bluuebunny/crossref_metadata_2025 huggingface dataset:
Finite element modelling of thermoviscous acoustics in closed cavities
Authors: Joly Nicolas
Published: May 2008

A numerical methodology is presented to compute the acoustic field in a closed domain filled by a thermoviscous fluid, using the finite element method. The formulation based upon temperature variation and particle velocity is well suited for both (i) modelling the acoustic, thermal and viscous effects in the fluid bulk and (ii) accounting for the non-slip and thermal boundary conditions on the solid. The set of coupled equations presents usual scalar- and vector- operators for finite element modelling. Due to the development of thin thermal- and viscous- boundary layers compared to the acoustic wavelength, very different scales are present in the computed fields: the finite-element mesh needs to be dramatically refined along the direction normal to the boundary, while a coarse mesh is often sufficient along the directions parallel to the boundary, and far from the boundaries. The optimised mesh is obtained iteratively, using a loop procedure computing (a) the thermoviscous solution, (b) the metric map suited for this field, and (c) the unstructured anisotropic adapted mesh based on this metric, until convergence. Results are presented for two-dimensional and axisymmetric three-dimensional applications, illustrating thermoviscous effects for various geometries of cavities.
Detection and Cortical Representations of the Break in Interaural Correlation of Narrowband Noises Are Affected by Center Frequency and Interaural Delay
Authors: Huang Ying , Kong Lingzhi Z. , Fan Silu , Wu Xihong H. , Li Liang
Published: May 2008

When either broadband or narrowband arbitrary noises presented at the two ears are correlated, a fused noise image is perceived inside the head if the interaural interval (interaural time difference, ITD) is sufficiently short, indicating that acoustic-waveform information can be binaurally integrated. At both the perceptual level and neurophysiological level, this study investigated whether the binaural integration of correlated noises is affected by center frequency (for narrowband noise) and ITD. Results of the psychophysical experiment show that the duration threshold for detecting a break in correlation (BIC) in the correlated noises at the two ears was higher for high-frequency noises than for low-frequency noises, and dramatically elevated with the increase of the ITD from 0 to 4 ms. Moreover, the ITD-induced threshold elevation was much larger for high-frequency narrowband noises than for low-frequency narrowband noises. Results of the neurophysiological experiments show that the cortical components of scalp event-related potentials to the BIC were markedly modulated by both the center frequency (for narrowband noise) and ITD. Thus, temporal integration of acoustic waveform details of correlated noises at the two ears is both frequency and ITD dependent. Supported by the National Natural Science Foundation of China.
Spatiotemporal interactions between audition and vision
Authors: Bronkhorst Adelbert W. , Koelewijn Thomas , Van Der Burg Erik
Published: May 2008

It is well known that spatial auditory cues can affect performance on spatial visual tasks. One example is crossmodal cueing: response times for a visual spatial discrimination task are shortened when the location of the visual stimuli is cued auditorily, and prolonged when an invalid location is cued. Another example is auditory support of visual search: we normally find lower search times for visual items when the location of the target item is indicated by an auditory cue. In our research we have explored two questions: one related to crossmodal cueing, and the other to audiovisual search. In our crossmodal cueing paradigm we have looked at the interaction between (exogenous) cueing and endogenous information given before each trial. In this way we wanted to determine whether auditory cues are able to capture visual attention even when this attention has already been focused endogenously. In our audiovisual search paradigm we have studied whether nonspatial auditory cues, presented simultaneously with the visual target item, are able to improve visual search. Because of the temporal synchrony, the auditory stimuli do not act as actual cues but they are probably fused with the visual stimuli into an integrated percept.
Towards an hearing threshold prediction model in car noise
Authors: Sciabica Jean-François , Richard Florent , Roussarie Vincent
Published: May 2008

Sound design is an important challenge at PSA Research Center. Engine noise is rich in harmonics and interacts with aerodynamic and road noise. In this context, comprehension of masking phenomenon is indispensable to obtain a good description of sound attributes like roughness and booming. In this paper, we present a study about interaction between engine harmonics and road and aerodynamic noise. Thresholds for the detection of one harmonic in noise are firstly measured as a function of noise level. We propose also a prediction model for thresholds as a linear function of noise level in critical band. Results were compared to data available in the literature. Secondly, masking of harmonics by a harmonic and noise is measured. The masker frequency is 100 Hz and its level varies from 65 to 85 dB SPL in 5 dB steps. Noise level is 80dB SPL. Masking patterns are traced for 8 frequencies (50 to 225 Hz).
Parametric quality prediction for IP-based audio
Authors: Raake Alexander , Garcia Marieneige , Feiten Bernhard , Möller Sebastian
Published: May 2008

