bluuebunny/crossref_metadata_2025 huggingface dataset:
ZYP—An Automatic Impedance mad Admittance Plotter
Authors: Schenck Harry A.
Published: November 2005

A versatile instrument for measuring and recording complex impedance and admittance has been constrncted. A variable-frequency two-phase oscillator supplies a reference signal to the unknown immittance and reference and quadrature signals to a two-channel synchronous detector. For the measurement of impedance, a constant current is supplied to the variable load, and the voltage across the unknown is applied to each channel of the synchronous-detector. The two filtered dc-output voltages are then proportional to the real and imaginary parts of the unknown impedance. For the measurement of admittance, a constant voltage is supplied to varying loads, and the current through the unknown becomes the input signal to the synchronous detector. The present model of the Z-Y plotter (ZYP) was constructed for the measurement of driving-point immittance of electroacoustic transducers. For unknowns whose impedance magnitude ranges from 100 Ω–100 kΩ, either impedance or admittance can be measured and plotted as a function of frequency between 30 and 30 000 cps. Full-scale deflection on any range was made to correspond to 10 μW power dissipated in a resistive unknown, but operation at other power, voltage, or current levels could be achieved. ZYP can also be adapted to measure complex transfer functions. [This research was supported in part by funds made available under a contract with the U. S. Office of Naval Research.]
Pure-Tone Diffuse-Field Reciprocity Calibration of a Dynamic Microphone
Authors: Barnett Norman E. , Trasher Donald B.
Published: November 2005

In order to determine the total radiated sound power as a discrete function of frequency from acoustic sources of arbitrary and unknown characteristics operated in a reverberation room, it is essential to know the diffuse-field calibration of the measuring microphone. The diffuse-field reciprocity method demonstrated by H. G. Diestel [J. Acoust. Soc. Am. 33, 514 (1961)] for bands of white noise has been applied successfully to the pure-tone calibration of dynamic microphones (Altec type 633A). Thus, the «diffuse-field” correction usually postulated on theoretical grounds can be replaced by a direct absolute calibration. As a consequence, given a reverberation room and a pair of similar microphones, one may arrange to make absolute sound-power measurements without recourse to acoustic secondary standards or outside calibration of microphones.
Sound Reinforcement for a Touring Musical Show
Authors: Kaye David H.
Published: November 2005

A sound-reinforcement system for a touring musical show was designed using an adjustable central theater-type loudspeaker cluster, supplementary time-delayed distributed loudspeakers, a frequency shifter, a custom-designed control console with vertical attenuators, a graphic equalizer, and directional condenser microphones. The objective was to provide speech reinforcement and reproduction of sound effects equal to those provided by state-of-the-art permanent systems. This paper discusses the system design, plus installation and operational problems and results in seven large auditoriums having varied configurations and acoustical environments.
Crossover Problems in Two-Way Loudspeaker Systems
Authors: Veneklasen Paul S.
Published: November 2005

A detailed quantitative study is presented of crossover effects due to space-phasing, crossover-network characteristics, loudspeaker impedances, mutual diffraction effects, and listening angle.
Investigation of a Reverberant Sound Field in a Concrete Sonar Test Tank
Authors: Batchelder Laurence
Published: November 2005

The acoustic behavior of a water-filled tank 30×12×7 ft deep was investigated at audio frequencies. Such an enclosure differs in theory from the usual reverberant chamber because of the phase reversal in the nearly lossless reflection at the water-air interface. The experimental results differ markedly from the theory. The average absorption coefficient decreases with frequency to a minimum of 0.06 near 5 kc/sec. The number of peaks in the frequency-response characteristic corresponds to only the normal modes tangent to the end planes, as if at least one end wall were completely absorbing. Nevertheless, standing waves do exist in all three dimensions. The standing-wave pattern is extremely complicated and quite unlike the rectangular pattern that mode theory would predict. Considerable sound appears to be transmitted through the concrete walls and floor to the surrounding earth. It is suggested that this transmission may vary greatly from point to point, depending on the inhomogeneous nature of the back fill. It is concluded that a tank of this construction is neither sufficiently reverberant nor sufficiently absorptive to be useful for the calibration of transducers. [The experimental work was performed at the U. S. Navy Electronics Laboratory and supported by the U. S. Office of Naval Research.]
Computation of Farfield Characteristics of Phased Portions of a Cylindrical Transducer from Nearfield Measurements
Authors: Baker D. D. , James D. B.
Published: November 2005

