bluuebunny/crossref_metadata_2025 huggingface dataset:
Impaired- and normal-hearing listeners’ recognition of [b, d, g, v, z, edh] in [ΛCΛ] isolated versus in sentence contexts.
Authors: Kozma-Spytek L. , Revoile S. , Holden-Pitt L. , Pickett J.
Published: October 2005

Recognition of consonants in connected speech has received limited study for severely hearing-impaired listeners. Such listeners were tested (n=7) for identification of [b, d, g, v, z, edh] in [ΛCΛ] tokens, each embedded in a spoken carrier sentence (male talker) and extracted from the sentences. The severely hearing-impaired listeners’ recognition was poorer for the consonants in the [ΛCΛ]’s when in the sentence context than when extracted. In contrast, neither a group of moderately impaired (n=11) nor of normal-hearing listeners (n=6) showed a significant effect of context on consonant recognition. Performance by the severely hearing-impaired listeners was not improved by insertion of an artificial pause preceding the word containing the test consonant in the sentence.
Anchoring: A potential mechanism for processing relative amplitude of frication.
Authors: Hedrick Mark S. , Ohde Ralph N.
Published: October 2005

The results from earlier research suggested that listeners use vowel amplitude as a reference or anchor from which to determine the amplitude level of a consonant for identification of place of articulation [Ohde and Stevens, J. Acoust. Soc. Am. 74, 706–714 (1983)]. The anchoring hypothesis was tested for a /s ɑ /–/sh ɑ / synthetic fricative contrast by inserting temporal gaps of silence between the fricative and vowel for several fricative durations. The findings revealed that temporal gaps affected perception of place of articulation, supporting the anchoring hypothesis. An interaction of frication duration with temporal gap duration suggested two stages of perceptual integration, with selection of the integration stage dependent on fricative duration. The results indicate that the relative amplitude property is a component of spectral prominence, which is comprised of a frication spectral peak and a fricative/vowel peak comparison [Stevens, J. Phon. 17, 3–45 (1989)]. [Supported by NIH, DC00464.]
Scattering from a fluid–solid interface with periodic surface roughness: An exact solution.
Authors: Bishop Garner C. , Smith Judy
Published: October 2005

A solution, based on the technique developed by Holford [J. Acoust. Soc. Am. 70, 1116–1128 (1981)], is obtained for the far-field scatter of a plane wave that is propagated in a fluid medium and is scattered from an infinite fluid–solid interface with periodic surface roughness. The Helmholtz–Kirchhoff integral equations are used to represent the scattered pressure field in the fluid and the displacement field in the solid. The boundary conditions are applied and the surface fields are represented by an infinite series of Floquet plane waves. To determine the complex amplitudes of the spectrum of Floquet plane waves for the surface fields, the Helmholtz–Kirchhoff integral equations are evaluated on the surface. This process produces a system of three coupled Fredholm integral equations of the second kind that are solved by the method of moments. This solution may be used to calculate scattering from any surface that is continuous and at least twice differentiable or any surface that is represented by a truncated Fourier series. An exact solution is obtained for scattering from sinusoidal surface roughness.
Iterative direct and inverse scattering in multilayered elastic background media.
Authors: Aymé-Bellegarda Eveline J. , Habashy Tarek M.
Published: October 2005

This work is concerned with the modeling of elastic wave scattering by solid- or fluid-filled inclusions buried in multilayered elastic background media. A general framework, encompassing both direct and inverse problems, is developed based on vector integral equations for elastic scattering. The entire medium, comprising inclusion plus background, is probed by a time-harmonic body force and the field scattered by the inclusion is computed (direct problem) or observed (inverse problem) at some known receiver locations. The scattering framework developed is independent of the problem geometry and the transmitter–receiver characteristics. A Born approximation for inhomogeneous background is invoked (i) in the direct problem to obtain a closed-form expression for the scattered field, and (ii) in the inverse problem to linearize for the inclusion multiparameters. To circumvent the limitations of the Born approximation, both direct and inverse scattering are tackled iteratively. In the case of direct scattering, a recursive algorithm that recovers the Born series is developed. In the case of inverse scattering, a two-step iterative inversion procedure alternating direct and inverse modeling to improve on the resolution and accuracy of the reconstruction is proposed. The aim is to calculate a better estimate of the inclusion given a better estimate of the actual field inside the inclusion. Preliminary simulations results are presented.
Advances in digital holographic reconstruction of an odd-shaped source.
Authors: Huang Yanmin , Maynard J. D.
Published: October 2005

