bluuebunny/crossref_metadata_2025 huggingface dataset:
Absolute tonality versus absolute piano
Authors: Ward W. Dixon
Published: October 2005

Earlier studies of the ability of individuals to recognize when excerpts from Bach's “Well-Tempered Clavichord” has been shifted upward or downward from the original key by 1 or 4 semits suggested that almost all musicians were able to recognize the direction of these shifts at a better than chance level, even those who professed little or no ability to identify isolated tones (absolute pitch). In an effort to determine to what extent this ability represents (1) “absolute tonality”—long-term memory for the specific compositions, (2) detection of a disparity between sound and printed score, and (3) a vague recognition that the series of tones, no matter who wrote them, are played too high or too low to be good music, ten musicians were given the test “cold”—i.e., they were asked to identify the shifted excerpts with no prior exposure to examples of extreme shifts and without being shown the score. All succeeded in identifying the ± 4-semit transpositions, though not the ± 1-semit ones. A week later, a second test was given, but with the printed score (in the original key) available. Now, even the ± 1-semit shifts were recognized. The results provide little support for the absolute tonality hypothesis.
A study of angular distributions from submerged spheroidal and finite cylindrical shells
Authors: Werby M. F. , Green L. H. , Wooten R. P.
Published: October 2005

In the investigation of submerged objects, one generally examines backscattered form functions. In theoretical studies, this has been standard practice in the investigations of spheres and infinite cylinders; and this usually limits one to the determination of resonances. When such studies are carried out, one must already know the orientation of the incident field relative to the object (this is obvious for spheres); and calculations must be performed over a broad frequency range in small increments. However, for elongated objects, the orientation of the incident field relative to the object as well as the actual shape of the object is of importance and cannot be determined from backscattered measurements alone. We examine the feastibility of determining shape information as well as orientation by studying bistatic angular distributions from spheroids and finite cylinders for a variety of orientations relative to the axis of symmetry. In addition, we examine the effect resonances have on angular distributions, particularly in the forward direction.
Scattering from submerged solid-, fluid-, and air-filled elastic shells to the high ka region
Authors: Gaunaurd G. , Feuillade C. , Werby M. F.
Published: October 2005

We examine scattering from submerged, solid-, fluid-, and air-filled elastic spheres for ka values ranging from 0 to 100 (where k is the wavenumber of the incident waves and a is the radius of the sphere). The analysis includes interpretation of forward and backscattered form functions for a variety of materials and shell thicknesses. An analysis of resonances, particularly their change in character with increasing ka, will be illustrated using numerous examples.
The interaction of acoustic waves with multi-layered shells
Authors: Green L. H. , Werby M. F. , Wooten R. P.
Published: October 2005

We report on our investigation of steady-state scattering of acoustic waves by multi-layered shells with an elastic inclusion. Numerical results will be presented for scattering from spheroidal shells and finite cylindrical shells with hemispherical endcaps. Comparisons will be made with similar results for scattering from elastic solids, thin shells, and impenetrable targets. The frequency of the incident acoustic wave will be in the resonance region of the materials studied. The identification of prominent effects with the elastic waves propagating in the shell will be made. The dependence of the results on frequency, shell and fluid parameters, and aspect ratio is discussed. A physical interpretation of the results will be provided.
Analysis of the impedance boundary condition in terms of an exact scattering calculation
Authors: Ayres Virginia M. , Gaunaurd Guillermo C.
Published: October 2005

The scattering of incident acoustic waves by convex fluid-loaded bodies can be analytically modeled at various levels of realism. In our approach to scattering from elastic/viscoelastic bodies, the coefficients of the exact Rayleigh series for the scattered field are determined by the application of three boundary conditions (b.c.). These boundary conditions are statements of continuity of stresses and displacements across the body's surface. An often-used approximation assumes the body to be impenetrable and subjected to a single impedance-type or Cauchy b.c. on its convex surface. The specific surface impedance of the body's material, usually determined by other means, is introduced into the formulation in an “ad hoc” way. The details and limitations of how a single simple b.c., on a body assumed impenetrable, can model the effects of three quite complex ones, on a realistically modeled penetrable object, are studied here for a simple (spherical) shape. Comparison of the two methods yields an expression for the surface impendance which is dependent on frequency, on mode order, and on other parameters of the acousto-elastic solution. We find this dependence to be quite complicated. The outcome, which is amply supported by numerical displays, provides a more basic understanding of the uses and limitations of the impedance approach to predict echoes from any fluid-loaded real object.
Bistatic echoes from elastic bodies in the resonance scattering regime
Authors: Tsui C. Y. , Reid G. N. , Gaunaurd G. C.
Published: October 2005

