bluuebunny/crossref_metadata_2025 huggingface dataset:
Noise exposure in children and youths.
Authors: Clark William W.
Published: October 2005

Over the past several years there has been an increasing concern about excessive noise exposure in children and youths. Noise-induced hearing loss, once considered to be an occupational hazard, can be caused by any acoustic environment with sufficient energy to damage the cochlea. This presentation reviews available data on exposure levels and patterns from the most common sources of nonoccupational noise in children: Impulse noise from toy cap pistols and firecrackers in children and small-caliber firearms in youths, attendance at rock concerts, listening to amplified music through stereo headphones; and noisy school and leisure activities. For each exposure, consideration is given to the ‘‘dose’’ of noise experienced by the typical consumer, and the distribution of ‘‘doses’’ across users. The risk of each noisy activity causing or contributing to noise-induced hearing loss is determined by considering these factors and the prevalence of the activity in children and youths. Strategies are proposed to educate young consumers about the potential hazards to hearing from these exposures. [Work supported by NIOSH.]
Noise-induced hearing loss in children and adolescents.
Authors: Brookhouser Patrick E.
Published: October 2005

Occupational noise exposure remains the most commonly identified cause of noise-induced hearing loss (NIHL) in the general population. The issue of NIHL in children has received scant attention. Parents and public policy makers have expressed increasing concern about hearing risks to children and adolescents posed by noise sources such as: rock music, ‘‘boom’’ cars, loud toys, recreational vehicles, and personal cassette players. This report focuses on 114 children and adolescents (ages 19 and under; 90.3% males) who were diagnosed as having probable NIHL on the basis of clinical history and audiometric configuration. Forty-two children had unilateral losses, while 72 had sensorineural losses of varying degree in the contralateral ear. The mean age of referral for evaluation was 12.7 yr (range 1.2 to 19.8 yr, s.d. 4.21 yr), although 26% of the children were age 10 or younger at the time of referral. Inherent weaknesses of currently available clinical studies of NIHL in nonoccupational settings include imprecision in diagnosis and lack of quantification in characterizing the nature and duration of exposure. The occurrence of irreversible but potentially preventable hearing loss among a preschool and school-age sample should prompt improved observational and interventional, prospective longitudinal studies of NIHL in children and adolescents. Advances in molecular medicine as related to hearing loss, incorporated into such studies, could enhance understanding of individual variation in susceptibility to hearing damage from noise associated with age, gender, racial or ethnic status, and genetic make-up. Psychosocial measures should address the issue of improving compliance in the utilization of ear protectors among the younger age group.
Human auditory response to low-flying aircraft noise.
Authors: Nixon Charles W. , West Denise M. , Allen Nancy K.
Published: October 2005

Noise from low-flying aircraft may constitute a potential risk of noise-induced hearing loss. In this study, hearing threshold levels were measured following human exposures to audio recordings at ground level of noise from low-flying aircraft. The noise was characterized by sudden onset, durations of about 10 s, maximum A-weighted sound pressure levels in the frequency region of 1200–2400 Hz, and A-weighted peak levels up to 130 dB. The primary independent variables were A-weighted peak levels and successive repetitions of the noise. In phase I, subjects experienced single noise bursts at a maximum level of 130 dB with no observable changes in hearing, as reported at the ASA meeting in Houston. In phase II, a single noise burst was repeated at the same sound level up to eight successive repetitions or until a TTS was observed. Hearing threshold levels were measured at 4 and 6 kHz immediately following each individual noise burst. The results of phase II and the overall study will be discussed in terms of noise-induced hearing loss relative to damage risk criteria and ISO 1999.
New facility for very-low-frequency acoustic testing.
Authors: Kugler Andrew , Sneddon Matthew
Published: October 2005

A very-low-frequency test facility has just been completed for experimentation on the effects of impulsive and other low-frequency noise exposure on structures, animals, and people. The facility is a cube of steel-reinforced concrete with outside dimensions of 4.6 m (wide) by 6.7 m (long) by 3.2 m (high). The interior volume can be partitioned into test volumes ranging from 3.6 to 32.4 m3. Sound pressure levels as great as 155 dB can be generated between 0.5 to 100 Hz by 72 servomotor-driven transducers. Computer-generated test signals of arbitrary shape and duration can be produced, and a fully automated data acquisition and analysis system is available to aid efficient experimentation.
A group contribution analysis of the integral of the linear loss modulus-temperature curves.
Authors: Sperling L. H.
Published: October 2005

The integral of the linear loss modulus-temperature curves, termed the loss area, or L.A., is controlled in part by the chemical structure of the mers making up the polymers being examined. Another contribution arises through the morphology of multipolymer materials, particularly phase domain continuity. Research to date has centered on acrylic, styrenic, and vinyl polymers and interpenetrating polymer networks made from these polymers. The relationships among chemical structure, morphology, and damping capability will be discussed. The requirements for instrumental background corrections to the data will be examined, and several proposed corrections will be compared. Possible methods of maximizing (or minimizing!) damping will be explored. [Work supported by ONR.]
Modeling and analysis of echo reduction mechanisms in a composite material containing inclusions.
Authors: Hackman Roger H.
Published: October 2005

