bluuebunny/crossref_metadata_2025 huggingface dataset:
Masked loudness functions and their relation to intensity discrimination
Authors: Hellman R. , Scharf B. , Teghtsoonian M. , Teghtsoonian R. , Hellman W.
Published: October 2005

One way to assess the relation between intensity discrimination and the slope of the loudness function is to measure the jnd for a sound that falls on two distinctly different loudness functions. Two such functions were generated by presenting a 1000-Hz tone in narrow-band noise (NBN, 925–1080 Hz) at 70 dB SPL and in wideband noise (WBN, 75–9600 Hz) at 80 dB SPL. Over a range from near threshold to about 75 dB SPL, the tone's loudness function is much steeper in the NBN than in the WBN. At 72 dB SPL, where the two loudness curves cross, we measured the tone's jnd in each noise by a two-interval forced-choice procedure. Despite the differences in slope (and in sensation level), the jnd was the same in both NBN and WBN. The value of ΔI/I was 0.22, which is close to that interpolated from Jesteadt, Wier, and Green [J. Acoust. Soc. Am. 61, 169–176 (1977)] for a 1000-Hz tone at the same loudness in quiet as our 72-dB tone in noise. These and other data suggest the possibility that the size of the jnd for intensity depends more on loudness magnitude than on the slope of the loudness function. [Supported by funds from the Medical Research Service of the VA to RH and by a grant (BMS 73-06944) from NSF to RT.]
Evoked potentials: An automated threshold-tracking procedure using an objective detection criterion
Authors: Ahroon William A. , Saunders Samuel S. , Arnold Sally A. , Salvi Richard J.
Published: October 2005

The estimation of auditory sensitivity or “threshold” using various evoked-potentials generally has been made by visually inspecting a series of waveforms obtained over a range of stimulus intensities. As intensity is decreased, the object waveform gradually disappears into the background of neural activity and the experimenter/clinician must make some arbitrary decision about the intensity region in which this occurs (i.e., threshold). Difficulties can arise in making these judgments; waveform amplitudes may be unstable over time, different observers may have substantial biases which affect the threshold judgments or the range of data that is ultimately collected. This paper illustrates how a relatively simple quantitative technique (correlation) can be used to reliably estimate evoked-potential thresholds in the chinchilla. Evoked potentials were recorded from chronic electrodes implanted in the inferior colliculus. Correlations among prestimulus and poststimulus waveforms were used by a computer-automated threshold-tracking procedure to increase or decrease stimulus intensity. The objective procedure yielded comparable results to visual threshold determinations by highly practiced observers. The practical aspects of using an automated threshold-tracking procedure are discussed.
Auditory OFF and ON evoked potentials have different frequency- and intensity-specific properties
Authors: Henry Kenneth R.
Published: October 2005

An auditory brainstem response (ABR) can be elicited to both the onset and offset of a sufficiently long duration tone. In the laboratory mouse, the onset and offset ABR are similar enough in waveform and latencies to assume a similar source for these two measures. However, the shape of the visual detection level audiograms of onset and offset ABRs are quite different. Onset ABR threshold audiograms reflect behavioral audiograms, having their lowest thresholds at midfrequencies. By contrast, offset ABR thresholds from normally hearing CBA/J mice are highest at midfrequencies. Genetic factors can differentially affect these two responses. The slope and shapes of the onset and offset ABR amplitude input-output functions also differ, as does the effect of frequency on these measures.
Studies of the auditory steady-state response I. Minute-rhythms
Authors: Makeig Scott , Galambos Robert , Stapells David R.
Published: October 2005

A train of discrete stimuli (clicks or tonebursts) delivered to the ear at rates near 40 per second evokes a corresponding train of roughly sinusoidal brain potentials (the 40 Hz ERP, the high-rates or steady-state response, etc.). Since the response lasts as long as the stimulus train continues (minutes, hours) it may be used as an ongoing probe of changes within the auditory CNS. We have been studying both the spontaneous and event-related modulations of the complex response (amplitude, phase) during behavioral experiments (threshold estimation, masking, vigilance, etc.). In resting subjects, quasirhythmic fluctuations in response amplitude and phase occur spontaneously, with periods of 20–120 s (“minute-rhythms”). Amplitudes may vary by 100% or more, especially as the subject drifts into or out of sleep. We have also observed minute-rhythms in responses to visual and tactile stimulation at similar repetition rates, and in concurrent measures of performance and EEG-band amplitudes during a vigilance task. Evidently, minute-rhythms are a natural feature of brain electrophysiology. [Work supported by NIH.]
Studies of the auditory steady-state response III. Threshold estimation
Authors: Stapells David R. , Makeig Scott , Galambos Robert
Published: October 2005

