bluuebunny/crossref_metadata_2025 huggingface dataset:
Computational studies of different laser intensity modulation techniques to generate underwater sound via thermal expansion
Authors: Willett Thomas C. , Berthelot Yves H.
Published: October 2005

The thermalization of water, due to the absorption of laser radiation, offers a way to generate underwater sound from an airborne platform. However, the thermal expansion mechanism that produces sound is notoriously inefficient and the standard technique of amplitude modulation (AM) of the laser intensity is generally inadequate for practical applications such as bathymetry. With the advent of ultra-high repetition rate pulsed lasers, it is now appropriate to take a fresh look at some alternative schemes for modulating the laser intensity. Computer simulations will be shown for three different modulation techniques in comparison with the standard AM technique. These modulation schemes are, respectively, frequency modulation (FM), pulse density modulation, and pulse amplitude modulation. In each case, the parameters controlling the far-field acoustic pressure amplitude will be identified in an attempt to optimize the system. [Work supported by NSF.]
Fitting of a digital hearing aid using in-the-ear measurement data
Authors: Chang J. S. , Tong Y. C.
Published: October 2005

Acoustic tones were delivered using a hearing aid receiver in the custom earmold of the subject. A probe-tube microphone was used to measure the sound pressure level (SPL) in the ear canal with the earmold and receiver in place. In-the-ear SPLs corresponding to threshold, soft, comfortable, loud, and uncomfortable listening levels were recorded. Using these data, a wearable speech processing hearing aid (WSPHA) was programmed as a linear amplifier for two subjects with rapidly sloping profound high-frequency hearing loss. The WSPHA was implemented using a 24-bit digital signal processor (DSP56001) with 14-bit analog-to-digital and digital-to-analog converters. The WSPHA is adjustable to within 1 dB of any specified frequency response. The fitting of the WSPHA was based on matching the long-term speech spectrum to the contour of the comfortable listening level. The maximum power output was programmed to be below the uncomfortable listening level. Preliminary speech perception results indicated that the subjects' performance for WSPHA was better than for their own conventional hearing aids fitted on the basis of the NAL procedure.
Enhancement of speech intelligibility for hearing-impaired subjects for a telelphone bandwidth
Authors: Terry Mark , Bright Kathryn E. , Sweetman Richard H. , Kepler Laura , Krauss Tom , Durian Michael
Published: October 2005

Speech intelligibility scores from hearing-impaired subjects were evaluated using a computerized version of the California consonant test. The speech signal was presented through a 300-to 3000-Hz bandpass filter (to simulate a telephone signal) via earphones. Each subject was tested with an unprocessed signal, with a signal that was amplified and equalized to compensate for the individual's hearing loss, and with a signal that was amplified, equalized, and compressed using compressor compression. Subjects were tested at three sound pressure levels above threshold. Two modes of digital processing were examined: processing in the frequency domain, that is, modification of the short-term spectrum obtained through an FFT; and processing in the time domain, that is, passing the signal through a bank of bandpass FIR digital filters whose gains had been adjusted to fit the hearing loss of the subject. Results for 12 subjects with sensorineural hearing loss show significant improvement in intelligibility scores (15%–30%) using both processing techniques. [Work supported by US West.]
Resonant acoustic scattering as a singular perturbation phenomenon
Authors: Norris Andrew
Published: October 2005

The resonant scattering of acoustic waves from very rigid or dense elastic targets can be viewed as a singular perturbation from the case of a perfectly rigid scatterer that displays no interior resonances. The method of matched asymptotic expansions is used to develop the scattered field as a combination of the rigid background with an “inner” solution valid for frequencies close to the modal frequencies of the target in vacuo. The general form of the inner solution shows that the strength of the response at a given modal frequency is very sensitive to the spatial distribution of the mode, with the strongest resonances occurring if the modal displacement is confined to the vicinity of the exterior surface. These general conclusions are illustrated for a spherical target, and a comparison of the matched asymptotic results with the predictions of the resonant scattering theory for the sphere sheds some light on the asymptotic validity of the latter.
The thin-shape breakdown (TSB) of the Helmholtz integral equation
Authors: Martinez R.
Published: October 2005

A number of numerical implementations of the Helmholtz integral equation exist today that can predict routinely the field scattered by a volume-holding body, such as the ellipsoidal “core” of a typical airborne or submerged vehicle stripped of its thin appendages, i.e., stripped of control surfaces, etc. The reason for these exclusions has often been an inherent limitation of the cited modeling tools, rather than a rational dismissal of the potential effect of the neglected protrusions on the complete body's expected scattering cross section. The limitation of existing techniques is this: The standard form of Gauss' theorem on which they are based, which leads to the Helmholtz integral, becomes meaningless when the volume of the shape addressed tapers down to zero even over only part of the structure. This paper explains, analytically, the origin of this thin-shape breakdown (TSB), and develops an alternate boundary element formulation for its cure.
Critical band distortion analysis: Part II
Authors: Cudahy Edward , Levitt Harry , Link Cyndi , Jiang Wei Hua
Published: October 2005

