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/*
* 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 <memory>
namespace webrtc {
class ApmDataDumper;
// Saturation protector. Analyzes peak levels and recommends a headroom to
// reduce the chances of clipping.
class SaturationProtector {
public:
virtual ~SaturationProtector() = default;
// Returns the recommended headroom in dB.
virtual float HeadroomDb() = 0;
// Analyzes the peak level of a 10 ms frame along with its speech probability
// and the current speech level estimate to update the recommended headroom.
virtual void Analyze(float speech_probability,
float peak_dbfs,
float speech_level_dbfs) = 0;
// Resets the internal state.
virtual void Reset() = 0;
};
// Creates a saturation protector that starts at `initial_headroom_db`.
std::unique_ptr<SaturationProtector> CreateSaturationProtector(
float initial_headroom_db,
int adjacent_speech_frames_threshold,
ApmDataDumper* apm_data_dumper);
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
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