Different multimedia services are more and more transmitted over a common network infrastructure, e.g. using the Internet Protocol (IP). Examples are the widespread voice over Internet Protocol (VoIP), and Internet Protocol Television (IPTV). The streaming of pure audio over IP even has a longer tradition, with applications such as internet radio. For an efficient development, planning and monitoring of such services, models can be used that predict user-perceived quality based on technical service characteristics. Speech quality models for telephony are among the most advanced ones in this context, with different model types like the signal-based PESQ (ITU-T Rec. P.862, 2001) or the parametric E-model (ITU-T Rec. G.107, 2005). In this paper, we describe a parametric approach for predicting the quality of IP-based audio. The main parameters are the audio codec, codec bitrate, packet loss characteristics and the audio content. We base our considerations on own listening tests conducted in the framework of IPTV quality assessment, on approaches and test data described in the literature and on complementary knowledge from the fields of speech and video quality models. In this context, we identify similarities and discrepancies between different types of services in the light of a common model framework.
Blind source separation and sound source localization on time-frequency domain considering time lag information
Authors: Ueda Shogo , Sasaki Fumio , Tanaka Osamu , Yasuoka Masahito
Published: May 2008

The blind source separation and sound source localization based on independent component analysis on time-frequency domain considering time lag information between source signals and observation signals are conducted. The formulation based on the independency of time-frequency domain and the linearity of source signal is presented. The method which can be conducted not only the separation of source signals but also the specification of location of source signals is proposed through the consideration of time lags. Using this method, it can be analyzed even if observation signals include an intermittent noise, under the assumption of some independency of source signals. First of all, the number of source signals is specified through the quotient of complex valued time-frequency information of two observation signals. Next, the locations of source signals are specified using the relationship of relative time lags between source signal and observation signal. Then, the source signals are obtained by use of the Fourier information. The numerical test is conducted to confirm our method, and then the locations of source signals and source signals are obtained by high accuracy.
Verifying the attenuation of earplugs in situ: variability of transfer functions among human subjects
Authors: Bockstael Annelies , Botteldooren Dick , Vinck Bart
Published: May 2008

The use of in situ measurements of hearing protectors' (HPD's) attenuation following the MIRE-protocol (Microphone In Real Ear) is increasing. The attenuation is hereby calculated from the difference in sound levels outside the ear and inside the ear canal behind the HPD. Custom-made earplugs have been designed with an inner bore that allows inserting a miniature microphone. A thorough understanding of the difference, henceforth called 'transfer function', between the sound pressure of interest at the eardrum and the one measured at the inner bore of the HPD is indispensable for optimizing this technique and extending its field of application. Of particular interest is the variation of these transfer functions among humans. This was checked experimentally on 19 subjects. Differences in sound pressure were measured at the HPD's inner bore, by the MIRE-microphone, and at the eardrum by inserting an extra tube microphone in the ear canal. All transfer functions showed a comparable shape, however variability was substantial for the exact frequency and amplitude of the resonance peaks. The link between this variability and the morphology of the individual's HPD and ear canal was addressed using FDTD-simulations (Finite-Difference Time-Domain) of the outer ear canal occluded by an earplug with inner bore.
Binaural loudness summation in and out of the laboratory
Authors: Florentine Mary , Epstein Michael
Published: May 2008

Textbooks state that a tone presented binaurally is louder than the same tone presented monaurally. This is called Binaural Loudness Summation, BLS. Recent data and classroom demonstrations of BLS using speech stimuli from a visually present talker yield much less BLS than previously reported. Are conclusions about BLS drawn from tones presented via earphones in the laboratory applicable to live- voice speech in a room? To answer this question, eight normal listeners were presented three types of stimuli [monitored live voice (MLV) spondees, recorded spondees, and tones] monaurally and binaurally across a wide range of levels. The same stimuli were presented via earphones and loudspeakers. Loudness was measured using magnitude estimation. Results show that the amount of BLS was significantly less for MLV spondees than for tones or recorded spondees. It was also significantly less for loudspeaker presentation than for earphone presentation. The amount of BLS was least for MLV spondees presented via loudspeakers. Results indicate that BLS in loudspeaker conditions is significantly less than BLS in typical laboratory test conditions using earphones. A new phenomena-called "Binaural Loudness Constancy"-will be described that may result from expectations about loudness of a visually present talker. [Work supported by NIH-NIDCD grant R01DC02241]
Loudness of impulsive sounds as a function of duration: a parametric study of the effect of level and frequency
Authors: Meunier Sabine , Rabau Guy
Published: May 2008

