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.