Distortion analysis typically focuses on the physical measurement of distortion. There is little information on the relation between these measurements and perception. A series of studies has been conducted as part of the development of a general method for measuring distortion, based on a simple critical band model of hearing, that links physical measurement with the perception of distortion. Results from both detection and ranking experiments using paired comparison measures of speech processed through a hearing aid will be reported. These results suggest that a critical band analysis of distortion can be useful for relating physical and perceptual measures of distortion, at least for simple stimuli. The limitations of the measure will also be discussed. [Research supported by NIDRR.]
The removal of noise from a voicelike signal
Authors: Vaidya P. G. , He Rong
Published: October 2005

This paper deals with a technique to remove noise from a voicelike signal. It is assumed that the listening device has attained a certain familiarity with the source of the signal. In fact, the first part of the paper deals with certain measurements to be performed to identify and measure some of the characteristics of the source. Once these characteristics are determined, the source produces a sequence of signals (most of which were not used during the familiarization process) that are mixed with noise. The task now is to remove the noise and identify the signals. A model of voice production (described at an earlier meeting of the ASA) has been used, which begins with a nonlinear section followed by a linear section. The nonlinear section characterizes the source and the linear section determines various vowellike sounds. It has been assumed in this paper that the nonlinear section does not change (which would hold the pitch constant) and that the linear section is characterized by two unknown formants. Within the limitations imposed by this rather simplistic model, a large amount of noise can be removed from the signal.
Hearing acoustic details of speech. II. Detection of temporal gaps in frication noise
Authors: Stein Ramona , Espinoza-Varas Blas
Published: October 2005

This paper focuses on the detection of temporal gaps in the frication noise of the syllable /t∫a/. A token of /t∫a/ produced by a female talker was digitized at 20 kHz, output via a D/A converter and presented at ∼65 dB SL on each interval of a 2IFC task. In one randomly chosen interval, the syllable contained a 1–20-ms temporal gap at the temporal center of the 183-ms frication noise. Gaps were obtained by multiplying the corresponding sample points by (1 − k). Values of k were 0.8 and 1.0 for partial and silent gaps [Forrest and Green, J. Acoust. Soc. Am. 82 1933–1943 (1987)]. Three normal-hearing listeners were asked to decide which interval contained a (silent or a partially filled) temporal gap. Gap detection was studied: (a) for frication noise excerpted from the syllable and presented in isolation; and (b) for the same frication noise presented in syllabic context. Gap-detection thresholds were interpolated at the 75%-correct level from probits fitted to 7–8 point psychometric functions. Thresholds in isolation were 3.5 and 4.0 ms, for k of 1.0 and 0.8. Thresholds in context were similar to those in isolation. Gap thresholds for the final position fricative in the word “watch” /wat∫/ will be described also. [Work supported by OCAST Grant HSO-005 and Presbyterian Health Foundation.]
Characterization of the surface velocity distribution using the angular spectrum method
Authors: Lewin Peter A. , Schafer Mark E.
Published: October 2005

A field expansion technique developed for analyzing the acoustic fields and vibrating structures will be presented. The technique is based on the angular spectrum method of wave-field analysis. In this approach, acoustic propagation between parallel planar surfaces is modeled using the two-dimensional Fourier transform of the wave field, with each element in the spatial frequency domain multiplied by the appropriate phase factor. The technique developed is applicable to both monochromatic and wideband pulsed waves, and allows the effects of acoustic parameters such as absorption, dispersion, refraction, and phase distortion to be accounted for. The backpropagation results demonstrate the ability of the extended angular spectrum method to reconstruct the surface velocity distribution of complex acoustic radiators. The forward propagation results illustrate the effects of transmission through layers with different propagation properties. The application of the angular spectrum method to visualize vibrating structures and to analyze their features will be demonstrated. Also, the use of the technique in both characterization of acoustic fields and optimization of acoustic sources design will be discussed. [This work was partially supported by the NSF Grant BMS-899044579, and the Advanced Technol. Ctr. of Southern Pennsylvania, Ben Franklin Partnership.]
Near-field acoustical holography in an enclosed space
Authors: Konzelman Charles J. , Burroughs Courtney B.
Published: October 2005

In wind and water tunnels, the interaction of flow with rotating and nonrotating structures is a source of radiated noise. To characterize both the source and noise field radiated by sources operating in wind or water tunnels, near-field acoustical holography algorithms for the nonintrusive reconstruction of the velocity of the source and the complete acoustic field radiated by the source into a free field are presented for use in an enclosed cylindrical space, such as that often encountered in wind and water tunnels. Two reconstruction algorithms have been developed, both of which account for reflections from the cylindrical walls that surround the source. The first algorithm models the space as an infinite open duct and the second models the space as a closed duct where measurements on the ends of the duct are included in the simulation. Acoustic fields inside the ducts are simulated for different sources. These simulations are used to model a series of measurements taken on the holography measurement surfaces. From these simulated measurements, known sources are reconstructed and the acoustic radiation field, from the surface of the source to the far field, is computed. The sensitivity of the algorithms to the type of source, measurement configuration, measurement errors, and background noise are investigated. Examples of near-field acoustical holography reconstructions in enclosed cylindrical spaces are presented for different types of sources.
Time-based energy analysis of acoustic radiation and structural vibration using generalized near-field acoustical holography measurements
Authors: Mann J. Adin , Williams Earl G. , Washburn Karl B. , Grosh Karl
Published: October 2005

