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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
1.2 KiB
C++
45 lines
1.2 KiB
C++
/*
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* Copyright (c) 2015 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_REAL_FOURIER_OOURA_H_
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#define COMMON_AUDIO_REAL_FOURIER_OOURA_H_
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#include <stddef.h>
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#include <complex>
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#include <memory>
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#include "common_audio/real_fourier.h"
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namespace webrtc {
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class RealFourierOoura : public RealFourier {
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public:
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explicit RealFourierOoura(int fft_order);
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~RealFourierOoura() override;
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void Forward(const float* src, std::complex<float>* dest) const override;
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void Inverse(const std::complex<float>* src, float* dest) const override;
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int order() const override;
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private:
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const int order_;
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const size_t length_;
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const size_t complex_length_;
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// These are work arrays for Ooura. The names are based on the comments in
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// fft4g.c.
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const std::unique_ptr<size_t[]> work_ip_;
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const std::unique_ptr<float[]> work_w_;
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};
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} // namespace webrtc
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#endif // COMMON_AUDIO_REAL_FOURIER_OOURA_H_
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