bluuebunny/crossref_metadata_2025 huggingface dataset:
The effect of double hearing protection on speech intelligibility in a simulated aircraft noise environment
Authors: Shilling Russell D. , Thomas Gerald B.
Published: October 2005

The ability to receive radiocommunications in a noise field while wearing double hearing protection (i.e., circumaural muffs and ear plugs) was assessed under simulated aircraft noise conditions. A version of the modified rhyme test [i.e., the tri-word modified rhyme test (TMRT)] was administered to nine student aviators in three noise level conditions [90, 100, and 110 dB (SPL)] and three hearing protector device (HPD) conditions. The HPD conditions included the traditional single protection (i.e., flight helmet with no earplugs) and double protection (helmet with properly fitted earplugs) configurations. A third condition sought to mimic the anecdotally reported more typical condition of ‘‘sloppy fit’’ earplugs with flight helmet; a ‘‘real life’’ double protection configuration. A repeated measures ANOVA provided significant main effects for noise level (p<0.001) and for hearing protector (p<0.001). Post hoc tests revealed the expected inverse noise level-speech score relationship (p<0.05) and significantly (p<0.05) better speech scores as the degree of hearing protection declined. The trade-off between hearing protection and speech reception in noise is discussed.
Effect of piezoelectric sensor size and geometry on fidelity of sensor response when embedded in a structure subjected to fluid loading
Authors: Varadan Vasundara V. , Kim Jaehwan , Varadan Vijay K.
Published: October 2005

Piezoelectric transducers are commonly used to measure the response in a structure to impressed acoustic fields and other forces. By definition, an ideal sensor must be nonintrusive and its output must be a close representation of the response of the structure at that location. Thus the design of the sensor—its shape and size, material properties relative to the structure, its location in the structure—all contribute to sensitivity and fidelity. In addition, material damping in the plate and the transducer must be taken into account. The sensor response involves computation of the voltage induced in the transducer by an incident acoustic wave in the fluid or a force to which the plate may be subjected to. A hybrid numerical method involving finite element modeling in the structure and transducer region and a plane wave representation in the fluid region will be used. Canonical structures such as plates will be considered for which the structural response can be obtained in the absence of the sensor for purposes of comparison. A parametric study will be presented.
A transfer matrix model for analysis of noise and vibration in automotive power steering systems
Authors: Hastings Mardi C. , Chen Chuan-Chiang
Published: October 2005

Noise and vibration generated in automotive power steering systems significantly influence the sound pressure level measured in the passenger compartment and other disturbances such as steering wheel shudder. To address these problems, a distributed parameter, transfer matrix model of a complete power steering hydraulic transmission line was developed and implemented on a personal computer. The model includes not only the axisymmetric viscous fluid mechanics, but also the fluid–structure interaction created by the three-dimensional nature of the bends and twists in the tubing, and structural supports. The computations account for 14 state variables throughout the line. Computational results show that the effectiveness of devices such as expansion hoses with or without coaxial tuning cables, employed to reduce structurally transmitted noise and vibration, is limited by the complex coupling between fliud and structure. In fact, proper use of hoses, bends, and structural supports can attenuate noise and vibration and potentially eliminate the need for special devices. Use of this computational model could reduce the amount of experimental testing required to evaluate these systems. [Work supported in part by NSF.]
On the interaction of sound from a jet with a nearby structure
Authors: McGreevy Jennifer L. , Bayliss Alvin , Maestrello Lucio
Published: October 2005

The excitation of a flexible beam by acoustic disturbances generated by a spreading subsonic jet and the resulting interaction and acoustic radiation is investigated numerically. The formulation consists of solving the nonlinear Euler equations coupled to a nonlinear equation governing the vibration of the panel due to acoustic loading. The incident sound waves result from both disturbances due to the initial source as well as disturbances generated by instability waves in the jet. Under appropriate conditions, instability waves can also be generated upstream of the nozzle exit and may contribute additional sound. Results are obtained for the panel response as well as the transmitted and reflected acoustic field from the flexible panel.
Speech perception by normal children
Authors: Nittrouer Susan
Published: October 2005

Since the pioneering work of Eimas etal. [Science 171, 319–324], numerous experiments have shown that infants discriminate speech-relevant acoustic parameters in the same way adults do. Both infants and adults discriminate an acoustic difference if it places two stimuli on opposite sides of a phoneme boundary, but fail to discriminate a difference if it places two stimuli within the same phoneme category. Such results have led to the suggestion that the relation between acoustic and phonemic structure is innately recognized. However, studies of phoneme identification with preschool and young school-aged children call into question the intransigent nature of this proposed, innate mechanism. When two acoustic parameters are varied across stimuli, the contributions of each to phonemic decisions appear to differ for adults and children. Such results have led to the suggestion that the weights assigned to various acoustic properties of the speech signal are retuned as children gain experience with their native language, a proposal called the developmental weighting shift. Evidence for this proposal will be presented, as well as a discussion of its potential relation to developmental changes in auditory capacities and phonemic awareness.
Speech perception by less-skilled readers
Authors: Brady Susan A. , Fowler Anne E.
Published: October 2005

