bluuebunny/crossref_metadata_2025 huggingface dataset:
Improving the bandwidth of 1-3 connectivity composite receivers using mode coupling
Authors: Gachagan Anthony , Hayward Gordon
Published: July 2002

Temporal resolution in an ultrasonic system may be enhanced by the application of mechanical damping to the transducer rear face, thereby reducing internal reverberation and increasing effective bandwidth. However, for thickness drive transducers, this is accompanied inevitably by a reduction in sensitivity and, moreover, manufacture of suitable damping blocks can be difficult, particularly for lower frequency, small signal applications such as the detection of gas coupled ultrasound. This work describes an interesting alternative approach that utilizes the relatively strong coupling between the fundamental thickness mode and first lateral mode in 1-3 connectivity piezocomposite transducers. Finite element modeling is used to evaluate the influence of mode interaction on electromechanical coupling efficiency, surface displacement, sensitivity, and bandwidth as functions of the ceramic pillar dimensions for operation into both water and air load media. A range of composite devices was constructed and close agreement between theory and experiment is demonstrated, with a measured device bandwidth of 130% centered at 1.15 MHz. An example of using such a device within a gas-coupled ultrasonic system is presented and the response is shown to compare favorably with alternative transducer configurations.
Comments on “A method for forward and inverse solutions of a three-dimensional model of the cochlea” [J. Acoust. Soc. Am. 103, 3725 (1998)]
Authors: Duifhuis Hendrikus
Published: July 2002

This comment specifies reservations about the generalizability of de Boer’s letter [J. Acoust. Soc. Am. 103, 3725 (1998)] in which he claims that the real (nonlinear) cochlea can be analyzed using a straightforward extension of a common linear analysis. Two points are elaborated. Basically they regard physical aspects of boundary conditions at the cochlear partition and at the helicotrema.
Perceptual evaluations of spectral and temporal modifications of deaf speech
Authors: Yeni-Komshian Grace H. , Bunnell H. Timothy
Published: July 2002

The perceptual effect of modifying speech produced by deaf talkers was investigated to discover the changes necessary for disordered speech to be judged normal. Recordings of passages read by three deaf talkers were used as material. For the first two experiments, a three-syllable word was extracted from the deaf talkers’ passages and from a similar passage recorded by a hearing talker. Each of the deaf speech samples was paired with the normal speech sample to generate various continua that differed in the spectral and temporal modifications applied to them. Within each continuum, the individual stimuli varied in the shape of the spectrum envelope and were produced by linear interpolation of LPC analysis parameters between the deaf and normal speech end points. Results suggest that correcting the temporal component of deaf speech alone is not enough to make it sound normal. Spectral corrections that approximate about 70% of normal appear to be necessary for the deaf speech samples to be judged normal. A third experiment made use of a 10-syllable segment of speech in which the relative contributions of spectral and temporal adjustments were investigated. The general conclusion of these three experiments is that spectral adjustments are more important to perceptual judgments of normality than temporal adjustments.
On loudness at threshold
Authors: Buus So/ren , Müsch Hannes , Florentine Mary
Published: July 2002

Absolute thresholds for and loudness matches between pure tones and four- and ten-tone complexes were used to assess the form of the function relating loudness to sensation level, SL, at low and moderate levels. The components of the tone complexes had equal SLs and were separated by one, two, four, or six critical bands. Six listeners with normal hearing were tested. The thresholds for the multitone complexes indicate that they generally can be detected even when the level of a single component is a few dB below the threshold. The average detection advantage is consistent with predictions for multiple observations in independent, frequency-selective auditory channels, but differences among listeners are apparent. The loudness matches also vary somewhat among listeners. Five of the six listeners matched tone complexes composed of subthreshold components to a pure tone a few dB above threshold. This indicates that the loudness of tones at or even below threshold is greater than zero for these five listeners. A simple model of loudness summation was used to obtain loudness functions from the individual listeners’ loudness matches. The slopes of the loudness functions [log(loudness) plotted as a function of log(intensity)] generally exceed unity at low levels and are near 0.2 at 40 dB SL. This shallow slope at moderate levels agrees with loudness functions derived from data on temporal integration of loudness. The average loudness function derived from the present data also is in good agreement with a variety of previous data obtained by magnitude estimation, magnitude production, ratio production, and measurements of binaural loudness summation.
Modal analysis of wave fields in range-independent oceanic media
Authors: Parastates Elias
Published: July 2002

A maximum-likelihood (ML) method for modal analysis of time invariant two-dimensional wave fields in a deterministic layered waveguide is introduced. The Cramer–Rao bounds (CRBs) of the variances of the ML modal parameter estimates are derived from the likelihood function of the wave field sampled in a vertical plane by a rectangular receiver array. The ML estimates are shown to be consistent for a finite array size and a fixed number of sensors as the number of snapshots tends to infinity. The ML method is applied on data collected in the Swedish archipelago, is shown to resolve a high number of modes from moderate length apertures, and to achieve a good agreement between modeled and observed wave fields.
Range-dependent acoustic radiation from line-driven plates above the bending coincidence frequency
Authors: Rumerman M. L.
Published: July 2002

