bluuebunny/crossref_metadata_2025 huggingface dataset:
Distribution, characteristics, and some acoustic implications of oceanic fronts in the North Atlantic
Authors: Khedouri E. , Cheney R. E.
Published: October 2005

Oceanic fronts and eddies occur throughout the North Atlantic, but are most evident in the western and northern regions. The Gulf Stream is the dominant front in the Atlantic and is also the source of both cyclonic and anticyclonic eddies. These features are 100–200 km in diameter and have life-times of 1–2 years; as many as 13 have been observed to exist at one time in the vicinity of the Gulf Stream. Acoustic ray traces and intensities for various receiver depths were calculated across the Gulf Stream using AXBT temperature profiles and historical sound velocity data. When sound intensities across the Gulf Stream were compared to intensities for similar situations without a front, it was found that the Gulf Stream North Wall could significantly increase or decrease propagation loss depending on distance of the receiver from the front; the maximum difference was 18 dB. The South Wall of the Stream had no significant effect on sound intensity. Three-dimensional propagation loss diagrams show that effects of the North Wall on propagation loss are evident at all depths from sea surface to 1000 m. It is clear that oceanic fronts and eddies must be included in any realistic acoustic model.
Attenuation of northern dwellings to a linear source of noise
Authors: Driscoll D. A. , Dulin J. P. , Keast D. N.
Published: October 2005

Insertion loss in octave bands was measured for 14 residences in upstate New York using a carefully developed procedure similar to ASTM E336-71 A1.2. Although other dwelling attenuation data are available [e.g., House Noise-Reduction Measurements for Use in Studies of Aircraft Flyover Noise, Soc. Automative Engrs., AIR 1081; Oct. 1971] none was clearly applicable to rural areas of upstate New York and to sources of noise located near the ground. The source was simulated by a three-speaker array using broadband noise. Attenuation of transmission line noise was derived mathematically by applying the measured octave-band attenuations to a typical broadband transmission line noise spectrum and converting to A-weighted and sound levels. Average attenuation was about 15 dBA with windows open 2 ft2, and about 30 dBA with windows closed. [The study was conducted by Rochester Gas and Electric Corporation and their consultant Bolt Beranek and Newman in consultation with the New York State Departments of Public Service and Environmental Conservation.]
Sound localization with impaired hearing
Authors: Hausler R. , Mart E. M. , Colburn H. S.
Published: October 2005

Discrimination tests were made on persons with hearing impairments and on patients with multiple sclerosis. Results include estimates of horizontal minimum audible angle (MAA) at eight azimuths, vertical MAA straight ahead, and just-noticeable differences in interaural time delay (time JND) and interaural amplitude ratio (amplitude JND). The standard stimulus was broadband (0.25–10 kHz), pulsed (1 s) noise. Conductive cases (>35-dB loss) gave abnormal values in almost all tests. Symmetric sensori-neural cases with speech discrimination scores above 90% gave roughly normal values in all tests, while the corresponding group with scores below 80% gave elevated values for the vertical MAA and the horizontal MAA on the sides. Meniere's cases had normal MAAs and time JNDs and an amplitude JND that was normal only at high levels. Neurinoma cases gave abnormal values in at least one measurement each, with large intersubject variability. Persons with only one functional ear showed no ability to discriminate interaural parameters but some gave MAAs within the normal range. For multiple sclerosis patients, time JNDs, amplitude JNDs, and vertical MAAs were affected independently. [Work supported by NIH and Hopital Cantonal, Geneva, Switzerland.]
Experiments on the “prosodic intelligibility” of low-frequency speech codes
Authors: Allen Donald A. , Strong William J. , Palmer E. Paul
Published: October 2005

We have reported previously on the intelligibility and discriminability of low-frequency speech codes for the severely hearing impaired. In recent experiments the following versions of speech and speech codes have been tested to explore how well they carry prosodic information: (1) natural speech; (2) speech severely low-pass filtered at 900 Hz; (3) a formant code consisting of three sinusoids scaled to the formant frequencies; (4) a largest harmonic code consisting of four harmonic sinusoids closest to the formants; and (5) an all harmonic code consisting of many harmonic sinusoids. Fundamental frequency and formant frequencies are scaled by different amounts in the various codes. Normal hearing subjects were asked to determine which of five versions of the sentence, “John drove to the store,” was heard. Speech and speech codes were tested for two female and two male talkers. “Prosodic intelligibility” results are presented and compared for the various talker, speech, and speech code combinations.
Passive vocal tract enlargement during voiced stops
Authors: Ohala John J. , Riordan Carol J.
Published: October 2005

