bluuebunny/crossref_metadata_2025 huggingface dataset:
Acoustic imaging for medicine and commerce: A survey of new quantitative ultrasound methods
Authors: Johnson S. A. , Greenleaf J. F. , Rajagopalan B. , Christensen D. A.
Published: October 2005

Examples are given of how new acoustic imaging techniques developed at Mayo and elsewhere provide quantitative images in two or three dimensions of useful acoustic interaction parameters such as refractive index, acoustic absorption, reflection coefficient, acoustic impedance, acoustic speed dispersion, material temperature, fluid turbulence, and even the three-vecter components of fluid velocity. Newly developed transmission techniques include methods to (1) provide quantitative images of refractive index and absorption in the human breast for diagnosis of breast cancer, (2) provide two- and three-dimensional (non-Doppler) images of the three-dimensional vector components of fluid motion itself, (3) potentially permit reconstruction of tissue temperatures for monitoring of control cancer hyperthermic therapy. Newly developed reflection techniques include (1) a real-time medical B-scan and Doppler flow profile carotid artery scanner and (2) methods for obtaining images of absorption, attenuation, reflection, and impedance (potentially at a resolution of about one-half wavelength). Methods are given to correct for degradation due to refraction and absorption. Potential medical and commercial applications for these techniques will be discussed. [Work supported by NCI CB-64041, NHLBI HT-4-2904, RR-00007, and NIH HL-00170-03.]
Optical video disk review
Authors: Whitman R. L.
Published: October 2005

Transmissive and reflective optical video disk systems will be described. Physical characteristics of the disks and players will be discussed in some detail. Video disk mastering and replicating processes will also be covered. The optical readout process will be analyzed in a manner similar to that used to explain the operation of optical detection systems for acoustic surfacĕ waves. Some mention will be made of applications other than consumer television which can utilize the high storage density, low replication cost, and random access capability of the video disk.
Thin-film acousto-optic interactions and devices
Authors: Tsai Chen S.
Published: October 2005

Acousto-optic interactions involving surface acoustic waves and optical waves in a planar optical waveguide has been a subject of considerable interest in recent years. [See, for examples, C.S. Tsai et al., Proc. IEEE, 64, 318 (1976); C.S. Tsai, 1975 Ultrasonics Symp. Proc.; IEEE Cat. 75 CHO-994-4SU, pp. 120–125.] This interest results from the fact that diffraction efficiency of the resulting thin-film acousto-optic devices can be substantially higher than that of their bulk-wave counterparts and that such devices are more compatible with future wide-band integrated and fiber optic systems. In this paper a selective review of the more recent progress in this subject area is made. First, basic interaction mechanisms, configurations, and key acoustical and optical parameters are discussed. A number of wide-band device configurations which employ isotropic/anisotropic Bragg diffractions and multiple-surface acoustic waves together with the results obtained using optical waveguides in LiNbO3 substrates are then described in detail. Also to be given are the typical performance figures that have been obtained with applications using the resulting wide-band devices, emphasizing those being carried out at the author's institution, including high-resolution light beam deflectors, spectrum analyzers, convolvers, pulse compressors, and tunable optical filters. Finally, the comparative advantages of guided-wave acousto-optic devices over their bulk-wave counterparts will be discussed. [Work supported by NSF, ONR, AFOSR, and BMDATC.]
Application of digital inverse filtering techniques to sound field spectral flattening
Authors: Talkin David , Opauski Michael
Published: October 2005

An acoustic noise field with an essentially flat spectrum from 200 Hz to 5 kHz was needed for a hearing-aid response study. The impulse response of a Brüel and Kjaer hearing-aid test box was determined by driving its speaker with a short-duration pulse and digitizing the output of the reference microphone amplifier. This impulse response was then transformed via an autocorrelation linear predictor analysis into filter coefficients which were used to filter the output of a random number generator. This sequence was desampled and used to drive the test box speaker. The resultant noise field, as measured through the reference microphone, was found to be flat within ±1.5 dB from 80 Hz to 5 kHz. Considerations in the use of this method and its applicability to other sound field spectral flattening situations will be discussed.
Physical model of the external ear
Authors: Wojcik J. J.
Published: October 2005

Little physical information has been published on the size and shape characteristics of the pinna. These details are necessary to establish a standardized replica of the ear for test purposes; and to ensure accurate relation between psychoacoustic measurements and the ear's physical properties. This paper describes anthropometric measurements on 35 human ears. A detailed description of the pinna is qualified with a statistical analysis of the data. The orientation of various parts of the pinna relative to the side of the head are computed. The concha opening is measured. The variation of the concha volume is calculated when a supra-aural earphone is applied against the ear. Data is computed for all 35 ears and also for groups with respect to age and sex (10 females, 25 males).
Preliminary analyses of speech sounds with a digital model of the ear
Authors: Miller J. D. , Engebretson A. M. , Spenner B. F. , Cox J. R.
Published: October 2005

