Open source (under the MIT license) high-quality professional audio sample rate converter (SRC) (resampling) C++ library. Features routines for SRC, both up- and downsampling, to/from any sample rate, including non-integer sample rates: it can be also used for conversion to/from SACD sample rate and even go beyond that. SRC routines were implemented in a multi-platform C++ code, and have a high level of optimality.
The structure of this library's objects is such that they can be frequently created and destroyed in large applications with a minimal performance impact due to a high level of reusability of its most "initialization-expensive" objects: the fast Fourier transform and FIR filter objects.
The SRC algorithm at first produces 2X oversampled (relative to the source sample rate, or the destination sample rate if the downsampling is performed) signal and then performs interpolation using a bank of short (14 to 24 taps, depending on the required precision) polynomial-interpolated sinc function-based fractional delay filters. This puts the algorithm into the league of the fastest among the most precise SRC algorithms. The more precise alternative being only the whole number-factored SRC, which can be slower.
C++ compiler and system with the "double" floating point type (53-bit mantissa) support. No explicit code for the "float" type is present in this library, because as practice has shown the "float"-based code performs considerably slower on a modern processor, at least in this library. This library does not have dependencies beside the standard C library, the "windows.h" on Windows and the "pthread.h" on macOS and Linux.
The sample rate converter (resampler) is represented by the r8b::CDSPResampler class, which is a single front-end class for the whole library. You do not basically need to use nor understand any other classes beside this class. Several derived classes that have varying levels of precision are also available.
The code of the library resides in the "r8b" C++ namespace, effectively isolating it from all other code. The code is thread-safe. A separate resampler object should be created for each audio channel or stream being processed concurrently.
Note that you will need to compile the "r8bbase.cpp" source file and include the resulting object file into your application build. This source file includes definitions of several global static objects used by the library. You may also need to include to your project: the "Kernel32" library (on Windows) and the "pthread" library on macOS and Linux.
The library is able to process signal of any scale and loudness: it is not limited to just a "usual" -1.0 to 1.0 range.
By defining the
R8B_IPP configuration macro it is possible to enable Intel
IPP front-end for FFT functions, instead of Ooura FFT. IPP FFT makes sample
rate conversion faster by 23% on average.
#define R8B_IPP 1
If a larger initial processing delay and a very minor sample timing error are
not an issue, for the most efficiency you can define these macros at
the beginning of the
r8bconf.h file, or during compilation:
#define R8B_IPP 1 #define R8B_FASTTIMING 1 #define R8B_EXTFFT 1
If you do not have access to Intel IPP then you may consider enabling PFFFT
which is only slightly slower than Intel IPP FFT in performance, however since
PFFFT works in single-precision resolution, the overall resampler's precision
will be limited to 24-bit sample rate conversions. To use the PFFFT, define
R8B_PFFFT macro, and compile and include the supplied
to your project build. Note that, subjectively speaking, the sound produced
with such lower-resolution FFT is not as good as with the higher-resolution
Ooura FFT or Intel IPP FFT.
#define R8B_PFFFT 1
The code of this library was commented in the Doxygen
style. To generate the documentation locally you may run the
doxygen ./other/r8bdoxy.txt command from the library's directory.
Preliminary tests show that the r8b::CDSPResampler24 resampler class achieves
61.2*n_cores Mflops (
83.3*n_cores for Intel IPP FFT) when
converting 1 channel of 24-bit audio from 44100 to 96000 sample rate
(2% transition band), on an Intel Core i7-7700K processor-based 64-bit
AVX2-enabled system without overclocking. This approximately translates to a
real-time resampling of
868*n_cores) audio streams, at 100%
CPU load. Time performance when converting to other sample rates may vary
greatly. When comparing performance of this resampler library to another
library make sure that the competing library is also tuned to produce a fully
linear-phase response, has similar stop-band characteristics, and similar
sample timing precision.
Dynamic Link Library
The functions of this SRC library are also accessible in simplified form via the DLL file on Windows, requiring a processor with SSE2 support (Win64 version includes AVX2 auto-dispatch code). Delphi Pascal interface unit file for the DLL file is available. DLL and C LIB files are distributed in the DLL folder at the project's homepage. On non-Windows systems it is preferrable to use the C++ library directly. Note that the DLL was compiled with the PFFFT enabled.
The resampler class of this library was designed as an asynchronous processor: it may produce any number of output samples, depending on the input sample data length and the resampling parameters. The resampler must be fed with the input sample data until enough output sample data was produced, with any excess output samples used before feeding the resampler with more input data. A "relief" factor here is that the resampler removes the initial processing latency automatically, and that after initial moments of processing the output becomes steady, with only minor output sample data length fluctuations.
