Index of content:
Volume 82, Issue S1, November 1987
- PROGRAM OF THE 114TH MEETING OF THE ACOUSTICAL SOCIETY OF AMERICA
- Session A. Engineering Acoustics I: Structure Interaction and Transducer Arrays
- Contributed Papers
Application of a variational principle for fluid‐structure interaction to the analysis of the response of an elastic disk in a finite baffle82(1987); http://dx.doi.org/10.1121/1.2024694View Description Hide Description
A variational principle derived from the Kirchhoff‐Helmholtz integral theorem was previously employed to predict the surface pressure and vibrational response of an unbaffled elastic disk to harmonic excitation [J. H. Ginsberg and A. D. Pierce, J. Acoust. Soc. Am. Suppl. 1 79, S35 (1986)]. Here, the analysis is extended to describe a finite rigid baffle by using assumed mode functions for the pressure distribution along the vibrating and rigid surfaces. Versions using various analytical functions, as well as finite element modes, are developed and examined for convergence, numerical accuracy, and efficiency. Dependence of the surface pressure on the baffle radius is examined; results in the limit of an infinite baffle are shown to agree well with a prior analysis by Alper and Magrab [J. Acoust. Soc. Am. 48, 681–690 (1970)]. [Work supported by the Office of Naval Research, Code 1132‐F.]
82(1987); http://dx.doi.org/10.1121/1.2024695View Description Hide Description
The purpose of this work is to compare the modal contribution of a finite elastic plate to the power spectrum and the flexural wave contribution of an infinite elastic plate to the power spectrum. Flexural waves in an infinite plate submerged in a fluid produce pressurewaves in the fluid that travel parallel to the plate and decay away from the plate surface in the fluid. These waves may be called evanescent waves and are strongest in a region near the plate where they may represent noise. When a rectangular plate submerged in a fluid is excited, the vibrating plate generates pressurewaves that decay away from the plate surface in the fluid and an acoustic wave that radiates into the fluid. The analytical model considered here is a two‐sided fluid‐loaded rectangular plate simply supported at its boundaries and excited by a forcing function on one side of the plate. Theoretical analyses are discussed and the wavenumber‐dependent transfer functions that are used for calculating power spectra are presented. [Work supported by NUSC.]
Impulse responses and radiation impedances of cylindrical vibrators using wave‐vector‐time domain (k‐t) methods82(1987); http://dx.doi.org/10.1121/1.2024742View Description Hide Description
A wave‐vector‐time domain (k‐t) method is presented to evaluate the harmonic and time‐dependent loading on a cylindrical shell that is vibrating with a specified spatial and time‐dependent velocity. The method is based on utilizing a specified time‐dependent modal expansion for the radial velocity of the vibrator. The acoustic loading on the vibrator is also expressed as a modal expansion in which each coefficient is a summation of convolution integrals of the modal velocity coefficients with mode‐dependent radiation impulse responses. In contrast to an earlier work [D. D. Ebenezer and Peter R. Stepanishen, J. Acoust. Soc. Am. 81, 854–860 (1987)], the radiation impulse responses are evaluated using a (k‐t) method based on a time‐dependent Green's function for a baffled cylindrical vibrator. A comparison of the new method with the earlier method is presented along with numerical results for various axial mode shapes and circumferential mode numbers. [Work supported by ONR.]
An array to produce convected normal velocity and pressure fields for wavenumber calibration and boundary layer modification82(1987); http://dx.doi.org/10.1121/1.2024743View Description Hide Description
A transmitting array is described that is capable of producing specified normal velocity distributions at the surface of the array. Such an array is useful in the production of convected normal velocities and pressure fields that may emulate turbulent boundary layerpressure fields. The pressure fields are convective in nature, moving parallel to the plane of the array with a phase velocity that is slow, i.e., less than the speed of sound in the medium. A slow phase velocity corresponds to a spatial wavenumber k that is greater than the corresponding acoustic wavenumber in the medium. Applications include boundary layer modification, testing of sensors in controlled TBL flows, and wavenumber calibration of sensors. Results are presented for several prototype arrays involving different construction methodologies. Shading schemes for the production of uniform convected pressure fields are discussed relative to the physical limitations of the device and electronics. Measured pressure fields and phase velocity are contrasted with those predicted theoretically. Plans for producing larger and more complex arrays are discussed relative to current prototype results. [Work partially supported by ONT.]
