text: /**********************************************************************
  
    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 AUDACITY_DLL_API InterpolateAudio(float *buffer, size_t len,
                                         size_t firstBad, size_t numBad);
  
  #endif // __AUDACITY_INTERPOLATE_AUDIO__
  