bluuebunny/crossref_metadata_2025 huggingface dataset:
Detection of simultaneous streams of running speech
Authors: Tufts Jennifer , Frank Tom
Published: July 2002

The purpose of this study is to determine how many simultaneous streams of running speech a listener can detect and correctly enumerate. Thirty normal-hearing subjects will participate. Stimuli will consist of three sets of digitally mixed recordings of female talkers where the number of talkers in each set ranges from two to ten. Subjects will be asked to identify how many speakers they hear and to rate the certainty with which they make their judgments. Each subject will be tested in the left ear, right ear, and binaurally. Results are expected to demonstrate the limits of the auditory system in resolving individual speakers or streams of speech without localization cues.
Implementation of a time-reversal cavity by confining a time-reversal mirror in a waveguide
Authors: Tanter Mickaël , Thomas Jean-Louis , Fink Mathias
Published: July 2002

Time reversal is an original method to focus ultrasonic waves through heterogeneous lossless media. Nevertheless, to conduct a perfect time-reversal experiment, the wave field coming from an acoustic source should be recorded on a closed surface covered with transducers and surrounding the region of interest. This led to the concept of time-reversal cavity. In practice, its implementation is not easily realizable and the time-reversal operation is only performed on a restricted area named a time-reversal mirror. Some aberrating medium may diffract a part of the wavefield coming from the acoustic source outside the limited aperture of the time-reversal mirror. Hence, focusing quality may be strongly degraded. To approach the optimal focusing obtained with a time-reversal cavity, a channel bounded by totally reflecting walls was used to guide the wave field between the source and the time-reversal mirror. The previously lost information was then recovered to focus on the source location. Therefore, the combination of a time-reversal mirror with a waveguide can simulate a time-reversal cavity with only a small number of transducers. Experimental results give good agreement with a wave propagation model using a finite differences scheme. Both show that the focusing is now the same as that in a homogeneous medium.
Relaxor-based single-crystal materials for ultrasonic transducer applications
Authors: Shrout Thomas R. , Park Seung-Eek , Lopath Patrick D. , Shung Kirk K.
Published: July 2002

Relaxor ferroelectric single crystals of Pb(Zn1/3Nb2/3)O3 (PZN), Pb(Mg1/3Nb2/3)O3 (PMN) and their solid solutions with normal ferroelectric PbTiO3 (PT) were investigated for ultrasonic transducer applications. Longitudinal coupling coefficients (k33) as high as 94% and dielectric constants (K3T) in the range of 3500–6000 were achieved with low dielectric loss (<1%) using 〈001〉 oriented rhombohedral crystals of (1−x)PZN-xPT and (1−y)PMN-yPT, where x<0.09 and y<0.35. Dicing direction as well as poling direction was critical for high coupling under laterally clamped conditions. Dicing parallel to the (001) yields 90% of laterally clamped coupling (kbar) out of 94% longitudinal coupling (k33) for PZN-8%PT. Thickness coupling (kT) as high as 64% and low dielectric constant (K3T)<600 with low loss (<1%) could be achieved using tetragonal crystals of (1−x)PZN-xPT and (1−y)PMN-yPT, where x>0.1 and y>0.4. The performance gains associated with these ultra-high coupling coefficients and range of dielectric constants are evident in relations to broader bandwidth and electrical impedance matching. Using the KLM model, a pulse/echo response simulated a bandwidth of ∼124% for a subdiced array element with a center frequency of 10 MHz. An optimized array design of the same geometry constructed of PZT 5H displayed 87% bandwidth. [Work supported by ONR, DARPA.]
The role of vowel duration in the perception of /E/ and /æ/
Authors: Sawusch James R. , Palmer Nancy J.
Published: July 2002

Perceptual studies have consistently shown that the dominant acoustic correlates of vowel perception are the frequencies of the first three formants. The present studies used natural syllables to explore the role of vocalic duration in vowel perception. The first study examined the vocalic duration and frequencies of the first three formants of American-English vowels. Out of 48 talkers, 15 were identified for whom the formant frequencies were virtually identical in the CVC words ‘‘had’’ and ‘‘head’’ at the vowel mid-point. These tokens differed in duration and were highly intelligible. The second study used edited natural tokens from 4 of the 15 talkers to investigate the influence of vowel duration in perception. Pitch pulses from ‘‘had’’ were removed to create tokens varying in vowel duration from the natural ‘‘had’’ to a shorter duration similar to ‘‘head.’’ Pitch pulses from ‘‘head’’ were reduplicated to create a second series for each talker. Duration was used by some listeners for some talkers, but was not consistently used to differentiate these vowels in perception. This implicates other acoustic qualities, such as formant movements, as determinants of perception. [Work supported by NIDCD Grant No. R01 DC00219 to SUNY at Buffalo.]
A phonation-type effect in vowel perception
Authors: Johnson Keith , D’Imperio Maria Paola
Published: July 2002

