bluuebunny/crossref_metadata_2025 huggingface dataset:
Telephone listening ability in hearing impaired subjects
Authors: Holmes Alice E. , Frank Thomas
Published: October 2005

The purpose of this study was to investigate telephone listening ability of the hearing impaired. Forty-five subjects with bilateral sensorineural hearing losses were divided into three groups (N = 15/group) based on the audiometric configuration of their test ear. The subjects were evaluated using conventional and alternative telephone listening strategies by assessing their discriminations of word lists presented at 86-dB SPL (simulating the output of a standard telephone handset) and at their Most Comfortable Loudness (simulating the output of a telephone amplifier handset). The results indicated that subjects with precipitous drop hearing losses had similar word discriminations across the listening modes which included a TDH-39 earphone, unaided telephone, and their hearing aids acoustically coupled to a telephone handset and at both 86-dB SPL and their Most Comfortable Loudness (MCL). Subjects with gradual slope or flat hearing losses had similar word discriminations across the listening modes at either 86-dB SPL or MCL. However, these subjects with gradual slope and flat hearing losses had better word discriminations in the listening modes when the word lists were presented at MCL compared with 86-dB SPL.
Recognition of selected French vowels and consonants by deaf children: lipreading and coarticulation
Authors: Gentil Michèle
Published: October 2005

The visual recognition of selected French vowels and consonants by deaf children was studied. Thirteen vowels were associated with [l] in the final position and/or placed between [p] and [l]. Sixteen consonants were combined with the extreme vowels on the vocalic triangle [i-a-u] in nonsense words of consonant-vowel type. Two speakers were chosen for this investigation. This research determined confusion matrices. Concerning vowels, the extreme vowels on the vocalic triangle were the most easily recognized visually. Concerning consonants, the results of visual confusions revealed four basic homophenous categories or viseme groups. Three of these categories were based on lip movement [p-b-m] [f-v] [∫-ȝ]. The fourth category consisted of consonants produced inside the mouth [t-d⋅n-s-z-k-g-ɲ-R]. Besides in this study emphasis was placed on coarticulatory effects which make lipreading difficult. The difficulty of recognizing depended on the vocalic environment: most difficulty when followed by [u], intermediately difficult when followed by [i], least difficult when followed by [a]. Inter-speaker variability was evident.
Development changes in perceived stimulus structure in consonant-vowel and vowel stimuli
Authors: Bernstein Lynne E. , Talkin David , Condino Rosemary , Stark Rachel E.
Published: October 2005

It was hypothesized that developmental changes take place in the ability to extract acoustic-phonetic structure from the speech signal. In order to demonstrate developmental change, Garner's two-choice, speeded-classification paradigm was adapted for use with children [Am. Psych. 25, 350–358 (1970)]. In this paradigm, two stimulus dimensions are presented under control, correlated, and orthogonal experimental conditions. Patterns of results across conditions are interpreted as indicative of certain perceptual structures, viz., integral versus separable processing. Stimulus dimensions being tested with adults and eight-year-olds are (1) consonant identity (/bɑ/ versus /dɑ/) and pitch (125 vs 165 Hz F0) [cf Wood, Percept. Psychophys. 15, 501–508 (1974)]; and (2) vowel identity (/ɑ/ versus /i/ and pitch (125 vs 165 Hz F0). It was hypothesized that adults process these two stimulus sets in terms of, respectively, integral and separable structures, but that children process both sets in terms of similarity structure (integrally). Preliminary results confirm predictions for children but suggest that adults process pitch integrally for both stimulus sets. Implications from a larger group of subjects will be discussed in terms of a theory of speech perceptual development. [Work supported by MCH Training Grant Project ♯917 and March of Dimes Grant ♯12-84.]
Emergence of vocants in infant utterances
Authors: Stark R. E. , Talkin D. , Heinz J. M. , Bond J.
Published: October 2005

This preliminary study was designed to determine if the point vowels /ɑ/, /i/ and /u/ emerge in a particular order and within specific age ranges in the productions of infants. Vowel-like sounds (vocants) were selected randomly within certain general criteria from the cooing, expansion, and babbling periods of vocal development in two normal female infants. The formant frequencies of these vocants were estimated, at several points, by inspecting the wideband spectrogram, a 32-ms spectral section, an inverse LPC spectrum, and the frequencies and bandwidths obtained by finding the roots of the denominator polynomial of the LPC model. The order of the LPC model was chosen adaptively to provide frequency resolution consistent with the fundamental frequency of voicing for each frame in order to minimize the interaction between harmonic and resonance locations. The results indicated (1) only modest changes in formant structure from the cooing to the expansion period and (2) a frequent lowering of the first formant and raising of the second in the babbling period, suggesting that an /i/-like vocant does not emerge until the babbling period. Perceptual data for these vocants will be reported. [Work supported by NICHD grant HD 11970.]
Backscattering of sound in shallow water
Authors: Muir T. G. , Thompson L. A. , Shooter J. A. , DeMary T. E.
Published: October 2005

