bluuebunny/crossref_metadata_2025 huggingface dataset:
Finite-Amplitude Distortion in Air at High Acoustic Pressures
Authors: Burns Stephen H.
Published: September 2005

Measurements have been made at the harmonic distortion of progressive waves having initial peak pressures as high as 0.1 atm at a fundamental frequency of 445 cps. These signals were propagated in tubes of diameters 1.587, 2.66, and 5.26 cm. The tubes were terminated after seven wavelengths or before the experimental shock-formation distance, whichever came first. Typically, twenty harmonics, whose amplitudes were within 60 dB of the initial pressure, were measured near the shock-formation distance. The measurements of the first four harmonics are compared with an extension of a theory presented by Blackstock [Paper O15 in Proceedings of the Fourth International Congress on Acoustics, Copenhagen, 1962 (Organization Committee of the 4th ICA and Harlanz & Toksvig, Copenhagen, 1962)]. The theory is confirmed in all essential details. [Work supported in part by the National Science Foundation and by the U. S. Office of Naval Research.]
Masking-Level Differences for Monaural Signals Binaurally Masked
Authors: Dolan Terrence R. , Deatherage Bruce H. , Hafter Ervin R.
Published: September 2005

The changes in detectability (MLD) of a partially masked, monaurally presented, 500-Hz gated sinusoid that occur as a function of interaural intensive differences in a 100- to 3000-Hz band of masking noise were investigated at five interaural correlations of the noise. The correlations were +1.0, +0.5, 0.0, −0.5, and −1.0; these are symbolized as N0, N+0.5, NU, N-0.5, and Nπ, respectively. The shapes of the functions relating MLD to interaural differences of intensity are similar for the nonzero correlations, increasing as noise is added in the nonsignal ear to some maximum value in a range where the level of that noise is like the level of the noise in the signal ear. The maximum MLD's were approximately 10 dB for N0, 9 db for Nπ, 4 dB for N+0.5, and 2.5 dB for N-0.5. The MLD's obtained for zero correlation NU were less than 1 dB over the range of interaural intensive differences investigated, negative for lower levels of noise in the nonsignal ear, and positive for higher. [Work supported in part by the U. S. Office of Naval Research.]
Interaural Time and Intensity Difference and the MLD
Authors: Zerlin Stanley
Published: September 2005

Masking-level differences (MLD) for a 600-cps low-pass transient were explored as a function of (1) interaural time difference (Δt); (2) interaural intensity difference (ΔI); and (3) concomitant time and intensity disparity. Masking-level difference is the difference between two levels of a noise (N0): one required to mask a given condition of click, the other to mask a binaural in-phase click (S0). Results show that: (1) The Δt function has two distinct segments: (a) for short-time separations, the MLD is governed by interaural phase relations, and has a maximum that approximates the Sπ condition; (b) for longer Δt, the MLD shows a systematic decrease and ultimately approaches the signal monaural (Sm) condition. (2) For increasing ΔI, the MLD increases monotonically and reaches a maximum of 6–7 dB at signal monaural. A ΔI of 24 dB yields an MLD that is still 1–2 dB below Sm. (3) When ΔI is added to a fixed Δt, the effect, in most cases, is a degradation of the MLD that would have resulted from time difference alone. In general, for the combination, MLD decreases as ΔI increases.
Binaural Masking-Level Difference and Frequency Discrimination
Authors: Henning G. Bruce
Published: September 2005

Two tones of different frequency were presented successively to observers who were required to indicate whether the first or the second tone was the higher-frequency tone. Each tone was presented binaurally for 250 msec in a background of continuous white noise of average spectrum level 30 dB. The noise was always presented in phase at the two ears; however, in one condition, the tones were in phase at the two ears (N0S0) and in the other condition 180° out of phase at the ears (N0Sπ). Functions relating the percentage of correct discriminations (in 200 trials) to signal-to-noise-ratio were obtained for several different values of frequency separation (ΔF) between tones centered at 250 cps. For large values of ΔF (15 cps), the functions in N0S0 and N0Sπ differ only by a displacement (14 dB) on the log signal-to-noise ratio axis, as is the case with binaural-masking level differences (BMLD) in simple detection performance. For small values ΔF (1 cps), however, the masking difference depends on signal-to-noise ratio as well, with no masking difference at large signal-to-noise ratios and a maximum difference of 5 dB. No apparent masking-level difference occurs when ΔF is 0.5 cps, even though 62% correct discrimination is possible. [Research supported by the National Institutes of Health, Public Health Service, U. S. Department of Health, Education, and Welfare.]
Laser-Beam-Induced Ultrasonic Waves in Glass
Authors: Lester William W.
Published: September 2005

