bluuebunny/crossref_metadata_2025 huggingface dataset:
An acoustic telemetry system for use on undersea ranges
Authors: Jarvis Susan , Blackmon Fletcher
Published: July 2002

The underwater digital acoustic telemetry (UDAT) system is a set of underwater acoustic modems developed by the Engineering, Test and Evaluation Department at the Naval Undersea Warfare Center Division, Newport. The modems were designed to provide robust underwater acoustic communication at throughput data rates of approximately 900 and 1800 bits per second. Since February 1996 the modems have been used in nine exercises in five diverse deep-water and shallow-water environments. The results from the tests are very encouraging. Continuous, low-error transmission has been achieved over many minutes (∼45 min). Detection ranges for the system have exceeded 4 km in shallow water and 8 km in deep water. In April and September 1997, the acoustic modems were used for continuous, full duplex, encrypted communications at the Atlantic Undersea Test and Evaluation Center (AUTEC). This paper will review the design of the UDAT system, discuss the results from the field tests, and present the current status of the modems. [Development of the modems was sponsored by Naval Air Systems Command, PMA-248.]
Ranges as undersea acoustic networks
Authors: Catipovic Josko
Published: July 2002

Contemporary undersea acoustic ranges can be used as undersea local area networks to support cellular acoustic communication with mobile and fixed platforms and for associated local and remote sensing measurements. Bidirectional range nodes, particularly those containing multiple receiver hydrophones, inherently support high-speed acoustic communication with SSNs, UUVs, or other network users, and the range area of coverage can be used as an undersea cellular local area network. This talk will present an overview of acoustic local area network (ALAN) technology and highlight its compatibility with deployed and proposed ranges. A network communications protocol suitable for simultaneous communication with multiple platforms will be presented and shown to be compatible with the proposed East Coast Shallow Water Training Range. Experimental results in acoustic communication and networking relevant to range applications will be summarized. Adding an acoustic communication capability to a range greatly expands its usefulness. The talk will describe several applications ranging from SSN communications to cooperative UUV behaviors. Several related efforts are currently under development and ready for transition to an undersea range network. High-resolution signal processing required for acoustic communications also allows the use of range nodes for undersea remote sensing. Several applications of ranges to remote sensing of sound speed, current, and turbulence are realizable as extensions to processing required to support acoustic networking. An example network application for mapping current flows will be discussed, with extensions to travel-time tomography for sound speed inversion and scintillation tomography for volume mapping of turbulence.
Error-rate performance of convolutionally encoded, interleaved QPSK and 8PSK TCM underwater acoustic telemetry
Authors: Blackmon Fletcher
Published: July 2002

A preliminary error-rate performance of comparison was conducted between two promising signal architectures for underwater acoustic telemetry modem use. The first candidate signal is a convolutionally encoded and interleaved QPSK modulation. The second candidate signal is an 8PSK trellis coded modulation (TCM). In both cases, the convolutional encoder is the same, i.e., k=1, n=2, and const=7. A short test in April of this year was performed at the Atlantic Undersea Test and Evaluation Center (AUTEC) in approximately 1000 m of water. Signals were transmitted from a seabottom mounted UQC transducer and were received by a hull-mounted transducer on a drifting surface vessel. One-hundred telemetry packets from each signal type were transmitted and were adaptively equalized within the receiver prior to decoding. The QPSK signal outperformed the TCM signal with bit error rates of 5e-6 and 1.4e-5, respectively.
Long-range signaling in shallow ocean transmission channels
Authors: Rice Joseph A. , Shockley Richard C. , McDonald Vincent K. , Green Dale , Proakis John G.
Published: July 2002

Undersea acoustic signaling typically exploits the slowly varying, sparse impulse response of the physical channel. This paper addresses channels where the range is much greater than the water depth and where propagation ducts are absent. Here the existence of stable, discrete multipaths cannot be assumed. Transmission in these channels relies on forward-scattering from the rough seafloor and sea-surface boundaries. The Rayleigh and Ricean fading channel models are used to statistically describe the unpredictable transmission as a complex-valued Gaussian random process. Motion of the faceted scatterers at the time-variant sea surface further affects the transmitted signal by smearing the spectral content with Doppler spread. The prolonged signal duration and broadened signal bandwidth characterize a doubly spread transmission channel. An additional serious impediment is the dynamic and nonwhite channel noise. Coherent demodulation, channel-adaptive equalization, and matched-filter processing are inhibited in such doubly-spread, noisy channels. Robust 1- to 1000-bit/s undersea acoustic telemetry and ranging (telesonar) techniques are being developed by the U.S. Navy using digital coding, spread-spectrum modulation, diversity signaling, and noncoherent demodulation. Incidental benefits include CDMA networking, transmission security, and low power. [Work supported by ONR-321 and the SPAWAR SBIR Program.]
The neutral network analysis of false killer whale (Pseudorca crassidens) vocalizations
Authors: Murray Scott O. , Mercado Eduardo , Roitblat Herbert L.
Published: July 2002