Measurements are reported on the backscattering of sound from an ocean surface and sand bottom overlying the continental shelf near San Diego, California. Data were acquired in the 1–25 kHz band with a directive parametric source at a grazing angle of 10°. The bottom was a ∼5-m layer of coarse grained sand and clay (mean grain size 0.57 mm) that contained many shell fragments and sublayers. The bottom backscattering coefficient was found to be −40 dB re: 1 m2, independent of frequency. Surface backscattering was measured in a shoaling sea containing biologic scatterers. The surface heights were Gaussian distributed (rms wave height = 0.35 m) and had a power spectral density that peaked near 80 MHz and decayed as the fifth power of frequency. The surface backscattering coefficient was found to be −43 dB re: 1 m2, with no frequency dependence. Statistical data in the form of probability distributions (peak acoustic amplitudes in 500 sample ensembles at each frequency) show a Rayleigh distribution for bottom backscattering and high-frequency (10–25 kHz) surface backscattering. Low-frequency surface backscattering (1–5 kHz) departs from the Rayleigh form due to the prevalence of high-amplitude echoes from fish. [Work supported by the Office of Naval Research.]
Multiple scattering with applications to fish-echo processing
Authors: Stanton T. K.
Published: October 2005

A general expression has been derived and evaluated for the backscattered echo energy of an acoustic pulse due to a collection of identical randomly distributed isotropic scatterers. Excess attenuation of the signal due to the extinction cross section of the scatterers as well as second-order scattering have been taken into account. Special attention is focused toward the numerical evaluation of second-order scattering effects. The expression is evaluated for three scattering geometries. It is shown in each geometry that when the absorption cross section of the scatterers is small, second-order scattering at least partially offsets effects due to excess attenuation in the low-to-moderate attenuation region. It is shown that second-order scattering can be important in fish-echo processing where the echo energy is quite often analyzed. The directional characteristics, acoustic frequency of the pulse in relation to the resonance frequency of the swimbladder (if any), and the degree of randomness of the spatial distribution of the fish determine the degree to which second-order scattering plays a role in this area. [Work supported by Office of Naval Research N00014-80-C-0667.]
Con- and de-convolution of the fish scattering PDF and echo PDF for single transducer sonar
Authors: Clay C. S.
Published: October 2005

This paper is a continuation of our studies of acoustic methods to measure fish abundance. The integral equation for the probability density function (PDF) of fish echoes is in Clay and Medwin [Acoustical Oceanography, pp. 476–482 (1977)]. It relates the PDF of the output of the sonar wE(e) to the PDF of the sonar for a target at random location in the beam wI(b), and the PDF of the scattering process at the fish wR(e), where e is echo amplitude and b is the transducer response. Our procedure is to adjust parameters in the two-parameter Rice PDF (the amplitude of the envelope of a sine wave plus noise) until wE(e) matches experimental data. Numerical computations are somewhat tedious. In this paper I use nonlinear transformations b = exp(x) and e = e0 exp(y) to change the integral to a standard form of the convolution integral. One can do a Fourier transformation of the result and then express the convolution as the product of z−− transforms. Computations with the z−− transforms are simple and may become the basis of ways to determine wF(e) directly. [Work supported by Office of Naval Research N00014-80-C-0667.]
Prediction of ocean basin bottom reverberation using a range-dependent reverberation model
Authors: Nicolas K. R. , Franchi E. R. , Tolstoy A.
Published: October 2005

The NRL model for underwater acoustic reverberation incorporates range dependent bathymetry and sound speed profiles. An extended ray theoretic approach for propagation estimates is combined with boundary scattering models to predict, among other measures, reverberation time histories. This paper presents model predictions of ocean basin and margin bottom reverberation for a monostatic source-receiver geometry at a deep water site in the Northeast Pacific. The reverberation time histories are calculated for a number of radials which are selected to include prominent features such as seamounts, fracture zones, islands, and continental slopes. These time histories are azimuthally convolved with a receiver beam pattern to approximate the total basin reverberation response. The strong backscattered returns from both the continental shelf of North America and large fracture zones are of particular interest. Their strength can be 30 to 50 dB above the normal reverberation decay levels, making them a serious interference, even when located in the beam sidelobes. These predictions provide valuable information for source-receiver design, signal selection, and pulse repetition rates to be used in future deep sea experiments. [Work supported by Naval Electronic Systems Command, Code 612.]
Deterministic signal processing of acoustic array outputs
Authors: Stocklin Philip L.
Published: October 2005

