Completed

  • Objective:

    Head-related transfer functions (HRTF) describe the sound transmission from the free field to a place in the ear canal in terms of linear time-invariant systems. Due to the physiological differences of the listeners' outer ears, the measurement of each subject's individual HRTFs is crucial for sound localization in virtual environments (virtual reality).

    Measurement of an HRTF can be considered a system identification of the weakly non-linear electro-acoustic chain from the sound source room's HRTF microphone. An optimized formulation of the system identification with exponential sweeps, called the "multiple exponential sweep method" (MESM), was used for the measurement of transfer functions. For this measurement of transfer functions, either the measurement duration or the signal-to-noise ratio could be optimized.

    Initial heuristic experiments have shown that using Gabor multipliers to extract the relevant sweeps in the MESM post-processing procedure improves the signal-to-noise ratio of the measured data even further. The objective of this project is to study, in detail, how frame multipliers can optimally be used during this post-processing procedure. In particular, wavelet frames, which best fit the structure of an exponential sweep, will be studied.

    Method:

    Systematic numeric experiments will be conducted with simulated slowly time-variant, weakly non-linear systems. As the parameters of the involved signals are precisely known and controlled, an optimal symbol will automatically be created. Finally, the efficiency of the new method will be tested on a "real world" system, which was developed and installed in the semi-anechoic room of the Institute. It uses in-ear microphones, a subject turntable, 22 loudspeakers on a vertical arc, and a head tracker.

    Application:

    The new method will be used for improved HRTF measurement.

  • Beschreibung:

    Es wird ein Formantsythesizer, basierend auf dem Klatt Synthesizer, implementiert, der sowohl zur Erzeugung stationärer Vokale und auch zeitvarianter Formant- und Grundfrequenzspuren verwendet werden kann. Die Implementierung erfolgt als SP-Atom.

    Anwendung

    Die Synthese wird als Kontrollwerkzeug in die Anwendungen Viewer2 (Spektorgramm und Parameter Plot) und SPEXL (Segmentations-Tool) eingebunden. Dazu wird eine graphische Steuerung implementiert, die geeignete Funktionen zur Eingabe von Formantdaten (Vokalsynthese) und zur graphischen Auswahl von Parametersätzen (Resynthese von Parameterverläufen) zur Verfügung stellt.

  • Objective:

    So-called Gabor multipliers are particular cases of time-variant filters. Recently, Gabor systems on irregular grids have become a popular research topic. This project deals with Gabor multipliers, as a specialization of frame multipliers on irregular grids.

    Method:

    The initial stage of this project aims to investigate the continuous dependence of an irregular Gabor multiplier on its parameter (i.e. the symbol), window, and lattice. Furthermore, an algorithm to find the best approximation of any matrix (i.e. any time-variant system) by such an irregular Gabor multiplier is being developed.

    Application:

    Gabor multipliers have been used implicitly for quite some time. Investigating the properties of these operators is a current topic for signal processing engineers. If the standard time-frequency grid is not useful to the application, it is natural to work with irregular grids. An example of this is the usage of non-linear frequency scales, like bark scales.

    Partners:

    H. G. Feichtinger, NuHAG, Faculty of Mathematics, University of Vienna

    Project-completion:

    This project ended on 28.02.2008 and is incorporated into the 'High Potential'-Project of the WWTF, MULAC (WWTF 2007).

  • Objective:

    An irrelevance algorithm based on simultaneous masking is implemented In STx. In the years following its first development by Eckel, the efficiency of this algorithm has been clearly shown. In this project, this irrelevance model will be based on modern mathematic and psychoacoustic theories and knowledge.

    Method:

    This algorithm can be described as a Gabor multiplier with an adaptive symbol. With existing related theory, it becomes clear that a high redundancy must be selected. This guarantees:

    • perfect reconstruction synthesis
    • an under-spread operator for good time-frequency localization
    • a smoothing-out of easily detectable quick on/off cycles

    Furthermore, it can be shown that the model used for the spreading function here is mathematically equivalent to the excitation pattern.