The sound attenuating mechanisms of a class of composite materials containing dense inclusions are analyzed. The basis for this analysis is the calculation of the elastic wave scattering from a single spherical inclusion embedded in a microinhomogeneous medium. The role of the rigid body translational mode of the inclusion and its relationship to PS conversion in the coating are described. Multiple scattering effects and the potential role of collective modes in a many inclusion system are also discussed. A parametric T-matrix analysis of the scattering addresses the potential for tailoring the frequency response of the coating through variation of the material properties of the inclusion and of the host medium within allowable constraints. The effects of varying the inclusion shape are likewise addressed by a spheroidal coordinate based T-matrix analysis of the scattering from prolate spheroidal inclusions of varying aspect ratio. Recent improvements in the treatment of multiple scattering aspects of this problem will also be reviewed.
Attenuation of sound in porous materials.
Authors: Plona Thomas J.
Published: October 2005

This paper gives a brief review of the attenuation properties of porous materials when the pore space is filled with either liquids or gases. Liquid-saturated materials (i.e., rocks, sediments, synthetics) will be shown to exhibit three primary mechanisms of sound attenuation: local flow, global flow (i.e., Biot theory), and scattering. How these mechanisms depend on liquid and solid properties (frame stiffness), porosity, pore shape, fluid permeability, pressure, and temperature (etc.) will be emphasized. Attenuation in air-saturated materials (i.e., soils, snow, synthetics) will be discussed in a similar manner with an emphasis on the nearly rigid nature of the solid. The unique two compressional wave behavior of porous media (fast and slow waves or type I and type II waves) will be reviewed as well as their history from Rayleigh to Biot.
Exploration of turbulence by nonlinear acoustic scattering.
Authors: Parker James E. , Korman Murray S.
Published: October 2005

A system has been developed to explore the nuances of nonlinear scattering in the presence of turbulence. The computer-controlled scattering apparatus makes translational and rotational ‘‘crossed beam’’ measurements of the nonlinear interaction. This is made possible by the geometry of the transducer setup, resolution of the focused transducers, and mobility of the transducer mounting unit. Two ultrasonic focused transducers (of sinusoidal driving frequencies 2.0 and 2.1 MHz) are mounted mutually perpendicular and with their focal points overlapping. The enhanced spatial resolution results from the minimal interaction volume defined by the intersection of the focused sound beams. The transducer mount is allowed to pivot about the vertical axis while independently translating across a submerged water jet. The 1/4-in. circular jet is the source of turbulence in this system. By making several cat scans of the turbulent jet at all angles, one is able to map out several statistical properties of the randomly fluctuating turbulent velocity components. In particular, one can experimentally determine the energy spectrum of turbulence E(k,r) (where k is the wave number of the turbulent eddies and r the radial position in the jet) [M. S. Korman and R. T. Beyer, J. Acoust. Soc. Am. 85, 611–620 (1989)]. [Work supported by Naval Academy Research Council.]
Materials characterization via acoustic levitation in air.
Authors: Boulware Bob , Everbach E. Carr
Published: October 2005

A system for acoustically levitating small objects in air at 25 kHz has been built. The system consists of two PZT crystals driving an aluminum horn to produce a vertical standing wave field between the horn end and a curved reflector. Both the solid particles and liquid drops were levitated, and the deformation and motion of these objects were measured in response to an excitation. The detection system consists of a He–Ne laser illuminating the object, which casts its shadow on a photodiode. Light extinction variations cause a fluctuating voltage signal from the photodiode that can be correlated with the motion or deformation of the object in response to impulsive or continuous acoustic excitations. Quantitative information about the mass, viscosity, and surface tension of objects has been derived from this signal for a variety of materials.
Consulting applications for an acoustical simulation room.
Authors: Pierson Richard G.
Published: October 2005

An acoustical simulation room was constructed for acoustical consulting use. The design criteria called for a space that could simulate the background noise, reverberation, spectral, time based, and directional response of proposed and existing spaces. Readily available professional audio and acoustical products were utilized for implementation into a dedicated space. These include programmable delays, reverberation processors, equalizers, and mixers as well as numerous amplifiers and loudspeakers. Significant use of digital audio interfacing was utilized to maximize system signal-to-noise ratios. Applications have included simulation of the acoustical response to architectural spaces and mechanical system noise for subjective evaluation; simulation of diffuse sound fields for attentuation evaluations; and simulation of a sound reinforcement system and masking response for intelligibility studies and subjective evaluation. Binaurally recorded demonstrations of the Orfield Associates Acoustical Simulation Room will be available at the end of the session.
High-speed digital, fiber-optic-linked, telemetry considerations for underwater acoustic arrays.
Authors: Desrosiers David A.
Published: October 2005

