bluuebunny/crossref_metadata_2025 huggingface dataset:
High-intensity ultrasonic spectra in water and brain tissue
Authors: Jambrosic Kristian , Ivancevic Bojan , Zoric Igor
Published: July 2002

A common method of brain tumour surgery is using CUSA (cavitation ultrasonic surgical aspirator). While using CUSA, there is a strong focusing of the ultrasonic field within the cranium. This can cause damage to healthy brain tissue. Therefore, the ultrasonic field intensity at the critical spots must be measured. These measurements cannot be made in vivo, thus raising the need for a substitute model. In this paper, frequency spectra of the CUSA probe was measured at its working frequency of 25 kHz in an oval container (simulating the cranium). The measurement was made at few field intensities, K hydrophone in the container, and the B&K microphone outside it. The medium temperature has been controlled. Calf brain tissue, physiologically similar to human, was used as a substitute model. The same measurement was repeated in water to examine the spectrum difference in high- and low-attenuation media. During the measurement, the occurrence of the cavitation was observed by spectra changes.
Biased lead zirconate titanate as a high-power transduction material
Authors: Powers James M. , Moffett Mark B. , Jevnager Michael D. , Gilardi Stephen S.
Published: July 2002

High-power, underwater transducers using polarized piezoelectric ceramic material are usually limited in drive amplitude so that depolarization does not occur, but application of a dc bias field in the polarization direction allows the use of higher ac drive fields. To demonstrate the feasibility of a biased operation as a means of achieving higher power, a thin-walled, spherical-shell transducer was constructed of Channel 5800 and tested in NUWC’s Acoustic Pressure Tank Facility. The transducer was successfully driven to 33 V/mil (1.3 MV/m) rms with an accompanying bias field of 31 V/mil (1.2 MV/m) and a hydrostatic pressure of 1400 psig (9.65 MPa). The source level was 206 dB//mPa-m at 61 kHz, corresponding to a 10-dB improvement over the usual Navy standard unbiased drive limit of 10 V/mil (0.4 MV/m) rms. [Work supported by ONR.]
Recent low-frequency sonar transducers and arrays
Authors: Boucher Didier
Published: July 2002

Research in the field of high-power, low-frequency underwater transducers has been sustained by advances in submarine noise reduction and the promise of long-range detection at low frequencies. The earliest idea came up to use the already available passive towed array sonars as receive arrays in conjunction with additionally towed sound sources. The frequency of the active adjunct idea was of course given by the existing passive arrays. Very promising trials at sea, even in the shallow water environment, soon led to the consideration of full low-frequency active sonar systems (LFASS). Those have now been investigated for nearly two decades. With the emergence of new high-power transduction materials and the rapid evolution of fast computers and powerful modeling techniques, this research field has been fertile ground for transduction innovation. Specifically, high-power transducer developments include performance improvements to conventional piezoelectric ceramic projectors, progress in the exploitation of new active driver materials, and advances in transduction design techniques. While high power, low frequency, wide bandwidth, and high efficiency remain the primary performance objectives behind these developments; low weight, small size as well as medium to deep submergence are desirable physical attributes that require equal attention by the designer. This paper emphasizes the key characteristics of two sea-proven concepts for LFASS: Janus Helmholtz transducers and multihead arrays (international patents). The attention will be particularly focused on measured values of the sound sources. Finally, this paper addresses future trends in terms of sonar array designs for LFASS.
Application of numerical techniques in teaching technical acoustics
Authors: Augusztinovicz Fulop
Published: July 2002

Due to their highly extended visualizing capabilities, numerical methods are very apt at demonstrating acoustic effects and phenomena. Finite-element methods (FEM) are mainly used to describe the sound fields in closed spaces, and boundary element methods (BEM) are well suited to predict radiation from vibrating bodies. Coupled systems can also be analyzed by some hybrid approaches. The paper shows applications of these tools in undergraduate and postgraduate courses of technical acoustics. Starting from the well-known concentrated parameter approach, a more realistic description of basic acoustic elements and systems consisting of acoustic masses, acoustic compliances, Helmholtz resonators, waveguides, etc. can be presented. Based on these examples, the dynamic behavior of acoustic systems such as the existence of standing waves, eigenmodes, and resonance effects can be visualized and understood. Radiation problems can be solved for elementary acoustic sources, demonstrating the basic rules of fluid–structure interaction phenomena. The most advantageous feature of numerical—FEM/BEM—calculation and visualization techniques is its capabilty to handle a large amount of data. Problems can be approached in three dimensions, both in time and frequency domain. Apart from the application areas, the lecture addresses the necessary hardware and software tools to prepare these sorts of demonstration, too.
A library of visualization and auralization examples to support understanding
Authors: Blomqvist Jens , Kihlman Tor , Kropp Wolfgang
Published: July 2002