Kirchoff's formula, with a simple approximation for the normal pressure gradient, was used to compute directivity patterns of two phased portions of a large cylindrical transducer from near-field-pressure measurements made over a cylindrical surface. In one case, 16 staves of the 48-stave transducer were phased to a plane and shaded to give 20-dB minor lobe suppression in the farfield. Nearfield-pressure amplitude and phase mensurements were made by receiving with the 16-stave array while transmitting with a line hydrophone that was rotated around the transducer. The transducer was mounted with the axis vertical, and the horizontal directivity pattern for the plane passing through the center of the transducer was computed. In the second case, several staves of the transducer were driven with phasing to produce a beam depression of about 30° Nearfield measurements were made by receiving with a probe hydrophone that was rotated around the transducer at several vertical levels. There is good agreement between the computed patterns and the equivalent measured ones in both cases. This is significant because one of the original restrictions on the gradient approximation in use was that the measurement surface closely fit the radiator. The approximation has been shown to be better than it was thought to be because in the present work the surfaces used were cylindrical and the radiators—due to phasing—were portions of a plane and a truncated cone.
Parametrically Excited Surface Waves and Associated Time-independent Circulation
Authors: Brand Ronald P. , Nyborg Wesley L.
Published: November 2005

It has been established that surface waves are readily set up on vibrating gas bubbles and have significance with respect to noncollapse cavitation. In this research, studies are made of such waves by means of a large-scale model at low frequencies (25–1500 cps). Plastic-walled rectangular vessels partially filled with liquid are used, set into vibration vertically with a commerical MB Electronics vibration exciter. When the vibration amplitude is increased from zero at a given frequency ω0, a definite threshold A0 is found at which the waves first appear on the liquid surface, as predicted by theory given by Benjamin and Ursell (1954). Theory has been given by Sorokin (1957) and Eisenmenger (1959), yielding the expression (4 νk/ω0) for A0. This predicted amplitude is much less than our observed values in water in the lower-frequency range, but approaches agreement with observations at the higher frequencies. Improved agreement is attained by modifying the theory to include viscous dissipation at the container walls. Time-independent circulatory motions are observed, associated with the waves. Counter-rotating eddies spaced at intervals of λ/4 are observed, evidently of the kind predicted by Longuet-Higgins (1953). The speed is much greater than would be expected on the assumption of zero tangential stress at the surface. [This work was supported in part by the U. S. Air Force Office of Scientific Research.]
Hydrodynamical Considerations in the Theory of Wake-Vortex Sound
Authors: Ross Donald
Published: November 2005

Theoretical expressions for the dipole radiation associated with wake-vortex shedding have been developed by Curle, Phillips, and others. These well-known relations show that the sound depends on the lift coefficient associated with the oscillating force, the Strouhal number, and the spanwise correlation length of the vortices. Using hydrodynamical relations for vortex flows, interrelationships between these factors can be developed, and the relationship for the radiated sound can be reduced to an expression involving only the geometry of the vortex pattern and the steady-state drag force. By applying the Karman-Sears theory for the lift force on oscillating airfoils, an explanation is derived for observed effects of trailing-edge geometry on the production of wake-vortex noise. [This research has been partially supported by the U. S. Bureau of Ships.]
Direct Comparisons of the Autophonic Scale and the Loudness Scale
Authors: McMahon Lee , Taylor James
Published: November 2005

It has been shown that a speaker's estimate of his own vocal effort grows more rapidly as a function of the sound-pressure level of his voice than does a listener's estimate of the loudness of the speech. In order to test this effect directly, the loudness of speech was judged simultaneously by a speaker and by one or more listeners. The speaker pronounced single words at different intensities; both he and the listeners estimated the loudness of the speech. Each observer served as both listener and speaker. The speaker's estimates of his loudness were plotted against the listeners' estimates. In log-log coordinates, the points approximated a straight line whose slope provided an estimate of the ratio between the exponents of the power functions relating loudness and autophonic level to SPL. The results show that the listeners' estimates grow approximately as the 0.8 power of the speaker's estimates. An exponent less than unity is a necessary condition for a stable system when two speakers try to match the loudness of their voices.
Effects of Noise on Dichotic Temporal Resolution
Authors: Bankoff Harvey , Sutton Samuel
Published: November 2005

The effect of broadband (125–8000 cps) background noise on the dichotic temporal interval between two clicks required to report them as two was investigated. As noise level is increased up to 30 dB SL (re monaural threshold), the dichotic temporal interval required for a report of two (Δt) is decreased. For noise levels from 30–50 dB SL, the dichotic temporal interval for a report of two increases. The extent of this reversal seems related to mask level, although the effect of noise in decreasing Δt is not related to signal-to-noise ratio. With narrow-band low-pass noise, the Δt required for a report of two continues to decrease as noise level increases up to 50 dB SL, whereas a narrow-band high-pass noise shows the same reversal as does the broadband noise. Increasing the click level without adding any background noise decreases the Δt required for a report of two. Additional experiments that elucidate these findings are presented.
Estimates of Autophonic Level Are Not Estimates of Loudness
Authors: Lane Harlan
Published: November 2005