A time-based energy analysis of structural vibration and radiation is carried out with near-field acoustical holography measurements. By using the force driving the structure as a reference for the measured pressure, the impulse response can be determined over the frequency range of the force signal. Assuming that the signals are linear and shift invariant in time, the structural vibration and radiated acoustic field due to a synthetic force are determined. This is done by multiplying the frequency domain descriptions of the measured impulse response and the synthetic force. The signal processing, which will be outlined, occurs in the frequency domain. The results are then transformed into the time domain via discrete Fourier transforms where the analysis is done. Thus, from a single measurement, the response to force time histories other than the original measurement force can be studied, assuming that the new force signal can be described with the frequency range of the original measurement. Results for a fluid-loaded, point driven finite cylindrical shell will be shown, with emphasis placed on examining the energy radiated to the acoustic far field, the energy in the acoustic near field, the energy delivered by the driver to the structure, and the energy absorbed by damping in the structure. For finite time length forces, a large amount of acoustic energy enters the structure rather than radiating to the far field. This time-based analysis provides many insights into the mechanisms of fluid/structure interactions that are not observable by a purely frequency based analysis.
Development of a speech intelligibility test for hearing aid research
Authors: Nilsson Michael , Sullivan Jean , Soli Sigfrid D.
Published: October 2005

An English language speech comprehension test has been developed to address shortcomings of current speech tests as measures of hearing handicap [R. Plomp, J. Acoust. Soc. Am. 63, 533–549 (1979)]. The Bamford-Kowal-Bench sentences, as revised to conform to American English usage, were modified to create 28 lists of 12 sentences of approximately equivalent length, difficulty, and phonetic content. These materials were then sampled at 20 kHz with 8-kHz bandwidth for measurement, editing, and analysis. Following procedures used with Dutch materials [R. Plomp and A. M. Mimpen, Audiology 18, 43–52 (1979)], Leq for each sentence was matched as closely as possible. Speech reception thresholds were measured in the presence of masking noise that matched the average long-term spectrum of all sentences as a means of norming the difficulty of each sentence list. Results of the norming tests will be presented and the use of these materials in hearing aid research will be discussed.
Temporal and spectral cue use for initial-stop voicing perception by hearing-impaired and normal-hearing children and adults
Authors: Hazan Valerie L. , Holden-Pitt Lisa D. , Revoile Sally G. , Edward Donna M. , Droge Janet
Published: October 2005

VOT cue use was assessed using a DAD-TAD stimulus continuum developed from computer-edited spoken [tæd] and [dæd]. Within continua, stimuli contained progressively longer durations of the [t] burst extracted from [tæd]; use of vowel onset cues was assessed via continua differing for the vowel stem (from [tæd] or [dæd]) to which these bursts of progressively increased duration were appended. Identification of these stimuli was assessed in hearing-impaired children (n = 16) and adults (n = 20), and normally hearing children (n = 10) and adults (n = 5). Of the hearing-impaired, only 6 children and 15 adults correctly identified the endpoints of the VOT continua; for these listeners, voicing boundaries assumed longer VOT values than for normal-hearing listeners. The width of the phonemic boundary region widened from the normal-hearing adults to the hearing-impaired children. Cues in the vowel did appear to affect VOT boundaries for all listener groups, but less for the normal-hearing children and hearing-impaired adults than for the normal-hearing adults, and least for the hearing-impaired children.
Perceptual features for normal-hearing listeners' confusions of initial and medici consonants in a reverberant lecture hall
Authors: Danhauer Jeffrey L. , Johnson Carole E.
Published: October 2005

Twelve normal-hearing listeners responded to a nonsense syllabic test consisting of 25 consonant-vowel-consonant-vowel (CVCV) items while seated at a 0-deg azimuth to and at three distances (i.e., 1 m, 4 m, and 8 m) from a loudspeaker within a 415-m3 lecture hall with T = 1.9 s, S/N > + 17 dB, and an estimated critical distance of 3 m. Consonant scores were significantly better within the critical distance (1 m) than at distances beyond the critical distance (4 and 8 m), which were not significantly different from each other. Subjects' confusions of initial and medial consonants were submitted to a sequential information analysis (SINFA) to document a priori perceptual features. More information was transmitted within than beyond the critical distance for both consonant positions. However, more information was transmitted for initial consonants than for medial consonants. Further, the relative effects of masking by reverberation on perceptual features was more evident for medial consonants than for initial consonants. Beyond the critical distance, high-frequency features (i.e., sibilancy) were transmitted somewhat better than those based more on low-frequency energy (i.e., voicing). [Work supported by UC General Research Grant ♯8587529-19900-7.]
Statistical distributions of speech for various languages
Authors: Pavlovic Chaslav V. , Rossi Mario , Espesser Robert
Published: October 2005

