bluuebunny/crossref_metadata_2025 huggingface dataset:
Minimum dynamic lateralization for multiple moving sources
Authors: Neelon Michael F. , Jenison Rick L.
Published: February 2014

It is still unresolved whether auditory motion is perceived via specialized motion detectors or inferred from static samples of changes in spatial position. This question has been investigated by comparing minimum audible movement angles for single sources moving discretely or continuously [Perrott and Marlborough, J. Acoust. Soc. Am. 85, 1773–1775 (1989)]. But using one source does not force the listener to determine movement by only one of the aforementioned processes. To better measure sensitivity to pure auditory dynamics, the following study simulates multiple overlapping sources moving in the same direction across random sections of the auditory hemifield. Experimental stimuli are composed of different portions of the trajectories of circling resonant sources, which are individually created by dynamically varying interaural time and level differences. The multiple, variable endpoints in the composite stimulus should inhibit the listener from relying solely on such cues to determine movement direction. Pilot studies using from 1–5 concurrent sources show dynamic lateralization thresholds are lowest for one moving source. This implies movement is best perceived when stimulus endpoints can be sampled. However, thresholds across multiple sources do not significantly differ regardless of number, which may represent a measure of sensitivity to pure auditory dynamics.
Finite-difference computation of head-related transfer functions for human hearing
Authors: Xiao Tian , Liu Qinghuo
Published: February 2014

Modeling the head-related transfer function (HRTF) is a key to many applications in spatial audio. To understand and predict the effects of head geometry and the surrounding environment on the HRTF, a three-dimensional finite-difference model has been developed for the head-related impulse response. A perfectly matched layer (PML) is used to absorb outgoing waves at the truncated boundary of an unbounded medium. An external source is utilized to decrease the computational domain size. This numerical model has been validated by analytical solutions for a spherical head model. The FD code is then used as a computational tool to predict the HRTF for various scenarios. In particular, a simplified spherical head model is compared to a realistic head model up to about 7 kHz. The HRTF is also computed for a realistic head model in the presence of a wall. It is demonstrated that this FD model can be a useful tool for spatial audio applications.
Cepstrum of the scattering response of a cylinder
Authors: Sarkissian Angie , Dragonette Louis R.
Published: February 2014

The cepstrum of the scattered field of a ribbed, hemispherically end-capped cylindrical shell is computed for various incident and scattered angles, for 2≤ka≤30, where k is the wave number and a is the radius of the cylinder. Highlights observed in the cepstrum domain are related to various scattering mechanisms. The cepstrum representation is compared to the frequency domain and the time domain response of the cylinder. Benefits and disadvantages of the cepstrum representation are discussed. [Work supported by ONR.]
Acoustic scattering from a fluid-loaded membrane with periodic discontinuities using local–global homogenization (LGH)
Authors: Bliss Donald , Franzoni Linda , Danilov Pavel
Published: February 2014

A homogenization method for complex structures, applicable over the entire frequency range, is being developed. The method utilizes a local–global decomposition facilitated by adding and subtracting canceling smooth forces. The low-wave-number global problem has an infinite-order structural operator, and periodic discontinuities are replaced by an equivalent distributed suspension. The rapidly varying local problem, which provides transfer function information for the global problem, is solved separately and independently, except for amplitude information from the global problem. Once formulated for a specific structure, the self-contained global problem is solved first, and the local solution can be reconstructed afterwards. The LGH reformulation allows the global problem to be solved at much lower resolution than the length of flexural waves on the original structure. The application of this approach to scattering from a fluid-loaded membrane is described. The effects of fluid radiation are transferred entirely to the smooth global problem, whereas evanescent fluid modes are contained within the local problems. Sample calculations are presented comparing the method with classical solutions. [Work supported by ONR.]
Reconstruction of transient sound radiation from impulsively accelerated objects
Authors: Singh Manjit , Wu Sean
Published: February 2014

This paper presents reconstruction of transient acoustic radiation from impulsively accelerated objects using the HELS method [Wu, J. Acoust. Soc. Am. 107, 2511–2522 (2000)]. The radiated acoustic pressure is expanded in terms of the spherical wave functions and spherical harmonics in the frequency domain. The coefficients associated with these expansion functions are determined by matching the assumed solution to the measured acoustic pressures. The errors incurred in this process are minimized by the least squares. Once the frequency-domain solution is obtained, the transient acoustic pressure signal is reconstructed by taking an inverse Fourier transformation via the reside theorem. The objects considered include an explosively expanding sphere, impulsively accelerated rigid sphere, and impulsively accelerated baffled sphere. The acoustic pressure signals thus obtained are compared with the analytic solutions. It is shown that this methodology can be extended to a nonspherical object subject to an arbitrarily time-dependent excitation. The resulting transient acoustic pressure can be reconstructed by a convolution integral of the impulse response function to the time history of the measured acoustic pressure signals. [Work supported by NSF.]
The onset of the flexural mode in fluid loaded elastic shells revisited, and its value in object identification in both the frequency and time domain
Authors: Uberall H. , Werby M. F.
Published: February 2014