Loudness of brief signals increases when signal duration increases up to about 100 ms, this increase is called temporal summation. The question examined in this study is to determine how loudness change as a function of duration. Literature shows different results that lead to different conclusions: when duration increases (i) the loudness increases in direct proportion to energy (intensity x duration); (ii) the loudness increases faster than energy; (iii) the loudness increases slower than energy. We have shown, in previous experiments, that loudness change with duration can follow either (i) or (ii) depending on signal level. The present study provides a parametric examination of this question. Loudness was measured for signals whose energy was kept constant while the duration was varied. The parameters of the experiment were the signal frequency and level. The results allow us to explain part of the discrepancies found in literature. Moreover, the results of this study are used into a model of impulsive sounds that calculate with a good precision the loudness of environnemental short-duration sounds.
Acoustic source identification in an enclosed space using the inverse phased beam tracing at medium frequencies
Authors: Ih Jeong-Guon , Jeong Cheol-Ho
Published: May 2008

When the source is enclosed by room surfaces, it is not easy, though not totally impossible, to apply conventional modal methods for source identification. This is because there are too many complicated wave interferences and effects of wall impedance, in particular at medium frequencies. The phased beam tracing method was suggested as a fast and efficient acoustic simulation tool at the medium frequencies in an enclosure, which overcomes the defects of geometrical acoustics techniques. In this study, the phased beam tracing method, implemented in its inverse form, was applied to the identification of the acoustic sources inside a chamber. In the simulation tests, spherical and rectangular shaped sources operating in a room were taken as examples. When the source pressure distribution was reconstructed by this numerical inverse technique, the sound power spectrum radiated from the source could be estimated by eliminating the contribution from surface reflections. Reconstruction error was investigated by the Monte Carlo simulation in terms of the signal-to-noise ratio.
Identification of source velocities with Inverse Patch Transfer Functions method
Authors: Aucejo Mathieu , Totaro Nicolas , Guyader Jean-Louis
Published: May 2008

The identification of source velocities remains an important problem in noise control. For this purpose, several methods were developed such as Near-field Acoustic Holography (NAH) or inverse Boundary Elements Method (iBEM). An alternative method, based on the double measurement of pressure and particle velocity fields surrounding the source is presented. This method has been developed in the SILENCE European project framework. In this method, called inverse Patch Transfer Functions method (iPTF), measurement and identification surfaces are divided into elementary areas called patches. Then, source velocities are computed from acoustic field and inversion of impedances matrices obtained by FEM. Theoretically, this method presents two main advantages: it can be applied to sources with complex 3D geometries and measurements can be carried out in a non-anechoic environment, even in the presence of other sources. In the present paper, theoretical background of iPTF is exposed and results are presented on a source with simple geometry (an L-shaped plate) and discussed.
Detection of acoustic radiating areas of a generic helicopter cabin by beamforming
Authors: Martin Gérard , Simon Frank , Biron Daniel
Published: May 2008

Improvement of the helicopter internal noise is essential to decrease the fatigue effects of passengers. This requires having a metrological tool, able to give information on acoustic radiating areas in cabin in order to target appropriate acoustic passive or active solutions. In this context, the authors have shown, in previous papers, capability of modified beamforming using a cross-shaped array of microphones, associated with an acoustic mask, to localize isolated sources in a generic composite helicopter cabin (VASCo), thus, in spite of free field's hypothesis. The present paper deals with the feasibility of this measurement method to identify main acoustic pressure areas radiated by an helicopter mechanical deck whose vibration is generated by gear box beams between 800 and 5000 Hz. This configuration is performed, thanks to simulations and experiments on a sandwich composite panel of VASCo, excited by 4 correlated shakers. It appears that beamforming can be used to identify main acoustic area coming from the radiating of many coupled vibration modes. Nevertheless, the dynamic range decreases with the number of pressure anti-nodes and it is necessary to introduce an inverse method to reject the ghost images and to reconstruct amplitude and phase of synthesized sources.
Design of air springs for improved frequency response characteristics using computational fluid dynamics
Authors: Kessenich Grace R. , Pierce Allan D. , Pletner Baruch
Published: May 2008

High performance air springs can be limited as vibration isolation devices by aeroacoustic cavity resonances. Such systems require detailed knowledge of their frequency response over all frequencies of interest. Transient computational fluid dynamics (CFD) simulations can derive quantitative magnitude and phase data early in the design cycle, before manufacturing. CFD models are unaffected by ambient noise, resonances from other sources, and sensor sensitivity, major experimental concerns. Unlike analytical methods, CFD modeling is applicable to complex as well as simple geometries. The modeling methodology is illustrated in the present study with 2D and 3D simulations of three different prototypical chambers. A chamber wall section, representing a piston, is moved with a sinusoidal displacement at constant frequency. Measurement of the magnitude and phase from the resulting sinusoidal force on this surface provides one point of a frequency response. By running multiple simulations over a wide frequency range, the null and resonant frequencies are readily apparent. Comparison with analytical predictions for simple geometries validates this technique. Insertion of porous membranes in the air spring cavity is modeled and is shown to be an effective method for mitigating aeroacoustic effects on air spring performance. Simulations provide crucial information for membrane parameter and placement optimization.
Influence of Constructional Parameters Of Small Reciprocating Compressors on Sound Power Emissions
Authors: Kral Peter J.
Published: May 2008