The experimental technique that we described earlier [C. Y. Tsui et al., J. Acoust. Soc. Am. Suppl. 1 77, S79 (1985)] which extracts the active resonance spectrum of any scatterer from the echo it returns, was applied to solid elastic cylinders and to cylindrical shells of various thicknesses immersed in water. This technique consists in sampling the tailend of the returned pings, which contain the elastic (whispering gallery) resonances of the bodies, in order to isolate them from the (rigid) backgrounds they are usually mixed with. We now study various experimentally obtained differential scattering (bistatic) cross sections for the above mentioned objects. It is found that at certain resonance frequencies, the background-suppressed, bistatic cross section, which is a measure of the free-vibration target reradiation, ideally consists of a symmetric rosetta pattern, having twice the number of lobes as the modal order n of the body resonance being excited. Using theoretically computed plots of the actual spectral shapes of the body's individual modal resonances, we explain that the anomalous angular plots, which have more than 2n lobes, are caused by a superposition of several modes for which other broad modal resonances having wide tails, interfere with the chosen resonance frequency within the nth mode. Theory and experiment complement each other to elucidate the resulting aliasing effect.
Resonance spectra of elongated elastic objects
Authors: Überall H. , Nagl A. , Stoyanov Y. J. , Numrich S. K. , Brown S. H. , Dickey J. W. , D'Archangelo J. M.
Published: October 2005

It is well known [H. Überall, L. R. Dragonette, and L. Flax, J. Acoust. Soc. Am. 61, 711 (1977)] that the complex eigenfrequencies at which smooth convex objects resonate under the incidence of an acoustic wave are those at which circumferential waves generated by the incident signal phase match over a closed orbit. This principle was verified by us, by analytically obtaining the resonance frequencies of elastic cylinders and spheres [E. D. Breitenbach et al., J. Acoust. Soc. Am. 74, 1267 (1983)] and deriving phase and group velocities of the surface waves from this. In the present study, this approach is inverted, and applied to elastic prolate spheroids and to cylinders with hemispherical endcaps. From the known phase velocities and trajectories (geodesics) of the surface waves, we were able to predict the elastic resonance frequencies of these bodies. [Work supported by the David W. Taylor Naval Ship R & D Center, the Naval Research Laboratory, and the Office of Naval Research.]
Resonances of a spherical shell covered with a constrained viscoelastic layer
Authors: Husson D. E.
Published: October 2005

A computer model has been written to predict the interaction of a plane acoustic wave with an elastic or viscoelastic layered sphere. This model is based on the decomposition of the acoustic potentials in each layer in spherical harmonics. It can compute the scattered near- and farfields of spheres formed of up to ten layers, at frequencies up to ka = 120, a being the radius of the sphere and k the wavenumber of the incident wave. This model is used to study the influence of a constrained layer, consisting of a viscoelastic layer and a constraining outer shell, on the resonances of a metallic hollow sphere. Viscoelastic materials are modeled according to the equations of the Kelvin-Voigt model, with complex elastic moduli. Plots of the acoustic pressure backscattered by the sphere, of the total pressure on its surface, of its surface displacements, and color-coded maps of the pressure around the sphere are presented to show the influence of the thickness of the viscoelastic layer and of its absorption coefficients on the resonances of the metallic shell, and the effect of the resonances of the three layers on the scattered field.
Scattering from multiple compliant tube gratings in an elastic layer
Authors: Radlinski Ronald P. , Janus Robert S.
Published: October 2005

From previous studies of scattering from multiple gratings of resonant compliant tubes in water [R. P. Radlinski and M. M. Simon, J. Acoust. Soc. Am. 72, 607–614 (1982)], the excitation of noncompliant, antisymsmetric structural modes by nearfield evanescent waves was found to severely degrade the reflectivity of closely packed gratings in the bandwidth of excitation of the compliant symmetric modes. Also, transmission resonances due to the spring-mass-spring configuration of the two gratings separated by a fluid mass diminished low-frequency performance. In this paper, encapsulating the gratings in a low-stiffness elastomer is shown experimentally to have a minimum effect on single and widely separated gratings with respect to fluid but enhances the performance of closely packed gratings. Comparison of insertion loss performance with a high stiffness encapsulant indicates dramatic frequency response differences for closely packed arrays. Two different grating configurations will be considered and comparisons with a mathematical model will be discussed.
A real-time digital signal processing evaluation system
Authors: McCollough W. Vance , Knight Diane M. , Erickson David , Jannsen Raymond A.
Published: October 2005

This presentation will describe a test-bed processing system which is based on four serially connected Texas Instruments TMS-320 signal processors. Programs can be developed using standard TI development tools, and loaded into the test-bed processor for execution. Any one of the processors can be replaced by the TI EVM in-circuit emulator, for more flexibility in algorithm testing and development. A variety of status, control, and interrupt inputs and outputs are provided to interface the processors to each other and to external devices. Built-in A/D and D/A converters provide ports for data acquisition and output, and can be synchronously clocked from external sources. Digital data can also be input and output—the digital output port can directly interface to a line scan recorder. The primary uses for the test-bed processing system are to test software and algorithms on real data and in a realistic hardware environment, and as a laboratory tool to analyze experimental data. An example will be discussed which shows how the test-bed processor has been used in the above ways to evaluate a correlation processor, and to process experimental data.
Hardware for time delay beamforming, demodulation, and processing
Authors: Metzger K. , Birdsall T. G.
Published: October 2005

