bluuebunny/crossref_metadata_2025 huggingface dataset:
Prediction of flow-induced structural vibration and sound radiation using energy flow analysis
Authors: Han F. , Bernhard R. J. , Mongeau L.
Published: July 2002

The energy flow analysis method (EFA) was used to predict the vibration response and the associated radiated sound power of a plate excited by turbulent boundary layer flows and separated–reattached flows. The power input from the turbulent flow into the structure was calculated using the mechanical impedance of the structure and the cross-power spectral density or the wave-number-frequency spectral density of the wall pressure field. The Smol’yakov–Tkachenko model was used to describe the pressure field under the turbulent boundary layer, and the Corcos model was used to describe the wall pressure field beneath the separated–reattached flow. Experiments were performed using a clamped aluminum plate mounted flush with the floor of a low-speed, closed test section wind tunnel. A fence was used to trip the flow upstream of the plate. The wall pressure fluctuations, the plate velocity, and the acoustic pressure radiated from the plate on its quiescent side were measured using flush-mounted microphones, a scanning laser vibrometer, and a microphone located in a soundproofed enclosure located beneath the test section floor, respectively. Frequency-averaged EFA predictions compared favorably with the measured data in the frequency range where the plate modal density was sufficiently high.
Effect of vocal tract inertance on phonation threshold pressure: theory and measurements
Authors: Chan Roger W. , Titze Ingo R.
Published: July 2002

In a previous analytical study [Titze, J. Acoust. Soc. Am. 83, 1536–1552 (1988)], it was shown that vocal fold oscillation is facilitated by the presence of a vocal tract, i.e., phonation threshold pressure is lowered by an inertive acoustic impedance of the vocal tract. Results of a recent simulation study [Titze and Story, J. Acoust. Soc. Am. 101, 2234–2243 (1997)] support the theory and show that a narrow ‘‘epilarynx tube’’ (area <1.0 cm2) in the lower vocal tract provides impedance-matching and gives the biggest facilitative effect. This small-amplitude oscillation theory was revised in the present study to simultaneously account for vocal fold mucosal wave propagation and vocal tract inertance, as energy transfer mechanisms for flow-induced oscillation. Analytical expressions of phonation threshold pressure were derived showing the relative contributions of the two mechanisms. Empirical data on the effect of vocal tract on phonation threshold pressure were also obtained in a physical model of the vocal fold mucosa [Titze et al., J. Acoust. Soc. Am. 97, 3080–3084 (1995)]. Both analytical and experimental results show that the facilitative effects of vocal tract on the ‘‘ease’’ of phonation are the most prominent under specific glottal geometry and biomechanical conditions. [Supported by NIH Grant No. P60-DC00976.]
Articulatory speech recognition using hierarchical, interpolating hidden Markov models
Authors: Fain Daniel C. , Barr Alan H.
Published: July 2002

Articulatory speech recognizers infer the configuration and movement of a speaker’s vocal apparatus as an intermediate representation between acoustics and phonetics. Articulatory modeling makes coarticulation processes explicit, as opposed to the conventional triphone technique of elaborating all coarticulation combinations. In the present study, hierarchical, interpolating hidden Markov models (HMMs) transform acoustics into articulation and articulation into phonetics. Articulatory and acoustic data with transcriptions are used for both training and testing. The articulatory data, collected by the University of Wisconsin, consist of synchronized recordings during speech of x-ray microbeam positions, throat vibration, and acoustics. State variables of the Markov model represent articulator configurations. These variables change continuously between targets to represent coarticulation. In this model, their state changes are driven by a top-level state variable which changes from phone to phone. The model emits both articulator positions and acoustic analysis coefficients, when viewed as a generator (in recognition, these emissions are the inputs). Variational techniques make the training problem tractable. The latest phonetic-recognition results of the model will be contrasted with baseline HMM systems trained on acoustic and articulatory data. [Work supported in part by NIMH, the NSF/DARPA STC for Computer Graphics and Scientific Visualization, and Hewlett–Packard.]
Optimal performance bounds for the classification of acoustic sources by depth in an oceanic waveguide
Authors: Tantum Stacy L. , Nolte Loren W. , Wazenski Michael T.
Published: July 2002

The classification of acoustic sources by depth is an application of matched-field processing which is important for both civilian and military surveillance operations, where the emphasis is placed on the correct classification of surface and submerged sources rather than the accurate estimation of the source’s depth. This classification task can be posed as a composite hypothesis decision problem and approached using signal detection theory. The strength of this approach is that it is robust to uncertainties in the propagation environment and the specific source position. The likelihood ratio for this detection problem has been derived and used to determine through simulation the performance bounds for this problem under varying degrees of uncertainty regarding the acoustic environment. Results are presented in the form of receiver operating characteristic (ROC) curves for a shallow water channel with various levels of a priori knowledge concerning the source position. [Research supported by SPAWAR.]
Diffraction from simple shapes by a hybrid asymptotic-finite element method
Authors: Montgomery Joshua M. , Barbone Paul E.
Published: July 2002

