bluuebunny/crossref_metadata_2025 huggingface dataset:
Dynamic information in French vowel identification by native and non-native listeners
Authors: Gottfried Terry L. , Neville Jonathan S.
Published: July 2002

French /t/–vowel and /t/–vowel–/t/ syllables were spoken by a native speaker in a constant sentence frame. Target syllables were edited so that native French and non-native listeners heard intact syllables, only the initial and final portion of syllables (silent-center), only initial (initial-only), or only center portions (center-only). Research in American English [e.g., W. Strange, J. Acoust. Soc. Am. 85, 2135–2153 (1989)] has demonstrated that listeners accurately identify vowels in silent-center consonant-vowel-consonant syllables; identification of initial- or final-only portions is very poor. However, French syllables are more commonly consonant-vowel in structure; for this reason, native listeners might identify initial-only portions nearly as accurately as silent-centers. By contrast, Americans whose second language is French might perform more poorly on silent-center French syllables than on intact syllables, and most poorly on initial-only. Findings suggest that indeed native French listeners identify initial-only syllables as accurately as silent-center and intact syllables, except for confusions of oral-nasal contrasts (e.g., /a/ vs /ε/). Native listeners perform significantly better on TV syllables than TVT syllables, especially in silent-center and initial-only conditions. The pattern of errors for non-native listeners is different, particularly with respect to less familiar vowel contrasts (e.g., /u/ vs /y/). [Work supported by Lawrence University Faculty Grant.]
Self-calibrating microphone array
Authors: Meyer Jens
Published: July 2002

A popular application for microphone arrays is speech pick up in situations where a close spacing between talker and microphone is not suitable. A beamformer is used to achieve a directivity pattern that maximizes the ratio between the desired signal and noise sources. Here a fixed beamformer is evaluated. A fixed beamformer means that the coefficients are set at one time and then remain constant. Many design methods are known for fixed beamformers. Most of these methods assume ideal microphone array characteristics like precise sensor positions, perfect phase and gain of the sensors, and free-field conditions of the acoustical field. In many practical applications these conditions are not fulfilled. The disregard of the free-field condition in some speech pick-up applications (e.g., array on the top of a monitor, array as a hearing aid) especially can cause severe distortions of the desired beam pattern. This presentation suggests an approach that includes the nonideal effects in the design process of the beamformer. The directivity pattern is therefore measured at some discrete angles. The measured data are then compared with a predetermined directivity pattern and the beamformer coefficients are optimized such that this pattern is approximated as closely as possible.
Audio teleconferencing systems with sound localization effects
Authors: Aoki Shigeaki , Okamoto Manabu
Published: July 2002

In teleconferencing, conversation should flow naturally between separate sites. The acoustic design should, of course, be able to at least convey the correct volume. However, sound localization is also very important for realizing the kind of virtual reality world that is teleconferencing. Coordinating the visual image (a speaker’s face) and the sound image (his or her voice) creates a kind of virtual environment in which the participants at each site feel as if they are having a conversation in the same room. Two prototype teleconferencing systems with sound localization effects have been developed. The more primitive of the two is a small teleconferencing system that can connect two sites with stereo reproduction in a way that utilizes psychological acoustic phenomena. The other is a high-presence system that aims at achieving a high level of ambience among three separate sites. Its features and key technologies are natural images on a large super-high-resolution display, good sound localization from remote locations through the use of a multichannel sound image localization system, and high-definition images as a result of HDTV coding and transmission. The proposed multichannel sound system comprises four subsystems, one each for sound input, reproduction, acoustic feedback control, and transmission.
A third octave real time VXI analyzer
Authors: Martin Alberto , Recuero Manuel , Manuel Lopez Juan , Ruiz Mariano
Published: July 2002

A digital signal processing system with VXI interface has been developed in a C-size board conforming to VXI 1.4 specification, and is made up of two blocks. The first one implements a message VXI interface and the second the digital processing itself, based in a 49-MHz TMS320C31. The system includes a complete software to customize for a specific application. The VXI interface has been gifted with a real time operating system and a software parser for building a SCPI (standard command for programmable instruments) translator. The software developed for the digital signal processing system has been built in several modules written in C language and C30 assembler in order to improve the velocity. The utility of this system is to realize customizable and specific digital signal processing without the necessity of acquiring complex and expensive VXI boards that do not allow modification of the commands, the specific algorithm, or the firmware. In order to show the versatility of the system, a one-third-octave real time analyzer has been developed in agreement with ANSI S1.11.
Infrasound source location at the prototype international data center
Authors: Brown David J. , Katz Charles N. , Le Bras Ronan , Wang Jin , Gault Anna
Published: July 2002

