bluuebunny/crossref_metadata_2025 huggingface dataset:
The development of a local micro-computer network for an acoustics laboratory
Authors: Kerivan J. E. , Holmer C. I.
Published: October 2005

Local computer networks (LCN's) lend themselves as a cost-efficient, growth-oriented, powerful answer to the problem of real-time control and processing of acoustic signals in a laboratory. The availability of inexpensive scientific and engineering “off-the-shelf” software which can be structurally integrated in a shared resource environment permits the realization of large data-base management systems with micro-computers. The comparison of LCN alternatives, both hardware and software options, as it relates to specific acoustic measurement problems will be contrasted in showing E-A-R's solution for a mini/micro-computer network. Modal analysis, reverberation-time measurements, damping data reduction, hearing protection attenuation, and analytical modeling are some of the tasks which are integrated in the system. In this presentation we will discuss in detail some of the alternatives considered to date in our system development and their perceived advantages and limitations. Listings of structured programs currently under development will be provided.
Computers in acoustics education at Lehigh
Authors: Ochs John B.
Published: October 2005

In the late 1970's the Industrial Engineering Department and the Mechanical Engineering Department of Lehigh University joined forces to develop and promote an integrated approach to the progression from mechanical design to manufacturing with computer graphics systems providing the man/machine interface. The primary focus of the CAD/CAM program is the education of the approximately 600 undergraduates in the two departments with the emphasis on the design and manufacturability of products so that an engineer can understand the ramifications of his design decision in an integrated CAD/CAM environment. The aim of the program is to provide state-of-the-art computer graphics systems as tools to improve design and manufacturing innovation. Educationally the goal is to implement the computer graphics technology into every undergraduate course in the two departments, which include such diverse topics as statics/dynamics, thermodynamics, mechanical vibration, statistics, operation research as well as the classical design and manufacturing. Some courses stress extensive use of these computer systems and the means to manage them, while others use the facilities for projects or homework assignments. By hard wiring the CAD and CAM laboratories to classrooms, the color real-time dynamic displays can be used as dynamic blackboards to visualize complex three-dimensional geometry or to illustrate multi-dimensional mathematical relationships. Growing from a strong undergraduate base the program has attracted graduate research from several engineering disciplines supported mainly by Lehigh's industrial partners.
Approaches for digital processing acoustic and vibration data in a teaching laboratory
Authors: Sachse Wolfgang , Cochran R. , Yu S. , Chen C. P. , Chang C.
Published: October 2005

In this paper we describe the signal acquisition and processing approaches we have implemented using a portable mini-computer system in laboratory experiments of acoustics and vibrations courses. Also described will be examples of programs we have developed to data which include the simulation of 1- and 2-degree-of-freedom vibrating systems, the analysis of displacement, velocity, or acceleration signals of such systems; the digital processing of speech and other transient acoustic signals to obtain spectrograms; the determination of reverberation time; two-dimensional ray tracing in homogeneous and inhomogeneous layered media; and a general data acquisition, processing, and display program in which any one of more than 70 commands can be called with a three-letter mnemonic. Key to all the examples is the versatility of the system and the ease with which new experiments or examples can be implemented by inexperienced users.
Intermittence and the total-energy hypothesis
Authors: Ward W. D. , Turner C. W. , Fabry D. A.
Published: October 2005

The theory that auditory damage is proportional to the total A-weighted acoustic energy that has entered the ear has been shown to hold for single uninterrupted exposures to steady 700–2800-Hz noise in the chinchilla, provided that a critical level of 110–115 dBA is not exceeded. The theory is not correct, however, if any appreciable interruptions in exposure occur. Earlier research [W. D. Ward, T. Kiester, and C. W. Turner, J. Acoust. Soc. Am. Suppl. 1 66, S61 (1975)] had established that breaking up a 220-min exposure at 114 dB SPL into 22 10-min exposures, two per week, reduced both permanent threshold shift (PTS) and cochlear damage (missing outer hair cells, MOHC) by an amount equivalent to a reduction of at least 7 dB in level. Present results primarily compare the damage produced by a 15-day continuous exposure at 92 dB SPL to that generated by nine weeks of “work-week” exposure (8 h/day Monday through Friday). The average PTS at 1, 2, 4, and 8 kHz was 11 dB for the continuous exposure, 4 dB for the work-week exposure, and the MOHC count dropped from 640 to 340. The work-week values were similar to those observed in a group exposed continuously for 15 days to 85 dB SPL, indicating again a “saving” equivalent to a 7-dB reduction, though this value is surely coincidental. Implications for exposure criteria will be discussed. [Work supported by NIH grant NS 12125.]
Alterations in the auditory brainstem response produced using a forward-masking paradigm
Authors: Ananthanarayan A. K. , Gerken George M.
Published: October 2005