The application of a hybrid asymptotic/finite element method to the problem of scattering from two-dimensional, submerged objects is considered. The hybrid method is based on patching a short-wavelength asymptotic expansion of the scattered field to a finite element interpolation of the near field. In patching, the diffracted field shape functions with unknown amplitude are forced to agree smoothly with the solution in the near field along a curve at a prescribed distance from the diffraction points. A new hybrid finite element on this artificial boundary represents the effect of the outer domain on the solution within this new boundary. This allows the replacement of the original boundary value problem with an asymptotically equivalent boundary value problem, the domain of which is small and efficiently discretized. The method is applied to diffraction by a blunted wedge, which in this context represents a degenerate prism. The hybrid scattering solution shall be compared to an analytic field representation found using an exact Dirichlet-to-Neumann map.
An overview of the shallow water geoacoustic inversion workshop97
Authors: Chapman N. R. , Tolstoy A.
Published: July 2002

In June of 1997 a workshop was held in Vancouver BC with the focus on shallow water geoacoustic inversion methods. The intent was to benchmark a wide variety of inversion methods all applied to common data sets. For that purpose a number of simulated data sets consisting of acoustic fields for vertical and horizontal arrays were prepared and distributed to participants but the values of the geoacoustic parameters were NOT supplied to users. Such parameters included: water depth, sediment layer thickness, sound-speed profile, density, and attenuation, half-space sound-speed, density, and attenuation, and source range and depth. This talk will discuss those test cases and summarize the results obtained by inversion methods based on simulated annealing, genetic algorithms, cw and broadband approaches, neural nets, iterative parameter sequencing, and random searches with gradient descent. Propagation models included normal modes, PE, ray, and full field wave-number integration where cost functions were generally based on the Bartlett MFP but also included high resolution Capon and an approximate maximum likelihood MFP. Most methods were quite successful in estimating sensitive model parameters, but a number of interesting questions remain. These issues and possible future efforts will be discussed. [Work supported by ONR.]
Matched-field bottom property inversion and sensitivity analysis for the SWellEX series experiments
Authors: Baxley Paul A. , Booth Newell O.
Published: July 2002

The Shallow Water evaluation cell Experiment (SWellEX) series, conducted off California between 1993 and 1996, has demonstrated the feasibility of performing matched-field localization in shallow-water (∼200 m) environments. A critical issue in these analyses has been the accurate determination of bottom geoacoustic properties. A ‘‘best guess’’ for these parameters resulted in model predictions possessing an excessive number of propagating modes compared to that observed in SWellEX-1 (August 1993) data. It was hypothesized that the most significant parameters causing this discrepancy were the compressional sound speed and intrinsic attenuation. The values of these properties were then derived via inversions with a single-frequency (70 Hz) SWellEX-1 source-tow tonal at three stations along a radial track of nearly constant water depth. The use of these optimal parameters greatly enhanced matched-field correlations and localizations at all times along the track, as well as at other frequencies. A sensitivity study performed for an identical track and environment in SWellEX-3 (July–August 1994) reaffirmed that the geoacoustics properties obtained for SWellEX-1 are optimal, and suggested that simpler representations of the bottom may be permissible. A similar inversion approach was also performed for the high-slope SWellEX-4 (March 1995) environment, located off San Clemente Island. [Work supported by ONR/321US.]
Environmental inversion using the singular value decomposition modes of multiple frequency vertical line array data
Authors: Neilsen Tracianne B. , Westwood Evan K.
Published: July 2002

A method is presented for geoacoustic inversion that uses the depth-dependent normal modes extracted from vertical line array (VLA) data. The normal modes for multiple frequencies are obtained by forming a cross-spectral density matrix (CSDM) for each frequency of VLA data, measured as a source moves outward in range. The singular value decomposition (SVD) of each CSDM yields the depth-dependent normal modes of the ocean environment at the time the signal was recorded [T. B. Neilsen and E. K. Westwood, J. Acoust. Soc. Am. 101, 3025 (1997)]. The data-extracted mode functions are then used to invert for the environmental parameters. The Levenberg-Marquardt nonlinear optimization method is used to find the best fit between the data-extracted and modeled mode functions by adjusting the environmental parameters. The algorithm requires computation of the first partial derivatives of the mode functions with respect to the varied parameters, which is achieved using the ORCA normal mode model. The method is useful for determining the sound speed profile and the characteristics of the uppermost bottom layers. Results using simulated data will be presented. [Work supported by U. S. Army Research Office, AASERT Grant number DAAH04-95-1-0486, and ONR.]
Broadband geoacoustic inversion using mode traveltime dispersion
Authors: Potty Gopu R. , Miller James H.
Published: July 2002