    Application:

    This algorithm has been used for several years already for things such as:

    • automobile sound design
    • over-masking for background-foreground separation
    • improved speech recognition in noise
    • contrast increase for hearing-impaired persons

    Partners:

    • G. Eckel, Institut für Elektronische Musik und Akustik, Graz

    Publications:

    • P. Balazs, B. Laback, G. Eckel, W. Deutsch, "Introducing Time-Frequency Sparsity by Removing Perceptually Irrelevant Components Using a Simple Model of Simultaneous Masking", IEEE Transactions on Audio, Speech and Language Processing, Vol. 17 (7) , in press (2009) , preprint

    Project-completion:

    This project ended on 01.01.2010, and leads to a sub-project of the 'High Potential'-Project of the WWTF, MULAC.

  • ITD MultEl: Binaural-Timing Sensitivity in Multi-Electrode Stimulation

    Binaural hearing is extremely important in everyday life, most notably for sound localization and for understanding speech embedded in competing sound sources (e.g., other speech sources). While bilateral implantation has been shown to provide cochlear implant (CIs) listeners with some basic left/right localization ability, the performance with current CI systems is clearly reduced compared to normal hearing. Moreover, the binaural advantage in speech understanding in noise has been shown to be mediated mainly by the better-ear effect, while there is only very little binaural unmasking.

    There exists now a body of literature on binaural sensitivity of CI listeners stimulated at a single interaural electrode pair. However, the CI listener’s sensitivity to binaural cues under more realistic conditions, i.e., with stimulation at multiple electrodes, has not been systematically addressed in depth so far.

    This project attempts to fill this gap. In particular, given the high perceptual importance of ITDs, this project focuses on the systematic investigation of the sensitivity to ITD under various conditions of multi-electrode stimulation, including interference from neighboring channels, integration of ITD information across channels, and the perceptually tolerable room for degradations of binaural timing information.

    Involved people:

    Start: January 2013

    Duration: 3 years

    Funding: MED-EL

  • Objective and Method:

    Current cochlear implant (CI) systems are not designed for sound localization in the sagittal planes (front-back and up/down-dimensions). Nevertheless, some of the spectral cues that are important for sagittal plane localization in normal hearing (NH) listeners might be audible for CI listeners. Here, we studied 3-D localization with bilateral CI-listeners using "clinical" CI systems and with NH listeners. Noise sources were filtered with subject-specific head-related transfer functions, and a virtually structured environment was presented via a head-mounted display to provide feedback for learning. 

    Results:

    The CI listeners performed generally worse than NH listeners, both in the horizontal and vertical dimensions. The localization error decreases with an increase in the duration of training. The front/back confusion rate of trained CI listeners was comparable to that of untrained (naive) NH listeners and two times higher than for the trained NH listeners. 

    Application:

    The results indicate that some spectral localization cues are available to bilateral CI listeners, even though the localization performance is much worse than for NH listeners. These results clearly show the need for new strategies to encode spectral localization cues for CI listeners, and thus improve sagittal plane localization. Front-back discrimination is particularly important in traffic situations.

    Funding:

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

    Publications:

    • Majdak, P., Goupell, M., and Laback, B. (2011). Two-Dimensional Localization of Virtual Sound Sources in Cochlear-Implant Listeners, Ear & Hearing.
    • Majdak, P., Laback, B., and Goupell, M. (2008). 3D-localization of virtual sound sources in normal-hearing and cochlear-implant listeners, presented at Acoustics '08  (ASA-EAA joint) conference, Paris
  • Objective:

    Humans' ability to localize sound sources in a 3-D space was tested.

    Method:

    The subjects listened to noises filtered with subject-specific head-related transfer functions (HRTFs). In the first experiment with new subjects, the conditions included a type of visual environment (darkness or structured virtual world) presented via head mounted display (HMD) and pointing method (head and finger/shooter pointing).

