bluuebunny/crossref_metadata_2025 huggingface dataset:
Characteristics of shakuhachi signals
Authors: Seltzer Linda A.
Published: October 2005

Selected portions of a recorded performance of shakuhachi (Japanese bamboo flute) music have been analyzed using spectrograms, and waveforms have been plotted by computer. The study has revealed interesting and unusual features of the signal during special performance practices. These effects add variety to the timbre, as well as subtlety in the conception of pitch in the music. The effects include: nonharmonic partials which are anticipations of harmonics of the subsequent note; harmonics which linger and become nonharmonic partials of the subsequent note; 90-deg phase shifts when a lateral opening is repeatedly hit with a finger; subharmonics during transitions between notes; gradual, smooth micro-tonal variations of pitch in the course of long notes; and abrupt changes in timbre.
Instrumentation for intensity measurements
Authors: Nielsen Torben G.
Published: October 2005

The concept of sound intensity has been known for many years, and the usefulness of an intensity measuring device has been understood for just as long a time. The factor that has drawn the attention of the noise control community towards intensity measurements within the last decade is the development of instruments making direct measurement of sound intensity a feasible task in the field as well as in the laboratory. Today, a selection of analyzers and probes of different types are being used for practical intensity measurements. The analyzers fall mainly into three groups: dual-channel FFT analyzers, real-time analyzers based on digital filters, and small battery-operated systems usually equipped with analog filters. Most probes are based on the two-microphone principle, but a system using a pressure and a velocity transducer is also available. From a user's point of view, these different types of instruments should provide identical measurement results—but this is not always the case in practice. In the presentation, the various instrument principles are briefly reviewed and the performance of the instruments when subjected to different sound fields and signal types is discussed.
One-microphone sound intensity method
Authors: Mielnicka-Pate Anna L. , Bai Robin M.
Published: October 2005

A new method based on the cubic spline approximations of pressure gradient was developed for particle velocity and acoustic intensity calculations. In this method, a cubic spline is interpolated for every four consecutive pressure measurements performed with equal spacing. In contrast to conventional approximate methods, only one microphone and synchronized scanning are required in this technique. A numerical simulation was performed in order to evaluate the accuracy of this method. A monopole and a baffled piston were chosen as the acoustic sources in this simulation since the analytical solutions are easily available for comparison purposes. The cubic spline interpolation technique provided substantially better accuracy than the finite difference method. In addition, some preliminary experimental data will be presented for the cubic spline and finite difference techniques. [Work supported by NSF.]
Complex wave fronts and generalized phase speed described by the active intensity vector
Authors: Mann Adin , Tichy Jiri
Published: October 2005

The active intensity vector is directed normal to the surfaces of constant phase in an acoustic field. The surfaces of constant phase are wave fronts; consequently, the active intensity vector characterizes the resultant wave propagation at a point in space. This physical description provides a powerful means to organize a complicated nearfield of a radiator or a scatterer. In the farfield, the phase speed of the resultant wave fronts is the plane-wave phase speed c. However, in general, the phase speed varies throughout an acoustic field. The general phase speed Cp is described by taking the total derivative of the phase of the pressure. The resultant wave fronts and their phase speed will be discussed with examples of acoustic vorticies, and the measured fields diffracted by a rigid circular disk or generated by a loudspeaker.
On the adequacy of linear acoustics for prediction of acoustic pressure transients created by laser beams moving over water surfaces at sonic speeds
Authors: Pierce Allan D. , Berthelot Yves H.
Published: October 2005

Previous discussions of laser-generated sound, particularly with reference to themoacoustic effects, are often based on linear equations. However, seemingly plausible arguments can be developed to the effect that the peak acoustic pressure predicted by a linear theory will grow without bound when the sound is generated by a laser beam that travels for an indefinitely long period of time over a water surface at exactly the speed of sound. One consequently asks (1) whether the linear model does indeed predict a pressure singularity for a sonically moving laser beam, and (2) whether nonlinear effects can be neglected. Analysis by two different methods (Fourier transforms and method of retarded potentials using the transient (Green's function) shows that the linear acoustic framework does not predict any singularity in the acoustic pressure generated by a thermoacoustic source moving at the speed of sound over a water surface. Additional theoretical considerations strongly suggest that cumulative nonlinear effects can be ignored during the buildup phase of the acoustic pulse. The key ingredient in the mathematical model, which holds the peak pressure to a finite value, is the pressure release boundary condition at the water surface. However, once the acoustic pulse is launched (i.e., after the laser is turned off), accumulative nonlinear steepening effects may alter the received pressure waveform. In the latter case, a quasi-one-dimensional nonlinear acoustics model based on Burger's equation is expected to be adequate to predict the nonlinear distortion in farfield sound propagation. [Work supported by ONR.]
Numerical solution of Burgers' diffusion equation for arbitrary waveforms: Convolution method
Authors: Mellen R. H.
Published: October 2005