Ultrasonic waves have been produced in samples of various glasses by passing a Q-spoiled laser beam through the sample. The ultrasonic waves were observed at a frequency arbitrarily chosen as 266 kc/sec, but in general they consist of a broad spectrum. Various theoretical explanations have been explored, and the most palusible one has been found to be that of Chiao, Townes, and Stoicheff [Phys. Rev. Letters 12, 592 (1964)] and Garmire, Pandarese, and Townes [ibid. 11, 160 (1963)], originally formulated to explain the production of hypersonic waves through electrostriction at larger energy densities. The theoretical prediction is that the effect should grow linearly with the laser-beam intensity and increase very rapidly with the index of refraction.
Distorted-Wave Interaction at Boundaries
Authors: Van Buren A. L. , Breazeale M. A.
Published: September 2005

The shift of the relative phase of the harmonics of a finite-amplitude ultrasonic wave on reflection from a plane interface can produce waveforms that are distinctly different from those generated by nonlinear effects in a single medium. Since this phase shift is a function of the angle of incidence, it is possible to produce distorted waves having an arbitrary phase relationship between the fundamental and the second harmonic. Measurements of the phase relationship between the fundamental and the second harmonic have been made using various media. These data are compared with theory. [Research sponsored by the U. S. Office of Naval Research.]
Propagation of Transients in Waveguides
Authors: Coppens A. B.
Published: September 2005

A theoretical development of the behavior of transient acoustic signals in waveguides, based on a Laplace transform solution of the time-dependent wave equation, yields relatively simple expressions for the acoustic pressure or particle velocity in certain cases, including that of a gated sine wave. A general prediction of the theory that the initial portion of the acoustic packet propagates with the free-field speed of sound, whether the walls of the waveguide are “soft” or “hard,” is experimentally confirmed. This disagrees with the experimental results of Lastovka and Carome [J. Acoust. Soc. Am. 35, 1279 (1963)]. Estimates are also made for the errors introduced by waveguide effects in measusing the speed of sound by echo-superposition techniques. [Work supported in part by U. S. Office of Naval Research.]
Free-Molecule Propagation and Sound Propagation in Rarefied Gases
Authors: Sessler G. M.
Published: September 2005

At sufficiently low gas pressures and small separations between sound transmitter and receiver, the propagation of acoustic waves in gases is basically different from ordinary sound propagation. The propagation in this case is called “free-molecule propagation.” In terms of the fundamental parameters—namely, the reference wavelength λ0=2πc0/ω (c0 is the adiabatic sound velocity)—the mean free path Λ of the gas molecules, and the transmitter-receiver separation s—free-molecule propagation and sound propagation can be distinguished as follows. Free-molecule propagation occurs at sufficiently small values of s/Λ. In this case, k⋅λ0 (k is the propagation constant) depends on λ0/s and is independent of Λ. Sound propagation occurs at sufficiently large values of s/Λ. Then k⋅λ0 depends on λ0/Λ and is independent of s. Additional information on free-molecule propagation and sound propagation has been obtained from measurements taken in air and in argon [E. Meyer and G. Sessler, Z. Phys. 149, 15–39 (1957)]. For example, the range of free-molecule propagation and sound propagation are found to be approximately s/Λ<1 and s/Λ>5, respectively. These ranges are about the same for air and for argon. Also, the measurements confirm the predicted dependence of free-molecule propagation on λ0/s.
Generalized Scattering Theory for One, Two, and Three Dimensions
Authors: Harrison Mark
Published: September 2005

In the acoustical literature, one notices that for a wide variety of boundary conditions the expressions for the scattering cross section can be cast into a similar form. This suggests that the form of the results are independent of the boundary conditions. It can be shown that, by using asymptotic arguments, the usual expressions for the scattering cross section can be derived. The phase angles are obtained, in general, by an integral-equation technique. When boundary conditions are available, this method reduces to the usual expressions found in the literature. When the index of refraction of the medium varies continuously and there are no boundary conditions, this method permits one to solve the problem by the same general method. This method can be used for problems where the index of refraction varies either continuously or discontinuously in one, two, or three dimensions. The style of arguments is the same for all number of dimensions and the form of the results exhibits a unity of form. [Work supported by the U. S. Office of Naval Research.]
Volume-Scattering Measurements at 12 kc/sec in the Western Pacific
Authors: Batzler W. E. , Vent R. J.
Published: September 2005

A 12-kc/sec directional transducer aimed vertically downward was used to obtain underwater volume scattered data in 14 deep-water areas in the Western Pacific. The geographical region visited lies between 5° and 45°N latitude and 115° and 175°E longitude; included are observations in the Philippine, Celebes, and Sulu Seas, the Kuroshio Current system, and in the Main Pacific Ocean Basin. The results obtained are presented graphically as volume-scattering strength versus depth. Approximately 90% of the scattering strength values obtained fall in the range −100 to −60 dB. Considerable variations in scattering strength were found from area to area; in general. the more southerly areas have the lower scattering strengths and the less pronounced scattering layers. Most-intense scattering occurred at a station northeast of Honshu, Japan, where layers 500 ft thick with a peak scattering strength of −60 dB were observed. Comparison of the Western Pacific results with deep-water data off Southern California shows, with few exceptions, that in the latter case peak scattering is higher, layers are more pronounced, and there is more evidence of diurnal migration than was observed in the Western Pacific.
Scattered Fields from the Ocean Bottom
Authors: Thompson K. P. , Flowers Kenneth D. , Hurdle Burton G.
Published: September 2005