The leaky waves generated on plates driven in bending above the bending coincidence frequency radiate as extended sources; therefore, the standard saddle point evaluation of the radiated pressure, which yields a far-field result, may not be appropriate at the range of interest. The calculation can be modified by removing the contribution of the leaky waves from the original integrand, evaluating the remaining regular part by the saddle point method, and evaluating the effect of the leaky waves in terms of error functions and residue. Results showing range dependence are presented for radiation by a steel plate in water driven by normal and tangential line forces. The correctness of the results is made plausible by comparing independent calculations of acoustic power passing through a semicircle at finite range.
Vortex sound in bass-reflex ports of loudspeakers. Part II. A method to estimate the point of separation
Authors: Roozen N. Bert , Bockholts Marije , van Eck Pascal , Hirschberg A.
Published: July 2002

In part I of this paper, the vortex shedding that may occur in a bass-reflex port of a loudspeaker system was discussed. At the Helmholtz frequency of the bass-reflex port, air is pumped in and out at rather high velocities, vortex shedding occurs at the end of the port, and blowing sounds are generated. It was explained that the key in the design of a port with a minimum of blowing sounds is the point of flow separation from the wall at which vortices are formed. This paper presents a method for estimating the point of separation for an unsteady flow like the flow through a bass-reflex port. Assuming that the flow can be described by a potential flow up to the point where flow separation occurs, it was found that the point of separation can be estimated on the basis of measurement of the sound pressure inside the loudspeaker box and measurement of the sound pressure at a distance of 1 m from the port exit. Application of the proposed technique to a cylindrical port with rounded edges at both port ends revealed that the point of separation is determined by the particle displacement rather than by the particle velocity. It was also found that a good indicator of the onset of severe vortex shedding is the Strouhal number based on the radius of curvature of the port edges.
The ring-type all-fiber Fabry–Pérot interferometer hydrophone system
Authors: Li Donglin , Zhang Shuquan
Published: July 2002

A compact theory and the detailed design of the ring-type all-fiber Fabry–Pérot interferometer hydrophone system has been developed here. The system is composed of four parts: the optical part with the laser source, the ring interferometer of all-fiber suited for the acoustical structure, the acoustical structure, and the stabilization maintaining the feedback servo mechanism and signal demodulation part. For this reason it differs greatly in many ways from the traditional concept of hydrophone which is commonly made up with one or several functional material elements such as the piezoelectric ceramic or the PVDF ones. So, on accuracy ground it is named the Fiber Hydrophone System instead of Hydrophone. The fiber-optic hydrophone itself, not including the laser source and the electric servo part, has the compact size of φ40×80 mm, specific gravity of less than 1, and the system has a free-field open-circuit voltage sensitivity level of above −140 dB re: 1 V/μPa. Experimental results show an exciting future of this kind of fiber-optic hydrophone system for the application of all optical towed arrays.
Automobile airbag impulse noise: Otologic symptons in six patients
Authors: Saunders James E. , Slattery William H. , Luxford William M.
Published: July 2002

Automobile airbag safety systems have successfully reduced the number of occupant injuries from motor vehicle accidents. Unfortunately, airbags are also associated with some inherent risk, including a high-amplitude, short-duration noise from airbag deployment. A review of the available research in the automobile industry indicates that the peak amplitude of this noise may exceed 170 dB sound-pressure level. Despite the increasingly wide application of airbags in automobiles, there have been no previous reports of airbag-related otologic injuries. The authors have encountered six patients with otologic symptoms that appear to be related to airbag impulse noise. Five of these patients have documented hearing loss, one patient reported persistent tinnitus, and two patients have significant dysequilibrium. Although permanent hearing loss from airbag noise appears to be rare, temporary threshold shifts are probably much more common. It is important, therefore, that the clinician be aware of the noise associated with airbag inflation and the possibility of acoustic trauma from these safety devices.
Otologic effects of air bag deployment
Authors: Yaremchuk Kathleen , Dobie Robert
Published: July 2002

The Medical Aspects of Noise Subcommittee of the American Academy of Otolaryngology/Head and Neck Surgery published two requests, in the Academy’s monthly journal the Bulletin, for information from otolaryngologists on patients that had been exposed to airbag deployment and developed otologic compliants. There were 20 otolaryngologists that replied with reports of 29 patients that had been evaluated for hearing loss, tinnitus, and or vertigo. Seven patients had pre- and post-airbag deployment audiograms that document sensorineural hearing loss. Eighteen patients had asymmetrical sensorineural hearing loss on post airbag deployment audiogram. The most common symptoms after airbag deployment were tinnitus (17) and hearing loss (16). There were two reports of tympanic membrane perforations and one report of a perilymph fistula after exposure to airbag deployment. Two patients complained of vertigo. Two patients complained of recruitment.
Numerical simulations of xylophones. II. Time-domain modeling of the resonator and of the radiated sound pressure
Authors: Doutaut Vincent , Matignon Denis , Chaigne Antoine
Published: July 2002

