bluuebunny/crossref_metadata_2025 huggingface dataset:
Wave-vector filter measurements of the nearfield pressure produced by a vibrating plate
Authors: Strawderman Wayne A.
Published: October 2005

The wave vector, or Fourier conjugate of the spatial separation variable, is commonly used by theoreticians in the solution and interpretation of spatially distributed dynamic systems. Previous experiments utilizing one-dimensional wave-vector filters to directly measure the wavenumber-frequency characteristics of turbulent wall-pressure fluctuations have demonstrated that wave-vector-frequency measurements are a feasible alternate, and a complement, to the measurement of the more common space-time characteristics. This paper describes the measurement of the two-dimensional wave-vector characteristics of the nearfield pressure produced by a vibrating plate. A two-dimensional, 24-element planar array, arranged in a 6×4 dement grid, was utilized as the wave-vector filter. Examples of measured wave-vector spectra are presented. The measured wavevector characteristics of the nearfield pressure are consistent with theoretical predictions.
Experimental system for teleconferencing
Authors: West J. E. , MacLean D. J. , Flanagan J. L.
Published: October 2005

Travel to distant conferences conferences consumes time, energy, and hemfin resources. An alternative, should it prove an adequate replacement for face-to-face communication, is “teleconferencing.” While the necessary and desirable elements of conferenoing by telephone are yet to be quantified, we describe a rudimentary system which provides a communication versatility that belies its simplicity. We establish a dedicated four-wire connection between distant conference rooms. Conventional speakerphones (Model 4A) are modified by disconnecting the electronic hybrid to give separate transmit (microphone) and receive (loudspeaker) leads. We give individual instruments to each conferee. In a given conference room, we connect (sum) all individual transmit leads to the transmit path of the four-wire. Similarly, we connect (distribute) all individual receive leads from the receive path of the four-wire. The voice switch and the noise gain-adjusting device (NOGAD) are left intact in each individual speakerphone. Complementary connections are made in the other conference room. As a result of these features, the teleconferencing system is able to cope well with room reverberation and background noise, both at transmit and receive locations, and to provide effective and economical conferencing.
Method for measuring noise-canceling microphone characteristics
Authors: Mozo B. T. , Patterson J. H. , Camp R. T.
Published: October 2005

A test method was developed using fast Fourier techniques to determine noise-canceling microphone nearfield frequency response and noise-canceling properties. This procedure utilizes transfer function analysis between a standard laboratory microphone and a noise canceling microphone. The standard microphone and the noise canceling microphone are mechanically coupled and move thru a pink noise sound field produced by an artificial voice. Transfer functions were measured at a point defined as the source of the sound field which yield the nearfield response and at fixed distances from that point which yield measures of noise canceling properties. A mapping of the sound field was completed by moving two standard laboratory microphones thru the sound field and measuring their transfer functions.
Fluctuating surface pressure measurements on USB wing:Two types of transducers
Authors: Reed James B. , Schoenster James A.
Published: October 2005

Measurement of noise levels in the vicinity of a jet exhaust, such as is desired on surfaces of propulsive lift devices, requires high-temperature transducers that will withstand intense low-frequency flow buffeting, yet will be sensitive to lower-pressure loads at high frequencies. For flight test installation in an aircraft assembly. such as the USB YC-14 prototype, rugged transducer integrity and attachment are critical for periodic installation and removal, and for flight exposure occurring in a test program of several months duration. Two transducer types used in such previous test environments were Kulite and Photocon units. These two transducers systems were compared by pairing three wing surface locations in a NASA Langley JT15D upper surface blowing (USB) static test. This USB static arrangement was an inverted simulation of the overwing section of a small airplane used in Langley wind tunnel USB experiments. In addition to transducer comparisons, the experience of measuring full-scale surface-fluctuating pressures of attached flow and jet interaction noise in the nearfield was desired. Some of the noise sources and characteristics of overwing nozzles and Jet interaction are discussed.
Measurement of acoustic impedance via reflected pulse
Authors: Chalupnik J. D. , Clark C. D.
Published: October 2005

This paper explores a technique for using a reflected acoustic pulse to determine acoustic impedance. A Fourier transform of the acoustic wave reflected by the test material is used to find the reflection coefficient as a function of frequency. An expression is derived which relates the reflection coefficient to the acoustic impedance of the sample. Experiments were performed to test the procedure on samples of typical acoustical materials. The acoustic pulse was generated by an electrical spark. The effect of the sphericity of the wavefront is taken into account in the derivation. The pulse was repeatable at most of the audible frequencies, for which the results of this study agree with values obtained by classical impedance tube techniques; however, at low frequencies, results were at variance with the impedance tube results. This is attributable to the low energy density in the test pulse at low frequencies. The pulse technique is unuseable.
Comparison of the reliability of an auraldome headset and a standard headset in obtaining pure-tone audiometric threshold measures
Authors: Vogen C. S. , Copeland A. B.
Published: October 2005

