bluuebunny/crossref_metadata_2025 huggingface dataset:
Acoustical and psychoacoustical analyses of simultaneously sounding wind instruments
Authors: Carterette Edward C. , Kendall Roger A.
Published: October 2005

Previous studies of timbre largely have employed amusical signals, including widely cited Grey [J. Grey, “An Exploration of Musical Timbre,” Stanford: CCRM Report STAN M-2 (1975)]. The salient features of his brief (ca. 0.33 s) signals may not inhere in musical phrases. Six phrase-length musical contexts played by pairs [dyads] of orchestral wind instruments were investigated. Perceptual data from scalings and verbal ratings showed that principal attributes of wind dyads are (1) nasality and (2) its complement, rich/brilliant; these are correlated with total harmonic distortion, which in turn is connected to the lattice tone-hole cutoff frequencies. Cepstral distances of dyad spectra have a weak correlation with perceptual distances; however, the cluster analyses of perceptual distances and cepstral distances provide similar dyad pairings and orderings. Evidence suggests that individual spectral contributions vary by instrument combination. This demonstrates that mere mixtures of separately recorded voices are different acoustically from those of ensemble playing.
A 19th century trumpet in F
Authors: Birkemeier Richard P. , Ayers R. Dean
Published: October 2005

There is a tendency these days to think of “the trumpet” as the familiar brass instrument seen commonly in orchestras and bands. In reality, the instrument that is now called the trumpet is a hybrid design that more closely resembles the old band instrument called the cornet. The last true trumpet disappeared from the musical scene almost 100 years ago. This instrument was built with a majority of cylindrical bore tubing, was longer in overall length than the modern instrument, and was pitched in F. One example is this model made by Courtois and Mille of Paris late in the 19th century. The tone color is different from that of the modern trumpet, and it has some specific performance problems, but despite its lower pitch, this was the soprano brass instrument preferred by composers of a century ago. A comparison with the modern trumpet in terms of responses in the time and frequency domains will be correlated with live demonstrations. [Work supported in part by a CSULB Summer Research Grant and Equipment Grants from Hewlett-Packard.]
Correlations of frequency variation of spontaneous otoacoustic emissions in multiple-emitting subjects: Evidence for an underlying physiological concomitant
Authors: Haggerty Holly S. , Schubert Earl D.
Published: October 2005

Recently, 24-h and monthly variabilities of spontaneous otoacoustic emission (SOAE) frequency have been observed [H. S. Haggerty, J. Acoust. Soc. Am. Suppl. 1 86,31 S44 (1989); Assoc. Res. Otolaryngol. Abstr. 268 (1990)]. These frequency variabilities may reflect modifications of the basic properties of SOAEs that occur secondary to cyclical changes in some other underlying physiological concomitant. If the concomitant is fairly localized, then the likelihood exists that frequency variations for two different emissions produced by the same ear might be correlated. However, if the underlying physiological concomitant is a more generalized phenomenon, rather than localized to a given cochlea, then a correlation of frequency variation may hold even for SOAEs produced by different ears of a given subject. To check these hypotheses, Pearson product-moment correlation coefficients were computed, comparing the SOAE frequency values for seven subjects who presented with multiple SOAEs. Results showed that the majority of coefficients, whether for within-ear or between-ears SOAEs comparisons, was statistically significant. These findings of correlations of frequency variability for SOAEs both within and between ears support the idea that changes in this property of SOAEs occur as a function of an underlying, generalized, physiological concomitant. [Work supported by the Knowles Research Foundation.]
Brain potentials evoked by voiced and unvoiced consonant-vowel combinations differ
Authors: Michalewski Henry , Starr Arnold
Published: October 2005

Auditory-evoked potentials were recorded from normal hearing subjects in response to voiced and unvoiced phonemes (e.g., “ta” vs “da”). The negative component generated by primary auditory cortex (N100) is markedly reduced in amplitude to unvoiced compared to voiced signals. The effects of the initial consonants' intensity, duration, and proximity to the vowel were defined. It is the initial unvoiced 10–20 ms portion that is responsible for the attenuation of the vowel-evoked N100 component. This attenuation persists even when the consonant is attenuated 20 dB relative to the vowel's intensity; the attenuation persists with as much as a 150-ms silent period inserted between the consonant and vowel. “Noise masking” as the basis for the unvoiced consonants ability to attenuate vowel evoked potentials were evaluated using a brief noise burst to mask tone evoked potentials; there was little similarity between the two paradigms suggesting that noise masking is not the mechanism accounting for the effects. The N100 component in human reflects auditory cortical processes that are sensitive to simple phonemic features of language is proposed (i.e., voicing versus unvoicing).
A feasibility study of acoustic tomographic monitoring of the atmospheric boundary layer
Authors: Wilson D. Keith , Thomson Dennis W.
Published: October 2005

