• Randelementemethode im Zeitbereich


    Die Randelemente-Methode (BEM) wird oft zur Simulation von akustischen Abstrahl- und Reflektionsproblemen benutzt. Im Allgemeinen wird eine Formulierung im Frequenzbereich verwendet, wenn jedoch kurze Impulsantworten oder eine Kopplung mit nichtlinearem Strukturverhalten in Interesse sind, ist eine Formulierung im Zeitbereich zielführender.


    Die für die BEM notwendigen Randintegralgleichungen und Fundamentallösungen werden mittels inverser Fourier-Transformation der äquivalenten Formulierungen im Frequenzbereich ermittelt. Diese Gleichungen werden dann mittels Galerkin-Methode im Ortsbereich und Kollokation im Zeitbereich diskretisiert. Die MOT (Marching-On-in-Time) Methode wird verwendet um das durch die Diskretisierung erhaltene lineare Gleichungssystem zu lösen. Die bekannten Stabilitätsprobleme der MOT-Methode werden mittels einer Burton-Miller Formulierung im Ortsbereich und höhere Interpolationsordnung im Zeitbereich behandelt.

    Zusätzlich ist geplant, die Effizienz des Codes mittels eines modifizierten Plane-Wave-Time-Decomposition Algorithmus zur erhöhen.

  • Start des FWF-Projekts "Time-Frequency Implementation of HRTFs"

    The FWF project "Time-Frequency Implementation of HRTFs" has started.

    Principal Investigator: Damian Marelli

    Co-Applicants: Peter BalazsPiotr Majdak

  • Acoustic Holography


    Acoustic holography is a mathematical tool for the localization of sources in a coherent sound field.


    Using the information of the sound pressure in one plane, the whole three-dimensional sound field is reconstructed. The sound field must be coherent and the half-space in which the sources are situated must be known.


    Acoustic holography is used to calculate the sound field in planes parallel to the measured plane. Normally, a plane near the hull of the structure is chosen. Concentrations in the plane are assumed to be the noise source.

  • Acoustic Measurement Tool at Acoustics Research Institute (AMTatARI)


    The Acoustic Measurement Tool at the Acoustics Research Institute (AMTatARI) has been developed for the automatic measurement of system properties of electro-acoustic systems like loudspeakers and microphones. As a special function, this tool allows an automatic measurement of Head Related Transfer Functions (HRTF). 

    Measurement of the following features has been implemented so far:

    • total harmonic distortion (THD)
    • signal in noise and distortions (SINAD)
    • impulse response

    The impulse responses can be measured with the Maximum Length Sequences (MLS) or with exponential sweeps. Whereas, in case of the sweeps, the new multiple exponential sweep method (MESM) is available. This method is also used to measure HRTFs with AMTatARI.

  • Adaptive Audio-Visuelle Sprachsynthese von Dialekten (AVDS)


    The aim of this project is to conduct basic research on the audio-visual speech synthesis of Austrian dialects. The project extends our previous work on


    10 speakers (5 male and 5 female) will be recorded for each dialect. The recordings comprise spontaneous speech, read speech and naming tasks, eliciting substantial phonemic distinctions and phonotactics. Consequently, a detailed acoustic-phonetic and phonological analysis will be performed for each dialect. Based on the acoustic-phonetic and phonological data analysis, 600 phonetically balanced sentences will be created and recorded with 4 speakers (2 male, 2 female) for each dialect. In these recordings the acoustic and the visual signal, resulting from the same speech production process, will be recorded jointly to account for the multimodal nature of human speech. The recorded material will serve as a basis for the development, training, and testing of speech synthesizers at the Telecommunications Research Center.


    FWF (Wissenschaftsfonds): 2011-2013

    Project Manager: Michael Pucher, Telecommunications Research Center, Vienna

    Project Partner: Sylvia Moosmüller, Acoustics Research Institute, Austria Academy of Sciences, Vienna

  • ADesc Command in STx


    ADesc is a facility (technically, a class library) for storing numeric parameters with an unlimited number of independent and dependent axes and a large - and theoretically unlimited - amount of data. It has been developed as a part of the Noidesc project, whose large amounts of numeric data have been expected to stress the existing, purely XML-based APar class to-and-beyond its limits. In practice, ADesc has proven to be highly efficient with parameters consisting of hundreds of thousands of values, thereby fully meeting the demands of Noidesc. It is expected to meet the demands of challenging future projects as well.

