bluuebunny/crossref_metadata_2025 huggingface dataset:
The Intelligibility of Hydrogen Speech
Authors: Sergeant Russell L.
Published: October 2005

Deep sea divers substitute helium-rich breathing mixtures for air in order to alleviate nitrogen narcosis at depths of approximately 100 ft or more. Some people propound that hydrogen would be an even more suitable substitute for air during extremely deep dives. Theoretical explanations of the marked change noted for helium speech would indicate an even more drastic change should occur for hydrogen speech. This study evaluated effects of breathing hydrogen on speech intelligibility. Taped recordings were obtained within an hyperbaric diving chamber while a diver was breathing compressed hydrogen, helium, and air mixtures. Responses by six panels of listeners revealed, unexpectedly, that there was no appreciable difference in mean intelligibility of speech between the helium and hydrogen conditions. In fact, the hydrogen speech was slightly more intelligible than the helium speech. Interpretation of these results will be made according to theory of speech production and application to deep-diving communication systems.
Flow Noise Prediction and Control in Steam Piping Systems for Nuclear Power Plants
Authors: Brittain F. H. , Giampapa S. W.
Published: October 2005

The flow noise of steam in pipe lines, particularly in power plants, is a major noise source and contributor to OSHA noise problems. The ability to predict flow noise levels is vital to efficient and economical noise control. Octave-band measurements of flow noise in the main steam piping system of a nuclear power plant were made. To determine the effect of velocity, measurements were conducted for a wide range of velocities during plant start-up. Results in the form of plots of measured flow noise as a function of velocity were compared with limited data that have been recently published. An empirical formula for prediction of flow noise and corresponding design techniques for control of noise by proper pipe sizing have been developed. Alternate methods of noise control are reviewed.
Transmission of Sound Through Pipe Walls as a Function of Flow Within the Pipe
Authors: Holmer Curtis I. , Andersen Douglas W.
Published: October 2005

The transmission of sound through pipe walls is an important component of the problem of determining the noise generated by valves. In this paper we present a progress report on an experimental program to determine the transmission loss (TL) of a series of pipes, as a function of flow within the pipe. The experimental program is outlined, and the results of measurements on a 6-in.-diam., schedule 40 (0.29-in. wall thickness) pipe are discussed in detail. The experimental data for this pipe indicate three different frequency regimes, where different processes controlled the TL. At low frequency (below ka ≈ 3), the TL shows a −12 to −18 dB per octave slope, and is relatively unaffected by flow. Between this frequency and the ring frequency (∼10 kHz), the TL is roughly independent of frequency, but is strongly affected by flow. (TL in this frequency range decreased by approximately 10 dB with a flow change from 0 to 400 ft/sec.) Above the ring frequency, the TL exhibited a 3 to 6 dB per octave slope, and a less strong flow dependence. These data are then compared with theoretical TL estimates by Cremer, Heckl, and Morfey. [;Work supported by Westinghouse Corporation, Steam Turbine Division, Lester, Pennsylvania.]
A Systems Approach for Determining the Feasibility of Industrial Noise Control Through Engineering
Authors: Bailey J. Ronald
Published: October 2005

When employees are exposed to sound exceeding permissible levels, employers are required under the Williams-Steiger Occupational Saftey and Health Act of 1970 to maintain a continuing effective hearing-conservation program and to utilize feasible administrative or engineering controls. While many aspects of compliance such as hearing protection, audiometric programs, administrative controls, etc., are generally understood, one question remains unanswered, namely, what constitutes feasibility for engineering control? The complexity of this problem is clearly recognized. Problems of this magnitude are amenable to the systems approach, which has been adopted for this investigation. The result is a flow diagram which can be used as a “road map” for evaluating noise control feasibility. General suggestions are made for controlling the noise produced by air sources, oscillating bodies, frictional sources, and gears. Absorption, enclosures, and barriers have also been considered as means of interrupting the path between the source and the receiver. Noise exposure has been included as an important factor, with economic feasibility as a final consideration in the flow diagram.
Characteristics of Octave-Band Noise Spectrums for Several Different Industrial Environments
Authors: Royster L. H. , Stephenson James E. , Scott C. E.
Published: October 2005

Over 2000 octave-band noise spectra were obtained from several different industries. These spectrums were analyzed with respect to the existence of pure tones, approximate flatness, allowed exposure times as compared with the 90-dBA and 85-dBA criteria, maximum SPL variation, etc. The preliminary results of these analyses will be presented.
Effectiveness of Different Ear Protectors in Protecting the Employee from Overexposure in Industrial Environments
Authors: Stephenson James E. , Royster Larry H.
Published: October 2005

Several different ear protectors were selected for evaluation. In order to rate the protection ability of each protector, 1000 different octave-band noise spectrums were selected, representing different types of industrial environments. The effective protection provided was then determined by using the “published” attenuation data and comparing then allowed exposure times using the 90-dBA and 85-dBA criteria and the CHABA damage risk contours for broad-band and pure-tone environments. The preliminary results of these analyses will be presented.
Experimental and Theoretical Study of Propagation of Finite-Amplitude Noise in a Pipe
Authors: Pestorius F. M. , Blackstock D. T.
Published: October 2005