    Results:

    The results show that the errors in the horizontal dimension were smaller when head pointing was used. Finger/shooter pointing showed smaller errors in the vertical dimension. Generally, the different effects of the two pointing methods was significant but small. The presence of a structured, virtual visual environment significantly improved the localization accuracy in all conditions. This supports the idea that using a visual virtual environment in acoustic tasks, like sound localization, is beneficial. In Experiment II, the subjects were trained before performing acoustic tasks for data collection. The performance improved for all subjects over time, which indicates that training is necessary to obtain stable results in localization experiments.

    Funding:

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

    Publications:

    • Majdak, P., Goupell, M., and Laback, B. (2010). 3-D localization of virtual sound sources: effects of visual environment, pointing method, and training, Attention, Perception, & Psychophysics 72, 454-469.
    • Majdak, P., Laback, B., Goupell, M., and Mihocic M. (2008). "The Accuracy of Localizing Virtual Sound Sources: Effects of Pointing Method and Visual Environment", presented at AES convention, Amsterdam.
  • Localization of sound sources is an important task of the human auditory system and much research effort has been put into the development of audio devices for virtual acoustics, i.e. the reproduction of spatial sounds via headphones. Even though the process of sound localization is not completely understood yet, it is possible to simulate spatial sounds via headphones by using head-related transfer functions (HRTFs). HRTFs describe the filtering of the incoming sound due to head, torso and particularly the pinna and thus they strongly depend on the particular details in the listener's geometry. In general, for realistic spatial-sound reproduction via headphones, the individual HRTFs must be measured. As of 2012, the available HRTF acquisition methods were acoustic measurements: a technically-complex process, involving placing microphones into the listener's ears, and lasting for tens of minutes.

    In LocaPhoto, we were working on an easily accessible method to acquire and evaluate listener-specific HRTFs. The idea was to numerically calculate HRTFs based on a geometrical representation of the listener (3-D mesh) obtained from 2-D photos by means of photogrammetric reconstruction.

    As a result, we have developed a software package for numerical HRTF calculations, a method for geometry acquisition, and models able to evaluate HRTFs in terms of broadband ITDs and sagittal-plane sound localization performance.

     

    Further information:

    http://www.kfs.oeaw.ac.at/LocaPhoto

     

  • Objective:

    It is known in psychoacoustics that not all information contained in a "real world" acoustic signal is processed by the human auditory system. More precisely, it turns out that some time-frequency components mask (overshadow) other components that are close in time or frequency.

    In the software S_TOOLS-STx developed by the Institute, an algorithm based on simultaneous masking has been implemented. This algorithm removes perceptually irrelevant time-frequency components. In this implementation, the model is described as a Gabor multiplier with an adaptive symbol.

    In this project, the masking model will be extended to a true time-frequency model, incorporating frequency and temporal masking.

    Method:

    Experiments have been conducted (in cooperation with the Laboratory for Mechanics and Acoustics / CNRS Marseille) to test the time-frequency masking properties of a single Gaussian atom, and to study the additivity of these masking properties for several Gaussian atoms.

    The results of these experiments will be used, in combination with theoretical results obtained in the parallel projects studying the mathematical properties of frame multipliers, to approximate or identify the masking model by wavelet and Gabor multipliers.

    The obtained model will then be validated by appropriate psychoacoustical experiments.

    Application:

    Efficient implementation of a masking filter offers many applications:

    • Sound / Data Compression
    • Sound Design
    • Back-and-Foreground Separation
    • Optimization of Speech and Music Perception

    After completing the testing phase, the algorithms are to be implemented in S_TOOLS-STx. 