The Prototype International Data Center (PIDC) is developing an operational system to provide temporal and spatial source location information for highly impulsive sources with an equivalent nuclear yield of at least 1 kT in any terrestrial environment. This source information is deduced by a combination of automatic and interactive processing. Detecting the infrasound released by large-scale atmospheric explosions is the most suitable method for the immediate location and characterization of atmospheric sources. The PIDC treats the source location problem uniquely in that a given source can be located using any combination of infrasonic, seismic, or hydroacoustic signals when these signals exist. The highly dynamic aspect of the atmosphere, however, creates problems not traditionally encountered in the seismic or hydroacoustic source location problem. The dynamic nature of these effects will limit the ability to apply detailed propagation models in an automatic operational system, such as is being developed at the PIDC. A seasonal atmospheric model is being used in conjunction with near real-time meteorological data to provide the ambient atmospheric conditions required by the physical propagation model. Results obtained from a ground-truth data set will be presented that demonstrate the operation of the PIDC source location procedure.
Transient, tonal, and background noise filtering with wavelet and cosine transforms
Authors: Delory Eric , Potter John R.
Published: July 2002

In underwater acoustic signal processing, a major concern has been to extract information from a noisy time series, without over-constraining the estimation. Often, very little is known about the signal or noise and any strict assumption increases the risk of failure in real applications. Here, a new denoising method is presented, tested on three types of real underwater noise sources (n1(t),n2(t),n3(t)), each added to an arbitrary signal s(t). Noise class indices respectively refer to snapping shrimp noise, shipping noise, and underwater background noise, which are technically considered as transient, tonal, and broadband pink noise. s(t) is defined as any signal that does not belong to these three classes, thus imposing no strict constraint. The extraction process is performed in three steps, each performing a noise-specific orthogonal block transform. First, a one-level Haar wavelet-packet transform is applied to detect transient noise components, using an additive information cost function as a hypothesis test and a simple dot-product to identify and extract them. Second, the denoised signal is reconstructed and decomposed by a one-level cosine-packet transform to identify and extract tonal components in a same manner. After reconstruction, pink noise is removed by thresholding a near-optimal Daubechies wavelet-packet basis.
Moths’ ears evolved first for bat defense and later for communication
Authors: Surlykke Annemarie
Published: July 2002

Many moths possess simple ears primarily to alert them to the echolocation calls of hunting bats. Since large targets reflect stronger echoes than small ones, bats should detect moths with large wing and body surface areas at greater distances than small moths. The increased conspicuousness of large moths would reduce their safety margin over bats, unless this is offset by a greater sensitivity of their ears. Results seem to support this: The sensitivity of ears of noctuid moths is significantly dependent on size, with larger moths possessing lower thresholds. If moth ears exist for bat detection, moths that live in bat-free niches should exibit reduced acoustic sensitivity. Studies of moths temporally isolated from bats (flying either in the winter or during the day) have confirmed this expectation. Another effect of the release of predation pressure from bats may be the use of the preexisting ears for sexual acoustic communication. All sonorous noctuoid moths described so far produce sounds in the same frequency range as that emitted by bats, presumably because bat-sensitive ears are the ancestral (plesiomorphic) condition and acoustic communication is a facility aquired later. [Work supported by the Danish National Research Foundation.]
Gas cavities in phytoplankton
Authors: Selivanovsky Dmitry A. , Didenkulov Igor N. , Stunzhas Pavel A.
Published: July 2002

Acoustical and related properties of phytoplankton suspensions were investigated in laboratory conditions and in situ. A review of studies is present. It was found that cells have gas cavities which allow them to keep a neutral buoyancy. Under sharp compression gas cavities dissolve and recover only under the light exposition. Such properties can explain an anomalous migration of some sound scattering layers. The use of the compression/decompression method allowed us to find gas cavities in blood erythrocites also. The 6.5-MHz acoustic backscattering from phytoplankton suspensions was applied for characterization of reverberation level under the natural condition and compression (up to 5 atm), which allowed us to evaluate a relative volume of gas cavities in cells. Gas cavities increase an acoustical cross section by 2–3 orders. A dispersion of sound velocity in phytoplankton suspensions with the phase and the resonance methods (from 200 Hz to 11 MHz) for six kinds of living cell suspensions was also studied. Both the increase and decrease of sound velocity dispersion with frequency was observed. The effect is different for cells of different kinds. [Work supported in part by RFBR, Russia.]
Mechanoreception for food fall detection in deep sea scavengers
Authors: Klages Michael , Muyakshin Sergey I.
Published: July 2002

