Linux Audio projects at Kokkini Zita

Jconvolver is a command line JACK application using the zita-convolver library. Complex convolution matrices can be defined in a configuration file. The same configuration file can also be used by fconvolver which performs the same operation on files rather than real-time.

Jmatconvol is a command line JACK application similar to jconvolver, but optimised for dense convolution matrices with relatively short impulse responses, e.g. for processing microphone array signals. Fmatconvol provides the same for processing files rather than real-time signals.

Main features:

• Any dense matrix of convolutions of up to up 64 inputs to 64 outputs, as long as your CPU(s) can handle it.
• Zero-delay processing, the partition size is Jack's period size.
• The entire matrix can be defined by a single audio file.
• Processing can be multithreaded to reduce the processing time of the Jack client.

Libzita-convolver is a real-time C++ convolution library using FFT based convolution with non-uniform partition sizes: small ones at the start of the impulse response and building up to the most efficient size further on. It can perform zero-delay processing of very long IRs (e.g. reverbs) with moderate CPU load. The calculations are distributed over up to five threads, one for each partition size, running at priorities equal to or just below that of the calling thread. This ensures that even when multiple instances are active at the same time, CPU resources will be allocated in the optimum way.

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.

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.

The source distribution includes the resample application. Input format is any file readable by libsndfile, output is either WAV (WAVEX for more than 2 channels) or CAF. Apart from resampling you can change the sample format to 16-bit, 24-bit or float, and for 16-bit output, add dithering. Available dithering types are rectangular, triangular, and Lipschitz' optimised error feedback filter. Some examples of dithering can be seen here.

Libzita-alsa-pcmi is a C++ library for using ALSA audio devices with minimal latency and using ALSA's memory-mapped mode (as used by Jack). It hides most of the complexity of configuring and using ALSA devices in this way.

Input and output audio streams are converted to / from 32-bit floating point format, regardless of the hardware sample format. By calling the read and write functions from a real-time thread, it is easy to provide a callback interface, and to write applications that work with both ALSA and Jack using the same processing code. See jaaa or japa for examples.

First presented 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:

• Jack client with graphical user interface.
• 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.

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.

Main features:

• Jack client with graphical user interface.
• 1st, 2nd and 3rd order 2-D or 3-D decoding.
• Up to 36 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.
• Unlimited number of presets.

Avaiable filters:

• Flat - no filtering.
• 20 kHz. Lowpass with a noise bandwidth of 20 kHz.
• A and C weighting filters.
• ITU-R468.
• ITU-R468, Dolby variant.

Avaiable Detectors:

• RMS. Root-mean-square (i.e. 'power') meter. The time constant is 125 ms as per IEC standard, or 1 second in slow mode.
• Average. This measures the average of the absolute value. The one used in jnoisemeter has actually the exact ballistics of a VU meter. A 10 times slower version is also provided.
• ITU-R468. This is a 'pseudo-peak' detector designed specifically to measure noise and S/N ratios. For a peak meter it is quite slow, as it should be for noise measurements, but at the same time it is much more sensitive to short impulsive noise than its speed would suggest.