An Auraldome headset and standard headset (TDH-39/MX-41/AR) were compared via threshold measures, probe tube microphony, and coupler measures. The same TDH-39 receiver was used in each headset. Statistical and descriptive comparisons of headset performance were carried out. For each frequency, intersubject variances of the two headsets were compared statistically, as were their intersubject variances. Descriptive comparisons were also made of the probe tube and coupler data. The threshold data and physical data, considered together, permit comparison of the reliability of the two headsets, as well as allowing comparisons of their real-ear-to-coupler errors. Results demonstrated the suitability of the Auraldome headset for audiometric purposes and of the probe tube mierophone methodology used in this study for transferring threshold standards from conventional to unconventional earphones.
On the sampling and modeling of urban noise
Authors: Porter Michael A.
Published: October 2005

This paper summarizes the methodology and results of an environmental noise survey of the City of San Francisco. The noise survey program and the urban-noise-modeling technique derived from the survey program offered a combination of ease of use and accuracy not previously available to the urban planner. The noise survey portion of the study was designed to provide statistically reliable data on the emission levels of the major urban noise sources (the various surface transportation elements). The measurement locations and procedures were chosen to best sample the range of these sources rather than the typical geographic sampling technique most often proposed for urban noise studies. The model developed to predict noise levels at a specific point in the city uses source emission level information to account for both the immediate nearby sources and for the distant or background sources typical to most urban environments. The model provides an accuracy of ± 2.5 dBA on the Ldn at a given point in the city.
Community noise survey technique for large cities
Authors: Harris B. , Grantner L.
Published: October 2005

Land use planning and environmental impact statements require as a data base knowledge of the current community noise levels. However, many community noise evaluation methods for large cities are not survey techniquesper se, but rather data collection and processing for specific typical sites. Such techniques, because they are data intensive, do not permit the efficient survey of large areas. The technique presented here, which was successfully used over a 40-square-mile area of New York City, comprised sampling in time at fixed stations and in space by the use of mobile stations. The results were 5-min histogram data that could be accurately interpolated in time and space to give one-hour dBA percentiles over the area with only a few hundred hours of data. This presentation describes the statistical basis of the survey procedure in terms of the joint space-time probability distributions, the necessity for the histogram approach in the interpolation procedures, and the final attainment of noise contour levels for the city.
Meaningful and accurate noise contours cannot be obtained
Authors: Benjamin Kaplan O.
Published: October 2005

The formula E(S) ≃ ± E(L)% describes the relationship between the percent error of S (the distance between a contour and noise source) and the dBA error of L (the estimated sound pressure level). E stands for error, or tolerance, or standard deviation. The constant k assumes the values of 11.5, 17, and 23, for respectively point, intermediate, and linear noise sources. The formula is more and more exact as E(S) approaches zero. The formula can be used to prove that noise contours are not really contours, but bands with minimal amplitudes equal to ± 0.115 S (point source condition), ± 0.17 S (intermediate condition), or ± 0.23 S (linear source condition). This poses certain problems that call for modifications in the California noise laws.
Connection between the solutions of the Helmholtz and parabolic equations for sound propagation
Authors: DeSanto John A.
Published: October 2005

Using a conformal mapping technique in a rectangular waveguide, we present an exact integral relation between the solutions of the Helmholtz equation ψ whose sound speed c(x,y) is a function of depth y and range x, and the solution p of a parabolic equation whose sound speed varies in the mapped depth coordinate and which satisfies a related boundary-value problem. The integral relation is a Fourier-Mellin-inversion transformation of p. The conformal transformation interrelates the sound speeds of the two equations. Examples presented include the special case of Polyanskii when c(x,y)=c(y), and profiles which are sinusoidal in depth and exponentially decrease to a constant in range. The exact relation between ψ and p also clarifies the use of p as an approximation to the asymptotic value of ψ. For example, in multimode propagation problems in cylinderical coordinates, p preserves at most one modal amplitude of the asymptotic value of ψ, the others being scaled by an eigenfrequency dependent factor. Also, because of the exact relation between ψ and p, their respective incident fields cannot be independently chosen.
Range-dependent ray-tracing model with a linear, hyperbolic cosine composite vertical profile
Authors: Weinberg N. L. , Zabalgogeazcoa X.
Published: October 2005

A model based on the four section composite vertical profile reported in a previous paper [Weinberg, Clark, and Flanagan, J. Acoust. Soc. Am. 56, 447–458 (1974)] has been extended to include the effect of profile range dependency. The technique is computer efficient as each segment can be represented by a closed form solution. The parameters of interest, such as travel time, boundary losses, boundary phase shifts, and spreading losses, are included so that amplitude and phase of each arrival can be determined. By the use of variable range steps, the desired degree of accuracy can be achieved. This method is proving useful in the study of underwater acoustic propagation under the influence of ocean dynamic processes which exhibit range-dependent behavior.
Applicability of the parabolic equation method (PE method) to sound propagation modeling
Authors: Jensen F. B.
Published: October 2005