Although knowledge about functional principles of deep-sea ecosystems is rather scarce, it is assumed that the energy supply for scavengers is restricted to large food falls of dead vertebrates. It is generally accepted that chemoreception is one of the major tools for marine organisms to detect food sources. However, another major source of information may come from hydroacoustical feeding noises produced by scavengers appearing on a cadaver reached the seafloor. The aim of the present study was to investigate whether scavenging crustaceans—pandalid shrimps Pandalus borealis—are able to detect such rare food fall events via mechanoreception or not. These results are based on 228 single experiments indicating that these animals possess the sensitivity to the particle displacement of 0.1–10 mkm in frequency range 30–250 Hz. Therefore, acoustic feeding noises offer a possibility for animals to detect such rare events but only at distances of a few meters. At such small distances chemoreception is presumably more important. However, based on theoretical calculations on the relevance of various types of waves, originating on the water-sediment interface from any object falling on the seafloor, it is proposed that such ‘‘micro seismic events’’ may allow resting scavengers even some hundred meters away the detection of this event, most likely followed by chemoreceptive tracking.
Acoustic studies of the large-scale ocean circulation
Authors: Menemenlis Dimitris
Published: July 2002

Detailed knowledge of ocean circulation and its transport properties is prerequisite to an understanding of the earth’s climate and of important biological and chemical cycles. Results from two recent experiments, THETIS-2 in the Western Mediterranean and ATOC in the North Pacific, illustrate the use of ocean acoustic tomography for studies of the large-scale circulation. The attraction of acoustic tomography is its ability to sample and average the large-scale oceanic thermal structure, synoptically, along several sections, and at regular intervals. In both studies, the acoustic data are compared to, and then combined with, general circulation models, meteorological analyses, satellite altimetry, and direct measurements from ships. Both studies provide complete regional descriptions of the time-evolving, three-dimensional, large-scale circulation, albeit with large uncertainties. The studies raise serious issues about existing ocean observing capability and provide guidelines for future efforts.
Reciprocal tomography between ATOC transmitters
Authors: Mercer James A. , Dushaw Brian , Howe Bruce M. , Spindel Robert C. , Worcester Peter F. , Metzger Kurt
Published: July 2002

The Acoustic Thermometry of Ocean Climate (ATOC) sound sources are located 8 miles north of Kauai, Hawaii and 50 miles west of Half Moon Bay, California. The sources are connected to shore-based electronics by undersea coaxial cables. Recent modifications in the shore-based electronics have permitted the ATOC transducers to serve as hydrophones as well as transmitters. Data collected over this 4-Mm reciprocal path will be presented and discussed in terms of current and internal wave effects. [Work supported by ONR.]
Modal structure of low-frequency, broadband receptions at megameter ranges
Authors: Wage Kathleen E. , Baggeroer Arthur B. , Preisig James C.
Published: July 2002

Data from the Acoustic Thermometry of Ocean Climate (ATOC) experiment show that at 3515-km range, each low (axial) mode signal contains a series of arrivals, rather than a single dispersive arrival, which would characterize adiabatic propagation. Short-time Fourier transform (STFT) analysis demonstrates that this multipath structure exhibits frequency-selective fading and significant temporal variability. Despite the complexity of individual receptions, averaging the mode estimates for transmissions at successive four-hour intervals reveals some of the expected dispersion characteristics of a deep water channel. Although this evidence suggests that the modes retain travel-time information at megameter ranges, the complicated nature of the arrivals indicates that a better understanding of mode coupling due to internal waves is required for tomography and matched-field processing applications. This study investigates the temporal and frequency coherence of broadband modes by comparing PE simulations with the ATOC results. Since ambient noise and environmental uncertainty affect the accuracy of mode signal measurements for experimental data, this research also addresses certain practical issues and limitations involved in estimating mode coherence. [Work supported by ONR.]
Diagnostic ultrasound for two-dimensional temperature mapping in hyperthermia and laser-induced tissue coagulation
Authors: Willig Hendrik , Pfennig Annette , Haenel Volker , Kleffner Bernhard , Schwerdtfeger Karsten
Published: July 2002

Laser-induced tissue heating and coagulation is a common technique in minimally invasive surgery to achieve localized thermal tissue damage. Precision and efficiency of treatment could be improved markedly by noninvasive spatio-temporal temperature monitoring. A method for two-dimensional mapping of tissue temperature as well as tissue structural changes has been investigated in vitro. The algorithms are based on the temperature dependence of speed of sound and a combination of additional acoustical parameters in order to consider the expansion of the heated volume, creation of gas bubbles, and carbonization. An approach for motion compensation, as it is necessary for clinical application, will be described in short.
The calibration of extended stationary low-frequency hydroacoustic antennae in the real conditions
Authors: Agranovski Alexander V. , Osman Byneev R. , Cygankov Sergei G. , Rosenberg Alexander V.
Published: July 2002