This study reports the use of unsupervised, self-organizing neutral networks to categorize the repertoire of false killer whale vocalizations. Self-organizing networks are capable of detecting patterns in their input and partitioning those patterns into categories without requiring that the number or types of categories be predefined. The inputs for the neural networks were two-dimensional characterizations of false killer whale localizations where each vocalization was characterized by modulations in duty cycle and peak frequency [Murray et al. (in prep.)]. The first neural network used competitive learning, where neurons in a competitive layer distributed themselves to recognize frequently presented input vectors. This network resulted in classes representing typical patterns in the vocalizations. The second network was a Kohonen feature map which organized the outputs topologically, providing a graphical organization of pattern relationships. The networks performed well as measured by (1) the average correlation between the input vectors and the weight vectors for each category, and (2) the ability of the networks to classify novel localizations. The techniques used in this study could easily be applied to other species and facilitate the development of objective, comprehensive repertoire models.
Can one allocate noise-induced hearing loss retrospectively for individuals?
Authors: Clark William W.
Published: July 2002

For decades, physicians and other clinicians have been asked by the legal profession to answer two simple questions about individuals presenting with marked or measurable hearing loss: (1) How much of the observed hearing loss was caused or contributed to by excessive exposure to occupational noise? (2) How did this hearing loss accumulate over time? Until recently, answers provided represented no more than clinical guesswork. However, with the adoption of the international standard ISO 1999, ‘‘Acoustics-Determination of Occupational Noise Exposure and Estimation of Noise-Induced Hearing Impairment,’’ and its heterozygous twin, ANSI Standard S3.44, of the same title, clinicians and scientists now have a tool, or framework from which they may construct answers to the two questions. The precision of the answers, however, depends entirely upon the precision of the framework and data underlying the standard. This presentation will summarize the weaknesses in the ISO standard and will support the conclusion that one still cannot, with any reasonable degree of scientific certainty, answer either question for an individual.
Modal densities of simple dynamic systems
Authors: Maidanik G. , Becker K. J. , Dickey J.
Published: July 2002

Modal densities are parameters that participate in the specification of a statistical energy analysis (SEA) model of a complex composed of a number of coupled dynamic systems. The modal density n(ω) of a dynamic system in this context, is defined by the equation: (Δω)n(ω)=N(ω), where a frequency band {ω,Δω} is identified by the bandwidth (Δω) centered at the frequency (ω) and N(ω) is the number of resonant modes, in this dynamic system, for which the resonance frequencies lie within this frequency band. Usually, it is assumed that the modal density n(ω) is a continuous and/or a monotonic function of the frequency (ω). Moreover, it is often argued that the modal densities are identical for dynamic systems that possess the same propagation speeds and the same spatial extents, i.e., the same volumes, areas, or lengths. The implication that the assumption and the argument are particularly valid for the higher frequency range, has designated SEA suitable for ‘‘a high-frequency analysis only.’’ The modal densities are examined, for a few simple dynamic systems, in an attempt to reassess the assumption, the argument, and the designation.
The effect of wave energy trapping in inhomogeneous elastic structure
Authors: Grinchenko Victor T.
Published: July 2002

In contrast to an acoustical waveguide the distribution of characteristics of normal modes with respect to the cross section in elastic waveguides is strongly dependent on frequency. Such dependence gives rise to several interesting wave phenomena, which take place in inhomogeneous structures. One of them, namely, energy trapping, is quantitatively and qualitatively considered in this report. In terms of the theory of elastic waveguides, this phenomenon manifests itself in an anomaly of the frequency functions of the reflection (transmission) coefficients of energy in the composite waveguides. These functions are calculated for both in-plane and flexural motions in waveguides consisting of two semi-infinite layers with equal thickness. The results are systematized with respect to the ratio of wave impedances of the layers. The frequency bands where the energy trapping is effective are determined. The dependence of these bands on the mechanical characteristics of materials is different for the in-plane and flexural waves. The mechanism of redistribution of energy among the incident and reflected waves is shown. The function of the normal mode with opposite signs of phase and group velocities is illustrated.
Effects of the size and form of the orofacial structure on vowel production
Authors: Honda Kiyoshi , Hashi. Michiko , Wu Chao-Min , Westbury John R.
Published: July 2002

Individual difference of the orofacial structure and its articulatory and acoustic influences have been studied based on the University of Wisconsin x-ray microbeam database [Honda and Wu, J. Acoust. Soc. Am. 100, 2598(A) (1996)]. This work describes a subsequent statistical analysis on the same datasets. The outlines of the orofacial structure of 20 subjects (10 Japanese and 10 English speakers) were extracted from lateral x-ray scan images and mapped onto a midsagittal reference space defined by the palatal plane and anterior nasal spine. Vowel records in the microbeam data were analyzed to reveal the effect of orofacial morphological difference on vowel articulation and acoustic parameters. In the main results, the size parameters were commonly correlated with the three formant frequencies. In particular, the lower facial height or the size of the mandibular symphysis showed a clear tendency to inversely correlate with the first formant. Among the parameters of form, oral cavity aspect ratio and mandibular symphysis shape thought to be possible factors affecting vowel gestures, which were reflected in the spread of the second formant in all the vowels analyzed. A multiple regression model of speaker characteristics will be discussed.
Acoustic characteristics of perceived vowel quantity and quality in English and Norwegian vowels
Authors: Nylund Anne , Behne Dawn M.
Published: July 2002