Noise in the environment is an increasing problem in industrial countries and is often strongly restricted by governmental laws. Therefore reducing sound levels must be one target in every design process. This paper describes in detail the influence of constructional parameters on the total sound power emitted by a small reciprocating compressor (VH=159 m3) with reed valves. Measurements (sound power levels, sound pressure levels, FFT-analysis, time-signal analysis) on a test bed situated in an anechoic room by varying the operating conditions - for example - rotational speed, maximum pressure, and suction terms, were done. Additionally sound power measurements with regard to reed valve parameters (thickness of the reed, area of the inlet and outlet port of the valve plate, material of the valve plate) were achieved. Theoretically researches with the aid of computational methods (FEM-Analysis with Ansys, Calculations with MatLab) are presented and show very good accordance compared to the sound level measurements. These results should lead to a better understanding of the causal connection between constructional (valve) parameters/operating conditions and the resulting sound power emissions. Finally the differentiation of airborne and structure born noise may be helpful to minimize the noise emissions of compressors already within the design process.
Hydroacoustic modeling of the butterfly valves - some industrial applications
Authors: Villouvier Véronique
Published: May 2008

The French PWR units include a large number of control valves which are sometimes at cavitation working for certain operating conditions. The very high excitations which are then created may lead to the cracking of some small lines located nearby. These devices have been tested on laboratory test loops. An hydroacoustic behaviour model has been set up on the basis of very precise identifications obtained from numerous experimental tests. The model obtained describes the acoustic transfers through the valve and the sources generated by turbulence and cavitation in the low frequency domain, according to the main operating parameters: aperture, flow rate, cavitation number, upstream pressure and diameter. Hydraulic and vibratory measurements were carried out on several Residual Heat Removal circuits for many control configurations. When this circuit is functioning under certain conditions, the two valves which regulate the hot and cold flow rates operate according to a cavitation system. The hydroacoustic valve model is used to simulate of the valves under the hydraulic conditions of measurements. The results are compared at the vibratory levels recorded on site. The model also makes it possible to determine the optimal operating conditions in order to limit the vibrations of the circuit.
The wavelet analysis the acoustic emission signals generated by multi-source partial discharges
Authors: Cichon Andrzej , Borucki Sebastian , Boczar Tomasz
Published: May 2008

Within the research work, the results of which are presented in this paper, a comparative analysis of the acoustic emission (AE) signals generated by multi-source partial discharges (PDs) was carried out. The research investigations were carried out in a model system in which PDs were generated with two spark-gaps. A spark-gap in the surface system and a point-plane spark-gap were used due to the fact that these are the most commonly occurring PD forms in power transformers. The AE signals were registered with a contact transducer placed on the external part of the tub. The analysis in the time-frequency domain using a short-time Fourier transform (STFT) was carried out for the AE signals generated by PDs. The results of the analysis are shown in the form of two- and three-dimensional spectrograms of the power spectral density and three-dimensional spectrograms of the amplitude spectrum.
Using wavelet transform to locate acoustic emission source in composite plate with one sensor
Authors: Refahi Oskouei Amir , Ahmadi Mehdi
Published: May 2008

In this paper a new method for locating the AE source with one sensor is presented. When acoustic emission signals propagate in wave-guided, they have multi-mode and dispersion characteristics. The separation of the modes at the sensors could make it possible to extract the exact information about the source that produced the waves. Based on the modal nature of AE, it can be understood that a good location would have two methods to determine the arrival times. One is determined on the same part of the waves (not only the same mode, but also the same frequency) at all sensors. The other is determined on the different modes at one sensor, which makes it possible to reduce the number of sensors needed. First, through modal analysis of the propagation of elastic waves in a thin plate, the dispersion characteristics of the modes are predicted. Second, the wavelet transform theory is briefly outlined and its application in elastic waves is explained. It is shown that by using the peak of the magnitude of the wavelet transform, the arrival times of the different modes can be determined. Additionally, experiments were undertaken using a lead break on the edge of the plate. These demonstrated the feasibility of the one sensor linear location scheme.
Progress in Primary Calibration of Acoustic Emission Sensors
Authors: Keprt Jiri , Benes Petr
Published: May 2008