Statistical distributions of speech spectral energy were obtained for Danish, German, Italian, English, and French. The variable whose distribution was analyzed is termed “perceived spectral energy.” It represents the energy of speech contained within a band 1 Hz wide at the output of an exponential time window. Various window time constants were used in the analysis. Speech material was obtained from the EUROM. O database. The results indicate that there are no significant differences between the languages. Therefore, the normative values can be specified across the languages. However, significant sex differences are found and norms are given for each sex. Finally, these distributions will be compared to those of synthetic speech. [Work supported by a grant from the EEC Esprit SAM project.] Also at the Speech and Hearing Ctr., Univ. of Iowa, Iowa City, IA 52242.
Listeners' judgments of intelligibility of normal and Lombard speech
Authors: Junqua J. C.
Published: October 2005

Noise is one of the factors that influence speech intelligibility. The listening conditions, the context, the speaker, or the listener can also affect speech intelligibility. In presence of noise, speech is masked and its production is modified by what is called the Lombard reflex. The purpose of this work is to study the influence of psychoacoustic and psycholinguistic factors on speech produced in noise (Lombard speech) and normal speech. Through a series of perceptual tests the influence of the following factors on the judgments of intelligibility of normal and Lombard speech is investigated: (1) nature of the vocabulary; (2) size of the vocabulary; (3) sex of the speakers; (4) speech quality evaluation measure (global and segmental signal-to-noise ratio); (5) type of noise (white Gaussian noise and unintelligible speech (babble noise)]; and (6) ability of the listener to perform the task. A 49-work database (alphanumeric plus control words produced by ten speakers) was used, for which some subsets can be classified as similar or not acoustically. The main conclusions of this work are the following. (1) The type of noise is an important factor that influences the intelligibility. Babble noise decreases the intelligibility compared to white Gaussian noise. (2) When additive babble noise is used, Lombard speech is more intelligible than normal speech. (3) When additive white noise is used, Lombard speech was found to be less intelligible than normal speech for the E-SET vocabulary. Results about the influence of the different factors on the intelligibility of normal and Lombard speech will be presented and discussed.
The effects of phonetic training on the identification of normal and articulatory disordered children's naturally produced glide sounds
Authors: Ohde Ralph N. , Eisen Stephanie L.
Published: October 2005

One type of articulation disorder is a sound distortion that has been defined as an allophonic variation within the perceptual boundary of a target phoneme. In order to determine the effects of phonetic training on the identification of naturally produced glide variants, 12 speech pathology students identified normal and articulatory disordered children's /w/ and /r/ productions before and after a course in narrow transcription. In addition, 12 students with no training in phonetics identified the same productions twice with the time interval between sessions comparable to that for the speech pathologists. The speech sample consisted of productions from a normal child and five articulatory disordered children characterized as producing either /w/ for /r/ or distorted /r/. Although the results showed that standard phonetic training improved distorted /r/ identification in some vowel contexts, the inherent phonetic ability of some subjects was a critical factor in the identification of sound variants. [Work supported by NIH, DC00464, and BRSG, RR-05424.]
Four-year-olds' perception of nonnative contrasts differing in phonological assimilation
Authors: Insabella Glendessa , Best Catherine T.
Published: October 2005

Young infants discriminate most native and nonnative phonetic contrasts. Adults, however, often have difficulty with nonnative contrasts, as do infants by 10–12 months [Werker, Am. Sci. 77, 54–59 (1989)]. Best et al. proposed that phonological development fosters perceptual assimilation of nonnative contrasts to the most similar native phoneme(s), according to four possible patterns [J. Exp. Psychol.: Human Percept. Perf. 14, 345–360 (1988)]: (1) Both phones assimilate to a single category (SC); (2) they differ in category goodness (CG); (3) they assimilate to two categories (TC); or (4) they are nonassimilable (NA). Only SC contrasts should pose difficulty for phonologically sophisticated listeners. Recently, however, 10–12 month olds failed to discriminate nonnative CG and TC contrasts, which were discriminated by adults and younger infants [Best, Int. Conf. Infant Studies (April 1990)]. The present study examined further perceptual development in 4 year olds, who discriminated a native control contrast and the CG contrast, but not the TC contrast. Two possible explanations will be discussed: (1) still-immature perception of phonological contrast; or (2) specific difficulty with the fricative voicing feature employed in the TC contrast. In either case, the results suggest that 4 year olds do not yet show mature phonological assimilation. [Work supported by NIH Grant DC-00403.]
Semantic activation using prosodic variables: An electrophysiological test
Authors: Pratarelli Marc E.
Published: October 2005

