bluuebunny/crossref_metadata_2025 huggingface dataset:
Prediction and attenuation of noise resulting from construction activities in major cities: The case for modernizing construction noise codes
Authors: Raichel Daniel R. , Dallal Michael
Published: July 2002

Noise arising from operation of construction machinery in major cities is generally most annoying at street and below-ground levels, where the loudest equipment is used. Both residential and business neighborhoods are adversely affected by the presence of excessive noise, particularly if construction activity occurs during early hours. In some cases, vehicular traffic patterns become altered with a resultant change in the traffic noise levels. A survey was conducted as to which type of machinery is most commonly used in New York City, the amount of noise pollution generated by specific equipment, and commercially available means of attenuating the noise, preferably through retrofitting of existing equipment. On-site sound-level measurements were taken and the results compared with published data. Methods of predicting noise levels through the use of computer models, based on the effect of surrounding topology, were also evaluated. Noise codes in several major cities in the United States and abroad were compared with respect to their compliance with U.S. federal regulations. This was done for the purpose of developing a proposal for a new set of construction noise regulations for the City of New York. [Work supported by the New York City Department of Design and Construction.]
Propagation of noise from petrochemical activities in the Taranaki region of New Zealand
Authors: Prasad Daryl
Published: July 2002

Noise is a significant environmental concern facing the petrochemical industry in New Zealand. Typical inland oil and gas exploration involves operations that can run continuously for 70 days. Exposure to the resulting steady and impulsive noise produces undue stress and loss of amenity to neighboring residents. This paper reports on an investigation into the propagation of noise in the Taranaki region of New Zealand. Of the many factors that influence the propagation from drilling, atmospheric conditions (especially wind, temperature, and humidity) create the most variability. This study aims to provide a better understanding into the way noise is propagated in this region. Results of wind and temperature profile measurements will be presented as well as the related frequency and seasonality of events such as temperature inversions. Actual noise propagation data combined with the meteorological data will be used to test a model developed for the prediction of noise levels based on the CONCAWE and ISO 9613-2 methods of calculation.
An acoustic impedance measurement of a parallel-pore stack used to determine the thermoviscous functions of the stack
Authors: Simmons Timothy G. , Raspet Richard , Brewer Jeremy
Published: July 2002

Recently, thermoviscous functions have been measured for single pores of various cross-sectional geometry [L. A. Wilen, J. Acoust. Soc. Am. 103, 1406–1412 (1998)]. This investigation consists of measuring the acoustic impedance through a stack with parallel pores, inverting to yield the thermoviscous functions F(λ) and F(λT), and determining the accuracy with which the inversion can be accomplished. The measurement shall be performed using a standard impedance tube with the typical measurements taken, i.e., pressure maximum, pressure minimum, and the positions of two pressure minima. In order to be able to solve for the thermoviscous functions, a second measurement must be taken with the stack positioned away from the rigid termination, leaving a region of air. The impedance translation theorem (modified to account for the stack porosity) will have a functional form containing both F(λ) and F(λT) and shall be employed for both stack placements to provide two equations which can be solved for the two unknowns, F(λ) and F(λT), for a given frequency. If the method proves successful, modifications shall be made to the experimental setup so that it can accommodate a temperature gradient imposed across the stack. [Work supported by ONR.]
Characterization of the nuisance heat load in a thermoacoustic refrigeration demonstration device
Authors: Wakeland Ray Scott
Published: July 2002

The heat load in a thermoacoustic refrigeration demonstration device is studied. In the demo, which acts only to create a low temperature at the position of a thermocouple, the only heat load is nuisance heat, which is not well known. The method described here for estimating experimentally the size of this heat load assumes that heat transport associated with acoustics depends on acoustic amplitude p and on the magnitude of the deviation from ambient temperature |ΔT|, but is independent of the sign of ΔT. That is, the heat load due to acoustics-induced transport is taken to have the form Q̇transport=-F(p)ΔT, where F(p) is some function of p only, and ΔT can be either positive (refrigerator) or negative (heater). Direct thermoviscous heating at the site of the thermocouple Q̇tv(p) is also a function of p only. At steady state, Q̇heater=F(p)ΔT-Q̇tv(p). In the experiment, the refrigeration stack is replaced by an electric heater, the power of which is easily measured. The steady−state temperature rise above ambient ΔT is measured as a function of heater power and acoustic amplitude. The experiment generates a family of straight lines, the slopes of which determine the function F(p), and the intercepts give Q̇tv. [Work supported by ONR and NSF.]
Separation of gas mixtures in the acoustic boundary layer
Authors: Spoor Philip S. , Swift Gregory W.
Published: July 2002

Spatial separation of binary gas mixtures in the presence of a sound wave has been reported by various experimenters in the last half-century, and is usually attributed to static pressure or temperature gradients. The authors suggest an additional mechanism, a ‘‘concentration pumping’’ in the acoustic boundary layer analagous to thermoacoustic heat pumping. Data from a simple acoustic mixture-separation experiment will be compared with predictions of this new theory. Observation of mixture separation between two acoustically coupled thermoacoustic engines will also be discussed.
Finding the missing fundamental: A connectionist model of harmonic detection
Authors: Lewis Clifford F. , Fountain Stephan B. , McBeath Michael K.
Published: July 2002

