bluuebunny/crossref_metadata_2025 huggingface dataset:
Comparison of the Debye and Born Approximations for One-Dimensional Sound Propagation
Authors: Neubert J. A.
Published: October 2005

The one-dimensional stochastic Helmholtz equation is solved first by a two-variable expansion. This gives the one-dimensional equivalent of the Debye asymptotic expansion. Then the one-dimensional Born result is derived for the stochastic Helmholtz equation. It is shown that imposing the traditional single-scattering condition on the Debye result produces the Born result. The coefficient of intensity variation V is calculated for both approximations, and it is again shown that the Debye result reduces to the Born result under the single-scattering condition. It is emphasized that the Debye result avoids the long-range and high-frequency restrictions of the Born result, which arise from the cumulative phase effects.
Reflection and Scattering from a Time-Varying Rough Surface—the Nearly Complete Lloyd's Mirror Effect
Authors: Parkins B. E.
Published: October 2005

The effect of a rough, moving surface upon transmission between an omnidirectional source and receiver in an isovelocity medium is described using the method of physical optics. This is done by relating the spectral density of the source and the wave spectrum of the surface and also the geometry. The dependence of the received spectrum upon the interference between the direct and reradiated waves and the characteristics of the “frequency smearing” caused by scattering are described. A favorable comparison of the theory with the results of an experiment performed under near-ideal conditions in a model tank is made. [Work supported by the U. S. Navy Electronic Systems Command.]
Project Massawa: Attenuation of Low-Frequency Sound in the Red Sea
Authors: Browning David G. , Jones Everett N. , LaPlante Robert F. , Mellen Robert H. , Thorp William H.
Published: October 2005

Extensive measurements in the Atlantic Ocean (4°C) have shown the attenuation of sound below 10 kHz to be anomalously high. In order to determine the temperature dependency of this anomaly, a similar experiment has been conducted in the warmer water (22°C) of the Red Sea. The results show a shift in the apparent relaxation frequency from the 1 kHz observed in the Atlantic to 1.5 kHz, resulting in higher values of the attenuation coefficients in the range 1–6 kHz. The computed activation energy is comparable with that expected for hydrogen bonding.
Nonlinear Spectral Interactions
Authors: Fenlon F. H.
Published: October 2005

A method for computing the nonlinear interactions between the spectral lines of progressive finite-amplitude waves in fluid media is developed via Burger's equation. By taking the Fourier transform of the particle velocity, this equation is reduced to a coupled set of ordinary nonlinear differential equations, which are then solved by means of Airy's algorithm. The solution thus obtained consists of a spectral vector, which initially contains the signal spectrum at the source, and is subsequently enriched by nonlinearly generated spectral lines as the signal propagates through the medium. The utility of the method consists in the ease with which it can be implemented on a digital computer and its applicability to different types of source waveform. As an example, the conversion efficiency of parametric interaction between the spectral lines of a finite amplitude source driven simultaneously at two different frequencies and subject to spherical spreading loss is considered. It is shown that the sound pressure level of the difference-frequency signal increases linearly as the source level is increased until a point of diminishing returns is reached when the high-intensity pump and/or mixer waves go into shock.
Propagation of 127-, 400-, and 1700-Hz Acoustic Energy in the BIFI Range
Authors: Sussman B. , Kanabis W. G.
Published: October 2005

Comparisons are made of propagation loss and frequency smear at the three frequencies. In a previous paper, it was noted that in the summer the measured propagation loss values at 1700 Hz exceeded the predicted values and were independent of sea state. The current summer tests at 127 Hz showed that while the losses are still independent of sea state, they are close to the predicted values. Ray tracing showed that at 127 Hz the ray undergoes considerably fewer bounces than at 1700 Hz, thus explaining the relative magnitudes of the losses. Application of Rayleigh's criterion showed that, even at the highest sea state encountered in the tests, the surface was smooth for the lower frequencies, thus making the loss independent of sea state. The effect of surface roughness was also apparent in the frequency smear measurements which showed higher dispersion at the highest frequency than at the lower ones. An asymmetry noted in all dispersion spectra is explained by means of diffraction theory.
The Calculation of Optimal Linings for Jet-Engine Inlet Ducts
Authors: Wilkinson J. P. D.
Published: October 2005

A numerical method is presented for calculating the optimal lining impedance for the inlet duct of a let engine. The calculation is based on the condition that the total radiated power from the duct shall be a minimum. Several situations are considered: first, the case of an axially symmetric duct in a stationary acoustic medium; second, that of a body of arbitrary geometry in a stationary medium; and finally, the case of an axially symmetric duct in the presence of spinning modes and axial air flow. In each case, the method may yield the optimal lining impedances for a given discrete frequency, flow velocity, and duct geometry. To demonstrate the application of the method, a number of cases are investigated. First, a cylindrical duct closed at one end by a vibrating piston is used to make comparisons with predictions obtained analytically by Rice. Then a more complex problem of a bell-mouth compressor is investigated. A discussion of the results obtained is presented together with a critique of the potential difficulties of the method as applied to complex duet configurations. [This work was carried out under contract with NASA, Washington.]
Subsonic Waves and Troposphere Properties
Authors: Bowman H. S.
Published: October 2005

