bluuebunny/crossref_metadata_2025 huggingface dataset:
Impulse responses and radiation impedances of cylindrical vibrators using wave-vector-time domain (k-t) methods
Authors: Stepanishen Peter R. , Ebenezer D. D.
Published: October 2005

A wave-vector-time domain (k-t) method is presented to evaluate the harmonic and time-dependent loading on a cylindrical shell that is vibrating with a specified spatial and time-dependent velocity. The method is based on utilizing a specified time-dependent modal expansion for the radial velocity of the vibrator. The acoustic loading on the vibrator is also expressed as a modal expansion in which each coefficient is a summation of convolution integrals of the modal velocity coefficients with mode-dependent radiation impulse responses. In contrast to an earlier work [D. D. Ebenezer and Peter R. Stepanishen, J. Acoust. Soc. Am. 81, 854–860 (1987)], the radiation impulse responses are evaluated using a (k-t) method based on a time-dependent Green's function for a baffled cylindrical vibrator. A comparison of the new method with the earlier method is presented along with numerical results for various axial mode shapes and circumferential mode numbers. [Work supported by ONR.]
A diffuse field sonar tank for measurement of projector sound power level and directivity index
Authors: Lubman David
Published: October 2005

Though widely used in air acoustics, the diffuse field technique has not been extensively used in underwater acoustics. This paper describes a small (2 m3) diffuse field sonar tank facility intended for production testing of the steady-state sound-power level and directivity index of wideband sonar projectors. When evaluated by standards used in air acoustics to ensure adequate diffuse field measurement accuracy, this compact and inexpensive facility appears to be qualified for the determination of band-averaged sound-power level over a frequency range exceeding five octaves (from about 4 to 160 kHz). When compared with a free-field measurement facility, the practical advantages of this approach include large savings in cost and space. Long transducer rise times are permitted since measurements are steady state. Moreover, the implicit spatial integration of radiation patterns makes results of diffuse field testing more appropriate than free-field testing for some applications.
Development of higher-order Zernike polynomials employed in the analysis of acoustic sensing antennas
Authors: Adekola S. Adeniyi
Published: October 2005

Zernike circle polynomials have been found to be invaluable analytical tools for investigating the characteristics of acoustic sensing antennas [S. A. Adekola, Acustica 47, 114–131 (1981)]. New analytical results obtained from an investigation of the generalized Rnm(r) Zernike circle polynomials, summarized in the above-mentioned reference, are of higher degrees of accuracy than those presently available in the literature. Indeed, these polynomials are generated to 30° and 29° of freedom describing the even- and odd-numbered modes, respectively; and the field radiated by the acoustic sensing antenna, which is then expressed in terms of the polynomials, is evaluated through direct routine integration. Some of the new analytical results evaluated and numerically computed are here displayed in graphical forms for completeness. Also, illustrative examples given show not only how the Zernike polynomials can be employed in the interpretations of the radiation characteristics of the acoustic echosounding systems, but also how use can be made of the polynomials to describe significantly tapered source distributions essential in successful acoustic remote sensing of the lower atmosphere. [Work supported by Lagos State University, Badagry Expressway, Ojo, Lagos, through a Visiting Professorial Appointment.]
Temporal aspects of the auditory system: Gap detection and the modulation transfer function
Authors: Forest T. G. , Green David M.
Published: October 2005

A three-stage model consisting of a bandpass filter followed by a nonlinear element and a low-pass filter is often used to model temporal processing in the auditory system [e.g., N. F. Viemeister, J. Acoust. Soc. Am. 66, 1364–1380 (1979)]. Two experimental paradigms, the detection of gaps in noise and of sinusoidal modulation of noise, have been used to estimate the parameters of this model. A simulation of this model was used to “listen” to waveforms in gap detection and modulation detection experiments. Responses of the simulator were based on statistical calculations at the output of the low-pass filter stage of the model. Our decision statistic was the maximum-to-minimum ratio of the waveform from the low-pass filter. Good agreement between the data produced by normal-hearing human observers and the simulator occurred when the bandwidth of the initial filter was 4000 Hz and the time constant of the low-pass stage was 3 ms. [Research supported by NSF.]
Discrimination of envelope phase disparity
Authors: Wakefield Gregory H. , Edwards Brent
Published: October 2005

