NAME

       zresample - resample and change sample format of audio files

SYNOPSIS

       zresample [options] input-file output-file

DESCRIPTION

       zresample  copies  an  audio  file, changing the sample rate and/or the sample format. For
       16-bit output it can also dither the audio signal. Input can be any audio file readable by
       the libsndfile library. The output file type is either WAV, WAVEX, CAF, AIFF or FLAC.

OPTIONS

       --help Display a short help text.

       --rate sample-rate
              Set  the output sample rate.  Zresample uses the zita-resampler library which means
              that not all combinations of  input/output  sample  rate  will  be  accepted.   The
              resample ratio must be a reducable to a fraction a/b with both a and b less than or
              equal to 1000.

       --gain gain
              Gain in dB, default zero.

   Output file type
       --wav  Produce a WAV file, or for more than  2  channels,  a  WAVEX  file.   This  is  the
              default.

       --amb  Produce  a  WAVEX  file  with  the  Ambisonic GUID. Such files should have the .amb
              filename extension.

       --caf  Produce a Core Audio file.

       --aiff Produce an AIFF file.

       --flac Produce a FLAC file.

   Output sample format
       --16bit
              Output sample format is  signed  16-bit.  This  option  also  enables  the  use  of
              dithering, described below.

       --24bit
              Output sample format is 24-bit. This is the default.

       --float
              Output sample format is 32-bit floating point.

   Dithering
       --rec  Add  white dithering noise with a rectangular distribution. This is the best option
              if the output data is going to processed again,  but  in  that  case  it  would  be
              advisable to use 24-bit or float.

       --tri  Add  filtered  noise  with  a  triangular distribution. Compared to the rectangular
              dither this reduces the noise density in the lower frequency range.

       --lips This uses the optimal error feedback filter described by Stanley Lipschitz. This is
              recommended is the output is the final distribution format, e.g. for a CD.

   Timing
       --pad  Insert  zero  valued input samples at the start and end so that the output includes
              the full symmetric filter response even for the first and last samples.

EXIT STATUS

       Zero in case there are no errors, non-zero otherwise.

AUTHOR

       Fons Adriaensen (fons (at) linuxaudio.org)