82(1987); http://dx.doi.org/10.1121/1.2024744View Description Hide Description
Nearfield calibration arrays (NFCAs) are used to determine the farfield acoustic radiation from a transducer by measurements made in the nearfield. The original Trott NFCAs were planar arrays. More recently, cylindrical NFCAs have been developed. This paper describes a spherical NFCA for determining the full three‐dimensional radiation pattern of an enclosed transducer. In theory, the NFCA consists of a large number of discrete hydrophone elements arranged over a spherical surface in N equi‐spaced constant latitude bands. In practice, the array can be synthesized by use of a single semicircular are containing relatively few elements and by either rotation of the transducer or revolution of the are about the polar axis. Individual hydrophone sensitivities (both amplitude and phase) are selected so that the NFCA serves as a plane‐wave filter for acoustic radiation from inside the sphere. The relative sensitivities, called shading coefficients, are computed using the NFCA reciprocity principle and a least‐squares procedure. Only a few sets of N/2 shading coefficients are required for full three‐dimensional operation of the NFCA over a frequency range of three octaves or more. Design criteria are discussed and sample numerical results are presented.
Design of an experiment to measure the acoustic mutual impedances of a resonant, close‐packed sonar array82(1987); http://dx.doi.org/10.1121/1.2024745View Description Hide Description
Knowledge of the acoustic mutual impedance coefficients for a sonar array is necessary to achieve an accurate analytical description of the performance of the array. In the early stages of a design, analytical approximations, e.g., of an array in an infinite rigid plane baffle, provide adequate estimates. When prototype hardware is available, it may be appropriate to derive refined estimates of the impedance coefficients. This allows more accurate prediction of the effects of design changes on performance, and also provides an analytical baseline for a design that is entering production. The measurement of the mutual impedance coefficients in a resonant, close‐packed array is difficult. The only previously reported attempt [Stephen C. Thompson, J. Acoust. Soc. Am. Suppl. 1 68, S34 (1980] achieved inconclusive results, due to inaccurate estimates of the element electromechanical transfer matrices and strong element interactions that occur in the vicinity of the element resonance frequency. These measurements are being repeated, with greater care given to the previously noted deficiencies. The experiment will be described, concluding with the requirements placed on the special transducer elements that were necessary for the measurement. The design of these transducer elements will be described in a companion paper in this session [M. P. Johnson, J. Acoust. Soc. Am. Suppl. 1 82, S2 (1987)].
82(1987); http://dx.doi.org/10.1121/1.2024746View Description Hide Description
A companion paper in this session has described an experiment to measure the acoustic mutual impedances in a resonant, close‐packed sonar array [S.C . Thompson, J. Acoust. Soc. Am. Suppl. 1 82, S2 (1987)]. This experiment requires the use of special measurementtransducers in place of the elements in the array under study. There were several special requirements for these transducers: (1) high mechanical input impedance at the element radiating face; (2) accurate knowledge of the transducer electromechanical impedance matrix; and (3) radiating face dimensions that are identical to those of the elements of the array under study. The first requirement reduces the effects of interelement coupling and, consequently, reduces the sensitivity in the calculations to errors in the measurement. The second acknowledges that the electromechanical impedance matrices of the transducer elements are needed in the calculation. The third requirement provides a measurement array with the same mutual impedance coefficients as the array under study. A prototype transducer element that seems to meet these requirements is a doubly resonant piston element. The element is designed so that the operating band of the array under study falls between the two resonant frequencies of the measurementtransducer. The design and performance of this element will be described.
82(1987); http://dx.doi.org/10.1121/1.2024793View Description Hide Description
Though widely used in air acoustics, the diffuse field technique has not been extensively used in underwater acoustics. This paper describes a small (2 m3) diffuse field sonar tank facility intended for production testing of the steady‐state sound‐power level and directivity index of wideband sonar projectors. When evaluated by standards used in air acoustics to ensure adequate diffuse field measurement accuracy, this compact and inexpensive facility appears to be qualified for the determination of band‐averaged sound‐power level over a frequency range exceeding five octaves (from about 4 to 160 kHz). When compared with a free‐field measurement facility, the practical advantages of this approach include large savings in cost and space. Long transducer rise times are permitted since measurements are steady state. Moreover, the implicit spatial integration of radiation patterns makes results of diffuse field testing more appropriate than free‐field testing for some applications.
