bluuebunny/crossref_metadata_2025 huggingface dataset:
Investigation of scaling law models for a rank of stopped diapason pipes
Authors: Smith Theoren P. , Elder S. A.
Published: October 2005

The acoustical characteristics of a rank of closed diapason pipes have been investigated experimentally to provide a basis for testing theories of scaling laws. Measured quantities determined for each pipe include pipe dimensions, sounding frequency, radiated sound pressure, jet speed at the lower lip, phase of jet crossing at upper lip, and pipe Q as a function of amplitude. Comparison with current models of organ-pipe mechanism will be attempted.
Noise and vibration ride comfort criteria
Authors: Dempsey Thomas K. , Leatherwood Jack D. , Clevenson Sherman A.
Published: October 2005

A program is underway at Langley Research Center to develop a comprehensive ride quality model based upon the various physical and psychological factors that most affect passenger ride comfort. Two of the most important factors, namely vibration and noise, were studied in a previous investigation in which the relative contribution of each factor to overall passenger discomfort was determined. This earlier study utilized a category scale to elicit responses from subjects exposed to combined sinusoidal vibration and octave band noises. The present study is an extension of this work and used a magnitude estimation procedure to obtain subjective responses to combined noise and vibration where the vibration stimuli now included random vibrations. The specific purposes of this paper are to (1) determine the absolute contribution of noise and vibration to passenger discomfort in terms of the discomfort units associated with the ride quality model, and (2) determine if passenger discomfort responses to combined noise and vibration depend significantly upon the nature of the vibration stimuli, i.e., deterministic as opposed to random. The stimuli for this study consisted of octave bands of noise centered at 500 and 200 Hz and vertical vibration composed of either 5-Hz sinusoidal vibrations or random vibrations centered at 5 Hz and with a 5-Hz bandwidth. The noise stimuli were present at levels ranging from ambient to 95 dBA and the vibration stimuli at levels ranging from 0.02 to 0.13 grms.
Correlation measurements of airframe noise in a wind tunnel
Authors: Meecham W. C. , Regan D. R. , Ahtye Warren
Published: October 2005

Preliminary airframe noise measurements have been performed on a 19-ft-high model consisting of a half-fuselage and one wing with a three-flat system (nine flaps total). Pressure taps were placed on the middle of the underside flap surfaces, with the flaps full deployed during the test. Farfield microphones were placed approximately 12 feet away from the flap system and were mounted about 10 feet above the 40 × 80 ft. NASA—Ames wind tunnel floor. During these test, the overall sound pressure level in the wind tunnel varied from 100 to 113 dB for flow speeds ranging from 130 to 230 ft/sec. Cross-correlations between hydrodynamic pressure fluctuations on the flap surface and farfield sound yield normalized cross-correlation coefficients of up to 0.2. It is found that the hydrodynamic pressure fluctuation at the flap surface is of order 12 ρ0u′2, where ρ0 is the ambient gas density and u′ is the turbulent fluctuation velocity. Using the relation between the correlation function and the far-field radiated sound pressure level, it is found that the flap region (correlated with the pressure tap) radiates at sound pressure levels ranging from 70 to 93 dB. A numerical factor yet to be evaluated probably reduces these levels by up to 10 dB. It is significant that the radiated sound is correlated with the hydrodynamic pressure measured at the center of the flap surface thus casting some doubt on the often stated proposal that sound radiation takes place only at the trailing edge of the flap surface. Estimating the effects of all eddies radiating from the flap structure yields a total sound pressure level ranging from 80 to 90 dB. Such a plane flying overheat at an altitude of 500 ft would give rise to sound pressure levels ranging from 55 to 65 dB for the speeds encountered in the present work. It is interesting to note that the sound pressure levels measured from the center region of the flap are generally 20 dB below the tunnel background noise level (100–113 dB), hopefully establishing the usefulness of cross-correlation techniques within a wind tunnel.
OTW noise correlation for several nozzle/wing geometries using a 5:1 slot nozzle with various external reflectors
Authors: von Glahn U. , Groesbeck D.
Published: October 2005

The aerodynamic design requirements of the nozzle boattail for OTW configurations are different for cruise and powered lift. For cruise, a small nozzle boattail angle (<10°) is required to minimize drag; while for powered lift, a large angle (25°–30°) is needed to provide jet flow attachment to the lifting surfaces. A solution to these design requirements is to use an external flow-deflector downstream of the nozzle for powered lift and to store (retract) this deflector for cruise. For powered lift, however, the jet/deflector/wing noise transmitted to ground observers can be a problem. In the present paper, representative acoustic spectral data obtained from a model scale study of several OTW configurations with a 5:1 slot nozzle using various external deflectors are correlated in terms of deflector geometry and flow parameters. Variations in the deflector geometry include deflector size and deflector angle. In addition, geometry variations in flap setting and nozzle chordwise location are included. Three dominant noise sources are correlated: fluctuating lift noise, flap-trailing edge and deflector related noise, and jet mixing noise. Lift and thrust measurements obtained in the study for the various configurations are summarized.
Flyover noise reduction for general aviation aircraft
Authors: Campanella Angelo J.
Published: October 2005