Inversion of sediment compressional sound speeds using the data from SUS charge explosions acquired during the Shelf Break Primer Experiment in 1996 is presented. A hybrid approach combining a Genetic Algorithm and Levenberg-Marquardt nonlinear least squares method is used for the inversion. This inversion is based on the time-frequency dispersion of the acoustic signal calculated using a wavelet approach. Group speeds for modes 1–7 and for frequencies ranging from 10–200 Hz were obtained from the wavelet scalograms of the acoustic signal. The ocean waveguide depth was approximately 90 m. The minimized objective function is weighted to incorporate the quality of the data at various frequencies and modes. Forward modeling is done using adiabatic normal mode theory assuming a mildly range-dependent environment. Resolution and error of the inversions are expressed in terms of the Hessians and a posteriori model standard deviations. Inversion results are compared with compressional speeds based on the data from the nearby AMCOR-6012 site using Biot–Stoll theory. In addition, gravity cores taken at selected locations in the experimental site provide some corroboration of the sediment compressional speeds in the top 4 to 5 m of the sediment. [Work supported by ONR.]
Inversion of broadband ambient noise in shallow water for the geoacoustic parameters of the seabed
Authors: Buckingham Michael J. , Deane Grant B.
Published: July 2002

Ambient noise propagates through the ocean in much the same way as sound from a man-made source. The noise is a stochastic phenomenon, but some of its statistical measures, for instance, the spatial coherence, may be influenced by the environment in a systematic and stable manner. Such statistical measures may then be inverted to obtain information on the environment. In particular, over the continental shelf, where water depths are less than 200 m, the nature of the seabed has a significant effect on the vertical coherence, or, equivalently, vertical directionality, of the noise in the water column. This suggests that simple measurements of the vertical coherence of the noise, taken with just two hydrophones in the water column, may be inverted to yield the geoacoustic parameters of the bottom. An essential element of such an inversion procedure is an ambient noise model which includes a geoacoustic model of the sediment. In the presentation, a noise inversion model based on spatial coherence will be described. Examples of inversions for the bottom properties, obtained from real data taken at shallow-water sites in the northern and southern hemispheres, will be presented. [Research supported by ONR.]
Numerical simulation of the transverse current monitoring in the Fram Strait
Authors: Naugolnykh Konstantin A. , Wang Yun-Yu , Shang E. C.
Published: July 2002

The numerical simulation of transverse current monitoring in the Fram Strait environment is performed using horizontal refraction modal tomography (HRMT). The simplified West Spitsbergen Current profile is constructed in the framework of geostrophic approach and calculation of the modal phase difference between signals arriving along the two paths of acoustical interferometer due to horizontal refraction is made for the sound signal frequencies 100–200 Hz and different sound source depth. It was demonstrated that the measurable modal phase difference in a range of 80–160 can be expected for an averaged current velocity in the strait cross section of 0.3 m/s. At higher frequency of the sound signal the sensitivity of the HRMT increases but the space coherency decreases so the optimal frequency has to be chosen. The strong mode coupling effect presented in such a complex environment was apparently weakened by using the signals propagating along the closed paths.
Noise impact model
Authors: Burn Melissa M. , Bradley Kevin A.
Published: July 2002

A noise impact model (NIM) has been developed that allows the user to examine the impact that quieter aircraft technologies might have on air carrier operating efficiency at major U.S. airports. The analyst selects one of 16 available airports and one of three case years for study, chooses a set of flight tracks, and has the option of reducing the noise of one or more aircraft. Two sets of flight tracks are available for most airports; one that represents actual current conditions, including noise abatement tracks, and a second set that offers more efficient routing. NIM computes the resultant noise impact and the time and distance saved for each operation on the more efficient, alternate tracks. In addition, NIM offers the option of examining the effect that runway use patterns have on capacity and delay for three airports. Noise impact is characterized in three ways; the size of the noise contour footprint, the number of people living within the contours, and the number of homes located in the contours. A World Wide Web implementation of NIM will be available this year through the Aviation System Analysis Capability at www.asac.lmi.org.
Development of a ‘‘quiet by design’’ approach to reduced-noise gas flow systems for experimental research facilities at NASA Lewis Research Center
Authors: Cooper Beth A. , Nelson David A.
Published: July 2002

At the NASA Lewis Research Center, occupational and community noise exposures associated with experimental research in aircraft and space propulsion are managed via a hearing conservation program that prohibits unprotected exposure to A-weighted noise levels above 85 dB. In support of hearing conservation goals, a ‘‘buy quiet’’ program requires project designers and engineers who purchase equipment expected to generate A-weighted noise emission levels above 80 dB to consider noise emissions along with other performance criteria. A Guide to Specifying Equipment Noise Emission Levels provides the means to effecitvely pursue this approach. A recently implemented ‘‘quiet by design’’ initiative complements this program and provides design guidance for new facilities, systems, and equipment designed in-house rather than purchased from an outside vendor. A Reduced-Noise Gas Flow Systems Design Guide serves as a companion to the earlier document and covers design practices for reducing noise emission and methods of estimating noise emission from piping and atmospheric vents associated with gas flow systems. A computer spreadsheet guides the user through the design process from noise emission specification through predictive equations and selection of noise control elements, greatly facilitating the development of noise control strategies for new experimental research facilities and the associated servives required for their operation.
Acoustic measurements in an airport environment and the identification of the sound radiated by aircraft wake vortices
Authors: Fine Neal , Rees Frank , Von Winkle William
Published: July 2002