The autophonic scale relates the speaker's estimate of his own vocal level to the sound-pressure level produced. It has been suggested that estimates of autophonic level are, in fact, estimates of loudness; these “loudness” judgments can be made in the absence of sidetone (due to masking) as well as in its presence, for S can infer loudness from correlated effort [R. M. Warren, Am. J. Psychol. 75, 452 (1962)]. Findings reported previously [J. Acoust. Soc. Am. 33, 160 (1961)] and subsequent research show that estimates of autophonic level are not estimates of loudness: increased or attenuated voice sidetone has no appreciable effect on the autophonic scale; the outcome of crossmodality matches, in which Ss equate their vocal level to sound presented at various intensities, is predicted by the disparate slopes of the autophonic and loudness functions (1.1 and 0.6, respectively), as is the outcome of crossmodality compensation, in which Ss adjust their vocal level to maintain sidetone loudness constant; total masking of sidetone has only a slight steepening effect on the autophonic scale (slope 1.3); the autophonic scale for Ss with total, congenital hearing loss also has a slope of 1.3.
Use of the Aural Reflex to Measure Ear-Protector Attenuation in High-Level Sound
Authors: Nixon Charles W. , Sommer Henry C. , Cashin John L.
Published: November 2005

A technique of tympano-manometry developed for study of the aural reflex was adapted for use as an objective psychophysiological method for measuring attenuation of ear protectors in high-level sound. Four representative ear protectors (an insert, a semiinsert, a small-volume muff, and a large-volume muff) were evaluated by this aural-reflex method and by the ASA REAT method. Attenuation was designated in both instances as the mean difference between threshold values with and without ear protection (threshold shift). The aural reflex with ear protection was measured only for frequencies ranging from 400–4000 cps, due to limitations of the signal source. A comparison of the results shows that for the frequency range used the aural-reflex threshold-shift method gives attenuation data comparing favorably with other above-threshold data. The absolute-detection threshold-shift method indicates somewhat greater attenuation than does the aural-reflex threshold shift, the differences varying both with the type of ear protector and as a function of frequency. The results suggest that a single-value correction factor for adjusting standard threshold-shift attenuation data to estimate attenuation in high-level sound may not be appropriate.
Acoustic Correlates of Subglottal Activity
Authors: Ladefoged Peter
Published: November 2005

The source of power for most speech sounds is the pressure of the air below the vocal cords. This pressure is regulated by the action of the muscles of the respiratory system. A series of electromyographic experiments has shown that bursts of muscular activity, particularly of the internal intercostal muscles, occur during speech. These bursts produce peaks of subglottal pressure that can be correlated with the perceived stress of an utterance. The subglottal pressure variations are perceived also as pitch changes, since they affect the frequency of vibration of the vocal cords. A second set of experiments has shown that an increase in the subglottal pressure causes an immediate increase in frequency, which could not be caused by any adjustment of the laryngeal musculature, since such a reflex action could not have occurred except after a short reaction time. It is shown that the complex variations in fundamental frequency recorded during a phrase are often the product of a varying pressure drop across the glottis and a comparatively simple variation in vocal-cord tension.
Laryngeal Activity and the Analysis and Synthesis of Speech
Authors: Lieberman Philip
Published: November 2005

The activity of the larynx is a function of the tensions of the laryngeal muscular system, the mass of those parts of the vocal cords that are in motion, the losses caused by the collision of the vocal cord's margins, and the aerodynamic forces that act upon the vocal cords. The larynx is coupled through these aerodynamic forces to the rest of the respiratory system. The subglottal air pressure is affected by the resonances of the lungs and trachea. When the fundamental frequency of the glottal waveform or the lower harmonics of the fundamental frequency excite these resonances, the peak subglottal air pressure builds up. The superglottal air pressure is a function of the air flow through the vocal tract. Pressure transients result when constrictions occur during voicing. Variations in the glottal waveform, therefore, result from period to period, which causes rapid variations in the glottal source's spectrum and periodicity. This paper discusses the results of analyses of the acoustic waveform and high-speed ,notion pictures of the larynx and the relevence of these experimental results to the preservation of natural quality in speech synthesis.
Motor Patterns of Speech Production
Authors: MacNeilage Peter F.
Published: November 2005

There is evidence from studies of speech perception that distinctions among the segmental phonemes are made by reference to some neural representation of the articulatory movements in speech. More-direct clues to the nature of this representation may be provided by electromyographic measures than by the analysis of vocal-tract configurations or their acoustic accompaniments. Accordingly, techniques have been developed for recording the electrical activity of the muscles during articulation. The results of several such studies are discussed with reference to the broad aspects of speech production and perception. [This investigation was supported in part by PHS Research Grant DE 01774 from the National Institute of Dental Research, National Institutes of Health, Public Health Service, U. S. Department of Health, Education, and Welfare.]
Background-Noise Levels in Suburban Communities
Authors: Goodfriend Lewis S. , Cardinell R. L. , Ostergaard Paul B. , Donley Ray
Published: November 2005

Noise surveys have been made in a number of recently developed, as well as established, suburban communities. These communities are not penetrated by major expressways or railroads, and show low long-term background noise levels. These levels are presented and compared with earlier published data. The findings indicate the need for considerable care in developing noise regulations for zoning and nuisance-control statutes.
Normalized Acoustic Output of an Air Impeller
Authors: Potter Allen C.
Published: November 2005