The onset of the excitation of flexural resonances for fluid loaded evacuated elastic shells produces a striking event. This issue has been interpreted theoretically in 1988 by means of a partial wave decomposition which showed a very narrow-peaked subsonic wave and a broad-peaked weak wave that begins at the speed of sound of the entraining fluid and increases. The narrow peaks are identified with subsonic water-borne waves that resonate in the fluid along the surface of an elastic object, and the broad peaks correspond to the inception of flexural waves. They exist over a small interval in wave number at the point when the flexural wave begins to couple with the fluid. Here we examine the pulse solution. With increasing frequency the partial waves change phase (a 90 degree phase change) and the overlapping flexural waves transition from a partially coherent constructive signal to one that is partially destructive. This leads to an envelope or hump effect, also called mid-frequency enhancement; it is a function of shell thickness as well as material property. We demonstrate how this effect may be employed to identify a submerged elastic shell in either the pulse or frequency solution.
The importance of the first and second symmetric Lamb modes in target identification of submerged elastic shells
Authors: Werby M. F. , Uberall H.
Published: February 2014

Resonances excited on any shell of constant thickness are one large resonance at a frequency inversely proportional to the shell thickness. Hence for thin shells it occurs at very high frequency. This effect occurs at the inception of the S1 resonance, which may be shown to be an amplitude-modulated wave at inception. Its critical frequency may be determined by the condition: ka=3.14 VLa/(2 daVW) or ka=3.14 VTa/(daVW) where VW, VT, VL, a, da, and k are the speed of sound in the ambient fluid, transverse and longitudinal velocities in the elastic material, largest dimension of the object, object thickness relative to a and the wave number in the fluid. If 2VT>VL the first condition defines the S1 critical frequency and the second that of the S2. The converse is true otherwise. The S2 resonance is not striking but may be identified as such. Thus, ratios of the two resonances lead to ratios of the bulk velocities and other considerations can isolate the shell thickness. This offers for any shell of constant thickness a means to determine the presence of certain submerged objects. We discuss the reasons for this and illustrate results in both time and frequency domains.
Reconciling static and dynamic aspects of the speech process
Authors: Lindblom Björn , Diehl Randy L.
Published: February 2014

In speech, movement and spectral change are pervasive whereas steady-states are rare. Yet descriptive frameworks (e.g., IPA) focus mainly on static properties. Similarly, in quantitative models, dynamics is generally attributed to response characteristics while the underlying control consists of static units, e.g., spatial ‘‘targets,’’ ‘‘equilibrium points,’’ or ‘‘stable auditory goals.’’ Accordingly, the input to the production of a diphthong would not be some dynamic aspect of the event, but an ordered string of two articulatory states. This approach creates a paradox when placed in the context of findings on visual and auditory perception that show that perceptual systems are more sensitive to changing stimulus arrays than to purely static ones. Clearly nervous systems have evolved to detect change. We are led to ask: If perception likes change, why assume production control in terms of static targets? We maintain that the two perspectives can be reconciled by according a significant role to positional targets in movement control and, at the same time, fully acknowledging the importance of dynamics in perception. In support of this position we present some recent computational work on simulating vowel systems by selecting optimal sets of vowels seen as objects of spectral change. [Work supported by NIH.]
Dynamic units in speech production: Evidence from speech production errors
Authors: Goldstein Louis , Byrd Dani
Published: February 2014

Measurements of vocal tract activity during speech production show many distinct parts, all exhibiting different patterns of near-continuous motion. This description stands in sharp contrast to the description of speech as composed of a small number of discrete phonological units. In recent years the tools of dynamical systems have been brought to bear on the problem of relating these descriptions. The continuous vocal tract activity can be theoretically decomposed into discrete units of action, or gestures, each of which is a dynamical control regime for vocal tract articulators. These gestures can also serve as primitive phonological elements. Evidence for gestures as units of the speech production process has recently been obtained in experiments that measure the kinematics of speech errors produced when a simple phrase (e.g., ‘‘cop top’’) is repeated. In errors, one of the gestures comprising the phrase may be duplicated in an anomalous position in the utterance. Since the anomalous gesture is often reduced in magnitude, such errors could not be detected without articulatory movement data. The nature of the errors and their asymmetries are consistent with the hypothesis that they represent phase transitions of a system of coupled dynamic units. [Work supported by NIH.]
The synergy between speech production and perception
Authors: Shamma Shihab A.
Published: February 2014

