| /* |
| * 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_SATURATION_PROTECTOR_H_ |
| #define MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_ |
| |
| #include <array> |
| |
| #include "absl/types/optional.h" |
| #include "modules/audio_processing/agc2/agc2_common.h" |
| |
| namespace webrtc { |
| |
| class ApmDataDumper; |
| |
| class SaturationProtector { |
| public: |
| explicit SaturationProtector(ApmDataDumper* apm_data_dumper); |
| SaturationProtector(ApmDataDumper* apm_data_dumper, |
| float initial_saturation_margin_db); |
| |
| void Reset(); |
| |
| // Updates the margin by analyzing the estimated speech level |
| // `speech_level_dbfs` and the peak power `speech_peak_dbfs` for an observed |
| // frame which is reliably classified as "speech". |
| void UpdateMargin(float speech_peak_dbfs, float speech_level_dbfs); |
| |
| // Returns latest computed margin. |
| float margin_db() const { return margin_db_; } |
| |
| void DebugDumpEstimate() const; |
| |
| private: |
| // Ring buffer which only supports (i) push back and (ii) read oldest item. |
| class RingBuffer { |
| public: |
| void Reset(); |
| // Pushes back `v`. If the buffer is full, the oldest item is replaced. |
| void PushBack(float v); |
| // Returns the oldest item in the buffer. Returns an empty value if the |
| // buffer is empty. |
| absl::optional<float> Front() const; |
| |
| private: |
| std::array<float, kPeakEnveloperBufferSize> buffer_; |
| int next_ = 0; |
| int size_ = 0; |
| }; |
| |
| float GetDelayedPeakDbfs() const; |
| |
| ApmDataDumper* apm_data_dumper_; |
| // Parameters. |
| const float initial_saturation_margin_db_; |
| // State. |
| float margin_db_; |
| RingBuffer peak_delay_buffer_; |
| float max_peaks_dbfs_; |
| int time_since_push_ms_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_ |