bluuebunny/crossref_metadata_2025 huggingface dataset:
Improved acoustic coupling between the hearing aid and the telephone
Authors: Arndt H. , Wojick J. J.
Published: October 2005

Technological evolution in telephone-receiver design has necessitated a closer examination of the ways in which hard of hearing telephone users employ their hearing aids. All hearing aids pick up acoustic signals with a microphone. In addition, some hearing aids have a magnetic pickup coil (telecoil) for sensing magnetic fields. Magnetic leakage from telephone receivers is usually a by-product of a particular design and does not exist for some designs and for receivers which incorporate nonmagnetic technologies. In order to maintain a uniform level of access to the telephone for hearing-aid users. one needs to make acoustic coupling the accepted and preferred method of interfacing between a hearing aid and the telephone. Three approaches for acoustic coupling between hearing aids and the telephone are investigated (1) conducting acoustic energy from the receiver-ear cavity to the hearing-aid microphone with an acoustic tube, (2) placing a microphone in the ear cavity (ear mold) and connecting it electrically into the hearing-aid amplification circuitry, and (3) placing the microphone behind the ear in the body of the hearing aid and coupling to it through the pinna. The results are discussed in this paper.
Multilavalier microphone arrangement for teleconferencing
Authors: Curtis T. H. , Mitchell O. M.
Published: October 2005

Microphone arrangements for teleconferencing from rooms which are particularly noisy or reverberent pose difficult problems for the acoustic engineer. The Washington D. C. Room of the General Service Administration Teleconference Network was such a room. Pronounced end-to-end flutter echos were apparent to the casual listener, the ambient noise level was 45 dBA and corridor noise, air-conditioner noise and airplane fly-over noise were present. Using the acoustic guidelines for teleconferencing (Stockbridge et al.) a microphone-talker separation of 12 in. was determined. To satisfy this 12 electret, tie-clasp microphones were mixed together and fed to a teleconferencing terminal. Walnut cabinets were designed to house the mixers and provide table-top storage of the microphones when not in use. The use of voice gates to prevent noise buildup from multiple live microphones was explored and found to be unnecessary. The final system configuration eliminated the problems inherent in this room for teleconferencing.
Process of holograph: a graphic interpretation
Authors: Cohen R. L.
Published: October 2005

Holography is a method of recording and regenerating wave fronts. A motion picture illustrating the fundamental principles of holography was made with the aid of a high-speed interactive computer graphics environment. The concepts presented in the movie are as follows. The interference pattern of a spherical object wave and a planar reference wave is recorded as a hologram. Illuminating this wave-front recording with a plane wave produces three transmitted components, namely, the central diffraction order, the real image, and the virtual image. Huygen's principle is used to show how these components are formed.
Vibratory modal structure of a cantilevered elliptic plate
Authors: Eschenberg K. E.
Published: October 2005

The measured amplitude and phase of the acceleration of the surface of a cantilevered elliptic plate are shown using an interative computer graphics environment. The data are shown as a cross-hatched surface, which slowly oscillates with the measured amplitude and phase, as it is rotated, providing insight into the nature of the mode shapes which cannot be practically obtained by other methods. A short color film will be used to demonstrate the effectiveness of this technique.
Trading relation in perception between silence and spectrum
Authors: Erickson Donna M. , Fitch Hollis L. , Halwes Terry G. , Liberman Alvin M.
Published: October 2005

We have investigated the perceptual effects of independent variation in two quite different cues for the presence or absence of the medial stop consonant in [split] versus [slit]. One is a temporal cue—specifically, the duration of silence between the s noise and the vocalic part of the syllable; the other is spectral—specifically, the presence or absence of those formant transitions that distinguish [plit] and [lit]. A result is that the amount of the temporal cue—the duration of silence—needed to convert [slit] to [split] is about 25 msec less (more) when the spectral cue is present (absent). Thus, there is here a trading relation: 25 msec of silence is equivalent in these phonetic percepts to the spectral difference between [plit] and [lit]. We should wonder what underlines that equivalence. Perhaps it is that the two cues are the common products of the same linguistically significant gesture. [Work supported by NICHD.]
Discrimination of the “tempo of frequency change” cue
Authors: Godfrey John J. , Millay Kathleen K.
Published: October 2005

The stop/glide distinction, e.g., /b/ versus /w/, is cued by the rate of spectral change in the formant transitions [Liberman et al., J. Exp. Psychol. 52, 127–137 (1956)]. A series of 12 stimuli from /be/ to /we/ were synthesized, differing only in the rate of formant frequency change. Five subjects took identification and two-step ABX discrimination tests with these stimuli, resulting in functions typical of “categorical perception.” Other two-step tests altered the discrimination of differences both within and across phoneme boundaries. The three-interval forced-choice (3IFC) or “oddity” test, which makes more demands on memory, produced generally poorer discrimination than ABX, with lower peaks at the boundary. The less demanding four-interval paired test (4IAX) enhanced discrimination of stimulus pairs across and near the boundary, with little effect elsewhere. In another oddity test, each judgement was preceded by repeated presentations of one of the stimuli, which facilitated discrimination markedly both within and across categories. Under these conditions, the discrimination functions of some subjects are not so clearly categorical.
Separability of place and voicing cues for initial consonants in natural speech
Authors: Soli Sigfrid D. , Strange Winifred , Jenkins James J.
Published: October 2005