This paper presents a time-domain modeling for the sound pressure radiated by a xylophone and, more generally, by mallet percussion instruments such as the marimba and vibraphone, using finite difference methods. The time-domain model used for the one-dimensional (1-D) flexural vibrations of a nonuniform bar has been described in a previous paper by Chaigne and Doutaut [J. Acoust. Soc. Am. 101, 539–557 (1997)] and is now extended to the modeling of the sound-pressure field radiated by the bar coupled with a 1-D tubular resonator. The bar is viewed as a linear array of equivalent oscillating spheres. A fraction of the bar field excites the tubular resonator which, in turn, radiates sound with a certain delay. In the present model, the open end of the resonator is represented by an equivalent pulsating sphere. The total sound field is obtained by summing the respective contributions of the bar and tube. Particular care is given for defining a valid approximation of the radiation impedance, both in continuous and discrete time domain, on the basis of Kreiss’s theory. The model is successful in reproducing the main features of real instruments: sharp attack, tuning of the bar, directivity, tone color, and aftersound due to the bar-resonator coupling.
High-frequency asymptotics and random matrix theory in reverberant rooms
Authors: Legrand Olivier , Mortessagne Fabrice
Published: July 2002

It is now recognized that in the low-frequency range below Schroeder’s cutoff a modal approach of the spectral fluctuations in a reverberant room is appropriate. Indeed, in his experimental study of the modal frequencies in a reverberation room, Davy [J. L. Davy, Proc. Inter-Noise, 159–164 (1990)] confirmed that the level repulsion is in good accord with the predictions of the Wigner–Dyson’s random matrix Theory (RMT). In Quantum Chaology many works have tried to understand the link between the observed spectral statistics of wave cavities which are known to be chaotic in the limit of rays and the statistics of eigenvalues of random matrices. Here this matter is addressed in the context of room acoustics by considering an asymptotic representation of the response in chaotic rooms in terms of periodic rays. Various results are established concerning universal aspects of the spectral fluctuations recovering predictions of RMT both in the moderate and large modal overlap regimes [O. Legrand and F. Mortessagne, Acustica–acta acustica 82 (Suppl. 1), S150 (1996)]. The results are illustrated in a two-dimensional chaotic enclosure using a semiclassical Green’s function formalism [F. Mortessagne and O. Legrand, Acustica–acta acustica 82 (Suppl. 1), S152 (1996)].
Sound-absorption properties of suspended ceilings with plastic foils
Authors: Bosmans Ivan , Lauriks Walter , Lombaert Geert , Mermans Joris , Vermeir Gerrit
Published: July 2002

Suspended ceilings using plastic foils can be used as wall or ceiling finishing systems. The cavity behind the foil (plenum) is used for hiding technical appliances. Without sound-absorbing material in the plenum, the absorption coefficient of the suspended ceiling can be very low, and even with absorbing material in the cavity, the presence of the membrane can seriously degrade the acoustic performance. The sound-absorption properties of suspended ceilings using a plastic impervious membrane are investigated theoretically and experimentally. The sound absorption coefficient has been modeled using a general model for the acoustic transmission and absorption of a multilayered structure. Measurements have been performed in a reverberant room using different sound-absorbing materials in the cavity. Due to the impervious foil, the structure behaves as a resonant system. The different physical mechanisms of these resonances will be highlighted and simple equations to estimate the resonance frequencies are put forward. Under certain conditions, the absorption coefficient can become exactly one, but at other frequencies the sound absorption can be quite low, unless sufficient sound-absorbing material is provided in the cavity. Using a perforated membrane increases the sound absorption drastically.
Acoustic attraction of the parasitoid fly Ormia ochracea to the song of its host
Authors: Ramsauer Nathalie , Robert Daniel
Published: July 2002

The parasitoid fly O. ochracea locates its cricket host by acoustically detecting its calling song. The phonotactic abilities of the fly’s auditory system were investigated behaviorally. Phonotaxis was investigated in a large (w 2.5×14.5×h 3m), sound-proofed, flight cage equipped with loudspeakers. Simulated cricket calls with different carrier frequencies including the standard song (4.8 kHz, 45 pps, 13-ms pulse length) were played back to attract the flies. Behavioral threshold curves were established by varying the sound pressure level (SPL) for songs with different frequencies (3.5 to 6.0 kHz). Threshold was determined when phonotaxis failed to occur but was validated only if the fly subsequently reacted to the next higher SPL in the intensity series. Between 4.5 and 5.2 kHz, thresholds were observed to be around 40 dB (re: 20 Pa; i.e., at 4.8 kHz: 38.3 dB, s.d.=3.0 dB, n=5 animals). The sensitivity of phonotaxis to the level of masking noise was also tested by masking songs of different frequencies with varying noise levels. Noise-threshold curves show the behavior to be least affected by noise at 4.8 kHz, where successful phonotaxis to a 60-dB SPL song occurs up to a masking noise level of 45.3 dB SPL. However, at the frequency of best physiological threshold (5.0 kHz), masking occurs at 39.0 dB SPL. [Work supported by the Swiss Science Foundation.]
Sound localization in shallow waters in the clawed frog Xenopus l. laevis
Authors: Tuschick Sebastian
Published: July 2002