The post-stimulatory effects of a masker on the probe-evoked auditory brainstem response (ABR) were evaluated in 18 normal-hearing subjects using a tone-on-tone forward-masking paradigm. The masker consisted of a 4 kHz tone burst of 60 ms duration and the probe was a 4 kHz tone burst of 10 ms duration. The probe was either centered in the masker or followed the masker by various offset-to-onset intervals (Δt). Stimulus intensity was specified for each subject re: threshold for the masker. The latency versus, Δt functions for waves III and V and the amplitude versus Δt function for wave III were compatible with a partial forward-masking interpretation. However, the wave V amplitude versus Δt function showed amplitude increments re: the unmasked wave V amplitude for some values of Δt (enhancement). Other stimulus variables manipulated included the masker frequency, the probe frequency, the relative intensity of the masker, the absolute intensity of the probe and masker, and the rise/fall times of the masker, Enhancement of wave V is a robust phenomena obtainable with a broad but definite set of stimulus conditions.
Auditory nuclei in the brainstem exhibit hypersensitivity to electrical stimulation as a result of sound-induced hearing loss
Authors: Gerken George M. , Paul Robin E. , Saunders Samuel S.
Published: October 2005

Thresholds for the detection of electrical stimulation applied to auditory brainstem nuclei were previously demonstrated to decrease by 6 to 15 dB in most cases following mechanical destruction of the cochleas. In the present work electrodes were located in cochlear nucleus and in the nucleus of inferior colliculus, and detection thresholds for electrical stimulation were measured behaviorally. A 48-h exposure to a 1-kHz sine wave of approximately 110 dB SPL resulted in temporary threshold shifts of 50 dB or more and permanent threshold shifts of 10 to 30 dB as evaluated using evoked potential techniques. Behaviorally measured electrical-stimulation thresholds were obtained for each electrode for a number of sessions prior to and following noise exposure. Stimulation thresholds decreased by 2 to 17 dB (hypersensitivity) immediately following the exposure to sound and for most electrodes this hypersensitivity was maintained even after recovery from the temporary portion of the threshold shift. These results indicate that a moderate hearing loss may alter the condition of central auditory mechanisms. [Work supported by NIH Grant NS 16411.]
An automatic electric response audiometer
Authors: Galambos Robert , Makeig Scott
Published: October 2005

Available evoked potential procedures vary widely in how fast and how accurately they assess sensitivity at the audiometric frequencies (250 through 4000 Hz). We will describe a device that can extract this information quickly and automatically. As in Bekesy audiometry, it uses patient responses to control attenuator settings which are then plotted as a function of time. The responses used, however, are the sine-wave like brain potentials recorded at the scalp during monaural presentation of tone pips at a rate around 40 Hz [Galambos et al., Proc. Natl. Acad. Sci. USA 78, 2643 (1981)]. Cross correlating several hundreds of these with a sinusoidal template yields a number which, depending on whether it is high or low, causes a 5 dB decrease or increase, respectively, in the sound intensity. The procedure is then repeated at the new sound level. A computer processes the responses, performs the statistical computations and resets the attenuator every 5–15 s. A threshold estimate can be obtained in a few minutes; these agree well with the behavioral for alert adults, but when the mind wanders, and during drowsiness and sleep, they can rise by 20–30 dB.
Steady-state evoked potentials in humans to continuous amplitude modulated tones
Authors: Rickards Field W. , Clark Graeme M.
Published: October 2005

Phase locking a computer to the modulation envelope of a continuous sinusoidally modulated amplitude modulated tone, enables an averaging technique to be used to record a periodic, steady-state evoked potential from the scalp of normal human subjects. The periodic evoked potentials had the same period as the modulation waveform. Fourier analysis was used to quantify the amplitude and phase of the first and second harmonics of the response. Responses were recorded for modulation rates from 4 to 448 Hz, for carrier frequencies from 250 Hz to 4 kHz and for sound pressure levels from 30 to 100 dB SPL. In general, the response amplitude increases with SPL. Estimates of latencies of these steady-state evoked potentials can be made by measuring the phase change of both harmonics as the modulation frequency is varied. The latencies appeared to have one of 14 discrete values from 3 ms up to 104 ms depending on the stimulus parameters. The majority of the responses had latencies from 9 to 33 ms.
Asymmetries in the auditory brainstem response: Estimates of reliability and variance
Authors: Berlin C. , Allen P. , Hughes L.
Published: October 2005

At the previous meeting of the society we presented data on binaural asymmetries in the auditory brainstem response. Individual data show that the binaural interaction was difficult to replicate and resisted systematic reliability studies. In the present studies a factorial design was used to reduce contribution of earphone and electronic channel assymmetries. Although binaural interaction (defined as the sum of binaurals minus the sum of monaurals [Σ BIN-(Σ MON R + L)]) was seen in group data it was not reliably obtained from all subjects under all conditions, confirming R. Dobie [J. Acoust. Soc. Am. 71, 1031–1033 (1982)]. However, differences between responses acquired from the right and left sides of the head were quite systematic, and reliably duplicable from subject to subject and from condition to condition. A statistical program to perform point-by-point analysis of each of the auditory brainstem responses was written and designed by one of us (LFH) and allows an exact probability determination that specific similarities occur by change. [Supported by NINCDS ♯NS 11647.]
An afterhyperpolarization model for vestibular afferents
Authors: Smith Charles E. , Goldberg Jay M.
Published: October 2005