In the field of reading disability, the major finding of the last 25 years is that poor readers have difficulty attaining explicit awareness of the phonemic components of language. This insensitivity to the phonemic structure of words makes it correspondingly hard to master an alphabetic writing system representing those phonemes. An important question still under investigation is whether difficulty achieving phonemic awareness is entirely a metacognitive problem or whether it might stem from more basic deficits in perception. Using several paradigms (e.g., categorical perception tasks, speech perception in noise, word repetition), the research on this issue has generally supported two conclusions. First, less-skilled readers apply similar strategies but do so less efficiently and/or less accurately than better readers. Second, these difficulties are restricted to the linguistic domain: less-skilled readers perform as well as better readers on nonspeech control tasks. Current work is exploring the association between the emergence of phonemic awareness and developmental change in the underlying form of the phonological representations of words.
Speech perception by language-impaired children
Authors: Schwartz Richard G.
Published: October 2005

Children with language impairments, particularly those defined in an exclusionary fashion as having a specific language impairment, have been reported to have auditory and speech perception deficits in comparison to age-matched peers. Investigators have suggested characterizations ranging from generalized temporal deficits, causing difficulty with brief or rapidly changing cues, to limited processing capacity, resulting in high demands for difficult discriminations. Findings concerning perceptual abilities and limitations of children with developmental language disorders will be critically reviewed. The review will focus on the nature of the stimuli and the tasks employed in order to provide a comprehensive characterization of these auditory and speech perception limitations. The relationship between these deficits and known speech and language deficits and their hypothesized relationship to limitations in working memory will be considered. The problem of comparing auditory and phonetic perception to production data drawn from children’s production of meaningful words and phrases will also be addressed. A methodological alternative for assessing phonological perception and some preliminary data will be presented. A discussion of the importance of levels of data in understanding the relationships among speech perception, speech production, and the linguistic deficits observed in specific language impairment will conclude the presentation. [Work supported by NIDCD.]
Speech perception by hearing-impaired listeners
Authors: Boothroyd Arthur
Published: October 2005

Speech perception requires the generation, by the perceiver, of language patterns believed to underlie the speech actions of a talker. At any moment, the perceiver’s decisions are based on both direct sensory evidence and indirect contextual evidence. Depending on the perceiver’s prior knowledge and perceptual skill, the value of contextual evidence can contribute 5 to 10 times as much information as sensory evidence in normal conversation. Impaired hearing reduces auditory sensory evidence by its effects on threshold, dynamic range, resolution, and susceptibility to noise, leading to increased dependence on visual and contextual evidence. Hearing aids and cochlear implants offer only partial solutions. Adult-acquired impairments, which affect only sensory evidence, are managed primarily by sensory assistance. Congenital and prelingually acquired impairments, however, also affect acquisition of knowledge and perceptual skill. As a result, sensory assistance, though necessary and important, is only the first step in a comprehensive program of management for hearing-impaired children. In this presentation, the foregoing outline will be supported with empirical data on the sensory capabilities of aided and implanted individuals and on the roles of vision and context in speech perception by the hearing impaired. [Work supported by NIH Grant No. 2PO1DC00178.]
Speech perception by second language learners
Authors: Strange Winifred
Published: October 2005

Cross-language studies of speech perception by adults have shown ‘‘language-specific’’ patterns of perception of phonetic categories and contrasts. In general, phonetic categories that are distinctive (phonemic) in the listener’s native language are differentiated easily and effortlessly, while non-native phonetic categories/contrasts present perceptual difficulties. Thus learners of a second language (L2) often have persistant difficulty learning to perceive (and produce) ‘‘foreign’’ consonants and vowels. However, recent research has shown that not all non-native phonetic categories and contrasts are equally difficult to differentiate perceptually. Current theories that attempt to predict and explain these relative perceptual difficulties in terms of the relationship between native language (L1) and L2 phonetic categories will be discussed. In addition, results of perceptual training experiments with L2 learners which explore the effects of subject, stimulus, and task variables on perception of non-native phonetic categories will be summarized. Finally, implications of this research for general theories of speech perception and perceptual learning will be suggested. [Work supported by NIDCD.]
The resonance scattering theory and acoustic transients
Authors: Überall Herbert
Published: October 2005