Voiced stops should become devoiced within 5 to 10 m of stop closure if there is no vocal tract enlargement to delay the inevitable reduction in transglottal pressure drop. Since stops may be voiced longer than this, some cavity expansion must take place, either passive through tissue compliance or active through larynx lowering and the like. To estimate the duration of voicing in stops when only passive enlargement occurs, subjects produced isolated nonsense words of the form VC:V, where V was one of a variety of English vowels and C: was an artificially prolonged /b/, /d/, or/g/. Oral air pressure was vented through a catheter leading from the pharynx to the atmosphere via the nasal cavity. At unpredictable times, a solenoid-activated valve closed the catheter and the consequent build-up of oral pressure extinguished the voicing. Voicing continued after this closure longer during /b/ than /d/ or /g/, and longer when coarticulated with high vowels than with low vowels. These results can most plausibly be explained by reference to differences in compliance of the surfaces on which oral pressure impinges during the various consonants and vowels. [Supported by NSF and NIH.]
Identification, discrimination, and selective adaptation of simultaneous musical intervals
Authors: Halpern Andrea R. , Zatorre Robert J.
Published: October 2005

Four experiments investigated perception of major and minor thirds whose component tones were sounded simultaneously. Effects akin to categorical perception of speech sounds were found. In the first experiment, musicians demonstrated relatively sharp category boundaries in identification and peaks near the boundary in discrimination tasks of an interval continuum where the bottom note was always an F, and the top note varied from A to A flat in seven equal logarithmic steps. Nonmusicians showed these effects only to a small extent, with wide individual differences. The musicians showed higher than predicted discrimination overall, and reaction time increases in identification at boundaries. In the second experiment, musicians failed to consistently identify or discriminate thirds which varied in absolute pitch but retained the proper interval ratio. In the last two experiments, using selective adaptation, consistent shifts were found in both identification and discrimination, similar to those found in speech experiments. Manipulations of adapting and test stimuli showed that the mechanism underlying the effect appears to be centrally mediated and confined to a frequency-specific level. A multistage model of interval perception, where the first stages deal only with specific pitches may account for the results. [Work supported by NICHD.]
Computer-aided imaging of sound fields and objects
Authors: Dardy H. D.
Published: October 2005

Conventional acoustic imaging techniques implemented to visualize sound fields and objects have been aided through the current availability and power of newly developed computer architectures. Application of this processing power in the laboratory has transformed the basic visualization of sound fields from a qualitative representation of the process to a quantitative one where we can now directly obtain absolute data from the visualization experiment. Linearization of the acoustic image, signal enhancement, weak signal extraction, and video color display under control of a minicomputer processor helps establish an immediate visual understanding of the fundamental acousto-optic process under study. Further capability in the area of animation and three dimensional imagery can extend the techniques' useful application. The utility of computer-aided processing as applied to schlieren visualization will be discussed: results from both schlieren and other representative imaging applications will be presented.
Review of acoustical holography
Authors: Ahmed Mahfuz
Published: October 2005

Acoustical holography today represents several different processes and techniques, ranging from the direct analog of optical holography to schemes not possible in the optical domain, with some very general hybrid forms in between. These different processes have played an important role in advancing the entire field of acoustic imaging. This paper will review those aspects of holography which have had a significant impact on acoustic imaging with emphasis on concept. It will deal with the changing identity of holography, the applications of optical and acoustical holography, holography by square-law detection, holography by linear detection with or without an electronic reference, holography by scanning techniques, temporal reference holography using an analog or a digital clock, and quasi-holographic systems such as synthetic aperture sonar and Bragg-diffraction imaging. The front-end embodiment of these concepts, and the problems inherent in acoustical holography will be discussed. Some speculations about the future of acoustical holography will be forwarded.
Review of acousto-optic interactions
Authors: Young E. H.
Published: October 2005

This paper reviews the theory of acousto-optic interactions. From a qualitative approach the acousto-optic interactions can be explained from the scattering of the phonons. By examining the conservation of momentum and energy in the particle interactions, the basic relations of the acousto-optic interaction can be obtained. From the rigorous wave theory approach, the details of the acousto-optic interactions are obtained. Acousto-optic interactions are generally divided into two dominant regions, the Raman-Nath and the Bragg regions and their differences will be discussed. Acousto-optic interactions with a sound beam generated by a transducer array will also be discussed. The basic theory of acousto-optic interactions presented in this paper is the basis for applications in acoustic microscopy and acoustic imaging.
Microstructure of the detection threshold curve
Authors: Cohen Marion F. , Schubert Linda P.
Published: October 2005

Typically, the threshold of audibility curve is based on threshold determinations at octave intervals and the presumption is generally made that the threshold curve is fairly smooth at frequencies between those at which threshold is measured. In 1975, Thomas [J. Acoust. Soc. Am. 57, 526 (1975)] reported findings which indicated that in fact there exist fluctuations in detection threshold of as much as 10 dB which occur with relatively small changes in frequency. The experiments reported here consist of determinations of the detection threshold of sinusoids with no added noise using the 2IFC procedure. The stimulus tone ranged in frequency from 1000 to 2000 Hz. The results indicate that in fact changing the frequency by as little as 30 Hz can cause an increase in the 75% detection level by as much as 12 dB. and further that if the frequency is shifted in the same direction, the detection threshold will decrease. This periodic threshold fluctuation, or the microstructure of the detection threshold curve, will be shown in detail.
A model of psychophysical suppression
Authors: Shannon Robert V.
Published: October 2005