Burgers' nonlinear equation for plane acoustic waves in a dissipative fluid reduces to the diffusion equation by Hopf-Cole transformation. A Bessel series solution has been obtained for an sinitially sinusoidal wave [D. T. Blackstock, J. Acoust. Soc. Am. 36, 534–542 (1964)]. Solutions for initial waveforms of arbitrary shape are also possible by numerical methods using the FFT method [R. H. Mellen, J. Acoust. Soc. Am. Suppl. 1 75, S32 (1986)]. However, the method is limited to relatively small Goldberg numbers because of the roundoff errors. Gaussian convolution is shown to be an equivalent method that does not softer this limitation. Examples of calculated waveform distortion for typical cases including random noise are presented.
Parametric wave interactions near reflecting surfaces
Authors: Darvennes Corinne M. L. , Hamilton Mark F.
Published: October 2005

Parametric wave interactions offer the possibility of synthesizing directional transmitting and receiving arrays by exploiting the acoustical nonlinearity of the fluid. Use of parametric arrays is frequently suggested as one way to reduce problems involving multipath in both the transmission and reception of sound in reverberant environments. Novikov et al. [Sov. Phys. Acoust. 29, 138 (1983)] have analyzed the case of parametric radiation parallel to an adjacent reflecting surface. Here their work is reviewed and the effects of reflection on the radiation pattern are discussed. The analysis is then modified to include the case of parametric reception. Of interest is the use of a parametric receiver for calibrating the sound field radiated by surface ships. The proposed application involves mounting the pump on the ship and placing the hydrophone in the farfield. Although the proposed system has recently been analyzed for a freefield environment [M. F. Hamilton et al., J. Acoust. Soc. Am. Suppl. 1 78, S40 (1985)], the effects of reflected sound in the interaction region must be understood. [Work supported by ONR.]
Reflection from an anechoic duct termination
Authors: Tanzosh J. P. , Hayek S. I.
Published: October 2005

A theoretical model was developed which describes the farfield reflection from various absorbent terminations of an acoustic duct with rigid walls. The absorbent termination is utilized to reduce the reflected sound from the ends of ducts and pipes. Such sound absorbing terminations can be used to minimize the shock wave (water hammer) travel in water pipes, reduce the flow noise at flow-through pressure valves and at stations along the hydraulic lines where cross-sectional changes occur. A parametric study of the terminations's geometry and material properties revealed the effects of longitudinal and shear wave resonances inside the termination on the frequency response of the reflection. A partially cemented lower boundary achieves much greater echo reduction than a uniform rigidly attached or stress-free boundary. Peaks in the frequency response result not only from compressional thickness resonances, but also from shear and lateral compressional resonances induced by the discontinuity. A slight impedance mismatch promotes multiple reflections within the termination, hence increased resonance effects. [Work supported by the Naval Sea Systems Command.]
Convergent transient acoustic fields from pulsed planar contracting ring sources
Authors: Warshaw Stephen I.
Published: October 2005

If a pulsed, radially symmetric planar source imbedded in a rigid baffle of infinite extent is described by a normal surface acceleration history of the form an = δ(t + r/vs), where r is radial surface distance from the source center, v, is the speed of uniform radial contraction, and the motion proceeds from r = a to r = 0, then convergent and intersecting wave fronts can be produced in the nearfield of this source. Fields from this source have been calculated using techniques described previously [S. I. Warshaw, J. Acoust. Soc. Am. Suppl. 1 77, S60 (1985); and 79, S90 (1986); Proc. 12th ICA, Toronto, 1986] in the time domain. The calculation technique will be reviewed briefly and compared for contracting and expanding ring sources. Details of the time-evolving two-dimensional wave field will be presented in the form of contours of constant acoustic amplitude calculated at different times from source onset to propagation past the convergence zone. The various dynamically changing wave front patterns will be indicated and discussed. [Work supported by U.S. Department of Energy.]
Simple theoretical model for dynamic bulk modulus of elastometers and liquids
Authors: Burns Jay , Dubbelday Pieter S.
Published: October 2005

A free-volume theory for the temperature dependence of the dynamic bulk modulus is presented which is based upon the Cohen-Grest theory of the liquid-glass transition [M. H. Cohen and G. S. Grest, Phys. Rev. B 20, 1077–1096 (1979)]. The theory leads naturally to the WLF method for constructing “master curves” for the bulk modulus analogous to those for shear and Young's moduli. It also provides for the transition at sufficiently high temperatures from the WLF regime to the range where the bulk modulus is dominated by an Arrhenius-type temperature dependence. The frequency dependence of the bulk modulus was developed from an approach used by Fuoss and Kirkwood in their theory of dielectric losses in polymers [R. M. Fuoss and J. G. Kirkwood, J. Am. Chem. Soc. 63, 385–394 (1941)]. It leads to a hyperbolic tanget dependence of the storage modulus upon the logarithm of frequency and to a hyperbolic secant dependence of the loss modulus upon the log frequency. Experimental data on elastometers support the theory closely. [Work supported by ONR.]
Discrimination of tonal complexes on the basis of which component is interaurally delayed
Authors: Dye Raymond H.
Published: October 2005

