bluuebunny/crossref_metadata_2025 huggingface dataset:
Coding of auditory information in pigeon vestibular nerve fibers
Authors: Wit H. P. , Bleeker J. D.
Published: October 2005

After fenestration of the lateral semicircular canal of the pigeon auditory stimuli (e.g., tone bursts) evoke phase-locked responses in the vestibular nerve. Single unit activity was measured in the ampullary branch of the vestibular nerve innervating the crista in the horizontal semicircular duct. The sound stimulus was delivered to the eardrum of the pigeon through a silastic tube. Most units show best phase lock for frequencies of about 700 Hz. Few units however seem to be “tuned” to lower frequencies. The lowest sound pressure level that has an observable influence upon the firing pattern of a unit is about 95 dB SPL at the units best frequency. The low-frequency slope of the tuning curve for a unit (sound pressure level versus stimulus frequency for a present synchronisation criterion) is approx. 18 dB/oct. The high-frequency slope is 25 dB/oct. If the roof of the membranous ampulla wall is directly stimulated with a piezoelectric vibrator, tuning of most fibers is not different from the tuning measured with stimulation by (sound) pressure variations in the ear canal. This rules out the middle ear system of the pigeon as the primary source for the observed tuning. The threshold for observable phase lock in direct stimulation is a vibrator amplitude of 20 nm, corresponding to a velocity of 0.1 mm/s at 700 Hz.
Real-time determination of multi-phase excitation for LPC speech synthesis
Authors: Hanson R. J. , Atal B. S. , Kohut J.
Published: October 2005

We describe a real-time implementation of a multi-phase excited LPC speech coder using Bell Laboratories DSP chips. A major advantage of multi-pulse excitation is that it produces natural-sounding synthetic speech and avoids both pitch analysis and voiced-voiceless decision. Pulse amplitudes and locations are determined by a noniterative, analysis-by-synthesis procedure which minimizes a perceptual distance criterion. The multi-pulse analysis is performed in fixed-point arithmetic on two DSPs. The analysis yields four excitation pulses for every 5 ms of speech, which are necessary for producing high quality synthetic speech. A companion real-time LPC area-parameter analyzer using DSP chips was described previously by Hanson and Olive [J. Acoust. Soc. Am. Suppl. 1 69, S18 (1981)].
Evaluation of a model for generating and displaying synthetic animated lip shapes
Authors: Montgomery Allen A.
Published: October 2005

This paper summarizes the development of the capability to synthesize on a vector-driven graphics system the lips of a talker producing phrase length utterances. Details of the algorithm for generating smoothly varying sequence of images using a phoneme string as input are given, including methods for controlling timing and specifying form and extent of coarticulation. Also, data on the intelligibility of real and synthetic utterances are presented and compared. The model seems to account satisfactorily for changes in lipshape due to variation in phonetic context and rate of speaking, and the implementation of the model is shown to produce patterns of intelligibility close to those obtained with natural talkers. [Work supported by Department of Clinical Investigation, WRAMC.]
Dielectric losses in a large signal hydrophone far from resonance evaluated from its hysteresis
Authors: Porat Yariv , Browder Margaret E.
Published: October 2005

A large signal hydrophone may deviate from the linear dielectric behavior which is characterized by the hysteresis curve. If the hydrophone is operated far from its mechanical resonance, the only losses evident in the hysteresis loop will be dielectric. This work describes a method to calculate the hysteresis pattern using Landau phenomenological theory [F. Jona and G. Shirane, Ferroelectric Crystals (McMillan, New York, 1962)] and network analysis of a Tower Sawyer's bridge. The algorithm has the Landau free energy coefficients, the physical dimensions of the hydrophone, and the presumed lossy elements of the network as data. An iteration method was developed for calculating the blocked capacitance of the hydrophone to less than 1% convergence. The method also enables one to predict the temperature and pressure dependence of losses from Landau theory. The algorithm is programmed on a HP9815S desk calculator and plotted for comparison with the oscilloscope recorded pattern of the hysteresis.
Simulation of an underwater load for transducers using a lossy polymeric layer
Authors: Nicholls B. H.
Published: October 2005

The availability of in air test facilities for sonar arrays, particularly regarding validation procedures for the drive amplifiers of the arrays, offers a convenient and cost effective approach for carrying out these tests. An investigation, based on the use of lossy polymers normally employed in anechoic applications, has identified a material with the appropriate properties. This results in a matched impedance with the required loss parameters to enable a close simulation of a water load. The study was considered, initially, from single transducer element characteristics and developed further as an array configuration dummy load. The design philosophy and salient features of a specific system are discussed and features of the test procedure are highlighted to demonstrate a unique approach to material characterization.
Computation of the sound field near baffled and unbaffled planar sources using the fast Fourier transform
Authors: Williams Earl G.
Published: October 2005