A model will be presented in which suppression and excitation are treated as two separate processes. A template is assumed for each based on neurophysiological data. Cochlear output patterns can be computed for an arbitrary input stimulus. In the first step of the model, the excitation templates for each input are added in power. Then the resulting excitation pattern is convoluted with the suppression template to produce a pseudocochlear output distribution across the basilar membrane. Psychophysical decision rules operating on this output are able to predict many aspects of psychophysical two-tone suppression data. As a single tone is increased in level, the output of the model shows that suppression can act as an amplitude compression process, representing the full range of input in a smaller range of outputs. Correspondences between the model and other psychophysical data will be discussed.
Temporal factors in forward masking for sinusoidal and noise maskers
Authors: Weber Daniel L. , Moore Brian C. J.
Published: October 2005

Forward-masked thresholds for sinusoidal signals were estimated with an adaptive 2IFC procedure as a function of signal duration following both sinusoidal and narrow-band noise maskers. Both masker and signal had 5-ms linear rise and fall ramps; the masker had a 500-ms steady-state portion and the signal duration varied by changing the steady-state portion. The center frequency of the noise and the frequencies of the sinusoidal masker and signal were all equal. The two maskers had essentially equal power within the critical band. Given no silent interval between masker and signal the noise produces more masking than the sinusoid if the signal has a short duration (10 ms). However, the reverse is true if the signal duration is longer (40 ms). When signal duration increases by varying the signal onset for a fixed offset-offset delay between masker and signal, threshold decreases following the noise masker, but remains constant or increases following the sinusoidal masker. We argue that these differences between noise and sinusoidal maskers probably reflect the operation of two different detection processes.
Forward masking as a function of frequency
Authors: Jesteadt Walt
Published: October 2005

The time course of forward masking of a sinusoidal probe by a sinusoidal masker of the same frequency was determined at 125, 250, 1000, and 4000 Hz. A 20-ms probe was presented at Δts of 5, 10, 20, or 40 ms after the offset of the 250-ms masker. Four levels of the masker were used at each frequency. Masked thresholds in dB SPL are linearly related to log Δt in all cases. The slopes of the functions for 125 and 250 Hz are shallower than those for the higher frequencies. The difference in slope between the 250-Hz function and those for 1000 and 4000 Hz is greatly reduced when the data are replotted in terms of amount of masking rather than masked threshold. The slope of the function for 125 Hz is markedly shallower than those for the higher frequencies in either coordinate system. [Work supported by NIH.]
Theoretical aspects of internal and external noise
Authors: Siegel R. A.
Published: October 2005

The concept of internal noise occupies a central role in psychophysics. In experiments in which the external physical stimuli contain random noise, the subject's response will vary in part due to the noisiness of the stimulus and in part due to his own internal noise. Several investigators have attempted to measure the relative magnitudes of internal and external noise in various experimental situations. However, there have been no rigorous definitions made for internal and external noise. Thus, when interpreting results of these experiments, one must be careful in ascertaining what is measured. We present mathematical definition of internal and external variability, and discuss their advantages and limitations. Applying these definitions to experimental noise-density discrimination data [D. H. Raab and J. J. Goldberg, J. Acoust. Soc. Am. 57, 437–447 (1975)], we estimate relative internal and external variance components for that task. Our estimates differ from those calculated by Raab and Goldberg, who assumed that auditory processing is characterized by an energy detector and an independent internal noise source. [Work supported by NIH.]
Wave development in heated human erythrocytes
Authors: Crum L. A. , Coakley W. T.
Published: October 2005

Heated human erythrocytes gradually lose their form-maintaining structure as the temperature is increased to 49°C and behave in many respects as a viscous liquid. We have discovered that heated erythrocytes that are stressed by shear forces in liquid flow develop periodic disturbances along their length that are wavelike in appearance. We have developed a technique for heating and stressing these cells that is novel, simple, and quantitatively precise. We have applied this technique to heated human erythrocytes and have caused them to be drawn out into long thin cylinders that show wave development similar to that observed in liquid jets. From an analysis of the instability development in liquid jets as applied to the erythrocyte system we can obtain more precise values for the membrane surface tension than previously reported by other techniques. [Work partially supported by Office of Naval Research.]
Sex differences in most preferred listening level for telephone speech
Authors: Kames Audrey J.
Published: October 2005

Female listeners were found to prefer to hear speech significantly lower in volume than male listeners. A series of three studies was conducted to evaluate the preferred listening level for speech heard over a standard telephone handset [G-type]. Equal numbers of men and women listened over a G-type handset to recorded speech samples. Hearing threshold and age of the listener were obtained. The sex of the talker, the transmitter device and room conditions, ambient noise and reverberation time, were varied systematically during preparation of the speech samples. Subjects adjusted the loudness level with a slide attenuator. In the three studies the mean preferred listening levels were as follows.