In order to reconstruct surface vibrations on a source with arbitrary shape from its field measurements, a new computational method, the equivalent surface source (ESS) method, is developed. The ESS method, which is based on the Helmholtz integral equation (HIE), assumed there is an equivalent surface source that is inside the real source surface but may be a different shape and treats this equivalent source as if it is the real source for field outside the actual source. The ESS produces the same field as the actual source for field outside and on the surface of the actual source. The ESS method can be used in both radiation and reconstruction problems. This method is implemented by using the finite-element method with the second-order interpolation function. The singular value decomposition (SVD) method is employed to do the matrix inversion. The SVD method coupled with the finite-element method implementation of HIE enabled the extension of the near-field acoustical holography technique from two- to three-dimensional sources. Test results are given.
Mode localization in a fluid-loaded plate.
Authors: Ginsberg Jerry H. , Pham Hoang
Published: October 2005

The effects of fluid loading on the response of nearly periodic structures is a topic of current concern. Several dynamic features of such systems are closely related to the vibrational properties of a general self-adjoint system in which two eigenvalues are close. This often leads to mode localization, in which eigensolutions are confined to subregions. A two-span beam with a torsional spring at its intermediate pin support displays such behavior when that support is slightly off-center. This paper examines the degree to which such phenomena are important for fluid-loaded structures by considering a two-dimensional plate in an infinite baffle with a similar type of intermediate line support. In addition to the fluid loading, the plate is subjected to point forces. Acoustic field pressure and the plate displacement are determined by a coupled formulation of the surface variational principle and Hamilton’s principle based on the assumed modes technique. Results for surface displacements are examined for evidence of mode localization in the presence of fluid loading. Radiated power as a function of frequency is also examined for the significance of the center support offset. [Work supported by ONR.]
Review of high-power transducers.
Authors: Tito Frank A.
Published: October 2005

This paper will present a review of underwater high-power transducers. Various transduction types will be presented with applications for each type discussed. Specific design information and trade-off considerations will be evaluated. Active material for various applications and transduction types will be evaluated and applicable parameters discussed. Various new active materials will be presented and compared with current well-known existing materials. Various modeling approaches and computer analysis of various types will be presented.
Mutual coupling effects between elements in a spark source array.
Authors: Cook Jeffrey A. , Rogers Robert L. , Gleeson Austin
Published: October 2005

Experimental and theoretical investigation of an array of spark sources has yielded information about the mutual effects of sources in close proximity, and about the acoustic spectra and radiation pattern. High-speed photographs have been taken of several shots to supplement the calculations and acoustic data. Preliminary results indicate that the enhancement of particular output characteristics is feasible, and that this enhancement hinges on the interaction of multiple radiating bubbles in a particular way. Theoretical calculations and experimental data are compared.
Perceptual and performed accents in musical sequences.
Authors: Drake Carolyn
Published: October 2005

Some events in musical sequences are perceived as accented due to changes in physical characteristics of events. These perceptual accents are also often performed with additional emphasis (played louder and/or longer). These performance variations could result from a musician’s desire to add musical expression to important events (accents are played louder and/or longer because it sounds more musical), or they may result from psychophysical constraints of the human auditory system (accents are played louder and/or longer because they are heard softer and/or shorter). Two psychophysical experiments are presented that distinguish between these two explanations. Subjects heard short musical sequences containing rhythmic grouping and melodic accents. Their task was to detect and identify the position of one event, which was either slightly longer (or shorter) or louder (or softer) than other events in the sequence. The detection of intensity variations was unaffected by the presence of accents in the sequences, but the detection of timing variations varied in relation to both rhythmic grouping and melodic accents. These results provide support for the independent processing of timing and other physical characteristics of acoustic events, and they suggest that not all systematic variations observed in music performance are perceived by performer and listener even though they may be above threshold levels described for single auditory events.
Vocal intensity in speakers and singers.
Authors: Titze Ingo R. , Sundberg Johan
Published: October 2005

Vocal intensity is studied as a function of fundamental frequency and lung pressure. A combination analytical–empirical model is used to predict sound-pressure levels from five professional tensors and twenty-five normal control subjects. The empirical model describes features of the glottal flow waveform (peak flow, peak flow derivative, open quotient, and speed quotient) in terms of lung pressure and phonationthresholdpressure, a key variable that incorporates the F0 dependence of many of the features of the glottal flow. The analytical model describes the contributions to SPL by the vocal tract. Results show that SPL increases with F0 at a rate of 8–9 dB per octave provided that lung pressure is raised proportional to phonation threshold pressure. SPL also increases at a rate of 8–9 dB per doubling of excesspressureoverthreshold, a new quantity that appears to assume prominence in vocal intensity calculations. For the same excess pressure over threshold, the professional tensors studied produced 10–12 dB greater intensity than the male nonsingers, primarily because their peak airflow was much higher for the same pressures.
Increased backscattering from rough surface at the Rayleigh angle.
Authors: Pecorari Claudio , Nagy Peter B. , Adler Laszlo
Published: October 2005

