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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.
59 lines
2.0 KiB
C
59 lines
2.0 KiB
C
/*
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* Copyright (c) 2011 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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/*--------------------------------*-C-*---------------------------------*
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* File:
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* fftn.h
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* ---------------------------------------------------------------------*
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* Re[]: real value array
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* Im[]: imaginary value array
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* nTotal: total number of complex values
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* nPass: number of elements involved in this pass of transform
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* nSpan: nspan/nPass = number of bytes to increment pointer
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* in Re[] and Im[]
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* isign: exponent: +1 = forward -1 = reverse
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* scaling: normalizing constant by which the final result is *divided*
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* scaling == -1, normalize by total dimension of the transform
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* scaling < -1, normalize by the square-root of the total dimension
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*
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* ----------------------------------------------------------------------
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* See the comments in the code for correct usage!
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*/
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#ifndef MODULES_THIRD_PARTY_FFT_FFT_H_
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#define MODULES_THIRD_PARTY_FFT_FFT_H_
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#define FFT_MAXFFTSIZE 2048
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#define FFT_NFACTOR 11
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typedef struct {
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unsigned int SpaceAlloced;
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unsigned int MaxPermAlloced;
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double Tmp0[FFT_MAXFFTSIZE];
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double Tmp1[FFT_MAXFFTSIZE];
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double Tmp2[FFT_MAXFFTSIZE];
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double Tmp3[FFT_MAXFFTSIZE];
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int Perm[FFT_MAXFFTSIZE];
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int factor[FFT_NFACTOR];
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} FFTstr;
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/* double precision routine */
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int WebRtcIsac_Fftns(unsigned int ndim,
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const int dims[],
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double Re[],
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double Im[],
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int isign,
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double scaling,
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FFTstr* fftstate);
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#endif /* MODULES_THIRD_PARTY_FFT_FFT_H_ */
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