In Norwegian and English, as in many languages, vowels can be characterized in terms of distinctive phonological vowel quantity and vowel quality. Norwegian is described as having phonological distinction between long and short vowels, whereas in English vowel quality is generally seen as more distinct. However, both vowel quantity and quality can acoustically be realized in vowel duration and the accompanying characteristic resonance of a vowel. The goal of this project is to examine the perceptual weight of vowel duration and the first two formant frequencies for Norwegians when hearing Norwegian vowels versus English vowels. Six sets of words based on three Norwegian vowel pairs and three English vowel pairs were resynthesized. The words in each set were resynthesized using the members of the pairs as the end points of a matrix with five degrees of vowel duration and five degrees of F1 and F2 adjustment. Based on judgments by native Norwegian listeners, results to date suggest that native listeners of Norwegian will be more perceptive to differences in vowel duration than to spectral characteristics. The results for the English and Norwegian vowels will be compared and discussed in terms of native language influence on vowel perception.
Combining frication and glottal constriction: Two solutions to a dilemma
Authors: Maddieson Ian
Published: July 2002

In ejective consonants the airflow generating the sound comes from raising the larynx with the glottis closed. This pistonlike action only moves a small volume of air. Ejective constants are usually stops, and raising the larynx provides sufficient pressure for a salient burst. A handful of languages are described as having ejective fricatives, but how can one sustain a continuant, such as a fricative, using the ejective mechanism? Two languages reported to have sounds of this type are Tlingit (Na-Dene) and Yapese (Austronesian). Acoustic and aerodynamic data on the fricatives in these languages have been collected in recent fieldwork. This demonstrates two different solutions to the problem of combining glottal closure and frication. Tlingit has genuine ejective fricatives, glottal closure overlaps the entire frication duration, and intraoral pressure reaches a higher peak but for a shorter duration than in a pulmonic fricative. The duration of frication appears to be prolonged by narrowing the constriction more than for a pulmonic fricative. In Yapese the frication and glottal constriction are sequenced; intraoral pressure is the same in plain and glottalized fricatives. Both patterns provide clear contrast by solving the problem of combining frication and glottal closure in different ways.
The relation between the age of learning a second language (L2) and the degree of foreign accent
Authors: Liu Hua , Flege James E. , Yeni-Komshian Grace H.
Published: July 2002

A 1995 study by Flege, Munro, and MacKay [J. Acoust. Soc. Am. 97, 3125–3134 (1995)] examined the degree of perceived foreign accent in English sentences spoken by Italians whose ages of arrival (AOA) in North America ranged from 2 to 22 years. Two surprising findings were obtained: (1) A linear relation was found to exist between AOA and the degree of perceived foreigen accent; and (2) many of the native Italian subjects who began learning English as young children were found to have detectable foreign accents. To determine if these findings were generalizable, a comparable population of native Korean subjects was examined. English sentences spoken by 240 Koreans and 24 native English (NE) controls were rated for degree of foreign accent by ten native English-speaking listeners. As in the earlier study, a linear relation was found to exist between AOA and degree of accent; and sentences spoken by Koreans who arrived in the United States as young children received significantly lower ratings than those spoken by the NE controls. It appeared, however, that the native Korean subjects generally had stronger foreign accents than did the native Italians subjects. Possible explanations for this cross-language difference will be discussed. [Work supported by NIH.]
The influence of prosodic phrasal boundaries on the resolution of semantic lexical ambiguity
Authors: Schafer Amy J. , Speer Shari R.
Published: July 2002

Two experiments explored the effect of two levels of prosodic phrasing, the intonational phrase (IPh), and phonological phrase (PPh), on the interpretation of ambiguous words in sentences. In experiment 1, polysemous words were presented in semantically neutral sentence-initial clauses, followed by either an IPh or PPh boundary. A second clause resolved the ambiguity. End-of-sentence judgment times showed that reanalysis to the subordinate meaning of the ambiguous word took longer following IPh than PPh boundaries, suggesting that more extensive interpretive processing had taken place following the higher-level boundary. In experiment 2, strongly-biased polysemous words were presented in clauses weakly biased toward their subordinate interpretation, followed by either an IPh or PPh boundary. Following IPh boundaries, visually presented semantic associates of the subordinate meaning of the polysemous word were named faster than targets associated with the dominant meaning. This difference was not found for PPh conditions. Phonological and phonetic analyses will be presented. It will be argued that these processing effects cannot be explained solely on the basis of durational differences between IPh and PPh boundaries. The results support the view that the human sentence processor builds a prosodic representation and is guided by it during semantic interpretation. [Work supported by NIMH.]
Source identification—Theory and experiment
Authors: Hartmann William M. , Rakerd Brad , Gaalaas Joseph B.
Published: July 2002

