bluuebunny/crossref_metadata_2025 huggingface dataset:
How do humans process and recognize speech?
Authors: Allen Jont B.
Published: October 2005

Until the performance of automatic speech recognition hardware surpasses human performance in accuracy and robustness, one stands to gain by understanding the basic principles behind how humans recognize speech. This problem has been studied extensively at Bell Labs between the years of 1920 and 1950 by Harvey Fletcher and his colleagues. The motivation for these studies was to quantify speech sounds in the telephone plant and to improve speech articulation and quality. To do this they studied the effects of filtering and noise on the average speech articulation errors for nonsense CVC syllables, words, and sentences. During WWII these studies were continued at Harvard, but under much more adverse conditions of filtering, noise, and distortion. These studies have recently been extended to include the effects of reverberation degradation due to changes in the modulation transfer function. Fletcher and his colleagues derived a linear speech information density function D(ω) for the CVC’s and found a formula that accurately predicts the average CVC errors. The area under D is called the articulation index. Many people are not familiar with Fletcher’s derivation, which is very insightful. He then went on to find the relations between the errors for these nonsense speech sounds and words and sentences. This work has recently been reviewed and extended by Boothroyd. Taken as a whole, these studies can tell us a great deal about how humans process and recognize speech sounds.
Auditory front ends in speech recognition systems
Authors: Hirahara Tatsuya
Published: October 2005

The performances of several auditory front ends were evaluated in a phoneme recognition task using a VQ-HMM or an LVQ2 back end, and in a word recognition task using a DTW back end. The auditory front ends used in the experiments were different combinations of a fixed-Q cochlear filter (FQF), an adaptive-Q circuit (AQC) [T. Hirahara etal., Proc. ICASSP, 496–499 (1989)], an inner-hair-cell model (IHC) and a lateral inhibition (LINH). Traditional DFT-based and LPC-based front ends were examined for comparison. In the VQ-HMM/LVQ2 phoneme recognition task, the AQC and the LINH improved the performance in both noise-free and noisy conditions. However, the best performance provided by an auditory front end was the same as that of the DFT front end. Unexpectedly, the IHC degraded performance in most cases. Hence, it was concluded that the auditory front ends provide little benefit. In the DTW word recognition task, the AQC improved the robustness for noise and speaker variation. The LINH greatly improved robustness for noise but degraded that for speaker variation. The best performance for noisy speech was obtained by the combination of FQF, AQC, and LINH, which outperformed traditional front ends thoroughly. Similarly, the best performance for multiple speakers was by the FQF in conjunction with the AQC. This combination completely outperformed other front ends. Hence, it was concluded that the auditory front end pays off. These results indicate that the difference in back ends and the task difficulty strongly affect the evaluation of the front ends.
Acoustic correlates of vocalic nuclei of syllables
Authors: Miller James D. , Gottfried M. J. , Meyer D. J. , Frederick E. J.
Published: October 2005

Values for the fundamental frequency and F1, F2, and F3 were obtained for a corpus of 1248 vocalic nuclei from CVCs (26 phonemically different vocalic nuclei×4 speakers×2 stress rate conditions×3 consonantal contexts×2 repetitions of each token) at 25 equally spaced times within each vocalic nucleus. This corpus included monopthongs /i, i, (eh), æ, a, (inverted vee), (open oh), u, u/, diphthongs /au, ai, ei, ou, (open oh)i/, rhotic vowels /ir, er, ar, or, (hooked backward he)/, and vowels followed by /(el with tilde)/—/i(open oh), i(el with tilde), i(eh), a(el with tilde), (ivertd vee)(el with tilde), (open oh)(el with tilde), u(el with tilded)/. These were spoken in /b—d/, /d—d/, and /g—d/ contexts. A standard back-propagation neural network with one hidden layer was trained to identify which of these 26 vocalic nuclei was spoken. Input data for the neural network was presented as (1) an auditory-perceptual space using (x’y’z’) or (2) other implementations of the fundamental frequency and the first three formants. Preliminary results indicate that the neural network is able to identify these nuclei on the basis of acoustic parameters alone. [Work supported by NIDCD.]
Encoding amplitude modulation in the cochlear nucleus: A model
Authors: Lorenzi Christian , Berthommier Frederic , Tirandaz Nosrat
Published: October 2005

A physiological model of a post-cochlear mechanism responsible for enhancing the phase-locked activity of auditory neurons is proposed, when excited by sinusoïdally amplitude-modulated (AM) tones. In the cochlear nuclei, chopper units respond to the summated low-frequency components of their peripheral inputs. This feature is due to the time constant of the excitatory postsynaptic potential. It is also the result of the specific time course of the cellular membrane potential during the refractory period. The physiologically plausible model is derived from Hodgkin–Huxley equations. AM tones and white noises were applied into the chopper model. In both experiments, the responses of the model showed a strong phase-locking for modulation frequencies between 100 and 300 Hz. The mean gain of the chopper model suggests that chopper units enhance the phase-locked AM responses that they receive as inputs from auditory-nerve fibers. These results agree with earlier electrophysiological studies.
Involvement of medial olivocochlear system in detection in noise
Authors: Micheyl Christophe , Collet Lionel
Published: October 2005