Vowel perception is influenced by the fundamental frequency of voicing (F0 normalization). This paper describes a similar effect resulting from a manipulation of phonation type. Acoustic characteristics of two phonation types were produced in formant-synthesized and LPC resynthesized stimuli. In the formant-synthesized stimuli, spectral tilt and open quotient were used to produce a flat-spectrum continuum (low-amplitude H1 and high-amplitude spectrum above 1000 Hz) and a tilted-spectrum continuum (high-amplitude H1 and low-amplitude higher frequencies). The flat-spectrum tokens sound like a pressed or creaky voice while the tilted-spectrum tokens sound breathy. In the LPC resynthesized stimuli a similar effect was achieved by replacing the LPC residual with voices synthesized using the LF model of voicing. Identification responses to both sets of stimuli showed reliable boundary shifts on an F1 ‘‘hood’’–‘‘hud’’ continuum. There were more ‘‘hood’’ responses to the flat-spectrum continua than to the tilted-spectrum continua (higher F1 boundaries). This voice quality effect is reminiscent of F0 speaker normalization, but the creaky voice (which sounded more like a male speaker) was associated with a higher F1 boundary. Data on perceived speaker characteristics of the stimuli will be reported. [Work supported by NIDCD R29-DC01645.]
Perceptual evaluation of alternate formant normalization models
Authors: Nearey Terrance M. , Galama Debora A.
Published: July 2002

Two current theories of vowel normalization imply that vowel formant frequencies of one speaker can be related to those of another by a simple translation along a tonotopic frequency axis [T. M. Nearey, J. Acoust. Soc. Am. 85, 2088–2113 (1989)]. Theory A uses a Bark scale, while theory B uses a log-frequency (musical) scale. To compare these theories, a baseline continuum is first constructed that spans the range appropriate for the front vowels of an adult male speaker. In test series A, the formant frequencies are translated upward by 1.2 Bark, corresponding to the difference between average adult male and average female and child values from a vowel data base. In test series B, an upward translation of 0.2 natural log units is made, based on the corresponding average values in a log space. (Fundamental frequencies are also changed by the appropriate average value in the appropriate scale.) Each theory predicts its associated continuum will be identical to the baseline set, while that of the rival theory will show systematic distortion. Results of perceptual experiments will be presented that evaluate the phonetic distortion of the two test continua empirically.
Influences of aging and calcium-channel blockers on noise-induced hearing loss
Authors: Boettcher Flint A.
Published: July 2002

Two issues will be reviewed: (1) combined effects of noise and aging on hearing and (2) protective effects of calcium-channel blockers (CCBs). There are no empirical data on the auditory effects of noise exposure and the aging process in humans. The only sources of information on combined effects of noise and age have been epidemiological studies. In recent experiments, gerbils were exposed monaurally at middle age to an intense tone which caused approximately 20 dB of noise-induced permanent threshold shift (NIPTS). At 36 months (lifespan of the gerbil), the threshold in the unexposed ear was considered the age-related loss and that in the exposed ear was considered the loss caused by aging plus noise. Data were examined in terms of published models which attempt to discriminate between NIPTS and presbycusis in older individuals. Most models significantly overestimated the noise-induced component of the hearing loss in the aged subjects. In the second series of experiments, gerbils were exposed to an intense noise while concurrently receiving CCBs or placebos. Nimodipine (a potent CCB) reduced the NIPTS. Furthermore, epidemiological data suggest that aged women taking CCBs have hearing levels 10–12 dB better than average. [Work supported by NIH-NIDCD and NOHR.]
On-product noise labeling: Design and efficacy
Authors: Snyder Harry L.
Published: July 2002

Labeling issues pertinent to both nonhazardous and hazardous noise levels are considered and discussed. Among these are pertinent ANSI and engineering standards for warning label design; recommended label content and design criteria; format and presentation variables that affect the likelihood of noticing, reading, and complying with the labels; the effect of user cost of compliance (e.g., comfort) on the efficacy of labels; and conditions that indicate high versus low likelihood of label effectiveness. Particular emphasis will be placed on the experimental literature that has addressed warning label efficacy.
A model of silence duration adaptation in the perception of consonant cluster pairs
Authors: Cohen Michael A. , Govindarajan Krishna K.
Published: July 2002

Listeners are able to adapt to the silence duration distribution between two different stop consonants VC1−C2V [K. K. Govindarajan and M. A. Cohen, J. Acoust. Soc. Am. 95, 2978(A) (1994)]. This adaptation affects the likelihood of hearing just one stop consonant or the consonant cluster. An adaptation model is presented that emulates most of the individual subject data reported earlier; and based on Chi square goodness-of-fit tests, is capable of accounting for 98.5% of the variance of the psychometric functions. The model consists of a one-stage leaky integrator that averages the silence duration of prior tokens. This average is subtracted from the current silence duration which is then corrupted by Gaussian noise of constant mean and variance. If the result is positive, the predicted response is hearing two stops, otherwise one stop is heard. The decay rate, the initial average, and the mean and variance of the noise were fit to the six subjects in a variety of experiments and conditions using maximum likelihood estimation. Reasons for the reliability of the constructed fit, and variability of parameters across subjects and conditions will be discussed. [Work supported by AFOSR, NIH.]
Non-natives’ identification of English consonants in noise
Authors: MacKay Ian R. A. , Meador D. , Flege James E.
Published: July 2002