A combination of circumstances is considered to account for the almost sinusoidal subsonic pressure waves measured at the earth's surface in the Washington, D. C., area on 18 March 1969. The elusive generating mechanism is not known exactly; however, the available experimental data and analyses show interesting observations. The waves had a horizontal phase velocity of 45 m/sec from a direction 274° clockwise relative to North and periods of 300 sec. The maximum zero-to-peak pressure amplitude measured greater than 275 dyn/cm2. The duration of the event was approximately 1 h. The troposphere properties measured within 2 h of the infrasonic event include vertical profiles of winds, temperatures, humidity, and barometric pressures. The winds measured zero at the earth's surface and a maximum of 85 m/sec from 267° at an altitude of 12.1 km. The altitude of the tropopause was at a distance equal to one full wavelength of the infrasound and the winds at that level measured 70 m/sec from a direction corresponding to that of the surface waves. Also a strong nonfrontal low-level temperature discontinuity existed at 975 mbar of barometric pressure.
Interstimulus Interval and Recovery Time in Averaged Evoked Responses
Authors: Tomsic R. T. , Elfner L. F.
Published: October 2005

The effect of stimulus duration and stimulus repetition rate on the amplitude of the N1−P2 wave of the averaged auditory evoked response was measured. A 1-kHz tone was presented in four different patterns to four subjects. The four different stimulus patterns were: (1) a tone of 1-sec duration repeated every 2 sec; (2) a tone pulse of 50-msec duration repeated every 2 sec; (3) a tone pulse of 50-msec duration repeated every 1 sec; and (4) ten 50-msec tone pulses presented in 1 sec, followed by 1 sec of silence. Condition 1 evoked a smaller response than condition 2 and conditions 1 and 2 were both larger than conditions 3 and 4. Condition 4 was not significantly smaller than condition 3. The results show that duration of a stimulus affects the amplitude of the response and that the rate of stimulation does not affect the amplitude under certain conditions.
Finite-Amplitude Acoustic Waves in Nonlinear Relaxing Media
Authors: Klinman R. , Chi L. K. , Kraus L.
Published: October 2005

The problem of propagation of finite-amplitude acoustic waves in nonlinear media is studied by the method of small perturbations in acoustic Mach number M. Depending upon the dispersion parameter σ = (C∞2 − C02)/C∞2, which measures the coupling between acoustic and relaxation processes, the system of pertubation equations takes two distinct forms. This distinction is analogous to that which occurs in viscous fluids, where the critical parameter is the Reynolds number [Z. A. Goldberg, Sov. Phys. Acoust. 2, (1956)]. For σ≪M, the perturbation equations are typical of those for non-dissipative media whose solutions are characterized by continued harmonic amplitude growth with propagation distance, indicative of shock formation. For the case where σ≫M, solutions are obtained correct to second order, taking account of the second-order boundary conditions. The second harmonic and phase-shifted wavenumbers agree with those obtained by Polyakova [Sov. Phys. Acoust. 6, (1960)]. The amplitude of the second harmonic grows with distance to a maximum and thereafter decays. The position and amplitude of this maximum as a function of frequency will be exhibited. Also, the amplitude exhibits dependence on the parameter of nonlinearity B/A. The equivalence of relaxation and viscous effects in the nondispersive region, ωτ≪1, is demonstrated for finite-amplitude waves [Research supported by the U. S. Office of Naval Research.]
Mode-Coupling for Finite-Amplitude Acoustic Waves
Authors: Yen Nai-Chyuan
Published: October 2005

The hydrodynamic equations describing finite-amplitude acoustic wave motion in a multiply resonant system have been solved by a perturbation method applied to the “time” scale (a procedure sometime referred to as the “two-timing” or “two-variable” method). The approximate solution so obtained indicates that the nonlinear properties of the medium can produce coupling among the oscillation modes of the system. For example, energy exchange can occur among the modes of a resonator corresponding to the frequencies f1, f2, and f3 when the relation f3 = f1 + f2 holds. Furthermore, it can be shown that under proper conditions a weak signal can be pumped to a higher level by a strong driving source. Such amplification may also be sufficient to enable a transient disturbance to initiate a pair of subharmonics which, in addition to satisfying the f3 = f1 + f2 relation, must also be resonance modes of the system. The nonlinear parameter, the dissipation mechanism, and the driving-signal frequency determine the threshold for generating such subharmonics. [Work supported by the Office of Naval Research.]
Experimental Observation of Acoustic Saturation in Water Caused by Nonlinear Propagation Effects
Authors: Shooter J. A. , Muir T. G. , Blackstock D. T.
Published: October 2005