A digital time delay beamformer/demodulator/processor was designed and built as part of the U of M's involvement in ocean acoustic tomography measurements. This paper describes the overall system design, the criteria on which design decisions were made, and how well the resulting equipment performs. The beamformer can form up to eight simultaneous beams using up to 24 hydrophone inputs per beam. The beam outputs are coherently shifted from any center frequency in the range from 10 Hz to over 500 Hz down to baseband. Center frequencies can be set in steps of 1 mHz. Special hardware was included in the system to speed up commonly performed processing operations. The fast arithmetic device uses serial multipliers and adders to aid in performing double precision complex additions and various multiplication operations. The sequence removal hardware speeds up the computations involved in pulse compressing the linear maximal sequence coded transmissions commonly used in making ocean acoustic multipath measurements. The speed-up factor is about 100. [Work supported by ONR.]
Fast sequential closed-phase glottal inverse filtering based on optimally weighted recursive least squares
Authors: Luk T. C. , Deller J. R.
Published: October 2005

Presented in this paper is the theoretical basis, with simulation verification, for a sequential method of deconvolution of the glottal waveform from voiced speech. The technique is based upon a linear predictive model of the vocal tract, and the assumption of a “pseudo-closed phase” (PCP) (noisy closed phase) of the glottis during each pitch period. Existing techniques for closed phase glottal inverse filtering (CPIF) employ “batch”-type methods which are generally slow, highly user interactive, and restricted to the use of one cycle of data in the analysis. The basic ideas underlying CPIF, and a brief review of existing methods, will be presented in the first part of the paper. In the second part of this paper, requisite theoretical results and the new method will be developed. In particular, these will include a unified theory of CPIF in which the selection of closed phase points is viewed as a data weighting process. This viewpoint readily admits the use of more than one cycle of data in the analysis (advantageous when the data fare noisy), and further leads to the use of optimal weighting of the accepted data. The theory of “membership set” identification is used as a basis for optimization of weights. Novel weighting strategies employed in a conventional recursive least-squares algorithm form the basis of the improved technique. The last part of the paper contains simulation studies and computational considerations. The new method is shown to result in significant increases in both accuracy and computational efficiency.
Measurement of aliasing in cepstrum analysis
Authors: Fahmy Mahmoud N.
Published: October 2005

In acoustical signal processing, cepstrum analysis is used to deconvolve an output signal into a path impulse response and an input source wavelet. The cepstrum of an input wavelet will occupy the first portion in the quefrency domain while the path response will occupy the last portion. Aliasing in the quefrency domain is of course an ever-present problem. This aliasing is caused by introducing a nonlinear complex logarithm in the frequency domain. The aliasing error has a significant effect on the last portion in the quefrency domain, which represents the path response. In this work, a special proposition is introduced to measure the error due to aliasing caused by sampling of nonband-limited function in the quefrency domain. By appending zeroes to the frequency domain or multiplying the time domain by an exponential window, the aliasing error in a path response can be reduced. Different examples using a real signal will be presented. A special type of long pass lifter was used to extract the path response with a minimum of aliasing errors measured by the new proposition.
Statistical signal analysis for systems with mutually related input functions
Authors: Bai M. Robin , Mielnicka-Pate A. L.
Published: October 2005

Statistical signal analysis approaches have been successfully used in analyzing linear and constant parameter multiple input-output systems. These methods have been particularly useful when input signals are not correlated. However, there are a number of systems that have measurable input signals that are related. An example is a vibrating plate under two excitation forces. When both the acceleration and force are monitored at a particular input, then they are correlated. In this paper three different approaches are used to investigate a two input-one output system with mutually related input signals. These approaches are: (1) conventional multiple input-output statistical signal analysis, (2) conditioned spectral analysis, and (3) a new approach, based on alternations in one of the input signals. In the last approach, the system of equations for the multiple input-output model is solved twice for two different input signal cases. The results when these three methods are used to analyze an electrical two input-one output model and a plate with two excitation forces will be presented and discussed in terms of the accuracy and required signal processing.
Experimental study on geophysical diffraction tomography
Authors: Lo Tienwhen , Toksöz M. Nafi
Published: October 2005

