bluuebunny/crossref_metadata_2025 huggingface dataset:
Transient Behavior of Nonlinear Isolation Mountings
Authors: Snowdon J. C.
Published: September 2005

The response of damped nonlinear isolation mountings to step-like foundation displacements has been determined theoretically. Both “stiffening” and “softening” mounts have been considered. Utilization of the techniques of numerical integration (performed by a digital computer) rather than perturbation methods has made possible the consideration of large departures of mount stiffness from linearity. For brevity, the results presented relate only to a value of mount damping (linear) equal to five percent of critical damping. For the majority of step rise times considered, the softening mount is shown to reduce both the acceleration and the displacement of the mounted item below the values observed for a linear or a stiffening mount. The reduction in acceleration can be comparable with 10 db, and it is frequently the magnitude of the acceleration to which the likelihood of damage to the contents of the mounted item may be related. (This investigation was conducted at the University of Michigan, Ann Arbor, Michigan, under the sponsorship of the U. S. Navy, Bureau of Ships.)
Combined Bendings—Torsion of a Closed Beam with the Addition of Axial Loading
Authors: Saunders Herbert
Published: September 2005

The basic Myklestad method for the determination of combined bending-torsion loads on a beam is supplemented by the addition of axial loading. The beam is considered to consist of finite lumped masses and their associated springs. The procedure employed is the “transfer matrix” approach making the technique readily applicable to the digital computer. The author extends a note originally published by him [J. Acoust. Soc. Am. 32, 409 (1960)] concerning the addition of rotary inertia and shear deformation. As examples, the author determines (1) the buckling load of a static column and (2) the critical length to prevent buckling of a thin column loaded by its own weight.
Power Flow between Structural Modes and Reverberant Fields
Authors: Lyon Richard H.
Published: September 2005

The response of a resonant structure in a reverberant noise field is viewed as the attempt of a set of damped oscillators to come to equilibrium with a thermal bath. The asymptotic (in time) energy in the modes is evaluated from a simple energy equivalent circuit and depends on the ratio of internal to radiation resistance and the “temperature” of the sound field. The connection between these results and those of P. W. Smith, Jr. [J. Acoust. Soc. Am. 32, 929(A) (1960)] is demonstrated. Experimental illustration of the method is included. (Supported in part by Wright Air Development Division.)
Some Structural Design Features of the WADD Sonic Fatigue Test Facility
Authors: Neubauer Donald J. , Williams Lloyd J. , Ungar Eric E.
Published: September 2005

Construction of a large high-intensity sonic fatigue facility is under way at Wright-Patterson Air Force Base, Dayton, Ohio. No previous large structure has been required to withstand long term intense excitation at all frequencies in a wide band, so that structural design of this facility was faced with some unprecedented problems. The present paper outlines the test chamber size and performance requirements and how these led to the present design. The present structure is described, and design of viscoelastically damped configurations is discussed. The sizes and design stress levels of typical beams and panels in this structure are compared to those encountered in conventional structures. Some interesting details of the structure are pointed out.
Vibrations in a Large Softly Clamped Glass Plate
Authors: Lange James N.
Published: September 2005

The 3×2.5 square-meter glass plate was driven electrodynamically at the center and the amplitude distribution as a function of the frequency was recorded for positions out to and along the edge of the plate. The loss factor was measured by both bandwidth and decay methods. Transients were excited by a shot from a blank pistol and photographs of the resulting oscilloscope pattern were taken. (This work was sponsored by the U. S. Atomic Energy Commission and Bureau of Naval Weapons.)
Noise and Vibration Measurements of a Large Ventilating Fan for Various Degrees of Vibration Isolation
Authors: Miller Laymon N.
Published: September 2005

We were recently presented with the opportunity of making a series of noise and vibration measurements on a 25-hp ventilating fan located above an occupied floor in an office building. The measurements were carried out during the process of trying to diagnose the cause of excessive noise levels in the office floor beneath the fan. Vibration levels at various parts of the fan base and noise levels in the office area below are presented for various degrees of isolation of the fan on its mounting.
Measurement of the Musical Interval Sense
Authors: Pikler Andrew G. , Harris J. Donald
Published: September 2005

Ten generally unmusical Ss were used in an effort to dissect and quantify the hierarchy of cues which enable the trained instrumentalist or singer to create a progression of musical intervals with great precision, speed, and consistency. Experiment A required Ss to create one at a time, by adjustments, the eight intervals of the major scale (C fixed at 262 cps). Errors were commonly half a semitone. The Octave was almost always significantly and seriously sharp. Evidently the intervals in isolation sustain an average error of up to a quartertone. Experiment B required Ss to set seven oscillators arranged in a rudimentary electric piano to create the scale. The intervals showed improvement in placement and variance, though deviations were still commonly a fourth of a semitone. Evidently the context of even a very primitive musical line is a distinct additional cue in pitch intonation. Experiment C was as B except that the tonic and octave were both fixed. With this additional anchor, placement of all tones was improved; for example, the fifth improved from 48 cents sharp to 21.5 cents sharp. Evidently errors increase with distance from an anchor. The further cue of harmony, not studied here, would suffice to bring intonation errors within limits of the DL.
Computation of Far-Field Characteristics of a Transducer from Near-Field Measurements Made in a Reflective Tank
Authors: Baker D. D. , McCormack K.
Published: September 2005