The effective design of an underwater cable-to-shore based telemetry system for large hydrophone arrays centers on efficient use of the following: the bandwidth of the transmission media, the information coding scheme, and the power at the remote end. Previous systems, using submarine coax cables 6 mi long, were bandwidth limited by cable attenuation to a 4 Mb/s transmission rate. This rate supported 20 data channels at a 5-kHz bandwidth (BW) with 8-bit resolution (48-dB dynamic range). The need for 100 channels in deeper water 20 mi offshore initiated the change to fiber-optic cable. The greater bandwidth of the fiber allows a transmission rate above 100 Mb/s, with 40 Mb/s actually being used. Dynamic range at 5 kHz BW was increased to 12-bit resolution (72 dB). Utilizing advancements in communications, data acquisition and low power electronics permitted a modular, bus-structured system, designed with off-the-shelf electronics. These improvements were first implemented in March 1991 in 1200-ft-deep water on the end of a 20-mi-long fiber-optic submarine cable, with a similar system still in operation today.
A precision measurement system for use in underwater acoustics measurements.
Authors: Forsythe S. E. , Hansen G. L.
Published: October 2005

A Precision Measurement System (PMS) has been developed at NRL-USRD to serve both as a production measurement system and as a platform for ongoing research in new signal generation and measurement techniques. This system is intended as a single-platform solution to most of USRD’s underwater acoustic test needs. Calibration of transducers and measurements of acoustic coatings are included in the tests performed by the PMS. The precision measurement system offers the following features: wide range in both frequency and amplitude; built-in calibration for traceability to NIST; multiple channels for simultaneous signal measurements; a flexible combination of analog gain and filtering stages; digital technology to allow generation of complex waveforms as well as sophisticated processing of measured data; an integrated programming language interface based on VAX/VMS workstations; a variety of signal processing algorithms for wideband versus narrow band, continuous versus transient, parametric versus nonparametric signal processing techniques. Four copies of the system have been built to insure uniformity of measurements across all USRD measurement facilities. Because the system is based on VAX/VMS, a variety of software is available for plotting and reporting as well as more sophisticated offline mathematical analysis.
A three-dimensional sound intensity probe.
Authors: Oguro Shun , Anzai Masazou , Suzuki Hideo , Ono Takahiko
Published: October 2005

A commonly used sound intensity probe is one-dimensional, that is, only the sound intensity in the direction of the probe axis is measured. This makes the sound intensity measurement very time consuming if two- or three-dimensional intensities are needed. Very rarely a two- or three-dimensional intensity probe is used for practical applications. A conventional three-dimensional intensity probe consists of six (three pair) microphones, making the probe itself very clumsy. Due to its non-negligible size, the sound field is disturbed and an accurate measurement becomes difficult. A three-dimensional intensity probe was developed that solves this problem. Four 1/4-in. microphones are located at each apex of a regular tetrahedron. Each of them are attached at the tops of four parallel tubes with a 4 mm diameter. When viewed from the front, the four microphones are located at the three apexes and the center of a triangle. The pressure at the center of the tetrahedron is given by the average of the four pressure outputs. The intensities from the center to the four apexes are obtained by use of the average pressure and individual pressure output. These four components are distributed to three (x,y,z) components. Results of numerical calculations that show the accuracy of the algorithm will be given.
Using acoustical measurements to monitor corn moisture.
Authors: Wheeler Paul A. , Gopalan Anush , Budge Scott
Published: October 2005

The moisture content of grain, such as corn, is usually measured by weighing a sample before and after drying. This paper reports the feasibility of measuring corn moisture on-line by sampling and processing the acoustic signals of corn with different moisture content. Acoustic measurements were made in an anechoic chamber. A condenser microphone connected to an Omnidata Polycorder was used to sample the acoustic signal of corn dropping into a bin. An adaptive filter was used to discriminate corn moisture by comparing frequency information from signals of different corn samples. The results demonstrated the feasibility of using acoustical measurements for on-line monitoring of corn moisture.
A general purpose multimedia system for real time transcription of acoustic data to digital storage.
Authors: Leybourne Allen E. , Pogue Anthony T.
Published: October 2005

Activities of the Arctic Acoustic Branch of the Naval Research Laboratory at Stennis Space Center, MS often require transcription of field recorded multichannel acoustic data into digital streams. Insertion of time codes, synchronization words, and check fields into this stream is difficult because of the real time requirement of the transcription process. This paper describes our experience with the integration of components to obtain a multimedia general purpose 2- to 110-channel system whose end product is stored on 2.3 Gigabyte, 8-mm (exabyte) tapes. The system has successfully processed direct analog data, analog data originally recorded on magnetic tape, pulse code modulated (PCM) data tapes, and digital data streaming tapes. Major components of the system are the LORAL System 500, Racal Storehorse, Panasonic Hi-Fi VCR, an in-house developed EVCR (expanded video cassette recorder) system, DATATAPE model DCR8-L5 and a Digital MicroVAX II which continuously receives the LORAL data stream and writes the exabyte tapes. These may then be processed off-line on computers that support the exabyte tape drives. As header information is easily included at the beginning of these tapes, the archival value of this storage format is significantly enhanced.
Real time wave vector filtering in one-dimensional media.
Authors: Gibbs Gary P. , Fuller Chris R. , Silcox Richard J.
Published: October 2005

