bluuebunny/crossref_metadata_2025 huggingface dataset:
Assessment of noise impact in transportation planning
Authors: Fidell S.
Published: October 2005

As part of a comprehensive reference book on environmental impacts in transportation corridors, a practically oriented guide to assessment of noise effects has been prepared for the California Department of Transportation. Among the major areas covered are (1) review of current knowledge of noise and vibration effects associated with highway and rail traffic; (2) provision of methods of estimating exposure levels associated with varying amounts of traffic; (3) relation of expected levels of exposure to anticipated social impacts; (4) explanation of a quantitative Noise Impact Index for evaluating alternative routes; and (5) discussion of measures to mitigate noise impacts. This paper summarizes the findings contained in the guide to noise impact assessment to bring it to the attention of transportation planners and others concerned with assessment of noise effects.
Workship on noise control and urban planning
Authors: McCord Dickman Donna , Anderson Patrick
Published: October 2005

In recognition of the fact that unrestricted land development and planning decisions made without regard to noise have left a legacy of noise problems which are just now being addressed by noise abatement progams, the Metropolitan Washington Council of Governments and the Environmental Protection Agency, Region III, sponsored a two day Workshop on Noise Control and Urban Planning. The purpose of the workshop was to prepare local, state, and Federal government land use planners to identify and quantify potential noise problems using current analytical techniques and solutions to noise problems through compatible land use planning. The Workshop was divided into several major elements. Problem definition, assessment alternatives and problem solutions presented the land use planner with a logical, sequential path of attack. A discussion of the integration of Federal, State, and local legislative parameters into the planning process was coupled with the latest analytical techniques and noise prediction programs. Potential solutions to the noise problems included planning as well as operational controls. Comprehensive and master planning, zoning, architectural controls, building codes, and capital improvement planning were among the solutions covered. The purpose of this paper is to present highlights of the major elements of the Workshop.
Perceptual invariance and onset spectra for stop consonants in different vowel environments
Authors: Blumstein S. E. , Stevens K. N.
Published: October 2005

In this series of perception experiments, we have attempted (a) to determine if a brief stimulus in which only the spectral information at onset is preserved provides sufficient cues for identification of place of articulation across vowel contexts, and (b) if it does, to define further the nature and size of the spectral window. Subjects were randomly presented with synthetically produced stimuli consisting of a 5- or 10-msec noise burst followed by a brief voiced interval containing three formant transitions with onset and offset characteristics appropriate to the consonants [b, d, g] in the environment of the vowels [a, i, u], as well as stimuli with steady second- and third-formant transitions. The length of the voiced interval was systematically varied from 40 to 5 msec. The results indicate that an onset spectrum consisting of the burst plus the initial 5 or 10 msec of voicing provide sufficient cues for the identification of the stop consonant, and that vocalic information can be reliably derived from these brief stimuli containing only one or two glottal pulses. [Research approved by an NIH grant.]
Secondary acoustic characteristics and vowel identification
Authors: Bischoff D. M.
Published: October 2005

The relation between primary acoustic cues and secondary acoustic characteristics of vowels was investigated by asking subjects to identify vowels in different contexts and under different conditions. Six isolated vowels, /a/, /i/, /u/, /æ/. /I/, and /ε/ were presented under nine conditions: naturally spoken, synthesized, synthesized with no differential fundamental frequency information, synthesized with no differential durational information, synthesized with no formant structure, synthesized with no differential fundamental frequency or durational information, synthesized with neither formant structure nor differential fundamental frequency information, synthesized with neither formant structure nor differential durational information, and synthesized with no formant structure and no differential fundamental frequency or durational informational information, synthesized with neither formant structure contexts and presented to subjects for identification. These contexts were chosen to either minimize or maximize the consonant-vowel transitional information. The results will be discussed in terms of the relative importance of primary and secondary acoustic characteristics for vowel identification.
Speech perception in early infancy: Perceptual constancy for vowel categories
Authors: Kuhl Patricia K.
Published: October 2005

The young infant discriminates a single token of /a/ from a single token of /i/ and can successfully ignore variations in the pitch contour of these vowels [Kuhl and Miller, J. Acoust. Soc. Am. 59, S54(A) (1976)], but demonstrations that an infant perceives an inherent similarity in a male's /a/ and a female's /a/ are lacking. Three infants, 512- to 6-months old, were tested in a task in which a head-turn response was reinforced with a visual stimulus in the presence of a speech sound from one category but not in the presence of a speech sound from a second category. The infants were trained with a single synthetically produced /a/ and /i/ whose formant frequencies and fundamental frequency were appropriate for a male voice (rise-fall pitch contour) and then tested with synthetic /a/ and /i/ tokens whose formant frequencies and fundamental frequencies were appropriate for female and child talkers, with both rising and rise-fall pitch contours. Generalization to novel tokens was often times immediate. The six-month old's behavior is suggestive of either an inherent, or easily learned, perceptual constancy for vowel categories when both the critical (formant frequencies) and noncritical (fundamental frequency, pitch contour) acoustic dimensions are randomly varied.
Stimulus correlates in the perception of voice onset time (VOT): II. Discrimination of speech with high and low stimulus uncertainty
Authors: Sachs R. M. , Grant K. W.
Published: October 2005