Acoustic signal processing currently relies heavily upon probabilistic assumptions concerning data and upon statistical properties of data to develop efficient processing principles. In contrast, deterministic signal processing uses only the actual sensor output amplitudes over a finite time (a data set) to estimate signal waveform and input power for that data set, employing no probabilistic assumptions or statistically based techniques. The major result presented is a demonstration of the improvement in output S/N and in signal waveform and power estimation of deterministic processing versus probabilitistic averaging under conditions of a priori unknown signal waveform and input power, very low input S/N, short processing time and small number of sensors. For example, with input S/N in the region of −15 to −30 dB, order of 3 to 10 sensors, and 5 to 20 time sampling points, improvements in output S/N on the order of 5 to 15 dB are achieved. These results are preceded by a summary of relevant deterministic properties of data sets and derivation of the deterministic algorithm used. Areas of further necessary work and implementation concepts are discussed. [Work supported by NUSC.]
Generation of multiple target data sets
Authors: Bucker Homer P.
Published: October 2005

Standard analysis of acoustic detection systems is based on the sonar equation in terms of array signal gain, array noise gain, signal level, noise level, etc. In most cases, results from the sonar equation are too optimistic because of real ocean effects such as multipath arrivals, array bending, and environment induced fluctuations. In addition, most modern high-resolution techniques, such as cross-sensor processing, cannot be analyzed in terms of the sonar equation because needed quantities such as directivity index cannot be defined. To solve this problem a more basic approach is required. Experiments should be conducted using multiple sources for tests in areas where the acoustic system will operate. Then it is a simple matter to determine the performance of the acoustic system. Because a multisource experiment may be too expensive, a good approximation to the desired experiment can be obtained by combining data collected at different times during a single source experiment. An example will be shown using data collected during a recent sea test.
Doppler properties of pulse bursts
Authors: Cox Henry , Frentz Michael
Published: October 2005

Impulsive sound sources are attractive because they can generate very high energy levels in short pulses with good range resolution properties. However, short pulses have relatively poor Doppler resolution properties and in a number of measurement applications Doppler resolution is important. This paper discusses the design of sequences of pulses called bursts which are processed coherently. New results are presented for the Doppler properties of nonuniformly spaced pulses. The effectiveness of coarse quantization achieved by varying the number of sources per pulse in achieving good side-lobe behavior in the Doppler domain for uniformly spaced pulses is illustrated in several examples. Results for narrow-band pulse bursts (2VTW/C<1) are generalized to broadband pulse where the wideband ambiguity function must be used.
Beampattern synthesis and computation for cylindrical arrays
Authors: Cox Henry , Elmer Robert , Thurston John
Published: October 2005

The synthesis and calculation of beampatterns for cylindrical arrays can be greatly simplified by factoring the problem into a line array of identical hoops under very mild restrictions on array geometry. For example, elements may be directive and hoops may be nonuniformly spaced. This result has particular significance for large arrays of many elements. The design approach consists of using one of several techniques to design a good hoop beampattern combined with a standard technique for designing a good line array beampattern. The technique is illustrated by application to an array of more than 2000 directional elements where other design techniques are impractical and straightforward computation of beampatterns is several orders of magnitude more time consuming. [Work supported by ONR.]
Frequency-dependent refocusing in phase shift beamforming systems
Authors: Lynch James F. , Stokes Bob , Garcia Al
Published: October 2005

It is shown that the focal point of a spatially focused phase shift beamforming system changes as one changes the operating frequency of the system. This change is small for linear arrays, but is quite large for curved arrays. The advantage of a system of this type is that it can be focused on any desired range simply by making a slight change in operating frequency. Theoretical predictions, computer simulation, and experimental results are presented, all showing good agreement. [Work supported by Naval Coastal Systems Center.]
Rules of thumb in designing an array beam pattern in the presence of random shading and phasing errors
Authors: Quazi Azizul H. , Porter David T.
Published: October 2005

Random errors that include shading, phase, and element failures affect the beam pattern of an array, especially the side-lobe level, shape and width of main beam, beam gain, and beam pointing direction. The sources of these random errors may originate in the construction of the array, beamforming operations, and wavefront distortions or in a combination of these factors. We have developed rules of thumb to estimate average side-lobe level, peak side-lobe level, and an error in main beam pointing direction of an array which is subject to random shading, phasing errors, and element failures. We also compared the estimated values calculated by rules of thumb with observed simulated results and found they agree well within 2–3 dB.
Array aperture selection in the presence of inhomogeneous noise
Authors: Buehler Barry G.
Published: October 2005

Sonar slide rules tell us that the directivity index of an antenna array increases 3 dB per length doubling per array dimension. This directive gain is against isotropic noise an it is clear that calculation of the true antenna signal-to-noise gain must consider the actual noise environment. A less obvious consequence of some noise fields is the existence of finite, optimum array dimensions. This paper presents array gain versus aperture computations for a number of representative inhomogeneous noise environments. The important points are the existence, for some environments, of well defined optimum dimensions beyond which gain actually decreases and other environments in which the rate of return (gain increase versus aperture growth) becomes insignificant. [Work supported by ONR.]
Range estimation statistics utilizing Beta density functions
Authors: Perruzzi Joseph J. , Hilliard Edward J. , Pinkos Robert F.
Published: October 2005