In a project financed by the Swedish Council for renewal education a library has been produced containing visualization and auralization examples concerning physical phenomena in acoustics. The main goal of the library is to support the development of a thorough understanding of basic acoustic phenomena. The work was supported by studies in the form of interviews and questionnaires to understand—What are the genuine problems students have in acoustics, mathematics, and physics? How do the students’ and teachers’ view of acoustics, physics, and mathematics differ? What are suitable measures to achieve a common basis between teacher and student for a successful communication? During the project it became obvious such a library has to be developed together with students from our courses. The library has to be adapted to the students needs and to their mental picture of physics, mathematics, and acoustics. In order to achieve this, a group of students was asked to take part in the project. They directed both content and layout of the library from their own perspective. The results of this work will be presented together with typical examples from the library.
Teaching boundary element methods in acoustics
Authors: Gaul Lothar , Wagner Marcus , Wenzel Wolfgang
Published: July 2002

The boundary element method (BEM) provides a powerful tool for the numerical solution of sound fields, because (1) it maps the domain equation on the boundary and thus reduces the problem dimension by one, (2) it provides a nonreflecting boundary on an arbitrary radiating surface, and (3) pressure, velocity, and intensity can be calculated at every field point from the boundary data. The author’s concept of teaching BEM in a graduate engineering course is to provide a link with subjects taught in undergraduate engineering mechanics such as rod vibrations, reciprocity principle, influence functions, and finite elements (FE). These subjects are generalized to the field equations of acoustics and elastodynamics. The associated integral equations are derived from the dynamic reciprocity principle. Their kernels are fundamental solutions which turn out to be influence functions of a full space. Boundary formulations of rods and beams recall well-known FE matrices. The fluid–structure interaction of interior and exterior acoustics is formulated by adopting the unsymmetric allocation approach of BEM and a symmetric hybrid BEM which is based on a multifield varational principle. Waves and vibrations in a flexible pipe and the sound radiation from a vibrating tire are treated as examples.
An adaptive, active noise reduction system in closed space
Authors: Golas Andrzej , Ciesielka Wojciech
Published: July 2002

In the demonstration an adaptive system of noise reduction is presented. Both the acoustic signals phase compensation and the adaptive digital filtering theory are employed. An incident acoustic wave is identified and a compensation signal is generated using a finite impulse response filter which coefficients are calculated in real time employing an LMS algorithm. The algorithm is implemented on a dSpace card with a signal processor (TMS320C31 Texas Instrument). In the receiving point (artificial head, real listener, receiving transducer) the disturbing signal (white noise filtered in range 50–250 Hz) is reduced by an active adaptive system. The damping coefficient that is obtained in the system varies from 10 to 20 dB. The experiment can be conducted and observed in any interior.
The use of computers in acoustical engineering: A computer-aided electroacoustics course
Authors: Gomez Alfageme Juan J. , Recuero Manuel , Sanchez Jose L.
Published: July 2002

The teaching of electroacoustics in the audiovisual engineering curriculum has sustained important changes during the last 4 years. The increasing use of computers in the classroom has allowed a new approach to this subject, which needs vast knowledge in physiscs, mathematics, and signal analysis. In the last years our department has developed a computer-aided course in electroacoustics. This course is made in the Windows operating system and programmed in Matlab, for its extended use in engineering departments all over the world. At the moment, the implemented modules are electroacoustic transducer analysis, direct radiation loudspeaker systems, array directivity, sound intensity, electrodynamic loudspeaker distortion, and speech index transmission calculus. All the modules have the same structure using the Matlab object-oriented user interface facilities and use the Matlab resources for graphical representation. Each module makes transducer behavior simulation based on equivalent circuits. The student does not need to have a great deal of knowledge in programming to work with the modules. The structure of different modules will be explained and the results with students in our curriculum will be provided.
Courses on acoustics: What is needed?
Authors: von Estorff Otto
Published: July 2002

The constant reduction of allowable noise levels and consumer demand for a better acoustic performance of technical systems put a tremendous pressure on manufacturers. As a consequence, engineers who know about acoustics, and the measures to improve it, are currently looking at numerous job openings. In fact, the industry is seeking intensively in this field. The present paper is related to the contents of courses on acoustics at the university as well as in industry. Its focus is to discuss the question ‘‘What is needed?’’ in order to meet the requirements of future acoustic challenges and, of course, the expectations of the student’s future employer. Based on the author’s experience and a little questionnaire, a possible course content is setup and discussed.
On the function of the violin bridge
Authors: Jansson Erik V. , Niewczyk Benedykt K.
Published: July 2002

