bluuebunny/crossref_metadata_2025 huggingface dataset:
Signal alternation disrupts perception of music as well as speech
Authors: Samuel Arthur G.
Published: October 2005

Studies over the last 40 years have demonstrated that, when a speech passage is periodically alternated between the right and left ears, intelligibility is disrupted. The perceptual degradation is quite selective: Alternation rates of approximately 3–4 cps are disruptive, while slower or faster rates are not. In addition, the maximally disruptive alternation rate covaries with speech rate: When speech is presented at a faster rate, the critical alternation rate increases proportionally. Models suggested for the phenomenon include attentional limitations, disruption of perceptual units (syllables), and multifactor psychoacoustic theories. A recurring and previously unanswered issue has been whether or not the effect is speech specific, as would be expected if disruption of a unit such as the syllable causes the degradation. To answer this question, a successive same-different discrimination task was run using piano melodies. On each trial, a monaural melody was followed by a melody that was alternated between the ears. The pattern of performance mirrored the traditional speech results: Selective perceptual degradation is a general property of auditory perception. The theoretical implications of this finding will be discussed. [Work supported by AFOSR.]
Synthesis of speech from the low-dimensional PLP representation
Authors: Hermansky Hynek , Cox Tony L.
Published: October 2005

Low-dimensional perceptual linear predictive (PLP) analysis [H. Hermansky, J. Acoust. Soc. Am. 87, 1738–1752 (1990)] was developed as an analysis technique for speaker-independent automatic speech recognition. It is claimed to preserve the linguistic part of information in speech and to suppress the speaker-dependent part. Its low dimensionality also makes PLP a good candidate for speech synthesis. However, a problem that remains to be solved in PLP-based synthesis is restoration of the speaker-dependent information. In the current work, use of a fixed speaker-dependent transformation [Broad and Clermont, J. Acoust. Soc. Am. 86, 2013–2017 (1989)] is investigated between PLP cepstral coefficients and poles of the high-order LPC model. The speaker-dependent transformation is derived from the set of training data. This technique allows for synthesis from relatively few parameters (five parameters per frame were used). In principle, this proposed technique could allow for synthesis of different voices from one set of PLP templates or for modification of speaker voice personality in the analysis/synthesis system.
Acoustical conditions in three well-known classical concert halls
Authors: Bradley J. S.
Published: October 2005

Measurements of newer auditorium acoustics parameters obtained in the Vienna Musikvereinssaal, the Amsterdam Concertgebouw, and the Boston Symphony Hall are compared to show the principal characteristics of these well-known concert halls. Hall averages as well as the within-hall variations of reverberation times, early decay times, early/late sound ratios, overall strength or level, and lateral energy fractions are presented. Values of these parameters are also calculated to be representative of occupied conditions. The results give an indication of the values of these newer parameters found in these famous concert halls, as well as an indication of the differences among these halls.
Low-noise terminal design for HVAC systems
Authors: Kahn David W. , Greenberg David
Published: October 2005

In the design of noise-critical spaces such as music and theatre performance and rehearsal spaces, low background noise levels are essential. Therefore, noise emanating from the supply and return air terminals of the HVAC system must be kept to a minimum. This requirement often dictates the need for many large terminals. This approach can be a very expensive alternative. In addition to the expense, the large area of duct terminals can have a negative impact on the room acoustics, particularly in a concert hall where sound absorptive materials/openings must be kept to a minimum. A number of different supply and return air terminals that create a minimum amount of noise at relatively high outlet velocities (up to 600 fpm) have been designed and tested. The results of these tests and guidelines for the design of low noise terminals will be discussed.
An experimental study of the oscillations in a flue organ pipe
Authors: Wijnands A. P. J. , Hirschberg A. , van Steenbergen A. , de Vries J. J. , Fabre B. , Gilbert J.
Published: October 2005

Stationary oscillations of a simple flux-type organ pipe have been studied. Internal acoustical pressure measurements in combination with modification of pipe geometry indicate that the sound source at the labium is a dipole. This observation is explained by Coltman's analysis [J. Acoust. Soc. Am. 60, 725–733 (1976)]. The jet oscillation observed by flow visualization is compared with theories by Cremer and Ising [Acustica 19, 144–153 (1967/68)] and Fletcher [Ann. Rev. Fluid Mech. 11, 123–146 (1979)]. Both theories lack a firm mathematical background; there are, however, no better theories available at the present time. Vortex shedding at the labium is observed in the course of the starting transient and for high-blowing pressures during stationary oscillations. Theory indicates that these vortices can induce a significant nonlinear damping for stationary oscillations.
Computer processing of audio signals by exclusion filters
Authors: Maher Robert C.
Published: October 2005