Traditional analysis of the triangulation ranging problem utilizes Gaussian distributions to represent angle or time-delay measurement errors. This approach leads to mathematically intractable solutions due to infinite limits on the Gaussian distribution. To resolve these problems, this analysis can be carried out by replacing the Gaussian measurement error statistics by an error model with finite limits (Beta density function). This type of probability density function presents a credible error distribution over a finite range that maps into an estimated range density function from which the range error moments can be generated. This approach is intuitively useful because not only can the validity of the linearized model be evaluated, but this analysis also yields realistic bounds on ranging performance.
Signal conditioning alternatives for hydrophones near a low-impedance surface
Authors: Berkman E. F. , Martin N. C.
Published: October 2005

Low-impedance acoustic baffles are often used to shield hydrophones from unwanted noise. However, when the hydrophones must be located close to such low-impedance surfaces, the resulting loss in signal sensitivity can lead to a decrease in signal-to-noise ratio. Several techniques have been devised to maintain the signal sensitivity of hydrophones in these instances. The theoretical performance characteristics of several such signal conditioning techniques have been examined. Results are presented for the following types: (1) mass layer, (2) hydrophone standoff, (3) a single homogeneous absorptive layer, (4) combinations of mass layers, standoff, and tuned compliant layers, and (5) superdirective hydrophones. In every case the low-frequency signal conditioning performance is determined by the total mass of the signal conditioning treatment.
Acoustic tracking of expendable bathythermographs
Authors: Dammann W. Paul
Published: October 2005

This paper describes an experiment conducted to determine the feasibility of acoustically monitoring the fall rate of expendable bathythermographs (XBT's). Several XBT casts were monitored using a 20-kHz narrow beam echo sounder. The XBT's were dropped at approximately 5-min intervals near the transducer towbody. A magnetostrictive transducer with a conical beam pattern measuring 15° by 25° total beam width was used with a pulse length of 500/μs, an input power of 750 W, and a pulse repetition frequency of 2.08 pulses per s. Receiver output was recorded on analog tape and later digitized at a rate of 5120 samples per s. Results of the experiment prove feasibility of acoustic tracking of XBT's to depths in excess of 200 m. Further studies will be conducted to determine actual fall rates of these devices using acoustic methods. [Work supported by NORDA.]
Hydrophone size versus cover layer thickness in a simple blanket dome
Authors: Martin Nathan C. , Martin Mark M.
Published: October 2005

Previous investigations have examined the filtering action of a fluid-like blanket which separates hydrophones from the flow interface [G. Maidanik and W. T. Reader, J. Acoust. Soc. Am 44. 497–502 (1968)]. When the sound velocity in the blanket is greater than or equal to that of the external fluid, the filtering action of the blanket can be described as a low-pass filter in wavenumber with a cutoff wavenumber that decreases with increasing layer thickness. However, the filtering action of a hydrophone can also be described as a low-pass filter in wavenumber with its cutoff wavenumber varying inversely with hydrophone length. Thus there exists a performance tradeoff for flow noise rejection between layer thickness and hydrophone length. These tradeoffs have been evaluated numerically for simple one-dimensional characterizations of hydrophone and array response, fluid-layer transmission, and TBL wall-pressure wavenumber-frequency spectra. Results are presented to illustrate the nature of the tradeoffs and the relative importance of hydrophone size versus layer thickness.
Spectral broadening in turbulent wakes and jets
Authors: Chotiros Nicholas P. , Culbertson C. Robert
Published: October 2005

A continuous acoustic signal propagating through turbulence is scattered by variations in the flow velocity. The scattered signal is Doppler shifted in a random manner due to the turbulent motion of the scatterers, thus giving rise to spectral broadening. Since the scattered component is usually quite small in underwater acoustics, the method of small perturbations was used. A theoretical estimate of the power spectrum of the scattered signal was derived for a sound wave propagating in a wake, and also for the case of propagation in a jet. Some experimental results were obtained, and compared with the theoretical estimates. [Work supported by Naval Sea Systems Command.]
Ambiguity design for the recognition of spatially extended acoustic scatterers
Authors: Ricker D. W.
Published: October 2005

Acoustic scatterers which have spatial extent in range (R) and azimuth (θ) space may be mapped into an equivalent delay (τ) and frequency (φ) space if they are observed with an active sonar which has a relative cross-range velocity component. It is shown that the mean-square output of a simple correlation receiver may be maximized by seeking signal waveforms whose ambiguity functions most closely match the support of the scatterer image as mapped into (τ,φ) space. Spatially extended scatterers may be recognized and an estimate of their shape and orientation may be obtained by a comparison of receiver responses to waveforms whose ambiguity functions have distinct support in (τ,φ).
Effects of auditorium acoustics on musical performance
Authors: Cox John Charles
Published: October 2005