Sound localization in waters by small animals like Xenopus was considered impossible due to the following physical facts: (i) The sound is not attenuated by going around the animal’s head, thus preventing interaural intensity difference. (ii) In water sound propagates five times faster than in air, yielding too little interaural time difference for detection. Tests showed, however, that Xenopus can locate underwater sound. To investigate the biophysical mechanisms of sound localization in Xenopus, tympanic vibration was measured with laser vibrometrics. The measurements were conducted in an artificial pond (6×9 m2, 1 m deep) that was especially constructed for analysis of underwater sound. To produce equal sound from any direction relative to the animal, a platform was built which could be rotated with the animal while sound was broadcast from a constant position. Initial experiments show a higher amplitude of tympanic vibration if sound is applied from the ipsilateral side compared to the contralateral side. The highest tympanic responses were observed between 1.6 and 2.1 kHz, the dominant frequency of the Xenopus advertisement call. Thus, obviously special biophysical adaptations of the ear enable Xenopus to localize underwater sound.
Effect of seismic waves on the acoustic field interference structure in the shallow sea
Authors: Borodina Elena , Egorychev Sergei , Kurin Vasily , Pronchatov-Rubtsov Nikolai , Petukhov Yury
Published: July 2002

The influence of lateral waves on forming of the acoustic field space-frequency interference structure in a shallow sea was investigated theoretically and experimentally. Within the scope of the simple model of shallow sea as a fluid layer of thickness H on an elastic half-space, the analytical frequency and horizontal range dependencies of amplitudes of compressional and shear lateral waves were derived. Numerical simulation was carried out for natural and laboratory experiment conditions. The numerical results showed the resonant character of frequency dependencies of compressional and shear wave amplitudes, with abrupt narrow maximums at critical frequencies of leak and propagating modes, respectively. It was shown also, that below the critical frequency of the first mode the interference structure is formed mainly by the fundamental mode and the shear wave, which substantially exceeds the compressional one; besides, the shear wave amplitude is almost two times less than the zero mode at distances of about 10 H. Thus it was proved that at ranges up to 200 H realized at both natural and laboratory experiment conditions, only the shear wave significantly influences forming of the space-frequency interference field structure in a shallow sea. [Work supported by RFBR Grant No. 98-02-16402, CCPE Grant No. 95-0-8.1-34.]
Orthogonal acoustical factors of sound fields in a bamboo forest
Authors: Sakai Hiroyuki , Shibata Shozo , Ando Yoichi
Published: July 2002

In order to discuss acoustical quality of sound fields in a bamboo forest, acoustical measurements were conducted for orthogonal acoustical factors of a sound field. Results at a distance of 40 m from the source show that the IACC was 0.16 at 2 kHz. This value is smaller than that in the forest previously investigated [Sakai, Sato, and Ando, J. Acoust. Soc. Am. 104, 1491–1497 (1998)]. The measured subsequent reverberation time Tsub was about 1.5 s in the frequency range above 1 kHz. For a certain music source with a higher frequency component, therefore, it is found that sound fields in the bamboo forest have excellent acoustic properties.
Acoustic levitation, a theoretical and experimental round up
Authors: Lierke E. G.
Published: July 2002

A resumé of terrestrial levitation of solid samples and drops in one-axial acoustic standing wave levitators is presented. It concentrates on a normalized theoretical description of drop displacement and deformation in the full dynamic range of the levitation safety factor, the ratio of axial levitation force and sample weight. The range of the initial sample radius a0 follows from the wave number product 10−2<ka0<1.5. Sample properties density ρ and surface tension σ are represented by the Bond number Bo=ρ/σ⋅a20⋅g0<2 (g0 is gravity constant). The critical Weber number Wecrit=10 (ratio of Bernoulli stress and capillary pressure) defines the upper limit of levitation at acoustic Mach numbers Macrit=√5/P0κ0⋅σ/a0 (P0 is static pressure, κ0 is heat capacity ratio). Resulting temperature variations and acoustic convections, represented by Nusselt and Sherwood numbers, are briefly discussed. A new family of modular, one-axial, acoustic levitators (20, 60, and 100 kHz) with sensor-controlled, calibrated resonance tuning is introduced. The levitators can easily be changed from a ‘‘Bessel mode’’ tube version, with minimized gas volume, into an ‘‘open’’ version with free radial access to the sample, both optional with superimposed axial gas flow. Some experimental techniques with compatible commercial components, like temperature and humidity sensors, optical accessories (observation and spectroscopy), and microdrop injectors are briefly discussed.
Iconic prosody and information in spontaneous oral narrative
Authors: Toledo Guillermo Andrés
Published: July 2002