The PE method is the latest major development in the field of acoustic modeling [S.T. McDaniel, J. Acoust. Soc. Am. 57, 307–311 (1975); R.M. Fitzgerald, J. Acoust. Soc. Am. 57, 839–842 (1975)]. In the investigation described here, the PE method has been applied to various propagation cases in both deep and shallow water, with and without bottom absorption, and for both range-dependent and range-independent environmental conditions. It is shown that the predicted transmission losses provided by the PE program are in close agreement with beth experimental data and with predictions obtained from mode and ray programs. However, there are practical limitations to the PE method. For instance, as shown theoretically and confirmed by the numerical tests, the method cannot handle steep sound speed gradients without running into excessive calculation times. This means that in practice the PE method is inapplicable to acoustic modeling in shallow water, where there are big changes in sound speed at the water/bottom interface. The major advantage of the method is the simple way in which it handles range dependence. We may therefore conclude that the PE method is primarily applicable to low-frequency acoustic modeling in deep water (no bottom effects) under range-dependent environmental conditions.
Airborne noise requirements in building codes
Authors: Schultz Theodore J.
Published: October 2005

The business of devising a rating for airborne sound attenuation of building elements that closely corresponds to people's subjective evaluation of the acoustical performance, and of devising test methods for demonstrating satisfactory acoustical performance in a finished building, is very complicated indeed. Until recently, the tendency has been toward increasing complication of the test and rating procedure, in an attempt to make the single procedure account in near-complete detail for the complexities of human response to noise. The result has been a test procedure that is too complicated for routine use in evaluating finished buildings. An alternative approach would be to divide the complicated task into a number of simpler tasks, each of which can be handled by relatively untrained staff, and to assure by regulation that all of the simpler tasks will be done for each evaluation. This paper proposes an example of the latter approach.
Assisted resonance in the United States
Authors: Berry G.
Published: October 2005

The paper describes Assisted Resonance installations designed to provide variable reverberation time in two different types of situation. The first, in Concord, California, consists of a 3500-seat pavilion with a backstage wall, but with the other sides open. The acoustic consultant has designed this space to provide ideal acoustic conditions for a full range of activity from drama through to symphonic presentations. The second, in Scottsdale, Arizona, is a smaller enclosed space but once again it is designed to accomodate a wide range of presentations. At the time of writing the abstract, the Concord installation is completed and believed to be successful but the Scottsdale installation is still under way and the outcome will not be clear until the paper is presented. The paper does not discuss the overall acoustic design, but is restricted to the role of Assisted Resonance, which deals with only one parameter, and the potential and limitations of the technique will be discussed.
Amplitude fluctuations of sound transmitted through Gulf Stream rings
Authors: Porter R. P. , Spindel R. C. , Baxter L.
Published: October 2005

During September 1974, ten neutrally buoyant floats were placed in two of the cold eddies spun off from the Gulf Stream. The floats, designed to be neutrally buoyant near the axis of the sound channel, transmit an 80-sec continuous tone with a signal level of 176 dB re 1 μPa every 6 h. These floats have been tracked by the U.S. Naval Oceanographic Office and the positions determined once per day. Since December we have been monitoring the amplitude of the signal at Bermuda on a digital data logger. Approximately, three months of continuous data have been collected and processed. A 15-dB processing gain is achieved by performing a running average over the raw digital data. The data exhibit quasiperiodic variations with a time scale of 300 h. Temperature profiles, collected along the paths between some of the floats and Bermuda, have been used in calculations of transmission loss. Correlations between measured and calculated signal levels are discussed.
Omnidirectional and beam statistics of the temporal fluctuations in low-frequency CW tones
Authors: Heitmeyer R. M. , Deihl D. T.
Published: October 2005

The temporal fluctuations of CW receptions at frequencies of 16, 29.5, and 115 Hz on a horizontal array have been analyzed to obtain a statistical description of the instantaneous power at both the individual hydrophone outputs and a frequency domain beamformer output. The signals were transmitted from a towed source and received en a ten-hydrophone bottom-mounted array in deep water. Fine resolution spectra of the individual hydrophone signals are compared with the beam-former frequency-azimuth spectra. Time series of instantaneous signal power from the individual hydrophones as well as the beamformer are obtained as the outputs of both wideband filters (350 mHz) and narrowband filters (50 mHz) which are realized using FFT processing techniques. The narrow bandwidth is selected to pass only the slow signal fluctuations and to reject the fast variations which appear in the spectra as the surface interaction sidebands, whereas the wide bandwidth is selected to pass the total signal. Noise power in comparable bandwidths is also measured to generate time series of signal-to-noise and array gain. Estimates of probability density functions, fluctuation spectra, and correlation functions are presented for selected time series and comparisons are made between the hydrophone and the beam statistics for the different transmitted frequencies.
Photoacoustic spectroscopy of solids
Authors: Rosencwaig Allan
Published: October 2005

