
private:
// Part of the level estimator state used for check-pointing and restore ops.
struct LevelEstimatorState {
int num_frames;
float sum_of_levels_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_;
float tracking_level_dbfs_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_SPEECH_LEVEL_ESTIMATOR_EXPERIMENTAL_IMPL_H_