    Publications:

    • P. Balazs, B. Laback, G. Eckel, W. Deutsch, "Introducing Time-Frequency Sparsity by Removing Perceptually Irrelevant Components Using a Simple Model of Simultaneous Masking", IEEE Transactions on Audio, Speech and Language Processing (2009), in press
    • B. Laback, P. Balazs, G. Toupin, T. Necciari, S. Savel, S. Meunier, S. Ystad and R. Kronland-Martinet, "Additivity of auditory masking using Gaussian-shaped tones", Acoustics'08, Paris, 29.06.-04.07.2008 (03.07.2008)
    • B. Laback, P. Balazs, T. Necciari, S. Savel, S. Ystad, S. Meunier and R. Kronland-Martinet, "Additivity of auditory masking for Gaussian-shaped tone pulses", preprint
  • Objective:

    This project is part of a project cluster that investigates time-frequency masking in the auditory system, in cooperation with the Laboratory for Mechanics and Acoustics / CNRS Marseille. While other subprojects study the spread of masking across the time-frequency plane using Gaussian-shaped tones, this subproject investigates how multiple Gaussian maskers distributed across the time-frequency plane create masking that adds up at a given time-frequency point. This question is important in determining the total masking effect resulting from the multiple time-frequency components (that can be modeled as Gaussian Atoms) of a real-life signal.

    Method:

    Both the maskers and the target are Gaussian-shaped tones with a frequency of 4 kHz. A two-stage approach is applied to measure the additivity of auditory masking. In the first stage, the levels of the maskers are adjusted to cause the same amount of masking in the target. In the second stage, various combinations of those maskers are tested to study their additivity.

    In the first study, the maskers are spread either in time OR in frequency. In the second study, the maskers are spread in time AND in frequency.

    Application:

    New insight into the coding of sound in the auditory system could help to design more efficient audio codecs. These codecs could take the additivity of time-frequency masking into account.

    Funding:

    WTZ (project AMADEUS)

    Publications:

    • Laback, B., Balazs, P., Toupin, G., Necciari, T., Savel, S., Meunier, S., Ystad, S., Kronland-Martinet, R. (2008). Additivity of auditory masking using Gaussian-shaped tones, presented at Acoustics? 08 conference, Paris.
  • Objective:

    Many problems in physics can be formulated as operator theory problems, such as in differential or integral equations. To function numerically, the operators must be discretized. One way to achieve discretization is to find (possibly infinite) matrices describing these operators using ONBs. In this project, we will use frames to investigate a way to describe an operator as a matrix.

    Method:

    The standard matrix description of operators O using an ONB (e_k) involves constructing a matrix M with the entries M_{j,k} = < O e_k, e_j>. In past publications, a concept that described operator R in a very similar way has been presented. However, this description of R used a frame and its canonical dual. Currently, a similar representation is being used for the description of operators using Gabor frames. In this project, we are going to develop and completely generalize this idea for Bessel sequences, frames, and Riesz sequences. We will also look at the dual function that assigns an operator to a matrix.

    Application:

    This "sampling of operators" is especially important for application areas where frames are heavily used, so that the link between model and discretization is maintained. To facilitate implementations, operator equations can be transformed into a finite, discrete problem with the finite section method (much in the same way as in the ONB case).

    Publications:

    • P. Balazs, "Matrix Representation of Operators Using Frames", Sampling Theory in Signal and Image Processing (STSIP) (2007, accepted), preprint
    • 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
  • Objective:

    In this project, head-related transfer functions (HRTFs) are measured and prepared for localization tests with cochlear implant listeners. The method and apparatus used for the measurement is the same as used for the general HRTF measurement (see project HRTF-System); however, the place where sound is acquired is different. In this project, the microphones built into the behind-the-ear (BtE) processors of cochlear implantees are used. The processors are located on the pinna, and the unprocessed microphone signals are used to calculate the BtE-HRTFs for different spatial positions.

    The BtE-HRTFs are then used in localization tests like Loca BtE-CI.

  • Objective:

    The Acoustic Research Institute was mandated to do measurements with the acoustic 64-channel microphone array using the beam forming method to derive a source model for high speed trains according to the new guideline CNOSSOS-EU.