The present studies investigate the use of “phase coherence” of steady-state responses for the estimation of thresholds in normal hearing adults [Makeig and Galambos, J. Acoust. Soc. Am. Suppl. 1 74, S65 (1983)]. In the first study, rate-series data (19/s–54/s) were obtained in ten subjects. Phase coherence was high in the 34–44/s range in all ten subjects, even when their maximum response amplitudes occurred at lower presentation rates. We then recorded responses at near-threshold intensities (39.1/s rate) to two-tone frequencies in six subjects and found significant responses (phase coherence, p < 0.01) within 6 dB of behavioral threshold at 500 Hz (mean 1.0 dB SL) and at 2000 Hz (mean 2.3 dB SL). The recording time necessary to obtain a significant response was then determined in ten subjects. At 25 dB SL, 17 of 20 recordings were significant within 50 s, and all 20 were significant within 7 min. At 10 dB SL, 29 of 40 recordings were significant within 4 min, and 38 of 40 were significant within 15 min (mean-time-to-significance 2.5 min.) These results indicate that the use of steady-state response phase coherence provides good estimates of threshold in adults within a practical recording time for audiometry. [Work supported by NIH.]
Mechanical vibrations and bone strength: A review
Authors: Smith Suzanne D. , Kazarian Leon E. , von Gierke Henning E.
Published: October 2005

The fact that the skeletal system possesses the inherent ability to adapt its architectural array to meet the structural demands placed upon it has intrigued and perplexed scientists since time immemorial. In the past century extensive theoretical and scientific literature has appeared on the stress-oriented structure of bone, and numerous theories have been put forth to explain the phenomenon of bone remodeling and the mode of bone induction under the periodic daily stress exposures varying in direction and intensity. This communication will review the various theories and experimental findings governing our current state of knowledge concerning the role of mechanical stress on bone resorption and deposition. Attention is directed to bone tissue as a material and to bones as structural elements. Bone tissue is described anatomically and from a microstructural point of view. Morphological and histological adaptations due to external static and vibrational loading are presented.
Application of the Articulation Index model to hearing-impaired listeners
Authors: Dirks Donald , Bell Theodore , Humes Larry
Published: October 2005

This presentation will review results from several experiments and retrospective analysis in which the applicability of the Articulation Index (AI) model for characterizing the speech recognition performance of listeners with sensorineural hearing loss was investigated. These investigations were conducted on subjects with a variety of audiometric configurations using nonsense syllables (CUNY-NST) and sentence (SPIN) material. The AI procedure predicted performance for listeners with mild to moderate hearing loss who had potential for high speech recognition scores. For hearing-impaired listeners with moderate or more severe loss, the AI procedure was often a poor predictor of speech recognition. These studies have raised several issues regarding various procedures and measurements recommended in ANSI S3.5-1969. The applicability of the standard AI procedure for hearing-impaired listeners will be explored. [Work supported by NINCDS.]
Application of articulation theory to the study of speech reception by the hearing impaired
Authors: Braida Louis D. , Zurek Patrick M. , Durlach Nathaniel I. , Milner Paul
Published: October 2005

Roughly 40 years have elapsed since articulation theory was applied to the analysis of speech reception measurements on impaired listeners made in conjunction with the “Harvard” and MEDRESCO hearing aid studies. We review three recent applications of articulation theory for similar purposes. In one study, it was used to predict the effect of varying the frequency-gain characteristic of amplification systems used by listeners with sharply sloping high-frequency losses. In a second study, it was used to evaluate the results of high-, low- and bandpass filtering applied to speech for five listeners with flat and sloping losses. In a third study, it was used to predict speech reception in noise for hearing-impaired listeners with a variety of audiograms. The successes and limitations encountered in these applications, and possible modifications of the theory, will be discussed. [Work supported by NIH.]
Factors underlying errors in the recognition of stop consonants by listeners with cochlear hearing impairment
Authors: Dorman Michael F. , Lindholm Julie Mapes , Hannley Maureen
Published: October 2005

In a series of experiments we have assessed the contribution of selected stimulus factors to the recognition of stop consonants by listeners with mild to severe hearing impairments. Errors in recognition do not appear to be conditioned in a significant manner by a high amplitude first formant or by brief duration formant transitions. Errors are conditioned by the spectral tilt at signal onset and by the duration of the steady-state spectrum at signal onset. In severely hearing-impaired listeners, discrimination among the stops appears dependent on the presence of distinctive temporal properties among the signals. In general, our results point to the critical role played by accurate temporal resolution in maintaining accurate speech intelligibility.
The effect of noise-induced hearing loss on speech intelligibility in quiet and in noise
Authors: Smoorenburg G. F. , van Golstein Brouwers W. G. , Mimpen A. M.
Published: October 2005