Often it is necessary to measure the wave vector components of propagating waves in vibrating media. Some examples are: measurement of traveling flexural and extensional waves in beams; measurement of acoustic waves in ducts; and determination of boundary conditions in structural or acoustic systems. In previous work, a time domain technique for wave vector filtering of plane waves in ducts was developed which required that the sensor spacing relative to the wavelength of interest be small. Frequency domain techniques have also been developed to perform wave vector filtering in both dispersive and nondispersive media. These techniques cannot be used in real time because of the signal processing requirements. In this work, a new time domain technique is presented that utilizes two sensor locations in conjunction with a digital filter, for each wave type, to provide a real time estimate of the complex amplitude of each of the traveling wave components for both dispersive and nondispersive media. The technique estimates the wave components over a band-limited spectra whose limits are dictated by the spacing between the sensors (up to a spacing of half a wavelength). A simulation is used to determine the performance of the new technique in both dispersive and nondispersive media over a range of frequencies.
A matched-mode processing technique for localizing a transient source.
Authors: Lu I-Tai , Chen Han-Yang
Published: October 2005

A matched-mode processing method separates the range and depth estimation tasks, thereby obtaining a sequential procedure that circumvents the need for a two-dimensional search over all hypothesized source locations. Therefore, when applicable, it is more efficient than a matched-field method. Here, the conventional frequency-domain, matched-mode method is extended to deal with more realistic nonharmonic, narrow-band signals directly in the time domain. The waveform of each mode is derived by a least-squares estimating procedure in the time domain. This procedure is essentially equivalent to the conventional modal filtering approach which is carried out by inner products of the received signals and the modal function in the frequency domain. Then, the source range and depth information can be estimated by the arrival times and magnitudes of some clean modal arrivals. The advantage of this extension is that the arrival times of the modal wave packets are less sensitive to model uncertainties than the phase of the different modes in the purely harmonic case. Moreover, this approach does not require a prior knowledge of the source waveform. This extension is applicable when the source signal is narrow band and the range dependence in the ocean environment is weak. In numerical simulation, the source and receiver arrangement and the ocean parameters are chosen from the experimental data measured in the Mediterranean Sea [S. M. Jesus, J. Acoust. Soc. Am. 90, 2034–2041 (1991)]. [Work supported by ONR.]
Matched-field localization of acoustic backscatter in the Arctic Ocean.
Authors: Hayward Thomas J.
Published: October 2005

Matched-field processing has been studied extensively in recent years as a method for localizing underwater sound sources. For a source in the water, matched-field processing is frequently implemented assuming a point source for the replica field, ignoring the directional radiation pattern of the source. Generalization of the method to boundary backscatter from ocean surface features requires computation of the scattered field of an extended-body scatterer; in general, one expects that the directional properties of the scattered field may need to be included in the replica generation. In this work, localization of surface (ice) backscatter using data collected during the CEAREX 89 Arctic field experiment is demonstrated, using first a point-source replica. Then, the possibility of improving resolution and sidelobe levels is investigated by using a replica for the scattered field generated by using methods of coupled normal mode theory. Performance improvement is first investigated in simulations, one purpose of which is to test the sensitivity of the method to the scatterer shape. The improved field replicas are then applied in matched-field processing using the real data. [Work supported by the ONR Arctic Sciences Program.]
A tribute to Richard N. Hamme: The father of open office acoustics.
Authors: Harris David A.
Published: October 2005

The late Richard (Dick) Hamme, proprietor and founder of Geiger and Hamme Laboratories in Ann Arbor, Michigan was a Fellow of the Acoustical Society of America, and a leader and pioneer in the development of procedures for the evaluation of acoustical materials and architectural systems. Dick and his associates corrected major misconceptions about open plan office acoustics. Many said that achieving speech privacy in the open plan was impossible. Not only did his efforts prove entirely the opposite, he invented, developed, and nurtured a whole technology that is just now being understood by the acoustical community. Procedures developed by Hamme for the U.S. General Services Administration, Public Building Services were used by public and private sector specifiers alike. Known as PBS C.1, C.2, and C.3 with several ancillary standards, these documents led to the re-thinking of the acoustical specifications in terms of system performance. After many years of use, these documents have been recently adopted by ASTM Committee E-33, Environmental Acoustics. This paper describes the evolution of the achievement of confidential speech privacy, a process that began in the late 1960’s and has been refined to a technology of its own.
Sabine reverberation equation revisited.
Authors: Young Robert W.
Published: October 2005