Most of the diffraction tomography experiments that can be found in the literature focused on medical or NDE applications. With medical or NDE applications, these experiments allowed us to measure the scattered wave field in all desired. directions. This full coverage of the targets by insonifying sonic waves, however, is impossible for objects of geophysical interests. Diffraction tomography experiments with geophysical applications have not been done yet. In order to experimentally test the applicability of diffraction tomography with geophysical applications, the authors conducted an experiment with a target immersed in water and with a source-receiver geometry similar to the source-receiver geometry of the offset VSP techniques currently being used by exploration geephysicists in the field. We measured the scattered wave field by this poor coverage but realistic source-receiver arrangement. With scattered wave field data, we reconstructed the images of our targets by a diffraction tomography algorithm. The results of our experiment are encouraging. Objects of simple geometry can be successfully imaged. In theoretical studies of diffraction tomography, most reconstruction algorithms developed so far are based on two assumptions: (1) weak scattering approximation and (2) linear behavior of scattered wave field assumption. These two assumptions impose limits on the applications of diffraction tomography and the authors took an experimental approach to delineate these limits.
The perceptual reality of a formant frequency
Authors: Klatt Dennis H.
Published: October 2005

A series of vowels similar to [ɪ] was synthesized using a different constant value of fundamental frequency (f0) for each member. As f0 ranged from 133 to 200 Hz, harmonics either fell exactly at the first formant frequency (F1) or a pair of harmonics straddled F1 with varying degrees of skew and thus differing relative amplitudes. The presumed task of the auditory system is to interpret harmonic amplitudes in the neighborhood of F1 in order to estimate the true F1 frequency of the vowel. It will first be shown that (1) energy-based methods of interpreting filter-bank outputs, (2) non-pitch-synchronous linear prediction analysis, and (3) synchrony measures based on auditory models all systematically mis-estimate F1 frequency by as much as plus/minus 10% as f0 varies. Secondly, a perceptual experiment was performed to determine whether these systematic estimation errors are present in auditory judgement data. Comparison of the standard stimulus series with one in which F1 is “corrected” so that a filter bank sees a constant F1 indicates that the uncorrected fixed-F1 series has more nearly constant phonetic quality. Accounting for the perceptual data in terms of simple processing strategies is not yet possible, but if the central auditory system is provided with the frequencies and amplitudes of harmonics near F1, a moderately complex calculation could yield F1 with minimal error. [Work supported in part by an NIH grant.]
Nonlinear auditory coding of vowel formants at high sound pressure levels
Authors: Donnan M. F. , Lindholm J. M. , Hannley M. T. , Leek M. R.
Published: October 2005

To study vowel intelligibility in the absence of the acoustic reflex, ten synthetic vowels whose duration was shorter than the effective latency of the acoustic reflex were presented to normal hearing listeners at levels ranging from 72 to 106 dB SPL. Signal intelligibility varied as a function of both SPL and vowel identity. Vowels with widely spaced formants, i.e., /i/ and /ɪ/, were unaffected by presentation level. Vowels with more proximal formants began to show a decrement in intelligibility at 96 dB and suffered as much as a 40% decrement at 106 dB. Confusion errors were not symmetrical in F1-F2 space suggesting a systematic distortion in the auditory coding of formant location. These data indicate another source of distortion for many severely hearing-impaired listeners who commonly listen to speech amplified to greater than 100 dB and who do not have measureable acoustic reflexes. [Research supported by NIH.]
Consonant confusions and perceptual spaces for natural and synthetic speech
Authors: Yuchtman Moshe , Nusbaum Howard C. , Pisoni David B.
Published: October 2005

Earlier research in our laboratory has demonstrated that synthetic speech is less intelligible and more capacity demanding than natural speech. These differences appear to be related to processes responsible for encoding the input signal into a segmental phonemic representation. There are several hypotheses that could account for the greater difficulty involved in synthetic speech perception. One hypothesis is that synthetic speech is structurally equivalent to natural speech degraded by noise. An alternative hypothesis is that the acoustic-phonetic structure of synthetic speech is impoverished in comparison to natural speech in that a minimal set of acoustic cues are used to implement phonetic segments. The two hypotheses lead to different predictions about the nature of synthetic consonant confusions in relation to confusions of natural speech degraded by noise. To resolve this issue, we carried out multidimensional scaling analyses of confusion matrices for synthetic consonants produced by DEC-Talk, Prodse-2020, and the Votrax Type-N-Talk, and for natural speech at several S/N ratios. The results of the analyses tend to support the second hypothesis: The properties of the perceptual spaces obtained for the synthetic consonants differed considerably from those obtained for the natural consonants. [Work supported by AFOSR.]
Applications of the Apple computer for classroom demonstrations and laboratory experiments
Authors: Bolen Lee N. , Crum Lawrence A.
Published: October 2005

The Apple computer game port can be used as a timing circuit for inexpensive interactive demonstrations and laboratory experiments. Examples of the use of this technique to measure the velocity of a baseball thrown in the classroom by the most adept students as well as the measurement of speed and acceleration of a slot car racer in the laboratory will be demonstrated.
Microcomputer as a laboratory instrument in the basic physics lab
Authors: Shelby R. , Nordling D. , Sadler D.
Published: October 2005

