Papers and technotes

Audio measurements workshop
Fons Adriaensen
LAC2014 Slides (pdf, 177k)
Some things you need to know when doing audio measurements.
Design of an audio oscilloscope application
Fons Adriaensen
LAC2013 Paper (pdf, 203k)
This paper documents some aspects of the design of zita-scope, an Audio Oscilloscope application for the GNU/Linux system. It is designed to permit accurate display and measurements on audio waveforms captured from any source via the Jack audio server. Topics covered include performance requirements, an analysis of some problems that need to be considered, and an overview of the implemention structure. The software will be available at the time this paper is presented at the 2013 Linux Audio Conference in Graz.
Controlling Adaptive Resampling
Fons Adriaensen
LAC2012 Paper (pdf, 203k) Slides (pdf, 253k)
Combining audio components that use incoherent sample clocks requires adaptive resampling - the exact ratio of the sample frequencies is not known a priori and may also drift slowly over time. This situation arises when using two audio cards that don't have a common word clock, or when exchanging audio signals over a network. Controlling the resampling algorithm in software can be difficult as the available information (e.g. timestamps on blocks of audio samples) is usually inexact and very noisy. This paper analyses the problem and presents a possible solution.
Loudness measurement according to EBU R-128
Fons Adriaensen
LAC2011 Paper (pdf, 341k) Slides (pdf, 205k)
This paper introduces a new Linux application implementing the loudmess and level measuring algorithms proposed in recommendation R-128 of the European Broadcasting Union. The aim of this proposed standard is to ease the production of audio content having a defined subjective loudness and loudness range. The algorithms specifed by R-128 and related standard documents and the rationale for them are explained. In the final sections of the paper some implementation issues are discussed.
The WFS system at La Casa del Suono, Parma
Fons Adriaensen
LAC2010 Paper (pdf, 203k) Slides (pdf, 1.12M)
At the start of 2009 a 189-channel Wave Field Synthesis system was installed at the Casa del Suono in Parma, Italy. All audio and control processing required to run the system is performed by four Linux machines. The software used is a mix of standard Linux audio applications and some new ones developed specially for this installation. This paper discusses the main technical issues involved, including sections on the audio hardware, the digital signal processing algorithms, and the software used to control and manage the system.
Listening tests of the localization performance of Stereodipole and Ambisonic systems
Andrea Capra, Simone Fontana, Fons Adriaensen , Angelo Farina, Yves Grenier
AES 132nd Convention Paper (pdf, 985k)
In order to find a possible correlation of objective parameters and subjective descriptors of the acoustics of theatres, auditoria or music halls, and so to perform meaningful listening tests, we need to find a reliable 3D audio system which should give the correct perception of the distances, a good localization all around the listener and a natural sense of realism. For this purpose a Stereo Dipole system and an Ambisonic system were installed in a listening room at La Casa dell Musica (Param, Italy). Listening tests were carried out for evaluating the localization performances of the two systems.
A Digitally Controlled Two Dimensional Loudspeaker Array
P. Marignon, F. Adriaensen, D. Torelli, A. Farina
32nd AES International Conference Paper (pdf, 1M55)
A “sonic chandelier” has been designed and manufactured to be installed in the new “Casa del Suono” in Parma. This museum will be an exhibition of sound reproduction devices from the 20th century, and a multi-channel audio laboratory as well. The chandelier is a 64-channel, dome shaped array of 228 speakers suspended in the center of the main hall of the museum, at a height of 4 meters. It creates, by means of Wave Field Synthesis, virtual sources moving above the listener’s heads, and will reproduce an original spatial music composition. This listening experience plays the role of a very advanced music reproduction system, in contrast to the other, historic devices exhibited. An important requirement is that the sound produced must be confined to a restricted area beneath the system itself, in order not to interfere with other exhibits present in the hall. The physical structure and algorithm design are described, as well as some listening tests, performed for now on a reduced linear version of the array itself.
Three-Dimensional Acoustic Displays in a Museum emplying WFS and HOA.
Angelo Farina, Andrea Capra, Paolo Martignon, Simone Fontana, Fons Adriaensen, Paolo Galaverna and Dave Malham
ICSV14, Cairns, Australia Paper (pdf, 599k)
The paper describes the sound systems and the listening rooms installed in the new "museum of reproduced sound", actually being built in Parma, restoring an ancient church. The museum is devoted to the exposition of a huge collection of antique radios and gramophones, but it will also exploit the frontiers of modern methods for immersive surround reproduction: WFS and HOA.At the end of the exposition path, a special HOA space is employed for showing the recent developments of recording/reproduction methods started from the Ambisonics concept, capable of creating natural reproduction of sport events, live music and other immersive acoustical experiences; in this room also a binaural/transaural system is available. A second, larger listening room capable of 30seats is equipped with a horizontal WFS array covering the complete perimeter of the room. The paper describes the technology employed, the problems encountered due to the difficult acoustical conditions (the museum was formerly a church), and the novel software tools developed for the purpose on LINUX platforms.
A Tetrahedral Microphone Processor for Ambisonic Recording
LAC2007 Paper (pdf, 129k) Slides (pdf, 418k)
This paper introduces a Linux audio application that provides an integrated solution for making full 3-D Ambisonics recordings by using a tetrahedral microphone. Apart from the basic A to B format conversion it performs a number of auxiliary functions such as LF filtering, metering and monitoring, turning it into a complete Ambisonics recording processor. It also allows for calibration of an individual microphone unit based on measured impulse responses. A new JACK backend required to make use of a particular four-channel audio interface optimised for Ambisonic recording is also introduced.
Near Field filters for Higher Order Ambisonics
(2006) Technote (pdf, 112k)
A digital implementation of the near-field filters used in Higher Order Ambisonics may introduce some problems with numerical precision and stability. This is due to the fact that these filters mainly operate at the very low end of the audio range. This technical note documents a simple practical solution to this problem, and introduces a set of C++ classes implementing the filters up to fourth order.
Acoustical Impulse Response Measurement with ALIKI
LAC2006 Paper (pdf, 125k) Slides (pdf, 295k)
The Impulse Response of an acoustical space can be used for emulation of that space using a convolution reverb, for room correction, or to obtain a number of measures representative of the room's acoustical qualities. Provided the user has access to the required transducers, an IR measurement can be performed using a standard PC equipped with a good quality audio interface. This paper introduces a Linux application designed for this task. The theoretical background of the method used is discussed, along with a short introduction to the estimated measures. A short presentation of the program's features is also included.
Design of a Convolution Engine optimised for Reverb
LAC 2006 Paper (pdf, 104k) Slides (pdf, 117k)
Real-time convolution has become a practical tool for general audio processing and music production. This is reflected by the availability to the Linux audio user of several high quality convolution engines. But none of these programs is really designed to be used easily as a reverberation processor. This paper introduces a Linux application using fast convolution that was designed and optimised for this task. Some of the most relevant design and implementation issues are discussed.
Using a DLL to Filter Time
LAC 2005 (cancelled) Paper (pdf, 149k)
A new mechanism to obtain an accurate mapping between samples and system time was recently introduced into JACK .It is based on a the use of a Delay Locked Loop (DLL). This paper discusses the problem that was solved, and introduces the reader to the concept of control loops in general, and to the DLL solution adopted in particular.
Aeolus - A Church Organ in your PC
LAC 2004 Slides (pdf, 319k)
Audio Measurements using Jaaa
LAC 2004 Slides (pdf, 103k)