In an earlier paper [J. Acoust. Soc. Amer. 53, 383(A) (1973)], the authors presented a computer algorithm for predicting the distortion of plane, finite-amplitude acoustic signals of arbitrary waveform. Experimental measurements of 160-dB periodic waves in a 2-in.-diam, 100-ft-long pipe confirmed predictions after the algorithm had been modified to account for attenuation and dispersion caused by the acoustic boundary layer at the pipe wall. The algorithm has now been used to compute the distortion of high-intensity noise. Bursts of broad-band noise of SPL 160 dB and higher have been used. Again, the computed time waveforms closely resemble the observed ones. Qualitatively, the noise waveform becomes simpler in appearance (for example, the number of zero crossings decreases) as the signal travels down the pipe. This change is due to the fact that as the shock waves form, grow, and decay, they “eat up” minor irregularities in the waveform. Information on the spectral content of the noise as a function of propagation distance will also be presented. [Work supported by ONR, AFOSR, and USN Postgraduate School.]
Acoustic Doppler Wind Measuring System
Authors: Beran D. , Carsey F. , Willmarth B.
Published: October 2005

A system using the Doppler shift of audible sound waves to measure atmospheric winds to heights of 1 km above the surface is described. Planned application of the device at a commercial airport required that optimum transducer and antenna shielding be provided in order to reduce the effect of ambient noise. A description of the antenna and shielding is followed by an outline of the remaining system components. Comparisons of winds measured by both the acoustic Doppler system and standard radiosondes are presented. These results indicate that a rather high degree of accuracy and resolution can be achieved and that the wind profiles can be continually monitored with relatively simple ground-based equipment.
A Retro-Reflection Calibration of Acoustic Antennas
Authors: Carsey F. , Hall F. F. , Neff W. D.
Published: October 2005

Two parabolic reflector antennas are calibrated: one a converted WWII searchlight and one a fiberglass “sugar scoop.” The technique of calibration is to aim the main lobe of each antenna at a corner reflector mounted atop a 300-m distant 150-m high tower instrumented with meteorological sensors. Acoustic pulses 40 to 200 msec in length at several frequencies between 1 kHz and 5 kHz are propagated toward the reflector and subsequently reflected back toward the same antenna for reception. Data from the meteorological sensors are used to compute the expected spreading of the beam due to atmospheric turbulence, and this predicted beam shape is compared with the antenna gain pattern (measured a few Rayleigh distances from the apertures) by moving the reflector down the tower. To obtain a reasonable estimate of the mean log intensity, averaging must allow the wind variance to refractively move the main beam through a total of several Fresnel zones. This requires the averaging of 10 to 20 pulses.
Measurement of Axially Symmetric Directivity Patterns near a Rough Surface
Authors: Barger J. E.
Published: October 2005

Accurate measurement of underwater radiated-sound directivity patterns is often frustrated by unwanted surface scatter. When the surface is acoustically rough, a simple image source correction is not appropriate. This paper describes a rather special case whereby surface scatter-contaminated sound measurements are sufficient to determine the free-field directivity pattern—without the need for rotating the sound source. This case requires measurements along the axis of pattern symmetry, which is to be parallel with the surface. The technique is illustrated with examples of directivity patterns determined from surface scatter-contaminated data taken along the maximum response axis of a sound source located beneath a diffuse-scattering surface. These directivity patterns compare closely to patterns obtained by rotating the source in essentially free-field conditions.
Steady-State Transducer Calibration in a Reverberant Water Tank
Authors: Watters B. G.
Published: October 2005

A nonanechoic water tank can be used for transducer calibration in either a pulsed “free-field” or steady-state “diffuse-field” mode. The diffuse-field mode may be preferred if the desired output is a random incidence calibration (e.g., for a hydrophone to measure noise in a diffuse environment). The diffuse-field mode may also be a desirable alternate technique for nondirective transducers. We have evaluated some of the sources of calibration variance for the BBN reverberant water tank (dimensions 14×22×33 ft) both for small, nondirective transducers and for transducers with DIs of approximately 20 dB. The degree of diffusivity at high frequencies has been measured using a directive hydrophone. Both secondary (comparison) and primary (reciprocity) hydrophone calibrations will be discussed.
Some Notes on the Evaluation of the Acoustic Parameters of Sound-Absorbing Materials by Stationary-Wave Methods
Authors: Ianniello C. , Vanoli R.
Published: October 2005

This paper describes some impedance-tube methods used for the determination of the absorptive qualities of acoustical materials. Several materials have been investigated at different frequencies by each method and results are compared.
Measurement of Phase Angle of the Velocity of a Radiating Piston
Authors: Miller Harry B. , Vogelsong Richard D.
Published: October 2005

