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Journal of the Acoustical Society of America

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Nov 1959

Volume 31, Issue 11, pp. 1407-1588

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Measurement of Sound Diffusion in Reverberation Chambers

Manfred R. Schroeder

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1407-1414 (1959); (8 pages) | Cited 7 times

Online Publication Date: 11 Jul 2005

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This paper describes a variety of methods for the measurement of the diffusion of sound fields in reverberation chambers. Diffusion is defined on the basis of the angular distribution of sound energy flux, in accordance with the definition that has found its visual expression in the “sound hedgehog” of Meyer and Thiele. The theoretical foundations of the methods proposed here are: normal mode expansion, the sampling theorem (both in time and two‐dimensional space), and either Fourier or correlation analysis. The quantities to be measured are sound pressures and, in some cases, sound pressure gradients at a number of sampling points on the measuring wall. Results of these measurements are suitably transformed to give the sound energy fluxes for all possible angles of incidence. The accuracy of measurement is determined by the Q (frequency times reverberation time) of the chamber and is typically of the order of 1°. This extra‐ordinary directivity is achieved without substantial perturbation of the sound field. Methods applicable to both single frequencies and finite frequency bands are described.

Scaling Human Reactions to the Sound from Aircraft

Karl D. Kryter

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1415-1429 (1959); (15 pages) | Cited 4 times

Online Publication Date: 11 Jul 2005

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Studies were undertaken to (determine how “noisy” commercial jet aircraft will sound to people on the ground in comparison to present‐day commercial aircraft with reciprocating engines. Three experiments involved the flyover sound made by reciprocating‐engine aircraft (a DC‐7 and a Super‐Constellation) and jet aircraft (a Caravelle, a Comet, and a 707). People were asked to judge the “annoyance” or potential bother‐someness of the sounds relative to one another.
It was found that the over‐all sound pressure level (“C” scale) and the speech interference level (SIL) of the sounds from the aircraft as measured on the ground, either outdoors or indoors, bear little relation to the judgments. The loudness level in phons, the values obtained with the aid of the “equal listener‐response curve” of Miller and Beranek, and over‐all sound pressure level “A” scale and “P” scale correlate better with the findings. However, the results of the judgment tests are predicted best when the SPL's of the various octave bands are weighted in a newly prescribed manner. This calculation scheme is based on previously published equal‐annoyance contours and the procedures developed by Stevens for calculating loudness level.

Noise Stress in Laboratory Rodents. I. Behavioral and Endocrine Response of Mice, Rats, and Guinea Pigs

Adam Anthony, Eugene Ackerman, and James A. Lloyd

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1430-1437 (1959); (8 pages)

Online Publication Date: 11 Jul 2005

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Mice, rats, and guinea pigs were exposed daily to high intensity noise in the frequency range of 150–4800 cps at a sound pressure level of about 140 db, re 0.0002 μbar. It was found that the behavioral and endocrine responses differ in the three species studied. The adrenal response of animals exposed to noise supports the interpretation that noise acts as a physiological stress, but does not induce harmful extra‐auditory effects. Thus, the endocrine system appears to act as a multiloop feed‐back system which compensates for the effects of noise on the central nervous system.

Noise Stress in Laboratory Rodents. II. Effects of Chronic Noise Exposures on Sexual Performance and Reproductive Function of Guinea Pigs

Adam Anthony and Jack E. Harclerode

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1437-1440 (1959); (4 pages)

Online Publication Date: 11 Jul 2005

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Chronic exposure of male guinea pigs to intense, low‐frequency noise (139–144 db, 300–4800 cps) for six weeks did not affect their sexual performance. Histological and histochemical analyses of various organs indicated noise is followed by increased activity of the adrenal glands. However, the absence of pathology in the reproductive and digestive organs indicated that the tolerance limits of animals were not exceeded.

Role of Interaural Time and Intensity Differences in the Lateralization of Low‐Frequency Tones

George Moushegian and Lloyd A. Jeffress

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1441-1445 (1959); (5 pages) | Cited 2 times

Online Publication Date: 11 Jul 2005

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There has recently been a renewal of interest in the roles of intensity and time in the lateralization of sounds. The technique generally employed is to offset the effect of an interaural time difference in one direction with a difference of level in the opposite. In the present series of experiments a different method was used, that of having the subject adjust the interaural time relation for a noise until it appears to be in the same lateral position as the stimulus tone. Using this procedure, we have obtained results which support the findings of other recent workers that increasing the intensity of the stimulus to an ear will cause it to transmit earlier in time. We have found additional evidence, however, which shows that the central nervous system, too, responds to interaural intensity differences, and that its response is different from that of the peripheral system. When time and intensity are opposed, time has less effect on it than when time and intensity both favor the same side.