As known, the complex problems of a sea polygon calibration must include solutions of coordinate problems after an antennae systems placement was completed. At calibration, a necessity more frequently arises in a solution of other kinds of coordinate tasks: geometrical form determination for a receiving system; geometrical form determination for a receiving system and coordinates of the main receiving element in the selected immovable coordinate system (absolute fixing); and the self-coordinated determination of a mutual position for emitting points and receiving system elements. The first and second tasks assume that point coordinates are known (e.g., ones were determined by navigation methods), but at a third task they are not known. In the last case, a mutual location of hydrophones and emitting points ought be determined only by means of hydroacoustic measurements. At calibration, other kinds of acoustic signals are used: continuous signal (tonal or narrow-band) and pulsed signal (explosion, pneumatic gun, etc.). In this paper some algorithms for the decision of the above-mentioned tasks at the calibration of extended stationary low-frequency hydroacoustic antennae by using different kinds of emitted signals are described.
Experimental and theoretical studies of the modes of stringed instruments and their relevance for quality control of instrument manufacture
Authors: Richardson Bernard E.
Published: July 2002

This paper reviews various methods for measuring and predicting the modes of vibration of the wooden structures found in instruments of the violin and guitar family. These methods include holographic and speckle interferometry, modal analysis by impact hammer excitation, and the finite element method. Theoretical modeling of stringed instruments has now advanced to the stage where the models can be ‘‘played,’’ allowing listening tests to be used to link construction of the instrument with the perceived sound quality in a formal way. Early indications of this work suggest that mode frequencies play a much smaller role in determining the sound quality of an instrument than has previously been assumed, and that greater attention should be paid to the ‘‘ease’’ with which modes can be driven by the strings and to the ‘‘source strengths’’ of the modes. This paper will put the case for measuring additional modal parameters, such as effective mass and effective stiffness, and discuss their relevance to quality control by the maker during manufacture of an instrument.
Dynamical behavior evolution of the guitar harmonic plate in the successive construction phases
Authors: Elejabarrieta María J. , Ezcurra Amaya , Santamaría Carlos , Bretos José
Published: July 2002

The fundamental piece for the quality of the sound of a guitar is its harmonic plate. From the initial phase (gross plate without hole) until the last construction phase (the addition of the structures system), the craftsman has established a series of stages where the acoustical behavior is modified until the definitive guitar is created. This is the reason to establish the effect of each modification in the vibratory response of the harmonic plate. With this purpose an experimental modal analysis and a finite element modeling of all the construction phases in the low-frequency range have been carried out, obtaining information about vibration patterns, eigenfrequencies, quality factors, and spectra. In the three first construction phases, the plate goes from a uniform thickness distribution and no hole to a mouth and variable thicknesses. These changes do not modify the vibration patterns, but they lead to a gradual decrease of the eigenfrequencies and an increase of the vibration amplitude. The vibration patterns are modified when the structures are added. Then, the rigidity of the structure is increased and the resonances present a smaller damping. Likewise, the frequency response function decreases remarkably and the resonances tend to be more uniformly distributed in frequency.
Comparison between modal analysis and finite-element modeling of a marimba bar
Authors: Chaigne Antoine , Bork Ingolf
Published: July 2002

The sound spectrum of low-tuned marimba bars contains many components that cannot be explained by a simple one-dimensional model of flexural vibrations. A modal analysis has thus been performed on a bass marimba bar C3 with a fundamental frequency of 130 Hz [outer dimensions: 46×6×2.35 cm (x,y,z), material: rosewood], taking into account two spatial components of vibration. The 20 lowest modes were extracted from this analysis (modal frequencies and mode shapes). A 3-D finite-element analysis of this bar, based on an orthotropic model for the material, has been conducted in parallel. Modal analysis and finite-element modeling yield very similar mode shapes and modal frequencies for the first 12 modes, between 130 and 4000 Hz, which confirms the experimental results and validates the theoretical approach. The discrepancies between measured and calculated frequencies are less than 4% in this frequency range. For higher frequencies, between 4000 and 8000 Hz, the main flexural modes along 0x, such as the (7,0) and the (8,0) modes, are still correctly predicted. In this range, the three components of the bar motion become significant and the finite-element modeling shows a number of modes which were not detected in the modal analysis.
Deconstructing the violin—The road from B to A
Authors: Bissinger George
Published: July 2002

A recently initiated project to measure the elastic and damping material properties, normal mode frequency and shape vibrational properties, and the angular distribution and efficiency of radiation from assembled violins is described. Combining accurate geometries and densities, experimental modal analysis with a scanning laser vibrometer, measurements of bridge-corpus transfer mobilities, and far-field radiation over a sphere, one potential application is to extract the relevant material and vibrational properties for all substructures. Another goal is to generate a finite-element model ‘‘particularized’’ to each violin to calculate normal mode radiation from the violin and simulate its sound. Other aspects and applications of the model will also be briefly discussed. [Research supported by the National Science Foundation under Grant No. DMR-9802656.]
Implementation of adaptive digital ANR in hearing protectors
Authors: Steeneken Herman J. M. , Verhave Jan A.
Published: July 2002