An experiment was undertaken to determine the extent to which information arising from interaural differences of time (IDTs) presented in one component of a complex is correlated with information arising from other components. Observers discriminated between multicomponent stimuli that consisted of the same spectral components but differed in terms of which component was interaurally delayed. Correlations were estimated by comparing discrimination d's to those predicted on the basis of the d's obtained for the detection of IDTs in each of the to-be-discriminated stimuli (versus diotic complexes). Discrimination d's were measured for three-component complexes consisting of 653,753, and 853 Hz for the following conditions: 653T-753-853 vs 653-753T-853 Hz, 653T-753-853 vs 653-753-853T Hz, and 653-753T-853 vs 653-753-853T Hz, where T indicates an interaurally delayed component. In addition, d's were measured for discrimination of the three varieties delayed three-tone complexes from diotic complexes. Results show that interaural information arising from different components in these tonal complexes is highly correlated. [Work supported by the Air Force Office of Scientific Research, Life Sciences, the National Institutes of Health, NINCDS, and the National Science Foundation.]
Localization of sound: Source discrimination and source identification
Authors: Rakerd Brad , Hartmann W. M.
Published: October 2005

The ability of human listeners to localize sound sources can be measured by discrimination experiments, such as the minimum audible angle (MAA) paradigm, or by source identification experiments. In principle, these two kinds of experiments can be unified by a statistical decision theory model, incorporating a process whereby a sensory scale is referenced to an absolute scale. To test this idea we performed several kinds of free field discrimination and source identification experiments, both in the light and in the dark. Among our conclusions are the following: (1) The variance of the referencing process depends upon the width of the absolute scale required by the experiment. It doubles as the width increases from 2°–33°. (2) The MAA experiment, nominally a measurement of discrimination, probably includes a source identification component as well. In fact, our data in the light are consistent with a model in which the MAA experiment is treated exclusively as a source identification task. [Work supported by the NIH and NIMH.]
Ear position affects the auditory space representation in the inferior colliculus of bats
Authors: Sun Xinde , Jen Philip H. S.
Published: October 2005

Using free field acoustic stimulation conditions, the spatial response areas of single neurons and auditory space representation in the inferior colliculus (IC) of the big brown bat Eptesicus fuscus, was studied under different ear positions. A best frequency stimulus was delivered from the loudspeaker which was moved manually across the frontal auditory space of the bat to determine the response center of the neuron. At the response center, the neuron had its maximal directional sensitivity. The spatial response area of each neuron was determined for acoustic stimuli set at several intensity levels above the minimum threshold of the neuron. The spatial response areas of each IC neuron expanded unsymmetrically with the stimulus intensity. The size of the spatial response area was not correlated with the minimum threshold, best frequency, or recording depth of each neuron. Although each neuron had a point of maximal directional sensitivity, the point-to-point representation of the auditory space was not systematically organized. This representation was not correlated with the recording site of each neuron in the mediolateral, posteroanterior and dorsoventral axes of the IC. Both the response centers and spatial response areas of individual neurons varied with ear position but individual variations were complementary. It is likely that a mobile auditory space representation may provide the bat with versatility in maximizing the directional sensitivity of its echolocation system for accurate prey capture. [Work of Philip Jen supported by NIH.]
Laryngeal contributions to the difference between two types of whisper
Authors: Pearl Nancy B. , McCall Gerald N.
Published: October 2005

The laryngeal behaviors that contribute to perceptual differences (primarily loudness and noisiness) between whispers produced in a relaxed manner and in a forced manner are not well understood. To address this issue, we examined vocal fold configuration, glottal size, and supraglottal constriction during ten adults' productions of relaxed and forced whisper for three vowels in consonant-vowel syllable strings. Video recordings of the laryngeal structures were obtained through fiberoptic endolaryngoscopy. Results provided evidence that the differentiation of relaxed (quiet) and forced (loud) whisper was related to some extent to changes in supraglottal constriction (more for forced than relaxed whisper) but not to changes in glottal configuration or size. The failure of supraglottal constriction to occur for all instances of forced whisper implies that this strategy is an incomplete explanation for perceptual differences between whisper types. Simultaneous variations in both laryngeal and respiratory behaviors is a more tenable model, and deserves further consideration.
Effect of nerve section on spastic dysphonic voice: Implications for phonatory motor control
Authors: Shipp Thomas , Izdebski Krzysztof
Published: October 2005

The results of three laryngeal EMG studies on patients with spastic dysphonia (SD) show a similar pattern of inappropriate increased adductory muscle activity bilaterally during speech, resulting in phonation that is momentarily interrupted or severely reduced in amplitude. Following these moments of “strained-strangled” phonation, the involved muscles resume activity at the previous levels and normal phonation is resumed. After temporary unilateral recurrent laryngeal nerve block, phonation is uniterrupted and no inappropriate buildup of activity is observed in muscles contralateral to the nerve block. A model of the SD process explaining these data identifies normal peripheral nerve and muscles, but abnormal afferent/efferent interaction at the brainstem level. Further, SD symptoms developed in post-surgical voice appear in patients with midline position of the paralyzed fold who produce voice at low F0 values.
An investigation of motor equivalence in the speech of children and adults
Authors: Smith Bruce L. , McLean-Muse Ann
Published: October 2005