The three Lisker—Abramson CV series (ba-pa, da-ta, ga-ka) were computer synthesized at C.I.D. in 1-msec steps. For each 84-trial block of a high-uncertainty, same-different task (no feedback), one of the three continua was chosen at random. On each trial, the ability of three naive listeners to resolve for differences among VOT pairs ranging from VOT = 10–80 msec, in 10-msec steps was tested. As expected, there was a local maximum performance for each of the three speech continua, thus satisfying one criterion for categorical perception. A minimum-uncertainty AX task (with feedback) was run with three highly trained subjects for the ga-ka continuum. Only one standard VOT was presented in a block of trials. Results were not categorical: the minimum detectable ΔVOT was less than 2 msec at VOT = 10 msec and increased as VOT increased. Our results indicate that trained listeners can utilize subtle stimulus cues in synthetic speech if stimulus uncertainty is minimized. [Supported by a grant from NINCDS.]
Perception of vowel duration
Authors: Wang William S.-Y. , Lehiste Ilse , Chuang Chin-Kuang , Darnovsky Nancy
Published: October 2005

Lehiste and Pisoni have independently shown that the perception of vowel duration is influenced by the F0 pattern that accompanies the vowel. Here we report some related results. Sixteen subjects are asked to judge the relative duration of pairs of synthesized stimuli. The stimulus pairs are made up of the vowels /i/, /a/, /ai/, and nonspeech, where nonspeech is a single formant at 1500 Hz. The F0 patterns are: level = 120 Hz, rising = 105–135 Hz, and falling = 135 to 105 Hz. The durations are 220, 240, 260, and 280 msec. The orthogonal combination of these factors makes up a total of 512 stimulus pairs. In comparing duration judgements of stimuli of rising (R), falling (F), and level (L) F0's, we found that the following relation obtains: R > F > L, where “>” denotes “perceived as having greater duration.” This relation holds to the same extent for both speech and nonspeech stimuli. There is no signficant difference in the perceived duration between /i/ and /ai/. The most consistent effect, found also earlier in a pilot experiment we performed with 58 subjects, is that /i/ is heard as longer than /a/, in a ratio of 60% : 40%. Since /i/ is normally shorter than /a/ in natural speech, we posit a compensation mechanism mediating between perception and production that is responsible for this negative correlation between vowel height and duration. It is interesting to note that although the processes of duration judgment and pitch judgement (cf. the abstract by Chuang and Wang) are different, they seem to share a common compensation mechanism between production and perception. [Work supported by NSF Grant No. BNS 76-00017-Wang-2/78J34.2.]
How a composer looks at organization
Authors: Chadabe Joel
Published: October 2005

Two discrete levels represented one after another on a graph are normally considered to be connected by a vertical line which has no time value. But the realities of engineering teach us that any vertical line has a time value, no matter how short. If that graph is charting the sound from a single source, we may observe that even extremely dissimilar sounds are connected in a continuum; that they are not items that are arranged, but rather a continuous process of change. Changes may be periodic, transient or random, and they exist at different time scales from “audiotime” (changes in milleseconds) to “conscious time” (changes in seconds). These changes attribute to a sound's “meaning.” The problem for a composer of computer music is how to generate these changes meaningfully.
Octave equivalence and the processing of melodic sequences
Authors: Deutsch Diana
Published: October 2005

It is clear that a certain perceptual equivalence exists between tones standing in an octave relationship. However, it was previously found that subjects were unable to recognize a well-known melody when its component tones were placed randomly in any one of three octaves. In the present experiment, subjects listened to a standard melody, and then to a comparison melody; and they judged whether the comparison melody was an exact transposition of the standard or not. Exact repetition of the standard melody produced a significant improvement in performance, which also occurred when the repeated melody was displaced entirely to the octave above or the octave below that of the standard. However, when the melody was repeated such that its individual components were placed alternately in the octave above and the octave below, performance was significantly poorer than when the melody was not repeated at all. It appears that octave equivalence does not operate in short term melody acquisition any more than in long term melody recognition.
Effect of wavelet processing on stratigraphic resolution
Authors: Rives R. G. , Patch J. R.
Published: October 2005