A major shortcoming of the place-coding principle of pitch perception has been that it does not account for the phenomenon of virtual pitch, or the missing fundamental, because there is no physical energy on the basilar membrane at the location corresponding to the fundamental frequency. A connectionist model provides an example of a simple mechanism for virtual pitch that is consistent with place-coded input. An autoassociator network was trained using place-coded amplitudes of harmonics that corresponded to tones produced by a variety of musical instruments. After training, the network exhibited the missing fundamental illusion. Specifically, activation was present in the output unit corresponding to the fundamental frequency, even when the input energy at the fundamental was removed. This activation was approximately an order of magnitude larger than the activation of other units receiving no input energy, and it was the same order of magnitude as the activation when the fundamental frequency was present. The model demonstrates how virtual pitch can result from learned associations between harmonics that typically occur simultaneously in nature. When one of the harmonics is missing, the network simply fills in the missing part of the pattern.
Subjective evaluation of audio compression algorithms: Do conservatory students have a ‘‘golden ear?’’
Authors: Carrier Sam , Quackenbush Schuyler
Published: July 2002

Digital audio compression algorithms are typically based upon psychoacoustic models of hearing; thus they are known as perceptual coders. For commercial applications such as providing music over the Internet, where bit-rate reduction is essential, these lossey codecs have customarily been evaluated through subjective testing using professional audio engineers as listeners. This study replicated the MPEG-2 AAC Stereo Verification Tests (ISO/IEC JTC1/SC29/WG11 N2006) with the 20 listeners drawn from undergraduates enrolled in the Oberlin Conservatory of Music. These tests used the triple-stimulus hidden-reference double-blind methodology with eight codecs and 10 test stimuli. In this procedure, listeners compare two test signals (one coded, the other not) to an uncoded reference, then rate the ‘‘audio quality’’ of both test stimuli on a 1–5 scale in 0.1-step increments. The test stimulus selected as the hidden reference is graded 5.0, the coded signal graded on the degree of impairment. Difference scores thus reflect the discriminability of the stimuli. Taking the magnitude of Studentized t-statistic values as a measure of acuity, MPEG listeners averaged 4.45, while Oberlin listeners averaged 9.86. While there were some methodological differences between the two experiments, subjects differed several ways Oberlin listeners were younger and were engaged in formal musical training.
A computational acoustic field reconstruction process based on an indirect variational boundary element formulation
Authors: Zhang Zhidong , Vlahopoulos Nickolas , Raveendra S. T. , Allen T. , Zhang K. Y.
Published: July 2002

The objective of this work is to develop a computational capability based on the indirect variational boundary element method (IVBEM) to evaluate appropriate velocity boundary conditions on an assembly of piston-type sources such that they can recreate a prescribed acoustic field. Information for the acoustic pressure of the original acoustic field at certain field points constitutes the input to the developed process. Several new developments associated with the IVBEM are necessary for completing the field reconstruction process. The developed computational capability targets exterior radiation applications. Thus a new formulation for treating irregular frequencies in the IVBEM is developed and implemented in the field reconstruction process. The computation of the appropriate velocities for the piston-type sources is based on deriving transfer functions between the field points, where the acoustic pressure of the original field is prescribed, and the velocity assigned on each element of the generic source. A singular value decomposition solver is integrated with the IVBEM computations in order to evaluate the velocity boundary conditions from the transfer functions. Finally, an algorithm that identifies the optimum field points from a set of candidate points for prescribing the acoustic pressure of the original field is developed. [Research supported by a University Research Program award from Ford Motor Co.]
Love wave surface acoustic wave sensor for ice detection on aircraft
Authors: Varadan Vasundara V. , Gangadharan Sunil , Varadan Vijay K.
Published: July 2002

This paper presents the design, fabrication, experimental results, and theoretical validation of a Love wave surface acoustic wave sensor for detecting the phase change from liquid water to solid ice. The sensing of this phase transition is due to the shear horizontal nature of Love waves which couple to a solid (ice) but not to a liquid (water). An SiO2 film of thickness 3.2 μm deposited on an ST cut quartz wafer via plasma-enhanced chemical-vapor deposition acts as the guiding layer for Love waves. Testing is carried out with the water or ice placed directly in the propagation path of Love waves. An oscillation frequency shift of 2 MHz is observed when water on the sensor is frozen and melted cyclically. The contribution to the frequency shift is explained in terms of the acousto-electric effect (high permittivity and conductivity of water relative to ice), mass loading, and elastic film formation (solid ice). An arrangement for wireless interrogation of the sensor is proposed which is particularly attractive for aircraft and rotorcraft applications obviating the need for complex wiring and local power sources. t
Bifurcation and chaos in flexural vibration of a baffled plate in mean flow
Authors: Wu Sean F. , Zhu Jason
Published: July 2002