Larry Flax conceived the idea that provided the impetus for the development of what is now called the acoustic resonance scattering theory (RST), as described in Chap. 4 of AcousticResonanceScattering, edited by H. Überall (Gordon and Breach, New York, 1992). The explicit development of this theory employed an approach used in the quantum mechanical theory of nuclear scattering, the so-called Breit–Wigner resonance theory. The interplay between target resonances and transient signals, hardly accessible to nuclear scattering experiments, is easily observed in acoustics and now forms the basis of experimental resonance studies that are most vigorously being carried out in France (see the above reference, or N. Gespa, LaDiffusionAcoustique (CEDOCAR, Paris, 1987). Pulses of long duration generate surface waves that carry out multiple circumnavigations of the target, their overlap leading to final transients or ringing of individual resonances (Carbó-Fite, Veksler, Maze, Ripoche). Short pulses combine the ringing of many resonances, the corresponding circumferential waves synthesizing surface-wave pulses in the fashion of beats, as demonstrated by Werby. For the interpretation of time domain experiments, a methodology is developed that is based on RST for a unique interpretation of results with a variety of target. [Work supported by ONR.]
Symmetry and axial focusing in backscattering by elastic objects in water
Authors: Marston Philip L.
Published: October 2005

Early calculations [L. Flax, J. Acoust. Soc. Am. 62, 1502–1503 (1977)] of the exact partial-wave series of elastic objects in water for ka from 900 to 950 suggest that elastic contributions to backscattering are relatively more important for spheres than for circular cylinders. The main distinction between the two situations is the axial symmetry of the sphere that produces backwards directed toroidal wave fronts associated with off-axis rays [P. L. Marston and L. Flax, J. Acoust. Soc. Am. Suppl. 1 68, S81 (1980)]. In addition to geometrical considerations and diffraction theory leading to a (ka)1/2 factor for such focusing, early support for this hypothesis was provided by calculations [L. Flax (personal communication, 1980)] for an aluminum sphere with ka=938 showing the form function magnitude ‖f‖ as a function of the backscattering angle γ=π−θ. The enhancement of ‖f‖ is localized to small values of γ as expected from the theory of axial focusing. The analysis of axial focusing [P. L. Marston, PhysicalAcoustics (Academic, New York, 1992), Vol. 21, pp. 1–234] is not limited to such extremely large values of ka and includes elastic shells and other symmetric objects. [Work supported by ONR.]
Relation between the landing-launching coefficient and the attenuation of surface waves propagating around an elastic cylindrical object
Authors: Conoir Jean-Marc , Millerat P. , Derem A.
Published: October 2005

One approach for developing a physical understanding of the scattering of sound from elastic objects in water is to use the Sommerfeld–Watson transformation (SWT). Following the works of Überall etal., it is known that elastic objects resonances are due to a phase matching of the surface waves propagating around the scatterer. What is more, resonance widths are related to surface waves attenuation, thereby showing the close link between resonance and surface waves propagation phenomena. According to the geometrical theory of diffraction, surface waves are also characterized by a landing coefficient and a launching one, both identical in the present case. This landing–launching coefficient calculated from the SWT gives an indication, as resonance widths and surface waves attenuation, on the coupling between the object and the surrounding fluid. The purpose of this work is to determine an analytical relation between the landing–launching coefficient and the attenuation of surface waves. This relation comes from an identification between the Breit–Wigner approximation of resonances and the Mittag–Leffler expansions related to surface waves contribution. Thereafter, a method of identification of the background introduced by Flax etal. in the acoustic scattering theory has been developed.
Scattering by an elastic shell with nonconcentric cylindrical surfaces: Generalized Debye series expansion
Authors: Danila E. , Conoir J.-M. , Izbicki J.-L.
Published: October 2005

The scattered pressure by a solid elastic shell defined by two nonconcentric cylindrical surfaces is considered; the cylinder axes are parallel. The shell is immersed in a fluid and is excited by a monochromatic plane wave impinging on it normally to the axis. The generalized Debye expansion (GDE) formalism is used to physically analyze all the local interactions involving in the scattering process. Two cylindrical coordinate systems, each associated with one cylindrical surface, are introduced. With this choice, all the needed cylindrical wave functions can be written either in one system or the other thanks to the Graf addition theorem and the boundary conditions can be simply expressed; the scattered pressure can be calculated. It is also possible to relate the elements of the S matrix to the local interaction coefficients, i.e., reflexion and transmission modal coefficients coming from the GDE. In next future, it allows to obtain an expression of the background which is useful to calculate the target resonances.
Observation of guided waves from a hollow cylindrical shell with the MIIR ‘‘in propagation.’’
Authors: Léon Fernand , Maze Gérard
Published: October 2005

The observation of the guided waves p=1 and p=2 from an isotropic elastic hollow cylindrical shell of infinite length excited by an obliquely incident plane acoustic wave is investigated. The use of ‘‘classic’’ MIIR (method of isolation and identification of resonances) is limited because of the refraction effect of the surface wave: The state of stationarity generated at a resonance frequency occurs out of the insonified area of the cylindrical shell. This problem has been solved by a new method called MIIR ‘‘in propagation.’’ Then, an isolation and an identification are possible in a large range of incidence, contrary to the configuration of transducers according to Snell–Descartes’ law (‘‘classic’’ MIIR). The resonances of the guided wave p=1 can be detected in a broad range of incidence angle. Contrary to the guided wave p=1, the experimental results achieved with incidence angles varying between 0° and the second critical angle show a particular evolution of the resonance frequencies of the guided wave p=2. The analysis of the theoretical width Γ justifies the presence or the absence of résonances generated at an incident angle.
Adaptive feedforward modal space control
Authors: Clark Robert L.
Published: October 2005