Acoustic cues in natural speech specifying place and manner of articulation for the initial stop consonants /bdpt/ were examined with a reaction-time task to determine their perceptual separability. In condition one, subjects discriminated pairs of natural CV syllables differing by a single cue, place (e.g., /b/ versus /d/) or voicing. In condition two, syllables differing on both cues redundantly (e.g., /b/ versus /t/ were discriminated, and in condition three, discriminations were based on a single-cue difference although nonrelevant variations in the order cue were also present (e.g., /b/ versus /d/ and /t/). If cues are separable, redundant differences should not facilitate discriminations and decrease reaction times, nor should nonrelevant variations increase. reaction times. For single-cue contrasts, voiced place cues (i.e., /d/ versus /b/) and alveolar voicing cues (i.e., /d/ versus /t/) were best discriminated. Redundant cues facilitated discriminations slightly; however, nonrelevant cue variations did not increase reaction times. The partially separable place and voicing cues are contrasted with the inseparability of cues for place and vowel identity reported by Wood and Day [Percept. Psychophys. 17, 346–350 (1975)]. [Supported by NIMH and NICHD.]
Properties of feature detectors for place of articulation
Authors: Miller Joanne L. , Eimas Peter D. , Root John
Published: October 2005

An experiment was conducted in order to determine whether the sensitivity of a detector tuned to place of articulation is reduced by selective adaptation. The basic stimuli were three syllables, [ba], [da], and [ga]. Each of these syllables was synthesized with F2 and F3 at nine levels of attenuation relative to F1, for a total of 27 stimulus tokens. The task of the subject was to indicate which of the three syllables was heard on each trial. As expected, prior to adaptation the identification of [ba] and [ga] (although not [da]) was a function of attenuation level, with performance systematically declining as F2 and F3 were attenuated. Adaptation with [ba] resulted in a depression of the [ba] response function and similarly, adaptation with [ga] resulted in a depression of the response function for [ga]. Thus after adaptation with a particular syllable, an increase in the critical energy (i.e., less F2−F3 attenuation) was required in order to yield a given level of performance on that syllable. These results strongly suggest that prolonged exposure to a stimulus with a given place value diminishes the sensitivity of the detector tuned to information specifying that value, rendering it a less effective processor of that critical information. [Supported by NIH and NSF grants.]
Occluded ear simulators
Authors: Burkhard M. D. , Zuercher J. C.
Published: October 2005

With the aid of a computer program, accurate models of acoustical elements were used to study and investigate both the Zwislocki-type ear simulators and proposed alternative designs. An occluded ear simulator is the part that remains adjacent to the eardrum when an earmold is inserted into the ear and, in most designs, contains the acoustic networks that establish an equivalent eardrum impedance. The designs were optimized using parametric methods. This led to a better understanding of Zwislocki-type four-resonator structures for simulating real ear eardrum impedance and also to the design of a new two-resonator construction. The agreement between the ear simulator designs (both theoretical and experimental) with available data on ears is discussed. Tentative electrical analog values of constants for the resonators of the new design are as follows: Resonator 1: resistance 1030 ω, compliance 0.13 μF, and inertance 0.16 H. Resonator 2: resistance 300 ω, compliance 0.34 μF, and inertance 0. 0043 H. The canal portion of the occluded ear simulator is a cylinder 12.7 mm long and 7.5 mm in diameter. The external diameter is 19.4 mm. The design avoids placement of resistance elements in the canal simulation portion which would prevent its use for experiments that might require access to this region.
Dual processes for loudness
Authors: Atkinson W. H.
Published: October 2005

A body of anatomical, electrophysiological, psychophysical, and behavioral research supports the proposition that there is a division in audition similar to the rod/cone dichotomy in vision. Evidence will be presented for the presence of discrete processes for low and high intensity sounds. Both can be identified in electrophysiological measurements at all levels of the auditory nervous system. Both processes can also be discerned in magnitude estimates of loudness. If subjective loudness is plotted as a function of the cube root of sound pressure, the curve consists of two linear sections with a discontinuity of slope at approximately 65 dB SPL. Both segments are present for frequencies up to at least 3 kHz. Linearity is preserved in the presence of a masking noise but the variation in slope and intercept constants suggests the operation of different mechanisms of masking for each process.
Pitch strength of filtered ripple noise: evidence for a dominant pitch region?
Authors: Yost William A. , Hill Richard
Published: October 2005