In order to investigate the potent role of the efferents in perception in noise, medial olivocochlear feedback and signal-in-noise detection abilities were both examined in 30 normally hearing subjects. Medial olivocochlear feedback was assessed in terms of reduction in click-evoked otoacoustic emission amplitude under a contralateral acoustic stimulation consisting of a 30-dB SL broadband noise. Monaural detection thresholds of 1- and 2-kHz tones embedded in broadband noise were measured with and without the same contralateral stimulation sucessively. The finding of a correlation between the reduction in otoacoustic emission amplitude and the detection threshold of the 2-kHz tone in the presence of the contralateral stimulation suggests that the detection abilities are partly related to the operation of the medial olivocochlear feedback. The results indicate moreover that in these subjects with the strongest medial olivocochlear feedback, the detection performance is improved by the addition of noise in the opposite ear whereas it decreases or remains constant in the other subjects.
Eye-closed positional and positioning nystagmus in an abnormal population—A comparison with two previous normative data studies
Authors: Shiau Yea-Wen , Keith Bobert W. , Kretschmer Laura W. , Pensake Myles , Weiler Ernest M.
Published: October 2005

This retrospective study investigated the diagnostic power of two previous built sets of normative data (Barber and Wright, 1973; Schwallie, 1988) by examining eye-closed positional and positioning tests in 100 clinical subjects who had balance problems. The research questions were: (1) What is the prevalence of eye-closed positional and positioning nystagmus in an abnormal population? (2) Which normative data set is more useful in diagnosis? (3) Are the two sets of normative data useful in diagnosis? The findings were: (1) Essentially, the occurrence rate of pathologic nystagmus was higher in the abnormal population than normals. (2) Female subjects had more imbalance problems than males. The age range of female patients was 31 to 70 years, while most male subjects were between the age of 21 to 30. (3) Both sets of normative data were found to be of diagnostic use; however, there was no evidence to show which is more useful. (4) Horizontal nystagmus data were more applicable in ‘‘peripheral’’ diagnostic categories while vertical data were applicable to both CNS and peripheral categories. It is suggested that eye-closed positional and positioning tests with both vertical and horizontal nystagmus deserve more emphasis in clinical situations.
Detection of amplitude modulation with linear-frequency-modulated carriers
Authors: Pitton James W. , Burns Edward M. , Atlas Les E.
Published: October 2005

Results are presented for detection of sinusoidal amplitude modulation (AM) superimposed upon linear-frequency-modulated, 120-ms tone bursts. Thresholds for detection of AM were obtained using an adaptive 2IFC procedure over a range of AM frequencies from 25 to 400 Hz, and over a range of frequency modulation (FM) rates from 0 to 30 Hz/ms. The stimuli are a simplification of speech transitions, with both AM and FM, but consisting of only one signal component. These experiments explore the ability to detect modulations that are distributed across critical bands. For fast rates of FM, the stimuli will sweep through a critical band within one to two periods of the AM. A rapid frequency transition also broadens the spectrum of the signal, so that these stimuli do not possess the distinct sidebands of an amplitude-modulated, stationary tone. Thus, both temporal cues from periodicities within individual channels and place cues derived from the excitation pattern will weaken as the FM rate increases. Nonetheless, data indicate that listeners readily detect amplitude modulation for all modulation frequencies at the highest rates of FM. These data will be discussed in light of some current physiologically based models of auditory processing, wherein both within- and across-channel temporal information is utilized. [Work supported by the Virginia Merrill Bloedel Hearing Research Center, University of Washington.]
Temporal resolution of frequency-modulated signals
Authors: Madden John P.
Published: October 2005

Normal-hearing subjects were asked to discriminate between two sinusoidal signals. One, a 50-ms glide, moved from its initial frequency over a linear path to its final frequency. The other, the step signal, was the same except that its trajectory followed a series of discrete steps in frequency. The smallest frequency increase per step (FIS) at which the step signal could be distinguished from the glide was determined. These FIS thresholds (FISTs) were obtained for signals with 2 to 11 steps at center frequencies of 0.25 0.5, 1.0, 2.0, 4.0, and 6.0 kHz. At center frequencies from 0.5 to 2.0 kHz, the FIST was roughly constant for signals with up to about 8 steps. Beginning at about 9 steps, the FIST rose sharply, eventually reaching the limits of the possible FIS. It is inferred that beginning at this point, the duration of the individual steps (5–6 ms) limits discrimination. At center frequencies above 2000 Hz, performance deteriorated markedly, unlike the results of gap detection studies. The implications of these results for mechanisms tracking FM signals are discussed. [Work supported by NIH.]
A difference limen for periodicity pitch perception related to spectral change
Authors: Hawks John W. , Sapp Christine L.
Published: October 2005

