bluuebunny/crossref_metadata_2025 huggingface dataset:
An intermediate-range sonar for fish detection
Authors: Farmer David , Trevorrow Mark , Pederson Bjarke
Published: July 2002

A primary challenge in acoustic fish stock assessment is the limited sampling volume of vertically oriented sonars. While long-range imaging with powerful shore-based systems has demonstrated interesting possibilities, practical and environmental constraints motivate an intermediate technology with the goal of mapping fish distributions to ranges of a few km. This work describes a 40-element 12-kHz array deployed both for subsurface towing and also for mounting on the sea floor with motor drive allowing azimuthal rotation. Field tests in the towed configuration have been carried out in the Strait of Georgia, with the array maintained on the sound channel axis at a depth of 40 m. While there was some sensitivity to towing depth, fish could be detected out to the 7-km sampling limit at the same time as fishing activity showed the presence of migrating sockeye salmon. Sea floor deployment in a shallow coastal area demonstrated the sonar’s ability to track herring and provides experience in the use of a rotating mount for sector imaging of fish schools. Based on experience gained with this implementation, both the potential applications and limitations of this approach for intermediate range fish detection are identified.
Multibeam sonar image processing and three-dimensional analysis of fish schools
Authors: Hamitouche Chafiaa , Fracasso Valerie , Scalabrin Carla
Published: July 2002

Digital data from a high-resolution multibeam sonar were processed to provide 3-D fish school descriptors. The multibeam sonar operates at a carrier frequency of 455 kHz, with 60 beams of 1.5° each, allowing an observation angle of 90°. The envelope of the signal is sampled every 5 cm. The usual operating range is 100 m with a ping rate of 7 pings/s. This huge amount of data provides larger biovolume samples with higher data resolution than standard fisheries research vertical echosounders used until now to describe fish school structure [C. Scalabrin et al., ICES J. Mar. Sci. 53, 181–188 (1996)]. Furthermore, the use of a multibeam sonar allows construction of 3-D images of fish schools. Data were processed by specially designed image algorithms in order to account for anisotropy in the range and vessel movement direction planes. Three-dimensional analyses such as smoothing, segmentation, visualization of the segmented volume, and its quantification were performed. Three-dimensional patterns description tools (invariant with respect to translation, rotation, and scaling) were studied. These tools might lead to the classification of the detected structures using pattern recognition methods.
Nonlinear effects in the paraxial region of diffracting sound beams radiated by cylindrical sources
Authors: Babenkova Svetlana V. , Khokhlova Vera A. , Kargl Steven G.
Published: July 2002

Intense cylindrical sources are widely used in underwater acoustics and medical ultrasound; however, nonlinear-diffraction phenomena in acoustic fields of this geometry are not studied yet as well as the fields produced by unfocused and weakly focused circular piston sources. The present work is based on a special analytical method that has been developed recently [M. F. Hamilton, V. A. Khokhlova, and O. V. Rudenko, J. Acoust. Soc. Am. 101, 1298–1308 (1997)] to model nonlinear and diffraction effects near the axis of an acoustic beam radiated by apodized circular source. This method combines the advantages of the parabolic approximation with nonlinear geometrical acoustics, and it permits an exact solution of the nonlinear problem. The approach is extended to acoustic waves radiated by a finite-length cylinder. A system of nonlinear coupled equations describing waveform distortion in the paraxial region of the beam is derived from the modified KZ equation. Analytical solutions are obtained in the time and frequency domains for an initially sinusoidal wave radiated by the cylinder with Gaussian shading of the amplitude along the axis. Nonlinear waveform distortion, shock formation distance, harmonic propagation curves, directivity pattern, and effect of steering are investigated for various parameters of the source. [Work supported by ONR, RFBR, and CRDF.]
Use of pseudo-random codes in pulse compression
Authors: Shoning Paul , Sarkady Antal
Published: July 2002

Pseudo-random, or pseudo-noise (PN), codes are numeric sequences which appear random to the uninitiated observer yet are easily predictable to those who know the generation algorithm. Traditionally, PN codes have been generated by a hand wired sequence of binary shift registers, but are now created using programmable logic chips wired from user defined netlists. A properly implemented logic chip can generate PN codes of maximum length L=2n−1 where n is the number of shift register segments. The code’s period is limited only by the number of shift registers available. Bipolar PN codes have a normalized autocorrelation function that is unity at zero shift and 1/L at all other shifts. Thus PN codes can be used to amplitude modulate (AM) the ‘‘main bang’’ of an active RADAR or SONAR system. This modulation produces compression which vastly improves the range resolution and the sensitivity of the system. The properties of PN codes and their use in a simulated active SONAR system will be demonstrated using matlab programs.
Nonlinear acoustics of solids: History and modern trends
Authors: Krasilnikov Vladimir A.
Published: July 2002