The directivity pattern and source level of a large, cylindrical transducer were computed by using Kirchhoff's formula with a simple approximation for the normal pressure gradient. The integration was performed numerically over a finite cylinder. Near-field pressure amplitude and phase measurements were made over the surface of integration by rotating the transducer and measuring the output of a line hydrophone less than 1 wavelength from the active face. The measurements were made on the leading edge of the received pulses—before steady-state conditions were reached. This was done to simulate this transducer operating in a tank of diameter only about twice that of the transducer. Under such conditions, interference due to tank-wall reflections would begin when the amplitude of the pulse envelope was as much as 2–3 dB below the steady-state level. Computed directivity patterns agreed well with equivalent measured ones, but, naturally, computed source levels were 2–3 dB low. A correction term was derived from the first portion of the pulse risetime curves, and was applied to the source-level values to give good agreement with measured ones. The present results indicate that accurate transducer calibration can be obtained in reflective tanks considerably smaller than those used in previous work.
Sequential Phenomena in Signal Detection
Authors: Carterette Edward C. , Atkinson Richard C.
Published: September 2005

A series of experiments designed to assess the limit of a human observer's level of sensitivity is reported. A mathematical theory of the detection process is introduced which explicitly considers the sequential effects observed in psychophysical data. This variable-threshold model for signal detection is an axiom system based on a stimulus-sampling theory of the learning process. The theory generates predictions for all aspects of an observer's response protocol, e.g., mean response probabilities, associated variances, sequential predictions such as autocorrelations, runs, etc. It thereby permits a detailed treatment of individual trial-by-trial data. Experimental data are presented for both yes-no and forced-choice auditory-detection situations, with and without information feedback to the observer. The model is compared to signal-detection theory [Tanner and Swets (1954); Green (1960)] and to another threshold model [Luce (1961)].
Masking with Two Sinusoids
Authors: Green David M.
Published: September 2005

The masking produced by 2 continuous sinusoids located equidistant in frequency from the signal, a pulsed sinusoid of 110 sec, was studied. The 2 sinusoidal maskers were of equal energy. Their separation in frequency Δf was varied. The main parameter of the experiment was the frequency of the pulsed signal. The results can conveniently be summarized in one graph. Along the abscissa, we plot the frequency separation of the maskers (Δf). Along the ordinate, we plot the intensity of the pulsed signal (Is) needed for some constant level of detectability. As the frequency separation of the maskers increases, the curve descends, indicating the pulsed tone is becoming easier to hear (Is∝1/Δf). For moderate frequency separations (Δf<100 cps), the results are independent of center frequency. For example, essentially the same signal intensity in SPL is needed at 250, 1000, or 4000 cps to overcome the masking produced by 2 continuous tones when these maskers are located 20 cps above and below the frequency of the signal. At much wider frequency separations, the results are dependent on center frequency, and a critical band of a kind may be measured. The width of the “critical band” inferred from this experiment is about the same size as the half-response points of the curve relating the amplitude of the traveling-wave envelope to frequency; i.e., the “critical band” is nearly 600 cps wide at 1000 cps. [This work was supported in part by the U. S. Army Signal Corps, the U. S. Air Force ESD Contract AF19(604)-7459, the Office of Naval Research, and in part by the National Science Foundation (Grant G-21807).]
Onset Discrimination for White Noise
Authors: Pollack Irwin
Published: September 2005

The discrimination of the rate of onset of a white noise was examined for exponential onsets. Upon a given trial, either one of two onsets was presented to the listener. His task was to vote whether a given onset was “fast” or “slow.” On successive trials, the shorter onset time was held constant. The longer onset was varied to achieve 75% correct discrimination. The average ratio of the longer to the shorter onset was determined for each of 5 listeners. In terms of the interval between 10%–90% of asymptotic level, the average ratio varied from about 4.5 to about 1.5 as the shorter onset time varied from 1 to 500 msec. Over the range of onsets from 5–100 msec, the ratio was about 2.0. That is, an increment of 100% in the onset of a white noise is required over a wide range of onsets for 75% correct discrimination.
Low-Frequency DL's in the Presence of a High-Frequency Noise
Authors: Campbell R. A.
Published: September 2005