In measured frequency responses of violins of soloist quality a broad maximum is found at about 2.5 kHz which has been called the bridge hill as the violin bridge on rigid support has its major in-plane resonant frequency nearby [E. V. Jansson, Acust./acta acust. 83, 337–341 (1997)]. Experiments show, however, that the hill rather lies in combined properties of the bridge and of the top plate. It can be influenced both by the properties of the top plate between the f-holes but also by the bridge. The maximum is efficiently excited in the normal main direction of bowing. The force along the bow is transformed into a force couple at the bridge feet which excites the bridge and body. The broad maximum involves resonant motion which is not confined to the bridge alone. [Work supported by KTH, Swedish Natural Science Research Council, Swedish Institute, and Wenner-Gren Center Foundations for Scientific Research.]
An investigation of vibrational and acoustical properties of the violin using MLS and optical holography
Authors: Morset Lars H.
Published: July 2002

In this investigation a multi-channel MLS (maximum length sequence) measurement system is used for vibrational and acoustical measurements on the violin. MLS measurements are useful since both phase and magnitude can be measured with great accuracy. In these measurements, both the phase and magnitude influence of the transducers are corrected. Using this tool, the input admittance, the velocity at the feet of the bridge, and the sound pressure at the far field of the violin are measured simultaneously. These parameters give important information about the vibrational and acoustical properties of the violin. A more extensive investigation is also performed. The sound level pressure and phase are measured in several equally spaced positions in the far field of the violin situated in an anechoic chamber. Radiation patterns are found for each mode and the radiated power is calculated. Optical holography is then used to measure the vibrational pattern for the most important modes of the violin body. These acoustical and vibrational measurements are correlated to determine which modes are most important to the radiated sound power.
Characteristics of the noise found in day-care centers
Authors: Michel Picard , Chantal Boudreau
Published: July 2002

Today’s day-care centers are apparently extremely noisy environments. Given the paucity of data on this particular issue, noise measurements were collected for 25 sites with groups of 1.5- to 5-year-old children (N=4−16). Noise levels associated with three types of children’s activities were studied, namely: unstructured activities of the children; structured ones (e.g., story telling); and lunch time. In addition, measurements were collected for the unoccupied case. A-weighted levels (10–90 percentiles, average levels, LAeq), and third octave levels will be presented for each measurement condition. Statistical analysis to model the noise in all conditions will also be reported. Of special concern, LAeq (10 min) indicated frequent readings in excess of 85 dB for all measurement conditions except the unoccupied case. This suggests that children may be at risk of noise-induced hearing loss dependent on the length of stay in the day-care centers. They would only be able to initiate or sustain any meaningful verbal communication activity by using extreme vocal effort to speak loud enough to raise their voice above the din. Obviously, this is quite an alarming situation.
Occupational noise in Spanish textile industries
Authors: Garcia Amando , Garrigues Jose V. , Garcia Ana M.
Published: July 2002

This study has been carried out in 20 small-, medium-, and large-sized textile industries located in the provinces of Valencia and Alicante (Spain). A series of noise-level measurements has been carried out in order to evaluate the daily noise exposure levels (Lep,d) for a wide variety of workplaces. About 33% of the workers present during the measurements were exposed to daily mean A-weighted noise levels Lep,d higher than 85 dB. It has been noticed that personal protection against noise was used only by a minority of these workers. Available data from routine medical examinations have been also analyzed. The observed hearing loss is related to noise exposure level in the corresponding workplaces. In particular, about 48% of the workers exposed to Lep,d levels higher than 85 dB showed acoustic trauma (hearing losses higher than 40 dB for the 4000-Hz frequency in either ear).
Artificial heads versus miniature microphones for measuring active noise-reduction hearing protector performance
Authors: Fedtke Thomas , Richter Utz
Published: July 2002

The results of sound attenuation measurements on hearing protectors with head and torso simulators (HATSs) are quite encouraging if suitable corrections for the bone conduction and the occlusion effect are applied. Yet, the real-ear-at-threshold (REAT) method is still the standardized and most often used procedure to get appropriate measures for the protection provided by passive earmuffs and passive earplugs. With the development of active noise reduction (ANR) hearing protectors, however, the REAT method fails because of the background noise generated by the electronic components of the devices. Furthermore, the limits and the efficiency of such devices cannot be tested with sound-pressure levels near the hearing threshold. Thus there is an urgent need for using either HATSs or probe microphones on humans for the measurement of the ANR performance of a hearing protection device. Attempts to determine the effective attenuation with a combination of REAT and artificial head measurements will be compared with the results of microphone in real ear (MIRE) measurements for several commercially available ANR systems, including noise cancelling headphones. Aspects of validity and uncertainty of the various procedures, as well as practical relevance and ethical considerations, will be discussed.
Acoustic emission signals from the sol–gel transition of tetraethoxysilan (TEOS) and sodium silicate solutions
Authors: Rzeszotarska Jadwiga , Rejmund Feliks , Ranachowski Jerzy
Published: July 2002