    ADesc fits into the existing STx design by offering an alternative to the existing APar class. Just like APar, the new ADesc stores parameters in the existing STx XML database. There are two ways of storing the numeric data:

    1. In-place in the XML database: This is the conventional way. It keeps all the benefits of XML storage (readable and editable, simple export and import to/from other software) without impairing performance for small and medium-sized parameters.
    2. Binary storage: For large parameters, there is an optional feature for binary storage. With ADesc binary storage, the parameter itself is still part of the XML database, keeping the advantages of the XML organization fully intact. Only the numeric data values of the axes themselves are stored as an external binary file. The XML axis data contains only a reference to that file and the position within the file. This keeps the XML database small and allows for very fast random access to data values.

    The user must decide which kind of storage to use. For large parameters containing hundreds of thousands of numerical values, the performance gain of binary storage may be significant (up to a factor of three for loading and saving the data). At the same time, the saving of space in the XML database by about the same factor (or, more accurately, quotient) increases the speed of the general handling of the XML database.

    Aside from performance, the main design criteria for the ADesc class library were flexibility and ease of use. ADesc provides for automatic unit conversion with most regularly used and predefined domains and units. More unusual situations may be handled with user-defined converter classes. There is even room for completely user-defined axes, thereby enabling things such as dynamically supplied data (e.g. live spectrogram) or data calculated on-the-fly.

    As a result of the positive experiences with the ADesc class and its performance, plans are in place to fully replace the existing APar class over time.

    Object Model:

    Each parameter is modeled by an instance of the ADesc class or of one of its derivations. There are several such classes derived from ADesc, each one optimized for a number of common cases. At this time, the following ADesc classes exist:

    1. ADesc: ADesc is the most general parameter class. It handles parameters with an arbitrary number of independent and dependent axes. It is also prepared for handling even infinite axes and dynamic axes, like axes whose values are supplied or computed at run-time.
    2. ADescX: AdescX is a simpler, less general variation of the most general ADesc, supporting neither infinite nor dynamic axes. Its internal storage is organized such that it matches the current way STx handles large tables. In the long run, it is expected to optimize the STx table handling, thereby possibly rendering ADescX redundant.
    3. ADesc0: ADesc0 models the special case of parameters without any independent axes.
    4. ADesc1: ADesc1 optimizes handling of parameters with exactly one independent axis and an arbitrary number of dependent axes. Storage organization is much simpler, rendering ADesc1 by far the fastest kind of ADesc parameter.
    5. ADesc2: ADesc2 efficiently handles parameters with exactly two independent axes and an arbitrary number of dependent axes. Storage organization is simpler and hence faster than with the general classes. The dedicated ADesc2 class has been supplied, because most parameters encountered so far have proven to have two axes.

    The axes of a parameter are modeled by classes derived from AAxis. In general, each axis has a domain (e.g. time or frequency), a unit (e.g. ms or Hz) and, if applicable, a reference value, i.e. a constant value based upon the axis values that have been computed. At this time, the following kinds of axes exist:

    1. AOrderAxis: The AOrderAxis is the only axis without a domain and unit. Its only property is its cardinality.
    2. AIncrementAxis: The AIncrementAxis has a fixed start value, a fixed offset, and a cardinality. Each value of the axis equals the sum of its predecessor and the offset value.
    3. AEnumerationAxis: The AEnumerationAxis stores a finite number of arbitrary values.
    4. ASegmentIncrementAxis: The ASegmentIncrementAxis is an AIncrementAxis whose values are relative to the beginning of a given STx audio segment.
    5. ASegmentEnumerationAxis: The ASegmentEnumerationAxis is an AEnumerationAxis whose values are relative to the beginning of a given STx audio segment.#
    6. ADependentAxis: Each dependent axis of a parameter is modeled by an instance of an ADependentAxis. The number of dependent axes and their data are restricted by the choice of the respective ADesc class used.

    The hierarchy of the most important classes making up the ADesc library is the following:

    Programming Interface:

    The ADesc programming interface is as orthogonal a design as possible. The basic access functions are called getValue, setValue, getValues, setValues, getValueMatrix, setValueMatrix, getNativeValues, and setNativeValues. They are available both for the whole parameter and for its individual axes. Depending on which object they are called upon, they also set or retrieve one or more values of the desired axis or axes.

    If the parameter modeled by ADesc is considered to be an n-dimensional space (n being the number of independent axes), each point in this space is uniquely described by an n-tuple of coordinates which is the argument to the respective get and set function. The coordinates may be supplied either as an STx vector or as a textual list.

