45 lines
1.4 KiB
C++
45 lines
1.4 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_GAIN_APPLIER_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
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#include <stddef.h>
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#include "modules/audio_processing/include/audio_frame_view.h"
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namespace webrtc {
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class GainApplier {
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public:
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GainApplier(bool hard_clip_samples, float initial_gain_factor);
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void ApplyGain(AudioFrameView<float> signal);
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void SetGainFactor(float gain_factor);
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float GetGainFactor() const { return current_gain_factor_; };
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private:
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void Initialize(size_t samples_per_channel);
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// Whether to clip samples after gain is applied. If 'true', result
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// will fit in FloatS16 range.
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const bool hard_clip_samples_;
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float last_gain_factor_;
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// If this value is not equal to 'last_gain_factor', gain will be
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// ramped from 'last_gain_factor_' to this value during the next
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// 'ApplyGain'.
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float current_gain_factor_;
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int samples_per_channel_ = -1;
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float inverse_samples_per_channel_ = -1.f;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
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