Results are presented of an effort to measure the sound generated by aircraft wake vortices. The objective of the research is to estimate the spectral shape and source levels characteristic of the sound generated by aircraft wakes during approach and landing. Measurements were made at JFK airport in August 1997, for various aircraft, using three Bruel & Kjaer outdoor microphones. The data was recorded to tape and spectrum analysis was performed using a B&K portable spectrum analyzer. The three microphones were placed near the beginning of runways 13L and 22L, arrayed in several configurations parallel to and perpendicular to the runway centerline. Initial spectrum analysis of the data revealed a prominent broadband ‘‘hump’’ in the one-third octave sound-pressure level spectra, centered at approximately 50–60 Hz for a B-747 landing (with a bandwidth of roughly 20–30 Hz). The timing of the hump is consistent with that expected for wake vortex sound and it is hypothesized to be generated by the wake vortices. Cross correlations between microphones arrayed parallel to the runway will test this hypothesis. Additional data, which were recorded in January 1998, will also aid in identifying the wake vortex sound characteristics.
Developments in aircraft interior acoustics—Recent progress and directions
Authors: Silcox Richard J.
Published: July 2002

Aircraft interior noise is a continuing problem for a wide range of aircraft types. High interior noise levels in general aviation aircraft make any communication difficult, significantly impact pilot endurance, and are considered important to the reemergence of this class of aircraft. For passengers in propeller-powered commuter aircraft, conversation is difficult at best and becomes oppressive on the passengers for long periods. It is also regarded as one reason for the replacement of propeller commuter aircraft by the emerging regional jet. In business jets, the passengers are often interested in conducting meetings during flights. This calls for low SPL and SIL for which the aircraft incurs both space and weight penalties. Large commercial aircraft pay a significant weight penalty for current interior noise levels considered acceptable. With the drive for more weight-efficient airframes, the challenge becomes even more difficult. This paper will overview current efforts to develop both design and noise control techniques. These range from improvements in the quantification of the noise sources, to considerations in the overall structural design, and to the use of both active and passive control tools. In addition, the emerging use of predictive and measurement tools will be discussed and applications to aircraft highlighted.
Turbulent boundary layer noise transmission into an aircraft fuselage—Modeling and analysis
Authors: Mathur Gopal P. , Chin Cliff L.
Published: July 2002

The turbulent boundary layer (TBL) pressure fluctuations on the exterior of an aircraft excite the fuselage structure which in turn radiates noise inside the passenger cabin. In most of the previous analytical and experimental studies of sound transmission loss of structures, only acoustic excitation is considered. It is then assumed that the transmission loss of a structure due to the TBL pressure field will be the same as that for the acoustic field. The main objective of this paper is to examine the effects of the TBL pressure field on the transmission loss of a fuselage structure and compare it with the acoustic transmission loss. The vibration and acoustic response of a circular cylindrical shell due to the TBL excitation is first modeled with a finite-element-based computational approach. The finite-element model of the cylindrical shell panel is validated with the results from classical analytical model. The vibration and acoustic response of an aircraft fuselage due to the TBL pressure field is then calculated using the finite-element model. The incident TBL pressure field is simulated by a cross-spectral formulation of the excitation field. The noise reduction characteristics of the fuselage due to acoustic and TBL pressure fields are compared.
Phonotactics, phonological neighborhoods, and working memory in spoken word recognition
Authors: Vitevich M. S. , Luce Paul A.
Published: July 2002

Spoken word recognition has often been described as a competitive process among activated words in memory. Work in this area has demonstrated that words with many similar sounding words (i.e., dense neighborhoods) are recognized more slowly and less accurately than words with few similar sounding words (i.e., sparse neighborhoods). Evidence will be presented from several experiments that suggests that activation at a sublexical level may also play a prominent role in the recognition process. In particular, it is demonstrated that lexical and sublexical activation in spoken word recognition has demonstrably different effects on processing. In particular, it is demonstrated that, when processed at a sublexical level, nonwords composed of phonological segments with high-probability phonotactic patterns are responded to more quickly and accurately than nonwords composed of phonological segments with low-probability phonotactic patterns. However, when processed primarily at the lexical level, nonwords behave much like real words. That is, nonwords having high probability phonotactic sequences and dense neighborhoods are processed more slowly and less accurately than those with low probability phonotactic sequences and sparse neighborhoods. It is also proposed that the differential effects of these two levels of processing may be mediated by short-term verbal memory.
Angle-frequency spectra for broadband pulses
Authors: Chambers David , Lewis D. Kent
Published: July 2002