A high-pass filtered noise will increase the difference limen (DL) for frequency for a simultaneously presented low-frequency, unmasked sinusoid. Since only the basal end of the basilar membrane should be interfered with by such filtered noise, this would seem to indicate the participation of the basal end in the perception of a low-frequency sinusoid, contrary to a traditional place theory. These data are from 20 normal-hearing listeners making either unilateral, alternate-pitch matches or constant-method judgments for a 150-cps sinusoid at a 5- to 50-dB sensation level (SL). Two interfering noises, one periodic and one aperiodic (either unilateral or contralateral to the sinusoid), were presented at the maximum level possible without masking the sinusoid at its threshold. Both noises contained essentially no energy below 400 cps. The frequency DL's were increased approximately 20% by all the noises, regardless of ear of presentation. The increase would thus appear to be attributable to a general, bilateral phenomenon, such as distraction, rather than the interference with the basal end of the ear receiving the sinusoid. [This work is based on a Ph.D. thesis completed at the State University of Iowa under the direction of Dr. Arnold M. Small, Jr., and was supported in part by the National Science Foundation.]
Acoustics of Philharmonic Hall, New York
Authors: Beranek Leo L. , Johnson Russell
Published: September 2005

Philharmonic Hall, part of the new Lincoln Center for the Performing Arts in New York City, seats 2644 and has a cubic volume of 865 000 cu ft. It is designed exclusively for symphonic concerts, although during its early years it will serve a variety of purposes. This paper describes the acoustical design, the activities of “tuning week,” 28 May to 2 June 1962, and the experiences during the first month of its operation following its dedication on 23 September 1962. Data taken during “tuning week” indicate that its midfrequency reverberation time will be approximately 2.0 sec, and the ratio of reverberation at low frequencies to that at midfrequencies approximately 1.25.
Jacksonville Civic Auditorium
Authors: Klepper David L.
Published: September 2005

The new Civic Auditorium in Jacksonville, Florida, consists of 3 major spaces: a 3200-seat auditorium, a 600-seat theater, and 20 000 sq ft of exhibition space. The main auditorium will accommodate a variety of speech and music activities. It has a volume of 1 million cu ft and a design reverberation time (full) of 1.7 sec at midfrequencies. Essential to the acoustical design is a demountable, plywood stage enclosure for orchestral concerts. The theater is plannned primarily for amateur dramatic productions, lectures, and occasional small recitals. It has a volume of 118 000 cu ft and a design midfrequency reverberation time of 1.1 sec. “Central” sound-amplification systems in the auditorium and the theater supplement the “natural” acoustics. The sound-absorbing treatment for the exhibition hall is also discussed, with the details of the distributed loudspeaker system which provides flexibility and adequate coverage for the varied exhibition-hall activities. The paper compares the predicted reverberation times for the auditorium and theater with measurements obtained following completion of the building. Recommendations and later compromises for the sound-isolating construction are discussed. Also discussed is the interrelation of room acoustics, electroacoustics, and noise-control design considerations, and the importance of attention to details in achieving the successful over-all acoustical environment.
Vibration of a Rotating Beam with Tip Mass Analyzed by a Variational Method
Authors: Craig Roy R.
Published: September 2005

Hamilton's principle has been used in determining a variational principle suitable for moderately large displacements of a beam which is initially stress-free or the small displacements of a beam having initial stresses. The principle is stated in a manner that allows the displacement and the moment to be varied independently as originally suggested by Reissner. In approximate solutions, the actual moments can be more closely approximated than is possible when the moment is obtained by twice differentiating assumed displacement functions. Initial stresses and, in the given example, boundary stresses which depend on the frequency of vibration are conveniently handled by the given principle. Forced vibration likewise may be treated. Conclusions regarding the frequencies of a rotating beam with tip mass are in agreement with those obtained by other methods by Boyce and Handleman. For all speeds of rotation, the fundamental frequency is decreased slightly by the addition of a tip mass. However, the addition of a tip mass greatly increases the frequency of the 2nd mode when the rotational speed is sufficiently great.
Transverse Vibration of Damped Beams
Authors: Snowdon J. C.
Published: September 2005

Representative computations of the response to sinusoidal vibration of coupled systems composed of damped beams and lumped elements of mass are described. The beams vibrate in their transverse modes, and are excited only at their midpoints. They are assumed to possess damping of the solid type. Discussed in detail is the response to vibration of a simply clamped beam excited at points other than its center.
Time Relationship between the Glottal and Sound-Pressure Waveforms—Measurements from Sound-Synchronized High-Speed Laryngeal Motion Pictures
Authors: Lieberman Philip , Soron Henry I.
Published: September 2005

High-speed motion pictures of the human vocal cords during phonation were taken at a maximum rate of 8800 frames per sec by the techniques of indirect laryngoscopy. The Fastax camera was blimped to isolate its operating noise, and the acoustic waveform was picked up at the lips by an Altec 21B condenser microphone. The acoustic waveform was recorded directly on the film in accurate time relationship to the pictures of the vocal cords by an optical variable-area track. The area of the glottal opening and the displacement of the vocal cords from each other were measured frame by frame. The relationship between these measurements of the vocal cords' vibratory cycle and the resulting acoustic pressure wave was determined for the onset, middle, and end of phonation. When the time delay between the recorded sound at the lips and the glottis was considered, it was found that maximum sound pressure occurred during the closing phase of the vocal-cord vibratory cycle. Other aspects of the vibratory cycle—in particular, those relevant to the initiation and end of phonation—are also discussed. The relationship between the displacement of the vocal cords from each other and the area of the glottal opening is also discussed in detail.
Contribution of the Pharyngeal Cavity to Vowel Resonance and in General
Authors: Truby H. M.
Published: September 2005