We believe that current lack of simple, invariant relations between the acoustic description of speech sounds and their phonetic features is due, in part, to the failure of current analyzers to adequately model the analysis performed by the auditory system. Our approach is to use models of the outer, middle, and inner ears for speech analysis. The sound-field-to-stapes velocity transformation is simulated with a digital filter consisting of two complex-conjugate pole pairs with center frequencies of 2.4 and 4.0 kHz and bandwidths of 0.7 and 2.0 kHz, respectively. This filter is consistent with descriptions of the outer ear provided by Shaw and of the middle ear by Zwislocki. The stapes is then used to drive a digital, 50-section implementation of the two-dimensional cochlear model developed by Spenner and Cox. Because the solutions of the model are in the time domain, its transient response is included in the speech-wave analysis and the need for arbitrary data windows and assumptions of periodicity are eliminated. Vowels and plosive consonants spoken by female and male talkers have been processed by the model. Sample traveling waves and their envelopes will be shown. [Supported by RR 00396 and NS 03856.]
Limitations in the application of cochlear microphonic recordings in the study of acoustic trauma
Authors: Durrant J. D.
Published: October 2005

Recordings of the cochlear microphonic were made in the cochleae of guinea pigs utilizing differential intracochlear electrodes placed either in the first or third turns. The animals were exposed to pure tones of either 10 kHz (turn I group) or 750 Hz (turn III group) typically presented at 120 dB SPL (monitored in front of the eardrum in a closed sound system) for a duration of 15 min. These frequencies thus approximated the best frequencies of the respective recording sites. The CM pseudothresholds (1-μ V isopotentials) observed following exposure were quite variable between subjects despite the homogeneity of the subjects within each group. This variance existed not only in the overall postexposure sensitivity of the preparations but also in the pattern of the observed CM “losses” across frequency and between turns (groups). In general, the findings reflect the involvement of factors in the gross microphonic pickup which operate, largely, independently from the pathology itself to the extent that the so-called electrical audiogram should not be expected necessarily to mimic the behavioral audiogram (that is, accurately reflect the behaviorally determined threshold shifts). The cochlear microphonics are perhaps best suited for merely ranking the effects of acoustic trauma. Certainly, any attempt to infer much more information from such data must be made cautiously. [Work supported by DRF.]
Effects of oral structures on airway conductance during normal speech
Authors: Warren Donald W. , Davis J. U. , Allen G. D.
Published: October 2005

The production of fricative consonants requires precise articulatory movements. The purpose of this study was to develop a technique for measuring articulatory movements in normal speech so that comparisons could be made with defective speech. Specifically, the influence of tongue position and oral airway opening was studied in terms of effects on airway pressure and air flow. Measurement of pressure differential was made by appropriately placing two catheters, connected to a pressure transducer, on both sides of the structure(s) under study. Air flow was recorded by a pneumotachograph connected to a face mask. The data reveal a high correlation between size of oral constriction and volume rate of air flow, suggesting that respiratory effort increases to maintain a certain magnitude of pressure for each fricative sound. The results also demonstrate that pressure magnitude is not related to size of constriction. In normal speakers oral airflow dynamics appears to be controlled primarily at the anterior portion of the oral cavity with the middle and posterior aspects of the tongue demonstrating a minimal influence. [Work supported by NIDR.]
Induced transglottal pressure variations during voicing
Authors: Rothenberg M. , Mahshie J.
Published: October 2005

The effect of variations in the transglottal driving pressure on the glottal air flow waveform is basic to the understanding of the functioning of the laryngeal sound source during speech. A number of laboratories have reported experimentally manipulating the transglottal pressure by mechanically varying the subglottal or supraglottal pressures. If the variations occur in about 100 msec or less it can be assumed that there were no significant compensatory changes in laryngeal or respiratory muscle activity. In this study the supraglottal pressure was varied by having the subject speak into a box having four 15-in. loudspeakers in its walls. The loudspeakers generated variations in supraglottal pressure at preset points in the vocalization. The subject spoke through a circumferentially vented pneumotachograph mask which yielded the glottal air flow waveform after inverse filtering. Reported here are the first attempts at using this apparatus to map out the relationships between transglottal pressure, voice fundamental frequency, peak air flow during the glottal pulse, and glottal pulse width.
Peak oral air flow magnitudes associated with affricate and plosive consonant cognate productions in Hindi
Authors: Dixit R. Prakash , Brown W. S.
Published: October 2005

Peak values of air flow rate were obtained for the productions of plosive and affricate consonants in CVCVC nonsense utterances embedded in the carrier phrase, “ye-hae” by three normal adult speakers of Hindi. Air flow values for these consonants were found to be generally higher for stressed position when compared to unstressed position. Aspirated stops showed substantially higher air flow rates than their unaspirated counterparts. Magnitudes of air consumption for voiced versus unvoiced stops under unaspirated conditions were quite similar, whereas under aspirated conditions the magnitudes were somewhat variable. These results suggest that the rate of oral air flow for various stop consonants depends on the condition of the glottis at or immediately after the release of articulatory occlusion (assuming that subglottal respiratory activity for these stops is uniform).
Mechanical properties of single motor units in the anterior belly of the digastric
Authors: MacNeilage Peter F. , Sussman H. M. , Powers R. K. , Westbury J. R.
Published: October 2005