A survey of noise emission reduction techniques is presented for propeller driven aircraft. Most of the audible noise is generated by the propeller tip traveling with a local velocity near that of the speed of sound. The feasibility of mechanical design steps (larger geared propeller and longer landing gear) required to reduce this speed was investigated. It was found that only new aircraft designs can be so configured. Existing aircraft will continue to operate. The attrition rate of these existing aircraft (1%–3% per year) is small, so that certain operational methods are desirable to achieve quieter airport environs. Procedures of administrative control, operational rules, a noise abatement power adjustments are described which decrease the noise exposure experienced on the ground. Enlightened land planning by county and state officials is described. [Work supported by FAA.]
Analog to Curle's sound formula for closed cavities
Authors: Burton T. E.
Published: October 2005

Although S. N. Curle's very useful formula for aerodynamic sound radiated from a rigid surface can be applied to closed fluid circuits by the method of images, this procedure is not practical for cavities of complex shape. Sound fields in such cavities are best described with standing waves. However, substitution of an orthonormal mode expansion into the inhomogeneous wave equation results in volume integrals of dilatation source terms for the generalized model loads [N. A. Cumptsy and D. S. Whitehead, J. Sound Vib. 18, 353–369 (1971)]. Since the dilatation field is usually not known, the formal solution is difficult to apply. This paper presents an expansion of the sound field in a cavity in terms of nonorthogonal functions which results in a coupled set of equations with the generalized loads in terms of surface pressures (as in Curle's derivation), plus quadrupole terms. When the equations can be conveniently uncoupled and the quadrupole terms neglected, this formalism is very useful. Some of the cavity and flow geometries for which these two simplifications can be made and discussed.
Analytical and experimental studies of folded cavity duct acoustic liners
Authors: Beckemeyer R. J. , Sawdy D. T.
Published: October 2005

Low-frequency noise suppression in flow ducts can be achieved without excessive acoustic liner backing depth if the backing depth if the backing volume is formed as a long shallow cavity, folded so as to lie along the duct axis. Initial studies of such configurations revealed that sound transmission characteristics could not be accurately predicted when locally reacting impedance models were used. Consequently, an extended reaction model was developed for folded cavity liners by use of eigenfunction expansion techniques. Accuracy of the model has been verified by comparing analytical and experimental results for liners mounted in a 6-in.-square flow duct test facility. Several face sheet impedances and cavity geometries were tested at Mach numbers of 0.0 and 0.2 and frequencies from 500 to 1000 Hz. Good agreement was obtained between theoretical and experimental values of panel insertion loss as well as between analytical and measured duct centerline axial standing wave patterns.
Experiments on the intelligibility of speech codes for the severely hearing impaired
Authors: Stewart S. E. , Strong W. J. , Palmer E. P.
Published: October 2005

We have reported previously (San Francisco ASA meeting in November 1975) on the intelligibility and discriminability of a low-frequency formant speech does with both normal hearing and hearing impaired subjects. Since then we have endeavored to discover just how much degradation the speech suffers in the process of coding and at what stage of the coding it occurs. Using Form IV of the Diagnostic Rhyme Test as test material, we have presented a modified speech code (dealing with partials of the speech rather than formants as such) at various stages of coding to normal hearing subjects. (This was done to assess the adequacy of the code and to initially avoid training hearing impaired subjects.) The test is used in the usual “intelligibility” mode in which the subject hears a single stimulus and is asked to identify it as one member of a word pair for the following conditions: (1) speech directly as recorded, (2) speech low-pass filtered (4.5 kHz) followed by analog-to-digital and digital-to-analog conversions, (3) speech synthesized from largest amplitude partials. In addition, the test is run in a “discrimination” mode in which the subject hears a pair of rhyming stimuli and is asked to judge whether they are the same or different for the above conditions and for conditions after low-frequency transposing by a factor of four to one. Work is in progress with hearing impaired subjects.
The relationship between speech discrimination ability and self-assessed hearing-handicap of adults with sensorineural hearing losses as a function of reverberation and noise
Authors: Mason D. , Asp C. W.
Published: October 2005

The relationships between the Modified Rhyme Test which was administered in four listening situations—quiet and +5-dB signal-to-noise ratio with reverberation times of 0.2 and 0.5 sec—and hearing handicap as determined by the Self-Assessed Hearing-Handicap Scale was investigated in a sample of sensorineural hearing impaired subjects. Analysis of the data revealed that there was a significant correlation between self-assessed hearing handicap and MRT scores, but only when the test words were presented in a +5-dB signal-to-noise ratio with a reverberation time of 0.2 sec. Changes in correlation coefficients were explained by the effects of noise and reverberation on speech-discrimination scores as a function of the severity of hearing handicap. Noise affected the speech-discrimination scores of the more severely handicapped subjects to a greater degree than the less severely handicapped subjects. The opposite was true when reverberation time was lengthened.
Tactual speech reception using a cochlear model
Authors: Sparks D. W. , Kuhl P. K. , Edmonds A. E.
Published: October 2005