If the amplitude of a plane originally sinusoidal wave is high enough, the wave will distort as it propagates, until a sawtooth is formed. Dissipation then slowly smooths the sawtooth. Eventually, at long range, the sinusoidal wave shape is restored. Beyond this range, according to the solution of Burgers's equation, the amplitude of the wave is independent of the source level. In other words, acoustic saturation occurs. In the case of spherical waves, no analytical solution of Burgers's equation is known, but it is possible to estimate the distance at which shocks form and the distance at which the wave loses its sawtooth shape. Curves showing these distances as a function of source level, with frequency as a parameter, have been computed for spherical waves in fresh water. An experiment to investigate the saturation phenomena has been carried out in a fresh-water lake. A piston was used to generate a high-intensity narrow beam at a frequency of 450 kHz. Data was taken along the axis in the farfield. For each range point, a curve of sound-pressure level of the fundamental versus source level has been plotted. Saturation is clearly evident at the longest range (36 yd = 104 wavelengths). The effect of saturation is also seen in the beam patterns, which become blunt at long range as the source level is increased. [This work was sponsored by the U. S. Office of Naval Research.]
Pulsed Laser Imaging System for Direct Visualization of Ultrasonic Waveforms
Authors: Newman Dennis R.
Published: October 2005

A pulsed laser with an extremely short pulse duration (≈3 nsec) has been used in a schlieren system to resolve individual wavefronts within high-frequency pulse trains. Wavefronts have been resolved at frequencies of 1–20 MHz. The system and its components are briefly described; photographs of basic ultrasonic wave propagation effects such as refraction, diffraction, and distortion in liquids are shown. Several possible applications to acoustic problems such as lens design, transducer and transducer array evaluation, critical effects, and propagation in biological specimens are discussed. The system's potential as a very graphic teaching aid is presented.
Effect of Physical Detail on the Response of the External Ear
Authors: Gardner Mark B. , Hawley Melville S.
Published: October 2005

The purpose of the present paper is to show how differences in detail in the shape, size, and other physical characteristics of the outer ear, including differences in the impedance of the tympanic membrane, affect the transmission characteristics of this division of the hearing mechanism. Toward this end, lathe-constructed models of the various external cavities as well as models made from castings of actual ears have been used, both in connection with and in the absence of a manikin head. The former type of model has the advantage that changes in dimensional detail can be readily achieved and as easily specified. The latter has the advantage that the shape, size, and texture of an actual ear can be more nearly simulated.
Intensity Function of the Cochlea and Cortex
Authors: Walloch Richard A. , Brummett Robert E.
Published: October 2005

Amplitude changes of the cochlear potential due to increases in sound intensity were compared in the same animals with amplitude changes in the evoked potential from the cortex. As is well known, the cochlear potential is linear over a 30–50 dB range. Above this linear region, the cochlear potential plateaus, and then it actually declines when the sound intensity is further increased. The cortical-evoked potential behaves in a similar manner. The amplitude of the cortical-evoked potential increases monotonically with increases in sound intensity as long as the response of the cochlea is linear. When the cochlear potential departs from linearity, the increases in the evoked potential also tend to plateau. At the point the cochlear potential decreases in response to an increase in sound, the evoked potential from the cortex also decreases. This decrease in the amplitude of the evoked potential stresses the need for careful measurement of the sound intensity and/or the cochlear potential. In several studies, units have been observed to be inhibited by loud tones. True neural inhibition can be differentiated from overdriving the cochlea only if the proper measurements are made.
Action of Some Cholinomimetics and Cholinolytics on the Effect of the Crossed Olivocochlear Bundle (COCB)
Authors: Bobbin R. P. , Konishi T.
Published: October 2005

The methods used to record the cochlear potentials, stimulate the Crossed Olivocochlear Bundle (COCB), and perfuse the scala tympani in guinea pigs were those described by Konishi [J. Acoust. Soc. Amer. 47, 1055–1062 (1970)]. Drugs were dissolved in a solution containing: NaCl, 139 mM; KCl, 5.6 mM; CaCl2, 2.2 mM; NaHCO3, 21 mM; 95% O2-5% CO2; with pH of 7.2–7.4. The perfusion solution increased the endocochlear potential (EP) and the. change in the scala-media potential evoked by COCB stimulation (DCS). The following doses of agents were found to reduce the DCS by approximately 50% at 5 min of perfusion through the scala tympani: carbamylcholine chloride, 50 μM; acteylcholine chloride, 550 μM; eserine sulfate, 400 μM; acetylcholine chloride, 250 μM plus eserine sulfate, 10 μM; choline iodide, 550 μM; nicotine tartrate, 1 mM; d-tubocurarine chloride, 5 μM; styrchnine sulfate, 5 μM; atropine methyl bromide, 25 μM; and atropine sulfate, 100 μM. Arecoline hydrobromide (1 mM) and eserine sulfate (10 μM) alone were inactive. The data are consistent with hypotheses that the COCB is cholinergic. [Supported by a Public Health Service grant and an NIH fellowship from NDS.]
A Digital Simulation of the Time-Varying Vocal Tract
Authors: Ruiz P. M.
Published: October 2005

