bluuebunny/crossref_metadata_2025 huggingface dataset:
Spectral acoustics of speech in the ear
Authors: Swartz Bradford L. , Church Gerald T.
Published: October 2005

The spectral acoustics of the sounds of speech as measured upon production are well known. But the spectral features of these sounds as they exist in the ear canal immediately prior to auditory reception are somewhat different from those typically cited, due most likely to the resonance characteristics of the external meatus. A recent pilot study using probe microphonics and spectral analysis revealed that the formant structures of the vowel sounds /i, u, ɜ/ as recorded in the ear canal were similar to those recorded outside of the ear, but the amplitude of the second formant was reduced in each case. These findings have potential implications for the importance of the second formant in the perception of these and other vowel sounds. The spectral structure of more vowels, consonants, and sounds in speech are yet to be explored when recording is performed from inside the external ear canal.
The acoustic and perceptual basis of stress shift
Authors: Gow David W. , Gordon Peter C.
Published: October 2005

The acoustic and perceptual basis of stress shift was examined in 24 noun phrases in sentence context. Acoustic analyses involved the comparison of word-initial syllable duration, amplitude, and fundamental frequency in words which either undergo or do not undergo stress shift as required by the rhythm rule. In addition, the time course of a perceptual stress effect found in earlier work with a short-term memory probe task [D. W. Gow and P. C. Gordon, J. Acoust. Soc. Am. Suppl. 1 86, S80 (1989)] was investigated. Subjects responded to a probe syllable either the offset of target syllables, the offset of target-bearing words, or 450 ms after the offset of target-bearing words. Responses associated with probes at different post-target latencies and stress conditions were analyzed to examine the role of post-syllable context on stress assignment. The results of both sets of analyses were used to evaluate the psychological reality and processing implications of the rhythm rule. [Work supported by AFOSR.]
The time course of auditory segregation of simultaneous vowels
Authors: McKeown Denis
Published: October 2005

The perceptual segregation of simultaneous speech is known to benefit from differences in pitch or loudness between the speech sounds. It is likely that different cues operate on different time courses. A series of experiments examined subjects' ability to identify both members of simultaneously presented steady-state vowel pairs. The duration of each vowel pair varied from 1 to 8 cycles. A “dominant” vowel was identified with near 100% accuracy even in the single-cycle vowel pair condition (one pitch period), whereas the recognition of the “nondominant” vowel showed a slow improvement up to the 8-cycle condition. Asynchrony between the pitch periods of the vowels improved performance when they had a low fundamental frequency (80 Hz), but about 3 cycles of the vowels were necessary for this advantage. The effects of changing the amplitude of one vowel relative to the other and the effects of differences in frequency modulation were also studied. The results support previous research in showing that phonetic decisions “await” organizing mechanisms that operate in the auditory periphery. [Work supported by Esprit BRA 3207.]
Source imaging and sidelobe suppression using time-domain techniques in a shallow-water waveguide
Authors: Feuillade C. , Clay C. S.
Published: October 2005

Source localization in a shallow-water waveguide, using harmonic (cw) techniques, is typically complicated by the repetitive sidelobe structure of the acoustic field. For this reason, much interest has been shown recently in the development and implementation of time-domain methods, which should achieve better performance because of their additional frequency-averaging capability. In a previous publication [C. S. Clay, J. Acoust. Soc. Am. 81, 660–664 (1987)], the basis for optimum signal transmission and source localization in a waveguide using time-domain techniques was described. In the work reported here, using a simple two-layer model with shallow water depth (20 m) and short range (1500 m), the variation in time-domain source image resolution and sidelobe suppression as a function of frequency bandwidth, mode selection, and hydrophone receiver emplacement has been investigated. The results indicate that source localization performance is most strongly dependent upon the bandwidth and number of modes available to carry the signal transmission. The physical mechanism for narrowing the source image and suppressing sidelobes is spatial averaging of the acoustic field due to variation of the horizontal wave numbers for the modes as a function of frequency. Improvements in mode sampling, by increasing the number of receivers, also served to reduce the level of sidelobes, but did not improve the resolution of the image. [This work was sponsored by the Naval Oceanographic and Atmospheric Research Laboratory.]
Stability of incoherent frequency-averaged matched-field processing against random phase errors
Authors: Smith G. B. , Feuillade C.
Published: October 2005