The author had the pleasure to be involved in the pioneering experiments on the nonlinear acoustic wave propagation in crystals and metals (1960). It was found that even for small wave amplitudes acoustic harmonics generation took place for longitudinal waves. The results obtained were interpreted from the point of view of Hooke’s law nonlinearity due to nonparabolic interaction potential taking into account accumulation effects in nondispersive media. Besides that, transversal acoustic waves under some external effects like pressure, heating, etc., also demonstrated nonlinear behavior despite that it was forbidden in a framework of crystal lattice theory. It was accounted for by the development of structural disturbations in a material and seemed to be the first attempt to demonstrate the importance of acoustic nonlinear nondestructive testing of different materials. It was shown later that some construction materials and rocks have much higher nonlinear parameter values than those for crystals and metals. The paper will also present an historical review of recent developments in other areas of nonlinear acoustics in solids: measurements of nonlinear acoustic parameters, surface acoustic waves nonlinearity, ‘‘clapping’’ nonlinearity, defective solids and rocks, etc.
Theoretical bases of the vibroacoustical nonlinear forced oscillation diagnostics method
Authors: Bouraou Nadezhda I. , Gelman Leonid M. , Ossokina Nataliya Yu.
Published: July 2002

A new passive vibroacoustical forced oscillation diagnostics method, based on nonlinear dynamics of tested objects, is considered. The method uses nonstationary vibration excitation with changed frequency. For the first time, theoretical research of the nonlinear oscillation system is given with mentioned excitation and the presence of the crack in the tested object. New analytical expressions between the estimates of spectral density of signals, radiated at tested object forced oscillations, and the relative size of the crack, duration of diagnostics, and speed of frequency excitation changing are received. On the basis of mentioned expressions, testing data are proposed and investigated. Comparison with the case of stationary excitation is given.
Acoustic-phonetic features for the automatic recognition of stop consonants
Authors: Ali Ahmed M. Abdelatty , Van der Spiegel Jan , Mueller Paul
Published: July 2002

Despite the recent successes in the field of automatic speech recognition, more research is still needed in order to understand the variability of speech and the acoustic characteristics of speech sounds in different contexts and for different speakers. In this paper, the acoustic-phonetic characteristics and the automatic recognition of the American English stop consonants are investigated. The acoustic features that exist in the literature are evaluated and new features are proposed. To test the value of the extracted features, a knowledge-based acoustic-phonetic system for the automatic recognition of stops, in speaker independent continuous speech, is proposed. The system uses an auditory-based front-end processing and incorporates new algorithms for the extraction and manipulation of the acoustic-phonetic features that proved to be rich in their information content. Several features, which describe the burst frequency, formant transitions, relative amplitude, spectral shape, and duration, are combined in the recognition process. Recognition accuracy of 95% for voicing detection and 90% for place of articulation detection are obtained for TIMIT database continuous speech of multiple speakers from different dialect regions. The obtained results are analyzed and compared to previous work. [Work was supported by Catalyst Foundation.]
Convergence characteristics of frequency-domain LMS adaptive filters
Authors: Reichard Karl M. , Swanson David C.
Published: July 2002

The frequency-domain implementation of the adaptive LMS algorithm offers several advantages over the time-domain LMS algorithm. The primary advantage is that the orthogonality of the individual bins in the signal spectra permit the implementation of a single complex LMS filter for each FFT bin independently, each with a separate convergence parameter. The result is a uniform rate of convergence for all FFT bins as opposed to the time-domain algorithm which converges only as fast as the dominant eigenvalue of the input data. An additional parameter which affects frequency-domain LMS convergence is the overlap between successive FFT buffers. It is shown that the maximum advisable overlap between FFT buffers is 50%. Greater overlap between FFT buffers can lead to oscillations (and even instability) in the adaptive algorithm. Experimental results are presented which illustrate the convergence characteristics of a frequency-domain adaptive active noise control system for several different types of reference inputs.
A rejection method for the Korean isolated word recognition system
Authors: Kim Dong-Hwa , Kim Hyung-Soon , Kim Young-Ho
Published: July 2002

The small vocabulary isolated word recognition systems as well as large vocabulary continuous speech recognition systems can be applicable in many areas. For the isolated word recognition systems to be deployable in actual applications, the ability to reject the out-of-vocabulary is required. This paper presents a rejection method which uses the clustered phoneme modeling combined with postprocessing by likelihood ratio scoring. Our baseline speech recognition was based on the whole-word continuous HMM. Six clustered phoneme models were generated using the statistical method, monophone clustering algorithm, from the 45 context independent Korean phoneme models, which were trained using the phonetically balanced Korean speech database. The performance of this method was assessed in terms of the out-of-vocabulary rejection rate and the accuracy on the pre-defined-vocabulary. The performance test for speaker independent isolated words recognition task on the 22 section names shows that this method is superior to the conventional postprocessing method, which is performing rejection according to the likelihood difference between the first and second candidates. Furthermore, these clusters phoneme models do not require retraining for the other isolated word recognition systems with different vocabulary sets.
Signal fluctuation statistics and performance of phase coherent underwater acoustic communications: An experimental evaluation
Authors: Al-Kurd A. , Yang T. C.
Published: July 2002