Sol–gel processes have been used extensively to prepare oxide materials (glasses and ceramics). The acoustic emission (AE) method with broadband piezoelectric transducer (up to 500 kHz) was used to control the evolution of slow gel formation from organic TEOS and fast formations from sodium silicate as the precursors of silica. The obtained dependencies of the AE count rate and rms values versus time characterize the dynamic of the cross-linking and desolvation processes under ambient conditions. The frequency distribution analysis of individual power spectra provides an activity estimation of integral parts of processes. In sodium silicate solutions the spectra of low frequency (17–70 kHz) were mainly observed. On the other hand, the most representative spectra from the gelation of TEOS display at three basic ranges; 16–20 kHz, 200–230 kHz, and 400–470 kHz. The relative intensities of spectra in these ranges were changed considerably during the gel formation. [Work partly supported by KBN Grant No. T 07B 047 15.]
Computer models of the auditory periphery
Authors: Meddis Ray
Published: July 2002

Enough is now known about the auditory periphery to allow the development of useful computational models. These accept an arbitrary acoustic stimulus and predict the response of auditory nerve fibers. Current models are largely based on linear filters representing the mechanical frequency selectivity of the cochlear. Nonlinear models would be more appropriate if they could be developed in a form that replicated physiological measurements. This paper describes one approach that combines a nonlinear model of frequency selectivity with a revised inner hair cell model. The nonlinear filters are evaluated using published data collected at three sites on the basilar membrane. They behave nonlinearly at frequencies close to best frequency and linearly at remote frequencies. They show appropriate impulse responses and patterns of two-tone suppression. The hair cell model incorporates a simple account of the receptor potential and retains a previously developed reservoir model of synaptic flow. An innovation includes a model of the discrete release of transmitter quanta that explicitly assumes a presynaptic origin for the stochastic nature of the auditory nerve response. The combined model gives a useful account of the differences between low-, medium-, and high-spontaneous rate fibers that is consistent with earlier physiological speculation.
Recent advances in models of binaural detection and localization
Authors: Colburn H. Steven
Published: July 2002

There has been a significant increase in the mathematical modeling of binaural phenomena in the past couple of years by a number of research groups. Available review chapters [e.g., Colburn, in Auditory Computation (Springer-Verlag, New York, 1996); Stern and Trahiotis, in Handbook of Perception (Academic, New York, 1996)] do not include this new material. Recent advances were stimulated in part by the availability of more powerful computational tools and in part by recent data that include more complex stimulus situations such as noise waveforms with interaural time differences but no interaural intensity differences and vice versa [van de Par and Kohlrausch, J. Acoust. Soc. Am. (1998)]. This presentation reviews recent developments in binaural modeling with particular attention given to models of binaural detection and sound localization. Both pink-box models, which incorporate available information from auditory physiology, and black-box models are considered. [Work supported by NIDCD Grant No. R01 DC00100.]
An exploration of coherence masking protection in a spectral profile discrimination task
Authors: Bailey Peter J.
Published: July 2002

Gordon [P. C. Gordon, Percept. Psychophys. 59, 232–242 (1997)] has reported a form of masking protection, in which identification thresholds for noise-masked target signals are lower when a nondistinctive cosignal is present. Gordon suggested that target signals are protected from masking as a result of their perceptual coherence with the cosignal. The experiment reported here sought to show whether cosignals enhance performance in a task involving discrimination of spectral profiles. The target signals were increments in the level of components in a complex tone (harmonics 1−7, F0=125 Hz), with the third harmonic incremented in one trial interval and the fifth harmonic in the other. Discrimination threshold increments were measured for four well-practiced listeners in three conditions. In one condition, the complex tone was presented alone, without a cosignal. In the two cosignal conditions, harmonics 8–40 were added, with either a vowellike or a flat spectral profile. Average threshold increments were similar in the three conditions, and no listener showed reliably smaller threshold increments when a cosignal was present. Cosignals may confer protection from the effects of noise masking, but they do not necessarily enhance performance in other auditory tasks. [Work supported by UK MRC.]
An overview of the inverse problem for sound reconstruction in interior spaces
Authors: Williams Earl G.
Published: July 2002

Calculation of the sound field between noise sources and a closed two-dimensional measurement surface is an inverse propagation problem. A powerful and popular method for the reconstruction of sound fields in this space is based on boundary element methods applied to the Helmholtz integral equation. Unfortunately, due to the numerical nature of these solutions, the physics associated with backtracking (backpropagating) the field from the measurement surface to the source surfaces is hidden. Analytic models are used to view this backpropagation as a deconvolution operation on the measured field. The nature of this deconvolving function (called the inverse propagator) is examined; it is a singular function, it is spatially local, and it unsoothes the measured pressure field. An understanding of the nature of the inverse propagator is critical to successful solution of the inverse problem by BEM methods (especially the regularization process), as well as the sucessful development of three-dimensional sound projection methods for sound synthesis. Furthermore, this understanding leads to new insights into the solution of the inverse problem. [Work supported by ONR and NASA.]
Problem of reconstructing the full vibration field from limited data
Authors: Bobrovnitskii Yuri I.
Published: July 2002