Matched-field processing (MFP) must frequently be employed under conditions of significant environmental and array element location mismatch. This necessitates the development of techniques that stabilize MFP with respect to mismatch errors for successful matched-field localization of submerged sources. A technique that offers this kind of stability (as well as some noise rejection) is incoherent frequency averaging [A. Baggeroer, W. Kuperman, and H. Schmidt, J. Acoust. Soc. Am. 83, 571–587 (1988)]. This technique incoherently averages a number of single-frequency matched-field ambiguity surfaces together to simulate broadband averaging. Its ability to suppress signal and noise sidelobes has already been demonstrated [G. B. Smith, D. R. DelBalzo, and C. Feuillade, J. Acoust. Soc. Am. Suppl. 1 83, S101 (1988)]. In this paper, the ability of incoherent frequency averaging to suppress false peaks caused by destabilization due to mismatch errors is investigated. In this preliminary study, mismatch errors are modeled as random perturbations of the phases of the complex pressures. It is demonstrated that this technique does offer significant stabilization with respect to these random phase errors. The relationship between the number of surfaces needed in the average and the amount of random phase error introduced will be investigated. [This work is supported by the NOARL basic research program.]
Acoustic holography in a range-dependent ocean: Using a reference source to minimize oceanic variability in source localization problems
Authors: Mourad Pierre D. , Porter Robert P.
Published: October 2005

A mathematical process based upon imaging techniques that remove a significant percentage of the random oceanic variability between an acoustic receiving array and an unknown source is described. This procedure is based upon combining measured data from the unknown source with measured data from a known reference source. It is assumed that the latter is in the vicinity of the former relative to the receiving array. The closer the sources are to one another, the larger the percentage of oceanic variability that may be removed by this technique. The heart of the concept is the realization that both sets of measured data are contaminated in the same way by the integrated effects of the random ocean environment, for the stretch of ocean that they share in common between their positions and the receiving array. These integrated effects may be made to cancel one another, leaving the much smaller and more manageable integrated effects of the oceanic variability between the reference source and unknown source.
An iterative scheme to reduce the monitoring time interval for localizing a moving source
Authors: Chen Han-Yang , Lu I-Tai
Published: October 2005

When the velocity of a moving source is much smaller than the sound velocity of the waveguide, the acoustic field generated by this moving source can be represented in terms of normal modes with different Doppler-shifted frequencies and, hence, different changes of modal eigenvalues. Using these Doppler's shifts, matched-mode processing schemes for estimating the frequency, velocity, depth, and range information of a time-harmonic point source moving uniformly in stratified oceanic waveguides has been presented [H. Y. Chen and I. T. Lu, J. Acoust. Soc. Am. Suppl. 1 87, S153 (1990)]. The processing schemes over time domain is especially promising because only one receiver is required. Here, an iterative algorithm to reduce the monitoring time is used [A. Papoulis, IEEE Trans. Circuits Sys. 25, 735–742 (1975)], which is essential in practical applications. The central problem here is to determine the frequency spectrum F(ω) of measured data f(t) only known in finite time interval. Due to finite Doppler shift, f(t) is a band-limited function that is analytic in the entire t axis. Therefore the unknown part of f(t) can be determined by some forms of extrapolation. Since the received f(t) is only known in finite time interval, its frequency spectrum F̂(ω) will not be band-limited. Those spectral constituents outside the frequency band of F(ω) are not supposed to exist. Performing inverse transform of the truncated F̂, some of the unknown portions of f(t) were extrapolated. Combining these unknown portions with the known portions of f(t), new “received” data defined over the entire time domain were obtained. The above procedure can be iterated until the solution converges. The method involves only discrete Fourier series and makes use the N log N economy of the FFT. Numerical calculations show very promising results.
A new method for range and depth source localization in shallow water
Authors: Jesus Sérgio M.
Published: October 2005

Range and depth source localization in shallow water amounts to the estimation of the normal mode structure of the acoustic field. As “seen” by a vertical array, and from a modeling point of view, the normal mode structure appears as a set of nonplane coherent waves closely spaced in angle. The technique presented in this paper uses the eigen decomposition of the data cross-correlation matrix in order to resolve this set of nonplane waves and to estimate the subspace spanned by the normal modes that are significantly excited by the source(s): the mode subspace. The projection of the replica pressure vector onto the mode subspace for all possible range/depth source locations gives the final ambiguity surface. Results obtained on realistic environments show that its performance is always greater or equal to that of the generalized MLM processor [A. B. Baggeroer, W. A. Kuperman, and H. Schmidt, J. Acoust. Soc. Am. 83, 571–587 (1988)]. Good performance results have been obtained on short time records; this suggests the method may be applicable to localizing moving sources and/or sources emitting transient signals. Preliminary results obtained on real data will be discussed and compared with those obtained by the conventional and MLM processors.
Performance bounds on the passive localization of a moving source for ocean acoustics
Authors: Song Hee Chun , Baggeroer Arthur B.
Published: October 2005