The Kelly-Lochbaum model for the simulation of the vocal tract is based upon the solution of Webster's horn equation for a static tube. This model has been found to be adequate for the synthesis of steady vowels; however, for time-varying vocal tract shapes, the model gives rise to discontinuities in pressure and volume velocity. These discontinuities were found to produce noise in the speech synthesized by the model. A solution of Webster's horn equation for time-varying tubes was obtained by representing the vocal tract by a sequence of uniform cylindrical sections. Recursion equations for pressure and volume velocity at the junctions of adjacent sections were derived. Sentence-length utterances synthesized by this model were found to be free from noticeable noise.
Some Prosodic Details of American English
Authors: Umeda N. , Coker C. H.
Published: October 2005

General principles for prosodic rules for English, reported previously [C. Coker and N. Umeda, “Acoustical Properties of Word Boundaries in English; On the Vowel Duration and Pitch Prominence,” J. Acoust. Soc. Amer. 47, 94 (1969)] are now implemented in a computer program. Surprisingly, the rules depend mainly on the surface structure of the sentence and on properties of the words. To provide better prosidic features for automatic synthesis, however, additional rules have been added. A good measure of the difficulty of speech material is found to be the proportion of content words to function words. According to this ratio, stress and pause assignment should be different. A paragraph redundancy rule, promoting first occurrences of content words and demoting subsequent repetitions, is found to improve naturalness and reduce monotony of the synthetic speech. Several special verb classes are assumed for assignment of clause, phrase, and word boundaries. These verbs, although different from auxiliary verbs, behave like auxiliaries in the sense that the main idea is conveyed by the remainder of the predicate. Stress and boundary assignment are altered according to verb class. Local phonetic interactions require adjustments in durations of phonemes according to specific contexts. Details of these rules and examples from synthesis are presented.
A Method for On-Line Interactive Transcription of Quantized Speech Data
Authors: Dixon N. R. , Tappert C. C.
Published: October 2005

An interactive procedure has been developed to facilitate the assignment and location in the time domain of phonetic symbols to utterances of quantized continuous speech. Conventional methods for transcribing speech data are often based on talker intention and/or listener bias rather than on physical evidence and, generally, symbol location cannot be related to subsequent machine processing. The procedure described operates on-line, in an interactive mode, utilizing a graphic display with an IBM System/360 Model 40. Available for display are a digital sound spectrogram, the power spectrum at a given time sample, an average spectrum for a given sound class, and correlations of a current spectrum with average spectra. With this technique, location and assignment of phonetic symbols are less dependent upon operator expertise (and/or bias) and are highly related to subsequent processing by machine. [Supported in part by Rome Air Development Center.]
Propagation
Authors: Williams A. O.
Published: October 2005

These 20 years have seen much progress in understanding of underwater sound propagation. Few new topics have appeared, but far more complex problems can now be handled—in considerable part, thanks to computers. Emphasis has shifted to lower frequencies, longer ranges, and wave theoretical methods. We are still limited by extreme variablity of the medium and its boundaries, and by a gap of understanding between mechanical-chemical and acoustical properties. Efforts to narrow this gap should continue, as should the search for “invariants” of problems and for statistical theories to match averaged data.
Acoustics for Music Students: A Lecture-Laboratory Course
Authors: Proctor D. G.
Published: October 2005

A course in acoustics has been designed and tested with two groups of students whose interests are in musical performance or music education. A survey of their interests before and during the source of study reveals areas of acoustics which are relevant to their goals but lacking in their studies of music. Conservatory faculty suggested areas of acoustics which they felt should be in the course of study. Student and faculty inputs are similar, but a few differences occur and are discussed. The topics studied in the course include basic principles of acoustics and applications in instruments, auditoriums, hearing, voice, and electronic production and reproduction of sound. Laboratory experience for the students has proven valuable to them, especially those experiments on determination of the resonance modes of vibrating bars and of the resonant frequencies of the strings on a violin. An analysis of the effectiveness of different aspects of the course with respect to the students involved is presented.
The Ear as an Analyzer of Musical Tones
Authors: Gibson Lee
Published: October 2005

Prior investigations by Plomp and Mimpen have concluded that the ear can hear and identify separately only the first five to seven particles of a tone, in agreement with limitations deduced from the critical band theory. With somewhat different equipment and procedure, a set of experiments was conducted for the purpose of discovering whether such variations of timbre as are typical of musical performance may affect a replication of the conclusions of Plomp and Mimpen. Oscillators were phase locked to a common fundamental frequency and adjusted to uniform sound levels, which were checked by a wave analyzer; the resultant complex tone could be deprived of any partial (except its fundamental partial) by means of interruption by an electronic switch; an observer was asked to duplicate the frequency of the interrupted partial, and the chosen frequency was checked with a frequency counter. Results indicate a less categorical limitation of the ear's capability, in which, at least in the middle frequencies, partials as high as the 11th and the 12th may be separately heard with some consistency. (Research sponsored by North Texas State University.)
A Comparison of “Frequency Flexibility” in the Oboe and Shehnai
Authors: Hadden W. James
Published: October 2005