A stochastic afterhyperpolarization model, modified from Kernell (Brain Res. 11, 685–687 (1968)], simulates many features of the steady-state discharge of vestibular afferents, and may be relevant for spike initiation mechanisms in auditory afferents. Major assumptions are: (a) a single trigger site and firing occurs when the transmembrane voltage reaches a fixed critical level; (b) postsynaptic voltage fluctuations are responsible for interspike-interval variability and are due to random times of occurrence of quantal synaptic inputs; (c) postpike recovery is due to a time-dependent decrease in the potassium conductance (gk), i.e., an afterhyperpolarization; (d) cumulative after hyperpolarization is responsible for the lengthening of the mean interval following an interposed spike; and (e) regular and irregular afferents differ only in the magnitudes of their afterhyperpolarization. The model predicts a stronger dependence of galvanic sensitivity on normalized coefficient of variation of interspike intervals than is actually observed. The discrepancy can be explained by assuming a fourfold variation in the size of synaptic quanta characterizing different vestibular afferents. [Work supported by NSF, S.C. Appropriation for Biomedical Research.]
Acoustic radiation from point-driven, doubly periodic ring-supported, fluid-loaded circular cylinders
Authors: Burroughs C. B.
Published: October 2005

Closed form analytic expressions are derived for the farfield acoustic radiation from a point-driven, fluid-loaded circular cylindrical shell reinforced with doubly periodic ring supports. The ring supports interact with the cylindrical shell only through normal forces; moments for the supports are not included. The analytic expressions include the effects of fluid loading and interactions between supports. Characteristics of the acoustic radiation are examined by example and some physical interpretation of significant features are offered.
Effects of a rigid coupling on wave propagation in fluid-filled elastic cylinders
Authors: Menton Robert T.
Published: October 2005

In order to more closely approximate the true nature of pertinent experimental facilities, the effects of a short, fluid-filled rigid coupling on wave propagation into a coaxial, fluid-filled, semi-infinite elastic cylinder are modeled. A vibrating piston is located at the end of the rigid coupling that is remote from the elastic cylinder. Linear plane-wave propagation is assumed, with the rigid coupling supporting fluid-borne waves at the speed of sound in the fluid, and the elastic cylinder supporting hydroelastic breathing and extensional waves. Compared to the case where the vibrating piston is located right at the elastic cylinder (i.e., no coupling), it is found that the amplitude of breathing waves in the elastic cylinder is nearly doubled. In addition, breathing waves propagate down the cylinder as though they originated at a position slightly in front of the vibrating piston. These results are useful for correlating elastic cylinder modeling with experimental data.
Application of two-variable Taylor series to the ray theory of propagation in an unbounded medium
Authors: Hall Marshall
Published: October 2005

The ensonified acoustic field near the initial caustic in the first convergence zone is considered. This field is formed by a pair of rays that are heading downward at the source and upward at the receiver. The ray parameter (namely the phase velocity) of each ray is expressed as a two-variable Taylor series in horizontal range and depth relative to a fixed expansion point. The terms of the series are calculated for derivatives up to fifth order. The method is applied to a typical oceanic example in which the caustic of the rays passes the expansion point at a vertical distance of 15 m and at a horizontal distance of 200 m. The Taylor-series phase velocities are inserted into the usual ray theory expressions for the relative energy and travel time. It is found that satisfactory accuracy is obtained over a region of the ocean that is sufficiently large for several sonar applications. The advantage of this new approach is that, to calculate the sound field at a grid of points within a region, it is sufficient to determine (by iteration) the ray parameters of only the two rays that pass through the expansion point.
Sound propagation in a surface duct: Can the deep water profile be neglected?
Authors: Hall Marshall
Published: October 2005

In calculating normal-mode acoustic propagation in an ocean surface duct, the sound-speed profile is often represented by the simple bilinear profile. The question arises as to whether the positive gradient in the sound-speed profile that occurs beneath the SOFAR axis has a significant effect on surface duct propagation. Calculations have been made for typical oceanic profiles and it has been found that the bilinear profile and the complete profile yield the same result as a function of range (even though the individual normal modes are different) until the convergence zone effect becomes significant. The threshold range at which this effect becomes significant decreases as frequency is decreased below the surface duct's cutoff frequency (since the energy propagated within the duct decreases while the width of the convergence zone increases). For a duct that has a cutoff frequency of 200 Hz for example, the threshold range decreases from 30 to 15 km as the frequency decreases from 200 to 100 Hz.
A coupled mode solution for acoustic propagation in a waveguide with stepwise depth variations
Authors: Evans Richard B.
Published: October 2005