Matched-field processing (MFP) for locating a point acoustic source in the ocean using a vertical array is extended to treat a moving source problem. The extension involves both temporally nonstationary and spatially inhomogeneous structure of the sound field generated by a time-harmonic point source moving uniformly in a stratified oceanic waveguide. Using normal mode description of the sound field, the focus was on the effect of source motion on MFP. An optimum receiver based on maximum likelihood method is developed in the presence of spatially and temporally white noise. The generalized ambiguity function (GAF) was used to analyze problems of accuracy, ambiguity, and resolution. The principal result is the demonstration that a moving source problem can be treated as a stationary source problem if the source travel distance (uncompensated speed × time window) is less than half the wavelength of trapped modes. Also, a closed-form expression for the optimum potential resolution is derived based on the Cramer-Rao bound. The lower bound provides physical insight of how each mode contributes to the localization process, and can be easily evaluated for a wide range of source positions in any sound channel using sound channel eigenfunctions, eigenvalues, and the number of modes involved. Simulations of GAF and the bounds for Arctic environment illustrate the coupling of ocean environment to the localization performance.
Detection and time-delay estimation for hand-limited transients using higher-order correlations
Authors: Pflug Lisa A. , Ioup George E. , Ioup Juliette W. , Barnes Kenneth H. , Field Robert L. , Rayborn Grayson H.
Published: October 2005

Previously reported research examining the performance of higher-order correlation detectors for band-limited transient signals using hypothesis testing has been extended to include alternate detection methods, detector performance as a function of signal-to-noise ratios (SNR's), and time-delay estimation. In particular, frequency-domain detection has been compared to detection in the time domain. Probability of detection is determined for example transients at a fixed false alarm rate over a domain of SNR's. The accuracy of time-delay estimation is investigated for these signals using higher-order correlation estimators for the same noisy transients. Signals affected by the environment are also included in the analyses. [Work supported by NOARL and the Navy/ASEE Summer Faculty Fellowship Program.]
A method for estimating an acoustic source signature using deconvolution
Authors: Finette Steven
Published: October 2005

The signal received at a hydrophone located at some distance from an acoustic source can be regarded as a temporal convolution of the source function with the ocean's impulse response (or Green's function). The latter quantity is the solution of a wave equation describing propagation from a point source to the receiver location(s) with appropriate estimates for environmental parameters, e.g., source location, sound-speed profiles, etc. In this paper, a method for separating the impulse response from the source signature by a two-step process is considered. First, the source location in range and depth is estimated by using the received signals on a vertical array of hydrophones in conjunction with matched-field processing. This information is incorporated in a normal modes propagation model that allows us to obtain an estimate of the impulse response. Next, the received data and the impulse response are combined in a least-mean-squares deconvolution algorithm in order to estimate the source time signature. We will discuss the experimental data and some results using chirp source signals in a deep ocean environment. [Work supported by ONR.]
Optimum signals having specified frequency spectra
Authors: Johns Murray D.
Published: October 2005

A common requirement in active acoustic measurements is to generate a periodic signal having a specified frequency spectrum and maximum average power. The signal amplitude is usually limited by a constraint on the peak signal power. A new algorithm for designing such signals is presented. An initial constant amplitude periodic chirp with approximately the desired spectrum is refined by alternate-frequency and time-domain operations until the process converges. The resulting signals are commonly within 0.1 dB of the theoretically optimum signals.
Characteristics of sound radiation from large raindrops
Authors: Snyder David E. , Jacobus Peter W. , Medwin Herman , Nystuen Jeffrey A.
Published: October 2005

Drop diameters from 2.7 to 4.5 mm are common in heavy rainfall. By using a hydrophone with a flat response up to 300 kHz and A/D conversion up to 1 MHz the components of the sound radiation for these large terminal-velocity drops in the time and frequency domains can be identified. At these high sampling rates it is possible to see notches on the lagging edge of the impulse trace. These notches at 100 kHz and higher frequencies are probably caused by internal drop reflections. Lower frequency oscillations with a range of frequencies less than 10 kHz occur after a time lag of 30 ms or more. Both the frequency of oscillation and the time lag appear to be the functions of drop size. Although the amplitudes of these components are comparable for individual drops the energy radiated by the low-frequency oscillation is greater due to its longer duration. These early results suggest that it may be possible to ascertain the drop spectrum (number of drops as a function of diameter) in rainfall from the spectrum of the underwater sound of the rain. [Research supported by ONR.]
Broadband undersampled synthetic aperture arrays: Targets stay sharp, aliases smear
Authors: Rolt Kenneth D. , Milgram Jerome , Schmidt Henrik
Published: October 2005

Synthetic aperture sonars (SAS) or radars (SAR) that transmit narrow-band signals are sampled arrays and therefore must satisfy sampling requirements to avoid aliasing. The sampling requirements for a narrow-band SAS or SAR may be relaxed if a broadband signal is used. From a beamforming view, broadband transmit signals result in a mainlobe that always points in the same direction and has a width that varies as the transmit signal sweeps through the broadband, but having aliasing lobes that change in both width and direction. A composite broadband beam pattern would show constructive mainlobes but smeared aliasing lobes. Broadband undersampled synthetic aperture images similarly show a target image having the same x–y image space position for all frequencies, but alias target images that have azimuth positions that vary with frequency. A synthetic aperture image uses the entire transmit band, and so, targets appear sharp and aliases become smeared. An approximate relation for the level of smearing of the alias targets may be shown, which depends on the signal bandwidth, the platform speed, the sonar along-track length, center frequency, and the pulse rate. This allows the design of broadband undersampled synthetic aperture sonars with higher speeds and greater mapping rates. [Work supported by the Charles Stark Draper Laboratory, Cambridge, MA.]
Frequency importance functions
Authors: Studebaker G. A.
Published: October 2005