The fast Fourier transform is used to evaluate Rayleigh's formula and the resulting bias errors are carefully analyzed. The unacceptably large errors which often occur are nearly completely eliminated by the use of an “integrated” Green function to replace the usual Neumann Green function in the Rayleigh integral. This new Green function is shown to cause a large spatial attenuation of the replicated sources and to provide highly accurate results. Furthermore, it is shown that FFT techniques can be used to obtain not only the nearfield pressure, but also the nearfield vector velocity and vector intensity for a planar source of arbitrary shape (with specified surface velocity). An iterative technique is presented using the FFT to evaluate the radiation from unbaffled planar sources. In particular the computation of the pressure on the surface of vibrating, unbaffled disk is shown to be in excellent agreement with the exact solution using oblate spheroidal functions. The radiated fields from unbaffled sources of any geometry and surface velocity may be evaluated using this technique. The FFT provides an extremely fast and economical numerical solution for radiation problems, especially if an array processor is used to speed the calculations. Solutions using the iterative technique with an array processor are estimated to take only in the order of 10's of seconds to compute 16 000 field points.
Computerized measurement and tracking of acoustical resonances
Authors: Conte D. , Garrett S. L.
Published: October 2005

A system will be described which incorporates a Hewlett-Packard 85 “desk-top” computer to control a frequency synthesizer and read the output of a lock-in analyzer to measure, display and record the resonant frequencies, amplitudes, and quality factors of several modes of an acoustical resonator. The system is capable of locating, measuring, and tracking the resonant modes as parameters which effect sound speed and attenuation are varied. [This work has been supported by a grant from the NPS Foundation Research Program.]
Low cost digital data acquisition system for measuring the free decay of standing waves in continuous media
Authors: Meredith Roger W. , Zuckerwar Allan J.
Published: October 2005

A low-cost digital system based on an 8-bit Apple II microcomputer has been designed to provide on-line control, data acquisition, and evaluation of sound absorption measurements in gases. The measurements are conducted in a resonant tube, in which an acoustical standing wave is excited, the excitation removed, and the sound absorption evaluated from the free decay envelope. The free decay is initiated from the computer keyboard, after the standing wave is established and the microphone response signal is the source of the analog signal for the A/D converter. The acquisition software is written in assembly language and the evaluation software in BASIC. This presentation will discribe the acoustical measurement, hardware, software, and system performance. Also, measurements of sound absorption in air will be presented as an example. [Work supported by NASA under Grant NAG 1-234.]
Modal analysis and its application
Authors: Ebbitt Gordon , Suzuki Hideo , Abbagnaro Lou
Published: October 2005

Modern modal analysis techniques provide a method for the rapid determination of structural mode shapes and frequencies. This analysis utilizes the impulse measurement of the transfer inertance (acceleration/force) between a reference point and points on the surface of a structure. The procedure involved in the analysis will be outlined and potential problems in its implementation will be discussed. These potential problems include the improper location of the reference position, the number of measurement points chosen, the resolution of the processing system, and the spectral content of the impulse. The application of modal analysis to musical instruments will be considered and some results will be presented.
Experimental study of vibration and radiation characteristics of a violin
Authors: Strong William Y. , Torick Emil , Maynard Julian D.
Published: October 2005

A study of a violin's vibration and radiation characteristics has been undertaken using two techniques. The goals of the study are to perform a modal analysis of the instrument and to use the nearfield acoustical holography (NAH) system located at the Pennsylvania State University to reconstruct the instrument's radiated sound field. The modal analysis will locate the various resonances of the top and back plates and show the relative motion of the plates. The NAH system will locate those areas on the instrument that produce the energy that radiates to the farfield. This paper will discuss the work completed to date and present the results obtained from this work.
The “4X” real-time digital signal processor
Authors: DiGiugno Peppino , Kott Jean , Armand Jean-Pierre
Published: October 2005

We have realized a real-time digital signal processor for applications in real-time analysis and synthesis of sound and music. The 4X processor works as a peripheral connected to a host computer. A main feature of the 4X processor is its modularity. It consists of up to 16 elementary processors working in parallel. Each of these processors contains a data memory, an address memory, a microprogram memory, a function memory, a multiplier, and an arithmetic and logic unit. All the computations are done in 24-bit fixed point arithmetic. The 4X processor also contains 256 programmable clocks and a large dual buffer memory for recording and reproduction of signals to and from the host computer disk. The programming of the 4X, which is done in high level languages (FORTRAN and C), allows an easy programmation of the different possible structures of the machine, as oscillators, envelope generators, filters and so on. These capabilities make it possible to implement all known audio synthesis and processing techniques. The 4X processor is now used for musical composition, synthesis of aircraft noise for flight simulators, synthesis of underwater noise of submarines, audio mixing, and psychoacoustic stimulus generation.
Scattering of ultrasonic waves in the MHz range from rough surfaces and rough defects in metals
Authors: Quentin G. , de Billy M. , Jungman A. , Tenoudji F. Cohen
Published: October 2005