The microcomputer-based measuring station is proving to be a versatile, high-performance tool for use in data gathering, plotting, storage, and analysis in undergraduate physics laboratories. This paper will discuss the large scale use of microcomputer-based measuring stations at the Naval Academy during the last year. Topics considered will include our reasons for changing to measuring stations and reasons for specific equipment choice, our experience in the laboratories with over 1000 students, necessary elements for basic and more advanced stations, the requirement for simple but versatile software, and the philosophy necessary to keep the focus of the laboratories on physics rather than computer science. The specifies of the equipment and software used in our program will be reviewed and a brief look at the future will be given.
Microcomputers and the undergraduate laboratory: An ideal match
Authors: Saltsburg Howard , Heist Richard , Olsen Thor
Published: October 2005

Making the undergraduate laboratory a useful educational experience requires that the tedious data acquisition and reduction of raw data be minimized in favor of enabling the student to concentrate on the scientific and engineering aspects of the study. The use of the microcomputer not only permits that approach to be implemented but offers new opportunities for improved laboratory studies. For the past 5 years, we have been engaged in the introduction of the microcomputer into the undergraduate chemical engineering laboratories. Primary emphasis was placed upon improving the technical content of the program. To provide for efficient operation of the laboratory, every computer-aided experiment has a dedicated computer. This required that we utilize inexpensive computers and minimize the cost of the interfaces needed for appropriate experimental variables. In no sense, however, has a computer laboratory been developed. The computer is simply a tool and only those studies which can profit from the use of the computer have been interfaced. A brief description of the development of the program will be given, several examples of laboratory studies will be described, and some experiments will be demonstrated.
A personal computer workstation for music education
Authors: Dolson Mark
Published: October 2005

Courses in acoustics and psychoacoustics are becoming increasingly common within the music curriculum. An introduction to these subjects enables students both to better understand the structure of traditional music and to explore contemporary alternatives to that structure. But the information in these courses is generally imparted to students exclusively by means of lecture/demonstration. This mode of presentation makes the material remote and difficult to absorb. A sufficiently powerful personal computer with analog-to-digital and digital-to-analog converters can overcome this problem by providing the student with a rich environment in which to explore the material directly. In this report we describe a major software-development project aimed at producing a coherent personal-computer-based workstation for sound analysis, synthesis, and modification. The software is written in C and intended for a UNIX environment. Much of it runs at present on a VAX 11-780 and is used for computer music composition and research. The package currently under development, though, has education as its primary focus. Since this development may continue for some time to come, an important goal of this report is to solicit informed opinions and contributions from a diversity of sources.
Effect of pore fluid viscosity on the velocities of acoustic waves in porous rocks
Authors: Wang Zhijing , Nut Amos
Published: October 2005

Compressional and shear velocities of acoustic waves propagating in porous rocks saturated with viscous fluids were measured both in the kHz and MHz frequency range. The experimental results of the velocities were plotted as a function of pore fluid viscosity ranging from 1 to 1010 centipoise. It was shown that the measured velocities increased with increasing pore fluid viscosity in the kHz frequency range, while in the MHz range, the pore fluid viscosity did not have much effect on the acoustic wave velocities. The experimental results were also discussed in terms of theories of wave propagating in viscous fluids and in the saturated porous solids. Applications of the results to oil production and exploration, as well as to the low velocity zone in the earth, were also presented.
Measurement of normal-incidence impedance of outdoor ground surfaces in the frequency range 20–500 Hz
Authors: Daigle G. A. , Stinson M. R.
Published: October 2005

The impedance of natural ground surfaces has been measured extensively at frequencies above about 200 Hz. At lower frequencies, however, measurements are scarce and have been inaccurate. We have begun using a two-microphone, phase-difference technique similar to that of Nicolas and Legouis [Proc. 2nd Int. Congress on Acoustic Intensity, Senlis (1985)], which promises to overcome some of the difficulties at low frequencies. A point source is suspended above the ground and the sound field is measured with two phase-matched microphones along the vertical line below the source. The reflection coefficient, and hence the impedance, is obtained from the variation of the phase difference between microphones as a function of height above the ground surface. The use of two closely spaced microphones has the advantage of minimizing the effects of random fluctuations due to turbulence. Preliminary measurements have been made down to 50 Hz over grass covered ground. Further measurements are underway in the range 20–500 Hz.
A comparison of optical and numerical methods for analyzing structural vibrations
Authors: Evans Tammy , Clark Joseph A.
Published: October 2005