The relative phase angle between the velocity of a transducer head and the driving voltage or current can be measured directly by an accelerometer or indirectly by a pressure hydrophone. Some nondestructive and quasi-nondestructive methods will be discussed.
Engineering Acoustics Workshop on the Accuracy of Calibrating Underwater Sound Transducers
Authors: Bobber Robert J.
Published: October 2005

Measurement accuracy in the calibration, test, or evaluation of underwater sound transducers is a somewhat neglected subject. It is the purpose of this workshop to shed some light on the subject and the reasons for its neglect. A panel of four, representing three naval laboratories and an observer of European laboratories will address several questions. What do the laboratories claim for accuracy? What is meant by “accuracy”? What is the claim based on? Can more accurate measurements be made? What would it cost? Is there a need or demand for more accuracy? After a ten-minute informal presentation by each panel member, the floor will be open for comment and contributions by others who perform measurements, use the data, or are otherwise interested in the subject. Representatives of several industrial and university, as well as naval, organizations have been invited to comment. The panel members are: Joseph Blue, Underwater Sound Reference Division, Naval Research Laboratory, Orlando, Florida; Charles Green, Naval Undersea Center, San Diego, California; Joseph Navin, Naval Underwater Systems Center, New London, Connecticut; W. James Trott, Acoustics Division, Naval Research Laboratory, Washington, D. C., and Chairman, Working Group on Hydrophones, International Electrotechnical Commission Technical Committee on Ultrasonics.
Electrostatic Transduction
Authors: Janszen Arthur A.
Published: October 2005

Electrostatic transduction became one of Professor Hunt's lively interests in 1947, when it was suggested as an alternative means of producing audio-frequency tone bursts of predictable shape, to be used in a study of the transient response of linear arrays of receiving transducers. The first electrostatic loudspeaker built in that year produced waveforms that were free of evidence of multiple uncoupled or loosely coupled vibrational modes. A subsequent series of experiments disclosed means of increasing the range of useful frequency response and the maximum per-unit-area acoustic power output. This paper will describe some of the experiments that proceeded under Professor Hunt's direction during eight years of the author's participation, and some of the consequences of the program that are detectable in current transducer technology.
The Behavior of Sound in a Bounded Space
Authors: Berman J. M.
Published: October 2005

Consideration of the currently received theories of sound behavior in a room brought to light some severe anomalies. A concrete model of a rectangular room was constructed such that the number of boundaries could be varied. The steady-state and transient responses were investigated for different boundary configurations, and these were compared, first with predictions from the classical normal-mode model, and second with digital computations from the impulse response, based on image model. The normal-mode concepts were found to give a misleading picture of sound behavior, whereas the time-domain model was able to account for all results including those at frequencies comparable with the boundary dimensions where the normal-mode model is usually substantiated. This model has also led to the prediction and verification of some unexpected phenomena in the fine structure of transient waveforms which, in the light of recent work on the hearing mechanism, may well be responsible, in part, for our subjective response to an environment.
Loudspeaker Design and the Mathematical Theory of High Fidelity
Authors: Mawardi O. K.
Published: October 2005

An attempt is made to develop an objective procedure to assess the quality of loudspeakers. The method is based on the premise that the evaluation of loudspeakers can be viewed as a problem in experimental esthetics. Borrowing on techniques used in psychology, a scale of merit is devised which is adapted to prescribe loudspeaker-design guide lines.
Measurements of Acoustic Intensity in Reactive Sound Fields
Authors: Schultz T. J. , Smith P. W. , Malme C. I.
Published: October 2005

The usefulness of direct measurements of vector intensity for the calibration of acoustic transducers has been studied by laboratory experiments in anechoic and reverberant chambers and by analysis. Determination of total power output is feasible by intensity measurements in the nearfield and in reverberant spaces, situations where pressure measurements would be inadequate without elaborate calculation. However, the directional distribution of intensity differs from the farfield, free-space directivity both near a source and far from a source but in a reverberant space. Analysis shows no theoretical way to correct for these differences. In some circumstances, even though intensity measurements do not yield the exact directivity pattern, the differences may be acceptable. [The intensity measurement instrumentation was developed at Harvard University, and both the original work and the study reported here were supported by the Office of Naval Research.]
Radiation and Scattering by Submerged Elastic Structures
Authors: Junger M. C.
Published: October 2005

Analysis of submerged vibrating structures and of the associated sound fields requires the simultaneous solution of the vibrational and acoustical problems, because surface pressures exerted by a dense medium are typically comparable to inertia and damping forces in metal structures. The scattered field generated by a submerged structure may be similarly altered by the elastic response to the incident wave. This feedback phenomenon, whereby forces exciting the structure must be combined with radiation loading dependent on the structural response, is unimportant in the atmosphere. Consequently, as underwater sound approached airborne sound in practical importance, acousticians were forced to develop new analytical and experimental techniques for studying structure-water interactions. The development of this hybrid branch of acoustics and structural mechanics is reviewed with regard to radiation and scattering. Hunt, even though never personally active in this field, was interested in its applications. He performed his characteristic function of a catalyst by sponsoring theses and encouraging postdoctoral fellows of the Harvard Acoustics Research Laboratory to venture into the novel area of structural acoustics. [Paper prepared under contract with the Office of Naval Research, Acoustics Programs.]
Upstream and Downstream Sound Radiation into a Moving Fluid
Authors: Ingard Uno , Singhal Vijay K.
Published: October 2005

