| /* |
| * 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_ |