bluuebunny/crossref_metadata_2025_split huggingface dataset:
The Influence of Interaural Phase Relations upon the Masking of Speech by White Noise
Authors: Licklider J. C. R. 
Published: September 2005

If a communication engineer, confronted with a sound wave consisting of speech mixed with audible random noise, were requested to build a device to separate the speech from the noise, he would be hard pressed to produce a mechanism as effective as the human auditory system. But if he were given two waves, one a sample of speech plus a sample of random noise, the other the same speech minus the noise, he would invoke the elementary mathematical (or electronic) processes of addition and subtraction and oblige in short order with noise-free speech and with speech-free noise.
This paper examines the performance of the (binaural) human auditory system in handling the two-wave problem. The effectiveness of the solution is judged in terms of the intelligibility of speech heard against a background of white noise. If monaural intelligibility is taken as a standard of comparison, it is found that the advantage of binaural presentation of the speech and the noise depends upon the interaural phase relations. The auditory system handles best the problems that are easiest for the engineer, though not as effectively as the engineer would handle them. Intelligibility is highest with noise plus speech in one ear, noise minus speech (i.e., the noise wave plus the inverted speech wave) in the other. Words are understood almost, but not exactly, as well with speech plus noise in one ear, speech minus noise (i.e., the speech wave plus the inverted noise wave) in the other.
These modes of presentation, in which either the speech waves or the noise waves in the two ears are 180 degrees out of phase, yield word articulation scores as much as 25 percentage units higher than the more conventional mode of presentation in which both the speech waves and the noise waves in the two ears are in phase. Observations with other interaural phase relations and with monaural-binaural presentation of speech and noise are also described.
The results suggest a means of providing a small but probably significant improvement in reception whenever speech is heard through earphones in the presence of ambient noise. The scheme is simply to reverse the connections of one of the earphones.
The significance of the results for the theory of masking is discussed.
The Perception of Short Bursts of Noise
Authors: Miller George A. 
Published: September 2005

Short bursts of random noise were used to study the perceptual growth and decay of the noise experience. Noise, unlike sinusoidal tones, does not change materially in spectrum as the duration of the sound is varied.
A short burst of noise must be more intense in order to be equal in effectiveness to a longer noise. Thus the threshold of hearing is lowered by increasing the duration of the noise up to durations at least as long as 1 sec. The loudness of an intense noise, however, depends upon its duration up to durations of only 65 msec. Judgments of the slowest rate of decay of a noise which sounded indistinguishable from an abrupt termination also led to the concept of a critical time of approximately 65 msec. for the sensation to decay to threshold from any steady magnitude of stimulation.
From these data it is inferred that the auditory system acts as if the growth and decay periods of the noise perception depend upon differences in latencies among the various neural paths in transmitting the cochlear activity to the higher centers in the brain. According to this hypothesis, the activity in the slowest pathways arrives at the higher center 65 msec. after the activity of the fastest pathways. By means of this hypothesis it was possible to predict the judgments of time of termination of noises in the two ears when the intensity of the noise differed between the ears.
A Study of Ultrasonic Velocity and Absorption in Liquid Mixtures
Authors: Burton Charles J. 
Published: September 2005

The absorption and velocity of sound in binary mixtures of water with alcohols, glycols, and glycol ethers have been measured at frequencies in the range of 5–25 megacycles, using the optical diffraction method originally developed by Debye, Sears, Lucas, and Biquard. However, the method of absorption measurement has been improved by utilization of a high gain, secondary multiplier-type photo-tube viewing the first-order diffraction spectrum, the change in intensity of this spectrum being an index of the sound absorption in the liquid. This method has proved more rapid and precise than other diffraction techniques described in the literature.
Ultrasonic absorption peaks at intermediate concentrations have been found in mixtures containing acetone, ethanol, propanol, isopropanol, and tertiary butanol and in the monoethyl and monobutyl glycol ethers. In both series, the magnitude of the peaks increases as one progresses from the ethyl to the butyl derivative, and there is a simultaneous shift of the peak toward higher water concentration. No absorption peaks were found in mixtures containing methanol, glycol, monomethyl glycol ether, or dioxane. Velocity peaks at intermediate concentrations have been found in all of the mixtures listed. Mixtures of tertiary butanol and methanol show neither absorption nor velocity peaks.
The observed absorptions are considerably greater than would be expected from classical theory. Consequently, the experimental data appear to require a new, or at least modified, theoretical approach. The problem is discussed briefly in the light of modern theories of molecular association, liquid crystal formation, etc.
Effects of High Pass and Low Pass Filtering on the Intelligibility of Speech in Noise
Authors: Pollack Irwin 
Published: September 2005