Speech intelligibility is relatively unaffected by certain spectro-temporal deformations of the signal spectrogram. These include spectral and dynamic translations, stretching or contracting dilations, and shearing of the spectrogram along its temporal or logarithmic frequency axis. We shall argue that this stability results from a synergy between the dynamics and characteristics of the auditory cortex (the receiver) and of the vocal tract (the source). Specifically, on the perception side, neurophysiological evidence suggests that the acoustic spectrogram is represented in the primary auditory cortex along multiple spectro-temporal scales, and that the above deformations correspond to simple translations of the representation of the dynamic acoustic spectrum along these axes. On the production side, these spectro-temporal distortions can be caused by common variations among talkers in speaking rates, slurred articulatory dynamics, vocal tract length, cross-sectional profile, and losses. Using a simplified sinusoid model of the vocal tract, it is possible to relate ‘‘articulatory’’ parameters (e.g., the extent and location of a vocal tract constriction) directly to the formants, and hence to the cortical representation of the spectrum. A similar remarkable synergy exists between the dynamics of perception (e.g., via the perceived spectro-temporal modulation transfer functions) and the speed of articulation and hence the syllabic rates of speech. These results suggest an intimate link between perception and production, a view which offers new insights into the organization of speech phonemes (such as vowels) in terms of different vocal tract configurations.
Vowel identity change: Trajectory length, transition velocity, or effort?
Authors: Divenyi Pierre L.
Published: February 2014

To what degree does the vocal tract shape need to be changed for a listener to perceive a vowel change? A 100-ms vowel V1 was followed by a transition toward vowel V2. V1 and V2 were two vowels chosen from the triplet /a/, /i/, and /u/. Subjects had to indicate their confidence level that the transition, in fact, reached V2. Data were converted to response-operating-characteristic (ROC) curves from which the transition velocity and transition duration were calculated for a fixed criterion of V2 percept. Results indicated that, regardless of the actual V1V2 trajectory, the criterion was reached when the product of transition duration and transition velocity (=the trajectory length) was approximately constant. Since there is no logical explanation that could account for an invariance of trajectory length across all V1V2 pairs, and since the criterion was reached with transition velocities that are not invariant, it appears that a constant percept is obtained when the transition encounters an elastic, opposing force that is inversely proportional to the acceleration at the start of the transition. This explanation, based on inferred articulatory effort, suggests the existence of a link between production and perception dynamics. [Work supported by NIH and the VA Medical Research.]
Incorporating uncertainty in ocean acoustics for optimum signal detection
Authors: Nolte Loren W. , Tantum Stacy L. , Sha Liewei
Published: February 2014

An overview of detection, classification and localization of signals in an uncertain ocean environment is presented from a Bayesian perspective. The Bayesian approach allows one to both model and incorporate, in an optimal way, the inherent uncertainty that often exists in the knowledge of ocean acoustic environmental parameters. The relationship between environmental parameter estimation and signal detection is illustrated, and it is shown how environmental parameter estimation can be viewed as an inherent part of the optimum Bayesian detector. Furthermore, by using the Bayesian approach, one can design optimum detection algorithms that are robust with respect to precise knowledge of environmental parameters. Using the receiver operating characteristic (ROC), one can illustrate the trade-off among detection performance, environmental uncertainty, and signal-to-noise ratio. Several examples are taken from an uncertain shallow water environment to illustrate these trade-offs. Finally, it is shown how depth classification can be viewed as an optimal detection problem. [Work supported by ONR.]
Towed array processing as a Bayesian problem
Authors: Sullivan Edmund J.
Published: February 2014

The towed array range and bearing estimation problem is cast in the form of a Bayesian estimation problem. It is shown that by casting the problem in the form of a MAP estimator, the Kalman estimator naturally follows. There are two advantages to this. First, the problem becomes recursive, resulting in an adaptive processor. Second, once the problem is in the Kalman form, it becomes possible to include sophisticated models of the signals and noise in a natural way. Thus, performance can be enhanced, since the models essentially provide a priori information to the processor. In this work an algorithm, which explicitly contains the forward motion of the array, has been developed. It is capable of performing bearing and range (wavefront curvature) estimation with a short array. Here, short means that the ratio of the physical aperture of the array to the range of interest is small. An example using simulated data is given where an array with a length of 45 meters is capable of determining the range and bearing of a narrow band source. Results are shown for several array speeds, signal-to-noise ratios bearings and ranges.
Separation and localization of acoustic signals
Authors: Knuth Kevin H.
Published: February 2014

The cocktail party problem is considered where there are several sound sources present in a free-field environment and an observer, who has the ability to record only linear mixtures of these sounds, is interested in obtaining estimates of the original source signals. Previous application of Bayesian inference to the general problem of blind source separation has demonstrated that it is possible to include additional prior information such as the transmission of the signals to the detectors and the possible locations of the sources to improve the source estimates. The derivation of prior probabilities describing such information will be discussed as will the construction of useful models and application of a general Bayesian methodology allowing one to deal with noninstantaneous inverse square transmission, noisy environments and any number of sources and detectors. Methods for dealing with environments in enclosure will also be described. [Work supported in part by NARSAD Young Investigator Award.]
Acoustic benchmark analysis of GRASP optimum sonar search plans
Authors: DelBalzo Donald R. , Wagstaff Melvin A. , Collins Mona A. , Leclere James H. , Hempsteter Kevin P.
Published: February 2014