There are a great many substances, both organic and inorganic, that, because of their physical state, cannot be readily studied by conventional absorption or reflection techniques. In photoacoustic spectroscopy, light absorbed by the sample is converted into a measurable acoustic signal, and spectra closely corresponding to optical absorption spectra, can be obtained on any type of solid or semisolid material, whether it be crystalline, powder, smear, gel, etc. The physical processes underlying the photoacoustic effect in solids, and a mathematical analysis of the production of the photoacoustic signal will be presented. The methodology of photoacoustic spectroscopy will also be described. Various applications of this technique in the fields of physics, chemistry, biology, and medicine will be discussed. These applications will be treated under three main headings: bulk studies, surface studies, and de-excitation studies.
Photophone—an optical telephone receiver
Authors: Kleinman D. A. , Nelson D. F.
Published: October 2005

The photophone (name originally given by A.G. Bell) is an optical telephone receiver based on the optoacoustic effect. J. Acoust. Soc. Am., Vol. 58, Suppl. No. 1, Fall 1975.S53 90th Meeting: Theory will be given for a modern version intended to operate on a few mW of power modulated light from an optical fiber. The receiver consists of an absorption cell, a response-equalizing device such as an air column or diaphragm, a tapered acoustic tube acting as a transformer, and an earpiece similar to a conventional telephone earpiece including a response equalizing device. In an alternative version, the cell is directly coupled to the earpiece without a transformer. The photophone gives a flat (3 dB) response to power modulated light over the telephone voice band 300–3300 Hz. The optoacoustic effect is treated theoretically on the basis of a pseudogas model for solid absorbers in the form of a fine mesh. We find that the sensitivity (the optical power modulation required to produce a sound pressure level SPL =81 dB in a closed volume of 6 cm3, representing the earpiece pressed against the ear) is 2.8, 0.9 and 51 mW for the air-filled, xenon-filled, and direct-coupled photophones, respectively.
Wavefront reconstruction using the adage system
Authors: Cohen R. , Eschenberg K. , Watson E. E.
Published: October 2005

Sound sources from vibrating objects are detected by holographic means using a microphone scanner and associated photographic equipment. Reconstructions are done digitally on an IBM 370/165 computer and displayed on an adage graphics terminal. The adage system is a hybrid computer with a cathode-ray tube display device. Conjugate image reconstructions are presented in three dimensions with relief representing the relative strength of far field sound sources. Program support allows the image to be scaled, translated, and rotated in real time. In addition, background noise can be eliminated from the image by removing all parts of the display below a prescribed cutoff level. With this holographic system, wavefront reconstructions have been made of circular, elliptical and ribbed rectangular plates as well as other objects. In all cases, the plates were driven below their coincidence frequency. Farfield sound sources, which were not evident in nearfield scan displays have been found readily visible in these holographic images.
Optoacoustic transducer for high-sensitivity orthographic imaging
Authors: Wang K. Y. , Shen H. , Wade G.
Published: October 2005

A possible embodiment of a real-time, orthorgraphic, acoustic imaging system with piezoelectric detection which employs an electronically controlled, scanning, focused beam to probe the target has been described. {K. Wang and G. Wade, Proc. IEEE 62, (2), 650–651 (1974)]. The key element of this embodiment is an optoacoustic transducer (OAT) addressed by diffracted light emerging from a properly designed Fourier-transform hologram. The OAT under study is a sandwich structure containing a piezoelectric layer which is differentially activated from point to point across its surface via photoconductive switching. To be useful in the final system, the OAT has to be 20 cm × 20 cm in dimension and has to meet certain key requirements. For example, real-time capability requires that the photo-conductive relaxation time be no longer than 1 μsec. In addition it is essential that when a zone-plate pattern of light is incident on the OAT, the output acoustic amplitude distribution be such that the generated ultrasound converge to a good focus through propagation. Preliminary considerations and simulations on various OAT structures show that certain available materials may have the proper characteristics to produce good focusing. [Work supported by NSF ENG 74 22184.]
Frequency bands for binaural interactions
Authors: Neutzel J. , Hafter E.
Published: October 2005