Ripple noise is produced when a broadband noise is delayed (T) and either added to or subtracted from the original noise. The resulting waveform has a power spectrum in which power varies cosinusoidally as a function of frequency. Ripple noise has been used to study pitch perception and echo processing. In the present experiments subjects were asked to discriminate between two noises which differed in the amount of delay introduced (one stimulus contained a delay, T, the second stimulus a delay, T + 0.1T). The discriminability of these two ripple noises was studied as a function of the amount of attenuation added to the delayed noise. Increases in attenuation of the delayed noise decreases the strength of the pitch of ripple noise. Delays (T) of 1, 2, 5, and 10 msec were used to produce ripple noises which were passed through very steep, one-third octave, digital filters centered at 500, 1000, 2000, and 4000 Hz. The pitch of the ripple noise produced with any delay (T) was strongest (i.e., the greatest amount of attenuation was added to the delayed noise before discrimination performance was near threshold) when the center frequency of the filters were at four times 1/T. For center frequencies lower or higher than 4×1/T the pitch was weaker. The pitch of the ripple noise when the delayed noise was substracted from the undelayed noise was weaker than when the delayed noise was added to the undelayed noise. These data will be discussed in terms of recent models of pitch perception and the assumption that pitch is determined in a dominant spectral region. [Work supported by NSF.]
Pitch of dichotically presented amplitude modulated sinusoids
Authors: Robinett L. N. , Nabelek I. V.
Published: October 2005

Periodicity pitch resulting from dichotic presentation of harmonically related tonal complexes generated by amplitude modulation of pure tones was investigated. In the first experiment, AM signals were presented dichotically in such a way that if the individual components from each ear were combined and rank ordered, a single pitch was heard. Otherwise, separate pitches were heard at each ear. Pitch fusion was investigated as a function of frequency spacing between components and frequency region. In the second experiment, loudness balances were attempted at each ear for the periodicity pitch percept as a function of the intensity of tonal complexes. (identical to those used in the first experiment) at the other ear. The purpose of this experiment was to determine if pitch strength of tonal complexes could be increased at one ear by the simultaneous presentation os a complementary complex at the other ear. The results of the experiments are discussed in terms of the time fine-structure theories, the pattern-transformation theory, and the optimum processor theory.
Acoustic signal processing based on the short-time spectrum
Authors: Callahan M. W.
Published: October 2005

This paper discusses modification of acoustic signals by two-dimensional (time/frequency) processing of the short-time Fourier transform. This approach is suggested by time/frequency processing in the human auditory system, and is appropriate for signals such as speech and music. Time/frequency processing has two potential advantages (1) the signal processing domain is similar to the perceptual domain, so that signal modifications are related to perceived changes, and (2) visible displays of the short-time spectrum (spectrograms) can be used as an aid in developing and evaluating specific processes. The mathematical basis for time/frequency processing is developed, and two examples are discussed removal of broadband background noise in old recordings and compression and expansion of speech for transmission through noisy channels. [Work supported by ARPA.]
Computer graphics as an instructional aid in demonstrating acoustical concepts
Authors: Nobile M. A. , Thomas R. W.
Published: October 2005

A short color film will be shown illustrating how computer-generated and animated displays can firmly establish a visual understanding of some fundamental acoustic principles. These displays were produced on a CRT screen by an Adage Model AGT/30 graphics terminal, with color being introduced in the subsequent filming process. The importance of visual imagery in conveying fundamental acoustic principles cannot be exaggerated and tradition has found professors struggling with multicolored chalk and “hand-waving” techniques in an attempt to depict these on a blackboard. Since three-dimensional animation is inherent to the computer graphics package, its introduction into the classroom as support for the lecture material (never as replacement for it) should be welcome. The present film segment explores the concept of simple harmonic motion and is intended for use in an introductory “nontechnical” acoustics course.
Digital video color display for real-time analysis of sodar signals
Authors: Coulter R. L. , Underwood K.
Published: October 2005

Techniques utilizing a digital color monitor and minicomputer for the display of both amplitude and frequency information derived from acoustic scattering in the atmosphere have been developed. Signal processing, including digitizing, is performed on line by a Data General “NOVA” computer. The digital signal, averaged over the desired time interval (height in the atmosphere), is displayed on the color terminal in the familiar height-time section format; intensity variations appearing as color changes, height in the atmosphere proportional to y displacement and elapsed time proportional to x displacement. A roll-up feature allows the continuous display of the last 190 acoustic pulses (typically 16 min). With nine color levels available, the resolution in both amplitude and height is dramatically improved. Similar processing is done with Doppler-frequency information to yield a vertical time section of wind speeds throughout the sensed volume. It may be possible to display the two (amplitude and frequency) simultaneously through use of characters available to the color terminal.
Elicitation of the acoustic reflex with synthetically produced consonant and vowel-like stimuli
Authors: McCarty T. A.
Published: October 2005

Acoustic reflexes were elicited from seven normal-hearing subjects with synthesized isolated consonant and vowel-like stimuli. Stimulus intensity and duration were controlled so that equivalent amplitudes and specific durations were produced. The recorded stimulus items were presented over an earphone to the subject's right ear, while susceptance changes in response to bilateral acoustic reflex contractions were monitored in the subject's contralateral (left) ear. Analysis of variance was performed on the data for acoustic reflex thresholds. No significant difference between isolated consonant and vowel-like reflex threshold values was demonstrated. The main effects of stimulus duration and subjects, and the two-way interaction of phoneme type by subject were found to be significant. [Work supported by NIH.]
Spatial and temporal sampling methods for community noise surveys
Authors: Sutherland L. C. , Stearns John , Pies Don , Konheim A.
Published: October 2005