The sensitivity of the auditory system to changes in ongoing temporal structures across different spectral regions is investigated using sinusoidally amplitude-modulated (SAM) carriers. Previous results are extended for the detection of phase disparity between the envelopes at two carriers [J. Acoust. Soc. Am. Suppl. 1 81, S34 (1987)] to the discrimination of such disparity when a nonzero relative phase offset between the envelopes is present. For a standard SAM two-carrier complex with a relative phase offset of φ, the relative phase disparity of the comparison φ + Δφ was adjusted to measure discrimination threshold as a function of φ. The results show that Δφ varied by more than a factor of 2 as φ ranges from 0–180 deg. Modulation frequency (8–64 Hz) or carrier separation (500 Hz, 1–4 kHz) has little effect on this phase dependence. When expressed as a change in envelope correlation (ρ), however, we find that Δρ is constant as a function of ρ. Additional experiments using harmonic complexes for envelopes support the hypothesis that envelope correlation is the primary determinant of performance. A neural-based coincidence-detector model of envelope correlation is proposed to account for these findings. [Research supported by AFOSR.]
Evidence for cross-channel processing in detection of envelope phase disparity
Authors: Strickland Elizabeth A. , Viemeister Neal F. , Fantini Deborah A. , Garrison Margery A.
Published: October 2005

The sensitivity to the correlation between the envelopes of two sinusoidally amplitude-modulated sinusoids was investigated in two experiments. Correlation was manipulated by varying the phase disparity between the envelopes. In each experiment, two carriers (either 1 and 2 kHz or 1 and 3.2 kHz) were modulated at the same frequency (8 or 128 Hz) and added together either in phase or, on signal intervals, with an adaptively determined phase disparity. The starting phase of the envelope of the lower frequency carrier was randomized across trials. In the first experiment, the lower frequency carrier was presented at 65 dB SPL and the higher frequency carrier was presented at levels from 65 to 25 dB SPL. No decrement in threshold was noted until the higher frequency signal was 20–40 dB below the lower frequency signal. In the second experiment, the lower frequency carrier was presented at 65 dB SPL to one ear, and the higher frequency carrier was presented at the same level to the other ear. Thresholds were slightly above those obtained monotically. These results suggest that subjects are able to use information across frequency channels to detect envelope phase disparity and do not need to use envelope interactions occurring within a single channel in performing this task. [Work supported by NINCDS Grands NS12125 and NS07889.]
The acoustic stochastic inverse problem
Authors: Ewart Terry E.
Published: October 2005

It has been decided that the statistics of the phase, complex amplitude, and intensity fluctuations that result from waves propagating in media with random index of refraction inhomogeneities can be predicted. In fact, the emphasis has been refocused from the development of the theory to a requirement to understand more fully the statistics of the ocean medium. An exciting result of the success of the theoretical predictions is the possibility of inverting acoustic measurements to obtain the statistics of the ocean media. In the stochastic inverse problem, one obtains coefficients of statistical models of the acoustic index or refraction using optimization techniques. These techniques require that deterministic signals, e.g., tides, be treated formally in a different manner from stochastic signals, e.g., internal waves. Examples of the inverse problem will be given from ocean and numerical experiments. Models of the oceanic internal wave, finestructure, and tidal processes, obtained from measured acoustic fluctuations, are compared with those measured. A discussion of the future expectations including 3-D considerations will be presented.
A discussion of three-dimensional wave propagation effects
Authors: Lee Ding , Scully-Power Paul D. , Botseas George , Siegmann William L.
Published: October 2005

In most environments, the prediction of acoustic wave propagation is carried out with two-dimensional propagation models. This should NOT mislead anyone to believe that three-dimensional models are not needed. In this paper, sound-speed information from eddy models is used to examine three-dimensional effects. This examination was carried out employing the IFD (two-dimensional) and the FOR3D (three-dimensional) models. A set of numerical results from these two models will be discussed. Interesting findings will be reported. These findings provide stimulation for continued three-dimensional model development.
A graphic representation of acoustics using ray tracing
Authors: Thompson Philip R.
Published: October 2005

This research details the development of a computer model that displays the propagation and reverberation of sound within an architectural enclosure. The visual rendering of acoustics is done by simulating sound fields using graphical ray methods. Unlike many other acoustical ray-tracing methods, in this concept, rays are emitted from a receiving point and trace the paths to the sound source so as to produce a picture of what a listener hears. This three-dimensional recursive ray-tracing algorithm has been combined with an acoustical model, utilizing specular and first-order diffuse reflections, to produce an image (or series of images) depicting the locations of real and virtual sources, the time delays in which they arrive, and their approximate energy levels. The free-path lengths of the rays are sorted into frames that produce an animation sequence demonstrating the propagation of a sound field towards the listener. The sources' emitted energy and subsequent attenuations along a ray's path result in an intensity, which is then depicted as a color at a pixel on the computer's monitor to compose an image. While the program is meant to run on most any machine, a workstation is preferred since a relatively large amount of computation (involving hundreds of thousands of rays) is required. The purpose of this analysis tool is to elucidate and verify our perceptions of room acoustics through the most readily understandable visual means. [Work supported by MIT Project Athena through the Computer Resource Lab in the School of Architecture and Planning and by the Robert B. Newman Memorial Fellowship.]
RASTI for speech isolation evaluation
Authors: Hojbjerg Klaus
Published: October 2005