Women Men

Study 1 −33.5 VU −29.3 VU

Study 2 −41.7 VU −39.1 VU

Study 3 −49.8 VU −45.3 VU
Acoustic imaging by spatial correlation
Authors: Jackson Darrell R. , Moravan Kou-Ying
Published: October 2005

Analysis and computer simulation based on the point scatterer model have been used to show that high-resolution images can be produced by forming the temporal cross-correlation of the echo signals received by two or more transducers, with angular resolution on the order of the pulse resolution divided by the receiver spacing. With two receivers, the correlator output constitutes a one-dimensional image, with correlation magnitude corresponding to “brightness” and correlator shift proportional to azimuth. The image contains a noisy background having a level approximately equal to (T/T0)1/2, where T is the pulse resolution, T0 is the duration of the echo from the object to be imaged, and unity is the maximum possible correlator output. The desired image is a convolution of the signal autocorrelation with the angular density function for the acoustic highlights of the object. This convolution results in a smearing of the image, which is reduced if relative bandwidths on the order of one decade are used. Possible applications of this technique are underwater imaging for divers and ultrasonic medical diagnosis. [Funded by Naval Sea Systems Command under Contract N00024-78-C-6018.]
Accuracy of a series of ambient noise measurements
Authors: Driscoll D. A. , Webster W. J.
Published: October 2005

It has been shown that a single 15-min or 1-h measurement sequence is a very poor representation of the long-term ambient noise at a measuring location, even though the data may be a highly accurate representation of the noise during the sample period. For any particular short-term statistic (e.g., 15 min, Leq, L10, etc.) there is little chance of being within ±3 dB of the corresponding long-term statistic (based on a month or more of data); ±3 dB limits can be claimed with only 25%–50% confidence depending on the long-term variability of the noise at the site. However, by taking the composite of several independent 15-min measurements, ±3 dB accuracy can be claimed with substantially improved confidence. At a suburban site, for example, five measurements will give 70% confidence, seven will give 80% and ten will give 90% confidence. This and similar information for urban and rural sites was derived using Student's t distribution and evaluated experimentally using 100 ways of data from a rural site, 40 days from a suburban site and 60 days from a quiet urban site.
Motorcycle noise control through use of a stationary sound level test
Authors: Walsh John B. , Marcus Wayne E.
Published: October 2005

New motorcycles have gotten significantly quieter throughout the 1970's. Motorcycle noise control is now a problem of keeping the quiet motorcycles quiet. This problem involves control of both after-market motorcycle exhaust systems and user modifications of stock exhaust systems. Various stationary sound level tests have been suggested as a means of ensuring that motorcycle exhaust systems are quiet when built, and remain quiet in use. Sophisticated test methods have been proposed, but suffer because of their complexity; simpler tests, however, have lower correlation than the complex test methods. A new approach to the use of simple stationary sound level tests was evaluated, and produced promising results.
Transmission loss of a steel noise barrier coated with vibration damping material
Authors: Behar A. , May D. N.
Published: October 2005

This is a study into a more economical way of obtaining the required structural sound transmission loss (TL) in steel highway noise barriers by using vibration damping material. When a layer of a high inherent damping is applied to a relatively lightly damped panel, the result is a composite panel with higher damping than the original one. The resulting TL is higher, even below the critical frequency, than that of a single lightly damped panel of the same weight. Adding vibration damping material to a steel barrier improved the Sound Transmission Class (STC) by 3, for only a 20% increase in weight, whereas about a 50% increase in weight would have been required to increase the STC by the same amount through the addition of more steel. The cost of damping material is also lower than of steel, and its use can provide savings of up to about 20% in the cost of noise barrier panels.
Vehicle directionality and propagation characteristics as errors in traffic noise predictions
Authors: Bradley John S.
Published: October 2005

Several traffic noise prediction equations assume vehicles act as point sources, and radiate sound spherically. The energy average sound level (Leq) is then obtained by integrating over the passby of such a source. Measured vehicle passby sound levels under controlled conditions and at a number of microphone locations, for both grass and asphalt covered ground indicate considerable deviations from these assumptions. More complex models are considered and the resulting reductions in Leq errors are presented. Computer calculations using more exact propagation theory for grass and asphalt covered ground indicate further errors in Leq predictions can occur. The magnitude of the errors depend on the source spectrum and microphone location.
A graphic display for combined presentation of acoustic and articulatory information
Authors: Miller Joan E. , Fujimura O.
Published: October 2005