    If there is only one dependent axis, the value for each given coordinate is the value of this axis at the respective coordinate. If there is more than one dependent axis, the value for a given coordinate is a vector of length m, such that m is the number of dependent axes. By specifying the index or the name of a desired dependent axis, the user gets the value of this axis at the respective coordinates. By not specifying this information, the caller gets the whole vector of dependent values at the respective coordinates. This maximizes the flexibility for the ADesc user and requires awareness of fewer distinct functions.

    Other than functions for retrieving one or more parameter values for a specific coordinate, there are also functions for retrieving a larger number of data at the same time. For example, with two-dimensional parameters (i.e. parameters with exactly two independent axes), there are the functions getValueMatrix and setValueMatrix for efficiently setting all of the data of an independent axis. For all parameters with at least one independent axis, there are the functions getValueVector and setValueVector for accessing the whole of an axis.

  • Amadee: Frame Theory for Sound Processing and Acoustic Holophon

    S&T cooperation project 'Amadee' Austria-France 2013-14, "Frame Theory for Sound Processing and Acoustic Holophony", FR 16/2013

    Project Partner: The Institut de recherche et coordination acoustique/musique (IRCAM)

  • BanachFrameMul: Bessel and Frame Multipliers in Banach Spaces


    Another project has investigated the basic properties of frame and Bessel multipliers. This project aims to generalize this concept so that it will work with Banach spaces also.


    As the Gram matrix plays an important role in the investigation of multipliers, it is quite natural to look at the connection to localized frames and multipliers. The dependency of the operator class on the symbol class can be researched.

    The following statements will be investigated:

    • Theorem: If G is a localized frame and a is a bounded sequence, then the frame multiplier Ta is bounded on all associated Banach spaces (the associated co-orbit spaces).
    • Theorem: If G is a localized frame and a is a bounded sequence, such that the frame multiplier Ta is invertible on the Hilbert space H, then Ta is simultaneously invertible on the associated Banach spaces.

    The applications of these results to Gabor frames and Gabor multipliers will be further investigated.


    Although Banach spaces are more general a concept than Hilbert spaces, Banach theory has found applications. For example, if any norm other than L2 (least square error) is used for approximation, Banach theory tools have to be applied.


    • K. Gröchenig, NuHAG, Faculty of Mathematics, University of Vienna


    This project ended on 28.02.2008 and is incorporated into the 'High Potential'-Project of the WWTF, MULAC.

  • BE-SyMPHONic

    BE-SyMPHONic: French-Austrian joint project granted by ANR and FWF

    Principal investigators: Basilio Calderone, Wolfgang U. Dressler
    Co-applicants: Hélène Giraudo, Sylvia Moosmüller

    Start of the project: 13th January 2014


    Language sounds are realized in several different ways. Every language exploits no more than a sub-set of the sounds that the vocal tract can produce, as well as a reduced number of their possible combinations. The restrictions and the phonemic combinations allowed in the lanquage define a branch of phonology so-called phonotactics.

    Phonotactics refers to the sequential arrangement of phonemic segments in morphemes, syllables, and words and underlies a wide range of phonological issues, from acceptability judgements (pseudowords like <poiture>in French or <Traus>in German are phonotactically plausible) to syllable processes (the syllabic structure in a given language is based on the phonotactic permission in that language) and the nature and length of possible consonant clusters (that may be seen as intrinsically marked structures with respect to the basic CV template).


    Exploring the psycho-computational representation of the phonotactics in French and German is the aim of this research project.

    In particular, our researh will focus on the interplay between phonotactics and word structure in French and German, and investigate the behavioural and computational representations of phonotactic vs. morphonotactic clusters.

    As a matter of fact, the basic hypothesis underlying this research project ist that there exist different cognitive and computational representations for the same consonant cluster according to its phonotactic setting. In particular, the occurence of a cluster across a morpheme boundary (morphonotactic cluster) is considered as particularly interesting.


    Our research will focus on the interplay between phonotactis and morphology and investigate the behavioural and computational representations of consonant clusters according to whether they are: a) exclusively phonotactic clusters, i.e. the consonant cluster occurs only without morpheme boundaries (e.g.Steinin German); b) exclusively morphonotactic clusters, i.e. the consonant cluster occurs only beyond morpheme boundaries (e.g.lach+st), c) both are true with one of the two being more or less dominant (e.g. dominantlob+stvs.Obst)[1]. Thus we test the existence of different ‘cognitive and computational representations’ and processes for the same and for similar consonant clusters according to their appartenance to a) or b) or c).