Practical acoustic beams and synthesized source fields for turbulence detection at the lower atmosphere82(1987); http://dx.doi.org/10.1121/1.2024794View Description Hide Description
Computer simulations of new analytical results that are important extensions of an earlier work [S. A. Adekola, J. Acoust. Soc. Am. 76, 345–368 (1984)] are here presented for the echosonde (acoustic echosounding) system. It is shown that the source distributions synthesized from the preassigned directive pressure fields are not only confined within finite antenna aperture regions, but are also characterized by maximum intensities at the boresight region of the aperture and are considerably attenuated towards the rim of the antenna cuff. It is also shown that if the beamwidth of the pattern prescribed is extremely narrow, then the source distribution synthesized from it tends to produce an undesirable strong field or singularity at the rim of the antenna cuff. The paper then focuses attention on the factors governing the realizations of practical echosonde patterns; which are suitable for turbulencedetection of atmospheric irregularities, such as thermal structures, dynamics, and turbulent velocity fields at the lower atmosphere; and from which physically realizable source fields exhibiting no abrupt discontinuities across the antenna aperture can be synthesized. Last, a comparative analysis shows that the approximate patterns generated from the synthesized source distributions manifest good mean‐square fits to the idealized acoustic beams originally specified almost all over the entire visible ranges of the patterns prescribed. [Work supported by Lagos State University, Badagry Expressway, Ojo, Lagos, through a Visiting Professorial Appointment.]
Development of higher‐order Zernike polynomials employed in the analysis of acoustic sensing antennas82(1987); http://dx.doi.org/10.1121/1.2024795View Description Hide Description
Zernike circle polynomials have been found to be invaluable analytical tools for investigating the characteristics of acoustic sensingantennas [S. A. Adekola, Acustica 47, 114–131 (1981)]. New analytical results obtained from an investigation of the generalized Zernike circle polynomials, summarized in the above‐mentioned reference, are of higher degrees of accuracy than those presently available in the literature. Indeed, these polynomials are generated to 30° and 29° of freedom describing the even‐ and odd‐numbered modes, respectively; and the field radiated by the acoustic sensingantenna, which is then expressed in terms of the polynomials, is evaluated through direct routine integration. Some of the new analytical results evaluated and numerically computed are here displayed in graphical forms for completeness. Also, illustrative examples given show not only how the Zernike polynomials can be employed in the interpretations of the radiationcharacteristics of the acoustic echosounding systems, but also how use can be made of the polynomials to describe significantly tapered source distributions essential in successful acoustic remote sensing of the lower atmosphere. [Work supported by Lagos State University, Badagry Expressway, Ojo, Lagos, through a Visiting Professorial Appointment.]
- Session B. Psychological and Physiological Acoustics I: Hearing‐Impaired Speech and Auditory Perception
The effect of varying the amplitude‐frequency response on the masked SRT for sentences in hearing‐impaired listeners82(1987); http://dx.doi.org/10.1121/1.2024796View Description Hide Description
A multichannel gain‐control hearing aid, in which the frequency‐dependent amplification is adapted automatically to the fluctuations of the incoming signal,. may optimally deliver speech to an impaired ear. Such a system requires that the speech‐reception threshold (SRT) in noise is, within limits, unaffected by dynamic variations in the amplitude‐frequency response of the hearing aid. For normal‐hearing listeners, van Dijkhuizen et al. [J. Acoust. Soc. Am. 81, 465–496 (1987)] found that the masked SRT for sentences is remarkably resistant to dynamic variations in the slope of the amplitude‐frequency response when it shaped both speech and noise. In this experiment, we studied corresponding conditions for hearing‐impaired listeners. Again, the amplitude‐frequency response shaped both speech and noise and thus left speech‐to‐noise ratios untouched. The results obtained so far also indicate that, for hearing‐impaired listeners, dynamic variations of the amplitude‐frequency response do not affect the SRT.
82(1987); http://dx.doi.org/10.1121/1.2024842View Description Hide Description
For moderate hearing impairment, the speech‐reception threshold (SRT) for sentences in stationary noise is roughly up to 5 dB higher than for normal hearing. However, a fluctuating interfacing sound seems more typical for everyday listening conditions than stationary noise. For a group of young hearing‐impaired listeners, the difference in SRT with normal‐hearing listeners increased from 2.7 dB for stationary noise to 7 dB for 100% sinusoidally intensity‐modulated noise, and even to 10.5 dB for interfering speech. In this experiment, the interfering speech was the time‐reversed voice of the same speaker as the signal, which may have introduced an extra difficulty. In a subsequent experiment, the SRT for sentences from two speakers (one male and one female) was measured with a variety of fluctuating maskers covering the range from stationary noise to interfering speech. The two voices were mutually used as masker, giving a meaningful interference. Meaningless interference was obtained by time reversal of the interfering speech or by modulating noise with the envelope of speech. Two types of modulation were used: one broadband and one in which high‐ and low‐frequency noise (separation frequency 1000 Hz) was modulated with the high‐ and low‐frequency envelope of speech, respectively.