A tactual speech aid employing an electrical analogue of the cochlea is being evaluated on normal and hearing-impaired listeners. The cochlea has 36 frequency channels spaced logarithmically between 90 Hz and 10 kHz. The power envelope is extracted from each channel and submitted to a processor where it is transformed into eight discrete intensity steps. The outputs from this processor, then, are displayed as an intensity by frequency matrix consisting of 288 points (8 intensity × 36 frequency). This spectral “plot” changes, as the speech spectrum, in real time. The matrix itself consists of 288 silver electrodes and is applied to the abdomen. The electrocutaneous stimulation scheme employs biphasic pulses for zero net current flow; the stimulators are constant-current limited for safety and comfort. Vowel, consonant, and connected discourse recognition are compared in three modes; (1) tactual alone, (2) visual alone, and (3) tactual with visual. Results are discussed in relation to data obtained from other tactual speech aids.
Analysis of speech produced with an artificial larynx
Authors: Rothman H. B. , Goldstein L. P.
Published: October 2005

For those individuals unable to learn esophageal speech and for the period of time until adequate esophageal speech is developed, the successful use of an artificial larynx is of primary importance. The purpose of this paper is to present the results of an investigation which measured and analyzed the speech parameters of (1) rate, (2) duration, (3) pitch, (4) intensity, and (5) articulation of both good and poor artificial larynx users. In addition to the analysis of the above parameters, perceptual judgments of speech intelligibility and proficiency were obtained. The conclusions reached from this investigation have resulted in the establishment of criteria for training laryngectomized individuals in the use of an artificial larynx. The material presented can also be used as a diagnostic basis to determine why an individual is not functionally intelligible us using the artificial larynx.
Multidimensional classification of abnormal voices
Authors: Singh Sadanand , Murry Thomas , Sargent Margaret
Published: October 2005

Twenty samples of voice disordered speakers were randomly selected from a library of 200 tape recordings to determine their perceptual attributes. Twenty-three subjects judged the similarity of the ordered pairs of these voices on a seven-point scale. One of the three groups of seven judges rated the hoarseness, breathiness, and nasality. The judged similarities were analyzed by INDSCAL. The physical measures of airflow, F0, F1, F2, F2/F1, F2−F1, and the acoustic spectrum and the psychophysical measures of hoarseness, breathiness, and nasality were used to interpret a five-dimensional INDSCAL solution. Dimension 1 was explained by absence versus presence of periodicity and by judgments of “hoarseness.” Dimension 2 was explained acoustically by elevation versus nonelevation of F2, and diagnostically by presence versus absence of mass lesion. Dimension 3 was explained physiologically by airflow rate and psychologically as “breathiness” and to a lesser extent, as “nasality.” Dimension 5 was interpreted acoustically in terms of speakers' F0. The five-dimensional solution accounted for 50% of the variance in the data.
Linguistic processing and control strategy in the HWIM speech understanding system
Authors: Woods William A. , Bates Madeleine , Brown Geoffrey , Wolf Jared J.
Published: October 2005

The principal source of higher-level linguistic knowledge in HWIM, the BBN speech understanding system, is an augmented transition network parser, which embodies the syntactic, semantic, and part of the factual sources of knowledge of the system. It parses sentences or sentence fragments in either direction, and it can, for a sentence fragment, enumerate the words and syntactic/semantic classes permissible at the ends of the fragment. The control component of the system is a program that calls on the other sources of knowledge of the system in order to formulate, evaluate, and extend hypotheses about the interpretation of the utterance. It is responsible for guiding the system to the most likely interpretation as efficiently as possible. [This research was supported by the Advanced Research Projects Agency of the Department of Defense and was monitored by ONR under Contract No. N00014-75-C-0053.]
Use of intonational phrase boundaries to select syntactic hypotheses in a speech understanding system
Authors: Lea Wayne A.
Published: October 2005

A procedure has been developed for using prosodically detected phrase boundaries to weight word and phrase hypotheses in the Bolt Beranek and Newman (BBN) SPEECHLIS speech understanding system, so that correct words and structural hypotheses will be proposed at earlier stages in parsing, and erroneous theories can be avoided. The state-transition arcs of the augmented transition network grammar were specially marked if they were expected to be immediately preceded by intonationally detected phrase boundaries. The scores on words associated with the arcs were increased if expected boundaries were detected, or decreased if expected boundaries were missing in the acoustic-prosodic darn. Fifteen BBN sentences were processed through a computer program that detected phrase boundaries at fall-rise valleys in fundamental frequency contours. Analysis of simple traces of the hypothesizing, testing, and constructing of syntactic structures by the SPEECHLIS system showed that prosodic adjustment of scores would increase the likelihood of correct words and phrases being selected before incorrect ones. These ideas are being refined and tested further, for implementation in the SPEECHLIS system.
Labeling speech events for acoustic and linguistic processing
Authors: Hanson R. J. , Pfeifer L. L.
Published: October 2005