Spatial and temporal sampling methods from over 40 previous community noise surveys have been reviewed in the course of selecting optimum noise-survey methods for use by local communities. Factors which cause variance in the spatial or temporal samples are considered in developing practical sampling methods which minimize this variance. Results of pilot applications of improved spatial sampling methods developed out of this study are presented for two communities. [Work supported by U. S. Environmental Protection Agency.]
Outdoor noise of fossil-fuel power plants
Authors: Dennison E. E. , Kanistanaux D. C. , Ying S. P.
Published: October 2005

Outdoor noise of several fossil-fuel power plants was measured for varying purposes. Typical continuous noises were generated by water circulation pumps, forced-draft fans and induced-draft fans, emergency diesel engine generators, stacker-reclaimers, SO2 systems, and gas-turbine peaking units. Transient noises included the bumping of coal cars and coal unloading. Another transient noise source was the main steam and cold reheat lines during steam purge. All noises were recorded on magnetic tapes and analyzed using a 1/3-octave band-filter system and in terms of statistical data. Noise propagation was interpreted from topographic conditions, meteorologic effects and sound attenuation factors. The community noise impacts for both continuous noises and transient noises are discussed along with necessary noise controls. One outdoor noise survey of an existing power plant served as a baseline ambient noise measurement for further expansion of the plant. The environmental noise of the expanded plant was also predicted.
Experimental study of environmental effects on truck noise measurements
Authors: Hosier Robert N.
Published: October 2005

A unique experiment has been conducted to identify the dependence of variations in truck noise emission measurement data upon changes in vehicle performance and sound propagation properties. A comparison of data measured at different sites and at different times at the same site by previous investigators [e.g., J.E. Piercy et al., “Mechanisms Causing Variability in Noise Testing of Light Motor Vehicles,” NRC, Ottawa, Ontario (1976)], has shown a variability in the mean maximum A-weighted sound pressure level of up to 4 dB. Little similar data is available for medium and heavy-duty trucks. In this experiment, preliminary data have been obtained for one gasoline and two diesel-fueled trucks performing SAE J366b drivebys, simulated drivebys, and idle-maximum idle runups. The results obtained to date, as part of a one year series of tests, are presented as a function of several of the various performance and environmental variables.
Historical highlights of the development of free-flooded magnetostrictive ring transducers
Authors: Sinsky J. A.
Published: October 2005

The historical highlights of research activity on magnetostrictive free-flooded ring transducers are presented with emphasis on recent work. The significant achievements of the research, their background, and the ongoing materials research that has been stimulated are summarized. They may have to use at some future time the technology of magnetostrictive ring transducers for high-power active surveillance because of changing conditions affecting undersea surveillance. The highlights include the advantages and disadvantages of free-flooded magnetostrictive transducers. A detailed summary of all of the work done on the subject would be lengthy and far beyond the scope of this. Nevertheless, an attempt is made to present an accurate and complete account of some of the more significant contributions to the technology.
Effects of ultrasound on bilayer lipid membranes
Authors: Rohn K. R. , Rooney J. A.
Published: October 2005

The effects of ultrasound of a frequency of 1 MHz on an oxidized cholesterol bilayer lipid membrane (BLM) were observed for the intensity range from 0 to 1.5 W/cm2. In the experimental arrangement the BLM was formed in a 0.9-mm hole in a teflon support positioned in the farfield of a plane-wave sound field. In a typical experiment the BLM was formed in its support separating two chambers containing KCl or NaCl in aqueous solution. The electrical parameters of the BLM were monitored using platinum electrodes on both sides of the teflon support. At an intensity of 1.5 W/cm2 the sound caused the membrane to rupture, in the intensity range from 0.5 to 1.5 W/cm2 it accelerated the draining of the bulk cholesterol from the teflon during the formation of the BLM and at all intensities the sound had no effect on the conductance, capacitance, or the dependence on each on the voltage applied across the membrane. These results will be discussed in relation to determining the effects of ultrasound on nerve membranes.
Auditory evoked responses in neonatal chicks after auditory deprivation
Authors: Saunders J. C. , Houlihan R.
Published: October 2005