Following a pilot study (Smoorenburg et al., Scand. Audiol., Suppl. 16, 123–133) based on 44 ears (22 subjects) with noise-induced hearing loss (NIHL) we conducted a study based on 400 ears with different degrees of NIHL. The study included speech reception thresholds for sentences presented in quiet and in noise, tone audiograms measured in quiet and against a low-frequency background noise, and self-rating of the hearing handicap according to the Social Hearing Handicap Index. The results show virtually no correlation between the hearing loss for speech presented in quiet and the loss for speech presented in background noise. Speech thresholds found in quiet show the highest correlation with losses in the tone audiogram at 0.5 and 1 kHz, whereas speech thresholds in noise show the highest correlation with the losses at 2–4 kHz. The speech thresholds will be compared with predictions from the audiogram using the Articulation Index procedure and with the audiograms measured against a low-frequency background noise. Only those questions in the Social Hearing Handicap questionnaire that refer directly to practical situations show a correlation with the speech thresholds. The results for speech perception in noise show that the AAOO fence of 25-dB average hearing loss across 1/2, 1, and 2 kHz implies a severe auditory handicap. The fence should be set at a lower level and it should include audiometric frequencies.
The importance of numeral and vowel perception tests for classifying severely/profoundly deafened young adults
Authors: Webster J. C.
Published: October 2005

Using a six-choice response format six lists of 10 numeral items, 10 vowel items, and 15 initial (IC) and final consonant items were adminstered to 250 NTID students. Analyses showed that students who scored at chance levels on CID sentences obtained meaningful scores on one or more parts of this Webster/Pickett (W/P) Mod 4 test. Error analyses showed word familiarity effects on IC items and place confusions on all non-numeral items. Vowel confusion analyses compared articulatory features, tongue advancement (A), and height (H), with distinctive features: (A), (H), and tenseness (T). No recorded stimulus items included the retroflex (R) dimension but a followup subexperiment is in progress where (R) errors can be made by substituting such words as curt, lurk, etc. for cot, lack, etc. on answer sheets for lists where this is possible. These results and other principles for maximizing the efficiency of this type of test will be discussed.
The response characteristics of auditory neurons: A review
Authors: Ryan Allen F.
Published: October 2005

Auditory neurons typically respond to a wide range of auditory stimuli, with responses which can vary across several stimulus dimensions. This has made classification of individual auditory neurons on a functional basis somewhat difficult. Auditory neuronal responses are therefore often described by the manner in which neural populations appear to encode such stimulus dimensions as frequency, intensity, interaural differences in signal phase or intensity, and signal envelope. Of course, with most neurons encoding several dimensions, these classifications overlap extensively. Auditory neurons also vary in the rate at which they discharge spontaneously, in the absence of external stimuli. These response characteristics can vary dramatically depending upon the state of the organism. A high degree of response plasticity may be observed with changes in attention, behavior, sleep/waking state, or with anesthesia. Due to the complexity of the auditory system, the relationships between auditory neuronal responses and their underlying anatomical substrates are just beginning to be explored. [Supported by NIH/NINCDS and the Research Service of the VA.]
Comparison of vertical line array data taken at two depths in the deep ocean
Authors: Lawrence Mimi Zebrick , Ramsdale Dan J.
Published: October 2005

Vertical line array data were acquired in the Northeast Pacific using a 31-element vertical line array placed with the center of the array first at the sound channel axis and second at a depth approximately equal to the conjugate depth of a cw source, which was towed at a nominal depth of 100 m for ranges up to 600 km from the array. These time series data have been transformed to the frequency domain and beamformed for frequencies near the cw source frequency to determine the signal arrival structure at each array depth. Preliminary results show that there is a strong persistent multipath arrival structure at both depths even at these long ranges. Although this vertical arrival structure exhibits fluctuations with range, three or four strong arrivals are present during most of the data. Qualitative agreement was found between the measured vertical arrival structure and that computed using range-independent acoustic models. Values of array signal gain computed for both depths were very similar, showing an approximate 3.5-dB degradation from that predicted theoretically. [Work supported by Naval Electronic Systems Command, Code 612.]
Matched mode method of passive source localization
Authors: Wilson Gary R. , Vidmar Paul J.
Published: October 2005

Source range and depth localization in a waveguide are accomplished by matching the measured mode amplitudes to predicted mode amplitudes. The matching function is F(r,z) = Σn = 1M an(r0,z0)an*(r,z), where an(r0,z0) is the measured amplitude of the nth mode due to a source at a range r0 and depth z0, an(r,z) is the predicted mode amplitude due to a source at a range r and depth z, and the sum extends over M measured modes. The values of r and z that produce the maximum of the magnitude of F(r,z) are taken to be the estimates of the true range and depth r0 and z0. Simulations were performed for an ocean environment using a range invariant normal mode propagation model. The sound field measured by a vertical string of hydrophones was simulated and a least-squares method was used to estimate the measured mode amplitudes from the simulated hydrophone outputs. The predicted mode amplitudes were computed from the propagation model. The effects of array length, hydrophone spacing, number of modes resolved, and the presence of multiple sources were examined. It was found that accurate ranging could be achieved, but only if the vertical aray extended over essentially the entire water column. It was also found that accurate depth resolution required the measurement of a larger number of modes than was necessary for accurate range resolution. [This work was supported by Naval Sea Systems Command, Code 63R.]
Effects of acoustic multipath interactions on source range and bearing estimation for a focused line array
Authors: Monti Joseph M. , Koenigs Paul D.
Published: October 2005