    The central hypothesis which we test is that speakers not only reactively exploit the potential boundary signaling function of clusters that result from morphological operations, but take active measures to maintain or even enhance this functionality, for example by treating morphologically produced clusters differently than morpheme internal clusters in production or language acquisition. We call this hypothesis, the ‘Strong Morphonotactic Hypothesis’ (henceforth: SMH) (Dressler & Dziubalska-Koɫaczyk 2006, Dressler, Dziubalska-Koɫaczyk & Pestal 2010).

    In particular, we suppose that sequences of phonemes exhibiting morpheme boundaries (the ‘morphonotactic clusters’) should provide the speakers with functional evidence about the morphological operation occurred in that sequence; such evidence should be absent in the case of a sequence of phonemes without morpheme boundaries (the ‘phonotactic clusters’).

    Hence our idea is to investigate the psycho-computational mechanisms underlying the phonotactic-morphonotactic distinction by approaching the problem from two angles simultaneously: (a) psycholinguistic experimental study of language acquisition and production and (b) language computational modelling.

    We aim therefore at providing, on one hand, the psycholinguistic and behavioural support to the hypothesis that morphologically produced clusters are treated differently than morpheme internal clusters in French and German; on the other, we will focus on the distributional and statistical properties of the language in order to verify whether such difference in clusters’ treatment can be inductively modelled by appealing to distributional regularities of the language.

    The competences of the two research teams overlap and complement each other. The French team will lead in modelling, computational simulation and psycholinguistic experiments, the Austrian team in first language acquisition, phonetic production and microdiachronic change. These synergies are expected to enrich each group in innovative ways.

    [1] An equivalent example for French language is given by a)prise(/priz/ ‘grip’, exclusively phonotactic cluster), b)affiche+ rai(/afiʃʁɛ/ ‘I (will) post’, exclusively morphonotactic cluster) and c)navigue+ rai(/naviɡʁɛ/ ‘I (will) sail’) vs.engrais(/ãɡʁɛ/ ‘fertilizer’), the both conditions are true with morphonotactic condition as dominant.

  • BesselMult: Basic Properties of Bessel and Frame Multipliers


    The applications involving signal processing algorithms (like adaptive or time variant filters) are numerous. If the STFT, the Short Time Fourier Transformation, is used in its sampled version, the Gabor transform, the use of Gabor multipliers creates a possibility to construct a time-variant filter. The Gabor transform is used to calculate time frequency coefficients, which are multiplied with a fixed time-frequency mask. Then the result is synthesized. If another way of calculating these coefficients is chosen or if another synthesis is used, many modifications can still be implemented as multipliers. For example, it seems quite natural to define wavelet multipliers. Therefore, for this case, it is quite natural to continue generalizing and look at multipliers with frames lacking any further structure.


    Therefore, for Bessel sequences, the investigation of operators

    M = ∑ mk < f , ψk > φk

    where the analysis coefficients, < f , ψk >, are multiplied by a fixed symbol mk before resynthesis (with φk), is very natural and useful. These are the Bessel multipliers investigated in this project. The goal of this project is to set the mathematical basis to unify the approach to the Bessel multipliers for all possible analysis / synthesis sequences that form a Bessel sequence.


    Bessel sequences and frames are used in many applications. They have the big advantage of allowing the possibility to interpret the analysis coefficients. This makes the formulation of a multiplier concept for other analysis / synthesis systems very profitable. One such system involves gamma tone filter banks, which are mainly used for analysis based on the auditory system.


    • Balazs, P. (2007), "Basic Definition and Properties of Bessel Multipliers", Journal of Mathematical Analysis and Applications, 325, 1: 571--585. doi:10.1016/j.jmaa.2006.02.012, preprint


    This project ended on 01.01.2007. Its completion allowed the sucessfull application for a 'High Potential'-Project of the WWTF, see MULAC.

  • BestApprMult: Best Approximation of Matrices by Frame Multipliers


    Practical experience has shown that the concept of an orthonormal basis is not always useful. This led to the concept of frames. Models in physics and other application areas, including sound vibration analysis, are mostly continuous models. Many continuous model problems can be formulated as operator theory problems, as in differential or integral equations. An interesting class of operators is the Hilbert Schmidt class. This project aims to find the best approximation of any matrix by a frame multiplier, using the Hilbert Schmidt norm.