The effect on intelligibility of eliminating either the high frequency speech sounds or the low frequency speech sounds was determined by standardized articulation testing procedures under the special circumstance of a background of white masking noise. In general, it was found that intelligibility increased as the frequency range and the intensity level of the speech signal were increased. Contours are presented which describe the interrelations among (1) the frequency range of the speech signal, (2) the intensity level of the speech signal, and (3) articulation efficiency.
For each of the experimental conditions, the articulation index—a measure developed by the Bell Telephone Laboratories—was computed. It was found that the scatter of the computed indices, when plotted against the scores actually obtained, was of the same order of magnitude as the intrinsic variability of typical articulation scores. This result indicates that the computational procedures are an acceptable substitute for time-consuming articulation testing when an approximate assessment of the relative efficiencies of a series of communication systems is required and when a background noise is encountered like the white noise of this study.
It was found that the relative contribution of the various speech frequencies was not constant, but rather was a function of the intensity of the speech signal relative to the constant white noise masking signal. More specifically, as the intensity level of the speech signal was increased, the relative contribution to intelligibility of the higher speech frequencies increased.
Theoretical Analysis of the Mercury Delay Line
Authors: McSkimin H. J. 
Published: September 2005

A mathematical expression for the pressure distribution in a circular mercury delay line is derived. It is found that many waves having slightly different phase velocities can exist, and under certain circumstances may interfere with each other to produce distortion effects.
An analysis is made of the voltage developed by the piezoelectric pick-up crystal, and of the distortion that might result when the carrier is pulse modulated.
Reaction of Small Enclosures on the Human Voice
Authors: Morrow Charles T. 
Published: September 2005

A peculiar problem in communication arises when a talker is wearing an oxygen mask or a gas mask, in that both react on the vocal mechanism and distort the voice at its source. Thus, the loss of intelligibility when a gas mask is worn is much greater than would be predicted from the amount of muffling, and, even when the oxygen mask contains a microphone, the talker's enunciation is not, in general, clear. The reaction on the voice is in part mechanical, by constraining the facial muscles and obstructing the flow of the breath, and in part acoustical, by adding a new cavity onto the vocal system so that the talker has less control over the character of the sound. The requirements for good intelligibility during the use of such enclosures have been discussed in some detail in a previous paper (see reference 1).
The present paper concerns itself with an experimental analysis of the distortion produced by some small enclosures, and an evaluation of present theories of vowel production by comparing their predictions with the experimental results. Such a comparison is probably as severe a test of the theory of vowels as can be devised.
In general predictions from the present state of the theory of vowels give a useful qualitative picture of the nature of distortion by small enclosures, but are not in quantitative agreement with experiment. A criticism is given of the assumptions involved in the present theory, and some suggestions are made toward the future improvement of our knowledge of speech.
The Loudness of Repeated Short Tones
Authors: Garner W. R. 
Published: September 2005

The relation between loudness and the repetition rate of short tones was experimentally determined. The effects of intensity, frequency, and tone duration on this relation were also studied. The results can be summarized as follows. (1) A series of repeated short tones can be louder than a steady tone of the same peak intensity. (2) The difference between the loudness of repeated tones and a steady tone of the same intensity is dependent on both frequency and intensity. (3) Two series of repeated tones, of the same frequency and intensity, will not have the same loudness unless their repetition rates and durations are the same. At high intensities, the tones with the faster repetition rate and shorter duration are louder, and conversely at low intensities. (4) The rate at which loudness increases with repetition rate is dependent on the duration. The shorter the duration, the greater the change in loudness with a change in repetition rate; and conversely, the slower the repetition rate, the greater the change in loudness with a change in duration.
All of the relations shown are understandable in terms of the shape of the loudness function, and the complex spectra of repeated short tones. Calculated loudness levels, according to the method of Fletcher and Munson for calculating the loudness of multicomponent tones, agree very well with the observed loudness levels.
Amplitude Cross-Section Representation with the Sound Spectrograph
Authors: Kersta L. G. 
Published: September 2005

Spectrograms as normally produced by the “sound spectrograph” are three dimensional. Time is portrayed along the length of the spectrogram paper, frequency across it, and amplitude by the blackness of the marking. The time and frequency scales thus provided are adequate for most analysis purposes. The amplitude scale, however, is poorly defined, first because inherent characteristics of the spectrogram paper limit the amplitude portrayal to a highly non-linear and limited range, secondly, because reading degrees of blackness by eye is inaccurate. This inadequacy in determining amplitude led to the development of a method for portraying amplitude which was independent of the characteristics of the paper or of the ability of the eyes to delineate grades of blackness of marking.
The device developed and herein described provides an auxiliary presentation which portrays, at any preselected point in time, a two-dimensional spectrogram with axes of amplitude and frequency. This device delineates a 35-db amplitude range with the scale in uniform logarithmic steps.
Criteria for Normal and Abnormal Ultrasonic Light Diffraction Effects
Authors: Willard G. W. 
Published: September 2005

