Aeolus |
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The Aeolus page |
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Aeolus is a high quality pipe organ emulator using additive synthesis.
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Jconv |
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Jconv is a Convolution Engine for JACK, based on FFT convolution and using
non-uniform partition sizes: small ones at the start of the IR and building
up to the most efficient size further on. It can perform zero-delay processing
with moderate CPU load. Jconv uses the convolution engine designed for Aella,
a convolution application for reverberation processing (to be announced later).
This distributes the calculation over up to five threads, one for each
partition size, running at priorities just below the the one of JACK's
processing thread. This engine will become a separate library as soon as I
can find the time to write the user documentation.
Main features:
- Any matrix of convolutions between up to up 64 inputs and 64 outputs, as
long as your CPU(s) can handle it.
- Allows trading off CPU load to processing delay, and remains efficient
even when configured for zero delay.
- Sparse and diagonal matrices are handled as efficiently as dense ones.
No CPU cycles or memory resources are wasted on empty cells in the matrix,
nor on empty partitions if IRs are of different length.
- Accepts the same configuration files as Jace (see below).
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Zita-resampler |
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Documentation |
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Libzita-resampler is a C++ library for resampling audio signals. It is designed
to be used within a real-time processing context, to be fast, and to provide
high-quality sample rate conversion.
The library operates on signals represented in single-precision floating point
format. For multichannel operation both the input and output signals are
assumed to be stored as interleaved samples.
The API allows a trade-off between quality and CPU load. For the latter
a range of approximately 1:6 is available. Even at the highest quality
setting libzita-resampler will be faster than most similar libraries, e.g.
libsamplerate.
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TetraProc / TetraCal |
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Screenshots |
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To be released at the
2007 Linux Audio Conference.
The paper and pressentation can be found
here.
TetraProc converts the A-format signals from a tetrahedral Ambisonic
microphone into B-format signals ready for recording. Main features:
- A-B conversion using a classic scalar matrix and minimum phase filters, or
- A-B conversion using a 4 by 4 convolution matrix using measured
or computed impulse responses, or a combination of both.
- Individual microphone calibration facilities.
- 24 dB/oct higpass filters.
- Metering, monitoring and test facilities.
- Virtual stereo mic for stereo monitoring or recording.
- Unlimited number of stored configurations.
- Jack client with graphical user interface.
TetraCal is the microphone calibration application for TetraProc. Given a number
of impulse response measurements of a tetrahedral microphome (captured using
Aliki, see below) it computes the matching configuration file for TetraProc.
Main features:
- Automatic determination of gain and directivity errors of the A-format microphones.
- Automatic computation of an A-B matrix compensating for those errors.
- Frequency and phase response display of all A and B format signals.
- Interactive adjustment of pre- and post matrix equalisers.
Future versions of TetraCal will also exploit the convolution matrix
implemented in TetraProc.
TetraProc is available now, please contact me by e-mail. Release of TetraCal
awaits some required changes in Aliki, and completion of the manual that
explains the complete calibration procedure.
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Calibration service for Core Sound's TetraMic |
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As one result of a NDA signed with Core Sound, I can offer a free
calibration service for anyone using their
TetraMic on a Linux system. Given the serial number, I will provide a
TetraProc configuration based on the impulse response measurements performed
by Core Sound on your microphone. Currently this configuration does not
use the convolution capabilities of TetraProc, but the results are excellent
even without that. The combination of a very well designed microphone with
IR-based calibration provides an affordable Ambisonic recording system of
really suberb quality.
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Aliki |
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Manual (PDF) |
The Aliki page |
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Aliki is an integrated system for Impulse Response measurements,
using the logaritmic sweep method developed by Prof. Angelo Farina.
Release 0.0.3-beta is available on the downloads page. It's still
very incomplete but it has been used for real measurement work.
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AmbDec |
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Manual (PDF) |
Screenshots |
An Ambisonic decoder for first and second order. Release 0.0.1 is available
on the downloads page. Main features:
- 1st or 2nd order, 2-D or 3-D decoding.
- Up to 24 speakers (could be extended).
- Optional dual frequency band decoding.
- Optional speaker delay and gain compensation.
- Optional Near-Field effect compensation.
- Built-in test and Mute/Solo for each speaker.
- Graphical display of polar patterns, rV, rE (not yet in this release).
- Unlimited number of presets.
- Jack client with graphical user interface.
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Jaaa |
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Screenshots |
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Jaaa (JACK and ALSA Audio Analyser, is an audio signal generator and
spectrum analyser designed to make accurate measurements.
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Japa |
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Screenshots |
Japa (JACK and ALSA Perceptual Analyser), is a 'perceptual' or
'psychoacoustic' audio spectrum analyser.
In contrast to JAAA, this is more an acoustical or musical
tool than a purely technical one. Possible uses include
spectrum monitoring while mixing or mastering, evaluation
of ambient noise, and (using pink noise), equalisation
of PA systems.
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Yass |
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Screenshots |
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Yet Another Scrolling Scope. Main features: up to 32 channels, variable
scrolling speed, automatic gain control, and very light on CPU usage.
Beta release available.
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Jdelay |
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This is a small command line JACK app you can use to measure the latency of
your sound card. It uses a phase measurements on a set of tones to measure
the delay from the output to the input. Accuracy is about 1/1000 of a sample.
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HOA NF filters |
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A set of IIR filters combining forward and inverse Near Field correction
for Higher Order Ambisonics, up to 4th order. A 'textbook' implementation of
such filters will not work when using single precision floating point.
There is a simple solution for this explained in
this note which also comes with the sources.
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ACweight |
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High precision A- and C-weighting filter for sound measurements. IIR
implementation working at 44.1, 48, 88.2 and 96 kHz. Accuracy should
be in the order of 0.1 dB over most of the frequency range (errors occur
only close to half the sample frequency). The tarball contains: C++ class,
JACK in-process client and LADSPA plugin.
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Jnoise |
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A command line JACK app generating white and pink gaussian noise.
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LADSPA plugins |
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The LADSPA page |
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Various sets of plugins, both synth modules and audio processing.
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