The increased backscattering from liquid–solid interface at the Rayleigh angle is a well-known indication of leaky Rayleigh wave generation. A comprehensive experimental study [de Billy etal., J. Acoust. Soc. Am. 72, 1018 (1982)] showed much stronger backscattering signals than predicted later by theoretical work based on back reflection. It was later demonstrated [P. B. Nagy and L. Adler, J. Acoust. Soc. Am. 85, 1355 (1989)] that back reflection is negligible in the backward signal, which is simply backscattering from material in homogeneities greatly enhanced by surface wave generation. In this work, recent theoretical developments using second-order perturbation theory was adapted to study backscattering from liquid–solid rough interfaces. It was shown that at the Rayleigh angle there is an increased backscattering independent from the rms roughness. Theoretical results were experimentally verified. [Work supported by the Office of Naval Research contract N00014-88-K0452.]
Voice analysis of spasmodic dysphonia patients.
Authors: Fisher Kimberly V. , Scherer Ronald C. , Owen Ann S.
Published: October 2005

Adductor spasmodic dysphonia (SD) is a focal laryngeal dystonia resulting in closure spasms of the larynx during speech. Botulinum toxin (occulium A) injection into the thyroarytenoid muscle is currently used to temporarily ameliorate gross voice symptoms of this condition. Studies of subtle voice change are needed to describe and track the toxin effects. Aerodynamic, kinematic, and acoustic recordings of SD patients were obtained over a three-month period. In one patient, glottal adduction immediately decreased after toxin injection and subglottal pressure steadily decreased over the three-month period, suggesting an adaptation to reduced respiratory effort. The course of peak glottal airflow, maximum flow derivative, and glottal adduction show variations reflecting physiological and behavioral adjustments. Data for other subjects will be discussed. The significance of the study pertains not only to toxin effects on communication, toxin dose administration, and therapy concerns, but also to laryngeal function response to neurophysiological interventions, codependence of phonation parameters, and limits of phonatory behavior. [Work supported by NIDCD.]
Heard Island feasibility experiment: Analysis of the Pacific path data obtained with a horizontal line array.
Authors: Heard G. J. , Chapman N. R.
Published: October 2005

This paper presents preliminary results from the analysis of the Pacific path data that were obtained by the Defence Research Establishment Pacific during the Heard Island feasibility experiment. The data were recorded using a 64-element, 610-m horizontal line array that was towed about 240 km off Southern California, a distance of nearly 18 Mm from the Heard Island source. The ship’s track was designed to cross the ray bundle of a set of predicted propagation paths with source angles of 133°–136°. A total of 20 1-h periods were recorded from 29–31 January, and the signal intensity and the horizontal directivity were measured for all types of signals transmitted. Initial analysis on board the ship indicated that the beam signal-to-noise ratio varied from 10 dB ( for one source) to 25 dB (for five sources), corresponding to a maximum level of −10 dB for a single hydrophone in a 1-Hz band. The signal arrival angle in the beam of greatest intensity was 215°. Further analysis of the data has been carried out to study the horizontal multipath structure, the variation of intensity with array location, and the time spreading of the signal. These results are presented in this paper.
Horizontal multipath in transoceanic acoustic propagation: Model results and data.
Authors: McDonald B. E. , Kuperman W. A. , Heaney K.
Published: October 2005

Adiabatic normal mode predictions are presented for selected acoustic transmissions in the Heard Island project. Results from our model for the 1960 Perth-to-Bermuda transmission experiment [Shockley etal., J. Acoust. Soc. Am. 71, 51 (1982)] portray the arrival structure as strongly affected by horizontal multipath and secondarily by modal dispersion along individual paths. Different horizontal eigenray bundles, derived from the geographical distribution of individual modal phase speeds computed from archival oceanographic data, were found to sample different ocean volumes; within a single eigenray bundle, the volume swept out depended on frequency and mode number. The possibility of horizontal multipath could impact signal processing for Heard Island. Model results for the Heard Island feasibility test (January 1991) have shown agreement with preliminary data for transit times and for launch angle as inferred from known source motion. General features of our results are that transmissions from Heard to either California or Ascension Island are less dispersive than was Perth–Bermuda (1960), partly due to reduced contact with Antarctic waters. Signal paths to various receiver stations are discussed in the light of available data. [Work supported by ONR.] a)Permanent address: Planning Systems Inc., McLean, VA 22101.
Dispersion of acoustic modes over very long transmission paths.
Authors: Munk Walter
Published: October 2005

Dependence of group velocity on mode number and frequency cannot be ignored over transmission paths of 10 000 km. Some simple rules are given for estimating dispersion in a standard midlatitude sound channel, in an Arctic duct, and as a function of variable depth.
Digital implementation of an audio frequency Fourier analyzer (AFFA).
Authors: Koenig David , Dixon Warren
Published: October 2005