The shehnai is an Indian instrument which is similar, in its basic physical characteristics, to the oboe. One of the most striking features of the shehnai is the great degree of “frequency flexibility,” i.e., the possibility of playing a relatively wide band of frequencies with a single fingering. An unsophisticated experimental procedure, in which the maximum playable bandwidth is measured, indicates that an ad hoc Q (ratio of bandwidth to center frequency) for the oboe is two to three times larger than that for the shehnai. Its effect will be demonstrated and its origins—both physical and socioesthetic—will be discussed. [This work was performed when the author was a Special Guest Lecturer in the Department of Physics, Rajasthan University, Jaipur, Rajasthan, India.]
Impulsive Sound from Helicopters Due to Compressibility Effects
Authors: Lyon Richard H.
Published: October 2005

The tip regions of the lifting rotor of a helicopter in forward flight may reach nearly sonic speeds with respect to the surrounding air. When this occurs, compressibility effects will cause an impulsive sound wave to be generated. Both lift and thickness distributions are responsible for this component of the sound. Calculations of the sound radiated from blade tips of various shapes and lift distributions have been made. Some results of the calculations and correlations with experimental data are presented.
Sound Radiation from a Reciprocating Compressor Orifice without Valve
Authors: Trella Tom J. , Soedel Werner
Published: October 2005

As a first step toward the acoustic radiation modeling of a reciprocating air compressor, the suction and exhaust valves were removed and the compressor was made to cycle air through a single circular orifice. The theoretical model incorporates the aharmonic and large piston motion by thermodynamically modeling the flow in the cylinder. The flow through the orifice was modeled as compressible. Contraction coefficients on an ON-OFF basis were used to account for streamline effects in the orifice section. The farfield acoustic spectrum was predicted by treating the orifice as a monopole source radiating into an infinite half-space. Agreement between theoretical prediction and experimental measurements was encouraging. A small rise in both the predicted and measured frequency spectra was hypothesized to be a Helmholtz resonator effect since it occurred in the expected frequency band. The theoretical calculations were carried out using both a conservation of mass and momentum approach and a conservation of mass and energy approach. Small differences in the results of these two approaches were observed. Finally, because of our desire to simplify future models, influences of individual terms in the derived equations on instantaneous mass flow rates were evaluated.
Threshold Tracking and Negative Threshold Shift
Authors: Melnick William
Published: October 2005

Melnick [J. Acoust. Soc. Amer. 46, 1585–1586(L) (1969)], in a study of auditory sensitization using a threshold-tracking procedure, reported a negative threshold shift of 2 dB for a 1000-Hz signal following 3 min of quiet as well as following 3 min of exposure to 1000 Hz at 40 dB SL. This could not be due to an assumed sensitizing effect of a previous sound. A series of experiments was conducted to isolate factors contributing to this negative shift. Prolonged threshold tracking at 500 Hz (10 min) following 3 min of quiet revealed a negative shift which did not return to thresholds measured prior to the quiet period. When three tracking periods (3 min) were interspersed by 2 periods (3 min) of quiet or 40 dB SL of 500 Hz, negative threshold shift persisted following both the tone exposure and the quiet periods. In another experiment, six repetitions of a tracking task did not eliminate negative threshold shift following quiet. Test-tone mode (pulsed or continuous) and direction of approach to the initial tracking period (ascending or descending) did not matter. Negative threshold shift seems to be a minute but hardy phenomenon.
Temporary Threshold Shift in Hearing from Exposure to Different Noise Spectra of Equal dBA Level
Authors: Cohen A. , Anticaglia J. R. , Carpenter P. L.
Published: October 2005

The intent of this study was to determine whether sound-level measurements in dBA could adequately depict harmfulness to hearing from exposures to noises differing widely in spectra. Each of 11 male listeners, aged 20–26 yr, were exposed on two separate occasions to three noises whose spectral slopes were −6, 0, and +6 dB per octave band. All such noises were presented for 30 min via an earphone at 100 dBA. Analyses of variance of the post-exposure threshold data (corrected to 2 mm, TTS2) found insignificant differences due to the spectral variations for both test and retest conditions as averaged across audiometric frequencies 250–8000 Hz. Interactive effects between spectra and test frequencies were significant, the −6 and 0 dB spectral conditions causing relatively more TTS2 at frequencies below 3000 Hz, and the +6 dB spectrum more TTS2 at frequencies above 3000 Hz. Implication of results for noise-hazard rating schemes using dBA measures are noted.
Methods Proposed for Use in Assessing A-Scale Auditory Risk Limits for Persons Wearing Ear Protection
Authors: Gasaway Donald C.
Published: October 2005

Recent auditory-risk criteria and limits employ A-scale measures of computations as the basis with which varying degrees of potential auditory risk can be estimated. Since most auditory-risk criteria account for unprotected exposures, persons who must use these newly proposed criteria must employ some method with which attenuated conditions of exposure can be evaluated. This report provides insight and specific guidance required to evaluate conditions of noise exposure when three types of personal ear protection are worn (headsets, earmuffs, and earplugs). Noise spectra typical of Air Force operations are categorized into five groups (flat, drop-off of 2–4 dB/oct, drop-off of 4–6 dB/oct, rising spectrum of 3 dB/oct, and spectra containing pure-tone components), and differences which result between attenuated and nonattenuated conditions related to spectral distribution are reported.
Descending Békésy Threshold with Various Off-Times
Authors: Young I. M.
Published: October 2005