Acoustic tomography is a promising method for monitoring atmospheric wind and temperature fields. It may be possible to achieve better spatial and temporal resolution than offered by existing technologies. In this paper, three different schemes, all based upon travel-time measurements, are proposed and analyzed: a horizontal array for determining structure of boundary-layer thermals and gravity waves; a vertical array for high-resolution monitoring of the mean surface-layer profiles; and a long baseline array for studying the evolution of stable boundary layers. The schemes are evaluated in terms of resolution capability, practicality, and interest in atmospheric science. Some preliminary results from experimental and numerical studies of the horizontal array are given. The numerical studies use wind and temperature fields generated by a large-eddy simulation (LES) model.
Incorporation of a large-scale wind-driven turbulence model into a range-dependent parabolic equation
Authors: Noble John M. , White Michael J.
Published: October 2005

Atmospheric turbulence contains numerous scales ranging from a few centimeters to a few kilometers. As shown by Daigle et al. [J. Acoust. Soc. Am. 64, 622–630 (1978)], the effects of small scale turbulence (L<10 m) can be handled fairly well from a statistical standpoint. This approach works because for most of the time, small-scale turbulence can be treated as isotropic over short ranges. However, large-scale turbulence is anisotropic in nature. This results in variations in temperature and wind speed with range. In order to account for the effects of wind-driven large-scale turbulence on acoustic propagation, a large-scale turbulence model was developed. This large-scale turbulence model assumes elongated longitudinal vortex pairs roughly aligned with the mean wind speed with height and range due to the passage of an eddy pair. The wind speed variations are used to calculate range-dependent sound-speed profiles. The sound-speed profiles are used by a range-dependent parabolic equation to calculate variations in the received amplitude and phase of an acoustic wave as the eddy pair passes through the field of propagation.
The acoustic signature of thunder from seismic records
Authors: Kappus Mary E. , Vernon Frank L.
Published: October 2005

Thunder, the sound wave through the air associated with lightning, transfers sufficient energy to the ground to trigger seismometers set to record regional earthquakes. Instruments located within a few meters of the surface of the Earth register ground motion directly coupled to air motion. Instruments in a 100-m-deep borehole are more than one wavelength from the boundary, so recordings there have amplitudes reduced by the transmission coefficient, a factor of 1000. More than a dozen records of individual lightning strikes were made during 2-h period at a seismic station in Karasu, USSR. The near-surface instrument records were analyzed for frequency content using multitaper techniques that deal appropriately with nonstationary signals. The spectra are broadband from approximately 5–100 Hz, with small peaks at frequencies ranging from 6–13 Hz. No peaks are observed in the audible range as previously predicted by theory based on rapidly expanding gases. Analysis of the frequency content as a function of time shows that the individual claps have similar frequency distributions, with no evidence of dispersion.
Tactile aids with deaf-blind children
Authors: Franklin Barbara
Published: October 2005

This paper will present the results of a 3-yr study (1986–1989) comparing the effect of two tactile devices on communication skills of deaf-blind children—a 2-channel vibrotactile (Tactaid II) and a 16-channel electrotactile aid (Tacticon). A total of three communicative behaviors were selected for each child. A single-subject alternating treatment design was used and data from each of three substudies were analyzed using three combined Wilcoxin tests. In the first substudy, subjects displayed significanfly more of the desired behaviors when wearing the Tactaid than the Tacticon (0.05 level). In the second and third substudies, and in the replication-across-settings study, no differences were found in any of the conditions. In the replication-across-change agents study, both tactile devices produced significantly more of the desired behaviors than no device (0.05 level). One of the most important findings was that the children accepted both devices despite the tactile defensiveness often exhibited by so many of them. A second study to replicate the research using only the Tactaid II + device with deaf-blind infants and preschoolers is underway (1989–1992), and the results of the first year of this study will also be presented. [Work supported by OSERS, U.S. Dept. of Educ.]
Vowel acoustics in cochlear implant patients
Authors: Perkell J. , Lane H. , Svirsky M. , Webster J.
Published: October 2005

Acoustic parameters were measured for vowels spoken in /hVd/ context by three postlingually deafened cochlear implant recipients. Two female subjects became deaf in adulthood; the male subject, in childhood. Recordings were made before and at intervals following processor activation. The measured parameters included formant frequencies, F0, SPL, duration, and amplitude difference between the first two harmonic peaks in the log magnitude spectrum (H1–H2). A number of changes were observed from pre- to post-implant, with intersubject differences. The male subject showed changes in F1, vowel durations, and F0. These changes were consistent with one another; however, they were not necessarily in the direction of normalcy. On the other hand, the female subjects showed changes in F2, vowel durations, F0, and SPL, which were in the direction of normal values and, for some parameters, tended to enhance phonemic contrasts. For all three subjects, H1–H2 changed in a direction that was consistent with previously made flow measurements. [Work supported by NIH.]
Effect of age of onset of deafness on speech production before and after cochlear implantation
Authors: Osberger Mary Joe
Published: October 2005

