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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
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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