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Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.h | 81 |
1 files changed, 81 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.h b/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.h new file mode 100644 index 0000000000..4d9f106ba9 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.h @@ -0,0 +1,81 @@ +/* + * 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_SPEECH_LEVEL_ESTIMATOR_H_ +#define MODULES_AUDIO_PROCESSING_AGC2_SPEECH_LEVEL_ESTIMATOR_H_ + +#include <stddef.h> + +#include <type_traits> + +#include "modules/audio_processing/agc2/agc2_common.h" +#include "modules/audio_processing/include/audio_processing.h" + +namespace webrtc { +class ApmDataDumper; + +// Active speech level estimator based on the analysis of the following +// framewise properties: RMS level (dBFS), peak level (dBFS), speech +// probability. +class SpeechLevelEstimator { + public: + SpeechLevelEstimator( + ApmDataDumper* apm_data_dumper, + const AudioProcessing::Config::GainController2::AdaptiveDigital& config, + int adjacent_speech_frames_threshold); + SpeechLevelEstimator(const SpeechLevelEstimator&) = delete; + SpeechLevelEstimator& operator=(const SpeechLevelEstimator&) = 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 is_confident() const { return is_confident_; } + + 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 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_; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_AGC2_SPEECH_LEVEL_ESTIMATOR_H_ |