It is widely held that certain parts of the auditory spectrum are more important than others for speech recognition. However, the concept of frequency importance for speech recognition has received relatively little attention from researchers over the years. This was probably because most assumed that published frequency importance functions for nonsense syllables were satisfactory for most applications. Recent research suggests that different types of speech materials have different importance functions and that predictions of speech recognition for one speech material cannot be made accurately under all conditions using an importance function for a different material. Research on this topic will be reviewed and some recent data presented. An important remaining question is the effect that size of the frequency importance function has under different listening conditions. Early findings suggest that, although the importance function has relatively little influence on outcomes under some conditions, it can be substantial in others.
Predicting audible distortions in speech
Authors: Levitt Harry
Published: October 2005

The reproduction of speech by public address systems, or other means of amplification, is typically accompanied by electronically generated distortions. These distortions can be both linear (e.g., modification of frequency response) or nonlinear (e.g., peak clipping) and may also include the addition of background noise. Nonlinear distortion is of particular concern since it can be highly disruptive of speech quality. A generalized index for predicting distortion in amplified speech signals has been developed. It is similar in concept to the articulation index and can be used to predict the audibility of nonspeech components in the speech signal, such as noise and/or nonlinear distortion. Linear distortions are treated as changes in the speech spectrum. An earlier version of this index was used to predict binaural release from masking and gain in intelligibility [H. Levitt and L. R. Rabiner, J. Acoust. Soc. Am. 42, 820–829 (1967)].
Species-specific perceptual processes in budgerigars, canaries, and zebra finches
Authors: Brown Susan D. , Dooling Robert S. , Okanoya Kazuo
Published: October 2005

Budgerigars, canaries, and zebra finches were tested for their ability to discriminate among distance calls of each species. Response latencies to detect a change in a repeating background of sound were taken as a measure of the perceptual similarity among these stimuli. Each species showed an enhanced ability to discriminate among conspecific distance calls compared to those of the other two species. Budgerigars and zebra finches also showed an enhanced ability to discriminate between conspecific calls and the calls of the other two species. Perceptually salient acoustic characteristics of these calls for each species were identified using multidimensional scaling and regression analyses. There were differences in the acoustic characteristics used by each species to discriminate among the calls. This suggests that the same acoustic features may have different communicative significance for each species. These findings support the notion that specialized processing mechanisms may be a common feature of avian acoustic communication systems.
Swimbladder modeled as a finite gas-filled cylinder
Authors: Clay C. S.
Published: October 2005

Traditionally, swimbladders of fish have been modeled in two regions. In the acoustic long wavelength region, i.e., λ much larger than the dimensions of the swimbladder, the gas-filled bubble has been used. Since a spherical bubble has a very sharp resonance, damping terms have been included to broaden the resonance peak. The spherical bubble has been stretched into a prolate spheroid without reformulating the bubble theory [M. Strasberg, L Acoust. Soc. Am. 25, 536-537 (1953) and R. H. Love, J. Acoust. Soc. Am. 64, 571-580 (1978)]. At short wavelengths compared to the dimensions of the swimbladder, the scattering process depends on the shape of the swimbladder. Experimental and theoretical studies of the sound scattered by swimbladders include measurements of fish and numerical integrations of the Kirchhoff diffraction integral. The Kirchhoff approximations give “good” approximations to the backscattering cross sections at short wavelengths [K. G. Foote, J. Acoust. Soc. Am. 78, 688-700 (1985) and M. A. Do and A.M. Surti, J. Acoust. Soc. Am. 87, 1588-1596 (1990)]. For a single model, a finite gas-filled cylinder can be used over the whole frequency range. Stanton's approximation for the finite fluid cylinder was used and the fluid used was a gas [T. S. Stanton, J. Acoust. Soc. Am. 83, 55-63 (1988)]. It will be shown that the gas-filled cylinder model fits a well-documented set of data [D. V. Holliday, J. Acoust. Soc. Am. 51, 1322-1332 (1972)]. [Work supported by ONR.]
Fundamental frequency declination is not unique to human speech: Evidence from nonhuman primates
Authors: Hauser Mare D.
Published: October 2005