Preliminary estimates to determine the relative amplitude relationship necessary for one spectral pattern to influence another in terms of the resulting periodicity pitch were made. The threshold of discrimination (DL) for a change in periodicity pitch was estimated for complex tones whereby either the even- or odd-order harmonic components were simultaneously decremented in amplitude. An adaptive up-down procedure employing a 2AFC task with two standards was utilized with minimally trained subjects. Stimuli were digitally constructed by summing sine waves spaced at 100-Hz intervals. Three stimulus parameters were evaluated: (1) total number of components (5 or 6); (2) position of varied components (even or odd order); and (3) frequency range (1400–1900 Hz or 2000–2500 Hz). Difference limens obtained ranged from −8.2 to −23.1 dB relative to the fixed component amplitude with an overall mean of −13 dB. The implications of these findings will be discussed.
Properties of the structure of multi-tone sequential patterns that determine the difficulty of perceptually isolating single target components
Authors: Watson Charles S. , Kidd Gary R. , Surprenant Aimée M. , Drennan Ward R.
Published: October 2005

A method of adjustment was used to establish the importance of each of several structural properties of the context tones, in nine-tone sequences, in determining the perceptual isolability of target components. Successful ‘‘perceptual isolation’’ of a target tone was assumed to be achieved when frequency matches were as accurate as those achieved for tones presented in isolation, generally meaning less than 1%–2% for the 50-ms tones in these sequences. The context property that was found to primarily affect the frequency matches was the separation, in Hz, between the target tone and both the local and (to a lesser degree) the remote context tones. Other than its bandwidth, the form of the local pitch contour (the target tone plus the single tones immediately before and after it) had no clear effect on the ability to ‘‘hear out’’ the target tone, i.e., whether the local context was ascending, descending, concave up, or concave down. The contours of the remote context tones (first and last three in the patterns) likewise had no effect on performance. Performance ranged from 25% target tones isolated for the most difficult conditions to 90% for the easiest.
Temporally directed attention in the detection and discrimination of auditory pattern components
Authors: Kidd Gary R.
Published: October 2005

Thresholds for detection and frequency discrimination were determined for tones that occurred at unexpected temporal locations within 12-tone sequences. Expectancies were established by repeated presentations of a standard pattern on each trial. Temporal deviations were introduced in comparison patterns by advancing or delaying the onset of a single ‘‘target’’ tone while maintaining its serial position within the pattern. Rhythmic patterns consisting of 350- and 150-ms intertone intervals were used to allow for a large range of temporal displacements. Thresholds were determined for target tones that were advanced (‘‘early’’ targets) or delayed (‘‘late’’ targets) by various degrees. Thresholds for displaced targets were elevated with respect to nondisplaced targets for both detection and discrimination. However, for most listeners, there was little or no effect of temporal displacement on detection except when targets were advanced by 200 ms or more. Temporal deviations had more consistent effects on frequency discrimination for both early and late targets. The results are consistent with the view that, when presented with an auditory pattern, a listener’s attention is dynamically allocated in time in response to the temporal structure of the pattern. [Work supported by NIH.]
Use of the psychophysical method of adjustment in tonal pattern discrimination
Authors: Watson Charles S. , Kidd Gary R. , Surprenant Aimée M. , Drennan Ward R.
Published: October 2005

A difficulty in tonal-pattern research is that several thousand trials are typically required to approach asymptotic discrimination performance under minimal-uncertainty testing conditions. One solution to this problem is to use the method of adjustment to determine thresholds, rather than a forced-choice psychophysical method. In this study the extremely brief times that are required for a listener to achieve perceptual isolation for single components of a multi-tone patterns using the method of adjustment instead of a forced-choice method (minutes as opposed to hours) are demonstrated. A quantitative criterion for ‘‘perceptual isolation’’ reached when a frequency match is made that is as close to the standard as can be achieved when the standard and variable tones are both presented in isolation, rather than in pattern contexts. Not all adjustments are this accurate, however. The most useful distinction between difficult and easy adjustments is shown to be the percent of all the adjustments, for a given combination of target and context tones, that meet this perceptual-isolation criterion.
The effects of training method on frequency discrimination for individual components of complex tonal patterns
Authors: Port Robert F. , Rogers Catherine L. , Watson Charles S. , Kidd Gary R.
Published: October 2005

It has been assumed that subjects trained to detect increments in the frequency of all components of complex tonal patterns (broad focus) would be less accurate in detecting changes in a single target tone than subjects who have been trained to detect changes in only that component [e.g., Watson etal., J. Acoust. Soc. Am. 60, 1176–1186 (1976)]. In several experiments, using a number of 750-ms ten-tone patterns, subjects were trained using one of three methods: in the first two, a S/2AFC procedure was used to train subjects to detect frequency increments in a specific target tone (group one) or to detect frequency increments that could occur in any of the ten components (group two), and in the third, subjects were trained only to identify the individual patterns. Subjects trained using these methods were tested on their ability to detect changes in various components of the patterns, including the target tone for the first group. In all of these experiments, only very slight differences in performance were found among the different groups. These results suggest that lengthy experience with a given pattern allows a listener to discriminate small differences in frequency in any of the individual components of that pattern, relatively independent of the nature of that experience.
Evaluation of a digital adaptive filter, noise-reduction scheme for normal and hearing-impaired listeners
Authors: Jamieson Donald G. , Brennan Rob , Cornelisse Leonard E.
Published: October 2005