Plug-in digital signal processing (DSP) boards provide personal computers with a level of processing power that was once restricted to all but the most powerful mainframe computers. These boards transform the PC into a parallel processing environment, delivering 33 million floating point operations per second (MFLOPS) or more to numerically intensive tasks. Couple this with high-quality A/D converters and you have a powerful DSP workstation for applications ranging from digital audio to underwater acoustics to vibration and modal analysis. This paper details a real-time DSP application using a Texas Instruments TMS320C30-based plug-in board to process audio frequency (DC to 20 KHz) information. A real-time audio frequency Fourier analyzer (AFFA) was implemented using a Macintosh II computer equipped with a digital signal processing board and an audio frequency input/output board. AFFA provides the Macintosh with enough power to replace several stand-alone signal analyzers. To test the power and functionality of AFFA, three stand-alone audio frequency analyzers were used for comparison. AFFA met or exceeded these signal analyzers in many key specifications, including broadband accuracy, dynamic range, amplitude flatness, and cost. Several real-time applications of AFFA are also detailed.
Performance of an electrohydraulic shock wave lithotripter measured with thin film PVDF transducers.
Authors: Sturtevant Bradford , Hartenbaum Bruce
Published: October 2005

Measurements of the operation and performance of a research lithotripter have been carried out using a new thin film polyvinylidene fluoride (PVDF) pressure transducer. The research lithotripter is modeled after a commercial electrohydraulic machine. Experimental techniques to match nanosecond-response transducers to digital sampling oscilloscopes are described. Shock rise times at F2 of about 45 ns are measured. Techniques have been developed for characterizing lithotripter focal properties by a combination of remote measurement and calculation to minimize damage to transducers. The results of numerical calculations of focusing shock waves in water by the method of shock dynamics are presented. The measurements show that the strength of shock waves in electrohydraulic lithotripters varies from shot to shot by as much as 40%, even though the discharge current of the shock generator is repeatable. Thus every shot in experiments carried out with these machines should be monitored with a reference transducer. It is suggested that the variability is due to the large sensitivity of ellipsoid optics to small changes of the symmetry of the spark discharge. The effects of tissue phantoms on focusing shock waves are reported.
Mid-frequency enhancement of the backscattering of tone bursts from thin elastic shells: Fourier synthesis and ray model.
Authors: Zhang L. , Sun N. H. , Marston P. L.
Published: October 2005

A broad enhancement of the form function for these shells is known to be present in the mid-frequency range. A ray model of the form function in this region was previously demonstrated for cylinders and spheres [N. H. Sun and P. L. Marston, J. Acoust. Soc. Am. 89, 1949 (1991)] and the enhancement was found to be associated with contributions to the scattering from the subsonic surface guided wave frequently designated as A0−. The consequences of this enhancement were investigated for spherical shells by synthesizing the response to tone bursts from an evaluation of the exact partial wave series over the sampled spectrum of the burst [L. Zhang, M. S. dissertation, Washington State University, 1991]. This was compared with a ray approximation for the earliest contribution for this wave, which circumnavigates only the backside of the shell. The amplitude from the ray model is ‖f0,l‖ ≊ 8πβl(c/cl)exp(−πβl), where the surface wave parameters cl and βl are found from the exact elastic equation for the roots of Dν(ka)=0. Both the Fourier synthesis and ray models show a broad enhancement which for a 2.5% stainless steel shell peaks at ka≊46 where ‖f0,l‖ ≊ 3.1 and the exact results differ by only 3%. Neglecting dispersion the peak occurs where βl=π−1. [Work supported by ONR.]
Effects of noise on the discriminability of distinctive features in normal and whisper speech.
Authors: Voiers William D.
Published: October 2005

The Diagnostic Rhyme Test (DRT) and Diagnostic Alliteration Test (DALT), a final-consonant counterpart of the DRT, were used to study the effects of band-limited Gaussian noise on normal and whispered speech. Although possibly an artifact of the method of measuring speech-to-noise ratio, whisper speech appeared generally more robust than normal speech over much of the range of conditions involved. In normal speech, and in both final and initial consonants, the features, voicing and nasality are highly resistant to degradation; sibilation and compactness, moderately resistant; graveness and sustention, least resistant. In the case of whispered speech, sibilation and compactness appeared most resistant to noise in both initial and final consonants. In initial consonants, voicing was most affected; nasality, sustention, and compactness, moderately affected. In final consonants, graveness was the feature most affected by noise. These results attest to the robustness of whisper speech and suggest that in some circumstances it could provide a more effective means of communication than normal speech.
Extraction of pitch using the autocorrelation of scalp-recorded responses.
Authors: Dolphin William F. , Mountain David C.
Published: October 2005