The problem of expanding the vibration field from the measured points to unmeasured (inaccessible) points is posed and solved. This field reconstruction problem is one of the inversion problems of structural dynamics and acoustics. The procedure of handling it is the following: the full vibration field under study is described by a mathematical model containing a certain number of parameters (e.g., a modal model with the mode amplitudes as the model parameters); the values of these parameters are identified from the limited data; the field of the structure is then computed using this model. The main result of the present submission is that there exists the best mathematical model (containing a finite number of parameters) which gives the minimal error of describing the field in unmeasured points. Another result presented is that there also exists the optimal volume of the measurement data. The paper contains a general theory, equations relating the error to the measurement accuracy and the data volume, and some computer simulation results. Discussion of how to choose the best field mode using only the measurement data is also presented.
Iterative feedback control design for active vibration control
Authors: Veres Sandor M.
Published: July 2002

Model-based robust feedback control design is widespread: a parametric model is identified with some measure of the unmodeled dynamics present in the most advanced approaches. In off-line design, often open-loop experiments are used which can fail to provide satisfactory models because the identification criterion does not serve the aims of feedback robustness. Another frequent problem is possible large bias of the parametric part because of wrong model orders. In the best case large bias comes with large bounds of the unmodeled dynamics in the H-infinity or the L-1 norms. The conclusion is that the use of a plant model is the source of considerable problems while it has so far seemed a natural approach to controller design. In this paper a new ‘‘model-free’’ approach will be outlined for designing feedback vibration controllers. This will be based on a direct search using a sequence of iterative experiments. The experiments will alternate between two types: with and without the vibration source being active. The iterations of the controller parameters allowed by the sequence of experiments will lead to a sequence of improvements of control performance. This will be verified experimentally with reduction of sound-induced low-frequency vibration of a glass plate.
Predictive feedback and feedforward control for systems with unknown disturbances
Authors: Juang Jer-Nan , Eure Kenneth
Published: July 2002

This paper presents the theory and implementation of a hybrid controller for general linear systems by incorporating a feedforward path in the feedback control. The generalized predictive control is extended to include a feedfoward path in the multi-input multi-output cases. There are cases in acoustic-induced vibration where the disturbance signal is not available to be used by the hybrid controller, but a disturbance model is available. In this case the disturbance model may be used in the feedback controller to enhance performance. In practice, however, neither the disturbance signal nor the disturbance model is available. This paper presents the theory of identifying and incorporating the noise model into the feedback controller. Implementations are performed on a test plant and regulation improvements over the case where no noise model is used are demonstrated.
Digital filters used in the feedback loop of an ANR earplug
Authors: Zimpfer Veronique , Buck Karl , Gache Nicole
Published: July 2002

Commercially available active hearing protectors are usually implemented as ear muffs with an analog driven feedback loop. The active attenuation bandwidth of these devices is limited to about 800 Hz. Moreover, it is not possible to adapt the analog system for specific noise events or to the user’s morphology. Active earplugs allow one to extend the bandwidth of the active protection to frequencies higher than 2 kHz. A digital feedback system has been implemented for active earplugs, in order to allow an adaptation to different noises and users. As the acoustic delays are too short to obtain a causal system when using FIR filters, the filters have been implemented as IIR type filters. The paper presents numerical simulations of different possible algorithms, used for the cascaded IIR filter of the ANR system. It shows how numerical errors are propagated in cascaded Bi-Quad implementations, and how an optimum signal-to-noise ratio may be obtained. The numerical predictions and experimental results will be presented and compared. As these filters will be used in a closed-loop feedback system, a numerical simulation of the behavior of the closed-loop system, including the electroacoustic transfer function of the ear plug, will be presented.
Methods for subjective assessment of service quality: Milestones to customer-centered quality engineering
Authors: Klaus Harald
Published: July 2002

In a liberalized market, the opinion of the customer about the quality of an entire service is a key issue for success. In the telecommunication market, technology evolves very rapidly to intelligent and integrated applications. Therefore, new views and methods for the assessment of quality aspects have to be developed in time. The first part of the presentation gives an overview of subjective assessment methods for speech as well as for audio and video applications. Furthermore, practical aspects of quality testing are discussed. In the second part, issues and trends for customer-centered quality engineering are outlined.
Blind separation of speech signals in the frequency domain
Authors: Anemüller Jörn
Published: July 2002

Blind source separation algorithms try to reconstruct original signals, e.g., multiple speakers, from knowledge of their superpositions, using solely the mutual statistical independence of the source signals as criterion for separation. However, application of existing algorithms to acoustic superpositions is limited by the complex nature of room transfer functions and by the use of nonlinear computations. Expanding on our previous work, we linearize the acoustic source separation problem by moving to the frequency domain [Anemüller and Gramss, DAGA (1998)] and eliminate the need for computation of nonlinear functions by using a multiple decorrelation approach. Thus, our algorithm exploits the highly redundant structure of, e.g., speech signals in order to reduce the computational cost. Results of separation experiments are presented.
Phonetic recognition of Arabic figures based on formant analysis via artificial neural network
Authors: El Mallawany A. I. , Abdallah M. I. , Abd El Gawad M. , Sedik H. S.
Published: July 2002