Békésy threshold measurements were made with various off-times in normal-hearing subjects. Signals with a constant on-time of 200 msec consisted of various off-times ranging 0 and 1000 msec at 10-msec intervals below 100-msec off-time and at 100-msec intervals above 100-msec offtime. Rise and decay time of stimulus envelope was 10 msec. Attenuation rate was 1 dB/sec and 4 dB/sec with 0.2-dB steps. For ascending method (starting at infrathreshold level), there were no observable differences in threshold in audibility for signals with various off-times for two attenuation rates. For descending method (starting at suprathreshold level), with the off-times 40 msec or shorter, thresholds in audibility were essentially the same as the steady tone, with mean value of 21.5 dB re 0.0002 μbar. As the off-time increased from 50 to 700 msec, monotonic improvement in threshold was obtained. When the off-time was 700 msec or longer, there were no further changes in threshold, but an asymptote in hearing level (approximately 7.0 dB SPL). These results refer to an attenuation rate of 1 dB/sec. For an attenuation rate of 4 dB/sec, the pattern was essentially similar and thresholds were approximately 2.0 dB better than for 1-dB attenuation rate throughout the off-time range. The results of this study indicate that offtimes of at least 700 msec are required for complete recovery from adaptation on descending method in normal ears.
An Application of Digital Computer Logic to Pure-Tone and Speech Audiometric Procedures
Authors: Wood T. J. , Wittich W. W. , Mahaffey R. B.
Published: October 2005

The purpose of this study was to develop a computer program and the necessary interfacing systems which would enable a PDP/8-I digital computer to (1) control all aspects of administering pure-tone air- and bone-conduction and speech-reception thresholds, and speech-discrimination tests; (2) recognize the need for masking, determine the appropriate level of masking, and regulate the presentation of the masker to the non-test ear; (3) analyze the subject's responses for the purpose of regulating the rates and intensities of the test stimuli presentation; and (4) present the obtained responses in an audiogram format at the conclusion of the computerized testing procedure. The efficiency of this automated design was evaluated in terms of Pearson product-moment correlation coefficients obtained between masked and unmasked thresholds and discrimination scores, as well as effective masking levels employed by the computer and an audiologist when each performed these audiometric tests on the same hypacusic subjects. The results of this study indicate that the computerized procedures employed to determine pure-tone air- and bone-conduction thresholds, speech-reception thresholds, and speech-discrimination scores are, in terms of the subjects tested, a clinically feasible means of assessing these auditory functions.
Time-Intensity Trade in Measurements of Auditory Function
Authors: Hopkinson N. T.
Published: October 2005

Threshold for periodic tone pulses as a function of duration and repetition rate in the normal auditory system is controlled by a time constant of the order of 200–300 msec. Thresholds measured in the neurally impaired ear are greatly influenced by the fact that neural adaptation increases as (1) the time interval between pulses decreases (below 200 msec), and (2) the duration of the signal is prolonged. Threshold measures further reflect variations in response behaviors such as reaction time. For any one individual, the time constant of the ear is related to a critical “decay” time that depends on both signal duration and interstimulus interval. In standard audiometric practice, lengthened signal durations combined with short off-intervals strain already altered time constants in neurally impaired ears, leading subjects to trade intensity for the critical times necessary to perceive the signal. An adaptive method in which signal duration, off-interval, repetition rate, and time for subject response are controlled and defined will avoid the adverse intensity-for-time trade that can occur during conventional psychoacoustic measures of threshold in auditory pathology. [Research supported in part by grants from the NINDS of the U. S. Public Health Service.]
Thresholds for Spectrally Complex Stimuli in Pathological Ears
Authors: Thomas Ian B. , Sparks David W.
Published: October 2005

The purpose of our investigation was to determine if, for pathological ears, it is possible to predict thresholds for spectrally complex stimuli from pure-tone thresholds, i.e., to see if, in this restricted sense at least, the pathological ear behaves in a manner consistent with Fourier analysis principles. Subjects having pure-conductive, pure-sensorineural, and mixed losses were drawn from a clinical population. Pure-tone thresholds were determined from 1-min Békésy tracings by using continuous tones for a wide range of frequencies. Thresholds were then determined by a similar procedure for continuous signals consisting of 50-μsec rectangular pulses having repetition rates corresponding to each of the audiometric frequencies. Thresholds for pulse signals were then predicted from the pure-tone thresholds after considering critical-band effects and after taking into account the effects of the audiometer system on the pulse spectra. In almost all cases and for all types of loss, experimental and predicted thresholds for pulses agreed within ±5 dB. [Work supported by NIH.]
A Multiband Center-Clipping Echo Suppressor
Authors: Berkley D. A. , Mitchell O. M. M.
Published: October 2005