Most natural sounds consist of a large number of harmonics. Thus, in any real situation sounds impinging on a listener are spectrally and temporally complex. The initial processing of acoustic stimuli is to split the signal into its component frequencies via spectral analysis performed by the cochlea, yet, when one listens to a multitone complex a single tone having a pitch corresponding to the fundamental frequency (F0) is perceived, even if F0 is not present. This phenomena has been attributed to a labeling process by which the auditory system collects certain neural inputs, using such features as harmonic relatedness and temporal patterns, and tags them with a pitch identifier. The physiological correlates of pitch perception were examined using complex harmonic and inharmonic tone stimuli and recording neurophysiological responses from the scalp of gerbils. It is hypothesized that, based on the model of Licklider [Experentia 7, 128–133 (1951)], the auditory system uses across-frequency-channel processing and performs an autocorrelation function (ACF) on the summed output of each neural channel. The pitch of the stimulus was assigned as that frequency corresponding to the first largest peak in the ACF of the evoked scalp potentials. Using this simple rule our results have compared very favorably to data from a wide variety of psychophysical pitch studies. [The research was supported by an NIH Individual NRSA postdoctoral award to W.F.D.]
Classifying dolphin echoes using an integrator gateway artificial neural network.
Authors: Moore P. W. B. , Roitblat H. L. , Nachtigall P. E. , Penner R. E.
Published: October 2005

Echoes from targets used in an echolocation matching-to-sample task were captured, digitized, processed, and fed to a new artificial neural network (ANN) which classified the target by combining information from successive echoes. The ANN emulates a natural sampling strategy used by dolphins to obtain multiple representations from the same target. The data processed by the ANN were 30-dimensional vectors of spectral information obtained from the dolphin during echolocation. Echoes were recorded from a hydrophone placed in the water between the dolphin and the targets. Signals were recorded, digitized, and fast-Fourier transformed into 30 frequency bins, each 3.84 kHz wide ranging from 31.25 to 146.5 kHz. The network was trained to classify these vectors according to the target (sphere, cone, or tube) that produced the echoes. One network output corresponded to each stimulus class. The resulting activations of each of these output classes were taken to be an estimate of the likelihood that the echo resulted from the particular stimulus type. The network was trained on a small subset (4%) of the total set of available echoes (1335) and produced about 90% correct classifications. In contrast, a standard backpropagation network (not using successive echoes) performing the same task performed with only about 63% accuracy.
The scattering of low-frequency sound from random bubble distributions.
Authors: Carey W. , Fitzgerald J.
Published: October 2005

The scattering of sound from a random distribution of bubbles is discussed for the cases for which the excitation frequency is low compared to the resonance frequencies of the individual bubbles. The cases of incoherent, coherent, and compliant backscatter are characterized by the bubble size distribution, number of bubbles per wavelength, and volume fraction. It is shown that the compact cloud of bubbles acts as a single bubble with mixture density and sonic speed. Furthermore it is shown that this case of the compact cloud can be treated as a collection of poles thus easing the treatment of sound scattering from bubble distributions near pressure release and rigid boundaries. An important factor in the scattering of sound from bubble distributions is the damping at the lower frequencies. Estimates of the coherent and compliant (damped resonant) backscatter from spherical and cylindrical clouds are presented and the two regimes differentiated with respect to volume fraction. [Work supported by ONR and NUSC IR.] a)Also at Kildare Corp.
The effect of the background bubble population on scattering from the sea surface.
Authors: Norton G. , Keiffer R. , Dubberley J. , Novarini J. C.
Published: October 2005

A known effect of the unorganized background bubble population that exists near the sea surface is on the sound-speed profile (SSP) and hence on the illumination of the air/sea interface. In this paper a new algorithm that makes uses of the latest and best estimates of the bubble population (CBUBBLY) to generate a perturbed SSP is employed to examine the effect of the bubbles on scattering from the air/sea interface. The numerical example considered will simulate convergence zone backscatter experiment, the sea surfaces considered are two-dimensional and are due to nonstationary, nonuniform forcing conditions. [Work supported by ONT.]
Optimizing the performance of the adaptive beamformer noise reduction scheme for hearing aids in everyday listening situations.
Authors: Kompis Martin , Dillier Norbert
Published: October 2005

The adaptive beamformer approach for noise reduction in hearing aids is known to be very successful in anechoic chambers and other experimental situations, but the performance decreases rapidly in realistic environments. In order to optimize the adaptive beamformer for everyday use, the algorithm was analyzed by means of a novel theoretical approach, which estimates the performance as a function of several acoustic and design parameters. According to our findings, adaptive filters of at least 50 ms in length have to be used in order to achieve a gain in signal-to-noise ratio of approximately 4 to 6 dB in ordinary office-sized rooms with typical reverberation times of around 0.4 s. Furthermore, a reliable target signal detection algorithm is necessary in order to prevent the adaptation of the filter in the presence of strong target signals. An optimized version of the adaptive beamformer has been implemented in real time on a TMS320C30 digital signal processing system. Preliminary results of intelligibility tests will be presented. [Work supported by the Swiss National Research Foundation.]
Speech recognition with a new DSP-multiband loudness correction hearing aid algorithm.
Authors: Fröhlich Thomas , Dillier Norbert
Published: October 2005