An application of a genetic algorithm and Monte Carlo simulation with Bayesian detection statistics is used to optimize sonar search tracks in nonhomogeneous environments. The optimization metric is maximum cumulative detection probability for a specific sonar (passive or active) against a target with specified characteristics (acoustic and tactical) during a fixed time period. This new search planning capability is named GRASP for genetic range-dependent algorithm for search planning. The sensitivity of GRASP solutions to various ocean environments is examined under benchmark conditions, i.e., fairly simple synthetic environments and a simple target model. The results indicate that the genetic algorithm produces intuitive tracks that correlate highly with acoustic signal excess predictions, as expected.
Extracting acoustic observables from the NPAL billboard array data
Authors: Dzieciuch Matthew , Colosi J. A. , Cornuelle B. D. , Dushaw B. D. , Dzieciuch M. A. , Howe B. M. , Mercer J. A. , Spindel R. C. , Worcester P. F. ,   NPAL Group
Published: February 2014

Low-frequency (75-Hz) acoustic signals were repeatedly transmitted over a 1 year period and sampled vertically (with up to a 1400-m aperture) and horizontally (with a 3600-m cross-range aperture) by a distant bill-board array (3900-km range) as described by the NPAL Group. The data are complicated by the fact that the sound interacts with the bottom near both the source and receiver. Vertical beamforming is used to filter the bottom interacting energy, and thus allow analysis of the fundamental acoustic properties. Subband and subarray processing is used to produce estimates of arrival times and angles or resolved ray arrivals. Time-series of acoustic travel-times and arrival angles are then developed.
Analysis of mode coherence and intensity at megameter ranges
Authors: Wage Kathleen E. , Colosi J. A. , Cornuelle B. D. , Dushaw B. D. , Dzieciuch M. A. , Howe B. M. , Mercer J. A. , Spindel R. C. , Worcester P. F. ,   NPAL Group
Published: February 2014

The low-order modes constitute some of the most energetic arrivals at long ranges. Understanding fluctuations of these mode arrivals is crucial to their use as observables in matched field processing and tomography. Both simulated and experimental data indicate that at megameter ranges, the low modes have complex arrival patterns due to internal-wave-induced coupling. Analysis of broadband receptions at 3515 km from the ATOC experiment has shown that mode coherence times are on the order of 6 minutes and that centroid statistics provide useful measures of arrival time trends over the course of several months [Wage et al., IEEE Sensor Array and Multichannel Signal Processing Workshop Proceedings, pp. 102–106, 2000]. The North Pacific Acoustic Laboratory (NPAL) experiment presents an opportunity for further research on broadband mode arrivals at megameter ranges. This study examines temporal coherence, intensity variations, and other mode statistics using data from the 40-element NPAL vertical line array. Experimental results are compared with PE simulations of propagation through internal waves of varying strengths, and the impact of the up-slope propagation near the receivers on the mode statistics is discussed.
Observing horizontal wave fronts from the NPAL billboard array data
Authors: Dzieciuch Matthew , Colosi J. A. , Cornuelle B. D. , Dushaw B. D. , Dzieciuch M. A. , Howe B. M. , Mercer J. A. , Spindel R. C. , Worcester P. F. ,   NPAL Group
Published: February 2014

One of the main objectives of the NPAL experiment is to investigate the horizontal refraction and coherence of the acoustic wave fronts at long range. Given time series of acoustic arrival times and angles of resolved ray arrival arrivals, a detailed look at the acoustic wave fronts is possible. First and second order statistics (density functions and coherences) of the wave fronts are investigated. The wave fronts are shown to vary with time, frequency, depth and across the horizontal aperture.
The terminal problem: Model-data comparisons of basin scale transmissions including effects of shallow bathymetry near source and receiver
Authors: Heaney Kevin D. , Worcester P. F. , Colisi J. A. , Cornuelle B. D. , Dushaw B. D. , Dzieciuch M. A. , Howe B. M. , Mercer J. A. , Spindel R. C. ,   NPAL Group
Published: February 2014

Acoustic transmissions on basin scale ranges are being used to determine depth-dependent temperature variability. With travel time being the primary observable, stationary sources and nearly stationary receivers are experimental requirements. This has led to the use of bottom-mounted sources and receivers to reduce travel time variability. The NPAL (North Pacific Acoustics Laboratory) experiment has transmitted broadband acoustic pulses from two bottom-mounted sources near the sound channel axis. Recordings have been taken on the NPAL billboard array, a linear series of five vertical line arrays moored in 1800 m of water near Monterey, CA. Additional recordings have been taken from the SOSUS system throughout the Pacific basin. The effects of the near source and near receiver scattering are examined. In particular, near source scattering leads to excess high-angle energy entering deep water with a travel time delay of nearly 1 s due to the low group speeds of high-angle rays/modes in shallow water. We also compare the energetics of the arriving rays that have bounced once on the rising seafloor near the NPAL receivers. Comparisons of models and data for bottom interacting acoustics lead us to the perennial issue of geoacoustic parameters.
Bottom grazing acoustic arrivals: Explanation of anomalous arriving deep multipaths
Authors: Heaney Kevin D. , Cox Henry
Published: February 2014

