mirror of
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.
33 lines
947 B
C++
33 lines
947 B
C++
/*
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* Copyright (c) 2014 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 COMMON_AUDIO_WINDOW_GENERATOR_H_
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#define COMMON_AUDIO_WINDOW_GENERATOR_H_
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#include <stddef.h>
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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// Helper class with generators for various signal transform windows.
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class WindowGenerator {
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public:
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static void Hanning(int length, float* window);
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static void KaiserBesselDerived(float alpha, size_t length, float* window);
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private:
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WindowGenerator);
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};
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} // namespace webrtc
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#endif // COMMON_AUDIO_WINDOW_GENERATOR_H_
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