Detection of a Pulsed Sinusoid in Noise as a Function of Frequency

David M. Green, Mary J. McKey, and J. C. R. Licklider

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1446-1452 (1959); (7 pages) | Cited 13 times

Online Publication Date: 11 Jul 2005

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The detectability of a pulsed sinusoid (0.1 sec) in white noise was measured at sixteen frequencies ranging from 250 to 4000 cps. The measurements are compared with the results previously obtained from experiments in which continuous sinusoids of indefinite duration were used. The dependence of detectability on frequency appears to be very similar in all the experiments. We also measured the detectability of compound signals, i.e., signals with 12 and 16 sinusoidal components. A comparison of the detectability for the single and combined sinusoids allows us to determine approximately how the auditory system sums energy over frequency.

Experimental Study of Bone Conduction in Ears with Mechanical Impairment of the Ossicles

J. P. Legouix and S. Tarab

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1453-1457 (1959); (5 pages) | Cited 1 time

Online Publication Date: 11 Jul 2005

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The cochlear‐microphonic potential of guinea pigs was recorded during stimulation by bone‐conducted sounds. The amplitude and the phase of the response were measured while the ossicles were altered in various ways.
At low and middle frequencies, fixation of the ossicular chain produces a decrease of amplitude and a phase advance of about 70°, while an increase of the mass yields an increase of amplitude and a phase lag greater than 180°. A progressive increase of the mass, produced by introducing paraffin oil in the middle‐ear cavity, yields a progressive variation of amplitude and phase and, at a particular moment, a complete disappearance of the response. Symmetrical results were obtained by altering the intrabullar pressure. All these results are interpreted as showing the participation of several mechanisms contributing to the whole response. At low frequencies a translational mechanism involves two components: one related to the motion of the ossicular chain and the other related to the motion of the perilymphatic fluid. These two components contribute to the whole response according to their amplitude and phase. At higher frequencies a compressional mode of bone conduction, independent of the motion of the ossicles, is responsible for the stimulation. These findings agree with the alteration of bone conduction observed in clinical cases and provide an explanation for the lateralization of the sound source in an ear affected by conductive deafness, as shown in the Weber test. This lateralization may be accounted for by the observed phase shifts.

Second‐Order Instrumentation Systems with Frequency‐Dependent Stiffness and Damping

Eric Rule, Fred J. Suellentrop, and Thomas A. Perls

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1457-1462 (1959); (6 pages)

Online Publication Date: 11 Jul 2005

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The theory of the dynamic properties of thin fluid films is reviewed and it is shown how these properties can be used to introduce frequency‐dependent damping and stiffness factors into the equations governing the response of second‐order instrumentation systems such as transducers for the measurement of dynamic accelerations, vibration, pressure, sound, etc. By proper design, the frequency‐dependent nature of these factors can be utilized to provide substantial improvement in the frequency‐response characteristics of such systems over the characteristics obtainable with the usual second‐order system with constant viscous damping. For the specific case when the fluid is air at normal temperature and pressure, curves are given from which transducers can be designed to have optimum frequency‐response characteristics.

Surface and Groove Noise in Disk Recording Media. I

D. H. Howling

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1463-1472 (1959); (10 pages)

Online Publication Date: 11 Jul 2005

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A theoretical and experimental investigation into the causes and mechanism of playback surface and groove noise of recorded media has been made. It is shown that such noise can be considered as arising from the random superposition of voltage pulses produced by local changes in the number and size distribution of welded friction junctions or asperities supporting the playback stylus load. A generalized noise equation, involving the physical parameters of the recorded medium and the playback stylus, has been derived and shows reasonable agreement with the measured playback noise spectra for several cold‐flowing and rigid plastic surfaces. From the noise equation emerges a concept of modulation noise and prediction of its dependence on modulation velocity. Examination of the playback noise level as a function of recording and playback conditions has also been included. It is also shown that surface noise measurements can be used to give information concerning the n/a ratio associated with the theory of friction.
The measured lower limit of wide band playback noise for most plastic surfaces corresponds to an equivalent rms lateral noise velocity of 10−2 cm/sec or equivalent lateral noise amplitude of 1 μin. peak at a linear velocity of 15 cm/sec. The equivalent roughness is thus half of that associated with an optical flat. Using this information, it is shown that at 1000 lines/in. information density a signal‐to‐noise ratio of 50 db at a linear velocity of 15 cm/sec should be realized.

