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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 that differentiates background speakers and
// low-activity segments to avoid adapting to secondary speech or corrupting the
// tracked target speaker level.
//
// Once the estimator reaches confidence on the primary speaker's level,
// target speaker inactivity is detected in two ways:
// 1. Level drop (background speaker): If accumulated reliable speech is
// quieter than the tracked level by at least
// `background_speaker_offset_dbfs`, it is classified as a background speaker
// and the tracked level is retained.
// 2. Low-activity segment: A primary speaker typically triggers the VAD
// consistently, whereas a distant speaker or pauses between speech segments
// produce sporadic bursts. If accumulating the required speech frames
// exceeds `kMaxTimeToUpdateMs`, accumulation is reset and the segment is
// flagged as low activity.
class SpeechLevelEstimatorExperimentalImpl : public SpeechLevelEstimator {
public:
static constexpr float kDefaultBackgroundSpeakerOffsetDbfs = 7.0f;
static constexpr int kDefaultMaxTimeToUpdateMs = 5000;
SpeechLevelEstimatorExperimentalImpl(
ApmDataDumper* apm_data_dumper,
const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
int adjacent_speech_frames_threshold,
float background_speaker_offset_dbfs,
int max_time_to_update_ms);
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_; }
// Returns true if the target speaker is actively speaking.
// Returns false if speech is classified as a background speaker or if
// speech activity is too sporadic to reliably update the target level.
bool IsTargetSpeakerActive() const override {
return !is_background_speaker_ && !low_activity_segment_;
}
// Returns the threshold offset in dBFS for background speaker detection.
float GetBackgroundSpeakerOffsetDbfs() const {
return background_speaker_offset_dbfs_;
}
// Returns the maximum time in ms to update the level before resetting state.
int GetMaxTimeToUpdateMs() const { return max_time_to_update_ms_; }
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();
void DumpDebugData() const;
ApmDataDumper* const apm_data_dumper_;
const float initial_speech_level_dbfs_;
const int adjacent_speech_frames_threshold_;
const float background_speaker_offset_dbfs_;
const int max_time_to_update_ms_;
const int max_frames_to_update_;
LevelEstimatorState preliminary_state_;
LevelEstimatorState reliable_state_;
float level_dbfs_;
bool is_confident_;
bool is_background_speaker_;
bool low_activity_segment_;
int num_adjacent_speech_frames_;
int num_frames_in_current_update_window_;
float tracking_level_dbfs_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_SPEECH_LEVEL_ESTIMATOR_EXPERIMENTAL_IMPL_H_