Project

  • 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

    Introduction:

    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).

    Objective:

    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.

    Method:

    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.

  • Objective:

    The beam forming method focuses an arbitrary receiver coil using time delay and amplitude manipulation, and adds to the temporal signal of the microphones or the short time Fourier transform.

    Method:

    64 microphones are collected by a microphone array with arbitrary shape. For compatibility with acoustic holography, equal spacing and a grid with 8 x 8 microphones is used.

    Application:

    Localization of sound sources on high speed trains is a typical application. The method is used to separate locations along the train and especially the height of different sound sources. Typical sound sources on high speed trains are rail-wheel contact sites and aerodynamic areas. The aerodynamic conditions occur at all heights, especially at the pantograph.

  • Objective:

    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.

    Method:

    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.

    Application:

    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.

    Publications:

    • 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

    Project-completion:

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

  • Objective:

    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.

    Method:

    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.

    Application:

    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.

    Publications:

    • 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

    Project-completion:

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

  • 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.

    Funding

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

    Publications

    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.

    References

    • 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

    Potion

  • Objective:

    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.

    Method:

    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.

    Application:

    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.

    Funding:

    internal

    Publications:

    • 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 Beschreibung
    Workflow Biotop

    Introduction

    Localization of sound sources plays an important role in our everyday lives. The shape of the human head, the torso and especially the shape of the outer ear (pinna) have a filtering effect on incoming sounds and thus play an important role for sound localization. This filtering effect can be described using the so called head related transfer functions (HRTFs). By calculating the distribution of the sound pressure around the head with numerical methods like the boundary element method (BEM), these HRTFs can be calculated numerically.

    Aim

    In BIOTOP the numerical calculations shall be made more efficient by using adaptive wavelet- and frame techniques. Compared to commonly used BEM basis functions, wavelets have the advantage that wavelets can adapt better to a given distribution of the acoustic field on the head. As a generalization of wavelets, frames allow for an even more flexible construction method and thus for a better adaption to the problem at hand.

    BIOTOP combines abstract theoretical mathematics with numerical and applied mathematics. It is an international DACH (DFG-FWF-SFG) project between the Philipps-Universität Marburg (Stephan Dahlke), the University Basel (Helmut Harbrecht) and the ARI. The expertise of all three research groups shall be combined to develop new strategies and numerical methods. The project is funded by the FWF: Pr. Nr. I-1018-N25

     

  • Objective:

    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.

    Method:

    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.

    Application:

    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.

    Funding:

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

    Publications:

    • 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.
  • Objective:

    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.

    Method:

    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.

    Application:

    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

  • Objective:

    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.

    Method:

    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.

    Application:

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

  • HIGH SPEED TRAINS

    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.

  • Objective:

    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.

    Method:

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

    Application:

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

  • Objective:

    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.

    Method:

    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.

    Application:

    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.

  • Objective:

    In certain measurement setups, such as the measurement of gear mechanism behavior undergoing load reversal, the fine structure of the rotation speed function within a single rotation is interesting. In these situations, measurement errors caused by irregular cog intervals or by other failures of cogwheels are disturbing and must be corrected.

    Method:

    From a reference signal, the distribution parameter of the rotation angle for each cog of the cogwheel is assigned as a cogwheel model. This cogwheel model can minimize the measurement failures caused by the cogwheel if its cog is implemented in synch with the measurement signal. If the reference signal and the measurement signal come from different measurements, the synchronicity has to be established first. The calculation of the shift between the two signals is determined by the cog index, which has the maximum correlation of the rotation angle allocation between the reference signal and the measurement signal.

    Application:

    The developed method will be used as a module in the acoustic measurement and analysis system PAK.

  • Objective:

    This study explores the adaptation of localization mechanisms to warping of spectral localization features, as required for CI listeners to map those features to their reduced electric stimulation range.

    Methods and Results:

    The effect of warping the stimulation range from 2.8 to 16 kHz to the range from 2.8 to 8.5 kHz was studied in normal-hearing listeners. Fifteen subjects participated in a long-time localization-training study, involving two-hour daily audio-visual training over a period of three weeks. The Test Group listened to frequency-warped stimuli, the Control Group to low-pass filtered stimuli (8.5 kHz). The Control Group showed an initial increase of localization error and essentially reached the baseline performance at the end of the training period. The Test Group showed a strong initial increase of localization error, followed by a steady improvement of performance, even though not reaching the baseline performance at the end of the training period. These results are promising with respect to the idea to present high-frequency spectral localization cues to the stimulation range available with CIs