This paper presents the results of an investigation of flexural vibration of a finite plate clamped to an infinite baffle in mean flow. The plate flexural displacements are solved by the Galerkin method. The critical mean flow speed is determined by a general stability theory and the Routh algorithm. Results show that the instabilities of a baffled plate are mainly caused by an added stiffness due to acoustic radiation in mean flow, but controlled by structural nonlinearities. The added stiffness is shown to be negative and increase quadratically with the mean flow speed. Without the inclusion of structural nonlinearities, the plate has only one equilibrium, namely, its undeformed flat position. Under this condition, the amplitude of plate flexural vibration grows exponentially in time everywhere, known as absolute instability. With structural nonlinearities, the plate may possess multiple equilibria. When the mean flow speed exceeds the critical value, the plate becomes unstable around all equilibria, jumping from one equilibrium position to another. Since this jumping is random, the plate flexural vibration may seem chaotic. This chaotic behavior disappears when viscous damping is introduced. Accordingly, the plate settles down to a ‘‘buckled-down’’ position owing to the hydraulic fluid-loading effect. [Work supported by ONR.]
A real-time audio–video front-end for multimedia applications
Authors: Zotkin Dmitry , Duraiswami Ramani , Hariatoglu Ismail , Davis Larry , Otsuka Takahiro
Published: July 2002

A real-time system combining auditory input received by a microphone array with video input is designed and implemented as a prototype smart video-conferencing system. The audio part of the system uses accurate algorithms for localization of the source in typical noisy office environments, and performs dereverberation and beamforming to enhance signal quality. The coordinates of the sound sources obtained are used to control the pan, tilt and zoom of a video camera. The video-processing part segments people in the image using an obtained background model, and can label and track them and get a close-up view of the head of the talker. Heuristic algorithms for intelligent zoom and tracking of people in video-conferencing scenarios are incorporated into the system. The current implementation is performed on a commercial off-the-shelf personal computer equipped with a data acquisition board and does not require expensive hardware. The speed, precision and robustness of the system are superior to existing ones. Demonstrations of the system in different scenarios will be presented, along with quantitative measures of its performance. The system is designed as a general purpose front-end to systems for speech and person recognition, and for newer human-computer interfaces. [Work supported in part by DARPA.]
Vowel intelligibility in clear and conversational speech
Authors: Ferguson Sarah Hargus , Kewley-Port Diane
Published: July 2002

Several studies have demonstrated that instructions to speak clearly yield significant improvements in speech intelligibility along with a wide variety of acoustic changes relative to conversational speech. The current study explored the relationship between acoustic properties of vowels and their identification in clear and conversational speech, as well as the effects of hearing loss on this relationship. The goals were, for both normal and impaired listeners: (1) to determine what acoustic factors underlie the intelligibility differences observed between speaking styles and (2) to explore the relative contribution of various acoustic cues to vowel identification, using stimuli that vary naturally in intelligibility. Monosyllabic words were excised from sentences spoken either clearly or conversationally and presented in a background of 12-talker babble for vowel identification by young normal-hearing and elderly hearing-impaired listeners. Vowel identification performance was correlated with the results of acoustic analyses to assess the relative importance of spectral, durational, and dynamic cues to vowel perception. Preliminary analyses suggest that spectral target and dynamic formant information are primary cues to vowel identity, while the importance of duration is significant but small. [Work supported by NIHDCD-02229 and the ASHA Foundation.]
Is the perceptual magnet effect a matter of auditory sensitivity?
Authors: Radhakrishnan Sreedivya , Hawks John W. , Otto Robert J.
Published: July 2002

Efforts to validate and extend Kuhl’s [P. K. Kuhl, Percept. Psychophys. 50, 93–107 (1991)] perceptual magnet effect for vowel prototypes in adult listeners has been somewhat elusive and particularly difficult to replicate in other than the /i/ vowel space. However, there is converging evidence that suggests the magnet effect, attributed to assimilation by a prototype, may more likely be due to differences in auditory sensitivity across the vowel space. Hawks’ [J. W. Hawks, J. Acoust. Soc. Am. 95, 1074–1084 (1994)] data on discrimination of formant changes for 17 synthetic vowels supported this notion, indicating that difference limens (DLs) based from an exemplar token for /i/ was among the largest (poorest) found, while tokens neighboring /i/ yielded the smallest DLs. If the perceptual magnet effect is a demonstration of differences in auditory sensitivity and Hawks’ data adequately reflects auditory sensitivity in the vowel space, then a likely location to find a magnet effect is within the space for /u/, which yielded the poorest DLs in Hawks’ study. The results of an experiment investigating this possibility through replication of Kuhl’s original protocol in the vowel space for /u/ will be presented.
Effects of first formant onset properties on voicing judgments of prevocalic stops without F1 cutback
Authors: Benkí José R.
Published: July 2002