An experimental demonstration of the influence of fluid motion in a duct on the sound-pressure field radiated in the upstream and downstream directions from a stationary sound source is presented. The ratio between the upstream and downstream sound-pressure amplitudes is obtained as a function of the flow Mach number in the range M=0–0.5. The results are discussed in the light of a mathematical analysis of the problem. An observed irregularity in the Mach number dependence of the pressure ratio is interpreted as a result of a flow transition which in effect alters the boundary conditions at the source. [This work was supported by the U. S. Navy (Office of Naval Research) under Contract N00014-67-A-0204-0019.]
Nonlinear Interaction of Sound with a Resonator in a Duct
Authors: Galaitsis Anthony G.
Published: October 2005

An acoustic resonator with a resistive load is attached to a side wall of a duct and is excited by a sound wave. For wavelengths greater than the maximum dimension of the duct cross section, the plane-wave treatment is employed to calculate the absorbed, transmitted, and reflected fractions of the incident power. The results are subsequently compared to experimental findings. It is found that as the sound level increases, the absorbed and reflected fractional powers decrease while the transmitted power increases. [This work was supported by Grant Agreement DOT-OS-30011 with the U. S. Department of Transportation.]
Sound Transmission in a Duct with Nonuniform Boundary Conditions
Authors: Vaidya P. G. , Beard J. , Herrick Ray W.
Published: October 2005

Nonuniform mufflers have been used in practice for a long time. However, a systematic analytical and experimental evaluation of such mufflers is lacking. This paper presents an account of a theory for analyzing sound transmission under nonuniform boundary conditions, along with details of an experimental setup for investigating sound fields under such conditions.
Acoustic Backreaction on Sound Sources in Ducts
Authors: Vaidya P. G. , Baade P. K.
Published: October 2005

Analysis of Acoustic backreaction on a sound source is of fundamental importance for measurements in duct acoustics. In this paper, a preliminary analysis of the problem is presented. For the purpose of analysis, a source is considered to be a combination of an ideal source (such as monopole, dipole, or quadrupole), in series with a constant impedance. It is shown that the relative magnitude of the source impedance would lead to the more familiar idealizations such as constant velocity or constant pressure source. The backreaction is shown to be strongly dependent on the length of the duct. The backreaction for higher-order modes is more prominent near their cutoff frequencies and wall absorption strongly influences its magnitude.
Calculation of Perforated Plate Liner Parameters from Specified Acoustic Resistance and Reactance
Authors: Guess A. W.
Published: October 2005

A mathematical procedure is developed to calculate the parameters of a single-layer perforated plate acoustic liner in order to achieve a specified acoustic resistance θ and reactance χ for single-frequency excitation. The procedure includes the standard impedance terms (due to viscous, radiation, and backing effects), and emphasis is on terms due to high sound-amplitude and steady airflow (“nonlinear” effects). Specification of θ and χ (at one frequency) means that two quantities are used to determine the four liner parameters σ (fraction open area of plate), t (thickness of plate), d (diameter of holes), and b (depth of backing cells). The developed method has the following sequence of steps: (1) convenient values are assumed for t and d; (2) the resistance equation is solved for σ (is a cubic algebraic equation in general); and (3) the reactance equation is solved for b (involves inverse cotangent in general). Various special cases give simplifications in the solution procedure. Higher values of sound amplitude and/or steady airflow velocity necessitate higher values of σ. While the procedure applies strictly for single-frequency excitation, it is probably also valid for a highly periodic excitation. The possibility of solving for values of σ, t, d, and b to achieve specified values of θ and χ under two separate environmental conditions is also discussed.
Attenuation and Redistribution of Sound Energy in Lined Annular Ducts with Shear Flow
Authors: Mikhail M. N. , Abdelhamid A. N.
Published: October 2005

The effects of shear flow and duct lining on the attenuation and energy distribution of complex sound waves in annular ducts are theoretically investigated. Detailed quantitative results show that the extent of redistribution of modal energy due to shear flow effects increases with boundary-layer thickness, flow Mach number, and importantly with the difference between the sound frequency and the mode cutoff frequency. For the frequency range and liner impedances considered in this study, the attenuation of sound propagating with the flow increases with boundary-layer thickness and Mach number. This trend is reversed for sound propagation in the upstream direction. The effects of liner impedance on sound attenuation and mode shapes are also investigated in detail. The information presented in this paper leads to a better design of annular duct liners and accurate prediction of the directivity patterns of sound radiated from duct ends. [Research supported by National Research Council of Canada.]
The Sounds of Artificial Lighting
Authors: Remillard W. J.
Published: October 2005