A single dimensionless curve representing the approximate acoustic output of air impellers due to turbulence is presented. By the use of this curve, predictions of the acoustic output in 13-octave bands of various blowers are developed and compared to actual measurements made on these blowers. The normalization and, therefore, the prediction technique depend only upon the impeller dimensions and the customary performance parameters for air-moving devices. They work equally well for axial-, centrifugal-, and transverse-flow blowers whether operating at design point or far removed therefrom. Examples of all three types are presented ranging in tip speed from 2000–15 000 ft/min.
Noise due to Surface Turbulence on Printed-Circuit Cards
Authors: Lang William W. , Wise Russell E.
Published: November 2005

The surface turbulence produced by air moving over printed-circuit cards sets the lower limit on the acoustic noise levels of a computer cooling system. The sound power generated by two types of components mounted on printed-circuit cards was compared by measuring acoustic power levels in a 230-m3 reverberation chamber. The components were mounted on phenolic cards, 1.5×65×100 mm. Over 200 cards were installed in a flow chamber so the cooling air moved between parallel rows of cards spaced 11 mm apart. Miniature components (transistors, etc.) were mounted on one type of card; microminiaturized components were mounted on the other. The air velocity was varied up to the maximum normally encountered in air-cooled circuitry: 25 m/sec. The fraction of energy radiated as sound power to total energy of moving air stream, as well as the acoustic conversion efficiency per card, was determined for both card types.
High-Performance, Automatic Level-Controlling Amplifier
Authors: Hellwarth George A. , Beetle David
Published: November 2005

In the laboratory analysis of speech signals, there is often a critical need to hold the amplitude function constant over intervals as short as a pitch period or as long as words or sentences. To fill this need, an all solid-state variable-gain amplifier has been developed that automatically controls the peak amplitude of these signals. The gain-controlling element is a symmetrical transistor. By operating this nonlinear device with a constant ratio of signal current to bias current, signal distortion and thump are maintained at very low and nearly constant levels over a 60-dB range of gain. A sudden increase in input level can result in a full 60-dB gain reduction in less than 1 msec. A decrease in input level results in a logarithmic gain recovery at a constant dB-per-second rate, which is manually adjustable over a very wide range. Component selection and bias adjustments are not critical for high performance of this circuit. Gain ranges exceeding 120 dB are feasible, using the same basic techniques.
Improved Pressure-Gradient Noise-Canceling Microphone
Authors: DiMattia A. L. , Kaulins E. , Bauer B. B.
Published: November 2005

A miniature pressure-gradient microphone exhibits improved noise discrimination and extended crossover frequency. Ability of a cosine pressure-gradient microphone to discriminate against the pickup of unwanted noise is known to be limited by the acoustical pathlength between the two sound entrance points in front and back of the diaphragm. Noise discrimination is further limited by unwanted phase shifts occurring in the acoustical treatment surrounding the diaphragm. A miniature, efficient magnetic transducer actuated by a slender drive pin connected to a pressure-gradient diaphragm permits a reduction of diaphragm size with corresponding decrease of front-to-back distance. Added symmetrical acoustical networks avoid differential phase shifts. Further enhancement of noise discrimination stems from a mechanically balanced system that diminishes sensitivity to mechanical vibration. Prototypes exhibit noise discrimination of 40 dB at 200 cps and a crossover frequency above 5 kc/sec. [This work was supported by the U. S. Air Force under Contract 30(635) 18989.]
Calibration of a Miniature Condenser Microphone at Elevated Temperature and Reduced Pressure
Authors: Meyer Norman J. , Van Houten John J.
Published: November 2005

A miniature condenser microphone has been developed for operation at temperatures to 525°F and altitudes to 100 000 ft. Absolute sensitivity at room temperature was determined with the pistonphone, and comparison calibrations were perforlned over the frequency range from 20 cps–40 kcps. The frequency response of the microphone at elevated temperature and altitude was established by use of an electrostatic actuator. Sound-pressure linearity over the range from 90–170 dB (relative to 0.0002 dyn/cm2) was verified by use of a standing-wave tube. The linearity and frequency response of the cathode follower and connecting cable were investigated by voltage insertion between the microphone and microphone base. In view of the extended temperatures and altitudes, the accuracies involved during these procedures were carefully reviewed. A summary of the probable error encountered during each of the above calibrations is presented in detail.
Analysis of a Finite-Time Crosscorrelated Direction Finder
Authors: Leblanc Lester R. , Hill Richard F.
Published: November 2005

An analysis is presented that considers the effect of noninfinite integration time in a passive sonar direction finder utilizing the crosscorrelation technique. The source is assumed to be a bandlimited Gaussian signal in a sea of Gaussian noise. The reliability of the detected direction of the source is obtained as a function of the integration time, the direction of the source, and the received source to sea noise ratio.
Susceptibility to Temporary Threshold Shift as a Function of Age and Sex
Authors: Loeb Michael , Fletcher John L.
Published: November 2005