Present state-of-the-art personal hearing protectors with ANR are mainly based on analog control circuitry. The performance of these systems is fair to good; an additional attenuation for low frequencies of 10–25 dB is obtained. By using a digital control circuitry for feedback gain control and compensatory filtering, a similar performance can be obtained. Digital signal processing allows for real-time control of the system parameters. By using an adaptive algorithm, the system performance can be optimized for individual (acoustic) frequency transfer and for the spectrum of the specific noise to be suppressed. A prototype system has been developed, in which the feedback gain and the frequency response parameters are adaptively controlled. The system uses an (additional) probe to determine the stability of the feedback loop. Other features of digital ANR are: continuous monitoring of the noise dose, transparency at low noise levels in order to improve direct communications and warning signals, and advanced signal processing of speech signals from an intercom. Preliminary results obtained with an add-on system (attached to a standard earmuff) will be presented and compared to the performance of 20 commercially available systems.
Effect of wall adsorption on the photoacoustic detection of ammonia with a 1.53-μm DFB diode laser
Authors: Hebach Andreas , Miklos Andras , Hess Peter
Published: July 2002

The adsorption of ammonia molecules (and other polar molecules such as nitrogendioxide, water, etc.), on the walls of the photoacoustic (PA) detector influences the accuracy of the concentration measurement. This effect becomes larger for small concentrations and depends on the direction of the concentration change. This means that a hysteresis of the PA signal is observed between measurements carried out by decreasing and by increasing the concentration. The adsorption problem was reduced considerably by performing the measurements in a continuous flow. For measurements with high flow rate a new photoacoustic detector was developed. The differential design provides good flow and electronic noise suppression. The PA signal has been generated by tuning a 40-mW, 1.53-μm GEC-Marconi DFB laser to the absorption peak of ammonia at 1527 nm. The dependence of the PA signal on the flow rate of ammonia–nitrogen mixtures with different concentrations was investigated. In another experiment the hysteresis of the PA signal was investigated by increasing and decreasing the concentration in several steps, but maintaining the same total flow rate. It was found that a higher flow rate results in a smaller error of the concentration measurement and a shorter response time of the PA detector.
Modeling within- and across-channel processing of amplitude modulation
Authors: Dau Torsten
Published: July 2002

Recently, a model was presented in which a modulation filterbank was introduced to analyze the envelope fluctuations at the output of each excited peripheral auditory filter [Dau et al., J. Acoust. Soc. Am. 102, 2892–2905 (1997)]. As a decision stage, an optimal detector was applied which combines all modulation-filter outputs linearly, assuming independent observations. The model accounts for temporal modulation transfer functions (TMTF) with narrow band and broadband carriers as well as for modulation masking data. The model does not cover conditions which require across-channel processing, such as modulation detection interference (MDI). In this case, the auditory system seems to combine information across frequency channels whose responses are temporally modulated. A new modeling approach is presented that integrates the modulation-filter outputs across frequency channels prior to the decision stage, while preserving the original model’s ability to describe TMTFs and modulation masking. To further ensure the possibility of predicting spectral masking, a low-frequency part of the modulation spectrum, including the dc component, is processed separately within each peripheral channel. The model can account for the main features of MDI, but it does not account for comodulation masking release.
Physiological measurements and mathematical models for interaural pathways
Authors: Rangol Hans-Peter
Published: July 2002

The middle-ear cavities of birds, reptiles, and anurans are often connected through a tube, which can act as an interaural pathway between both eardrums. Such an arrangement of acoustically effective structures is described as a pressure-difference receiver. It is generally assumed that pressure-difference receivers enhance the skills for directional hearing in animals. Sound pressures and phases were measured with probe-microfones in the interaural pathway from Zebra Finches during azimuthal rotation. The results show pressure differences between right and left middle ear, which are dependent on the azimuth and the frequency of the applied sound. A mathematical model was constructed, which gives the resulting pressure and phase of two overlaying waves that travel different distances. Assuming one wave traveling through the pathway and a second around a sphere, the resulting wave from dependency on azimuth and frequency was calculated. The pattern shows the same characteristic pressure differences as the physiological measurements. The same procedures were performed with a sphere as a hardware model for a head with an interaural pathway. The results show again the expected characteristics. The combination of physiological measurements, mathematical modeling, and testing this model with hardware models gives a meaningful explanation for the acoustical properties of middle ears as a basis for psychological acoustics.
Comodulation masking release in bit-rate reduction systems
Authors: Vestergaard Martin D. , Rasmussen Karsten B. , Poulsen Torben
Published: July 2002