The information interchange on cognitive categories of the mental representation of referents (concepts) interrelated with prosodic marking (levels of phonological prominence) have been studied. Speakers actualize a limited amount of information inactivating backwards. These cognitive processes appear in three states of the referents: activation, semi-activation, and inactivation. In activation, a concept would be focalized in actual stream of consciousness. In semi-activation, the information is situated in a peripheral zone of consciousness: the concept was already activated. In inactivation, the concept would be activated in the hearer’s conscience for the first time. These states were analyzed through data extracted from spontaneous narrative discourses obtained from an Argentinean Spanish speaker. The iconicity of prosodic prominences, a correlation between the phonetic encoding and the mental status of information weight, was explored acoustically. Measurements were made through F0 data normalized by a logarithmic z-score transformation calculated in the highest value of an item classified among one of the three states. Findings were similar to previous research on semi-spontaneous materials [G. Toledo, 16th International Congress on Acoustics, 4aSC1]: lower prominences were the marking of activation and semi-activation, and higher prominences were the feature of inactivation, i.e., a degree of prosodic iconicity.
Analysis of pitch accent categories in Welsh
Authors: Kibre Nicholas J.
Published: July 2002

This paper describes the results of a study of pitch accent categories in Welsh, and their distribution in natural speech. In addition to these descriptive findings, it describes the development of a data-driven approach to intonation. To be useful for the analysis of natural speech, an intonational model must provide means for determining an unambiguous phonological representation for any observed speech token. One approach which will be addressed is the Tilt Model [K. Dusterhoff and A. Black, ESCA Workshop on Intonation (1997); P. Taylor, ICSLP (1998)], amended as necessary to account for the accent patterns of Welsh. This will be compared with an alternative of using statistical methods, particularly factor analysis, to determine the underlying descriptive dimensions of pitch accents in the language. The resulting model will be applied to a corpus of spoken Welsh utterances, and initial findings on the distribution of accent types will be reported.
Time-interval information in the auditory representation of speech sounds
Authors: Patterson Roy D.
Published: July 2002

In the auditory system, the primary fibers that encode mechanical motion of the basilar partition are phase-locked to that motion, and this information is preserved, to varying degrees, up to the inferior colliculus. It is known that this timing-interval information is used in localization, and it is probably also used to separate sources from diffuse background noise. The time intervals are on the order of milliseconds, and so traditional speech preprocessors (like MFCC systems) with frames on the order of 15 ms, remove the time-interval information from the representation. The performance of these systems deteriorates badly when the speaker is in a noisy environment. This suggests that time-interval processing will eventually need to be integrated into speech recognition systems if they are to achieve the kind of noise resistance characteristic of human speech recognition. An auditory image model (AIM) will be presented that is designed to stabilize repeating time-interval patterns like those produced by voiced speech, and results from experiments where AIM has been used as a preprocessor for automatic speech recognition. [Work supported by UK MRC (G9703469).]
On temporal smoothing and speech perception
Authors: Ghitza Oded
Published: July 2002

Neurophysiological and psychophysical studies provide evidence for the existence of temporal integration mechanisms in the auditory system. These may be viewed as low-pass filters, parametrized by their cutoff frequencies. It is of interest to specify these cutoffs, particularly for tasks germane to the effect of temporal smoothing on speech quality, within the context of speech-coding strategies (e.g., what is the smallest number of bits with which to represent temporally smoothed speech that is perceptually indistinguishable from the original?). In order to answer such questions, speech-processing rules have been derived so as to create synthetic speech capable of generating model auditory-nerve (AN) firing patterns temporally smoothed in a manner commensurate with that observed in neurophysiological studies of the AN (e.g., at low characteristic frequencies, neural discharges of AN fibers are phase locked to the underlying driving critical-band signal, i.e., synchrony is maintained; at high CFs, temporal information is preserved by the instantaneous average rate of the neural firing pattern, which is related to the temporal envelope of the underlying driving signal). This presentation describes the processing rules and the results of psychophysical experiments performed to measure the cutoff frequencies of these filters within the context of speech quality.
Implications of a sensory-motor theory for the representation and segregation of speech
Authors: Todd Neil
Published: July 2002