Wavelet processing has proved to be a viable technique for improving the resolution and character of seismic events. Substantial improvement in signal-to-noise ratio, better continuity and stability of events are obtained with this technique. As a result of this improvement a more meaningful stratigraphic interpretation is available to the geophysicist. Evidence of this improvement in data quality has been the close agreement between well data and wavelet processed data. This agreement is possible because the effects of ghosting, instrument and system responses are removed from the seismic data. Also, accurate stacking velocities obtained with this technique normally help to improve the primary-to-multiple ratio on the stacked sections. Owing to this technique there is increased stratigraphic resolution in marine data which has resulted in more reliable quantitative methods for relating the reflection coefficients and the polarity of reflections to lithology. Current subsurface modeling techniques and other recent developments along these lines are providing insights which are principally of a stratigraphic nature.
Band elimination processor for an experimental parametric acoustic receiving array
Authors: Rohde D. F. , Goldsberry T. G. , Olsen W. S. , Reeves C. R.
Published: October 2005

In the Parametric Acoustic Receiving Array (PARRAY), a low-frequency acoustic wave modulates a locally generated high-frequency carrier wave. The information in the low-frequency signal wave then appears as low-level, modulation side bands of the carrier. The function of the receiver electronics is to suppress the high-level carrier while simultaneously amplifying the low level near side-band signals. A band elimination processor has been developed that is capable of detecting signals with carrier to side-band ratios approaching 180 dB. Cascaded crystal filters and low-noise amplifiers provide the high order to carrier suppression. Additional circuitry is incorporated to permit independent observation of the upper and lower side bands. The band elimination processor is described in detail and test data for side-band signals 40 Hz–4 kHz from a 65-kHz carrier are presented. [This research was supported by the Advanced Research Projects Agency of the Department of Defense and was monitored by the Naval Electronic Systems Command under Contract NO0039-76-C-0231.]
Research in listener related problems for high fidelity sound reproduction
Authors: Nakayama T. , Miura T.
Published: October 2005

Since the end of the 1950's, active research into the subjective assessment of reproduced sound quality has been conducted in Japan. After a brief review of these efforts, one research of the sound quality assessment process is introduced as an example of recent developments. In this study, a multiple regression model is adopted for quantitatively describing the sound quality assessment process by listeners. In this model, a psychometric scale expressing the total quality judgement (preference) of a listener to a reproduced sound is explained using a weighted sum of multidimensional scale values. These values are related to the physical characteristics of the reproduction system. The weight for each sensory scale is different for each listener and is thought to change with advances in the audio field. Subjective sound quality for various reproduction systems differing in lowpass and highpass frequency characteristics, nonlinear distortion and dyanmic range are experimentally analyzed. Then, four-dimensional sensory scales, corresponding to these physical characteristics, are constructed. The estimated preference scale values, calculated from these sensory scales using a multiple regression model, coincide fairly well with observed values.
On the effective stiffness of corrugations and its contribution to loudspeaker cone
Authors: Nomoto Isami , Suzuki Kiyoaki , Yamamoto Makoto , Tomita Yoshio
Published: October 2005

Formation of corrugation on the loudspeaker cones has been commonly used for a long time to control vibrational characteristics of the cones, however, no practical formula for designing the corrugation has ever been developed. The authors have developed an approximate formula for the corrugation design with the aid of the Finite Element Method. Since the corrugation provides a lower stiffness to the loudspeaker cone at its lumped portion, the effectiveness of the corrugation depends upon the stiffness ratio between the corrugation and the cone. Two equations are introduced to obtain the stiffness ratio. One is for the effective stiffness of corrugation which involves the material, the shape, and the dimensions of the corrugation, with the ratio of the corrugation curvature to the width as a parameter and the other is for the effective stiffness of cone as a function of its shape. The ratio of the two equations gives a practical formula for corrugation design. Graphical displays of the cone vibration obtained by using the F.E.M. are close to authors' expectation. The effects of corrugation on the cone vibration and on the on-axis frequency responses are also discussed.
Some recent developments in acoustic remote sensing
Authors: Brown Edmund H.
Published: October 2005

New initiatives for real-time quantitative acoustic determinations of atmospheric parameters and structure are described. Included are methods for measurement of profiles of turbulence and temperature fluctuation strengths, inversion heights, the strength of capping inversions, and the characteristics of atmospheric gravity waves. The talk will briefly describe an echo interferometer using coherent detection of acoustic mean phase that may lead to a new method for remote sensing of mean temperature profiles. The talk will also include some preliminary experimental results.
Event rates and array design for DUMAND
Authors: Bradner Hugh , Blackinton J. G. , Learned J. G. , Parvulescu A. , Stenger V. F.
Published: October 2005

We present here our consensus reached at the 1978 Summer Workshop of DUMAND (24 July-1 September 1978, held at Scripps). This abstract predates the Workshop hence the inability to go into further details.
Acoustic radiation pattern from a particle beam in water
Authors: Gualtieri F. J.
Published: October 2005