This paper describes a speaker recognition for dependent and independent speaker systems based on the formant analysis technique. Also, this system is used for Arabic figure recognition. Linear predictive code (LPC) is applied for extracting the formants of the Arabic digits from 0–9. After training and selecting the proper network structure, the recognition accuracy is found to be 98.8% for the dependent system while it is 78.8% for the independent case. Recognition accuracy for independent speaker is improved to 95.5% when grouping the Arabic digits into three groups.
Virtual concerts in virtual spaces—in real time
Authors: Lokki Tapio , Savioja Lauri , Hiipakka Jarmo , Hanninen Rami , Pulkki Ville , Vaananen Riitta , Huopaniemi Jyri , Ilmonen Tommi , Takala Tapio
Published: July 2002

The DIVA system is an experimental interactive real-time virtual environment with synchronized sound and animation components. The system provides real-time automatic character animation and visualization, dynamic behavior control of virtual actors, interaction through motion analysis, sound generation with physical models of musical instruments, and three-dimensional sound auralization. The combined effect of 3-D visual and acoustic elements creates stronger immersion than would be possible with either alone. As a demonstration, a virtual band with four artificial musicians has been implemented. The user interacts with the virtual musicians by showing the tempo with a baton, like real conductors do. The animated band follows the gestures of the conductor and another user controls the viewpoint of the audience. Due to the real-time acoustic modeling and sound rendering, both users hear the auralized music in a real concert hall.
Sonification of range information for 3-D space perception
Authors: Milios Evangelos E. , Kapralos Bill , Stergiopoulos Sotirios
Published: July 2002

A device is presented that allows 3-D space perception by sonification of range information obtained via a point laser range sensor. The laser range sensor is worn by the user, who scans space by pointing the laser beam in different directions. The resulting stream of range measurements is then converted to an auditory signal whose frequency or amplitude varies with the range. This device differs from existing navigation aids for the visually impaired. Such devices use sonar ranging whose primary purpose is to detect obstacles for navigation, a task to which sonar is well suited due to its wide beam width. In contrast, the purpose of this device is to allow users to perceive the details of 3-D space that surrounds them, a task to which sonar is ill suited, due to artifacts generated by multiple reflections and to limited range. Preliminary trials demonstrate that the user is able to detect corners and depth discontinuities accurately with ease and to perceive the size of the surrounding space.
Development of underwater acoustics in China and L. M. Brekhovskikh’s early activities in China
Authors: Guan Dinghua
Published: July 2002

In this paper the early activities and influence of L. M. Brekhovskikh in the development of underwater acoustics in China are reviewed. The activity of Brekhovskikh as an invited underwater acoustics expert in the formulation of the 12-year plan of S&T development of China in 1956 is described. This paper reviews the early development of underwater acoustics in China, including the visit of Chinese experts to the Institute of Acoustics in Russia, the first joint Sino–Russian experiment in the South China Sea, and the training of young Chinese scientists. The monograph ‘‘Waves in Layered Media’’ has had a large influence on Chinese underwater acoustics. This paper also reviews part of the research on underwater acoustics in China. The ray-mode theory was developed, improved, and used in propagation, reverberation, and noise field modeling in shallow water. The relationship between rays and normal modes was analyzed and proved. The influence of sea bottom in shallow water was expressed in terms of angular dependence of bottom reflection loss. Wave theory for long-range reverberation was developed. MFP, MMP, and other methods of positioning and inversion was developed. Internal waves and their influence on sound propagation were studied.
The tangent plane approximation and related approaches in rough surface scattering theory
Authors: Voronovich Alexander G.
Published: July 2002

The tangent plane approximation (TPA) was introduced into the theory of wave scattering from rough surfaces by L. M. Brekhovskikh in 1952. This method corresponds to the semi-classical (WKB) approximation and is the second ‘‘classical’’ method (after the small perturbation method introduced into this realm by Rayleigh in 1907) that is widely used for the solution of wave propagation problems. Development of this approach significantly increased the capabilities to solve practical tasks and produced a long-lasting effect on other theoretical developments. For example, the operator expansion method or the small-slope approximation developed recently could be considered as a generalization of the TPA. The first part of the talk will be devoted to a demonstration of the relationship between TPA and different ‘‘nonclassical’’ approaches developed later. In many but not all instances, TPA could be obtained as a zeroth-order iteration of an appropriate integral equation for surface sources (for this reason TPA is also called the Kirchhoff approximation). However, some other approaches to the approximate solution of this integral equation based on smoothness of the rough surface are also possible. Corresponding enhancements of the TPA will be considered also for the case of impedance surface.
Leonid M. Brekhovskikh and his scientific school
Authors: Lysanov Yury P.
Published: July 2002