Descriptions of vocal-tract configuration commonly neglect the pharynx. Side-view sketches of the “supraglottal cavities” traditionally offer little distinction of pharyngeal outline from vowel to vowel, or from sound to sound of whatever nature. Actually, the pharyngeal cavities assume shape and sizes highly characteristic and predictable for given sounds—especially vowel sounds. When viewed cineradiographically, in sagittal-plane projection, the mesopharynx and hypopharynx appear to be maximally constricted during the [a] vowel, maximally extended during the [u] vowel, and similarly extended but with a decidedly different shape—during the [i] vowel, for example. Articulatory adjustments in anticipation of an associated vocalic nucleus are also predictable and characteristic in terms of the particular approaching vowel, especially, but exclusively, in the same “syllable.” Thus, the pharyngeal configurations for the respective consonantal portions of [la] and [lu], for example, are radically different. Static x-ray mews are at best only partially descriptive, but cineradiography provides the dynamics prerequisite for reliable estimates, and correlated sound-spectrographic analysis confirms the time-axis particulars. Lateral x-ray projections never truly define the vocal tract, and anteroposterior views reveal the consistent asymmetries of cross-sectional areas and disclose postulated “midlines” as illusionary. Preadjustments of the epipharynx, too, are out of phase with classically described articulatory gestures. (A short cineradiographic film and individual extracted frames correlated with sound, sound spectrograms, over-all intensity, oscillograms, and frame marking are offered in demonstration of statements and criticisms above.)
Speaker Identification and Vocal Variability
Authors: Starkweather John A. , Hargreaves Wm. A.
Published: September 2005

In the course of speech processing for the machine recognition of verbal content, it is desirable to minimize the effects of speaker differences. On the other hand, expressive differences between speakers and an individual's variability from one time to another are of psychological interest for a number of applied possibilities. Such measures would be particularly useful if accomplished under conditions of natural speech without artificial control of speech content. Twelve speakers, all young college women, were recorded 6 times on each of 3 experimental days. Measurement of these recordings was made with a real-time spectrum analyzer which integates the output of a bank of one-third-octave filters over successive 2-sec intervals. Linear discriminant functions were derived from the spectrum-analysis data, and a decision tree developed for individual speaker identification. Considerable success was obtained in the identification of new speech samples of 60 sec or less from the same speakers. An error matrix of 16 speech samples per subject showed a mean error of 10.4%, with 5 subjects identified correctly from all samples.
Error Analysis of a Formant-Locating Procedure Utilizing a Bank of Bandpass Filters
Authors: Pinson E. N.
Published: September 2005

Spectrum-matching has been suggested as a promising technique for formant-frequency estimation. In this procedure, a functional form for the speech spectrum is assumed in which formant frequencies appear as adjustable parameters. A reference spectrum is constructed using this functional form, and the adjustable parameters are varied until the error between the input-speech spectrum and the reference spectrum is minimized. A method is outlined for estimating the accuracy of such a procedure when a weighted-squares error criterion is used. It is assumed that errors are small, and a linearized expansion of the observed measurements is made about the estimated parameter values. The method is specialized to the case where formant frequencies are estimated by minimizing the squared error between the log spectrum of an input-speech signal (as measured by a bank of bandpass filters) and a synthesized set of filter outputs in which formant frequencies appear as adjustable parameters. The values of these parameters which minimize the error are taken as estimates of the formants in the speech signal. Numerical results for several cases of interest have been obtained by a digital-computer program. Some effects of harmonic structure, additive noise, and filter bandwidths are described.
Voiceprint-Identification Infallibility
Authors: Kersta L. G.
Published: September 2005

Previously reported work demonstrated that a high degree of speaker-identification accuracy could be achieved by visually matching the Voiceprint (spectrogram) of an unknown speaker's utterance with a similar Voiceprint in a group of reference prints. Both reference and unknown utterances consisted of single cue words excerpted from contextual material. An extension of the experimental showed identification success in excess of 99% when the reference and unknown patterns were enlarged to 5- and 10-word groups. Other experiments showed that professional ventriloquists and mimics cannot create voices or imitate others without revealing their own identities. Attempts to disguise the voice by use of various mechanical manipulations and devices were also unsuccessful. Experiments with a population expanded over the original 25 speakers showed that identification success does not decrease substantially with larger groups.
Statistics of Spoken English
Authors: Denes P. B.
Published: September 2005