A computer simulation technique was developed to investigate the effects of acoustic multipath interactions on source range and bearing estimation for a focused line array. To simulate the multipath environment the Genetic Sonar Model [Weinberg, NUSC TD 5971 19 Dec. 1981)] was utilized to modify source signals and emulate oceanic transmissions to individual sensors. Characteristics of a narrow-band frequency domain focused beamformer were implemented incorporating wavefront curvature techniques and a maximum-likelihood estimator. Simulations for sparsely populated colinear arrays were conducted at several frequencies for differing sound speed profiles, bottom loss conditions, and source/receiver configurations. The beamformer output was utilized to obtain relative array pressure response and analyzed to obtain source range and bearing estimation errors. For a given environmental case the results indicate no bearing and relatively small range errors for a source located at broadside for the simulated range and frequency parameters. Bearing and range estimates degrade as the source moves to off-broadside locations. For a given source/receiver geometry the results indicate range and bearing errors can be dependent on the environment due to the resulting complexity of the multipath structure.
Autocorrelation processing in convergence zones
Authors: Hauck Walter S.
Published: October 2005

The sensitivity of first and second convergence zone arrivals to small changes in source location has been examined using two simple analytic models and the NUSC Generic Sonar Model (GSM). All three models indicate that the number of arrivals and the interpath time differences vary greatly with the location of the source in a convergence zone. For single point of inflection sound speed profiles, results obtained using the GSM show the number of first zone arrivals range from 1 to 10, and 1 to 12 in the second convergence zone. Travel time differences vary from less than one millisecond to several hundred milliseconds, with these time differences clustering in well-defined groups. Analysis of GSM produced autocorrelograms show several interpath time differences may occur within a millisecond of each other, requiring received signal bandwidths of several kHz to resolve individual autocorrelogram peaks. [Work supported by NAVSEA PMS 409.]
FM synthesis in musical instruments
Authors: Gatts John , Lewis Don
Published: October 2005

The objective of the presentation will be to explain the development and implementation of FM synthesis in musical instruments. Topics to be covered include: (1) development of FM synthesis; (2) introduction of FM synthesis in musical instruments; (3) explanation and demonstration of principles of FM synthesis; (4) implementation of FM synthesis in current Yamaha instruments: the DX7; (5) demonstration of the musical capabilities of FM synthesis.
Current state of the art as embodied in the Kurzweil 250
Authors: Moog Robert A.
Published: October 2005

The first objective of this presentation is to describe the Kurzweil 250 and to compare it with other electronic keyboard musical instruments. The second objective is to provide some personal projections on where the future lies for electronic keyboard musical instruments.
A low cost digital synthesizer system for music applications and psychoacoustic research
Authors: Maher Robert C. , Scandrett John H. , Crawford Robert E. , Grant Kenneth W.
Published: October 2005

The availability of microcomputers has done little to give a musician or acoustic researcher of modest means access to computer-assisted sound generation. To help remedy this situation, our work has resulted in an integrated digital sound synthesis system using a popular microcomputer and commercially available plug-in synthesizer, all for less then $4000. One software package for music generation and another for psychoacoustic research viii be described. For music, the program VERSATRACS allows versatile manipulation of sound in real time or artificial time. Musical note entry is from a piano-style keyboard. All notes and parameter modifications are held in memory for subsequent playback, editing, and permanent storage on floppy disk. For psychoacoustics, the program AUDIO EDITOR uses frequency-vs-time and amplitude-vs-time histories to control independently each of 16 available digital oscillators. A novel time window representation provides powerful editing features. Sound samples of music and research audio will be presented. [This work is supported partially under a National Science Foundation Graduate Fellowship.]
Real-time music synthesis: The importance of ease and completeness of flexible control
Authors: Gibian G. L. , Harnden E. N.
Published: October 2005

One definition of a real-time music synthesizer is one which allows spontaneous and immediate specification or alteration of all available acoustic parameters, including the configuration or “patch” of module interconnections. This definition does not exclude the ability to use preprogrammed patches, parameter values, note sequences, or stored sounds. In any case, the goal is to reduce the delay time between conceptualization and realization by increasing the ease and completeness of real-time control. Many commercially available synthesizers allow performances of preprogrammed note sequences with real-time selection of preset timbres which may include sampled sounds. Less common is the ability to simultaneously modify several of the stored parameters. Musical examples and circuit diagrams will be presented to illustrate a modest microcomputer-based real-time polyphonic synthesizer using inexpensive VLSI chips. The chips implement repetitive processes in hardware rather than software, generating and modifying sounds according to parameter tables, while functioning in parallel, leaving the computer free to read control-surface inputs in real time.
On the use of personal audio headsets and table radios by employees and students
Authors: Royster Larry H. , Skrainar Stephen F. , Royster Julia Doswell , Berger Elliott H.
Published: October 2005