An experiment was performed which was designed to measure and record some salient features of thunder in the absence of reflections, strong temperature gradients, and turbulent air motion. A 10-ft spark with a current-versus-time waveform similar to that of a typical lightning stroke was produced. Measurements were made of the direct sound waves from the spark. A description of this experiment will be presented and the results will be compared with those predicted by mathematical models of lightning.
Stochastic E-L Wave Theory: Perturbation Techniques
Authors: Neubert J. A.
Published: October 2005

Although stochastic E-L (Eulerian-Langrangian) wave theory does not require random perturbation techniques in principle to provide solutions, in practice, ensemble expectations are usually resolved by perturbing the random medium. The fundamental concepts for the evaluation of wave functionals are first treated in one dimension, then a central limit theorem for Lagrangian functionals permits these concepts to be treated in three dimensional analysis via stochastic perturbation. The resultant relation for calculating the pertinent Eulerian ensemble expectation corresponding to the Lagrangian functional in question is shown to depend in a natural manner upon the Lagrangian path spreading and, in the case of a statistically isotropic medium, to reduce to the proper nonstochastic limit. Thence, stochastic E-L wave theory can be demonstrated via stochastic perturbation techniques applied to secular Lagrangian relations. In the stationary Markovian limit, the comparable Weiner integral results.
Cues Required for Externalization and Vertical Localization
Authors: Searle C. L.
Published: October 2005

Frequency response measurements of the human pinna for a sound source at a number of different elevation angles in the vertical median plane show surprising 30-dB dips in the 8- to 12-kHz frequency range. For some subjects, these dips or notches tune down in frequency almost linearly as a function of elevation angle. Detailed measurements indicate that there is a systematic frequency difference between the left- and right-ear notches. Psychophysical experiments show that these notches are perceptually significant and can be used to synthetically change the apparent angle of arrival of a sound source. On the basis of these experiments we postulate that vertical median plane localization is achieved by taking the ratio of the left-ear to the right-ear frequency response in certain key frequency bands. By so doing, the brain could deduce the incident angle of signals in the vertical median plane without knowledge of the spectrum of the incoming signal.
Sound Localization in the Bottlenose Porpoise, Tursiops truncatus: Signals Presented on the Horizontal Plane
Authors: Renaud Donna L. , Popper Arthur N.
Published: October 2005

Sound localization was investigated in the porpoise using a method of localization resolution (minimum audible angle, MAA). The animal was tested in a pen acoustically open to Kaneohe Bay at the Naval Undersea Center facilities, Oahu, Hawaii. The MAA was determined using an up-down staircase method of threshold determination at the 70% level in a spatial two-alternative forced-choice design. The animal maintained a fixed spatial relationship to the transducers by biting a stationary bar placed 61 ft from and midway between two underwater transducers. The pure tones tested varied as a function of frequency, pulse duration, number of pulses, and rise-decay times. Total signal duration (pulse plus interpulse intervals) was constant at 1.0 sec. The MAAs were between 2° and 3° for pure-tone pulses (80 pulses, 3 msec on, 10 msec off, 0.25-msec rise-decay time) from 20–100 kHz. [Supported by the Office of Naval Research.]
An Investigation of Underwater Sound Localization in Shallow and Noisy Water
Authors: Smith Paul F. , Yonovitz A. , Dering Gilbert H.
Published: October 2005

The ability of divers to discriminate the angular separation of two sound sources in the presence of high ambient noise in a reverberant environment was tested. In one experiment it was found that divers could not discriminate directionality with separations as large as 90° for low signal-to-noise ratios. However, when the signal to noise ratio was increased to about 19 dB, all four divers could discriminate a 30° angle with 100% accuracy. Two of the four divers could discriminate a 15° separation of sources. In a second experiment, a procedure was used in which the angular separation between sound sources could be continuously varied between about 35° and 1.4° in an “up and down” method. Six divers were tested but the data for one diver were uninterpretable. For the remaining five divers the underwater minimum audible angle for a 46-Hz band of noise centered at 1 kHz varied between 3.1° and 9.0° over all trials. Some suggestion that experience in underwater listening enhances localization skills appeared.
Vocal Responses of “CF-FM Bats” to Electric and Acoustic Stimuli
Authors: Suga N. , Simmons J. , Shimozawa T.
Published: October 2005