A new algorithm for a multiband loudness correction hearing aid was implemented in real-time on a digital signal processor (DSP) (TMS320C25, Texas Instruments). Selective amplification of the short-time amplitude spectrum is used to compensate individual hearing losses. The fitting procedure is based on the estimation of loudness growth functions for narrow band signals via a scaling procedure. Frequency and intensity dependent gain factors are calculated for every hearing impaired subject to yield similar loudness growth functions as average normal hearing listeners. To prevent loudness summation across frequency bands while processing broadband signals such as speech, a simplified psychoacoustic loudness (SPL) perception model was incorporated to estimate the loudness of the incoming signal and correct the gain tables accordingly. Speech tests in quiet (at 70 and 60 dB SPL) and noise [at 0 and −5 dB signal-to-noise ratio (SNR)] were performed with 11 hearing aid users. Significant improvements of recognition scores were obtained with the new DSP compared to the subjects own conventional hearing aids [mean improvements in consonant tests: 12% (70 dB), 29% (60 dB), 14% (SNR=0 dB), 12% (SNR=−5 dB); vowel tests: 6% (70 dB), 13% (60 dB), 6% (SNR=0 dB), 11% (SNR=−5 dB); all t-tests significant at p<0.025]. [Work supported by the Swiss National Research Foundation.]
Levels of quiet in a neonatal intensive care unit.
Authors: Gray Lincoln , Philbin M. Kathleen
Published: October 2005

Sound levels were measured in a level III neonatal intensive care unit (NICU). Data include tabulations of peak, average, and minimum levels of sound every hour for 5 days. Probit analysis was adapted to depict trends documented by dosimeter measures. Noise never reached levels considered to cause ear damage. An average 67 dB Leq is within acceptable limits [Environmental Protection Agency, Report No. 550/9-74-004 (1974)]. The noise may be harmful, nevertheless, because there are no periods of quiet. Levels exceeded 90% of the time, L90s, were always above 63 dB, exceeding the standard for a ‘‘very noisy urban residential area’’ (EPA, 1974). There is no diurnal rhythm in the NICU noise, meaning no periods of respite. Levels of quiet may be an important and previously ignored aspect of the acoustic environment around premature infants. Recent research with neonatal animals show that exposure to NICU noise detrimentally affects attentional and perceptual development.
Temporary threshold shift by low and medium intensity exposures.
Authors: Young I. M. , Lowry L. D.
Published: October 2005

Temporary threshold shift was measured by Békésy audiometry in subjects with normal hearing and sensorineural hearing impairment. The 20-dB SL exposure of 1-min duration at 4000 Hz resulted in similar shift for the same exposure frequency as 80-dB SL exposure in a normal subject. This finding is in good agreement with results of Lierle and Reger, Hirsh, and Bilger. When the exposure level was higher than 90 dB SPL, shift at a frequency 1/2 octave above the exposure frequency exceeded that produced at the exposure frequency. The 20-dB SL (77-dB SPL) and 30-dB SL (87-dB SPL) exposures of 1 and 3 min durations at 4000 Hz for a subject with sensorineural loss showed similar shift as in a normal subject. It is assumed that the area of basilar membrane affected by adaptation remainder is smaller after an exposure to 20 dB SL than to 80 dB SL. Since adaptation is maximal for a given neurone, those neurones concerned with threshold intensities are adapted equally for both 20 and 80 dB. During adaptation remainder, however, the ear is not responsive to a threshold sound. Stimulation of neurones adjacent to those concerned with threshold is required.
Effects of first formant onset properties on voicing judgments result from processes not specific to humans
Authors: Kluender Keith R.
Published: October 2005

Both first formant (F1) transition duration and F1 onset frequency have been proposed to be perceptually significant in categorization of voiced and voiceless syllable-initial stops. Transition duration per se may not, however, explain the fact that, for longer transitions, longer F1 cutback is required in order to perceive a stop as voiceless. Longer transitions result in lower F1 onsets at any duration of cutback greater than zero, and it is possible that the major effect of F1 is determined by its frequency at onset. In this study, F1-transition duration, onset frequency, and slope were varied across four types of F1 transition in which one of the three variables (onset frequency, duration, slope) was held constant while the other two were allowed to vary. Each of the four F1 types was used in syllables with higher formants appropriate for labial, alveolar, and velar places of articulation. By far, the best predictor of identification of these stimuli by human listeners was F1 onset frequency. F1 duration, F1 slope, and place of articulation had little or no effect on labeling boundaries. In a second experiment using Japanese quail (Coturnix coturnix japonica), birds were trained to respond differentially to voiced versus voiceless stops. The differential effects of F1 onset frequency on the ‘‘labeling’’ behavior of these birds was strikingly similar to that of humans listening to the same stimuli. These results are taken to provide strong evidence that F1 onset frequency is the primary determinant of shifts in voicing boundaries across place of articulation, and that general mechanisms not unique to humans appear adequate to account for the effects of F1 onset frequency on perception of voicing for syllable-initial stops.
A study of barrier attenuation by using acoustic impulses
Authors: Papadopoulos A. I. , Don C. G.
Published: October 2005