Earlier reports [Henning, J. Acoust. Soc. Am. 55, 84–90 (1974); Neutzel and Hafter, J. Acoust. Soc. Am. 56, S56(A) (1975)] have shown that temporal information derived from the envelopes of amplitude-modulated signals can be used for lateralization even when the carrier frequencies are interaurally discrepant. This study was intended to examine that fact more closely. Lateralization was measured using digitally generated bursts of amplitude-modulated tones of duration 200 msec and amplitude 40 dB SPL. A 50–1600 Hz bandpass noise with an overall level of 60 dB SPL masked low-frequency distortions, while the signals themselves were filtered at 200 Hz below and above the lower and upper sidebands. The digital methods employed allowed arbitrary choices of frequency and phase relations; consequently, inharmonic frequency relations between the modulation and carrier frequencies could be used. These were valuable, both as a precaution against monaural cues and as a means of assessing the degree of interaural spectral overlap necessary for lateralization. We find that the binaural overlap bands obtained with anharmonic stimuli are narrower than those reported for carriers which are multiples of the modulator. However, lateralization with interaurally indentical carriers is excellent, regardless of whether the carrier to modulation relation is harmonic.
Effects of vocalic context, place of articulation, and speaker's sex on “voiced” stop consonant production
Authors: Smith B. L.
Published: October 2005

A study was made of voice-onset time in initial “voiced” (prevoiced and short lag) labial, dental and velar stop consonants. Ten male and ten female English-speaking subjects produced isolated CVC syllables. Prevoicing occurred more frequently and was greater in duration before high vowels than before low vowels. Prevoicing also occurred more frequently and was of greater duration for labial and dental stops than for velar stops. Short-lag duration was greater preceding high vowels than preceding low vowels, but short-lag stops occurred more frequently in the context of low vowels than in the context of high vowels. Short-lag productions were of greater duration and occurred more frequently in the case of velar stops than in the case of dental and labial productions. Male subjects exhibited both more frequent occurrence of voicing lead and greater prevoicing duration than females. Although females employed the short-lag mode more frequently than males did, female speakers were found to produce shorter voicing lags than males.
Comparative performance study of several pitch detection algorithms
Authors: Cheng M. J. , Rabiner L. R. , Rosenberg A. E. , McGonegal C. A.
Published: October 2005

A comparative performance study of five pitch detection algorithms was conducted. A speech data base, consisting of eight utterances spoken by three males, three females, and one child was constructed. Both telephone and wideband recordings were made of each of the utterances. For each of the utterances in the data base a “standard” pitch contour was semiautomatically measured using a highly sophistocated interactive pitch detection program. The “standard” pitch contour was then compared with the pitch contour that was obtained from each of the five programmed pitch detectors. The algorithms used in this study were (1) a center clipping, infinite-peak clipping, modified autocorrelation method; (2) the cepstral method; (3) the SIFT method; (4) the parallel processing time domain method; and (5) the data reduction method. A set of measurements were made on the pitch contours to quantify the various types of errors which occur in each of the above methods. Included among the error measurements were the average and standard deviation of the error in pitch period during voiced regions, the number of gross errors in the pitch period, and the average and standard deviation of the error in choosing onset and offset of voicing. By pooling the various error measurements, the individual pitch detectors could be rank ordered as a measure of this relative performance.
Dichotic “masking” of voice-onset time
Authors: Repp B. H.
Published: October 2005

When consonant (CV) “targets” are presented to one ear and isolated vowel “masks” to the other ear, the perception of the voicing feature of the target is biased by the temporal relationship between target and mask. This onset asynchrony acts like a “pseudo voice-onset time” and completes with the actual voice-onset time of the target. The effect is especially strong when there is high uncertainty about the voicing category of the target, and it is more pronounced when the masking vowel lags behind than when it leads in time. The bias is strongest when the masking vowel is the same as the vowel of the target, but it is present with a different vowel mask as well. There tends to be a right-ear advantage that is strongest when the masking vowel leads in time. The fundamental frequency at masking vowel onset has an additional influence: the lower it is, the stronger is the tendency to identify the target as voiced. These experiments demonstrate that a temporal relationship (voice-onset time) may be masked by another temporal relationship, that the voice-onset time across ears may act as a phonetic cue, and that the effect is more complex than a simple weighted-averaging model would predict.
Effect of varied response mode upon reaction times in a dichotic target-challenge paradigm
Authors: Hughes Larry F. , Harston Carol M.
Published: October 2005

Reaction times for two right handed subjects who received 6000 trials of dichotic CVs were obtained under response conditions involving the left hand, right hand, and axial movements [Geschwind, Am. Sci. 63 (2), 188–195 (1975)]. The simultaneous CVs were presented in a target-challenge paradigm (Berlin et al., IEEE Trans. Audio Electroacoust. AU-21, 189 (1973)]. The results indicated that when the targets were presented to the right ear, the latencies were shorter (150+ msec) than when presented to the left ear. The large and stable differences in latencies remained invariant with the mode of response. Phonetic effects involving manner and place of articulation will be discussed. The relationship between the obtained measures of latency and percent correct are also included. [Supported by National Institutes of Health, USPHS Grant No. NH 11647-02.]
Dichotic listening in musically naive and musically trained subjects
Authors: Van Lancker Diana , Fromkin Victoria A.
Published: October 2005