About 1900 W. C. Sabine reported the ‘‘sound absorbing power’’ of materials in a reverberant room as proportional to a logarithm of the ratio of an initial to a final ‘‘residual sound,’’ after the source is stopped. In present-day terms, sound pressure level (a logarithm) decays at the rate d decibels per second, after the source is stopped. At least as far back as 1965, ASTM C423, ‘‘Sound Absorption⋅⋅⋅Reverberation Room Method,’’ obtained the Sabine absorption A in metric sabins by A=0.9210 Vd/c; 0.4 ln 10=0.9210 is a pure number. The volume of the reverberation room is V, cubic meters; the speed of sound of air therein is c, meters per second. On substituting reverberation time T=60 dB/d, Sabine sound absorption A=55.3(dB)V/cT; this dimensionally complete equation shows the decibel explicitly. The Sabine absorption of a surface of area S in square meters, and Sabine absorption coefficient a, isA=Sa. The unit of the Sabine coefficient a is thus the decibel; it is dimensionless in the sense of length, mass, or time; the unit is usually omitted. Being derived from a decay rate, the Sabine sound absorption coefficient is not ‘‘Ideally the fraction of the randomly incident sound power absorbed... .’’
On the presentation of large amounts of architectural acoustics data.
Authors: Campbell Richard H.
Published: October 2005

With the proliferation of data acquisition systems that process time domain impulse responses taken in auditoria, there is an overwhelming quantity of (useful) data that can be computed using the digital signal processing algorithms ordinarily supplied with such systems. The example given is a 507-seat lecture hall where two different speech-reinforcement systems were evaluated at 10 spatial locations using MLSSA, generating over 4.8 megabytes of time domain data in less than 2 h. One additional hour of semiautomated postprocessing resulted in 960 important acoustic parameters from a field of 2100 pieces of analytical data, and 1960 modulation transfer function parameters for STI calculations. These multitudinous data are organized into a series of colorful graphs that show a large amount of interconnected information in a single view, and in a way that provides maximum insight to the experienced acoustician, yet can be created easily in a laboratory having a modest software budget and rather ordinary color display and printing means.
Acoustic scale modeling of auditoria.
Authors: Potter Whitney N. , Everbach E. Carr
Published: October 2005

Acoustic scale modeling was explored using a 1/48th scale model of the Lang Performing Arts Center at Swarthmore College [M. H. Soon and E. C. Everbach, J. Acoust. Soc. Am. 89, 1898 (1991)]. A system was designed to test the scaled acoustic properties of modeling materials, and appropriate materials were selected to model to acoustic treatments of the auditorium. Measurements of acoustic descriptors made from the model were compared to measurements made in the actual auditorium. This modeling technique can be applied to testing designs of unbuilt auditoria, or to testing possible acoustic treatments in existing auditoria.
The developmental role of spectral relations in processing place of articulation of nasal consonants from short duration segments.
Authors: Ohde Ralph N. , Ochs Marleen T.
Published: October 2005

In processing place of articulation of prevocalic nasal consonants, the spectral change from the murmur to vowel onset is a critical cue for both child and adult speech. In order to further our understanding of the nature of this cue in the speech of children and adults, three children, ages 3, 5, and 7, and an adult female and male produced CV syllables consisting of either /m/ or /n/ in the context of four vowels /i,æ,u,open aye/. Ten adults identified the place of articulation of the nasal before and after the insertion of periods of silence between 25-ms segments of the murmur and vowel transition. The major findings were as follows: (1) across speakers, the average identification of isolated segments was low ranging from 70% to 53% for the murmur, and 71% to 49% for the vowel transition; (2) across speakers, the average identification of the murmur+transition segment was above 90%; and (3) the insertion of temporal gaps caused a reduction in the identification of nasals with the greatest effects occurring for children, the alveolar place of articulation, and a silence gap of 150 ms. The findings will be discussed relative to differences between children and adults, and the potential mechanisms involved in processing spectral relations. [Work supported by NIH, DC00464.]
The developmental precedency of spectral and temporal cues in the perception of the /r-w/ phonemic contrast.
Authors: Fitzgerald L. K. , Slawinski E. B.
Published: October 2005

The present study examined acoustical and developmental influences on the categorical perception of /r/ and /w/. One group of adults, and three groups of normally developing preschool children, aged 3, 4, and 5 years, respectively, participated in three experimental conditions. These conditions involved performing a video-game identification task employing three different synthetic ‘‘rock-walk’’ continua. In condition I, the acoustic tokens varied according to both spectral and temporal cues; in condition II, the acoustic tokens varied only with respect to spectral cues; in condition III, the acoustic tokens varied only with respect to temporal cues. As hypothesized, changes in perceptual responses were seen as a function of both age and acoustical cue availability. Temporal cues, relative to spectral cues appeared to be the more salient cue for 3 yr olds, whereas for the 4 yr olds, 5 yr olds, and adults the two types of cues appeared to be equally salient. Other developmental trends evidenced included findings that children, relative to adults, demonstrate greater /r/ perceptual categories, more variable perceptual performance, and a tendency to base their phonemic categorizations on primarily only one cue.
Vowel boundaries for steady-state and transient formants.
Authors: Nabelek Anna K. , Czyzewski Zbigniew , Crowley Hilary
Published: October 2005