Fifty “normal” men and 50 “normal” women, divided into five equal age groups, were exposed to 12 min of 110-dB SPL 1200- to 2400-cps random noise, and temporary threshold shift (TTS) at 2000 and 4000 cps was measured approximately two minutes after the exposure. A significantly greater TTS for females than for males was noted at 2000 cps. Though no general trend with age was observed, the smallest difference in TTS between sexes was noted for the youngest group (those under 30). No significant differences in TTS attributable to age or sex were noted at 4000 cps. Interpretations of the findings are discussed. The general conclusion is that no evidence for biological difference in susceptibility as a function of age or sex may be adduced on the basis of data from this investigation.
Response Patterns of Superior-Olivary Units in Cat to Sounds
Authors: Moushegian Goerge , Rupert Allen L. , Whitcomb Milton A.
Published: November 2005

Previous studies from this laboratory have disclosed some interesting characteristics of units in the superior-olivary complex to tones and clicks. Recent findings confirm and supplement these results. Monaurally stimulated by clicks, some units show a decrease, whereas others show an increase in response latency with increasing intensities. Furthermore, those units that show a decrease in latency with an increase in intensity also show a significant reduction in their response variations. In another type of unit, the response evoked by a click in one ear may be inhibited by a click, presented simultaneously or with some time difference, to the opposite ear. Occasionally, however, if a response occurs, its latency is increased significantly. The amount of latency shift, in such cases, depends upon the interaural time difference of the clicks. Finally, monaurally presented tones show that a unit's spontaneous activity (measured in pulses per second) is progressively depressed as intensity is increased; but, as the intensity is further increased, the unit now discharges above its spontaneous rate. A model utilizing well-known parameters, excitation and inhibition of neurophysiology, is presented.
Olivocochlear and Auditory-Unit Responses in Brain Stem of Cat
Authors: Rupert Allen L. , Moushegian George
Published: November 2005

Rasmussen has shown that fibers of the olivo-cochlear bundle arise from regions of the superior-olivary complex and project into both cochleae. Approximately 80% of these fibers cross to the contralateral cochlea; the uncrossed fibers join the crossed fibers of the opposite side and leave the brain. Single units of this bundle were isolated with tungsten microelectrodes in the region of the medial superior-olivary nucleus and their responses to tones were recorded. These units have a low, regular, spontaneous activity and when excited by tones show only a slight increase in their rates, maintaining throughout the stimulation a regular but slowly adapting spike output. In marked contrast to these units, the usual auditory units of this region of the brain exhibit an irregular, spontaneous activity and when stimulated by a tone may be excited further or suppressed. The degree of excitation and suppression is dependent upon the intensity and frequency of the sound. Furthermore, the level at which the spike output settles during the stimulation may be flat or show considerable variation, suggesting a complex interaction of excitatory and inhibitory influences.
Dominant Formants in Phoneme Perception
Authors: Focht Louis R.
Published: November 2005

The 1860 Helmholtzian theory of fixed-frequency vowel perception has at times been neglected in recent years because of discrepancies between it and the observed data. The von Helmholtz theory was reexamined during a preliminary study of the minimal stimulus requirements needed to produce a desired phonetic response in the human. Results indicate that the introduction of a single dominant formant concept that plays a major role in the perception of phonemic sounds leads to a better agreement with data. An effective frequency can be associated with this dominant formant that is near the actual formant frequency, but is modified somewhat by two factors. The first of these relates to the other formants present, the second to the phonemic environment. This sheds light on such factors as the alterations in the perceived phonemic response during the speeding up of a phonograph and during the prolonged listening to a sustained phoneme. It also accounts for the shifts in response due to the varying phonetic environment. Experimental results are shown.
Investigation of Speech Sounds with an Electronic-Analog Ear
Authors: Parker Donald E. , Mundie J. Ryland
Published: November 2005

A good deal of attention has been devoted to the analysis of speech sounds into intensity-frequency-time patterns. The present investigation is concerned with the related problem of analysis of speech sounds in terms of movement of the basilar membrane. Sounds have been characterized in terms of patterns of velocity of movement at 36 locations along the “basilar membrane” of an electronic-analog ear [W. F. Caldwell, E. Glaesser, and J. L. Stewart, “Design of an Analog Ear,” in Biological Prototypes and Synthetic Systems, edited by E. E. Bernard and M. R. Kare (Plenum Press, New York, 1962)]. Following conversion into digital form, the response of the analog ear was stored in a PDP-1 computer. For visual analysis, the digitized response was displayed on a 16-in. oscilloscope in three-dimensional patterns (location-velocity-time). Initial studies with the analog ear had two main goals: (1) specification of velocity patterns for the various sounds present in speech, and (2) simplification of the velocity patterns by determining the minimum number of basilar-membrane locations required to discriminate the patterns. The second goal has been pursued on the assumption that the locations that have the greatest variability (σ2) as determined across speech sounds would be most important for the discrimination of those sounds.
Physical and Psychophysical Descriptions of Sixteen Equally Speech-Interfering Noises
Authors: Webster J. C. , Klumpp R. G.
Published: November 2005