So, while for offline resampling a "push" method can be used demonstrated in
example.cpp, for real-time resampling a "pull" method should be used
which calls resampling process until output buffer is filled.
When using the r8b::CDSPResampler class directly, you may select the transition band/steepness of the low-pass (reconstruction) filter, expressed as a percentage of the full spectral bandwidth of the input signal (or the output signal if the downsampling is performed), and the desired stop-band attenuation in decibel.
The transition band is specified as the normalized spectral space of the input signal (or the output signal if the downsampling is performed) between the low-pass filter's -3 dB point and the Nyquist frequency, and ranges from 0.5% to 45%. Stop-band attenuation can be specified in the range 49 to 218 decibel. Both transition band and stop-band attenuation affect resampler's overall speed performance and initial output delay. For your information, transition frequency range spans 170% of the specified transition band, which means that for 2% transition band, frequency response below 0.966*Nyquist is linear.
This SRC library also implements a much faster "power of 2" resampling (e.g. 2X, 4X, 8X, 16X, 3X, 3*2X, 3*4X, 3*8X, etc. upsampling and downsampling).
All code is fully "inline", without the need to compile many source files. The memory footprint is quite modest.
r8brain-free-src is bundled with the following code:
- FFT routines Copyright (c) 1996-2001 Takuya OOURA. Homepage
- PFFFT Copyright (c) 2013 Julien Pommier. Homepage
This library is used by:
- Combo Model V VSTi instrument
- Boogex Guitar Amp audio plugin
- Zynewave Podium
- Red Dead Redemption 2
Please drop me a note at firstname.lastname@example.org and I will include a link to your software product to the list of users. This list is important at maintaining confidence in this library among the interested parties.
- Added //$ markers for internal debugging purposes.
- Backed-off max transition band to 45 and MinAtten to 49.
- Implemented Wave64 and AIFF file input in the
r8bfreesrcbench tool. The tool is now compiled with the
R8B_EXTFFT 1macros to demonstrate the maximal achievable performance.
- Updated allowed ReqAtten range to 52-218, ReqTransBand 0.5-56. It is possible to specify filter parameters slightly beyond these values, but the resulting filter will be slightly out of specification as well.
- Optimized static filter banks allocation.
- A major overhaul of interpolation classes: now templated parameters are not used, all required parameters are calculated at runtime. Static filter bank object is not used, it is created when necessary, and then cached.
- Implemented one-third interpolation filters, however, this did not measurably increase resampler's speed.
- Used ippsMul_64f_I() in the CDSPRealFFT::multiplyBlockZ() function for a minor conversion speed increase in Intel IPP mode.
- Added memory alignment to allocated buffers which boosts performance by 1.5% when Intel IPP FFT is in use.
- Implemented PFFFT support.
- Improved resampling speed very slightly.
- Updated the
r8bfreesrcbenchmark tool to support RF64 WAV files.
- Added a more efficient half-band filters for >= 256 resampling ratios.
- Made minor fix to downsampling for some use cases of CDSPBlockConvolver, did not affect resampler.
- Converted CDSPHBUpsampler and CDSPHBDownsampler's inner functions to static functions, which boosted high-ratio resampling performance measurably.
- Minor fix to the latency consumption mechanism.
- Reoptimized fractional delay filter's windowing function.
- Implemented a new variant of the getInLenBeforeOutStart() function.
- Reimplemented oneshot() function to support
- Considerably improved downsampling performance at high resampling ratios.
- Implemented intermediate interpolation technique which boosted upsampling performance for most resampling ratios considerably.
- Removed the ConvCount constant - now resampler supports virtually any resampling ratios.
- Removed the UsePower2 parameter from the resampler constructor.
- Now resampler's process() function always returns pointer to the internal buffer, input buffer is returned only if no resampling happens.
- Resampler's getMaxOutLen() function can now be used to obtain the maximal output length that can be produced by the resampler in a single call.
- Added a more efficient "one-third" filters to half-band upsampler and downsampler.
- Optimized 2X half-band downsampler.
- Optimized power-of-2 upsampling.
- Optimized half-band downsampling filter.
- Implemented whole-number stepping resampling.
- Fixed initial sub-sample offseting on downsampling.
- Improved sample timing precision.
- Increased CDSPResampler :: ConvCountMax to 28 to support a lot wider resampling ratios.
- Removed getInLenBeforeOutStart() due to incorrect calculation.