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Fast, free sample rate conversion

WARNING!  LibFResample is not release quality!

1. LibFResample does not report errors yet!

2. LibFResample only works on mono or stereo 16-bit data!  (Support
for 32-bit floating point as well as automatic data conversions is in
the works.)

3. LibFResample has an "in-progress" build system!  (Autoconf and
Xcode support is currently broken.)

4. LibFResample has not been ported to Windows!

LibFResample is a library for resampling audio with a permissive
(FreeBSD style) license.  Like other audio resampling libraries, it is
based on the bandlimited interpolation algorithm described by Julius
Orion Smith.  LibFResample is designed to use SIMD operations where
available.  Currently, LibFResample supports SSE2 and AltiVec.

If you can hook me up with some hardware, I may be able to optimize
this library for other architectures - ARM NEON, Cell SPE, et cetera.

Resampling speed

LibFResample gets its speed by precalculating filter coefficients for
the desired conversion ratio, reorganizing coefficients for maximum
cache locality, and using SIMD operations as much as possible.

System A: 1.6 GHz Intel Atom, GCC 4.4 (circa 2008)
System B: 2.0 GHz IBM PowerPC G5, GCC 4.0 (circa 2005)
System C: 3.2 GHz AMD Phenom II, GCC 4.7 (circa 2010)
System D: 1.7 GHz Intel Core i5, Clang 4.0 (circa 2012)

Task: Resample 16-bit stereo audio from 48 kHz to 44.1 kHz
Speed: 1x is realtime, 2x is twice realtime, etc.
SRC: Secret Rabbit Code 0.1.8, also known as libsamplerate

    Settings    System A    System B    System C    System D

    Q=5 Medium  197x        545x        1276x       1247x
    Q=8 High    47x         141x        318x        303x
    Q=10 Ultra  13x         51x         107x        87x

    Secret Rabbit Code
    Sinc Fast   13x         20x         68x         77x
    Sinc Med.   5.9x        9.3x        35x         39x
    Sinc Best   0.69x       1.9x        8.5x        11x

That's fast!  To put it in perspective, System D, an entry level
MacBook Air from 2012, can resample an hour of audio in under 3
seconds at medium quality (Q=5).  In fact, LibFResample is so fast, it
even beats Secret Rabbit Code's linear interpolator, which the docs
note is "blindingly fast"...

Task: Resample 16-bit mono audio from 48 kHz to 44.1 kHz

    Settings    System A    System B    System C    System D

    Q=2 Low     484x        887x        2759x       3195x
    Q=5 Medium  286x        721x        1875x       1811x

    Secret Rabbit Code
    ZOH         119x        250x        986x        1326x
    Linear      113x        242x        895x        1112x

Notes: SRC throughput figures were calculated by dividing the output
of SRC's throughput tests by 44100.  LibFResample throughput figures
were calculated by resampling two minutes of pink noise twenty times
after warming the cache.

The '' script in the tests folder will benchmark
LibFResample, it requires SoX.

Audio quality

LibFResample uses band-limited interpolation and dithering to achieve
high-quality output.  The filter is a simple windowed sinc filter with
a Kaiser window, the window size and beta parameter are adjusted to
achieve the desired SNR and transition band.  There are better ways to
design filters, but this works.

An included test script finds the bandwidth and signal to noise ratio
of the resampler at various quality settings.  The bandwidth and SNR
will vary depending on the exact sample rates used.  In particular,
the bandwidth decreases if the input sample rate increases -- but if
you record 96 kHz or 192 kHz audio, you probably want to use higher
quality settings anyway.

    Settings        Bandwidth   SNR

    Q=2  Low        13.3 kHz    29 dB
    Q=5  Medium     16.3 kHz    84 dB
    Q=8  High       19.5 kHz    92 dB
    Q=10 Ultra      21.2 kHz    92 dB

Proper dithering introduces a noise floor of 1 ULP peak-to-peak, which
at 16 bit resolution puts the noise floor at -96.  The test dithers
both input and output, giving a noise floor of -93 dB.  At high
quality settings, the measured SNR of 92 dB is nearly perfect.

Bandwidth is measured by doing a binary search to find the 3 dB
attenuation point.  The attenuation is measured by resampling windowed
sine waves.

The SNR is measured by repeatedly resampling sine waves and taking the
FFT of the resampled result.  The original sine wave is zeroed from
the FFT bins and the signal power in the other bins is computed.  Sine
waves at 40 different frequencies are tested and the worst SNR is

The '' script in the tests folder will compute these quality
figures, it requires SciPy.

Theory of operation

Resampling audio takes two steps: first you design a low-pass filter,
then you use the filter to resample the audio.  In general, a new
filter must be created for each combination of sample rates, filter
parameters, and bit depth.

The low-pass filter is used to remove aliasing from the resampled
output -- aliasing is undesirable noise created during the resampling
process.  Adjusting the filter parameters is the only way to change
the quality of the audio output.  There are two major parameters to
filter design:

1. Signal to noise ratio.  A filter with a higher SNR more efficiently
excludes aliasing noise, but increasing the SNR also makes the filter
larger.  Note that the noise is not pleasant noise like white noise;
the noise has a gritty, low-fi sound.

2. Bandwidth.  A filter with a higher bandwidth includes more of the
original signal, but increasing the bandwidth also makes the filter
larger.  (The crucial parameter is actually transition bandwidth,
between the filter's pass band and stop band.  Decreasing the
transition bandwidth increases the filter size.)

LibFResample includes a function which creates a filter with the
desired SNR and transition bandwidth, and there are also some preset
SNR and transition bandwidth combinations.  The function also performs
a number of adjustments of the design parameters to ensure that the
pass band is large enough and that the filter isn't overdesigned to
preserve ultrasonic frequencies.

Internal details

The filter is a simple sinc filter windowed with the Kaiser window.
Choosing the parameter for the Kaiser window allows us to adjust the
level of the side lobes, and higher side lobes contribute to aliasing
noise.  Increasing the filter size decreases the width of the window's
main lobe, which decreases the transition bandwidth in the resulting

Many copies of the filter are generated, each with a fractional (less
than one sample) offset from each other.  Choosing a copy with a given
offset allows us to sample the original audio at that offset.
Interpolating between copies allows us to sample the audio at finer
intervals without a significant increase in memory usage.

Filters may be created with 16-bit integer or single precision
floating point coefficients.  The cutoff is typically above Q=5.  As
filter sizes increase, coefficient quantization begins to dominate
stopband attenuation.

At low quality settings, the fractional copies are individually
normalized.  Otherwise, variations in DC gain will modulate the input


Fast audio sample rate conversion with simplified BSD license




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