In human speech, declination of the fundamental frequency of the voice (F0) spans syntactically coherent units of an utterance and is used by listeners to determine where syntactic units begin and end. Because declination is present in a number of languages, it has been suggested as a language universal. Theories of language universals describe phenomena that are claimed to be particular to speech and thereby exclude the possibility of similar phenomena in nonhuman animal communication systems. Acoustic analyses reveal that vocal utterances produced by vervet monkeys (Cercopithecus aethiops) and rhesus macaques (Macaca mulatta) show declination in F0. Moreover, in certain respects declination appears to participate in communicative exchanges in a way analogous to its role in human speech. These results argue against the claim that declination is unique to speech. [Work supported by NIH and the National Geographic Society.]
Hand-arm vibration syndrome and hearing loss in hard rock miners
Authors: Pelmear P. L.
Published: October 2005

Heating loss in forestry workers from hand-arm vibration exposure has been reported in several studies. To evaluate this association further, a hand-arm vibration survey of hard rock miners in five mines in Ontario included the collection of audiometric data. Both air conduction and impedance audiometry were used. The presence and severity of hand-arm vibration syndrome were assessed by an occupational history and clinical tests. The hearing levels of both ears were combined, and by multiple regression analysis it was determined that there was a significant association between hearing level (at 4 kHz) and stage of hand-arm vibration syndrome; hearing loss and age; and hearing loss and vibration exposure years. The hearing loss was significantly greater in miners with Raynaud's phenomenon who had less than 11-years hand-arm vibration exposure. The mechanism is still uncertain but it would seem that the accelerated hair cell destruction in such patients occurs because of reflex vasospasm of cochlea vasculature through the autonomic system.
Changes in TTS2 values and tinnitus sensations due to complex exposure conditions among different human age groups
Authors: Manninen Olavi J.
Published: October 2005

The study took place in an exposure chamber as a factorial experiment yielding a total of 48 exposure combinations, with 192 healthy, nonsmoking, male volunteers who were under 25 yr, 26-39 yr, and over 40 yr old participating. The noise categories were (1) no extra noise and (2) stable 90-dBA broadband noise. The vibration conditions included (1) no vibration and (2) whole-body vibration with a frequency range of 2.8–11.2 Hz (along Z axis). The classes of physical work loads were (1) work at 2 W and (2) work at 7 W. The mental work was (1) no competition and (2) competition on least errors in mental arithmetic. All experiments were carried out at 35 °C. The TTS2 values were determined in both ears at 4 and 6 kHz. Tinnitus sensations were obtained through a five-point rating scale. Results showed that noise, vibration, and physically loading work and their corresponding combinations influenced TTS2 values and tinnitus sensations most markedly. On the other hand, the combined effects of noise and vibration on TTS2 values were age-dependent. In general, changes in tinnitus sensations seemed to be associated with the changes in hearing thresholds; the greater the TTS2 values, the greater were the means of tinnitus rating scores. [Work supported by the Academy of Finland.]
Effects of frequency and acceleration of whole-body vibration combined with ototraumatic noise
Authors: Byrne Clyde D. , Henderson Don
Published: October 2005

The effects of simultaneous exposures to whole-body vibration and steady-state noise were observed for thresholds of hearing in chinchillas. Four groups of animals were exposed to one of the following conditions: (a) noise only (500 Hz centered octave band, 100 dB SPL); (b) vibration alone at body resonance (63–89 Hz, 0.1 grms, 1.0 grms); (c) noise + resonant vibration; and (d) noise + nonresonant vibration (14–20 Hz, 0.1 grms, 1.0 grms,). The exposures lasted 6 h per day over 20 consecutive days. No interaction was demonstrated with the combination of nonresonant vibration and noise, but small increases in TTS and PTS were observed when the resonant vibration was added to the noise. Increasing the acceleration from 0.1 grms to 1.0grms did not result in a corresponding increase in TTS. Thresholds in the group exposed only to vibration improved slightly.
Interaction of noxes: Amelioration, summation, or synergism
Authors: Ward W. Dixon
Published: October 2005

That two noxious agents A and B act synergistically cannot be stipulated merely because their combined effect is greater than the effect of either separately. Instead, it must be shown that the effect is greater than predicted by the exposure-equivalent principle, which can be accomplished only if the rate of growth of effect with increasing exposure to at least one noxious agent (A, say) is known. In that ease, the value of exposure to A that would have produced the effect that was in fact produced by B can be calculated; adding this exposure equivalent and the actual exposure to A will give a total exposure to A whose effect can be predicted from the growth function. Only if the combined effect actually observed exceeds this value can synergism be claimed. This principle was applied in order to show that there is no synergism between steady 90-dBA noise and impulse noise with a peak level of 139 dB SPL, which means that crest factor is not an important parameter in the assessment of hazard from noise exposure. [Work supported by NIH.]
The development of interior noise and vibration criteria
Authors: Stephens David G. , Leatherwood Jack D. , Clevenson Sherman A.
Published: October 2005