Articulation data obtained from the x-ray microbeam system [Fujimura, Kiritani, and Ishida, Comp. Biol. Med. 3, 377–381 (1973)] is coded and presented in combination with a time-frequency representation of the corresponding speech. The articulatory features such as velum height for nasalization, lip constriction and tongue position for place of articulation, and mandible height for stress [Fujimura, Miller, and Escolar, J. Acoust. Soc. Am. 61, S48(A) (1977)] are depicted along with the spectral display in order to reveal the time correspondence between the articulation and acoustic events. These displays have been further annotated by phonetic transcriptions derived from pattern matching techniques [Nelson, Miller, and Macchi, J. Acoust. Soc. Am. 63, S32(A) (1978)] to be discussed elsewhere in this meeting. This joint presentation of the two types of data plus phonetic annotation affords a useful means of studying the temporal characteristics of speech dynamics.
Speech analysis by pole-zero decomposition
Authors: Yegnanarayana B.
Published: October 2005

A new method of speech analysis based on a pole-zero decomposition technique is presented. The technique enables determination of pole parameters and zero parameters of a linear system model in an identical manner. Computation of the parameters is done in a straightforward recursive manner without the need for solution of either linear or nonlinear equations. The new method overcomes the limitations of the simplified all-pole models proposed for speech spectra. A pole-zero model for the vocal tract system not only provides an accurate description of the short-time spectral envelope but it also provides a means of obtaining accurately the source characteristics. The method has been applied to several difficult cases of real data such as voiced fricatives, nasals, transitions, etc., and the results indicate that the spectra derived from the pole-zero model match very well with the envelope of the short-time spectrum. It is shown that accurate and reliable voiced/unvoiced decision and pitch period estimation can be made by using the pole-zero decomposition technique.
Three-channel time-domain pitch detector using different non-linear digital filters
Authors: Hess Wolfgang J.
Published: October 2005

Nonlinear distortion is an adequate means to enhance the first harmonic of speech signals for time-domain pitch determination. To evaluate the usefulness of nonlinear functions (NLFs) in pitch extractors, 16 NLFs were investigated: odd functions including linear processing, even functions (e.g., full-wave rectification), “mixed” functions (e.g., half-wave rectification), and rectified single sideband (SSB) modulation. These functions were applied to three categories of signals; undistorted male speech, band-limited male speech, and high-pitched female speech. No single NLF was found to perform well for all signal categories. Simultaneous application of three NLFs (odd, even, and SSB) with equal amplitude characteristics, however, gives good results. Using these NLFs, a three-channel time-domain pitch extractor is implemented. After application of the NLFs, the partial signals in the individual channels are linearly filtered by a crude approximation of the inverse filter, and preliminary period boundaries (markers) are derived by threshold analysis. At least in one channel a strong first harmonic and thus a correct marker sequence can be expected. The algorithm finally selects this channel by checking the regularity of the obtained markers.
Some effects of stimulus rise time on responses of auditory-nerve fibers
Authors: Smith R. L.
Published: October 2005

Post-stimulus-time histograms (PSTs) were obtained for responses to tone and wide-band-noise bursts with stimulus rise times (RTs) between 0–25 ms. For the longer RTs, the PSTs built up to a peak and then decayed, and the results resembled those reported by Rigden et al. [J. Acoust. Soc. Am. 63, S77(A) (1978)]. At low-sound intensities the time delay to the peak approached the stimulus RT, and the delay decreased as intensity increased. In some units, during certain time intervals, firing rate was a nonmonotonic function of sound intensity even though overall firing tale increased monotonically. These shape changes and nonmonotonicities were not present for the shortest RTs, suggesting that they do not reflect a time-dependent nonlinearity and may result from an interaction between the long stimulus RT and a monotonically increasing nonlinearity followed by adaptation. Adding a suppressing tone above CF produced PST shape changes and rate changes similar to those caused by decreasing the sound intensity, further suggesting that intensity-dependent nonlinearities are located peripherally to the site of adaptation.
Simulation of distributions of observed effective scattering strengths of individual fish
Authors: Foote K. G.
Published: October 2005

A knowledge of the probability distribution of effective scattering strengths of observed fish is important in determining the reliability of acoustic estimates of fish abundance. The effective scattering strength is defined here as the product of backscattering cross section of observed fish and transmit and receive beam patterns of observing sonar. The distribution depends on the geometry of ensonification and fish behavior, or joint distribution of spatial and orientation states of fish, in addition to the sonar beam patterns and fish target strength as a function of orientation. Simulation of the distribution is discussed and illustrated for the case of observation of gadoids, for which measurements of the dorsal aspect target strength functions at two ultrasonic frequencies are available, by a directional downward-looking sonar.
A simple way to find resonances in sound scattering from water droplets in air
Authors: George Jacob
Published: October 2005

An especially simple equation presented here reduces the task of locating the resonances in sound scattering from water droplets in air to only a matter of looking up the zeroes of spherical Bessel functions from mathematical tables. We demonstrate this technique by predicting and plotting twelve of the resonances, including the lowest in frequency. The use of these resonances for remote sonar determination of the sizes of the droplets is discussed. A complete derivation of the formulas for scattering cross section and for the resonance condition is given. [Supported in part by ONR.]
Field experiments of the transmission of helicopter noise from air to water
Authors: Book Wayne J. , Main Geoffrey L.
Published: October 2005