An optical holographic interferometry method has been used to determine the resonant frequencies, mode shapes, and peak displacement amplitudes of an irregular, clamped, vibrating plate. The plate was driven by an electromagnetic shaker with swept sine wave execution and resonances were observed by time holographic interferometry. Mode shapes at several resonant frequencies were then photographed by a time-average holographic interferometry method. The observed resonant shapes were compared with predictions obtained by two finite element programs (GIFTS and NASTRAN). Facilities required for each method of solution and procedures for collecting and interpreting mode shape data will be described. Applications of the optical and numerical methods to problems involving damped vibrating structures will be discussed. [Research supported by DTNSRDC and ONR.]
Use of DP matching for evaluating synthesized speech
Authors: Saito Takashi , Matsuda Yasuhiro , Kaneko Toyohisa
Published: October 2005

We examined the applicability of DP matching as a tool for evaluating methods for connecting synthesis units. In speech synthesis by rule, speech quality depends greatly on how to connect synthesis units (e.g., syllables). The most widely used method is to use human ears. Although it is justifiable as the final judgment, it is time-consuming as well as fairly listener-dependent. The method proposed here is to use the amount of residual errors of DP matching, which has been widely known in speech recognition. When we confront the problem of selecting parameters of a particular connecting method, we record words naturally spoken (presumably continuous) and then operate DP matching between the natural speech and synthetic speech. The parameter set that produces the least amount of residual error is selected. We devised DP matching particularly suited to this purpose. We found that the optimized parameters with this method agreed well with those obtained with human ears and believe that this proposed automated method is applicable as a substitute of human ears for sizable classes of evaluation problems.
Subphonemic mismatches and perceiving by segments or by syllables
Authors: Johnson Keith
Published: October 2005

Reaction time measurements to tokens composed by means of categorial phonetic mismatches have indicated that information which is distributed throughout a syllable is used in making segment identification judgments [Whalen, Percept. Psychophys. 35, 49–64 (1984)]. This raises the possibility that speech perception proceeds from syllable perception “backwards” to segment identification. The mismatches created using tokens similar to those of Whalen's experiment 1 are not identical in their demands on the listener. Thus the mismatched fricative may be closer or further from the expected fricative depending on the degree of coarticulatory rounding. For instance, if the formant transitions lead the subject to expect a palatal fricative, the fricative noise of [us] will be closer to the expected fricative than that of [is]. It is proposed in this paper that the perception by syllables approach would predict that subjects would recover from the mismatch more quickly when the actual fricative noise is close to the expected fricative while the perception by segments would predict the opposite. The results of an experiment testing these hypotheses support the perception by segments approach. The interaction (in an ANOVA) between actual fricative and original vocalic context of the fricative was significant [F(3,42) = 3.52, p < 0.05] in the appropriate direction.
Perception of stress contrasts by the hearing impaired
Authors: Rubin-Spitz Judith , McGarr Nancy S. , Youdelman Karen
Published: October 2005

Waveform manipulation and LPC resynthesis techniques were used to develop test stimuli in which the F0, amplitude, and durational cues for first versus second syllable stress were systematically and independently controlled. Both normal and exaggerated stress contrasts were represented in the test corpus. Twenty severely to profoundly hearing-impaired students served as subjects for a perceptual experiment and were asked to indicate whether the stress was on the first or second syllable of each two-syllable test item (data were also collected on a group of normal hearing controls). Results revealed that the hearing-impaired listeners were able to correctly identify the stressed syllable only for stimuli in which the amplitude cue was available. That is, they scored below chance on stimuli in which the durational and/or F0 cues were present but where the amplitude cue had been neutralized. Exaggerating the amplitude cue for stress resulted in higher identification scores. Exaggerating F0 and/or durational cues did not improve performance. The implications of these results will be discussed. [Work supported by PHS grant No. NS17764-05 to the Graduate Center, C.U.N.Y.]
Children's discrimination of CV syllables differing in VOT: II
Authors: Elliott Lois L.
Published: October 2005

Normal children aged 6–8 and 8–11 years and young adults were tested on a same-different task using a [ba]-[pa] continuum with VOT's that ranged from 0–35 ms. Data were analyzed in terms of d′, using tables of Kaplan et al. [Behav. Res. Methods Instrum. 10, 1978 (1978)]. The adult subjects' discrimination was better than the children's for pairs of CV's differing by 10 and 20 ms; this supported results previously obtained using a different test paradigm and different subjects (Elliott et al., Child Der. (in press)]. Calculated k values indicated that adults demonstrated greater bias than children. Results were interpreted as indicating true age differences in discriminating these stimuli. [Work supported, in part, by NINCDS and NIH.]
Some aspects of the statistical energy analysis—SEA
Authors: Maidanik G.
Published: October 2005

Recently SEA has been gaining recognition as a bonafide noise control tool. A few of its past critics are not only rescinding, but are joining in its advocacy. Naturally SEA has, since its inception and development in the early 1960s, been improved, extended, and even reformulated. The advantages and limitations of its use are now more clearly understood and defined. In this paper some recounting of SEA is presented, removal of a few early limitations reviewed, and a few items for further consideration are discussed.
Energy accountancy concept
Authors: Richards E. J.
Published: October 2005