Electromyograms were recorded from three subjects during steady, low-rate discharge of single motor units of anterior belly of digastric, accompanying maintenance of isometric tension. The overall tension developed in muscles subserving mandibular depression was recorded from a force transducer situated inferior to the mandibular symphysis. The force transducer output for the 100-msec period immediately following each of several hundred discharges of a single unit was averaged to obtain a characteristic tension curve for the contractile response (twitch) of that unit. Contraction times of units (time from initiation of tension increase to peak tension) ranged from 16 to 42 msec. Recorded peak twitch tensions ranged from 0.1 to 1.8 g. Consistent with the Size Principle [Henneman et al., J. Neurophys. 28 (1965)] which is based primarily on findings from animal and human skeletal muscle, and also with a study of human masseter and temporalis [Yemm, J. Physiol. 265 (1977)] later recruited (larger) units possessed higher twitch tensions. Contrary to the Size Principle, but in agreement with results from human masseter and temporalis, contraction times were not shorter in later recruited motor units.
Physiological control of sibilant durations
Authors: Hamlet S. L. , Stone M. , Cullison B.
Published: October 2005

In an earlier paper [S. Hamlet and M. Stone, J. Acoust. Soc. Am. 61, S7 (A) (1977)] tongue overshoot was reported to occur on /s/ and /z/, when subjects first attempted to speak with a dental prosthesis with 4 mm of alveolar thickness. The duration of frication for /s/ under that condition was no longer than for /s/ spoken when wearing a thin prosthesis, or when speaking without a prosthesis. Duration of /z/ was greater than in natural speech. Multiple explanations exist: (1) Timing of tongue contact for /s/ may have been altered as a compensatory attempt. (2) Duration of /s/ frication may be governed more closely by laryngeal adjustments than by tongue gestures. Special modifications of a palatography system were made so that measurements of duration of tongue contact could be made with an accuracy of 1 or 2 msec. Measurements of cessation and onset of voicing were made from pretracheal accelerometer signals. Preliminary results are discussed, based on a small number of subjects. [Work supported by NIH.]
Computational model of the tongue: A revised version
Authors: Kakita Y. , Fujimura O.
Published: October 2005

A previous model of the tongue as a three-dimensional static elastic system [S. Kiritani, K. Miyawaki, O. Fujimura, and J. E. Miller, J. Acoust. Soc. Am. 57, S3(A) (1975)] has been revised for studying the physiological functions of intrinsic and extrinsic muscles in articulation of speech sounds, including apical consonants. The new version of the finite-element computation uses 86 tetrahedra to represent one-half of the tongue, assuming symmetry with respect it the midsagittal plane. The entire set of tetrahedra is subgrouped into 30 functional units, each of which can be given independently specified stress and strain in any direction to simulate muscle contraction. In addition, any node points can be pulled by external forces. The computation procedure is substantially improved over the previous model, in both accuracy and speed as well as in convenience for the user. A preliminary study of articulatory gestures using the model will be reported on with particular emphasis on the stability of the acoustic consequence when articulatory gesture is slightly varied.
Role of rugae in speech adaptation to dental appliances
Authors: Garber Sharon R. , Speidel T. M.
Published: October 2005

Palatal rugae may aid in lingual placement during speech production by providing landmarks for the tongue. The role of palatal rugae was assessed in the present study by asking 16 normal speakers to read while wearing two dental appliances, one with rugae and one without rugae. Each speaker was tape recorded before each appliance was placed, immediately following placement, one hour after placement and immediately after removal of the appliance. With both appliances, articulation errors increased following placement of the appliance, decreased slightly after the appliance had been worn for an hour and decreased further when the appliance was removed. Presence of rugae had no significant effect on the number of errors produced when the appliance was in place. However, following removal of the appliance with no rugae, errors decreased to the preappliance level. By contrast, following removal of the appliance with rugae, errors remained substantially higher than the preappliance level. [Work supported by an NIDR post-doctoral fellowship awarded it S.R. Garber.]
Anticipatory coarticulation: Some implications from a study of lip rounding
Authors: Bell-Berti F. , Harris K. S.
Published: October 2005

The anticipation of articulatory features, in particular, lip rounding in anticipation of a rounded vowel, has been reported to occur as many as four segments before the segment for which the feature is specified. Recently we reported [F. Bell-Berti and K. S. Harris, J. Acoust. Soc. Am. 60, S63(A) (1976)] that the motor commands for the rounding gesture for /u/ begin a fixed time before voicing onset for the vowel. This timing is unaffected by the number of preceding consonant segments or the location of syllable boundaries in the preceding string. Thus, the initiation of lip rounding appears to be linked to other features of the vowel articulation, much as movement of the velum has been shown to be linked to lingual articulation [R. D. Kent, P. J. Carney, and L. R. Severeid, J. Speech Hearing Res. 17, 470–488 (1974)]. Analysis of these and additional data lead us to postulate a system of motor organization which is segment rather than feature based. [Work supported by NINCDS grant NS 13870, NIDR grant DE 01774, and NINCDS postdoctoral fellowship 1 F32 NS 05332-01A1.]
On the relationship between the geometric and parabolic propagation models and an extension of both
Authors: McCoy John J.
Published: October 2005