An alternative method of implementing adaptive feedforward control is presented in this work based upon independent modal space control of reverberant systems. Multivariable control theory is used to derive the optimal feedforward compensators, and when the error sensors are constructed in modal coordinates, the optimal compensators reduce to a single coefficient filter per control channel (assuming the reference signal is available). If the control inputs are also implemented in modal coordinates, then the control system is termed noninteracting since each mode is orthogonal to the other. Adaptive feedforward modal space control is ideal for implementation of time-averaged gradient descent algorithms such as Newton’s method since the optimal filter coefficient is obtained from a single iteration of the algorithm, and is insensitive to changes in the resonant characteristics of the reverberant system. Formulating the control system in modal coordinates also clearly illustrates the effects of causality and nonminimum phase characteristics on the ultimate performance of any adaptive control system, whether it is implemented in physical coordinates or modal coordinates.
Adaptive compensation of piezoelectric sensoriactuators
Authors: Cole Daniel G. , Clark Robert L.
Published: October 2005

An adaptive filter is used to estimate the feedthrough capacitance of a piezoelectric sensoriactuator in this work. The mechanical response of the piezostructure is resolved from the electrical response of the piezoelectric device through standard adaptive signal processing techniques. Two common adaptive algorithms are reviewed for the given application: the LMS and the RLS. For spectrally white inputs the adaptation of the digital compensator yields a filter output which is proportional to the electrical response of the piezoelectric device. Thus the remaining electrical signal consists of the charge due to the mechanical response of the piezostructure. An additional adaptive filter must be employed if a harmonic input is present since the harmonic input yields a bias term on the adaptive filter equal to the real part of the piezostructure response. Preliminary results from the theoretical analysis and numerical simulations indicate that adaptive signal processing can be employed to realize an adaptive, sensoriactuator for simultaneous strain actuation and sensing.
Active damping of enclosed sound fields through direct rate feedback control
Authors: Clark Robert L. , Cole Daniel G.
Published: October 2005

A new method of controlling enclosed sound fields subjected to impulsive or broadband disturbances is presented based upon direct rate-feedback control. Results from this preliminary study suggest that if a pressure transducer is collocated with a volumetric source in an enclosed sound field and rate-feedback control is employed, then the global properties of the sound field can be modified, specifically the modal damping ratio of the acoustic modes. Asymptotic stability of the control approach was demonstrated for positive feedback gains (neglecting specific sensor and actuator dynamics), making it model insensitive and thus ideal for actively tuning the transient characteristics and thus the reverberation time of the enclosure. Significant levels of global reduction in the acoustic potential energy can be achieved with the proposed control approach for stationary random disturbances or impulsive disturbances.
Model insensitive control of transient and persistent disturbances
Authors: Clark Robert L.
Published: October 2005

A hybrid, model-insensitive control algorithm for active damping and control of a persistent disturbance is presented. Direct velocity feedback control is implemented to suppress the transient response of the system and adaptive feedforward control is used to minimize the system response due to a persistent disturbance. A time-averaged gradient descent algorithm was selected to adapt the feedforward filter, rendering the controller model insensitive and thus adaptive for the chosen application. The feedback control loop serves to increase the damping of the controlled structure, decreasing the overall convergence time associated with the adaptive feedforward filter. Thus the integration of the two-control approaches affords increases in performance previously not possible with the adaptive feedforward controller alone.
Study on the method of active noise control in space
Authors: Jiong Zhong Yan , Shan Zhang Zhen , Kai Yue
Published: October 2005

A mathematic model of the impulse response of a MIMO speaker–microphone system in the spatial field space is obtained by means of a pseudo-random correlation identification. Two methods are selected for space active noise control. One method is the developed adaptive filter presented by the authors, which is a recursive algorithm with interlacing cycle and increasing phase. The other method proposed by the authors is to find a match point in the spatial field, which ensure that an active noise signal has the same shape and advanced phase in comparison with that of the canceling point. The single frequency signal, multifrequency signals, and the noise sampled from a motor driver cabin are used as the tested noise for digital simulation, electricity analogy experiment, and a test of the sound field in space. All the results prove the effectiveness and practicability of the two methods. Thus regardless of the qualities of the electricity-sound instruments, better results will be reached by way of a match point in the spatial field than by use of the adaptive filter.
Evaluation of a prototype beamforming binaural hearing aid
Authors: Terry Mark , Schweitzer Chris , Lindemann Eric , Melanson John
Published: October 2005

