/* * 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 #include "modules/audio_processing/agc2/agc2_common.h" // kFullBufferSizeMs... #include "modules/audio_processing/agc2/saturation_protector.h" #include "modules/audio_processing/agc2/vad_with_level.h" #include "modules/audio_processing/include/audio_processing.h" namespace webrtc { class ApmDataDumper; class AdaptiveModeLevelEstimator { public: explicit AdaptiveModeLevelEstimator(ApmDataDumper* apm_data_dumper); AdaptiveModeLevelEstimator( ApmDataDumper* apm_data_dumper, AudioProcessing::Config::GainController2::LevelEstimator level_estimator, bool use_saturation_protector, float extra_saturation_margin_db); void UpdateEstimation(const VadWithLevel::LevelAndProbability& vad_data); float LatestLevelEstimate() const; void Reset(); bool LevelEstimationIsConfident() const { return buffer_size_ms_ >= kFullBufferSizeMs; } private: void DebugDumpEstimate(); const AudioProcessing::Config::GainController2::LevelEstimator level_estimator_; const bool use_saturation_protector_; size_t buffer_size_ms_ = 0; float last_estimate_with_offset_dbfs_ = kInitialSpeechLevelEstimateDbfs; float estimate_numerator_ = 0.f; float estimate_denominator_ = 0.f; SaturationProtector saturation_protector_; ApmDataDumper* const apm_data_dumper_; }; } // namespace webrtc #endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_