Discussed is a computational model which attempts to simulate the principal brain structures involved in sensory-motor sequencing, including auditory cortex, posterior parietal cortex, cerebellum, motor cortex, and the motor output system. A particular focus in this presentation is on the auditory cortical component responsible for computing a 3-D spatio-temporal power spectrum of the auditory image flow [N. Todd, Proc. 16th ICA (1998)], which captures information topographically on three levels, roughly corresponding to timbre, pitch, and rhythm, and reflects the temporal resolution of the cochlea, inferior colliculus, and auditory cortex, respectively. By considering the spatio-temporal orientations of cortical receptive fields (RFs), it is possible to delineate a set of stationary and moving primitive features upon which the higher-order constructs of pitch and timbre are built. A separate model population of secondary auditory cortical neurons, which receive inputs from the population of primary RFs, provides a potential means of accounting for a number of important phonological and melodic processes. Implications of this model are discussed for the development of new algorithms for the representation and segregation of speech via self-organization of primary and secondary RFs by means of the principle of information maximization.
Signal processing algorithm for sound source localization with discretionarily distributed multiple receiving sensors
Authors: Yen Nai-chyuan
Published: July 2002

The deployment of strategically placed multiple receiving sensors has many practical ways to detect, localize, and classify a sound source in situations such as monitoring an isolated street disturbance in an urban environment with posted microphones or identifying a submerged object in the ocean with air-dropped sonobuoys. A simple processsing algorithm based on signals received from multiple sensors is derived from the information contained in the cross-spectrum matrix formed from the receiving sensors. Using the magnitude distribution of eigenvalues across the frequency bandwidth as the indication for the strength of signal excess in the frequency domain, their corresponding eigenvectors provide the required complex number weightings to select the related coherence components of the signal. By examining the phase angle data from each sensor’s weighting factor, a set of simultaneous algebra equations can be formulated to estimate the location of the sound source. The computation procedure of this approach is uncomplicated in comparison with the traditional triangulation method. Furthermore, the additional accompanied computed eigenvalues can be utilized for the characterization of the detected sound source, thereby establishing its identification reference. Simulated examples are used to test out the effectiveness of this multiple sensor sound source evaluation methodology at various operational scenarios.
Localization of the sources using blind identification of signal subspace
Authors: Ammar Bendjama , Salah Bourennane
Published: July 2002

To find the direction of arrival (DOA) of N sources impinging on an array of M sensors, actual fourth-order direction-finding methods try to solve an N-dimensional problem from the statistics of the data. Current narrow-band array processing techniques are based on the second-order statistics of the received signals. In many situations, the received signals are non-Gaussian, so that they contain valuable statistical information in their moments of order greater than two. In these circumstances, it makes sense to develop array processing techniques which are designed to use their higher-order information. Of particular interest are the algorithms based on higher-order cumulants, which are asymptotically insensitive to Gaussian noise: it is thus neither necessary to know, to model, or to estimate the noise covariance, which is a reasonable assumption in many practical situations. In this paper, an approach of direction finding is proposed, based on the first step of blind identification of source steering. As a second step, the direction-finding concept is now possible due to the development, these last years, of several blind source separation methods which allow the blind identification of source-steering vectors.
A look at the modal phase structure function in shallow water
Authors: Sperry Brian J. , Lynch Jim
Published: July 2002

Experimental and theoretical work in shallow-water acoustic propagation has invariably encountered the complexities of the shallow-water environment. A useful measure of the propagation complexity for array processing is the ‘‘modal phase structure function.’’ The modal phase structure function makes it possible to study how environmental fluctuations affect point-to-point phase differences, quantities which are directly related to modal coherences. A method has been developed, using perturbation theory, that directly relates the modal phase structure function to the environmental sound-speed correlation function. Despite its relative simplicity, this method can address fully coupled mode range dependence, as well as anisotropy in the environmental correlation function. The phase structure function may be applied to time and frequency coherence problems as well. Another strength of the technique is that contributions from various processes, such as mesoscale and finescale variability, interface roughness and geoacoustic variability, may be separated out under reasonable assumptions. Early calculations show good agreement with experimental results. [Work supported by ONR.]
Contemplations about the numeric transposition of the generalized Kirchhoff integral with the intention of active noise reduction
Authors: Trimpop Mattias , Krahe Detlef
Published: July 2002

The method of feedback is often used for reduction of noise in defined areas because feedback is able to compensate chances of the transfer function. However, the feedback system cannot work in some special cases, e.g., in single events or impulse noise no preceded signals parts are existing and usable for prediction. Therefore, the feedforward system has to be used in these cases and the prediction is realized by acquisition of the sound wave before it reaches the interesting point of immission. After the presentation of the one-dimensional feedforward system presented in the last year, now the two- and three-dimensional cases and their mathematical treatments will be discussed. For that purpose acoustic wave fields will be analyzed by the generalized Kirchhoff integral and minimized using sound sources in a defined wave field area. Further, the expected problems in the practical application of the numerical solution will be discussed.
Analysis of the neuro-fuzzy classifiers for fish species identification from acoustic echoes
Authors: Stepnowski Andrzej , Tran Van Dung , Maciolowska Joanna
Published: July 2002