At 18 to 19 days of incubation the heads of chick embryos were removed from the egg. Three groups of subjects were formed by injecting a silicone ruber adhesive into either the left, right, or both ear canals. A fourth nontreated group acted as controls. Embryos were returned to the incubator and allowed to hatch normally. Between 24 and 72 hours the chick neonates were anesthesized, the ear plug(s) removed, and the animals were mounted in a head holder. A sound-probe tube system was sealed over the left and right tympanic ring. The neostriatum was exposed to reveal the optic tectum and a concentric electrode was inserted into the right Mesencephalicus Latoralis pars Dorsalis (MLD). Evoked-response thresholds for test frequencies between 0.125 and 5.0 kHz and the relation between stimulus SPL and evoked-response amplitude at 0.25, 0.5, 1.0, and 2.5 kHz were measured in every subject for left- and right-ear stimulation. The results indicate no differences in threshold sensitivity between all sound-deprived and control animals. However, the relation between SPL and evoked-response amplitude at 0.25 and 0.5 kHz was significantly different between deprived and control subjects. The results indicate that an early conductive hearing loss results in abnormal auditory development which is probably located in the nervous system. [Work supported by NSF.]
Development of a clinical procedure for assessment of auditory localization
Authors: Cook W. G. , Frank T.
Published: October 2005

A clinical test of auditory localization ability was administered to 45 normal hearing subjects. The subjects were seated in the center of a circular array of seven ceiling-mounted speaker assemblies. They were required to identify the speaker through which each of a series of 24 presentations of the stimulus “where is this?” was directed. The localization test was performed employing three stimulus presentation levels of 5, 10, and 15 dB SL, re sound-field SRT, and three speaker placements which had radii of 2′7″, 4′, and 5′ from the center of the audiometric test booth. Localization ability improved as the presentation level and speaker placement increased, reaching a maximum mean score of 76% at 15 dB SL and a 5′ speaker radius. The results agreed with previous findings [F. Malpica, M.S. paper (Penn State, 1976) (unpublished)] but did not agree with the results obtained in the initial Investigation of this localization test [G. R. Bienvenue and B.S. Siegenthaler, J. Speech Hear. Disord. 39, 469–477 (1974)].
Role of the acoustic reflex in the acquisition of temporary threshold shift
Authors: Olsen C. C. , Brandt J. F.
Published: October 2005

Acoustic reflex responses were monitored during the presentation of fatiguing stimuli in a temporary threshold shift (TTS) experimental paradigm. This method yielded a measure of TTS and a record of acoustic reflex activity elicited by the fatiguing stimuli. The fatiguing stimuli were interrupted and continuously presented octave noise bands (500–1000 and 1500–3000 Hz) presented at 98 dB SPL. Test frequencies for the low and high band-pass noises were 1500 and 4500 Hz, respectively. Noise exposure conditions having shorter off-times produced significantly more reflex activity over time than did exposures having longer off-times. In addition, greater impedance changes were elicited by the high-band noise exposure than by the low-band exposure. Though the high-band noise exposure provided greater reflex activity it also produced significantly more TTS than did the low-band noise. Differences in the amount of TTS produced by the two noise bands could not be attributed to the effects of acoustic reflex contraction. It may be that under certain stimulus conditions the acoustic reflex plays no role in the acquisition of TTS.
Rate of convergence in adaptive fitting of wearable master hearing aid
Authors: Levitt Harry , Dubno Judy R. , Freeman Lisa , Slosberg Ronald
Published: October 2005

In many adaptive strategies the rate of convergence is, on the average, greatest at the start of testing and gradually fails to zero as the target is approached. Preliminary data on the adaptive fitting of a wearable master hearing aid show the same trend. In particular, the rate of improvement in the subjects' performance (on a nonsense syllable test) was found to be greatest, on the average, during the first test session and then tapered off rapidly during subsequent test sessions. Considering the cost of extended testing, an important practical question is when to terminate testing. The implications of different stopping rules in the adaptive fitting of hearing aids will be discussed. [Research supported by Contract No. NIH-NO1-NS-4-2323 from the National Institute of Health.]
Perception of auditory temporality in several species of birds
Authors: Dooling R. J.
Published: October 2005

Thresholds for the detection of gaps in noise, temporal integration, and difference limens for duration were measured in several species of birds using avoidance conditioning and the method of limits procedures previously described for the parakeet. The threshold for the detection of gaps in white noise bursts is about 3–5 msec for the house finch. Temporal integration functions showing absolute threshold as a function of tone duration for field sparrows and parakeets reveal a “time constant” of 200–300 msec. Difference limens for duration obtained at six durations of a 2.86-kHz tone for the parakeet show Weber fractions, ΔD/D, in the range of 0.10–0.20. These data are compared to similar data available for the human and the results of these experiments are discussed in relation to the characteristics of avain vocalizations and the role of hearing in song learning and song production. [Work supported by NSF grant BNS75-19431.]
Critical bands in the parakeet
Authors: Denny R. M. , Saunders J. C. , Bock G. R.
Published: October 2005

Critical bands (CB's) were measured in the parakeet Melopsittacus undulatus by obtaining masked thresholds of a pure tone in the presence of noise bands of varying bandwidth. Six subjects were tested at each of five tone frequencies (0.315, 0.630, 2.0, 4.0, and 5.0 kHz) and nine masked thresholds were determined for each subject in the presence of variable bandwidths of noise, each with a spectrum level of 30 dB. Two regression lines were fitted to the mean masked threshold values for each bandwidth and the critical band was defined as their point of intersection. The relationship between CB's and tone frequency was found to be comparable with the corresponding relationship seen in previous (mostly mammalian) data. However, the shape of the curve relating CB's to frequency in the present data differs significantly from the shape of the curve relating critical ratios (CR's) to tone frequency in parakeets. Such a difference is not found in other species, where CB and CR functions are quite similar. It is suggested that CR's may not be predictive of CB's in avian species. [Work supported by NSF.]
Influence of semantics on speech perception
Authors: Garnes Sara , Bond Z. S.
Published: October 2005