It has been suggested that the level dependence of the upper masking slope be utilized in perceptual models in bit-rate reduction systems. However, comodulation masking release (CMR) phenomena lead to a reduction of the masking effect when a masker and a probe signal are amplitude modulated with the same frequency. In bit-rate reduction systems the masker would be the audio signal and the probe signal would represent the quantization noise. Masking curves have been determined for sinusoids and 1-Bark-wide noise maskers in order to investigate the risk of CMR, when quantizing depths are fixed in accordance with psycho-acoustical principles. Masker frequencies of 500 Hz, 1 kHz, and 2 kHz have been investigated, and the masking of pure tone probes has been determined in the first four 1/3 octaves above the masker. Modulation frequencies between 6 and 20 Hz were used with a modulation depth of 0.75. A CMR of up to 10 dB was obtained at a distance of 6 Bark above the masker. The amount of CMR was found to depend on the presentation level of the masker; a higher masker level leads to a higher CMR effect. Hence, the risk of CMR affecting the subjective performance of bit-rate reduction systems cannot be ruled out. a)Currently at Oticon Research Centre Eriksholm, 243 Kongevejen, DK-3070 Snekkersten, Denmark, mve@oticon.dk
Identification of forces by the RIFF method
Authors: Pezerat C. , Guyader J. L.
Published: July 2002

The goal of this presentation is to show the application of a test bed which should permit the identification of forces created by an engine when it is not possible to measure them directly. The interest of this experimentation is to have an idea of the forces in the real situation. The mechanism consists of fixing the engine on a known plate which has characteristics related to those of the real situation. To identify forces exciting the plate, the method used, the RIFF method, is based on the computation of the force distribution, i.e., the second member of the equation of motion of the plate, from the knowledge of the flexural displacements. The main difficulty of this inverse problem comes from uncertainties in data that produce a high noise level in the result. To overcome this instability, two regularization techniques are presented. The first approach is based on singular value decomposition (SVD), consisting of eliminating the singular values of the operator applied on displacements. The second approach is based on high wave number filtering. Formulations of techniques are presented and experimental results are shown in order to validate both approaches.
Inverse scattering of acoustic waves by two elastic shells in water
Authors: Gaunaurd G. C. , Huang H.
Published: July 2002

The key problem of inverse scattering consists of the identification of an unknown target from its scattering pattern (form-function). For a single, simple-shaped target in an infinite medium, the problem has been solved elsewhere. There is present interest in the case of two elastic shells, with centers a distance d apart, insonified by a (distant) sonar, operating at frequency f, in a boundless sea. The determination of the scattering pattern of such a double-shell target was analyzed by this group earlier [viz., J. Acoust. Soc. Am. 98, 2149–2156 (1995)]. That direct solution is now built upon to solve the associated inverse problem. From the form-function of the two shells—assumed equal—which is also assumed given in two arbitrary directions, it is determined: (a) that there are actually two interacting shells (i.e., spatial resolution), (b) their separation d, (c) the pair orientation angle α, (d) the shape and size of the shells, (e) their thickness, h, and (f) their material composition. This information is all extracted from features in the form-functions, using the detailed procedure explained here, which accounts for all orders of multiple scattering, since the shells could be close to each other, and thus be strongly interacting. A way to obtain/measure the two needed form-functions is using ultra-wideband (UWB) interrogating waveforms or UWB processing. [Work partially supported by the ILIR Program of NSWC-CD.]
Force spectra identification by FRF matrix inversion: A sensor placement criterion
Authors: Blau Matthias
Published: July 2002

Excitation force spectra are a key quantity in structure-borne sound characterization. In this paper, the identification of multiple broadband force spectra by FRF matrix inversion is considered. Because of the notorious sensitivity of inverse methods to small errors in measured data, such measurements must be carried out with extreme care. One major decision that could heavily affect the accuracy of the final results concerns the placement of the vibration response sensors. To date, this is often accomplished by just putting them as close as possible to the force input points, or by subjective judgment, or by considering the condition number of the FRF matrix as a criterion. In this paper, an alternative criterion is proposed which takes into account that, especially in the frequency region up to a few hundred Hz, the resulting errors are often dominated by measurement noise-induced bias on response spectra estimates.
Hands-free car kits—A performance discussion of ‘‘state-of-the-art’’ solutions
Authors: Kettler Frank , Gierlich Hans Wilhelm , Krebber Winfried
Published: July 2002

The increasing demand for hands-free solutions in vehicles partly results from safety reasons. In several countries, the use of handsets is already prohibited for drivers. From the technical point of view, not only the ‘‘hands-free problem’’—acoustical stability and echo—has to be solved. The environmental conditions, i.e., noise level and position of microphone and loudspeaker, are extremly critical. Technical implementations include various signal processing methods such as echo cancellers, noise reduction algorithms, and level switching devices. Typically they work in conjunction with codecs (GSM or others). Consequently, speech transmission quality is influenced in various ways. Comparative measurements carried out with eight different hands-free car kits are discussed in this presentation. The test setup guarantees reproducible conditions using the auditory part of a driving simulator, artificial head technology, and a codec simulator. The determination of important objective parameters was carried out with adapted test signals, while real speech samples were used for subjective evaluation. Measurement setup and relevant parameters concerning speech transmission quality are briefly discussed. A number of listening examples demonstrate differences between various implementations.
Expectation in quality assessment of Internet telephony
Authors: Möller Sebastian , Riedel Joachim
Published: July 2002