Some of the variety of classification methods have been recently implemented also for fish species identification from their acoustic echoes. The classification systems employing artificial neural networks and fuzzy logic separately are not satisfactory, so the combined neuro-fuzzy approach were carried out. A couple of neuro-fuzzy classifiers were built and tested using the NEFCLASS program, which discovered the rules and found the shape of membership functions for determining correct class categories for a given input data set. The systems were tested on fish echoes acquired by DT4000 digital echo sounder. The results are promising with reference to the classifiers performance and their generalization abilities.
Bergen acoustic buoy (BAB)—A tool for remote monitoring of marine resources
Authors: Godo Olav Rune , Totland Atle
Published: July 2002

Marine resource abundance monitoring is often accomplished by means of research vessel acoustic surveys. Vessel time is limited and expensive, and optimal and cost-efficient utilization is required. Such vessels have clear limitations in observing the true acoustic distribution, density, and acoustic properties of marine organisms due to noise and light emitted during surveying and sampling. A potential solution to some of the emerging problems is a remote operated acoustic device, which can be used to monitor distribution, density, movements, and properties of marine organisms under natural and survey vessel affected conditions. In this paper a new concept for a remote operated acoustic data logger is presented which can give an online presentation of the acoustic information. The system is built on the new Simrad General Purpose Transceiver (GPT) operated through a standard windows NT interface. The technical specification and operational features will be presented and exemplified with experience from the first sea trials. The system is designed to meet demands from other applications, e.g., a drop sonde and towed underwater function, and future development of the concept is discussed.
A microphone without a membrane
Authors: Sennheiser Joerg
Published: July 2002

In all microphones used today, the component which affects the quality most is the membrane. The frequency response and the signal-to-noise ratio of a microphone depend on the best possible coupling of this mechanical component to the acoustic sound field and on its oscillatory behavior. In the mass production of membranes within narrow tolerances, properties such as geometry, mass, stiffness, native modes, etc. have a great effect on the quality and are often problematic. There is therefore great interest in finding a working principle for microphones which does not require a membrane. A possible way out is to utilize the well-known thermal anemometer principle: An electrically heated wire is cooled down in an air flow. The resistance change in the wire is a measure of the air speed and thus of the velocity. If the heat capacity of the heated sensor is reduced by micromechanical techniques, this principle can also be used to measure the acoustical volume velocity at higher frequencies. The expected limits and first readings are shown via the models realized.
Optical measuring and calibration techniques for microphones
Authors: Behler Gottfried K. , Vorländer Michael
Published: July 2002

Microphone measurement techniques are normally based on acoustical excitation and electrical derivation of either the voltage or the current output of the transducer. With condenser microphones, electrical excitation can be used as well. Calibration of microphones based on acoustical excitation therefore is quite time consuming, and sensitive for several errors originating in inaccurate sound-field properties and environment conditions. Measurements of the microphone membrane velocity with optical means like the laser Doppler vibrometer technique offer an accurate, touchless method to derive the local movement at almost each point of the membrane. Excitation of the membrane can be accomplished by direct feeding of the transducer (due to its reversible construction), which provides highest accuracy for the derivation of either current or voltage at the terminals of the transducer. From the knowledge of the vibration characteristics and the excitation properties, the ‘‘differential transducer sensitivity’’ defined at each surface point can be derived. With numerical methods, like the discrete solution of the Helmholtz equation or BEM, the radiated sound can be calculated, and finally the overall microphone radiation pattern and sensitivity can be calculated. The described techniques were applied to different types of microphones. The measurement and calculation results will be shown.
Acoustic loading exercises in singing training
Authors: Titze Ingo R.
Published: July 2002

Singing teachers have devised vocal exercises that present variable acoustic loads to the sound source. These exercises consist of lip and tongue trills, nasal consonants, and vowels with small mouth openings (like /u/). Some pedagogues and clinicians have advocated phonating into long and narrow tubes. Since none of these exercises produce either a strong or aesthetically pleasing acoustic output in and of themselves, their merit must lie in facilitating the respiratory and phonatory systems. Recent Hopf bifurcation analysis has shown that inertive acoustic loads, resulting from deliberate constricting and lengthening of some portion of the vocal tract, can lower the oscillation threshold pressure, thereby facilitating the ease of vocal onset. Resistive acoustic loads, either upstream or downstream of the vocal folds, have a less beneficial effect on oscillation threshold pressure. [Work supported by Grant No. P60 DC00976 from the National Institute on Deafness and Other Communication Disorders.]
Measurement of performance response differences when striking different areas of a marimba bar
Authors: Larkin Barry J. , Roberts Ronald A.
Published: July 2002

