source: src/musicutils.h @ 39a7b26

/*
  Copyright (C) 2003-2009 Paul Brossier <piem@aubio.org>

  This file is part of aubio.

  aubio 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 3 of the License, or
  (at your option) any later version.

  aubio 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 aubio.  If not, see <http://www.gnu.org/licenses/>.

*/

/** @file
 *  various functions useful in audio signal processing
 */

#ifndef MUSICUTILS_H
#define MUSICUTILS_H

#ifdef __cplusplus
extern "C" {
#endif

/** create window 
 
  References:
    
    - <a href="http://en.wikipedia.org/wiki/Window_function">Window
function</a> on Wikipedia
    - Amalia de Götzen, Nicolas Bernardini, and Daniel Arfib. Traditional (?)
implementations of a phase vocoder: the tricks of the trade. In Proceedings of
the International Conference on Digital Audio Effects (DAFx-00), pages 37–44,
Uni- versity of Verona, Italy, 2000.
  (<a href="http://profs.sci.univr.it/%7Edafx/Final-Papers/ps/Bernardini.ps.gz">
  ps.gz</a>)

*/
fvec_t *new_aubio_window (char_t * window_type, uint_t size);

/** set elements of a vector to window coefficients

 */
uint_t fvec_set_window (fvec_t * window, char_t * window_type);

/** compute the principal argument

  This function maps the input phase to its corresponding value wrapped in the
range \f$ [-\pi, \pi] \f$.

  \param phase unwrapped phase to map to the unit circle
  
  \return equivalent phase wrapped to the unit circle

*/
smpl_t aubio_unwrap2pi (smpl_t phase);

/** 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);

/** clean up cached memory at the end of program
 
  This function should be used at the end of programs to purge all cached
  memory. So far it is only useful to clean FFTW's cache.

*/
void aubio_cleanup (void);

/** zero-crossing rate (ZCR)

  The zero-crossing rate is the number of times a signal changes sign,
  divided by the length of this signal.

  \param v vector to compute ZCR from

  \return zero-crossing rate of v

*/
smpl_t aubio_zero_crossing_rate (fvec_t * v);

/** compute sound level on a linear

  This gives the average of the square amplitudes.

  \param v vector to compute dB SPL from

  \return level of v

*/
smpl_t aubio_level_lin (fvec_t * v);

/** compute sound pressure level (SPL) in dB

  This quantity is often wrongly called 'loudness'.

  This gives ten times the log10 of the average of the square amplitudes.

  \param v vector to compute dB SPL from

  \return level of v in dB SPL

*/
smpl_t aubio_db_spl (fvec_t * v);

/** check if buffer level in dB SPL is under a given threshold
 
  \param v vector to get level from
  \param threshold threshold in dB SPL

  \return 0 if level is under the given threshold, 1 otherwise

*/
uint_t aubio_silence_detection (fvec_t * v, smpl_t threshold);

/** get buffer level if level >= threshold, 1. otherwise

  \param v vector to get level from
  \param threshold threshold in dB SPL

  \return level in dB SPL if level >= threshold, 1. otherwise

*/
smpl_t aubio_level_detection (fvec_t * v, smpl_t threshold);

#ifdef __cplusplus
}
#endif

#endif