A source identification experiment measures the ability of a listener to identify the location of a sound source. The experimental results are in the form of an rms error and a variability. The experiment can be described in terms of a statistical model in which the listener’s responses are determined by the width and bias of an internal distribution. Model calculations were performed to aid the design of source identification experiments. Calculations in the limit of small spans (≪180 deg) showed that the experimental variability provides a good estimate of the width of the internal distribution if both the bias and the source spacing are not greater than the width. The results are insensitive to the number of sources in the array. Specific models were developed for dealing with large biases, a frequent experimental occurrence. Calculations in the limit of large spans showed that experiments that ignore front-to-back reversals may lead to misleading results. Experimental tests of the models, using four human listeners in experiments having 3, 6, 12, or 24 sources, generally corroborated the models for both easy and difficult localization conditions. [Work supported by the NIDCD, DC00181.]
Estimating the spectrum of a moving sound source
Authors: Neelon M. F. , Jenison Rick L.
Published: July 2002

A recent analysis of moving sound sources has revealed that information in the dynamic acoustic array can specify fundamental attributes of the source, such as its distance, speed, and spectrum [R. L. Jenison, Ecol. Psych. (1997)]. A model based upon the Kalman filter is introduced which successfully estimates such distal properties using only observed proximal information. The experiment investigated whether 3 specific forms of acoustic information (amplitude, frequency, interaural phase differences) are necessary and sufficient to estimate the intrinsic frequency of a simulated moving sound source. Simulations show the model most accurately estimates intrinsic frequency when all three sources of acoustic information are included in the signal; when dynamic changes in one or more of these parameters are removed, the model’s performance degrades. To test human subjects, a technique was developed to generate moving signals from a sample of a natural moving sound source. This process allowed for precise control over the signal’s virtual and acoustic parameters. Subjects in a 2AFC design are required to guess the unshifted spectra of the moving stimuli as dynamic changes in the acoustic parameters are systematically removed. [Work supported by NSF.]
Multipath compensation in shallow-water environments using a virtual receiver
Authors: Siderius Martin , Jackson Darrell R.
Published: July 2002

An acoustic technique for the compensation of signal distortion due to propagation is developed in analogy with an astronomical technique in which light from a bright ‘‘guide star’’ is used to correct atmospheric aberration of weaker objects that are nearby in the angular sense. The acoustic technique investigated here uses a vertical array to receive both the signal from a broadband ‘‘guide source’’ and an unknown ‘‘objective source’’ which propagates over a partially shared path. The algorithm is a spatial-temporal cross correlation of the two signals and is termed a ‘‘virtual receiver,’’ as the output approximates the signal that the unknown source would produce at the location of the guide source. By strategically locating the guide source, many of the distorting effects of an unknown propagation region can be removed, including mode coupling and multipaths. Results are given for a variety of numerical experiments in both the time and frequency domains. The numerical simulations are used to illustrate the virtual receiver algorithm in range-dependent, shallow-water environments exhibiting lossy propagation and mode coupling.
Three-dimensional propagation effects near the Mid-Atlantic Bight
Authors: Smith Kevin B. , Chiu Ching-Sang , Miller James H. , Lynch James F. , Gawarkiewicz Glen G.
Published: July 2002

A recent set of experiments sponsored by the Office of Naval Research has been conducted in the vicinity of the Mid-Atlantic Bight shelf break region off the coast of New Jersey. The environment in this region is highly variable in both range and bearing due to the shelf break slope and its associated front. One of the goals of this Shelf Break Primer project is a determination of the influence of azimuthal coupling on the 3-D propagation. By using a fully 3-D PE model and comparing with results from the associated 2-DxN PE approximation, a quantitative assessment of the influence of azimuthal coupling can be made. Some sample benchmark cases are performed to illustrate this method. Using this approach, the significance of both the sloping bathymetry and the shelf break front will be considered separately and combined. In addition, the influence on the arrival time structure of a broadband pulse, similar to a signal employed during the experiments, will be determined. Characteristic features of such 3-D effects in the pulse propagation will then be explored in the available data sets. [This work is supported by the Office of Naval Research, Code 321OA.]
Predictions of ambient noise directionality and array response in range-dependent environments with CANARD and RANDI-2
Authors: Brind Richard J. , Harrison Chris H. , Hamson Rachel M.
Published: July 2002