The results of a dichotic listening study reported on at the 84th ASA meeting by Van Lancker and Fromkin [J. Acoust. Soc. Am. 53, 343(A) (1973)] showed a right ear advantage for Thai words differing only in tone and only in initial consonant, but not for hummed pitches, when native Thai speakers served as subjects. Subsequent analysis of individual POE scores yielded significant correlations between performances on consonant-words and tone-words, and between tone-words and hums, suggesting that two modes of processing may have been used by Thai subjects. Preliminary testing of American English speakers resulted in no ear advantage in any of the three conditions. A study by Bever (1974) on nonlinguistic (musical) stimuli supports the hypothesis that hemispheric specialization is correlated with different processing modes, not with stimulus type, This paper reports on a study aimed at further investigation of this hypothesis. Two groups of American English speakers, musically naive and musically trained, were tested to determine whether ear advantages for recognizing tones (on Thai words) and pitches (on hums) differ between them. Implications for hemispheric specialization for linguistic and other acoustic stimuli will be discussed.
Electroacoustic modeling of magnetostrictive shells and rings
Authors: Hanish S. , King B. J. , Baier R. V. , Rogers P. H.
Published: October 2005

The mathematical model of the electroacoustic performance of a force-driven free-flooding magnetostrictive cylinder shell used as an underwater sound transducer, originally presented at the 86th Meeting of the Acoustical Society of America in 1973 [J. Acoust. Soc. Am. 55, 471(A) (1974)] and later developed in NRL Report 7767 (Dec. 1974), has been coded into a computer program called EIGSHIP. This program is designed to predict electrical and mechanical impedances of the loaded shell, transmitting responses, electroacoustic efficiency, surface velocities, farfield beam patterns, and other relevant performance parameters. An experimental check on the predictive capabilities of EIGSHIP was undertaken using three specially constructed magnetostrictive shells which were tested under water load conditions in an indoor test facility. A discussion of the comparison of predicted and measured performance is presented. Also samples of typical computer runs are displayed and commented on.
Measurements of lining impedance in a duct with flow
Authors: Bauer A. B.
Published: October 2005

The two-microphone technique has been used to obtain data from which values of acoustic impedance have been calculated for perforated-sheet linings. The magnitude and argument of the impedance values have been plotted as a function of frequency, wave amplitude, and duct Mach number for five different lining geometries. These results are compared to theoretical values of lining impedance obtained from a set of equations based on past work [E.J. Rice and C.E. Feiler, NASA TN D-6178 (1971); E.J. Rice, AIAA Paper 73–995 (1973); W.A. Bell, B.R. Daniel, and B.T. Zinn, AIAA Paper 74–61 (1974); A.S. Hersh and T. Rogers, AIAA Paper 75–495 (1975)]. Although the agreement between experiment and theory is good in general, it is not good in detail because the complicated flow field near the orifices is not fully understood nor completely modeled by the theory. Suggestions for possible improvements to the impedance model are discussed.
Experimental investigation of perforated- and woven-fiberglass duct linings
Authors: Marsh Alan H.
Published: October 2005

Acoustically absorptive duct linings for jet engine nacelles usually consists of a single porous layer backed by air-filled cavities, suitably partitioned, and an impervious backing sheet, Two common types of porous surfaces are perforated sheets and resin-impregnated woven-fiberglass sheets. The depth of the backing cavities and the size of the core-support structures are chosen to give the desired acoustical reactance yielding maximum attenuation at a frequency corresponding to maximum perceived noisiness at a given operating condition. The presumed advantage of the distributed-resistance design is wider attenuation bandwidth. Tests were conducted with four different porous surface designs successively installed in the fan-discharge ducts of a full-scale turbofan engine. There were two different perforated aluminum surfaces (nominal open areas of 8.6% and 15%) and two different fiberglass surfaces (nominal flow resistances of 70 and 150 mks rayls). The results of the tests indicated that the distributed-resistance linings had a wider attenuation bandwidth than the discrete-resistance (perforated) linings. Maximum attenuation values were nearly the same.
Magnetostrictive ring transducer technology manual
Authors: Sinsky J. A. , Baier R. V. , King B. J.
Published: October 2005

A comprehensive magnetostrictive ring transducer designer's manual will be described. The manual consists of three main parts: (1) the highlights of the history of magnetostrictive ring transducer research and development, (2) parameter studies of a large-flow-frequency ring transducer and a discussion of related design considerations, and (3) an extensive bibliography from which information in the manual is derived. The manual is written in a style which is appropriate for use by technical administrators and systems analysts as well as transducer engineers. Design curves, tables, and formulas will be discussed which enable the manual user to estimate with “state-of-the-art” accuracy the dimensions of ring transducers which achieve desired performance specifications. Limitations of the manual and the technology is included in the discussion.
Three-dimensional finite-element analysis of electroelastic normal modes of a tangentially polarized, segmented ferroelectric shell
Authors: Sabbagh Harold A. , Krile Thomas F.
Published: October 2005