In speech studies, data base access has been significantly improved through the use of label files which relate portions of the speech waveform with a phonetic (or phonemic) transcription as well as other linguistic and nonlinguistic information. In this way the acoustic and linguistic “realities” of speech can be associated and checked against each other since they were derived independently. Because of the lack of a simple correspondence between the acoustic structure of a portion of the speech waveform and the linguistic label assigned, additional principles must be invoked to make labels useful. As an example of the labeling process and the accompanying problems, a sound identification study is described in which 675 vowel nuclei were labeled using the transcription derived from a sophisticated multitranscriber system. Despite precautions taken to insure correct labeling, many of the vowels are judged in a pattern recognition experiment to be more similar to neighboring vowels in the vowel space or to schwa than the vowel they were labeled as. These supposedly misidentified vowels were found to have acoustic structures very different from the ideal for their label, at least partially due to the loss of coarticulatory and vowel reduction information when assigning transcription symbols. Directions for resolving these problems and for increasing the utility of labeled speech events in acoustic and linguistic processing are discussed. [Work supported by AFOSR Contract No. 44620-74-C-0034.]
An algorithm for speaker verification
Authors: Shridhar M. , Vidalon M.
Published: October 2005

An analysis of the spectral properties of male speech reveals that most of the spectral energy is distributed in the frequency range of 50–2000 Hz. The authors attempted to develop a computer algorithm for speaker verification that utilized parameters extracted directly from the digitized speech. The original speech signal was sampled at 4 kHz after filtering it with a 2-kHz low-pass filter. The investigations reveal that by the use of low-order linear predictor model a feasible set of parameters could be realized for application to speaker verification. A simple warping procedure modifies the parameter contour of the unknown speaker so that the correlation with the reference contour is maximized. The verification decision was based on the distance of the test sample contour from the reference contour for the claimed speaker. If the distance was less than a fixed threshold, the speaker was accepted. Among all the parameters investigated, the reflection coefficients and the autocorrelation coefficients were found to be the most effective, providing a verification accuracy of 98% for speech, 2 sec in duration, which increased to more than 98% for a duration of 3 sec. In conclusion, a procedure for speaker verification has been developed and it is fairly simple, reasonably fast, and reliably accurate.
Diagnostics of machinery by acoustic source location
Authors: Schulz M. W.
Published: October 2005

Signature Analysis has long been used to detect faults in a wide variety of machines. Recent efforts to locate faults by acoustic methods is illustrated by the location of corona in large power transformers. Excessive corona can damage insulation and require costly shutdown and repair. Corona generates both high-velocity electrical signals and ultrasonic signals which travel at the velocity of sound in the insulating oil. The acoustic signals are sensed by accelerometers attached to the exterior of the transformer tank. By triggering a signal average from the electrical signals, measurements of the acoustic travel time from the corona source to several transducers can be obtained. In simple cases these can be transformed to distance, and triangulation calculations yield a corona source location. More generally, the acoustic waves encounter obstacles, are reflected, and the triangulation techniques fail. A new technique called Space Interrogation systematically assumes a series of allowed corona locations and calculates the mean-square fit of the data to each assumed source location and set of permitted acoustic paths. The result is a more powerful means of locating corona sources.
High-frequency acoustic signals for condition monitoring
Authors: Darlow M. , Shinaishin O. , Winn L.
Published: October 2005

Acoustic signals. structure borne, have long been used to detect the integrity of various machinery. This presentation discusses the state of the art of using the ultrasound (high frequencies beyond the audible range) of machine elements and structures to detect failure or malfunction conditions. The paper will describe a recently developed technique, the High-Frequency Resonance Technique (HFRT), its principles, and results of applications in detecting rotating machine element failure especially in bearing inspection, as well as its potential for extensive machinery condition monitoring. A brief discussion of the use of fiber optics in the monitoring of vibrations and detection of flaws in rolling element bearing will be presented. This relatively new technique shows promise for fault detection in the areas of inspection and vibration measurement in general. The paper will also discuss the use of high frequency signal detection of pulses generated in structures failing under dynamic loads and the potential of the method for nondestructive testing of various structures in important applications.
Development of an on-board seismic derailment detector
Authors: Wassmann William W.
Published: October 2005