A review is presented on the study of the scattering of ultrasonic waves from water-metal interfaces. These surfaces are either random, or periodically rough. For random rough surface the inverse problem concerns the determination of the rms roughness, the autocorrelation length of the surface's profile and the probability density function of heights. Comparisons will be shown with theoretical scattering diagrams. For periodic surfaces we use ultrasonic spectroscopy in order to study the frequency dependence of the diffracted waves. At normal incidence “anomalies” are observed in the spectra and are explained by mode conversion to waves propagating along the surface. A comparison is made between experimental and theoretical spectra. The inverse problem is also discussed. We shall also describe experiments made on a rough penny-shaped crack embedded in a titanium block and show that one can ultrasonically “measure” not only the size of the crack, but its roughness and the autocorrelation length of its profile.
Scattering by a periodic rough surface
Authors: Johnson Richard K.
Published: October 2005

An exact theory based on the extended boundary conditions method [S. L. Chuang and R. K. Johnson, J. Acoust. Soc. Am. 71, 1368–1376 (1982)] for acoustical scattering from a periodic rough interface between a fluid and a solid is reviewed. Good agreement between theoretical calculations and laboratory measurements is demonstrated. Theory and measurements are used to explore the effects of roughness height and shape as well as material properties on the scattering behavior. Interesting phenomena involving surface wave generation and finite aperture transducer effects are also discussed.
Scattering by surfaces with multiple and infinite hierarchies of roughness scales
Authors: Moritz Elan
Published: October 2005

The theories of scattering from surfaces with multiple roughness scales and surfaces with an infinite hierarchy of roughness scales are reviewed. The statistics of scattered waves are developed from the statistics of the rough surface for single frequency, narrow-band and broadband signals. The influence of variations of ratios of dominant (wavelength) scales will be explored for rough surfaces with finite multiplicity.
Scattering from statistically rough surfaces: A 3-D numerical study of the Kirchhoff approximation
Authors: Berman David H. , Perkins John S.
Published: October 2005

The angular spread of acoustic radiation scattered from a rough surface is computed using the Kirchhoff approximation. Both the dependence on azimuth and elevation of the scattered intensity are plotted for model surface statistics. Results are compared to those of perturbation theory and to those of an extended Kirchhoff approximation which formally reduces to the perturbation results in the small surface height limit. In all cases studied, the perturbation result for the scattered intensity near the specular direction dominates the Kirchhoff result. Both hard and soft surfaces are considered and an example of non-Gaussian statistics is presented. The Kirchhoff integral is also compared to an approximation of that integral proposed by M. V. Berry [J. Phys. A 8, 566–584 (1975)].
Acoustic energy transfer in the presence of a temperature gradient
Authors: Barmatz M. , Allen J. , Gaspar M.
Published: October 2005

The transmission of acoustic energy through a temperature discontinuity region of a long tube resonator has been investigated. The temperature discontinuity divides the tube into two homogeneous regions. With a constant velocity generator at the cold end, the condition for maximum power transform to the termination lead at the hot end is that the source transducer see as high an impedance as possible. The boundary condition at the temperature discontinuity is that the interface impedance be the same as seen from the hot or cold region. For the case of negligible attenuation, this condition will be met if the characteristic impedance in each region (ρc/s) is the same. For example, in the case of the interface impedance being zero, the lengths of the two regions must be odd multiples of λ/4, while if this interface impedance is infinite, the lengths must be integral multiples of λ/2. The effects of non-negligible attenuation on the energy transfer will also be presented. Experiments are now underway to test the predictions of this model. [Work supported by NASA.]
Analytic expressions for resonant and anti-resonant frequencies in a composite viscoelastic material
Authors: Kligman Ronald L. , Madigosky Walter
Published: October 2005

General analytic representations for resonances and anti-resonances in the effective dilatation modulus and effective density have been found. These are valid in the Rayleigh regime k1a < 1, where k1 is the compressional wavenumber in the host medium and a is the radius of the spherical inclusion. The resonance frequency with respect to the dilatation modulus can assume values that are real or purely imaginary which correspond to the cases of an underdamped and overdamped system, respectively. A similar situation holds true for the case of the anti-resonance frequency. The effective density shows resonant and anti-resonant behavior analogous to the effective dilatation modulus. All these frequencies are shown to be functions of the weighted ratio of the Lamé constants of the host and inclusion media. Some specific examples will be considered.
Profile analysis for irregular intensity spectra
Authors: Kidd Gerald , Green David M.
Published: October 2005

In our previous experiments with the simultaneous spectral discrimination process called profile analysis, we measured the just detectable increment in the center component of a multitone complex. In those experiments, the multitone background was uniform in level. The present paper describes the results from conditions where the center component of the complex (the one to which the signal was added) was varied in level relative to the other components, thus forming a “pedestal” on the other-wise flat spectrum. The pedestal was varied from 18 dB below to 24 dB above the level of the other background components. Four equilog multi-tone backgrounds were used consisting of 3, 5, 11, or 21 components including the pedestal component. The overall level of the sound was varied ± 20 dB within trials. The threshold of an increment in the middle component of the complex was measured as a function of the pedestal The resulting function had a rather broad, shallow “U” shape. The minimum in the function usually occurred at a small positive pedestal value. In keeping with our earlier findings, the more components in the background, the lower the increment threshold. An asymmetry was observed between positive and negative pedestal levels that may be attributed to masking. [Work supported by NIH.]
Twenty-year life hydrophones
Authors: Tims A. C.
Published: October 2005