A common complaint of hearing aid wearers is that they have difficulty when listening in a background of noise. The ability of a digital adaptive noise reduction filter to reduce background noise without degrading speech perception has been evaluated. Each of four normal-hearing and six hearing-impaired subjects completed two behavioral tests of speech perception—the speech reception threshold (SRT) and the modified distinctive features differences test [DFD(m)]—and one subjective measure of listener preference (paired comparisons). Each test was performed in three types of background noise presented at a level of 65 dB: white noise, low-pass filtered at 8.0 kHz; white noise, low-pass filtered at 1 kHz; and multitalker speech babble. For hearing-impaired subjects, the speech-plus-noise signal was filtered and amplified to provide the NAL real-ear insertion gain. Results indicated that all subjects preferred the noise-reduction-processed speech-plus-noise signal over the unprocessed signal at positive signal to noise ratios. The effect of the adaptive digital filter on speech perception depended on hearing loss, noise type, and SNR. Feature analysis of DFD(m) responses indicated that processing had different effects on the error patterns for normal and hearing-impaired listeners.
Discriminating the direction of auditory motion
Authors: Saberi Kourosh , Hafter Ervin R.
Published: October 2005

Dichotic click trains presented through headphones were used to simulate changes in the interaural delay produced by a source moving horizontally in space. The interaural time-difference (ITD) in each click was increased by a constant ΔITD relative to the click immediately preceding it. Left motion was simulated with delays favoring the left ear and visa versa. The d’s for direction measured in a single interval design were linearly related to the number of clicks in the train and, by definition, the extent of change in ITD from first click to last. In experiment II, the middle clicks were eliminated; leaving only the first and last clicks. Performance was essentially unchanged, suggesting that subjects relied on only the onset–offset position of the moving image in experiment I. To examine this hypothesis, experiment III presented a variety of conditions for the middle clicks. If motion detection were simply based on the onset and offset of the total stimulus, then the interaural configuration of the middle clicks should be inconsequential. It was not. These results are explained in terms peripheral adaptation and an integration time of about 65 ms [Work supported by NIH.]
Relative auditory distance perception of synthetic reflections and reverberation presented over headphones
Authors: D’Angelo William R. , Ericson Mark A.
Published: October 2005

Relative auditory distance perception of a direct path signal by itself, with synthetic reflection(s), and with reverberation presented over headphones was measured using a 2AFC task. The stimulus, either a 500-ms pink noise burst or a 3-s phrase of male speech, was presented twice, first at a reference distance and second at an incremental distance from the reference. Reference distances included 5, 14, and 22 ft, and the incremental distances included multiples of 0.125, 0.25, and 0.5 ft, respectively. Stimuli pairs at each of the three distances were presented from four directions: front, back, left, and right. For each stimuli pair, the task of eight subjects was to indicate which of the two sounds appeared closer to himself/herself. The results of all locations indicated that the jnds for the direct path signal were 7%, jnds for the reflections were around 6%, and jnds for the reverberant sounds were about 12%. All of the sounds appeared out-of-head, however, almost all of the sounds presented from behind were perceived to be in front. Side sounds appeared to be biased towards the front when presented at the further distances.
Signal analysis and representation for speech recognition
Authors: Juang B. H.
Published: October 2005

The acoustic front end that converts the signal waveform into a parsimonious representation is a critical component of a speech recognizer. Traditionally, this front end conversion process has relied on the short time spectral analysis framework in which power spectra of speech segments, on the order of 10 ms, are successively and independently estimated, via all-pole modeling or bank of filters. In this paper, several key considerations in front end design are elaborated, ranging from the computational structure to the associated dissimilarity measures, in the context of prevalent short time spectral representations and the strengths as well as weaknesses of various speech analysis models are discussed, with an attempt to point out possible ways to improve the existing methods. It is further suggested how auditory modeling may help enhance the robustness of a speech recognizer by providing perceptually reliable measurements of speech.
Aerodynamic noise of high-speed ground vehicles
Authors: Brooks Bennett M.
Published: October 2005

A potentially serious problem for high-speed ground transportation vehicles is the aerodynamic self-noise generated simply by the vehicular movement through the air. This noise constitutes the floor to the total noise produced by the vehicle regardless of the method of propulsion. Several methods of calculating the sound pressure levels of this aerodynamic noise are developed based on the interaction of turbulent boundary layer flows with solid surfaces. These calculations are applied to a model for a high-speed train. Calculated noise levels compare favorably with the levels measured for unpowered vehicles operating at typical high speeds. [Work partially supported by United Technologies Corporation.]
Vibroacoustical design for the connecting structure between a vibrating source and a lightweight envelopping structure
Authors: Lessard Nicolas , Laville Frédéric , Charron François
Published: October 2005