During the 1996 Acoustic Engineering Test (AET) of the ATOC project, receptions were recorded on deep SOSUS arrays in the central Pacific. With the given SNR, several pairs of doublet arrivals are visible. These doublets have the form one would expect from an axial source, but to a receiver much shallower in the water (∼500–1000 m). The presence of this energy, well below the turning point of the corresponding rays has been termed the ‘‘acoustic shadowzone phenomena’’ [Dushaw et al., IEEE J. Ocean. Eng. 24, 202–214 (1999)]. Range independent normal mode modeling for an axial source reveals bottom grazing acoustic energy at all ranges. The arrival envelope (pairs of doublets) is independent of range. These arrivals are present in ray theory calculations when bottom grazing ray arrivals are included. With a critical angle of 18 deg, there are many subcritical bottom interacting rays with very little attenuation. This phenomenon has been understood in shallow-water acoustic problems where the reflection coefficient is linear with angle. The time spread of the pulse (number of rays) grows linear with range. The attenuation of the pulse decays linearly with range. The result is an envelope that is independent of range.
Scintillations in long-range acoustic transmissions
Authors: Colosi John , Tappert Frederick , Dzieciuch Matthew
Published: February 2014

In the Acoustic Thermometry of Ocean Climate (ATOC) program’s Acoustic Engineering Test (AET), broadband 75-Hz center frequency transmissions were recorded on a 700-m-long vertical array, 3252 km distant. Previously reported results from the AET using 12.7-min. averaged data by Colosi et al. [J. Acoust. Soc. Am. 105, 3202–3218 (1999), hereafter referred to as Colosi99] revealed surprisingly weak acoustic scattering for early arriving identifiable wave-fronts; these results have been confirmed using 1.8-min averaged data. It is shown that scintillation index (SI) is a strong function of position along the pulse with smallest values occurring at the peak and larger values occurring at the tails. Intensity PDFs of identifiable wave-fronts are reanalyzed in terms of both peak intensity and integrated pulse energy (IE), but both PDFs are very closely log-normal. Regarding multipathing along the wave-fronts it is found that on average there are 1.7 peaks per wave-front segment per hydrophone. The combined observation of weak scattering and multipathing is a novel result. Colosi99 analyzed the finale in terms of peak scintillations and found a near log-normal intensity PDF. Reprocessing the full field without limiting data to intensity peaks and accounting for mean intensity nonstationarity yields an exponential intensity PDF.
Noise field statistics and coherence on the NPAL array
Authors: Baggeroer Arthur B. , Scheer Edward K. , Colosi J. A. , Cornuelle B. D. , Dushaw B. D. , Dzieciuch M. A. , Howe B. M. , Mercer J. A. , Spindel R. C. , Worcester P. F. ,   NPAL Group
Published: February 2014

While the NPAL array was primarily deployed to examine the spatial coherence of the Hawaii source, it is also a rich data set for ambient noise studies. Shipping noise, earthquakes and biologics all have been identified in the NPAL data. Moreover, ambient noise coherence is the primary issue in maximizing the SNR output of a sonar system. The first and second order statistics of data from the NPAL ‘‘noise only’’ segments have been analyzed with the following results: (i) There is a wide spread in the peak levels, most likely due to the proximity to shipping lanes. The maximum peak level in the recording band is 117 dB. (ii) Full broadband coherences tend to be low because of the presence of many ships. (iii) If one examines frequency bands of 1–2 Hz, then lines of individual ships can be identified and associated and they are very coherent across NPAL aperture. (iv) Vertical beamforming indicates relatively highly directional spectra at low grazing angles and ‘‘noise notch’’ for the spectra at higher frequencies. Horizontal beamforming has been difficult to implement due to element positioning errors and the large array transit time.
A model for acoustical evaluation of rectangular concert halls
Authors: Bayazit Nurgun Tamer
Published: February 2014

The studies on room acoustics have shown the importance of hall geometry on the acoustical performance perceived subjectively by the audience. The aim of this study is to examine the effects of the geometrical parameters on the objective acoustical parameters and to develop an analytical model which will help designers in the early stage of design. The acoustical parameters of the model halls designed, selected as rectangular forms with varying dimensional ratios, were calculated with the aid of the ODEON room acoustical simulation program. In order to obtain realistic results with the proposed model, the geometrical parameters of model halls were chosen so as to allow comparison with those of existing halls and the set of data used for the calculation of acoustical parameters were chosen to reflect real conditions in the best possible way. Calculated acoustical parameters were statistically analyzed to derive an analytical model based on the relationship between the hall geometry and the acoustical parameters. The geometrical parameters are fed into the model as independent variables, and the acoustical parameters are calculated on the basis of them.
Calculation of diffuse sound reflections in predicting room acoustics using the image method
Authors: Lee Hee-Won , Ko Il-Doo , Doo Se-Jin , Oh Yang-Ki , Jeong Dae-Up , Choi Chul-Min , Kim Lai-Hoon
Published: February 2014