Event-related brain potentials (ERPs) were employed to assess whether meaningful distinctions based on prosodic variations rather than lexical/phonetic variations result in similar patterns of cortical activations. Subjects matched pictures with auditory presentations of compound words in a reaction time paradigm while ERPs were collected from over various midline and lateral scalp locations. Prosodic experimental conditions were as follows: Subjects would view a picture of a “greenhouse” and respond after the presentation of the word greenhouse (a match condition), or “green house” (a mismatch condition). Similarly, the picture of a green house could be paired with the correct match (green house) or mismatch (greenhouse). Another set of subjects received the reverse conditions wherein the words preceded (primed) the pictures that were similarly responded to. Both behavioral and electrophysiological results showed that semantic priming using phonologically identical words differing only in their stress form results in similar patterns of scale-recorded neural activations, namely, the N400 ERP. These results strongly support the notion that semantic priming regardless of the form in which it takes place (e.g., picture priming, or word priming, or prosodic priming, etc.) is conducted by a specialized system concerned only with priming meaningful relations and not with their physical forms.
Results of tonotopic and echo delay-tuning mapping of the auditory cortex of the Big Brown Bat
Authors: Ferragamo M. , Dear S. P. , Haresign T. , Fritz J. , Moss C. F. , Simmons J. A.
Published: October 2005

Characterization of the response properties of neurons in the auditory cortex of Eptesicus fuscus is an initial step toward defining the neural mechanisms that mediate echolocation in FM emitting bats. Multiple- and single-unit recordings from the cortex were obtained during multiple presentation of pure tones, FM sweeps, and FM sweeps in pulse-echo pairs decreasing in delay from 40 to 0 ms. The existence of a distribution of high to low frequencies from the anterior to posterior poles of auditory cortex suggests a tonotopic map. The frequencies comprising the first harmonic of the echolocation emission appear to be magnified in their cortical representation. Frequency tuning and onset latencies in response to FM sweeps with a 20-kHz bandwidth are similar to responses to pure tones. While the majority of cells sampled are not echo delay tuned, delay-tuned cells have been found with best delays ranging from approximately 3–30 ms. These neurons cluster into two distinct groups suggesting regions of functional subdivision.
Feedback reduction in a digital hearing aid
Authors: Krokstad Asbjørn , Svean Jarle
Published: October 2005

A hearing aid with a partially open ear canal provides improved comfort and sound quality at the cost of increased acoustic feedback. Power supply limitations exclude the use of sophisticated adaptive cancellation methods. Two power-saving feedback reduction techniques are used in our concha bearing aid. (a) Two microphone cancellation system: a microphone close to the vent opening generates the cancellation signal. The necessary time delay is obtained by delayed sampling. Only one AD converter is used. (b) Allpass digital filters for phase correction: The phase shift in the feedback loop of a concha hearing aid using high-quality transducers is so small that the feedback may be partially canceled by digital filters in the forward signal path. Results from measurements and simulations will be presented demonstrating that digital allpass filters of low complexity may increase the stable insertion gain considerably.
Interaction of context and target uncertainty in the discrimination of random tone sequences
Authors: Raman Indira M. , Lutfi Robert A.
Published: October 2005

Two experiments examined the effects of context and target uncertainty on the discrimination of brief (300-ms) ten-tone sequences. On each trial, the listener heard a randomly generated pair of sequences; the target was identified as the sixth tone in each pair with remaining tones as context. The frequencies of the tones were selected at random in the first experiment. Context tones were drawn from a single normal distribution with μc = 2000 Hz and σc = 50, 100, or 200 Hz. The two target tones were drawn from two normal distributions differing in means μt2>μt1 centered around 2000 Hz. The standard deviation of target distributions was σt = 50, 100, or 200 Hz. In the second experiment, the frequencies of the tones were fixed at 2000 Hz and the levels of the tones were selected at random. For context tones, μc = 65 dB SPL and σc = 1, 5, or 10 dB SPL. For target tones, μt2>μt1 = 65 dB SPL, and σt = 1, 5, or 10 dB. In both experiments, the listener's task was to select the sequence for which the target was drawn from the distribution with the greater mean, μt2. Performance of four listeners was compared to that of a theoretical ideal observer which, for all conditions, was constant. Listener performance appeared to vary monotonically with σt/σc, approaching ideal performance near σt/σc = 10. The data are consistent with a model in which listeners weigh multiple observations nonoptimally.
An information-theoretic analysis of tone-in-noise masking
Authors: Lutfi Robert A.
Published: October 2005