Our investigation of “The relationship between semantic expectation and acoustic information” [forthcoming in the Proc. Third Int. Phonology Meeting, Vienna (1976)] shows that subjects' perception of the place of articulation of voiced plosives is influenced by the semantic content of a sentence in which those plosives appear. Although subjects do perceive semantically anomalous sentences such as Bring the fishing gear and the date, if the hub of the second formant falls within the cross-over frequencies between, e.g., [b] and [d] in synthesized speech, the subjects report hearing [b] (p < 0.025) which results in the nonanomalous sentence, Bring the fishing gear and the bait. In this study we discuss the results of a current experiment in which natural productions of a variety of anomalous and non-anomalous sentences is masked by two levels of babble. If the responses are similar to those obtained using synthesized speech we expect that as the spoken stimuli become more difficult to perceive, subjects will report hearing fewer anomalous sentences.
Cascade/parallel terminal analog speech synthesizer and a strategy for consonant—vowel synthesis
Authors: Klatt Dennis H.
Published: October 2005

A FORTRAN subroutine has been developed for the synthesis of speech and speech-like waveforms appropriate for psychophysical experimentation. The program accepts, as input, a set of control parameter values every 5 msec and produces, as output, samples of a synthetic speech waveform. Formant resonators simulating the transfer function of the vocal tract may be configured in cascade and/or in parallel (depending on the requirements of the user) by changing the value of the logical switch. There is a minimal change in the output speech when the switch value is changed. The parallel version of the synthesizer is both easier to control and a better model of the vocal tract transfer function than a number of earlier parallel synthesizers. A FORTRAN listing of the synthesizer is available upon request. Soon to be completed is a manuscript that describes simplified control strategies for the synthesis of various consonant—vowel syllables of English. [Work supported by NIH Grant NS 04332.]
Analysis of the performance of a tandem connection of a 16-Kbps CVSD coder to a 2.4-Kbps LPC vocoder
Authors: Rabiner L. R. , Sambur M. R. , Crochiere R. E. , Goodman D. J.
Published: October 2005

In this paper the tandem link of a 16-Kbps CVSD waveform coder and a 2.4-Kbps LPC vocoder are studied. Of prime concern are the effects of the CVSD coder on the LPC vocoder analyzer. In particular the problems involved in making a reliable voiced-unvoiced decision, estimating pitch period, and estimating LPC coefficients from the coder output are studied. It is shown that LPC coefficients estimation from the CVSD output is highly inaccurate. An analytical distortion measure (an LPC distance) is used to show the magnitude of the distortion introduced by the coder as a function of the signal gain into the CVSD coder. Although the remainder of the LPC analysis (i.e., pitch detection, voiced-unvoiced decision, and gain calculation) can be performed reasonably accurately, the magnitude of the distortions in estimating the LPC coefficients is sufficiently large to make the vocoded speech barely intelligible and of poor quality.
Vibrations of a circular rubber membrane
Authors: Anderson C. A.
Published: October 2005

The properties of a vibrating gum rubber dam membrane are compared to those of an ideal membrane. The modes of vibration, obtained by driving the membrane at controlled frequencies in the steady state, are identified by employing chladni patterns and by observing phase differences over the surface with a microphone probe. The frequencies thus obtained are measured at different tensions of the membrane and the wave velocity is calculated and plotted as a function of frequency. The deviations from theory are discussed.
Industrial sewing machine noise
Authors: Bernhard D.
Published: October 2005

A sewing machine manufacturer came to the College of Engineering of the University of Hartford asking for assistance in quieting their latest model of high-speed industrial sewing machines. When the machine was installed in the anechoic chamber, initial investigations using a precision sound-level meter and weighting networks showed the sound level generally to be in the 90-dB range on the A scale. Measurements were taken at 3 ft. from the unit. Noticeably lower levels were observed in the low-frequency range. Further investigation using a real-time analyzer confirmed the initial observations. Readings were taken around the machine in a hemispherical pattern while the machine was in an anechoic chamber. All data observed on the real-time analyzer was recorded on paper charts for detailed analysis. Testing is incomplete at this point, but predominant frequencies have focused attention on certain drive mechanisms. As extensive redesign is not possible at this stage of the machine's development, only minor modifications will be allowed by the manufacturer. Probe microphones and further frequency analysis will be used to determine if redesign of external linkages will be helpful or not. All possible improvements will be outlined and submitted to the designer for future consideration. These improvements will be reported in this paper. [Student Research Project.]
Harvard Underwater Sound Laboratory and the cylindrical array
Authors: Wallace R. H.
Published: October 2005