Broadband pulses representing localized waves are emitted from an array. A receiver is scanned in angle at a constant distance from the array. The power spectra of the data are displayed as an image with frequency along one dimension and angle along the other. The resulting angle–frequency images are compared with simulations of the array performance. The images show at a glance the changing frequency content of the signal as the angle changes. This method of displaying broadband data greatly assists interpretation of the total beam pattern.
Time-frequency analysis of low diffraction beam pulses
Authors: Lewis D. Kent , Chambers David H. , Ziolkowski Richard W. , Henderson Terry L. , Krueger Ken
Published: July 2002

Recent work in low diffraction beams has given our group some new sets of wave fields to investigate. Theoretical predictions matched experimental measurements closely, and several comparison techniques were included in the analysis. The field experiments have now been conducted at the NUWC Keyport facility and, most recently, at the University of Texas’ Applied Research Laboratory facility in Lake Travis. In cooperation with ARL, a series of low diffraction beam pulses was launched to a wide band width receiver 600 feet distant. Among the comparison methods used was a time-frequency analysis of the received pulse. For each angular position the frequency arrival times via time-frequency analysis are investigated. It is hoped this analysis will shed light on this very interesting phenomena.
Sonar detection and classification of buried targets using broadband dolphin-like signals
Authors: Au Whitlow W. L. , Shizumura Reid H. , Nachtigall Paul E. , Moons Jerry , Roitblat Herbert L. , Hicks Robert C.
Published: July 2002

Dolphins have been observed foraging for prey that bury themselves into a sandy bottom. The dolphins swim about 1 to 2 m above the bottom and scan in a circular motion or swim immediately off the bottom, scanning from side to side with their beams pointed approximately normal to the bottom. The dolphin has been emulated by using dolphinlike sonar signals (120-kHz peak frequency, 39-kHz bandwidth, 50-μs duration) in order to classify proud and buried targets in real time. The transducer was attached to a bottom-crawling remotely operated vehicle. Target echoes were received via a cable and digitized at 1 MHz. Short-time Fourier transform and the Morlet wavelet were used to obtain time-frequency representations of the echoes. Echoes were processed in a hierarchical neural network system to perform target classifications. Six targets (cast iron pot, stainless steel sphere, glass jar, concrete tile, and coral rock) were placed either on the ocean bottom or buried into the bottom. Echoes were separated into three target categories: (1) cast iron pot; (2) stainless steel sphere; and (3) remaining four objects. The neural network was able to correctly identify 74%, 97%, and 88% of the category 1, 2, and 3 test echoes.
A geometrical fuzzy clustering-based solution to glottal wave estimation
Authors: Shapira Yair , Gath Isak
Published: July 2002

Accurate estimation of the glottal waveform (GW) is required for purposes such as natural speech synthesis, speaker recognition, physiological speech processing, etc. Most methods available for GW estimation are based on inverse filtering of the speech signal through the vocal tract, and they all suffer from inaccuracies due to incorrect assumptions. The method for GW estimation developed in the present study is based on fuzzy clustering of quasi-linear geometrical substructures, represented within the signal shifts hyperspace. Algorithms for estimation of the driving function to the vocal tract are presented and evaluated on simulated and real data. Comparison of the fuzzy clustering-based method with the PSIAIF and Wong’s closed-phase algorithms shows that the present method is superior with respect to both the GW estimation and determination of GW event time instants.
The perfect fifths tempered scale, a proposal or already a practice?
Authors: Gillessen Klaus
Published: July 2002

An alternative tempered scale is proposed, the perfect fifths tempered scale (PFTS), which is stretched by an amount of 3.4 cents per octave in comparison to the common (perfect octaves) tempered scale (POTS). The PFTS is a better approximation to the musical practice of stretched intonation than the POTS.
An acoustical frequency ratio technique for detection of a bifurcation hemiocclusion in a symmetrical branching structure
Authors: Raphael David T. , Dean Marshall H.
Published: July 2002

Within a symmetrical branching cylindrical structure open at one end and closed at the other, the ratio of the first overtone to the fundamental is either greater or less than 3, depending on the ratio of the distal branch length to the parent trunk length. For an unoccluded structure, the ratio is generally above 3. For an occluded structure, the ratio is generally below 3. This acoustical frequency ratio technique may be used to infer the extent of unilateral distal branch occlusion in a branching structure.
Second-order perturbations of peak-arrival times due to sound-speed variations
Authors: Skarsoulis E. K. , Athanassoulis G. A.
Published: July 2002

The concept of peak arrivals is used to study the second-order arrival-time behavior of broadband tomographic signals due to sound-speed variations. Peak arrivals, defined as the local maxima of the pressure amplitude at the receiver in the time domain, are the exact theoretical counterparts of the experimental observables in ocean acoustic tomography, and they offer a uniform description of all the peaks in the arrival pattern, independent of whether they can be resolved as ray/modal arrivals or not. Closed-form expressions for the first and second sound-speed derivatives of peak-arrival times are derived in terms of sound-speed derivatives of the Green’s function. Using normal-mode theory the sound-speed derivatives of the Green’s function are analytically expressed for the case of a range-independent medium in terms of eigenvalues and eigenfunctions at the reference state. Numerical examples are given for range-independent environments motivated from the Thetis-2 experiment, showing different kinds of nonlinear behavior of the arrival times in the large and the fine scale and demonstrating the performance of the second-order approach.
Ultrasonic velocity and attenuation in models of dry and wet sands with an embedded object
Authors: Tavossi Hasson M. , Tittmann Bernhard R.
Published: July 2002