82(1987); http://dx.doi.org/10.1121/1.2024843View Description Hide Description
Two specific applications of single‐channel compression are studied separately: whole‐range compression as a compensation for recruitment, and compression limiting as an elegant method to limit excessive output levels. In cases of compression limiting, compression proved to be superior to peak clipping, especially for the perception of consonants: An average improvement in identification scores of 18% was found and less roll‐over for the perception of final consonants at high presentation levels. In cases of whole‐range compression, a small, but significant, increase in identification scores was found for settings with the compression knee‐point at the lowest level. A detailed analysis of the patterns of confusions revealed clear qualitative differences in the perception of phonemes with and without compression. These differences can be brought into relation with an improved perception of temporal cues, like the preburst silent interval of plosives, and with spectral changes due to the activation of the compression circuit.
82(1987); http://dx.doi.org/10.1121/1.2024844View Description Hide Description
To date, hearing levels and pyschoacoustic measures have shown factor structure and correlations with speech perception in noise among the hearing impaired, but not among clinically “normal” ears. A pseudofree‐field sentence‐in‐noise test (PFFIN) from a dummy‐head recording, simplified psychoacoustic tuning curves, and tests of linguistic processing were administered to 60 people with hearing better than 20 dB HL (0.5–2.0 kHz); one‐third had complained of auditory difficulties. Significant correlations were found among the psychoacoustic variables, which also predicted PFFIN score. Correlations between hearing levels and PFFIN remained significant after partialing age, noise exposure, upward spread of masking at 2 kHz, and educational and linguistic measures. Systematically truncating the HL distribution showed that this correlation lies mostly in the 15–25 dB HL range and implicates low‐ as well as high‐frequency hearing. Thus the 17% of the adult population between 15 and 25 dB HL (0.5–4.0 kHz) should be considered marginal, both clinically and experimentally. Scientifically, if not pragmatically, a “low fence” at 15 dB HL is justified.
82(1987); http://dx.doi.org/10.1121/1.2024845View Description Hide Description
The current standard for calculating the Articulation Index (AI) includes a procedure to estimate the effective AI when hearing is combined with speechreading [ANSI S3.5‐1969 (R1978), “Methods for the Calculation of the Articulation Index” (American National Standards Institute, New York, 1969)]. This procedure assumes that the band‐importance function derived for auditory listening situations applies equally well to auditory‐visual situations. Recent studies have shown, however, that certain auditory signals that, by themselves, produce negligible speech reception scores (e.g., F0, speech‐modulated noise, etc.) can provide substantial benefits to speechreading. The existence of such signals suggests that the band‐importance function for auditory and auditory‐visual inputs may be different. In the present study, an attempt was made to validate the auditory‐visual correction procedure outlined in the ANSI‐1969 standard by evaluating auditory, visual, and auditory‐visual sentence identification performance of normal‐hearing subjects for both wideband speech degraded by additive noise and bandpass‐filtered speech presented in quiet. The results obtained for auditory listening conditions with an AI greater than 0.03 support the procedure outlined in the current ANSI standard. [Work supported by NIH.]
82(1987); http://dx.doi.org/10.1121/1.2024846View Description Hide Description
The frequency difference limen (DL) for short‐duration second‐formant transitions (F2) presented alone and in speechlike environments was examined in normal hearing and hearing‐impaired subjects. Four stimulus conditions were included: F2 transition alone; F2 transition in the presence of all formants (full formant); full‐formant stimuli preceded by a burst and followed by a vowel; and full‐formant stimuli preceded and followed by a vowel. For the simplest stimulus condition (F2 alone), all hearing‐impaired subjects had DLs within the 95% confidence interval around the mean of the normals. For all other conditions, however, the hearing impaired demonstrated extreme intersubject variation. Some hearing‐impaired subjects could not detect a formant transition of even 800 Hz, whereas others continued to perform like normal hearers. Consonant identification performance was also examined in CV and VCV environments. Consonants whose identification is believed to involve formant transitions were selected for study. In spite of modifications to the frequency discrimination task and use of relevant syllables, relatively weak correlations continue to be observed between F2 DL and consonant recognition ability, as well as between F2 DL and degree of hearing loss. [Work supported by NIH.]
