University of Music and Performing Arts, Graz
Graz, Austria
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Agency: European Commission | Branch: FP7 | Program: CP-CSA-Infra | Phase: INFRA-2012-1.1.24. | Award Amount: 23.40M | Year: 2013

Research accelerators are facing important challenges that must be addressed in the years to come: existing infrastructures are stretched to all performance frontiers, new world-class facilities on the ESFRI roadmap are starting or nearing completion, and strategic decisions are needed for future accelerators and major upgrades in Europe. While current projects concentrate on their specific objectives, EuCARD-2 brings a global view to accelerator research, coordinating a consortium of 40 accelerator laboratories, technology institutes, universities and industry to jointly address common challenges. By promoting complementary expertise, cross-disciplinary fertilisation and a wider sharing of knowledge and technologies throughout academia and with industry, EuCARD-2 significantly enhances multidisciplinary R&D for European accelerators. This new project will actively contribute to the development of a European Research Area in accelerator science by effectively implementing a distributed accelerator laboratory in Europe. Transnational access will be granted to state-of-the-art test facilities, and joint R&D effort will build upon and exceed that of the ongoing EuCARD project. Researchers will concentrate on a few well-focused themes with very ambitious deliverables: 20 T accelerator magnets, innovative materials for collimation of extreme beams, new high-gradient high-efficiency accelerating systems, and emerging acceleration technologies based on lasers and plasmas. EuCARD-2 will include six networks on strategic topics to reinforce synergies between communities active at all frontiers, extending the scope towards innovation and societal applications. The networks concentrate on extreme beam performance, novel accelerator concepts with outstanding potential, energy efficiency and accelerator applications in the fields of medicine, industry, environment and energy. One network will oversee the whole project to proactively catalyze links to industry and the innovation potential.

Guldenschuh M.,University of Music and Performing Arts, Graz
2013 3rd International Conference on Systems and Control, ICSC 2013 | Year: 2013

The Filtered-x-Least-Mean-Square (FxLMS) is an efficient algorithm for active-noise-control-headphones. It relies on a correct model Ŝ of the secondary-path S which, in the case of headphones, is above all determined by the acoustic path form the loudspeaker to the error-microphone. If the headphones are abruptly lifted or put on, the phase of S changes more than 90° and the formerly correct model Ŝ will suddenly be wrong and the FxLMS might diverge. This paper presents three methods how the divergence of the FxLMS can be avoided. All three methods rely on laboratory measurements under different conditions from tight headphones to completely lifted headphones. First, it is shown how a stable secondary-path model can be derived from the phase information of the measurements. For the second and third method, two secondary-path models are implemented. One for the tight use case and one for the lifted headphones. The current state of the secondary-path is then detected either via an online noise-cancelling-analysis or via an infrasonic test-signal. Comparison with existing approaches shows the robust stability and efficiency of the proposed methods. © 2013 IEEE.

Guldenschuh M.,University of Music and Performing Arts, Graz | Holdrich R.,University of Music and Performing Arts, Graz
IET Signal Processing | Year: 2013

Digital active noise control (ANC) for headphones usually has to predict the noise because of the latency of common audio converters. In adaptive feedback ANC, the prediction is based on the noise that entered the headphone. This noise is lowpass filtered because of the physical barrier of the ear cups. In this study, this low-pass characteristic is exploited to define a prediction filter which does not require real-time updates. For broadband noises, the prediction filter performs better than adaptive prediction methods like the least mean squares algorithm or iterated one-step predictions in the relevant frequency band. This is shown in simulations as well as in measurements. In addition, the authors show that their prediction filter is more robust against changes in the acoustics of the headphone. © The Institution of Engineering and Technology 2013.

Zotter F.,University of Music and Performing Arts, Graz | Pomberger H.,University of Music and Performing Arts, Graz | Noisternig M.,French National Center for Scientific Research
Acta Acustica united with Acustica | Year: 2012

Ambisonics with height is a three-dimensional sound field reproduction technique for spherical loudspeaker arrangements surrounding the reproduction area. It employs spherical harmonics up to a given order to expand incident sound fields with a limited angular resolution. The expansion coefficients describe the spatial sound scene. For reproduction, these coefficients are decoded to a set of surrounding loudspeakers. Common decoding approaches either sample the spherical harmonic excitation at the loudspeaker positions or match the excitation modes of a continuous sound field to those of the loudspeakers. For well-designed spherical loudspeaker arrays, both decoding approaches achieve good perceptual localization of virtual sound sources. However, both approaches perform unsatisfactorily with non-uniformly arranged arrays. Sounds from directions with only sparse loudspeaker coverage appear with altered loudness levels. This distracting effect results from variations in the decoded energy. The present article demonstrates an improved decoding technique, which preserves the decoded energy. Using available objective estimators, the localization qualities of these energy-preserving decoders are shown to lie between both common decoding approaches. © S. Hirzel Verlag · EAA.