There has been recent interest in the prediction of array performance in shallow-water environments. Several models have been developed in the UK over the last few years for predicting ambient noise directionality and array response in complex range-dependent environments. One model, CANARD, has been developed to provide quick and reliable predictions in realistic environments. It is a ray-based model but avoids using a propagation model in the usual sense. Environmental information can be read in from databases or specified manually. The model can treat nonuniform distributions of noise sources, wide vertical separation of hydrophones, cross slopes, and near-surface and duct leakage effects. Another model, RANDI-2, is a full-wave model based on the SUPERSNAP propagation model. It has been improved to calculate the wind noise component using virtual modes rather than the previous numerically intensive fast-field program method, and to improve accuracy at short range for the calculation of source and near-field shipping contributions. Software to generate input parameters from databases has also been produced. Details of the theory and capabilities of the models are explained. Examples of predictions in the SW approaches for a number of different types of array are presented. [Work supported by MOD(UK).]
Sound absorption by nonhomogeneous thin porous layers
Authors: Atalla Noureddine , Panneton Raymond , Allard Jean-Franois
Published: July 2002

Sound absorption and sound transmission of nonhomogeneous porous materials made of random or regular arrays of porous media is investigated using the finite-element method. The configuration studied consists of a column of porous materials inserted in an infinite rectangular waveguide. In the finite-element model, an equivalent fluid model for the porous material is coupled to the modal behavior in the waveguide. By comparison with analytical calculations, the difference between absorption mechanisms in three-dimensional porous patchworks and simpler stratified materials is investigated. In particular, the importance of nonpropagative mechanisms is enlightened. Also, the transmission loss performance of nonhomogeneous porous materials compared to homogeneous porous material is discussed. [Work supported by Bombardier, Inc., Canadair and NSERC.]
Conversational speech recognition
Authors: Crystal Thomas H.
Published: July 2002

A hot topic in speech recognition is developing technology for the automatic transcription of telephone conversations. The recognizer must contain robust language, pronunciation, and acoustic models that embody the world and topic knowledge and the understanding of syntax and pronunciation, which the talkers have and use in decoding each other’s acoustic signals. Partly because of the talkers’ shared knowledge and the casual, unprepared nature of the speech, the signals have dysfluencies, incomplete and ungrammatical expressions, and ‘‘lazy,’’ reduced articulation of words. Conversational speech recognition error rates, measured in the NIST Hub-5 evaluations, are 45% for English and 66% to 75% for Spanish, Mandarin, and Arabic. To improve this performance, the shared knowledge must be represented in a mathematical framework, which facilitates the efficient search of the sentences of a language to decode the speech. Recent work, including workshops at Rutgers CAIP and Johns Hopkins CLSP, has included the investigation of, among other techniques, multistream processing, frequency warping, adaptation of pronunciation and acoustic models of phones, pronunciation modeling, syllable-based recognition, dysfluency and discourse-state language models, and link grammar parsing. This talk will review how knowledge is represented in the recognizer architecture, searching procedures, and the results of the various investigations.
Robert Young and the measurement of community noise
Authors: Galloway William J.
Published: July 2002

Throughout his career in acoustics Robert Young has maintained a major interest in the development of good measurement practices for a variety of acoustical measurements. One of his major interests has been in the measurement of outdoor sounds as experienced by community residents. He has been one of the long-term, principal advocates of the use of an A-weighted sound level as a primary measure for noise as it relates to the response of people. He has shown that the A-weighted sound level can be used to describe the acoustical environments produced by sound sources as diverse as those of the quasisteady noise at an industrial site, the intermittent noise such as that produced by sequences of aircraft operations around an airport, to the impulsive sound produced by a sonic boom from the landing of a space shuttle orbiter.
On the units of Sabine absorption and the Sabine absorption coefficient
Authors: Pope J.
Published: July 2002

A century ago, W. C. Sabine demonstrated that sound absorption is fundamental to room acoustics, and introduced the idea for a measure of absorption that has become known as Sabine absorption. Because Sabine absorption and area have the same dimension, e.g., square-meters, many acousticians presume that Sabine absorption has the units of area. For a number of years Robert W. Young has been trying to convince us otherwise, arguing that a dimensionless logarithmic unit distinguishes Sabine absorption from area. [By dimensionless logarithmic unit is meant a scale factor that defines unity on a log scale. The neper (Np) and the decibel are two such quantities.] The metric sabin (Sa) is a unit of Sabine absorption. Defining the metric sabin to be one-half neper square-meter (1 Sa=0.5 Np m2) introduces a logarithmic unit while preserving the numerical scaling of current measurement practice. It follows that the unit of the Sabine absorption coefficient is the half-neper, and a difference between the Sabine absorption coefficient and coefficients from other methods becomes explicit. Among other things, a partial explanation for the experimental observation that Sabine absorption coefficients can exceed unity is provided.
The adaptive silicon cochlea
Authors: Sarpeshkar Rahul
Published: July 2002

Low-power wide-dynamic-range systems are extremely hard to build. The biological cochlea is one of the most awesome examples of such a system: It can sense sounds over 12 orders of magnitude in intensity, with an estimated power dissipation of only a few tens of microwatts. An analog electronic cochlea that processes sounds over six orders of magnitude in intensity, while dissipating less than 0.5 mW, is described. This 117-stage, 100-Hz to 10-KHz cochlea has the widest dynamic range of any artificial cochlea built to date. This design, using frequency-selective gain adaptation in a low-noise, traveling-wave amplifier architecture, yields insight into why the human cochlea uses a traveling-wave mechanism to sense sounds, instead of using bandpass filters.
Emergent collective behaviors in a simplified model of dolphin echolocation and predation
Authors: Hayward Thomas J.
Published: July 2002