Physical and psychophysical measurements and/or calculations were made on 16 equally speech-interfering noises. Speech-interference levels (SIL) based on octaves with center frequencies of 500, 1000, and 2000 cps or at 425, 850, and 1700 cps predicted very well that the noises were equally speech-interfering. These SIL's predicted better than the more conventional SIL based on octaves between 600 and 4800 cps. Articulation-index (AI) scores based on 5 or 6 weighted octaves were equivalent to the 20-band AI methods and were not necessarily better than SIL methods. Fitting the peaks of plotted spectra to noise-criteria (NC or NCA) curves was not adequate, although averaging that part of the spectra centered on octaves from 500 through 2000 cps through NC or NCA curves gave good prediction. Equally speech-interfering noises were not equal in loudness as calculated by Stevens' method, or in PNdB. There was evidence to show that the frequency that divided noise-masked speech into two equally intelligible frequency regions was 850 or 1000 cps, and was not 1700 and 1900 cps, which is the dividing frequency when the speech in quiet is progressively restricted in bandwidth by high- or low-passed filtering.
Noise Reduction of Small Enclosures
Authors: Lyon Richard H.
Published: November 2005

The noise reduction produced by a small rigid enclosure with one flexible wall is computed. The wall has an assumed coincidence frequency above the first acoustic resonance of the enclosure. The noise reduction is computed at very low frequencies where both wall and enclosed volume are stiffness-controlled, at frequencies where the wall has resonant behavior and the volume is stiff, and at frequencies where both the wall and the acoustic space have resonant behavior. In the last case, a comparison is made with the usual noise-reduction calculations based on a “field-incidence” transmission loss. The noise reduction computed in this paper turns out to be less than that computed by the more “classical” approach. [This research has been supported by the National Aeronautics and Space Administration.]
Frequency Dependence of Different Ways to Obtain Diffuse Sound Fields in Reverberation Chambers
Authors: Venzke Günter H.
Published: November 2005

In the reverberation chamber of the Physikalisch-Technisehe Bundesanstalt (PTB), the absorption coefficient of two absorbing arrangements has been measured for testing different states of equipment of the room with diffusing elements. The room has fixed convex cylindrical segments on two walls and the ceiling. In addition, movable spherical segments can be fastened to the walls and/or diffusing elements can be suspended in the space. With uncovered mineral-wool slabs as the testing material, additional space diffusors change the resulting absorption coefficient more or less in the frequency range above 200 cps. Decrease or increase of the absorption coefficient depends on the test-area size and on the number of diffusors. In the second arrangement, a cover of linoleum was fixed on the same slabs, thus performing an efficient low-frequency absorber with a resonance at 140 cps. With diffusors of 2 m2 each present in the space (their total area amounting to at least 20 m2), reasonable results were gained. The available cylindrical and spherical segments on the boundaries could not produce sufficient randomness in the low-frequency range.
Comparative Data of Field and Laboratory Measurement of Partition-Construction Transmission Loss
Authors: Hansen R. A. , Kodaras M. J.
Published: November 2005

The field-measurement data of transmission loss of partitions, permanent or movable, often do not agree with laboratory transmission-loss data of similar partition constructions. Generally, two methods are available for use in field measurements: (1) quasi ASTM E90-61T and (2) the A. London method [as described in Natl. Bur. Std. J. Res. 26, 419 (1941)]. However, each method has certain advantages and disadvantages associated with its performance in the field. The results of the various field tests, using both methods, are compared to laboratory tests for specific partition construction. Reasons for unsatisfactory agreement between field and laboratory TL measurements are presented. A composite method for field laboratories of TL is proposed for further investigation, which may result in the essential formation of a method that would provide data that are reasonably comparable to laboratory measurement, and can be considered a reliable evaluation of TL of the partition construction in the field.
The Marina as a Two-Dimensional Auditorium
Authors: Morrow Charles T.
Published: November 2005

The problem of reducing the waveheight in a marina or artificial harbor by comparison with that of the open ocean is treated by analogy to the classical derivations of auditorium acoustics. A formula is obtained that relates the effectiveness of the marina to the breadth of the mouth and the total absorption of wave energy by the interior shoreline.
Acoustics of St. Luke's Episcopal Church, Dallas, Texas
Authors: McCandless David , Lane R. N.
Published: November 2005

This paper presents an unusual wall treatment, plus trial and adjustment of reverberation time, in a new church building, St. Luke's Episcopal Church in Dallas, Texas. The architectural concept and the original acoustical design of this church-in-the-round are presented first. The organist and music committee insisted on a high reverberation time, and, since access for future placement of absorbent blanket was available in the walls, it was decided to install the blanket only after obtaining the congregation's reaction to the relatively long reverberation time (2 sec plus, occupied). Two years after the dedication of the church, the reverberation time was adjusted downward to improve speech intelligibility. Fiberglas blanket was installed in random patches, 10–16 ft long, both vertically and horizontally, just behind the curved, screened side walls, with an 8- to 10-ft airspace behind that, back to the building walls. Data are presented on the response time of the church before and after the installation of the absorbent material, and the reactions of the congregation, before and after treatment, are reported. A comparison of original calculations and final results is made, and the absorption coefficients are presented for the Fiberglas blanket as installed.
Electronic Room-Acoustic Fundamentals
Authors: Volkmann John E.
Published: November 2005