A parabolic wave theory is considered in which the energy flux through an elemental aperture is required to lie in directions measured relative to a “local” principal propagation direction that can be termed narrow angled. This can be compared to the more commonly encountered theory in which a “global” principal propagation direction is used. A propagation model is then developed which is formulated in terms of an acoustic field measure termed the acoustic intensity angular spectral density. The relationship between this propagation model and the geometric theory is considered, with the latter being recovered in the appropriate limit. A sample calculation is carried out for the behavior of a beamed signal in the vicinity of a point caustic.
Path integral approach to the parabolic approximation
Authors: Palmer D. R.
Published: October 2005

We have examined the parabolic approximation using Fradkin's path integral formalism [E. S. Fradkin et al., Rev. Nuovo Cimento 2, 498–559 (1970)]. The results confirm a previous study [D.R. Palmer, J. Acoust. Soc. Am. 60, 343–354 (1976)] where the parabolic approximation was shown to be equivalent to two approximations: (1) a straight-line geometric optics approximation applied in the horizontal plane (HGOA) and (2) a stationary-phase approximation (SPA). For the random component of the sound speed we obtain the usual results [V.I. Klyatskin and V.I. Tatarskii, Sov. Phys. JETP 31, 335–339 (1970)]; namely, if the Markov approximation is satisfied, the first-order correction to HGOA vanishes and SPA is valid. For the deterministic component of the sound speed we find the first-order correction to HGOA does not, in general, vanish and SPA implies additional constraints which go beyond the Markov approximation. Moreover, we find, even in the simple situation considered by Klyatskin and Tatarskii, that range-dependent terms appear in the first-order correction to SPA and the parabolic approximation is equivalent to SPA only if the sound speed is range independent.
Propagation in random oceans using a radiation transport code
Authors: Wilson H. L. , Tappert F. D.
Published: October 2005

A wave kinetic formulation of acoustic propagation in random media was recently developed by Besieris and Tappert [J. Math. Phys. 14, 1829 (1973); 17, 734 (1976)] which leads to an equation for the propagation of the mutual coherence function having the form of a radiation transport equation. A numerical method for solving such equations which has been successfully employed in other applications is described in Carter and Cashwell [“Particle Transport Simulation with the Monte Carlo Method,” ERDA Critical Rev. Ser., TID-26607 (1975)]. We have adapted this technique and developed an acoustic propagation model that takes into account both scattering from rough surfaces and scattering from volume fluctuations. Code output was successfully compared to ray-tracing predictions of transmission loss in the absence of scattering. We have used the code to calculate the effects of wind-driven waves on scattering acoustic energy out of surface ducts and have compared to the results of measurements reported by Pedersen and Gordon [J. Acoust. Soc. Am. 37, 105 (1965)]. [Work supported by ONR]
Pulse propagation in the ocean. II: The parabolic equation method
Authors: Kutschale Henry W. , DiNapoli Fred R.
Published: October 2005

The Fast Field Program (FFP) is a convenient method to compute the exact integral solution of the wave equation derived from harmonic sources in multilayered media. Solid layers as well as liquid layers may be included in the model. Thus, elastic subbottom layers are included in the model and, for Arctic propagation, an elastic ice sheet is also included. The sound field is computed as a function of range for each detector depth employing the fast Fourier transform (FFT) algorithm. The FFP has been extended to compute waveforms from broadband sources as a function of range and depth by Fourier synthesis employing the FFT algorithm. The input for this synthesis as a function of frequency is computed by the FFP. The computer program has been used to model SOFAR propagation in the central Arctic Ocean. Computed waveforms for the Arctic channel are in excellent agreement with field data. The computations clearly show the evolution in range of the dispersive signals observed in the Arctic channel. At any selected point in range and depth the waveform may be reversed in time and propagated back through the channel, showing the evolution of pulse compression to the range at which the ocean filter is matched and pulse spreading beyond this range.
Mode coupling in a sound channel with range-dependent parabolic velocity profile
Authors: Chwieroth F. , Überall H.
Published: October 2005

The sound propagation problem is separable in a channel with adiabatic range dependence, and has been solved analytically to this approximation for the case of a parabolic sound velocity profile with linear range dependence [R. D. Graves, A. Nagl, H. Überall, and G. L. Zarur, Acustica (to be published)]. The depth functions are here Hermite functions, and the range functions Airy functions. For the nonadiabatic case (nongradual range dependence), the normal modes couple, and the coupling terms can be evaluated as the harmonic oscillator matrix elements of quantum mechanics. The coupled problem is then soluble in an almost exact fashion,. by carrying out a series of diagonalizations of the coupled range equations. We have evaluated the coupled range equations in this fashion, and examined the energy transfer between modes. The transmission loss was calculated and compared with that of the uncoupled case. [Supported in part by NRL Code 8120.]
Comparison of experimental broadband and narrow-band propagation-loss data versus receiver depth
Authors: Jensen E. P. , DiNapoli F. R.
Published: October 2005