The Naval Surface Weapons Center/White Oak Laboratory, after conducting a feasibility study in 1972/73, has embarked on a two year program for the Federal Railroad Administration's Office of Research and Development to develop an on train system for train accident reduction (STAR). It will be installed for test on a 124-car unit ore train of the Duluth, Missabe and Iron Range (DM&IR) Railway Company. The system includes a seismic wheel-on-the-ground detector which senses an abnormal vertical motion of the unsprung mass of a freight car. The sensor then initiates a sequence of events which causes the train to stop. This paper describes the development of the seismic sensor: the initial feasibility study, establishment of design criteria, fabrication, installation, and evaluation of the sensors Initial calculations of a freight car wheel-drop trajectory suggest that there may be a sufficient velocity change above an acceleration threshold to make a discriminating spring-mass system feasible. Actual full-scale derailment tests and over-the-road vibration measurements support this theory. Sensors have been built and tested during actual freight train derailments and are currently in regular service on the DM&IR between the loading plant and ore docks to obtain false-alarm data.
Special Session in Honor of Hallowell Davis
Authors: Galambos Robert
Published: October 2005

It has been known for many years that sound stimuli can modify the electrical activity of the brain. The possibility of developing useful clinical tests of hearing based upon this fact has also long been recognized. Efforts to create such tests, however, have until recently yielded relatively unimpressive results. The developments of the past five years are another matter, and this session will review them for interested psychologists and physiologists, and for specialists in instrumentation related to audiometry. New information on brain events time locked to stimulus presentation—the so-called evoked responses—will be emphasized, along with the recording methods presently used to visualize them. Speakers will describe the way these evoked potentials are being used in the testing of hard-of-hearing patients. By emphasizing certain limitations of the methods currently employed they will also point to the future developments by which electric response audiometry may become an even more powerful clinical tool.
Panel discussion
Authors: Davis Hallowell , Galambos Robert , Moushegian George
Published: October 2005

The Chairman will describe the auditory-evoked response as this is recorded through scalp electrodes from normal subjects in the laboratory. It consists of some 15 successive waves varying in latency from about 1.5–300 msec poststimulus. Hallowell Davis will then present the principles of electric response audiometry and show how the so-called late, waves (latency 100 msec and later) can be used to test hearing in adults and children. The way the early waves (latency 1.5–10 msec), which are generated by the auditory nerve and brainstem nuclei. aid diagnosis of hearing problems in infants and other difficult-to-test patients will be described by Galambos. Still another electrical event, the frequency following potential, which also is generated within the brainstem, will be dealt with by George Moushegian, who will provide a description of the phenomenon and show examples of its clinical utility. Following this review of the present status of ERA the panel will discuss its limitations. The question of whether hearing at frequencies below 2000 Hz is testable using the brainstem response will receive particular attention.
Comparison of gross neural waveforms recorded from various electrode sites in the guinea pig
Authors: Mitchell Curt , Fowler Cynthia G.
Published: October 2005

Different waveforms of auditory evoked potentials were found when recording from locations in the cochlea, middle ear, and vertex of the skull in the guinea pig. The first negative potential N1 maintained its latency and waveform from a variety of sites, However, the N2 potential changed its peak latency and waveform considerably with location. At certain electrode sites, double-peaked N2 waveforms were found. The data suggest that the N2 potential is a complex potential reflecting activity from more than one neural population. Thus, N2 is composed of repetitive firing of cochlear afferents [Özdamar, Ph.D. dissertation, Northwestern University (1976)] and neurons in the cochlear nucleus or brainstem. The data do not suggest the apical portion of the cochlea as an origin of N2. The N2 potential was found to be most prominent from the vertex; and N1, N2, and N3 appeared at nearly the same amplitudes from certain positions. The potentials maintained their characteristic wave shapes over a range of intensities. [Supported by grants from the NINCDS.]
Bilateral brainstem responses to monotic click
Authors: Zerlin S. , Naunton R. F.
Published: October 2005

Brainstem-evoked responses (BER's) from both sides of the human head were simultaneously recorded by means of a vertex electrode with separate left and right earlobe references. Monotic clicks were presented via earphone at 75-dB sensation level and the BER's recorded bilaterally. Latencies of corresponding peaks of the BER from stimulated and unstimulated sides were compared, with these results: wave III, with latency of about 3.5 msec was always somewhat delayed on the stimulated side; waves IV–VI, with latencies in the 4.5–7.5-msec region, appeared simultaneously on the two sides. Studies on cat have associated wave III with the crossed neurons of the superior olivary complex and successive waves with other of the nuclei in the ascending auditory pathway. [J. S. Buchwald and C-M Huang, Science 189, (1975).] The present data support their finding that wave III is generated at the superior olivary complex, because of the latency effects manifested by wave III. [Work supported by NIH.]
Brain response to intracranial auditory motion
Authors: Ruhm H. B.
Published: October 2005