A computerized version of the California consonant test [Terry etal., Ear Hear. 13, 70–79 (1992)] was used to evaluate a wearable digital beam-forming binaural hearing aid. The binaural aid uses an analysis–synthesis method where phase and magnitude cues, derived from microphone placement at the ears, are used to attenuate sounds from locations other than the 0-deg azimuth position. Normal hearing subjects, with their left ear occluded to reduce binaural cues, were used together with hearing impaired subjects for evaluation. Speech at 0 azimuth was presented via loudspeaker at 50 dBA. In the noise condition four talker babble at 35-deg azimuth (54 dBA) was mixed with the target speech. The hearing aid was programmed to give a basic 6-dB/oct preemphasis. Overall gain for the hearing impaired subjects was initially adjusted to give approximately a 70% intelligibility score for the speech alone condition. Subjects were instructed to maintain body and head position while responding to word choices shown on a screen positioned above the target speaker. Both subject groups showed a relative improvement in intelligibility and response times for the beam processing mode in the speech in noise condition.
Noise classification for hearing-aid adjustment
Authors: Kates James M.
Published: October 2005

As a first step in designing an automatic hearing-aid adjustment system, a procedure is being developed for classifying common background noises. The intelligibility and quality of amplified speech depend on the type and level of the background noise as well as on the individual hearing loss. Thus to realize the maximum benefit from a hearing aid, the frequency response and processing characteristics should be set according to the type of background noise, and should change as the noise changes. The noise-classification procedure used 200-ms segments of noise, from which a parameter vector describing the spectral shape and the temporal envelope fluctuations is extracted. Parameter vectors for the different noises, averaged over 2 s of data, have been subjected to a cluster analysis to measure their similarity. Individual 200-ms noise parameter vectors are then assigned to the cluster having the nearest centroid, and performance results will be presented in terms of the percent of noises correctly classified. [Work supported by NIDCD.]
Recruitment compensation based on a simplified loudness estimation model
Authors: Hohmann V. , Kollmeier B.
Published: October 2005

To compensate for the frequency-dependent, nonlinear component of the hearing loss (recruitment), multichannel dynamic compression yields advantages over broadband compression. However, two unsolved problems remain: Due to the nonlinear additivity of perceived loudness across critical bands, the compression characteristic cannot be fitted correctly for narrow-band and broadband signals likewise. Furthermore, independent compression in different frequency bands introduces artifacts, which may limit the benefit of the algorithms. To overcome the problems, a 24-channel compression algorithm was developed and tested using critical bandwidths and time constants of ∼10 ms. The compression characteristic for broadband signals of arbitrary spectral shape is calculated by a loudness estimation model based on data from normal listeners. The algorithm should provide correct gain factors for narrow-band signals as well as for broadband signals. Furthermore, it introduces a psychoacoustically motivated interaction between frequency bands that reduces the processing artifacts. The complete algorithm was implemented in real time on a multi-signal processor setup and tested with sensorineurally impaired patients. First results of speech intelligibility measurements in quiet show the capability of the algorithm to compensate for the ‘‘distortion component’’ of the hearing loss. Further results of speech quality assessments and speech intelligibility tests in noise are presented.
Discrimination of multichannel-compressed stop consonants
Authors: Yund E. William , Crain Thomas R.
Published: October 2005

The effect of full-range multichannel compression (MCC) on stop consonant discrimination was studied. For normal-hearing subjects MCC was fit to three hypothetical flat losses, with thresholds ranging from 70 to 90 dB SPL, and one hypothetical sloped loss, with thresholds normal at 500 Hz and 90 dB SPL at 4 kHz. Hearing-impaired subjects were tested with the flat MCC as well as MCC fit to the subject’s hearing loss. The MCC had compression ratios from 1.75 to 7.00, numbers of channels from 2 to 31, and Robinson–Huntington compression [C. E. Robinson and D. A. Huntington, J. Acoust. Soc. Am. 54, 314 (1973)] with 10-ms time windows. Unprocessed stimuli and frequency-equalized linear amplification were control conditions. Deleterious effects of MCC on discrimination of voice onset time (VOT) were minimal. Normal-hearing subjects had difficulty with stop-consonant place discrimination in the most severe compression conditions. Hearing-impaired subjects needed either fitted MCC or frequency-equalized linear amplification to perform consistent place discriminations on these synthetic vowel–consonant–vowel (VCV) syllables. Complete results and their implications for the application of MCC in hearing aids will be discussed. [Work supported by Department of Veterans Affairs.]
The effect of compression ratio on speech intelligibility and quality
Authors: Fikret-Pasa Selda
Published: October 2005