Numerous investigations have demonstrated that diffuse reflection is one of the most important factors in predicting room acoustics by computer simulation. Recent studies have suggested several computational algorithms in order to account for diffuse reflections in the ray-tracing or beam-tracing method. In this study, a computational algorithm for the calculation of diffuse sound reflections in the image method is suggested and a computer simulation system is developed based on the suggested algorithm. The methodology adopted in our computer simulation system is similar to the extended radiosity method, which was developed for computer graphics. Various descriptions of room acoustics, including spatial distribution of image sound sources, impulse responses and other commonly used room acoustical measures, can be obtained from the MLS based monaural room acoustics measurement system. The measured results in a midsized rectangular classroom with/without chairs on the floor were compared with the predicted results using the computer simulation in which diffuse reflection coefficients of the floor were varied. [Work supported by Korean Research Foundation Grant KRF-1999-1-310-004-3.]
Interactive and aliasing-free acoustic modeling of reflections and diffractions in architectural environments
Authors: Funkhouser Thomas , Tsingos Nicolas , Carlbom Ingrid , Elko Gary , Pingali Gopal , Sondhi Mohan , West Jim
Published: February 2014

A primary challenge in geometrical acoustic modeling is computation of reverberation paths from sound sources fast enough for real-time auralization. This paper describes an aliasing-free beam tracing algorithm based on precomputed spatial subdivision and beam tree data structures that enables real-time acoustic modeling and auralization for sound sources in interactive applications. The proposed method traces convex polyhedral beams from the location of each sound source and receiver through a precomputed spatial subdivision data structure, constructing a beam tree representing the regions of space reachable by potential sequences of transmissions, diffractions, and specular reflections at surfaces of a 3D polygonal model. By computing beam trees asynchronously (off-line), our system can generate reverberation paths between sources and receivers at interactive rates and spatialize audio signals in real-time as sources and receivers move under interactive user control. Unlike previous geometrical acoustic modeling work, our beam tracing method: (1) supports evaluation of early reverberated paths at interactive rates, (2) scales well for large, densely occluded architectural environments, and (3) computes paths of diffraction without aliasing using the uniform theory of diffraction. This system is being used to develop interactive applications in which a user experiences a virtual environment immersively via simultaneous auralization and visualization.
Analysis of frequency structure of sperm whale clicks as a function of dive depth and animal orientation
Authors: Thode Aaron M. , Mellinger David K. , Stienessen Sarah , Martinez Anthony , Mullin Keith
Published: February 2014

During overnight tracking of a pod of sperm whales in the Gulf of Mexico in July 2000, the NOAA ship GORDON GUNTER recorded their characteristic ‘‘click’’ sounds on a five-element towed hydrophone array. Multiple reflections from the surface and ocean bottom were also recorded. Analysis of the arrival times and bearings of the reflections allowed the computation of three dimensional dive profiles for several animals. By assuming the orientation of the animal was aligned with its velocity, the relative orientation of the animals relative to the array could also be estimated. A visual examination of the frequency content of the received clicks versus dive time suggested that the double resonances in the 1.2 and 2.2 kHz band increased 20%–30% during depth changes of 1000 m. In this presentation the possible relationship between click structure and whale depth and orientation is rigorously analyzed, and the observed relationships are compared with predictions from various sound production and resonator models, in an attempt to gain insight into the sound production mechanism. [Work supported by Minerals Management Service, National Marine Fisheries Service, and ONR.]
The underwater whistle repertoire of wild false killer whales (Pseudorca crassidens)
Authors: Nester A. E. , Thomas J. A. , Acevedo-Guiterrez A.
Published: February 2014

There is limited information on the underwater repertoire of false killer whales (Pseudorca crassidens). Underwater recordings of feeding and traveling whales were made near Isla de Coco, Costa Rica in January and February 1994. This study classified the whistle repertoire, analyzed acoustic characteristics, examined differences associated with feeding and traveling whales, and examined whistles in varying noise environments. Whistles were categorized by contour into eight types; 197 were during feeding and 172 during traveling. Whistles were 1220 to 17891 Hz, had harmonics with intervals between 1797 and 13125 Hz, and were 0.03 to 2.07 s long. Some differences in whistles from traveling and feeding whales were found. Cluster analysis, principle component analysis, and stepwise discriminant function were used to classify whistle types. Ambient noise was measured at 1 and 2 kHz in recordings of seven different ambient environments (over a 25-dB range). Quieter environments were associated with feeding whales. When ambient noise was high, whales shifted the frequency of whistles higher and increased the duration and total number of whistles in a series. Results provide evidence for whales adapting frequency and time structure of their whistles in different noise environments.
Separation of the acoustic ‘‘sound space’’ by nearshore fishes in the Great Barrier Reef, Australia
Authors: McCauley Robert D. , Cato Douglas H.
Published: February 2014