The NASA Langley Research Center has completed a comprehensive research program that resulted in the development of a generalized model for estimating passenger discomfort response to combined noise and vibration. This model accounts for multiple frequency and multiple axes of vibration as well as the interactive effects of combined noise and vibration. The model has the unique capability of transforming individual components of a noise/vibration environment into subjective comfort units and then combining these comfort units to produce a total index of passenger discomfort and useful subindices that typify passenger comfort within the environment. This paper presents an overview of the model development including the methodology employed, major elements of the model, model applications, and a brief description of a commercially available portable ride comfort meter based directly upon the model algorithms. Also discussed are potential criteria formats that account for the interactive effects of noise and vibration on human discomfort response.
Systems theory and the active control of duct noise
Authors: Eriksson L. J.
Published: October 2005

The behavior of an acoustical system may be represented using either a transfer function or impulse response. This representation enables an electrical model of the acoustical system to be created. The output of this model may be used to drive an acoustical source and produce a secondary wave to cancel an undesired primary wave. A system identification process generates the required model using an adaptive parameter estimation scheme. A variety of alternatives are available for the model structure as well as the parameter estimation scheme. These include a finite impulse response (FIR) or infinite impulse response (IIR) digital filter with parameters obtained using a least-mean-square (LMS) or recursive least-mean-square (RLMS) adaptive algorithm, respectively. Since the process is one of adaptive control, the filtered-X or filtered-U algorithm can be used to compensate for various transfer functions, identified on an off-line or on-line basis, following the adaptive model. Results will be presented from a variety of applications of this technology.
Computational optimization of preditive filter for noise cancellation
Authors: Harley T. R. , Hendrix James E. , Shields F. D.
Published: October 2005

Noise cancellation of stationary random signals has been obtained at an isolated point in space. This has been done without using adaptive techniques, an error microphone, or any sort of “upstream” preview of a signal correlated with the noise source. The antinoise speaker is driven by a digital signal processor implementing an IIR filter using sampled input from a microphone placed at the point where noise cancellation is desired. By using a simple physical model of the total noise at the microphone, the coefficients of the DSP IIR filter can be defined. A linear prediction has been incorporated into the IIR filter in order to anticipate the level of sound that will be produced by the noise source in the future. Experimental cancellation results were attained by using an initial guess of the optimal IIR DSP implemented filter. The work presented here involves computationally finding the set of filter coefficients that gives the theoretically optimal reduction in noise. The total reduction in decibels of the entire noise spectrum is minimized subject to the constraint of feedback stability.
Active noise control of time-varying signals in ducts
Authors: Ng Kam W. , Leinhos Henry A.
Published: October 2005

A sound cancellation system has been developed to reduce the flow-induced noise in piping systems under both steady-state and transient conditions. A hybrid controller that combines both open-loop and closed-loop responses with adaptive digital signal processing and filtering techniques is utilized to speed up the response of the system controller for the transient condition. Autoregressive, moving average with exogenous input (ARMAX) alogrithm has been utilized in the on-line and off-line modeling of signal characteristics. Off-line model data are introduced into an on-line adaptive system model via a parameter leakage mechanism. The convergence of the resulting algorithm is improved at the expense of bias in the final system parameters. The bias depends on a weighting or leakage factor, which controls the relative amount of off-line data to be used. A preliminary test has been conducted to demonstrate the active cancellation concept for synthesized steady-state signals. Results showed that up to a 30-dB reduction has been obtained for pure sinusoidal waves ranging from 100 to 500 Hz. For confined jet noise, the noise reduction is about 10 dB, and similar results have been observed for the random broad-band noise.
Hot spots and supermodes: The theory of ambient sound in the ocean-surface bubble layer
Authors: Buckingham Michael J.
Published: October 2005

An elegant series of experiments was conducted recently by Farmer and Vagle [J. Acoust. Soc. Am. 86, 1897–1908 (1989)], in which the sound from individual wave-breaking events was recorded by a hydrophone located a few meters beneath the ocean surface. Between 1 and 20 kHz, the observed spectra from La Perouse Bank contain relatively sharp, isolated peaks (hot spots) at intervals of approximately 3 kHz, whereas the data set from the FASINEX experiment shows energy spread over broad spectral bands (supermodes), perhaps 3 kHz wide, and separated by fairly well-defined energy gaps (i.e., spectral minima). Both types of spectrum are a manifestation of waveguide propagation in the upward refracting, ocean-surface bubble layer. A new, modal theory of sound in an upward refracting duct shows that, in general, the mechanism responsible for the observed hot spots and supermodes is intermode interference. The qualitative differences between the two data sets arise because the number of modes supported by the profile at La Perouse is significantly less than the number in FASINEX. In both cases, the detailed features displayed by the theoretical and experimental spectra show an almost one-to-one correspondence. [Research supported by ONR.]
Modal analysis of vibrations in musical instruments
Authors: Rossing Thomas D. , Hansen Uwe J.
Published: October 2005

Modal analysis describes the properties of an elastic structure in terms of its modes of vibration. Videotape is a convenient medium for presenting the results of experimental modal analysis of structural vibrations, both in a lecture and in the laboratory. Some modes of vibration of several musical instruments, including a guitar, a handbell, a Caribbean steel drum, a snare drum, and a plane soundboard, will be demonstrated on videotape.
Sound and wave motion film excerpts available on videodisc
Authors: Wright Wayne M. , Winch David M.
Published: October 2005