The present investigation examined the effect of different compression ratios (compression ratios of 2:1, 3:1, 8:1 and wide-dynamic-range compression) on speech intelligibility and quality in compression-limiting systems. Speech intelligibility and quality were evaluated for sentences in four-talker babble. Sentences were presented at a signal-to-noise ratio of 7.5 dB at six different input levels, and at two of these levels (80 and 100 dB SPL) four different signal-to-noise ratios. Speech intelligibility was evaluated in terms of percent correct words and the quality of the conditions was rated on a scale of 0 to 100%. Across subjects, across input levels, and across signal-to-noise ratios, different compression ratios did not differ in terms of speech intelligibility and quality. Individual subject analysis revealed that for three of the subjects 8:1 compression ratio conditions gave the best speech intelligibility and quality results. One of the subjects achieved his best speech intelligibility performance with the wide-dynamic-range compression condition. For four of the subjects, different compression ratios did not effect speech intelligibility and quality differently. Overall, results indicated that there were considerable variations in performance with different compression ratios among individuals with similar hearing sensitivity. [Work partly supported by Northwestern Univ. Doctoral Student Research Grant.]
Speech recognition of normal-hearing and hearing-impaired listeners in amplitude-modulated noise
Authors: Eisenberg Laurie S. , Dirks Donald D. , Bell Theodore S.
Published: October 2005

Nonsense syllables were presented in high-pass, steady-state, and amplitude-modulated (AM) noise to determine whether the release from masking in AM noise was significantly different between normal-hearing and hearing-impaired subjects when the two groups listened under equivalent masker conditions. The normal-hearing subjects were tested in noise under two conditions: (1) in a spectrally shaped noise that produced 1/3-octave band thresholds equivalent to the hearing-impaired subjects; and (2) without the spectrally shaped noise. The release from masking in AM noise was significantly greater for the normal-hearing group than for either the hearing-impaired or masked normal groups. In a second experiment, normal-hearing and hearing-impaired subjects identified nonsense syllables and sentences in steady-state and AM noise at levels adjusted to approximate the spectral shape and gain of a hearing aid prescription. The release from masking was significantly less for the hearing-impaired subjects. These data suggest that hearing-impaired listeners are unable to benefit from the valleys in the amplitude envelope of a fluctuating masker even when both the speech and masker are presented at levels that are audible over much of the speech frequency range. [Work supported by NIDCD and VA R&D.]
Temporal asynchrony effects in listeners with cochlear hearing loss
Authors: Grose J. H. , Hall J. W. , Mendoza L.
Published: October 2005

Temporal asynchrony appears to play a dominant role in the perceptual organization of multi-source sounds. Two paradigms in which temporal asynchrony results in perceptual segregation are comodulation masking release (CMR) and modulation detection interference (MDI). Both of these paradigms involve amplitude-modulated stimuli. Because listeners with hearing loss of cochlear origin often exhibit reduced temporal resolution for modulated stimuli, this study examined whether these listeners are as susceptible to asynchrony effects as are listeners with normal hearing. The CMR task was similar to one described previously [J. H. Grose and J. W. Hall, J. Acoust. Soc. Am. 93, 2896–2902 (1993)]. The modulation interference task was novel. The baseline condition was a measure of sequention envelope discrimination for a 20-Hz wide band of noise centered at 1125 Hz. Modulation interference was measured as a reduction in performance brought about by the synchronous presentation of a second independent band of noise centered at 3784 Hz. The second band was then presented asynchronously with the target band: recovery from interference was interpreted as reflection perceptual segregation. While the listeners with hearing loss tended to perform more poorly overall, the pattern of their results was similar to that of the listeners with normal hearing. [Work supported by the NIDCD R01-DC01507.]
Frequency glide discrimination and frequency resolution as a function of age among a normal hearing population
Authors: MacNeil Jane F. , Slawinski Elzbieta B.
Published: October 2005

The relationship between frequency glide discrimination and frequency resolution was investigated among a group of normal hearing individuals aged 20 to 75 years. Frequency glide discrimination for 60-ms stimuli was determined at two frequency regions (1000 and 2500 Hz) for both upward and downward trajectories; converging or diverging offset frequencies; and in quiet and background noise. Psychophysical tuning curves were measured in a forward masking paradigm at both 500 and 4000 Hz. Relative to young adults, the elderly displayed larger just noticeable differences, greater effects for frequency region, noise, and transition direction. Though there was a significant effect for age, the results indicate that hearing sensitivity was the most important variable explaining more of the variance in the data than age. Even within normal limits, sensitivity still decreased with increasing age and therefore discrimination performance and shape of the tuning curves varied as a function of age. Consequently, the effect of age is confounded with sensitivity levels despite normal audiological profiles. Altered tuning curves are present before clinical evidence of cochlear damage indicating an impaired processing of complex signals.
Additivity of notched-noise and broadband-noise masking in normal-hearing and hearing-impaired listeners
Authors: Leek Marjorie R. , Summers Van , Jesteadt Walt
Published: October 2005