The echo perception thresholds of listeners and performers are compared. Performer's minimum detectable delay (MDD) and maximum tolerable delay (MTD) are presented. The effects of several auditoria on the performers of music are shown.
Finite amplitude method for the determination of the acoustic nonlinearity parameter B/A
Authors: Cobb Wesley N.
Published: October 2005

The acoustic nonlinearity parameter B/A is determined using a method based on the finite amplitude distortion of a sine wave emitted by a piston. We measure the growth of the second harmonic component of this wave using a piston receiver, which is coaxial with and the same size as the source. In order to determine B/A, the experimental measurements are compared to a theory which incorporates the nonlinearity parameter. The theory developed for this study accounts for the influence of both diffraction and attenuation on the experimental measurements. For this reason, the method is more accurate than previous techniques that employ plane wave theory for a lossless medium. To test the measurement method, we compare the experimental results for B/A in distilled water, ethylene glycol and glycerol to established values. The agreement between these values suggests that the measurement accuracy is plus or minus 4% for common liquids. [Work supported by ONR.]
Pulsed parametric array revisited
Authors: Trivett D. H. , Rogers Peter H.
Published: October 2005

In a previous paper [J. Acoust. Soc. Am. Suppl. 1 71, S30 (A)] we presented the results of an investigation of the pulsed parametric array. We found that while the primary pulse exited the transducer a sum and difference frequency signal was scattered. However, once the interaction region was freely propagating (i.e., both boundaries moving at the sound speed) no further scattered signal was generated. This results in the observed scattered signal having a pulse length equal to the primary pulse length and the appearance of direct radiation from the face of the transducer. We also concluded that the directivity pattern, identical to the cw parametric array, was not due to a volume distribution of virtual sources. On re-examination this conclusion has been found to be incorrect. In this paper we demonstrate that the observed signal at a given farfield point is generated by a volume distribution of virtual sources, as long as contributions to the field at that point are still being received from the stationary boundary. Once signals from the stationary boundary cease there is no further scattering to that point. The extent of the scattering region is dependent upon the primary pulse length and the observation angle and is given by Leff = L/(1 − cos φ), where L is the primary pulse length and φ is the observation angle. This result may have some practical applications.
Masked pure-tone thresholds of the bottlenosed dolphin (Tursiops truncatus) at extended frequencies
Authors: Moore Patrick W. B. , Au Whitlow W. L.
Published: October 2005

Masked underwater pure-tone thresholds were obtained for a bottlenosed dolphin (Tursiops truncatus) using an “up-down staircase” method of stimulus presentation and a “go/no-go” response procedure. Thresholds were determined at two continuous broadband masking noise levels at test frequencies of 30, 60, 90, 100, 110, 120, and 140 kHz. The Critical Ratio (CR), defined as the ratio of signal power to noise spectrum level at masked threshold, was calculated for both noise levels. A psychometric function relating CR's to frequency showed parallel functions with CR's previously reported for the dolphin [Johnson, J. Acoust. Soc. Am. 44, 965–967 (1968)] for frequencies up to 100 kHz. Results for 30, 60, 90, and 100 kHz were 31, 32, 45, and 40 dB, respectively, supporting previous findings for the dolphin. Results beyond 100 kHz, where no other values had previously been obtained, show a sharp increase at 110 kHz (51 dB) followed by a slight decline at 120 kHz (46 dB). Generally, the CR function suggests that the frequency resolving ability of this echolocator is not optimal at these higher frequencies even though peak energies of the outgoing echolocation pulse have been recorded in this higher range.
Natural modes, coupled modes, and transients of vibratory systems
Authors: Skudrzyk Eugen J.
Published: October 2005

Resonances of a point driven vibrator occur when the reflected waves reach the driver in phase with the outgoing wave, antiresonances occur when these two waves are in antiphase. The frequency response for the outgoing wave field determines the geometric mean line through the frequency response curve of the vibrator. Reflections at the boundaries of the vibrator, at ribs and appendages, or at material variations, have no effect on the “mean-line response.” Plate modes are scalar functions of the position. In contrast, shell modes are represented by three-dimensional functions. But they are also described by uncoupled and orthogonal mode functions. Isolated vibrating systems always have orthogonal modes. It may sometimes be convenient to describe a vibrator by mode functions of some simpler vibrator. Every mode then is represented by a sum of mode functions which are coupled (e.g., plate with a mass load). The transients of a complex vibrator can be built up from the transients of its modes. An understanding of the theory of transients is important, for instance, when frequency modulated pulses are generated; increasing the damping may lead to long duration transients; the aim here is to cancel the switching-on transient with the switching-off transient which can only be done if damping is small. The theory of natural modes and transients will be illustrated by practical examples. [This work was sponsored by the Office of Naval Research, Code 474.]
“Figural” and “tonal” aspects of melody discrimination
Authors: Watkins Anthony J.
Published: October 2005