The following items will be discussed: The first meeting with L. M. Brekhovskikh (1951), the Acoustics Laboratory at the Lebedev Physics Institute, L. M. Brekhovskikh and the organization of the Acoustics Institute, a collection of young scientists and post-graduates from many Russian cities for the new Institute, a post-graduate studentship, new problems in ocean acoustics, wave scattering by rough surfaces, Brekhovskikh’s approximation (the tangent plane method), the growth of international relations, the Second and Fourth International Congresses on Acoustics (USA, 1956; FRG, 1959), all-union Acoustical Conference in Moscow (1958), the great role of L. M. Brekhovskikh in educating young scientists in ocean acoustics and oceanology (lectures at the Moscow State University and the Moscow Institute of Physics and Technology), scientific seminars under the supervision of L. M. Brekhovskikh (Sukhumi, Zvenigorid, Moscow), and Brekhovskikh’s books and papers and their fundamental contribution to ocean acoustics and oceanology.
Room acoustic simulation in the planning of concert halls and opera houses
Authors: Mommertz Eckard
Published: July 2002

In room acoustic planning and design of concert halls and opera houses, computerized prediction of sound transmission gains an increasing significance. The room acoustic simulation may give a more reliable prediction of the room acoustic conditions to be expected. Furthermore, the visualized and auralized results have a positive effect on the communication between acoustician and architect. In this paper, the requirements as well as present possibilities and limitations of room acoustic prediction in practice are described by means of exemplary current building projects. Some important aspects such as coupled volumes, diffuse wall reflections, curved surfaces, or grazing sound transmission above the audience are discussed.
Scale model experiences in concert and opera hall design
Authors: Hidaka Takayuki
Published: July 2002

Scale model experiments were executed for two concert halls and two opera houses which recently opened in Tokyo. Based on the geometrical acoustic analysis by using CAD model, effective room shapes were studied so as to minimize the extra labor at the scale model test. Then, 1:10 scale models made of wood and proper acoustical materials were constructed. Up-to-date digital signal processing technique was applied in order to improve the accuracy of the experiment. Also, several new ideas concerning the sound source, receivers, scale modeled seatings, and how to calibrate the air absorption were introduced, intended for the precise prediction of the room acoustical parameters. Thus, monaural parameters RT, EDT, C80, D, G, and ST, and binaural parameter IACC were measured in the scale models, and they were compared with those measured in real halls. The accuracy and problems of this method were discussed with the acoustical data found here.
Links between concert hall geometry, objective parameters, and sound quality
Authors: Essert Robert
Published: July 2002

For decades, the design of concert halls was driven by considerations of time history alone (T60, C80, ITDG), and as a result, little importance was attached to room geometry. The subjective importance of binaural dissimilarity has been a strong, though often simplistic, influence on recent designs. While listening experience has shown which fundamental room forms sound better than others, computer modeling and statistical analysis have enabled systematic investigation of the degree to which the geometry affects the sound. Using simple parametric models, this study will investigate effects of surface parallelism, concavity, and convexity on spatial and monaural objective acoustical parameters and on essential subjective attributes.
Excellent acoustics and a low RT? A case study of the recent renovation of the Kennedy Center Concert Hall, Washington, DC
Authors: Jaffe Christopher , Scarbrough Paul , Cooper Russell , Clements Pamela
Published: July 2002

In October 1997, the renovated Concert Hall at the Kennedy Center reopened to critical acclaim. The hall is designed on the shoebox model, but with an audience of 2759 and a volume of 250 cubic feet per person, its occupied midfrequency reverberation time was below the 2.0 s that is now considered fundamental to acoustical excellence. The musicians were also having difficulties hearing on stage. The acoustic redesign involved rebuilding the stage with reshaped walls, an acoustical canopy, resonant floor, orchestra risers, seating at stage sides and rear, and refurbished organ. Parterrre walls were added at orchestra level to increase side reflections. Since the ceiling could not be raised, reverberation was enhanced by building small reverberation chambers beside the stage and reducing seat count. Removing parts of the upper side balconies created an area near the stage where the sound field could develop without audience absorption. Changes also included echo control, new seats, and new stage HVAC system. The paper presents data from acoustical measurements and computer models made before and after the renovation, and discusses the combination of acoustical parameters that enables the hall to achieve such high acclaim despite its relatively low reverberation time.
CAAD—Computer aided architectural and acoustical design
Authors: Smigelschi Marius , Badescu Alexandru , Pana Radu
Published: July 2002