A National Interactive Media Project for Secondary Physical Science Courses, supported by U. S. Department of Education Award No. R168D90059 to the American Association of Physics Teachers, is scheduled to be completed by the summer of 1992. The focus of this effort, which is being managed by the AAPT Instructional Materials Center, is to make portions of existing archival physics films available for use by teachers as interactive lessons on an inexpensive videodisc system. Professional film-to-video transfer is being supervised by R. Fuller at the University of Nebraska—Lincoln, while supervision of the generation of interactive software is by D. Winch at Kalamazoo College. Students are expected to be able to interact directly with visual images of natural phenomena on a TV screen. Materials that are already available in a video format, such as “Tacoma Narrows Bridge Collapse,” are not included. Instructor's hints are placed on the disc as text frames in front of each chapter. The two sound tracks on the disc consist of an inquiry audio track and an expository, more advanced track. Associated with each disc will be computer software (PC and Macintosh) to support the educational uses of the video images. Examples of the acoustics-related film material will be demonstrated.
Laboratory measurements of wind-wave dynamics
Authors: Mellen R. H. , Leykin I. A.
Published: October 2005

Doppler measurements can help to discriminate between waves and bubbles as mechanisms for surface backscattering. However, the analysis depends on knowledge of the near-surface dynamics. Wind-wave measurements of wave height versus time in a laboratory flume indicate that conventional linear wave models are not adequate at the higher wave numbers involved. Frequency autospectra have F−3 asymptotic dependence (compared to F−5 for the Pierson-Moskowitz spectrum) and cross spectra show nearly dispersionless downwind propagation. the speed increasing with wind speed. The F−3 spectral dependence appears to be due to an effective discontinuity of slope at the wave fronts caused by the finite-amplitude overtaking effect. Backscattering calculations based on a nonlinear wave model similar to that used for convective turbulence are in fair agreement with flume measurements.
Determination of sea surface characteristics from acoustic tomography data
Authors: Lynch James F. , Miller James H. , Worcester Peter
Published: October 2005

For many years, acoustic tomography avoided using acoustic paths that interacted with the ocean's surface, due to scattering loss and stability considerations. However, experiments off Spitzbergen, in Monterey Bay, and in the Greenland Sea have shown that such paths contain useful information about both the ocean's interior and surface. Transmissions off Spitzbergen and in Monterey Bay used rapidly sampled time series of multipath travel times to recover sea surface frequency spectra, as well as tidal, internal wave, and wave group signals. The results from these two experiments, and their applications to coastal oceanography, will be discussed. For larger scale tomography, such as the Greenland Sea experiment, one samples the surface only every few hours, thus precluding surface wave frequency spectrum measurements. However, by monitoring the decrease in amplitude of surface reflected rays due to scattering loss, one can still map rms wave height over the area of the array. Simulated and actual data from the Greenland Sea tomography will be presented to demonstrate this technique. Comparisons of the tomographic results to the output of wave models inputting wind stress will also be made. [Work supported by ONR and NSF.]
Models for acoustic scattering by trees
Authors: Rogers James C.
Published: October 2005

Acoustic scattering by trees has most often been considered in the context of the attenuation caused by scattering as sound propagates through a forest. Single cylinder scattering has been used as a model by Aylor [J. Acoust. Soc. Am. 51, 197 (1972)]. Price et al. [J. Acoust. Soc. Am. 84, 1836 (1988)] modeled the process as plane waves multiply scattered by a randomly spaced array of cylinders of infinite length. Their work, based upon theory by Embleton [J. Acoust. Soc. Am. 35, 1119 (1966)], showed that a single arithmetic mean diameter provided an adequate approximation to predictions obtained with a diameter distribution. However, the scatterer densities required were approximately 60% less than that measured. Recently, Rogers and Lee [Proc. 13th Int. Congr. Acoust. 3, 277 (1989)] have taken a different approach to studying the scattering problem and have directly measured the backscattered component. Additional measurements have provided scattering versus azimuth angle [in Proceedings of the 4th Long-Range Sound Propagation Conference, Hampton, VA, 1990 (in press)]. These data are modeled using a statistical approach with the fundamental element being the cylinder. The theoretical predictions are compared with experimental results from the literature.
Observation of enhanced backscattering of ultrasound by disordered media
Authors: Loewenherz James H. , Weaver Richard L.
Published: October 2005

Recent experiments demonstrating the enhanced backscatter of 5-MHz ultrasound of a random medium are reported. The effect has been observed in a sample composed of a highly concentrated slurry of 250-μm glass spheres in water, as well as in a sample of polycrystalline brass with randomly oriented grains. The scattered intensity is found to be enhanced within a wide angular cone, thereby indicating a parameter regime that approaches the Ioffe-Regel criterion for the Anderson localization transition. Prospects for the observation of strong ultrasonic Anderson localization are discussed. [Work supported by NSF.]
What reflected SH waves can reveal about anisotropy
Authors: Schoenberg Michael , Costa Jesse
Published: October 2005