This research is on the design of the connecting structures between a vibrating source and a lightweight structure with application to ventilation devices with a 120-Hz excitation by an electrical motor. The design method had two objectives: a low noise level and little variability in level (robust design). Three complementary approaches have been used to identify the various design, manufacturing, and assembling parameters that affect the low-frequency noise level. First, acoustical engineering judgment lead to basic design concepts. Two other approaches were then used to analyze the effect of these basics design concepts and various manufacturing and assembly parameters on the noise level. The first approach was a statistically designed experimentation to identify key parameters. The second approach was a finite element modelization including a sensitivity analysis of the quadratic velocity as a function of key parameters. The design approach applied to the case of an electrical motor in a box-like structure lead to the choice of a low-noise connecting structure as well as the identification of important manufacturing and assembly parameters to be controlled to ensure a low noise variability.
Micro-perforated sheet metal and its application to silencer design
Authors: Wu Mei Q.
Published: October 2005

Micro-perforated sheet metal is used to make no-media silencers, i.e., silencers without fiberglass, mineral wool, or any kinds of fibrous filling materials. The advantages of no-media silencers are that they will not release small fibers to pollute the air and they will not be contaminated by the moisture and bacteria in the air. No-media silencers are extensively used in the air-handling system ducts of hospitals, pharmaceutical factories, and chemical laboratories. This paper summarizes the research work of developing a theory to predict the performance of no-media silencers. The theory is based on Maa’s micro-perforation theory. Two modifications have been made to Maa’s original theory so it can be applied to air duct silencer design. One of the modifications is to introduce the effect of grazing incident sound wave, and the other is to introduce the effect of air flow. Based on the modified theory, a computer program was developed to calculate the insertion loss and the pressure drop of a silencer. The calculated results were compared with test results. The comparison indicated a reasonably good agreement between the theoretical prediction and the test results. [Work supported by BVA Systems.]
A new number-theoretic acoustical concrete masonry unit
Authors: D’Antonio Peter
Published: October 2005

A new number-theoretic structural acoustic concrete masonry unit will be described. These new acoustical building blocks uniformly disperse reflections created by the flat surfaces of conventional absorbing block over a broad range of frequencies, while simultaneously providing extended broadband low-frequency absorption, load-bearing reinforced construction, fire-safety, high transmission loss, economy, and an aesthetic appearance. Random-incidence acoustical testing yielded an NRC of 0.85 for the unpainted block and 0.40 for the painted block, with 100% efficiency at 100 Hz. The STC for painted block was 55. Structural testing for both compressive strength and flexural mid-span deflection were in compliance with ASTM C90-90. A novel cavity design, which provided extended low-frequency absorption will be described. The system consists of three blocks. The A and B blocks facing the noise source each contains a partial quadratic residue sequence and the rear mounting C block forms a structural cavity. The acoustical properties of the system address the needs of all musical facilities, movie theaters, auditoriums, school rehearsal rooms, and noise control applications including highway barriers, airports, power plant transformer substations, etc.
‘‘Wave automaton’’ for wave propagation in the time domain
Authors: Sornette Didier , Sebbah Patrick , Vanneste Christian
Published: October 2005

The new model introduced recently [Vanneste etal., Europhys. Lett. 17, 715 (1992)] for the dynamical propagation of waves in arbitrary heterogeneous media, which is efficient for calculations on large systems (1024×1024) over long times (several 106 inverse band widths) will be discussed. Instead of starting from a wave equation or a Hamiltonian that needs to be discretized for numerical implementation, the model is defined by the set of S matrices, one for each node, describing the interaction of the wave field with the scatterers. The different results are shown on wave packets in random media, which have been obtained using extensive numerical simulations on a parallel computer and these numerical results are compared with weak localization predictions. Finally, ‘‘wave automaton’’ is shown to be equivalent to a discretized version of the hyperbolic time-dependent wave and Klein–Gordon equations, when restricted to a suitable subclass of the control parameters, and the relationships between the two formulations of the wave propagation problem are made explicit. Compared to finite-difference versions of hyperbolic equations, the wave automaton is shown to be much more flexible for implementing arbitrary boundary conditions.
Anderson localization in the time domain: Numerical studies of waves in two-dimensional disordered media
Authors: Weaver Richard L.
Published: October 2005

Experimental work in the time domain on Anderson localization and anomalous diffusion in dimensions greater than one is scant, being confined to some ultrasonic work by Weaver and, inasmuch as measurements of frequency–frequency intensity correlation functions imply measurements of time domain behavior, some microwave work by Genack and co-workers. Numerical work in the time domain is similarly poorly developed. No studies have been conducted with sufficient detail for determination of the effective transport behavior. This paper presents results obtained from new numerical experiments in an attempt to rectify this lack. A numerical model is introduced that allows the tracking of evolving wave energy density in a two-dimensional linear classical wave equation version of an Anderson model diagonally disordered Hamiltonian. The behavior of the evolving average wave energy density in effectively infinite large samples of this system (typically 300 sites by 100 sites) excited by tone burst line sources is presented and the results shown to collapse to an apparent single universal response function with interesting asymptotic character like exp(−r/λ−r2/4 β t) where r is the distance from the source, λ is the localization length, and β is a constant not equal to the bare diffusivity. The effect of dissipation is also investigated, and, consistent with previous demonstrations, found to be trivial. [Work supported by NSF MSS-91-14360.]
Pulses, nonlinearity, and Anderson localization
Authors: Maynard J. D.
Published: October 2005