The concept of motor equivalence in speech production suggests that the contributions of various articulators can vary considerably across multiple productions of a segment, yet the end product is perceptually consistent. Since motor equivalence presumably represents a sophisticated ability in speakers, one purpose of the present study was to determine whether developmental trends could be observed toward more frequent occurrence of motor equivalence with increasing age. A strain gauge transduction system was used to monitor inferior-superior upper lip, lower lip, and jaw movements produced by a group of adults and three groups of children, as they spoke normally and in two experimental conditions (bite block and fast rate). Assuming that negative correlations between pairs of articulators constitute evidence of motor equivalence, there was not a strong indication of motor equivalence in the speech of the children or the adults for a variety of comparisons. In addition, the children tended to show more negative correlations than the adults. All four groups also tended to exhibit negative correlations most frequently in the bite block condition. [Work supported by NINCDS.]
Computer program for weapons noise propagation with terrain attenuation effects
Authors: Wrobel John , Lewis Nelson , Russell William , Baldridge Robin , Guth Peter
Published: October 2005

Whenever we are dealing with noise propagation at distances greater than 100 m from the source, there are many factors affecting the propagation. For weapons firing, a major factor affecting propagation is the tendency of terrain to diffract and deflect the sound. A computer program has been written in Turbo PASCAL to run on an IBM-XT computer. The program consists of digitized terrain data for a specific area along with a predictive sound propagation model for 25- and 7.62-mm guns along with the TOW anti-tank missile. The user selects locations of firing point, target point, and observer, along with number of rounds fired, and the start and end time of firing. The computer program provides the A-weighted day-night average sound level at the observer's location. The significant feature of NOISE.PAS is that it computes excess attenuation caused by terrain for specific sites. Additionally, NOISE.PAS takes the tedium out of doing sound calculations, which incorporate excess attenuation caused by terrain, for weapons firing.
Control of noise from kidney lithotripters
Authors: Weissenburger J. T.
Published: October 2005

The kidney lithotripter is a new medical device for removing kidney stones without requiring surgery. The patient is submerged in a tub of water and bombarded with repetitive high-energy acoustical impulses focused on the kidney. This not only disintegrates the kidney stone, but also results in significant airborne and structure-borne sound. This sound can be quite disruptive in surrounding areas if permitted to reach them. Installations are typically in existing facilities and this limits available options for noise control measures. Techniques that have been used successfully are discussed.
Acoustic energy propagation in noise control foams: Approximate formulas for surface normal impedance
Authors: Bolton J. S. , Green E. R.
Published: October 2005

Relatively stiff, partially reticulated polyurethane foam as used in noise control applications supports two longitudinal wave types having distinct propagation factors: the “frame” wave and the “airborne” wave. Both waves appear in both the solid and porous phases of the foam. Considered here is the division of acoustic power flow (within a semi-infinite foam layer) between the two wave types and between the two phases. These divisions are strongly dependent on surface boundary conditions. Energy is carried predominantly by the frame wave and almost exclusively via the solid phase when the surface is scaled by an attached impermeable membrane; in this configuration foam behaves like a homogeneous elastic material having enhanced structural damping owing to the contained air. In contrast, when the surface is open, energy is carried by both wave types and is observed to oscillate between the solid and air phases. These observations have allowed the derivation of simplified formulae for the surface impedance of foam layers.
Increasing accuracy of coronal tongue imaging using ultrasound
Authors: Stone Maureen , Shawker Thomas H. , Tipton Harold W. , Talbot Thomas L. , Rich Alan H. , Lederman Joseph N.
Published: October 2005

Coronal tongue images made using ultrasound have been limited by lack of precision in transducer placement. To improve precision and accuracy of data collection, especially in the coronal plane, a transducer holder and a head holder have been developed. The transducer holder maintains the transducer at a specified submental location in either the sagittal or coronal plane. Beam angulation is determined relative to the mandibular rami. An acoustic standoff prevents transducer interference with mandibular movement. The head holder consists of three cup-shaped cushions attached to the back of a dental chair which form a triangular support against the posterior head. A Velcro strap, placed around the head, is tightened by means of an eccentric. The head holder also contains a removable arm which can hold up to three strain gauge transducers allowing simultaneous lip and jaw measurements to be made. With increased instrumental accuracy, more subtle tongue shape comparisons can be made. The effects of coarticulation on midsagittal tongue groove, as seen coronally in /sVs/ and /pVp/ context will be presented at several transducer angles.
Japanese quail categorize [d] across allophonic and talker variation
Authors: Kluender Keith R. , Diehl Randy L.
Published: October 2005