We extend the variational method described in the preceding companion paper to calculate the short-circuit and open-circuit electroelastic normal modes of a tangentially polarized, segmented, ferroelectric shell. The basis functions used in the present variational approach are first-order, three-dimensional piecewise polynomials, whose method of generation is described. A discussion of the “condensation” scheme for satisfying boundary conditions is presented, and comments relevant to programming and numerical convergence matters are made.
Pitch confusions and the optimum processor theory
Authors: Houtsma Adrianus J. M.
Published: October 2005

Goldstein's optimum processor model for complex tones [J.L. Goldstein, J. Acoust. Soc. Am. 54, 1496–1516 (1973)] allows one to compute the multimodal probability density function of the estimated fundamental frequency for any two-tone complex with only one variance parameter which depends on stimulus frequency and subject. From a given set of complex tone stimuli a set of receiver-operating characteristics can be computed which have a staircase shape because of the discrete nature of the underlying density functions. Data from musical interval identification experiments using dichotically presented harmonic two-tone complexes are processed and compared with computed ROCs. Bias effects in the criteria and other bias caused by the particular musical context of the stimulus set are discussed. [Work supported by the National Institutes of Health, grant 1RO1 NS 11680-01.]
Some white-spectrum stimuli with distinct time structures
Authors: Pierce J. R. , Lipes R. , Cheetham C.
Published: October 2005

Responses to acoustic stimuli which differ in both spectrum and time structure may depend on both the time and the place mechanisms of hearing. Green [D.M. Green, “Temporal acuity as a function of frequency,” J. Acoust. Soc. Am. 54, 373–379 (1973)] has used short stimuli which differ in time structure only. Here we describe and analyze two sorts of white-spectrum stimuli with a periodic time structure. These are stochastic sequences of positive and negative short pulses of equal shapes and areas, and white noise multiplied by a periodic signal. While such stimuli have a white power spectrum, the short-term power spectrum exhibits periodicity and symmetry. The energy distribution function of the short-term power spectrum can differ for different frequencies. The time structure can be obliterated without changing the spectrum, and the spectrum can be changed without obliterating the time structure. [Work supported by Bell Telephone Laboratories and Xerox Corporation.]
Sequential effects in category judgments of noise
Authors: Nagel D. C.
Published: October 2005

Categorical annoyance judgments of the annoyance produced by flyover and other noises have been shown to be systematically affected by the stimulus context in which the sounds are heard. Such judgments also show pronounced sequential effects; that is, judgments depend on both the sequence of previous responses and stimuli. In the present study, subjects assigned category ratings without feedback to each of ten broad-band noises, presented at A-weighted sound levels of from approximately 65 to 90 dBA. The mean ratings assigned showed consistent contrast to the previous stimuli in the sequence but were assimilated to previous responses. Taken together, these judgment tendencies resulted in an overall bias to utilize categories with equal frequency. This bias, in turn, is responsible for the overall contextual effects displayed in the molar judgment data.
Project TASMAN TWO—low-frequency sound propagation measurements in the southern ocean
Authors: Bannister R. W. , Denham R. N. , Guthrie K. M. , Browning D. G.
Published: October 2005

The seas to the west and south of New Zealand provide unique conditions for the measurement of low-frequency sound propagation. Due to the dominant influence of the antarctic circumpolar current, a series of water masses are stratified between latitude 40 °S and 60 °S. This allowed the design of an experiment to extend a result of Project KIWI ONE—that sound propagation at frequencies below 200 Hz is significantly affected by variations in the sound velocity profile—by relating sound propagation to the oceanographic properties of water masses. From the receiving hydrophones, aircraft flew along five acoustic propagation paths out to a range of 3000 km. These paths were chosen to sample specific water masses and to cross water mass boundaries, which in one case was a large bathymetric feature. Detailed oceanographic data was obtained along the tracks. The results show that below 100 Hz there is an order of magnitude variation in the rate of attenuation from location to location and this may be related to the measured oceanographic properties. [This project was partially funded by ONR Code 486.]
Comparison between predicted and measured long-range transmission loss in the Arctic Ocean
Authors: Diachok O. I. , Long G. K. , Lohanick A. W.
Published: October 2005

Measurements of transmission-loss and sea ice ridge parameters were made in the Arctic Ocean to test predictions of an under-ice reflection loss model, which assumes that ridges may be represented as randomly distributed elliptical half cylinders. The acoustic measurements were accomplished with explosive signals deployed from aircraft, and detected on hydrophones deployed from ice island T-3, and at other sites on sonobuoys monitored from a second aircraft. Ice ridge keel characteristics were inferred from airborne laser profiler measurements of ridge sails, using geometrical and statistical models. Numerical values of reflection loss as a function of keel parameters and grazing angle were incorporated into ray theoretical computations of transmission loss for comparison with measurements. The good agreement at high frequencies (f>200 Hz) is in accordance with previous such comparisons, and may be considered a demonstration of the validity of the model. The good agreement at low frequencies (f<50 Hz), however, may in part be fortutitions due to uncertainties in the ridge model, reflection-loss model, ray theoretical computations, and source levels.
Scattering of sound from a point source by a rough surface progressing over a refracting ocean
Authors: Harper E. Y. , Labianca F. M.
Published: October 2005