A study is presently underway to define the potential effects on hearing that might result from employees' and students' use of personal audio headsets or table radios. Data for the employee population was obtained at one industrial site where approximately 50% of the all-female work-force utilizes personal radios. The typical employee TWA is between 85 and 90 dB(A) and hearing protectors are not required. Student utilization of personal radios was investigated by studying their radio volume adjustments in different background noise levels and by sampling the levels at which students play their personal radios on campus. Background information as to the degree of attenuation or amplification of the noise environment resulting from personal audio headsets (radio off) was established for several brands of headsets using KEMAR.
Preliminary analysis of tone-excited two-stream jet velocity decay
Authors: von Glahn U. H.
Published: October 2005

Acoustic research related to jet flows has established that sound, by amplifying the naturally occurring large-scale structures in turbulent shear layers, can cause a more rapid decay of the jet plume velocity and temperature and jet spread. One possible application of this sound-flow interaction phenomenon is to future STOL aircraft that may require modified jet plume characteristics in order to reduce the loads and temperatures on the deflected flaps during takeoff and landing operations. A preliminary analysis is made herein of the tone-excitation effect on the velocity decay of model-scale, two-stream jet plumes. Measured data are correlated in terms of parameters that include excitation sound level and outer-to-inner steam velocity ratio. The effect of plume tone-excitation on farfield jet noise is examined briefly and its implication for large-scale two-stream jets is discussed.
Demonstration of sound power determination using sound intensity measurements
Authors: Pope J. , Rasmussen G.
Published: October 2005

Sound power determination using direct measurement of sound intensity is presently under study by ANSI and other standards organizations. When compared with the conventional methods of sound power determination that are based on sound pressure measurements (e.g., ANSI S1.31–S1.36), the intensity approach offers significant advantages. Precision sound power determinations can be made without a special test chamber, and the result can be as much as several orders of magnitude less sensitive to ambient noise. Also, measurements can be made over any convenient control surface in the nearfield or farfield. After a brief introduction, the authors will demonstrate these advantages by performing a series of sound power determinations on a known source using commercially available equipment.
The target characteristics of submerged bodies
Authors: Dragonette L. R. , Numrich S. K.
Published: October 2005

The problem of identifying a submerged target by its active acoustic response is of major interest in several areas of endeavor, including ocean acoustics and medical applications of ultrasonic detection. The problem itself is complex, involving such issues as the interaction of pressure waves at the solid-fluid interface, the generation of vibrational modes in the target bodies, the presence of secondary scattering bodies, and a host of mathematical difficulties attendant upon solving elastic scattering problems for finite bodies of arbitrary shape. For a number of years, the Naval Research Laboratory has approached the investigation of elastic target characteristics by a unique combination of experimental measurements and theoretical analyses. Exact mathematical models, theoretical formulations that retain the physical detail needed to identify fundamental mechanisms, were developed for simple, canonical problems. Experimental measurements were performed to validate the results and algorithms were constructed to isolate in the experimental measurements the mechanisms under investigation. Measurements were then extended beyond these simple targets to finite bodies that could not easily be handled analytically. Using the physical insights drawn from the exact analytical models and algorithms developed from the experimental evaluations, mechanisms that provide the dominant scattering effects in the more complex targets were identified. The clues to size, shape, and material that can be drawn from the scattering results form the basis for classification concepts which can then be tested on additional experimental targets. Physical mechanisms that have been investigated include the isolation of the geometric background response, the identification of unique resonance behavior, and ping tailoring to uniquely identify particular targets. Examples of this analysis methodology will be shown for a variety of solid and hollow targets.
Analysis of the acoustic scattering from high aspect ratio prolate spheroidal shells of constant thickness
Authors: Green L. H. , Todoroff Douglas G. , Hackman Roger H.
Published: October 2005

Previously, an analysis [Roger H. Hackman and Douglas G. Todoroff, J. Acoust. Soc. Am. Suppl. 1 76, S8 (1984)] of scattering from a solid aluminum cylinder with an aspect ratio of 10:1 was presented based on a spheroidal coordinate T-matrix formalism. We now consider the acoustic scattering from a copper, spheroidal shell with an aspect ratio of 5:1 and a constant shell thickness of 5% of the broadside dimension, immersed in water. Experimental measurements are compared with calculations based on the spheroidal formalism and a physical explanation is provided for the more prominent features. Particular emphasis is placed on associating prominent effects with the elastic waves propagating in the shell.
Target characteristics from high aspect ratio objects
Authors: Werby M. F. , Gilbert K.
Published: October 2005

A systematic study of scattering from submerged objects of high aspect ratio (ratio of length to width) will be presented for a variety of targets over a broad frequency range. The purpose of the study is to determine features characteristic of or peculiar to elongated objects. Bistatic and monostatic angular distributions will be examined as well as target strengths and resonance effects.
Effects of fluid loading on air-backed thin shells
Authors: Green L. H. , Trivett D. H. , Flax L.
Published: October 2005