A particular hydrophone design application has been chosen to demonstrate that high reliability can be achieved through the use of reliability modeling, an in-depth knowledge of transducer design, careful selection of materials, and good construction techniques. The reliability prediction for the design has been based upon random component failure with the assumption of a constant hazard rate. No consideration has been given to wear-out failure. The design features that increase reliability are presented; alternate design approaches, predictions of lifetime based upon reliability modeling, choices of materials, and successful operation of hydrophones to date are discussed. It is expected that this hydrophone will demonstrate that sonar equipment can be made with reliability equal to space-age equipment.
Discrimination of modulation depth of SAM noise
Authors: Wakefield Gregory H. , Viemeister Neal V.
Published: October 2005

Difference limens (DLs) for modulation depth of sinusoidally amplitude modulated (SAM) wideband noise were measured as a function of the modulation depth of the standard, the frequency of modulation (25, 100, 400 Hz), and the spectrum level of the noise (0, 20, 40 dB SPL). Preliminary results for one subject show that psychometric functions are linear and parallel in Log d′ and 10 Log(m12 − m02), where m1 and m0 are the modulation depths of the variable and standard stimuli, respectively. In the main study, three subjects were tested using a 2IFC adaptive procedure. The DL [10 Log(m12 − m02)] is at, or slightly above, the modulation detection threshold when m0 is small. For m0 above modulation threshold, the DL is a positively accelerating function of 20 Log(m0). Weber's Law does not describe the discrimination of modulation depth over any appreciable range. Across all conditions, DLs tend to converge to values around −9 dB when the level of the standard is −5 dB (m0 = 56%) and generally cannot be measured beyond this level. The effects of noise level on the DL are, in general, small. [Supported by NIH grant NS12125 and by USPHS HD07151 traineeship from the Center for Research in Human Learning.]
Monaural phase effects: Release from masking
Authors: Schroeder Manfred R. , Mehrgardt S. , Strube H. W.
Published: October 2005

Masked thresholds of tones, masked by multicomponent harmonic stimuli, depend critically on the phase spectrum of the masker. Monaural phase effects in excess of 40 dB (release from masking) are reported. Based on these results, intelligible speech has been created by phase manipulation of harmonic signals having a time-invariant flat spectrum.
Formant discrimination and speech intelligibility at low signal-to-noise ratios in subjects with a sensory-neural hearing loss
Authors: Horst J. W. , Ritsma R. J.
Published: October 2005

Formant discrimination experiments were performed with a group of hearing-impaired subjects who according to an audiological diagnosis had a purely sensory-neural hearing loss. The subjects were selected for their relatively fiat audiograms. Most of these subjects showed abnormal susceptibility to ambient noise with regard to speech intelligibility. Apart from discrimination in noise of signal with a formantlike spectrum, the frequency selectivity as measured by means of psychophysical tuning curves and speech intelligibility in noise was measured. Significant correlations were found between formant discrimination in noise, frequency selectivity, and speech intelligibility in noise.
Frequency discrimination and signal detection in band-reject noise
Authors: Emmerich David S. , Brown William S. , Fantini Debbie A.
Published: October 2005

It is known that sinusoidal stimulation creates a broad pattern of activity in the inner ear. In an attempt to assess the importance of activity remote from the peak of this pattern to subjects given a frequency discrimination task, the tones to be discriminated were present in a “notch” in a background of band-reject noise. The width of the notch was varied in different blocks of trials. For purposes of comparison, the subjects were also given the task of detecting tones presented in the notches of various widths in band-reject noise. The results indicate that when subjects are given frequency-discrimination and signal-detection tasks (adjusted to be of approximately equal difficulty in a white noise background), information is integrated over a wider range in the frequency-discrimination task. A further experiment was conducted in which a noise floor was present in addition to band-reject noise. Again, information from regions remote from the peak of the pattern of activity was found to be more important for frequency discrimination than for detection.
A possible physiological correlate of frequency discrimination
Authors: Mills John H. , Schmiedt Richard A.
Published: October 2005

The ability of an observer to detect a change in the frequency of a signal is called frequency discrimination (Δf). Frequency discrimination is dependent upon both the signal level and signal frequency. Here, the hypothesis is that frequency discrimination at 1.0 kHz and above is based upon a principle of edge detection of excitation patterns present in the auditory nerve. For lack of more complete data, the high-frequency slopes of tuning curves of primary fibers are used as the criterion measures of the edge detector. There is excellent qualitative agreement between psychophysical data, which describe Δf as a function of signal level and signal frequency, and physiological data, which describe the high-frequency slope of a tuning curve as a function of signal level and characteristic frequency. A quantitative test requires additional physiological and psychophysical data. [Supported in part by an NINCDS contract and a grant from NIEHS.]
Frequency discrimination for single channel electrocutaneous signals
Authors: Bull Dale , Eilers Rebecca E. , Oller D. K. , Mandalia Baiju D.
Published: October 2005