Zotter F.,University of Music and Performing Arts, Graz | Frank M.,University of Music and Performing Arts, Graz
AES: Journal of the Audio Engineering Society | Year: 2012

All-Round Ambisonic Panning (AllRAP) is an algorithm for arbitrary loudspeaker arrangements, aiming at the creation of phantom sources of stable loudness and adjustable width. The equivalent All-Round Ambisonic Decoding (AllRAD) fits into the Ambisonic format concept. Conventional Ambisonic decoding is only simple with optimal loudspeaker arrangements for which it achieves direction-independent energy and energy spread, the estimated phantom source loudness and width. AllRAP/AllRAD is still simple but more versatile and utilizes the combination of a virtual optimal loudspeaker arrangement with Vector-Base Amplitude Panning.

Frank M.,University of Music and Performing Arts, Graz
Proceedings of Forum Acusticum | Year: 2014

Ambisonics is a recording and reproduction method that is based on the representation of the sound field excitation as a decomposition into spherical or circular harmonics, respectively. This achieves physically accurate sound field reproduction restricted within a sweet spot in the center of a loudspeaker array in an anechoic room. However, experiments show that a perceptually defined sweet spot is far less restrictive, even with a small number of loudspeakers in non-anechoic listening rooms. In this case, Ambisonics is rather understood as a simple amplitude-panning method based on the psychoacoustic phenomenon of a phantom source as it is known from stereophony. Taking the current opportunity, this contribution gathers recent experimental results, brings them together with the concept of quality, and hereby discusses the effect of quality elements (e.g. reproduction room, number and equalization of loudspeakers, order weighting, and decoder design) on perceived quality features (e.g. localization, source width, and coloration). The discussion reveals that a physically accurate reproduction does not necessarily yield good perceived quality. For this reason, the contribution puts optimal quality elements of Ambisonics in perspective that ensure optimal sound.

Frank M.,University of Music and Performing Arts, Graz
Archives of Acoustics | Year: 2013

Phantom sources are known to be perceived similar to real sound sources but with some differences. One of the differences is an increase of the perceived source width. This article discusses the perception, measurement, and modeling of source width for frontal phantom sources with different symmetrical arrangements of up to three active loudspeakers. The perceived source width is evaluated on the basis of a listening test. The test results are compared to technical measures that are applied in room acoustics: the inter-aural cross correlation coefficient (IACC) and the lateral energy fraction (LF). Adaptation of the latter measure makes it possible to predict the results by considering simultaneous sound incidence. Finally, a simple model is presented for the prediction of the perceived source width that does not require acoustic measurements as it is solely based on the loudspeaker directions and gains. Copyright © 2013 by PAN - IPPT.

Zotter F.,University of Music and Performing Arts, Graz | Frank M.,University of Music and Performing Arts, Graz
Archives of Acoustics | Year: 2013

We present a highly efficient filter structure to create power-complementary filter pairs for phantom source widening. It either introduces frequency-dependent phase or amplitude differences in a pair of loudspeaker signals. We evaluate how the perceptual effect is influenced by off-center listening positions in a standard ±30° loudspeaker setup. The evaluation of the phantom source widening effect is based on measurements of the inter-aural cross-correlation coefficient (IACC), which is justified by its pronounced correlation to the perceived phantom source width in prior listening test results. Copyright © 2013 by PAN - IPPT.

Goudarzi V.,University of Music and Performing Arts, Graz
IEEE Multimedia | Year: 2015

This article presents a user-centered design approach for creating an audio interface in the context of climate science. The author's team used contextual inquiry to gather information about scientists' workflows and focus groups to assess data about the scientists' specific use of language. The goal was to realize a domain-specific sonification platform and to identify climate metaphors to build a metaphoric sound identity for the sonification. In a separate set of experiments, participants were asked to pair sound stimuli with climate terms extracted from the initial interviews and to evaluate the sound samples aesthetically. They were also asked to choose sound textures (from a given set of sounds) that best express the specific climate parameter and to rate the relevance of the sound to the metaphor. The author's team assessed correlations between climate terminology and sound stimuli for the sonification tool to improve the sound design. Results show a tendency toward natural sounds by climate scientists. © 1994-2012 IEEE.

Guldenschuh M.,University of Music and Performing Arts, Graz
European Signal Processing Conference | Year: 2014

Adaptive filters in noise control applications have to approximate the primary path and compensate for the secondary-path. This work shows that the primary- and secondary-path variations of noise control headphones depend above all on the direction of incident noise and the tightness of the ear-cups. Both kind of variations are investigated by preliminary measurements, and it is further shown that the measured variations can be approximated with the linear combination of only a few prototype filters. Thus, a parallel adaptive linear combiner is suggested instead of the typical adaptive transversal-filter. Theoretical considerations and experimental results reveal that the parallel structure performs equally well, converges even faster, and requires fewer adaptation weights. © 2014 EURASIP.

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