While good acoustics has long been the goal of room-acoustic designers, the enhancement and simulation of natural acoustics by means of active electronic transducers is seldom fully utilized. Ideally, every link in the chain of events represented by the energy terms in the room-acoustic equation should be considered separately in the analysis or synthesis of the complex sound waves in a space; however, practically it is sufficient to group the energy terms into three major components: namely, the original direct energy, the early reenforcement energy, and the later reverberant energy. Criteria, principles, and other factors related to this design concept are discussed. Two early demonstrations of electronic room-acoustic systems—namely, “An Outdoor Electronic Orchestra Shell with Synthetic Hall Acoustics” and “Stereo Panoramic Sound Reproduction,”—both in freefield environments, are described briefly, along with other applications. Ordinary sound reproduction, either monophonic or stereophonic, does not completely satisfy the basic requirements for spatial sound realism because of false directional effects and the lack of an enveloping sound field. The possibility of simulating hall acoustics in the reproduction of sound in the home is covered. A short demonstration of “electronic room acoustics” is contemplated.
Air-Borne-Sound-Isolation Requirements for Dormitories
Authors: Curtis J. A. , Cavanaugh W. J.
Published: November 2005

The dormitory is an important facility for individual student study at most college and in-residence types of secondary schools. Ineffective air-borne sound-transmission loss between rooms often limits the dormitory's usefulness for the study, as well as for its other functions. The results of field measurements in a number of completed college dormitories are presented. Some conclusions are drawn as to suitable criteria for air-borne-sound isolation.
Central Factor in Pure-Tone Auditory Fatigue
Authors: Wernick Joel S. , Tobias Jerry V.
Published: November 2005

A long accumulation of psychophysical and physiological evidence indicates that auditory fatigue has its locus of effect in the cochlea; transfer studies with negative or questionable results, and studies of cochlear chemistry and potentials with positive results all lead to the same conclusion. However, Galambos' reports of inhibition through efferent stimulation taken together with his finding of changes in cochlear potentials corresponding to differences in “attentiveness” provide a basis for testing the existence of a central factor operative in auditory fatigue. The present study was designed to evaluate the possibility that a listener's level of mental activity constitutes an adequate central factor. Subjects were presented with a 4000-cps, 40-dB SL tone for 3 min under conditions of (a) a mental task or (b) reverie. Pre- and post-fatigue thresholds were measured with a switched (0.2 sec on, 0.2 sec off) 4000-cps tone. Subjects consistently showed greater temporary threshold shift and longer recovery time when the fatiguing tone was presented in the mental-task condition. Results thus indicate a central factor in pure-tone auditory fatigue. It is suggested that the degree of effect of a fatiguing tone is a function of the listener's degree of mental activity during stimulation.
Cyclical Masking of Clicks by Low-Frequency Tones
Authors: Shickman Gerard M.
Published: November 2005

In 1934, Derbyshire and Davis reported that auditory-nerve responses to clicks vary in size with different timing of the clicks relative to the cycles of a masking tone. Recent tests with pure-tone maskers have demonstrated a psychophysical parallel to this finding. The masked-threshold curve for clicks presented at about 3/sec, as traced out in time by a modified Békésy recording method, changes cyclically as the clicks occur later and later in recurrent tone cycles. With very small increments in arrival time, the method is essentially equivalent to the expansion of a few cycles of the masker to fill a 10-min listening session. The threshold undulation is observed mainly under masking by low-frequency tones at fairly high sensation levels. As the masking frequency rises, the maximal threshold shift continues to accompany the falling phase of the sinusoidal pressure wave produced by the PDR-10 earphone, but threshold peaks lag pressure peaks by diminishing phase angles. The minimum or “steady” threshold shift increases, and the undulatory or “cyclical” threshold shift gradually disappears, as the masking frequency rises from 100 to about 700 cps.
Spread of Masking in Ears Showing Abnormal Adaptation
Authors: Harbert F. , Young I. M.
Published: November 2005

Threshold measurements were made by Békésy sweep-frequency audiometry for pulsed and steady tones in the presence of wide- and narrow-band noise introduced in the same receiver. Threshold shifts produced by various intensities of noise were compared for normal and pathologic ears. Most pathologic ears showed findings similar to normal ears. In six cases, all of whom had marked abnormal adaptation, the masking effect was strikingly different. Masking effects at the center of the narrow band were 5 or more dB greater than in normal ears and the lateral spread of masking approached the effect of white noise with narrow-band masking.
Continuous Changes in Cochlear-Microphonic-Distortion Processes under Normal and Abnormal Physiological Conditions
Authors: Wolsk David
Published: November 2005