It is common to assume that the masking effectiveness of a noise on a tone is determined by noise energy falling in close spectral or temporal proximity to the tone. Another potential source of masking, however, is that resulting from uncertainty associated with trial-to-trial variation in the noise waveform, so-called informational masking. How much informational masking might exist in traditional tone-in-noise masking experiments? An estimate is derived based on a theoretical analysis of data from studies in which two equally effective noise maskers are combined. A weak version of Weber's law is shown to yield a scalar relationship in which masker uncertainty is proportional to the amount of masking in dB. The combined masking data are well described by this relation if it is assumed that 22% of tone-in-noise masking is informational. Implications for other tone-in-noise masking results are discussed. [Work supported by AFOSR and ONR.]
Post-operative fitting of the Nucleus cochlear implant: Bandwidth considerations
Authors: Takahashi Gail A. , Abbas Paul J.
Published: October 2005

Two different Nucleus F0F1F2 processor maps were compared, one in which equally spaced bandwidths were used and one in which bandwidths were based on each individual's psychophysical judgments. The aim of the study was to see whether the use of processor maps with individually tailored bandwidth specifications would result in improved consonant, vowel, and word recognition scores. Four subjects listened to various pairs of loudness-balanced stimuli and made judgments about the dissimilarity of each pair. Each stimulus pair consisted of sequential stimulation of adjacent electrodes in the individual's electrode array. The judgments were used to set the bandwidths associated with each electrode in the subject's active electrode array. Subjects had at least 1-month experience with each map before the speech tests were administered. The bandwidth changes made in the present study did not have a significant effect on speech perception scores. However, the two better users preferred the map with the altered bandwidths over their original maps. [Work supported by NIH.]
Experimental estimation of parameters used in a limit-cycle-oscillator model of spontaneous otoacoustic emissions: Effects of aspirin administration on time constants for suppression
Authors: Murphy W. J. , Talmadge C. L. , Tubis A. , Long G. R.
Published: October 2005

Limit-cycle-oscillator models of spontaneous otoacoustic emissions have been used to provide qualitative descriptions of the interactions of spontaneous emissions with external tones and their response to aspirin administration. The parameters of several classes of such models are estimated from data on the relaxation dynamics of spontaneous emissions, associated with the onset of, and recovery from, suppression by external tones. Aspirin administration is used to control the emission levels, with the relaxation time being observed to increase with decreasing emission level. The models considered include a modified Van der Pol oscillator, with the possibility of linear passive damping in the limit of large displacements, and an oscillator with nonlinear time-delayed elasticity. [Work supported by NIDCD.]
The hearing of Bedouin living in the Jordan Desert
Authors: Arpini A.
Published: October 2005

In industrialized countries, noise exposure is a well-known cause of hearing loss, mainly affecting higher frequencies of pure-tone audiograms. It is important to know the influence of surrounding noise exposure on the presbyacoustics phenomenon as a factor of early aging on hearing. Many authors have studied the relationships between age-related hearing loss and exposure to environmental noise, but not many have studied people living in areas without noise or with low noise. Rosen et al. (1962) studied hearing thresholds of the Mabaan tribe in Sudan as a function of age; in 1981, the MRC Inst. of Hearing Res.—Nottingham conducted a study in Gambia concerning the role of blood pressure as a possible secondary influence on hearing impairment. Other studies are not reported in the literature. In this work, results are reported of the research conducted in the Jordan Desert among the Bedouin people because of the low-environmental noise in this area. For this reason the study was carried out using a special mobil unit equipped for monitoring the noise and hearing of 208 ears. No differences were found between male- and female-related hearing threshold, showing good conservation of hearing compared to people living in industrialized countries.
Measurements of internal-wave effects on 1000-km pulse propagation
Authors: Flatté Stanley M. , Colosi John , Duda Timothy F. , Rovner Galina
Published: October 2005

Wave fronts propagated over 1000 km in the North Pacific Ocean in July 1989 are analyzed for internal-wave effects. Approximately 1 week of pulses emitted once an hour and 20 h of pulses every 10 min are available from the SLICE89 experiment; the pulse width is about 12 ms. The receiver consisted of a vertical array of 50 hydrophones at 60-m separations. The pattern of arriving wave fronts fits globally to expectations from ray theory. Internal waves and other small-scale structures are expected to corrugate the observed wave fronts. Time and space scales of arrival-time variations (wave-front corrugations) will be described. Differences from ray theory caused by internal waves, particularly within the late-arriving part of the pulse, will be discussed. [Work supported by ONR, Code 1125OA.]
Acoustic propagation experiments in a deep lake
Authors: Arveson Paul T. , McKisic J. Michael
Published: October 2005