The normal or common type of ultrasonic light diffraction occurs when traveling sound waves act like a phase grating. The spacing and intensities of the respective orders of spectra are given by the Raman-Nath theory. Experimentally and theoretically, this type of diffraction is obtained when the product of (width of sound beam) × (frequency)2 × (sound amplitude) is sufficiently low. When the light beam is maintained parallel to the sound wave fronts, the right and left spectra of the same order are of equal intensity.
An abnormal or uncommon type of ultrasonic light diffraction occurs when the sound waves diffract the light in a manner satisfying the Bragg reflection law, well known in x-rays. The spacing of the spectra is essentially the same as before but right and left spectra do not appear simultaneously. Strong intensity in a given portion of the spectra is obtained only when the glancing angles of incidence φ and of reflection θ are equal, as in mirror reflection, and when λ, Λ, and θ satisfy the Bragg law nλ = 2Λ sinθ, (λ and Λ are the light and sound wave-lengths, respectively, and n is the order of spectra). This type of diffraction is obtained only when the single product (width of sound beam) × (frequency)2 is sufficiently large.
Experiments and theory indicate the above criteria for these two types of light diffraction. Theory also indicates the possibility of other types of abnormal diffraction, such as mirror reflection, diffraction satisfying ruled grating theory, nλ = Λ(sinφ + sinθ), where φ does not necessarily equal θ, and others.
The Patterns of Response in the Cochlea
Authors: Wever Ernest Glen , Lawrence Merle 
Published: September 2005

The general form of the intensity function for the electrical responses of the cochlea is now well known. As we raise the intensity of a sound applied to the ear, the cochlear potentials rise at first in simple proportion to the intensity, and then at a high level, as the stimulus is elevated further, they cease to follow this simple relation and give a bending curve. From the first departure from linearity onward the responses rise more and more slowly in relation to the stimulus until at last they reach a maximum value, a value beyond which they actually diminish as the intensity is raised.
So much of the form of the function is established on the basis of numerous experiments (see reference 2), yet there still are many details not yet worked out with exactness. The present study seeks in a measure to supply this lack. We have been concerned especially with the initial departure from linearity and the maximum value of the response for various frequencies over the tonal range. This information, as we shall show, has a pertinent application to the problems of the locus and spread of response over the basilar membrane.
Adaptation of the Ear to Sound Stimuli
Authors: Lüscher E. , Zwislocki J. 
Published: September 2005

The most important results of the determinations previously published on the adaptation of the ear to sound stimuli are briefly recapitulated. Adaptation is defined as the elevation of the auditory threshold by a previous sound stimulus. It may be determined by means of a short tone impulse (testing impulse) which follows the sound stimulus (stimulating impulse) causing the adaptation. The testing impulse is adjusted to the threshold value. By altering the interval between the stimulating and testing impulses, the curve of return adaptation, i.e., the return to normal sensitivity, may be ascertained. When plotted on a logarithmic db scale, this curve is approximately a straight line.
New experiments have shown that the entire process of adaptation requires less than 0.4 sec. The return adaptation is complete within a few tenths of a second. After stimulation by a tone of 80 db above threshold intensity, return adaptation is complete after 250 μsec.
The increase in adaptation is approximately proportional to the intensity of the stimulating tone when both magnitudes are expressed in db. With a sound impulse of 80 db above the auditory threshold, the adaptation reaches a value of 40 to 50 db.
Binaural determination of adaptation, by leading the stimulating impulse to one ear and the testing impulse to the other, shows that adaptation is a monaural and therefore a peripheral process.
A pure tone induces an elevation of the auditory threshold not only for its own frequency but also for neighboring ones. The adaptation spreads chiefly to higher frequencies than that of the stimulating tone at which it reaches a maximum. The maximum is more pronounced at high frequencies of the stimulating tone than at low ones.
Comparative experiments between adaptation and masking demonstrate a far-reaching similarity between the two processes. These show a similar behavior with respect to intensity and frequency. A quantitative comparison leads to the conclusion that masking depends chiefly upon the adaptation of the ear to sound stimuli.
A Theory of the Crystal Transducer for Plane Waves
Authors: Cady Walter G. 
Published: September 2005