The threshold elevations produced by a combination of two or more maskers can be modeled as the sum of nonlinear transformations of the individual masker effects, along with an additional compressed term representing the listener’s threshold in quiet [L. E. Humes and W. Jesteadt, J. Acoust. Soc. Am. 85, 1285–1294 (1989)]. In this study, this modified power law (MPL) model has been applied to data recently published by Leek and Summers [M. R. Leek and V. Summers, J. Acoust. Soc. Am. 94, 3127–3137 (1993)]. They used notched-noise maskers to measure auditory filter shapes at 500 and 2000 Hz in quiet and in 20- or 35-dB spectrum level broadband noise in both normal-hearing and hearing-impaired listeners. Thresholds measured in combined notched and broadband noise are well-predicted from the individual masker effects according to the MPL model. Best-fitting compression parameters indicate greater linearity at 500 Hz than at 2000 Hz, but no significant difference in linearity between listeners with normal hearing and those with hearing loss. We are currently exploring alternative analyses of the data for hearing-impaired listeners, based on treating their sensitivity loss as an additional internal masker added to normal quiet thresholds. [Work supported by NIDCD.]
Effects of vestibular neurectomy on intensity discrimination and speech perception in noise
Authors: Zeng Fan-Gang , Lehmann Kristina M. , Soli Sigfrid D. , Linthicum Fred H.
Published: October 2005

Olivocochlear efferents are assumed to be severed during vestibular neurectomy in subjects with severe vertigo. Effects of vestibular neurectomy on pure-tone intensity discrimination and speech perception in noise were examined in six subjects by comparing performance between the surgery ear and the nonsurgery ear, and when available, between the pre- and post-operative conditions. Five of the six subjects had normal or near normal pure-tone average thresholds (<30 dB HL). Broadband white noise was used as in intensity discrimination, and speech spectral-shaped noise was used in speech reception threshold measurements. Both types of noise were presented binaurally at several different levels, whereas the tone or speech was presented monaurally and its level was varied adaptively according to the subject’s response. Preliminary results show that (1) loudness dynamic range is not affected by the surgery, (2) intensity and speech perception in noise significantly worsen after the surgery in some subjects but not others. The variability of the results may be due to the fact that not all efferents are severed in every subject during vestibular neurectomy. An objective measure of the efferent nerve survival after vestibular neurectomy in these subjects is needed to further understand perceptual function of the cochlear efferent pathway. [Work supported by NIH-BRSG.]
Toward a model for clinical evaluation of hearing loss and hearing aids using articulation index
Authors: Hou Zezhang , Thornton Aaron R.
Published: October 2005

Routine clinical application of the articulation index (AI) has been reported previously [A. R. Thornton etal., ASHA 34, 148 (1992); Z. Hou etal., ASHA 35, 234 (1993); Z. Hou and A. R. Thornton, Ear Hear. 15(2) (1994)]. This paper reports further development of an AI-based clinical model for evaluating hearing loss and hearing aids and for maximizing hearing aid performance across listening conditions. AIs across a range of listening conditions are plotted on a series of two-dimensional graphs to provide visual evaluations of a patient’s residual speech ability under each condition with or without a hearing aid. The AIs are integrated to obtain an integrated articulation index (IAI) as a criterion measure for evaluating the overall speech ability across the conditions. The optimal IAI is suggested as a hearing aid prescription method. An audibility improvement index based on the IAI is introduced to evaluate the benefit a patient might gain from using a hearing aid. This model can address loss and amplification requirements in both quiet and noise conditions and provides a comprehensive and easy-to-use tool for routine clinic applications.
Bounds on the detection of underwater objects using surface noise
Authors: Makris Nicholas C. , Kuperman William A. , Ingenito Frank
Published: October 2005

The idea of using naturally occurring ambient noise to detect objects in the ocean [Flatte’ and Munk, JASON JSR-85-108 (1985)] has received much attention due to experimental results in the high frequency regime [Buckingham etal., Nature 356, 327–329 (1992)]. In this paper, full-field surface-generated noise and scattering models are combined to derive an expression for the total noise field covariance of a stratified ocean waveguide with a submerged object present. The Cramer–Rao upper bound on performance is computed for some shallow water examples using conventional sensor arrays. The results, along with some simulated detections using beamforming, indicate the high frequency (∼10 kHz) detection is limited to relatively short ranges due to rapid decrease in scattered to direct noise ratio. The idea may be of most practical use for remote detection at low frequency (∼100 Hz). Detection is enhanced by high correlation between direct and scattered noise fields at the sensor. When direct and scattered noise fields are independent, the relatively short coherence scale of the scattered noise severely limits detection and precludes detailed imaging of the objects regardless of frequency. Standard detection techniques such as beamforming and MFP generally do not meet the upper bound on performance. a)Present address: Scripps Institution of Oceanography, La Jolla, CA.
Active detection prediction using higher-order moments
Authors: Pflug Lisa A. , Ioup George E. , Ioup Juliette W.
Published: October 2005