Frequency discrimination of electrocutaneous signals was investigated for several different base frequencies at signal durations of 50, 100, and 250 ms. Bipolar pulses (height = 10 mA, width = 13 μs) were presented on a single electrode placed 1 cm above and 8 cm to either the left or right Three practiced subjects judged pairs of stimuli (ISI = 300 ms) in a same-different task. For signal frequencies below 160 Hz, psycho-metric functions were similar for all subjects and for all signal durations. In this range of frequencies, Δf/f was constant at approximately 0.3. By contrast, for stimuli greater than 200 Hz, the Weber fraction was found to increase both as a function of frequency and signal duration. The results will be discussed in terms of possible intensity coding schemes for electrocutaneous vocoder designs.
Interior noise levels in passive solar residences
Authors: Holger David K. , Pickett Mary S.
Published: October 2005

Typical single family dwellings with passive solar heating systems involve large open plan areas. Multilevel designs often incorporate open plans within and between levels. Current design practice for passive solar residences commonly utilizes hard interior surface treatments and furnishings—primarily wood, metal, glass, or masonry. The combination of open plan design and acoustically hard surfaces can result in unacceptable interior noise levels. The results of a detailed noise survey conducted in a typical passive solar home are presented. The data reveal that household appliances, HVAC equipment, and home entertainment equipment can all cause excessive interior noise levels. Sound levels in excess of 80 dBA and NC values greater than 60 were observed in living areas of the home. These results indicate that boundary absorption is as important in open plan residential design as it is known to be in open plan commercial and institutional design. [Work supported by the Agriculture and Home Economics Experiment Station and the Engineering Research Institute.]
Time separation pitch discrimination of signals consisting of three pulses with various periodicities
Authors: Ceruti Marion G. , Floyd Robert W. , Martin Douglas W.
Published: October 2005

The ability of human listeners to discriminate between equal and unequal pulse separations in signals containing pulse triplets was measured in psychoacoustic experiments. Stimuli were composed of three identical broadband pulses, each of 1.0-ms duration centered at 2.4 kHz. The time occurrence of the third pulse was held constant relative to the first (a delay of 4 ms) while the delay between the first and second pulses was varied between 1.75 and 2.25 ms. In the standard signal, the delays between the first and second pulses and the second and third pulses were both 2.0 ms, while in test signals these delays were made unequal by changing the temporal position of the second pulse. Subjects responded either “test” of “standard” upon presentation of the stimuli. The results show that signals with unequal pulse delays can be discriminated from a signal with equal delays when at least a 0.14-ms difference exists between the time interval between first and second pulse of the standard and the corresponding interval in a test signal. Some subjects reported perceiving two Time Separation Pitches (TSP) in most test signals, in contrast to a higher, single TSP in the standard.
Pitch of components of complex tones
Authors: Peters Robert W. , Moore Brian C. J. , Glasberg Brian R.
Published: October 2005

A current question in pitch theory is whether or not the low pitch of a complex sound is dependent on and derived from individual pitches of lower harmonics. If this is the case, pitch shifts due to masking, adaptation, and diplacusis should be similar for both types of pitch. Findings on this issue are equivocal. Some studies found a closer correspondence than others that the two pitches were affected differently. A basic problem in these studies was that pitch-shift effects for the harmonics were estimated by measuring effects on the pitches of pure tones that correpond to frequencies of the harmonics. In this study pitch matches were obtained to harmonics in complex tones and to the same harmonics in isolation. The results indicated that mean pitch values for harmonics in context did not differ markedly from their pure tone pitches. This is contrary to the findings of Terhardt (Seventh ICA, 1971, paper 20 S 13), and it indicates that the pitches of the lower harmonics in complex tones are not shifted by the presence of adjacent harmonics. Given that our subjects could make consistent and accurate matches to harmonics in context, this method is recommended for further studies of the relationship between component and complex tone pitches.
Effects of monaural stimulation by a low-frequency pure tone on binaural tinnitus
Authors: Young I. M. , Lowry L. D.
Published: October 2005