Fifteen-step /i–eh/ continua were generated to investigate effects of: (1) listening conditions (quiet, noise, and reverberation), (2) subjects’ hearing (normal and impaired), and (3) trajectories of F1 and F2 (all steady-state, F1 changing in upward or downward direction, and both F1 and F2 changing in upward or downward direction). Stimuli (each repeated 10 times) generated with the Klatt synthesizer were 200 ms long. Speech-spectrum noise was mixed at S/N=0 dB. Reverberation was generated by a computer program and convolved with the stimuli (T=0.9 s). Subjects, 10 in each group, were tested individually. Stimuli were delivered monaurally through an earphone at a comfortable level. Listening condition had no effect on the boundary location for either group of subjects and had no effect on boundary slope for the normal-hearing subjects but a significant effect on slope for hearing-impaired subjects. Type of trajectories had a significant effect on location of the boundary only for the hearing-impaired subjects and on boundary slope for both groups of subjects; the two continua with F1 and with F1 and F2 changing in downward direction produced different results than the continua with steady-state formants and with formants in upward direction. [Work supported by NIDCD.]
Intelligibility of speech sounds in amplitude-modulated noise.
Authors: Bell Theodore S. , Eisenberg Laurie S. , Dirks Donald D.
Published: October 2005

Two experiments describe the effect of amplitude-modulated (AM) noise on consonant recognition in normal-hearing and hearing-impaired listeners. The paradigm made direct estimates of release from masking due to fluctuating noise by comparing performance from steady-state and AM conditions while controlling signal level differences. In the first experiment, ten young audiologically normal adults identified nonsense syllables varying in vowel context (a, i, u), order (CV, VC), and speaker (male, female) at −2 dB S/N in a shaped noise. The noise was either at a constant level or amplitude modulated by a sinusoid varying in frequency (10, 31.5, and 100 Hz) and depth of modulation (50% and 90%). Depth of modulation produced a significant improvement in intelligibility over steady-state performance, with the 90% modulation depth showing greater release than 50% modulation. The greatest improvement (14.9%) occurred with the 31.5-Hz modulation rate. Frequency effects varied significantly by speaker, and the interaction between frequency and depth of modulation was also significant. Depth of modulation interacted in complex ways with vowel, order, and speaker. Experiment 2 compared normal and hearing-impaired listeners with 90% AM noise at 31.5 Hz. The hearing-impaired individuals displayed significant release from masking, although less than the control subjects. The amount of release varied by audiometric configuration. Overall, these results indicate that release from masking is evident in some hearing-impaired listeners and absent in others; audibility and temporal factors will be discussed. [Work supported by NIH.]
Acoustic analyses of contrastive stress production in children with normal and impaired hearing.
Authors: Carney Arlene E. , Weiss Amy L. , Fahey Richard
Published: October 2005

The purpose of this study was to determine the acoustic cues used by listeners making judgments of contrastive stress production in two groups of speakers: (1) children with normal hearing ranging in age from 3 to 6; and (2) children with severe-to-profound hearing loss ranging in age from 4 to 18 who were language matched to the normally hearing group. Children performed a picture-description task of pairs of photographs that varied in agent, action, or object. The second sentence of the pair was designed to have one semantic category as the stressed target. Perceptual judgments of stress in these speakers were reported in Weiss etal. [J. Speech Hear. Res. 28, 26–35 (1985)]. Subjects were divided into four groups according to hearing status and high or low listener agreement of produced stress. Measures of fundamental frequency (F0), relative intensity, and relative duration were made. Both normally hearing and hearing-impaired groups had speakers who increased F0 for stressed words within an utterance as well as those who did not vary F0 or varied nontarget words. In addition, F0 rise was observed less frequently in the object (word final) position for all speakers. Duration and intensity effects were more variable for both groups. [Work supported by NIH.]
Effects of distorted auditory feedback on speech.
Authors: Tartter Vivien C. , Abdelli Nassima , Economou Alexandra
Published: October 2005

A research program for acoustically measuring changes in speech production will be described. Here, F0 and spectrographic measurements target acoustic correlates of consonant, vowel, and syllabic stress features in 45 phonemically balanced words. To date, 5 Nucleus-22 multichannel cochlear implant patients have participated (at least) prestimulation, immediately after stimulation, 3–6 months after stimulation, and 1-year after stimulation. Commonalities among patients include: a rapid increase and stabilization of F0 (in females) to normal values; immediate and continuing changes in vowel and syllable durations and in relative amounts of breathy noise and low-frequency energy, gradual changes in vowel formant frequencies with movement still observed at 1 year; and changes in consonant properties (such as F1 cutback for voiceless stops) emerging for the first time at 1 year. Results relate to (1) the potential role of critical periods in the effect of auditory feedback on speech habits, (2) the relative susceptibility of speech features to modification, and (3) the likely auditory parameters available from Nucleus-22 WSP III stimulation. [Work supported by NIDCD.]
Acoustic characteristics of normal and dysarthric speech.
Authors: Fourakis Marios , Coelho Carl D.
Published: October 2005