A coupled mode solution is formulated for the problem of acoustic propagation in a cylindrically symmetric ocean divided, in range, into a finite number of adjoining Pekeris waveguides of differing water depths. The problem is discretized by assigning a pressure release boundary condition at some large finite depth and including sufficient attenuation to exclude any significant energy returning to the water. This formulation includes backscatter from the depth variations of the waveguide and full cross coupling between a finite number of propagating and nonpropagating modes determined by the range and strength of the depth variations. Numerical results based on an implementation of this solution are presented. [Work supported by NORDA.]
A comparison of two range-dependent parabolic equation models: IFD and split-step
Authors: Lee Ding , Schultz Martin H. , Gilbert Kenneth E.
Published: October 2005

The introduction of the IFD (implicit finite-difference) model for the prediction of propagation loss offers users an additional tool for solving the parabolic wave equation. Numerical results comparing the IFD and split-step models for a set of selected problems using the VAX 11/780 computer are presented. In particular, we compare these two models with respect to generality, accuracy, speed, user effort, flexibility, reliability, and other features. [Work supported jointly by NUSC and NORDA.]
The national noise control program: Needs and strategies—State and local perspectives
Authors: Borthwick Jesse O.
Published: October 2005

On October 1, 1982, the U.S. Environmental Protection Agency's noise control program came to an end. The program was phased out following a determination by the Reagan Administration that, “the benefits of noise control can be adequately carried out at the State and local level without the presence of a federal program.” This decision to defederalize noise control has raised many questions concerning the further roles and needs of state and local noise control programs, such as: What actions will the states and cities take in the area of new product noise regulation, and how will these actions affect industry? Where will noise control officials be able to turn for technical and financial assistance? The author will discuss these and other questions currently confronting state and local noise control professionals.
European commissions' noise abatement action program—A global approach on an international level
Authors: Junger J. M.
Published: October 2005

The purpose of this paper is to indicate the role of the EEC's Action Programme for Noise Abatement, to present its evolution and to place it in the context of the European Community. It is also intended more particularly to discuss the interaction between this programme and (1) the national, regional, or local programs; (2) the global economic interests, which result from the free circulation of goods and a reduced distortion of competition; (3) energy saving. In conclusion, it is attempted to point at some criteria which might possibly constitute a basis for a coordinated noise abatement policy at an international level. The key word will be “harmonization” which must be understood as an optimal relationship between different components of a whole.
Voluntary noise standards to meet national needs
Authors: Eldred Kenneth McK.
Published: October 2005

The apparent termination of a federal noise control program in the Environmental Protection Agency removes a potential for a coordinated Federal leadership in the field of environmental noise in the USA, except for the areas in which authority is vested in other Federal agencies, such as the Department of Transportation, Defense, Labor, and Housing and Urban Development. At the same time the needs for control of various aspects of environmental noise persist both here and throughout the world. It may be anticipated that many states and localities will attempt to meet some of these needs through regulations and other activities. The the extent that these regulations and controls require measurement, harmonization of noise scales and measurement procedures is required, if the result is to be nonchaotic. This paper gives a brief overview of the extent of the potential impacts of various parts of the environmental noise problem in terms of sound exposure and population exposed. It then addresses the potential roles and interaction between sound and operating controls and summarizes their current status with respect to the various parts of the environmental noise problem. From this summary it deduces the needs for measurement procedures and their harmonization, and explores the possible roles of the voluntary standards sector in fulfilling these needs.
Piezoelectric polymer microphones and receivers of improved efficiency
Authors: Murphy Preston , Maurer Guy , Davies Geoffrey
Published: October 2005

Piezoelectric plastic film transducers have been designed to operate as microphones and receivers in new communications equipment such as electronic telephones. A theory was developed for a telephone type receiver using a thin, circular membrane, curved by air pressure and driving a coupling volume with significant reaction effects. A figure of merit was derived where the acoustic output level per input voltage was proportional to d̄ {Y/[t(1 − v)]}1/2, where d̄ is the mean piezoelectric coefficient (d33, d32), t is the diaphragm thickness, Y is Young's modulus, and v is Poisson's ratio. Experimental results were in general agreement with theory. A new transducer design of improved efficiency was developed with two circular diaphragms bonded in the center, spaced at the periphery, and connected to operate in a push-pull mode. Either series or parallel connection could be used to match output level and impedance to following circuitry. The efficiency of receivers with correct telephone bandwidth was about 100 dB/V. This is about 10 dB higher than receivers with auto-supported or foam-supported diaphragms. Microphone sensitivities were typically − 52 dB/Pa. The design was shown to be thermally stable and resistant to severe environmental conditions.
Computer-mediated measurement of acoustic radiation fields
Authors: Weinreich Gabriel
Published: October 2005