Tone sequences were synthesized using a random process constrained to generate relatively close approximations to the tonal relationships found in melodies which occur in music. In control sequences these constraints are systematically inverted. Other melody variables were fixed (e.g., rhythm) or completely randomized. The perceptual task involved discrimination between “same transposed” and “different transposed” melody pairs. Alterations which changed the “contour” of the melody were compared with those which preserved contour. We find that contour violation and a low “5th span” of the melody intervals aid discrimination, and these operate independently. We also examined diatonic scale, contour and tuning, the latter being equal temperament (100 cents per semi-tone), “micro” (60 cents per semitone) or “macro” (140 cents per semi-tone). A discrimination advantage for melodies based on a diatonic scale is only found for equal temperament tuning, whereas contour violation aids discrimination regardless of tuning. We conclude that contour is a general perceptual property of serial auditory patterns (“figural”), whereas scale and 5th span are specific to sequences heard as music (“tonal”).
Automatic identification of musical chords
Authors: Martin Daniel W.
Published: October 2005

Various electronic and computer systems, which have been developed during the last fifteen years for identifying musical chords automatically, will be reviewed and compared. The purpose in most cases has been to facilitate easy-play, automatic musical accompaniment for musical beginners on electronic musical instruments. However the means used could also help provide low-cost instrumentation for real-time recording of new musical compositions, improvisations, and arrangements.
Multiple-event airplane noise annoyance
Authors: Powell Clemans A.
Published: October 2005

Two experiments were conducted in which subjects in a simulated living room environment judged the annoyance of sessions of airplane noise which contained different noise levels and numbers of flyovers. In the first experiment one, two, or four high noise level flyovers occurred at the beginning, middle, or end of 30-min test sessions each of which contained a total of eight flyovers. In the second experiment, one, four or 16 flyover noises occurred in 15, 30, or 60-min test sessions. The time-of-occurrence of the high noise level flyovers in the sessions did not significantly affect annoyance, but annoyance increased with the number of such flyovers. Annoyance decreased with test session duration but increased with the total number of flyovers in the test sessions. These results support an average energy model better than total energy, annoyance decay, or peak noise level models.
Community noise from air-cushioned landing craft
Authors: Burke Richard E.
Published: October 2005

The effect of noise on neighboring communities resulting from operating the U.S. Navy's proposed Air Cushioned Landing Craft (LCAC) at an initial training site was predicted and evaluated in this preliminary impact assessment. Farfield noise levels of two prototype vehicles, each with a maximum gross weight of 160 tons, were measured under various stationary and moving modes at the Naval Coastal Systems Center in Panama City, FL. These data and appropriate vehicle operating characteristics were entered into NOISEMAP, a noise contour generating computer program which had heretofore only been used in aircraft noise analyses. Four vehicles undertaking a total of 440 annual missions were found to generate a 65-dB DNL contour with an area of only 1/4 square mile. Nonetheless, single event maximums over 65 dB predicted near residential locations were identified as potential areas of adverse impact on the community. [Work supported by the U.S. Navy.]
Profiles of free amino acids in cochlear structures
Authors: Thalmann R. , Comegys T. H. , Thalmann I.
Published: October 2005

Godfrey et al. [J. Histochem. Cytochem. 24, 468–470 (1976)] reported levels of the putative amino acid transmitters aspartate, glutamate, glycine and GABA in cochlear structures. Moreover, Thalmann et al. [J. Acoust. Soc. Am. Suppl. 1 66, S47 (1979)] determined levels of aspartate and glutamate in cell layers of the organ of Corti. Both studies used catalytic fluorometric assays, in which one amino acid is determined at a time. Due to recent advances in high performance liquid chromatography, we have been able to determine profiles of the majority of common amino acids in different tissues of the inner ear. Our findings with respect to the putative amino acid transmitters agree with earlier studies. In the organ of Corti, for example, glutamate is abundant (60–70 mmoles/kg dry weight), as previously shown. The putative neurotransmitters aspartate and glycine, as well as serine, glutamine, alanine, and phosphoethanolamine, are also present at considerable levels (8–30 mmoles/kg dry weight). The other amino acids measured range from 0.3 to 6.0 mmoles/kg dry weight. Data on other cochlear structures will be presented and implications of our findings discussed. [Supported by NIH.]
Response phases of cochlear hair cells at low sound frequencies and their significance
Authors: Zwislocki J. J.
Published: October 2005

Hair-cell excitation depends on the deflection of their stereocilia toward their kynocilium or basal body. Since the stereocilia of the outer hair cells (OHCs) are embedded in the tectorial membrane, their deflection must be controlled by the shear displacement between this membrane and the reticular lamina. Inferring from cochlear microphonics, OHCs are depolarized at low frequencies when the basilar membrane is displaced toward scala vestibuli. Because of the geometry of the organ of Corti, this produces a radial displacement of the reticular lamina toward the limbus. The response phase requires that the tectorial membrane move radially either in phase or in phase opposition with the reticular lamina, but the latter possibility may be excluded on the basis of the mechanics of the system. Since, in general the depolarization of the inner hair cells (IHCs) does not follow the depolarization phase of OHCs, their stereocilia cannot be solidly embedded in the tectorial membrane. The known experimental results are compatible with a viscoelastic coupling. Such a coupling may result from the viscosity and surface tension of the surrounding fluid. In addition, the assumption made earlier that at low frequencies at the location of IHCs the amplitude of the radial tectorial-membrane displacement is nearly the same as of the reticular lamina must be accepted. [Work supported by NIH.]
Letter-to-phoneme rules: A semi-automatic discovery procedure
Authors: Klatt Dennis H. , Shipman David W.
Published: October 2005