In November 1973 a series of experiments were conducted in the Tongue of the Ocean, Bahama Islands, as part of Project BILL (Below and In Layer Propagation Loss). A 200–700-Hz pseudorandom-noise projector was slowly towed first in the surface duct at 76 m and then below the duct at 122 m. These runs were followed by a cw tow at 76 m during which the source was operated at 300 Hz. Ranged varied to 9 km and the transmitted signals were received at five hydrophones located in and below the surface duct for each of the tows. Comparisons of the results at the different receivers are made to show the effects of ln, cross, and below layer narrow-and broadband propagation. Results obtained when the source and the receiver were at the same depth showed losses which were noticeably less than those obtained at other depths. For this situation model predictions required the inclusion of complex source angles in order to obtain better agreement. [Work supported by NUSC.]
Effects of sound-speed equations on critical depths in the oceans
Authors: Mackenzie Kenneth V.
Published: October 2005

Critical depths (and consequently depth excesses required for convergence zone propagation) are generally computed from depth, temperature, and salinity data. In 1972, V. A. Del Grosso published a 19-term formula that differs, up to 1.5 m/sec at depth, from the widely accepted 23-term equation of W. D. Wilson (1960). A simple equation to determine sound speed from temperature, salinity, and depth, instead of pressure, was developed for computations, employing a pocket calculator [Joint Oceanographic Assembly, Edinburgh, Sept. 1976]. To generate a practical equation, 15 worldwide deep stations of high quality were selected, and depths corresponding to a number of pressures were computed for each profile. A polynomial of only nine terms fits Del Grosso sound speeds for all stations to depths of 8000 m with a standard deviation of only 0.07 m/sec. The simplified equation facilitates comparisons with Wilson, since depth, temperature, salinity, and Wilson sound speed appear in practically every profile tabulation. Results will be presented for a number of realistic situations.
Quantitative model/measurement comparisons of propagation-loss mean and fluctuating components
Authors: Lauer R. B.
Published: October 2005

In November 1973 a propagation-loss experiment was performed in the Tongue of the Ocean, Bahama Islands, under carefully monitored conditions with regard to environmental parameters and source/receiver geometry. A projector transmitting a 300-H tone was towed at a depth of 76 m in the surface duct and receiver to a range of 9 km. A model, the Fast Field Program, was used to simulate propagation-loss results using a measured sound-speed profile and bottom parameters selected to achieve a good eyeball fit in the region dominated by bottom-reflected energy. The measured and simulated results were then compared in terms of range-dependent mean and range-dependent standard deviation as obtained in moving windows. The mean level of the model oscillated with range in the bottom bounce region as contrasted to the measured data which exhibited an essentially constant mean level. Variability about the mean was greater for the simulated results than that of the measured data by 2–3 dB. This disparity is accounted for by fluctuation mechanisms shown to be present in the measured data besides multipath interference which is the sole cause of variability in the model simulated data. [Work supported by NUSC.]
Effect of onset asymmetry upon the perceived onset of a CV during dichotic presentation
Authors: Hughes Larry F. , Holtzapple Pat , Berlin Richard I.
Published: October 2005

Data were gathered from four well-practiced subjects in a 2AFC paradigm which was subject driven, and allowed as many repeats of the stimulus pair as the subject requested prior to making his response. The stimuli consisted of the six stop-consonants followed by the vowel /a/. These natural speech CV's have served as stimuli in numerous dichotic speech perception studies. The results indicated that when the stimuli were separated by no more than 15 msec, the subjects could be divided into two homogeneous groups. One group was able to reliably determine a parallel pair of perceptual onset functions (one for each ear) which were of moderate slope and were separated by approximately 13 msec. The other group of subjects showed a reversal of the ear asymmetry and their perceptual onset functions had a zero slope. The results will be discussed in terms of its relevance to the temporal variables involved in dichotic CV experiments. [Supported by National Institutes of Health, USPHS Grant No. NS 11647-03.]
Effects of lateralization on selective and divided attention
Authors: Friedman C. , Pastore R. E.
Published: October 2005

In most two-channel selective attention studies, each earphone has been assumed to represent a separate perceptual input channel. However, at least for the detection and recognition of relatively simple acoustic signals, separate earphones do not appear to represent separate perceptual input channels. In an effort to clarify the concept of “channel,” we ran a set of monaural and binaural experiments to investigate the effects of differences in earphone, frequency, and perceived lateralized location on the adequate definition of input channel. Observers monitored increments in intensity of two simultaneously presented frequencies (2000 and 4200 Hz). In the first of two binaural conditions, the two signals were equivalent in loudness and thus, theoretically, both occupied central locations. The second of the binaural conditions spatially separated the lateralized locus of the frequencies by reducing the intensity of each in one (but not the same) earphone so that the 2000- and 4200-Hz tones appeared to be lateralized to the left and right, respectively. Analogous monaural conditions were run. Observers ran monaural and binaural conditions involving signal stimuli and two stimulus presentations with selective and divided attention tasks. Preliminary results show equivalent performance across the two binaural conditions, with the binaural conditions tending to be poorer than the monaural conditions. This suggests that channel cannot be defined in terms of earphone or location of perceived image. Frequency appears to provide a more adequate definition for channel. [Research supported in part be a grant from NINCDS.]
Doppler shift in a medium with velocity gradients
Authors: Kolano R. , Rowlands R. O.
Published: October 2005