Underwater acoustic communications data using phase modulated signals were collected during the Littoral Warfare Advanced Demonstration 98-1 experiments. The signals were projected from both a drifting modem and from a towed frequency source. Data of several different baud rate were transmitted and used to (1) study the temporal variation of the acoustic impulse response of the ocean and (2) evaluate the performance of the phase coherent communication algorithm. The amplitude and phase variations are analyzed on a fast scale (within the transmission packet of 5 s) and a low scale (minutes to hours). The performance of the communication algorithm is correlated with the phase rate and amplitude fluctuation statistics, the signal-to-noise ratio, etc. The data are compared with simulation results and used to guide performance predictions. [Work supported by the Office of Naval Research.]
Doppler tolerant link
Authors: Matthews Anthony D.
Published: July 2002

A specialized probe signal is used for environmental adaptation, frequency scaling, and alertment in an underwater acoustic digital communications link. The link is designed to permit one-way message traffic to be sent from a static node to a moving node. The moving node creates an unknown range rate within known range rate limits. Link design goals include power conservation, a bit error rate of 1/10 000, and tolerance of a 25 dBs noise field. Design parameters, simulation experiment results, and field experiment results (if available) are presented. [Sponsored by ONR.]
Compromises in orchestra pit design: A ten-year trench war in The Royal Theatre, Copenhagen
Authors: Gade Anders Chr. , Mortensen Bo
Published: July 2002

The ‘‘old stage’’ of The Royal Theatre in Copenhagen—a classical horseshoe theatre with an almost flat audience floor and four balcony levels—is the primary opera and ballet theatre in Denmark. In the early 1980s the orchestra pit was enlarged and its size made flexible. However, in the following years this new flexibility caused a formal battle between the orchestra, the singers, and management, because each group had different interests as to how the pit should be configured for each new production. In the 1990s, increased concern about the musicians’ hearing and the increase in the popularity of opera finally opened the way for a new renovation of the pit, which solved most of the problems. The paper describes the many lessons learned during the ten-year period in which the authors were involved with the case as acoustic consultants. Questions of how pit configuration influences mutual hearing and exposure levels in the pit, balance between singers and orchestra as judged by the audience, communication between stage and pit, as well as the influence of pit floor construction on orchestra timbre will all be illustrated by results of numerous objective measurements, subjective surveys, and computer simulations.
Classical analysis of multilayered walls transmission loss in comparison with the F.E.M. and S.E.A. approaches
Authors: Fringuellino Marco , Guglielmone Claudio
Published: July 2002

Prediction models of the acoustical behavior of multilayered soundproofing walls are of increasing practical interest in the field of building trade, particularly for designing applications. This paper reports a classical approach to the problem, whose aim is the theoretical calculation of the transmission loss of acoustical insulating multilayered walls. The model is based on the straight compilation of the ‘‘progressive impedance’’ encountered by the acoustical wave in its propagation through the wall. The pressures ratios at the ending surfaces of each layer are then computable as a function of the progressive impedances obtained before. The same is true for the total ratio of the incident and transmitted pressure. The characteristic impedance and the propagation constant of each layer are needed as input parameters. The results of the classical analysis are compared with a set of measurements performed on multilayered soundproofing walls. At present an effort is being made to develop a Finite Element Method and a Statistical Energy Analysis approach of the problem, to compare these theoretical results with the classical ones on the same set of walls. The final aim is to show the accuracy, the advantages, and the main limits of each of these three methods.
Audibility of edge diffraction in auralization of a stage house
Authors: Torres Rendell R. , Kleiner Mendel
Published: July 2002

A recent model for edge diffraction based on physical acoustics was applied to the auralization of a source in a stage house. In this paper the results of two methods of comparison (computation of frequency responses and evaluation of binaural listening tests) are used to judge the audibility of diffraction effects associated with such geometries. The computed impulse responses are also compared via binaural listening tests with measured impulse responses from a physical scale model. Current results indicate that, from an auralization standpoint, first-order diffraction is significantly more influential than second-order diffraction, which can lie more than 45 dB below the direct or specularly reflected arrivals. Thus, higher-order (and computationally more prohibitive) edge diffraction calculations may possibly be simplified or even neglected in auralization programs. [Work supported by Axel & Margaret Axson Johnsons Stiftelse, Sweden.]
1000-km square scale propagation from ocean acoustic tomography experiment in the Kuroshio Extension region
Authors: Fujimori Hidetoshi , Nakano Iwao , Nakamura Toshiaki , Yuan Guan , Kamoshida Takashi , Kaya Akio
Published: July 2002

An ocean acoustic tomography observation was carried out at JAMSTEC (Japan Marine Science and Technology Center) in the Kuroshio Extension region, east of Japan, from July to September 1997. In this observation, five transceivers (consisting of a sound source and hydrophone array) were used and arranged 28°−36° N, 143°−154° E area. The longest distance between transceivers was about a 1000-km range and moored about a 1100-m depth. These transceivers were communicated to eight times (in August) or four times (in July and September) a day of each other and a time series of 1000-km square propagation data in this observation area were recorded and analyzed.
Three-dimensional classical dynamics of acoustic rays in a quasirealistic ocean
Authors: Wolfson Michael A. , Tomsovic Steven
Published: July 2002