From the equation for propagation of acoustic plane waves through a solid medium, expressions are derived for the vibrational amplitude, radiated power, and electrical admittance of a plane-wave crystal transducer. The transducer consists of crystals, backing, and diaphragm, radiating into any medium. The crystals may be an assemblage of flat bars vibrating lengthwise, or there may be a single flat crystal plate, or mosaic of plates, in thickness vibration. The equations take account of attenuation in the diaphragm, but not of attenuation in crystals and backing, nor of the effects of cement layers and lateral stresses between crystals.
The general equations, which are valid at all frequencies and for all thicknesses of backing, crystal assemblage, and diaphragm, are specialized to the case in which the backing is air. Special attention is given to those frequencies at which the crystals, or diaphragm, or both, are in resonance. Some typical theoretical curves are shown, and a graphical method is described for the rapid plotting of curves.
Measurement of Acoustic Impedance
Authors: Mawardi Osman K. 
Published: September 2005

In this paper a method is described of measuring the acoustic impedance of a sample of material forming one boundary of a shallow cylindrical cavity, by determining the sound pressure produced when a known volume current is injected into the cavity from a high impedance source. The volume current is effectively determined by observing the pressure when the cavity is rigidly terminated. The use of a ring source suppresses the first radial mode of the cavity and extends the frequency limit set by uniformity of pressure by at least an octave.
Secondary effects due to the finite impedance of the source and microphone and heat losses at the walls have been evaluated. The method appears to be simple, rapid, and precise.
Interactions between Microphones, Couplers and Earphones
Authors: Morrical Keron C. , Glaser John L. , Benson Robert W. 
Published: September 2005

The earlier work on the comparison of the JRB and the Massa Artificial Ear couplers has been continued to seek the explanation of certain differences observed. The later work, for the most part, has been undertaken on couplers under active consideration for adoption by the American Standards Association.
The difference in pressure in 6-cc couplers as measured by the Western Electric 640-AA and the Massa M-101 microphones appears to be identified with the strong earphone coupler resonance peak rather than directly with the shape of the coupler. The effect is apparently associated with the diaphragm impedance of the microphone more closely when the microphone is in use as a measurement device than when merely present, but inactive, in the coupler, except in the case of the M-101 microphone whose presence gives rise to an unexplained increase in pressure.
For general earphone calibration work, the results obtained by means of any of the coupler systems will be satisfactory. For some physical purposes, however, the effects may lead to significant errors.
The Vibration of the Cochlear Partition in Anatomical Preparations and in Models of the Inner Ear
Authors: v. Békésy Georg 
Published: September 2005

An experimental technique was developed to measure, for intact middle and inner ears, the volume displacement of the fluid in the cochlea produced by a given sound pressure on the eardrum. In order to determine whether it is possible to immobilize the stapes by increasing the static pressure in the cochlea, the transmission characteristics of the eardrum-to-round-window system were measured with the ear under the maximum static pressure that can be produced in the cochlea without bursting the blood vessels. Since no important change in transmission was observed, it was concluded that immobilization of the stapes cannot be produced by an increase in static pressure in the cochlea.
With the same technique the mechanical impedance of the movements of the stapes footplate, the round window, and the fluid in the cochlea were measured singly and combined as in the normal ear. Once the elastic properties of the cochlear partition were known, it was possible to construct an ear model that exhibited patterns of vibration similar to those found in the cochlear partition of the human ear. By changing the shape, the size, and the fluid viscosity in this ear model the relative importance of the different parts of the inner ear were studied.
These experiments made it plain that the pattern of vibration of the cochlear partition is determined primarily by the volume elasticity of the membrane and the fluid close to the membrane. Length, height, shape, and diameter of the cochlear canal do not influence the vibration pattern. A mathematical description of the hydrodynamic properties of the inner ear must take into consideration this independence of the physical dimensions of the cochlear canal.
The Effect of White Noise on the Loudness of Speech of Assigned Average Level
Authors: Pollack Irwin 
Published: September 2005

Continuous recorded speech passages were presented against a background of white noise and against a quiet background. The effect of the noise on the loudness of speech was determined by having listeners adjust the level of speech heard in quiet to sound equal in loudness to an assigned average level of speech presented against selected levels of background noise. In general, it was found that (1) low noise levels produce a proportionately lower depression in the loudness level of speech than higher noise levels, (2) as a first approximation, however, the effect of noise on the loudness of speech is a function of the speech-to-noise ratio rather than of the level of the speech alone or of the noise alone, and (3) under none of the experimental conditions did white noise actually increase the loudness of speech (the “Egan effect”).
The assumption that a given level of noise subtracts a constant number of loudness units from speech (regardless of the level of the speech) was found to be fairly adequate in describing the experimental results.