This study examined the identification of English consonants presented in noise. The subjects were English native (NE) speakers and four groups of Italian/English bilinguals who differed according to age of arrival (AOA) in Canada and/or self-estimated percentage daily use of Italian (%ITAL). The naturally produced stimuli, which differed according to their initial or final consonant (e.g., ‘‘bado,’’ ‘‘sado,’’ ‘‘hodab,’’ ‘‘hodas’’), were presented at decreasing S/N ratios (+12, +6, 0, −6 dB). Subjects identified each consonant by selecting one of five written alternatives, including the target and four likely foils. Percent correct identification scores decreased similarly across the four S/N ratios for all five groups. However, the bilinguals scored lower than did the NE subjects, despite their long residence in Canada (M=35.4 years). Their scores varied inversely with AOA and %ITAL, and were related to their performance on a test of phonological short-term memory. Which consonants were most difficult to identify varied across the five groups. Error analyses revealed that errors due to confusions of place and manner of articulation were similar across the five groups. Additional analyses will determine whether the influence of the L1 can be characterized in terms of finer-grained features.
Directional microphones for teleconferencing
Authors: Elko Gary W.
Published: July 2002

Directional microphones are spatial filters that can be designed to improve the sound quality of speech in audio telecommunication. At present, the most common type of directional microphone utilized in audio telecommunication is a simple first-order differential microphone having a hypercardioid directivity pattern. The directivity index (microphone gain in isotropic noise) for the hypercardioid microphone is 6 dB. For very noisy and reverberant environments more directional gain will be necessary. This talk will describe and discuss some of the microphone array systems that have been proposed in the last decade that address this problem.
The Huge Microphone Array (HMA)
Authors: Silverman Harvey F. , Patterson William R. , Flanagan James L.
Published: July 2002

The Huge Microphone Array (HMA) is a collaborative effort between Brown University and Rutgers University that started in February 1994 to design, construct, debug, and test a real-time 512-microphone array system and to develop algorithms for use on it. Analysis of known algorithms made it clear that signal-processing performance of over 6 Gigaflops would be required; at the same time, there was a need for ‘‘portability,’’ i.e., fitting into a small van, that also set an upper limit to the power required. It was essential that the array be able to be used in both large and small acoustic environments. These tradeoffs and many others have led to a unique design in both hardware and software. The hardware uses 128 fast, floating-point DSP microprocessors and is designed so that data flow is independent of data processing. This leads to an unusually simple software environment. This paper presents the full design and its justifications. Performance for a few important algorithms relative to usage of processing-capability, response latency, and difficulty of programming is discussed. [Work supported by NSF Grant MIP-9314625.]
Localized detection of cavitation generated by lithotripsy shock waves
Authors: Cleveland Robin O. , Sapozhnikov Oleg A.
Published: July 2002

Extracorporeal shock wave lithotripters generate intense cavitation which plays a role in both kidney stone destruction and tissue damage. The cavitation can be detected by listening for the characteristic ‘‘double-bang‘‘ acoustic emission, radiated by the bubbles during their rapid expansion and collapse phases. Previous workers have used a single focused transducer to detect the double-bang signature. However, the focal spot of the transducers is typcially 5 cm long and individual cavitation events can not be localized. A system has been developed that uses two confocal transducers (1 MHz, 20 cm radius of curvature, 10 cm diameter) to detect cavitation events in a localized volume. The system can be run in two modes: passive or active. In the passive mode both transducers listen to acoustic emissions generated by the cavitation field. In the active mode one transducer sends a tone and the second transducer listens to the sound scattered by any bubbles present in the focal region. In both cases the region to which the detectors are sensitive is approximately a sphere 5 mm in diameter. The passive system shows bubble rebound times far longer than predicted by theory for a single bubble. Scattering indicates that bubbles persist in the focal region for at least 6 ms. [Work supported by NIH.]
Strings or springs: Insight into the nature of links in chick cochlear hair bundles
Authors: Duncan R. Keith , Eisen Marc D. , Saunders James C.
Published: July 2002

The present study considers the issue of symmetry in relative hair bundle motion. The stereocilia on in vitro chick cochlear hair cells were viewed in profile, stimulated by an oscillating water jet, and observed under stroboscopic illumination. The angular deflection of the tallest hair was set to either 10, 15, or 20 deg. The angular displacements for the tallest and shortest hairs were measured in the excitatory and inhibitory directions relative to the rest position. Motion of the tallest hair was approximately symmetric for each stimulus magnitude, while short hair motion was greater in the excitatory than inhibitory direction (p<0.01). Moreover, short hair angular displacement with respect to the tall hair was greater in the excitatory direction (p<0.01), but equal in the inhibitory direction (p>0.01). These results suggested that stereocilia linkages were more analogous to strings than springs, in that they transmitted tension but collapsed under compression. Also, breakage of tip links in low calcium reduced relative motion between the tallest and shortest hairs, suggesting that motion in low calcium media was unchanged for the inhibitory direction, but dramatically reduced for excitatory displacements. [Work supported by NIDCD and NOHR.]
Perceptual grouping of tone sequences by normally hearing and hearing-impaired listeners
Authors: Rose Marina M. , Moore Brian C. J.
Published: July 2002