Responses, recorded from scalp electrodes on human subjects were evoked by the onset of three types of modulation of a 100-Hz click train—AM, FM, and time difference modulation (TDM). The last consisted of intracranial sweeps generated by successive increments in the interaural temporal difference between members of dichotic click pairs that composed the train. Slope and extent of TDM were independently varied. All types of modulation produced similar slow-wave responses. The direction of intracranial motion was reflected in the N1-P2 amplitude of the response to TDM in that, lateral-moving sweeps evoked larger response than medial-moving sweeps. Response latency decreased and amplitude increased with increasing TDM slope up to 5.55 msec/sec with little change tereafter. Amplitude also increased with increasing extent of sweep up to 0.55 msec. interaural time difference. Extent of sweep did not systematically affect the latencies of N1 or P2. No right-left differences were observed, [Work supported by Veterans Administration.]
Cross-sensor beam forming with sparse arrays
Authors: Bucker H. P.
Published: October 2005

The quadrature field of a narrow-band signal measured at a sensor, or sensor group, in an array is a complex number that rotates with time. Thus time averages of the quadrature field tend toward zero. However, elements of the cross-sensor field, i.e., the quadrature field at one sensor multiplied by the complex conjugate of the quadrature field at a second sensor does not rotate and can build up with time. By using the cross-sensor field, a beam pattern (for a one-wave, no-noise acoustic field) can be generated for a spare array which is the same as for a filled array with the same aperture. When real world effects, such as noise and multiwave acoustic fields, are considered the performance of the sparse array degrades more than the performance of the filled array. However, by time averaging the cross-sensor field the performance of sparse array is greatly improved. Calculations for a 23-element sparse line array in a realistic simulated acoustic field show significantly improved performance when compared to a 23-element uniform array.
Nearfield Fourier transform imaging
Authors: Murphy Donald A.
Published: October 2005

The two-dimensional Fourier transform provides an efficient method of forming beams for a line array. The field as sampled by the evenly spaced array is the Fourier transform of the farfield source. Its resolution is limited by the receiving aperture and the signal-to-noise ratio at the array elements. When the field is stationary, the resolution can be increased by integrating the Fourier transforms magnitude squared. If the field is also homogeneous the resolution can be increased by extending the aperture without adding additional elements by the method described by Nuttall et al. This method constructs the two-dimensional spectrum of objects in the farfield without grating lobes from the cross spectra of elements of the array. The array spacings must all be multiples of a basic spacing in order to make the calculation efficient. The method has been extended to apply to objects in the nearfield as well, and allows estimation of noise measured in the nearfield with sparce arrays with an efficient computer program.
Parametric receiving array beam patterns using a phase demodulator
Authors: Beard J. K. , Truchard J. J.
Published: October 2005

In this paper, closed form solutions are derived for several configurations of the parametric receiving array with a phase demodulator receiver. The expression for parametric nonlinear acoustic phase modulation of a high-frequency spherical wave from a point source by a low-frequency plane wave is calculated. The result is in turn used to generate the solution for parametric receiving arrays with various pump transducers, including a truncated line source, a rectangular piston, and a circular piston. The solution of the parametric receiving array and a truncated line receiver is also found. A series of experiments was conducted with a 48-ft parametric receiving array at a pump frequency of 90 kHz. These experiments duplicated several of the geometries used to obtain the theoretical results. Good agreement between theory and experiment was achieved. [Work supported by ONR]
Spectral estimation techniques with special reference to sonar signal processing
Authors: Arcese A. , Bergeron L. E.
Published: October 2005

In this talk we present a comparative evaluation of several spectral estimation techniques. The estimation procedures investigated are the FFT and autoregressive modeling of the time domain waveform. For the autoregressive modeling the estimation procedures considered are the Durbin recursion, Burg recursion, Kalman filtering, and a gradient technique. We also present a new maximum entropy gradient spectral estimation procedure. The results of applying these spectral estimators to real submarine target data are presented.
Signal excess and detection probability of fluctuating sonar signals in noise
Authors: Urick R. J.
Published: October 2005

Signal excess is defined as the excess or deficiency of signal level relative to that required for a detection probability of 50%; curves of detection probability versus signal excess are sometimes called “transition curves.” In this paper, transition curves for a nonfluctuating signal in Gaussian noise and for Gaussian, Rayleigh, and log—normal signal fluctuations are obtained. A comparison with one set of detection data obtained in a sea test with a towed sound source indicates that log—normal signal excess fluctuations with a standard deviation of 6–8 dB give a reasonable fit to the data. The method suggested here—of using transition curves along with a known or assumed relation between transmission loss versus range—permits quicker computation of detection probability against range than does the conventional way of employing ROC curves, and can accommodate a signal excess that fluctuates with time about a mean value. [Work supported by Naval Surface Weapons Center and ASW Systems Project Office, Code ASW-13.]
Generalized performance
Authors: Hodgkiss W. S.
Published: October 2005