A basic formula for use in calculating Doppler shifts is shown to be dT/dt; this being the rate at which the travel time of the sound between source and receiver is changing. Examples of the application of this formula in a medium having velocity gradients of varying degrees of complexity are then given.
Propagation-loss fluctuations at long ranges from towed low-frequency sound sources
Authors: Fraser Ian A. , Franklin J. B.
Published: October 2005

The propagation of underwater sound at 30 and 115 Hz along an 1800-km path has been studied using sound sources supplied by the U.S. Naval Research Laboratory and towed by their research ship HAYES. From consecutive 5-min averages of narrow-band spectra of the received signal, time series of propagation loss were formed for each frequency, and filtered to remove trends. Following statistical analysis, a three-parameter phenomenological model involving the convolution of two gamma distributions afforded a good fit to the resulting probability distribution functions. Data recorded on the few occasions when the tow speed varied significantly suggest that the fluctuation rate is primarily governed by source tow speed rather than by instabilities in the medium. In addition, a monotonic decrease in the average fluctuation rate was observed with increasing range. Physical mechanisms to account for these results will be suggested.
Underwater sound propagation in a range-dependent environment
Authors: Zarur G. L. , Nagl Anton , George Jacob , Überall H. , Haug A. J.
Published: October 2005

For an ocean sound channel whose environmental parameters depend not only on depth, but in a gradual fashion also on range, the wave equation may be separated by the adiabatic range variation method of Pierce. For cases of less gradual range dependence, the range equations (and hence the modes) become coupled, and may be solved by diagonalization methods. This approach is used here to calculate underwater sound propagation in a channel with arbitrarily given range dependence, and also with arbitrary depth dependence of the sound velocity profiles, by employing Airy function solutions of segmentwise linearized problems. (For the coupling aspects, the simplification of segmentwise constant problems is made.) Our results are illustrated for a realistic deep water case with profiles from the Western North Atlantic, as well as a shallow water example from the Norwegian Sea, and are compared against the experimental transmission-loss data, and the results of calculations using other methods for the same cases. [Supported in part by NRL Code 8120.]
Improved energy-flux model for predicting range-averaged propagation loss in deep water
Authors: Fraser Ian A.
Published: October 2005

A pocket calculator program has been developed to predict propagation losses smoothed over range in situations where energy incident on the bottom can be considered lost, as in many deep water scenarios at long ranges. Input parameters are range, source frequency, sound-speed profile gradients (assumed bilinear) and the sound speeds at source, receiver, channel axis, and bottom. To determine the effective channel depth, an extension of a familiar expression for propagation loss in a channel [D. E. Weston, J. Sound Vib. 18, 271–287 (1971)] averages over the rate at which ray paths intersect the receiver depth. The model also accommodates Lloyd mirror effects, some finite wavelength effects, and horizontal gradients of sound speed. Since individual ray paths are not considered, convergence zones and statistical fluctuations are not reproduced. Other limitations of the model, further simplifications, and cases in which it outperforms more familiar models will be presented.
Quantitative acousto-optic image processing
Authors: Dardy H. D. , Bucaro J. A.
Published: October 2005

Acoustic fields visualized through acousto-optic transformation employing conventional Bragg, schlieren, or holographic techniques map an optical image which, in most cases, is different from the acoustic pressure distribution due to the nature of the diffraction process. Studies utilizing these techniques are for the most part qualitative and limited to low acoustic intensity where an approximate linear behavior exists. Current video techniques, coupled with an on-line minicomputer processor, allow digitization, storage, and linearization of the analog video image in reasonable time; a quantitative reconstruction and reproduction of the acoustic image is then performed digitally. This paper will present a working technique and philosophy developed for the quantitative linear reconstruction of a video image obtained from a diffraction-limited acousto-optic system. Aspects of the quantitative digitization and linearization process will be treated as well as digital techniques developed to store and extract information from the raw data image through signal enhancements, averaging and filtering, and pseudocolor image presentation.
Acousto-optic reconstruction of ultrasonic fields using Gaussian-Laguerre/Hermite formulation
Authors: Cavanagh Eduardo , Cook Bill D.
Published: October 2005

The reconstruction of ultrasonic fields from acousto-optic data is usually obtained using projection techniques based upon the spatial Fourier expansion; however, the spatial distribution of an ultrasonic field can be expanded in terms of Gaussian-Laguerre/Hermite function [B.D. Cook and W.T. Arnoult III, J. Acoust. Soc. Am. 59, 9–11 (1976)]. This paper explores the possibility of obtaining the expansion coefficients from the acousto-optic projection data.
Diffraction of a plane N wave of finite amplitude by a slit
Authors: Cobb Wesley N. , Blackstock David T.
Published: October 2005