An experiment to determine the acoustic field scattered from the bottom has been conducted in the Blake Plateau. Pulsed 19.5-kc/sec transmissions, directed toward the bottom, were made from a submarine as it approached and passed three special radio sonobuoys. The buoys were spaced to provide the maximum coverage of the scatter field in the half-space above the bottom. Reception was made at each of the three buoys and at the source transducer. Sets of data were taken with the axis of the 8° source beam incident on the bottom at grazing angles between 15° and 90°. The data were analyzed on the basis of an isotropic-scatter model. The velocity gradient was taken into account in the computation of the propagation. Results of the spatial field are presented in polar form in terms of the scattering strength. When integration over the scattered field is performed, the ratio of power scattered from the botom to incident power shows a decrease as the incident grazing angle is decreased from normal incidence. This implies that the loss of acoustic energy into the bottom is increased as the incident grazing angle is decreased in the range investigated.
Reflection of Explosive Pulses from a General Layered Sea Floor
Authors: Hastrup O. F.
Published: September 2005

A further development of the matrix formalism suggested by W. T. Thompson in 1950 has been used to calculate the complex transfer function and reflection coefficient from a multilayered general sea floor, taking both compressional and shear waves and their damping into account. The method is used to calculate reflection-loss isolines for models with up to three general layers. The results are given in terms of the angle of incidence and of the wavenumber. The influence of shear waves and their damping has been investigated using a two-layer model and several wavenumbers. The theory of linear systems combined with numerical Fourier transformation and inversion is used to obtain the shape of a general pressure pulse after its reflection from a general multilayered sea floor. The method is used to calculate the reflected shape of both the shock and bubble pulses after reflection from models with up to three layers.
Cavitation Caused by Explosive Shock Pulses Reflected from the Sea Surface
Authors: Chapman R. P. , Scott H. D.
Published: September 2005

The characteristics of shock pulses reflected at normal incidence from the sea surface are examined as a function of charge weight and charge depth. The reflected tension pulse possesses a number of prominent features such as an initial spike, which is considerably shorter than the incident shock pulse, followed by a plateau that becomes longer for shallower or larger charges. These differences between the shapes of the incident and reflected pulses are attributed to cavitation effects near the sea surface. The acoustic results indicate that cavitation takes place for charge depths considerably greater than the maximum depth at which visible effects are caused at the sea surface. The acoustic effects indicate a breaking tension at least a factor of 10 smaller than that indicated by the visible effects.
Pulsed Creeping Waves in Acoustics
Authors: Überall H. , Doolittle R. D. , McNicholas J. V.
Published: September 2005

Scattering and diffraction of acoustic waves from spherical and cylindrical obstacles can be explained in terms of “creeping waves,” i.e., waves that run around the perimeter of the scatterer. The theory of “creeping waves” is reviewed and extended to pulse shapes of arbitrary form. With pulses incident upon a “soft” cylinder, it is found that the reflected pulse has the same form as the incident pulse and also that “creeping-wave pulses” are generated on the surface of the cylinder. These “creeping waves” arise from a Sommerfeld-Watson transformation on the classical normal-mode series. The “creeping waves” propagate around the cylinder any number of times with a velocity less than the velocity of sound. Formal studies of causality show, e.g., that outside the shadow region besides the incident and reflected pulses (which obey geometrical acoustics) we also find “creeping-wave pulses” that have encircled the scatterer and then radiated tangentially to distant observation points. [Work supported in part by the U. S. Office of Naval Research.]
Underwater Steam Sound Source
Authors: Wisotsky Serge
Published: September 2005

Various underwater full-scale steam-exhaust configurations were investigated. The greatest sound-pressure level (SPL) was obtained with steam pulsed through a long tube, i.e., with a length/diameter ratio>10. This intermittent technique generated a collapsing, coherent bubble that produced a SPL 50 dB higher than that obtainable with modulated steady-state flows. For example, the peak presure, near the surface, with a 4-in.-diam nozzle was from a 14-ms acoustic impulse having an on-axis SPL of 141 dB//1 μbar/1 yd. Air injection causes damped oscillations. A water-hammer phenomenon was observed with significant acoustic power output. Oscillograms and movies were made of the bubble phenomena. [Work supported by the U. S. Office of Naval Research.]
“On-Line” Computing in Speech Research
Authors: Denes P. B.
Published: September 2005

Computer-simulated models offer great advantages to speech research. However, normal computer operation involves delays in obtaining results whilst effective study of certain speech models requires that the experimenter immediately observe the effects of varying his model. A computer installation for speech analysis and synthesis that does provide such direct man—computer interaction is described and experience gained so far is discussed.
Analysis-Synthesis of Glottal Excitation
Authors: Michaels Sheldon B. , Strong William J.
Published: September 2005