The computer program ‘‘AUDIO’’ is intended to encourage architects to include (integrate) the room acoustics study in the architectural design from its very beginning, by using in both ways, interactively, the same computer descriptions (AUTOCAD environment) of the room. The main stages of the design are: first: – first architectural variants: – 2-D acoustical comparison, improvement of the room shape in plan, longitudinal, and transversal sections; second: – architectural 3-D model: – 3-D detailed shape analysis, proposed absorbtion characteristics of room bound- aries; third: – architectural details and specifications: – proposals for finishings and materials, taken from the database of the program. In both 2- and 3D stages the program is based on a very performant ray-tracing method. Entry: – room boundaries taken from the architectural model and internally processed (boundary area, volume), audience area, sources (characteristics and location), acoustical processing: – in almost real time, the complete history of impingements (collisions), that is almost statistical, – output: graphic representation of first 2–4 reflections, equal contours on the audience surface of up to ten chosen parameters, (reverberation time, sound level, clarity, spatial impression, lateral efficiency, time gravity point...), detailed impulse response, reverberation time, and the same ten parameters for any reception point.
Experiences with the application of a software for noise prediction in workrooms
Authors: Maue Jürgen H.
Published: July 2002

For a couple of years, the BIA has been using a computer program for the calculation of room acoustic parameters and the prediction of sound pressure levels at the workplace. Experiences gathered with the program in connection with industrial consulting activities are presented and illustrated by examples. The software is based on a calculation method referred to in the German VDI guideline 3760 (image sound source method) requiring a mean degree of complexity in the description of the room. The software comprises a large library, indicating sound absorption coefficients for all tested materials. The program is mainly used for planning room acoustic improvements and new workrooms. Noise prediction in connection with production changes (replacement of machinery, new productions) is also possible. In some cases of hearing damage, the program was used for a retrospective determination of noise exposure at former workplace. A comparison of the calculated spatial sound decay curves and the thus-derived room acoustic parameters with the measurement readings showed good correspondence. The development of noise maps and the consequent presentation of the achievable noise reduction by means of concrete measures has become an important instrument for motivating and convincing companies.
A rating system in search of hearing protectors
Authors: Murphy William J. , Franks John R.
Published: July 2002

Previous methods for rating the noise reduction of hearing protection devices (HPDs) have determined the mean attenuation and standard deviations at several octave-band frequencies using real-ear attenuation at threshold (REAT) measurements. Typically, some fraction of the standard deviation is subtracted from the mean attenuation to estimate an assumed protected value for a given percentage of the population. The subtractive correction assumes normally distributed numbers that are linear and not logarithmic. Data from several HPDs were analyzed with a statistical approach which first determined the A-weighted noise reduction for individual measurements. The cumulative distribution of A-weighted noise reduction (CDNRA) was then fit with a combination of two logistic regression curves. The Pearsons correlation coefficients for the model and the data were better than 0.99 for each of four devices analyzed. The CDNRA method yielded reasonable results from data collected from well-fit and poorly fit hearing protectors. In contrast, the NRR method would yield negative NRR values for poorly fit protectors. This paper describes the CDNRA method and looks at the results for an interlaboratory test of four HPDs, as well as additional data collected in the NIOSH Hearing Protector Laboratory. [Portions of work supported by the U.S. EPA IA 75090527.]
Forecasts of sound propagation by a mesoscale atmospheric model
Authors: Haugland Svenn O. , Hole Lars R.
Published: July 2002

A mesoscale atmospheric model has been used to simulate local wind and temperature profiles in an area with hilly terrain in eastern Norway. Simulated wind and temperature profiles have been used to determine the local sound velocity profiles. The predicted profiles were used as input in an acoustic model. Meterological measurements were carried out in conjunction with measurements of propagation of low frequency (<100 Hz) impulse sound. The predicted atmospheric profiles are compared with profiles measured by meteorological towers and a tethered balloon launched at the site.
Meteo-BEM: A new hybrid approach to predict noise barriers’ efficiency with meteorological effects, theory, and comparison with experimental results
Authors: Premat Eric , Gabillet Yannick
Published: July 2002

In a complex outdoor environment, meteorological factors have been found to affect the sound field. On one hand, boundary element methods are particularly well adapted to calculate the sound pressure above complex boundaries (various absorbing properties and shapes) in a homogeneous atmosphere. On the other hand, a few models have been recently developed to describe the acoustic propagation in inhomogeneous media (normal modes, residue series, parabolic equation, fast field program...). Meteo-BEM is a new hybrid formulation taking advantage of the power of the boundary element methods, and including meteorological effects using as the appropriate Green’s function these recent propagation models in inhomogeneous media. In this paper, the theory of BEM, together with sound propagation in inhomogeneous media, is briefly recalled. The new hybrid model is then presented. The case of a rigid thin-noise barrier on a flat ground under a known sound-speed gradient condition is studied. Results from Meteo-BEM are then compared to measurement data for validation. It is shown that this new model allows one to calculate accurately the sound field in a complex outdoor environment and offers promising further developments.