Broad phonetic transcriptions of speech samples were compared for 4- to 7-yr old children who had congenital deafness or acquired deafness during the first 3 yr of life. Analysis of the data revealed limited phonetic repertoires for all children. The diversity of the phonetic repertoires increased as the age of onset of deafness increased. With few exceptions, however, the sounds produced by the children with acquired deafness were ones that appeared to have been developed prior to the time that these children lost their hearing. A subset of children subsequently received a single- or multichannel cochlear implant. Children with acquired deafness continued to show superior speech production abilities following implantation even though there was no significant difference between the speech perception abilities of these children and those of the children with congenital hearing loss. [Work supported by NIH-NIDCD.]
Dorsal cochlear nucleus responses to electrical stimulation of the auditory nerve
Authors: O'Leary Stephen J. , Tong Yit C. , Clark Graeme M.
Published: October 2005

Extracellular events of single units were studied in response to acoustical and electrical stimuli in the barbiturate anaesthetized cat. The electrical stimuli were biphasic current pulses, 100–200 μs/phase, delivered at 100–400 pulses per second (pps) via a bipolar electrode implanted into the scala tympani. The first group of units, group 1, generated action potentials 1.6–2.4 ms following an electrical pulse, while the latency for group 2 was 2.4–4.8 ms. In both groups, the spontaneous activity and the response to a pure tone was commonly suppressed between 2.5–3.75 ms following an electrical pulse. It is possible that groups 1 and 2 received mono- and di-synaptically mediated excitatory inputs, respectively, and a disynaptically mediated inhibitory input. Both groups commonly generated action potentials at a second latency between 3.75–10 ms, which may have been due to long latency monosynaptically mediated or disynaptically mediated excitatory inputs. There was a difference in discharge rate between acoustic noise stimulation and electrical stimulation at high current levels for group 2, but not for group 1, suggesting temporal interaction between excitatory and inhibitory drives.
The nonlinear vibration of pretensioned strings
Authors: O'Reilly Oliver M.
Published: October 2005

The vibration of pretensioned linearly elastic strings subject to planar transverse harmonic excitation is discussed. Initially a brief history is given of the study of the vibration of strings that features the contributions of Euler, the Bernoulli's, and Kirchhoff, among others. Attention then turns to the nonlinear phenomena observed experimentally in the past 40 yr, in particular, nonplanar periodic and chaotic motions. The existence of these motions and the underlying mechanisms for their behavior in a set of ordinary differential equations that model the vibrating string are discussed. [Work supported by NSF.]
Time evolution of local complexity measures
Authors: Meyer-Kress Gottfried
Published: October 2005

Numerical algorithms are discussed for estimating dimensional complexity of an observed chaotic time series. Factors that enter the procedure are discussed and applied to a local estimate of the pointwise dimension or crowding index. The concepts are illustrated with the help of experimental time series obtained from speech signals. The temporal evolution of the crowding index shows oscillations that can be correlated with properties of the time series. In a mechanical context, think of a nonlinear oscillator which is kicked periodically. Because of the nonlinearity of the oscillators many modes will be excited and therefore the dimensionality of the signal will be large. Since the specific shape of the nonlinearities will determine the rate at which different modes decay; a monotonic decrease will be observed in dimensionality until only the resonant modes will still be active at small amplitudes before the next kick. Instead of observing the specific dynamics of the system, the dimensional complexity of the system is considered as a new dynamical variable. The time evolution of the dimensional complexity parameter is compared with the original time series and also with recurrence plots of the embedded time series.
Vibration control of flexible structures via nonlinear coupling
Authors: Golnaraghi Farid
Published: October 2005

In this paper, a rather unconventional active/passive vibration suppression technique for a cantilever beam based on simple principles in nonlinear vibrations is proposed. The controller is a sliding mass-spring-dashpot mechanism placed at the free end of the beam, introducing Coriolis, inertia, and centripetal nonlinearities into the system. These nonlinear terms may be used to eliminate the transient vibration of the beam when the natural frequency of the slider is twice the fundamental beam frequency (internal resonance). Once at 2:1 internal resonance, the oscillation energy is transferred from the beam to the slider and dissipated for sufficiently high slider damping. Numerical results show that this technique may be used to increase the effective damping ratio of the structure from 0.5% to ∼6%. The controller is particularly successful in reducing large amplitudes of oscillation to levels that may be handled by conventional methods. For lower or zero controller damping, subharmonic and chaotic transient oscillations occur depending on the amplitude of the initial disturbance of the beam. Physical experiments are currenlly underway to verify the numerical and theoretical results.
Synchronization of chaotic systems
Authors: Vaidya P. G. , He Rong , Anderson M. J.
Published: October 2005