    Funding:

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

    Publications:

    • Walder, T. (2010) Schallquellenlokalisation mittels Frequenzbereich-Kompression der Außenohrübertragungsfunktionen (englisch: Sound source localization with frequency-range compressed head-related transfer functions), Master thesis, Technical University of Graz & Kunstuniversität Graz.
    • Majdak, P., Walder, T., and Laback, B. (2011). Learning to Localize Band-Limited Sounds in Vertical Planes, presented at: 34st MidWinter Meeting of the Association for Research in Otolaryngology (ARO). Baltimore, Maryland.
  • Objective:

    This project aims to develop an independent modulus for the wavelet analysis that contains a simple program interface and can be used flexibly.

    Method:

    The implementation was in C++ in the form of a wavelet analysis class and a signal queue. Features:

    • The Input/Output data format can be chosen at run time. The Input and the Output are separately configurable.
    • There are several possibilities for choosing the array and distribution of the frequency bin. The frequency bin vector can also be transferred.
    • Seven wavelets are implemented.
    • A down-sampling method can be used for the acceleration (factor: 1.2 convert frequency bins are chosen automatically).
    • Because of the disjunction in signal queue and analysis, an asynchrony Input/Output is possible.
    • Compiling an optimized numerical library can be achieved. Currently, the application of the "Intel® Signal Processing Library" (SPL) or of the "Intel® Integrated Performance Primitives" (IPP) is possible.
    • The signal queue class can be used independently of the analysis class. It also implements the down-sampling function.

    Application:

    The developed classes are used as a modulus in the acoustic measurement and analysis system PAK. The analysis class was also integrated as a signal processing atom WLLIB in STx.

  • Derzeit stellen SprecherInnen aus Deutschland die größte AusländerInnengruppe in Österreich und insbesondere in Wien dar. In diesem vom Kulturamt der Stadt Wien geförderten Projekt wird untersucht, ob und inwieweit aufgrund des Kontakts mit der deutschen Standardaussprache diese einen Einfluss auf die österreichische Standardaussprache ausübt und umgekehrt. Es werden akustische Aufnahmen von mehreren SprecherInnengruppen mit unterschiedlich intensivem Kontakt zu deutschen SprecherInnen durchgeführt

  • Projektleitung: Michael Pucher

    Beginn des Projekts: 1. Februar 2019

    Projektbeschreibung:

    Um den aktuellen Zustand einer Sprache zu erheben, soll bekanntlich der Sprachgebrauch eines alten, ländlichen, nicht mobilen Mannes analysiert werden. Für Entwicklungstendenzen einer Varietät sollte man jedoch die Sprache einer jungen und gebildeten Frau im urbanen Bereich untersuchen. Der Sprachgebrauch von jungen Frauen stellt ein besonders interessantes Forschungsfeld dar: Sie gelten als Initiatoren und Treibkräfte linguistischer Neuheiten einer Sprache, lautlich wie lexikal, die sich von Großstädten aus in den weiteren Sprachraum verbreiten können. Ebenso wird angenommen, dass aufgeschlossene junge Frauen linguistische Innovationen rascher übernehmen als ihre männlichen Peers. Sie verleiben sich eine neue Art zu sprechen schneller ein und geben diese an ihre späteren Kinder weiter. Frauen tendieren auch dazu, sprachliche Merkmale als social identifier zu verwenden, um sich der gleichen Peergroup zugehörig zu zeigen und können dadurch zu einem Sprachwandel beitragen.

    Die Stadt Wien hat sich in den vergangenen 30 Jahren stark verändert; so ist die Bevölkerung um 15% gestiegen und mit ihr auch die Anzahl der gesprochenen Sprachen. Laut einer Erhebung der Arbeiterkammer werden in Wien ca. 100 verschiedene Sprachen verwendet und man kann Wien nicht absprechen, weiterhin als ein Schmelztiegel verschiedenster Sprachen und Kulturen in Mitteleuropa zu gelten. Dass sich diese gesellschaftlichen bzw. gesellschaftspolitischen Veränderungen nicht nur im lexikalischen Sprachgebrauch der WienerInnen widerspiegeln, sondern ebenso in ihrer physiologischen Stimme zum Ausdruck kommen, soll hier den Ausgangspunkt der Studie darstellen.