Simulations on multimegameter range scales are presented for the dynamics of classical acoustic rays in a three-dimensional ocean waveguide which contains sound-speed inhomogeneities derived from an ocean general circulation model. The inhomogeneities include scales equal to and larger than mesoscales (>20 km). Both horizontal and vertical two-dimensional deterministic ray propagation through mesoscale environments are known to exhibit chaotic behavior in the sense of exponential divergence of neighboring trajectories that are initially infinitesimally separated. The extra degree of freedom provided by an acoustic phase speed that is fully three dimensional allows a richer, more complex structure to be revealed. We use a Hamiltonian prescription for the ray dynamics to reveal this structure geometrically via the phase space evolution of the associated Lagrangian manifold. The linear transformation matrices for this system which describe how elements of phase space volume distort under the action of the dynamics are examined for a large subset of trajectories. It is shown that their associated exponential rates of divergence vary orders of magnitude. The relative importance of horizontal versus vertical folding in the trajectory manifolds is also discussed.
Tissue harmonic imaging in cardiology and radiology applications
Authors: Averkiou Michalakis A. , Jago James R. , Roundhill David N. , Powers Jeffry E.
Published: July 2002

Harmonic imaging, the formation of ultrasound images from the harmonic components of the sonification signal, has been used in diagnostic ultrasound applications that utilize microbubble contrast agents. In the presence of microbubbles the harmonic signals are mainly due to nonlinear bubble oscillations. Recently, harmonic imaging has been applied without contrast agents, the harmonic signals arising from nonlinear propagation. Near-field artifacts and aberrations are reduced, and tissue borders are enhanced. The harmonic signals are at a lower amplitude and suffer more attenuation than the fundamental. However, the wide dynamic range, digital architecture, and signal processing capabilities of modern diagnostic ultrasound systems make it possible to utilize this tissue-generated harmonic energy for image formation. The nonlinear parabolic wave equation is used to model tissue propagation. A time domain numerical solution for a nonaxisymmetric source function is presented. Theory and experiments demonstrate harmonic generation in tissues. In vitro animal tissue experiments and in vivo collected radio frequency data are shown. A novel broadband pulsing scheme that uses two pulses with a phase difference of 180° is introduced, which results in cancellation of the propagated fundamental component and doubling of the second harmonic. Cardiac and abdominal harmonic images are shown.
A filter-based pulse-echo coded-excitation system for real-time cardiac imaging
Authors: Ebbini Emad S. , Shen Jian , Elbakri Idris
Published: July 2002

A new filter-based approach to pulse-echo imaging capable of parallel processing of multiple image lines has been developed. A major application of this approach is in true real-time 3D imaging, e.g., in cardiac imaging. The new approach utilizes a set of coded waveforms on transmit and conventional beamforming on receive. The transmitted codes produce a coded wavefront in the region of interest (ROI) and the receive beam is designed to be wide enough to allow echoes from the ROI to be received while rejecting echoes from outside the ROI. Image formation is achieved using a filter bank structure (post-beamforming) with every filter designed to extract echoes from a given direction in the ROI. The filter bank is derived from a regularized pseudoinverse operator of the (discretized) propagation operator from the ROI to the receiving array. Image reconstructions from speckle-generating phantoms using the new approach demonstrate its robustness and its potential in medical imaging applications. A complete description of the algorithm and its fundamental spatial and contrast resolution limits will be presented with illustrative experimental data. System architectures for robust real-time implementation will be discussed. [Work funded by NIH Grant HL57167.]
Image-based calculation of elevational B-scan separation
Authors: Krücker Jochen , Tuthill Theresa A. , LeCarpentier Gerald L. , Fowlkes J. Brian , Carson Paul L.
Published: July 2002

A speckle decorrelation technique was extended to high-frequency imaging to calculate the elevational (out of plane) separation of B-scan images. Pure speckle images were obtained with a 13-MHz, 1.5-D array in an elevational sweep over a phantom in 25-μm steps. For each pixel the plane-to-plane autocorrelation function was computed for a number of frames and fit to a Gaussian to find the depth-dependent scaling factor between the rate of speckle decorrelation and the actual frame spacing. From this the frame separation in sets of images containing speckle and structure was determined. To avoid using specular signals the mean intensity over standard deviation was employed to automatically detect areas of speckle. Based on the speckle decorrelation in these regions and using a 200*300 pixel subimage the frame spacing (100 μm) was calculated with 2% accuracy. The error increased with decreasing number of pixels and increasing separation of the frames. The technique can now be combined with high frame rate techniques to determine the elevational transducer motion during complex-motion 3-D scanning of the breast. [Work partially supported by the U.S. Army, Medical Contract DAMD17-96-C-6061 and the German Academic Exchange Service.]
Acoustic detection of a non-metallic object embedded in a ground material
Authors: Tavossi Hasson M. , Tittmann Bernhard R.
Published: July 2002