Propagation of ultrasound in wet and dry sand, up to 6 in. in depth, is investigated. Velocity and attenuation, as a function of mean grain size, water content, and depth, are measured. The results can be used in the detection of the objects embedded in dry and wet sand. Experiments show that for a dry sand, at frequencies of the order of 500 kHz and above, no through-transmitted signal is detectable. This is due to a very strong attenuation, even for a sample having only a depth of 2 in. However, in a wet sand sample, attenuation is significantly smaller, therefore strong signal is detected at a depth of 6 in. Using low-frequency transducers it is possible to transmit and receive longitudinal ultrasonic pulses in dry sand model medium, for depths of up to several inches. Some application of these results to the detection of objects embedded at a shallow depth of the order of several inches will be discussed.
Pianolike tones of multiple strings having stiffness
Authors: Strong William J.
Published: July 2002

The strings on a piano have stiffness which gives rise to inharmonic partials that are stretched in frequency relative to harmonic values. In addition, most notes on a piano are strung with more than one string. Even though the multiple strings of a note are tuned to the same nominal frequency, they are never precisely in tune. When the multiple strings are sounded together the resulting tone has partials that increase and decrease in amplitude which results from their in-plane and out-of-phase action on the bridge. In order to illustrate these phenomena, pianolike tones for the note C4 were synthesized in which the parameters of 15 partials were specified for each of three strings. Each of the 45 partials was represented with a decaying sinusoid whose frequency, initial amplitude, and decay rate were specified. The amount of inharmonicity and out-of-tuneness of the strings were varied over a considerable range in order to make their effects apparent. Audio samples were recorded for various combinations of partial parameters.
Theoretical consideration on the musical scale of Xoomij singing based on the speech production process
Authors: Kakita Yuki
Published: July 2002

This paper presents a theoretical consideration on what kind of musical scale can be produced by Xoomij, a type of singing performed in Mongolia and also in some other parts of the Eurasian continent. In Xoomij singing, a tune is produced by selecting appropriate voice harmonics one after another while keeping the fundamental voice pitch constant. The harmonics are selected by changing the shape of the oral cavity which functions as a band-pass filter. Calculations were performed to see what musical scales would be realized when the scale started with one of the 1st to the 30th harmonics of voice, allowing the error of plus/minus a quarter semitone. Some of the findings are: (1) the structure of the scale varies depending on the harmonic that the scale started with, (2) starting the scale with the 5th harmonic produces the widest variety of tunes and the smallest number of misfitted tones, and (3) a relatively high voice pitch is appropriate for producing an efficient scale, when the cavity resonance suitable for playing music and also the spectral decay of voice source toward the high frequency are considered.
An investigation of different timbral attributes as markers determining metrical structure
Authors: Singh Punita G.
Published: July 2002

Timbre attributes such as spectral locus, spectral density, and temporal envelope slope were studied to determine their relative effectiveness as accent markers defining metrical structure. Complex-tone sequences comprising 12 tones were used as stimuli with points of timbre change at positions implying a triple meter, quadruple meter, compound meter (both triple and quadruple), or unaccented sequences with no timbre changes. Tones at accented positions were made to differ from the rest by using different spectral densities (1, 2, 4, or 8 harmonic components), by changing the locus of components, and by changing the slopes of a two-part temporal amplitude envelope. Listeners were asked to report if the sequence had a triple meter, quadruple meter, ambiguous meter, or no meter. Results thus far indicate that each of the timbre markers used is individually capable of bestowing metrical structure on a sequence. However, when competing cues exist to offer alternative interpretations of meter, more dense spectra appear to dominate over less dense, higher centers of gravity over lower, and sharper attack functions over more gradual rise times.
Human auditory response to an air bag inflation noise: Has it been 30 years?
Authors: McKinley Richard L. , Nixon Charles W.
Published: July 2002

Once upon a time, over 30 years ago, 100 subjects volunteered for exposure to high level impulsive noises generated by the inflation of prototype automobile safety air bags. The government mandated that passive restraint systems, beyond seat belts, be introduced in passenger automobiles within 2 years. The only reasonable candidate was the air bag system, however, none were fully developed for automobile production. One concern was the effect of inflation noise on the auditory systems of vehicle occupants. Ninety-one volunteers qualified for the study; nine were rejected by an otolaryngologist because of head colds, congestion, and upper respiratory infections. Subjects facing forward in the rear seat of a small automobile experienced the noise of simultaneous inflation of a 9-cf passenger air bag and a 2.5-cf driver air bag. The median peak pressure level of the impulsive noises was 168 dB obtained with an overall measurement system ranging from 30 Hz (microphone low frequency cutoff) to 40 000 Hz. The dynamic range was 180 dB. Post-exposure auditory system evaluations included external auditory canals, eardrum membranes, verbal communications-auditory perception, aural discomfort, TTS, and PTS. Results are discussed in terms of the conditions of the exposures, the measurements, and the study population. [Work supported by DOT, National Highway Safety Bureau.]
Continuous-flow thermoacoustic dehumidifier
Authors: Hiller Robert A. , Swift G. W.
Published: July 2002