Consider two identical chaotic systems that are started from virtually identical initial conditions. In a short time, they would be observed to diverge from one another. While it is true that, given ample passage of time, the systems will come within an arbitrarily close distance of one another, they would drift apart once again. In their fascinating paper, Pecora and Carroll have recently described a system of two Lorenz oscillators. They could be metaphorically described as a master and a slave system. The master system, undergoing chaos, drives a part of the slave system. Incredibly enough, the two systems remain in perfect synchronization. In our paper, further analysis of this system is presented. In addition, another set of oscillators are proposed. These can perhaps be described as sister-sister systems. Both influence one another, and control and modification is possible with the access to either one. Some practical implications of these results are discussed.
On the spectrum of random nonlinear vibration
Authors: Davies Huw G. , Liu Qiang
Published: October 2005

The response of a nonlinear oscillator excited by white noise is considered. A truncated series of Hermite polynomials is used as an approximation to the probability density function. This series and the associated Fokker-Planck equation are used to generate two sets of coupled differential equations for time-dependent moments. The first set is for moments of variables evaluated at the same time; the solution yields, for example, the nonstationary mean-square value. The second set is for moments of variables evaluated at two different times. This second set uses the solution of the first set as initial conditions. A single-sided Fourier transform of the second set yields coupled complex algebraic equations that can be solved numerically for the spectrum. Examples are shown of spectra for the Duffing oscillator showing an increase in effective resonance frequency and broadening of the peak as the excitation level is increased, and for the van der Pol oscillator showing an entrained limit cycle response at low excitation level that disappears as the excitation level is increased. [Work supported by NSERC.]
On the chaotic motion and acoustic characteristics of confined jet flows
Authors: Ng Kam W. , Lee Nigel
Published: October 2005

An experimental study was conducted to investigate the chaotic motion, and flow-induced noise and vibration of confined jet flows in a cylindrical duct. Results from flow and acoustic measurements showed that the motion in pipe flow is random or chaotic, whereas the motion in confined jets is deterministic or orderly. Correlations of wall pressure fluctuations and pipe wall acceleration showed that jet flows generate more deterministic features than pipe flow. The coherence functions of the wall pressure and pipe wall acceleration are relatively high near the exit of the jet. The high coherence is due to the large-scale coherent structures. Further downstream, the activity of jet mixing decreases, the large-scale turbulence cascades to fine scales, and the flow and acoustic characteristics approach that of pipe flow. Phase portraits of the wall pressure data were generated using the method of delays. The deterministic and coherent structures of jet flows were observed and strange attractors could be identified. However, phase portraits of pipe flow showed that the wall pressure fluctuations were chaotic and that no strange attractors occurred. Similar results were obtained using the Wignet distribution time-frequency analysis on the acoustic signals.
From algorithm to market: Real-world issues in speech development projects
Authors: Vitale Tony
Published: October 2005

Papers on speech technology have typically addressed issues in one of two disparate areas: (a) hypotheses and observations aboul articulatory, auditory, or acoustic aspects of speech production and perception, and the development of the algorithms that underlie those observations; and (b) marketing and sales issues somewhat or totally unrelated to (a) such as competitive analyses, industry-specific applications, and applications software or the user interface. There is, however, a complex world of logistics that afffects a speech technology project in the most fundamental way. Nothing describes this more accurately and effectively than a brief history of the DECtalk development project at Digital Equipment Corp. It is a realistic illustration of the evolution from laboratory system to product. In 1981, Dennis Klatt, a senior research scientist at MIT, was searching for an industrial partner to help develop and produce a text-to-speech system derived from Klatttalk, a set of algorithms for a speech synthesizer that he had been working on since 1970. After almosl 9 months of searching, he found individuals within Digital Equipment Corp. who were willing to risk the costs of an extensive development project for a technology which, at the time, had no market. This relationship between academia and industry turned out to be so mutually beneficial and enlightening that Klatt was prompted to discuss it publicly from the academician's perspective [D. H. Klatt, Proc. Speech Tech. '87, 293–294 (1987)]. This discussion outlines some of the more important logistical issues from the corporate point of view and a description of these constitutes a typical project history: problems in hardware design and development; dependence upon external engineering and support groups (e.g., manufacturing, technical documentation, etc.) and external vendors; and testing for design verification/maturity and physical robustness. In addition, speech development was proceeding simultaneously in two separate places (Maynard and MII), the need to run in real-time entailed the perennial decisions on cost versus performance, and debugging cycles required constant compromise.
Speech recognition: Technology and applications
Authors: Rabiner Lawrence R.
Published: October 2005