We assume that the particles in a high-energy beam from an accelerator or other source deposit heat in a water target. This localized heating causes thermal expansion, giving rise to a radiated pressure wave. We calculate the acoustic intensity at points far from the beam. Earlier work [P. J. Westervelt and R. S. Larson, J. Acoust. Soc. Am. 54, 121–122 (1973)], treated the specific case of a narrow, cyclindrical, exponentially attenuated, sinusoidally modulated laser beam depositing heat in water. The present model incorporates the following features: The beam diverges, it is not everywhere narrow, and the decay of the beam is not exponential. Directivity of the generated sound is discussed and compared with the Westervelt-Larson results. [Work supported by the U.S. Naval Ocean Research and Development Activity.]
Eardrum impedance effects on the free field frequency response of occluded ears
Authors: Gilman S. , Dirks D. , Stern R.
Published: October 2005

As part of a program for determining the effects of eardrum impedance on hearing aids, the frequency response of a KEMAR mannikin in a free field was measured using modified Zwislocki simulators. The modifications permitted the representation of approximately 80% of the impedance range exhibited by normal ears—one at the 10th percentile, one at the mean, and one at the 90th percentile. In one experiment with the simulators unoccluded, frequency response was essentially the same for all impedances except for an 8 dB maximum change at the earcanal resonant frequency. Subsequent experiments with the simulators occluded by earmold-type inserts showed differences in the frequency response up to 15 dB due to the eardrum impedance and up to 20 dB due to the diameter of the insert bore. Substantial low and midfrequency resonances which can amplify the sound field SPL by as much as 14 dB depending on the eardrum impedance are produced by such occlusions.
Hazard from gunfire impulses with peak energy at low frequencies
Authors: Price G. Richard
Published: October 2005

A tentative damage-risk criterion (DRC) for impulse noise was derived from theoretical and empirical considerations [G. R. Price, J. Acoust. Soc. Am. 62, S95(A) (1977)]. It predicted that gunfiretype impulses with their energy peaks at low frequencies would be less hazardous than impulses with peak energies in the midrange, which is contrary to the prediction from the currently accepted DRC [CHABA. Rep. WG-57 (1968)]. As a first step toward determining the validity of this position, 13 restrained, unanesthetized, cats were exposed to 60 impulses at 15 per min from a 105MM Howitzer at peak pressures from 158 to 165 dB. Auditory sensitivity was estimated by evoked response audiometry at 2, 4, 8, and 16 kHz. Pre-exposure measures were compared with measures made immediately after exposure, 1 day, and 10 days later. Shifts in sensitivity were small at 158 dB and much larger at the higher intensities. Higher intensities were accompanied by blood spots on the tympanic membrane, although without visible perforations. Later measures showed some recovery of sensitivity, although large initial losses are associated with long lasting shifts. The loss for this particular exposure begins, by extrapolation, at just under 158 dB and grows very rapidly with increases in intensity.
Statistics of sound power measurements in a reverberation chamber
Authors: Waterhouse Richard V.
Published: October 2005

The simplest theoretical case is analyzed, in which a monopole source radiates a pure tone in a reverberation chamber. We wish to estimate the free-field value W0 of its power output from measurements in which both source and receiver positions are varied. When the source position is fixed, and the receiver position is varied, the values of the mean-square-pressure p̄1 follow an exponential distribution. It is shown that, when both source and receiver positions are varied, the values of p̄2 follow a K0 distribution, where K0 is a modified Bessel function. The average value of this distribution is proportional to W0. The distribution is highly asymmetrical about the mean, and many values are needed to yield a precise estimate of W0. K0 distributions are not well known but are of some importance, as they hold for any variable which is the product of two independent variables which have identical normal or Gamma distributions.
Sound power measurements in reverberation chambers
Authors: Tichy Jiri
Published: October 2005

A great deal of effort was devoted in recent years to improve the accuracy of the reverberation chamber method for sound power measurements of small sources. The paper summarizes the results achieved. The accuracy problems exist in two areas. The actual radiation of sound power as related to the power radiated into the free field. This ratio is a function of source position and source size as well as the radiation pattern of the source. Quite recently the statistics of the power radiation for a monopole source has been found (R. Waterhouse). Another area affecting the accuracy of measurements is the determination of the time-space average of pressure squared. For this purpose, various means, primarily the rotating diffusers, have been employed. The recent results on the effect of rotating diffusers on the statistical properties of the sound field is presented. Finally, suggestions for future development are given.
An application of digital control technique for vibration fatigue test of automobile exhaust system
Authors: Sakagami J. , Miura N. , Monai K.
Published: October 2005

Recently application of random vibration technique is becoming popular in accordance with development of digital control and fast fourier transformation techniques. In this paper some application of them for automobiles were considered. To fulfill the emission control regulation, most of the automobiles introduced catalytic convertor into the exhaust system which is becoming very important component and whose reliability must be good enough to vibration fatigue. The vibration characteristic of exhaust system has many resonances at relatively low frequency range. Therefore it is necessary to excite by random wave form at the vibration fatigue test in order to simulate real driving condition. To control random wave form precisely, digital control technique was applied. As a result many improvements were obtained in excitation control. In this paper general outline of control system will be discussed theoretically and experimentally.
The motor control of speech rate
Authors: Gay Thomas
Published: October 2005