The importance of using linguistic information for automatic speech recognition and speech synthesis is becoming more and more apparent as our knowledge of speech increases. Among the different kinds of linguistic rules that could be used in this connection are the statistical relationships between the various units of the language. This paper describes the statistical analysis of a considerable body of English speech, using a digital computer. The principles for selecting suitable texts are considered. The speech material used for the present study was obtained from several “Phonetic Readers” compiled for the teaching of English; these phonetic transcriptions were punched on IBM cards to make them suitable for computer processing. Counts were obtained for phoneme monogram and digram frequencies, for word length in terms of phonemes and of syllables, etc. Stress was taken into consideration, and many of the statistics were obtained separately for stressed and unstressed cases. Particular attention has been given to finding the distribution of minimal pairs, i.e., the pairs of phonemes that distinguish meaning between 2 words. Implications of the minimal-pair results with regard to automatic speech recognition and also to the development of words in language are discussed.
Automatic Method of Plosive Identification
Authors: Forgie James W. , Forgie Carma D.
Published: September 2005

A computer program has been developed which attempts to identify plosive consonants. Input to the computer is sampled, quantized spectral data. Several measurements are made on the burst portion of the sound. The identification of final plosives is made from these. Initial plosives use the burst information combined with formant transition cues as described in our other paper (G10).
Statistical Properties of ∫p2dt for Pulsed CW Reflected from a Rough Sea Surface
Authors: Saenger R. Alfred
Published: September 2005

Experimental work of M. V. Brown and J. Ricard has shown that, when a succession of “monochromatic” pings of duration T are reflected from a rough ocean surface, the statistical character of the resulting aggregate, ∫0Tp2dt=E, depends on frequency, sea state, and source-receiver range. An approximate theory designed to account for principal features of these experiments has been constructed on the basis of some rather crude assumptions concerning the physics of the reflection process and the statistics of the effective wave height, as sampled by successive pings. Analytical and numerical results are presented, showing the effect of the relevant physical variables on the probability density p(E) and the coefficient of variation (E−E)rmsĒ−1. The validity of the assumptions used in the theoretical derivation is examined in light of the agreement between theory and experiment. [Columbia University, Hudson Laboratories Informal Documentation No. 7. This work was supported by the Office of Naval Research.]
Design and Calibration of Wide-Range Acoustic Transducers
Authors: Beavers Bob R.
Published: September 2005

Wide-range (5 cps-250 kcps) electrostatic transducers have been designed for the transmission and reception of acoustic signals through air. These transducers include both the solid dielectric-type transducer constructed from metalized plastic films, and conventional air dielectric condenser microphones of unusually high sensitivity. Techniques for designing an optimum transducer for various applications are discussed on the basis of signal-to-noise ratio. sensitivity, directivity, frequency response, and the complementing electronic equipment. Various methods for the calibration of these transducers over the frequency range from 5 cps to 250 kcps are discussed.
Comparison of Microphone Sensitivities Obtained by Reciprocity and Pistonphone Methods
Authors: Embleton T. F. W. , Dagg I. R.
Published: September 2005

Errors in such quantities as the volume of the coupling cavity or the mechanical compliance of microphone cartridges cannot be detected, let alone corrected for, by means of calibrations of microphones by the reciprocity method alone. Such systematic errors affect the absolute value of the microphone sensitivity. Three microphones of each of 3 types have been subjected to 7 reciprocity calibrations apiece. In every case, these were followed immediately by a calibration using a pistonphone which operated in turn at 3 different amplitudes of motion. The difference between members of a pair of sensitivities (one obtained by each method) should always be zero. The actual differences may be studied statistically to determine the probable existence and amount of each of the following errors: (a) those related either to constants of the pistonphone or to electrical constants of the reciprocity apparatus; (b) those associated with any one of the 3 coupling cavities used for the reciprocity calibrations: and (c) those associated with any one of the 9 microphone cartridges.
Bandwidth Error of Symmetrical Bandpass Filters Used for Noise and Vibration Analysis
Authors: Sepmeyer L. W.
Published: September 2005

This paper is an extention of the results obtained with a simple mathematical model of a bandpass filter reported in the October 1962 issue of The Journal of the Acoustical Society of America. The noise power passed by 2-, 3-, and 4-pole Butterworth filters has been determined for input-spectrum level slopes ranging from −12 to +6 dB per octave by using digital-computer numerical-integration techniques on the filter-transmission equations. The bandwidths error has been found for both the usual 3-dB down case as well as an optimum bandwidth based on zero bandwidth error for pink noise. For 3-pole Butterworth filters this optimal condition occurs when the filter attenuation is approximately 3.6 dB at the band edges. The effect of filter-cutoff characteristics and room-excitation methods on reverberation-time measurements is also discussed.
Experimental Investigation of Jet Noise Generated by Cold Air Flow through Standard and Unusual Nozzles
Authors: Sperry William C.
Published: September 2005