Field experiments conducted during August 1978 are described. In a first series of experiments, a large helicopter flew over level solid ground and the sound was recorded by two microphones. The helicopter flew at altitudes of 25, 50, and 100 ft; data was taken both for hover and for low speed flight. The second series of experiments was carried out with the helicopter flying over water (20 m in depth). The sound level was recorded by a nominally vertical string of hydrophones at depths of 1, 10, and 20 m. The FM analog tape data was transferred to digital tapes with a sampling rate of 20 000 samples per s. Computer programs were written to extract the frequency spectrum of the received noise. The present paper discusses the consistency of the data and its application to the problem of prediction of sound underwater from previous measurements over solid ground. The errors introduced by the deviation of the helicopter trajectory in one set of experiments from that in the second set are assessed quantitatively using a recent theoretical model developed by W. J. Hadden, Jr. [Work supported by Naval Coastal Systems Center.]
Perceptual trading relation between aspiration amplitude and VOT
Authors: Repp Bruno H.
Published: October 2005

Two experiments demonstrate that the amplitude of (appropriately band-limited) aspiration noise is a cue for the distinction between voiced and voiceless syllable-initial stop consonants in English, and that it can be traded for aspiration noise duration, i.e., voice onset time (VOT). In Experiment I, the category boundary on a synthetic VOT continuum (/da/-/ta/, without bursts) was found to be a linear function of the amplitude ratio between the aspirated and vocalic portions. In Experiment II, a natural release burst was added to the synthetic stimuli, and burst and aspiration amplitudes were varied orthogonally. Both factors affected the voicing boundary in the expected direction but not independently; however, their interaction may have been due to backward masking of weak bursts by strong aspiration noise. If the data are adjusted for this presumed masking effect, the effect of burst amplitude seems very small compared to that of aspiration amplitude. Thus, other things equal a synthetic stop sounds the more voiceless (i.e., aspirated) the more aspiration there is, the perceived amount of aspiration being a function of both noise duration and amplitude. [Work supported by NICHD and BRS.]
Keyboard and electronic musical instrument research: Status and outlook
Authors: Martin Daniel W.
Published: October 2005

An acoustical review and a limited projection of the status of keyboard musical instrument research and development, both mechanoacoustic and electronic.
A kinematic theory of the parametric array
Authors: Rogers Peter H.
Published: October 2005

The absorption limited parametric array is considered as a three-phonon interaction process. By examining the kinematics of such interactions an integral expression is obtained for the difference frequency beam pattern. The expression, which is valid whether the difference frequency is generated in the primary's nearfield, farfield, or both, involves only the primary's farfield directivity function. The expression agrees with experiment and reduces to all appropriate Limiting values with the “aperture factor” predicted by moffett and Mellen [J. Acoust. Soc. Am. 60, 581–583] arising in a natural manner. The result is obtained by integrating over all allowed (i.e., energy and momentum conserving) interactions of the primary wave phonons, with the primary wave vector distribution determined by consideration of the primary field distribution in an attenuating fluid. It is thus shown that parametric array beam patterus are independent of the nature of the interaction and can be obtained without recourse to “virtual sources.” An interesting corollary is that generation of difference frequency takes place entirely at the expense of the higher frequency primary since ωd(=ω1−ω2) phonons result from a process in which ω1 phonons are annihilated and ω2 and ωd phonons are created. [Work supported by ONR.]
Parametric acoustic sources of transient signals
Authors: Mello Peter , Moffett Mark B.
Published: October 2005

A theoretical treatment is formulated for transient, or broadband, parametric acoustic sources, i.e., parametric sources utilizing an arbitrarily modulated primary carrier signal. This treatment makes use of previously calculated characteristics for steady-state or narrow-band parametric sources. The theory does not include saturation effects. A FORTRAN computer program, using Fast Fourier Transforms, generates spectra and waveforms for specified modulation envelopes. Experimental results, in good agreement with the theory, are presented for a 30 μs pulse applied to a 4 mm diam, 2 MHz quartz crystal underwater projector. [Work supported by Naval Material Command and National Science Foundation.]
Using SEA to model vibration transmission in engine structure
Authors: DeJong R. G.
Published: October 2005

Understanding the vibration characteristics of engine structures is an important aspect of the overall effort to reduce engine noise. Recent advances have been made in the measurement and modeling of the vibration characteristics of the individual components in an engine structure for the purpose of developing an analytical model which predicts the vibration transmission of the complete structure. For components with a small number of resonant modes of vibration in the frequency range of interest, lumped parameter models can be developed which match the vibration response function quite accurately. However, for components with a large number of resonant modes in the frequency range of interest, it is more useful to employ Statistical Energy Analysis to model the average vibration response. This paper demonstrates the comparability of these two types of models and shows some examples of the use of a complete vibration model in identifying component design changes which will result in engine noise reductions. [Work supported by EPA/ONAC.]
High-frequency vibration isolation of parallel beams and plates
Authors: Powell Robert E.
Published: October 2005