Acoustic propagation experiments were conducted at Lake Pend Oreille, a deep (350 m) lake in northern Idaho. The goal of the experiments was to quantify the level and seasonal variability of transmission loss fluctuations of acoustic pulses propagated along wholly refracting paths in the lake. Continuous measurements of transmission loss at mid-depth were recorded for approximately 1-week periods during 5 months in 1988-1989. Most measurements were made in the 10- to 40-kHz range, using a fixed bottom-moored projector and bottom-moored hydrophones at ranges of 1 and 2 km. In May 1989, a series of temperature profiles was acquired concurrently with acoustic data over a 5-day period. Acoustic transmission in this short-range, high-frequency regime is subject to weak scattering due to thermal microstructure. The mean temperature at mid-depth is near 4°C with superimposed microstructure. During some periods, the temperature gradients were strong enough to cause several dB of short-term acoustic fluctuations. The acoustic data have been processed to produce numerous received signal statistics including the mean, standard deviation and variance of the received signal level, intensity, logarithmic variance of the received intensity, and intensity scintillation index. This paper will present selected results from the experiments and provide a comparison with stochastic (weak scattering theory) and deterministic modeling approaches. [Work supported by DARPA.]
Fourth-order spectra of filtered intensity modulated sinusoids
Authors: Dwyer Roger F.
Published: October 2005

The sinusoidal frequency of a propagating intensity-modulated sinusoidal signal can be extracted by fourth-order spectra. This has been demonstrated in an underwater transmit/receive experiment. The physical reason for this result is that fourth-order spectra are not degraded by the covariance of the modulating function. Moreover, a special case of the fourth-order cumulant spectrum is independent of the covariance and only depends on the variance of the modulating function. The current study concentrates on the convergence properties of fourth-order spectra and on properties of filtered intensity-modulated sinusoids. The convergence properties are given in terms of the mean and variance of the fourth-order moment and fourth-order cumulant estimators. Then, the properties of filtered intensity-modulated sinusoids are discussed for both fourth-order spectra and the special case.
Forward modeling and data inversion for beam propagation in an arctic underwater channel
Authors: Lu I-Tai , Chen Han-Yang
Published: October 2005

The theory of localizing a beam source [I-Tai Lu, J. Acoust. Soc. Am. Suppl. 1 85, S18 (1989)] is generalized to allow an arbitrarily stratified medium. Numerical results for forward modeling and data inversion of a beam source in an arctic channel are presented. When propagating in a waveguide, an initially collimated beam undergoes diffusion after successive reflections and refractions, and is converted eventually into the oscillatory pattern of one or more guided modes. However, the up-going and down-going plane-wave constituents of a given mode do not have the same excitation strength as in the case of isotropic sources. To generalize an existing modal code for a beam-type source, one needs to decompose each eigenmode into its up-going and down-going constituents and then superpose these constituents with proper spectral strength and phase. For a Gaussian beam source, this procedure can be simplified via the complex source technique. By the analytic continuation of the coordinate of a point source into a complex location, one is able to derive the Green's function in a waveguide due to a beam source with arbitrary width, direction, and location of its waist. Regarding source localization, although the source plane of a beam is not localized in the real space, it is localized in the complex space. (This can be viewed via the complex source point technique.) Therefore, the existing matched-field processes are still useful after proper modifications. The complex source coordinates whose real parts represent the location of the beam waist, and whose imaginary parts represent the width and the direction of the beam, are being sought. If the source is a collimated beam, the procedure can be further simplified into two sequential steps: (1) beam parameter inversion, and (2) beam-waist location inversion. The first step is to determine the direction and width of the beam source by matching the envelope of the mode spectra, which is the spectra of that source in unbounded space. The second step is to determine the beam-waist location by conventional matched-field processing. [Work supported by Joint Services Electronics Program and by the University's Foldes Fund.]
Scholte wave generation by vertically guided waves in steep topographic feature
Authors: Chamuel Jacques R.
Published: October 2005

Interaction of underwater acoustic waves with the Arctic ice cover is least understood when the wavelength is comparable to ice thickness and topographic feature dimensions. Generation and detection of Scholte waves along a flat water/ice interface require proximity of both source and receiver to the ice. Laboratory ultrasonic experimental results are presented demonstrating the generation of pulsed broadband Scholte waves at ice/water interfaces from interaction of remote underwater source with a single steep ice topographic feature. Examples are given from different bonded ice ridges and unattached floating scatterers near the interface. The effects of ridge height, ridge width, and relative source position on the generation of Scholte waves are revealed. Each topographic feature acts as a vertical acoustic waveguide and locally excites the ice/water interface. Focusing of guided waves by ridge geometry enhances the generation of Scholte waves. The results also provide physical insight into the excitation of the Scholte mode by scattering of plane waves from a rough liquid/solid interface [W. A. Kuperman and H. Schmidt, J. Acoust. Soc. Am. 86, 1511–1522 (1989)]. [Work supported by ONR.]
Vertical directionality of ambient noise and ice reflection loss at low (<50 Hz) frequencies
Authors: Yang T. C. , Giellis G.
Published: October 2005