Long-term sea noise recordings were made off Cowley Beach. Four fish sources dominated ambient noise. Each displayed unique patterns in calling: character, habits, timing, and spatial extent. Two sources, a knocking and drumming, were up to 2 s long, always occurred as distinct calls, had spectral peaks over 350–450 Hz and had respective mean squared pressure (msp) source levels of 132 and 149 dB re 1 μPa @ 1 m. Call rates to 120/10 min occurred during the Austral summer, the highest calling period for all sources. The other sources, a ‘‘trumpet’’ and ‘‘pop,’’ occurred in choruses and, respectively, were 100–220 ms and <20 ms long; had spectral peaks over 800–1500 Hz and 400–700 Hz; and had source levels of 150 and 141 dB re 1 μPa @ 1 m (msp). Trumpet choruses were up to 2 km across and detectable to 8 km from chorus center in water <10 m deep. Popping choruses were dispersed over larger regions. It was possible that in nearshore waters all sources occurred within a 24-h period. Sources avoided competition for the acoustic ‘‘space,’’ partly by offsetting spectral peaks and geographic separation, but primarily by separating the time of chorus or maximum call-rate.
Patterns of echolocation in a group of captive bottlenose dolphins in Roatan, Honduras
Authors: Heinemeyer Heather L. , Thomas J. A.
Published: February 2014

Patterns of echolocation in a group of bottlenose dolphins (Tursiops truncates) at the Institute for Marine Sciences in Roatan, Honduras were examined. While much is known about echolocation, the role of echolocation in the context of dolphin social or communicatory behavior has been, for the most part, speculative (Xitco, 1996). Two females and two males were paired and presented a variety of novel targets in the water. Approximately equal numbers of each pair combination were conducted over 22 sessions consisting of a 5-min no-target trial followed by a 5-min target trial. Data were obtained via an underwater video camera and by an underwater observer. Number of click trains, silent passes, passes by both dolphins, passes by a clicking dolphin, and times a dolphin approach and click on the target first were analyzed. Clearly, dolphins paid attention to the targets. The number of trains and passes significantly increased during target trials. Females clicked more often during target trials and were first to approach and click at the target. Target type had no effect on echolocation behavior. This study should be repeated at other facilities and with other dolphins to determine whether gender differences are real or due to small sample sizes.
Matched-field geoacoustic inversion in a range-varying waveguide
Authors: Gillard Chris A. , Thomson David J. , Ebbeson Gordon R. , Heard Garry J.
Published: February 2014

In recent years, geoacoustic inversion methods based on matched-field processing concepts have been devised for inferring the sub-bottom properties of shallow-water waveguides using measurements of acoustic signals received on hydrophone arrays. The development of these matched-field inversion (MFI) procedures is driven by two goals: (1) to obtain an accurate geophysical description of the morphology of the sub-bottom structure, and (2) to characterize the acoustic response of the sub-bottom so that its effect on sound propagation can be predicted. Knowledge of (2) can improve the effective ranges for using sound in underwater communication and/or source localization applications. Of practical interest is the question: How crude can our estimates of the sub-bottom structure be and still allow successful acoustic source localization using model-based signal processing methods? In this paper, a hybrid simplex simulated annealing MFI code [M. R. Fallat and S. E. Dosso, J. Acoust. Soc. Am. 105, 3219–3230 (1999)] is combined with a high-order parabolic equation (PE) model and applied to geoacoustic inversion in a shallow-water waveguide whose sub-bottom properties vary with range. In particular, we investigate simple sub-bottom characterizations for some test cases considered recently at the Inversion Techniques Workshop held in Gulfport, MS. a)Presently on exchange from the Defence Science and Technology Organisation, Australia.
Range-dependent geoacoustic inversion
Authors: Lapinski Anna-Liesa , Dosso Stan , Chapman Ross
Published: February 2014

This paper considers acoustic inversion for a minimum-structure geoacoustic model in a range-dependent environment. The general survey configuration consists of recording acoustic fields at a vertical array of sensors due to a number of sources distributed in range along a track. The unknown model consists of the sound speeds and thicknesses of sub-bottom layers, which vary with range in an arbitrary manner, and geometric parameters of the experiment. The goal is to determine a range-dependent geoacoustic model with the least structure that is consistent with the resolving power of the data. An under-parametrized approach is developed, which consists of solving the inverse problem a number of times, each time increasing the range variability allowed in the model. The optimal parametrization is subsequently obtained by examining the data mismatch and a norm of the model structure as a function of the number of parameters. The individual inversions are performed using an adaptive hybrid inversion algorithm applied to all data simultaneously.
Range-dependent seabed characterization by inversion of acoustic data from a towed receiver array
Authors: Siderius Martin , Nielsen Peter L. , Gerstoft Peter
Published: February 2014