    Method:

    The beam forming method was used, because the train is a fast moving vehicle and therefore a transient acoustic source. Five heights were used in the evaluation based on the CNOSSOS-EU and additionally five heights were evaluated that fit to the geometry of the trains.

    Application:

    Speeds from 200 km/h up to 330 km/h were tested for the ICEs and from 200 km/h up to 250 km/h for the Railjet. At the same speed both trains had the same acoustic level.

  • Objective:

    The Multiple Exponential Sweep Method (MESM) is an optimized method for the semi-simultaneous system identification of multiple systems. It uses an appropriate overlapping of the excitation signals. This leads to a faster identification of the weakly nonlinear systems that are retrieving the linear impulse response only. Using a Gabor multiplier in the post-processing procedure of the system identification may reduce the measurement noise. This may further improve the signal-to-noise ratio of the measured data.

    Method:

    A Gabor multiplier is used to cut the interesting parts out of the measured signals in the time-frequency plane. This allows a specific optimization of signal parts, independent of the frequency. Initial tests applying a Gabor multiplier to simulated data showed that the depth of spectral notches could be raised. A systematic investigation of this method is the main goal this project.

    Application:

    This method may improve the signal-to-noise ratio in system identification tasks of any weakly nonlinear system, such as those involving acoustic measurements with electric equipment.

    Publications:

    • P. Majdak, P. Balazs, B.Laback, "Multiple Exponential Sweep Method for Fast Measurement of Head Related Transfer Functions", Journal of the Acoustical Engineering Society , Vol. 55, No. 7/8, July/August 2007, Pages 623 - 637 (2007)

    Project-completion:

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

  • Objective:

    In Cooperation with National Instruments an implementation of MPEG4 features in the software package DIADEM is planned.

    Method:

    The application of MPEG4 features to noise is proven. Now the implementation of MPEG4 features into DIADEM is planned. In preparation of the project additional features were implemented into STX. The implementation into DIADEM is projected in the future.

    Application:

    DIADEM is a database that allows for a rapid search of measurement recordings. New search indexes can be generated based on the MPEG4 features of the recordings.

  • Basic Description:

    Time-variant filters are gaining importance in today's signal processing applications. Gabor multipliers in particular are popular in current scientific investigations. These multipliers are a specialization of Bessel multipliers to Gabor frames. These operators are interesting in regard to both theory and application:

    Theory of Multipliers:

    • Bessel and Frame Multipliers in Banach Spaces: In this project, the concept of frame multipliers should be generalized to work with Banach spaces.
    • Theory of Wavelet Multipliers: The concept of multipliers can be easily extended to wavelet frames. The influence of the special structures of these sequences will be investigated.
    • Basic Properties of Irregular Gabor Multipliers: Here multipliers for Gabor frames on irregular lattices are investigated.

    Application of Multipliers:

    • Time Frequency Masking: Gabor Multiplier Models and Evaluation: The symbol for the Gabor multiplier is calculated adaptively and the resulting model incorporates both time and frequency masking components. The goal is to obtain an algorithm using 2-D convolution.
    • Improving the Multiple Exponential Sweep Method (MESM) using Gabor Multipliers: The MESM is an efficient system identification method. Initial tests have shown that this method can be improved with a Gabor multiplier applied as a mask for the original sweep.
    • Wavelet Multipliers and Their Application to Reflection Measurements: One method to calculate the absorption coefficient of a sound proof wall requires separation of the impulse responses of different reflections. They can be easily separated in a scalogram and they can be extracted using a wavelet multiplier.
    • Mathematical Foundation of the Irrelevance Model: In this project, the theoretical foundation of the irrelevance algorithms implemented in STx is being developed.