Propagation of Band‐Limited Noise in a Layered Wave Guide

C. S. Clay

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1473-1479 (1959); (7 pages)

Online Publication Date: 11 Jul 2005

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The normal mode solution of the problem of the radiation field of a simple harmonic point source in a layered wave guide is extended to the case of a band‐limited noise source. This theory is used to calculate the radiation field of a simple harmonic point source and a point noise source in shallow water over a thick layer of unconsolidated sediments, and the calculations are compared with experimental data. The experimental dependence of the acoustical pressure upon source distance is in agreement with that calculated from the theory.

Results Obtained from a Vowel Recognition Computer Program

James W. Forgie and Carma D. Forgie

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1480-1489 (1959); (10 pages)

Online Publication Date: 11 Jul 2005

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As an initial step toward a general speech recognition computer program, program was developed to recognize ten English vowels in isolated words of the form ∣b∣ ‐vowel‐ ∣t∣. The input to the computer was real‐time spectral data. The program first determined the rough location of the first two formants. The remaining confusions were resolved by (1) finer determination of the F1 and F2 locations by the use of slope and/or valley information, (2) the use of pitch information, and (3) in certain cases the determination of the position of F3. The over‐all score for 21 subjects (11 male and 10 female) was 88%. By the use of duration information, the score was raised to 93%.

Minimal Rules for Synthesizing Speech

A. M. Liberman, Frances Ingemann, Leigh Lisker, Pierre Delattre, and F. S. Cooper

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1490-1499 (1959); (10 pages)

Online Publication Date: 11 Jul 2005

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It has been found to be extremely difficult to isolate phonemic elements from recorded utterances or to synthesize speech by assembling prerecorded phonemic segments. One reason for the difficulty lies in the fact that the perceptually discrete phonemes are typically combined, and in some cases encoded, into units of essentially syllabic dimensions. As a result, prerecorded elements must, in many cases, approximate syllables, and the synthesis of speech by this means will require a large inventory of recordings. By taking advantage of knowledge about the acoustic cues for speech perception, however, it is possible to write rules for synthesis in terms of phonemes (rather than syllables) and thus reduce considerably the number of separate rules or items needed. Indeed, one can reduce the number of rules still further by writing them at the level of subphonemic dimensions, viz., place and manner of articulation. Several complicating factors make it impossible to achieve an ideal minimum. First, rules or rule modifiers must be added to take care of certain prosodic and positional variations. Failure to do so not only affects naturalness, but also impairs intelligibility, even at the level of segmental phonemes. Second, it is necessary in a few special cases to have different rules for a single consonant phoneme (or a dimension of that phoneme) before different vowels. This reflects the occasionally complex relation between phoneme and articulation on the one hand and sound output on the other; presumably, this complication would not affect the rules of synthesis for an articulatory model.
A system of rules for synthesis framed largely in terms of subphonemic dimensions is described with reference to an example. Words and sentences of rather high intelligibility have been synthesized by such rules.

Message Uncertainty and Message Reception

Irwin Pollack

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1500-1508 (1959); (9 pages) | Cited 2 times

Online Publication Date: 11 Jul 2005

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The fundamental principle of language engineering—namely, that the probability of a correct message reception is critically dependent upon the size of the set of messages available for communications—was examined experimentally. Accuracy of message reception was found to be independent of the message‐source uncertainty, but critically dependent upon the response uncertainty—or more exactly, upon the size of the set of relevant response categories. Implications for a conceptual model of the listener's behavior are suggested.

Message Repetition and Message Reception

Irwin Pollack

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1509-1515 (1959); (7 pages) | Cited 1 time

Online Publication Date: 11 Jul 2005

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The improvement in word intelligibility associated with successive presentations of a word in noise was examined with recorded and with independent samples of the speech and/or its background noise. The listener's criterion for termination of a trial was shown to be a crucial determinant of the gain of intelligibility with successive presentations. The improvement in intelligibility was greater for independent samples of speech and noise than for successive presentations of a single recorded sample. The observed improvement, however, was less than predicted for the ideal observer with perfect memory by the theory of signal detectability. Indexes of response discriminability and signal discriminability are briefly discussed.