By parametrizing an acoustic field as an expansion in solutions of the wave equation, one can not only predict the field at points that have not been measured, but also obtain a powerful connecting link to the detailed physics of the mechanical systems with which the field interacts. To provide such a description of an actual field one must measure its amplitude and phase at sufficiently many points, and sufficiently many frequencies, to compute the necessary coefficients; even if both domains are intelligently truncated, the amount of data to be collected and processed remains enormous. Fortunately, modern electronics and computers make such an approach practical. Illustrations will be given of (a) a system that expands a measured radiation field in spherical waves, separating outgoing from incoming waves, and classifying each by its spherical harmonic angular dependence; (b) a system that measures the radiative monopole and dipole moments of a complex vibrating structure, whose frequency dependence can then be analyzed to yield the radiative contributions of the normal modes. [Work supported by NSF.]
Planning speech for execution at different tempos
Authors: Cooper William E. , Ham Audrey , Damon Kristina , Soares Carlos
Published: October 2005

A study was conducted to examine the influence of speaking rate on the occurrence of palatalization across word boundaries, fall-rise patterns of fundamental frequency, and pausing. Eighteen speakers, nine of whom were characterized as normally fast speakers, uttered preselected sentences at normal, fast, and slow rates. The frequency of palatalization was greater for characteristically fast speakers and for fast rates of speech within speakers. Palatalization was reduced in the immediate vicinity of emphatic stress, most markedly when such stress appeared on the /y/-word, replicating earlier results [W. E. Cooper, C. Egido, and J. M. Paccia, J. Exp. Psych.: Hum. Perc. Perf. 4, 264–272 (1978)]. Analyses of fundamental frequency and pausing for these utterances and for a related corpus of naturally elicited speech will be discussed in terms of implications for a model of speech planning and execution. [Work supported by NIH.]
Experimental manipulation of speaking rate for studying temporal variability in children's speech
Authors: Smith Bruce L. , Long Steven H. , Sugarman Michael D.
Published: October 2005

Children's speech typically exhibits both longer segment durations and greater inter- and intra-subject variability than the speech of adults. One hypothesis which has been proposed to account for the greater variability is that children have less-refined neuromotor control capabilities and do not move their articulators with the same accuracy that adults do. Another hypothesis is that children's increased variability is essentially a statistical artifact which occurs because of the greater range of durations likely to be associated with longer segment durations. In the past, statistical methods alone (e.g., “relative variance” and “coefficient of variation,”) have been used in an attempt to compensate for between-group duration differences when comparing variability in the speech of adults and children. In the present study, durational factors were also experimentally manipulated by having three groups of children (ages 5, 7, and 9) and a group of adults produce ten repetitions of a sentence at each of three different speaking rates: normal, fast and slow. Within- and between-group comparisons of variability under these conditions were investigated. Results will be discussed in relation to the above two hypotheses.
The influence of consonant environment on F0 from voicing onset to vowel target in the speech of children
Authors: Ohde Ralph N.
Published: October 2005

Although research on adult speech has manifested systematic effects of consonant environment on the fundamental frequency (F0) of vowels, little is known about the nature of these influences in children. In order to determine the effects of consonant environment on F0 in children, three girls and three boys between 8 and 9 years recorded five repetitions each of voiceless aspirated /ph,th,kh/ voiceless unaspirated /sp,st,sk/, and voiced /b,d,g/ stops in combination with the vowels /i,e,u,o,a/. Fundamental frequency was measured for the first five glottal periods and vowel target, and voice onset time (VOT) was determined for each utterance. The findings for VOT revealed nearly identical results for voiceless unaspirated stops and voiced stops, whereas differences between voiceless unaspirated and voiceless aspirated stops for labial, alveolar, and velar features were 61, 65, and 59 ms, respectively. Despite the similarity in VOT between voiceless unaspirated and voiced stops, F0 contours of voiceless unaspirated stops were more similar to voiceless aspirated than voiced stops. In general, there was a decrease in F0 from the first glottal period to the second glottal period for all voicing conditions. Moreover, F0 was usually higher in the environment of voiceless stops than voiced stops, although this was not unanimous across subjects. The findings will be discussed relative to developmental and coarticulatory processes, and sex differences. [Work supported in part by NINCDS.]
A comparative study of common chimpanzee and human infant sounds
Authors: Bauer Harold R.
Published: October 2005

Chimpanzees observed in Tanzania produced a graded, dynamic sound system ranging from short distance, softer, lower F0 sounds to long distance, louder, higher F0 sounds. A softer and a louder sound category, the whimper and the scream, were selected for further study of F0 and harmonics. From these acoustic analyses and behavioral observations, evidence for a variety of phonatory and resonance patterns were evident in their vocalic sound system which could carry information but lack the segmental contrastivity of human speech. Human infants about a year old were audio and video recorded in the laboratory or home. Vocalic sound durations, F0 contours, and range were much more variable than those observed for chimpanzees. The F0 of human infant sounds often ranged from 300–600 Hz, although occasionally F0 values many times higher were produced making harmonic-formant interactions a critical issue in the comparison with chimpanzee sounds. Fricative, stop, and other consonants not evident in chimpanzee sounds, were common in human infants at 13 months. Selection and reduplication of early word productions suggests the importance of self stimulation, caregiver responsiveness, imitation, vocal tract development, and the acquisition of species typical speech motor skills as some important prerequisites to the development of human words. [Support was provided by NIH, NSF and the A. von Humboldt Foundation.]
Confusion effects in forward masking
Authors: Neff Donna L.
Published: October 2005