This paper will discuss recently obtained physiological and acoustic data related to some theoretical issues surrounding the motor organization of speech rate. The specific question that will be addressed is whether vowel target undershoot, increased articulatory velocity, and greater coarticulation between adjacent segments, features associated with increased speaking rate, can be explained by a simple horizontal time-compression model. This question was explored experimentally in a number of utterances by studying changes in segmental duration and articulatory effort as reflected by changes in the duration and amplitude of the muscle contractions underlying the production of these utterances. The results will also be discussed in terms of general theories of speech motor planning and recent perceptual studies of speech rate. [Work supported by NIH Grant NS-10424 and NSF Grant BSN-7616954.]
Delay quantization, throughput requirements, and performance of the shifted band beamformer
Authors: Mucci Ronald A.
Published: October 2005

For conventional digital beamforming the beam steering delays dictate the rate at which the hydrophone data is sampled: This rate is generally much higher than the Nyquist rate. Recently, interpolation beamforming was proposed as an alternate beamforming technique which utilizes hydrophone data sampled at the Nyquist rate and hence achieves hardware savings. A new technique referred to as shifted band beamforming and primarily intended for signals possessing bandpass spectral characteristics also requires lower data rates than conventional digital beamforming. This novel beamformer operates directly on frequency translated input signals to produce a frequency translated (generally baseband) beam output. The new structure has the advantage that beamformer vernier delay and throughput requirements depend on the frequency content of the translated band rather than the original band. This paper gives a detailed analysis of the potential hardware savings which may be realized using this technique. In particular, the impact of delay quantization on beampattern structure is compared for a shifted band and a conventional digital implementation. Beamformer throughput is also analyzed for both implementations. A further reduction in the beamformer throughput requirement is demonstrated by the use of interpolation with the new shifted band beamformer structure.
A generalized frequency tracker
Authors: Young J. W.
Published: October 2005

In the classic paper by Berger on Doppler velocity measurement [F. B. Berger, IRE Trans. Aero. Nav. Elec. ANE-4, 103 (1957)] it is shown that the signal received by a Doppler sonar (or radar) has the character of narrow bandwidth noise.The desired information is the “frequency” of this process which normally means some characteristic of the spectrum such as the center of power or a higher order moment. A variety of hardware implementations of “frequency trackers” have been used for this purpose.The relationship of the processing algorithm of these devices to spectral analysis of the signal is not, however, always obvious. In this paper, the concept of a generalized frequency tracker which can compute the expected value of an arbitrary function of the signal power spectrum is presented. A real-time signal processing architecture for such a device is presented. Finally, it is shown that computation of the centroid of the power spectrum requires a system closely similar to the frequency-locked loop tracker.
Optimum signal processing with suboptimum realizations to reject turbulent boundary layer noise
Authors: Owsley N. L. , Fay J. W.
Published: October 2005

Optimum beamforming or filtering of a line array of either two- or three-element closely spaced hydrophone clusters is considered. The particular topic of the paper is the cancellation of Turbulent Boundary Layer (TBL) noise but the signal processing approach applies to the general subsonic noise problem. Array gain for clusters of unweighted hydrophones and clusters of optimally weighted hydrophones are compared. Array gain improvement for two-hydrophone optimally processed clusters with high interelement TBL noise coherence relative to conventional (time delay and sum) single element beam-forming is shown to increase linearly with the TBL-to-uncorrelated noise ratio and quadratically with the magnitude coherence of the TBL process. Suboptimum realizations of the optimum cluster-element combining schemes such as noise cancellation are proposed. Those realizations are shown to result in very simple hardware implementations. The general noise cancelling approach for high TBL-to-uncorrelated noise ratios uses outputs of a subset of hydrophones in a cluster to estimate TBL noise in a single hydrophone. The estimate is subtracted from the single hydrophone resulting in a subtractor output that is ideally insensitive to the TBL process. It is shown that using three or more hydrophones in an optimally processed cluster is relatively less effective than the two-element cluster as the TBL noise level increases. This suggests that a simple two-element optimally processed cluster would be a cost effective technique for suppressing high level TBL noise because processing complexity increases as the square of the number of hydrophones in a cluster.
Analysis of an underwater DPSK telemetry experiment
Authors: Gluth W. I. , Schmid C. E.
Published: October 2005

An experimental DPSK (differential phase shift keyed) system was used to transmit messages in Dabob Bay, Washington. The 75-kHz carrier was modulated by a 48-bit binary encoded block message. A bit error rate of less than 1% was achieved with a projector-to-hydrophone range of 1000 yards with a signal-to-noise ratio around 12 dB. Results of a lab simulation and theoretical considerations indicated that a 1% rate should occur at a signal-to-noise ratio around 6 dB. Possible causes for this discrepancy, along with effects due to Doppler and multipath, will be discussed. [Work supported by U. S. Naval Torpedo Station.]
Multivariate Linear Predictive Spectral Analysis (MESA) with Positive Definite Spectral Estimate and Stable Correlation Recursion
Authors: Nuttall A. H.
Published: October 2005