82(1987); http://dx.doi.org/10.1121/1.2024891View Description Hide Description
For measuring frequency selectivity, the notched‐noise procedure, introduced by Patterson and Nimmo‐Smith, is regarded at the moment as the best method to avoid cueing effects and to take off‐frequency listening into account. In this study, a similar approach was adopted for the measurement of lateral suppression. The amount of suppression for a two‐tone masker, as a function of the frequency of the suppressor, was measured in the presence of a notched noise. Also in this paradigm, a fair amount of suppression was found in a group of 15 normal‐hearing subjects. For hearing‐impaired subjects, and, surprisingly, also for subjects with a conductive loss, a gradual loss of suppression as a function of hearing loss was observed. Finally, the results are related to auditory filter shapes obtained in simultaneous and forward masking.
82(1987); http://dx.doi.org/10.1121/1.2024892View Description Hide Description
The presence of “excessive upward spread of masking” in high‐frequency sensorineural hearing‐loss (HFSNHL) subjects has been previously demonstrated by other investigators. This excess masking, defined by predictions from a “noise” model of hearing loss, occurs in the transition region between normal and impaired hearing. To examine this phenomenon further, frequency masking patterns (FMPs) were obtained from two normal‐hearing subjects and one HFSNHL subject. Using a 200‐Hz narrow‐band noise (NBN) masker (90 dB SPL) with an upper edge at 520 Hz, FMPs from the impaired ear demonstrated excess masking, which it was hypothesized might be explained by an inability to “listen in the valleys” of the masker envelope because of poor temporal resolution. To test this notion, the experiment was repeated using a 40‐Hz NBN masker (90 dB SPL), which has less rapid envelope fluctuations. No differences in the upward spread of masking were seen between the 200‐ and 40‐Hz bandwidth‐equal SPL maskers. When the experiment was repeated with equal spectrum level maskers, the 40‐Hz NBN masker produced less upward spread of masking in both normal‐hearing and hearing‐impaired ears. These data suggest that, for the HFSNHL subject, excessive upward spread of masking cannot be completely explained by an inability to “listen in the valleys.” [Work supported by NINCDS.]
82(1987); http://dx.doi.org/10.1121/1.2024893View Description Hide Description
In a group of 144 subjects (288 ears) medically treated with platinum drugs, high‐frequency audiometry was applied at frequent intervals in order to compare the effects of three different drug administrations: high‐dose cis‐platinum cures, low‐dose cis‐platinum cures, and carboplatin cures. In all subgroups, high‐frequency audiometry considerably enhanced the early detection of ototoxicity. Marked differences between cures were established, both in the pattern of onset of the damage and in the relation between dose and damage severity. For subjects treated with platinum derivatives, especially, the thresholds at 14 kHz prove to be important. The results suggest that, for these subjects, a single measuring frequency should be taken into consideration. With a minimum of effort, most of the increased sensitivity for a complete high‐frequency audiogram can be reached. Finally, the predictive value of a threshold deterioration above 8 kHz for a threshold deterioration in the conventional range of audiometric frequencies will be considered.
82(1987); http://dx.doi.org/10.1121/1.2024894View Description Hide Description
Amplitude compression techniques utilizing both single and multi‐channel filtering systems have been proposed for use in compensating for recruitment of loudness in hearing aids. Due to a variety of limitations, these techniques have shown only marginal success. The sinusoidal model of speech developed by Quatieri and McAulay [IEEE Trans. Acoust.Speech Signal Process. ASSP‐34, 744–754 (1986)] has been used to simulate and compensate for recruitment, using the general theory described by Villchur [J. Acoust. Soc. Am. 56, 1601–1611 (1974)] and others. Specifically, the model synthesizes speech as the sum of sinusoids with time‐varying amplitudes and phases. Amplitude compression and amplification can be performed on individual sinusoids to shape speech to fit an impaired person's residual hearing. Conversely, expansion can be used to Simulate the effects of recruitment. This model offers almost unlimited flexibility in areas such as consonant boosting and spectral shaping in any portion of the spectrum. With a real‐time implementation of this model, the parameters can be easily adjusted to suit each person's needs.