The experiment reported here examined the implementation of final lengthening and main sentence stress in the speech of one moderately ataxic dysarthric (AD) and one normal speaker. A set of C1VC2 syllables was constructed such that C1 varied over [b,d,g,p,t,k], C2 was always [d], and V was one of the nine nonretroflex vowels for the [b——d] context, and one of [i, eh, æ] for the other five initial contexts. These syllables were embedded in carrier sentences such that the target syllable occurred in two positions, final and medial, and bearing main sentence stress or not. Eight repetitions of each syllable in each position-stress condition were recorded and digitized. Measurements of initial closure, VOT (if any), vowel, and final closure durations were made directly from the waveform. Preliminary results indicate that the AD speaker implemented final lengthening of the target syllable in a manner similar to the normal speaker overall, but differed in the distribution of the lengthening effect over the acoustic segments comprising the syllable. This result is not in agreement with the results reported by Bell-Berti etal. [Proc. of the 12th Int. Congr. Phon. Sci. (1991)] for French AD speakers. Results on the implementation of sentence stress will also be presented. These will be discussed within the framework of how cerebellar deterioration can affect the fine control of speech timing.
Performance stability of high-resolution matched-field processors to sound-speed mismatch in a shallow-water environment.
Authors: Smith G. B. , Chandler H. A. , Feuillade C.
Published: October 2005

The effects of variations in water sound-speed parameters on the performance of four matched-field processing algorithms (Bartlett, maximum likelihood, sector focusing, and multiple constraint) have been investigated. The SNAP propagation model was used to generate the replica acoustic pressure field for a shallow-water channel with a depth variable sound-speed profile typical of a midlatitude summer environment. It was also used to simulate a ‘‘detected’’ field due to an acoustic source. These were then correlated using the four algorithms for selected degrees of mismatch of the water sound-speed profile. The maximum likelihood and multiple constraint estimators achieved good peak resolution and high accuracy for small degrees of mismatch. The maximum likelihood estimator deteriorated quickly as the mismatch increased. The multiple constraint performed significantly better, but both failed to give correct location estimates at the higher mismatch values. In contrast, the sector focusing estimator continued to give accurate location estimates over the whole range of sound-speed mismatch, with slightly less peak resolution at small mismatch values. It also performed much better overall than the Bartlett estimator, which was found to give accurate location estimates over the whole range of mismatch values used, but with poor peak resolution against a background containing many high sidelobes. [This work was supported by the Naval Res. Lab.]
Effects of sound-speed mismatch in the lower water column on matched-field processing.
Authors: Livingston Ellen
Published: October 2005

The sensitivity of matched-field processing to sound-speed mismatch in the upper water column (above 1000 m) has frequently been discussed [A. Tolstoy, J. Acoust. Soc. Am. 85, 2394–2404 (1989); Henrik Schmidt etal., J. Acoust. Soc. Am. 88, 1851–1862 (1990)]. In this work, the effects of sound-speed mismatch in the lower water column (3000–5000 m) are investigated. Such mismatch can arise from differences in climatology data or inaccuracies in the sound-speed equation. Low-frequency sources are simulated at shot depths in a range-independent deep-water Pacific environment and are processed with a mismatched replica field on a long vertical receiving array. Source range is varied from 50 to 500 km. For a sound-speed mismatch of 0.6 m/s at 5000 m, the effects on the mainlobe peak level are strongly and cyclically range dependent, varying by as much as 1 dB. A closed form analysis of this phenomenon is presented and the effects of source depth and of array placement in the water column is considered.
Limitation of reflecting irregular bathymetry on matched-field processing.
Authors: Diachok O. , Smith J. F.
Published: October 2005

Simulations of the potential of matched-field tomography for inverting ocean sound-speed structure (Tolstoy and Diachok, 1991) have to date neglected effects of bottom interacting modes. At very low frequencies these modes are significant over: thickly sedimented bottoms and thinly sedimented bottoms where V _ s≳V _ w. In thinly sedimented regions of the mid-pacific, V _ s is approximately 1570 m/s (Diachok, 1991). Results of a study to determine the effect of experimentally obtained bathymetry in the thinly sedimented Pacific on matched-field processing follow. The bathymetry exhibits an irregular variation in depth as a function of range and is assumed to have a constant sound speed. The range- and depth-dependent sound speed in the water column was obtained from tomography data. The acoustic field used as synthetic data ‘‘measured’’ at the array was calculated by the parabolic equation program FEPE (Collins, 1991). The program was shown to be highly accurate (0.15 dB at 1000 km at 15 Hz) in matched-field studies against ‘‘exact’’ results produced by the normal modes program krakenc in a range-independent environment. Simulation of the loss in signal gain was conducted at a frequency of 14 Hz at various ranges and constant bottom sound speeds of 1530, 1540, 1550, 1560, and 1570 m/s. The effects of irregular bathymetry and shifts in critical angle were to increase signal array degradation and reduce the accuracy of source localization. Effects of interaction with far and near sides of seamounts (relative to array position), and the overall trend of signal loss with range due to multiple interactions with the bottom will be discussed.
Matched-field inversion of geoacoustic parameters of elastic bottoms.
Authors: Wolf John W. , Diachok Orest I.
Published: October 2005

