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sndfile-resample (from libsamplerate) are already
included in the Planet CCRMA
If you are running Red Hat Linux, check out the Planet.
- There is also
libsoxr, the SoX resampler library) to change sampling
rates by this method. Say port search sox (Mac),
yum search sox (Linux), etc., to find it. See also
ssrc (from Shibatch) which is a fixed-point sampling-rate
conversion library installable
for Debian and Ubuntu. There is a project combining ssrc
and sox on
- Check out
for a large list of implementations and their relative performance. Seriously check it out--the coverage is amazing.
High quality command-line audio sample rate converter - LGPL-2.1 License
Python (Cython) implementation released under a BSD-style (ISC) license, since 2016
- Brick is an arbitrary-quality audio resampler,
pitch-shifter, and format converter written and maintained by
William Andrew Burnson:
- Smarc describes itself as a fast and high quality audio rate
converter, allowing conversion between any two sampling rates, and
optimized for standard audio rates, available as a command-line program or C
(502 Kbytes) (v1.8.1 released November 11, 2006)
The resample software package contains free sampling-rate conversion
and filter design utilities written in C, including a stand-alone
command-line sampling-rate conversion utility called
resample. The package compiles readily under Linux and most other
UNIX operating systems. It is released under the GNU Lesser
General Public License (LGPL).
- Erik de Castro Lopo's ``SecretRabbitCode''
- Same basic algorithm
- Floating-point (resample is fixed-point--sometimes needed on DSP chips)
- Configures for non-Linux platforms such as Mac OS X and Windows
- Includes the command-line program
sndfile-resample which works like resample
- Uses Erik's libsndfile (resample uses Bill Schottstaedt's sndlib)
- License is either GPL or commercial (resample is LGPL)
(LGPL) (version 1.3 released Jan. 7, 2004)
by Dominic Mazzoni (used in Audacity) is a real-time library for sampling rate
conversion providing several useful features relative to
resample-1.7 on which it is based:
- It should build ``out of the box'' on more platforms, including
Linux, Solaris, and Mac OS X
(using the included configure script).
There is also a Visual C++ project file for building under Windows.
- Input and output signals are in memory (as opposed to sound files).
- Computations are in floating-point (instead of fixed-point).
- Filter table increased by a factor of 32, yielding more accurate results, even
without linear interpolation (which also makes it faster).
- Data can be processed in small chunks, enabling time-varying
resampling ratios (ideal for time-warping applications and supporting
an ``external clock input'' in software).
- Easily applied to any number of simultaneous data channels
- README file for libresample-0.1.3
is a Java port of libresample for real-time sampling-rate
conversion by dnault-laszlo.
- The free Open Source Audio Library
(LGPL) contains a
C++ class based on resample.
- The Speex
speech coder/decoder (based on CELP) contains a variation of the
resample algorithm in the file resample.c, is free
and open-source, and is released under a BSD-style license (i.e.,
free for both commercial and noncommercial uses). The code is
characterized as ``working, but it's in a very early stage, so it
may have artifacts ... the API isn't stable ...''. However,
this appears to be the best BSD-licensed starting point.
(MIT License) is specialized for a factor of 2 stereo upsampling,
allowing it to achieve significant computational savings at a
perceptually non-degraded quality level.
- Website comparing various sampling-rate converters:
- Paper on optimal FIR interpolation filter design:
To report bugs, or contribute enhancements, please send email to
jos (jos at ccrma).
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