This study examines the effects of first formant (F1) onset properties on voicing judgments of prevocalic stop consonants in contexts that do not exhibit the F1 cutback covariation present in pretonic and utterance-initial voicing contrasts. Previous research [K. R. Kluender and A. J. Lotto, J. Acoust. Soc. Am. 95, 1044–1052 (1994)] strongly suggests that F1 onset frequency effects on voicing judgements are due to auditory factors, and not to listener experience with natural covariance with F1 cutback and F1 onset frequency. The present study extends the previous findings to voicing contrasts in contexts that are not signaled by F1 cutback, such as before unstressed vowels in English. Categorization data were collected for a continuum between rabid (/b/) and rapid (/p/) in which both closure duration and F1 onset properties were manipulated using LPC resynthesis of natural speech. Preliminary results indicate that lower F1 onset frequencies of the unstressed vowel condition more voiced percepts of the preceding stop consonant, consistent with findings for utterance-initial stop consonants.
Inhibition in phonological priming: Lexical or strategic effects?
Authors: Shoaf Lisa C. , Pitt Mark A.
Published: July 2002

Facilitatory (speeded) and inhibitory (slowed) response times are found in phonological priming experiments, in which the amount of word-initial phoneme overlap between a prime and target is varied (e.g., mark–must). While facilitation appears to be strategic, there is debate as to the nature of the inhibitory priming found when high-overlap prime-target pairs are used (e.g., musk–must). Some researchers propose that this inhibitory priming is due to lexical competition between simultaneously activated candidates; others suggest it is due to the use of a response strategy. Experiments in our laboratory attempted to resolve this debate. We tested for the presence of strategic effects by manipulating variables thought to influence strategy acquisition (i.e., isi, proportion of overlap trials), and then by examining participants’ RTs to trials of varying overlap (e.g., prone–must, mark–must, muff–must, musk–must) collected over the course of the experiment. Results suggest that inhibition in the phonological priming paradigm is determined in large part by the development of a response strategy.
The effect of amplitude modulation on periodic- and aperiodic-formant-based sentences
Authors: Carrell Thomas D. , Dunn Camille C. , Gutzwiller Jennifer
Published: July 2002

Amplitude modulation has been shown to increase the intelligibility of tone-analog sentences by 30%–60% [T. D. Carrell and J. Opie, Percept. Psychophys. 52, 437–445 (1992)]. Tone-analog sentences are constructed from three or four time-varying sinusoidal waveforms that trace the formant frequencies of the first several formants of a natural utterance [Remez et al., Science 212, 947–950 (1981)]. Because these sentences have no fundamental frequency an important auditory grouping cue is missing. It has been argued that the observed intelligibility increments were based on the ability of amplitude modulation to supply auditory grouping cues similar to those missing from tone-analog sentences. This explanation was tested with whispered sentences and with reduced-channel envelope-shaped noise (ESN) sentences [C. L. Bahner and T. D. Carrell, 138th meeting Acoust. Soc. Am. (1999)]. In an experiment comparing the effect of amplitude modulation on tone-analog sentences, whispered sentences, and reduced-channel ESN sentences, it was found that the intelligibility of tone-analog sentences was improved whereas the intelligibility of the other two types of sentences was reduced. This pattern of results calls for a distinction between the effects of amplitude modulation on sentences constructed with periodic-based formants versus those constructed with aperiodic-based formants.
Blind estimation of the relative travel times and amplitudes of multipath using auto- and cross-correlation functions
Authors: Spiesberger John L.
Published: July 2002

A location problem is considered where sound propagates along multipath which are impractical to model because the environment is poorly known. The acoustic bandwidth is assumed to be large enough so that the cross-correlation functions between pairs of receivers contain multiple peaks from multipath. The highest peak may not correspond to the difference in path lengths between the source and the receivers. Using similarities in the patterns of peaks in auto- and cross-correlation functions, an algorithm is developed to identify which cross-correlation peak corresponds to the difference in first arrivals, which can be used for locating the source if these arrivals are straight. The similarities are expressed with new ‘‘correlation equations.’’ The number of lag-type correlation equations is O(R2N2), where N is the typical number of multipath at each of R receivers. The correlation equations may be impractical to solve exactly. Accurate solutions are found in simulations for the numbers, relative travel times, and amplitudes of all the multipath with the aid of a new fourth-moment function which is a cross correlation of non-negative lags of an auto-correlation function with lags from a cross-correlation function. The technique relies on time series which are filtered to yield one dominant source.
Horizontal refraction and sound fluctuations caused by internal waves packets in shallow water
Authors: Katsnelson Boris , Pereselkov Serguey
Published: July 2002