A statistical energy model has been derived for the flexural vibrational power transmission from a shaft to a point-connected flat plate. Coupling loss factors from shaft to plate are calculated using infinite beam and plate impedances along with lumped-parameter impedance models of the shaft support. The support is initially treated as a simple mass, transmitting point forces only. Experimental measurements of coupling loss factor for this configuration show very good agreement with the predicted values. The model is then exercized by introducing a spring, first between shaft and mass, then between mass and plate, and finally between two halves of the mass. The implications of the results for vibration isolation problems are discussed. [Research supported by NSF.]
The intelligibility threshold level (ITL): A new approach for evaluating performance of digital speech communications processors
Authors: Smith Caldwell P.
Published: October 2005

Intelligibility performance of voice processors has been typically specified in average intelligibility scores: scores presumably equaled or exceeded by half the underlying population of scores, assuming a normal distribution. However, extensive multispeaker testing of a wide variety of processors over the past decade has shown conclusively that: (1) populations of scores for processors typically deviate significantly from normal distributions; (2) mean scores of individual talkers in any multispeaker test typically show highly significant differences (a = 0.001); and (3) significant differences in mean scores for phonetic features are also typical. To further an implied objective of intelligibility testing: estimation of the intelligibility level equaled or exceeded by, say 90% of a population of talkers and listeners, a new approach to evaluation has been established, in which the intelligibility data is analyzed to determine intelligibility threshold levels (ITL's): levels equaled or exceeded by specified fractions of the population of scores. The method, based on nonparametric statistics of the Kolmogorov-Smirnov type, involves rank-ordering the population of scores and establishing cumulative distribution functions and their confidence limits, from which ITL's and associated confidence levels can be readily ascertained. Comparisons of processors typically show larger variations in ITL's than in mean intelligibility scores.
Aural harmonics: Estimation consistency from tone-on-tone masking measurements
Authors: Clack T. Dean
Published: October 2005

Amplitudes of inaudible “subjective” signals are inferred from tone-on-tone masking measurements. Several methodological issues are involved, e.g., the problems of how many and which measurements to include. To explore such sampling questions, sets of masked thresholds using a 250-Hz masker and a 500-Hz maskee are determined at various phase angles. These measurements are distributed into subsamples of (n =) 13, 9, 7, 5, or 4 each, and repeated estimates of the second aural harmonic inferred. For n > 4, these estimates averaged to within ± 1 dB of one another, although individual estimates vary considerably with the smaller n's (to 13 dB with n = 4]. Sets of five consecutive measurements also were analyzed when the masker : maskee phase angle was incremented in 90° steps through two complete periods (of the maskee). Again, the reestimates for individuals can vary by at least 6 dB, but the averages over seven subjects remain within 2 dB of each other. Apparently, for n's of six or more, it matters little which masker: maskee phase angles are used to estimate the aural harmonic amplitudes.
Additional Data on Underwater Sound Localization
Authors: Hollien Harry , Lauer J. L. , Paul Patricia
Published: October 2005

Theory would predict that humans exhibit little or no ability to localize sounds underwater. In order to test this assumption, three experiments were carried out on diver/subjects positioned in DALS (diver auditory localization system) at a depth of 40 ft in the NRL Underwater Sound Reference Division Bugg Spring facility. In the first two studies, they responded to four different signals (250, 1000, 6000-Hz sinusoids and thermal noise) at a sound-pressure level of 110 dB (40 dB re mean underwater hearing threshold) transduced by five J-9 projectors 8 ft from the subject's head and at angles of 0°, 45°, 90°, 270°, and 315°. In the first experiment, 17 divers, who were free to move their heads, obtained mean correct scores of (1) 250 Hz: 50%; (2) 1 kHz: 39%; (3) 6 kHz: 34%; (4) noise: 53%; and (5) over all: 44%. In a second study, seven of the subjects replicated this procedure with their heads in a fixed position; scores were almost identical (over-all mean 41% vs 47%) except for the first (250-Hz) condition; here a drop-off from 61% to 37% was observed. Also, latency of response was less for the fixed-head procedure. In the third study, seven diver/subjects were given training in localization by providing them with information concerning the correctness of their responses (30 trials each) for 25 pps and 1000-Hz signals. Mean improvement on a post-test was noted; in fact scores were over 30% better than on the pretest. Implications of these findings with respect to theory are discussed.
Probe−tube microphone assembly
Authors: Villchur Edgar , Killion Mead C.
Published: October 2005

A method of making in−the−ear probe−microphone measurements is described, using readily available equipment. The procedure is designed to make such measurements easier and safer than they have been in the past.