It is now well known that disorder in one-dimensional systems results in normal modes that are exponentially localized (Anderson localization). The propagation of waves and pulses through a disordered system is dramatically affected by the localization, and there is some interesting exceptional behavior. The problem is made further interesting by the addition of nonlinearity. A basic question is whether or not nonlinearity destroys Anderson localization. Of all the references,[1] half predict that localization is destroyed, and half predict that it is not. Continuous wave and pulse experiments have been performed on nonlinear disordered acoustic systems and some interesting results have been obtained. An analysis of a nonlinear periodic system, which shows effects of basin crowding, will also be presented. [Work supported by NSF DMR 9000549 and ONR.] P. Devillard and B. Souillard, J. Stat. Phys. (1986); B. Doucot and R. Rammal, Europhys. Lett. (1987); C. Albanese and J. Frohlich, Commun. Math. Phys. (1988); Q. Li etal., Phys. Rev. B (1988); A. Soffer and M. I. Weinstein, Commun. Math. Phys.; R. Bourbonnais and R. Maynard, Phys. Rev. Lett. (1990); Yu. S. Kivshar etal. (1990); R. Knapp etal., DisorderandNonlinearity, edited by A. R. Bishop, D. K. Campbell, and S. Pnevmatikos (1989).
Acoustic waves in bubbly liquids: (1) Can strong localization of waves be attained by internal resonances? (2) Acoustic wave propagation in one-dimensional stratified gas–liquid media: The different regimes
Authors: Sornette Didier , Bouchaud J.-P. , Legrand O.
Published: October 2005

An inhomogeneous medium with scatterers presenting internalresonances has been suggested to be an efficient system to localize classical waves. Within a mean-field approach describing the internal degrees of freedom of the scatterers and their strong interactions in the near field regime, the existence of two longitudinal modes associated with the scatterers and embedding medium degrees of freedom is found. Anderson localization is shown to be reached in a rather large frequency interval for the ‘‘slow’’ mode, above the single scatterer resonant frequency, whereas the ‘‘fast’’ mode is never localized. An experimental situation is discussed where the localization of the ‘‘slow’’ mode could be observed. This work appeared in J. Phys. I France 2, 1861–1867 (1992). Also considered is the problem of the propagation of acoustic waves in a one-dimensional bubbly liquid, either periodic or random. The response of the bubbly liquid to an incident wave in any range of concentrations, bubble size polydispersity, and frequency range has been computed exactly. Three main regimes are found, depending on the average value of the bubble radius R, average distance d between bubbles, and wavelength λ in the liquid: (1) ‘‘phonon’’ or ‘‘mass-spring’’ regime; (2) ‘‘bubble’’ regime; and (3) ‘‘stratified’’ regime. In the random systems, the Anderson localization length is computed both numerically and analytically in the limit of uncorrelated disorder, for various system realizations.
Nondestructive evaluations in multiple scattering media
Authors: Vanneste Christian , Feng Shechao , Sornette Didier
Published: October 2005

Due to its tendency to scramble the structural information, multiple scattering blurs the distinction between different materials containing different kinds and numbers of defects, or their configurations, beyond the scattering mean-free-path distance from the observation point, thereby rendering most random systems cloudy in appearance and opaque in the sense of structure delineation. However, it has recently been shown theoretically [Feng and Sornette, J. Acoust. Soc. Am. 90, 1742–1747 (1991)] that imaging of a new defect in a multiple scattering medium is in principle possible using speckle pattern correlations. Extensive numerical simulations and analytical calculations were performed to demonstrate the feasibility of detecting a new crack in a single given sample already containing many cracks using speckle pattern averaging technique at many different frequencies. Potential applications of this approach can be found in medical and industrial imaging needs in highly scattering systems.
Computer simulation of auditory stream segregation in sequential-tone sequences
Authors: Beauvois Michael W. , Meddis Ray
Published: October 2005

A simple computer model is described that takes a novel approach to the problem of accounting for perceptual coherence among successive pure tones of changing frequency by using simple physiological principles that operate mainly at a peripheral, rather than a central level. Using a single set of parameter values, the model is able to reproduce a number of streaming phenomena found in the literature. These are: (1) the buildup of auditory streaming over time, (2) the temporal coherence and fission boundaries of human listeners, (3) the trill threshold, and (4) the stream-organization process exhibited by human listeners when presented with ABC tone sequences. Whereas streaming phenomena are generally accounted for in terms of an auditory scene-analysis process that works on the basis of Gestalt perceptual principles, the success of the model in reproducing experimental data obtained from humans justifies the potential value of a low-level analysis for explaining high-level psychological phenomena such as Gestalt auditory grouping, and suggests that some Gestalt auditory grouping may be the product of low-level auditory processing. [Work supported by SERC(UK) and CNRS(France).]
A computer implementation of psychoacoustic grouping rules
Authors: Ellis Daniel P. W.
Published: October 2005