This study examined the perceptual grouping of the rapid sinusoidal tone sequence ABA–ABA for normal and hearing-impaired listeners. Tone A was fixed at 250, 500, 1000, or 2000 Hz. Tone B started with a frequency well above or below that of tone A, and its frequency was swept toward that of tone A. Listeners had to indicate when they could no longer hear two separate streams (fission boundary). For 12 normally hearing listeners, the fission boundary was roughly independent of the frequency of tone A when expressed as the difference in number of ERBs (delta E) between A and B, consistent with a recent model of stream segregation [M. W. Beauvois and R. Meddis, J. Acoust. Soc. Am. 99, 2270–2280 (1996)]. For four unilaterally hearing-impaired listeners, there was no consistent difference in the delta E magnitudes across ears, even though the auditory filters were broader in the impaired ears. This is not consistent with the theory of Beauvois and Meddis. The ten bilaterally hearing-impaired listeners showed fission boundaries ranging from normal to well above normal. The results are discussed in terms of the factors that might influence perceptual stream formation in hearing-impaired listeners. [Work supported by the Hearing Research Trust.]
Forward masking patterns measured in cochlear implant subjects
Authors: Throckmorton Chandra S. , Collins Leslie M.
Published: July 2002

Several psychophysical procedures have been proposed as measures of channel interaction in subjects implanted with a multichannel cochlear prosthesis. In order to compare these different measures of channel interaction, and to assess their impact on speech recognition, two psychophysical procedures were performed to measure the forward masking and electrode discrimination patterns elicited by electrical stimulation in subjects implanted with the Nucleus device. In this study, forward masking was measured for all combinations of the masker and probe electrodes. These results are presented and compared with the electrode discrimination data. The masking patterns are also compared to those described by Lim et al. [J. Acoust. Soc. Am. 86, 971–980 (1989)] and by Shannon [R. V. Shannon, Hear. Res. 12, 1–16 (1983)]. Speech recognition data are also presented, and the relationship between channel interaction and speech recognition is explored. Finally, since the Nucleus device stimulates electrodes sequentially from base to apex, the potential loss in transmitted information that is indicated by the channel interaction data is discussed. [Work supported by NIH.]
Time-frequency analysis adapted to the spectral characteristics of the signal
Authors: Cerdá Salvador , Romero José
Published: July 2002

The introduction of the wavelet transform (WT) and the analysis of multiresolution around the end of 1980, assumed an increase in the interest in signal processing and analysis. The development of the WT can be thought of as an attempt to improve the traditional analysis accomplished through STFT. However, STFT and WT outline resolution problems too. Attempts at improving the WT are the wavelet packets, and the designated M-band wavelet analysis. These analysis methods are similar in that they do not consider the characteristics of the signal for purposes of analyzation. A method that takes into account these characteristics is the ‘‘matching pursuits’’ of Mallat and Zhang. This method selects from the dictionary of time-frequency atoms those atoms that are more representative in the signal. Introduced in this work is an analysis method that is adapted to spectral characteristics of the signals. First, the spectral characteristics are determined. The transform of adjustment of the window size (TAWS), is an ‘‘interpolation’’ between the WT analysis and the STFT. This interpolation is accomplished by taking into account the signal characteristics: If frequencies that are very well defined do not exist in the spectrum, the analysis is the WT.
Mode coupling by internal waves for multi-megameter acoustic propagation in the ocean
Authors: Colosi John A.
Published: July 2002

Broadband parabolic-equation simulations have been performed with and without internal-wave sound-speed fluctuations. The simulations have a center frequency of 75 Hz, a bandwidth of 30 Hz, and a propagation range R of 1000, 2000, and 3000 km. For these cases it is found that acoustic propagation through internal waves is strongly nonadiabatic. In terms of modal travel times, low modes have a negative bias because they couple into higher, faster modes, while the higher modes show a positive bias, indicating preferential coupling into lower, slower modes. The lowest modes show the least travel-time spread and bias, and these quantities increase rapidly with increasing mode number. Empirically and approximately it is found that bias grows like R2 and spread grows like R3/2. The modal power distributions over frequency are markedly different from the source distributions. Power is distributed roughly equally across the 30-Hz frequency band for each mode, with 5.6-dB scintillations consistent with an exponential probability distribution function for intensity. In spite of the dramatic spread and bias in the higher modes, it is found that a synthesis of these modes results in coherent wavefronts, whose characteristic timing fluctuations at 3000-km range are two orders of magnitude less than those of the corresponding modes.
Assessment of turbulent flow in arteries using Doppler ultrasound
Authors: Stroud John S. , Holland Christy K. , Disimile Peter J. , Fox Curt W.
Published: July 2002