Telephony from Internet terminals is getting more and more popular. Thus it is interesting to know which level of quality one user expects from a connection established from a computer terminal in relation to a normal wirebound telephone. For mobile situations, it has been shown that the lower expectation results in more favorable quality judgments in comparison with wirebound systems when assessing physically equal connections. The user’s expectation includes call motivation, attitude, experience, emotions, etc., and it will have an important influence on his/her perception of and judgment on quality. A conversation test has been carried out in order to investigate the quality impact due to Internet-typical impairments (absolute delay, low-bitrate codecs), both for users of a conventional wirebound telephone and an Internet terminal. In contrast to what can be observed for mobile phones, it turns out that the expectation level of potential users may result in a disadvantage effect, i.e., quality expectation is so high that Internet terminal connections are rated worse than conventional ones for physically equal connections. The situation is different for subjects who never operated a computer terminal before. Results are discussed regarding user groups and the consequences for the assessment of Internet terminal telephony.
Application of auditory contours to speech quality measurements
Authors: Daniel Peter , Zollner Manfred , Ellermeier Wolfgang
Published: July 2002

For the development of instrumental speech quality measures it is important to model human auditory signal processing. The ITU-T standard P.861 for telephone transmission quality, however, neither takes into account the time-frequency resolution of the human ear nor its masking properties. Moreover the abstraction or contorization processes occurring in speech perception are disregarded. Terhardt proposed to use spectral pitch as the primary auditory contour. In our approach it is derived from the part-tone time-pattern (PTTP) originally conceptualized by Heinbach [Acustica 67, 242–256 (1988)]. The PTTP is extracted from an aurally adequate spectral analysis. A new instrumental method based on differences of auditory contours will be presented. The results will be compared with other instrumental methods and with data from subjective listening tests.
Blind dereverberation using short-time cepstrum frame subtraction
Authors: van Eeghem Jean , Tohyama M. , Koike T.
Published: July 2002

A blind dereverberation method that can be used to separate reverberant speech into an impulse response and a dry speech contribution is presented by the authors. This method is based on the assumption that the contribution of an impulse response to reverberant speech varies slowly compared to that of the dry speech. Processing the reverberant signal for short-time frames and using the special properties of the cepstrum domain, allows a recursive scheme to remove the equal impulse response contributions. Although short-time frames are processed, the effects of an amply longer impulse response can be separated from the dry speech. The assumption of a constant impulse response in only two frames and the iterative processing make the method inherently robust to changes in the impulse response. An ambiguous linear phase addition when calculating the inverse cepstrum constitutes one of the problems to limit the dereverberation performance.
Leonid Brekhovskikh: The man and the scientist
Authors: Dubrovsky Nikolai A.
Published: July 2002

L. Brekhovskikh was one of the main founders and a first Director of the Acoustics Institute (AI) of the USSR Academy of Sciences (1954). He contributed a lot in the selection of young gifted scientists and engineers as AI staff, formation of the main branches of research in AI, construction of new buildings and experimental setups, and promoted a new approach to design and construction of the research vessels Sergei Vavilov and Petr Lebedev. The active creative work of L. Brekhovskikh in AI (1963–1980) became even more intensive after he left the director’s position. It relates to new theoretical achievements, fruitful and numerous experiments in the ocean, organizing the famous Zvenigorod conferences on ocean acoustics, and publishing the fundamental treatise ‘‘Acoustics of the Ocean.’’ Brekhovskikh possesses the personal features that facilitate his achievements in science. He is a very gifted, amicable, benevolent, reserved, purposeful, concentrated, and rational person. Sport exercises and yoga practice have considerably supported his creative efforts for decades.
Wave diffraction at reflection and propagation in layered media
Authors: Godin Oleg A.
Published: July 2002

In his theory of waves in layered media, L. M. Brekhovskikh has systematically employed integral representations of the field in terms of plane or quasi-plane waves, with all the other useful representations of the field derived from this exact solution. It is this approach that enabled him to efficiently study, and for the first time gain a deep insight into and a quantitative description of the fundamental diffraction phenomena accompanying total reflection, guided propagation, and formation of caustics in inhomogeneous media. In this presentation, L. M. Brekhovskikh’s work on wave diffraction in layered media, including his concept of diffracted rays, study of ray and beam displacement at reflection, investigation of ray theory and the WKBJ approximation domains of validity, and an exact solution for a range-dependent benchmark problem, are reviewed. Recent developments in the theory of acoustic propagation in inhomogeneous media are traced back to the ideas first introduced and explored by L. M. Brekhovskikh.
Consistency in acoustic design when building symphonic performance halls
Authors: Jafe Christopher , Holden Mark , Lilkendey Robert
Published: July 2002