A normal-mode theory for scattering in the presence of refraction is reviewed. The theory is based on a perturbation expansion for small-surface wave heights. New results, based on this theory, are presented for the Doppler-shifted signal sidebands. The results are derived by a saddle-point analysis of the exact integral solution of the first-order perturbation problem. Specific numerical results for the case of upward refraction display considerably more structure than in the case of an isovelocity ocean, due to the effects of backscatter and focusing. [Research supported by ONR, Code 480.]
Statistical properties of ocean bottom roughness
Authors: Berkson Jonathan M.
Published: October 2005

Statistical properties of ocean bottom roughness for use in models of acoustic scattering from the ocean bottom were investigated for a wide variety of bottom types. The roughness probability density functions and spectra were determined from submarine topographic data obtained by narrow-beamwidth multibeam echosounding and sea bottom photogrammetry. The resultant power spectra of the roughness were compared to the asymptotic expression, A(k)=ck−b, where A(k) is the power spectral density, c is a constant, k is the wavenumber, and n is a constant which is characteristic of the class of roughness. In analytical treatments of acoustic scattering from a randomly rough ocean bottom, the bottom roughness power spectra is often assumed to have a dependence of b=3. However, for these data the value of b ranges from about 0 to about 4. The geological and oceanographic processes that may cause the wide variety of roughness spectra and probability density functions are discussed.
Linear statistical equilibrium theory of wind-developed deep water ocean waves
Authors: Middleton David
Published: October 2005

A new statistical-dynamical model of wind-developed deep water ocean gravity and ultra gravity waves is constructed. The present model is a linearized extension of the combined Miles-Phillips theory, into the limiting equilibrium region where all (surface) wavenumbers are saturated for the stationary average wind speed driving the ocean surface over an assumed (very large) fetch. Equilibrium is maintained by the nonlinear action of wave breaking primarily, whose effects are represented here by an equivalent dissipation mechanism governing a now-linearized Langevin equation of surface elevation and motion. The wave-surface/atmosphere interactions are described by a Gaussian forcing term, proportional to the effective (random) atmospheric pressure component in phase with the wave surface, whose postulated intensity spectrum appears acceptable over the general frequency range from low to high wave-surface frequencies. Among other results, the model thus provides explicit point and directional wave spectra (and their associated covariances) for the surface elevation, velocity, and cross spectral cases. The surface elevation spectra exhibit the g2fs−5, v−4 behavior required at the high frequencies on dimensional grounds, and include a non-vanishing low-frequency “cutoff”, which permits the transmission of very long wavelength disturbances through the wind-developed region. In addition to oceanographic interest, the model has important applications in underwater sound, acoustic scattering, and underwater communications. [Work supported by the Office of Naval Research, Code 222.]
SP and CM following exposure to intense tones
Authors: Durrant J. D.
Published: October 2005

Utilizing intracochlear electrodes in the guinea pig, the effects of intense pure-tone exposures on the SP (specifically, the negative component of the summating potential recorded differentially) and the CM (cochlear microphonic) were examined. Experiments were carried out in the first three turns (separate preparations for each turn; N=5/turn group) under conditions in which the exposure stimulus was presented at a frequency near the best frequency of the recording site: 12 000 Hz for T1, 2828 Hz for T2, and 750 Hz for T3. The same duration and intensity of the exposure stimulus was used in all cases, namely 110 dB SPL for 5 min. Overall-comparable degrees of depression were seen in the CM in all three turns following exposure. However, during the first 10 min or more of recovery, the SP was depressed proportionally less in the upper turns than in the first turn. The relative changes in the CM and SP thus differed between turns as well. In the third turn, the CM was depressed more than the SP, the opposite of the situation in the first turn. The second-turn data reflected a trend towards approximately the same proportional changes in the two responses following exposure. Viewing the CM and SP as spectral components of the hair-cell output, viz., the fundamental and the dc component, respectively, it appears that recovery of the total hair-cell output from acoustic trauma is somewhat complex and differs between turns. [Work supported by DRF.
Perceptual features of aging male speech
Authors: Hartman D. E. , Danhauer J. L.
Published: October 2005

Spontaneous speech samples of 46 male speakers between the ages of 25 and 70 years were played to 40 untrained listeners who estimated the speakers' ages. Samples which showed agreement among untrained listeners were played to 20 trained listeners who described the perceptual features of the given perceived ages via an a posteriori schema. Results showed characteristic perceptual features for four perceived age decades which could be classified to pitch, rate of speech, quality, and articulation. It was concluded that the features identified should be useful for defining the criteria of “normal” aging speech, and in traditional speaker recognition research.