The technology of speech recognition has evolved for almost 2 decades since the introduction of sophisticated pattern recognition techniques such as dynamic time warping and clustering. Applications of the technology have been slower to evolve for several reasons, including system performance, system cost, and the general acceptance of voice technology by the public. Because of dramatic improvements in each of the areas that impeded use of speech recognition in real-world applications, the technology is beginning to be accepted as a reasonable alternative to keyboards or mouse-like devices for entering data, and to operators or attendants for tasks like call routing, enhanced services, and information retrieval. This talk reviews speech recognition technology, including a discussion of isolated word, connected word, and continuous speech recognition systems. Then there is a brief outline of the reasons why speech recognition is a relatively difficult problem to solve in its most general form, namely, unlimited vocabulary, unlimited syntax, unlimited talker population, unlimited environmental constraints, and unlimited tasks. It is shown that by appropriately restricting one or more of these factors governing performance, a highly reliable, robust speech recognizer for some limited, but interesting, tasks can be implemented. A number of these tasks are discussed in this talk and current performance via videotape demonstrations is illustrated.
Automatic recognition of connected digit strings in a credit card authorization task
Authors: Wilpon J. G. , Ramesh P. , McGee M. A. , Roe D. B. , Rabiner L. R. , Lee C. H.
Published: October 2005

An important area of speech recognition is automatic recognition of connected digit strings (i.e., strings composed of the digits 0 through 9—including “oh”). Applications of this technology include credit card authorization, catalog ordering, dialing of telephone numbers, and data entry, to name just a few. For the past 2 yr AT&T, in cooperation with American Express Travel Related Services, has experimented with a system for automatic recognition of 10-digit merchant identification codes, and 15-digit customer credit card numbers, for the purpose of authorizing purchases charged to an AMEX card. (The problem of recognizing the transaction amount has also been studied, but this is a much more difficult problem and will not be discussed.) Field trial experience with the recognizer using about 1000 customers who provided 2000 connected digit strings over 800-based dialed up telephone connections was correct recognition of 97%–98% of the strings with no rejections using constraints on the validity of both merchant identifications and credit card numbers. Several schemes for applying the constraints in a practical implementation were studied and will be discussed in the talk.
Considerations in utilizing synthetic speech as an aid for proofreading
Authors: Kasuya H.
Published: October 2005

One typical application of Japanese synthetic speech produced by rule is in a newspaper company where a text-to-speech system has extensively been used as an aid for proofreading a manuscript. Operators in the proofreading department work at computer voice terminals about 4 h a day with adequate intermissions. Several important human factors relevant to the voice quality and prosodic properties of synthetic speech were raised in our interview with the operators. To make this point clearer, perceptual experiments were performed in the laboratory on the subjects' preference for the voice quality and prosodic characteristics of synthetic speech, focusing on the average pitch frequency, dynamic range of pitch contours, average frequency spectral characteristics, speaker's sex, and speaking rate, in the task where subjects were required to proofread a printed text by listening to the speech. The experiments indicated that a male voice of a relatively low average pitch frequency with rather less pitch changes was accepted well by the subjects. The experiments further suggested that the ability of a variable speaking rate is indispensable to a speech synthesis system. These findings were thought to be strongly related to the fact that the users work with synthetic speech for a long period of time.
Detection and classification of nonspeech complex sounds
Authors: Solinsky J. C.
Published: October 2005

Complex sounds generated by nonspeech mechanisms contain acoustic energy represented in both temporal and spectral domains. Conventional energy detection and spectral classification approaches have inferior performance relative to non-Gaussian and nonlinear approaches when the sounds are from unknown sources. A two-stage approach will be presented that incorporates polyspectral and neural-network technologies in a sonar application that performs in a manner similar to human expert operators.
Some aspects of time, stimulus complexity, and gender in 40-Hz EEG responses
Authors: Weiler Ernest M. , Grayson Amy , Sandman David , Warm Joel
Published: October 2005

The present study investigated 40-Hz EEG responses of 17 listeners who received ipsilateral and contralateral story and click stimuli. An initial comparison between loudness adaptation and 40-Hz activity failed to show any change in central activity corresponding to change in loudness judgments. For the complete sample of listeners, a significant interaction was found between stimulus complexity, ear, and 40-Hz response quadrant. Subsequent investigation of gender differences across the temporal region suggests an extension of conclusions offered recently by Kimura. Implications will be discussed.
Frequency selectivity in noise-exposed chinchillas
Authors: Davis Robert I. , Hamernik Roger P. , Turrentine George A. , Ahroon William A.
Published: October 2005

Evoked-potential tuning curves (TCs) were recorded from chronic electrodes in the inferior colliculus of 102 chinchillas before and after acoustic overstimulation in order to relate the effects of changes in frequency selectivity to the condition of the cochlea. Pre- and post-exposure measures of auditory threshold and masked thresholds (simultaneous tone-on-tone paradigm) were obtained at 0.5, 1, 2, 4, 8, and 11.2 kHz. The three TC variables Q 10 dB, and the low- and high-frequency slopes were compared in two different groups of chinchillas: (1) animals (N = 54) with cochleas that showed distinctly normal and lesioned regions, and (2) animals (N = 48) that had near-normal thresholds, but either had lesions that were not manifested in the threshold measures or had nearly normal sensory cell populations. In the first group, the TCs measured from normal regions of the cochlea, in general, showed no statistically (t test) significant decreases in the group averaged low- or high-frequency slope, i.e., upward or downward spread of masking, for lesions located either apical or basal of the CF, respectively. In the second group, no significant differences in the TC variables were found in lesioned and nonlesioned cochleas with little or no PTS. These results show that (1) the quality of tuning is not necessarily altered in normal regions of the noise damaged cochlea, and (2) the TC cannot be considered as a reliable measure of sensory cell damage when the amount of PTS is small. [Research supported by NIOSH and USAARL.]
Recovery of auditory threshold following exposures to pure tones
Authors: Young I. M. , Lowry L. D. , Menduke H.
Published: October 2005

