NAME

       resample - resample a 16-bit mono or stereo sound file by an arbitrary factor

SYNOPSIS

       resample  [-by  factor]  [-to  newSrate]  [-f  filterFile]  [-n]  [-l] [-trace] [-version]
       inputFile [outputFile]

DESCRIPTION

       The resample program takes a 16-bit mono or stereo sound  file  and  performs  bandlimited
       interpolation  to  produce  an  output  sound  file have a desired new sampling rate.  The
       output file is in the same format as the input.

OPTIONS

       -toSrate
              This option or "-byFactor" is required.  Specify new sampling rate in  samples  per
              second.   The conversion factor is implied and will be set to the new sampling rate
              divided by the sampling rate of the input soundfile.

       -byFactor
              Specify conversion factor.  This option or "-toSrate" is required.  The  conversion
              factor  is  the amount by which the sampling rate is changed.  If the sampling rate
              of  the  input  signal  is  Srate1,  then  the  sampling  rate  of  the  output  is
              factor*Srate1.   For  example,  a  factor  of  2.0 increases the sampling rate by a
              factor of 2, giving twice as many samples in the output signal  as  in  the  input.
              The  fractional  part  of  the  conversion  factor is accurate to 15 bits.  This is
              sufficiently accurate that humans should not be able to hear any  error  whatsoever
              in the pitch of resampled sounds.

       -filterFile
              Change  the resampling filter from its default.  Such a filter file can be designed
              by the windowfilter (1) program (included with  the  resample  distribution).   The
              preloaded  filter  file  requires  an  oversampling factor of at least 20% to avoid
              aliasing (in other words, its "transition band" as a lowpass filter is at least 20%
              of the useable frequency range in the sampled signal); the stop-band attenuation is
              approximately 80 dB.

       -noFilterInterp
              By default, the resampling filter table is linearly interpolated  to  provide  high
              audio quality at arbitrary sampling-rate conversion factors.  This option turns off
              filter interpolation, thus cutting the number of multiply-adds in half in the inner
              loop (for most conversion factors).

       -linearInterpolation
              Select  plain  linear  interpolation  for resampling (which means resampling filter
              table is not used at all). This option is very fast, but the output quality is poor
              unless   the  signal  is  already  heavily  oversampled.   Do  not  confuse  linear
              interpolation of the signal with linear  interpolation  of  the  resampling-filter-
              table which is controlled by the "noFilterInterp" option.

       -terse Disable informational printout.

       -version
              Print program version.

EXAMPLE

       To  convert the sampling rate from 48 kHz (used by DAT machines) to 44.1 kHz (the standard
       sampling rate for Compact Discs), the command line would look something like

            resample -to 44100 dat.snd cd.snd or      resample -by 0.91875 dat.snd cd.snd

       Any reasonable sampling rate can be converted to any other.  (Note that, in this  example,
       if you have obtained a direct-digital transfer from DAT or CD, you probably have some pre-
       emphasis filtering which should be canceled using a digital filter. See  README.deemph  in
       the resample release for further information)

REFERENCES

       Source  code  and  further documentation may be found at the Digital Audio Resampling Home
       Page (DARHP) located at

            http://ccrma.stanford.edu/~jos/resample/

HISTORY

       The first version of this software was written by Julius O.  Smith  III  <jos  /at/  ccrma
       /dot/  stanford  /dot/  edu>  at CCRMA <http://ccrma.stanford.edu> in 1981.  It was called
       SRCONV and was written in SAIL for PDP-10 compatible machines  (see  the  DARHP  for  that
       code).  The algorithm was first published in

       Smith,  Julius  O.  and  Phil  Gossett.  ``A  Flexible  Sampling-Rate Conversion Method,''
       Proceedings  (2):  19.4.1-19.4.4,  IEEE  Conference  on  Acoustics,  Speech,  and   Signal
       Processing, San Diego, March 1984.

       An expanded tutorial based on this paper is available at the DARHP.

       Circa  1988,  the  SRCONV  program was translated from SAIL to C by Christopher Lee Fraley
       working with Roger Dannenberg at CMU.

       Since then, the C version has been maintained by jos.

       Sndlib support was added 6/99 by John Gibson <jgg9c@virginia.edu>.

       The resample program is free software distributed in accordance with the Lesser GNU Public
       License  (LGPL).   There  is  NO  warranty;  not even for MERCHANTABILITY or FITNESS FOR A
       PARTICULAR PURPOSE.