    In finite dimensions, every sequence is a frame sequence, so the best approximation of any element can be found only via the frame operator using the dual frame for synthesis. Furthermore, the present best approximation algorithm involves the following steps: 1) The Hilbert-Schmidt inner product of the matrix and the projection operators involved is calculated in an efficient way; 2) Then the pseudo inverse of the Grame matrix is used to avoid the so-called calculation of the dual frames; The pseudo inverse is applied to the coefficients found above to find the lower symbol of the multiplier.


    To find the best approximation of matrices via multipliers gives a way to find efficient algorithms to implement such operators. Any time-variant linear system can be modeled by a matrix. Time-invariant systems can be described as circulating matrices. Slowly-time-varying linear systems have a good chance at closely resembling Gabor multipliers. Other matrices can be well approximated by a "diagonalization" using other frames.


    • P. Balazs, "Hilbert-Schmidt Operators and Frames - Classification, Approximation by Multipliers and Algorithms" , International Journal of Wavelets, Multiresolution and Information Processing, (2007, accepted)  preprint, Codes and Pictures: here


    This project ended on 01.01.2009. Its completion allowed the sucessfull application for a 'High Potential'-Project of the WWTF, see MULAC

  • Binaural Hearing and the Cochlear Phase Response (BiPhase)

    BiPhase:  Binaural Hearing and the Cochlear Phase Response

    Project Description

    While it is often assumed that our auditory system is phase-deaf, there is a body of literature showing that listeners are very sensitive to phase differences between spectral components of a sound. Particularly, for spectral components falling into the same perceptual filter, the so-called auditory filter, a change in relative phase across components causes a change in the temporal pattern at the output of the filter. The phase response of the auditory filter is thus important for any auditory tasks that rely on within-channel temporal envelope information, most notably temporal pitch or interaural time differences.

    Within-channel phase sensitivity has been used to derive a psychophysical measure of the phase response of auditory filters (Kohlrausch and Sanders, 1995). The basic idea of the widely used masking paradigm is that a harmonic complex whose phase curvature roughly mirrors the phase response of the auditory filter spectrally centered on the complex causes a maximally modulated (peaked) internal representation and, thus, elicits minimal masking of a pure tone target at the same center frequency. Therefore, systematic variation of the phase curvature of the harmonic complex (the masker) allows to estimate the auditory filter’s phase response: the masker phase curvature causing minimal masking reflects the mirrored phase response of the auditory filter.

    Besides the obvious importance of detecting the target in the temporal dips of the masker, particularly of the target is short compared to the modulation period of the masker (Kohlrausch and Sanders, 1995), there are several indications that fast compression in the cochlea is important to obtain the masker-phase effect (e.g., Carlyon and Datta, 1997; Oxenham and Dau, 2004). One indication is that listeners with sensorineural hearing impairment (HI), characterized by reduced or absent cochlear compression due to loss of outer hair cells, show only a very weak masker-phase effect, making it difficult to estimate the cochlear phase response.

    In the BiPhase project we propose a new paradigm for measuring the cochlear phase response that does not rely on cochlear compression and thus should be applicable in HI listeners. It relies on the idea that the amount of modulation (peakedness) in the internal representation of a harmonic complex, as given by its phase curvature, determines the listener’s sensitivity to envelope interaural time difference (ITD) imposed on the stimulus. Assuming that listener’s sensitivity to envelope ITD does not rely on compression, systematic variation of the stimulus phase curvature should allow to estimate the cochlear phase response both in normal-hearing (NH) and HI listeners. The main goals of BiPhase are the following:

    • Aim 1: Assessment of the importance of cochlear compression for the masker-phase effect at different masker levels. Masking experiments are performed with NH listeners using Schroeder-phase harmonic complexes with and without a precursor stimulus, intended to reduce cochlear compression by activation of the efferent system controlling outer-hair cell activity. In addition, a quantitative model approach is used to estimate the contribution of compression from outer hair cell activity and other factors to the masker-phase effect. The results are described in Tabuchi, Laback, Necciari, and Majdak (2016). A follow-up study on the dependency of the masker-phase effect on masker and target duration, the target’s position within the masker, the masker level, and the masker bandwidth and conclusions on the role of compression of underlying mechanisms in simultaneous and forward masking is underway.
    • Aim 2: Development and evaluation of an envelope ITD-based paradigm to estimate the cochlear phase response. The experimental results on NH listeners, complemented with a modeling approach and predictions, are described in Tabuchi and Laback (2017). This paper also provides model predictions for HI listeners.
      Besides the consistency of the overall pattern of ITD thresholds across phase curvatures with data on the masking paradigm and predictions of the envelope ITD model, an unexpected peak in the ITD thresholds was found for a negative phase curvature which was not predicted by the ITD model and is not found in masking data. Furthermore, the pattern of results for individual listeners appeared to reveal more variability than the masking paradigm. Data were also collected with an alternative method, relying on the extent of laterality of a target with supra-threshold ITD, as measured with an interaural-level-difference-based pointing stimulus. These data showed no nonmonotonic behavior at negative phase curvatures. Rather, they showed good correspondence with the ITD model prediction and more consistent results across individuals compared to the ITD threshold-based method (Zenke, Laback, and Tabuchi, 2016).
    • Aim 3: Development of a ITD-based method to account for potentially non-uniform curvatures of the phase response in HI listeners. Using two independent iterative approaches, NH listeners adjusted the phase of individual harmonics of an ITD-carrying complex so that it elicited maximum extent of laterality. Although the pattern of adjusted phases very roughly resembled the expected pattern, there was a large amount of uncertainty (Zenke, 2014), preventing the method from further use. Modified versions of the method will be considered in a future study.


    This project is funded by the Austrian Science Fund (FWF, Project # P24183-N24, awarded to Bernhard Laback). It run from 2013 to 2017


    Peer-reviewed papers

    • Tabuchi, H. and Laback, B. (2017): Psychophysical and modeling approaches towards determining the cochlear phase response based on interaural time differences, The Journal of the Acoustical Society of America 141, 4314–4331.
    • Tabuchi, H., Laback, B., Necciari, T., and Majdak, P (2016). The role of compression in the simultaneous masker phase effect, The Journal of the Acoustical Society of America 140, 2680-2694.

    Conference talks

    • Tabuchi, H., Laback, B., Majdak, P., and Necciari, T. (2014). The role of precursor in tone detection with Schroeder-phase complex maskers. Poster presented at 37th Association for Research in Otolaryngology (ARO) Meeting, San Diego, California.
    • Tabuchi, H., Laback, B., Majdak, P., and Necciari, T. (2014). The perceptual consequences of a precursor on tone detection with Schroeder-phase harmonic maskers. Invited talk at Alps Adria Acoustics Association, Graz, Austria.
    • Tabuchi, H., Laback, B., Majdak, P., Necciari, T., and Zenke,K. (2015). Measuring the auditory phase response based on interaural time differences. Talk at 169th Meeting of the Acoustical Society of America, Pittsburgh, Pennsylvania.
    • Zenke, K., Laback, B., and Tabuchi, H. (2016). Towards an Efficient Method to Derive the Phase Response in Hearing-Impaired Listeners. Talk at 37th Association for Research in Otolaryngology (ARO) Meeting, San Diego, California.
    • Tabuchi, H., Laback, B., Majdak, P., Necciari, T., and Zenke, K. (2016). Modeling the cochlear phase response estimated in a binaural task. Talk at 39th Association for Research in Otolaryngology (ARO) Meeting, San Diego, California.
    • Laback, B., and Tabuchi, H. (2017). Psychophysical and modeling approaches towards determining the cochlear phase response based on interaural time differences. Invited Talk at AABBA Meeting, Vienna, Austria.
    • Laback, B., and Tabuchi, H. (2017). Psychophysical and Modeling Approaches towards determining the Cochlear Phase Response based on Interaural Time Differences. Invited Talk at 3rd Workshop “Cognitive neuroscience of auditory and cross-modal perception, Kosice, Slovakia.


    • Carlyon, R. P., and Datta, A. J. (1997). "Excitation produced by Schroeder-phase complexes: evidence for fast-acting compression in the auditory system," J Acoust Soc Am 101, 3636-3647.
    • Kohlrausch, A., and Sander, A. (1995). "Phase effects in masking related to dispersion in the inner ear. II. Masking period patterns of short targets," J Acoust Soc Am 97, 1817-1829.
    • Oxenham, A. J., and Dau, T. (2004). "Masker phase effects in normal-hearing and hearing-impaired listeners: evidence for peripheral compression at low signal frequencies," J Acoust Soc Am 116, 2248-2257.

    See also


  • Binaural Loudness Scaling with Cochlear Implant Listeners (LoudSca)


    The dependency of perceived loudness from electrical current in Cochlear Implant (CI) stimulation has been investigated in several existing studies. This investigation has two main goals:

    1. To study the efficiency of an adaptive method to determine the loudness function.
    2. To measure the loudness function in binaural as well as monaural stimulation.