The acoustic success of all concert hall designs depends on meeting the subjective response of the audience and the performers. Since it is now possible to correlate subjective preferences of listeners and players with quantitative physical acoustic criteria, it is axiomatic that the acoustician should be consistent in setting acoustic environmental criteria for all spaces programmed for symphony performances. This paper describes the application of key physical acoustic criteria related to hall liveness, warmth, and intimacy in the design of rectangular and surround concert halls, multiuse proscenium style performances spaces, and symphonic music pavilions. In addition, the authors will illustrate how the consistent application of these same acoustic criteria can enable the practitioner to develop successful symphony halls and virtual concert facilities utilizing electronic reflected energy systems (ERES). All of the facilities described in this paper incorporated the reflections corelated to listener and performer preferences and were extremely well received by audiences, performers, and the press.
Hearing protection in everyday practice—The gap between expert knowledge and the beliefs of the users
Authors: Barrenas M. L.
Published: July 2002

Many well-informed workers do not use hearing protectors (HPs) as they are taught to. Also, there are differences between what workers know, believe, and do. HPs are used whenever ‘‘necessary’’ and taken off when ‘‘the noise levels are low.’’ Workers know when the noise is measured at their workplace. However, when there is no feedback from these measurements, the noise levels will be regarded as being ‘‘all right.’’ Based upon their own noise rating scale, workers create their own norms for HP usage. By regarding their hearing as normal, even when they know their hearing is impaired, there is no incentive to use HP more often. The great disparity between the knowledge of the experts and the way workers use HP shows that the awareness of damage risk is low. Adaptation to noise and to hearing disability is tremendous! Thus, HP usage is determined by the worker’s own noise rating scale, by exchange of opinions between workers, and by their belief of having normal hearing, rather than by the objective noise measurements, information about the need of hearing protection and audiograms. Therefore, the hearing conservation procedures must be renewed according to modern pedagogic principles starting from the perspective of the users, not the experts.
The development of prediction models to control noise levels around UK firing ranges
Authors: Kerry Geoff , West Martin
Published: July 2002

Since 1981, many UK firing ranges have used computer-based prediction models to control noise levels in the surrounding countryside. These models take into account local surface conditions and meteorology to provide peak noise level contour maps over distances of up to 30 km. A feature of these models is that peak noise level maps can be produced within a few minutes of the input of meteorological data obtained directly from radiosonde ascents or from mesoscale models. Several new prediction models have been developed at Salford over the years, taking advantage of the ever increasing power of personal computers to improve accuracy and user input and output facilities. This paper will discuss the model developments, including the meteorological preprocessor, illustrate some of the techniques adopted to improve accuracy and speed, and outline future improvements.
Aerodynamic noise from vehicles on the Yamanashi Maglev Test Line
Authors: Miyamoto Masaaki , Yamazaki Mikio , Goto Yasuyuki , Ueno Makoto , Zenda Yasuo , Iida Masanobu
Published: July 2002

This paper is a report about aerodynamic noise from Maglev vehicles running at an ultra high speed. Maglev running tests have been carried out on the Yamanashi Maglev Test Line and 550 km/h was recorded in December 1997. The noise produced when Maglev vehicles run at high speed consists of almost only aerodynamic noise because of the levitated running of Maglev vehicles. Two types of nose shapes were developed for Yamanashi test cars in order to considerably reduce air drag and aerodynamic noise. The noise measured when Maglev vehicles running on the viaduct at the velocity of 500 km/h will be reported in this paper. Also the rate of increase of noise to velocity and the source of noise of running Maglev vehicles will be reported. In general, it is known that when a high-speed train enters a tunnel, a compression wave is generated ahead of the train and a pulse wave radiates from the exit of the tunnel, and an air pressure booming noise is generated in some cases. This air pressure booming noise is successfully suppressed by setting a hood at the tunnel entrance in this test line.
The relationship between the power level and different parameters of a single vehicle noise
Authors: Reig Antonio , Gaja Esteban , Manglano Jose Luis , Sancho Salvador
Published: July 2002

This paper is concerned with the description of various factors, such as velocity, gears, and type of vehicle affecting the power level of a vehicle noise; a resume of this dependence will be offered. This work has been done in the Universidad Politecnica de Valencia (Spain) by the Laboratorio de Ingeniera Acoustica within a Thesis Project, with the aim to develop a predictive mathematical model of the urban traffic noise. Measurements were taken with an analyst Bruel mod. 2236, which enables one to obtain power levels in third octave bands by a real-time spectral analysis. The measurements being used are extensively explained in this paper, in particular that special care was taken to avoid noise from other sources that could interfere with the results. The reason why the power level was chosen as the first stage of this study of traffic noise, starting from the single vehicle as the point source, was to comply with the methodology established in the second part of standards ISO 9613, leaving for a second stage the propagation of the urban traffic noise. The final results show the great correlation between speed of the vehicle and power level.