Recovery from temporary threshold shift (TTS) was measured for a subject with normal hearing following exposures to pure tones 250 and 1000 Hz. The intensity of the exposure tones was 125 and 123 dB SPL and duration was 30 and 21 min for 250 and 1000 Hz, respectively. Bekesy audiometer was used to obtain TTS and recovery at continuously variable frequencies between 100–10 000 Hz. Recovery of auditory threshold was measured at selected frequencies from 1 min to 47 h after an exposure to 250 Hz and from 1 min to 51 days after an exposure to 1000 Hz. Results indicated that (1) there was the greatest TTS at frequencies around 1/2 octave above the exposure frequency, (2) there was steady progression of recovery to pre-exposure threshold at all selected frequencies, and (3) recovery was neither a simple linear nor monotonic process.
The oceanic world of infant and adult microbubbles
Authors: Medwin Herman
Published: October 2005

It is now quite clear that vast number of microbubbles are produced by breaking waves and precipitation, and that they start their lives as shock-excited, damped, nonlinear, pulsating sound sources. These screaming infant microbubbles are active for only milliseconds and are, therefore, very close to the ocean surface while they radiate. In the presence of a rough ocean surface, their radiation patterns may appear to come from tilted dipoles or even monopoles, depending on the ocean wave spectrum, the frequency of the sound radiation, and the listening angle. In the absence of rain, it is those infant bubbles created by breaking waves that are responsible for the Knudsen sea noise spectrum from 500 Hz to 20 kHz or beyond. Those active bubbles have been used to determine the ocean surface bubble density, as a function of bubble radius, caused by intermittent spilling breakers during low sea states. Another major component of underwater sound, the infant bubbles produced by rainfall, show promise as keys to the remote determination of the rainfall spectrum (number of drops as a function of the drop diameter) at sea. On the other hand, the passive adult microbubbles, whose highly variable numbers must also depend on the dissolved organic and detrital character of the local ocean, are excellent tracers of near-surface ocean processes. Some recent laboratory and ocean experiments that lead to those understandings will be reviewed. [Research supported by ONR.]
A study of the damping of bubble oscillations
Authors: Longuet-Higgins Michael
Published: October 2005

The rate of damping of bubble oscillations provides us with a clue as to their precise origin and mode of generation. Data on rates of damping of acoustical pulses from various sources will be examined, particularly those from low sea states. At a given frequency, the observed rates vary by a factor of order 10; the minimum damping rate corresponds closely to thermal damping. Theoretical reasons for the higher damping rates will be discussed, including the transfer of some energy to shape oscillations.
Classification of underwater acoustic transients by artificial neural networks
Authors: Greene Ronald L. , Field Robert L.
Published: October 2005

Artificial neural networks have been trained using the backpropagation algorithm to classify a variety of model transient source signals. The networks were then tested on signals propagated to 25 different receiver sites by the time-domain parabolic equation model. Despite the interference effects from surface and bottom reflections, the classification accuracy is about 90% in the noise-free case, virtually identical to that of a nearest-neighbor classifier on the same problem. Classification in the presence of noise is considerably reduced; however, the redundancy provided by the multiple receivers in most cases allows the network to correctly classify all signals from sources on which it was trained. In addition, it shows a robustness in the presence of unknown signals not shown by the nearest-neighbor classifier. [Work supported by the Naval Oceanographic and Atmospheric Research Laboratory through the U. S. Navy/ASEE Summer Faculty Research Program.]
Matched-field processing using a neural network with preprocessing
Authors: Ozard John M. , Zakarauskas Pierre , Ko Peter W.
Published: October 2005

Model-based signal processing for source localization usually requires a comparison of the measured and replica acoustic fields at all possible source positions. It is anticipated that this problem may be solved more efficiently as a pattern recognition problem. In order to make such a solution applicable to arbitrary array shapes, the covariance matrix of the simulated data was preprocessed and represented as the excitations of the eigenvectors. The orthogonal bases for the total possible signal space was employed. Localization of a source in depth was performed using a linear perceptron with the excitations as input. The perceptron was trained with the excitations for a distribution of source depths. Precision of localization of the resulting processor and its robustness in the presence of noise were measured and compared to the performance of a minimum variance matched-field processor. Extensions of the processor to estimate range and azimuth are being investigated by using a multilayer neural network with backpropagation training.
The effect of wall elasticity on the properties of a Helmholtz resonator
Authors: Photiadis D.
Published: October 2005