For applications in gas cooling a thermoacoustic refrigerator may be simplified by using the gas stream itself as the acoustic medium and eliminating the cold heat exchanger by directing the input gas stream through the stack (R. S. Reid et al., ‘‘Cyclic Thermodynamics with Open Flow,’’ Phys. Rev. Lett. to appear May 25, 1998). This concept may be extended to dehumidification, with the incoming humid air stream being cooled to its dew point and condensing water in the thermoacoustic stack. The water will appear as liquid on the stack surface or as a fog and is removed from the resonator, while the dry air flows out of the system. Design problems include collecting and removing the condensate from the stack to the outside of the resonator, and optimizing stack design for a working gas in multiple phases. Experimental results and numerical models for a prototype compressed-air dehumidifier will be presented. [Work supported by DOE Office of Basic Energy Sciences.]
A joint time-frequency model of auditory masking
Authors: O’Donovan Jonathan J. , Furlong Dermot J.
Published: July 2002

The perceptual phenomena of temporal and spectral masking (critical band masking) [B. C. J. Moore, Frequency Selectivity in Hearing (Academic, New York, 1986)] are related to the ear’s limited temporal and spectral resolution. This resolution is known from psychoacoustic experiments but is not typically incorporated in the time-frequency representations used in audio signal processing. A joint time-frequency distribution, modeling both critical band masking and nonsimultaneous (forward and backward) masking, is described here. Applications for such a distribution include audio compression and the modeling of room acoustics. The distribution is developed within the framework of Cohen’s class of generalized time-frequency distributions [L. Cohen, ‘‘Time-frequency distributions—A review,’’ Proc. IEEE 77, 941–981 (1989)] and is a development from the constant Q modal distribution [W. J. Pielemier and G. H. Wakefield, ‘‘A high resolution time-frequency representation for musical instrument analysis,’’ J. Acoust. Soc. Am. 98, 2382–2396 (1996)]. The design and implementation of this auditory model is detailed.
Inverse identification, optimization, and active structural acoustic control of aircraft interior noise using piezoelectric actuators
Authors: Fuller Chris R. , Paxton Scott D. , Gibbs Gary P.
Published: July 2002

This paper discusses an inverse method for identifying the dominant aircraft fuselage normal vibration in terms of interior noise. The identified vibration fields are used in an optimal approach based on a genetic algorithm to locate arrays of piezoelectric actuators. The actuators are used in conjunction with microphone error sensors and an LMS controller to implement an ASAC approach to efficiently reduce the interior noise. The methodology is demonstrated on a business jet fuselage under a variety of excitation types and frequencies. The results demonstrate that the inverse identification method and transducer optimization approach leads to improved control performance in terms of the amount and spatial extent of the noise attenuation. [Work supported by NASA LaRC.]
Characteristics of the Structural Acoustic Loads and Transmission (SALT) facility at NASA Langley Research Center
Authors: Grosveld Ferdinand W.
Published: July 2002

The Structural Acoustic Loads and Transmission (SALT) facility at NASA Langley Research Center consists of a reverberant chamber, an anechoic chamber, and a common window. The 267-m3 reverberant chamber is structurally isolated from the rest of the building and is the source room in a transmission loss suite configuration. The receiving anechoic chamber is 4.57 m high, 9.63 m long, and 7.65 m wide, measured wedge tip to wedge tip. The window between the two rooms can accommodate 141×141-cm2 test structures for which sound transmission loss may be measured. Calibration of the transmission loss suite is discussed. The acoustic and structural flanking paths were evaluated by acoustic intensity and cross-spectral accelerometer measurements. Acoustic measurements in the reverberant room are accommodated by covering the window with a stiff, heavy insert. Measured ambient levels, reverberation times, sound field diffusivity, lowest usable frequency, and preferred test object dimensions are discussed. A transportable partition with foam wedges can be placed in front of the covered window in the anechoic chamber to facilitate acoustic free-field measurements. Anechoic room characteristics are presented including acoustic ambient levels, inverse square-law curves, cutoff frequency, and test object versus room size dimensions.
Context and identification of vowels in speech variation
Authors: Williams Sheila M. , Diehl Randy L.
Published: July 2002