In open offices, there is a demand for speech isolation. As it becomes more important to avoid surveillance from outside conference rooms, a reliable method of evaluating the speech security becomes necessary. Research has shown that transmission loss or the articulation index method does not give enough information. This is because speech intelligibility, or rather the lack of speech intelligibility, depends on more than the signal-to-noise ratio. The RASTI method takes into account the effects of reflections, reverberation, and noise. By using the low end of the RASTI scale, the lack of speech intelligibility can be determined. This paper describes the measurement, using elevated signal levels, as well as measurement results.
On the acoustic excitations of an elongated elastic solid
Authors: Williams Kevin L. , Todoroff Douglas G. , Hackman Roger H.
Published: October 2005

Previously, an analysis of the acoustically coupled, axisymmetric, elastic excitations of a fluid-loaded, prolate spheroidal target was presented [D. H. Trivett, G. S. Sammelmann, and R. H. Hackman, J. Acoust. Soc. Am. Suppl. 1 81, S13 (1987)]. At this meeting, a controversy arose as to the phase speed of the elastic excitations. Here, direct experimental and theoretical evidence is presented that supports our contention that the velocity of propagation of the elastic waves that propagate on an elongated solid target at low frequencies is essentially the bar speed.
Development of an undergraduate experimental acoustics center
Authors: Jarzynski Jacek
Published: October 2005

A center for experimental acoustics and ultrasonics was recently developed for use in a senior-level laboratory course that is required for the BSME degree at Georgia Tech. The center is a distinct facility with permanently assigned instrumentation and equipment available for hands-on performance of experiments and laboratory projects by students; possible experiments are related to current course offerings and on-going research activities within the School of Mechanical Engineering. The experiments that have been developed for the center allow the students to study some of the basic features of wave motion and to learn about some typical applications of acoustics. Representative experimental tasks that can be carried out within the center include: measurements of pressure levels and frequency spectra of sound from environmental sources; application of the ultrasound to nondestructive testing of materials; and measurement of fluid flow using an ultrasound Doppler technique. To the fullest extent possible, the instrumentation of the center has been chosen such that the students will be introduced to the latest state-of-the-art data acquisition, storage, and processing techniques. The objectives of this type of laboratory instruction are outlined, with some sample student comments. The illustrated talk will be accompanied by several live demonstrations of the experiments that students carry out within the center.
ARMA speech analysis as a means for studying articulation
Authors: Boves Louis , de Veth Johan
Published: October 2005

It is well documented that linear prediction analysis will only allow prediction coefficients to relate to vocal tract shapes under very special conditions, and that the problems in interpreting LP analysis results in articulatory terms are due to the simplifying assumptions underlying the analysis model. Usually, the assumption that the system is all pole [or auto-regressive (AR)] is thought to be the most important simplification. Therefore, it is hoped that the use of the less restrictive pole-zero [or auto-regressive moving average (ARMA)] analysis model will offer greater opportunities for an articulatory interpretation. A detailed study of a number of different ARMA analysis implementations has shown that the all-pole assumption may not be the most important restriction on the articulatory interpretation of the analysis results; the assumption of Gaussian excitation may prove to be at least equally impeding. [Research supported by the Foundation for Speech Technology, funded by SPIN.]
Invariant acoustic cues in stop consonants: A cross-language study using the Wignet distribution
Authors: Garudadri H. , Gilbert J. H. V. , Benguerel A.-P. , Beddoes M. P.
Published: October 2005

Acoustic invariance was investigated in 200 unaspirated stops from English, Telugu, and French, using the Wigner distribution. The bilabials were characterized either by absence of any noticeable burst, or by an energy concentration in the low-frequency region during the burst, when present. The burst of alveolars was mostly diffuse in nature, but sometimes became compact, especially when it was followed by a back vowel. The burst of velars had a compact spectrum centered near F2 of the vowel, when followed by back vowels, but was most often diffuse when followed by a front vowel. The burst of all dental from Telugu was found to be diffuse. When the burst of alveolars was compact, or that of velars diffuse, the duration from the start of the burst to the start of the vowel and F2 of the vowel were found to be reliable cues for place of articulation. The diffuse nature of bilabials and alveolars, reported by Lahiri et al. [J. Acoust. Soc. Am. 76, 391–404 (1984)] and by the authors referenced therein, can be partly attributed to the temporal averaging of the burst spectrum and the diffuse vowel spectrum, which is inevitable in Fourier-type analyses [Garudadri et al., J. Acoust. Soc. Am. Suppl. 1 79, S94 (1986)]. Unlike in other investigations, no apparent pattern could be found for the tokens wrongly classified. [Work supported, in part, by NSERC, Canada.]
Testing children and the aged with computerized audiometry
Authors: Picard Michel , Ilecki Henry J. , Baxter James D.
Published: October 2005