A common task in computer music and electroacoustic signal processing is additive mixing of two audio signals. If the two input signals contain discrete spectral components, their sum will typically contain amplitude beating and other interactions between pairs of components with similar frequencies. A new method is described that suppresses spectral interactions during mixing by deriving a time variant “exclusion filter” from the short-time spectrum of one of the signals in order to prefilter the other signal. This technique allows one of the signals (dominant) to pass through the mixing process with little modification, while the other signal (secondary) is prevented from interaction by attenuation of its conflicting spectral components. The exclusion filter is specified in a flexible manner, which can include such psychoacoustic criteria as critical bands. Audio examples will be presented.
Correlation between frequency-specific ABR and behavioral thresholds in 3-month-old infants
Authors: Werner Lynne A. , Rickard Lisa K. , Folsom Richard C.
Published: October 2005

The absolute sensitivity of 3-month-old human infants was estimated using ABR and a behavioral measure. The stimuli were tone pips with 2-cycle rise, no steady state, and 2-cycle fall. Frequencies of 1, 4, and 8 kHz were used. For the ABR, thresholds were determined by reducing the level of the tone pip from 20 dB nHL until the response was no longer observed, and then increasing the level until a response was again seen. Threshold was defined as the lowest level at which the ABR could be observed. Behavioral thresholds were determined using an adaptive procedure. Threshold was defined as the average of the last eight reversals obtained. A trial was considered “correct” if an observer said “yes” when a signal was presented to the infant, or ”no” when no signal was presented. The observer had no prior knowledge of the type of trial being presented, and none of the adults in contact with the infant could hear the tone pips. The correlation between ABR and behavioral thresholds depended on the frequency of the tone pip. This suggests that different factors contribute to between-subject variability in the two measures, but that under certain circumstances, infant behavioral threshold can be predicted from the ABR threshold with some degree of accuracy. [Work supported by NIH DC000520.]
Artificial viscosity effects on blast pulses in a nonlinear finite difference outdoor sound propagation program
Authors: Sparrow Victor W. , Raspet Richard
Published: October 2005

In a finite amplitude acoustic blast wave propagation program developed at the University of Illinois, Urbana-Champaign and at USA-CERL a fourth-order artificial viscosity numerical technique is used. The program is based on the method of finite differences, and the artificial viscosity acts to attenuate very high frequencies generated by nonlinear steepening effects. Such high-frequency components cannot be resolved easily by the finite difference grid due to computer memory limitations. In this presentation the physical basis for the fourth-order artificial viscosity will be given. Additionally numerical experiment results will bc described that were used to find minimum artificial viscosity coefficients appropriate for blast waves propagating in a free-field ANSI standard atmosphere. As expected, higher minimum artificial viscosity values were required for stronger blasts. [Work performed at the U.S. Army Construction Engineering Research Laboratory (USA-CERL), Champaign, IL. Supercomputer support from the National Center for Supercomputing Applications (NCSA), Champaign, IL.]
Scattering of sound by atmospheric turbulence: Predictions in a refractive shadow zone
Authors: McBride Walton E. , Bass Henry E. , Raspet Richard , Gilbert Kenneth E.
Published: October 2005

According to ray theory, regions exist in an upward refracting atmosphere where no sound should be present. Experiments show, however, that appreciable sound levels penetrate these so-called shadow zones. Two mechanisms contribute to sound in the shadow zone: diffraction and turbulent scattering of sound. Diffractive effects are more pronounced at lower frequencies and are negligible in the limit of high frequencies. In the short wavelength limit, then, scattering due to turbulence should be the predominant mechanism involved in producing the sound levels measured in shadow zones. There does not exist an analytical method for the prediction of sound-pressure levels in upward refractive shadow zones resulting from turbulent scattering. In order to obtain quantitative average sound pressure level predictions, a scattering center-based numerical simulation of the effects of atmospheric turbulence on sound propagation is performed. [Work supported by U.S. Army Construction Engineering Research Laboratory.]
Training paradigms for multichannel tactile aids
Authors: Weisenberger Janet M. , Brenner Christine , Grant Ken W.
Published: October 2005

Because connected speech presented through a multichannel tactile speech aid results in a complex and rapidly changing signal, it has been assumed that subjects having no prior tactile aid experience would require considerable amounts of “analytic” tactile training with phoneme and word-level single-item identification tasks, before they could usefully interpret connected speech. While it is the case that subjects who receive substantial amounts of analytic training with tactile aids perform better than subjects with no training, it may be the amount of prior experience, rather than the specific training tasks employed, that determines asymptotic performance levels. In the present study, groups of subjects were trained with a multichannel tactile aid in either an “analytic” condition (a graduated series of single-item tasks) or a “nonanalytic” condition (connected speech tracking only). Each group received the same number of hours of tactile aid experience. During the final 20 h of the study, all subjects were tested with connected speech tracking. Results comparing these 20 h of tracking for the two groups showed no significant differences in performance. However, substantial intersubject differences indicated possible influences of other variables, including talker effects and subject familiarity with phonetics. These issues were addressed in a second experiment. [Work supported by NIH.]
Evaluation of a wearable multichannel tactile aid
Authors: Brenner Christine , Weisenberger Janet M.
Published: October 2005