Interpretation of a particular vowel pattern may vary according to the perceived speech patterns of the speaker [P. Ladefoged and D. E. Broadbent, J. Acoust. Soc. Am. 29, 98–104 (1957)]. Formants from a given speaker, using a specific speech style, reflect both anatomical dimensions of the speakers vocal tract and the speakers preferred vocal settings for the context of the utterance. Speech samples of a set of nine vowels in a standard CVC frame [G. E. Peterson and H. L. Barney, J. Acoust. Soc. Am. 24, 175–184 (1952)], recorded from local speakers, were analyzed and resynthesized, manipulating formant values to derive similar formant frequency ratios from different vowel bases. Thus if the listener identifies the vowels according to the formant patterns, the perceived vowel will differ from the base vowel from which it was derived, whereas if the overall formant frequency shift is detected, then the vowel should maintain its identity, differences in quality being attributed to a change of speaker or of speech style. Using listeners from the same speech community, monosyllabic word identification is contrasted with identification of excised vowels to ascertain the effect of word context on distinguishing the relatively long-term formant ratios of the speaker. [Work supported by NIDCD.]
Learning changes in the recognition, repetition, and free recall of pseudowords
Authors: Barac-Cikoja Dragana , Goldberg Lyn , Siren Kathy , Ludlow Christy
Published: July 2002

Acquisition of a pair of three-syllabic (11-phoneme) pseudowords associated with meaningless ideographs was examined across 15 trials on three tasks. For each word, an ideograph-prompted recall was succeeded by the identification of each constituent syllable among two foils, ending with two repetitions of the recorded pseudoword. The words were alternated throughout the experiment. Subjects verbal productions were phonetically transcribed and scored for segmental and syllabic accuracy. The scores were used to compare changes in performance on: (a) the recall and the repetition tasks, (b) the syllable recognition and the repetition tasks, and (c) between two repetitions. Phonological errors in the three tasks were highly related with intra-syllabic differentiation increasing with learning. [Work supported by NIH ZO1 DC 00005-09.]
Sound source localization in multimedia applications
Authors: Settel Zack
Published: July 2002

With the arrival of real-time 3-D digital audio processing, techniques for acoustic space modeling and sound source spatialization have become of great interest for implementation in various multimedia applications, ranging from CD-ROM authoring to video conferencing. In surround sound applications, where sound source localization is a primary objective, various digital signal processing (DSP) techniques can be employed to heighten the sense of source localization, often creating a stronger link between sound and image. In several multimedia applications, the degree of live interactivity can play a crucial role in the success of the application. For example, the underlying software in video games, acoustic space simulators, and air traffic monitoring systems (using auditory display) must render audio objects in a real-time acoustically modeled space, based on varying position data supplied to these programs by the user(s). This paper will focus on sound source localization techniques and their use in various multimedia applications, and will examine key issues in implementation.
Reduction in computational requirements with matched field processing
Authors: Wazenski Michael
Published: July 2002

The sensitivity of matched field techniques to source location is a significant benefit from a localization accuracy perspective. Unfortunately, the implementation of matched field techniques in real-time systems is hampered by the computationally demanding hypothesis testing of source location in range and depth. Furthermore, operational requirements of such systems generally do not demand a high degree of accuracy. The Minimum Variance with Environmental Perturbation Constraints (MV-EPC) processor developed by Krolik [J. Acoust. Soc. Am. 92, 1408–1419 (1992)] has been used to reduce the sensitivity to environmental uncertainty. This same approach can be used to reduce the sensitivity to source location, thereby reducing the number of source location hypotheses. This produces a computational tractable approach to real-time, robust matched field processing. Illustrative simulations and real data results using SWellEX-96 data are presented.
Nonanalytic nonlinear oscillations: Christiaan Huygens, quadratic Schrödinger equations, and solitary waves
Authors: Denardo Bruce
Published: July 2002

An example of an oscillatory system with a time-reversible nonanalytic nonlinearity is shown to be a pendulum with a flexible cord sandwiched between two identical circular disks, in contrast to the analytic case of a pendulum interrupted by a single circular disk. The amplitude-dependent frequencies of both cases are perturbatively calculated, and are compared to numerical simulations over the entire range of amplitudes. The nonanalyticity causes the unusual effect of the frequency to vary linearly with amplitude for small amplitudes, which has also been observed in the resonant frequencies of compressional standing waves in sandstone. A general condition for a nonanalytic nonlinearity to yield this behavior is presented. The amplitude-dependent frequency for the double-interrupted pendulum allows an explanation for Huygens’surprising observation that circular interrupters were as effective as cycloidal interrupters in achieving isochronous motion. A lattice of linearly coupled double-interrupted pendulums is described near the lower and upper cutoff modes by quadratic nonlinear Schrödinger (NLS) equations, in contrast to cubic NLS equations which arise for analytic pendulum lattices as well as typical acoustic and surface waveguides. Solitary breather and kink solutions to the quadratic NLS equations are presented, and are compared to the known soliton solutions of the corresponding cubic NLS equations. Compressional waves in sandstone are shown to be modeled by the inclusion of a nonanalytic quadratic nonlinearity in the stress–strain relationship. Quadratic NLS breathers are predicted to occur in a waveguide of sandstone, and an analysis indicates that such an observation is feasible.