Previously, we presented [L. H. Green, R. H. Hackman, D. H. Trivett, and L. Flax, J. Acoust. Soc. Am. Suppl. 1 76, S65 (1984)] an analysis of the low-frequency resonances of thin spherical shells. The analysis demonstrated that the proper background to subtract (in order to identify the elastic and fluid responses) is that of a rigid sphere. Some questions were raised about the validity of this approach to very thin shells. In this paper the analysis is performed on an aluminum spherical shell with an inner to outer radius ratio of 0.99 and is extended to include thin cylindrical shells. In both cases, the effects of fluid loading on the in-vacuum modes are discussed.
Experimental implementation of the RST predictions for elastic objects in water
Authors: Tsui C. Y. , Reid G. N. , Gaunaurd G. C.
Published: October 2005

Selected predictions of the Resonance Scattering Theory (RST) were experimentally investigated at NSWC's Hydroacoustic Measurements Facility during the summer of 1984. Elastic cylinders and cylindrical shells of various thicknesses and aspect ratios were placed in a water-filled redwood tank 30 ft in diameter and 20 ft deep, and were insonified by trains of 1-ms sinusoidal pulses. The frequencies were varied between 50 and 500 kHz. The signal from a hydrophone, located between the projector and the objects, was sampled at three judiciously chosen time-gated windows to yield the strengths of the incident wave, of the backscattered echo, and finally of the free-vibration target reradiation. Spectral recordings showed that the penetration of sound energy into the object is maximized when the input frequency matches one of the natural frequencies of the object. This is manifested in large dips in the spectral displays of the reflected echoes. The spectral display of the reradiation plot shows sharp rises at these resonance frequencies. For the chosen object shapes, the experimentally observed phenomena, resonant amplitude, bandwidth, and frequency, match well with the theoretical predictions [G. C. Gaunaurd and D. Brill, J. Acoust. Soc. Am. 75, 1680–1693 (1984)]. The observations, in general, show that: (a) each object returns a characteristic resonant signature, (b) this signature contains shape and composition information about the object, (c) agreement between theory and experiment exists for various simple shapes, and (d) a signal processing method for active classification using this signature is possible.
Categorical tendencies in vowel imitation: II. Natural vowel stimuli
Authors: Repp Bruno H. , Williams David R.
Published: October 2005

From the vocal imitation responses recorded in our first experiment, we selected tokens approximating [u]-[i] and [i]-[æ] continua in each subject's own acoustic vowel production space. The imitation experiment was then repeated with these natural vowels as the stimuli. The results from the two experiments will be compared in this presentation. If the second experiment replicates the response patterns found in the first experiment, then the perceptual processing of synthetic vowels will have been ruled out as one possible source of nonlinear stimulus-response mapping in the imitation task. [Work supported by NICHD Grant HD-01994.]
Notes on the history of vowel theory
Authors: Miller James D.
Published: October 2005

The history of metrics used to describe the vowels will be reviewed. The early arguments concerned whether the absolute or relative pitches of one or more resonances were crucial for their specification. Special attention will be given to the work of R. J. Lloyd of Liverpool who forcefully argued in 1890 that ratios of the pitches of audible resonant peaks characterized the vowels. Arguments and experiments relating to these issues will be traced from the late 1800s to the present. [Supported by NINCDS 03856.]
TTS and psychophysical suppression effect
Authors: Mott John B. , Norton Susan J.
Published: October 2005

Robertson and Johnstone [J. Acoust. Soc. Am. 69, 1096–1098 (1981)] found that in single eighth nerve fibers, exposure to a brief, intense pure-tone stimulus in the region of the suppressor frequency produced changes in two-tone suppression without affecting the frequency threshold tuning curve. We wished to see if a corresponding effect could be observed in human psychophysical performance. Quiet thresholds, forward-masked thresholds, and suppression were measured before and after exposure to brief, intense tonal stimuli. The suppressor frequency was above the masker frequency and the exposure stimulus was at or below the suppressor frequency. Postexposure reductions in suppression were observed only when temporary threshold shift (TTS) occurred at the suppressor frequency. However, changes in suppression do not appear to be the result of a simple attenuation of suppressor or masker level. Pre- and postexposure masked thresholds were not found to be different, despite significant TTS at the masker frequency. [Work supported in part by Sigma Xi.]
Identification and discrimination of azimuth with passive and active hearing protectors
Authors: Howse William R. , Elfner Lloyd F.
Published: October 2005

Three studies are reported in which auditory localization ability is measured with and without hearing protectors. Assessment is made under two methods. Identification is indication on an analog of perceived azimuth of the stimulus. Identification data are presented as response frequency surfaces for 36 stimulus azimuths at 10° intervals. Discrimination measures were made under a two-alternative forced-choice matching paradigm for temporally and spatially disparate stimuli. Data are presented as psychometric functions of proportions of correct responding. Results of these studies indicate that passive circumaural devices degrade localization ability by eliminating pinna cues and by modifying spectral cues. Active devices tested severely disrupted localization and produced sometimes bizarre patterns of responding by eliminating interaural difference cues or by severely distorting signals. Stimuli were long duration broadband noise bursts.
Production of /t/ and /u/ by monolinguals and French-English bilinguals
Authors: Flege J. E.
Published: October 2005