The preliminary results of acoustic detection of a nonmetallic object embedded in a ground material model at a shallow depth is investigated. The frequency range of the acoustic wave is optimized to the composition of the ground material model to obtain minimum attenuation of the received signal. The investigation considers both theoretical and experiment aspects of the problem to provide a better understanding of the interaction of the acoustic waves with the ground model and the embedded nonmetallic object. One part of the theory is aimed at finding the optimal acoustic signal with maximum ground penetration as a function of frequency, band width, signal energy, and other wave characteristics. The other part of the theory considers ground material characteristics such as grain size and contact areas, density, rigidity, and porosity. In the experimental investigation both through-transmission and pulse-echo methods are used. The results include the relationship between the optimal frequency for maxi-mum amplitude of the received signal, wave velocity, band-width, incident direction, signal energy, attenuation, ground material, and the embedded object characteristics.
Head wave data inversion for geoacoustic parameters of the ocean bottom off Vancouver Island
Authors: Laidlaw Mark C. A. , Godin Oleg A. , Chapman N. Ross
Published: July 2002

As a part of the Pacific Shelf-93 experiment, small explosive charges were used as underwater sound sources on the continental shelf off Vancouver Island. Broadband acoustic signals received by a vertical line array show the rich structure of precursors arriving prior to the direct water wave. Kinematic properties of the precursors are found to be consistent with those of head waves with different numbers of reflections at the ocean surface and bottom. Power spectra of the precursors reveal complicated interference structure, including deep nulls at relatively low frequencies. Positions of the nulls in the frequency domain and their depth dependence are shown to be sensitive to the variation of compressional wave speed in the bottom. A combination of beamforming, spectral analysis, and travel-time considerations is applied to separate and identify individual head wave arrivals in the sequence of precursor signals. Range- and depth-dependence of the travel times for individual head waves are inverted to determine compressional wave speeds in two distinct subbottom layers and to estimate the thickness of the sediment layers. The frequency- and range-dependence of the head waves amplitudes are analyzed to estimate the compressional wave attenuation. [Work supported by NSERC.]
Signal processing techniques to analyze the effects of guitar geometry on musical timbre
Authors: DiDomenico Dominic , Morris Kevin , Keilson Suzanne
Published: July 2002

Musical acousticians seek to answer many questions. What makes a particular instrument sound like it does? How do listeners perceive musical sounds? As an undergraduate design project, three guitars of varying shapes (‘‘box,’’ ‘‘triangle,’’ ‘‘octagon’’) were constructed and then compared to the traditional design. Materials were not varied, only the geometry of the sound boxes. The overall volume of each sound box was approximately equal. Digitized samples of the plucked strings of the constructed and traditional guitar were obtained. Conventional Fourier signal analysis of the notes from the three constructed guitars showed similarities to those of the traditional guitar. Other signal processing methods, such as short-time Fourier transform spectrograms, appropriate for the transient signal of the plucked string, were also investigated. Although the signals were broadly similar, analysis revealed differences, which might be attributed to the different geometries. A survey was conducted to assess listener’s preferences for the various instruments. The survey was conducted from taped samples of the plucked strings to correlate the importance of psychoacoustics and the physical signal in assessing instrument quality. Listener preference rated the octagon-shaped guitar as highly as the traditional one. [Work supported by the Loyola College Department of Electrical Engineering and Engineering Science, and the Loyola College Hauber Fellowship Fund.]
New noise impact criteria for high-speed ground transportation systems in the United States
Authors: Hanson Carl E.
Published: July 2002

New noise impact criteria have been developed by the U. S. Department of Transportation for the assessment of environmental impacts of high-speed rail projects. The environmental impacts of new high-speed rail projects in the United States are the responsibility of the Federal Railroad Administration (FRA). Adoption of these criteria will assure a uniform treatment of noise impacts of new high-speed rail systems throughout the country. The noise impact levels were developed based on well-documented criteria, and on comprehensive social survey data concerning annoyance due to transportation noise. Impact levels are based on the Schultz response curve and include consideration of the combination of project and ambient noise levels. Correction terms are included in the case of high-speed rail (and maglev) to account for startle effects from fast rise times. This presentation discusses the basis for the curves used as thresholds of impact—where noise mitigation is to be considered—and severe impact—where noise mitigation must be implemented. These criteria are included in a new guidance manual published by the FRA on ‘‘High Speed Ground Transportation Noise and Vibration Impact Assessment’’ and will be used throughout the USA.
Effects of traffic noise within the Madrid region
Authors: Recuero Manuel , Gil Constantino , Grundman Jorge
Published: July 2002