The paper reviews the background of the primary calibration of acoustic emission sensors and the determination of uncertainty by this calibration. There are discussed the aims and the purpose of the primary calibration and main sources of uncertainty in practical usage of calibration results. The comparison of the results of the both method is presented. The shape of calibration characteristics corresponds well up to 500 kHz. The calculation of the uncertainty in the reciprocity and step function calibration is closely described. The problem of propagation of uncertainty in the fast fourier algorithm was solved. Uncertainty of measurement by primary calibration is determined and presented in the paper. The uncertainty in reciprocity calibration of UT 1000 in range from 60 kHz to 285 kHz was ±3dB and from 285 kHz to 1 MHz up to ±8 dB and the same results in the calibration by step function. The calculation of the uncertainty includes all the presumable sources. Such as variation of input coefficient of reciprocity parameter, impact of thickness of glass capillary Also the main problems, that were solved and have to be solved, are presented.
Dialectal effects in the perception of vowels produced by first and second language speakers: North Carolinian versus Southern Welsh listeners
Authors: Hoffmann Bion Ricardo Augusto , Escudero Paola , Morrison Geoffrey Stewart
Published: May 2008

This paper investigated the effect of listeners' regional origin on the perception of vowels. Listeners from North Carolina and South Wales were presented with natural tokens of the four English vowels /i/, /I/, /E/ and /{/ which were produced by L1 and L2 speakers. Randomization tests revealed a significant difference between the listener groups' confusion matrices. Examination of the confusion matrixes suggested that the dialectal differences were primarily in the vowel contrasts /I/-/E/ and /E/-/{/. In order to explore these findings, logistic regression models were fitted to each listener group's responses (predictor variables in the models were initial and final formant values and duration). Territorial maps based on the logistic regression models indicated that there was a substantial difference in the locations of the /E/-/{/, and a smaller difference in the location of the /I/-/E/ boundary.
Is there evidence of an allophonic mode of speech perception in dyslexic children
Authors: Messaoud-Galusi Souhila , Hazan Valerie , Rosen Stuart
Published: May 2008

Recent studies have suggested that dyslexics perceive phonemic categories in a less categorical fashion than individuals of average reading ability due to increased within-category discrimination ('allophonic' mode of perception). In the present study, 62 dyslexic and 51 typically-developing (TD) children aged between 6;6 and 13;7 years old were tested on the perception of a /pi-bi/ continuum using an adaptive identification task presented in clear and in background noise, and a fixed-level discrimination task presented in clear. They also undertook a battery of tests of reading, IQ, phonological awareness, phonological short-term memory and word perception in noise. The gradient of the identification function was significantly weaker for the dyslexic than TD group in clear, but the two groups did not differ in the noisy condition. In the discrimination test, there was no evidence of increased within-category discrimination in the dyslexic group but the TD group showed better cross-category discrimination. These results do not support the existence of an allophonic model of speech perception in dyslexic children. Also, they do not support the view that dyslexic children are more affected by noise as a result of more poorly specified phonemic categories. [Work supported by Wellcome Trust].
Are acoustic and articulatory changes of speech produced in noise only related to the increase in vocal intensity
Authors: Garnier Maeva , Henrich Nathalie , Dubois Danièle S.
Published: May 2008

Speech acoustics and articulation are modified in speech produced in noisy environments. Is this simply a result of the increase in vocal intensity or may these modifications be some communicative strategies to increase audiovisual intelligibility? We recorded a first acoustic database with ten speakers and a second acoustic and articulatory database with three speakers, for quiet and for two kinds of background noise. We first verified previous observations about speech modifications in noise and observed that some are not strongly correlated to the increase in vocal intensity and cannot be entirely explained by previous models of vocal effort. We also made additional observations. Pitch does not only rise in noise but also extends its dynamic. For female speakers, pitch not only rises but also fits spectrum "holes" of cocktail-party noise. Speech modifications are not only global over the whole utterance but also specific to some of its units. Articulatory modifications consist not only of an amplification of lip movements, but also correspond to an enhancement of the potentially visible contrast between vowels. Lastly, some articulatory changes may be compensation for formant modifications induced by the increase in vocal intensity, rather than a consequence of it.
A grouping approach to harmonic complexes
Authors: Krijnders Dirkjan J. , Niessen Maria E. , Andringa Tjeerd C.
Published: May 2008

Humans seem to perform sound-source separation for quasi-periodic sounds, such as speech, mostly on harmonicity cues. To model this function, most machine algorithms use a pitch-based approach to group the speech parts of the spectrum. In these methods the pitch is obtained either explicitly, in autocorrelation methods, or implicitly, as in harmonic sieves. If the estimation of pitch is wrong, the grouping will fail as well. In this paper we show a method that performs harmonic grouping without first calculating the pitch. Instead a pitch estimate is associated with each grouping hypothesis. Making the grouping independent of the pitch estimate makes it more robust in noisy settings. The algorithm obtains possible harmonics by tracking energy peaks in a cochleogram. Co-occuring harmonics are compared in terms of frequency difference. Grouping hypotheses are formed by combining harmonics with similar frequency differences. Consistency checks are performed on these hypotheses and hypotheses with compatible properties are combined into harmonic complexes. Every harmonic complex is evaluated on the number of the harmonics, the number of subsequent harmonics and the presence of a harmonic at the pitch position. By using the number of subsequent harmonics octave errors are prevented. Multiple concurrent harmonic complexes can be found as long as the spectral overlap is small.
Vocalizations of deaf infants before and after cochlear implantation
Authors: Fagan Mary K.
Published: May 2008