For a subject with binaural tinnitus pitch 10 000 Hz, loudness 50 dB SPL, left ear was stimulated by a continuous pure tone 250 Hz, 125 dB SPL, 30 min. Immediately following cessation of stimulation, tinnitus completely disappeared from stimulated ear. Twenty-three min after cessation of stimulation, tinnitus was heard as a pitch 12 445 Hz. One h after stimulation, tinnitus pitch changed to 12 131 Hz; 2 h later, 11 000 Hz; 312 h later, 10 953 Hz; 5 h later returned to original pitch 10 000 Hz and stayed on to date. For a period of 23 h after stimulation, however, stimulated ear continuously heard multiple frequency tones approximately 200–800 Hz. In nonstimulated ear, immediately after cessation of stimulation, tinnitus was heard as a pitch 5000 Hz and gradually increased in pitch to 7500 Hz for 312 h. Thereafter, tinnitus pitch has been fluctuating between 9000–10 000 Hz. Tinnitus never returned to previous pitch 10 000 Hz until 16 days after stimulation. Apparent differences in recovery pattern and process between two ears were compared with a previous experiment of 500-Hz monaural stimulation.
Localization of sound in rooms
Authors: Hartmann W. M.
Published: October 2005

We have studied the ability of human listeners to locate the origin of a sound in a room in a series of source azimuth identification experiments. All experiments were done in a small rectangular concert hall with variable geometry and acoustical properties. Subjects localized a 50-ms, 500-Hz sine pulse with an rms error of 3.3° (±0.6°) regardless of room reverberation time. Lowering the ceiling from 11.5 to 3.5 m decreased the error to 2.8° (±0.6°). Subjects localized broadband noise without attack transients with an rms error of 2.3° (±0.6°) if the reverberation time was 1 s. The error increased to 3.2° (±0.7°) if the reverberation time was 5 s. For complex tones without attack transients the localization error continuously increased as the intensity of spectral components decreased. Performance was nearly random for a 500-Hz sine tone, but was significantly better than random for a 5000-Hz sine tone. Our azimuth identification experiments revealed significant biases, as much as 2°; such biases are, of course, invisible in minimum audible angle experiments.
Binaural horizontal plane localization: Influence of monaural spectral cues
Authors: Musicant Alan D. , Butler Robert A.
Published: October 2005

A recent article [Butler and Flannery, Percept. Psychophys. 28, 449–457 (1980)] reported the existence of monaural spatial referent maps (SRMs). These orderly maps related the perceived horizontal plane locus of narrow bands of noise to center frequency, which ranged from 4–14 kHz. Perceived location was governed primarily by spectral composition and not actual source location. Similar SRMs under binaural listening conditions were obtained in the current experiments. Stimuli were presented from a loudspeaker at 270° azimuth. Subjects could report the perceived origin as any of 13 positions, each 15° apart and extending from 360°to 180°. Results were consistent with observations regarding the region of the horizontal plane most sensitive to spectral cues. [Musicant and Butler, J. Acoust. Soc. Am. Suppl. 1 71, S87 (1982)]. Localization tests utilizing specific features of the SRMs decreased binaural localization accuracy. A 4-kHz high-pass noise with a notch at a “map-connecting” region was the stimulus. Stimuli were presented from all 13 loudspeakers in the horizontal plane. Performance decrements were greatest within the 315°–225° arc. The present results strengthen our argument that monaural spectral cues are an important cue to binaural horizontal plane localization.
Growth rate of perceived magnitude as a function of tone-to-noiSe ratio
Authors: Hellman Rhona
Published: October 2005

The growth of overall perceived magnitude (loudness and annoyance) of single tones superimposed on broadband-flat, low-pass, and high-pass noise spectra was measured by absolute magnitude estimation (AME). Results disclose that, in contrast to noiseness, loudness and annoyance growth are dependent on the relationship between the frequency of the added tone and the spectral shape of the noise. Tones centered in noise produce nonmonotonic loudness and annoyance growth functions; those added to the noise skirt produce power functions. Detailed analysis reveals that the measured exponents are invariant across tone-to-noise ratio when the tones are positioned within the noise spectrum, but not when they are added to the skirt. A high-frequency tone combined with low-pass noise increases the growth rate of perceived magnitude as tone-to-noise ratio increases. Conversely, the addition of a low-frequency tone to high-pass noise decreases the growth rate of perceived magnitude. The obtained results are qualitatively compatible with excitation patterns known to be evoked by auditory stimuli at high intensities. [Work supported by NASA Langley Research Center.]
Aided and unaided localization on subjects with different degrees and types of losses
Authors: Gatehouse R. Wayne , Patee C. L.
Published: October 2005

Binaurally deaf persons categorized by (1) three degrees of ac measured losses (26–35 dB, minimal; 36–45 dB, mild; 46–55 dB, moderate) and (2) clinical categories (SN, mixed (M), or conductive C) were tested on accuracy in “simultaneous” localization (i.e., specification of signal position in both azimuth and elevation). For some subjects this was done with and without their hearing aids. Four signals (WN; 1.0- and 2.5-kHz tones; and speech) equated for 72 dB SPL at head position were localized. Three 24-trial randomized blocks of eight degrees azimuths (0°, 45°…315°)×3 elevations (0°, ± 30°) were presented for each signal under all conditions. The orders of signal presentations and blocks were randomly varied. The results like those reported by Tonning (1975) indicate that degree of loss as measured by pure tone audiograms may be useful predictors of localization performance—errors do increase (especially in azimuth) as the loss severity does. However, other factors (age; symmetry/not of audiogram) are also important. Additionally, loss type (SN, M, or C) seems to give some basis for prediction of localization ability. Finally, as in Gatehouse (1981) head-borne monaural aids do appear to produce inferior localization, especially for speech and WN signals. The results will be considered in light of Tonning's long series of articles (1970–75) and in terms of possible diagnostic utility of including localization tests. [Work supported by MRC Canada.]
Synthetic seismograms
Authors: Chapman C. H. , Orcutt J. A.
Published: October 2005