As early as 1912, an acoustic echo-ranging system for detection of obstacles in the water was proposed by L. R. Richardson. A system was demonstrated by R. A. Fessenden on 25 April 1914. These were search light type systems. For almost a quarter of a century progress was made in improving the components of the system; however, little was done to improve on the low information rate. Early in World War II it became imperative that a high information rate, 360° coverage system be developed. A literature research revealed an interesting German patent that proposed a cylindrical transducer with vertical elements which could be excited independently of one another. Electrical compensating networks could be used to give a unidirectional beam of sensitivity steered to any horizontal direction by means of a commutator. By June 1943 the Harvard Underwater Sound Laboratory (HUSL), under the direction of F. V. Hunt, had attached beam scanning, beam formation, and beam prediction problems relating to the cylindrical array and had developed and demonstrated a working model employing such an array. This HUSL work will be reviewed.
Thirty years' progress in transducer source and receive arrays
Authors: Martin Gordon E.
Published: October 2005

Early emphasis was on low-frequency source arrays. Work in this field led to pioneering computer activities. Interaction impedances involved various “simple” baffle conditions in small or infinite arrays. Operational troubles in arrays led to velocity control concepts. Computerized mathematical models of longitudinal vibrators were developed during the 1960's following advances in dissipative anisotropic piezoelectric devices. The behavior of source amplifiers was included in the matrix representation of amplifier and transducer arrays with electrical and radiation interactions. Using circulant and Toeplitz symmetry, complicated sonar array system analyses were completed with less computer storage and execution time than less realistic methods. Modeling of array structure and baffles has improved by combined Helmholtz integral equation formulations (CHIEF). Finite-element techniques model systems such as source transducer arrays and self-noise in the towed receiving arrays. Lately, increased effort has been on passive-receive-array improvements. The issue of optimal array design is of different character for source and receiver systems due to ambient noise anisotropy. Transducer progress for long-range sonar systems has been more significant than that for commercial loudspeakers and high-frequency sonars.
Design of optimum sonar transducer arrays using goal programming
Authors: Draus S. M. , Ignizio J. P. , Wilson G. L.
Published: October 2005

The design of acoustic antenna arrays often involves the satisfication of multiple, conflicting design specifications. The use of conventional mathematical programming techniques in this endeavor have been inhibited by their inability to consider these multiple objectives within their design formulations. Most present techniques formulate the design problem in terms of a single objective subject to a set of design constraints which become difficult, if not impossible, to solve if these constraints are conflicting in nature. Goal Programming is an effective alternative methodology. Array pattern synthesis through Goal Programming provides the optimal solution to the multiple criteria design of various classes of acoustic antenna arrays. The design problems are formulated in terms of the multiple objectives expressed by the system response functions and the desired characteristics of the design parameters. The solution of two reasonably complex planar array designs is presented. The details of their formulation and solution provide an indication of the ability of this approach to resolve even more complex array design problems. [Work supported by Naval Sea Systems Command Code 0342.]
Single-tone and two-tone responses of cochlear-nerve fibers: changes after kanamycin and noise treatments
Authors: Schmiedt R. A. , Zwislocki J. J.
Published: October 2005

Responses to single tones and two-tone inhibition (TTI) above CF were determined in untreated Mongolian gerbils and after kanamycin or impulse-noise treatment. Kanamycin produced maximum hair-cell damage in the basal cochlear turn, the noise, in the mid-cochlear region and sometimes in the base. Fibers associated with 100% loss of outer hair cells (OHC) in the basal turn showed elevated CF thresholds, reduced slopes of tuning curves, no TTI, steep rate-intensity curves above CF, and excitation during basilar-membrane displacement towards scala vestibuli. In the presence of 50–95% OHC loss restricted to 2–4 mm of the midcochlear region, corresponding fibers exhibited less homogeneous characteristics. CF thresholds corresponding to the basal boundary of the lesion were nearly normal and increased toward and beyond the apical boundary. The tuning curves tended to be somewhat broadened and TTI, though reduced, was still present in fibers with elevated CF thresholds. Fiber excitation usually occurred during basilar-membrane displacement towards scala vestibuli. Some fibers from both treated groups showed increased sensitivity below CF despite elevated CF thresholds. Tuning curves associated with no TTI had high-frequency slopes between 70–200 dB/oct.
Comparative structure of the fish ear
Authors: Popper A. N.
Published: October 2005

Comparative investigations of over 20 species of fish show that there is significant interspecific variation in several aspect of the anatomy and ultrastructure of the inner ear. Variation is particularly extensive in the sacculus and lagena, the two regions of the ear thought to be involved in sound detection. The most striking interspecific variation is in the orientation patterns of the sensory hair cells, as determined with the SEM. In some species the hair cells are found in one of two groups, with all cells in each group oriented in the same direction. In other species there are four orientation groups. The relationship between the orientation groups appears to be species specific. On the gross anatomical level variation includes the relative orientation between the two otolithic organs and the connections, if any, between the sacculus and the cranial cavity. There is also considerable variation in the relationship between the single otolith and the sensory maculas. In some species the otolith “covers” the whole macula while in other species only a small portion of the macula is in contact with the otolith. Variation found in the ear may have functional and taxonomic significance particularly with regard to comparative sound detection mechanisms. [Supported by NIH and Deafness Research Foundation.]
Evaluation techniques for industrial hearing conservation programs
Authors: Lilley D. T. , Royster L. H. , Thomas W. G.
Published: October 2005