The detection of atherosclerotic-related diseases in the early stages could potentially impact medical treatment in a cost-effective manner. Atherosclerosis is a common arterial disorder characterized by deposition of cholesterol, lipids, and cellular debris in the inner layers of large- and medium-sized arteries. As atherosclerotic deposition progresses, an area of narrowing, or stenosis, forms, resulting in reduced circulation to organs and other tissues supplied by the artery. When the stenoses become severe in critical arteries (e.g., the carotids or the coronaries) the result can be a catastrophic stroke or heart attack. In the early stages of the disease, the hemodynamics downstream of a stenosis become disturbed. The resulting flow disturbances produce characteristic features in Doppler ultrasound scans which are of diagnostic value. A technique has been investigated for extracting the streamwise turbulence intensity, or normalized velocity variance, using Doppler ultrasound in an arterial flow model. The model consists of a pump and 1.3-cm i.d. polyurethane tubing which is optically and acoustically transparent. Correlation between Doppler ultrasound and laser Doppler anemometry (LDA) measurements was examined in laminar as well as turbulent flow. Continuous flow has been employed initially and the work will be extended to pulsatile flow to replicate cardiac output.
Finite-element and boundary-element computer modeling for loudspeaker cone design
Authors: Shively Roger E.
Published: July 2002

The use of finite-element and boundary-element computer modeling for the design and optimization of loudspeaker cones is demonstrated. First, the specifics of the analytical model and theory are summarized. The experimental setup, which utilizes laser vibrometry is described, and the correlation of analytical FEM and BEM models to experimental models and anechoic measurements for a test case are reviewed. Three case histories are then illustrated: (1) correlation of a high-frequency model, (2) a design case for maintaining a current loudspeaker packaging size with improved frequency response and distortion over the current performance, and (3) a design case for evaluating a new material for a loudspeaker’s diaphragm.
Loudspeaker drive-unit measurements at excursion
Authors: Clark David
Published: July 2002

Loudspeaker drive units are conventionally modeled as linear devices. Evidence is presented that shows that this assumption leads to unacceptable modeling errors for practical conditions of use. This paper presents new techniques for accurate, nondestructive measurement of linear and nonlinear electromechanical parameters. These parameters are used in more complete models which improve accuracy of frequency response and distortion estimates. The measurement workstation uses pneumatic pressure to displace the diaphragm in order for position-dependent parameters to be measured over the full excursion range. Suspension compliance and mechanical damping are measured under conditions which simulate normal operation, thereby capturing the influence of creep, aerodynamic drag, and other second-order characteristics which are not measured separately at present. Moving mass and force factor measurements utilize a magnetic position servo and gravity to obtain data without displacing the suspension. Conventionally, maximum diaphragm displacement, xmax is calculated from voice-coil and magnet geometry. This ignores the magnetic fringe field and the effect of suspension nonlinearities. An improved value of xmax is derived from the new measurement data.
Effect of number and frequency spacing of masker components on multitone masking
Authors: Oh Eunmi , Lutfi Robert A.
Published: July 2002

Neff and Green [Percept. Psychophys. 41, 409–415 (1987)] reported that the masking of single tones by random-frequency multitone maskers is greatest when the masker is comprised of between 10 and 50 tonal components. In this paper we present data to suggest that such results are related to the frequency spacing of masker components. Two conditions were run. In the first, the number of tones comprising the masker was varied from 20 to 906, with frequencies chosen at random over a fixed range from 0.1 to 10 kHz. In the second, the number of tones was fixed at 20, and the frequency range over which components were chosen was varied to match the frequency spacing of components in the first condition. An adaptive two-interval, forced-choice procedure was used to measure masked thresholds for a 1.0-kHz signal. Both conditions yielded large elevations in signal threshold. This was true even after compensating for the effects of energetic masking produced by components nearest the signal. Compensated thresholds showed little dependency on component number, but decreased systematically with decreasing frequency spacing of components. The results are consistent with a model of multitone masking based on component-relative entropy [R. A. Lutfi, J. Acoust. Soc. Am. 94, 748–758 (1993)].
The adaptive value of connotation in speech perception
Authors: Wurm Lee H. , Vakoch Douglas A.
Published: July 2002

In previous work it was demonstrated that reaction times (RTs) in a lexical decision paradigm were related to words’ ratings on the dimensions of evaluation, potency, and activity (independent of word frequency). This was found to be true of words in general [D. A. Vakoch and L. H. Wurm, Cognition and Emotion (in press)], and also of words from the affective lexicon [L. H. Wurm and D. A. Vakoch, Cognition and Emotion 10, 409–423 (1996)], but there were differences in the way these dimensions appeared to be used in each case. For emotion words, RTs were quickest for words that were rated as bad, strong, and fast (the evaluation × potency × activity interaction was significant), a finding interpreted in terms of the adaptiveness of danger avoidance. For words from the general lexicon, RTs were quickest to words that were rated as either good and strong, or good and fast, as though these represented resources that had to be seized quickly or lost. The current study clarifies the earlier findings, and expands the existing theory to include concepts such as the danger and usefulness of referents. [Work supported by AFOSR and NIMH.]
Echosignal peculiarities in biological mediums
Authors: Genis V. , Oboznenko E. I. , Oboznenko I. L.
Published: July 2002

