view aubioFullOSXUni/include/aubio/mathutils.h @ 0:572c856e38ac

Starting up openFrameworks project for audio time warping. The ofxFileReader goes in addons of your OF folder, the libraries and source (chromogram, fftw and source code src+ timewarp) are probably best kept in the repository, then dragged into the project afresh. That way, as we update the repository, the code that the openFrameworks project looks for will be updated.

/*
	 Copyright (C) 2003 Paul Brossier

	 This program is free software; you can redistribute it and/or modify
	 it under the terms of the GNU General Public License as published by
	 the Free Software Foundation; either version 2 of the License, or
	 (at your option) any later version.

	 This program is distributed in the hope that it will be useful,
	 but WITHOUT ANY WARRANTY; without even the implied warranty of
	 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
	 GNU General Public License for more details.

	 You should have received a copy of the GNU General Public License
	 along with this program; if not, write to the Free Software
	 Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.

*/

/** @file
 *  various math functions
 *
 *  \todo multichannel (each function should return -or set- an array sized to
 *  the number of channel in the input vector)
 *
 *  \todo appropriate switches depending on types.h content
 */

#ifndef MATHUTILS_H
#define MATHUTILS_H

/** Window types 
 * 
 * inspired from 
 *
 *  - dafx : http://profs.sci.univr.it/%7Edafx/Final-Papers/ps/Bernardini.ps.gz
 *  - freqtweak : http://freqtweak.sf.net/
 *  - extace : http://extace.sf.net/
 */

#ifdef __cplusplus
extern "C" {
#endif

typedef enum {
	aubio_win_rectangle,          
	aubio_win_hamming,
	aubio_win_hanning,
	aubio_win_hanningz,
	aubio_win_blackman,
	aubio_win_blackman_harris,
	aubio_win_gaussian,
	aubio_win_welch,
	aubio_win_parzen
} aubio_window_type;

/** create window */
void aubio_window(smpl_t *w, ba_uint_t size, aubio_window_type wintype);

/** principal argument
 *
 * mod(phase+PI,-TWO_PI)+PI 
 */
smpl_t aubio_unwrap2pi (smpl_t phase);

/** calculates the mean of a vector
 *
 * \bug mono 
 */
smpl_t vec_mean(fvec_t *s);
/** returns the max of a vector
 *
 * \bug mono 
 */
smpl_t vec_max(fvec_t *s);
/** returns the min of a vector
 *
 * \bug mono 
 */
smpl_t vec_min(fvec_t *s);
/** returns the index of the min of a vector
 *
 * \bug mono 
 */
ba_uint_t vec_min_elem(fvec_t *s);
/** returns the index of the max of a vector
 *
 * \bug mono 
 */
ba_uint_t vec_max_elem(fvec_t *s);
/** implement 'fftshift' like function
 * 
 * a[0]...,a[n/2],a[n/2+1],...a[n]
 * 
 * becomes
 * 
 * a[n/2+1],...a[n],a[0]...,a[n/2]
 */
void vec_shift(fvec_t *s);
/** returns sum */
smpl_t vec_sum(fvec_t *s);
/** returns energy 
 *
 * \bug mono 
 */
smpl_t vec_local_energy(fvec_t * f);
/** returns High Frequency Energy Content
 *
 * \bug mono */
smpl_t vec_local_hfc(fvec_t * f);
/** return alpha norm.
 *
 * alpha=2 means normalise variance. 
 * alpha=1 means normalise abs value. 
 * as alpha goes large, tends to normalisation 
 * by max value.
 *
 * \bug should not use POW :(
 */
smpl_t vec_alpha_norm(fvec_t * DF, smpl_t alpha);
/**  dc(min) removal */
void vec_dc_removal(fvec_t * mag);
/**  alpha normalisation */
void vec_alpha_normalise(fvec_t * mag, ba_uint_t alpha);
/** add a constant to all members of a vector */
void vec_add(fvec_t * mag, smpl_t threshold);