Moore and Glasberg [J. Acoust. Soc. Am. 71, 941–948 (1982)] recently stressed that some forward-masked thresholds may reflect the listener's difficulty in discriminating between signal and masker, rather than the detectability of the signal energy. As an extreme example, if the masker were a long series of brief tonal pulses, the detectability of one more identical pulse added to the end of the series would be very low if the level of all pulses were identical. Psychometric functions for such stimuli should show a local minimum, with performance approaching chance in the region of greatest signal-masker confusion. These predictions were tested in several experimental conditions, using various masker-signal combinations, psychophysical procedures, and a contralateral cue (broadband noise) designed to resolve confusion about masker offset. Results confirm the predictions of the “confusion” interpretation of certain forward-masking effects. However, psychophysical procedure and listener training have a stronger influence on performance than the use of the contralateral cue. [Work supported by NSF, NIH, and the University of Nebraska.]
Interaction of simultaneous and forward masking for maskers and signals differing in frequency
Authors: Jesteadt Walt , Weber Daniel L. , Wilke S. Shane
Published: October 2005

The combination of a 1000-Hz forward masker and a broadband simultaneous masker produces more masking than would be predicted from the sum of their individual effects [Jesteadt and Wilke, J. Acoust. Soc. Am. Suppl. 1 71, S72 (1982)]. We have extended the original study to include forward maskers above and below the signal in frequency. We measured the threshold for a 1000-Hz signal as a function of masker level (30 to 80 dB SPL) for forward masker frequencies from 600 to 1200 Hz (100-Hz steps below the signal, 50-Hz steps above) with the signal and masker presented in quiet and in an 8-dB No background noise. Nonadditivity was greatest when the forward masker and signal were similar in frequency. We also obtained psychophysical tuning curves for a 45-dB SPL, 1000-Hz signal using forward maskers from 600 to 1200 Hz, with the signal and masker presented in quiet and in noise. As reported by Green et al. [J. Acoust. Soc. Am. 69, 1758–1762 (1981)], the forward-masked tuning curves were sharper in noise, presumably because the non-additivity of the two maskers was greatest in the region of the tip of the tuning curve. If the background noise produced an increase in masking by restricting off-frequency listening, we would predict greater nonadditivity when the signal and masker differed in frequency as well as broader tuning curves in noise than in quiet. [Work supported by NIH and NSF.]
The influence of the sound level of a steady-state broadband noise on the temporal response of the acoustic reflex
Authors: Lalande Nicole , Hétu Raymond
Published: October 2005

The influence of the sound level of a steady-state pink noise on the short- and the long-term response of the contralateral acoustic reflex was investigated with nine adult subjects. Three noise levels were selected (LpA of 95, 100, and 110) for a duration of 16 min (or 8 min occasionally). Based on previous data [R. H. Wilson et al., J. Acoust. Soc. Am. 64, 782–791 (1978)], it was hypothesized that when the reflex activity is expressed in absolute value of change in acoustic susceptance, the reflex decay is the same whatever the intensity of the noise; expressed in relative value of percentage of change in susceptance, the reflex adaptation is greater for a low level of noise than for higher ones. Results clearly confirmed the above hypothesis. These findings explain the conflicting results previously obtained in terms of the way data are analyzed. Practical implications concerning the protection afforded by the acoustic reflex under such noise exposures will be discussed.
An evaluation of the acoustics of the O'Connell Center, Gainesville, Florida
Authors: Kinzey Bertram Y. , Siebein Gary
Published: October 2005

The O'Connell Center is a mass seating facility on the University of Florida campus in Gainesville, Florida. Activities housed in this large, air-supported fabric structure range from rock concerts to basketball games to commencement exercises. The study will compare the design methods and programmatic restrictions used by Coffeen, Anderson and Associates, Mission, Kansas, the acoustical consultants for the space, with a post occupancy study including field measurements and user evaluations to understand the ability of acousticians to predict the acoustical performance of this relatively new construction technology. A similar building was constructed at the University of South Florida in Tampa, Florida. A comparison of construction techniques and acoustics between the two structures will also be presented.
Application of cepstral technique to remove organ pipe tones from combustion spectra
Authors: Miles J. H. , Wasserbauer C. A.
Published: October 2005