A modified method of Born’s approximation is proposed. The essence of the method is that in real situations, the incident wave gains the object’s speed, but not the medium’s speed; which follows from the Fresnel formula. A solution algorithm for the weak-reflecting bio-organ’s scattering problem is proposed in this task. The algorithm is based on a more precise definition of Born’s approximation. Numerical and experimental research shows that the pathologies can have dimensions of about 1 mm, which can be identified according to echosignal spectrums. It is shown that norm and pathologies, like pectoral glands, are distinguished. If the pathology is less dense, there will be potent destruction in the echosignal spectrum. If pathology is denser, there will be less potent destruction. The proposed method permits one to classify the pathologies in bio-tissues more subtly.
SAMI: A low-frequency wideband prototype for synthetic aperture mapping and imaging
Authors: Zakharia Manell E. , Chatillon Jacques
Published: July 2002

A low-frequency wideband (6–12 kHz) prototype has been developed in the framework of the SAMI (synthetic mapping and imaging) project. The prototype uses a subsurface towed fish. One transmitting array and two parallel receiving arrays 2 m long were installed on the tow-fish. Each array is associated to a synthetic aperture sonar channel. Thanks to the use of suitable transmitted signals (linear period modulated chirps), array synthesis could simply be achieved in the time domain; pulse compression and azimuthal matched filtering could be separated in order to reduce the processing complexity. Synthetic image resolution was measured at sea using reflectors on a frame; the results obtained were comparable to the predicted performance: 0.5×0.15 m. The cross correlation between both synthetic aperture channels output was used for measuring the elevation of the sea bottom reverberation cells. Synthetic aperture maps were then produced. Real-time processing was achieved on board the vessel for a swath of 500 m. Data were stored and post-processing was achieved off-line. Several experimental data obtained at sea in both shallow and deep (2400 m) water are presented. [Work supported by the EC-MAST (Marine Science and Technology) program.]
Vocal accuracy in performance of well-known melodies
Authors: Henderson Clare L. , Smith Lloyd
Published: July 2002

Vocalization is an important aspect of music making, regardless of the culture. Psychologists have carried out considerable research on human melody recognition and perception, which includes anaysis of musical cognitive processes. However, questions that relate directly to the ways in which people perform what they recall have been left largely unanswered. In this experiment, a number of people were asked to sing, from memory, ten well-known songs. Their performances were recorded on audio tape and analyzed and compared with notated versions of the songs. The results showed that pitch maintenance was more accurate when songs moved by steps rather than leaps and when intervals moved up rather than down. Pitch problems were more prevalent in songs using downward leaps or changes of tonality. Wide leaps, such as fifths or sixths, were often compressed, with subjects singing the top note of an ascending interval flat, or the bottom of a descending interval sharp. The results of this experiment have useful applications for musicologists, music educators, composers, and those involved with data retrieval processes.
Experimental studies of diffuse decay curvature
Authors: Burkhardt John
Published: July 2002

Reverberant, ultrasonic decay experiments were performed on an irregularly shaped aluminum block on which varying sized water drops were placed. The resulting nonproportionally damped systems were found to possess nonexponential decays in agreement with theoretically derived results. These findings support the author’s proposition that the features of nonexponential decays are a possible quantitative nondestuctive technique for the characterization of localized fatigue damage in metals.
Acoustic spectroscopy of deep levels in Cr-doped GaAs using light beam generated SAW
Authors: Bury Peter , Baják Ivan , Grondžák Karol
Published: July 2002

A light-beam-generated space-charge inhomogeneity was used for the first time as a probing tool to study the behavior of deep levels in high-resistivity, Cr- doped GaAs using acoustic transient spectroscopy. The space-charge inhomogenity produced by the illumination of photoresistive piezoelectric semiconductors in a high-frequency electric field can generate, in proper conditions, both the surface and transversal or longitudinal acoustic wave. The temperature dependence of the time development of such a generated surface acoustic wave (SAW) after illumination by pulsed monochromatic light was used to investigate deep levels in high-resistivity GaAs using the technique of acoustic deep-level transient spectroscopy (A-DLTS) [P. Bury and I. Jamnický, Proc. World Congress on Ultrasonics, Berlin, 1995, p. 535]. Deep-level parameters such as activation energy and capture cross-section were determined. [Work supported by a Grant of the Slovak Ministry of Education, No 1/1310/96.]
Magnitude estimation scaling of the loudness component of pure tones, narrow band noise, broadband noise, rock music, and babble speech
Authors: Fucci Donald , Petrosino Linda , Wyatt Denise , Wilcox Corry
Published: July 2002