Using computerized audiometry in a group of noise-exposed workers has been shown to be a reliable and valid means of assessing hearing sensitivity [M. Picard et al., Proceedings: Computer Applications to Canadian Health, pp. 127–130 (1987)]. Based on this study, an investigation was conducted to similarly determine the feasibility of implementing like-methodology in two clinical populations typically regarded as “difficult to test” (viz., geriatric and pediatric groups). Children were aged 8 to 12; seniors, 65–81 years. Reliability was assessed by relating air- and bone-conduction thresholds where there was no evidence of middle-ear pathology. Validity was measured by comparing air-conduction pure-tone thresholds (0.5, 1, 2, 3, 4, and 6 kHz) obtained under two test modalities, computerized audiometry and conventional testing performed by a trained examiner. Both procedures followed ANSI S3.21-1978. Findings in both age groups were comparable to those reported in the previous study. Coefficients of reliability remained equally high across frequencies regardless of testing method. As well, high correlations between conventional and computerized audiometry were found. Complementing this observation, a PCA supported the criterion-related validity of computerized audiometry. [Work supported by MRC.]
Detecting dynamic frequency shifts of simple and complex tones
Authors: Carlyon Robert P. , Stubbs Richard J.
Published: October 2005

Thresholds for the detection of frequency modulation by a single cycle of a 12-Hz sinusoid were obtained for carriers that were simple or complex tones. Two phases of modulator were used, producing percepts of either upward or downward frequency glides during the middle of the stimulus. To ensure that subjects based detection on frequency movement, both the phase of the modulating sinusoid and the starting frequencies were randomized from presentation to presentation. For pure-tone carriers, thresholds (= minimum detectable frequency swing/carrier frequency) were constant for carrier frequencies of 800–2400 Hz and increased below 800 Hz. Thresholds were lower for modulations producing the percept of an upward glide than for “downward” glides. This difference was much greater at low carrier frequencies. When the carrier was a three-component harmonic complex, thresholds were almost independent of its center frequency but increased with decreasing fundamental frequency. Thresholds for harmonic and for inharmonic complexes were compared. These results have implications for the recording of voice pitch (e.g., in signal-processing hearing aids).
Central spectrum models of dichotic edge pitches
Authors: Hartmann W. M.
Published: October 2005

The central spectrum model of Raatgever and Bilsen [J. Acoust. Soc. Am. 80, 429–441 (1986)] successfully accounts for a number of dichotic-noise pitch effects and for the lateralization of the pitches. The model is, however, physiologically implausible. An improved model, based upon a neural delay line having coincidence cells that obey the statistical rules of Stem and Colburn [J. Acoust. Soc. Am. 64, 127–140 (1978)], retains the predictive power of the Raatgever-Bilsen model. It also successfully predicts the results of new experiments where the Raatgever-Bilsen model fails. New experiments measure the relative strengths of Huggins pitches HP + and HP −, the lateralization of the binaural-edge pitch (BEP), the spectral density dependence of the BEP, and the strength of generalized BEP, where the interaural phase shift may have any magnitude. [Work supported by the NIH.]
Infants' extraction of pitch from inharmonic tonal complexes
Authors: Clarkson Marsha G. , Clifton Rachel K.
Published: October 2005

Previous research has demonstrated the infant's ability to extract the virtual pitch of harmonic tonal complexes. In the present study, we used a conditioning procedure to assess 7-month-old infants' perception of pitch for inharmonic tonal complexes. Infants initially discriminated a pitch change from 160 to 200 Hz or vice versa in harmonic tonal complexes, which contained eight consecutive harmonies but no fundamental frequency. After infants discriminated the harmonic complexes, each partial's frequency was shifted upward by 30 Hz to render the sounds inharmonic. All infants who discriminated the inharmonic complexes proceeded to a categorization stage in which a randomized sequence of three inharmonic complexes (containing six partials) comprised each pitch category so that the spectral characteristics of the tokens changed continuously while the pitch category changed only on change trials. The harmonic complexes were discriminated by 19 of 22 infants, while 16 infants successfully discriminated the inharmonic sounds. However, only 11 infants categorized spectrally different sounds consistently with their pitches, suggesting that infants had more difficulty extracting pitch from inharmonic complexes than from harmonic ones presented in previous research. [Work supported by NSF and NICHD.]
Low-frequency noise and bubble plume oscillations
Authors: Carey W. M. , Fitzgerald J. W.
Published: October 2005