Frequency Response of an Acoustic Air‐Jet Generator

James C. Gravitt

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1516-1518 (1959); (3 pages)

Online Publication Date: 11 Jul 2005

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The mechanism which gives rise to the intense vibrations of an acoustic air‐jet generator is investigated both experimentally and theoretically. The results indicate that the mechanism for establishing the vibrations in the generator consists of pressure instabilities in the nozzle air stream which act as the source of the forced oscillations of an air plug in the resonator cavity.

Performance of High Frequency Barium Titanate Transducers for Generating Ultrasonic Waves in Liquids

H. J. McSkimin

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1519-1522 (1959); (4 pages)

Online Publication Date: 11 Jul 2005

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A study is made of high frequency (10 mc/sec) barium titanate transducers radiating into water, with band width and efficiency being of principal concern. Optimum electrical damping alone is considered; however, it is demonstrated that the introduction of an intermediate ¼ wavelength thick composite mechanical element results in superior performance. Of the units constructed, the best exhibited a frequency band width (half‐power points) of 44% with 4‐db midband loss (equivalent insertion loss on a 2 transducer basis). Focusing radiators, as well as plane, were tested.

Some Ultrasonic Measurements in Normal and Superconducting Aluminum

R. W. Morse and H. V. Bohm

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1523-1526 (1959); (4 pages) | Cited 1 time

Online Publication Date: 11 Jul 2005

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Ultrasonic attenuation measurements in an aluminum single crystal near the superconducting transition temperature are described. These include both longitudinal and shear wave attenuations at frequencies of between 11 and 58 Mc/sec and at temperatures between 1.0 and 4.2°K, as well as attenuation as a function of magnetic field in the normal state. It is found that the attenuation of shear waves at the transition temperature shows a discontinuity of about 25% of the total attenuation, this fraction being relatively independent of frequency. The longitudinal wave attenuation shows a sharp but continuous drop at the transition temperature. It is shown that the electron mean free path l can be estimated from the magnetic field dependence of the attenuation and that the condition kl<1 (k = 2π/λ, λ being the ultrasonic wavelength) holds for the measurements. The possible significance of the shear wave discontinuity is discussed.

Propagation of a Pressure Pulse in a Compressible Flow

Alan Powell

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1527-1535 (1959); (9 pages) | Cited 2 times

Online Publication Date: 11 Jul 2005

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A one‐dimensional treatment of the propagation of a pressure pulse through a channel carrying a compressible flow is given. The method consists of summing the multiple linear reflections of increasing order, the “transmission” and “reflection” coefficients then being given in the form of power series whose terms depend only on the entry and exit Mach numbers of the mean flow. The series fails to converge under certain circumstances, but there is nothing corresponding to this in the equivalent results obtained from “before” and “after” steady‐state considerations. Both methods are shown to fail at a sonic throat, where it is suggested that the assumption of one‐dimensional perturbations is no longer tenable.

Measurements of Acoustic Radiation Force

K. Budal, E. Hoy, and H. Olsen

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1536-1538 (1959); (3 pages)

Online Publication Date: 11 Jul 2005

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The acoustic radiation force of a plane wave impinging on a rigid sphere of diameter 10 cm and on a circular disk of diameter 7.4 cm has been measured in the wavelength interval 4 to 40 cm. The agreement with theory is satisfactory.
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Auditory Perception of Temporal Order

Donald E. Broadbent and Peter Ladefoged

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1539-1539 (1959); (1 page) | Cited 1 time

Online Publication Date: 11 Jul 2005

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Contributed Papers
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A Note on Low‐Pass Filters for Retrieving Sampled Speech Signals (A)

Henry S. McDonald

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1563-1563 (1959); (1 page)

Online Publication Date: 11 Jul 2005

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In the case of retrieving a sampled speech signal using a nonideal low‐pass filter, some loss of quality is observed due to passing energy grouped as side bands about the sampling frequency. This paper describes a series of experiments designed to relate the filter characteristic to the speech quality. By subjective matching of signals using filters with variable out‐of‐band transmission, it was found that the transmission in the vicinity of the sampling rate was the most important correlate to signal distortion. The transmission in the frequency region adjacent to the base band was found to be relatively unimportant. An explanation in terms of the spectral characteristic of speech signals is offered. Presampling low‐pass filtering was also investigated and found to be much less important than accurate post sampling filtering.
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Formant Tracking by Computer (A)

C. G. Bell, F. Poza, and K. N. Stevens

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1563-1563 (1959); (1 page)