The effect of compliant walls on the properties of a Helmholtz resonator is examined. The effective stiffness of the resonator is decreased by the wall compliance, while the effective mass is unchanged to leading order. The radiation resistance is also decreased due to a cancellation between the radiation from the cavity opening and the radiation from the cavity walls. This leads to a reduction in the cross section at resonance by the factor [Zs/(Zs + Zf0)]4, where Zs is the wall impedance and Zf0 is the total fluid loading on the breathing mode of the cavity.
Anderson localization of one-dimensional wave propagation on fluid-loaded structures
Authors: Photiadis D.
Published: October 2005

One-dimensional wave propagation on a structure with irregularly spaced scatterers is known to be localized for any degree of irregularity if the underlying wave theory is local. Fluid loading introduces nonlocal interactions and tends to short circuit the multiple scattering effect leading to localization; in the asymptotic low-frequency limit one might expect localization effects to be washed out as a result of this effect. [D. Crighton, Proc. Inst. Acoust. 10(2), 1–37 (1988)]. When shifting to higher frequencies, fluid loading becomes weaker and the effects of Anderson localization are expected to be observed. In this paper, the effects of irregularity on wave propagation along a fluid-loaded structure are examined using numerical simulation as the primary tool. Results showing the spread of energy as a function of frequency and irregularity are presented.
Reflection of incident plane waves from plane ribbed panels
Authors: Maidanik G. , Dickey J.
Published: October 2005

The reflection of incident plane waves from plane uniform panels is specular and can be defined in terms of a reflection coefficient. When the panel is ribbed, the reflection, in addition, assumes a nonspecular term. In a model of the ribbed panel in which the ribs are noninteracting, the nonspecular term exhibits regular features when the ribs are regularly spaced. These features are commensurate with those of diffraction gratings in optics. When the natural interactions between the ribs are accounted for in the model, the correspondence with the diffraction grating breaks down in spectral regions. The phenomenon that underlies this breakdown is closely related to the “pass-and-stop-bands” phenomenon that characterizes the response of uniform panels to which regularly spaced ribs are attached. The results of computer experiments relating to the reflection of incident plane waves from the two models of ribbed panels will be cited and discussed.
Determination of structural coupling loss factors by modal coupling approach
Authors: Pan Jie , Bremner Paul
Published: October 2005

The coupling loss factors are important in SEA (statistical energy analysis) for predicting vibrational energy distributions and power flows between coupled structures. The traditional empirical and analytical methods for determining these parameters were either for high-frequency range or for coupled structures with very simple geometric shapes. Analytical results are presented in this paper, which can be used to determine the energy distributions and power flow between coupled structures and so the coupling loss factors at any frequency and for any structure shape. The numerical tests for the two coupled plate system is conducted by using the analytical results, and are compared with the SEA predictions and previous experimental results in the high-frequency range.
Rate representations of formant frequencies in the anteroventral cochlear nucleus
Authors: Sachs Murray B. , Blackburn Carol C. , Banks Matthew I.
Published: October 2005

Features of the spectrum of a complex stimulus may be encoded as a profile of discharge rate versus best frequency (cochlear place) across the population of auditory-nerve fibers. Such profiles have been shown to represent the formant frequencies of steady-state vowels and consonant-vowel syllables at low and moderate stimulus levels. At high stimulus levels, rate saturation and two-tone suppression degrade the representation in the population of high-spontaneous rate (SR) fibers, but not in the low-SR population. Therefore, the central auditory system could produce a representation that is robust across stimulus level by “selectively listening” to its high-SR auditory-nerve inputs at low stimulus levels and to its low-SR inputs at high levels. This possibility has been explored by measuring rate profiles across populations of cells in the anteroventral cochlear nucleus (AVCN). Specifically, the population of chopper units in the AVCN maintains a robust rate representation of vowel spectra across the entire range of levels tested. At low levels, rate profiles for these units are similar to those of their high-SR auditory-nerve inputs and at high levels, where the high-SR inputs are saturated, they maintain at least as good a spectral representation as do the low-SR inputs. New data and modeling results that relate to the mechanisms underlying this selective listening process will be considered. [Work supported by an NIDCD Award of Excellence.]
Formant coding in the cochlear nucleus—Is there a place for time?
Authors: Greenberg Steven , Rhode William
Published: October 2005