    Loudness functions are measured at single electrodes (or interaural electrode pairs) using the method of categorical loudness scaling. The efficiency of this method for hearing impaired listeners has been demonstrated in previous studies (Brand and Hohmann, JASA 112, p.1597-1604). Both an adaptive method and the method of constant stimuli are used. Binaural functions are measured subsequently to monaural function, including monaural measurements as control conditions.


    The results indicate the suitability and efficiency of the adaptive categorical loudness scaling method as a tool for the fast determination of the loudness function. This can be applied to the clinical fitting of implant processors as well as for pre-measurements in psychoaoustic CI studies. The measurement results also provide new insights into monaural and binaural loudness perception of CI listeners.




    • Wippel, F., Majdak, P., and Laback, B. (2007). Monaural and binaural categorical loudness scaling in electric hearing, presented at Conference on Implantable Auditory Prostheses (CIAP), Lake Tahoe.
    • Wippel, F. (2007). Monaural and binaural loudness scaling with cochlea implant listeners, master thesis, Technical University Vienna, Autrian Academy of Sciences (in German)
  • BIOTOP: Adaptive Wavelet and Frame techniques for acoustic BEM. FWF Project I-1018-N25

    Biotop Beschreibung
    Workflow Biotop


    Die Lokalisierung von Schallquellen spielt eine wichtige Rolle im täglichem Leben. Die Form des menschlichen Kopfs, des Torsos und vor allem des Außenohrs (Pinna) bewirken einen Filtereffekt für einfallenden Schall und spielen daher eine wichtige Rolle bei der Ortung einer Schallquelle. Dieser Filtereffekt kann mittels der s.g. head related transfer functions (HRTFs, kopfbezogene Übertragungsfunktionen) beschrieben werden. Diese Filterfunktionen können mittels numerischer Methoden (zum Beispiel der Randelemente Methode, BEM) berechnet werden. In BIOTOP sollen diese Berechnungen durch Anwendung adaptiver Wavelet und Frame Methoden effizienter gemacht werden.


    Verglichen mit den herkömmlichen BEM Ansatzfunktionen haben Wavelets den Vorteil, besser an gegebene Schallverteilungen angepasst werden zu können. Als Verallgemeinerung von Wavelets sollen Frames dabei helfen, eine noch flexiblere Berechnungsmethode und damit eine noch bessere Anpassung an das gegebene Problem zu entwickeln. BIOTOP verbindet abstrakte mathematische Theorie mit numerischer und angewandter Mathematik. BIOTOP ist ein internationales DACH-Projekt (DFG-FWF-SFG) zwischen der Philipps-Universität Marburg (Stephan Dahlke), der Unicersität Basel (Helmut Harbrecht) und dem Institut für Schallforschung. Die gemeinsame Erfahrung dieser drei Forschungsgruppen soll helfen, neue numerische Strategien und Methoden zu entwickeln. Das Projekt wird vom FWF (Proj. Nummer: I-1018 N25) gefördert.


  • Boundary Element Method (BEM) Model of the Head


    In order to numerically calculate individual head-related transfer functions (HRTFs), a boundary element model (BEM) was developed. This model makes it possible to calculate the sound pressure at the head that is caused by different external sound sources with frequencies up to 20,000 Hz.


    In engineering, the traditional BEM is widely used for solving problems. However, the computational effort of the BEM grows quadratically with the number of unknowns. This is one reason why the traditional BEM cannot be used for large models, even on highly advanced computers. In order to calculate the sound pressure at the head at high frequencies, very fine meshes need to be used. These meshes result in large systems of equations. Nevertheless, to be able to use the BEM, the equations must be combined with the Fast Multipole Method (FMM). With the FMM, the resulting matrices can be kept smaller, thus allowing the numeric solving of the Helmholtz equation with feasible effort and almost no accuracy loss as compared to the traditional BEM.


    The geometry of the head (especially the form of the outer ear or pinna) acts as a kind of filter. This geometry is very important in localizing sound in the vertical direction and distinguishing between sounds coming from the front or the back. The BEM model can be used to numerically calculate these filter functions, which are dependent on the position and the frequency of the sound source.