The evidence for a wind dependence in low-frequency ambient noise (< 200 Hz) is sparse due to the corrupting influences of the radiated sound from ships. Measurements made with hydrophones below critical depth, in sparse shipped basins, or at high sea states show indications of two distinct regimes associated with the occurrence of breaking waves and a variation of root-mean-square pressure with the square of the local wind speed. In a previous paper [W. Carey, J. Acoust. Soc. Am. Suppl. 1 78, S1 (1985)], the generation of sound by wave turbulence interaction at low sea states (WS < 10 m/s) and collective bubble oscillations driven by turbulence at high sea states was proposed (WS > 10 m/s). This paper shows that recently observed bubble clouds that penetrate to tens of meters below the surface of the sea result in regions of low sonic velocity described by Wood's treatment of air-bubble water mixtures, but with a density close to that of water. It is shown that such a region can be treated as a flexible body with mixture speed and density. The radiation from such a body would be monopole and dipole in nature; but due to the proximity of the sea surface, only the resulting dipole source would be of importance. These regions could also have a resonant characteristic, and when driven by the turbulence result in sufficient radiated sound. These results are similar to and consistent with the analysis by A. Prosperetti [Nato AS1, “Natural Mechanisms of Surface Generated Noise in the Ocean” (1987)1.
Transmission loss studies of muffler and duct systems by the boundary element method
Authors: Cheng C. Y. R. , Seybert A. F.
Published: October 2005

The present paper deals with transmission loss studies of acoustical muffler and duct systems using the boundary element method (BEM). Results obtained by using this approach are presented for both axisymmetric and fully three-dimensional models. The transmission loss of a muffler element can be computed using the transfer matrix approach. This method is ideally suited for the BEM because only the values of the unknowns at the inlet and outlet boundaries are required. The BEM values are compared to theoretical predictions and other available data such as finite clement data, and good agreement is obtained within a broad frequency range. It can be concluded that the BEM should give reliable estimates of transmission loss in practical situations. [Work supported in part by ASHRAE.]
The application of the cepstrum technique to the separation of the input signal and the structure response
Authors: Wu Mei Q. , Crocker Malcolm I.
Published: October 2005

In this paper, an iterative algorithm for the cepstrum is presented. The improved cepstrum technique is applied to separate the input signal from the response of a structure. It is proved that if the input signal is an impulse and the frequency response of the structure is bandlimited, then the frequency response function of the structure can be determined from the output of the structure without any preknowledge of the input signal. Furthermore, the spectrum of the input signal can be determined from the structure response function and the output spectrum. Some computer simulated experiments have been conducted. The experimental results show that the frequency response function recovered by using the iterative algorithm converges to the correct response function, and for about 30 times of iteration the recovered response function is very similar to the correct response function. [Work supported by IBM, Austin, TX.]
Subjective evaluation of a test earmuff exhibiting flat attenuation with nonlinear characteristies
Authors: Peters June L. , Royster Larry H. , Royster Julia Doswell , Pearson Richard G.
Published: October 2005

Subjective user responses were obtained for a test earmuff exhibiting approximately flat attenuation of about 25 dB from 500–8000 Hz for SPLs less than 115 dB, with attenuation increasing nonlinearly up to about 10 dB at higher SPLs. The study population consisted of police officers performing annual gunfire requalification, in which officers execute required target shots during two sequential relay exercises. In the first phase of the study, subjects wore either their regular hearing protector or the test muff during the first relay, then the alternate protector for the second relay (with order counterbalanced). In the second phase, additional subjects wore either a control earmuff that was identical to the test earmuff except for the nonlinear mechanism (also eliminating the flat response characteristics) or the test earmuff. Subjects completed questionnaires concerning their opinions of the hearing protectors worn during both phases. Preliminary results indicate a significant difference in favor of the test earmuff in three areas: speech understanding, perceived level of protection, and comfort.
Regional cerebral blood flow for singers and nonsingers while speaking, singing, and humming a rote passage
Authors: Formby C. , Thomas R. G.
Published: October 2005

Two groups of singers (n = 12,13) and a group of nonsingers (n = 12) each produced the national anthem by (1) speaking and (2) singing the words, and by (3) humming the melody. Regional cerebral blood flow (rCBF) was measured at rest and during each phonation task from seven areas in each hemisphere by the 133-Xe-inhalation method. Global, intrahemisphere, and interhemisphere rCBF were generally similar across phonation tasks and did not yield appreciable differences among the nonsingers and the singers. From these rCBF data, it was concluded that: (1) the normal production of a familiar passage by speaking, singing, or humming requires the interaction of both cerebral hemispheres more or less equally and (2) these tasks are relatively independent of musical training. [Research supported by NIH.]
Note coupling in Caribbean steel drums
Authors: Hampton D. Scott , Alexis Clifford , Rossing Thomas D.
Published: October 2005

The acoustical behavior of Caribbean steel drums is characterized by strong coupling between the different note areas [T. D. Rossing, D. S. Hampton, and J. Boverman, J. Acoust. Soc. Am. Suppl. 1 80, S102 (1986)]. This makes it difficult to maintain tuning stability, especially in the harmonics of the various notes. An improved new design makes use of double grooves to separate the various note areas. By means of holographic interferometry, the note coupling was compared in two comparable double second (alto) drums, one with single grooves and one with double grooves. The drum with double grooves exhibits less internote coupling at high amplitudes, and better tuning stability.
Cue salience in the perception of a stop voicing contrast by hearing and hearing-impaired children
Authors: Hazan Valerie , Holden-Pitt Lisa , Revoile Sally , Edward Donna , Droge Janet
Published: October 2005