The experiments were performed for the exploration of coupling between the auditory and vocalization systems of awake bats, Pteronotus parnellii and P. suaprensis. The electric stimulation of the midbrain elicited the emission of sounds identical to species-specific orientation sounds consisting of CF and FM components. When acoustic stimuli were delivered with the electric stimuli, the electrically evoked sounds increased in amplitude and number. In P. parnellii these vocal responses were selectively sensitive to 62–63 kHz sounds, comparable to the CF component in its orientation sound, and also to downward-sweeping FM sounds comparable to the FM components in the orientation sound. In P. suaprensis, the threshold of the vocal response was high for pure tones and upward-sweeping FM sounds, but low for downward-sweeping FM sounds. The vocalization system was thus coupled with the auditory system by particular types of auditory neurons. These data indicate that the responses to acoustic stimuli were related not only to these power spectra, but also to properties of single auditory neurons and information-bearing elements in an emitted signal.
Directionality of an Echo-Locating System of an “FM Bat.”
Authors: Schuetze S. , Hendler P. , Suga N. , Shimozawa T.
Published: October 2005

In order to understand the directional sensitivity of an echolocating system, directionalities of both emitted sounds and ears should be measured. The skull of Myotis grisescens was rigidly held with a nail cemented to it. FM sounds nearly identical to the species-specific orientation sounds were elicited by electric stimuli applied to the midbrain. The fields of 55-, 75-, and 95-kHz components in the FM sounds were measured with microphones moved around the bat's head. The amplitudes of these components were half at 30°–40° lateral, 25°–50° down, and 5°–20° up from the eye-nostril line. As expected, the higher the frequency, the narrower was the sound field. Directionality of the auditory system in terms of lateral lemniscal evoked potential was also measured at 55, 75, and 95 kHz. From these measurements, the directional sensitivity of the echolocating system was obtained, which was sharper for higher frequencies, but the most sensitive direction moved toward the midline with an increase in frequency, so that the most important cue for echolocalization, the interaural intensity difference, became smaller for higher frequencies.
Binaural Detection in Forward and Backward Masking
Authors: Wightman Frederic
Published: October 2005

Detection thresholds for an octave-band filtered click, masked by a 500-msec burst of octave-band filtered noise were measured as a function of the time delay, Δt, between the onset of the click and the onset of the masker. Signal and masker filters were identical and centered at 500 Hz; signals were presented in both S0N0 and SπN0 phase configurations. The thresholds for the S0 click agree with classical data: backward masking was quite short-lived; thresholds for Δt=0 and Δt=250 msec were virtually the same (i.e., no overshoot was observed); forward masking decayed approximately linearly with logΔt, reaching asymptote after about 250 msec (Δt=750 msec). The behavior of the Sπ thresholds was quite different. For Δt=0, the threshold for a Sπ click was as much as 15 dB below that for Δt=250 msec. Thus, the MLD for conditions in which click and noise started simultaneously was nearly 30 dB. In forward masking (Δt>500 msec), the Sπ thresholds dropped much slower than the S0 thresholds; after 20 msec (Δt>520 msec) the Sπ and S0 thresholds were essentially equal. These differences in the time course of the masking function are interpreted as suggesting that the time constant of the detection system is much longer when Sπ signals are being detected. [Research supported in part by a National Science Foundation Postdoctoral Fellowship and by a grant from the National Institutes of Health, U. S. Public Health Service.]
Binaural Masking-Level Differences for Pulse Train Signals of Differing Correlation
Authors: Sorkin R. D. , Pohlmann L. D.
Published: October 2005

Three trained observers performed a two-interval forced-choice detection task; the noise level in the earphones was controlled by a PEST program. In all conditions, the noise was equal in amplitude but inverted in phase (Nπ) in the earphones. Signal conditions included Sm, S0, and partially correlated (ρ) or uncorrelated signal waveforms in the contralateral ear. The signal in the ipsilateral channel was always a random pulse train composed of 100-μsec pulses of random (+1 or −1) polarity, with an interpulse period of 2 msec. All signals were low-pass filtered at 5 kHz. The signal conditions included two ρ=0 conditions in which the contralateral signal waveform was identical to the ipsilateral except it (a) had (randomly) half of the pulses reversed in polarity, or (b) was a deterministic signal composed of pulses of all negative polarity; and two ρ=0.5 conditions in which the contralateral signal (c) had (randomly) half of the pulses deleted, or (d) was composed of only the negative polarity pulses in the signal. We also ran ρ=0.75 versions of (a) and (c), and reverse polarity versions of (b) and (d). All the partial and zero ρ conditions resulted in psychometric functions which fell in the 6-dB range between those of NπSm and NπS0. The worst performance was obtained for condition (d).
Temporal Binaural Masking for Homophasic and Antiphasic Conditions
Authors: Osman E.
Published: October 2005