For telephone circuits which include synchronous satellites, conventional echo suppressors of the voice-switching type are unsatisfactory because of speech mutilation and the presence of echo during double talking. We have found that multiband center-clipping process may be used as an echo suppressor. The near-end signal plus echo of the far-end signal are divided into five or six contiguous bands, with each filter output going to a center clipper. Each clipping level is set equal to the echo level in that band. An analog implementation of this echo suppressor was evaluated in a simulated satellite circuit. Echo was completely eliminated, even during double talking, for return losses approaching 0 dB, and operation appeared to be full-duplex at all times with little distortion of the speech. For return losses greater than 15–20 dB, the center-clipping system was practically indistinguishable from a four-wire connection with no echo path.
Ultrasonic Scans of the Tongue
Authors: Kelsey C. A. , Minifie Fred D.
Published: October 2005

Ultrasonic scans taken along the longitudinal dimension of the tongue, depicting a saggital plane on the dorsal surface of the tongue, are compared with similar views obtained via conventional lateral head x ray. Also, transverse ultrasonic scans are presented which picture the dorsal surface of the tongue in coronal section, coronal oblique section, and horizontal section. Four adult male subjects participated in this experiment, producing the sounds /i, æ, a, u, s, ∫/ and holding the tongue in a rest position. A discussion is presented pertaining to the use of ultrasonic scans for visualization of structures used in speech production.
Quality of Speech Reconstituted from Reduced Spectral Data
Authors: Li K.-P. , Hughes G. W. , House A. S.
Published: October 2005

Characterizing speech spectra by means of statistical procedures reduces the amount of information needed to specify the data. When eigenvectors of the covariance matrix of short-time digital spectral data are chosen to represent the speech spectra at an average accuracy of 90%, continuous speech can be characterized by seven dimensions. However, a proper test of the adequacy of such a data reduction scheme involves perceptual studies of the reconstituted speech signals. In order to establish the relationship between statistical accuracy and perceptual intelligibility, a computer program to reconstruct the speech signal from spectral data has been prepared. Preliminary listening tests to judge the qualities of speech signals reconstituted from reduced data have been made. The experiments and results are demonstrated and discussed.
Recording and Analysis of Dolphin Echolocation Signals
Authors: Diercks K. J. , Trochta R. T. , Greenlaw C. F. , Evans W. E.
Published: October 2005

A new technique for recording dolphin echolocation signals during target-discrimination tests is described. The technique utilizes sensors attached to the animal. A brief description of the sensors is provided. A short color film of dolphin echolocation tests performed using the attached sensors is shown. Problems in recording the transient-like dolphin echolocation signals are discussed. A short summary of analyses of echolocation signals of two dolphins—an Atlantic bottlenose dolphin, Tursiops truncatus, Montagu, and an Amazon river dolphin, Inia geoffrensis—is presented. [This work was performed under contract with NURDC, San Diego, California.]
Target Discrimination by Dolphins using Acoustic Echolocation
Authors: Evans W. E. , Diercks K. J. , Trochta R. T.
Published: October 2005

Target discrimination performance by two dolphins—an Atlantic bottlenose dolphin, Tursiops truncatus, Montagu, and an Amazon river dolphin, Inia geoffrensis—is described. The targets were a series of pressure-release cylinders of equal length which varied in target strength (i.e., varied in diameter) in 1-dB and 0.5-dB increments about a median (reference) value. The dolphin's task was to identify the reference target when paired with any other target, either larger or smaller. Tursiops achieves 85%–95% correct discrimination with a 1-dB target-strength difference; performance drops to chance at 0.5-dB difference. Inia's, performance is somewhat poorer. The relationship between discrimination performance and echolocation signal characteristics is discussed. [This work was performed under contract with NURDC, San Diego, California.]
A High-Pressure Acoustic Measurement System
Authors: Higgs R. W. , D'Amico P. M. , Eriksson L. J.
Published: October 2005

Experimental systems designed to measure the acoustical properties of materials from 0.4 to 100 kHz under high hydrostatic pressure are presented. From 0.4 to 20 kHz, a resonant technique is used utilizing a 1.25-in.-i.d. 6-ft tube and a 0.625-in.-i.d. 2-ft tube. The range from 20 to 100 kHz is covered by using two impedance tubes. The above-2-ft tube is used from 20 to 40 kHz and a 0.300-in.-i.d. 2-ft tube is used from 40 to 100 kHz, both in an RF-pulse mode. The water-filled tubes are temperature controlled from 4° to 30°C and are capable of 10 000 psi. The measured quantities in the resonant system are the frequency at which resonance occurs and the Q of the tube, both with and without the sample. In the impedance tubes, discrete RF-pulses are allowed to reflect off the sample and a standard λ/4 reflector. Measurements are made on the phase and amplitude of the reflected pulse in comparison with a reference signal. Transducer designs, measurement techniques, and data analysis are described.
A 252-Channel Recorder and Analyzer for Large Acoustic Array Studies
Authors: Byrnes Francis X.
Published: October 2005