Signal reinforcement and cancellations due to standing waves may distort constant bandwidth combustion spectra. Cepstral techniques previously applied to the ground-reflection echo problem [J. H. Miles, J. Acoust. Soc. Am. 61, 35–38 (1977)] are used to obtain smooth broadband data. Internal fluctuating pressure measurements made using a J47 combustor attached to a 6-m-long duct are studied. Measurements made using jet A and hydrogen are compared. The acoustic power levels inferred from the measurements are presented for a range of operating conditions.
Rough analysis of installation effects on turboprop noise
Authors: Durbin Paul A. , Groeneweg John F.
Published: October 2005

A rough analysis of noise from a propeller operated at angle of attack, and in the nonuniform flow due to a line vortex approximating a wing flow field suggests installation can significantly affect turboprop noise levels. On one side of the propeller, where the blades approach the horizontal plane from above, decreases of noise occur; while on the other side noise increases. The noise reduction is due to negative interference of steady and unsteady sources. An angle of attack, or distance between propeller and vortex, exists for which noise is a minimum. The relative contributions of thickness and loading noise are discussed, and results are presented both as plots of directivity and as plots of maximum noise level.
Correlation of core noise obtained by application of three-signal coherence techniques
Authors: Von Glahn U. H. , Krejsa E. A.
Published: October 2005

The prediction of frequency content and noise levels of turbofan engine core noise is re-examined as a result of recent test data and a new diagnostic technique [E. A. Krejsa, NASA TM 82634 (1981)]. The diagnostic technique, utilizing a three-signal coherence method is used to obtain core noise spectra for several engines. Similarities and differences of the spectra are discussed. Finally, the three-signal coherencce data are correlated, leading to the development of an improved core noise prediction procedure.
Development to high-resolution side-look sonar
Authors: Nelkin Arthur , Gilmour George A.
Published: October 2005

About 28 years ago, the Navy had a requirement to scan the bottom of channels and harbors with sufficient resolving power to identify minelike objects on the bottom with cylindrical dimensions 112 in. in diameter by 6 in. long. A side-scan sonar was designed to provide a narrow fan shaped beam focused on a line on the bottom by means of a fixed arc array. The transducer array had to be towed a fixed distance (the radius of curvature of the arc array) off of the bottom for optimum focus. The system scanned the bottom by means of the vehicle movement in the direction of tow and perpendicular to the direction of tow by the travel time of the acoustic signal emitted from the transducer to the bottom. This system had a number of limitations: bottom coverage rate (m2/h), fixed height above the bottom, a limited display capability, and limited signal processing. These led to the development of multibeam arrays with electronic all-range focusing, CRT displays, and adaptive time varied gain tied into the background reverberation levels.
Interferometric array for high-resolution sonars
Authors: Grail Georges
Published: October 2005

The spatial and temporal properties required by a sonar array to obtain an acoustical picture of the sea floor are reviewed and illustrated by simulation techniques. A system using both an interferometer for transmission and a linear array for reception is described. The resolution gain obtained is considered, as well as the system limitations such as angular sector and bandwidth. The comparison with a classical sonar is discussed in detail, and experimental results achieved by this kind of array operating at 200 and 420 kHz are presented.
Odd/even beamformers for high-resolution sonar
Authors: Henderson T. L.
Published: October 2005

The use of odd/even beamformers is a well-known generalization of the phase comparison and amplitude comparison methods employed in monopulse radars and split beam sonars to locate targets precisely within the mainlobe of the beam pattern. However, it is not well known that target angle can be determined unambiguously, and independently of frequency, from the scale factor that relates the two beam outputs, if the odd/even aperture shadings satisfy the Kerr-Murdock conditions and if the odd beam output has been phase corrected by an ideal integrator. Applications of this technique to high resolution sonar are discussed, emphasizing the use of classical linear regression analysis to compare the outputs. The regression coefficient is used to estimate target angle, and the correlation coefficient gives an indication of the quality of the estimate as a function of time or range. A graphically based method is presented for determining the angular separation of two targets at the same range. The use of scale comparison to measure target angle, rather than channel-to-channel time delay as in split beam sonar, appears to offer several advantages, particularly where broadband signals are involved. [Work supported by NAVSEA.]
A study on FS-PDMP system
Authors: Zhang Shuying
Published: October 2005