In designing a tactile aid, the most important considerations are that a device provide useful information and that it be wearable. While electrotactile rather than vibrotactile devices are more easily made wearable, results with these devices to date have been disappointing. One exception to this has been the Tickle Talker (Univ. of Melbourne). The Tickle Talker is an eight-channel device, worn as rings on the fingers of one hand, with a electrode placed on either side of each finger. The processor performs a formant extraction, displaying an estimated fundamental frequency, second formant frequency, and the amplitude of the speech signal. In an attempt to corroborate early promising results [Cowan et al., J. Acoust. Soc. Am. 85, 2593–2607 (1989)], this device was evaluated with three normally hearing adult subjects. In initial testing with closed set tasks done under a tactile aid alone condition, subjects on the average were able to identify the manner of production of stimuli 76% of the time, presence or absence of voicing 74% of the time, and medial vowel 81% of the time. Wearing the tactile aid in conjunction with lipreading raised subjects' scores on a subsequent closed set identification task from 57% lipreading alone to 72% with the device. Results to date with connected discourse tracking also show improvements with the device. [Work supported by NIH.]
Discrimination of electrical trajectories by cochlear implant patients
Authors: Busby P. A. , Tong Y. C. , Clark G. M.
Published: October 2005

Psychophysical studies were conducted on eight cochlear implant patients to measure the discrimination of trajectories with time-varying electrode positions, repetition rates, or current levels. Electrical stimulation was by means of the University of Melbourne/Cochlear multiple-electrode prosthesis. Four patients were postlingually deaf adults and four were children who were deafened prior to 3 yr of age. In each study, the parameter values of the stimuli were systematically varied. The performance of the adults was, in general, superior to that of the children for the discrimination of electrode position and repetition rate trajectories. There were no major differences in performance between the adults and children for the discrimination of current level trajectories.
Nonlinear dynamical behavior of a free-standing metglas ribbon
Authors: Cawley Robert
Published: October 2005

Dynamical oscillations of a vertical, free-standing magnetoelastic ribbon clamped at the base are induced by parametric driving. The Young's modulus is a nonlinear function of an externally applied axial magnetic field, H = H0 + H1 cos 2πft, where H0 and H1 are dc and ac amplitudes, respectively, of the drive, and f is the applied frequency. Field parameters and ribbon length are chosen so that gravitational buckling occurs. A variety of nonlinear dynamical effects have been observed, including crisis-induced intermittency [W. L. Ditto et al., Phys. Rev. Lett. 63, 923 (1989)] and strange (fractal attractor), nonchaotic (no positive Lyapunov exponents) behavior [W. L. Ditto et al., Phys. Rev. Lett. 65 (1990)]. The experiments, some of the results, and a simple model (if time permits) are described.
Complex dynamics in a simple mechanleal system
Authors: Tongue Benson
Published: October 2005

The last decade has seen an explosion of interest in the field of nonlinear dynamics, most especially those dynamics associated with chaotic motions. A tremendous amount of analysis has been done, and is ongoing, with respect to differential and difference equations and the behavior that they can support. Unfortunately, the number of physical demonstrations of chaos in mechanical systems is nowhere near as great. In the area of structural mechanics, the most common demonstration is that of one or two relatively flexible beams that, when excited by an external force, give rise to large amplitude nonlinear oscillations. Although certainly useful, such constructs implicitly involve an infinite-dimensional system and are typically operated at amplitudes well beyond any likely to be found in practice. With these points in mind, a single degree of freedom slider crank was constructed, the motivation for which was to see if chaos could be found in such a simple and common mechanism. Interestingly, chaos turned out to be robust in the actual device, although it was more elusive in the numerical model. The reasons for this and some conclusions with regard to chaos in physical systems will be presented.
Application of neural networks to signal processing and control problems
Authors: Widrow B.
Published: October 2005

Neural networks can be used to make nonlinear transversal filters for signal processing. In some cases, signal estimation in noise can be enhanced by nonlinear filtering. Examples will be presented. Applications of neural networks in nonlinear control systems will also be presented. Self-learning neural control systems for continuous bang-bang control (the “broom-balancer”) and for transient analog control (The “truck backer-upper”) will be explained and demonstrated.
Application of artificial neural networks to the classification of underwater ambient noise signatures
Authors: Kriebel John N.
Published: October 2005