Last fall, the ability of Japanese quail to categorize allophonic variants of [d] was reported. In those experiments, quail were trained to discriminate [dæs], [das], [dus], and [dis] from tokens with the same vowels following [b] and [g]. Quail then correctly categorized novel [d], [b], and [g] tokens with the vowels [ε], [Λ], [U], and [I]. A possible criticism is that those novel nowel environments were not sufficiently novel, since [ε], [Λ], [U], and [I] may be loosely characterized as lax versions of the vowels used in training. To address this point, we tested the same birds on four additional vowel environments: [ɝ], [eI], [oI], and [ou]. Performance with these new vowel environments essentially mimicked that with the former novel vowels. Additionally, whether the quails' ability to categorize [d] transfers to tokens from a second male talker was examined. One quail was tested using unreinforced novel [d], [b], and [g] tokens with the vowels [æ], [i], [Λ], [U], [eI], and [oI] from the second talker. The quail again pecked significantly more to novel unreinforced [d] tokens than to novel [b] and [g] tokens. These data support the view that there exists significant acoustic/aduitory commonalities among the class of [d] allophones across talkers that may serve as the basis of category formation for both humans and animals. [Work supported by NICHD.]
Perceptual confusions in the speech of tracheoesophageal talkers
Authors: Haaf Robert G. , Doyle Philip C.
Published: October 2005

This project investigated the preception of speech stimuli produced by tracheoesophageal (TE) talkers. Four highly proficient TE talkers produced pre- and post-vocalic consonants within a 66-item word list comprised of CVC stimuli [M. S. Weiss and A. G. Basili, J. Speech Hear. Res. 28, 294–300 (1985)]. Stimuli were representative of all manner classes (stops, fricatives, affricates, nasals, and semi-vowels). Twelve normal-heating young adults served as listeners. Listeners were trained in IPA transcription and had no, or minimal exposure to alaryngeal speech. Listeners were presented TE stimuli via headphones and were required to record their responses using broad phonetic transcription. An open-response paradigm was employed. All talker stimuli were randomized. Confusion matrices for each talker were prepared and overall intelligibility for each talker was determined. Analyses of the data were conducted for individual talkers, and by talker group. Categories of perceptual confusions were analyzed across manner class by phonetic context (pre- and post-vocalic). Voicing distinctions were also quantified by manner and phonetic context. Predominant perceptual confusions will be discussed as related to the unique alaryngeal speech production method used by this clincal population.
Effects of acoustical context on perceived vowel quality
Authors: Pattison Helen , Gardner Roy B. , Darwin C. J.
Published: October 2005

Vowels along an /I/-/e/ continuum with added energy at a harmonic frequency near to F are embedded in sequences of tones and their phonetic quality assessed by the position of the phoneme boundary. If the sequence of tones has the same frequency as the added energy, then its contribution to vowel quality decreases as the number of tones preceding the vowel is increased, and is greater if the tone sequence continues after the vowel, indicating a greater tendency for the added tone to be perceptually segregated from the vowel. A similar effect is observed if the vowel is embedded in a sequence of tones which descends in frequency (by whole tones, or by half-harmonic steps), coinciding with the vowel at the harmonic frequency. However, if the tone sequence ascends in frequency, then the added tone does alter the perceived vowel quality, indicating that the tone sequence has not captured the embedded tone. In a further experiment a single tone at various frequencies preceded the vowel by various time intervals. The experiment showed that the difference between ascending and descending series was due to the cumulative individual effects of the two tones that immediately preceded the vowel, rather than to the configuration of the series. [Work supported by SERC.]
Perceptual compensation for effects of transmission channel characteristics on vowel quality
Authors: Watkins Anthony J. , Pattison Helen , Darwin C. J.
Published: October 2005

The phonetic quality of front vowels along a synthetic /I/-/e/ FI continuum can be influenced by passing the vowels through a filter that has a sloping characteristic in the FI region. The shift in phoneme boundary that occurs for different filter slopes was measured and whether the influence of various conditions listeners can compensate for the characteristics of the transmission channel was examined. These conditions include placing the vowel in a consonantal context and preceding the vowel with a carrier phrase that has been subjected to the same filtering as the vowel. Experiments have also been performed on the preception of natural vowels in isolation and in a consonantal context. [Work supported by SERC.]
Listener biases in diagnostic inteiliglbility testing
Authors: Smith Caldwell P.
Published: October 2005