Psychometric functions were obtained for a 10-msec 400-Hz tonal signal presented binaurally with a 125-msec broad-band noise burst, for homophasic (N0S0) and antiphasic (N0Sπ) conditions, and for varying temporal relations of signal and noise. The signal was presented before, during, or after the noise burst. Signal and noise were also gated on and off together for 10 or 125 msec. For simultaneous masking, detection of the signal was enhanced when its termination coincided with the termination of the noise burst, as compared to when the onsets or middurations of the two were coincident. This was found for N0S0 and N0Sπ. The magnitudes of the BMLDs for these temporal positions varied for different subjects. A large BMLD was found for signal onset following noise termination by 2 msec. When the signal ended 2 msec before noise onset, a small BMLD was found. The shapes of the psychometric functions are compared for N0S0 and N0Sπ, for the different temporal relations studied. Results are related to theoretical mechanisms of binaural analysis and temporal effects in masking.
Interaural Time Delay of Speech with Coherent and Incoherent Noise
Authors: Gerber Sanford E. , Milner Paul
Published: October 2005

The matter of the “magic number” for interaural delay has occupied the attention of numerous investigators for the past few years. Some have reported that delays around 50 msec between ears produce effects different from those found when the delays are some other value; others have reported the same phenomenon but with the “magic number” set around 60 msec. Also, there has been some question about the delay of the masker, i.e., whether the masking noise is also delayed may be of critical interest. In the study reported here, a panel of listeners heard speech delayed interaurally at all multiples of 25 and 30 msec up to a maximum of 150 msec. Furthermore, each amount of delay was heard with the masker delayed also (incoherent noise) and with the masker not delayed at all (coherent noise). Data are reported showing the effects of the amount of delay, the effects of two possible delay constants, and the effects of noise coherence.
Lateralization Threshold of Signals in Noise
Authors: Wilbanks W. A. , Elliott Thomas F. , Pate Michael W.
Published: October 2005

The minimal interaural time difference discriminable as a shift in the location of the sound image was determined for tonal signals in the presence of continuous noise. These lateralization thresholds were measured by means of a forced-choice method. Data will be presented showing how the accuracy of lateralization varies as a function of the frequency of the signal, signal-to-noise ratio, and the interaural correlation of the noise. Possible relationships between the results of this study and some phenomena of binaural masking will be discussed. [Work supported by National Science Foundation.]
Lateralization and the MLD: The Vector Summation Model
Authors: Gilliom J. D. , Hamm P. K.
Published: October 2005

At the Houston meeting, the first author reported the results of an experiment requiring discrimination between various combinations of interaural time and intensity cues. Performance could be described quite well by a simple vector summation of performances with each cue alone. The angle, η, between the vectors provided an estimate of the correlation of the two cue dimensions in the psychological discrimination space. Similar data reported recently by Jeffress and McFadden [J. Acoust. Soc. Amer. 52, 1197–1206 (1972); 50, 1484–1493 (1971)] provided the opportunity to extend the vector summation model into an MLD paradigm. As in our earlier work, one angle, η, along with an efficiency ratio, reflecting the relative sensitivity of each observer to interaural time and intensity differences, predicted performance well for all values of alpha; that is, performance for cue conditions in which time and intensity differences are consonant or dissonant can be described by the same vector summation process. An analysis of changes in efficiency ratio and η across observers and frequency, along with a comparison of these parameters in the detection and lateralization tasks will be presented.
Comparisons of Binaural-Interaction Models
Authors: Colburn H. S.
Published: October 2005

Comparisons of binaural-interaction models that have been proposed are made on the basis of overall structure, internal-noise postulates, areas of applicability, ability to describe and interrelate experimental results, and consistency with physiological data. The models considered include count-comparison models, phase-detector and lateralization models, the equalization-cancellation model, various correlation models, and a model based on auditory-nerve data. An attempt is made to describe the major problems which cannot be treated satisfactorily in terms of any model now available.
Flute Headjoint Cork Position and Damping of Higher Modes
Authors: Benade A. H.
Published: October 2005

The effect of embouchure hole size and cork position on tuning of the flute normal modes (input admittance maxima) is well known (Benade and French, 1964). The observation that the “stuffiness” of the higher played modes rises sharply as the cork is pulled is familar to flutists. The fact that perceived “stuffiness” correlates well with the physical damping parameter tanhαL suggests examination of the real part of the admittance function. For a tube of radius α and length L jointed to a head cavity of length r at an embouchure hole of radius b and length t, the imaginary part of the admittance has maxima at frequencies that are near fn = nc/2[L+r+(a/b)2t]. However the damping (contributed mainly by the pipe) has a maximum near fm = (2m−1)(c/4L). Going up the series, the fn's precess toward some fm. We find the damping (and stuffiness) rises rapidly, the resonances being completely obliterated in the immediate neighborhood of fm. The resonances reappear beyond fm but are not usually playable for other reasons. The highest notes attainable on any flute vary with the cork position in agreement with predictions based on measured and/or calculated admittance values at the embouchure hole. Also, a suitably chosen cork position will slightly raise the admittance of the first few flute resonances above simple pipe values, improving the response of the instrument.
Similarities an d Differences Between Cavity Resonators and Organ Pipes
Authors: Elder Samuel A.
Published: October 2005