More than 2500 ambient noise samples were collected in Behm Canal, Alaska, between July 1989 and January 1990. The data were obtained using a system deployed from a MINIMET buoy, and consist of one-third octave analyses covering the 50 Hz through 63 kHz bands. Signatures obtained on an hourly schedule, were transmitted via the ARGOS satellite and entered into a LOTUS 1-2-3 database. Attempts to categorize the data by conventional means, e.g., manual sorting of plots of the signatures, or on a purely statistical basis, were unsatisfactory: thus the application of artificial neural networks was investigated. This approach yielded reasonable classifications and allowed atypical signatures to be identified and deleted from the database prior to performing statistical analysis. The groupings assigned by the network parallel those which might have been made by an analyst in that obviously different signatures are assigned to different classes, but in some cases the network makes more subtle distinctions than an analyst might. One network separated the signatures into 22 major groups (those containing at least 26 signatures) that include 90% of the data.
Modal analysis of tennis rackets
Authors: Korte Rodney , Rossing Thomas D.
Published: October 2005

The modes of vibration of four tennis rackets are compared: two of the new “wide-body” rackets, a midsize racket with a graphite composite frame, and an inexpensive midsize racket with an aluminum frame. In all four rackets, the strings vibrate in modes that resemble those of a membrane, while the frames show a variety of bending and torsional modes. The coupling between the string modes and the frame modes is weaker than expected. The enormous popularity of wide-body rackets is attributed to their greater “power,” resulting from greater frame stiffness. This is especially evident in the higher frequencies of the torsional modes of the frame, although bending modes occur at higher frequencies in the wide-body rackets as well. The wide-body rackets in this study were furnished by Pro-Kennex.
Segmentation of the Helmholtz integral operator and its application to a boundary integral element code
Authors: Leyrat G. , Hamdi M. A.
Published: October 2005

One of the major drawbacks of any finite-element method applied to structural acoustics is its poor ability to deal with problems involving large values of ka, where k is the acoustic wave number and a is the largest length of the system considered. This paper presents an algorithm that overcomes this limitation by segmenting the global assembled matrix of the boundary integral element formulation into small matrices. The major difficulty in implementing this algorithm is that the assembled matrix, although symmetric, is full and complex, which is not shown that this algorithm drastically decreases the required computational time for repetitive structures, since the matrices corresponding to each subassembly are computed only once.
Poisson's ratio determination of foamed aluminum by laser Doppler vibrometry
Authors: Dubbelday Pieter S. , Amberntson Carolynn A.
Published: October 2005

Foamed aluminum is a promising candidate in hydroacoustics for pressure-release material in the dry state, and for sound absorption, when saturated with a viscous liquid. Previously the dynamic shear modulus measurement of this material has been reported, as a function of density and pore size [Proc. IEEE 1985 Ultrasonics Symp. 2, 1052–1055 (1985)]. To characterize the elastic properties, at least one other modulus measurement is required. It is difficult to conceive how the bulk modulus could be determined. The dynamic Young's modulus may be found by measuring the extension of a beam by means of accelerometers. In the present experiment the dynamic Poisson's ratio and Young's modulus of the same sample were determined by the use of laser Doppler velocimetry. The results are given for foamed aluminum of varying porosity and pore size, and the consistency with measurements of the shear modulus is discussed.
Masking of formant transitions and stop bursts
Authors: Alwan Abeer
Published: October 2005

In this study a model of the mechanisms of speech perception in noise is introduced and evaluated by comparing the outcomes of the model to results of perceptual experiments. The model is based on the theory of auditory masking. As a case study, the alveolar/labial distinction for the stop consonants /d,b/ is considered. Two experiments were conducted using synthetic /ba/ and /da/ syllables differing only in the shape of the F2 trajectory; a falling trajectory was used for /da/ and a rising trajectory for /ba/, and the transition duration for both stimuli was 40 ms. An appropriate burst was added at the release of the consonant in the second experiment. The stimuli were mixed with a white-noise masker at various signal-to-noise ratios and presented to subjects for identification. Results showed that when the first 24 ms or more of the F2 trajectory for the /da/ stimulus are masked, /ba/ is heard. Results of the second experiment show similar results, indicating that the F2 transition, rather than the burst, is the primary perceptual cue for the alveolar/labial distinction when the signal is corrupted by white noise. Results of similar experiments using different spectral shapes of noise will also be reported. [Work supported in part by NIH.]
An acoustic-phonetic diagnostic tool for the evaluation of speech-analysis methods
Authors: Ghitza Oded
Published: October 2005

Most speech processing systems (e.g., speech recognition systems or speech coding systems) contain a feature-analysis stage that extracts the required task specific information from the speech waveform. This study addresses the question of how to identify what part of the speech information is lost in this process. To answer this question, a diagnostic system has been developed that simulates the psychophysical procedure used in the standard Diagnostic Rhyme Test (DRT). In the psychophysical procedure, the subject has all the cognitive information needed for the discrimination task a priori. Hence, errors in discrimination are due mainly to inaccuracies in the auditory representation of the stimulus. In the simulation, the human observer is replaced by an array of recognizers, one for each pair of words in the DRT database. The recognizer used [O. Ghitza and M. M. Sondhi, J. Acoust. Soc. Am. Suppl. 1 87, S107 (1990)] was designed to keep errors due to the recognition procedure to a minimum, so that the overall detected errors are due mainly to inaccuracies in the front-end representation. The system provides detailed diagnostics in the form of a map that shows the error distributions among six phonetically distinctive features. Results are given for the behavior of two speech analysis methods, a cepstral representation, and an auditory-based representation, in quiet and in a noisy environment. These results will be compared with psychophysical results for the same database.
Smoothing formant motion with triggered temporal integration
Authors: Patterson Roy
Published: October 2005