    In dieser Untersuchung wird die Stimme als der physiologische und im Vokaltrakt modulierter Schall zur Lautäußerungen des Menschen gesehen. Die Stimme kann abgesehen davon auch als Ort des verkörperlichten Herz der gesprochenen Sprache gelten, die den Körper durch Indexikalität im sozialen Raum verankert. Als Vehikel der persönlichen Identität kann die Stimme nicht nur soziokulturelle, sondern auch gesellschaftspolitische Merkmale (bspw. „Frauen in Führungspositionen haben eine tiefere Stimme“) widerspiegeln. Hier übernimmt die Soziophonetik eine tragende Rolle, denn sie stellt ein wichtiges Instrument dar, das es ermöglicht, den sozialen Raum und seine gesellschaftsrelevanten Diskurse mit dem Individuum zu verknüpfen.

    Studien aus dem angloamerikanischen Raum wie legen nahe, dass sich die Stimme der jungen Frau in einem Wandel befindet. Das soziophonetische Stimmphänomen Vocal Fry hat sich inzwischen im angloamerikanischen Raum zum prominenten Sprachmerkmal junger, gebildeter und urbanen Frauen entwickelt.

    Basierend auf zwei Korpora soll eine Longitudinalstudie entstehen, die nachskizziert, inwiefern sich die Stimme der jungen Wienerin geändert hat. Soziophonetische Studien zu Frauenstimmen gibt es in Österreich nicht, vor allem in Hinsicht auf die angestrebte Qualität der Studie. Durch ihren longitudinalen Charakter kann sie aufzeigen, in wie weit das gesellschaftliche Geschehen Einfluss auf die Stimme der Frau ausübt.

    Darüber hinaus bietet diese Studie eine einmalige Gelegenheit, eine Momentaufnahme der Wienerin und ihrer Stimme zu erhalten und sie in einen historischen Kontext zu setzen.

     

    Informationen zur Teilnahme finden Sie hier!

  • Basic Description:

    This project line has the goal of finding efficient algorithms for signal processing applications. To apply the results of signal processing, Gabor or wavelet theory, the algorithms must be formulated for finite dimensional vectors. These discrete results are motivated by the continuous setting, but also often provide some insight. Furthermore, the efficient implementation of algorithms becomes important. For the consistency of these algorithms, it is useful to incorporate them into a maintained software package.

    Subprojects:

    • Double Preconditioning for Gabor Frames: This project develops a way to find an analysis-synthesis system with perfect reconstruction in a numerically efficient way using double preconditioning.
    • Perfect Reconstruction Overlap Add Method (PROLA): The classic overlap-add synthesis method is systematically compared to a new method motivated by frame theory.
    • Numerics of Block Matrices: In some applications in acoustics, it is apparent that block matrices are a powerful tool to find numerically efficient algorithms.
    • Practical Time Frequency Analysis: This project evaluates the usefulness of a time-frequency toolbox for acoustic applications and STx.

    Partners:

    • H.G. Feichtinger et al., NuHAG, Faculty of Mathematics, University of Vienna
    • B. Torrésani, Groupe de Traitement du Signal, Laboratoire d'Analyse Topologie et Probabilités, LATP/ CMI, Université de Provence, Marseille
    • P. Soendergaard, Department of Mathematics, Technical University of Denmark
    • J. Walker, Department of Mathematics, University of Wisconsin-Eau Claire

    Publications:

    • P. Balazs, H.G. Feichtinger, M. Hampejs, G. Kracher; "Double Preconditioning for Gabor Frames”; IEEE Transactions on Signal Processing, Vol. 54, No.12, December 2006 (2006), preprint
    • P. Balazs, H.G. Feichtinger, M. Hampejs, G. Kracher; "Double Preconditioning for the Gabor Frame Operator”; Proceedings ICASSP '06, May 14-19, Toulouse, DVD (2006)
  • Objective:

    The usual transformation in acoustics is the Fourier-Transformation. A fast and simple implementation is the windowed Fast Fourier Transformation. A disadvantage of the FFT is that all frequencies are equally spaced in the time frequency plane. A logarithmic spacing that allows keeps the relative resolution in the frequency plane constant is the Wavelet Transformation. This gives the possibility of a higher temporal resolution in the high frequency plane. Several types are implemented in STX and PAK.

    Method:

    A higher temporal resolution is possible, if quadratic transformations defined in the Cohen Class are used. The Windowed Pseudo Wigner Ville Distribution and a discrete version of the Choi-Williams Distribution are implemented in STX and PAK. Disadvantages of these transformations are the cross products that are reduced by smoothing in the different transformations of the Cohen class.

    Application:

    A handbook is written for or the practical use of the difficult transformations. The Handbook documents the possibilities and the limits of the transformations