This study was designed to extent the range of auditory stimuli used to study magnitude estimation scaling of loudness. The five stimuli chosen were: a 1000-Hz pure tone; narrow band noise (700–1300 Hz bandwidth); broadband noise (100–10 000 Hz bandwidth); rock music [Led Zeppelin, Atlantic Recording Corp., CD Recording No. 19127-2 (1969)]; and babble speech [D. Kalikow, K. Stevens, and L. Elliott, J. Acoust. Soc. Am. 61, 1337–1351 (1977)]. Subjects were 30 normal young adult women. During the auditory magnitude estimation task for each of the five stimuli, a subject was instructed to assign numers to that simulus which was presented in a randomly ordered series of nine sensation levels (10, 20, 30, 40, 50, 60, 70, 80, and 90 dB SL). An analysis of variance repeated measured design showed no difference in the numerical responses of the subjects for the five different stimuli. Results suggest the presence of an underlying stabilizing factor (internal scaling mechanism) that allows adult subjects to consistently scale loudness irrespective of the type of auditory stimulus presented [J. Zwislocki and D. Goodman, Percept. Psychophys. 28, 28–38 (1980)].
Linear and nonlinear sound wave propagation in turbulent and inhomogeneous media
Authors: Krasilnikov Vladimir A.
Published: July 2002

The author had the pleasure of being involved in the pioneering experiments on sound wave propagation in the turbulent atmosphere (1939–1950). The obtained results were interpreted from the point of view of the, then recently developed, Kolmogorov–Obukhov (KO) statistical theory of locally isotropic turbulence. These results were one of the first confirmations of this theory. It may be said that the KO theory and the experiments mentioned above had stimulated the broad developments in the problem of ‘‘waves and turbulence.’’ The resulting classical theory now provides a basis for the analysis of signals and the computer simulation of sound propagation through turbulence. This paper provides an historical overview of the development of the classical theory for sound propagation through turbulent media. Models of turbulence and experimental verification of these models will be discussed for sound propagation in the atmosphere, as well as in seismic and underwater media. Some examples of the problem of ‘‘wave turbulence and nonlinearity’’ also will be discussed.
Using principal component analysis of tongue surface shapes to distinguish among vowels and speakers
Authors: Stone Maureen , Cheng Y. , Lundberg A.
Published: July 2002

The present study uses principal component analysis (PCA) to examine sagittal tongue contours for five English vowels taken from ultrasound images. The vowels are repeated three to five times each in a /pVp/ carrier utterance. Of particular interest is the use of plots of the coefficients of the eigenvectors to distinguish both subjects and vowels, and the building of a linear model to fit their family of data. Data will be transformed to normalize surface length across contours. Short contours will be stretched in the x direction to the length of the longest curve. The x and y dimensions will be stretched equally for each curve to preserve scale. Preliminary data indicate fairly good success distinguishing among three subjects and four vowels using a linear model based on the first few components. Methods of improving this result are being explored using factor analysis or optimal fitting.
A computerized ultrasound transducer scanning system
Authors: Davis Edward , Stone Maureen
Published: July 2002

A computerized electromechanical ultrasound transducer holder is being developed to allow representation of three-dimensional tongue motion. Current three-dimensional ultrasound systems are comparatively slow (10 s/volume) and hence are not useful for the study of dynamic three-dimensional speech. The proposed mechanism will (automatically) both scan along and rotate about an axis, sensing transducer position and providing feedback. This motion will be motorized and controlled by a computer according to custom-developed software and parameters. One can program the computer to perform a synchronized series of approximately five sagittal or coronal slices. These slices can later be assembled and interpolated to form a three-dimensional surface model. Mechanically, this system is advantageous because it rotates the transducer about a virtual center, which is located along the axis of the center of the arc of the piezoelectric crystals within the transducer, and does not necessitate additional mechanical structures near the subject. The electrical and computer control system will allow automated imaging of a series of coronal or sagittal slices synchronized in time and located precisely in space. This system will be added to and used in conjunction with the HATS system described in Stone and Davis [J. Acoust. Soc. Am. 98, 3107–3112 (1995)]. [Work supported by Northrup Grumman and NIH.]
Acoustic evidence for featural and subfeatural errors in speech production
Authors: Frisch Stefan , Wright Richard
Published: July 2002

Phonological speech errors are a traditional source of psycholinguistic evidence for the representations of phonology theory. In an electromyographic (EMG) study of experimentally induced phonological speech errors, Mowrey and MacKay [J. Acoust. Soc. Am. 88, 1299–1312] found that speech errors frequently occur at a subfeatural, gestural level, with no apparent effect on the percept of the word. Mowrey and MacKay’s study considered the activity of a single muscle, and thus was unable to determine whether single gestures acted independently of gestural constellations. This study is a preliminary report from an ongoing acoustic analysis of speech errors. The data are tape recordings of an error-inducing experiment using nonsense tongue twisters. Recordings of six speakers producing four different tongue twisters targeting /s/ and /z/, e.g., sit zap zoo sip, were digitized and analyzed. Some errors involved multiple changes in acoustic properties, including simultaneous changes in periodicity, amplitude of frication, and duration, while others involved a subset of these properties. This evidence suggests that errors can occur at both the single gesture level, affecting noncontrastive acoustic properties, and at the level of the gestural complex or segment, creating a perceptible, linguistically contrastive change. [Work supported by NIH Grant No. DC00012.]
Finite-element analysis of both flexural and Helmholtz resonance modes in a flextensional transducer
Authors: Jones Dennis F.
Published: July 2002