Vowel transitions sound smooth even though the neural activity patterns (NAP) in the auditory nerve oscillate and glottal intervals change size in discrete steps. Temporal integration (TI) can be used to stabilize auditory images and to smooth transitions, but traditional TI mechanisms smear out timing details that we hear if the integration time is long enough to produce smooth motion in speech sounds. The problem can be solved by performing TI separately within NAP channels and triggering the TI to occur at times corresponding to the larger pulses in the channel. The triggering process converts the rapidly flowing NAP into a high-resolution auditory image in which “what you see is what you hear.” Formants that are audible are visible and global phase changes that are not heard are removed. Video illustrations show that vowel transitions produce smooth motion that occurs at the same rate as the motion that is heard in the sound. The auditory model requires about 50 MFLOPS for real-time operation, which makes it a candidate for ASR. [Work supported by U.K. RAE 2239 and Esprit BRA 3207.]
Self-surveying techniques for matched-field processing
Authors: Tran Jean-Marie , Hodgkiss W. S.
Published: October 2005

The classic problem of matched-field processing is the detection and localization of (unknown) sources assuming a known oceanic environment and a calibrated vertical line array of known shape. If a broadcasting source is deployed at a known location, the array shape can be determined through assumptions on the environment, or the environmental characteristics (such as the sound-speed profile) can be determined if the array shape is known. These two problems are investigated in the framework of matched-field processing. First, the array shape calibration is discussed and a simple procedure, using a correlation measure and a standard multidimensional optimization algorithm, is proposed. This procedure is used on simulated data (where convergence is observed) and on real data collected at sea in September 1987 with a 900-m-long array with 120 sensors. The experimental results are found consistent with those obtained through previous analysis. Then, the problem of calibrating the oceanic environment is studied using a normal mode formulation. Bucker's self-cohering approach to correct for tame environmental mismatch in the matched-field processor output [H. Bucker, 4th MFP Workshop, 6–8 Sept. 1989, Victoria, Canada], provides a convenient measure of the environmental mismatch for a range-independent medium. An approach to tomography, using a vertical line array that samples well the normal modes, is proposed. The preliminary simulation results show that the Bucker correction factors provide a smooth measure of sound-speed mismatch and that the inversion for the sound-speed profile should be possible through optimization. [Work supported by ONR.]
Application of phase conjugation to underwater acoustics
Authors: Jackson Darrell R. , Dowling David R. , Belcher Edward O.
Published: October 2005

Phase conjugation shows promise as a means of compensating for multipath propagation and array deformation in underwater acoustics. In acoustic phase conjugation, an initial probe signal is received by a large, multielement array. If the output of each element is time-reversed and retransmitted, the resulting field will converge on the location of the probe transmitter and will compensate for multipathing. Applications of phase conjugation to underwater acoustics will be illustrated. Constraints imposed by aperture size and time variation of the medium will be discussed. [Work supported by ONR.]
Bubble-related sources of sea-surface sound
Authors: Crum L. A. , Roy R. A. , Yoon S. W. , Kolaini A. R. , Nicholas M.
Published: October 2005

The observation that oceanic ambient noise is wind dependent over a broad frequency range [G. Wenz, J. Acoust. Soc. Am. 34, 1936 (1962)] suggests a major role for gas bubbles as sources of this sound. Bubble noise generation mechanisms in a variety of modalities and under both passive and active surveillance are examined. The preliminary studies have examined topics such as collective oscillations of bubble clouds, bubble entrainment by capillary waves, and precipitation noise. A brief overview of this research is presented and the most recent results are described in some detail. [Work supported by ONR and ONT.]
Acoustical tracking of breaking surface waves
Authors: Farmer David M. , Ding Li
Published: October 2005

The sound generated by breaking surface waves can be used to track their location, lifetime, and motion across the ocean surface. A six-hydrophone array of span 8.5 m and 5-kHz bandwidth has been developed. It forms part of a freely drifting and self-contained instrument that also includes various active acoustical systems for studying bubble clouds; it was first used in the Surface Wave Processes Project in February/March 1989. Preliminary results will be presented, focusing on the pattern and behavior of the breaking in different ambient wave environments.
Review of speech intelligibility measures
Authors: Pavlovic Chaslav V.
Published: October 2005