    FWF (Austrian Science Fund): Project #P18401-B15


    • Kreuzer, W., Majdak, P., Chen, Z. (2009): Fast multipole boundary element method to calculate head-related transfer functions for a wide frequency range, in: J. Acoust .Soc. Am. 126, 1280-1290.
    • Kreuzer, W.  and Chen, Z. S. (2008). "A Fast Multipole Boundary Element Method for calculating HRTFs," AES preprint  7020, AES Convention, Vienna.
  • Calm Tracks & Routes


    Upon first investigation, the design of new outward-curved noise barriers has an improved noise-shielding effect if absorbing material is applied. Further investigation shall prove this ability. Numeric simulations and measurements are being processed.


    Advanced boundary element methods (BEM) in two dimensions will prove the noise-shielding ability of the sound barrier. Different curvy and straight designs are compared to each other with respect to their shielding effect in the spectrum. Measurements at existing walls are processed and compared. Measurements are conducted without a noise barrier. A simulated softening affect of the noise barrier walls is used to simulate the noise signal behind the new barriers.


    Calma Tec has patented the designs and will offer new designs in practice.

    List of Deliverables:

    01. Traffic Noise Recording Plan. 02. Sound Data Storage, Retrieval and Spectrographic Description. 03. Descriptive Noise Statistics. 04. Pricipal Component Analysis. 05. Sound Barrier Mesh Models. 06. Simulation, Transfer Functions & Clustering. 07. Visualization. 08. Psychoacoustic Irrelevance. 09 Modulation Effects. 10. Subjective Preference Tests. 11. Conclusions

  • Cochlear Implants: Stimulation Sequences for Perception of Interaural Time Differences (StimSequ)


    A recently developed stimulation strategy for cochlear implants attempts to encode temporal fine structure information, which is known to be important in perceiving pitch and interaural time differences (ITD). So-called "sequences" of pulses are triggered with each zero-crossing of the acoustic input waveform. It is expected that adaptation effects at the auditory nerve level limit the information flow. The goal of this project is to find optimum parameter values for this new stimulation strategy, which is intended to be applied in clinical applications.


    The effects of a parameter's pulse rate within each sequence, the number of sequences per second, and the temporal shape of the sequence on ITD perception are studied systematically.


    The optimum parameter values determined in the experiments are intended to be used in the clinical application of the new stimulation strategy.

  • Content Based Description of Train Noise (NOIDESc)


    The Austrian OeBB-HL-AG company performed tests with high-speed train ICE-S in 2004. A test rail section was adapted to the for a time period of a week. The train was driven with speed from 200 to over 300 km/h.

    We had the opportunity to record the noise emissions caused by the train. This was a great chance to test our equipment such as microphone array and outdoor microphone recording system.

  • Content Management System


    Redesign the Institute's homepage using a Content Management System (CMS) to facilitate easy actualization by all Institute employees, easy extension of the homepage functionality and a consistent style.


    The CMS 'Mambo' (today: Joomla) was chosen from the available open-source systems. The homepage was redesigned. The homepage content was transferred.


    If employees can easily update their content from any web browser, the homepage will be more up-to-date.

  • Control of the Tsetse Fly


    The tsetse fly genus Glossina is a carrier of the sleeping sickness and of the Nagana epidemic, which affect the ungulates. Over the past years, the sicknesses carried by the tsetse flies has spread so rapidly that intensified disease-fighting measures were necessary. One of the most effective methods is the exposure of a sterile male. The sterilized flies are raised on a large scale using radiation and then released. During the culture, a continuous control of the quality of the flies is necessary. The objective of this project is to develop an acoustic quality check for the sterile males. The research has demonstrated that the quality control is only possible using the sound activity of the flies.


    The tsetse fly uses its flying apparatus to produce sounds in addition to flying. Whereas the flying noise consists mainly of low frequent parts (<2000Hz) with only a few tonal parts, the "singing" consists of tonal components in the range of ca. 1-8kHz. For the detection of the singing, a high-pass-filtered spectrum of the interested frequency range is calculated (using DCT). From this spectrum, three parameters are extracted (energy in local peaks, 95 percent energy bandwidth, variance of the amplitudes), which are suitable for the determination of sounds with distinctive components. These single parameters are converted in weight values between 0 and 1 by using trigger functions. Afterwards, they are merged. The thresholds of the trigger functions are investigated in a separate measuring run from the background signal. The test version of this method was implemented in STx.


    The program will be tested on the testse flies at the laboratory in the 2006/2007 winter semester. As a result of initial tests, it will probably be enhanced. As of 2007, the program is planned to be put into practice in an African institute.