The contribution of two acoustic cues, voice onset time (VOT) and vowel onset frequencies, to the perception of a TAD-DAD voicing contrast was examined for ten normal-heating and six hearing-impaired children (3 FA: 50–95 dB HL) between 6 and 12 years of age. Both natural stimuli and high-quality synthesized copies of these stimuli were used to create five VOT continua (three natural and two synthetic), in which the vowel onset cue was either appropriate or conflicting. An adaptive identification procedure was used to determine phoneme boundaries under these five test conditions. Changes in spectral characteristics at vowel onset did produce significant shifts in phoneme boundary for the normal-hearing group, but not for the hearing-impaired group. Similar phoneme boundaries were obtained by the normal-hearing group for natural continua and their synthetic counterparts. However, higher values of standard error were observed for synthetic stimuli for both listener groups.
Vowel discrimination using different frequency-to-electrode maps in cochlear implant users
Authors: Svirsky Mario A. , Cullen John K. , Ragland Howard P. , Walker Cedric F.
Published: October 2005

The Nucleus WSP-III is a speech processor for cochlear implants that stimulates two electrodes per fundamental period. The electrodes stimulated are chosen based on the first two formant frequencies: lower frequency formants excite electrodes that are more apical. This gives users a cue to discriminate vowels with different formants. But, if two particular vowels had second formants close enough to fall within the frequency band of some electrode, the user would miss that cue—those two vowels would “behave” as if they had exactly the same second formant. Cochlear implant users were tested under three frequency-to-electrode maps: the standard map (used by most Nucleus implant patients), the modified standard map (proposed by P. Blamey, private communication, 1987) and a new map designed to maximize contrast between different vowels. A discrimination test was used where subjects were not asked to identify the sounds they heard, but only to indicate whether the different word in a set of three was in the initial or final position. This tends to minimize learning effects due to prior experience with a given map. Also, if cochlear implant users can discriminate different phonemes, they could conceivably be trained to identify them correctly, whereas lack of discrimination makes identification impossible. [Work supported by Cochlear Corp. and Sigma Xi.]
A new technique of acoustic mode filtering in shallow sea
Authors: Chouhan Harish M. , Artand G. V.
Published: October 2005

A new technique of filtering acoustic normal modes, which overcomes many of the drawbacks of the earlier techniques, is presented in this paper. It is based on the fact that each normal mode in an isovelocity channel comprises a pair of plane waves with characteristic directions of propagation symmetrically disposed with respect to the channel axis. A vertical array of equispaced hydrophones is shaded so as to steer nulls in the directional response of the array along the directions of arrival of plane waves corresponding to the unwanted modes. All the shading coefficients are real, leading to simplicity in the hardware realization of the processor. The shading coefficients are invariant to a shift in the position of the array. The array need not span the entire depth of the ocean. Efficient filtering is possible even when the eigenfunctions of the modes have a significant penetration into the bottom. [Work supported by DOE, Government of India.]
Phase recovery and calibration with underwater acoustic arrays
Authors: Cohen Nathan
Published: October 2005

Underwater acoustic imaging provides unique challenges because of the nature of the medium and difficulties in accurate measurement of array observables. Baseline uncertainties and loss of phase locking make phase observables especially difficult to acquire in some cases. Here, linear and nonlinear imaging techniques that aid in the calibration and/or recovery of phase observables and the self-calibration of amplitude observables are described. Enhanced dynamic range and image accuracy are possible with these post-processing techniques, which include closure relations and phase referencing. Examples of the application of the techniques in other imaging disciplines, such as optics and radio astronomy, are discussed.
A practical VME-based sonar system development workstation
Authors: Sloat Shon
Published: October 2005

This paper describes present capability and furture development plans of a practical low-cost sonar system-development work station based on off-the-shelf VME equipment. The work station allows system engineers and algorithm designers to quickly develop sonar processing algorithms and systems. The work station can be used to (a) simulate and analyze the characteristics of real time environments for system and algorithm development, (b) concurrently develop final implementation software, and (c) generate analog test data for system test and operator instruction. The work station overcomes many of the difficulties experienced with developing sonar acoustic signal acquisition and processing systems: high design and maintenance costs of custom equipment; lack of controllable test data; inability to correlate simulation data with final system performance; and high software development costs, particularly with parallel, pipelined signal processing equipment. The sonar development work station hardware consists of a 16.67-MHz 68020 processor supported by a VME bus architecture. The hardware provides (a) interative color graphics, (b) Sky Warrior and Mecury Zip 3232 array processors. The work station software is based on the Unix development system, VRTX real time kernel, X-Windows standard color graphics software, and array processor sonar processing primitives. This foundation provides software portability and device independent graphics.
Airborne sound to seismic coupling: Present understanding
Authors: Attenborough K. , Sabatier J. M.
Published: October 2005