A multichannel recording and analysis system is now in use for beamforming, signal processing, and noise-field studies on large acoustic arrays. Each of the 252 data channels records and reproduces signals from dc to 20 kHz with a reproduced-signal to system-noise level difference of more than 40 dB and a maximum timing error between channels of less than 3 μsec. Timing signals giving 1-μsec time resolution and digital signals with housekeeping and manually set information are also recorded. A real-time playback system reproduces the original time and voltage relationships for all 252 channels simultaneously. A second playback system processes 40 channels at a time through a 40-channel A-to-D converter and a 40-channel 1024-point FFT processor. This system produces digital tapes containing the signal samples the auto-spectral-density data and the cross-spectral data for the 40 input channels. The data on these tapes is either plotted, further processed by a computer, or combined with other output tapes to produce the cross spectra between any pairs of the 252 input channels. Examples of current rises of this system are given.
Physiological Components of Vocal Vibrato
Authors: Mason R. M. , Zemlin W. R.
Published: October 2005

The physiology of vocal vibrato was studied in four adult subjects sustaining /ɑ/ vowels during four conditions: two in an erect sitting posture, and two in the usual posture for high-speed laryngeal photography. Subject performances were monitored by acoustic recordings, electromyographic recordings of selected areas of the neck and thorax, and by high-speed cinematography. The cricothyroid muscle appeared to be a primary regulator of vocal vibrato in all subjects. Vocal-fold activity, as observed on films taken at 4000 frames per second, resembled nonvibrato vowel productions. Acoustic analysis revealed synchrony of the fluctuations in pitch and loudness of vibrato. EMG data, gathered from morphological antagonists at rations sites on the body, suggests that the functional synergism of opposing muscles is responsible for vibrato, and is a rather basic physiological phenomenon.
An Experiment on Voice Identification by Visual Inspection of Spectrograms
Authors: Tosi O. , Oyer H. , Pedrey C. , Lashbrook B. , Nicol J.
Published: October 2005

An experiment aimed to test Kersta's claims on the reliability of speaker identification by visual inspection of spectrograms is being performed under a contract with the Michigan State Police, sponsored by the Federal Department of Justice. Thirty-eight examiners were selected, trained during approximately three weeks, and then employed to perform a total of approximately 32 000 trials of identification. Two hundred and fifty speakers, taken randomly from a homogeneous male student population of 25 000 twice recorded, one month apart, five utterances of nine clue words, spoken in three different conditions. Approximately 54 000 spectrograms (broad-band, 4-kHz, Voiceprint spectrograph) were obtained from these recordings and used in the trials of identification performed by the examiners. One-third of these trials were closed, one-third were open, including a matching spectrogram, and one-third were open trials not including a matching spectrogram. Several levels were tested in this experiment, involving 648 possible combinations, each combination defining a different type of trial. Each type of trial was repeated four times, by 18 examiners grouped into three panels and nine subpanels. The levels tested were: number of clue words (2 levels); number of utterances of each clue word (3 levels); phonetic boundaries of the clue words (3 levels); levels of transmission (3 levels); class of spectrograms (2 levels, contemporary and noncontemporary); number of known speakers (3 levels); and type of trials (3 levels). After the completion of this experiment, the data is analyzed through a computer.
Changes of Human Voice Caused by Age, Disguise, and Simulation
Authors: Endres W.
Published: October 2005

In order to clarify whether visible speech spectrograms can be used to identify human beings, two questions must be studied: Does the formant structure of phonomes uttered by a certain speaker change over a longer interval of time? Can the formant structure be changed by disguise or is it even possible to imitate the formant structure of another speaker? Spectrograms of phonemes uttered by seven speakers and recorded over periods of up to 29 years showed that the position of formants and of the pitch of voiced sounds shifts to lower frequencies with increasing age of test persons. Speech spectrograms of texts spoken in a normal and a disguised voice revealed strong variations of formant structure. Visbile speech spectrograms of utterances of VIPs have been compared with those of imitators. The imitators succeeded in varying the formant structure and the pitch frequency of their voices, but they were not able to adapt these parameters to those of the imitated persons. In addition, a report is given on the results gained by comparing the voices of twins and their relatives.
Speaker Verification Experiments
Authors: Das S. K. , Mohn W. S. , Saleeby S. L.
Published: October 2005

Speaker verification is defined as confirming a speaker's purported identity. The method of confirmation is to compare a speech sample of the person against his speech profile, which is based on previous utterances of the same phrase. Experiments have demonstrated the feasibility of an automated scheme of speaker verification under certain restrictions. Important among these restrictions are the use of only male speakers, wide-band (180–8000 Hz) audio channel, and noise-free environment. The experiments used a total of about 7000 utterances of the phrase “Check available terminals” from 118 speakers. An average misclassification rate of 1% with a “no decision” rate of 10% was obtained. Further work is directed toward removing some of the restrictions, such as reducing the bandwidth and including female utterances. The strategy for improving the system performance is to reduce the “no decision” rate without increasing the misclassification rate.