Over the past 40 years, general percussion education manuals have stated that the best place to activate a marimba or xylophone bar is to strike it in the exact center. Over the years, this has become known as the ‘‘best spot,’’ and percussion students have been blindly advised to strive always to perform in this manner. Recent re-publications of mallet methods from the early part of the 20th century show that it was a common practice to perform on the end of the accidental bars for speed and superior sound. Also, leading classical marimba performer, Leigh Howard Stevens, has stated his preference for this type of execution. This paper will present spectral analysis results from measurements made at nine different locations on a Great F sharp marimba bar. The results will show the diverse types of responses generated by location changes of as little as an inch.
Experience of students of composition and instrument construction at the National School of Music of UNAM in the auditory normalization of timbre with age and with exposure to different instruments
Authors: Castro-Sierra Eduardo , Gómez-Gama Martha , Zenker Miguel , Pérez-Ruiz Santiago J.
Published: July 2002

A study of tonal memory using as stimuli synthesized harmonic sounds with a quasi-square-wave spectrum has been carried out at ENM. The subjects were young boys and girls (age span=8:1–15:6), starting their musical education at the Cycle for Musical Initiation, and young adults of either sex (age span=13:11−42), with experience of from 1 to 10 years in singing, piano, violin, clarinet, flute, trumpet, saxophone, guitar, percussion, or composition at the regular courses of ENM. While the children were adept in retaining in memory tonal changes of a major or a minor second using the synthesized stimuli, the older students showed much greater difficulty with this test, particularly those whose instrument had a timbre closer to that of the square wave (clarinet or saxophone), or who were undertaking studies of composition. Students of composition or instrument construction at the Instrument Construction Workshop of ENM have embarked on a further analysis of auditory normalization of timbres of diverse synthesized or natural sounds of instruments. It is expected that such an experience will allow the students to better conceptualize different timbre combinations as well as the timbre of individual musical instruments.
Today’s railfreight noise situation
Authors: Hecht Markus
Published: July 2002

In recent years railway freight wagons have tended toward a gradual increase in noise radiation. The reasons have been the transition from slide to roller-axle bearings, more welded joints instead of bolted or riveted ones, the increased number of four-axled wagons instead of two-axled ones, and higher medium speeds. Meanwhile, the competitors on the road and in the air have made substantial progress in noise reduction. The first major step toward quieter trains in railway freight was the decision of the German and the Swiss freight operators DB-Cargo and SBB-Cargo to replace cast iron brake blocks by composite brake blocks. This was done in the US decades ago, but the stopping distances in Europe are much shorter, about one-quarter of those in the US. Therefore, the braking forces are much higher, which results in higher thermal thrust on the running surfaces. A special wheel shape had to be developed to withstand this impact and to reduce the risk of broken wheels. Noise emission from European modern standard freight cars must be reduced by 25 dB. Only then can the existing perception limits at densely used freight lines be reached at night. Measures to achieve this goal will be presented.
Cause research and assessment of the acoustic behavior of rail vehicles using SFE AKUSRAIL
Authors: Zimmer Hans
Published: July 2002

Many efforts have been made in the past to further develop knowledge of both the interior acoustics and sound emission of rail vehicles using numerical methods. One major advantage of numerical methods is the possibility of forecasting the acoustic behavior (without prototyping) and hence influencing the design of the vehicles, also in terms of acoustic criteria. Another advantage of this ‘‘virtual prototyping’’ is that it is possible to determine the contribution of individual components to the sound level within the frequency range analyzed, i.e., to analyze the causes of acoustic behavior. However, strict criteria must be met to satisfy the claim of a useable forecast of acoustic behavior. It must be possible to simulate important acoustic phenomena—such as structure/air coupling, absorption, reflection, emission, etc. The numerical model must be sensitive enough to permit the forecasting of trends. The interaction of rail and vehicle must be taken into consideration as a crucial precondition for identifying and analyzing acoustically critical frequencies. Another decisive criterion, however, is that it must be possible to assess not only the ‘‘actual’’ acoustic state, but also the underlying causes. This includes, for instance, knowledge of structure-borne sound paths and the structure and coupling situation.
Roughness growth on tread-braked railway wheels: e
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Authors: Nilsson Anders , Ross Urmas
Published: July 2002

Rail and wheel surface irregularities are the main source of rolling noise. One of the main causes for wheel surface roughness is tread braking. Wheels braked by the most common tread brakes—cast iron blocks—have high roughness levels and, as usual, corrugation with wavelength of 30–100 mm. The Brite Euram research project EUROSABOT deals with new solutions for tread brakes, which would lower the roughness level, prevent the buildup of periodic roughness, and consequently reduce the rolling noise, and at the same time would follow the safety requirements of railway operation. Field measurements, discussed in detail by Stefan Buehler in a companion presentation, in parallel with laboratory studies has distinguished that the interaction of thermo-elastic and vibrational instabilities is the main reason for corrugation generation. Computer simulations describing simultaneously thermodynamical, structural, and acoustic processes of the braking operation have proved to be unpractical for parametric studies and, instead, several simplified models are used. These models have shown that the main parameters affecting roughness growth are Young’s modulus and coefficient of thermal expansion of the block material and damping of the system.