Theories of outdoor sound propagation at near-grazing incidence on porous ground surfaces assume either that the ground surface is an impedance boundary or that it is one of extended reaction. Such theories are consistent with models of the acoustic behavior of the ground as a locally reacting fluid or as a semi-infinite or layered modified fluid supporting a single type of compressional wave motion. However, these models are unable to explain the response of buried geophones to airborne sound sources. Measurements of the ratio of a shallow buried geophone output to the output of a vertically separated microphone placed near the ground surface have been explained in a variety of soil types by means of a layered poroelastic ground model. The layered poroelastic ground model and its experimental validation are described, and its implications for the impedance-frequency characteristics of outdoor ground surfaces are explored.
Extrapolation of acoustic fields by means of source function estimation
Authors: Elias G. , Payen F.
Published: October 2005

Department of Physics, Office National d'Etudes et de Recherches Aérospatiales, BP 72, 92322 Châtillon Cedex, France Extrapolation of acoustic pressures is needed when farfield measurements are not possible. Most of the methods usually used are based on a direct application of Kirchhoff's integral or on spectral decomposition, which leads to a fine sampling of a closed surface. The present method requires only a limited number of microphones and consists of a source function estimation. In the first step, the microphone array is used as a focused antenna for measuring the overall extent of the source region. In the second step, a biorthogonal basis is computed for the expansion of any source function limited to the previous spatial extent and the pressure radiated on the array. Then, by projecting the measured pressure on this basis, an estimate of the source function can be obtained to extrapolate the acoustic pressure in the farfield. Simulations show a good agreement between real and estimated directivity diagrams and the precise limitations of this method. [Work supported by DCN.]
Energy partitioning for very-low-frequency, long-range acoustic propagation in the Arctic
Authors: Baggeroer A. B. , Duckworth G. L. , Scheer E. K.
Published: October 2005

In long-range acoustic propagation in the Arctic, very-low-frequency signals interact with both the ice canopy and the bottom, both of which introduce loss. Signals from a sequence of long-range (340-km) shots detonated at several depths were recorded on a multichannel array and resolved according to their horizontal phase velocity and frequency components to separate the loss effects of the these two boundaries. Source levels and geometric losses based upon a determined sound-speed model were subtracted to arrive at an absorptive/ scattered loss measure for each resolved path and frequency. These reduced data were used to estimate the partitioning of energy according to phase velocity and frequency from the ice canopy and the seafloor. At frequencies below 20 Hz, bottom interacting signals account for more than 50% of the received energy observed at the array on almost all the shots.
Rate versus duration of formant transition as a discriminative cue
Authors: Porter Robert J. , Collins Jane , Cullen John K. , Jackson Dena F.
Published: October 2005

The linguistic messages of speech cannot be clearly read in its time-by-frequency-by-amplitude signature nor is it manifest in the corresponding psychoacoustic domain. It is suspected that important aspects of speech messages may, instead, be revealed in higher-order, rate-of-change attributes of speech signals and their corresponding perceptual representation. In a recent exploration of rate-of-change-of-spectrum [Besing et al., J. Acoust. Soc. Am. Suppl. 1 81, S54 (1987)], 300-ms, second-formant-like signals were used, which were varied in onset frequency to produce variations in the rate of a 30-ms, initial transition. Results suggested listeners may use rate-of-frequency-change to discriminate among these signals. The present study examined thresholds for rate changes produced by transition duration variation instead of onset frequency. Transitions began at either 1500 or 2100 Hz and rose or fell to a steady state of 1800 Hz. Standard transition durations of 60 and 124 ms were used. Discrimination was compared for rising and falling transitions presented either alone, or presented together with first-formantlike resonances. Results for ten subjects suggest, as in the previous study, that listeners may use perceived rate change rather than perceived duration to discriminate these speechlike signals. [Work supported by NIH-NINCDS and Louisiana Lions Foundation.]
Pulse synchronous analysis of speech using an auditory representation
Authors: Glass James R. , Seneff Stephanie
Published: October 2005

Research Laboratory of Electronics, Massachusetts Institute of Technology, Cambridge, MA 02139 A procedure has been developed that automatically extracts pulse excitation information from the speech signal. This time-domain algorithm operates on the transduction stage outputs of an auditory model developed by Seneff [this meeting]. Specifically, robust inflection points are determined from a waveform obtained by summing the channel outputs. This pulselike representation provides information about periodic and aperiodic excitation, and can be used to extract periodicities in voiced regions. Thus the information provided by this signal complements the spectral information provided by mean-rate response outputs of the auditory model. Furthermore, this signal can be used to sample channel outputs pulse synchronously. This produces a spectral representation with sharper temporal onsets and offsets, and with an enhanced dynamic range in the frequency dimension. [Work supported by DARPA under Contract N00039-85-C-0254, monitored through Naval Electronics Systems Command.]
Alternate measures of response bias in speech recognition by older listeners
Authors: Zeller Robin T. K. , Gerstman Louis J.
Published: October 2005