The potential of matched-field inversion of ocean bottom parameters is demonstrated through simulation of signals transmitted by one ship and received on a 1.5-km-long horizontal ship-towed array in end-fire mode. The simulations use an ocean model that is representative of a thinly sedimented (36-m) basalt bottom. The output of a matched-field processor (Capon correlator) at a known source position is maximized by searching through bottom parameter differences from a geoacoustic profile. In particular, the shear velocities, Cs, in a model of the upper crust and an overlying thin sediment layer are varied about the true values. Relatively small differences (15% of true) in basement shear velocity produce a 4-dB degradation in processor output. Shear speed mismatch in the sediment produces thin film effects that appear in the processor output function as resonance peaks at several values of ΔCs. The values of ΔCs for most peaks vary strongly with source frequency except the peak at ΔCs=0. The results illustrate the potential of broadband matched-field processing to invert elastic bottom parameters, particularly shear speeds in thin sediments and upper crust, which to date have generally proven inaccessible with existing methods.
Replica partitioning and preselection for faster matched-field processing.
Authors: Ozard John M. , Zakarauskas Pierre , Berryman Don G. , Wilmut Michael J.
Published: October 2005

Matched-field processing requires the evaluation of the inner product of the measured and replica fields at a grid of points covering the search region. This process may require a significant computational time despite the increased speed of computers. A way of reducing the search space is described and evaluated. First, a basis for the signal space is found for all possible source positions. Then the replica fields for the search region are partitioned into groups with similar excitations of the basis vectors used to represent the replicas. One can then quickly select the group of replicas that is most like any data to be matched, and then do a detailed search over only that group to find the best match. An example is shown of the detection and localization performance of the Bartlett beamformer with and without preselection of the replicas. A reduction in computation time of 10 to 20 was achieved with only a slight loss in performance over a wide range of signal-to-noise ratios.
Matched-field processing speedup through fast nearest-neighbor search in excitation space.
Authors: Zakarauskas Pierre , Ozard John M. , Berryman Don G. , Wilmut Mike
Published: October 2005

By casting the matching portion of the MFP problem into a nearest-neighbor search, one may apply the fast nearest-neighbor search algorithm to obtain potentially considerable speedups. This algorithm makes it possible to find the replica closest to a test pattern (peak of the Bartlett processor output), in a time which is asymptotically constant with the number of replicas. The algorithm first partitions the excitation space, and finds which partition each of the replicas falls into. When a test pattern is supplied, one compares it to all replicas within the partitions which are within a given distance R to the test pattern. If the distance d between the closest replica found during the first pass and the test pattern is less than R, then the search stops. If not, then a second pass is done using d as a search parameter to select partitions. This algorithm guarantees that the nearest neighbor is found. The mean number of operations done to find the nearest neighbor is presented as a function of the density of patterns per partition and the dimensionality of the excitation space. A global minimum is shown to exist for the mean number of operations at a partition size corresponding to slightly less than one replica per partition.
Acoustic source localization in a wedge waveguide using time-domain techniques.
Authors: Feuillade C. , Clay C. S.
Published: October 2005

In previous work [C. Feuillade and C. S. Clay, J. Acoust. Soc. Am. Suppl. 1 88, S26 (1990)], the results of an investigation into time domain source imaging techniques in a range-independent, shallow-water waveguide were reported. The research reported here extends this method to consider the case of an impulsive source in a density penetrable wedge waveguide. Single-channel source location information is used by means of time domain transmissions to multiple receivers. The signals from the receivers are time reversed and transmitted through sources at the receiver locations. The field is mapped to locate the source. Particular attention is given to the peak amplitudes, sidelobe amplitudes, and spatial resolution as a function of receiver channels. The field calculations are performed by using an image approximation that is based on D. Chu’s exact solution for a wedge [J. Acoust. Soc. Am. 86, 1883–1896 (1989)]. S. Li and C. S. Clay have also given a discussion of single channel theory and experiment [J. Acoust. Soc. Am. 82, 1409–1417 (1987)]. The results show that source localization in a wedge is significantly promoted by the range dependency of the environment, which eliminates ‘‘range’’ sidelobes, even for a single hydrophone. The signal peak is more effectively resolved by locating the receivers as far from the wedge apex as possible. ‘‘Angle’’ sidelobes may be effectively eliminated by introducing additional receivers in an arc at constant range from the apex. [This work was supported by the Naval Res. Lab. and by the Office of Naval Res. under Contract No. N00014-89-J-1515 (CSC).]