A stochastic modeling framework for representing and simulating marine-mammal spatial distributions, collective motions, and vocalization occurrence times was previously developed [J. Acoust. Soc. Am. 101, 3197(A) (1997)]. Emergent behaviors in these models include clustered and polarized motions simulating collective motions of marine mammals. The present work explores the potential of these models for representing and simulating more complex acoustic emission and collective motion behaviors related to short-range echolocation and predation by dolphins. A simplified predation scenario is constructed by including predator, prey, and nonprey species in the collective motion model. Echolocation is simulated by a simplified, sonar equation-based representation of the backscattered acoustic energy, with different backscattering characteristics for predator, prey, and nonprey individuals. Parameters determining echolocation signal types and collective motion characteristics are then allowed to vary in a random search that favors parameter values leading to higher prey capture rates. Emergent collective behaviors, including echolocation signals and prey pursuit and capture, are then observed in computer-generated video animations. The potential of the models for representating sound generated by groups of marine mammals and for interpreting observed marine mammal collective behavior is discussed. [Work supported by ONR Base funding at NRL.]
Nonlinear effects in HIFU propagation and attenuation in biological tissue
Authors: Khokhlova Vera A. , Sapozhnikov Oleg A. , Crum Lawrence A.
Published: July 2002

For therapeutic applications of medical ultrasound, such as hyperthermia, acoustic surgery, and hemostasis, it is desirable, for precise targeting, to induce effective local absorption of the acoustic energy at a small focal area of the acoustic beam. A source with high-focusing gain can be used to achieve the most effective regime of focusing, in which nonlinear effects increase dramatically very close to the focal point. Sharp shocks are developed and the corresponding absorption rate increases at the focus without extra nonlinear attenuation in the prefocal region. In an earlier presentation [Khokhlova et al., J. Acoust. Soc. Am. 98, 2944 (1995)], a modified spectral approach was used that enables us to model propagation of nonlinear plane waves, including shocks, in biological tissue. Here, this method is extended for the description of high-intensity focused sound beams. The acoustic pressure field is calculated numerically for specific focusing gains and tissue absorption over a wide range of the source amplitudes. The effect of nonlinearity, focusing gain, absorption on the waveforms, amplification and spatial localization of the intensity, heating rate, and positive and negative peak pressure at the focus is discussed. [Work supported by ONR and CRDF.]
Nonlinear acoustic parameter of trabecular bone
Authors: Donskoy Dimitri M. , Sutin Alexander
Published: July 2002

This work is an attempt to measure the nonlinear parameter of porous trabecular bone. Two different methods were applied. One method involves an AM ultrasonic signal transmitted through a bone submerged into water. A part of the signal energy transforms into the low-frequency (the frequency of modulation) signal due to the nonlinearity of the sample. The larger the nonlinear parameter of the sample, the more intensive such a transformation. This low-frequency signal is then picked up with a hydrophone and measured. The other method employs the effect of modulation of higher-frequency ultrasound by lower-frequency vibration applied directly to the bone. The same high frequencies and low frequencies were used in both tests, so the independently measured nonlinear parameters of the samples (cut from the same bone) could be compared. High dissipation of ultrasonic waves in trabecular bone in the frequency range above 100 kHz made it difficult to measure the nonlinear effects. The nonlinear parameter in the frequency range below 100 kHz was able to be measured reliably. Thus the nonlinear parameters were measured for three ultrasonic frequencies 26, 37, and 60 kHz modulated by the vibration frequencies 4.6 and 9.8 kHz. In vitro determined values of the nonlinear parameter for bovine bone used in these tests were in the range 80–120, which is an order of magnitude higher than for nonporous media (e.g., compact bone and liquid). [Work supported by NASA.]
Tactile estimation of roughness and size
Authors: Verrillo Ronald T. , Bolanowski Stanley J. , McGlone Francis P.
Published: July 2002

Subjects estimated the subjective magnitude of roughness and the size of steel balls by the method of absolute magnitude estimation. Using the index finger and thumb, estimates were made of 11 grades of sandpaper with particle diameters ranging from 16–905 μm. The results describe a power function and show spatial summation when stimuli are applied simultaneously to two sites (finger and thumb). The same subjects judged the size of nine steel balls ranging in diameter from 0.4–6.4 mm. Estimates were made with combinations of the index finger, volar forearm, and thumb. Again, all of the results were power functions and spatial summation of tactile stimulation was evident. The results are discussed in terms of theoretical considerations and practical application. [Work supported by Unilever and NIH, PO1DC00380].
Regular interval stimuli: Are higher-order intervals necessary?
Authors: Yost William A. , Shofner William P.
Published: July 2002