/** compute adaptive threshold of input vector */ 
void vec_adapt_thres(fvec_t * vec, fvec_t * tmp, 
    ba_uint_t win_post, ba_uint_t win_pre);
/**  adaptative thresholding 
 *
 * y=fn_thresh(fn,x,post,pre)
 * compute adaptive threshold at each time 
 *    fn : a function name or pointer, eg 'median'
 *    x:   signal vector
 *    post: window length, causal part
 *    pre: window length, anti-causal part
 * Returns:
 *    y:   signal the same length as x
 *
 * Formerly median_thresh, used compute median over a 
 * window of post+pre+1 samples, but now works with any
 * function that takes a vector or matrix and returns a
 * 'representative' value for each column, eg
 *    medians=fn_thresh(median,x,8,8)  
 *    minima=fn_thresh(min,x,8,8)  
 * see SPARMS for explanation of post and pre
 */
smpl_t vec_moving_thres(fvec_t * vec, fvec_t * tmp, 
    ba_uint_t win_post, ba_uint_t win_pre, ba_uint_t win_pos);

/** returns the median of the vector 
 *
 *  This Quickselect routine is based on the algorithm described in
 *  "Numerical recipes in C", Second Edition,
 *  Cambridge University Press, 1992, Section 8.5, ISBN 0-521-43108-5
 *
 *  This code by Nicolas Devillard - 1998. Public domain,
 *  available at http://ndevilla.free.fr/median/median/
 */
smpl_t vec_median(fvec_t * input);

/** finds exact maximum position by quadratic interpolation*/
smpl_t vec_quadint(fvec_t * x,ba_uint_t pos);

/** finds exact minimum position by quadratic interpolation*/
smpl_t vec_quadint_min(fvec_t * x,ba_uint_t pos, ba_uint_t span);

/** Quadratic interpolation using Lagrange polynomial.
 *
 * inspired from ``Comparison of interpolation algorithms in real-time sound
 * processing'', Vladimir Arnost, 
 * 
 * estimate = s0 + (pf/2.)*((pf-3.)*s0-2.*(pf-2.)*s1+(pf-1.)*s2);
 *    where 
 *    \param s0,s1,s2 are 3 known points on the curve,
 *    \param pf is the floating point index [0;2]
 */
smpl_t aubio_quadfrac(smpl_t s0, smpl_t s1, smpl_t s2, smpl_t pf);

/** returns 1 if X1 is a peak and positive */
ba_uint_t vec_peakpick(fvec_t * input, ba_uint_t pos);

/** convert frequency bin to midi value */
smpl_t aubio_bintomidi(smpl_t bin, smpl_t samplerate, smpl_t fftsize);
/** convert midi value to frequency bin */
smpl_t aubio_miditobin(smpl_t midi, smpl_t samplerate, smpl_t fftsize);
/** convert frequency bin to frequency (Hz) */
smpl_t aubio_bintofreq(smpl_t bin, smpl_t samplerate, smpl_t fftsize);
/** convert frequency (Hz) to frequency bin */
smpl_t aubio_freqtobin(smpl_t freq, smpl_t samplerate, smpl_t fftsize);
/** convert frequency (Hz) to midi value (0-128) */
smpl_t aubio_freqtomidi(smpl_t freq);
/** convert midi value (0-128) to frequency (Hz) */
smpl_t aubio_miditofreq(smpl_t midi);

/** check if current buffer level is under a given threshold */
ba_uint_t aubio_silence_detection(fvec_t * ibuf, smpl_t threshold);
/** get the current buffer level */
smpl_t aubio_level_detection(fvec_t * ibuf, smpl_t threshold);
/** 
 * calculate normalised autocorrelation function
 */
void aubio_autocorr(fvec_t * input, fvec_t * output);
/**
 * clean up cached memory at the end of program
 *
 * use this function at the end of programs to purge all
 * cached memory. so far this function is only used to clean
 * fftw cache.
 */
void aubio_cleanup(void);

#ifdef __cplusplus
}
#endif

#endif