Many of the rules employed by the human auditory system to fuse and segregate acoustic energy into separately perceived sources are described by Bregman [A. S. Bregman, AuditorySceneAnalysis (MIT, Cambridge, MA, 1990)]. However, the precise application of these rules to anything but the most simple experimental stimuli is less well understood. This idealized model of low-level auditory processing [Ellis and Vercoe, J. Acoust. Soc. Am. 91, 2334 (A) (1992)]; [Ellis, J. Acoust. Soc. Am. 92, 2376 (A) (1992)] is specifically designed to facilitate automatic grouping for real stimuli. This is achieved by representing sounds as sets of sinusoid tracks, the discrete time-frequency elements required by such rules. This system is discussed for grouping tracks, and it is shown that the results are obtained by applying simple rules for harmonicity, common fate, and proximity to some real sounds. The means by which ambiguity has been resolved between different types of rule, along with other issues arising from the model, will also be described.
Interactive computational auditory scene analysis: An environment for exploring auditory representations and groups
Authors: Crawford Malcolm , Cooke Martin , Brown Guy
Published: October 2005

Computational modeling of auditory scene analysis (ASA) offers a new paradigm for experimentation. It permits a novel approach to the development of theories of grouping, and to the design of experimental stimuli. For example: (i) grouping algorithms can be implemented, and validated against experimental data; (ii) experimental data can be analyzed to suggest a possible representation in the auditory system, and to test conformance with expectations; (iii) computational implementation can expose deficiencies in current theory. Over the last 4 years, the Sheffield Auditory Group has developed a rich set of representations used for investigating computational ASA [G. J. Brown, ‘‘Computational Auditory Scene Analysis: A Representational Approach,’’ Ph.D. thesis, University of Sheffield (1992); M. P. Cooke, ModellingAuditoryProcessingandOrganisation (Cambridge U.P., Cambridge, UK, 1993)]. There representations include computational maps for onsets, offsets, frequency transitions, and periodicities, in addition to higher-level symbolic representations of acoustic components. Recently, an environment has been created that brings together this diverse collection into a uniform framework for display, resynthesis, and experimentation. The environment supports experimental investigation and allows the ‘‘debugging’’ of stimulus selection. Further, it acts as a canvas onto which the results of auditory grouping can be drawn. It also serves as a tutorial in this increasingly complex field. The practical application of these points is illustrated in a case study that maps the path from stimulus generation to grouping by listeners or machine.
The generation of high-speed train noise from roughness
Authors: Remington Paul J. , Webb Jonathon
Published: October 2005

The forces generated in the contact area between railroad wheels and rails due to small scale roughness has been estimated using three analytical techniques: a simple average of the roughness, a point reacting spring model, and a full elastic interaction model. Roughness data measured in a number of parallel tracks in the rolling direction on partially worn wheels and rails was used in these three models. The forces estimated by all three analytical techniques were remarkably similar in the range of wavelengths important for noise generation from high-speed (200-to 400-km/h) trains. Wave-number decomposition of the roughness data has shown it to be bandlimited in the direction transverse to rolling, a result consistent with the similarity in the predictions of the three analytical techniques. The influence of wheel and rail profile and high roughness amplitudes at low wave number will be discussed.
Full-wave modeling of the transmission of sound over theater seats: Near-field investigation
Authors: Chéenne Dominique J. , Kubik Robert D. , Maher Robert C. , Bahar Ezekiel
Published: October 2005

The full-wave theory [E. Bahar, J. Acoust. Soc. Am. 89, 19–26 (1991)] is applied to a computer simulation of sound transmission across a simplified model of theater seats. The acoustic response is derived for observation points above the seats for a range of incident angles and receiver heights. The results obtained from the model study are in good agreement with the experimental data recently obtained by J. S. Bradley [J. Acoust. Soc. Am. 90, 324–333 (1991)]. The full-wave solution accounts for the acoustic pressure diffusely scattered by the chairs as well as the zero-order field scattered by the finite floor and the direct wave from the source. It suggests that the ‘‘seat-dip effect’’ is mostly due to an interference phenomenon at the observation point between the direct field and the scattered field reflected off the floor. The model allows for a detailed analysis of the effect of scattered field by the chairs, both in front of, and behind the receiver. These results suggest the need for the use of high directionality microphones that can distinguish between the forward and backward scattered acoustic pressure during future experiments.
A comparison of objective measurements in concert halls and multi-purpose rooms
Authors: Raia Loren , Siebein Gary W. , Chiang Wei-hwa , Doddington Harold W. , Schwab Wilhelm K.
Published: October 2005

A series of monaural and binaural acoustical measurements were taken at multiple receiver locations from two source locations in four concert halls and in the concert configuration of four multi-purpose performing arts centers. Comparisons both within and among rooms were made to find similarities and differences in the measured acoustical properties of the two types of rooms. Specific investigations tried to relate how the architectural properties of the rooms affected the measured acoustical properties. [Work supported by the National Science Foundation and Concert Hall Research Group.]
A finite element compatible bond graph modeling approach for transducers applied to the piezoelectric thickness vibrator
Authors: Moon Won-Kyu , Busch-Vishniac Ilene J.
Published: October 2005