In the course of conducting numerous diagnostic rhyme tests for intelligibility evaluations of many voice processors over the past several years, several retests were run as checks of reliability. Retests involved assigning new identification numbers to test recordings and resubmitting them for evaluation by presentation to eight-member listener crews. Retests were interspersed with other (new) tests and were usually run about a week after the first presentation. Overall intelligibility scores from the two test administrations have been found to be in close agreement. However, analysis of differences in paired diagnostic scores from the two tests has revealed significant effects of listener biases. The pool of listeners consisted of subjects screened for normal hearing and having extensive familiarity and experience with the test materials and procedures. When the second test was conducted with the identical listener crew, none of the 54 feature scores or the 18 speaker scores differed significantly. When all eight listeners were different, 18 of the 54 feature scores and three of the 18 speaker scores differed significantly (a = 0.05). For six cases in which the number of common listeners ranged from one to six, the number of diagnostic scores that differed significantly in the two tests tended to vary inversely with the number of listeners in common (r = − 0.77). This finding suggests that it cannot be assumed that two eight-member listener crews of equivalent competence will have equivalent performance in discriminating among phonetic events having minimal contrast.
The influence of correlogram spread and wander on a broadband cross-correlation system
Authors: Bilazarian P.
Published: October 2005

Scattering or time-varying mechanisms in the oceanic medium can affect the performance of a passive broadband cross-correlation system. These mechanisms include internal waves, sea-surface motion, and angular spreading caused by rough boundaries. The system involves three horizontal, collinear sensors and source location parameters are determined from time-delay estimates obtained by cross-correlation techniques. Tests for measuring the amount of spread or wander in correlograms with sufficiently high signal-to-noise ratios are presented. These tests are useful for determining the magnitued of the combination of scattering and time-varying effects in correlogram data. Spread is measured by comparing the average main-lobe width of stabilized correlograms over a period of time to the width of an ideal, “frozen-ocean” correlogram. Wander is estimated by a comparison of the standard deviation of correlogram peak location values over a period of time to the average main-lobe width of these correlograms. Examples of the application of the tests to simulated data are discussed. Finally, there is an identification of several correlogram types that can be associated with large amounts of spread and wander.
Simulation of side-scan sonar
Authors: Anderson Aubrey L. , Dwan Shufa
Published: October 2005

Acoustic remote sensing is becoming widespread in oceanographic measurement programs. Acoustic systems for such measurements include seafloor subbottom profilers, inverted echo sounders, tomography systems, and side-scan sonars. Numerical simulation of such systems can provide valuable information for system design and for analysis of data from an operating system. A long-range side-scan system currently under development will operate in deep water over a frequency range from 12–150 kHz and will provide side-scan images, bathymetry, and seafloor reflectivity data. Simulations of the system have been performed. These simulations provide insight into the influence of various system parameters and dispalys on usefulness of the system as a tool for the seafloor geologist. This paper describes the models used in the simulations and illustrates their use in performance prediction for this scientific data acquisition system.
Optimum time domain signal transmission and source location in a waveguide: Ideal wedge waveguide
Authors: Li Saimu , Clay C. S.
Published: October 2005

Transmissions in an ideal wedge waveguide in air are described. The walls are rigid. The Biot-Tolstoy exact wedge solution is used. The diffraction arrival from the apex is not present. Signals transmitted in a waveguide are distorted by the multiple arrivals. A waveguide matched filter has an impulse response that is the time reverse of the impulse response of transmission between a source and receiver. This filter is an optimum filter for the reduction of transmission distortion. Since the waveguide matched filter is sensitive to source and hydrophone locations, the waveguide can be used to locate a source that transmits an unknown signal. [Supported in part by the Shangdong College of Oceanography People's Republic of China, the Geophysical and Polar Research Center of the University of Wisconsin, and NORDA.]
Use of rectangular vandermonde matrix decomposition to resolve damped sinusoidal oscillation components
Authors: Henderson T. L. , Bailey R. S. , Lacker S. G.
Published: October 2005

Signals that are sums of decaying sinusoids arise frequently in passive and active sonar applications, both as spontaneous acoustic transients and as echoes from elastic objects. When the oscillations are highly damped, identification of component frequencies and damping factors is more amenable to least-squares Prony analysis than to Fourier analysis, and is particularly suited to a related technique that we call “rectangular Vandermonde matrix decomposition.” Our results indicate that the Vandermonde method is robust in the presence of noise, but not in the presence of distortions created by ideal bandpass filters. On the other hand, analysis of simulated replica-correlated echoes of FM pulses bounced off an object having a pair of heavily damped resonances give different results. Surprisingly, the Vandermonde method identified the resonant frequencies even without replica correlation, despite the fact that the analyzed waveform looked not at all like a sum of decaying sinusoids. [Work supported by the ONR.]
Localization of geophone stations by maximum-likelihood estimation
Authors: Schmidt Henrik , Snoek Michael , Van Alphen Paul
Published: October 2005