Critical aspects of spoken language depend on perceiving and understanding speech sounds. For deaf infants, however, neither sound perception nor awareness of the sound-making consequences of their actions is available. The goal of this study was to uncover and document early, measurable effects of hearing loss on infant vocalization and changes in these behaviors following cochlear implantation. Participants were 8 deaf infants, 7-11 months old, and 8 infants with cochlear-implants, 12-20 months old and 1-6 months post-implantation. Dependent variables include number of vocalizations, mean vocalization duration, and mean frequency before and after cochlear implantation and in relation to reduplicated babble onset and word production. Post-implant changes in vocal behaviors may have clinical implications for decisions regarding age of implantation and auditory habilitation as well as for understanding post-implant variability.
Acoustic analysis of the vocal tract during vowel production by finite-difference time-domain method
Authors: Takemoto Hironori , Mokhtari Parham , Kitamura Tatsuya
Published: May 2008

An acoustic simulator based on the finite-difference time-domain (FDTD) method was evaluated by acoustic measurements on solid models of the vocal tract. Three-dimensional vocal tract (3D VT) shapes for a male subject during production of the five Japanese vowels were measured by magnetic resonance imaging. Transfer functions of the 3D VT shapes were computed by the acoustic simulator. The accuracy of the finite-difference algorithm was second-order in time and fourth-order in space. From the same 3D VT shapes, solid models were made of epoxide resin by a stereo-lithographic technique, and their acoustic transfer functions were measured using a time-stretched pulse signal. The calculated and measured spectra were compared up to 8 kHz. Although locations of major poles and zeros were common between calculated and measured spectra, the calculated spectra showed a larger number of small zeros, possibly caused by underestimation of VT acoustic losses in the simulation. However, the lower four formants were simulated remarkably accurately, with a mean absolute error of only 2.2 % compared with the acoustic measurements. These facts indicated that the acoustic simulator can reasonably account for acoustic phenomena within the VT.
Speech intelligibility determined with various tests presented against noise
Authors: Ozimek Edward , Kutzner Dariusz , Warzybok Anna
Published: May 2008

The paper describes three Polish tests for assessment of speech intelligibility under noisy conditions: the matrix test, the sentence test and the digit triplet test. The matrix test contains semantically unpredictable sentences of fixed syntactical structures. This test is very useful for long-term intelligibility measurements and is applicable for subjects with profound hearing loss or cochlear implants users. The sentence test comprises utterances taken from everyday speech and aims at clinical speech intelligibility measurements. The digit triplet test contains sequences of three digits that are spoken separately and is aimed at screening measurements, especially via telephone. The statistical and phonemic properties of the tests are analyzed and compared. The correlation between results obtained in the respective tests and the tests applicability are also discussed.
Coarticulation in CV sequences: Locus Equation data
Authors: Graetzer Simone
Published: May 2008

Background: Speech sounds are known to be influenced by the context in which they occur, and to be produced in an overlapping way. These processes are referred to by the term "coarticulation". Objective: The aim of the study was to compare the magnitude of anticipatory coarticulation of oral voiceless plosives by following vowels in four Australian languages, Arrernte, Burarra, Gupapuyngu, and Warlpiri. Methods: The corpus consists of approximately six hundred isolated real words spoken by eight female adult speakers of the four languages. Locus equations are calculated for intervocalic consonants /p,c,t,tr,k/ (where "/tr/" represents a voiceless oral retroflex plosive) and all following vowels in consonant-vowel sequences. Measurement points are 0.1 and 0.5 into the vowel. The relationship between these two measurement points is linear and is expressed as a locus equation. The slope of the fitted line associated with the equation is a measure of the magnitude of coarticulation of the consonant by the following vowels. Results: There was a general tendency towards relatively highly coarticulated bilabial and velar plosives, weakly coarticulated retroflex plosives, and moderately coarticulated alveolar plosives. Results were variable for palatal plosives. These results are for the most part in accordance with findings for other Australian languages.
Speed-curvature relations in speech production: a multilingual experimental and modeling study
Authors: Perrier Pascal , Fuchs Susanne
Published: May 2008