Currently, various procedures exist for physical measurements of speech intelligibility (AI, STI, RASTI, etc.). A large majority of them are either based on or use some concepts of the articulation theory. In this talk the articulation theory will be critically reviewed and parameters and procedures that give best results will be recommended. The specific recommendations will vary depending on the application. Therefore, the result will be a global and unified model incorporating the best elements of the contemporary procedures. Among the variables that will be discussed are the basic theoretical foundations of the theory, speech dynamic range, importance of various frequencies to speech intelligibility, ways to account for the effects of various distortion types on speech intelligibility, accurate measurement of the prediction parameters, possible errors, etc. Finally, the meaning of the term “speech intelligibility” will be reexamined. It will be hypothesized that the intelligibility of speech that occurs in daily communication activities is better specified by physical measures than by any speech recognition testing. In fact, in many circumstances, the latter could be completely misleading.
Application of the speech transmission index (STI) and the articulation index (AI) to the hearing impaired
Authors: Humes Larry E.
Published: October 2005

Many of the listeners for whom physical measurements of speech intelligibility are performed have a significant hearing loss. Some general principles regarding the methods used to modify physical measurements of speech intelligibility for use with the hearing-impaired population are reviewed in this paper. This paper focuses on two specific physical measurements of speech intelligibility: the speech transmission index (STI) and articulation index (AI). Applications of the STI and AI reviewed in this paper include measurements made for hearing-impaired listeners with or without the use of hearing aids and with or without the use of hearing-protection devices. Both the STI and the AI are found to be accurate predictors of the average speech-intelligibility performance of groups of hearing-impaired subjects. It is not possible, however, to accurately estimate the performance of individual hearing-impaired listeners, especially for those over age 65 or those having steeply sloping high-frequency hearing loss. [Work supported by NIH.]
Maximum likelihood estimates (MLEs): An alternative approach for assessing speech intelligibility data
Authors: Smith Caldwell P.
Published: October 2005

The Diagnostic Rhyme Test (ANSI S3.2-1989) for assessing speech intelligibility results in arrays of scores for phonetic features, talkers, and processor conditions that can readily be converted to arrays of frequency counts representing numbers of correct responses by listeners. This transformation creates an opportunity for analyzing intelligibility data as contingency tables and applying the methods of discrete multivariate analysis to separately assess effects contributed by phonetic features, by talkers, and by processor conditions. The method is one of testing hypotheses: Particular combinations of main effects and interactions are postulated, and maximum likelihood estimates (MLEs) for each cell in the array under those constraints are calculated. The model is compared with the source array by means of chi-squared or G-squared measures, with the result providing a basis for rejecting or not rejecting the hypothesized independence from those effects omitted in the model. A pilot study was conducted in which this method was applied in comparisons of speech intelligibility from various processing conditions, as well as data from replicated tests. Results were compared with those from traditional analysis of variance, and with the use of the t test for paired data items.
Psychophysically derived auditory filters in chinchillas
Authors: Niemiec Andrew J. , Shofner William P.
Published: October 2005

A positive-reinforcement behavioral tracking technique was used to measure detection thresholds for a pure-tone signal (f0) presented against a background of notched noise in a simultaneous masking paradigm. Notched noise was generated by band-reject filtering a white noise at a rolloff of 96 dB/oct. The spectrum level of the noise was 43 dB. Detection thresholds were measured for a number of conditions in which the notch width was varied from 0 to 0.8 times f0 in steps of 0.1 times f0. The derivatives of the masking functions were computed to obtain the filter functions [R. D. Patterson, J. Acoust. Soc. Am. 59, 640–654 (1976)]. Behavorial data from the simultaneous masking paradigm indicate that auditory filter shapes of chinchillas and humans are similar, both having a simple bandpass characteristic. However, the chinchillas' auditory filters are more broadly tuned than those of humans, reflecting poorer frequency selectivity. [Work supported by NIH-NIDCD center grant.]
Can cochlear maps predict behavioral performance in frequency analysis in mammals and birds?
Authors: Fay R. R.
Published: October 2005

A modification of a cochlear position-frequency map function [D. D. Greenwood, J. Acoust. Soc. Am. 33, 1344–1356 (1961)] permits tentative maps to be constructed for mammals and birds using two species parameters: basilar membrane length and the upper frequency hearing limit. The slope of the map function (Hz/mm) estimates the form of the critical bandwidth (and other behaviorally equivalent units of frequency analysis) function of frequency once the critical cochlear distance (CD) corresponding to a critical band (or other unit) is known. Fits of the slope function to critical bands, critical masking ratio bands, frequency discrimination thresholds, and psychophysical tuning curve widths for 10 mammal and 12 bird species estimate respective CDs, and support the notions that: (1) Most mammal and bird cochleas analyze frequency similarly, (2) CDs are about equal for middle frequencies within a species, and (3) the various behavioral measures of frequency analysis are related. The hypothesis that critical band CDs for most mammals are equal (about 1.3 mm) cannot be rejected, but birds show a significantly smaller CD (about 0.4 mm). [Work supported by a center grant from NINCDS.]
Applications of structured modeling and control in acoustics
Authors: Busch-Vishniac Ilene J.
Published: October 2005

