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.
26 lines
765 B
C
26 lines
765 B
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 COMMON_AUDIO_THIRD_PARTY_FFT4G_FFT4G_H_
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#define COMMON_AUDIO_THIRD_PARTY_FFT4G_FFT4G_H_
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#if defined(__cplusplus)
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extern "C" {
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#endif
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// Refer to fft4g.c for documentation.
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void WebRtc_rdft(size_t n, int isgn, float* a, size_t* ip, float* w);
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#if defined(__cplusplus)
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}
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#endif
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#endif /* COMMON_AUDIO_THIRD_PARTY_FFT4G_FFT4G_H_ */
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