In this work the results of a 20-question survey about the acoustic environment made on the population of 17 towns are presented. Up to 7141 questionnaires were distributed, where 3272 were conducted in population centers with more than 100 000 inhabitants, 2695 in towns below 100 000 and above 50 000 inhabitants, and 1174 in towns with less than 50 000 inhabitants. The aim of this statistical work is to estimate the citizens’ opinions about noise sources in their municipalities, the annoyance that these sources produce, where and when the effect seems stronger, and how the noise affects residential areas. There were also questions concerning the opinions about protection against noise in dwellings and the presumed effects of environmental noise. Finally, it was intended to know how important the residents think the environmental noise problem is, how deep their knowledge is about their rights and the law, and what do they think about measures to improve the situation.
Control of exhaust noise from diesel electric locomotives using a hybrid active/passive system
Authors: Remington Paul , Hanna Douglas , Knight Scott
Published: July 2002

Diesel electric locomotives in the US represent a significant source of environmental noise under a variety of operating conditions. There are two primary sources of noise on these locomotives that will have to be reduced before significant reductions can be achieved: exhaust noise and cooling fan noise. Here a system is proposed that uses both active and passive components to control broadband exhaust noise below 3 kHz. The active system utilizes eight actuators and eight residual microphones in a feedforward configuration to control tonal noise below 250 Hz. The passive system is a compact exhaust silencer designed to control exhaust noise above that frequency while still fitting within the limited space available beneath the locomotive hood. Performance estimates for both the active and passive system will be presented along with the results of field tests of a breadboard active system on an F59PHI locomotive. [Work was supported under contract to the Federal Railroad Administration.]
Robust feedback active noise control algorithm for impulsive additive noise
Authors: Lee Sang-Wook , Sung Koeng-Mo
Published: July 2002

Active noise control (ANC) is a method to reduce the unwanted noise level by introducing secondary noise which has the same amplitude and antiphase with the primary noise. The feedback ANC algorithm is a kind of linear predictor and it uses only one sensor to get the information about the error signal and reference signal. The filtered-x LMS (least mean square) algorithm is widely used for implementing the ANC system because of its simplicity and good performance. But when the additive noise is impulsive, the performance of LMS-type algorithms are known to be lowered. In this paper, the robust feedback ANC algorithm is proposed when impulsive additive noise is present. Instead of the LMS algorithm, the proportion-sign algorithm (PSA) which is a mixture of LMS and dual sign algorithm (DSA) is used to estimate the primary noise field robustly when impulsive additive noise is present. Computer simulations were performed under impulsive additive noise circumstances and showed better performance with the proposed algorithm than that with the conventional LMS algorithm.
Effect of the flat panel resonance on the absorption characteristics of a microperforated-panel absorber
Authors: Liu Ke , Jiao Fenglei , Ding Hui
Published: July 2002

The flat panel resonance can affect the absorption characteristics of a microperforated-panel absorber. This effect was not considered in the existing theory. In this paper, the method of measuring the no-hole flat panel resonance is applied to analyze this effect. The sample is a stainless steel microperforated panel, with the diameter of perforations d=0.2 mm, the separation of holes b=2.5 mm, the thickness of panel t=0.2 mm, and the thickness of cavity behind the microperforated-panel D=95 mm. Theoretically, its resonance frequency is 508 Hz, and the maximum absorption coefficient is 0.9. However, the experimental result shows that there are two absorption peaks at 280 and 680 Hz, respectively. Two resonance peaks also appear as a sample of a no-hole flat panel is measured. When the thickness of cavity is changed, the result is still similar. Moreover, when sound intensity is enhanced, two resonance peaks also exist, and move towards low frequency. Therefore, it is possible that the absorption frequency band can be widened, or the absorption coefficient within absorption frequency band can be enlarged, by using the resonance frequency of different materials. [Work supported by the National Natural Science Foundation of China.]
Theory on mass transport generated by ultrasonic Lamb waves
Authors: Du Gonghuan , Zhu Zhemin , Zhao Xiaoliang , Wu Junru
Published: July 2002

It was reported that acoustic streaming generated by a Lamb-wave sensor (LWS) can be used to pump fluids and transport solids in a small-scale system [Appl. Phys. Lett. 59 (1991)]. The speed, observed by attracting small polystyrene spheres suspended in water, is proportional to the square of the wave amplitude and is about 100 μm/s for 6.5-nm surface displacement of the plate. In this paper, a fairly practical model, i.e., a thin isotropic plate in contact with a viscous liquid layer on its side, is presented. Based on the rigorous propagation theory of Lamb waves and Nyborg’s [Phys. Acoust. B 2, 265–295] and Bradley’s [J. Acoust. Soc. Am. 100, 1399–1408 (1996)] theories of acoustic streaming, the first-order ultrasonic field in the viscous liquid layer and the second-order mass-transport velocity distribution of the liquid are calculated. Comparison between our theoretical values of the Ao-mode Lamb wave and Moroney et al.’s experimental result is given. It shows good agreement. [Work supported by NSF of China.]
Gap detection threshold in ears with and without spontaneous otoacoustic emissions (SOAEs)
Authors: Smurzynski Jacek , Probst Rudolf
Published: July 2002