A bond graph modeling method suitable for transducers and compatible with finite element formulations is developed. This formulation leads to a new definition of the generalized displacements for a continuous system. The newly defined coordinates are easily interpreted physically and easily used in analysis of the system performance. The method is demonstrated for a piezoelectric thickness vibrator. Compared to the Mason equivalent circuit models, the bond graph modeling method presented offers the primary advantage of physical realizability. Compared to circuit models based on standard discrete electrical elements, the main advantage of the bond graph modeling method is a greater physical accuracy (which is the direct result of using multiport energic elements).
Effect of an underlying compliant layer on transducer performance
Authors: Young Rose M. , Corsaro R. D.
Published: October 2005

The performance of acoustic transducers (sensors and actuators) can be degraded or enhanced by the structure on which they are mounted. For various numerical structural acoustics studies, a simple computationally efficient model is required for estimating the change in transducer performance resulting from variations in the front and back face impedances. The approach used here is based on simple physical principles and intuition, which makes it well suited for incorporation in more-complicated structural models since it is conducive to direct extraction of physically meaningful forcing functions. As an example of its use, the performance of pressure sensors, velocity sensors, and acoustic drivers on an air-backed steel plate are estimated, and then the change in performance when a layer of compliant material is inserted between the transducer and the plate is calculated. Results are compared with a Mason-type equivalent circuit model.
Three-dimensional finite element model of a high-power, low-frequency ring-shell flextensional sonar transducer
Authors: Costa Pinto Rogerio N. , Baker Steven R. , Wilson Oscar B.
Published: October 2005

A three-dimensional finite element model of a high-power, low-frequency ring-shell flextensional transducer [Sparton of Canada, Ltd., model 34A0610] has been developed for use with the ATILA code [Decarpigny etal., J. Acoust. Soc. Am. 78, 1499–1507 (1985)]. A three-dimensional model is needed to represent nonaxisymmetric vibrations that can occur in a dense sonar array. The model was derived from a two-dimensional axisymmetric model developed by Blottman [J. B. Blottman, Naval Undersea Warfare Center, New London Laboratory, Technical Memorandum No. 89-1090 (1989)]. In extending Blottman’s model to three dimensions, reasonable simplifications were made to minimize the number of elements required to faithfully represent the dynamical behavior of the transducer in the frequency range of interest. A description of the model and the results of in-air modal and in-water harmonic analyses will be presented. [Work sponsored in part by the Naval Postgraduate School and the Naval Undersea Warfare Center.] a) Lieutenant Commander, Brazilian Navy.
Piezoelectric composites of the 1–3 type used as underwater sound sources
Authors: Reader Wayne T. , Sauter David F.
Published: October 2005

Piezoelectric composites with 1–3 connectivity used in the thickness mode have traditionally used a relatively rigid polymer as the matrix encapsulant to force uniform surface motion. Finite element analyses are employed to show that these rigid polymers constrain the piezoelectric rod motion sufficiently to reduce rod displacements by 50% or more. The magnitude of the reduction depends upon the ceramic volume fraction, the rod aspect ratio, rod spacing, and of course the stiffness of the matrix. Further, it has been shown [W. A. Smith, ONR Tech Note (31 May 1991)] that the hydrostatic response, rather than the thickness mode alone, controls the performance of 1–3 composites used as underwater sound sources. It is shown in this paper that efficient lightweight underwater sound sources can be produced by: (1) using lightweight rigid cap plates and a very compliant matrix polymer to produce uniform surface motion comparable to the motion of the unconstrained rod, and (2) using rigid edge strips bonded to the cap plates to prevent edge motion, thereby effectively limiting the transmitting response to the thickness mode. [Work supported by ONR.]
Comparisons of two encapsulating methods on the acoustic performance of a passive cylindrical array
Authors: Walden Alan K. , Howarth Thomas R. , Pecoraro Mark L. , Tims Allan C.
Published: October 2005

The purpose of this study was to determine the effects of encapsulation methods on the acoustic performance of a passive, cylindrical, high gain, broadband (multi-octave) array. Both an oil-filled boot and a polyurethane coating were considered. The array consisted of 36 lead titanate elements mounted in a 3 by 12 matrix along 180 deg of the circumference of a steel cylinder. Original intentions were to rely on the oil-filled booting to ensure performance void of transverse mode coupling between neighboring elements. Measurements of the oil-filled array showed a ‘‘scalloping’’ effect in the radiation patterns as well as significant disturbances throughout the receiving response [Howarth etal., J. Acoust. Soc. Am. 91, 2325–2326 (A) (1992)]. This study took the same array and encapsulated it in a polyurethane elastomer. Experimental data indicate that many of the perturbations notes in the oil-filled configuration were considerably diminished after potting. Performance comparisons between the two encapsulating methods shall be presented and discussed. [Work supported by NSWS/Carderock Division.]