Flow-excited cavity resonance is often referred to as “organ pipe” oscillation. Similarities and differences are noted between the operation of a typical cavity resonator and the behavior of an actual organ pipe. Variations in spectra are traced to differences in the driving mechanism, it being shown that the richness of organ pipe sound is related to the dynamics of thin jets. The effects of instability and entrained flow are discussed. An assessment is made of the relative importance of feedback, in each case, from the fluctuating lift force at the trailing edge and from the standing wave field of the resonator. [Work supported by GHR contract from the Naval Ship Research and Development Center.]
Viola Tone Quality Study Using an Instrument with Synthesized Normal Modes
Authors: Gorrill W. S.
Published: October 2005

A viola with electronically synthesized normal body modes provides a flexible means for getting correlation between the subjective observations of tone quality and the known physical characteristics responsible for producing the tone. The Q and the amplitudes of all the normal modes are easily adjustable. It is desirable to have the frequency of a few of the modes variable over a small range. The viola used is somewhat similar to the violins described by Max Mathews and is a development from the instrument demonstrated in 1972. Both laboratory studies and the subjective results of extensive playing in chamber music groups will be described. The instrument will be demonstrated with settings to produce tone qualities typical of several different types of violas.
Two Violins Compared by Acoustical, Optical, and Musical Methods
Authors: Hutchins Carleen M. , Stetson Karl A. , Taylor Pauline A.
Published: October 2005

One violin constructed according to the best available violin-making practice, with its free plates tuned as well as possible by the Hutchins plate tuning method, is compared with another violin made to incorporate two intentional variations from such an optimum: the upper area of the black plate left 1.5 mm too thick, and the bass bar left about 4 mm too high. Two series of photographs made by hologram interferometry, one showing the vibration modes of the free plates before assembly, and another of the completed violins in playing condition displaying simultaneous modes in top, back, and sides will be discussed in comparison with response curves and musical performance.
Consonant Durations in a Cluster Environment
Authors: O'Shaughnessy Douglas
Published: October 2005

The durations of initial and final consonant clusters in monosyllabic words within a frame sentence were studied from spectrograms of reading by speakers. The durations of consonants within a cluster varied with the features of the consonant and its phonetic environment, such as voicing and point and manner of articulation. A durational model was proposed based on two mechanisms: an articulatory mechanism was responsible for effects involving motion of the tongue and lips during a cluster. Shortening of consonants in clusters was attributed to the shorter distances that the articulators travel in clusters. Lengthening of consonants before fricatives and aspiration duration effects were part of this articulatory mechanism. The other factor was a “learned” mechanism, related to the use of duration as an acoustic cue in consonant perception. Phonetically similar single consonants had substantial durational differences which could aid in distinguishing them. Phonological restrictions, which do not occur in VCV contexts, arise in clusters; and when adjacent consonants provided information about another consonant's features, durational differences were less. [This research supported by an NSF Fellowship.]
Some Within- and Between-Word Timing Effects
Authors: Huggins A. W. F.
Published: October 2005

Sixteen “sentences” were composed by the binary inclusion or exclusion of each of the four unstressed (and parenthesized) syllables in the sentence: “Cheese(s) (a)bound(ed) (ab)out.” Each sentence was read five times, in an irregular order, by two speakers, and segment durations were measured from spectrograms. Among other things, the results show (1) that the duration of a stressed syllable is reduced by the addition, within the same word, of either a preceding or a following unstressed syllable, or both; and (2) that the duration of a stressed syllable may be affected by the addition of an unstressed syllable to other words, either preceding or following it in the phrase. Since the effects depend upon where the unstressed syllable is added, the latter result cannot be explained simply as proportional shortening tending to maintain total utterance duration. [This research was supported by NIH Grant No. NS 04332.]
Vowel Duration as a Function of the Syllabic Structure of a Word
Authors: Klatt Dennis H.
Published: October 2005

A list of English words was recorded in the frame sentence “Say… instead” by three talkers. The list consisted of a family of words and two-word phrases which were constructed from a set of single-syllable base words (e.g., “lock”) by appending unstressed syllables to either the front or back of the base word (e.g., “unlock,” “the lock,” “locking,” “locker”). Segmental durations of these materials were measured from spectrograms. Results indicate that the stressed vowel in the base syllable of a two-syllable word is shortened by, 25% (relative to its duration in the single-syllable word) if followed by an unstressed syllable in the same word, but is not shortened if preceded by an unstressed syllable in the same word. These results indicate either that English behaves in a way similar to Swedish in that syllables in a word are shortened as a function of the number of syllables remaining to be produced in that word, or that syllable shortening in multisyllabic words is related to the realization of other aspects of the utterance. For example; it was observed that the primary stress, which always fell on the base word in our data, is accompanied by a rise-fall fundamental frequency. This contour was realized within the stressed syllable and any following unstressed syllable of the same word. Perhaps the base word must be long in duration if the fundamental frequency contour can only be realized on a single syllable, and it is shorter if the falling contour can be extended to a following unstressed syllable of the same word. [Work supported in part by the National Institute of Health and the Office of Naval Research.]