The interpretation of seismic data, in particular oceanic refraction data, has improved considerably, in recent years using synthetic seismograms. Several different methods have been developed for computing synthetic seismograms and within the limitations of the various approximations, shown to give equivalent results. The computation of theoretical seismograms can be divided into two parts: the evaluation of the response in a transform domain (normally frequency and horizontal slowness), and the evaluation of the inverse transforms (Fourier, Fourier-Bessel, or Legendre transforms). Developments of the Haskell matrix method are widely used to evaluate the transformed response. Problems with numerical stability have been solved, and the method has been extended to use asymptotic solutions in inhomogeneous layers. Often the complete response is expanded, at least at some interfaces, into “rays” and only a limited set of rays investigated. The ray expansion often simplifies the evaluation of the inverse transforms and takes advantage of separation in the time domain. The inverse transform methods fall into two catagories: spectral and slowness methods. In spectral methods, the inverse wavenumber transform is evaluated first and the intermediate result is the spectral response for a point source. In slowness methods, the inverse frequency transform is evaluated first and the intermediate result is the impulse “plane wave” response. For a general response function, neither method is numerically preferable but the intermediate result in the slowness method is convenient as it corresponds to slant stacking the data. With the ray expansion, the slowness method is often preferable as the inverse transforms can be evaluated exactly—the generalized ray method and the WKBJ seismogram. The last method has recently been extended to laterally inhomogeneous media.
Wave attenuation
Authors: Richards Paul G.
Published: October 2005

It is commonly observed that propagating waves lose amplitude sooner than might be expected for elastic media with smoothly varying properties. The explanation is in part that real materials are not elastic, and in part that real materials are not smoothly varying but have fine scale spatial fluctuations. We thus speak of anelastic attenuation, and attenuation due to scattering. Anelastic attenuation can be understood in a framework that allows for complex moduli relating stress and strain. A brief review will be given, of successful procedures for accommodating anelasticity in the computation of waveforms. The main part of this paper concerns attenuation of transmitted waves when forward scattering is dominant. This is the case for an elastic medium with a heterogeneous velocity structure when the typical fluctuation is spatially confined within less than a wavelength and the wave propagates over a sufficiently long path. The medium is found to have low-pass characteristics. The removal of high frequencies can often be summarized by a frequency-independent apparent Q. It is important in seismic prospecting to be able to determine whether observed amplitude decay is due to the presence of anelasticity and true dissipation, or to the presence of velocity fluctuations and resultant scattering. Numerical experiments indicate that when scattering dominates over anelasticity: (1) The coda of a transmitted wave contains relatively higher frequencies than the initial phase; (2) the attenuation deduced from the power spectrum of the transmitted wave is greater than that deduced from the phase spectrum; (3) compressional and shear wave apparent Q's are approximately equal; and (3) estimates of apparent Q made from reflected coda vary with frequency, while estimates made from the transmitted waves do not.
Inversion for sediment structure using multichannel data
Authors: Phinney Robert A.
Published: October 2005

The problem of inferring a velocity or impedance structure for a layered sediment has traditionally been posed in terms of a restricted data set, such as a normal incidence reflection seismogram or a refraction travel time curve. Data acquisition systems used for petroleum exploration record 96 or more channels in a linear array, and thus obtain a virtually unaliased representation of the wavefield. The data contains reflections from all angles of incidence, including postcritical angles, all normal modes, leaky modes, and head waves. A key tool in dealing with this kind of data is the Radon transform, which transforms the data set into a plane-wave representation. A raw data set u(x,t) is transformed into v(p,τ), where p, which parameterizes the plane waves, is the x component of slowness, and τ is a reduced (intercept) time. The ensemble of plane-wave seismograms is the basis for inversion of the data to obtain a one-dimensional velocity/impedance model. Modeling the lossy elastic transient plane-wave response is easily done by one of the modifications of the Thomson-Haskell matrix method. The algorithm can be vectorized to run very rapidly on a machine with an array processor. The object of the inversion is to obtain a model whose plane-wave decomposition agrees exactly with that of the observed wavefield. In the absence of noise, one plane-wave seismogram suffices to determine the model. An ensemble for a range of wave parameters overdetermines the model. In the presence of noise, the problem has the character of an ill-posed, overdetermined least squares problem. In order to carry through the solution by means of least squares, it is necessary to go through a construction procedure to make the trial model linearly close to the correct one in some parameter subspace. Several construction procedures proceed by stripping from the top down, although none have been reduced to a fully automatic alogrithm, and require some human intervention in the iterations.
Application of a Biot/Stoll model in acoustic simulations
Authors: Brunson Burlie A. , Molinelli Eugene
Published: October 2005