If adult foreign language learners judge the phones found in L2 as belonging to categories established during native language (L1) acquisition, the accuracy with which L2 phones are produced may be limited, and production of L1 phones modified. To test this, the VOT of /t/ and the F2 in /u/ produced by four bilingual groups, and by monolingual speakers of French and English, were examined. Experienced but not relatively inexperienced L2 learners produced phones differently in L1 and L2. Although the L2 learners “approximated” the phonetic norms of L2 to a greater extent as a function of increased experience, none of them perfectly matched monolingual speakers of L2. There was evidence that learning a foreign language affected production of L1 phones with a close counterpart in L2. The /t/ (and to a lesser extent /u/) produced by experienced bilinguals in L1 tended to resembled the /t/ (and /u/) of their L2.
Orchestra pit design considerations
Authors: Benade A. H.
Published: October 2005

The first design priority is to provide good intra-pit communication, permitting the orchestra to coordinate tuning, timing, and loudness. Secondly, performers on stage need clear signals from the pit, to coordinate their efforts with those of the orchestra (the reverse path is important, but the conductor can usually mitigate problems that arise). Finally, the good overall performance achieved by these means must be properly transmitted to the audience. These goals are chiefly met by providing an adequate supply of early reflections (t ≲ 30 ms), each of which is a reasonably coherent version of an originally radiated sound. Designer's resources include reflection, quasicoherent scattering, and absorption via properly oriented surfaces whose dimensions correspond to wavelengths in the musically critical 250 to 1500-Hz range. Electronic aids are risky if attention is not given to the need to assure multiple signal paths. [Work supported by NSF.]
Resonant frequencies of a musical wind instrument
Authors: Dudley J. Duane , Strong William J.
Published: October 2005

We consider four possible ways of defining and calculating resonant frequencies of a closed-open air column in a musical wind instrument: (1) maxima of an input impedance versus frequency curve, (2) maxima of a power input curve, (3) frequencies at which input reactance vanishes (i.e., zero phase angle, where input impedance is purely resistive), and (4) frequencies at which a pressure antinode occurs at the inlet. For many bore shapes (cylinder, trombone bell, etc.) these yield nearly the same frequencies. However there are others such as a simple truncated cone for which these methods give results that differ by several percent. A further complication that occurs with the truncated cone is that the input reactance doesn't vanish after the first few modes, nor does a standing wave occur with a pressure antinode at the inlet for more than the first few modes. Since experimental resonance curves are usually obtained by measuring the inlet pressure amplitude with a constant-flow source, it is concluded that the maximum input impedance criterion is probably the most meaningful for our purposes.
Multiple reflections in simple bore shapes
Authors: Ayers R. Dean , Eliason Lowell J. , Lee Anthony S.
Published: October 2005

The acoustic pulse generator described in the previous paper has been used to examine extended impulse responses in bores of simple geometry with relatively few, distinct discontinuities. These are pressure waveforms at the input end that result from the application of a single volume velocity pulse there. The complex input impedance curve of a bore is just the Fourier transform of its extended impulse response, so that curve can be displayed live on an FFT spectrum analyzer [A. H. Benade, J. Acoust. Soc. Am. Suppl. 1 70, S22 (1981)]. Theoretical impulse responses for individual reflections and transmissions have been convolved repeatedly for comparison with the multiple reflections observed experimentally. This time-domain approach is easy to grasp intuitively, and it sheds a different light on the regimes of oscillation that arise when a bore provides feedback to a nonlinear driver.
Simulating musical instruments using their impulse responses
Authors: Sleator Daniel D.
Published: October 2005

A function of the wooden box of a stringed musical instrument is to couple the vibrations of the string to the air, while filtering these vibrations to improve the sound. If we assume that this is all the box does, and that the box behaves linearly, then we can completely characterize its effect (at a fixed point in space) by an impulse response. An interesting question is whether or not the information necessary to capture the musical characteristics of the box of an instrument is really in the impulse response. I measured the impulse responses of some violins, violas, and cellos by damping the strings, tapping the bridge, and recording the resulting sounds. These were then convolved with digitized measurements of vibrations of bowed violin strings. Blind experiments indicate that most people familiar with the sounds of actual instruments can easily tell which sounds “came” from which instruments. I am developing efficient ways of computing these convolutions so that they can be done cheaply in real time. The applications include making more realistic electric violins, making portable electric cellos, and improving the synthesis of stringed instrument sounds.
Long-range downwind propagation of low-frequency noise
Authors: Willshire W. L.
Published: October 2005