As part of an ongoing study of the contribution of industrial noise exposure to hearing loss, the audiometric test results of a number of industrial populations were evaluated employing selected test criteria. These tests include (1) variations in overall hearing threshold; (2) percent of population experiencing improvement and/or deterioration in hearing level with subsequent audiometric tests; (3) hearing level shift versus frequency; (4) permanent threshold shift comparisons between discrete groups with “better than average” hearing and those with “worse than average” hearing; and (5) analysis of standard deviation fluctuations as an indication of conservation program management efficiency. The results of these analyses are presented. [Work partially supported by a Rockefeller Foundation Grant.]
Comparison of techniques for measuring very high values of surface impedance
Authors: Mansbach Peter A. , Holmer Curtis I.
Published: October 2005

Five techniques were used in an attempt to measure the very high acoustic impedance of an asphalt surface. These techniques were as follows: Impedance Tube, Direct Accelerometer Measurement, Pure-Tone Traverse, Pulse-Echo, and Broadband Cross-Correlation. The results of these measurements are used to illustrate the strengths and limitations of each of these techniques.
U-space amplitude shading of linear transducer arrays
Authors: Wang H. S. C.
Published: October 2005

In an earlier paper [H. S. C. Wang, J. Acoust. Soc. Am. 57, 1076 (1975)], it was demonstrated that amplitude shading of a transducer array can be derived from the minimum property of the total received power except that through a slice of main beam. We show here that in the special and fundamentally important case of linear arrays, the formulation can be simplified by considering the power or object function (OF) to be minimized as a function of a generalized variable u = cosθ − cosθ0, where θ denotes the colatitude angle and θ0 the steered direction. Although the total visible range of a linear array covers −2⩽u⩽2, the OF needs only to be minimized in 0⩽u⩽2 due to symmetry of the directivity function. Furthermore, the space 0⩽u⩽2 can be sub-divided into various regions of different weights to form multiparameter OF for the control of individual sidelobes. As an illustrative example, an endfire array of 12 elements logarithmically spaced over an aperture of 4.4 wavelengths is shown to have an unshaded sidelobe level of −10.4 dB, which was improved to −14.8 dB when synthesized with a single parameter OF and to −23.1 dB with a multiparameter OF.
Experimental analysis of array beams tilted to large angles
Authors: Hughes W. J. , Clarke H. J.
Published: October 2005

Beams were formed using the amplitude beam tilting scheme on an equivalent 13-element line array. These beams were tilted in increments of 10° from broadside to endfire. Because of the influence of the baffle, beams which were tilted close to end-fire exhibited the typical reduction in peak sensitivity. There appears to be no advantage gained by tilting a beam more than 70° from broadside due to these baffling effects. Peak sensitivity relative to the forward beam is reduced 9 dB at a 70° tilt and 11.4 dB at endfire while the −3-dB beamwidth increases from 16.5° at broadside to 24° at both a 70° and 90° tilt.
Adaptive cross-sensor beam forming with sparse and random planar arrays
Authors: Bucker Homer P.
Published: October 2005

Let the quadrature field measured at sensor j of a planar array, located at (xj, yj), be Fj. The cross-sensor field Gl is formed as the product Gl = FjFk*, where k represents sensor k and * denotes complex conjugate. The beam response B̃ of the array in look direction φ̂ is B̃(φ̂) = ΣlwlGφ,l*〈Gl〉, where {w̃l} are cross-sensor weights, Gφ̂,l is the expected value of Gl for a single incident wave of unit amplitude and bearing φ̂, and 〈〉 denotes a time average. To be specific, Gφ̂,l = expi[h(yj − yk) cosφ̂ + h(xj − xk)sinφ̂], where h is the horizontal wave number. A simple adaptive algorithm can be used to remove the strongest signal in the beam response function. Let φ̂1 represent the bearing of the strongest signal and B̃(φ̂1) = A12 be the maximum value of B̃(φ̂). The values of 〈Gl are modified by subtraction 〈Gl〉 = 〈Gl〉 − A12 expi[h(yj − yk) cosφ̂1 + h(xj − xk) sinφ̂1]. A new beam response can now be calculated and the second largest signal then subtracted, etc. Examples are shown for both sparse and random planar arrays. [Work supported by NAVELEX 320.]
Numerical method for calculating the nearfield of a parametric acoustic source
Authors: Mellen R. H. , Moffett M. B.
Published: October 2005

The nearfield of the difference frequency generated by a spherically spreading primary wave can be represented by a triple integral of an oscillating integrand. The computation is facilitiated by a transformation that results in rapid convergence. [Sponsored by Naval Material Command.]