    Partners:

    • H.G. Feichtinger, K. Gröchenig et al., NuHAG, Faculty of Mathematics, University of Vienna
    • R. Kronland-Martinet, S. Ytad, T. Necciari, Modélisation, Synthèse et Contrôle des Signaux Sonores et Musicaux of the LMA / CRNS Marseille
    • S. Meunier, S. Savel, Acoustique perceptive et qualité de l’environnement sonore of the LMA / CRNS Marseille

    Publications:

    • P. Balazs, B. Laback, G. Eckel, W. Deutsch, "Introducing Time-Frequency Sparsity by Removing Perceptually Irrelevant Components Using a Simple Model of Simultaneous Masking", IEEE Transactions on Audio, Speech and Language Processing, Vol. 17 (7) , in press (2009) , preprint
    • P. Majdak, P. Balazs, B.Laback, "Multiple Exponential Sweep Method for Fast Measurement of Head Related Transfer Functions", Journal of the Acoustical Engineering Society , Vol. 55, No. 7/8, July/August 2007, Pages 623 - 637 (2007)

    Project-completion:

    This project ended on 01.01.2010; most subprojects ended on 28.02.2008 and are incorporated into the 'High Potential'-Project of the WWTF, MULAC.

  • Basic Description:

    Signal processing has entered into today's life on a broad range, from mobile phones, UMTS, xDSL, and digital television to scientific research such as psychoacoustic modeling, acoustic measurements, and hearing prosthesis. Such applications often use time-invariant filters by applying the Fourier transform to calculate the complex spectrum. The spectrum is then multiplied by a function, the so-called transfer function. Such an operator can therefore be called a Fourier multiplier. Real life signals are seldom found to be stationary. Quasi-stationarity and fast-time variance characterize the majority of speech signals, transients in music, or environmental sounds, and therefore imply the need for non-stationary system models. Considerable progress can be achieved by reaching beyond traditional Fourier techniques and improving current time-variant filter concepts through application of the basic mathematical concepts of frame multipliers.

    Several transforms, such as the Gabor transform (the sampled version of the Short-Time Fourier Transformation), the wavelet transform, and the Bark, Mel, and Gamma tone filter banks are already in use in a large number of signal processing applications. Generalization of these techniques can be obtained via the mathematical frame theory. The advantage of introducing the frame theory consists particularly in the interpretability of filter and analysis coefficients in terms of frequency and time localization, as opposed to techniques based on orthonormal bases.

    One possibility to construct time-variant filters exists through the use of Gabor multipliers. For these operators the result of a Gabor transform is multiplied by a given function, called the time-frequency mask or symbol, followed by re-synthesis. These operators are already used implicitly in engineering applications, and have been investigated as Gabor filters in the fields of mathematics and signal processing theory. If alternative transforms are used, the concept of multipliers can be extended appropriately. So, for example, the concept of wavelet multipliers could be investigated for a wavelet transform.

    Different kinds of applications call for different frames. Multipliers can be generalized to the abstract level of frames without any further structure. This concept will be further investigated in this project. Its feasibility will be evaluated in acoustic applications using special cases of Gabor and wavelet systems.

    The project goal is to study both the mathematical theory of frame multipliers and their application among selected problems in acoustics. The project is divided into the following subprojects:

    Theory of Multipliers:

    1. General Frame Multiplier Theory
    2. Analytic and Numeric Properties of Gabor Multipliers
    3. Analytic and Numeric Properties of Wavelet Multipliers

    Application of Multipliers:

    1. Mathematical Modeling of Auditory Time-Frequency Masking Functions
    2. Improvement of Head-Related Transfer Function Measurements
    3. Advanced Method of Sound Absorption Measurements

    Partners:

    • H.G. Feichtinger et al., NuHAG, Faculty of Mathematics, University of Vienna
    • R. Kronland-Martinet et al., Modélisation, Synthèse et Contrôle des Signaux Sonores et Musicaux of the LMA / CNRS Marseille
    • B. Torrésani et al., LATP Université de Provence / CNRS Marseille
    • J.P. Antoine et al., FYMA Université Catholique de Louvain