A 20 000-word phonemic dictionary has been compiled from several sources and hand-edited so as to contain phonemic representations that are ideal input for the Klattalk synthesis-by-rule program or for an automated phonetic alignment system such as is planned for the Spire phonetics work station. The phonemic representation consists of 40 phonemes, primary and secondary stress symbols, and morpheme-structure symbols to indicate a prefix, suffix, or compound. Computer programs were then written to create letter-to-sound rules of the form A→ [b]/CD EF, i.e., the letter “A” goes to the phoneme [b] in the letter environment consisting of left context CD and right context EF. Each instance of the letter “A” in the dictionary creates a rule that may be new, the same as an old rule, or conflicting with an old rule (a letter goes to a different phoneme in the same context). Conflicting rules (which limit the performance of this procedure) are inevitable unless each whole word is stored in the left/right context. Conflicting rules arise because English spelling does not indicate where morpheme boundaries are located, which syllables are stressed, or the fact that the present work has undergone exceptional historical sound change or has been borrowed from another language. If only the most popular of each set of conflicting rules is used, the computer program can form the rules into a tree for rapid execution. The accuracy and tree size of this approach have been investigated as a function of (a) length of the left context, (b), length of the right context, (c) presence or absence of stress symbols in the orthography (which would require stress prediction from text, a problem that we have not studied as yet), and (d) presence or absence of morpheme boundary structure in the orthography. [Research supported in part by an NIH grant.]
Position constraints on interacting segments in elicited errors favor the word over the stressed syllable
Authors: Shattuck-Hufnagel Stefanie
Published: October 2005

On the hypothesis that the position constraint on segmental interactions in speech errors reflects some aspect of lexical access during speech production planning, we have analyzed corpora of both spontaneous and elicited errors to determine the unit over which the position constraint operates. In spontaneous errors, two interacting segments are likely to occupy similar positions in both their respective words and syllables. Moreover, since most of the interaction errors in the MIT error corpus occur in initial position in words with first-syllable stress, it is difficult to determine which unit (word or stressed syllable) is the most appropriate one over which to state the constraint. Results of error elicitation experiments suggest that the word is the appropriate unit, since, for example, /j/ and /d/ are more likely to interact in the tongue twister “July dog dock Gillette” (where /ǰ/ and /d/ are both word-initial) than they are in the twister “legit dog dock largesse” (where /ǰ/ and /d/ are both prestressed). Implications for models of lexical access during production planning will be discussed.
Towards an expert system for automatic stop consonant identification
Authors: Kopec Gary E. , Bush Marcia A. , Zue Victor W.
Published: October 2005

The adequacies of three sets of acoustic measurements for stop consonant recognition were assessed by examining the performance of a trained human spectrogram reader in a stop identification task. The data sets consisted principally of(1) gray-level pictures (wideband spectrograms); (2) two-dimensional line drawings (LPC spectra and short-time energy contours); and (3) tables of numerical measurements (e.g., frequencies, amplitudes, and bandwidths of spectral peaks). The reader's task was to identify the initial consonant in /CVb/ syllables selected from a corpus containing all combinations of the six stop consonants /b,d,g,p,t,k/ with the six vowels /i,e,ae,a,o,u/ as spoken by three male and three female talkers. The objective of the experiment was to identify an appropriate set of measurements for use in a knowledge-based expert system for automatic stop consonant identification. Results of the experiment will be presented and discussed in the context of this objective.
Computation of vocal tract sensitivity functions: A complex Lagrangian approach
Authors: Charpentier Francis
Published: October 2005

A method for computing vocal tract sensitivity functions is presented in this paper. Fant and Pauli [Speech Commun. Semin., Stockolm, 121–132 (1974)] have derived closed-form expressions relating the kinetic and potential energies in the tract to the sensitivity of formant frequencies with respect to small variations of the area function. In our approach, more general closed-form formulas are derived to compute the sensitivity of any quantity related to the vocal tract transfer function. A Lagrangian formulation allows a straightforward computation of the gradient of acoustic parameters with respect to area function parameters. This technique was used to replace the finite-difference gradient approximation in a procedure identifying articulatory parameters from formant parameters [Charpentier, ICASSP Proc., 1984–1987 (1982)]. The method takes into account the vocal tract losses (heat conduction, viscosity, wall vibrations, lip radiation). It was found effective for determining the articulatory parameters from outputs of the model itself.
Transportation noise: The federal viewpoint
Authors: Wesler John E.
Published: October 2005