A generalization of the very useful univariate maximum-entropy spectral-analysis technique [J. P. Burg, NATO Advanced Study Inst. Signal Proc., Enschede, Netherlands (August 1968)] has been made to the multivariate case. The essential analytical procedure is to minimize the trace of the sum of the weighted forward and backward error matrices by choice of the partial correlation coefficients, subject to a linear matrix constraint which guarantees that the forward-extrapolated and backward-extrapolated correlation matrix estimates are Hermitians of each other. The choice of error weighting is very important. The resulting spectral matrix estimate can be shown to be always positive definite, and the correlation recursion inherent in the maximum entropy technique can be shown to be stable. Thus, physically meaningful spectral estimates always occur; that is, all autospectral estimates are non-negative and all magnitude coherences are bounded by unity. A FORTRAN program for evaluation of the spectral matrix is available.
Real multipurpose halls and rooms
Authors: Satory R. J.
Published: October 2005

The author has long used ray-tracing techniques along with Lochner and Berger's works to improve speech understanding. To expand on this phenomena, the author has developed a system of dividing a hall into one area of many early reflections with an almost normal decay shape and a different area for the audience where there are high numbers of early reflections from the stage and few late arriving reflections. The end product has offered auditoriums with satisfactory reverberation for orchestra rehearsal and idealized speech communication to the audience. A smaller music room offers an area for idealised choir rehearsal while offering idealised speech to the rest of this classroom. Other rooms and phenomena will be discussed. Tape recorded data demonstrating the phenomena will be offered to those interested. It is hoped that an adequate instrumentation system will be developed to read the phenomena as heard before the paper is delivered.
Improved methods for calculation of array gain in anisotropic noise
Authors: Martin G. E.
Published: October 2005

The large differences of array gain in isotropic and anisotropic noise fields impacts sonar system design. For this reason, computer models for the calculation of array gain in various noise fields are needed. Initially, models with realistic propagation and anisotropic noise effects were prohibitively expensive for large-array analysis. Modifications to utilize in lattice arrays have reduced cost by a large factor. This paper describes various methods for cost-effective computation related to the evaluation of arrays with hundreds of hydrophones and beamformer look-directions. Improvements are given even for irregular array-element locations. Typical results are presented by comparing the performances of vertical and horizontal planar arrays. [Work supported by Navy Internal Research program.]
Improved performance of receiving arrays in the presence of localized nearfield noise sources
Authors: Van Buren A. L.
Published: October 2005

An element shading technique for improving the performance of a receiving array in the presence of localized nearfield noise sources is described. The shading reduces the array sensitivity over a prescribed frequency range to sound sources located within a prescribed nearfield volume while retaining desirable farfield characteristics. A numerical technique for computing the optimum (in a least squares sense) shading coefficients for a given array, frequency range, nearfield volume, and desired farfield characteristics is presented. The technique is used to obtain shading coefficients for a line array of 20 elements. The theoretical results are confirmed experimentally by measurements made using a specially constructed line.
Optimization of gain of a conformal array for torpedo sonar
Authors: Salisbury J. I.
Published: October 2005

A conformal array designed for torpedo sonar has been analytically investigated using several array optimization techniques. In particular, the complex as well as the real weights which maximize the (receive) gain of this array have been computed. The incident signal is assumed to be a narrowband, plane wave; however, the noise field is assumed to be torpedo flow noise radiated within the near field of the array. A theoretical model of the covariance matrix of this noise field has been determined. A comparison is made with the (similarly optimized) planar arrays conventionally used in torpedo sonars. The results indicate that a more comprehensive criterion than array gain is required for the torpedo sonar application. Several alternative criterion are discussed.
A low-frequency log-periodic acoustic array
Authors: Cuomo F. W.
Published: October 2005

The basic concepts derived from frequency independent and log-periodic antenna structures have been, in some instances, utilized in the construction of underwater sound devices [K. Walther, J. Acoust. Soc. Am. 35, 81–90 (1963) and R. L. Sternberg, W. A. Anderson and G. T. Stevens, J. Acoust. Soc. Am. 63, 1617–1621 (1978)]. In all investigations known to the author, however, the frequency range of interest has been in the ultrasonic regime. This paper shows how the log-periodic approach can be extended to audio frequencies in the design of acoustic arrays yielding quasiconstant directionality in excess of one decade. The analysis indicates how this can be accomplished with broadside and end-fire configurations. Computer-generated directivity patterns are included to illustrate the theoretical approach used and preliminary experimental results are added to indicate feasibility.
Personal exposure and reaction to noise in daily life
Authors: Kono S. , Sone T. , Nimura T.
Published: October 2005