Reliable interpretation of seismic data in ocean bottom surveys or crust and upper mantle investigations requires precise knowledge of the position of the seismic sensors. Special problems arise when operating with free-falling ocean bottom seismometers (OBS). Since the ship's position and its towed signal sources (air gun, uniboom) or explosives are fairly well known based on high-precision navigation systems, one can determine the geophone positions indirectly from their response to the source excitation. Here, a newly developed source parameter estimation code [A. B. Baggeroer, W. A. Kuperman, and H. Schmidt, J. Acoust. Soc. Am. Suppl. 1 79, S56 (1986)] is first used to determine the performance bounds for a localization based on seismic interface waves. Then, an actual shear speed profile is determined by inversion of a new data set obtained using explosive sources, and a maximum-likelihood estimation of the geophone position is performed. By comparing the results with the actual experimental geometry, it is demonstrated that in spite of the fairly long wavelength of the interface waves, a reasonable resolution can be achieved by such a technique.
Preliminary results of matched-field localization using a vertical array in the Tufts Abyssal Plain
Authors: Fizell R. G.
Published: October 2005

PACIFIC ECHO was an experiment conducted jointly by the Naval Research Laboratory (NRL) and Defence Research Establishment, Pacific (DREP) in May-June 1986. Vertical array measurements of a 15-Hz cw signal projected by the NRL Mk-Vl source, towed at a depth of 100 m on a circular arc of radius 3 nmi, were analyzed with matched-field processing. The DREP array was 675 m long with the top hydrophone at 400-m depth. Matched-field processing between theoretical and measured fields was accomplished by linear correlation and by a nonlinear maximum-likelihood method (MLM) estimator. Theoretical fields were computed using a normal-modes program, the measured sound velocity profile, and an assumed thick-sediment bottom. Successful localization of the source was achieved with both estimators. Sidelobes produced by the MLM estimator were significantly below the main peak but, for the linear estimator, sidelobes were sufficiently large to be taken as false targets.
Wave-height fluctuation effects on matched-field detection and localization in shallow water
Authors: Del Balzo Donald R. , Feuillade Christopher , Rowe Mary M.
Published: October 2005

Sea surface wave-height fluctuations cause a time-dependent mismatch in environmental conditions and therefore effect the detection and localization performance of a matched-field processor. A sensitivity study was conducted to examine this mismatch phenomenon for an idealized, range-independent, Pekeris channel of 100-m depth, with a 150-Hz source, using a centrally positioned vertical array of 21 hydrophones spanning 50% of the water column. Variations in the sea surface height of + 3.5 m were considered as an extreme, but realistic, case, and the output signal-to-noise ratio (SNR) and predicted range and depth of the source were determined from a series of range-depth maximum-likelihood ambiguity surfaces. Surface height variation caused a systematic error in range estimation, such that when the depth was increased due to a wave crest, the target range appeared shorter than the actual range. The opposite held for a wave trough. The corresponding calculations of target depth were consistently biased towards greater depth. Calculations indicated that small (1%) variations in surface height can cause a loss of up to 15 dB in detection performance at a single time. However, when the computations are properly normalized and then averaged over time throughout a complete cycle of the wave-height variations, the resulting detection is dominated by the zero wave-height maximum-likelihood surface, and localization estimates based upon the position of the main peak are unambiguous.
Normal-mode filtering with orthogonal functions to avoid mode leakage
Authors: Rayborn Grayson H. , Ioup George E. , Ioup Juliette W. , Carr Janet C.
Published: October 2005

Propagation of sound in shallow water is commonly analyzed in terms of trapped normal modes. The process of normal-mode detection by adjustment of the response of individual hydrophones in a vertical array to match the pattern of a particular normal mode is normal-mode filtering. Despite the nonorthogonality of the normal modes in the water column, normal-mode filtering is based on the assumption that the modes are orthogonal. The error induced by this assumption is known as leakage. It has been estimated to range from 3% to 10% of a measured mode. In this paper, the error involved in treating the modes as orthogonal for a variety of bottom types, water depths, and source frequencies is analyzed, and a method for avoiding the error is demonstrated. Density and sound-speed ratios are selected from models fitted to experimentally measured values. To render the results indicative of a broad range of water depths and frequencies, use is made of dimensionless variables. Leakages as large as 19% are calculated.
An all-frequency solution of sound scatter by hard angular surfaces
Authors: Tolstoy I.
Published: October 2005

In estimating plane sound-wave scatter by hard bodies of simple prismatic cross section or by corrugated surfaces with angular (i.e., sawtooth or facetted) profiles it is possible to replace the actual bodies by suitable systems of single (infinite) wedges and corners plus line sources (at the vertices) plus images thereof. This procedure differs fundamentally from approximate superpositions of rigorous wedge/corner solutions used by other writers in that all orders of multiple scatter interaction are taken into account via the standard self-consistent algorithm. An infinite series representation of the field was obtained; the coefficients of which are determined by an infinite system of simultaneous linear equations, which appear to be soluble in a number of cases. The resulting solution for the total field is convergent everywhere and satisfies all the boundary conditions. In the case of an angular depression in a plane, or in the troughs of a corrugated surface with a sawtooth profile, a simple, highly accurate solution is obtained for all frequencies. [Work supported by ONR.]
Backscatter at a model of the Arctic ice canopy
Authors: Johnson Kevin R. , Denny Patrick L. , Reitzel Ken J. , Medwin Herman
Published: October 2005