Methods described in a previous paper [J. Acoust. Soc. Am. 37, 1186(A) (1965)] were used to determine the time-dependent glottal volume velocity and external pressure waveform for a speaker phonating /a/. By comparing these two quantities. it was possible to determine the approximate time dependence of the first three formants of the vocal tract. The glottal volume velocity was applied to the input of a variable POVO having appropriate time-dependent formants and the output was compared with the pressure waveform. “Standard” triangular waveforms and “standard” waveforms in which the relative phases of the harmonics had been modified were also used to excite the POVO. Psychoacoustic evaluations were made of the natural quality of the various synthesized outputs.
Analysis of Labial Movement
Authors: Lindblom Bjorn E. F. , Soron Henry I.
Published: September 2005

Motion pictures of the activity of the lips in speech production have been taken with synchronized tape recordings. In the search for a limited set of lip parameters, advantage was taken of the fact that a three-dimensional approximation of the lip contours can be made by means of segments of power functions and straight lines. Thus the locations of the contours in the midsagittal plane and at the mouth corners were measured and found to make possible fairly accurate predictions of shape and opening area. Data on the jaw dependence of the parameters selected were acquired for each talker individually. By subtracting the contribution of the jaw opening in all parameters, an estimate of labialization proper was obtained. The interdependencies among these that appear on applying this correction are discussed and their time-varying behavior in vowels is illustrated and related to spectrographic indications of lingual articulation in adjacent consonants.
Environmental Influence on the Speech of Family Members Shown by Spectrographic Speech Matching
Authors: Kersta L. G.
Published: September 2005

Speech-spectrogram pattern-matching experiments were conducted on the members of an 8-family population. A 5-word vocabulary was excerpted from normal and whispered narrative speech for each member of the 8 families, which consisted of father, mother, and a 5- or 6-yr-old son. Trained subjects were able to select the proper parents of children for 28% of the attempts. When children were matched with parents separately, the success was 17.5% with mothers and 21.5% with fathers. Husbands were spectrographically matched with wives correctly in 23% of the attempts. The chance expectancy of a correct match in the above data was approximately 2.0%. Sufficient evidence of familial influence in speech patterns was found to suggest that a more sophisticated experiment be performed.
Application of Digital Inverse Filtering to the Systematic Variation of Vowel Parameters
Authors: Hemdal John F.
Published: September 2005

The Levin time-domain method of the least-squares determination of impulse functions and filter characteristics has been applied to the measurement and systematic variation of vowel parameters. This technique determines the optimum filter that equalizes or transforms a given time series into a related or derived series. For example, when given a typical vowel waveform and its corresponding estimate of the laryngeal excitation, the method produces the impulse response of the vocal tract during the production of that vowel. This results in a time-domain approach to analysis by synthesis. Alternately, this method may be used for the systematic variation of certain vowel parameters. In this instance, one vowel waveform of a given speaker and a second vowel waveform of the same speaker are considered as input and output waveforms; the impulse response obtained from this equalizing technique exhibits the characteristic difference between the vocal-tract configurations of the two vowels. In addition, this filter impulse response may be systematically varied and convolved with different vowel waveforms, producing quasinatural vowels. These quasinatural vowels, in turn, may be useful in the determination of a listener's perceptual bounds. This paper describes the mathematical formulation, its implementation on a digital computer, and some preliminary results.
Reduction of Plumbing Noise in a Multistory Building
Authors: Wetherill Ewart A.
Published: September 2005

In recent years, the existence of unwanted plumbing noise in multistory buildings has received considerable attention, and much effort has been devoted to the development of construction techniques for minimizing it. This paper describes a typical problem situation in a hotel/apartment-building type. Measured data are presented to illustrate the degree of improvement achieved by simple modifications.
Behavior of Three Floors for Different Types of Impact
Authors: Hehmann H. W. , Mariner T.
Published: September 2005

Impact-noise tests were conducted in the laboratory on three floor constructions: (a) 4-in. concrete slab; (b) and (c) two floated-wood-floor constructions on the same slab (all floors surfaced with the same plastic floor covering). Data were taken using the standard tapping machine and real footfall. Whereas the floated-floor constructions had impact-noise rating (INR) (FHA Bull. 750) 13 and 15 dB greater than the surfaced slab, they did not show a substantial improvement for footfall. Moreover, the floated floor with the higher INR was poorer under footfall.
Sound Transmission through a Periodically Supported Thin Plate
Authors: Barger James E.
Published: September 2005

The transmitted and reflected sound fields resulting from a plane sound wave incident upon a periodically supported thin plate have been computed. The supports are taken to lie parallel to the incident wavefronts and to offer an impedance to transverse plate motion that is uniform over their length. A measure of the sound transmission that is of particular interest is the transmission loss, defined as TL=−10 log [intensity of the main transmitted wave/intensity of the incident wave]. In this definition, the main transmitted wave is the component wave in the transmitted sound field that has the same direction as the incident wave. The effect of the supports on TL are presented as a function of acoustic frequency, incidence angle, support spacing, support impedance, and the density of the fluid in which the supported plate is. immersed. In addition, the effect of plate damping on TL is shown to be much larger for supported plates than for free plates. [Work supported by the U. S. Navy Underwater Sound Laboratory.]
Response of Acoustic Resonators to Sonic Booms
Authors: Simpson James D. , Lowery Richard L.
Published: September 2005