Many stimuli that contain regular temporal intervals are perceived as having a pitch equal to the reciprocal of the interval. Autocorrelation is one method of determining the interval in these regular interval stimuli (RIS). Autocorrelation analysis has been used as a ‘‘model’’ for calculating the pitch and pitch strength of RIS. Autocorrelation is an ‘‘all-interval’’ analysis as opposed to the first-order interval analysis that is often used to describe the temporal properties of neural spike trains. Both actual neural responses (from the AVCN of the chinchilla) and those of a simple neural model were measured to determine the order of intervals contained within RIS. In a psychoacoustic experiment, RIS were generated with no first-order intervals or with only first-order intervals, and listeners discriminated between these stimuli and stimuli with no regular intervals. The results of the experiments and the calculations indicate that the pitch and pitch strength of RIS probably depend on only lower-order intervals (no higher than fourth-order intervals). These results will be discussed in terms of theories of pitch and timbre processing. [Work supported by NIDCD, DC-000293.]
Evaluation of different techniques in matched-field processing using measured data
Authors: Abawi Ahmad T. , Booth Newell , Schey Phil
Published: July 2002

The data from May 1996 Shallow Water evaluation cell Experiment (SWellEx-96) are used to evaluate the performance of commonly used processors in matched-field processing (MFP). The objective of SWellEx-96, which was conducted west of Point Loma in shallow water, was to investigate source tracking using matched-field processing and low-level signal detection. Data were recorded on a 64-element 118-m vertical line array and on an identical 45-deg tilted line array anchored in 200 m of water. A cw source emitting tonals from 49–388 Hz was towed over tracks ranging up to 6 km from the array. Signals were transmitted in group of frequencies. The source level in each frequency group decreased by 3 dB every 3 Hz, thus making it possible to study MFP as a function of the signal-to-noise ratio (SNR). Various figures of merits have been used to evaluate the performance of the Bartlett, the minimum variance distortionless response (MVDR) as well as MVDR with white-noise gain-constraint (MVDRWNC) processors.
Paul S. Veneklasen: The War Years, 1941–1945
Authors: Beranek Leo L.
Published: July 2002

Paul was a senior research associate at the Electro-Acoustic Laboratory (directed by Leo L. Beranek) and the Psycho-Acoustic Laboratory (directed by S. Smith Stevens) during World War II. This paper discusses his contributions during this period to the hearing protection of military personnel and to the development of special electronic equipment for the precision measurement of the properties of sound fields.
Approximate model for sound generation due to unsteady flows in pipes
Authors: Krane Michael , Sinder Daniel , Flanagan James
Published: July 2002

An approximate model for the generation of sound due to unsteady vortical flow in pipes of varying cross-sectional area is described. This model, which is being used in speech synthesis research, is based upon Howe’s acoustic analogy. This formulation allows the sound generation to be specified in terms of properties of the unsteady behavior of a vortical flow and the potential flow solution for the pipe that would exist in the absence of vortical inhomogeneities. An approximate model for vorticity formation and evolution is presented, and the potential flow solution is obtained from the axial duct shape. Computed estimates of the time evolution of the sound generated by a confined jet passing through a pipe constriction are presented. These results are compared to experimental data. [Research supported by NSF/ARPA IRI-9314946 and ARPA DAST 63-93-C-0064.]
Aesthetic considerations for the composition and performance of computer-based jazz
Authors: Leonard Neil
Published: July 2002

Computers, critical doubts not withstanding, provide an important way to extend jazz improvisation and compositional resources. Jazz musicians have long explored the use of new technologies and have given a number of recent instruments their first virtuosos and repertoires. Just as jazz had given the saxophone its voice, through musicians like Coleman Hawkins, it also established the validity of the drum set in the work of Baby Dodds among others, and the electric guitar in the breakthroughs of Charlie Christian. I have sought to expand my work as a jazz improviser and composer through the use of interactive systems. For the past 12 years I have presented concerts that included original compositions for saxophone and computer controlled electronics. In the process I have explored several approaches to designing and performing with these systems, and addressed issues of how jazz is evolving. The lecture will include a demonstration of one such composition that uses an interactive system.
Large-eddy simulation of compressible free jet turbulence applied to computation of exhaust mixing noise
Authors: Schein David B. , Meecham William C.
Published: July 2002

A computational study of free, heated jet flow and resultant far-field sound was performed using large-eddy simulation (LES) and Lighthill’s acoustic analogy. A subgrid scale model for small-scale compressible turbulence was developed using a combination of the popular Smagorinsky model and a deductive model. The primary objective is to address large Reynolds number (Re), high subsonic (compressible) flow with realistic geometries more representative of aircraft engine exhausts than typically considered using direct numerical simulation (DNS). Flow field fluctuations are stored over a period of time and used to calculate rms turbulence within the computational domain. The far-field sound and directivity is computed using the time-derivative form of Lighthill’s source–integral result formulated in terms of quadrupole sources from the simulated flow field, which is integrated in time and contains the fluctuations set up by the time-varying stress tensor. A simulation for a WR19-4 turbofan engine exhaust (Re≊2×106 based on exit velocity and diameter) is presented, and propagated jet noise results are compared with experimental acoustics data. Methods to account for effects of source convection and thermal refraction to obtain realistic frequency spectra and directivity are considered.
Mutual interaction of a two-phase turbulent/bubbly submerged water jet with a single-phase jet for the measurement of the enhanced hydrodynamic near-field pressure spectrum
Authors: Korman Murray S.
Published: July 2002