Online Publication Date: 11 Jul 2005

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At the last meeting of the Acoustical Society of America a method of formant tracking using a digital computer was described. By this method synthetic speech spectra, constructed within the computer from a catalog of elemental spectra, are compared with measured spectra of a speech signal. The present paper discusses the evaluation, extension, and refinement of this method of analysis. Computer techniques which result in the display of a “synthetic sonagram” are described, together with examples of the use of the computer oscilloscope for displaying intermediate steps in the spectral matching process. Results obtained with a variety of speakers are presented. The problems of preparing a representative catalog of glottal spectra, evaluating the role of analyzing filter banks, and determining a meaningful “fit criterion” are considered. A method similar to the foregoing for synthesizing fricative spectra is also described. [[This work was supported in part by the U. S. Army (Signal Corps), the U. S. Air Force (Office of Scientific Research, Air Research and Development Command), and the U. S. Navy (Office of Naval Research); in part by Air Force Cambridge Research Center Contract AF 19(604)‐2061; and in part by the National Science Foundation.]
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Studies of a Method for Digital Voice Communication (A)

Caldwell P. Smith

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1563-1563 (1959); (1 page)

Online Publication Date: 11 Jul 2005

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The approach to speech compression evolved at AFCRC attempts to exploit in a maximal way the constraints that govern speech at the acoustic level. The problem requires a method for statistical encoding and decoding of voice signals in two stages. First, it is necessary to quantize speech patterns and encode these in terms of a minimal set of patterns, based on criteria of speech intelligibility, quality, and talker recognition. After converting speech to a code that specifies a sequence from the limited catalog of patterns, the data must be further reduced by recoding in terms of pattern clusters, that is, pattern sequences in pairs, triples, quads, etc. Since successive speech patterns measured every one‐fiftieth of a second are not independent events but highly interrelated, recoding in terms of clusters, new patterns in which time is one of the dimensions, will markedly economize on the data rate requirement. Although these message units are purely statistical, it is anticipated that their correlation with phonetic elements of speech will be high. A digital computer has been utilized in studying some statistics of speech patterns, and these results will be presented. To expedite the study of this voice communications system concept, a real‐time digital data processing equipment is being fabricated that will sort and accumulate statistics of speech patterns in real time. The occurrences of single patterns as well as clusters of up to five patterns in a sequence will not only be accumulated, but can be utilized in simulating a minimum data‐rate digital speech system operating in real time. Some features of the logical design of the system will be discussed.
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A Voice Analog (A)

W. C. Dersch

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1563-1564 (1959); (2 pages)

Online Publication Date: 11 Jul 2005

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An artificial voice was constructed using novel and simple techniques. An outgrowth of an experiment in speech recognition, this voice analog was used to ascertain the efficiency of the information extraction techniques used in a speech recognition machine. The extracted information curves were reconstructed to full speech by means of this voice analog. Four voice‐characterized noise generators were connected to four voltage‐controlled bandpass filters. Each noise generator and filter pair operated within an assigned major subband. Four slowly varying frequency‐describing information curves were connected, one to each of the filters. Each filter then passed a narrow minor subband of frequencies determined by the voltage of the frequency‐describing curve at the same instant. Three additional power‐describing information curves modulated the four filter outputs. The resulting curves were mixed, amplified, and connected to a loudspeaker. The voice analog speech was harsh and estimates of its intelligibility varied greatly. Plosive consonants were good and consonants in general were better than vowels. The complete system occupied only a few square feet of bench space. Although the voice analog was primarily an experiment which provided the desired data and was not fully refined at the time of its construction, its compactness and simplicity suggest the desirability of further development.
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Pitch Synchronous Spectral Analysis of Vowel Sounds (A)

Joan E. Miller, M. V. Mathews, and E. E. David, Jr.

J. Acoust. Soc. Am. Volume 31, Issue 11, pp. 1564-1564 (1959); (1 page)

Online Publication Date: 11 Jul 2005

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Fourier analysis of individual periods of sustained vowel sounds has been carried out on a digital computer. The spectral data show two characteristics (1) a periodic oscillation in amplitude and (2) many rapid phase shifts, which cannot be explained by simple vocal tract resonances. A theory is developed to show that these spectral characteristics can be attributed to zeros of the glottal excitation function and that typically there is a large number of these zeros. Such a representation of the excitation is consistant with the glottal wave‐form information from high‐speed photography. A computer program has been developed to fit a spectrum with a pole‐zero configuration and good agreement has been obtained between the observed spectrum and that computed from a pattern of glottal zeros and formant poles.
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