For some signals, higher-order moment detection methods have been shown to perform better than conventional methods for stationary power signals and deterministic finite-energy transients. The latter case has been investigated by the authors using Monte Carlo simulations for both active and passive detection and recently, using theoretical prediction formulas for passive detection. The corresponding prediction formulas for active detection are presented in this paper. From knowledge of the first four signal moments the signal-to-noise ratio (SNR) at the minimum detectable level is obtained for the cross correlation, bicorrelation, and tricorrelation central ordinate detectors for an arbitrary probability of false alarm. The prediction formulas require that the noise have Gaussianly distributed correlation central ordinate values. Comparisons are made between the theoretical predictions and results of computer simulations for a representative set of test signals in Gaussian noise. [Research supported by Office of Naval Research, Program Element 61153N, with technical management provided by NTLSSC.]
The application of the Bayesian fusion detection criterion to the case of two disparate signals
Authors: Perry Teenia T. , Sanders William M.
Published: October 2005

A Bayesian fusion criterion is described and applied to estimate improvements in detection range for two collocated sensors of disparate signals. The amplitude detection threshold selected for each sensor is expressed in terms of a likelihood threshold. The likelihood function is the ratio of the area under the probability density function beyond a threshold for a given signal to that for noise. The Bayesian criterion allows tying the likelihood threshold for one sensor to that for the other. When one can be set high, the other can be set low, thereby optimizing the use of the two sensor types. The assumption of Gaussian signal and noise probability density functions enables solving for the combined probability of detection versus range using an error function. An example is given for two signal types where each is subject to a different spreading loss. Estimates indicate that when the two individual detection ranges are comparable, the improvement in detection range through the application of a Bayesian criterion when compared to a simple ‘‘or’’ criterion (each sensor used separately with fixed nonrelated likelihood thresholds) can be as much as 30%. [Work supported by ONR.]
Optimum target detection in an uncertain reverberation-limited environment
Authors: Premus V. , Alexandrou D. , Nolte L. W.
Published: October 2005

In a reverberation-limited environment, can imperfect knowledge of the spatial correlation properties of the underlying seafloor geomorphology impact active target detection performance? This work seeks to address this question by examining the dependence of the joint probability density function of the scattered acoustic field at low to moderate incidence angles on the seafloor roughness correlation function. The objective is the development of an optimum decision theoretic approach to the problem of high-frequency active detection of a target suspended in the water column in the vicinity of the seafloor. Acoustic modeling is a geologically meaningful seafloor scattering environment is performed by applying the 3-D Helmholtz–Kirchhoff formulation to the anisotropic, power-law description of seafloor relief due to Goff and Jordan. The generality of the Bayesian approach permits uncertainty with respect to geoacoustic environmental properties to be defined probabilistically and incorporated into the detection algorithm. The method attempts to exploit the spatial coherence of the scattered sound field in constructing the joint pdf of the scattered pressure field measured at an N element array of sensors. Upper bounds on detection performance in terms of ROC curves are obtained as a function of environmental uncertainty. [Work supported by ONR, Ocean Acoustics.]
A ray travel time inversion scheme. Part I. Theory and simulated results
Authors: Lu I-Tai
Published: October 2005

A ray travel time inversion scheme is reported, not only for localizing the source but also for determining the array position and environmental parameters such as sound-speed profiles (in both water column and sediment), bathymetry, etc. The approach consists of the following four steps: (1) Sort out some ray arrivals and determine their arrival directions and times from the observed data; (2) backpropagate these rays in a nominal model to their source region; (3) compute the travel times of the corresponding eigenrays; (4) compare the simulated results with observed data and determine the new model. Some iterations of these steps may be required. Some key advantages of this approach are (1) most of the unknowns can be searched sequentially because rays characterize only local properties of the environment; (2) source localization ambiguity is minimized by utilizing ray travel times as well as path convergence; (3) ray tracing is a very efficient propagator in range-dependent environments; (4) the approach does not require full-wave matched field processing and the required knowledge about the environment is minimized; (5) ray chaos can be avoided simply by excluding chaotic rays. [Work supported by ONR and NRL.] a)On sabbatical leave from Polytech. Univ., Route 110, Farmingdale, NY 11735.
A ray travel time inversion scheme. Part II. Applications
Authors: Lu I-Tai
Published: October 2005

The inversion scheme reported in the companion paper is employed to analyze the data collected in the ICESHELF-89 experiment. Using the first SUS short data and assuming a range-independent environment, the location of the source and array and the sound-speed profiles have been obtained [J. W. Wolf, J. Acoust. Soc. Am. 94, 1769 (A) (1993)]. Here, not only the first shot is considered, but also other shots with larger separations between sources and receivers. For the first shot data, the same results are derived as reported when assuming a range-independent environment. However, the approach is two to three orders more efficient, due to the advantages discussed in the companion paper. Moreover, a better agreement is obtained between the simulated and the observed travel times if the water channel is allowed to have a sloping bottom. For other shots where the environment is strongly range dependent, this approach can still be applicable. The location of the source, the position of the array, the sound-speed profile in water, and the bathymetry are obtained. [Work supported by ONR and NRL.] a)On sabbatical leave from Polytech. Univ., Route 110, Farmingdale, NY 11735.