Methods appropriate for efficient design, analysis, and implementation of acoustic systems have been difficult to develop because most acoustic systems are multienergic (i.e., embracing more than one energy domain) and contain both discrete and continuous subsystems. Structured modeling refers to a unified approach to the development of models of subsystems so that they may be easily combined to form a coherent picture of a complex physical system. This approach views subsystems in terms of the transfer of power and energy into them, through their components, and out of them. Development of a formal heirarchical procedure permits either a top-down design approach, in which general specifications and tolerances are given and a computer code searches for the optimum consistent design, or a bottom-up analysis of a specified acoustical system. Specific research areas include means of modeling systems with more than one scale (such as a model of the molecular behavior of a material whose macroscopic characteristics define the port characteristics of a transducer), development of a logical means for construction of reduced order models suitable for control and design synthesis, and inclusion of the second law of thermodynamics in models so that thermal characteristics can be considered as an integral part of the model. Applications in acoustics include transduction, the development of models for nonlinear wave propagation (including thermodynamical effects), and shock and vibration analysis. [Work supported by Texas Advanced Research Program.]
Vibrating string motions on videotape
Authors: Hanson Roger J.
Published: October 2005

Optical detection of string motion in two orthogonal directions has been used to examine the trajectory of a point on a vibrating string (or wire). The outputs of the optical detectors are exhibited on an X-Y oscilloscope that, along with a frequency analyzer display, is photographed with a video camera. The system is used to show the complex motions that can result from nonlinear effects that occur if the usual simplified textbook assumption of constant tension is not valid. If a wire is driven electromagnetically with a planar force, the sudden onset of motion in the plane perpendicular to the driving force is observed to occur to a critical frequency. The resulting tubular motion commonly occurs in elementary student experiments on a driven vibrating string. A point on a plucked string can also be observed to follow the precessional motion predicted from nonlinear effects. Videotapes will be shown of these phenomena along with a short clip from the “What If…” series of video vignettes by John Wiley and Sons illustrating sound spectral effects in aged guitar strings.
Standing solitons
Authors: Denardo Bruce
Published: October 2005

Systems that support spatially periodic standing waves may also allow localized modulations of such modes. These are fundamentally nonlinear waves that can exist at any location of the medium. They are examples of solitons. Depending upon the nature of the mode and the nonlinearity, two varieties of standing solitons can occur: breathers and kinks. Observations of these solitons will be shown in a videotape. One system involves transverse motion of a lattice of coupled pendulums; the other involves surface waves on a liquid in a long channel. The systems are driven so that steady-state motion occurs. The existence of the solitons can be simply understood as a local balance between nonlinearity and curvature of the modulation. This understanding leads to the prediction of these solitons in a variety of systems.
The surface manifestation of internal waves: Doppler sonar observations from SWAPP
Authors: Pinkel Robert , Green Lloyd
Published: October 2005

During the Surface Wave Processes Program (SWAPP, February–March 1990), a ten-beam dual frequency Doppler sonar system was used to observe the sea surface velocity field. The sonars were mounted on the research platform FLIP and oriented at various angles in azimuth, forming a directional array. Measurements were obtained out to a range of 1100m. Here, a preliminary effort to use the array to observe that component of the sea surface velocity field associated with internal waves is presented. The internal wave signal exists beneath the far stronger signatures of the surface wave and Langmuir fields. The utility of this approach depends on the ability to isolate these competing motions from the total velocity field. The directional properties of the open ocean internal wave field remain largely unknown. The large horizontal scales of the waves, coupled with the great range of scales, render the measurement problem difficult. The desire for adequate internal wave directional measurements prompted the original development of Doppler sonar at S.I.O. in 1974. Sections of the 20-day Patchex time series are examined, through a variety of weather and surface wave conditions. Emphasis is placed on the problem of distinguishing small-scale waves from nonpropagating Langmuir cells. In addition, an attempt to use the full 2-km aperture of the array to estimate the directional properties of the baroclinic tide will be discussed.
Wave dynamics and surface statistics in the framework of the weak-turbulence theory
Authors: Glazman Roman E.
Published: October 2005