Regularities in the relations between tangential velocity and trajectory curvature have been found in a number of human movements. They were formalized in the 1/3 power law [Viviani and Terzuolo, Neuroscience 7: 431-437, 1982]. Viviani and colleagues also found that modifying the speed-curvature relations would influence the perception of synthetic movements in terms of naturalness and classification. Hence, these relations could be at the core of the perception-action link in human movements. In this paper, tongue movement collected from speakers of French, German and Mandarin are analyzed. They are compared with data from simulations using a biomechanical tongue model, which is controlled on a target-to-target basis. Our results for experimental data and simulations are very similar. It is found that the 1/3 power law offers a fair description of the velocity-curvature relations when a large amount of movements are considered together. However, a refined analysis of individual movements reveals that the exceptions to this law are numerous. It is concluded that the velocity-curvature relation is not controlled in speech movements, and that the 1/3 power law only accounts for general characteristics of movements, which could arise from dynamical properties of the articulators.
Robust harmonic grouping by octave error correction
Authors: Van De Vooren Hedde , Violanda Renante R. , Andringa Tjeerd C.
Published: May 2008

Harmonic grouping is a frequently applied technique in computational auditory scene analysis and automatic speech recognition systems. However, grouping is easily disrupted by noise and reverberation. For instance, a noise induced signal component positioned roughly between two harmonics, might undesirably be assigned to the harmonic complex (HC) as well. This results in an octave error: harmonics in an HC are assigned to harmonic numbers twice as high as the correct values. We propose a cost function based method to correct these octave errors. This function is designed to, on the one hand, improve the balance between odd and even harmonic numbers, and, on the other hand, minimize the amount of signal components to be rejected. As a preprocessing step we applied short-time Fourier analysis to derive an instantaneous frequency representation from which we obtained the signal components. We used these as input for our harmonic grouping algorithm to obtain the HCs. Then we selected the optimal solution from the cost function and modified the composition of the HCs accordingly. As long as enough harmonics are sufficiently above the local noise level, this octave error correction mechanism works well for various sorts of harmonic sounds including speech.
A cross-language familiar talker advantage
Authors: Levi Susannah , Winters Stephen , Pisoni David B.
Published: May 2008

Previous research has shown that familiar talkers are more intelligible than unfamiliar talkers. In the current study, we tested the source of this familiar talker advantage by manipulating the type of talker information available to listeners. Two groups of native English listeners were familiarized with the voices of five German-English bilingual talkers; one group learned the voices from German stimuli and the other from English stimuli. Thus, English-trained listeners had access to both language-independent and English-specific talker information, while German-trained listeners had access to language-independent and German-specific talker information. After three days of voice learning, all listeners performed a word recognition task in English. Consistent with previous findings, English-trained listeners found the speech of familiar talkers to be more intelligible than unfamiliar talkers, as measured by whole words and phonemes correct. In contrast, German-trained listeners showed no familiar talker advantage, suggesting that listeners must have knowledge of talker-specific, linguistically relevant information to elicit the familiar talker advantage and that knowledge of language-independent talker information - such as size and shape of the vocal tract - does not facilitate speech perception.
Efficient Replica Generation for Space Time Adaptive Processing of Acoustic Signals
Authors: Imam Neena , Barhen Jacob , Traweek Michael
Published: May 2008

This paper addresses an innovative method for real-time replica generation using generalized Green's function for Space Time Adaptive Processing (STAP) technology for sonar applications. In hypothesis processing for STAP algorithms, the first step is to generate replicas for various target hypothesis such as the infinite-range fixed-direction (IRFD) target hypotheses or the tracking (trajectory) hypotheses. Once generated, these replicas are used in multiple steps of the STAP algorithm for computing system parameters such as the signal to noise ratio. Replicas are traditionally pre-computed or hard-wired due to computational load constraints. Such a pre-computation precludes direct application of array shape corrections, and limits the ability to process arbitrary target trajectories. W propose to use a generalized Green's function for efficient and ultra-fast real-time generation of replicas on multi-core processors Specifically, for an arbitrary target trajectory rT(ti)=[xT(ti), yT(ti), zT(ti)], arbitrary receive array trajectory rR(k)(ti)=[xR(k)(ti), yR(k)(ti), zR(k)(ti)], (k=1,...,NEL), arbitrary vector of frequencies in homogeneous acoustic medium f=[f1,f2,....,fn,...], and sound speed c0, we can write G[rT(ti),rR(k)(ti),fn|c0]= exp[j(2πfn/c0)| rR(k)(ti)- rT(ti) |]/ | rR(k)(ti)- rT(ti) | In terms of rationale, this is the simplest possible replica set for any STAP algorithm. The efficient implementation of the generalized Green's function on a multi-core processor will allow real-time adjustment/modification of acoustical target tracking resulting in improved numerical accuracy and time performance.
Tracking of underwater acoustic tone sources with linear antenna arrays
Authors: Orlov Denis , Rodionov Alexander
Published: May 2008