The subject of the paper is to consider the propagation of a low-frequency sound field in shallow water in the presence of solitonlike internal waves (SLIW). It is assumed that SLIW spread approximately across the stationary acoustic trace. The problem of sound propagation is considered within the framework of 3-D model of the shallow water. The theory of ‘‘horizontal rays and vertical modes’’ is applied to take into account influence of horizontal refraction due to SLIW. As shown in the paper SLIW can lead (periodically in time) to ‘‘focusing’’ and ‘‘defocusing’’ of the horizontal rays or, in other words, to the periodic appearance of ‘‘horizontal waveguide.’’ Estimations of the sound fluctuations due to SLIW demonstrate that the amplitude of fluctuations can reach up to 10–20 dB (between moments of the ‘‘focusing’’ and ‘‘defocusing’’). The next effect conditioned by internal waves is horizontal filtering of vertical modes, where we have separate focusing and defocusing rays, corresponding to different vertical modes at different points of horizontal waveguide. The results presented in the paper can be used as the basis of remote sensing of the internal waves packets in shallow water. [Work supported by RFBR, Grant 97-05-64878.]
Midfrequency structural acoustics by DDSM. I. Beam-like structures
Authors: Oppenheimer Charles H. , Miller Russel D.
Published: July 2002

The direct dynamic stiffness method (DDSM) allows low and midfrequency structural acoustic analysis of complex structures containing beam- and plate-like members and may be used in conjunction with other models like finite-element (FE) models. The DDSM description of beam-like structures involves closed-form relationships between transverse, axial, and torsion loads and displacements, and is accurate at frequencies below a beam-like structure’s fundamental cross section resonance frequency. Transverse shear and rotational inertia effects are included. Modeling of free layer and granular material damping treatment is discussed, and measurements on treated tubes are compared to predictions. A comparison is made between DDSM and FE solutions of a truss structure. The predictions are essentially identical, but the DDSM model is 91% smaller and requires 90% less computation.
A single-degree-of-freedom approach to response perturbation due to feature variability
Authors: Cunefare Kenneth A. , De Rosa Sergio
Published: July 2002

This paper presents a technique to assess the impact on model predictions due to feature perturbations on harmonically forced fluid-loaded structural acoustic models. The perspective taken is one of knowledge of a reference state, with a desire to determine the impact on the total radiated acoustic power due to perturbations in the reference state. Such perturbations change the predicted resonance frequencies of a structure under consideration, and hence, change the predicted response amplitudes. The method uses a single-degree-of-freedom response model in the local region of each fluid-loaded resonance, coupled with eigenvalue sensitivities, to estimate the perturbation impact. The sensitivity of the eigenvalues to changes in model detail is derived based on variations in the spatial representation of attached features (e.g., point versus distributed attachments). Elements of the analysis method are not necessarily restricted to model perturbations nor acoustic power, rather they may be used to assess the perturbation of any quadratic response quantity of interest due to changes in resonance frequency. The analysis reveals that the bandwidth of response perturbation increases directly with increasing resonance frequency. This ‘‘proportional bandwidth’’ has bearing upon the so-called midfrequency gap in structural acoustics.
Localized wave beam forming tests
Authors: Lewis D. Kent , Candy James V. , Ziolkowski Richard W.
Published: July 2002

Much has been reported about work done creating the localized wave (LW) acoustic pulse using active beam creation. This presentation will show work done in passive beam forming using the LW source signals as array filter coefficients. The simple tests which were performed showed that the LW filter can detect wide bandwidth signals in an idealized medium, and that there is an apparent increase in signal resolution. These results suggest that there may be applications for other low-diffraction beam techniques in passive signal processing. [This work was performed by the Lawrence Livermore National Laboratory under the auspices of the U.S. Department of Energy under Contract No. W-7405-ENG-48.]
Three-dimensional tracking of two blue whales using MFP and a tilted vertical array
Authors: Thode Aaron M. , D’Spain Gerald L. , Kuperman William A.
Published: July 2002

Two methods are explored for obtaining the bearing, range, and depth from three different blue whales, recorded in 1996 by a tilted vertical array deployed from the research platform FLIP. The first method uses the global inversion program SAGA to invert for the range, depth, and projected array tilt along the 2-D matched-field processing (MFP) plane between the animal and FLIP. Using array inclinometer information, the projected tilt can be converted into a source bearing. A second method exploits the range-dependent bathymetry around FLIP to perform three-dimensional MFP. This method makes fewer assumptions about the array geometry, but is much more computationally intensive than the first approach, and is sensitive to bottom mismatch. DIFAR sonobuoy bearings from two whales provided an independent test of these MFP bearing estimates. It was found that the projected-tilt method could identify the correct azimuthal quadrant. The fully 3-D MFP method selected bearings within 20 deg of the true bearing, but exhibited a bias toward regions of low-range dependence. [Work sponsored by ONR.]
Applications of a high-quality sound manipulation algorithm s
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Authors: Kawahara Hideki
Published: July 2002