A signal detection system, which consists of a frequency-stepping signal generator and a corresponding post-detected matching processor, was studied both theoretically and experimentally. The characteristic function determining the detection performance of FS-PDMP system was derived and analyzed, and a set of equipment, which can generate and process an eight-unit long signal with various FS modes [n] (T = 8τ = 200 ms, fn = 1360 + n×40 Hz), was made for the experimental studies. Some results are as follows: (1) The output waveforms of PDMP matched with many FS modes were recorded. It was been shown that the output with a narrow mainlobe (the duration = τ) and a few side lobes (the level < p/8, p is the peak of mainlobe) can be obtained from PDMP with the FS modes composed of eight different frequencies and that if there are two or more same frequencies in the FS modes, either the mainlobe will widen or the side-lobe level will increase, or even both. (2) The response of the FS-PDMP system to Doppler frequency β was investigated. For mode [01234567], the output of PDMP gradually reduces from p to p/8 with the variation of β from 0 to ± 280 Hz, and the time of output appearance correspondingly changes from 0 to ± 7τ. For both modes [75310246] and [05274163], when β varies from 0 to ± 40 Hz, the reduction of the mainlobe of PDMP output is very fast, and no notable side lobe appears for the former while there are two remarkable ones for the latter. (3) The detection threshold of the FS-PDMP system is about −8 dB lower than that of a single unit envelope detector. The above results are in conformity with the theoretical analysis.
System dynamics of large rotating machinery and its foundation
Authors: de Estrada Michael B. , Ying Shuh-Pan
Published: October 2005

System dynamics of a single-speed centrifugal induced draft fan on an isolated pile foundation was investigated to determine the adequacy of foundation design and fan balance. The investigation involved dynamic soils and pile evaluation to establish stiffness and damping of foundation supporting medium. A dynamic computer analysis of complete fan, concrete foundation, soil-pile system was conducted to establish resonant frequencies and system sensitivity to rotating unbalance. Dynamic impact testing using the transfer function technique was performed to determine system natural frequencies and damping for comparison with analytical results. Vibration levels during fan startup and steady-state operation were monitored to determine system resonances, sensitivity, and residual unbalance. Resonant frequencies associated with foundation motion were observed in the 5- to 7-Hz and 10- to 12-Hz regions; and in the the 21- to 25-Hz region for the fan rotor. The fan operating speed is 14.7 Hz which is outside the resonance zones. The maximum bearing housing sensitivity was measured at 0.027 mils/oz and residual unbalance at 0.0021 lb-in./lb which falls within ANSI Standard S2.19-1975 limits.
Articulatory constraints on vocal tract area functions and their acoustic implications
Authors: Butler S. J. , Wakita H.
Published: October 2005

A mathematical model for generating vocal tract area functions for vowel-like sounds has been developed. Using computer simulation of a transmission line analog of the vocal tract, the relationship between acoustic features (in this case formants) and parameters of the model (i.e., articulatory parameters) has been investigated in order to develop improved algorithms for acoustic-articulatory and articulatory-acoustic transformations. The model has been found to be a useful tool for study of acoustic phonetics, examination of the applicability of speaker normalization techniques, and for the refinement of high-quality speech synthesis. Extensions of the model to include consonantal vocal tract configurations are also presented.
Speech synthesis using allophones
Authors: May Janet G. , Lee Eugene H.
Published: October 2005

Synthesizing speech by concatenating allophones is a new application of linear predictive coding (LPC) that provides an unlimited vocabulary. Since the stored units are individual speech sounds, any English word can be synthesized by concatenating the appropriate sounds. Our inventory contains 59 speech sounds and five pauses. Each allophone was created by extracting it from the digital waveform of a word, and then synthesizing it. The sounds are called allophones because some phonemes have two or three versions for different environments. For example, /k/ and /g/ each have three allophones: one to be used before front vowels, one before back vowels, and one in final position. In contrast, research revealed that the same /p/ can be used in all environments. Isolating final /r/ from preceding vowels was impossible because of important transitional information. Therefore there are five phonemically unique V + /r/ combinations. Due to the omission of many transitions, allophone synthesis suffers from reduced intelligibility when compared with the technique of synthesizing words as units. This compromise seems minor, however, when its flexibility and low cost are considered.
Formant synthesis: Technique to account for source/tract interaction
Authors: Yea J. J. , Childers D. G.
Published: October 2005

Research in speech synthesis is presently concerned with methods for improving intelligibility and naturalness. But most work to date has dealt with intelligibility, i.e., the synthesized speech is intelligible but monotone and machinelike in quality. Although this type of speech is useful in some applications, the users of speech synthesizers are demanding a more natural sounding synthesized speech. We are addressing this issue in our research. In particular we are studying the influence of the glottal source function on the production of synthesized speech. Our data base consists of inverse filtered speech, ultra-high-speed laryngeal films, and the electroglottograph waveforms, all temporally synchronized in our experiments. We use the experimentally derived source waveforms obtained from these various method as the excitation for a serial/parallel Klatt formant synthesizer to synthesize sentences. We then evaluate the naturalness of this synthetic speech by listening tests. Our goal is to rank order the contribution of different source parameters to the naturalness of synthetic speech. More specifically, we report on our results to date of the effect of source-tract interaction upon the production of natural sounding speech.
Extensions of Davis' hair cell model
Authors: Allen J. B.
Published: October 2005

By making a very simple modification to the Davis model of hair cell transduction, it is possible to accurately model neural period histogram data. The model data match the neural data over the entire range of experimental observations. Computed responses to pure tones and to tone bursts will be compared to experimental observations.