The mechanisms which may cause low-frequency sound to backscatter from the Arctic ice canopy include rough surface backscatter, diffraction at leads, and reradiation of flexural waves. These phenomena have been studied by observing underwater sound pulses backscattered from an acrylic model of the Arctic ice floating on water. The flexural wave velocity and the acoustic roughness are accurately scaled and the ρc impedance contrast is approximately modeled by the selection of the acrylic material and the dimensions of the plate and roughness elements. The 3-mm-thick plate backscattering at 50 kHz then represents an Arctic ice cover 1.5 m thick and about a kilometer in extent, backscattering 100-Hz sound. The physical contributors of backscatter from the Arctic surface are studied for models of plane ice, an Arctic ice ridge, edges of leads, and a rubble field of ice. [Work supported by ONR.]
Forward scatter at a model of the Arctic ice canopy
Authors: Denny Patrick L. , Johnson Kevin R. , Reitzel Ken J. , Medwin Herman
Published: October 2005

When low-frequency underwater sound “reflects” from the Arctic ice cover, not only will it be reflected from the plane, and specularly scattered from roughness elements, but it will also be diffracted at leads and reradiated from flexural waves in the ice. In the case of near-grazing forward scatter from closely spaced roughness elements there may also be a coherent forward scattered boundary wave detectable near the under-ice surface. These phenomena have been studied in an anechoic tank by pulse transmission from an underwater point source to a large floating acrylic model of the Arctic ice. Scaling of the flexural wave velocity B(ωh)1/2 at frequency ω and ice thickness h is accurately achieved because B, for the acrylic model, is well within the range of typical values in arctic ice. Scaling of the sound scattered from roughness elements is also accomplished in terms of frequency and the length dimension. Consequently, the laboratory model at frequency 50 kHz represents 100-Hz sound encountering a typical Arctic canopy which is 112 m thick and about a kilometer in extent. The physical contributors to the gross “reflection coefficient” are studied for models of a plane ice layer, an Arctic ice ridge, edges of leads, and a rubble field of ice. [Work supported by ONR.]
Single backscatter reverberation
Authors: Codona Johanan L. , Patton Richard S.
Published: October 2005

By making use of Green's function techniques, a self-consistent, nonperturbative, single backscatter model for acoustic reverberation in the ocean was constructed. The model accounts for multiple forward scattering on both the outgoing and return paths. The model can be made to use any one-way wave equation: e.g., the standard parabolic equation or the Thomson-Chapman high-angle equation. When the standard parabolic equation is used, the technique lends itself to a numerical implementation that is much faster than competing schemes. While the model is particularly useful at low frequencies, its validity does not depend on frequency. The model's numerical speed and versatility suggest its use for the study of low-frequency reverberation from rough ocean surfaces, deterministic and random variations in the ocean volume, and range dependence in the bottom (both bathymetric and compositional).
Re-examination of convergence of rough-surface perturbation theory
Authors: Jackson Darrell R.
Published: October 2005

Several numerical studies have been reported which suggest that the rough-surface perturbation series can give accurate results even when the Rayleigh hypothesis is untrue. The question of convergence or nonconvergence is important for applications of perturbation theory and other theories based upon it (composite roughness, smoothing, phase perturbation). Numerical convergence will be demonstrated for large-amplitude sinusoidal surfaces of sufficiently small slope. Partial theoretical arguments will be given in support of the convergence criterion suggested by the numerical results. [Work supported by ONR.]
Studies of the validity of the Kirchhoff approximation for rough-surface scattering
Authors: Thorsos Eric I.
Published: October 2005

The accuracy of the Kirchhoff approximation for randomly rough, pressure-release surfaces has been examined through comparison with exact numerical results based on solving an integral equation. A Gaussian roughness spectrum was chosen for the surfaces, and a Monte Carlo procedure was used to obtain the average bistatic scattering cross section. Results will be presented that illustrate the breakdown of the Kirchhoff approximation as the incident grazing angle is reduced and as the average radius of curvature is reduced. The accuracy of shadowing corrections to the Kirchhoff approximation will also be discussed. [Work supported by ONR.]
Shallow water reverberation: A heuristic ray-mode model
Authors: Ellis Dale D.
Published: October 2005

A shallow water reverberation model has been developed based on normal mode theory together with various ray-mode analogies. The model seems to be different from other shallow water reverberation models [K. V. Mackenzie, J. Acoust. Soc. Am. 34, 62 (1962); E. C. Shang, 12th ICA Symposium on Underwater Acoustics, Halifax, N.S., Canada (1986)] in that the reverberation at time t is due to contributions from a number of different ranges. Acoustic energy propagates from the source to the scattering element via mode n, and returns via mode m. The scattering is assumed to take place at the water-surface or water-bottom interface in accordance to the scattering function M(θn, θm). The angles are determined from the mode wavenumbers in accordance with the usual ray-mode analogy. The travel times for the different paths are obtained from the modal group velocities. The final result is a double sum over the modes, but is computationally quite efficient since the mode functions and scattering angles are independent of range.