Acoustic impulses with a duration of about 2 ms have been used to measure the attenuation of full-sized barriers, including thin edges, wedges, and wide barriers. A technique involving two microphones was used which permits the direct comparison of the diffracted and undiffracted pulse waveforms. This allows the attenuation to be determined with an accuracy of better than 1 dB over the range 700 Hz–12 kHz from a single measurement set. The experimental results are compared with predictions from a variety of approximate and exact diffraction theories. As impulses involve both phase and magnitude information, they form a rigorous test of the theories. One advantage of the experimental technique is that it permits effects due to nonideal features, such as supporting struts, to be rapidly observed.
Acoustic sources in the low Mach number turbulent boundary layer
Authors: Hardin Jay C.
Published: October 2005

The sources of sound production in a low Mach number turbulent boundary layer are examined. The sources are shown to be quadrupole in nature and to result from supersonically convecting wave-number components of the fluctuating Reynolds’ normal stresses. The primary Tollmien–Schlichting instability of the boundary layer is found to radiate no sound. Analysis of various vortical phenomena suggests that the primary source is the process of formation of horseshoe vortices, with viscous sublayer bursts a possible secondary source.
Echographic diffraction filters and the diffraction function for random media through an instantaneous time-frequency approach
Authors: Cardoso Jean-François , Fink Mathias
Published: October 2005

In the estimation of ultrasound attenuation in random scattering media, near-field and near-focus diffraction effects introduce bias because of their depth-dependent filtering behavior. A new approach is proposed that is aimed at the prediction and correction of these effects. It is based on the concept of instantaneous time-frequency representation of random signals, and it leads to a rigorous definition of the echographic diffraction filter. A thorough study is provided of this time-varying filter for axisymmetric planar and focused transducers. In this last case the diffraction filter depends on five parameters (time, frequency, transducer radius, focal length, and sound velocity). Through the study of invariance properties of the diffraction impulse response, it is shown that the five parameters combine into a unique dimensionless variable. Hence the complete diffraction behavior is derived from the knowledge of a universal one-variable function: the diffraction function. The physical origin of this property is discussed and a theoretical study of this function is presented. Also provided is a look up curve and a fitting rational approximation to allow easy diffraction-correction implementation. Experimental calibration of an echographic diffraction filter shows good agreement with the predicted diffraction function.
Speech-perception aids for hearing-impaired people: Current status and needed research
Authors:   Working Group on Communication Aids for the Hearing-Impaired
Published: October 2005

Both the overall aging of the population and its exposure to higher noise levels have increased the tendency to hearing loss and the importance of improved hearing aids for speech perception. This article reviews improvements in conventional electroacoustic hearing aids, as well as recently developed alternative classes of speech-perception aids, including surgically implanted cochlear stimulators, and vibrotactile, electrocutaneous and optical stimulating devices. It is concluded that the most effective aid for the vast majority of hearing-impaired persons is, and will remain for the immediate future, the electroacoustic hearing aid. In those cases for which no benefit is demonstrated for the electroacoustic aid, generally meaning persons with profound hearing loss, either the cochlear implant or a tactile aid may provide some improvement in the understanding of speech. In rare cases, some speech understanding in the absence of lip reading is achieved by patients with cochlear implants, for unexplained reasons. This and other pressing questions about speech processing need to be addressed by the research community if more effective aids are to be developed for the use of the 36.5 million hearing-impaired persons expected in the U.S. by the year 2050.
Dynamic behavior of a piezoelectric beam
Authors: Söderkvist Jan
Published: October 2005

A piezoelectric vibration can be characterized by the components of its electric equivalent. The dynamic nature of these components, as well as their dependence on the location of the electrodes on a piezoelectric structure, is studied in this article. General equations, based on a decomposition of a known vibration into its eigenmodes, are derived for a flexural vibration. A correction factor for the dynamic and geometric properties of the equivalent components is derived. The mechanical compliance of the beam is shown to depend on the choice of electrode configuration. The behavior far from resonance is also studied. It has been assumed that the mechanical and the piezoelectric properties can be treated as two quasiseparate problems.
Amplitude modulation detection for band-limited noise stimuli.
Authors: Eddins David A.
Published: October 2005