Far-field sound-pressure levels were measured in an anechoic room for a selected number of standard and unusual nozzle types, the latter being modifications of optimum configurations previously reported in ASD-TDR-62-303. The results are reported in terms of over-all acoustic power vs mass flow, and are compared with various 8th-power relations. Spectral analyses are presented as well. An attempt is made to normalize acoustic and flow parameters with respect to size and temperature. [Supported by Propulsion Laboratory, Aeronautical Systems Division.]
Method for the Accurate Measurement of Sound Velocity in Liquids
Authors: McSkimin H. J. , Holton H. B.
Published: September 2005

A discussion is given of the factors entering into accurate measurement of sound velocity in liquids. This subject remains of interest, partly because it has not yet been satisfactorily demonstrated that any one of the experimental methods in current use can be relied upon when accuracies to within 0.05% are required. Phase comparison at high frequencies (>20 Mc) is shown to have definite advantages, and preliminary results for distilled water are presented by way of illustration.
Propagation of Sound in Rarefied Air
Authors: Heckl Manfred A. , Maidanik Gideon
Published: September 2005

The characteristics of sound propagation in rarefied gases are given in terms of attenuation and phase parameters. Experimental values of the attenuation and phase were determined in the frequency range from 2 to 16 kc and at pressures as low as 6×10−5 mm Hg. The experimental results are compared with free-molecule theory. The dependence of the attenuation and phase on the spacing between transmitter and receiver are discussed. Areas of disagreement between theory and experiments are particularly emphasized. [This research has been supported by the Office of Naval Research.]
Reflections of Sound Waves from a Washboard
Authors: Uretsky Jack L.
Published: September 2005

The problem, originally proposed by Rayleigh, of the reflection of a plane acoustic wave from a pressure-release surface is given an exact formulation suitable for numerical computation. Calculations, carried out on the Argonne IBM 704 and the UCSD CDC 1601 are described and results presented. [This presentation represents results of research sponsored by the Office of Naval Research and the U. S. Atomic Energy Commission.]
New Method for Finite-Amplitude Wave Analysis
Authors: Lester William W.
Published: September 2005

A new method for the measurement of the frequency components of a finite-amplitude wave has been devised which makes use of fixed-distance measurements. A 2-transducer pulse technique has been developed so that the problems of alignment and traversal of the transducers can be eliminated. The method has been used to determine the fundamental-frequency component as a function of distance in the near field of a 5-Mc pulse-operated transducer in water for various initial pressures. The measurements are shown to yield results observed by other investigators; in addition, some other results are obtained. For example, there is a definite tendency for the finite-amplitude effect to limit the maximum sound-pressure amplitude which can be propagated over a given distance. In addition, it is found that it is incorrect to extrapolate curves of pressure vs distance to the transducer face from measurements made at a distance. The method is most useful for small values of the parameter αL [William W. Lester, J. Acoust. Soc. Am. 33, 1196 (1961)] when making measurements on the fundamental-frequency component. Transducer calibration is accomplished by means of pulse-optical techniques [W. W. Lester and E. A. Hiedemann, J. Acoust. Soc. Am. 34, 265 (1962)].
Effect of Near Field on Cavitation Limit of Transducers
Authors: Sherman Charles H.
Published: September 2005

The cavitation-limited acoustic power of a transducer depends on the structure of its near field. The spatial variations of sound pressure in the near field, as well as the relative magnitude of the pressure components in phase and in-phase quadrature with the transducer velocity, influence the power which can be radiated without causing cavitation. Calculation shows that this effect is often very important, and that its importance varies strongly with the size and type of radiator. Assuming that cavitation starts when the sound pressure reaches a certain value, the problem has been formulated such that the near-field effect is clearly separated from effects related to the properties of the medium, and such that one parameter determines the near-field effect. This parameter depends on the radiation resistance and the maximum sound-pressure amplitude in the near field. The general properties of this parameter have been investigated and numerical values have been calculated for a wide variety of radiators. Circular pistons of flat and hemispherical profile, flexural disks with various velocity distributions, multipole spheres, and other radiators have been compared in terms of this parameter. The results are important in design of high-power underwater transducers and also in measuring the acoustic-cavitation threshold pressure.
Scaling the Energy Spectra of Underwater-Explosion Shockwaves
Authors: Christian Ermine A.
Published: September 2005

When an explosion is used as an underwater-sound source, determination of the acoustic-source level of the charge is complicated by the nonlinear character of the shockwave. The power functions that represent the shockwave parameters, rather than ideal acoustic-propagation laws, must be used to relate the spectrum levels for different sets of experimental conditions. Generalized correction factors with which shockwave energy spectra for one set of conditions (i.e., charge weight, range, explosive composition) can be scaled to a second set of conditions are derived. Using the TNT similitude relationships reported by Arons, graphical functions are given for scaling Weston's data for the shockwave energy spectrum at 100 yards from a one-pound charge to the spectra for other charge weights and ranges. Such scaled values are compared with the spectra obtained by Fourier analysis of measured shockwaves recorded at ranges of 1–300 yards from small explosive charges weighing 1 or 2 pounds. The errors that would arise from using ideal acoustic laws for scaling the energy spectrum are also shown.
Nonspecular Reflections from Underwater Objects
Authors: Crews Arthur A. , Erath Robert L.
Published: September 2005