Applications of a high-quality speech analysis/modification/synthesis algorithm [Kawahara et al., Speech Commun. 27, 187–207] for animal voices are discussed. The proposed algorithm consists of (1) F0 adaptive spectral smoothing to eliminate periodic interferences on a time-frequency representation, (2) F0 and source information extractor based on a fixed point analysis of mapping from frequency to instantaneous frequency, and (3) group delay manipulation to control temporal structure of an excitation source. Each component has to be tuned to specifications of animal voices to take full advantage of the method. There are two types of applications: The first method is converting animal voices to sound more like human speech: This modification tries to enhance the researcher’s auditory inspection of animal voices by mapping their voice physical parameters onto a region which is ecologically valid for humans. The second method is manipulating specific parameters of an animal voice to test function/contribution of each physical parameter: This modification tries to keep the naturalness of the stimuli while keeping a high degree of control accuracy of specific physical parameters of stimuli. This strategy would enable measurements of behavior responses in the vicinity of natural stimulus conditions.
Time-frequency analysis: A tutorial review
Authors: Loughlin Patrick J.
Published: July 2002

Many signals, such as speech and other animal sounds, FM radio waves, machine vibrations, and sonar and radar echoes, exhibit frequency characteristics that change over time. Standard spectral analysis of such signals provides an incomplete description of the process, because the spectral density reveals only what frequencies existed in the signal, but not when they occurred. Time-frequency analysis, however, shows how the frequencies change over time and hence is a more complete characterization of these signals. This approach to time-varying spectral analysis has become standard and has revealed new physical properties of signals. Perhaps the most common method of time-frequency analysis is the spectrogram, developed over 50 years ago. Many other methods have since been developed, driven largely by a desire to overcome limitations of the spectrogram. Most of these methods were developed by effective use of Cohen’s 1966 formulation for generating time-frequency distributions (TFDs). A review of the spectrogram and these TFDs, including the Wigner, Choi–Williams, and Zhao–Atlas–Marks distributions, among others, is presented. A variety of applications are shown that illustrate the basic ideas and the different methods. [Work supported by ONR grant N00014-98-1-0680.]
Testing the acoustic adaptation hypothesis for eastern and spotted towhee song
Authors: Khanna Hitesh
Published: July 2002

The acoustic adaptation hypothesis predicts that signals adapted for long-distance communication in forested habitats should possess low rates of amplitude modulation (slow trills), whereas signals adapted to open habitats should possess high rates of amplitude modulation (fast trills). The songs of two songbirds, the eastern and spotted towhees, were used to test this hypothesis. These two species have only recently been recognized as distinct species. Differences in song are one of the reasons given for splitting the species. Eastern towhees occupy edges of deciduous forests and sing slow trills. Spotted towhees occupying open chaparral habitat sing fast trills. Songs of both species were broadcast in their native habitats, and recorded at several distances from the sound source. Spectrogram cross-correlation was used to quantify sound degradation with distance. The results support the predictions of the acoustic adaptation hypothesis. The slow eastern towhee trill degrades less than the fast spotted towhee trill in a forested habitat, whereas the fast spotted towhee trill degrades less than the slow eastern towhee trill in an open habitat.
The perfectly matched layer for elastic waves in poroelastic media
Authors: Zeng Yanqing , He Jiangqi , Liu Qinghuo
Published: July 2002

The perfectly matched layer (PML) as a material absorbing boundary condition (ABC) was first introduced by Berenger for electromagnetic waves, and later developed by Chew and Liu for elastic waves. In the continuous limit, an interface between a regular medium and a fictitious, lossy PML medium can be made perfectly matched so that there is no reflection from the PML to the regular medium. This property is independent of the incidence angle and the frequency of the incoming waves. Consequently, the PML provides an ideal ABC for the truncation of the computational domain in numerical methods such as the finite-difference, finite-element, and pseudospectral time-domain methods. Numerical experiments show that this ABC can reduce the reflection to several orders of magnitude below the level of the previous ABCs. In this work, the PML is further extended to elastic waves in poroelastic media through the approach of complex coordinates for Biot’s equations. This nonphysical material is used as an ABC at the computational edge of a finite-difference algorithm to truncate unbounded media. Numerical results show that the PML ABC attenuates the outgoing waves effectively.
Newly observed seismoacoustic wave characteristics in disturbed water-covered sand
Authors: Chamuel Jacques R.
Published: July 2002

Surprising new experimental results are presented revealing the characteristics of broadband transient seismoacoustic waves in water-covered naturally deposited sand and disturbed sand. In the naturally deposited sand, one compressional wave (∼1650 m/s) and one shear wave (∼140 m/s) were detected as expected. In the disturbed sand, the high-frequency (∼100 kHz) fast compressional wave was highly attenuated, and a slow low-frequency (∼20 kHz) compressional wave component was observed. In certain models, the slow low-frequency compressional wave component co-existed with the fast high-frequency compressional wave. The slow compressional wave velocity varied between 285 m/s and 210 m/s. The shear wave velocity and amplitude remained practically unchanged. Results from coarse and fine sand are compared. The findings are important for understanding the penetration and conversion of high-frequency acoustic waves in sandy seafloor critical for the detection of buried objects in littoral regions. [Work supported by ONR.]
A cross-relation matched field inversion for geoacoustic parameter estimation in shallow water
Authors: Dizaji Reza M. , Chapman N. Ross , Kirlin R. Lynn
Published: July 2002