The analysis of the data collected in the Oklahoma City sonic-boom program gives strong indications that various structures can be excited as Helmholtz resonators. Pressure oscillations were observed to persist in one test house for 1–2 sec after the passage of the shock wave. On occasion, the peak inside pressure exceeded the overpressure of the ground wave. Since the Oklahoma City tests had insufficient flexibility to allow a complete analysis of pressure response, additional studies have been conducted through the use of a 32-ft plane-wave tube, at Oklahoma State University. This paper discusses transient-response spectra of the Helmholtz resonator with the attendant structural response considerations. Experimental results from the Oklahoma City tests and from the laboratory simulations are presented. Mathematical projections are made of possible pressure responses resulting from the flight of a hypothetical supersonic transport.
Maximum Hearing-Aid Sound Pressures Attainable with Available Variable-Reluctance Receivers Utilizing Powers up to 150 MW
Authors: Sebesta G. J. , Carlisle R. W.
Published: September 2005

On a receiver having moderately wide range, an average level of 142 dB is attained with 100 MW input; on a narrower-band unit, a level of 148 dB is attained with the same power. Some constructional details are given.
Temporary Threshold Shifts due to Single Clicks
Authors: Silbiger H. R.
Published: September 2005

Previous research has concentrated on the effect or series of repetitive clicks on the auditory threshold. The present study investigated the effect of single clicks. The clicks were generated by discharging a capacitor across a dynamic earphone. Momentary peak sound-pressure levels of maximally 158 dB SPL were developed. Pre- and post-exposure thresholds were obtained at 2, 3, 4, 6, and 8 kHz, using a Békésy-type recording attenuator. Threshold measurements were made at 3-min intervals post exposure. Temporary threshold shift is normally in the direction of elevating the threshold. Results obtained in this study were in the direction of a lowering of the threshold, which was termed “negative TTS,” The usual large spread in TTS susceptibility was found. The maximum positive TTS was in the 6- and 8-kHz region, while the maximum negative TTS was observed at the 3- and 4-kHz region. The recovery from negative TTS followed the same time course as recovery from positive TTS. Subjects showing the greatest positive TTS yielded the smallest negative TTS.
Click Heard when a Musical Tone Makes an Abrupt Frequency Change
Authors: Neustadt Herbert M.
Published: September 2005

Development work on new electronic musical instruments has revealed that (1) the ear hears a click when a musical tone makes an abrupt frequency shift; (2) the click is prominently loud when the shift is over a range as large as an octave, but barely perceptible when the range is as small as a semitone; (3) the click is sometimes absent in the sound output of a wind musical instrument making a large abrupt frequency change; (4) the only way that such a clickfree frequency change can be made is by means of a simultaneous interruption of amplitude; (5) when a large frequency change is accompanied by a properly shaped simultaneous interruption of amplitude, the ear hears no click and does not perceive the amplitude interruption. These facts are demonstrated with the aid of an electronic tone source. It is shown that the click sensation can be plausibly explained by considering it to be the readout of a fast-acting frequency analyzer. [Electronic tone source used for elucidating the click was built with support from the U. S. Office of Naval Research.]
Differential Sensitivity for Duration
Authors: Milburn Braxton
Published: September 2005

Differential sensitivity to acoustic signal duration was determined for a group of normal-hearing, well-practiced listeners. A modified psychophysical method of limits was employed to obtain a difference limen for each of 18 combinations of stimulus frequency (250, 1000, and 4000 cps), sensation level (10 and 50 dB), and reference duration (300, 500, and 1000 msec). Treated as a randomized complete block 2×3×3 factorial design and submitted to an analysis of variance, the data permit the following conclusions: (1) the magnitude of the relative difference limen is considerably smaller than has been reported previously; (2) the magnitude of the relative difference limen of pure-tone auditory stimuli is related to duration of the reference stimulus and to sensation level; (3) the magnitude of the relative difference limen of duration is not highly dependent on stimulus frequency; (4) relative to duration judgments, Weber's law appears to be valid only in the case of midrange and relatively high stimulus frequencies and at a moderately high sensation level.
Three Time Parameters of Central Masking
Authors: Zwislocki Jozef J. , Buining E. , Glantz J.
Published: September 2005

Previous studies have demonstrated that central masking depends on the time pattern of stimulus presentation. When both the masking and masked stimuli are pulsed simultaneously, the masking effect amounts to about 10 dB, when the masker is continuous and the test stimulus pulsed the effect decreases to a few decibels. In order to study the temporal course of central masking more analytically, the contralateral threshold shift was investigated by means of 10-msec tone bursts. The masker consisted of 250-msec bursts of tone or random noise, repeated at a rate of one per second, and the test bursts were presented at various time intervals from the onset of masking bursts. The results show that the contralateral threshold shift may exceed 10 dB at the onset of masking bursts. It decreases rapidly within about 15 msec and more slowly thereafter. In a matter of 200 msec, it is reduced to a few decibels. When the intermittent masking and test stimuli are turned on simultaneously, the threshold shift gradually decreases by several decibels and reaches an asymptote in a matter of minutes.
Monaural Phase Effects in Auditory Signal Detection
Authors: Fricke James E.
Published: September 2005