The purpose of the study is to gain a knowledge of the relation between the amount of a person's exposure and his reaction to noise in his daily life. In this paper we discuss the details of noise exposure of 416 people who live in Sendai and Tokyo. The equivalent sound level Leq is recorded every ten minutes over 24 h and Leq24 is measured, too. Some of the findings from analysis of those data are as follows: (1) There is no difference in Leq24 among areas of residence, though the difference is seen among their daily activities. Especially for workers, the amount of Leq24 is dependent on the kinds of their work and their means of commutation. (2) Average difference in value between Leq during some activities when the respondent felt annoyed and his Leq24 is found to be about 4 dB. No significant difference is seen in this average value between the noisy group (Leq during working ≧80 dbA) and the non-noisy Group (Leq during working <80 dBA). This fact suggests that there might exist a hazard of insidious adaptation to noise in our daily life. [Work supported by the Ministry of Education.]
Effects of impulsiveness on noisiness of helicopter noise
Authors: Powell Clemans A.
Published: October 2005

An experiment was conducted to examine the effects of impulsiveness on the noisiness of helicopters. Subject groups located outdoors and inside two houses listened to and judged the noisiness of two helicopters and a propeller airplane, which was used as a reference. The impulsive characteristics of one of the helicopters was controlled by varying the main rotor rotational rate while maintaining a constant airspeed which resulted in other characteristics of the noise being held relatively constant. Other controlled variables included altitude and sideline distance. Preliminary analyses indicate that within each altitude and sideline distance condition, the level of impulsiveness is positively correlated with noisiness. Across helicopter types and flight conditions, however, the addition of an impulsive correction factor did not significantly improve the correlation between the noisiness judgments and noise units such as EPNL which is being considered as an international standard. Comparisons of results between subject groups within the houses, while reasonably correlated, indicate significant effects of the orientation of the rooms with respect to the aircraft flightpaths.
A unique facility for testing both structural and human response to high-amplitude, low-frequency sound
Authors: Schomer Paul D.
Published: October 2005

Heretofore, testing structural or human response to high-amplitude low-frequency impulse sounds (blasts, sonic booms, low flying helicopters) in a repeatable fashion and in natural surroundings has presented numerous technical problems primarily centered on the ability to faithfully reproduce the requisite acoustical stimuli. At the Construction Engineering Research Laboratory, the dynamic shock-test facility (shake table) has been modified to serve as a 3 × 4 m piston loud speaker. Four 15 in. speakers serve as the “tweeters” for this low-frequency “high-fi” system. Peak sound pressure levels in excess of 125 dB are produced at distances of 5 m from the table. A special test “home” has been built adjacent to the table and a standard mobile home is being added to the test system. Initial structural tests are under way and human response tests are in the process of being inaugurated.
Isolation of the resonant component in acoustic scattering from fluid-loaded elastic spherical shells
Authors: Murphy J. Diarmuid , George Jacob , Nagl Anton , Überall Herbert
Published: October 2005

The theory of resonance scattering [L. Flax, L. R. Dragonette, and H. Überall, J. Acoust. Soc. Am. 63, 723 (1978)], recently developed for acoustic wave scattering from elastic objects and elastic-wave scattering from cavities, is applied to the problem of sound scattering from fluid-filled elastic spherical shells in a fluid, in particular air-filled aluminum shells in water. The eigenvibrations of the shell appear as resonances in the scattering amplitude of each normal mode; these are superimposed on a smooth background which if the shell is made thinner, converts from the amplitude of a rigid to that of a soft sphere. We have isolated the resonant component from the non-resonant background by using an “intermediate background” formalism in order to exhibit the pure resonances. The sequence of resonance peaks in successive partial waves may be interpreted as due to circumferential waves (Regge poles) which resonate when n + 12 wavelengths span the sperical-shell circumference; they undergo a phase jump of π/2 when passing the north and south pole of the shell which explains the 12. [J. G., A. N., and H. U. supported by the Office of Naval Research, Code 421.]
Qualifying the acoustic environment for in situ noise measurements
Authors: Angevine O. L.
Published: October 2005

Much interest has centered in recent years in measurements of the noise of large machines as installed. Free-field conditions above a reflecting plane are desired, and a means for evaluating conditions in situ are a necessary part of the measurement process. A current trend in drafting ISO and ANSI standards is to employ the method of enveloping surfaces. [G. Hubner, “Qualification procedures for free-field conditions for sound power determination of sound sources,” in Internoise 73 Proceedings (Copenhagen, August 1973).] It has been found that spaces which fail to qualify by this method often pass a criterion like that of American National Standard ANSI S1.13-1971, which requires only that the sound pressure level measured at one or more points in a direction away from the machine be at least 6 dB less than that at 1 m. It is argued that the latter criterion insures adequate accuracy for engineering measurements, and that the former one requires an unnecessary number of measurements.
Waves scattering by spherical inclusions in a solid host
Authors: Flax L. , Neubauer W. G.
Published: October 2005