The performance of a Bayes optimal detector is summarized by its ROC curve. In the general ease where uncertain parameters exist under each hypothesis, knowledge about them is explicitly noted at the outset by an a priori probability density function conditional to H1 and one conditional to H0. The processor's performance then becomes a function of their detailed shape. Often, the functional form of these densities is chosen so that various levels of uncertainty are easily modeled and a family of ROC's is reported. The question then arises: What performance would have been achieved under different prior knowledge assumptions (particularly when outside the class of densities modeled)? Or, more deeply: Does some algorithm exist which will operate on a known ROC for a given pair of priors to yield the ROC for a new set of priors? And, if not: Does a canonical intermediate step exist between observation and likelihood ratio statistics which always may be used as a starting point for the calculation of an ROC based on an arbitrary pair of priors? The intent of this paper is to pursue these questions. The discussion will use as a basis the fundamental concepts of sufficient statistics and reproducing densities. A specific example is presented. [Work supported by ONR.]
Contributions of acoustic echo sounding to atmospheric research and services
Authors: Little C. Gordon
Published: October 2005

This paper reviews briefly the first few years of acoustic remote sensing that followed the introduction of the echosonde in 1966, and then surveys in more detail many aspects of recent developments in acoustic echo sounding. The early, qualitative, facsimile recordings have now been supplemented by attempts to make the echosonde data quantitative. This has involved careful comparison of data from calibrated sounders and instrumented meteorlogical towers. Evidence is presented supporting the claim that echosondes can supplement the facsimile records with height profiles of mean wind speed and direction, as well as profiles of the structure constant of velocity Cv and temperature CT. The former permits derivation of the height profiles of the turbulent energy dissipation rate ε and the latter the measurement of height profiles of the structure constant of optical refractive index Cn. Various realizations of Doppler echosondes are presented, together with an echosonde interferometer which may offer promise of measuring atmospheric temperature profiles. Some future directions for acoustic echo-sounding developments are summarized.
Acoustic sounding of atmospheric winds
Authors: Jensen D. R. , Richter J. H.
Published: October 2005

Acoustic echo sounders have recently come into their own as valuable remote sensors for determining atmospheric structures and winds. Kelton and Bricout [Bull. Am. Meteor. Soc. 45, 571–580 (1964)] first demonstrated that the acoustic Doppler shifted signals from scattered sound were related to the ambient winds. Later, McAllister [J. Atmos. Terr. Phys. 30, 1439–1440 (1968)] showed that a vertically pointing monostatic acoustic sounder could reveal, in intricate detail, the atmospheric boundary layer structure. An acoustic echo sounder and wind sensor system capable of simultaneously measuring temperature fluctuations in the lower atmosphere and the vertical profile of wind speed and direction has been in operation at the Naval Electronics Laboratory Center since 1973. It is the purpose of this paper to describe the capability of this acoustic echo and wind sensor system for determining atmospheric structure and winds.
Acoustic wind and wind-shear measuring system
Authors: Mandics P. A. , Beran D. W.
Published: October 2005

An acoustic wind-profiling system designed to detect hazardous wind-shear conditions in the airport environment has been developed during the past four years. The system installed at Dulles International Airport consists of a vertically pointed transmitter surrounded by three receivers 290-m distant and separated by 120° in azimuth. Electronically steered receiver beams track the upward propagating transmitted tone burst and collect the scattered acoustic signals. The Doppler frequency shift of the returns is analyzed digitally to determine the horizontal wind at 20 height levels in 30-m increments. Unique design features of the system such as the steered receiver antenna are described. A one-leg prototype of the Dulles system was installed and tested at Table Mountain near Boulder, CO. Winds measured by the prototype acoustic system compared well with those determined by an FM-CW radar and a balloon-borne anemometer. Noise generated by rain and surface winds exceeding 16 m sec−1 proved to be the major limitations for the acoustic system. Preliminary results from the Dulles system are also presented.
Confidence limits for measured supertanker-radiated noise levels
Authors: Cybulski J. , Wright E. B.
Published: October 2005

Sound pressure spectral levels were obtained for the supertanker class of ships using a developing technique of aircraft deployed Sonobuoys with a deep hydrophone at 305-m depth. These measurements were made in the deep oceanic waters over the Seine, Tagus, Iberian and Biscay abyssal plains. The received levels at individual omnidirectional sensors were processed to yield ship radiated noise in 0.2-Hz bandwidths over the 5–80-Hz band and transformed to sound pressure levels in dB ‖ μPa at 1 m. Confidence limits that are constructed for these values reflect the effects of navigation, calibration, geometry reconstruction. CPA determination and a transmission loss correction associated with the range between ship and sonobuoy sensor. The variation of these limits is illustrated as a function of CPA, aspect angle and frequency with standard errors as high as 3 dB. The primary sources of error are hydrophone/system calibration, CPA determination and the transmission loss values. The FACT acoustic-environmental model was employed to obtain the frequency-range dependence corrections. Directions toward increasing the confidence of these measurements are indicated. [Work sponsored by Naval Electronics Systems Command. Code 320.]
Ambient sea noise directionality: Measurement and processing
Authors: Yen N.
Published: October 2005