The study of noise sources on an engineering level calls for a good working knowledge of machinery mechanics, structural vibration, energy absorption or damping, and fluid mechanics as well as acoustics. The acoustic output is effectively a remainder term associated with inefficiency in one or many of these processes. Due to uncertainties existing between various subjects, new mathematical techniques used to obtain exact calculations of cause and effect cannot be exploited. It is the contention of this paper that mathematical methods dealing with special simplified cases should be associated with a systems analysis which is strong in concept and universality rather than in detail. The energy accountancy system is intended to be such, not specifying the details of how each component is studied, but how they are linked together. It is based on the idea that noise occurs only when compressibility occurs in the flow near to a body or flow, and that in noisy machines this can be equated to the high-acceleration portions of the motion. It uses an energy balance approach which allows one to ignore the exact modal vibratory front except where necessary, as well as ignoring coupling and directional effects. It looks at a single machine impact and adds a repetition effect, because a large number of high-frequency modes arise from the sharp accelerations which can only be dealt with statistically. Thus while SEA can and is used in some of the terms in the “energy accountancy equation,” the method is not dependent upon this.
Power flow through dynamically coupled subsystems
Authors: Brown Richard N.
Published: October 2005

In practical applications of the statistical energy analysis (SEA) or other power flow techniques, the calculation of the power flow between subsystems is of fundamental concern. When the subsystems are connected by multimodal structures, for instance, vibration mounts, values for the coupling loss factors as found in the literature are not suitable. In this paper we present a general theoretical framework for calculating the coupling loss factor when the coupler is numerically modeled by classical dynamics, notably by means of a finite element analysis (FEA). After presenting some pertinent results and definitions from SEA, FEA, and structural dynamical theory, an expression for the “coupling” impedance matrix, which relates the forces and free velocities at nodes along the subsystem/coupler boundaries, is derived. This impedance is formed from the admittance matrix of the coupler, calculated using FEA, and input impedances for the systems, analytically calculated. From the coupling impedance, expressions for the coupling loss factors are derived for the cases where the SEA subsystems are plates or beams and include flexural, longitudinal, and torsional forms of energy transmission. Finally, numerical examples are given for the application of this technique to structures consisting of two beams or plates connected by a multimodal coupler.
Energy flow and acoustic radiation for a fluid-loaded panel
Authors: Maillet P. L. , Cuschieri J. M. , Dunn S. E.
Published: October 2005

The results of studying the energy flow in and the acoustic radiation from a fluid-loaded panel using the energy accountancy method are reported. Energy components are defined for the excitation, structural dissipation, and acoustic radiation in terms of excitation parameters, the structure, the medium, and the fluid-structural coupling factor. Using the energy components, a balance equation is written. The method is evaluated for a point excited flat plate with clamped edges. The plate is subjected to air loading on one side and water loading on the other. A comparison is made between the predicted energy balance terms using the accountancy approach and experimentally measured results. The energy balance equation provided good results between 50 Hz and 12 kHz.
The modified Cagniard method and its application to the calculation of impulsive acoustic waves in a layered fluid
Authors: de Hoop Adrianus T.
Published: October 2005

Through a specific combination of a Laplace transformation with respect to time and Fourier transformations with respect to the “horizontal” spatial coordinates (the modified Cagniard method), explicit space-time expressions are obtained for the acoustic pressure and the particle velocity of an acoustic wave generated by an impulsive source in a “vertically” layered fluid. The expressions have the form of a time convolution of the wave shape of the source strength and a properly defined acoustic space-time Green's function. The representation for the Green's function is either an explicit algebraic one (for two-dimensional wave motion), or a finite integral of an algebraic expression (for three-dimensional wave motion). The general aspects of the method are presented and examples are discussed.
A comparison of normal mode, WKBJ, and ray theory predictions with measured pulse arrival patterns at the 300-km range
Authors: Romm Joseph J. , Masters Guy
Published: October 2005

Precise measurements of the impulse response of the ocean sound channel were made during the 1983 reciprocal acoustic transmission experiment at the 300-km range with source and receiver near the sound channel axis. The transmitted pulse was centered at 400 Hz and had approximately 10-ms resolution. Using measured XBT data and the technique of inverse theory, a range-dependent sound-speed field had previously been computed such that the measurements were consistent with the prediction of range-dependent ray theory (Howe et al., 1986). Predictions made for the range-averaged sound-speed profile using normal mode theory, the WKBJ approximation, and ray theory were found to be in excellent agreement (though a small discrepancy with the measurements remained, since range dependence was neglected). By allowing for range dependence through the use of adiabatic normal mode theory, this discrepancy was largely removed. We conclude that ray theory or WKBJ theory is an adequate prediction technique at 400 Hz in this geometry. [Work supported by NSF and ONR.]
A Hamiltonian approach to horizontal ray tracing in adiabatic normal mode modeling
Authors: Pitre Richard , Gragg Robert F.
Published: October 2005

