c6abf6cd3f
Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone. We're continuing to carry iSAC even though it's gone upstream, but maybe we'll want to drop that soon.
47 lines
1.5 KiB
C++
47 lines
1.5 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
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#include <memory>
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namespace webrtc {
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class ApmDataDumper;
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// Saturation protector. Analyzes peak levels and recommends a headroom to
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// reduce the chances of clipping.
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class SaturationProtector {
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public:
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virtual ~SaturationProtector() = default;
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// Returns the recommended headroom in dB.
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virtual float HeadroomDb() = 0;
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// Analyzes the peak level of a 10 ms frame along with its speech probability
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// and the current speech level estimate to update the recommended headroom.
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virtual void Analyze(float speech_probability,
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float peak_dbfs,
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float speech_level_dbfs) = 0;
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// Resets the internal state.
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virtual void Reset() = 0;
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};
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// Creates a saturation protector that starts at `initial_headroom_db`.
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std::unique_ptr<SaturationProtector> CreateSaturationProtector(
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float initial_headroom_db,
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int adjacent_speech_frames_threshold,
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ApmDataDumper* apm_data_dumper);
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
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