In this paper the application of a matched field (MF) processor for geoacoustic parameter estimation from a signal of opportunity such as the broadband random noise radiated by passing ships is considered. A novel cross-relation (CR) based matched field processor is introduced for the purpose of ocean tomography based on broadband random source. This class of MF processors is based on the cross-relation property of sensor outputs at an array and their corresponding transfer functions from the true source location to the array. The processors are developed for nonstationary (NS), and wide sense stationary (WSS) random signals. For each formulation, two processors are proposed, a self-CR and a cross-CR. The performance of the proposed MF processors for environmental parameter estimation is demonstrated for real ocean environments using data collected by a 16-element vertical line array during one of the experimental tracks from the Pacific Shelf experiment that was carried out in shallow water off the west coast of Vancouver Island in the Northeast Pacific Ocean. The high-resolution property makes the cross-CR processor an excellent candidate for inversion for model parameters of the ocean waveguide. The replica or modeled fields are calculated using the normal mode model, ORCA.
Use of surfseisms for determining near-surf-zone properties
Authors: D’Spain Gerald L. , McArthur K. Megan , Deane Grant B. , Melville W. Kendall
Published: July 2002

Geophone array recordings of ‘‘surfseisms,’’ i.e., seismic interface waves created by breaking waves in the surf zone, show that at times, a breaking wave creates a pair of arrivals. The first arrival has peak spectral levels between 10 and 15 Hz and rolls off sharply above this peak, and the second surfseism arrival occurs 5–15 s later with broader-band energy concentrated above 20 Hz. The first arrival is created offshore when wave breaking has a plunging character, rather than spilling, whereas the second is present during both breaking types and apparently signifies the final dissipation of the turbulent wave bore on the beach face. The time interval between arrivals measures the distance offshore that plunging occurs, their relative amplitudes is an indicator of the type of wave breaking, and their absolute amplitudes measure the breaking wave forces interacting with the ocean bottom. In addition, the frequency of the first arrival’s spectral peak is a sensitive function of the thickness of the unconsolidated sand layer overlying wave-cut terrace material, indicating that surfseisms are naturally occurring probes of sand layer thickness. [Work supported by ONR, Code 32.]
A two-dimensional array system for studies of ultrasonic imaging with aberration correction
Authors: Waag Robert C. , Phillips Daniel B. , Lacefield James C. , Draeger Carsten G. , Lin Feng , Tabei Makoto
Published: July 2002

A two-dimensional array system is described for pulse-echo studies of aberration correction. The transducer array is an 80×80 array with a center frequency of 3.0 MHz and a −6-dB bandwidth of 56%. At the center frequency, each element has a physical size of 1.04 wavelength and spacing of 1.2 wavelength. A multiplexer accesses any contiguous 128 elements for transmission and any contiguous 16 elements for simultaneous reception. Transmit electronics have independently programmable waveforms. Each receive channel includes a 20-MHz, 12-bit A/D converter, and a time varied gain programmable over 40 dB. Transmit and receive apertures up to the size of the array are formed synthetically. A method that iteratively predistorts transmit waveforms to produce a transmit focus compensated for aberration has been implemented. Point-spread functions have been measured for propagation through a water path and through a tissue-mimicking aberration path. Pulse-echo images have been formed through a water path, through a tissue-mimicking aberrator, and through the aberrator using aberration correction that consists of time-shift compensation in the transmit–receive aperture or backpropagation followed by time-shift compensation.The system is useful for pulse-echo measurements of aberration, development of adaptive focusing techniques, and formation of high-resolution ultrasonic images using aberration correction.
Novel breast imaging techniques using transmission ultrasound
Authors: Andre Michael P , Janee Helmar S. , Olson Linda K. , Lehman Constance D. , Fecht Barbara A.
Published: July 2002

Despite several limitations, ultrasound is an important adjunctive modality for detection and management of breast cancer, but it is not recommended for screening. We have explored novel approaches to breast imaging using transmission ultrasound to provide very large image fields, differentiation of breast masses, and improved detection of microcalcifications without the use of ionizing radiation or compression. One method uses a circular array and an iterative Born technique to reconstruct scatter properties in a series of coronal slices through the breast. The second method is based on acoustical holography in which an interference pattern is formed by combining a transmitted field with a reference beam. The signal is converted to the optical domain using a coherent laser detector. This latter method provides real-time images in a large fluoroscopy-like format but with pronounced soft tissue edge definition. A set of large acoustic lenses provides zoom and focal plane control for magnified imaging with enhanced resolution and depth. Both systems have undergone preliminary investigation in large groups of patients with known abnormalities to assess their suitability for detection and characterization of breast disease and to provide accurate guidance for biopsy or tumor ablation. Results of laboratory and clinical findings will be presented.
The perceptual consequences of overlap in /s/ and /∫/ productions within a talker
Authors: Clouse Sheryl A. , Burnham Jessica L. , Newman Rochelle S.
Published: July 2002