There is a long history of controversy concerning the locus and mechanism of harmonic-distortion processes in the cochlea. This controversy has importance for theories of frequency analysis along the basilar membrane and concepts of the transduction process in the hair cells, as well as for the clinical applicability of psychophysical tests of aural harmonics. The cochlear-microphonic response at the round window of guinea pigs to a pure tone was led into three wave analyzers tuned to the 1st-, 2nd-, and 3rd-harmonic frequencies, each component then driving one channel of a pen recorder. The various components exhibit shifts through time, which were studied as a function of stimulus frequency and intensity, recovery from fatiguing and damaging levels of tonal and random noise stimulation, anoxia, blood pressure, and ototoxic drugs. The data are discussed in terms of mechanical vs transformation concepts of harmonic distortion in the cochlea. [Supported in part by U. S. Army Contract.]
Consistency of Auditory-Detection Judgments
Authors: Green David M.
Published: November 2005

An observer's decision in a psychoacoustic detection experiment is governed by two broad classes of determinants: (1) external determinants, such as the likelihood of a particular waveform being a signal in noise, and (2) internal determinants, the momentary state of the observer's nervous system, his response biases, or biases as to certain sequences of responses. The aim of the study was to determine, via a measure of the observer's consistency, the relative contribution of these factors. To achieve this end, the audio information presented during a sequence of two-alternative forced-choice trials was taped and repeated to the observer at a later time. The consistency of the observer's judgments was measured by determining a percent agreement score: the percent of times the subject agreed with his previous response on those special trials of the sequence in which no signal occurred on either interval of the forced-choice trial. Percent agreements range between 80% and 55%, depending on the observer and, perhaps, on the signal occurring on the signal trials. A simple linear model is used to establish a lower bound on the ratio of internal to external noise. Unlike some previous experiments, little evidence could be found for large response dependencies in this type of task. That the observer could occasionally hear the signal probably explains why his behavior remained, to a large degree, under stimulus control. [This work was supported in part by the U. S. Army, the U. S. Air Force Office of Scientific Research, and the U. S. Office of Naval Research, and in part by the National Science Foundation (Grant G-21807).]
Human Perception of Objects, Using Pulsed Sound
Authors: Beil Ralph G.
Published: November 2005

The various animal echolocation systems that have been investigated in recent years, such as those of the porpoises and the bats, were examined in the light of object-perception efficacy and their application to human echolocation systems. This led to the quantitative exploration of the ability of human subjects to perceive cylindrical objects by means of acoustic echoes. The experiments were conducted in an anechoic environment, using an acoustic pulse-train generator as a signal source and unaided human ears as receptors. The effect of systematic permutation of parameters associated with the outgoing signal on the size threshold for object perception was measured. An attempt was made to define an optimum signal for purposes of human target detection. Subjects were able to perceive aluminum cylinders 120 in. in diameter at a distance of 10 ft using a near-optimum signal. A theoretical model was formulated explaining the experimental results in terms of acoustic-scattering theory and the parameters of human hearing. Additional experiments were performed that measured object size, shape, and distance discrimination.
Measurement of Tonal Density
Authors: Stevens S. S. , Guirao Miguelina
Published: November 2005

Recent success in the scaling of tonal volume [H. S. Terrace and S. S. Stevens, Am. J. Psychol. 75, 596–605 (1962)] has led to an attempt to determine how tonal density depends on frequency and sound pressure. Density refers to the apparent “compactness,” “concentration,” or “hardness” of a tone. Loud, high-pitched tones have high density; soft, low-pitched tones have low density. Thus, density increases with both frequency and sound pressure. An attempt was made to determine directly the form of the density surface by asking observers to assign to a series of tones numbers proportional to the subjective density. From these magnitude estimations, the forms of the equal-density contours were estimated. These equal-density contours were later checked by a direct-matching procedure.
Effect of a Background Noise on the Loudness Function at 1000 cps
Authors: Hellman R. , Zwislocki J.
Published: November 2005

Only a few studies have been concerned with the measurement of the loudness of a 1000-cps tone in the presence of noise; and there is a considerable discrepancy among the results, especially at low sound intensities. The discrepancy appears to be due to biases in procedures. In order to determine the form of the masked-loudness function, a series of experiments, which were designed to reduce the biases to a minimum, was performed. It has been found that, by decreasing the procedural biases and by obtaining measurements under similar experimental conditions, the average of magnitude estimation and magnitude production can be used to corroborate the results of loudness matching. By combining the loudness-matching field with the one determined by magnitude estimation and magnitude production, the best estimate of the masked-loudness function is obtained. Two loudness curves are constructed, one for a 40-dB threshold shift and the other for a threshold shift of 60 dB. The form of these curves is in good agreement with the results of other investigations in which balanced procedures are followed. The theoretical implications of these data are examined. (Work supported by a contract between the U. S. Office of Naval Research and Syracuse University.)