Most traditional notions concerned with equilibrium sea have been prompted (in the 1960s and 1970s) by field and laboratory observations characterized by a relatively low degree of sea development (wave age ξ = C0/U is well below unity, where C0 is the phase speed of waves corresponding to the spectral peak and U is the mean wind speed well above the surface). In contrast, in the case of open-ocean waves, it is more typical that ξ > 1. Consequently, some of the common notions, for instance, wave groups, fully-developed sea state (the Pierson-Moscowitz model), energy dissipation rate, sea surface skewness, and other non-Guassian statistics, etc., need considerable revision, as they acquire a very different physical meaning as well as different quantitative ratings. Based on the weak-turbulence theory for equilibrium sea (due largely to V. Zakharov), which is appropriate for ξ⩾1, the corresponding revisions are proposed and their implications for statistical geometry of the sea surface are demonstrated. The results include a fairly complete statistical description of the field of steep and breaking waves related to intermittent whitecaps and bubble clouds. Various data products yielded by satellite altimeter, scatterometer, and microwave radiometer at different ξ, as well as some recent field measurements of open ocean waves, appear to support the theory. It is hoped that additional data might be inferred from acoustical measurements.
The relationship of singular value decomposition to wave-vector filtering in acoustic radiation problems
Authors: Photiadis D.
Published: October 2005

Singular value decomposition (SVD) has recently been suggested as a filtering tool to analyze acoustic radiation (G. Borgiotti, J. Acoust. Soc. Am., submitted). The relationship between this filtering technique and the more familiar wave-vector filtering is examined. The SVD algorithm applied to the Green's function connecting a local source distribution to the farfield radiated pressure yields acoustic “source modes” and accompanying “radiation modes.” For a geometry with a preferred axis, the source “modes” are found to consist approximately of a single wave number, and the accompanying radiation modes are thus beams. Some simple examples are worked out to illustrate the general behavior of the filtering, and some of the properties of the beams are given.
Envelope-weighted average of instantaneous frequency (EWAIF) model applied to narrow bandwidth signals
Authors: Feth Lawrence L.
Published: October 2005

The ability of the auditory system to follow amplitude and frequency fluctuations in complex sounds is essential to the perception of speech and music. The EWAIF model was devised to study this ability using narrow bandwidth signals. Narrow bandwidth signals may vary in amplitude, frequency, or both (simultaneously). In one series of experiments, listeners were asked to distinguish between pairs of frequency-modulated sinusoids in which the frequency modulation was identical across the pair, but the amplitude modulation was systematically varied. This was accomplished by skewing the waveform envelope of linear FM glides. For another series, amplitude modulation remained the same, but the frequency modulation of the pairs of sinusoids was varied. Here, the frequency trajectory was made to deviate sinusoidally from a linear glide. Listeners are very sensitive to small differences in either envelope skew or frequency trajectory. Performance of listeners with normal hearing will be compared with predictions derived from the EWAIF model. [Work supported by a grant from AFOSR.]
Effects of adding frequency differences to temporal patterns on detection and recognition
Authors: Weber Daniel L.
Published: October 2005

Temporal patterns were generated as sequences of 1-kHz sinusoids with different durations and temporal separations. Tonal patterns were generated as sequences of 100-ms sinusoids with different frequencies. Testing four-and seven-component temporal patterns led to recognition performance below that predicted by the detection/recognition theorem [Starr et al., Radiology 116, 533–538 (1975)]; listeners demonstrated performance at predicted levels when tonal patterns in different frequency regions were tested. To examine how frequency differences might improve performance for the temporal patterns, hybrid patterns were created by (1) adding a new component with a different frequency to each temporal pattern and (2) changing the frequencies of different numbers of components in the temporal patterns to match those used in the tonal patterns. Although individual differences were large, a characteristic of research with the detection/recognition theorem, the addition of a different frequency component generally raised detection performance but did not improve recognition performance beyond the level predicted by this increase in signal detectability. In contrast, the combination of temporal and tonal patterns led to a relative improvement in recognition performance that was not accounted for by the theorem and resulted in performance much better than for either temporal or tonal patterns alone. [Research supported by AFOSR through WPAFB AAMRL/BBA.]
Overshoot in subjects with sensorineural hearing impairment
Authors: Bacon Sid P.
Published: October 2005

Overshoot was measured in both ears in subjects with normal hearing and in subjects with unilateral high-frequency, sensorineural hearing loss by measuring threshold for a 10-ms signal presented 1 or 195 ms after the onset of a 400-ms broadband noise masker. Masker spectrum level was 20, 30, or 40 dB SPL. At 4.0 kHz, where absolute thresholds were elevated in the impaired ears, overshoot (the difference between thresholds at 1 and 195 ms) was small or absent. This may indicate that hearing loss alters a mechanism contributing to overshoot, or it may reflect the fact that sufficient amounts of masking (greater than about 20 dB) could not be produced except at high masker levels, where overshoot can be small or absent in normal ears [S. P. Bacon, J. Acoust. Soc. Am. (in press)]. In addition, overshoot was smaller at 1.0 kHz in the impaired ear than in the normal ear of the unilaterally impaired subjects, despite the fact that absolute thresholds were normal at that frequency. This, too, may reflect a damaged peripheral mechanism, or it may reflect the normal consequence of a relatively narrow, “effective” masker bandwidth. [Work supported by NIDCD.]