A theoretical and experimental study of the diffraction of a plane N wave of finite amplitude by a slit is presented. The experiment was done in air. Plane N waves with peak pressures of 99, 47, and 29 mbar were normally incident on a 2.38-cm-wide slit in a rigid baffle. Schlieren and microphone measurements confirmed the following theoretical predictions: (1) The diffracted wave from each edge of the slit has an expansion front in the beam region and a compression front in the shadow region; (2) finite-amplitude effects cause the expansion fronts to catch up with the transmitted plane shock front; (3) the loci of intersection of the expansion fronts and plane shock are not lines parallel to the axis of the slit, as predicted in the linear theory, but rather slant inward and merge with the axis to terminate the beam region at a finite distance; and (4) an accelerated decay of the axial peak pressure occurs after the loci-axis merger. [Work supported by AFOSR and ONR.]
Wilson P. Tanner, Jr.: A general review of his scientific contributions
Authors: Sorkin Robert D.
Published: October 2005

Spike Tanner's scientific work, particularly his work with John Swets, Ted Birdsall, and David Green, had an immense impact on the development of theory and method in psychoacoustics. His research has also had a profound influence on experimental psychology, where his work forms one of the most frequently cited foundations of the area known as cognitive or information-processing psychology. Spike's research activities spanned a wide range of basic and applied science; from work in the psychology of hearing and vision, to applied work in education, criminal justice, and in military and space research. A common theme runs throughout: The human is a rational observer, continually testing hypotheses on the basis of environmental inputs, prior experience, and costs and payoffs. The work of his associates and students at Michigan has continued to expand the influence of Spike's ideas. The breadth of this work, and some of the research directions pursued by his students, will be discussed. Finally, some personal recollections of working with Spike will be shared.
Repeated observation of an uncertain signal
Authors: Swets J. A. , Birdsall T. G.
Published: October 2005

Data presented earlier on repeated observation of a signal with an uncertain parameter [partially described by J.A. Swets and J. Markowitz, J. Acoust. Soc. Am. 42, 1194(A) (1967)] are compared to a model not available then. The signal was a brief sinusoid; its uncertain parameter was frequency. On a given trial, containing 12 successive observation intervals with each followed by a detection response, the signal (if present) could be any one of eight frequencies. The primary model is that developed for an adaptive optimum receiver [L.W. Nolte, J. Acoust. Soc. Am. 42, 773–777 (1967)], which draws on a model for the detection of one of M orthogonal signals [L.W. Nolte and D. Jaarsma, J. Acoust. Soc. Am. 41, 497–505 (1967)]. In one comparison condition the signal frequency was constant over trials; in another, it varied within a trial. Results indicate that the human observer can adapt to changing knowledge of the stimulus—attending to a wide range of frequencies, or to a particular narrow range, as appropriate. [Work supported by NASA-Ames Research Center.]
Relationship between detectability and annoyance of low-level signals
Authors: Fidell Sanford
Published: October 2005

Annoyance is a strong function of signal level only at relatively high absolute levels. At levels below about 65 dBA, physical indices of exposure are poorly correlated with annoyance judgments. Since regulation of low-level-noise sources (which present negligible hearing damage risk and interfere little with speech) is now under consideration, a reliable index of the annoyance of low-level signals is needed. Preliminary evidence gathered in an empirical study of the relationship between detectability and annoyance of recorded noises heard in different backgrounds suggests that the detectability of low-level signals may provide such an index.
Reducing backward masking through temporal contralateral cueing
Authors: Puleo J. S. , Pastore R. E.
Published: October 2005

Our previous experiments showed that the contralateral cueing improved the detectability of a signal during monotic backward masking in the presence of a second, continuous masking noise [J.S. Puleo and R.E. Pastore, J. Acoust. Soc. Am. 60, S49(A) (1976)]. In the present experiment, we investigated whether any improvements in detectability during backward masking would be differentially affected by varying the amount of temporal (onset and offset) cueing information regarding the signal's occurrence. Temporal cueing information was provided independent of any correlation between other cue and signal parameters by introducing a gap in an otherwise continuous (Nμ) broadband masking noise in the ear contralateral to that performing the detection task. The data showed that while detectability during simultaneous masking was not significantly improved by any of the cues, there were significant improvements in detectability during backward masking—but only when the cue was simultaneous with and extended in time 1 msec before the observation interval. These findings are discussed in terms of their theoretical importance regarding the nature of backward masking. [Research supported by a grant from NINCDS.]
Standard errors and confidence intervals for measures of sensitivity
Authors: Dorfman Donald D.
Published: October 2005