This paper presents some recent results of experiments conducted to investigate nonspecular acoustic echoes from the edges of underwater objects. When energy is incident at the critical angle, it has been found that ordinary refraction theory can be used to predict the large-amplitude nonspecular echoes observed. However, echoes resulting from energy incident at other angles require a more complex explanation. Echoes from edges shadowed by the object also have been noticed. These echoes appear to be the result of several diffractions around the object. All these experiments were performed in a 15-ft-diam wooden tank employing broad-band pulses and 40- to 150-kcps pulsed sine waves.
Acoustic Energy Generation as a Function of Turbofan Engine Design
Authors: Garmanian H.
Published: September 2005

The development of mixing engines (such as bypass and turbofan engines) has opened a new area to be explored for possible means of reducing jet noise. Analytical studies and experimental developments were made so that the noise characteristics of various mixing engines could be identified in terms of engine-design parameters. The method of the program was based on an analysis of engine-flow parameters, such as fan-pressure ratio and bypass ratio, in terms of energy distribution in the primary and secondary jet streams. The results revealed that (1) in geometrically different exhaust-nozzle configurations (ranging from short duct through coplanar to fully mixed), jet noise decreases as the mixing of 2 flows becomes more complete; and (2) appropriate combinations of engine variables (such as high bypass ratio with high fan-input energy) will lead to a significant reduction in the jet noise from bypass engines. The results of this program will provide the menus of establishing acoustically optimum design criteria for bypass engines.
Measurements of Discrete Frequency Signals from Turbofan Engines
Authors: Divine G. L.
Published: September 2005

Turbofan engine noise is characterized by large-amplitude, discrete-frequency components which originate in the rotating-fan sections of the engine. The amplitude variations of discrete frequencies (commonly called “spikes”) are described, and samples of noise recording are shown. The data obtained from discrete-frequency measurements are analyzed in terms of the type of filter used and the frequency and amplitude of the “spike” relative to the background noise. Probability and statistical techniques are used to characterize the noise. The concepts, techniques, and the results obtained are described. The accuracy of short data records is compared with the accuracy of long records.
Origin of Low-Frequency Mechanical Noise from Diesel Engines
Authors: Ross Donald
Published: September 2005

In his book, Noise of Marine Diesel Engines, V. I. Zinchenko distinguishes between aerodynamic-noise sources, such as the intake blowers and the exhaust, and mechanical sources associated with the reciprocating mechanism. Zinchenko proposes that the dominant source of mechanical engine noise is the lateral slap of the piston against the cylinder wall which occurs close to the time of peak cylinder pressure, and he develops an expression for the impulse of such an impact. From this expression, using some simplifying assumptions, one can derive an expression for the radiated sound associated with piston impact. The results are of the right order of magnitude; and the resultant dependence on rpm, horsepower, cylinder pressure, number of cylinders, and engine-design parameters are all in reasonable agreement with experimental trends.
Application of Model-Scaling Techniques to Aircraft-Noise-Reduction Problems
Authors: Butler Robert H. , Gruner William J.
Published: September 2005

Model-scaling techniques have been successfully applied to the problem of determining the noise reduction of thin-walled enclosures. The method was studied for application to aircraft acoustical design. Noise-reduction measurements were made on 3 scales of a simple untreated enclosure placed in a reverberant field. When the data were collapsed to the full-scale frequency, there was good agreement of major features of the noise-reduction curve, such as resonances and critical frequency. The noise-reduction magnitude is scale-dependent due to the change in mean free path, and, when the data are corrected for this, the curves collapse to within experimental error. The limitations of the method are discussed and methods of extension to a more complex structure are suggested.
Quality of Organ Tones
Authors: Fletcher Harvey , Blackham E. Donnell , Christensen Douglas A.
Published: September 2005

This paper is concerned with finding an objective description of the quality of organ tones. The pipe-organ tones were produced by the organ which was formerly in the Salt Lake Tabernacle and now is in the Smith Auditorium at Brigham Young University. These tones were compared to electronic-organ tones and also with synthetic-organ tones produced in the laboratory. These tones were picked up by a microphone situated in various parts of the auditorium, and they were recorded on magnetic tape. Analyses were made from these tapes to find the partial tone structure, the attack, and the decay. Also, level grams were made which show the variation of the intensity level for each partial during the time that the key was pressed down. To show the complexity of the tone structure of organ tones, one example is given. For the full organ, 228 partials were measured when 3 keys were depressed to produce a major-triad tone. These analyses were then used to construct synthetic tones. Such tones were then compared with the real-organ tones by judgment tests by a jury of musicians and a jury of laymen. These juries were unable to distinguish the real from the synthetic tones. However, a comparison of equivalent tones from the pipe organ with those from a commercial electronic organ showed that these juries could distinguish the tones, and they definitely preferred those from the pipe organ. It was found that the warmth of the tone is definitely associated with the level variation of the partials. These variations are due to the frequencies of several components being close together, causing beats. Methods and apparatus were developed so that a tone could be warmed to any extent until its warmth matched that of the real-organ tones. An objective method of rating the warmth of any musical tone is proposed. In our future work, this method will be applied to other musical tones coming from a single instrument or from a combination of orchestral musical instruments.
Pitch and Loudness of a Binaural Subjective Tone
Authors: Guttman Newman
Published: September 2005