During the last decade, several types of compact piezoceramic-driven barrel-stave flextensional transducers were researched and developed at the Defence Research Establishment Atlantic (DREA) for low-frequency high-power sonar applications. These transducers were designed to operate at the fundamental flexural resonance where they radiate sound efficiently and omnidirectionally. However, an important limitation common to almost all underwater flextensional transducers is that they require some form of depth compensation for operation at water depths exceeding a few hundred meters. One technique that may eliminate the need for depth compensation (and the associated design complications) for the barrel-stave flextensional transducer is to free flood the cavity inside the cylindrical ring-stack piezoceramic driver with the surrounding acoustic fluid medium and exploit the resulting Helmholtz resonances. To examine the feasibility of such a design approach, the flexural and Helmholtz resonances of a theoretical barrel-stave flextensional transducer with a free-flooded driver have been studied using finite-element techniques, and the results are presented in this paper. In addition, it is shown that the Helmholtz resonance in an experimental transducer does exist and may be practical.
Ultra-low-frequency ride motions and kinetosis
Authors: Sussman E. D. , DiSario Robert , Mengert Peter , Pollard John K.
Published: July 2002

Proposed ultra-high-speed ground transportation systems, such as Maglev, may have motion characteristics affecting passenger comfort different from anything previously experienced in ground transportation. These motions not only include isolated vertical maneuvers but also repeated motions. For these repeated motions the power spectra is as important as the magnitude in predicting kinetosis. Segments of the New York State Thruway were simulated by flying an airliner through a series of several dozen roll maneuvers based on nine combinations of speed and curvature. The combinations were also simulated using a moving base simulator which provided a geometrically accurate ‘‘out of the window view.’’ Analysis of the data lead to the following conclusions: (1) More than 95% of the public would accept isolated maneuvers involving bank angles up to 37° and roll rates up to 7°/s. (2) The majority found the plane simulation comfortable and felt no motion sickness. Differences between subjects’ appear to have been greater than the physical effects of the differences in bank angle. (3) Cumulative dosage and duration of exposure showed significant correlation with the motion-sickness ratings procedure developed based on the work of M. J. Griffin and British Standard 6841:1987.
Comparison of directivity patterns measured in the long-line hydrophone calibrator to predicted data
Authors: Luker L. D. , Beason M. K.
Published: July 2002

This presentation describes the results of the first calibration of line arrays in the long-line hydrophone calibrator (LLHC). The LLHC is a water-filled traveling-wave tube that simulates a plane-wave pressure field arriving from any bearing. It provides an alternative to open-water sensitivity calibrations and measures both sensitivity and directivity patterns over the array’s operational environment. This is accomplished by measuring the electroacoustic transfer matrix between hydrophones and projectors that are placed within a steel pipe with an inner diameter of approximately 30 cm. This transfer matrix is then used to compute projector drives suitable for creating the desired sound field. The current LLHC configuration is capable of calibrating line arrays up to 48-m long at pressures from 0 to 6.9 MPa and frequencies from 5 to 700 Hz. The line arrays that were calibrated were approximately 43-m long containing up to 16 sensors. The calibrations were performed at pressures up to 2.7 MPa and frequencies ranging from 10 to 800 Hz. Measured directivity patterns for various frequencies will be presented and compared to predicted data. [Work supported by NAVSEA.]
The idiotone and spontaneous otoacoustic emissions
Authors: Bilger Robert C.
Published: July 2002

Although the preponderance of Dix Ward’s work was in the area of the effects of noise on hearing, his paper on tonal monaural displacusis [J. Acoust. Soc. Am. 27, 365–372 (1955)] has always been my favorite from among all of his papers. In this paper, published several years after he had completed his work at Harvard and was being introduced to noise-induced hearing loss through the Central Institute for the Deaf, he performed a consummate study of his own tonal monaural displacusis and that of a woman who was studying music at Harvard. Not only did he specify their diplacuses exhaustively, his conclusions from that study foretold of many future developments in psychoacoustics, hearing loss, tinnitus, and signal analysis. The import of this monumental work will be iterated.
Possible correction factors for intermittent noise exposure
Authors: Johnson D. L.
Published: July 2002

A standard on Occupational Hearing Loss, ANSI S3.44 (1996), has been recently published. One of its features, unlike ISO R-1999 upon which it is based, allows a trading rule other than one based solely on equal energy. To account for the fact that intermittent noise, as compared to steady noise of the same energy, may not be as harmful to hearing, equal energy with a correction factor is being investigated. ANSI Working Group S3-58 is evaluating at least seven different approaches. In order to evaluate the best approach for correcting for intermittency, W. Dixon Ward’s temporary threshold shift (TTS) data, from a variety of intermittent noise exposures [W. D. Ward, Effects of Noise on Hearing (Raven, New York, 1976), pp. 407–420], will serve as one evaluation criterion. Actual hearing loss data from exposures to various intermittent noises will serve as the other evaluation criterion.