Recently, Arnold and Felsen [J. Acoust. Soc. Am. Suppl. 1 78, S23 (1985)] and Kamel and Felsen [J. Acoust. Soc. Am. 73, 1120–1130 (1983)] have extended adiabatic mode theory to treat trapped-to-leaky mode transitions. These papers introduce a new spectral decomposition theory by identifying an adiabatic invariant for the discrete modes and then analytically continuing it to the whole complex horizontal wavenumber plane. The theory is formulated for waveguides composed of isovelocity layers with slowly varying thicknesses. Our work extends Felsen and Kamel's definition of the adiabatic invariant to arbitrary vertical sound-speed profiles by applying Milne's amplitude and phase representation of Sturm-Liouville solutions [Phys. Rev. 35, 865–867 (1930)]. This adiabatic invariant was employed to trace horizontal ray paths. In our approach, the horizontal ray paths are identified as the trajectories of a two-dimensional dynamical system in which the Hamiltonian is the adiabatic invariant and the momentum is the horizontal wavenumber. By analytically continuing that Hamiltonian, ray paths can be defined for modes which pass through cutoff. A numerical model has been developed for horizontal ray tracing in waveguides with arbitrary vertical structure and slow but otherwise arbitrary horizontal variations. With this model the horizontal refraction of adiabatic modes for some waveguides of interest is examined.
Adaptive acoustical tomography
Authors: Mansfield Adams William
Published: October 2005

Acoustical and seismological tomography differ significantly from tomography as practiced using energy which travels in straight lines, such as with x rays. The complication of refraction introduces a large number of new variables. This has the indirect result of reducing the variance for most of the variables. Usually overlooked, conveniently, is that the variances are no longer equal, i.e., the variance of the variances is not a constant. A novel approach to experimental design, adaptive in nature, is revealed. This permits, but does not require, the investigator to proceed adaptively in the taking of observations. Extreme canonical examples are illustrated.
Appraisal of an active noise control scheme in a 1-D waveguide
Authors: Silcox Richard J.
Published: October 2005

An active noise control scheme was implemented for broadband random noise in a 1-D reverberant environment. A frequency domain approach was used with an adaptive algorithm to provide noise control in a 50- to 400-Hz band using conventional microphones and loudspeaker A 256-tap transversal filter provided the necessary narrow-hand control for the system to define and maintain a 15- to 30-dB reduction in transmitted power. Measured system frequency response functions defining both performance and control functions are related to primary system response functions, i.e., propagation parameters, transducer characteristics, and termination conditions. Results indicate reduced noise control near resonance in this lightly damped system. The use of a simple omnidirectional source, even with highly directional sensors, yields strong resonances when highly reflective termination conditions exist. In addition, the presence of control source feedback weakens the discrete frequency nature of the controller frequency response. The effect of finite source impedance is also examined. Previous work has considered only constant volume velocity source in the development of active control schemes. Analytical results show the impact on system control functions and experimental results indicate that source impedance effects are important near speaker resonance.
An active noise reduction system for flying helmets and ear protectors
Authors: Rood Graham M.
Published: October 2005

One of the major limitations of flying helmets or hearing protectors is their inability to provide adequate acoustic attenuation at low frequencies. In both fixed and rotary wing aircraft and in armored fighting vehicles, the high levels of low-frequency noise cannot be suitably reduced by the use of passive hearing protectors alone. The development of active noise reduction for use in such protectors allows much of this limitation to be removed. This paper describes the development and testing, both in the laboratory and during flight trials, of an active noise reduction system fitted to the current UK military force's flying helmet. The system was developed under a Ministry of Defence contract by the Institute of Sound and Vibration Research at Southampton University, and assessed by the RAE in a number of helicopters and during high-speed, low-level flight in a strike aircraft. The results from both the laboratory testing and from the flight trials are discussed, and the operational and medical advantages of using this type of active system summarized, both for military and civilian use.
Prediction of optimal active noise controllers using boundary element methods
Authors: Bernhard Robert J. , Mollo Chris G.
Published: October 2005

Boundary element methods have been used as numerical approximations of boundary integral equations to predict the sound fields in complicated three-dimensional interior and exterior spaces. In this investigation the methods have been investigated for their capability to find and evaluate the optimal active noise controller for complex acoustic geometries. The boundary element methods are utilized to enforce the specified boundary conditions while solving for the optimal secondary source distribution necessary to minimize a giver performance equation. For the current work the performance equation is a weighted sum of mean-square pressures at specified points. Thus the method can predict the optimal controller required to achieve local control using a single point, or to achieve a space averaged reduction by using a distribution of points. The formulation allows multiple secondary sources and either pressure, velocity, or normal specific acoustic impedance boundary conditions. The optimal secondary source's strengths and the insertion loss performance of several active controller configurations are illustrated.