    Publications:

    • P. Balazs, J.-P. Antoine, A. Gryboś, "Weighted and Controlled Frames: Mutual relationship and first Numerical Properties",  accepted for publication in International Journal of Wavelets, Multiresolution and Information Processing (2009), preprint
    • P. Balazs, “Matrix Representation of Bounded Linear Operators By Bessel Sequences, Frames and Riesz Sequence“,SampTA'09, 8th International Conference on Sampling and Applications, May 2009, Marseille, France
    • A. Rahimi, P. Balazs, "Multipliers for  p-Bessel sequences in Banach spaces", submitted (2009)
    • D. Stoeva, P. Balazs, "Unconditional convergence and Invertibility of Multipliers", preprint (2009)
    • Monika Dörfler and Bruno Torrésani, “Representation of operators in the time-frequency domain and generalized Gabor multipliers”, J. Fourier Anal. Appl., 2009 (in press)
    • Yohan Frutiger: "Multiplicateurs de Gabor pour les transformations sonores" (Gabor Multipliers for sound transformations) Master thesis under the supervision of R. Kronland-Martinet, June 2008 
    • F. Jaillet, P. Balazs, M. Dörfler and N. Engelputzeder, “On the Structure of the Phase around the Zeros of the Short-Time Fourier Transform”, NAG/DAGA 2009, International Conference on Acoustics, March 2009, Rotterdam, Nederland
    • F. Jaillet, P. Balazs and M. Dörfler, “Nonstationary Gabor Frames”, SampTA'09, 8th International Conference on Sampling and Applications, May 2009, Marseille, France
    • P. Balazs, B. Laback, G. Eckel, W. Deutsch, "Introducing Time-Frequency Sparsity by Removing Perceptually Irrelevant Components Using a Simple Model of Simultaneous Masking", IEEE Transactions on Audio, Speech and Language Processing (2009), in press
    •  B. Laback, P. Balazs, G. Toupin, T. Necciari, S. Savel, S. Meunier, S. Ystad and R. Kronland-Martinet, "Additivity of auditory masking using Gaussian-shaped tones", Acoustics'08, Paris, 29.06.-04.07.2008 (03.07.2008)
    • B. Laback, P. Balazs, T. Necciari, S. Savel, S. Ystad, S. Meunier and R. Kronland-Martinet, "Additivity of auditory masking for Gaussian-shaped tone pulses", preprint
    • Anaïk Olivero: "Expérimentation des multiplicateurs temps-échelle" (On the time-scale multipliers) Master thesis under the supervision of R. Kronland-Martinet and B. Torrésani, June 2008
  • Objective:

    The Multilevel Fast Multipole Method, when used in combination with the Boundary Element Method (BEM), is a tool to significantly speed up the simulation of large objects almost without loss in accuracy.

    Method:

    The Fast Multipole Method subdivides the Boundary Element mesh into different clusters. If two clusters are sufficiently far away from each other (i.e. they are in each other's far field), all calculations that would have to be made for every pair of nodes can be reduced to the midpoints of the clusters with almost no loss of accuracy. For clusters not in the far field, the traditional BEM has to be applied. The Multilevel Fast Multipole Method introduces different levels of clustering (clusters made out of smaller clusters) to additionally enhance computation speed.

    Application:

    The MLFFM is used for the simulation of head related transfer functions. The diagram above compares the result of a classical BEM with the MLFMM.

  • This area is involved with the analysis of the acoustics of music and with human perception thereof.

    In close cooperation with em.o.Univ.Prof. Dr. Franz Födermayr (Inst. of Musicology, Univ.Vienna) historic recordings of Georgian multipart songs are analyzed and transcribed.

  • Objective:

    An important difficulty of ray-tracing and boundary element method is the fine grid, which is needed in the high frequency region.

    Method:

    By means of new alternating shape functions e.g. wavelets at the boundary it could be possible to define a grid on the boundary that is independent from the wave number.