The study was designed to investigate the detectability of a complex signal as a function of the relative phase of the components of that complex waveform. The complex waveform was composed of a fundamental of 525 cps and a second harmonic of 1050 cps. Six phase relations of the second harmonic with respect to the fundamental were used: 0°, 30°, 60°, 90°, 120°, and 150°. Four 100-msec bursts of noise, one of which contained the complex waveform, were presented to four subjects and they were forced to choose which of the four contained the signal. Ten different signal-to-noise ratios, covering a range of 10 dB around a subjective “threshold” value, were utilized. Forty judgments at each of these signal-to-noise ratios were used in each phase condition. The statistical analysis of the data revealed significant differences with regard to the detectability of the complex signal as a function of the relative phase of the components. It appears that relative phase of the components must be taken into account in a model for the detectability of complex signals—at least for a complex signal composed of two components. Three theoretical explanations of the results were suggested.
Time-Separation Pitch Associated with Correlated Noise Pulses
Authors: McClellan Max E. , Small Arnold M.
Published: September 2005

Previous research has shown that time separation pitch (TSP) is perceived when one train of ac or dc pulses is delayed with respect to another identical train and both are presented monaurally to a subject. TSP has been shown to be related to the reciprocal of time separation between leading edges of proximal pulses of the two trains. On the asumption that TSP is mediated by an autocorrelational process, it was predicted in the present study that, contrary to previous findings, TSP would obtain using noise pulses, providing that proximal pulses were highly correlated. Correlated noise pulses were produced by an “acoustic-delay” system. Uncorrelated noise pulses were used for a control condition. The subject's task was to match a pure tone to the pitches associated with the pulse trains. It was found that a much greater number of pitch matches was made to the reciprocals of time separations bounded by correlated than by uncorrelated noise pulses. The results of this study tend to support the notion that TSP is perceived by virtue of an autocorrelational process.
Pitch Discrimination of Jittered Pulse Trains
Authors: Rosenberg Aaron E.
Published: September 2005

Listeners are presented with pulse-train stimulus pairs and asked to judge whether they can hear a difference between them. The interval between pulses is a random variable, identically and independently distributed for each stimulus of a pair. Two distributions are observed: one, nominally Gaussian, and the other, nominally the distribution of the amplitude of a sinusoidal wave whose phase is uniformly distributed. The principal experimental parameters are the mean interval between pulses and rms deviation or jitter about this interval. The stimuli of each pair are identical in other respects, pulse shape (50-μsec pulse width), loudness (30 or 35 dB SL), but differ in polarity pattern. Two pattern combinations are observed. For the range of mean pulse intervals investigated, 5–15 msec, the stimuli of each pair are generally discriminable when unjittered. However, the results indicate that, when jittered in amounts greater than 1 or 1.5 msec, the stimuli may be rendered nondiscriminable. This critical amount of jitter coincides with a flattening of the power-density spectra of the stimuli for frequencies greater than 250 or 300 cps. On the basis of this result and the results of other investigations, it is hypothesized that the correlates of discrimination for unjittered or lightly jittered stimuli are distinct neural volley patterns associated with basilar membrane activity in the 300- to 1000-cps region.
Real-Time Formant Vocoder, Using a Filter Bank, a General-Purpose Digital Computer, and an Analog Synthesizer
Authors: Coker Cecil H.
Published: September 2005

A formant vocoder has been built as a combination of analog equipment and a digital computer. The time domain operations—spectrum filtering and synthesis—are done in analog form. The formant-extraction operations are done at modest sampling rates (about 100 scans/sec) with a DDP-24 computer. The program is a tightly coded form of an analysis method described earlier [J. Acoust. Soc. Am. 35, 1911 (1963)] and first written in FORTRAN to run slower than real time with a programmed synthesizer. Spectra from a 24-channel filter bank are scanned for a peak, a resonance curve of appropriate frequency and average bandwidth is subtracted, and the process repeated to obtain four maxima. The lower three of these, ordered in frequency, are transmitted to the synthesizer along with other control data. The analyzer, in minimal form, takes about 7 msec to run and 1 msec for input-output, leaving time for a few frills in a 10- to 12-msec scanning period. Tapes of the system are demonstrated.
On-line Computer Control of a Formant Synthesizer
Authors: Coker Cecil H. , Cummiskey P.
Published: September 2005

A solid-state formant synthesizer has been constructed and interfaced to a digital computer. The computer, a DDP-24, has a 12K memory, and a 5-μsec cycle time. The computer-synthesizer combination permits on-line, real-time synthesis of speech under program control. The synthesizer has a number of useful features: optional serial or parallel operation; high-accuracy, wide-range control of three formant frequencies (±0.50% over a 20:1 range for F1); individual control of three formant bandwidths; unusual fidelity in spectral representation; and exceptionally low noise level. Amplitude controls are accomplished by linear multipliers with 60-dB range, hence frequency responses may be run with sine waves. Programming of the system is especially simple. Controls are normalized so that formant frequencies, bandwidths, and amplitudes at resonance are specified independently and without interaction. A data scan is accomplished under program control with only three instructions additional to those for computation. Alternatively, most of the data outputting can be done through a simultaneous memory access channel, with automatic program interruption for periodic servicing of the channel. Synthetic speech produced by the computer-controlled synthesizer is demonstrated.
New Reference Signal for Speech-Quality Measurements
Authors: Rothauser E. H. , Urbanek Günther E.
Published: September 2005