It is quite simple to obtain maximum-likelihood estimates of the slope (b0) and intercept (a0) of the linearized ROC curve under the unequal-variance Gaussian assumption. If a measure of sensitivity is a function (f) of (a, b) and f is differentiable at (a0, b0), then it is not hard to show that f(â, b̂) is asymptotically normally distributed with mean f(a0, b0) and variance Var(f) = [σa2(∂f/∂a) + σb2(∂f/∂b) + 2σab(∂f/∂a)(∂f/∂b)]. This result was used to obtain large-sample variances for estimates of Δm, d′e, and A, the area under the ROC curve, based on maximum likelihood. Confidence intervals for these estimates are obtained in the usual manner. The maximum-likelihood method for estimating area is compared with standard non-parametric methods. The maximum-likelihood approach is recommended for a variety of good reasons.
Learning processes in perception
Authors: Fried Lisbeth S.
Published: October 2005

Signal detectability theory assumes observers who transform each observation into a form monotonic with likelihood ratio. Since evaluation of likelihood ratios presupposes knowledge of the stimulus distributions, preasymptotic variability in performance has been presumed to derive solely from shifts in decision criteria with feedback and not to shifts in the estimates of the distribution parameters themselves. These were assumed known [Dorfman and Biederman (1971); Dorfman, Saslow, and Simpson (1975); Kac (1962); Kubovy, Rapoport, and Tversky (1971); Kubovy and Healy (to be published); Larkin (1971); Schoeffler (1965); Thomas (1973, 1975)]. In this research a Gaussian Ideal Learner (GIL) is developed which estimates the five unknown parameters of two Gaussian distributions. Each observation is used to continuously update parameter estimates, while current parameter estimates are used to classify the observation. Estimates approach the true parameter values as n approaches infinity. Preasymptotic variability results from shifts in parameter estimates. Data from two experiments using normally distributed numerical stimuli and two using randomly distorted visual patterns are consistent with the model. Observers, operating without feedback, use the observations to estimate distribution parameters, and that preasymptotic variability is due to changes in parameter estimates and not merely to shifts in decision criteria. [Work supported by NSF Contract No. BNS-77-01211.]
Effect of stress on fundamental frequency of the voice
Authors: Scherer K. R.
Published: October 2005

The effect of stress on the fundamental frequency of voice was assessed in two studies with 16 and 15 subjects, respectively. In each study, student nurses either saw a pleasant nature film and had to give truthful reports about it and their feelings toward it to an interviewer (baseline situation), while in the stress situation they saw an extremely unpleasant medical film and had to deceive the interviewer about their feelings. Subjects' answers to seven interviewer questions were separately analyzed for fundamental frequency (using digital F0 extraction procedures). A methodological analysis showed that mean F0 for very short samples (20 sec) correlates very highly with mean F0 for the total interview. In both studies F0 was significantly elevated in the stress condition (mean difference +9.0 Hz, combined p = 0.0003). For 25.5% of the subjects a reversal was observed. A positive correlation between extent of F0 elevation under stress and low scoring on the CPI Achievement via Independence scale (partialing out Achievement via Conformance, r = 0.65, p < 0.01, N = 31) suggests that F0 response to stress may be determined by individual coping strategies. Mean F0 levels for different answers significantly differentiated between different stages of the interview, presumably related to changing transitory stress levels.
Uniform scaling as a method of vowel normalization
Authors: Lennig Matthew , Hindle Donald
Published: October 2005

Recent studies [P. Nordström and B. Lindblom, Proc. 8th Int. Congr. Phonetic Sciences (1975); G. Fant, STL-QPSR 2–3 (1975)] have raised the question of whether uniform scaling is an adequate method for vowel normalization. One weakness of these studies is that none of them controls for dialect differences between speakers. Vowels treated as the same may or may not have the same phonetic quality. In this study, we compare imitations of six steady-state vowels produced by linguists with phonetic training and subsequently selected for similarity of timbre by a panel of phoneticians. Vowel formant frequencies are estimated by linear prediction. The hypothesis to be tested is that there exists a monotonic transformation of the space defined by the first two formant frequencies such that the vowels of any speaker in transformed F1 × F2 space can be made to coincide with the corresponding vowels of any other speaker by multiplication with a constant speaker-dependent scale factor. This is a more general formulation of the uniform-scaling hypothesis than proposed by Nordström and Lindblom, who use untransformed F1 × F2 space. [Work supported by NSF.]
Formant frequencies and lip position in American English vowels
Authors: Linker Wendy
Published: October 2005

In order to determine which parameters of lip position could best be determined from formant frequencies, simultaneous frontal and lateral photographs were taken of the lips of ten speakers producing nine American English vowels, while their production was tape recorded. Formant frequencies were obtained using an LPC program and by measuring spectrograms. For the frontal view, horizontal and vertical distances of the mouth opening were measured, and the area was calculated. Laterally, protrusion was measured with reference to three points: (1) a fixed point at the upper front teeth, (2) the most forward point of lip contact, and (3) the corner of the mouth. Center-line measurements were taken from these points to one another and to (a) the most forward points of the upper and lower lips and (b) the line connecting these points. Also measured were the distances along the inner-lip surface from the most forward point of lip contact to the most forward points on the lips. Multiple regression techniques were employed to determine the correlation between the lip parameters and the formant frequencies.