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Audio measurements workshop
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Fons Adriaensen
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| LAC2014 |
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Slides (pdf, 177k) |
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Some things you need to know when doing audio measurements.
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Digital State-Variable Filters
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| (2020) |
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Technote (pdf, 190k) |
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Design equations for digital state-variable filters.
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Design of an audio oscilloscope application
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Fons Adriaensen
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| LAC2013 |
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Paper (pdf, 203k) |
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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. |
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Controlling Adaptive Resampling
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Fons Adriaensen
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| LAC2012 |
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Paper (pdf, 203k) |
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Slides (pdf, 253k) |
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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.
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Loudness measurement according to EBU R-128
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Fons Adriaensen
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| LAC2011 |
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Paper (pdf, 341k) |
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Slides (pdf, 205k) |
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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.
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The WFS system at La Casa del Suono, Parma
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Fons Adriaensen
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| LAC2010 |
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Paper (pdf, 203k) |
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Slides (pdf, 1.12M) |
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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.
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Listening tests of the localization performance of Stereodipole and Ambisonic systems
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Andrea Capra, Simone Fontana, Fons Adriaensen , Angelo Farina, Yves Grenier
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| AES 132nd Convention |
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Paper (pdf, 985k) |
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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.
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A Digitally Controlled Two Dimensional Loudspeaker Array
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P. Marignon, F. Adriaensen, D. Torelli, A. Farina
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| 32nd AES International Conference |
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Paper (pdf, 1M55) |
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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.
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Three-Dimensional Acoustic Displays in a Museum emplying WFS and HOA.
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Angelo Farina, Andrea Capra, Paolo Martignon, Simone Fontana, Fons Adriaensen,
Paolo Galaverna and Dave Malham
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| ICSV14, Cairns, Australia |
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Paper (pdf, 599k) |
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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.
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A Tetrahedral Microphone Processor for Ambisonic Recording
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| LAC2007 |
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Paper (pdf, 129k) |
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Slides (pdf, 418k) |
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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.
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Near Field filters for Higher Order Ambisonics
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| (2006) |
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Technote (pdf, 112k) |
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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.
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Acoustical Impulse Response Measurement with ALIKI
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| LAC2006 |
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Paper (pdf, 125k) |
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Slides (pdf, 295k) |
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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.
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Design of a Convolution Engine optimised for Reverb
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| LAC 2006 |
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Paper (pdf, 104k) |
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Slides (pdf, 117k) |
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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.
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Using a DLL to Filter Time
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| LAC 2005 (cancelled) |
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Paper (pdf, 149k) |
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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.
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Aeolus - A Church Organ in your PC
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| LAC 2004 |
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Slides (pdf, 319k) |
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Audio Measurements using Jaaa
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| LAC 2004 |
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Slides (pdf, 103k) |
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