Ever since Ohm and Seebeck, hearing theory has pondered whether “place” or “time” is the dominant means by which acoustic information is encoded in the ear. Over the past decade it has become increasingly clear that, at the level of the auditory nerve, both place and timing mechanisms participate in the representation of complex sounds. As a consequence, attention is now focused on the cochlear nucleus (CN), to which the auditory nerve projects. Within this nuclear complex occurs a remarkable degree of physiological diversity of potential significance to speech coding. The present survey reviews the temporal response properties of the major CN cell types to voiced CV syllables and the potential role played by each in the encoding of speech. All of the major response classes—primarylike, onset, chopper, onset-chopper, and pauser-buildup—are capable of synchronizing to the fundamental frequency or first formant. However, the range of acoustic conditions over which they do so, as well as the precision and upper frequency limits of phase locking, appreciably differ. Of particular interest are the onset-choppers of the posteroventral division, which temporally encode the lower formants and f0 with a degree of precision unrivaled by other CN cells or auditory-nerve fibers. Their superior performance is probably a consequence of integrating afferent activity spanning a broad tonotopic range. The majority of CN units retain their ability to temporally encode low-frequency information in all but the most intense masking noise, consistent with the hypothesis that phase locking provides an effective means of reliably encoding informationally significant elements under acoustically uncertain conditions. [Work supported by NIDCD.]
Visual representations of speech—A computer model based on correlation
Authors: Slaney Malcolm , Lyon Richard F.
Published: October 2005

The use of the cochleagram and the correlogram in speech and sound recognition is discussed. The cochleagram represents a sound as a pattern of neural firing probabilities at places along the basilar membrane versus time. It is roughly analogous to the spectrogram and its benefits have been described in several papers. But using the cochleagram as a basis for speech recognition is only a weak way to use knowledge of human auditory processing A richer representation of speech and sound is called the correlogram. Sound is represented as a two-dimensional picture versus time—the extra dimension allowing several interesting perceptual experiences to be modeled. Assembling correlograms into a movie and synchronizing them to sound allow the auditory and visual percepts to be compared. The correlogram is more useful than a cochleagram (or spectrogram) because it shows an orthogonal dimension that represents the fine-time structure and pitch in the auditory signal. The extra dimension provides the information necessary for auditory grouping and speaker separation. This talk will emphasize the advantages of a two-dimensional representation of sound and describe several auditory maps that might be used by the brain to do auditory scene analysis. Videotapes will be shown to demonstrate the advantages of the correlogram representation.
Pitch, vowel quality, and the perceptual segregation of competing voices
Authors: Assmann Peter F. , Summerfield Quentin
Published: October 2005

An important problem faced by models of the auditory and phonetic analysis of speech is describing how listeners extract information from speech when competing sound sources are present. When the competing source is another voice, the task for the auditory system is complicated by the high degree of similarity in the spectro-temporal structure of the two sounds. This situation imposes special constraints on models of auditory processing. Some of these constraints are revealed in recent studies of the role of fundamental frequency (f0) and formant structure for grouping and segregating the acoustic components of two competing voices. In one set of experiments, listeners attend to “double vowels” created by mixing the waveforms of pairs of simultaneous synthetic vowels with similar overall amplitudes. Both constituents of double vowels can be identified with an accuracy greater than chance, even when they have the same f0. However, when a difference in f0 is introduced between the constituents, listeners show a 15%–20% increase in accuracy, reaching a maximum when the difference in f0 is 1–2 semitones. Recent attempts to model these findings have assumed that the identification of double vowels involves a source-segregation stage, and that pitch guides the process of segregation. These assumptions are supported by finding a close association between the number of pitches reported in double vowels and the accuracy with which the vowel constituents are identified. When double vowels are reduced in duration from 200 to 50 ms, listeners hear two distinct pitches on fewer trials, and identification accuracy is reduced. Three possible factors are considered that may account for this finding: (i) the reduction in pitch strength at short durations; (ii) temporal limits on attention switching; and (iii) a waveform-interaction hypothesis.
Perceptual categorization of vowel sounds: Representation in a self-organizing network with auditory inputs
Authors: Kluender Keith R. , Jenison Rick L.
Published: October 2005

Increasingly realistic auditory representations are becoming available to speech researchers; however, the utility of these representations as tools for studying phonetic perception is dependent upon assumptions regarding processes consequent to the auditory periphery. The case for using distances between vowels that have undergone auditory transformation as a metric for perceptual distinctiveness is examined. Lindblom and his colleagues, for example, have generally used a Euclidean distance between outputs from a simple auditory model to predict vowel inventories among languages. Some of the shortcomings related to this approach are described, and an alternative procedure more closely related to categorization, which is conceptualized as a process by which a system comes to efficiently organize and encode events in a perceptual space, is recommended. The statistical technique of principle component analysis (PCA), which has been shown to be the linear algebraic equivalent to a class of unsupervised neural network models with inhibitory connections, is exploited. Having applied PCA to a set of spectra that uniformly sample the range of producible vowels and have undergone nonlinear auditory transformation, vowels on a small number (three-four) of orthogonal dimensions accounting for almost all of the variance in the auditory vowel space can be plotted. The efficacy of this approach is evaluated by using distances between vowels plotted on these derived axes to predict favored vowel inventories across languages. The relation between these dimensions and formant patterns will be described, and the relation between perceptual organization and categorization will be discussed. [Work supported by NIDCD.]