The propagation of low-frequency noise outdoors was studied using the noise of a large (75-m-diam) 4-MW downwind wind turbine. Acoustic measurements were made with low-frequency microphone systems placed on the ground at five downwind sites ranging from 300 to 10 000 m (6.3 mile) away from the wind turbine. Despite high background noise associated with the inherently hostile wind environment found around operating wind turbines, certain wind turbine harmonics were measured with signal to noise ratios exceeding 7 dB at the 6.3-mile site. The wind turbine fundamental was 1 Hz and the wind speed was generally 30 knots at the hub height (75 m) during the acoustic testing. The harmonic levels when plotted versus propagation distance exhibit a 3 dB per doubling of distance divergence. In addition, the lower harmonics were attenuated less than the higher harmonics. The cylindrical spreading and frequency dependence seen in the measured results are indicative of surface wave propagation. Theoretical predictions of a source above a finite impedance boundary are compared to the measured results.
Search techniques applied to middle ear models
Authors: Nassef A. M. , Finch R. D. , Gray L.
Published: October 2005

The response of the middle ear to acoustic signals at the eardrum can be described using a mechanical model [see for instance, Y. Onchi, J. Acoust. Soc. Am. 33, 794–805 (1961)]. The transfer function between the input to the middle ear (pressure at the eardrum) and the output (velocity at the stapes) is obtained for such a model. The appropriate values of masses, spring stiffnesses, and resistance of dampers are still not exactly known. A search technique using the Newton-Raphson method has been used to find the best values of these parameters to fit measured transfer functions [A. R. Moller, J. Acoust. Soc. Am. 35, 1526–1534 (1963)]. Preliminary results for the values of the various parameters using this procedure are compared with previously obtained experimental results. The closeness of the fit of the transfer function is also determined and the effects of modifications of the model are investigated.
Human auditory central nervous system: Visualization of cortical and subcortical centers with regional cerebral blood flow measurements on the PET-VI
Authors: Lauter Judith L. , Herscovitch Peter , Raichle Marcus E.
Published: October 2005

Previous use of regional cerebral blood flow (rCBF) to monitor activity in normal human brains has been largely restricted to observations of centers in the cerebral hemispheres, primarily cortex and basal ganglia. Retrospective analysis of all auditory stimulation studies done to date in our laboratory suggests that with the PET VI, responses to auditory signals can be observed in regions related to at least four levels of the auditory system: upper brainstem, thalamus, primary auditory cortex, and association cortex. Correlations between such responses and variables such as stimulus characteristics and ear of presentation can be visualized and quantified.
Acoustic overstimulation reduces 2f1 − f2 cochlear emissions at low and high primary levels in the cat
Authors: Wiederhold Michael L. , Mahoney Judy W.
Published: October 2005

The cubic distortion product, 2f1 − f2 (DP), was recorded in the cat ear canal as the level of two primary tones (f1 and f2,f1 <f2, L1 = L2) were varied from approximately 40 to 90 dB SPL. Primary frequencies were either 8.0 and 10.4 or 4.0 and 5.2 kHz. The DP level grows slowly at low primary levels and usually exhibits a sharp dip with primaries between 60 and 70 dB SPL. Above the dip, the growth is rapid, approaching a cubic relationship to the primary levels. After stimulation with a 90–100 dB SPL pure tone at f1 for 20–40 min, the auditory-nerve responses to tone pips near f2 are reduced, indicating a threshold shift and sensorineural hearing impairment. In addition, the DP levels are altered at all primary levels. Below the original dip, it is frequently difficult to detect DP after overstimulation. A dip is usually seen at a higher level than before. At the highest primary levels used (90–100 dB SPL), DP levels are also significantly reduced. In similar experiments in the alligator lizard reported by Rosowski [J. J. Rosowski, W. T. Peake, and J. R. White, Hear. Res. 13, 141–158 (1984)], the growth of DP level with increasing primary level was similar to what we sec in the cat, but in the lizard only the DP generated at low levels were affected by overstimulation. We agree with their interpretation that the DP level function indicates several sources of cubic distortion which partially cancel at the dip. However, whereas the high-level component appears to be resistant to overstimulation in the lizard, our data indicate that all components are susceptible to cochlear damage in the cat. [Work supported by VA.]
Relationships among pure-tone pitch anomalies, threshold finestructure, and otoacoustic emissions
Authors: Burns Edward M.
Published: October 2005

A number of studies [e.g., E. Schloth, Acustica 53, 250–256 (1983)] have demonstrated a strong correlation between minima in the behavioral-threshold finestructure and frequencies at which spontaneous or evoked otoacoustic emissions are measured. Other studies have reported a correlation between threshold finestructure and pure-tone pitch anomalies, such as binaural diplacusis [e.g., G. van den Brink, in Frequency Analysis and Periodicity Detection in Hearing, edited by R. Plomp and R. Smoorenberg (Siothoff, Leiden, 19701)]. The above results imply a possible relationship between pure-tone pitch anomalies and otoacoustic emissions. Accordingly, emissions, threshold finestructure (at 10-Hz increments), and diplacusis finestructure (also at 10-Hz increments) were measured in three highly trained listeners. Pitch-intensity functions were also measured at, and in the vicinity of, frequencies at which emissions were measured. Although the expected correlation between threshold finestructure and emission frequencies was found, no simple correlations between threshold finestructure and diplacusis finestructure, or between threshold finestructure and the form of the pitch-intensity functions, were found in any of the listeners.