Gap detection was measured for two groups of normally hearing young adults using broadband (0.1–12 kHz) noise stimuli presented monaurally through an insert earphone. Group I consisted of subjects who exhibited both strong SOAEs and click-evoked otoacoustic emissions (CEOAEs). Group II included individuals with no SOAEs in either ear and CEOAE levels that were <50th percentile of a laboratory normative database. An adaptive 2IFC procedure was used to determine the hearing threshold of the noise stimulus for each ear tested. Next, an adaptive 2IFC gap detection task was performed for the stimuli presented at either 10-, 20-, 30-, or 50-dB sensation level (SL). At the stimulus level of 10 dB SL, the ears in group I exhibited greater intersubject variability and higher mean gap detection thresholds than those in group II. The results at higher SLs did not show any differences between the two groups and were consistent with the data in the literature. A short gap in the broadband stimulus presented near the hearing threshold is masked by SOAEs and strong CEOAEs that are evoked by the stimulus. [Work supported by Swiss National Science Foundation.]
Spatial distribution of energy in SEA acoustic volumes
Authors: Davis Evan B.
Published: July 2002

Statistical energy analysis (SEA) methods predict the space average energy in acoustic volumes. The predicted SEA acoustic energy levels are converted to SPL levels using the assumption that the sound field is diffuse. The SEA prediction methods estimate a single, space averaged value for the acoustic energy in the enclosure, whereas the theory of diffuse sound fields in enclosures predicts a nonuniform energy distribution in the enclosure. The energy levels and SPLs increase near the boundaries of the enclosure. The theory of diffuse sound fields can therefore be applied to generate an estimate of the spatial distribution of sound energy from the predicted spatial average. The sound field distributions are based on the Waterhouse corrections for reverberation rooms [Waterhouse, J. Acoust. Soc. Am. 27, 247–258 (1955)]. The use of the Waterhouse correction in the interpretation of SEA predictions is most useful in acoustic volumes which are relatively small in relation to the acoustic wavelengths of interest. Enclosures which have dimensions on the same order as the acoustic wavelengths are commonly encountered in transportation systems such as automobiles and airplanes.
Familiarity and pronounceability of nouns and names
Authors: Surprenant Aimée M. , Hura Susan L. , Harper Mary P. , Jamieson Leah H. , Hockema Stephen A. , Hsueh Tsung-Hsiang , Johnson Michael T. , Srinivasan Pramila N. , Thede Scott M. , White Christopher M. , Long Glenis , Rout Ayasakanta
Published: July 2002

Proper names have several properties that create problems for speech recognition systems: the number of names is large and ever changing, names can be borrowed directly from other languages and may not conform to usual pronunciation rules, and the variety of pronunciations for names can be high. Because the set of proper names is so dynamic and machines are notoriously poor at phoneme recognition, a promising approach to designing a name recognition system is to incorporate statistical aspects of proper names (e.g., frequency, familiarity). Unfortunately, there exists relatively little data on the distribution of names. Ratings of familiarity and pronounceability were obtained for a randomly chosen sample of 199 surnames (from 80 987 entries in the Purdue phonebook) and 199 nouns (from Kucera–Francis). The ratings for nouns versus names are substantially different: nouns were rated as more familiar and easier to pronounce than surnames. Frequency and familiarity were more closely related in the proper name pool than the word pool, although the correlations were modest. Ratings of familiarity and pronounceability were highly related for both groups. The value of using frequency and the ratings of familiarity and pronounceability for predicting variations in actual pronunciations of words and names will be discussed.
Relation between discrimination and identification of English vowels
Authors: Kewley-Port Diane , Neel Amy T.
Published: July 2002

Although resolution for formant frequency is considerably better than needed for vowel identification in English, adjacent vowels in the F1-by-F2 plane may not be equally discriminable. Kuhl has reported better discrimination for poor vowel exemplars than good exemplars (‘‘perceptual magnet effect’’) for a few vowels in high stimulus uncertainty tasks. The purpose of this experiment was to determine the relation between discrimination thresholds and the judgment of vowels as good, confusable, or non-English. A large F1-by-F2 vowel space (190 tokens) encompassing five English front vowels was synthesized in equal bark steps. Based on an identification task with goodness ratings, two vowels representing each goodness category were selected for the discrimination task. Eight listeners participated in a minimal uncertainty discrimination task to estimate thresholds for F1 and F2. Analysis of variance of the 12 thresholds (converted to Δ Barks) found that thresholds for non-English vowels were significantly poorer than the other two categories, whereas thresholds for good and confusable vowels were similar. These results are not consistent with the perceptual magnet effect. Vowels within areas of the F1-by-F2 space judged as good exemplars are not discriminated more poorly than those in confusable areas. [Work supported by NIH-NIDCD-02229.]
Inversion of multimegameter-range acoustic data for ocean temperature
Authors: Dushaw Brian D.
Published: July 2002