Propagation in the plane of mirror symmetry of a monoclinic medium, with displacement normal to the plane, is the most general circumstance in anisotropic media for which pure shear wave propagation can occur at all angles. Because the pure shear mode is uncoupled from the other two modes, its slowness surface in the plane is an ellipse. Consider a stratified half-space of such a monoclinic medium where all the strata have a common mirror plane of symmetry containing the normal to the stratification, say the vertical axis, and let it be overlain by a half-space of an isotropic medium. When plane SH waves are incident on the stratified monoclinic medium from above in the common mirror plane of symmetry, it is shown both analytically and by numerical simulation that there exists an infinite number of different stratified half-spaces, including a unique isotropic stratified half-space, which give the identical reflection coefficient as the original stratified monoclinic half-space for all incidence angles and all frequencies. Thus no set of pure SH experiments, performed with sources and receivers above a stratified half-space in a mirror plane of symmetry of that half-space, can reveal anything of the nature of that half-space's anisotropy.
Coherent elastic waves in pair-correlated media containing randomly positioned spherical inclusions: Comparison of quasicrystalline and Roth's effective medium theories
Authors: Weaver Richard L.
Published: October 2005

While there has been extensive theoretical work on the effective propagation characteristics of scalar waves in random media, the more complicated case of elastic waves has not yet been treated with a method purported to be valid over a wide parameter range in inclusion concentration and wave frequency. Such a treatment is reported here. The on-mass-shell T-matrix method formulation of the Lax quasicrystalline equations as presented by Varadan, Varadan, and Pao is employed, and this formulation is also adapted to the Roth effective medium equations long hailed as superior in the case of density of state calculations for the electronic properties of liquid and amorphous metals. Effective propagation characteristics and modal densities are exhibited and compared. The effect of various choices for the pair correlation function is also investigated. [Work supported by NSF.]
Perception of transient noise at the abrupt onset of a pure tone
Authors: Hartmann William Morris
Published: October 2005

The transient noise caused by the instantaneous onset of a pure tone depends upon the phase of the tone at onset. Listening experiments show that if the frequency of the tone is low, the onset noise is perceived to be least if the onset is in “sine phase.” If the frequency of the tone is high, the onset noise is least if the onset is in “cosine phase,” a choice that maximizes the discontinuity in the waveform. These results find a simple physical explanation in terms of the power spectra of the abrupt tones. Experimental psychometric functions show that the transition from low-frequency perception to high-frequency perception is sharp but rather strongly subject dependent. There is good correlation between transition frequency and high-frequency thresholds as measured by individual audiograms. The transition frequency can also be manipulated experimentally by using electroacoustical transducers that have different frequency responses. [Work supported by the National Institutes of Health.]
Effects of energy and spectral cues on discriminability of bursts of reproducible noise
Authors: Fallon Susan M.
Published: October 2005

Discriminability of bursts of reproducible noise was measured using a same-different psychophysical method. Bursts in a pair were identical on “same” trials. On “different” trials, bursts were identical except for τ ms of independent noise located in the middle of the burst pairs. Three burst durations were examined: 1, 10, and 100 ms. Discriminability increased as the duration of the noise bursts increased. Randomly varying the level of each burst of noise impaired performance at all durations. Further experiments examined whether discrimination was based on spectral or energy cues. On each “different” trial, the order of the samples in the first digitized burst was randomized to generate the second burst in a pair. Results showed that, with the exception of the 1-ms bursts of noise, discrimination was based the spectral shape of the waveform. [Work supported by NIH and AFOSR.]
On the frequency regions governing overshoot
Authors: Bacon Sid P.
Published: October 2005

Detectability of a brief, high-frequency tone can improve as the onset of that tone is delayed relative to the onset of a long-duration, broadband noise masker. This effect, known as overshoot, is, in part, consistent with adaptation at the auditory nerve, although recent psychophysical experiments suggest that other mechanisms also may be involved. To gain more insight into the mechanisms underlying overshoot, several experiments were designed primarily to manipulate the spectral properties of a 200-ms noise precursor (N0 = 20 dB SPL) whose offset preceded the onset of a 400-ms noise masker (N0 = 20 dB SPL) by 1 ms. Depending upon its spectral composition, the precursor could reduce or eliminate overshoot by reducing threshold near masker onset. The assumption here is that the frequency regions of the precursor that reduce overshoot are those that are, in turn, responsible for overshoot. The results indicate that overshoot is governed by a relatively large frequency region that is clearly not confined to the 4.0-kHz signal frequency region, and that frequency regions above the signal are more important for overshoot than are frequency regions below the signal. [Work supported by NIDCD.]