Track determination is a necessary task for a variety of underwater measurement applications, such as measurement of acoustic fields of ships. The use of linear antenna arrays provides great advantages for measurements and tracking. The investigated ship may be equipped by a special sound source of either pulse or tone signal utilized for tracking. The work considers two scenarios involving the case of a tone source. The scenarios are conditioned by parameters of the array and the source, which may vary depending on task. When a source is of relatively high frequency (several thousands Hz), tracking can be based on the dependence of the Doppler frequency shift and bearing on time; in this case, the parameters to be estimated are the ones of the whole track (velocity, traverse instant, etc.). If the source frequency is relatively low (up to 300 Hz), the array is large (greater than a typical distance from the source to the array aperture), the array focusing ability allows to perform the determination of "instant" locations of the source at various instances; estimating of parameters of the track as a whole is then based on the set of the obtained coordinates. The developed methods have been successfully tested using a large amount of experimental data.
Identification of Noise Emission in a Gear Unit
Authors: Belsak Ales , Prezelj Jurij
Published: May 2008

Today it is very important to ensure a stable production without unscheduled outages. To achieve this objective it is required to use advanced production technologies, to ensure adequate maintenance of mechanical systems and to monitor the condition of a device or machine. Reliable and accurate operation of machines and devices with as few outages as possible is desired. The significance of a life cycle design of machines and devices is growing. Possible damages in gear units can be defined by means of monitoring acoustic emission. A crack in the tooth root is usually indicated by significant changes in tooth stiffness. A difference in dynamic responses of an undamaged gear and of a damaged gear can be noted. The possibility of the use of an acoustic method in the field of condition diagostics is dealt with. The noises produced by a gear unit have been analysed, the noise sources within a gear unit have been determined and the corresponding time-frequency analysis of these sources have been performed, using an acoustic camera.
Digital High Frequency Coded Imaging System
Authors: Lewandowski Marcin , Nowicki Andrzej
Published: May 2008

Coded transmission is an approach to solve the inherent compromise between penetration and resolution required in ultrasound imaging. It is universally acknowledged that this technique gives major improvement in SNR. A novel high frequency (20-35 MHz) ultrasound real-time imaging system for research and evaluation was developed. The digital programmable coder-digitizer module based on the field programmable gate array (FPGA) supports arbitrary waveform coded transmission and RF echoes sampling up to 200 MSPS, as well as real-time streaming of digitized RF data via a high speed USB interface to the PC. All RF and image data processing were implemented in the software. A novel balanced software architecture using CPU and GPU processing supports real-time processing and display at rates up to 30 frames/sec. The system was used to acquire data for sine burst and 16-bit Golay code excitation using a single element scanning head with thick film focused spherical transducer 25 MHz center frequency and 56% system bandwidth. SNR gain for the Golay codes (referenced to single burst) of 15 dB for 20 MHz and 16 dB for 35 MHz were obtained. In water the axial resolution for both single burst and the Golay codes was the same FWMH=20 ns at 20 MHz and FWMH=15 ns at 35 MHz.
Spectral properties of the backscattering matrix in random scattering media
Authors: Aubry Alexandre , Derode Arnaud
Published: May 2008

We investigate, both experimentally and theoretically, the spectral properties of the backscattering operator in random media. The experimental set up consists in an array of 64 programmable ultrasonic transducers placed in front of a random scattering medium. The impulse responses between each couple of transducers are measured and form the interelement matrix.The evolution of the singular values with time and frequency is computed by means of a short-time Fourier analysis. Our aim is to compare the mean distribution of the singular values with the predictions of Random Matrix Theory (RMT). Two kinds of random media have been investigated: a highly scattering medium in which multiple scattering predominates and a weakly scattering medium. In both cases, residual correlations that may exist between matrix elements are shown to be a key parameter. The mean distribution of singular values exhibits, surprisingly, very different behaviors in the single and multiple scattering regimes. The agreement between experimental results and RMT predictions is found to be very good in both cases. The results are applied to the detection of a target embedded in a random scattering medium with the DORT method.
Optimal estimation of aircraft noise with four microphones in a spatial configuration
Authors: Roosnek Nico
Published: May 2008

By using all time shifts between the arrivals of the acoustic signal at the four microphones - in a spatial configuration - the source direction is obtained. The used correlation technique reduces significantly noise from other sources and the error in the estimated level of noise from the tracked aircraft/source. By taking into account a motion model for the source, the history of that motion and the measured acoustic data a very robust sound monitor system is obtained. The sensitivity for other sources, wind and rain noise is greatly reduced - theoretically almost unlimited - in comparison with classical monitor systems. Four aircraft can be tracked simultaneously. Another system capability is the estimation of the true flightpath with a system error depending on the system configuration and the environmental conditions. Using this path information the soundscape can be calculated by inter- and extrapolation. The design goal of the system is truly met!