In a previous paper in 1963, a method was described for the measurement of speech quality. It was proposed to utilize a reference signal consisting of high-quality speech degraded by additive white noise. Extensive studies have shown disadvantages of this form of degradation. The reference signal differs considerably in its nature from the output speech signals generated by most speech-processing devices, and in spite of its apparently simple definition it cannot be reproduced very easily, because of the difficulties in measuring the sound level of speech signals. A new reference signal is proposed. Instead of adding a controlled amount of noise to a high-quality speech signal, the noise is multiplied into the signal. One way to generate the reference signal of the form s(t)[1+k⋅r(t)] is to modify this form to s(t)+k⋅r(t)⋅s(t) and to generate the product with a Hall multiplier. Test results are given and discussed. [Research reported in this paper sponsored by the U. S. Government.]
Comparative Performance Evaluation of Conventional, Whispering, and Monotone Vocoder Speech
Authors: Voiers William D.
Published: September 2005

Speech as processed by the AFCRL polymodal vocoder operating conventionally, as a whispering vocoder, and as a monotone vocoder was evaluated for intelligibility and speaker recognizability using the diagnostic rhyme test, the Fairbanks rhyme test, and a multidimensional voice-rating procedure. Predictions as to the effects of monotone and whisper vocoderization upon diagnostic rhyme-test scores and upon the information structure of voice ratings were confirmed by the test results. [Research reported in this paper sponsored (in part) by the U. S. Air Force Cambridge Research Laboratories, Office of Aerospace Research.]
Some Effects of List Familiarity and Synthetic Segment Familiarity on the Intelligibility of PB Words Produced by Diphone Synthesis
Authors: Dixon N. R. , Maxey H. D. , Aylsworth R.
Published: September 2005

In a previous paper [S. E. Estes, H. R. Kerby, H. D. Maxey, and R. M. Walker, “Speech Synthesis from Stored Data,” J. Acoust. Soc. Am. 34, 2003(A) (1962)], a method for unlimited vocabulary synthesis was presented and discussed. The method consists of computer assembly of digital data representing diphone-length segments, together with auxiliary data specifying intonation and timing, to produce synthetic speech. In order to evaluate the adequacy of this method of segmentation, intelligilibity tests have been conducted using PB words in a standard carrier phrase. The results of these studies have indicated that intelligibility scores on synthetically produced PB words are affected by at least two major factors: familiarity with the words under test and familiarity with the diphone components of the words. A study has been designed to test the effects of these two factors on the intelligibility scores of normal adult listeners in sound-field conditions. The stimuli used are selected C.I.D. W-22 lists. These results are presented and discussed.
Dynamic Response of Thin Elastic Shells to Internal Fluid Sources
Authors: Rentzepis G. M. , Jewell R. E.
Published: September 2005

The problem of propagation of sound waves through cylindrical shells filled with a fluid is analytically investigated here. The problem is approached by, first, considering the response of thin shells due to excitations of “elementary” type, such as radial pressure waves, etc. The complete problem is afterward recovered by proper superposition of the “elementary” solutions. The “elementary” solutions are obtained by introducing a set of auxiliary functions, which, after some manipulations, simplify considerably the formal part of the method. In this paper, isotropic shells and fluids with constant velocity and temperature are considered.
Correlative Properties of Deep-Water Ambient Noise at Bermuda
Authors: Urick R. J.
Published: September 2005

Time-delay correlograms of the ambient noise picked up by various hydrophone pairs of the deep vertical array at Bermuda have been obtained. These show, as a function of time delay, the clipped correlation coefficient of the noise for the selected hydrophone pairs. In the two 1-oct bands 200–400 and 1000–2000 cps, and for wind speeds between 4 and 30 kt, different types of correlograms have been found for different combinations of frequency and wind speed. In the lower-frequency band and at low wind speeds, they show a strong peak at zero delay, consistent with the findings of Fox [J. Acoust. Soc. Am. 36, 1537–1540 (1964)] and Axelrod et al. [J. Acoust. Soc. Am. 37, 77–83 (1965)] that the noise arrives horizontally, probably from distant shipping. In the higher band and at high wind speeds, the correlograms are what would be expected for noise generated at the sea surface. The correlograms permit computation of the array gain of any arbitrary vertical array steered at any angle to the horizontal. Considerable degradation of array gain relative to uncorrelated noise is to be expected under certain conditions. A simple example is given.
Some Acoustic and Physiologic Correlates of the Breath Group
Authors: Lieberman Philip
Published: September 2005