The MAPEX2000 experiments were conducted by the SACLANT Undersea Research Centre in the Mediterranean Sea in March 2000 to determine the performance of acoustic inversion for seabed properties using a towed, horizontal receiver array. Towed systems are advantageous because they are easily deployed from a ship and the moving platform offers the possibility of estimating spatially variable (range-dependent) seabed properties. Previous research has successfully applied matched field processing (MFP) geoacoustic inversion techniques to measured acoustic data, however, in nearly all cases the inverted data were collected on moored, vertical receiver arrays. In the MAPEX2000 experiments acoustic data were collected on both towed and moored vertical receiver arrays. Results will be presented showing that seabed properties (e.g., seabed sound speed, sediment layer thickness and attenuation constant) can be extracted using MFP inversion of acoustic measurements from a towed array of receivers. These seabed properties agree with those inverted using data received simultaneously on a vertical array. These findings imply that a practical technique can be developed to map range-dependent seabed properties over large areas using a towed acoustic system. An example of such a range-dependent inversion is given for two sites from the MAPEX2000 experiments.
Parameter coupling in broadband geoacoustic inversion
Authors: Perkins John S. , Collins Michael D. , Dacol Dalcio K. , Fialkowski Laurie T. , Lingevitch Joseph F.
Published: February 2014

For many geoacoustic inverse problems involving a single frequency data set, the search parameters are often strongly coupled to one another. This may mean that individual parameters are difficult to determine, but combinations of parameters can be resolved. In terms of the optimization process, the parameter landscape that is being searched has multidimensional valleys that hinder location of a global minimum. A coordinate rotation technique has been developed [J. Acoust. Soc. Am. 98, 1637–1644 (1995)] to speed the search for optimal parameters. This technique also gives valuable information about the parameter hierarchy and parameter couplings. One approach to decoupling the parameters and obtaining realistic physical parameter values is to process broadband data or multifrequency data. A new fast forward modeling technique [J. Acoust. Soc. Am. 106, 1727–1731 (1999)] makes it possible to consider broadband data for geoacoustic inversion. The coordinate rotation technique is combined with fast forward modeling to study parameter couplings for broadband and multifrequency problems. [Work supported by ONR.]
Chirp sonar inversion results from SWAT East China Sea and New Jersey Shelf experiments
Authors: Turgut Altan , Lavoie Dawn L. , Lambert Douglas N. , Walter Donald J. , Ohta Kazuhiko
Published: February 2014

Subbottom surveys have been conducted during recent shallow water acoustic technology (SWAT) experiments to invert bottom geoacoustic properties at the East China Sea and New Jersey Shelf sites. Sediment properties such as density, porosity, and sound-speed profiles are inverted by using reflection amplitude and phase data obtained from a hull-mounted 2- to 12-kHz chirp sonar, a deep-towed 2- to 12-kHz chirp sonar, and a 30-kHz acoustic sediment classifier system. The attenuation coefficient is estimated using the frequency shift method which seems to be relatively insensitive to reflection and transmission effects, source-receiver beam patterns, and instrument responses. All three systems provided high-quality reflection data and inversion results are in agreement with those of sediment core measurements. [Work supported by ONR and JDA.]
Synthetic array imaging for biomedical ultrasonics
Authors: O’Donnell Matthew
Published: February 2014

Real-time array imaging with ultrasonics has been one of the primary tools of medical diagnostics for over two decades. Scanning systems based on phased array principles require a completely independent, programmable electronic channel for each element in the active aperture. Such complexity is acceptable for clinical systems propagating ultrasound across the body surface at frequencies ranging from 1 to 10 MHz. However, for high-frequency systems integrated into therapeutic devices entering the body through catheters or needle probes, system complexity must be minimized. Indeed, a full phased array approach is nearly impossible for most of these systems. An alternative method is to use synthetic imaging principles, greatly simplifying system and array complexity. There are many practical problems in implementing synthetic imaging principles for real-time medical scanners. Theses issues, along with possible solutions, will be presented in this talk. In addition, a fully confocal, real-time intravascular synthetic array imaging system operating at ultrasound frequencies ranging from 10 to 30 MHz will be presented. Finally, methods for fully synthetic imaging systems operating at frequencies in excess of 50 MHz will be described, and prospects for implanting such systems in the human body will be discussed.
Effect of aperture size on compensation of arrival time distortion
Authors: Lacefield James C. , Waag Robert C.
Published: February 2014

The effects of aperture size on focus distortion and time-shift compensation were investigated using a two-dimensional array system and a tissue-mimicking distributed aberration phantom that produces 65 ns root-mean-squared arrival time fluctuation. Pulse-echo data were acquired at 3 MHz using six apertures with areas ranging from 12.6×12.6 to 47.4×47.4 mm2, each focused at 55 mm. Transmit beams produced with and without compensation for arrival time fluctuations estimated from random scattering data were measured using a hydrophone. Focus degradations were characterized using 25% deviation levels, i.e., the level below the peak at which the aberrator path beam becomes 25% broader than the water path beam. Without focus compensation, the 25% deviation level was above −6 dB in five of eight trials performed with the smallest aperture and seven of eight trials with the largest aperture. With compensation, the 25% deviation level dropped below −6 dB in five trials with the smallest aperture and six trials with the largest. The results demonstrate that larger apertures are more likely to experience a decrease in point resolution caused by aberration. Point resolution is restored by time-shift compensation more readily than contrast resolution, so that method is more useful with larger apertures.