The response of a receiving hydrophone array in a noise field is analyzed for the relationship between the array parameters and the directionality of the noise field. The base functions formulated by array element spacing are considered as the coordinates of a multidimensional space. By orthonormalization of these base functions, the cross-spectra between hydrophone outputs are transformed directly to noise field directionality. Applying the same method to the array's beamforming output also gives another noise field estimate. When the number of independent measurements from the array's beamforming output is the same as that of the cross spectrum, the noise fields estimated by either methods are equivalent and depend on the array's structure. For stationary noise fields with low variability in the measured data, a further improvement in the angular resolution of the noise field can be achieved by deriving a new set of orthogonal base functions through analytic continuation. Interferences caused by uncorrelated local noise together with limits in the precision of numerical computation are shown to impose a practical limit on the accuracy of the estimated noise field. Thie processing method for noise directionality is illustrated by application to a simulated noise field and sea data acquired from a vertical line array. [Work supported by NAVSEA.]
Stochastic modeling of community noise levels
Authors: Freytag J. C. , Brittain F. H.
Published: October 2005

A modeling technique is proposed for combining two or more noise histograms (i. e., noise level versus duration histories) into a new histogram. By considering the time varying nature of independent noise sources, much more accurate community noise measures may be computed. Logarithmic addition of statistical measures (e. g., Leq or L50) invariably overestimates the true result. The method is particularly useful in modeling a multisource environment such as a construction site where the duty cycles of independently operating equipment are accurately known. The analytical procedure is presented with a construction noise example.
Truck noise emission reductions and/or roadside barriers; the effects of each on population impact from motor vehicle noise
Authors: Manns Basil H. , Peppin Richard J.
Published: October 2005

The Revised Design Guide (NCHRP 3-7/3) noise prediction program was modified to include reduced source noise levels. Medium and heavy truck noise levels were reduced in steps of 2 dBA. The sound levels for receivers in two scenarios were calculated using roadside barrier heights, which varied from 0–21-ft. These scenarios included a flat, open area and an area which included two rows of single-story houses. Population impact, varying source level, scenario type, and barrier height was evaluated using a modified form of the method in the Revised Design Guide. This modified method allows the sound level to vary linearly with the logarithm of distance instead of the stepwise variation that is used now. Results presented show the effect on “impacted” population by reducing emitted noise levels of trucks and erecting roadside barriers.
New vehicle certification measurement for automobiles and light trucks
Authors: Schomer Paul D.
Published: October 2005

Current certification measurement procedures are designed to measure the maximum noise that a vehicle can produce. Studies by GM and others have shown that vehicles are rarely operated in their maximum noise producing mode. More typical are operations such as urban acceleration (normal acceleration from a stop light, etc.), cruise and idle. To provide more meaningful new vehicle certification procedure and standard for the State of Illinois, an urban acceleration measurement procedure has been tested and developed. The procedure is based upon methods proposed by Hillquist and can currently be applied only to vehicles having an automatic transmission. The procedure is simple to perform, fairly short in duration and the results are quite repeatable. This paper deals primarily with the experiments used to develop the procedure and the resulting recommended practice.
Development of a motorcycle exhaust system certification program
Authors: Walsh John B. , Hagie Roger , Harrison Robin T.
Published: October 2005

Excessive noise from modified motorcycles annoys many citizens and has damaged the reputation of the motorcycle industry. Enforcement action against excessively noisy motorcycles is necessary, and an exhaust system certification program could be used as a tool to reduce motorcycle noise impact. The Motorcycle Industry Council, representing most motorcycle and motorcycle accessory manufacturers, recognized the need for control of noisy motorcycle exhaust systems, and initiated development of an exhaust system certification program for motorcycles. The program was to provide the maximum amount of noise control possible through an enforcement program of low to medium intensity. The hope is that the program will help remove the excessively noisy exhaust systems from the market with minium disruption to responsible motorcycle and after market manufacturers. The manufacturers are taking steps to provide quieter motorcyles and exhaust systems. By working with enforcement agencies concerned with effectively controlling motorcycle noise, the industry hopes to get noisey bikes off of the streets and to finally reduce adverse noise impact from motorcycles.
Interior and exterior diesel locomotive noise measurements
Authors: Campanella Angelo J.
Published: October 2005

Diesel locomotive noise data accumulated over the past two years for hearing-damage assessment and land-use analysis are presented. Interior locomotive noise levels of the cab and engine areas were measured while the engine was developing all power levels was measured. Cab noise levels varied from 88 to 91 dBA at full power, depending on engine type, when the windows and doors were closed. Engine areas noise was found to be between 104 and 124 dBA at full power, depending on engine enclosure design. Extenal noise emissions were measured along an active east-west pair of tracks to determine land use classification for residences. This site typified enroute noise emission on flat agricultural terrain. Numerous observations were plotted versus distance from the tracks to derive an enroute emission model. Land use disposition was determined for future residences.