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/**********************************************************************
Audacity: A Digital Audio Editor
InterpolateAudio.h
Dominic Mazzoni
*******************************************************************//*!
\file Matrix.h
\brief General routine to interpolate (or even extrapolate small amounts)
audio when a few of the samples are bad. Works great for a few
dozen bad samples, but not so well with hundreds. Uses the
least-squares autoregression (LSAR) algorithm, as described in:
Simon Godsill, Peter Rayner, and Olivier Cappe. Digital Audio Restoration.
Berlin: Springer, 1998.
This is the same work used by Gnome Wave Cleaner (GWC), however this
implementation is original.
*//*******************************************************************/
#ifndef __AUDACITY_INTERPOLATE_AUDIO__
#define __AUDACITY_INTERPOLATE_AUDIO__
#include <cstddef>
// See top of file for a description of the algorithm. Interpolates
// the samples from buffer[firstBad] through buffer[firstBad+numBad-1],
// ignoring whatever value was there previously, and replacing them with
// values determined from the surrounding audio. Works best if the bad
// samples are in the middle, with several times as much data on either
// side (6x the number of bad samples on either side is great). However,
// it will work with less data, and with the bad samples on one end or
// the other.
void MATH_API InterpolateAudio(float *buffer, size_t len,
size_t firstBad, size_t numBad);
#endif // __AUDACITY_INTERPOLATE_AUDIO__
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