The interaction of mutually perpendicular submerged turbulent water jets is studied from the measurement of the enhanced hydrodynamic near-field pressure spectrum. One jet is a conventional single-phase, free turbulent shear flow circular jet, while the other jet is a two-phase bubbly jet of similar construction. The overlap region is located at four-nozzle diameters (4D) away from each orifice. A small hydrophone located outside the overlap region (4D away from the interaction region and 45 deg from the jet axis) measures the near-field pressure spectrum. Very small gas bubbles are required for this two-phase turbulent jet. They are generated by pressurized carbonated water flowing through a thin circular nozzle plate having an array of small holes. These CO2 gas bubbles are mixed within the entrance to the tapered jet nozzle. Alternatively, bubbles are generated by pressurized N2 gas passing through a fritted disk housed in a Buchner funnel located near the nozzle entrance. Spectra are measured and compared for different void fractions and different Reynolds numbers that are based on the jet diameter.
The effects of nonconcurrent tonal and noise distractors upon auditory spatial resolution: The temporal, spatial, and spectral parameters underlying the ‘‘disruption effect.’’
Authors: Erpenbeck Eric A. , Perrott David R. , Lim Raymond
Published: July 2002

The minimum audible angle (MAA) thresholds obtained by Mills [A. W. Mills, J. Acoust. Soc. Am. 30, 237–246 (1958)] involved two 1000-ms sinusoidal tone pulses presented sequentially with an interstimulus interval (ISI) of 1000 ms. In contrast to the generally good localization performance obtained by Mills, localization of two simultaneous events (concurrent localization or CMAA) results in relatively poor performance [D. R. Perrott, J. Acoust. Soc. Am. 76, 1704–1712 (1984)]. The current set of experiments attempts to bridge the gap between these two extreme paradigms. While the target tones (14-ms, 1000-Hz sinusoids) were presented sequentially (an MAA task), an attempt was made to systematically examine the consequences of the addition of a nonoverlapping, auditory event (tones or noise) during the interval between the target events. The results of this work indicate that localization performance is extremely sensitive to events that occur between the target events. The disruptive effect of a ‘‘nontarget’’ event during the ISI could even be observed when relatively long ISIs (814-ms) and very brief (14-ms) distractors events were used. Interesting interactions were encountered as a function of the spatial positioning of the nontarget (‘‘distractor’’) event. Implications of these results will be discussed.
Evaluation of simulations of the effects of hearing loss produced by combinations of dynamic expansion and spectral smearing
Authors: Graf Isaac J. , Lum David S. , Braida Louis D.
Published: July 2002

By processing sounds with combinations of multiband expansion (MBE) and spectral smearing (SS), listeners with normal hearing can experience the elevated thresholds, abnormal growth of loudness, and reduced frequency selectivity similar to those of listeners with sensorineural hearing impairments. The effects of simulations of flat 40-dB losses produced by such combinations on masking patterns and loudness summation, and of simulations of a more severe loss on speech perception in noise, were studied. Both MBE and SS reduce frequency selectivity as measured by masking patterns for tones produced by narrow-band noise, and both broaden forward-masked tuning curves for tones. However, for equally loud probe tones, MBE elevates the tips of psychoacoustic tuning curves much more than SS. As measured by loudness balances between narrow-band and wideband noises, MBE reduces loudness summation much more than SS. The effects of MBE and SS simulations of a severe hearing loss on consonant identification in speech-shaped noise depend on the SNR, with MBE more detrimental at high SNR and SS more detrimental at low SNR. MBE tends to degrade reception of cues for voicing; SS, cues for place of articulation; both degrade cues for manner of articulation. [Work supported by NIH.]
On the apparent source width for music sources related to the IACC and the width of the interaural crosscorrelation function (WIACC)
Authors: Sato Shin-ichi , Ando Yoichi
Published: July 2002

It has been shown that the ASW for 1/3 octave band noise can be described as functions of the IACC and the WIACC, which is defined by the time interval between values of 10% below the IACC [S. Sato and Y. Ando, J. Acoust. Soc. Am. 100, 2592 (1996)]. In this study, it is examined whether the ASW for music sources also can be calculated with the IACC and the WIACC. In order to control of both the IACC and the WIACC, the delay time of two reflections relative to the direct sound was changed in the range of 0–50 ms. Results of the scale value of ASW obtained by paired comparison are compared with calculated ASW.