Over the past year and one-half, the Federal program for transportation noise control has undergone a number of changes, both in direct and overall policy. Transportation noise control remains an important priority, especially since public reaction to transportation noise severely constrains the efficient growth of a national transportation system needed to meet forecast demands for the movement of people and goods. In some cases, noise is forcing reductions in system capacity, directly in the face of demands for system expansion. This paper will review recent changes in Federal noise control programs and policies, and will attempt a brave look into the future of those activities.
Proposed noise strategy for commercial aircraft
Authors: Linn Richard J.
Published: October 2005

Since recent court rulings have given the local airport proprietor more responsibility in reducing noise in airport communities without being discriminatory to any one airline, and in light of the continued difficulty in achieving federal pre-emption, this paper will present a strategy for achieving meaningful noise reduction without undo legislation by governing bodies.
A Fourier-synthesis system with audible and visible ouput
Authors: Hadden W. James
Published: October 2005

All too often, when teaching Fourier analysis in vibrations and acoustics courses, one encounters a spurious resistance (from students) partially attributable to the absence of illustrations with tangible results. This paper describes the use of a commercially available digital/analog converter and microcomputer to provide demonstrations of Fourier series in which the usual time- and frequency-domain representations are supplemented by an audible output. Several versions of the driving software will be discussed. In one, the user may alter amplitudes of spectral components “on the fly” and soon thereafter hear the change in the acoustic signal while choosing between graphic displays of waveform or Fourier spectrum. In another, the user may enter spectral amplitudes and phases or fill in parameters in formulas for Fourier series—this information is then processed and stored for display as in the previous case. Convergence of Fourier sums to simple periodic functions, effects of relative phase changes, and responses of resonant systems to periodic excitation are among the topics that can be illustrated using this system. A brief demonstration is planned.
Some experiments with broadband ultrasonic pulses
Authors: Hagelberg M. Paul
Published: October 2005

Several broadband ultrasonic pulse experiments will be presented which are suitable for use in the undergraduate laboratories. A brief description will be given of a set which requires only time domain measurements and for which the analysis is conceptually easy. Attention will be concentrated on a group which requires frequency domain analysis and for which digital data acquisition techniques are particularly suited. While these may be performed with relatively unsophisticated equipment they are readily adapted for microcomputer control and analysis.
Computer based laboratory instruction in acoustics and noise control
Authors: Holger David K.
Published: October 2005

A computer based instructional laboratory in acoustics and noise control is described. The semester length laboratory, which is currently being developed, is based upon analog to digital (A to D) conversion of transducer signals followed by micro- and/or main frame-computer analysis of the digitized signals. The proposed laboratory is compared with traditional laboratories in acoustics which normally require a number of stand-alone instruments. Extensions of the microcomputer based laboratory using the “micro” as a “smart” terminal in a main frame time sharing system are also discussed. The progression of experiments in the laboratory evolves through these phases. First phase experiments are designed to familiarize the student with transducers, A to D conversion and simple, stand alone, micro-computer analysis. Second phase experiments involve multichannel data acquisition and more extensive micro-computer analysis. Third phase experiments involve main frame software and data storage using the micro-computer as a “smart” terminal in a time sharing system. A phase two experiment pertaining to acoustic parameters describing the impulsive response of a room will be described and discussed in detail.
Impulse noise measurement: The physiological basis
Authors: Price G. Richard
Published: October 2005

In designing a system to measure noise there are many interacting considerations, among them the electrical/acoustical state-of-the-art, regulatory requirements, administrative concerns, and economic issues. In contrast to these fluctuant elements, the structure of the auditory system is for all practical purposes fixed and establishes limits to many technical issues. Although theoretical work in the noise effects area is in its infancy, this paper will discuss the developing information with respect to the issues inherent in the measurement of noise, such as high- and low-frequency cutoffs, temporal factors, “trading ratios,” and limits on the dynamic range.
Industrial impact noise: Description and definition
Authors: Erdreich John , Doyle Thomas , Spaeth Steve
Published: October 2005

In order to define the practical requirements of instrumentation for the measurement of industrial impact noise and to determine common parameters of industrial impact noise which should be considered in the study of the biological effects of such noise, survey measurements were made in industries which produce impulse noise. The measurements were made to include a bandwidth from 10 Hz to 40 kHz and a dynamic range of either 42 or 72 dB, for analog and direct digital recording, respectively. From the impact noise samples, we calculated true rms sound pressure level for the burst, peak-level, crest factor based on burst duration, and 1-s equivalent continuous level. Dose based on 3-dB/doubling and 5-dB/doubling A-weighted slow response and A-weighted fast response was calculated also. We will discuss the descriptors of impuse noise and also present preliminary data which address the variability of impulse noise signatures from the same process and the same machine.