Fifty-two listeners ranging in age between 50 and 81 years, listened to two R-SPIN Tests, responding to each stimulus verbally and with a numerical confidence rating. Twelve performance measures were created: for HP and LP items separately, percent correct, d′ and P(A), which both measure sensitivity, and B, which measures response bias; for all 100 items, a mean and standard deviation of the ratings; and for the errors on the LP items, a mean frequency of word usage, and a rating of contextual feasibility by a panel of 15 judges. Correlations with age revealed no effects for the HP items, but established that with advancing age, there was lower sensitivity on the LP items as measured by d′ and p(A), but no change in B. Additionally, the older listeners used a narrower range of the rating scale and made more contextually feasible errors. No age variation was found for word frequency of errors, but all the proposed responses were ten times more probable than the misheard LP words.
Recognition of speech and recognition of speaker sex: Parallel or concurrent processes?
Authors: O'Kane M.
Published: October 2005

If the same phonetic string is spoken by a male speaker and a female speaker, both drawn from a linguistically homogeneous population, then a listener perceives that both speakers are saying the same sounds but generally knows that the speakers are different and, in particular, that the speakers are of a different sex. What is not well understood is whether the listener has to determine the identity of the sex of the speaker in order to determine the identity of the sounds, or whether the two pieces of information are essentially independent and can be determined concurrently, or whether the identity of the sounds can be roughly determined without knowledge of the speaker's sex but fine recognition is possible only after the speaker's sex is known. In this presentation, evidence for and against these various possibilities from a large number of psycholinguistic studies, particularly those dealing with various sex-specific effects in a variety of languages, is surveyed and the implications for the design of automatic speech recognition and understanding systems are discussed.
Standardized presentation of complex modulus
Authors: Rogers Lynn , Fowler Bryce
Published: October 2005

The capabilities of a proposed standard for graphical presentation of complex modulus data for viscoelastic vibration damping materials will be described. A plot of material loss factor versus modulus magnitude indicates data quality; a frequency-temperature nomogram is included on this plot, thereby also making it sufficient for the designer. The more typical reduced frequency plot is also presented, but with experimental temperature and frequency ranges given to assist the designer in avoiding extrapolation of data. The state of the art in modeling temperature shift functions and complex modulus using fractional derivatives will be presented. An interactive computer program for processing a set of complex modulus data will be described. A system to store and retrieve damping material data will be described.
Using video tapes and audio tapes on and off the campus
Authors: Rossing Thomas D.
Published: October 2005

The use of prerecorded presentations offers several advantages to the teacher. One of the most attractive is to present demonstrations of acoustical phenomena that are difficult to implement in real time, especially in a remote classroom. Another is to substitute animated text for still text (the so-called “animated blackboard”). Excerpts are presented from several audio and video tapes and how they can be used to enhance acoustics instruction both on and off campus is discussed.
Distortionless piezoelectric transducers with nonuniform electroelastic parameters
Authors: Hazony Dov , Raya Justo R.
Published: October 2005

Of concern are the generation and detection of sound by piezoelectric transducers with nonuniform material properties along the principal axis. It is shown that subject to conditions of distortionless propagation [V. Burke, R. J. Duffin, and D. Hazony, “Distortionless Wave Propagation in Inhomogeneous Media and Transmission Lines,” Q. Appl. Math. 183–194 (July 1976)] it is possible to associate a circuit model with the transducer similar to the Mason model [W. P. Mason, Electromechanical Transducers and Wave Filters (Van Nostrand, New York, 1948)]. This work follows the development of spherical shell piezoelectric transducers [D. Hazony, “Nonuniformly Poled Spherical Shell Piezoelectric Transducers,” J. Acoust. Soc. Am. 81, 1624–1627 (1987)]. [Work supported in part by Tecsonics, Inc.]
Analytical modeling of an active noise control system in a duct
Authors: Munjal M. L. , Eriksson L. J.
Published: October 2005

Making use of the transfer matrices and electroacoustic analogies, a standing wave analysis of the active noise control system in a duct is presented incorporating the characteristics of the primary source, as well as the auxiliary source. Analytical expressions have thus been derived for the ratio of the two source pressures and certain other ratios or relations of interest for complete cancellation of noise downstream of the auxiliary source. These expressions are then compared with those obtained from transfer functions of the various blocks constituting the block diagram of the entire system. In the process, equivalence has been established between the different entities involved in the two approaches, one of which is used in acoustics and the other in system theory. In particular, it has been shown that the pressure generated by a source against a load impedance can be looked upon as a sum of two pressure waves, one generated by the source against the characteristic impedance and the other by reflecting the rearward wave (incident on the source) off the source impedance. This principle is seminal in linking the two approaches.