A physical sediment model based on Biot Theory as modified by Stoll is used to predict the geoacoustic properties of the sea floor. The sediment properties are predicted as functions of depth and frequency for use as inputs to acoustic simulation models. The geoacoustic predictions are compared with geoacoustic descriptions derived from inversion of acoustic bottom loss data. The acoustic significance of observed differences is assessed using acoustic simulations such as bottom loss and propagation loss. The sensitivity of geoacoustic predictions to the variability of sea floor geophysical properties is discussed. [Work supported by ONR and NORDA.]
Deterministic geoacoustic model for a randomly layered ocean bottom
Authors: Gilbert Kenneth E.
Published: October 2005

A deterministic wave equation for the mean acoustic field in a randomly layered medium is derived. The wave equation is used to define a “smooth” deterministic geoacoustic model which is acoustically equivalent at low frequencies to a randomly layered model. The wave equation derivation starts with fundamental relations (Newton's law and Hooke's law) and leads to deterministic expressions for effective sound speed, density, and absorption which are similar to those obtained for periodic medium. In particular, the effective sound speed of the deterministic model depends on the grazing angle, being higher for propagation parallel to the layering than for propagation perpendicular to it. When the angular dependence of the effective sound speed is included, the low-frequency (f < 100 –200 Hz) reflection coefficients for the deterministic model are in good agreement with the ensemble averaged coefficients for the associated randomly layered model. Without the angular dependence, the agreement is marginal.
Acoustic propagation in the ocean with a porous bottom
Authors: Yamamoto Tokuo
Published: October 2005

Biot's theory was directly incorporated with the normal mode analysis of acoustic propagation in the ocean with a porous elastic bottom. The complex dispersion relations was obtained in a closed form for the case of a homogeneous ocean with a poro-elastic half-space. The attenuation data from the shallow water experiments in the Mediterranean Sea were compared with the theoretical calculations. The 100-m water/sandy bottom data showed an optional transmission frequency of about 250 Hz. The theoretical analysis using the sediment data revealed that the attenuation of the high-frequency components are mainly due to the seepage of pore water through the sand (the slow compressional wave effect) and the attenuation of the low-frequency components are owing to the energy dissipation of the shear waves by the Coulomb friction in the consolidated sand. As a result, the intermediate frequency components have minimum overall attenuations and result into the optimal transmission frequencies. The agreements between the theory and the experiments are good. [Work supported by NSF.]
Wake Island hydrophone seismic array
Authors: Sutton George H. , Pomeroy Paul W. , Walker Daniel A. , McCreery Charles S.
Published: October 2005

The hydrophone array installed near Wake Island (WHA) was reactivated as a seismic station in June 1979 by the Hawaii Institute of Geophysics. WHA consists of six bottom phones, at 5.5-km depth, in an array 40 km across and an additional four pairs of phones, at large spacing, suspended at SOFAR depth. The bottom array lies in a flat area on high Q lithosphere > 108 years old. All of the deep and five of the SOFAR phones are operational. Until September 1982, generally data were recorded from only three phones, on a slow speed analog cassette recorder. The station is now being upgraded for digital recording of all 11 phones. Many circumpacific earthquakes and nuclear tests have been well recorded by WHA, its location being ideally suited for monitoring such events. Nearly pure oceanic paths between WHA and circumpacific sources facilitate studies of the ocean-lithosphere waveguide such as propagation and attenuation of high-frequency, teleseismic Pn and Sn. For frequencies above 3 Hz, WHA is quieter than most quiet continental stations. [Work supported by AFOSR and ONR.]
A corner-reflector parametric transducer for acoustic imaging in marine sediment
Authors: Sackman G. L. , Eyman R. L. , LaStrange P. J. , Seeman A. H.
Published: October 2005

An experimental transducer and pulse modulator has been constructed which employs the parametric transient effect [M. B. Moffett, P. J. Westervelt, and R. T. Beyer, J. Acoust. Soc. Am. 49, 339–343 (1941)] for locating and identifying objects buried in marine sediment [L. A. Thompson and T. G. Muir, J. Acoust. Soc. Am. 55, 429 (1974)]. It was desired to produce a probing acoustic beam with the best possible combination of short pulse length and narrow beamwidth for good object resolution while simultaneously using the lowest possible frequency for penetration of the sediment. A novel corner reflector configuration for the transducer is used which is well adapted to operation at great depths. The transducer is shock excited at its 80 kHz primary frequency, from which a 100 μs parametrically demodulated baseband pulse is generated in a collimated beam of approximately 20-cm diameter over ranges from 5 to 15 m. [Work supported in part of Naval Undersea Warfare Engineering Station, Keyport, WA.]