Given a binaural subjective tone (the Huggins phenomenon, heard when noises interaurally bear certain phase relationships) as challenge stimulus, listeners matched it for pitch and loudness by adjusting the frequency and level of a sinusoid in comparable noise. Challenge variables were polarity (monaural-signal inversion to put the phase-shift band center frequency, designated the target frequency, interaurally in either 180° or 0° phase) and rate of change of phase shift with frequency (90° shift with respect to the target accomplished in either 5 or 50 cps). In both the challenge and matching channels, the noise spectrum was flat, 1500-cps low pass. and over-all level was 60 dB re 0.0002 dyn/cm2. The target frequency was 500 cps. Results indicate that, despite some consistent listener deviations, on the average the target is matched in all but the 50-cps 180° condition, where it is heard as 520 cps. Loudness is slightly greater with narrower bandwidth; the 50-cps 0° tone is louder than the 50-cps 180° tone. Levels of sinusoid above masked threshold equivalent to the challenge tone range from 1 to 4 dB.
Foreign Accent and Speech Distortion
Authors: Lane Harlan
Published: September 2005

Speech distortion is defined as any operation that evokes inappropriate behavior by a listener in response to speech. Two categories of distorting operations are distinguished: (1) response-independent, in which the transfer function applied to the original speech signal is not determined by the probable response of the listener (e.g., masking, filtering); and (2) response-dependent, in which the distorting operation is related to the probable response of the listener during undistorted transmission and, therefore, may be specified in linguistic terms (e.g., foreign accent). Two experiments compare the effects of these 2 types of distortion. Twenty-four midwestern Americans listened to recorded articulation lists rendered by 1 American and 3 foreign-born speakers under 8 conditions of masking and filtering. Reducing the speech-to-noise ratio to 20 dB or the transmission bandwidth to 500 cps yields approximately 50% reduction in word articulation for both native- and foreign-accent speech. The latter was approximately 40% less intelligible than native speech under all experimental conditions. A “phonetic difference” measure of the degree of foreign accent holds promise as a predictor of the effects of accent on intelligibility; a significant correlation ratio of −0.66 was obtained.
Experiments in Automatic Speech Recognition
Authors: Meeker Willard F. , Nelson Alan L. , Scott Phillips B.
Published: September 2005

Use of the phoneme as a basic unit for automatic speech recognition is attractive from the standpoint of storage requirements and correspondence to conventional phonemic analysis. However, the segmentation of speech into phonemes and the specification of the identifying characteristics of a phoneme have presented difficulties. It has become clear that the physical speech signal does not consist of clearly defined units corresponding to phonemes. Likewise, perception of phonemelike units is influenced by adjacent units. Nevertheless, speech can be segmented with some degree of consistency at points which approximate phoneme boundaries, and these segments can, in many cases, be identified with moderate accuracy This paper describes the results of an automatic-speech-recognition research program oriented toward phoneme recognition. Input speech consisting of CVC monosyllables spoken in isolation was first analyzed with an 18-channel filter bank whose rectified outputs were quantized to 8 levels in 3.5 dB steps and recorded in digital form on magnetic tape. These digital recordings then served as input to an RCA 501 computer. A program for automatic segmentation and identification of phonemes was developed. Automatic vowel location and recognition accuracy averaged 45% for 10 vowels spoken in 60 CVC monosyllables by 10 male talkers, with 10 repetitions by each talker; scores for individual talkers as high as 85% were obtained. No adjustment was made for individual talkers. Initial consonant-identification accuracy averaged 60% for 6 consonants (/s, ∫, z, v, f, h/) with average scores for individual consonants ranging from 30% to 80%, and average scores for individual talkers as high as 75%. [This work was sponsored by the U. S. Air Force Systems Command.]
Technique for Varying the Ratio of Harmonic and Intermodulation Distortion in Speech-Communication Tests
Authors: Burnett Edwin D.
Published: September 2005

In evaluating the effects of nonlinear distortion upon the intelligibility and naturalness of a transmitted speech signal, it would be desirable to separate intermodulation and harmonic distortion components so that the effects of each could be studied separately. It is not feasible to produce harmonic distortion alone. We have developed a method which produces only difference frequency-intermodulation components without harmonic distortion. A slight modification produces a signal consisting of harmonic distortion and the sum intermodulation products. The nature of these distorted signals is demonstrated. [This work was supported by the Veterans Administration.]