In a recent investigation of temporal gap detection, Eddins, Hall, and Grose observed that gap-detection thresholds were highly dependent on stimulus bandwidth and were relatively independent of stimulus frequency region. The present study investigated the relative importance of these two stimulus variables using an amplitude-modulation detection task. Thresholds for modulation depth were obtained as a function of modulation frequency. The stimulus bandwidth was either 200, 400, 800, or 1600 Hz. The frequency region was varied by adjusting the high-frequency cutoff of the noise which was either 600, 2200, or 4400 Hz. Temporal modulation transfer functions demonstrated the typical low-pass characteristic with modulation thresholds decreasing monotonically with increasing modulation frequency. Neither stimulus bandwidth nor the frequency region of the noise influenced the form of the temporal modulation transfer function, implying that the time constant of the temporal process is essentially the same for all stimulus conditions. Absolute sensitivity to amplitude modulation was better at the larger noise bandwidths. The ability to detect amplitude modulation was essentially the same at 2200 and 4400 Hz and was slightly poorer at 600 Hz.
Biological oceanography and bioacoustics: Promises, pitfalls, and idiosyncratic perspectives.
Authors: Ortner Peter B.
Published: October 2005

Populations of marine animals vary significantly in abundance over a broad range of time and space scales. It is almost certainly the case that the time and space scales of biological and physical processes are related but are so in a highly complex and nonlinear manner. Documenting and understanding the dynamics of this linkage has become a major intellectual focus for biological oceanography. The technical challenge implicit is the development of appropriate sampling and measurement systems. Although fisheries acoustics has received considerable attention, acoustic sampling of microzooplankton, macrozooplankton, and micronekton (organisms ranging from ca. 0.05 mm to a few centimeters) is in its comparative infancy. Nonetheless, results obtained to date have both stimulated the biological oceanographic community and occasioned major new funding initiatives within the federal agencies. Perhaps the potential that has most greatly contributed to the general excitement is the possibility of sampling biological structure on scales heretofore beyond reach, of sampling biological variability concomitantly with physical variability. Not surprisingly, once these possibilities were explored, a set of inherent difficulties became obvious. Perhaps the least tractable of all is the problem of target identification. For some ecological inquiries this is a critical lacuna of acoustical methodology. For others it may turn out to be less critical than first thought. To some degree, our community has converged upon the notion that if acoustical methods can be efficiently combined with alternative technologies (and the principal candidates have to date been optical) the target identification problem can be contained if not overcome.
Acoustic determination of seafloor roughness with multibeam echo sounders.
Authors: de Moustier C.
Published: October 2005

Multibeam echo sounders have been used extensively to gather swath bathymetry in the world’s oceans with co-registered depth samples a few percent of the water depth apart in the athwartships direction. The resulting maps provide seafloor roughness information down to scales of a few hundred meters. A much finer roughness resolution (cm scale) can be achieved when analyzing the beamformed seafloor echoes received by such sonar systems. One method consists of processing the corresponding acoustic data to remove system dependencies and to correct for bottom slopes and for geometric effects due to the area insonified by the transmitted pulse within the confines of the cross-fan beam geometry, so as to derive an angular dependence function of the average power in the returns for a given patch of seafloor. By fitting this function with a Helmholtz–Kirchhoff model for acoustic backscatter from a rough surface [D. R. Jackson etal., J. Acoust. Soc. Am. 79, 1410–1422 (1986)], a bottom roughness estimation is obtained. The model assumes Gaussian statistics for the roughness with a spectrum expressed as a power law of the wave number k, W(k)=βk−γ, where β is a roughness parameter which enters in the fitting process along with the exponent γ. Results for various seafloor areas are presented and implications for automatic seafloor classification schemes are discussed. [Work supported by ONR.]
Human efficiency for visual detection of targets on cathode ray tube displays using a two-level multiple-channel time history format
Authors: DiVita Joseph C. , Hanna Thomas E.
Published: October 2005

Human observers were tested for their ability to detect targets on visual displays. The displays simulated the multiple-channel time history format, using two levels of intensity encoding. A target was presented on 50% of the trials at a constant bearing. Observers indicated their judgment as to whether a target was present by using a four-category rating scale. Receiver-operating characteristics (ROCs) were generated from the rating data and compared to ROCs for an optimal detector. In experiment I, data were collected for two signal-to-noise ratios and for 32, 64, and 128 lines of data. Results indicated that observers were 3–5 dB less sensitive than an optimal detector. Performance improved as the number of lines increased but not to the extent predicted by optimal integration of information across lines of the display. In experiment II, the bearing at which subjects judged whether a target was present, was clearly delineated from the background by encoding information presented at this bearing in red. The purpose of experiment II was to determine the degree to which human inefficiency in detecting targets is due to the inability to focus on a single column of data. Subject performance improved when the red encoding was utilized, but subjects still performed 2 dB below ideal. This suggests that subjects use a suboptimal decision rule for judging the presence of the target.