Subject Classification: 85.62; 65.80.
On a method of superresolution of a limited number of point targets using ultrasonic scanning
Authors: Ikeda Osamu , Sato Takuso
Published: October 2005

To get high−resolution reconstructed images under the conditions of a restricted scanning size and a long wavelength of ultrasonic waves, a method of superresolution imaging is proposed. The basic idea is to multiply computationally the phase of the detected signals and to eliminate the extra deteriorating terms also by means of computational manipulations, and to reconstruct the image using the resulting signals. The principle and some results by computer simulation showing the effectiveness of the method are shown, though the technique is useful only for a limited number of a few point targets at most, and it also requires a priori knowledge of the target.

Subject Classification :35.65.
Velocity−dispersion measurements in liquids between 3.3 MHz and 70.2 MHz
Authors: Rogez D. , Bader M.
Published: October 2005

An improved differential phase−comparison method for the measurement of small changes in ultrasonic velocity is described. Velocity dispersion measurements can be carried out in the frequency range from 3.3 to 70.2 MHz; the relative accuracy is one part in 105; only 12 cm3 of liquid are required. Results are presented for aqueous and organic solutions, which are in good agreement with known ultrasonic−absorption data.

Subject Classification: 35.24, 35.35.
Radial vibrations of axially polarized piezoelectric ceramic cylinders
Authors: Adelman Nachman T. , Stavsky Yehuda , Segal Emanuel
Published: October 2005

This paper is concerned with the steady−state radial−shear vibrations of an axially polarized piezoelectric ceramic tube of infinite length, whose cylindrical surfaces are either traction−free or subjected to a relative displacement. Open− and short−circuit resonant frequency equations are formulated and the deviation between frequency pairs examined numerically. Fundamental resonant frequency curves are given for traction−free PZT−4 cylinders having arbitrary geometry, which approach the infinite plate solutions for large radius/thickness ratios. The theory is applied to the vibrations of annular accelerometers operating in the radial−shear mode.

Subject Classification: 40.26.
Intensity fluctuations of aircraft flyover noise
Authors: Gonter Robert H.
Published: October 2005

Aircraft flyover noise was recorded and played back with a peak detector to detect the intensity fluctuations. The intensity fluctuations were digitized and plotted. Jet noise fluctuations appear more jagged, while prop noise fluctuations are smoother. Spectral analysis of fluctuations shows most of the power is at low frequencies and decreased by 80 dB between 0 and 5 Hz.

Subject Classification: 28.45; 50.50, 50.55.
Acoustic diffraction by two concentric coaxial rigid spherical caps
Authors: Vaid B. K. , Jain D. L.
Published: October 2005

The paper deals with the diffraction of time−harmonic axisymmetric low−frequency acoustic waves by two concentric coaxial rigid spherical caps. An integral equation technique is used to solve this boundary value problem. Formulas are derived for the farfield amplitude as well as the scattering cross section when the incident wave is a low−frequency plane wave traveling along the common axis of the two caps. By taking an appropriate limit, the solution for the corresponding problem for a rigid spherical cap is also presented.

Subject Classification: 20.30.
SOFAR transmission fluctuations produced by a Rossby wave
Authors: Baer R. N. , Jacobson M. J.
Published: October 2005

SOFAR ray geometry, travel time, and spreading loss are considered for a still−ocean region in which the sound speed is a bilinear function of depth. A sound source is placed above and a receiver below the SOFAR axis at depths corresponding to the same sound speed. Next, an assumed Rossby wave is introduced, and its effect on the phase of the total acoustic field is determined. It is found that the amplitude of phase variation can vary by a factor of three as the source and receiver depths are altered. Our theory is applied to sound transmission in the Sargasso Sea area, and phase variation values are determined which are in general agreement with experimental results.

Subject Classification: 30.20.
Evaluation of the intensity method of sound power determination
Authors: van Zyl B. G. , Anderson F.
Published: October 2005

If the sound power of a source is obtained by forming the surface integral of the normal component of the sound intensity vector, the shape of the surface of measurement is unimportant and projection error only occurs as a result of nonideal polar response of the composite microphone. Furthermore, the sound power of the source may be determined selectively in the presence of background noise originating outside the closed surface. If a sufficient number of points of measurement are taken, the error depends mainly on the signal−to−noise ratio at each point and on the accuracy of the measuring system.

Subject Classification: 50.25, 50.85.
Preservation of information in matrix filtering of a coherent signal correlated with noise
Authors: Bordelon Derrill J.
Published: October 2005

With the concept of mutual information about a coherent signal in additive noise, where signal and noise are assumed jointly Gaussian, it is shown that information is preserved, in a trivial way, by block diagonal matrix filters of full rank and by weighting and summing matrix filters of rank 1 which are identical, within a multiplicative scalar, with the following matrix estimator filters: (1) maximum signal−to−noise ratio estimator, (2) best linear unbiased estimator, (3) minimum mean−square error estimator, (4) maximum liklihood estimator, and (5) maximum a posteriori estimator.

Subject Classification: 60.XX.