A primary issue in speech perception is the apparent lack of invariance between the acoustic information in a signal and the listeners perception. Different intended phonemes may be produced with identical acoustic values. Previously, we examined fricative centroids and frication peaks for over 100 utterances beginning with /s/ and /∫/ from each of 20 different speakers, and found substantial overlap across talkers. In the present study, we examine the effect of this overlap on perception. Listeners in a phoneme identification task heard natural productions of /s/ and /∫/ syllables from speakers with either little or great overlap between categories. Although labeling accuracy was near ceiling, effects were found in listeners’ reaction times. Listeners needed more time to interpret the speech of talkers who had substantial overlap in their fricative centroids, even when these talkers showed no overlap in their frication peaks. Listeners also had slower reaction times for talkers with overlap in their frication peaks, but not their centroids. This suggests that category overlap has measureable consequences to perception, and that peaks and centroids are sensitive to different aspects of the speech signal, both of which are perceptually impotant to listeners. [Work supported by NIDCD Grant R01-DC00219 to SUNY at Buffalo.]
Effects of stimulus uncertainty, consonantal context and training on formant frequency discrimination
Authors: Kewley-Port Diane
Published: July 2002

Words in natural speech vary in their predictability, e.g., stimulus uncertainty ranges from low to very high across sentences. The ability to discriminate formants is strongly affected by longer phonetic context [Kewley-Port and Zheng (in press)]. This study investigates the effects of stimulus uncertainty from minimal to high uncertainty and the phonetic contexts /V/ or /bVd/. The primary formants were from four female vowels /ɪ, ε, æ, ə/. Using adaptive tracking, DLs for discriminating a small change in a formant was calculated in Delta Barks. In experiment 1, performance for five listeners, optimized by extensive training, began with minimal uncertainty, subsequently increasing uncertainty from 8- to 16- to 22 formants per block. Effects of higher uncertainty were less than expected, only decreasing performance by about 33%, although DLs for CVCs were 25% poorer than for isolated vowels. In experiment 2, performance in the 22-formant condition was tracked over 1 h for 37 listeners without formal laboratory training. DLs for untrained listeners were about 300% worse than for trained listeners, and comparable to untrained listeners in Kewley-Port and Zheng (in press). Results indicate that the effects of longer phonetic context degrade formant frequency discrimination more than higher stimulus uncertainty. [Work supported by NIHDCD-02229.]
Estimating radiation modes using wave-number filtering
Authors: Sommerfeldt Scott D. , Lin Dong
Published: July 2002

Acoustic radiation modes have been investigated in recent years as a means of estimating the far-field radiated acoustic power from a structure. The authors have been investigating two possible methods of estimating the amplitudes of the radiation modes associated with radiation from a baffled beam. The first method, based on using distributed sensors shaped according to the radiation modes, has been reported previously [S. D. Sommerfeldt and D. Lin, J. Acoust. Soc. Am. 105, 1088 (1999)]. This paper will report a second method, based on using distributed sensors designed to act as low-pass wave-number filters. The spatial filtering provided by the sensors allows the array to estimate the radiation mode amplitudes with only a small number of sensors, while still avoiding spatial aliasing problems. Numerical analysis indicates that the method is capable of providing good estimates of the radiated acoustic power for a normalized frequency range extending up to kl=10, using an array of only six sensors. Although the radiation mode shapes are frequency dependent, the method has demonstrated the ability to provide good modal estimates throughout this entire frequency range. Broadband estimates of the acoustic radiation are typically within 1 dB of the actual value.
Contrast enhancement improves the representation of /ɛ/-like vowels in the hearing-impaired auditory nerve
Authors: Miller Roger L. , Calhoun Barbara M. , Young Eric D.
Published: July 2002

This study examines the neural representation of the vowel /ɛ/ in the auditory nerve of acoustically traumatized cats and asks whether spectral modifications of the vowel can restore a normal neural representation. Four variants of /ɛ/, which differed primarily in the frequency of the second formant (F2), were used as stimuli. Normally, the rate-place code provides a robust representation of F2 for these vowels, in the sense that rate changes encode changes in F2 frequency [Conley and Keilson, J. Acoust. Soc. Am. 98, 3223 (1995)]. This representation is lost after acoustic trauma [Miller et al., J. Acoust. Soc. Am. 105, 311 (1999)]. Here it is shown that an improved representation of the F2 frequency can be gained by a form of high-frequency emphasis that is determined by both the hearing-loss profile and the spectral envelope of the vowel. Essentially, the vowel was high-pass filtered so that the F2 and F3 peaks were amplified without amplifying frequencies in the trough between F1 and F2. This modification improved the quality of the rate and temporal tonotopic representations of the vowel and restored sensitivity to the F2 frequency. Although a completely normal representation was not restored, this method shows promise as an approach to hearing-aid signal processing.