Intonation can be analyzed, in part, in terms of the phonologic feature breath group. The results of an experiment are discussed in which subglottal air pressure and lung air volume were measured as functions of time for four speakers, in addition to the acoustic signal. These data show that the unmarked or “normal” breath group of American English is characterized at the articulatory level by a relatively steady tension of the laryngeal muscles throughout a single-bounded expiration of air from the lungs. At the end of phonation, the subglottal air pressure falls and hence the fundamental frequency also decreases repidly. The intonation pattern that is a consequence of this breath group is, of course, the one that has been described as characteristic of declarative sentences and numerous other languages. The marked breath group is differentiated at the articulatory level by an increase in the tension of the laryngeal muscles at the end of phonation, which counteracts the falling subglottal air pressure to produce a not-falling terminal fundamental frequency contour. The intonation pattern of the marked breath group is the characteristic intonation pattern of the “yes-no” question in English and other languages.
Coarticulation of Some Disyllabic Utterances Measured by Electromyographic Techniques
Authors: Harris Katherine S. , Huntington Dorothy A. , Sholes George N.
Published: September 2005

The fact that coarticulation occurs has long been known. Recent work has shown, for example, that a subsequent vowel has spectrographic effects on a preceding vowel-consonant sequence. Furthermore, at least some of these coarticulation effects may be seen in the sequence of positions assumed by the articulators, as observed in x-ray motion pictures and by articulatory position sensors. These facts, hawever, still leave open the question of the extent to which the complex gestures of speech are reorganized at the level of the muscular contractions. The present study was designed to provide information on this question by measuring the muscle-contraction patterns of various consonant-vowel sequences. [Research supported in part by the National Institute of Dental Research, National Institutes of Health, U. S. Department of Health, Education, and Welfare.]
Formulation of the Brass-Instrument Bore Problem
Authors: Benade A. H.
Published: September 2005

The boundary conditions for waves in a brass instrument require a pressure amplitude large in the mouthpiece and small at the open end. The nth vibrational mode of the air column therefore consists of (2n−1) standing quarter-wave segments For a horn whose cross section is given by πr2(x), Webster's equation is [∂2/∂x2−U(x)]ψn=−(ωn/c)2ψn. Here, ψn=πr2pn(x), and pn(x) is the pressure distribution for the nth mode, whose oscillation frequency is ωn. U(x)=(d2r/dx2)/r is the product of the longitudinal and transverse curvatures of the horn at x. The similarity with Schrödinger's equation gives considerable insight. WKB methods show a useful relation between the length Ln of a cylindrical pipe (whose nth frequency matches that of the horn) and the value of x at the point where U(x)=(ωn/c)2. For music, the resonances should approximate a harmonic series, so Ln=L1⋅(2n−1)/n. The choice of useful horn types is discussed, including effects of bell shape on starting and maintaining self-sustained oscillations. Effects of bore discontinuities and estimates of radiation and wall damping are described. Experimental data are presented and musical implications drawn.
Describing the Speech Process in Motor Command Terms
Authors: Cooper Franklin S.
Published: September 2005

The events of speech production can be characterized variously by the sounds that are produced (whether assessed subjectively or spectrographically), by the movements and shapes of the vocal and pulmonary tracts, or by the neuromuscular activities that cause these movements and changes of shape. There is a growing interest in experimental studies of the neuromuscular events, owing in part to the fact that they occur early enough in speech production to escape some of the encoding imposed on the message. As a working hypothesis, one might suppose that some limited set of the neural signals to the articulatory muscles would retain a one-to-one correspondence with linguistic units of the message. Such characteristic neural signals, i.e., the motor commands, could then provide a most useful basis for a phonological description of spoken language; further, such motor commands might well play a significant rôle in speech perception. The experimental technique that is most directly applicable is electromyography. Early results from several laboratories seem highly encouraging, though many problems of both method and interpretation remain to be solved.
Relations among Behavioral Thresholds, Cochlear Potentials, and Cochlear Injury following Traumatic Exposures to a Wide-Band Noise in the Chinchilla
Authors: Eldredge Donald H. , Miller James D.
Published: September 2005

Four monaural chinchillas were exposed to white noise (0.25–10 kHz) at 108 dB SPL for 220 min. Behavioral thresholds were measured subsequently for three months. Permanent threshold shifts of about 60 dB were observed for frequencies above 1400 Hz and of about 10 dB for frequencies below 1000 Hz. Cochlear microphonics were measured differentially. In the three turns of the cochlea, the respective losses of sensitivity were more than 60, 30–60, and 15–50 dB. The losses of maximum available microphonic voltage were more than 40, 25–30, and 10–20 dB, respectively. The whole-nerve action potential in response to clicks contained no component that could be masked by high-frequency noise and is attributed only to more-apical parts of the cochlea. Hair cells were entirely or almost completely absent in the basal turn, markedly reduced in the second turn, and nearly normal in the third turn. Corresponding losses of neurons were observed at the different levels of the spiral ganglion. [Work supported by National Institutes of Health, Public Health Service, U. S. Department of Health, Education, and Welfare.]