The vertical directionality of ambient noise in a wavegrade such as the Arctic is determined by the weighted contributions of near and distant sources. The weighting is determined by the source strength and the amount of transmission loss which in the Arctic is determined by the ice reflection loss. On a quiet day, i.e., when the local noise is minimal, the vertical directionality of the ambient noise is dominated by the contribution of distant noise sources. In other words, the vertical directionality could not be properly modeled without first modeling the ice reflection loss. A simple analytic expression for the mode intensity of ambient noise is given to relate the mode intensity with the mode arrival angles. This mode spectrum with respect to arrival angle is compared with the vertical directionality data on a quiet day to determine the angular dependence of ice reflection loss at low frequencies.
Perceptual characterization of the room acoustical quality derived from dissimilarity judgments
Authors: Jullien Jean-Pascal , Warusfel Olivier
Published: October 2005

Although acousticians resort to objective criteria when measuring or predicting the acoustical quality of a room, the interpretation of the results still remains critical, because links between measurements and our perception are not completely understood. The room acoustics laboratory of IRCAM has performed a series of perceptual experiments dedicated to this problem. They were based on hypotheses confirmed by previous research [Schroeder, J. Acoust. Soc. Am. 56, 1195–1201 (1974); Lehmann, Acustica 45, 256–268 (1980)]. This perception is structured with a limited number of factors that may be derived from experimental methods. An attempt was made to overcome drawbacks encountered with tests based on ratings on a set of predetermined attributes, or based on preference judgments. Therefore, a nonverbal method was preferred, dealing with dissimilarity judgments collected from paired comparison tests and analyzed with the INDSCAL method [Caroll, Psychometrika 35(3) (1970)]. A sound field simulator with computer interface was designed to build the listening configurations with independent control of objective parameters (acoustical criteria or temporal and spatial distribution of reflections). This method provided (1) coherent representation of the perception of room quality, and (2) structured relations between factors and objective criteria. It also led to the design of an automatic translator between perceptual and objective domains.
Acoustical characteristics of some Serbian Orthodox worship spaces
Authors: Mijié Miomir M.
Published: October 2005

Compared to other Christian communities, the functions of the human voice in services of the Serbian Orthodox church call for a special view of acoustical characteristics of worship spaces in which they are held. This, as well as the fact that no such investigation has yet been undertaken, justifies the analysis of such spaces. Reverberation time and stationary state sound levels have been measured in several Serbian Orthodox worship spaces of different volume and form. Results are discussed from the point of view of worship services.
Evidence of diffuse surface reflection in rooms
Authors: Hodgson Murray
Published: October 2005

In order to determine how to accurately predict the reverberation time and sound propagation (the variation of steady-state level with distance from a source) in rooms, predictions have been compared with measurements for an empty scale-model room and in various nominally empty factories. In the case of rooms with disproportionate dimensions, predictions by the method of images, which accounts for room shape but assumes specular surface reflection, were found to deviate in a consistent way from experiment. The reverberation time is always too high; the short-distance sound propagation tends to be slightly low and the large-distance sound propagation is always too high. This result suggests that diffuse surface reflections were occurring. To test this hypothesis, further predictions were mad.e using a ray-tracing model that accounts for diffuse surface reflections. It is found that excellent prediction accuracy is obtained as follows: (a) scale model—if all surfaces are 10%–10% diffusely reflecting; (b) real factories—if the ceiling and walls are 70%–90% diffusely reflecting.
Evaluation of reverberant room impulse responses, including the effects of head shadow
Authors: Hoffman Michael W. , Link Michael J. , Buckley Kevin M. , Soli Sigfrid D. , Jayaraman Sriram
Published: October 2005

Simulation and analysis of multimicrophone hearing aids in realistic environments require accurate knowledge of room impluse responses at various points on or near the head. A two-step method is presented that provides the room impulse responses for arbitrary locations around the head. The first step uses the method of images [J. B. Allen and D. B. Berkley, J. Acoust. Soc. Am. 65, 943–950 (1979)] to compute the impulse delay and attenuation from each of the various images. The second step factors incorporate the measured responses at various points on the head (or manikin) into the total room/head impulse response. The obtained responses can be used to process speech or other materials to simulate reverberant environments, and to generate array response vectors for multimicrophone hearing aid processors. These vectors can in turn be used to evaluate the sensitivity of the microphone array response to changes in the soundfield and to indicate the level of robustness required for hearing aid processors.
Shock spectra of piano hammers
Authors: Russell Daniel A. , Rossing Thomas D.
Published: October 2005

It has been suggested [I. Bork, Appl. Acoust. 30 (in press)] that an effective way to characterize the frequency dependence of energy transfer efficiency from a piano hammer to a string is by means of the shock spectrum, which describes the vibration amplitudes of a simple oscillating system with a resonance frequency f, either during (initial shock spectrum) or after (residual shock spectrum) an impact. Shock spectra of piano hammers striking a string segment have been recorded by means of a force transducer and an FFT analyzer. Initial results indicate that as the hammer speed increases, the shock spectrum broadens and its maximum shifts systematically upward in frequency (fmax is nearly proportional to the square root of speed for most hammers). In a carefully voiced piano, fmax increases from bass to treble but does not double each octave. Experimental results are compared to various hammer models that include a nonlinear spring characteristic.