resample - resample a 16-bit mono or stereo sound file by an arbitrary
resample [-by factor] [-to newSrate] [-f filterFile] [-n] [-l] [-trace]
[-version] inputFile [outputFile]
The resample program takes a 16-bit mono or stereo sound file and a
"sampling-rate conversion factor" r, specified as a floating-point num-
ber, and produces an output sound file whose sampling rate is r times
that of the input file.
The output file is in AIFF format. If you omit the output file name,
resample will create a file using the input file name, with ".resamp"
Specify conversion factor. This option or "-toSrate" is
required. The conversion factor is the amount by which the sam-
pling rate is changed. If the sampling rate of the input signal
is Srate1, then the sampling rate of the output is fac-
tor*Srate1. For example, a factor of 2.0 increases the sampling
rate by a factor of 2, giving twice as many samples in the out-
put signal as in the input. The fractional part of the conver-
sion factor is accurate to 15 bits. This is sufficiently accu-
rate that humans should not be able to hear any error whatsoever
in the pitch of resampled sounds.
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.
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 alias-
ing (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.
By default, the resampling filter table is linearly interpolated
to provide high audio quality at arbitrary sampling-rate conver-
sion factors. This option turns off filter interpolation, thus
tion of the signal with linear interpolation of the resampling-
filter-table which is controlled by the "noFilterInterp" option.
-terse Disable informational printout.
Print program version.
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 -by 0.91875 dat.snd cd.snd
or, more simply,
resample -to 44100 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.deemphasis in the
resample release for further information)
Source code and further documentation may be found at the Digital Audio
Resampling Home Page located at
The first version of this software was written by Julius O. Smith III
<email@example.com> at CCRMA <http://www-ccrma.stanford.edu> in
1981. It was called SRCONV and was written in SAIL for PDP-10 compati-
ble machines. The algorithm was first published in
Smith, Julius O. and Phil Gossett. ``A Flexible Sampling-Rate Conver-
sion 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 Digital
Audio Resampling Home Page given above.
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 <firstname.lastname@example.org>.
The resample program is free software distributed in accordance with