Ocean sound speed (temperature) derived from ray travel times obtained from multimegameter-range acoustic transmissions may be inaccurate when the reference ocean state is inadequate for linearized inversion. When the reference (e.g., the Levitus atlas) is significantly different than the ‘‘true’’ ocean, the ray paths defined by that reference inaccurately represent the true sampling. In addition, natural oceanic variations, such as the evolution of a summer mixed layer, can significantly change the ray sampling. The guiding principle for inversion is that ray travel times associated with the inverse solution must match the measured travel times. Inversion of travel times relative to an arbitrary mean can be inaccurate. Accurate inversions may be obtained by iterative methods or by a better choice of reference ocean, but the solution of this nonlinear problem is not always obvious. Inadequate depth resolution for the vertical structure in the inverse solution can make calculation of travel times difficult. A linearizing time-dependent reference ocean can reduce the solution uncertainties since the a priori model variances are less. Simulations employing the Levitus ocean atlas, so that the ‘‘true’’ ocean state is known, are used to define the problems of nonlinear inversion and to find accurate inversion methods.
Flow function method for the display of streamlines on ultrasonic color flow image
Authors: Ohtsuki Shigeo , Tanaka Motonao
Published: July 2002

Ultrasonic color Doppler technique is useful for the observation of the blood flow in heart. The information on blood flow by this technique is Doppler velocity, which is the velocity component in the direction of the beam emitted from the ultrasonic probe. The ultrasonic color Doppler image displays the distribution of Doppler velocity on an observation plane using red and blue corresponding to toward flow and away flow, respectively. Streamlines overlapped on ultrasonic color Doppler image are considered to be suitable for the quantitative evaluation of the characteristics of the flow. In this investigation, a new concept of the flow function is proposed in order to obtain streamlines on an arbitrary section of three-dimensional flow for visualizing and evaluating the characteristics of the flow. Flow function at a point on a plane is defined as the flow rate along a path from a reference point. This function is a multivalued function. The section of the three-dimensional display of the function at a constant value shows streamlines. The streamlines separated by an arbitrary unit flow can be fixed with the approximated point sources of the unit flow. These point sources are named simple sources. These simple sources are start points or end points of streamlines.
Synthesis and control of synthesis using a generalized diphone method
Authors: Rodet Xavier
Published: July 2002

Well known in speech synthesis, a diphone is a segment of sound extending from the middle of a phone to the middle of the next phone. More precisely, in order to allow for time and pitch modifications, a diphone is the set of synthesis parameters representing this segment in some given synthesis technique. A speech sentence is synthesized by concatenating diphones found in a dictionary. The concept of diphone has been extended into generalized diphone control in order to build musical phrases from dictionaries of sound segments by concatenating and articulating them. Sound segments stored in dictionaries can comprise any recorded sound as well as instrumental sounds. Any analysis–synthesis method such as source-filter synthesis, additive synthesis, or physical model can be controlled in terms of generalized diphones. Presented here are several important improvements to diphone control and synthesis implemented in a software named ‘‘Diphone,’’ such as hierarchical structure of sequences and subsequences of diphones, variable speed of transitions, various smoothing functions at diphone boundaries, etc. ‘‘Diphone’’ proved itself a very flexible and inspiring tool for composition and synthesis, which composers are using at IRCAM and outside of the institution.
Efficient synthesis model for bowed strings using a nonlinear bow–string interaction model
Authors: Smith Julius O.
Published: July 2002

A digital simulation method for bowed strings was developed which combines features of preexisting ‘‘digital waveguide’’ and ‘‘commuted synthesis’’ methods. In both methods, sampled acoustic traveling waves are explicitly modeled in the string, and string losses are lumped at one or two points along the string for computational efficiency. In the digital waveguide case, the bow–string interaction model is based on the theory of McIntyre, Schmacher, and Woodhouse [J. Acoust. Soc. Am. 74 (1983)]. The commuted synthesis model [J. Smith, Int. Computer Music Conf., pp. 64–71 (1993)] interchanges the order of the string and body (assumed linear and time invariant), and therefore cannot support a nonlinear bow–string interaction model. However, according to theory and measurement of bow–string interaction, disturbances sent out by the stick–slip process along the string are fundamentally impulsive in nature. The proposed method consists of a full nonlinear dynamic model for the bow and string, without body, followed by an ‘‘impulse prioritizer,’’ which detects the most important impulsive events reaching the bridge, followed by a commuted synthesis model which drives a second string model with body impulse-responses initiated at the time and amplitude of each high-priority pulse.
Bubble dynamics in non-Newtonian fluids
Authors: Allen John , Roy Ronald A.
Published: July 2002

Nonlinear oscillations of gas bubbles in viscoelastic non-Newtonian fluids remains a relatively unexplored area, but one of increasing importance with the growing use of high intensity ultrasound and contrast agents for imaging enhancement. Previous work by the authors focused on an analytical and numerical study of bubbles governed by linear viscoelastic constitutive equations. This work was constrained by the small deformation assumptions inherent in the equations. New numerical studies of objective viscoelastic constitutive equations reveal the limitations of the linear viscoelastic models and show many novel results. In particular, the risk of bioeffects from medical ultrasound and harmonic imaging applications are stressed. Questions about the trace of the stress tensor are also discussed. [Work supported by NIH and DARPA.]