A theoretical study is presented of the scattering of waves from a single spherical elastic inclusion embedded in an isotropic elastic medium. Three special types of solutions are obtained: scattering cross section, differential cross section and form function. Some solutions for absorbing media are included. Results indicate the various forms of expression of the solution are useful for different purposes.
Target strength of spherical reflectors
Authors: Kikuchi T. , Yoshihara M.
Published: October 2005

The homogeneous and Morgan lenses [J. Appl. Phys. 29, 1358 (1958)], 9.0 cm in diameter, are constructed by molding the mixture of RTV silicon rubber and zinc or nylon powder in the form of sphere. The sound velocity of such a composite material is a function of the concentration of the mixture. The materials with a continuous variation of sound velocity cannot be produced for the construction of Morgan lenses. Then an investigation is made of stepped-index lenses approximated by a number of constant-index spherical shells. The focusing properties of those lenses are measured for lens diameter to wavelength ratios smaller than 30. Those lenses partially are coated with a pressure release cover. The target-strength of those reflectors are measured with respect to wavelength and the angle of incidence. Their target-strength is nearly independent of the angle of incidence and the maximum level is about the same compared to a pressure release plane of equal cross section.
Rate of transport of ototoxic drugs into perilymph
Authors: Juhn S. K. , Rybak L. P. , Morizono T.
Published: October 2005

By measuring concentration of kanamycin in perilymph after multiple doses, the accumulation of kanamycin in perilymph was observed. When the correlation between the concentration of kanamycin in perilymph and hair cell damage (by surface preparation technique) was pursued, a dose-related phenomenon was observed. Furosemide levels have been measured in chinchilla body fluids after 100 mg/kg intravenous injection. The drug has a much longer half-life in perilymph compared to other body fluid compartments. However, when this same dose is given to animals who had received 25 mg/kg injections every twelve hours for ten doses, no evidence of accumulation in any body fluids was noted. Changes in endocochlear potential (EP) in chinchillas receiving a single IV dose of furosemide 25, 50, of 100 mg/kg showed dose-dependent changes. The time after administration of drug to initiation of EP recovery was 6, 9, and 19 min for these doses, respectively. The perilymph level of furosemide is 5 μg/ml at the time that recovery of EP is beginning after the 100 mg/kg IV dose. This may represent a threshold level for this drug. These studies support the existence of a blood labyrinth barrier with selective permeability. Kanamycin tends to accumulate in perilymph but furosemide does not. The mechanism underlying these differences in drug accumulation may be a differential toxic effect on membrane permeability or excretion rate from the perilymphatic space. This basic information helps to understand the differences of degree of hearing loss (permanent or temporary) observed in patients receiving antibiotics or diuretics.
Effect of noise exposure upon staining of cochlear hair cells by nitro-blue tetrazotium
Authors: Ryan Allen , Wickham M. Gary , Bone Robert C.
Published: October 2005

The perilymphatic space of chinchilla cochleas was perfused for 30 min with nitro-blue tetrazolium (NBT) with NADH as a substrate in phosphate buffer. NBT with NADH is reduced to form a dark precipitate by NADH-cytochrome C reductase, a widely-distributed metabolic enzyme. In the normal inner ear, NBT selectively penetrated and stained both outer and inner hair cells, leaving other elements of the organ of Corti unstained. Exposure to a two-octave (1414–5656 Hz) band of noise for 1 h at 100-dB SPL, sufficient to cause temporary but not permanent threshold shift, did not affect subsequent NBT staining of hair cells. Exposure to 110-dB SPL, sufficient to cause permanent threshold shift but no immediate cell degeneration, eliminated or greatly reduced hair cell staining. Exposure to 120-dB SPL resulted in staining of cellular debris in the region of degenerating outer hair cells. Pillar cells and radial nerve fibers were also stained in regions of outer hair cell disintegration, but inner hair cells were not stained. These changes in enzyme availability for NBT staining suggest that noise alters either enzymatic activity or membrane permeability in cells of the organ of Corti.
Optimizing digital speech coders by exploiting masking properties of the human ear
Authors: Schroeder M. R. , Atal B. S. , Hall J. L.
Published: October 2005

In any speech coding system that adds noise to the speech signal, the primary goal should not be to reduce the noise power as much as possible, but to make the noise inaudible or to minimize its subjective loudness. “Hiding” the noise under the signal spectrum is feasible because of human auditory masking: sounds whose spectrum falls near the masking threshold of another sound are either completely masked by the other sound or reduced in loudness. In speech coding applications, the “other sound” is, of course, the speech signal itself. In this paper we report new results of masking and loudness reduction of noise and describe the design principles of speech coding systems exploiting auditory masking.